Attachment 356816 Details for Bug 192316 – Patch with GTK changes

[patch] Patch with GTK changes

bug-192316-20181207085700.patch (text/plain), 7.97 MB, created by youenn fablet on 2018-12-07 08:57:14 PST

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Creator: youenn fablet

Created: 2018-12-07 08:57:14 PST

Size: 7.97 MB

patch

obsolete

>Subversion Revision: 238900
>diff --git a/Source/ThirdParty/libwebrtc/ChangeLog b/Source/ThirdParty/libwebrtc/ChangeLog
>index 77516ad964359e32e300b150f31df881ae851a66..57055e3fbe23d4b0d04699d292230d9a07f71bd3 100644
>--- a/Source/ThirdParty/libwebrtc/ChangeLog
>+++ b/Source/ThirdParty/libwebrtc/ChangeLog
>@@ -1,3 +1,20 @@
>+2018-12-06  Youenn Fablet  <youenn@apple.com>
>+
>+        Update libwebrtc up to 0d007d7c4f
>+        https://bugs.webkit.org/show_bug.cgi?id=192316
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Updating to latest libwebrtc will allows cherry-picking important bug fixes.
>+
>+        * Configurations/libwebrtc.iOS.exp:
>+        * Configurations/libwebrtc.iOSsim.exp:
>+        * Configurations/libwebrtc.mac.exp:
>+        * Source/third_party/abseil-cpp: refreshed.
>+        * Source/webrtc: refreshed.
>+        * WebKit/0001-libwebrtc-changes.patch: Removed.
>+        * libwebrtc.xcodeproj/project.pbxproj:
>+
> 2018-12-01  Thibault Saunier  <tsaunier@igalia.com>
> 
>         [GStreamer][WebRTC] Build opus decoder support in libwebrtc
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 51b1a7de57895e4941e9099de20929516077cb71..7ee239ef0eea4baa2a90bb74719fe95313986815 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,14 @@
>+2018-12-06  Youenn Fablet  <youenn@apple.com>
>+
>+        Update libwebrtc up to 0d007d7c4f
>+        https://bugs.webkit.org/show_bug.cgi?id=192316
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Update include according new libwebrtc.
>+
>+        * platform/mediastream/libwebrtc/LibWebRTCProvider.cpp:
>+
> 2018-12-05  Youenn Fablet  <youenn@apple.com>
> 
>         Enable the possibility to do video capture in UIProcess
>diff --git a/Source/WebKit/ChangeLog b/Source/WebKit/ChangeLog
>index 790710d968e6dc64877da03002986aa60e5cfff7..a833418df3f2360258036b2f6115fd99275444b7 100644
>--- a/Source/WebKit/ChangeLog
>+++ b/Source/WebKit/ChangeLog
>@@ -1,3 +1,22 @@
>+2018-12-06  Youenn Fablet  <youenn@apple.com>
>+
>+        Update libwebrtc up to 0d007d7c4f
>+        https://bugs.webkit.org/show_bug.cgi?id=192316
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Update code base according new libwebrtc backend
>+        In particular, use directly int64_t for packet time.
>+
>+        * Configurations/WebKit.xcconfig:
>+        * NetworkProcess/webrtc/LibWebRTCSocketClient.cpp:
>+        (WebKit::LibWebRTCSocketClient::signalReadPacket):
>+        * NetworkProcess/webrtc/LibWebRTCSocketClient.h:
>+        * NetworkProcess/webrtc/NetworkRTCProvider.cpp:
>+        * NetworkProcess/webrtc/NetworkRTCSocket.h:
>+        * WebProcess/Network/webrtc/LibWebRTCSocket.cpp:
>+        (WebKit::LibWebRTCSocket::signalReadPacket):
>+
> 2018-12-05  Youenn Fablet  <youenn@apple.com>
> 
>         Remove a document entry from NetworkMDNSRegister::m_services when document goes away
>diff --git a/Source/ThirdParty/libwebrtc/CMakeLists.txt b/Source/ThirdParty/libwebrtc/CMakeLists.txt
>index 25478847fe771ac9f9251076838ef66e5353c880..6c0856e53cd68b8eba52a0eb10194e1cd1de6c68 100644
>--- a/Source/ThirdParty/libwebrtc/CMakeLists.txt
>+++ b/Source/ThirdParty/libwebrtc/CMakeLists.txt
>@@ -24,6 +24,11 @@ endif ()
> set(webrtc_SOURCES
>     Source/third_party/abseil-cpp/absl/base/dynamic_annotations.cc
>     Source/third_party/abseil-cpp/absl/base/internal/raw_logging.cc
>+    Source/third_party/abseil-cpp/absl/base/internal/throw_delegate.cc
>+    Source/third_party/abseil-cpp/absl/strings/ascii.cc
>+    Source/third_party/abseil-cpp/absl/strings/internal/memutil.cc
>+    Source/third_party/abseil-cpp/absl/strings/match.cc
>+    Source/third_party/abseil-cpp/absl/strings/string_view.cc
>     Source/third_party/abseil-cpp/absl/types/bad_optional_access.cc
>     Source/third_party/abseil-cpp/absl/types/bad_variant_access.cc
>     Source/third_party/abseil-cpp/absl/types/optional.cc
>@@ -364,12 +369,15 @@ set(webrtc_SOURCES
>     Source/webrtc/api/audio_options.cc
>     Source/webrtc/api/call/transport.cc
>     Source/webrtc/api/candidate.cc
>+    Source/webrtc/api/create_peerconnection_factory.cc
>+    Source/webrtc/api/crypto/cryptooptions.cc
>     Source/webrtc/api/datachannelinterface.cc
>     Source/webrtc/api/jsep.cc
>     Source/webrtc/api/jsepicecandidate.cc
>     Source/webrtc/api/mediaconstraintsinterface.cc
>     Source/webrtc/api/mediastreaminterface.cc
>     Source/webrtc/api/mediatypes.cc
>+    Source/webrtc/api/media_transport_interface.cc
>     Source/webrtc/api/peerconnectioninterface.cc
>     Source/webrtc/api/proxy.cc
>     Source/webrtc/api/rtcerror.cc
>@@ -380,13 +388,17 @@ set(webrtc_SOURCES
>     Source/webrtc/api/rtptransceiverinterface.cc
>     Source/webrtc/api/statstypes.cc
>     Source/webrtc/api/transport/bitrate_settings.cc
>+    Source/webrtc/api/transport/goog_cc_factory.cc
>     Source/webrtc/api/transport/network_types.cc
>     Source/webrtc/api/units/data_rate.cc
>     Source/webrtc/api/units/data_size.cc
>     Source/webrtc/api/units/time_delta.cc
>     Source/webrtc/api/units/timestamp.cc
>+    Source/webrtc/api/video/builtin_video_bitrate_allocator_factory.cc
>     Source/webrtc/api/video/color_space.cc
>+    Source/webrtc/api/video/encoded_image.cc
>     Source/webrtc/api/video/encoded_frame.cc
>+    Source/webrtc/api/video/hdr_metadata.cc
>     Source/webrtc/api/video/i010_buffer.cc
>     Source/webrtc/api/video/i420_buffer.cc
>     Source/webrtc/api/video/video_bitrate_allocation.cc
>@@ -401,6 +413,7 @@ set(webrtc_SOURCES
>     Source/webrtc/api/video_codecs/video_decoder_factory.cc
>     Source/webrtc/api/video_codecs/builtin_video_decoder_factory.cc
>     Source/webrtc/api/video_codecs/builtin_video_encoder_factory.cc
>+    Source/webrtc/api/video_codecs/create_vp8_temporal_layers.cc
>     Source/webrtc/api/video_codecs/sdp_video_format.cc
>     Source/webrtc/api/video_codecs/video_codec.cc
>     Source/webrtc/api/video_codecs/video_decoder.cc
>@@ -414,12 +427,9 @@ set(webrtc_SOURCES
>     Source/webrtc/audio/audio_state.cc
>     Source/webrtc/audio/audio_transport_impl.cc
>     Source/webrtc/audio/channel_receive.cc
>-    Source/webrtc/audio/channel_receive_proxy.cc
>     Source/webrtc/audio/channel_send.cc
>-    Source/webrtc/audio/channel_send_proxy.cc
>     Source/webrtc/audio/null_audio_poller.cc
>     Source/webrtc/audio/remix_resample.cc
>-    Source/webrtc/audio/time_interval.cc
>     Source/webrtc/audio/transport_feedback_packet_loss_tracker.cc
>     Source/webrtc/audio/utility/audio_frame_operations.cc
>     Source/webrtc/call/audio_receive_stream.cc
>@@ -515,7 +525,6 @@ set(webrtc_SOURCES
>     Source/webrtc/common_video/i420_buffer_pool.cc
>     Source/webrtc/common_video/incoming_video_stream.cc
>     Source/webrtc/common_video/libyuv/webrtc_libyuv.cc
>-    Source/webrtc/common_video/video_frame.cc
>     Source/webrtc/common_video/video_frame_buffer.cc
>     Source/webrtc/common_video/video_render_frames.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.cc
>@@ -583,7 +592,6 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_coding/acm2/acm_resampler.cc
>     Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
>     Source/webrtc/modules/audio_coding/acm2/call_statistics.cc
>-    Source/webrtc/modules/audio_coding/acm2/codec_manager.cc
>     Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>     Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc
>     Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
>@@ -775,7 +783,6 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
>     Source/webrtc/modules/audio_coding/neteq/random_vector.cc
>     Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.cc
>-    Source/webrtc/modules/audio_coding/neteq/rtcp.cc
>     Source/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
>     Source/webrtc/modules/audio_coding/neteq/sync_buffer.cc
>     Source/webrtc/modules/audio_coding/neteq/tick_timer.cc
>@@ -857,9 +864,7 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
>     Source/webrtc/modules/audio_processing/agc2/down_sampler.cc
>     Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
>-    Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.cc
>     Source/webrtc/modules/audio_processing/agc2/gain_applier.cc
>-    Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.cc
>     Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc
>     Source/webrtc/modules/audio_processing/agc2/limiter.cc
>     Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.cc
>@@ -921,6 +926,7 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
>     Source/webrtc/modules/audio_processing/voice_detection_impl.cc
>     Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
>+    Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc
>     Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
>     Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler.cc
>     Source/webrtc/modules/congestion_controller/bbr/bbr_factory.cc
>@@ -928,19 +934,18 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/congestion_controller/bbr/data_transfer_tracker.cc
>     Source/webrtc/modules/congestion_controller/bbr/loss_rate_filter.cc
>     Source/webrtc/modules/congestion_controller/bbr/rtt_stats.cc
>-    Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
>-    Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_factory.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.cc
>     Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc
>     Source/webrtc/modules/congestion_controller/receive_side_congestion_controller.cc
>-    Source/webrtc/modules/congestion_controller/rtp/pacer_controller.cc
>+    Source/webrtc/modules/congestion_controller/rtp/control_handler.cc
>     Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller.cc
>     Source/webrtc/modules/congestion_controller/rtp/send_time_history.cc
>     Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
>@@ -975,6 +980,7 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc
>     Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
>     Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
>+    Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_size.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
>@@ -1050,7 +1056,6 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/video_coding/codecs/i420/i420.cc
>     Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
>     Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
>-    Source/webrtc/modules/video_coding/codecs/vp8/vp8_temporal_layers.cc
>     Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
>     Source/webrtc/modules/video_coding/codecs/vp8/libvpx_interface.cc
>     Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc
>@@ -1087,7 +1092,6 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator.cc
>     Source/webrtc/modules/video_coding/utility/frame_dropper.cc
>     Source/webrtc/modules/video_coding/utility/ivf_file_writer.cc
>-    Source/webrtc/modules/video_coding/utility/moving_average.cc
>     Source/webrtc/modules/video_coding/utility/quality_scaler.cc
>     Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator.cc
>     Source/webrtc/modules/video_coding/utility/simulcast_utility.cc
>@@ -1108,6 +1112,7 @@ set(webrtc_SOURCES
>     Source/webrtc/p2p/base/basicpacketsocketfactory.cc
>     Source/webrtc/p2p/base/dtlstransport.cc
>     Source/webrtc/p2p/base/dtlstransportinternal.cc
>+    Source/webrtc/p2p/base/icecredentialsiterator.cc
>     Source/webrtc/p2p/base/icetransportinternal.cc
>     Source/webrtc/p2p/base/p2pconstants.cc
>     Source/webrtc/p2p/base/p2ptransportchannel.cc
>@@ -1130,15 +1135,12 @@ set(webrtc_SOURCES
>     Source/webrtc/p2p/base/transportdescriptionfactory.cc
>     Source/webrtc/p2p/base/turnport.cc
>     Source/webrtc/p2p/base/turnserver.cc
>-    Source/webrtc/p2p/base/udptransport.cc
>     Source/webrtc/p2p/client/basicportallocator.cc
>     Source/webrtc/p2p/client/turnportfactory.cc
>     Source/webrtc/p2p/stunprober/stunprober.cc
>     Source/webrtc/pc/audiotrack.cc
>-    Source/webrtc/pc/bundlefilter.cc
>     Source/webrtc/pc/channel.cc
>     Source/webrtc/pc/channelmanager.cc
>-    Source/webrtc/pc/createpeerconnectionfactory.cc
>     Source/webrtc/pc/datachannel.cc
>     Source/webrtc/pc/dtlssrtptransport.cc
>     Source/webrtc/pc/dtmfsender.cc
>@@ -1199,20 +1201,22 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/event_tracer.cc
>     Source/webrtc/rtc_base/experiments/alr_experiment.cc
>     Source/webrtc/rtc_base/experiments/congestion_controller_experiment.cc
>+    Source/webrtc/rtc_base/experiments/cpu_speed_experiment.cc
>     Source/webrtc/rtc_base/experiments/field_trial_parser.cc
>     Source/webrtc/rtc_base/experiments/field_trial_units.cc
>+    Source/webrtc/rtc_base/experiments/jitter_upper_bound_experiment.cc
>+    Source/webrtc/rtc_base/experiments/normalize_simulcast_size_experiment.cc
>     Source/webrtc/rtc_base/experiments/quality_scaling_experiment.cc
>     Source/webrtc/rtc_base/experiments/rtt_mult_experiment.cc
>     Source/webrtc/rtc_base/file.cc
>     Source/webrtc/rtc_base/file_posix.cc
>-    Source/webrtc/rtc_base/filerotatingstream.cc
>-    Source/webrtc/rtc_base/fileutils.cc
>     Source/webrtc/rtc_base/firewallsocketserver.cc
>     Source/webrtc/rtc_base/flags.cc
>     Source/webrtc/rtc_base/helpers.cc
>     Source/webrtc/rtc_base/httpcommon.cc
>     Source/webrtc/rtc_base/ifaddrs_converter.cc
>     Source/webrtc/rtc_base/ipaddress.cc
>+    Source/webrtc/rtc_base/key_derivation.cc
>     Source/webrtc/rtc_base/location.cc
>     Source/webrtc/rtc_base/logsinks.cc
>     Source/webrtc/rtc_base/logging.cc
>@@ -1228,10 +1232,11 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/nethelpers.cc
>     Source/webrtc/rtc_base/network.cc
>     Source/webrtc/rtc_base/networkmonitor.cc
>-    Source/webrtc/rtc_base/noop.cc
>+    Source/webrtc/rtc_base/network/sent_packet.cc
>     Source/webrtc/rtc_base/nullsocketserver.cc
>     Source/webrtc/rtc_base/numerics/exp_filter.cc
>     Source/webrtc/rtc_base/numerics/histogram_percentile_counter.cc
>+    Source/webrtc/rtc_base/numerics/moving_average.cc
>     Source/webrtc/rtc_base/numerics/sample_counter.cc
>     Source/webrtc/rtc_base/openssladapter.cc
>     Source/webrtc/rtc_base/opensslcertificate.cc
>@@ -1240,8 +1245,7 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/opensslsessioncache.cc
>     Source/webrtc/rtc_base/opensslstreamadapter.cc
>     Source/webrtc/rtc_base/opensslutility.cc
>-    Source/webrtc/rtc_base/optionsfile.cc
>-    Source/webrtc/rtc_base/pathutils.cc
>+    Source/webrtc/rtc_base/openssl_key_derivation_hkdf.cc
>     Source/webrtc/rtc_base/physicalsocketserver.cc
>     Source/webrtc/rtc_base/platform_file.cc
>     Source/webrtc/rtc_base/platform_thread.cc
>@@ -1285,7 +1289,6 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/time/timestamp_extrapolator.cc
>     Source/webrtc/rtc_base/timestampaligner.cc
>     Source/webrtc/rtc_base/timeutils.cc
>-    Source/webrtc/rtc_base/unixfilesystem.cc
>     Source/webrtc/rtc_base/virtualsocketserver.cc
>     Source/webrtc/rtc_base/weak_ptr.cc
>     Source/webrtc/rtc_base/zero_memory.cc
>@@ -1308,6 +1311,7 @@ set(webrtc_SOURCES
>     Source/webrtc/video/call_stats.cc
>     Source/webrtc/video/encoder_rtcp_feedback.cc
>     Source/webrtc/video/overuse_frame_detector.cc
>+    Source/webrtc/video/frame_dumping_decoder.cc
>     Source/webrtc/video/quality_threshold.cc
>     Source/webrtc/video/receive_statistics_proxy.cc
>     Source/webrtc/video/report_block_stats.cc
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>index d95d51ea0c817ad06ace80ea2d026c483ae7b2ab..62286edc552daadd29d3143738507175564deeb8 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>@@ -218,7 +218,6 @@ __ZNK6webrtc18RtpSenderInterface19init_send_encodingsEv
> __ZNK6webrtc18RtpSenderInterface17GetFrameEncryptorEv
> __ZN6webrtc18RtpSenderInterface17SetFrameEncryptorEN3rtc13scoped_refptrINS_23FrameEncryptorInterfaceEEE
> __ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEE
>-__ZNK3rtc14NetworkManager16GetMDnsResponderEv
> __ZTVN6webrtc18RtpSenderInterfaceE
> __ZN3rtc23RTCCertificateGeneratorC1EPNS_6ThreadES2_
> __ZN3rtc9KeyParams3RSAEii
>@@ -233,3 +232,9 @@ __ZN6webrtc26PeerConnectionDependenciesC1EPNS_22PeerConnectionObserverE
> __ZN6webrtc30PeerConnectionFactoryInterface20CreatePeerConnectionERKNS_23PeerConnectionInterface16RTCConfigurationENSt3__110unique_ptrIN7cricket13PortAllocatorENS5_14default_deleteIS8_EEEENS6_IN3rtc32RTCCertificateGeneratorInterfaceENS9_ISD_EEEEPNS_22PeerConnectionObserverE
> __ZN6webrtc29SetSessionDescriptionObserver9OnFailureERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEE
> __ZN6webrtc32CreateSessionDescriptionObserver9OnFailureERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEE
>+__ZN6webrtc13CryptoOptionsC1ERKS0_
>+__ZN6webrtc13CryptoOptionsD1Ev
>+__ZNK6webrtc20AudioSourceInterface7optionsEv
>+__ZTVN6webrtc20AudioSourceInterfaceE
>+__ZN6webrtc23PeerConnectionInterface21peer_connection_stateEv
>+__ZNK3rtc14NetworkManager16GetMdnsResponderEv
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>index 19d0fe6b201d96cbfd1c1c1d086740977107a6f0..e3126bf06fd9870aa9f3f99c4753ce1c80138c85 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>@@ -219,7 +219,6 @@ __ZNK6webrtc18RtpSenderInterface19init_send_encodingsEv
> __ZNK6webrtc18RtpSenderInterface17GetFrameEncryptorEv
> __ZN6webrtc18RtpSenderInterface17SetFrameEncryptorEN3rtc13scoped_refptrINS_23FrameEncryptorInterfaceEEE
> __ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEE
>-__ZNK3rtc14NetworkManager16GetMDnsResponderEv
> __ZTVN6webrtc18RtpSenderInterfaceE
> __ZN3rtc23RTCCertificateGeneratorC1EPNS_6ThreadES2_
> __ZN3rtc9KeyParams3RSAEii
>@@ -234,3 +233,9 @@ __ZN6webrtc26PeerConnectionDependenciesC1EPNS_22PeerConnectionObserverE
> __ZN6webrtc30PeerConnectionFactoryInterface20CreatePeerConnectionERKNS_23PeerConnectionInterface16RTCConfigurationENSt3__110unique_ptrIN7cricket13PortAllocatorENS5_14default_deleteIS8_EEEENS6_IN3rtc32RTCCertificateGeneratorInterfaceENS9_ISD_EEEEPNS_22PeerConnectionObserverE
> __ZN6webrtc29SetSessionDescriptionObserver9OnFailureERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEE
> __ZN6webrtc32CreateSessionDescriptionObserver9OnFailureERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEE
>+__ZN6webrtc13CryptoOptionsC1ERKS0_
>+__ZN6webrtc13CryptoOptionsD1Ev
>+__ZNK6webrtc20AudioSourceInterface7optionsEv
>+__ZTVN6webrtc20AudioSourceInterfaceE
>+__ZN6webrtc23PeerConnectionInterface21peer_connection_stateEv
>+__ZNK3rtc14NetworkManager16GetMdnsResponderEv
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>index 19d0fe6b201d96cbfd1c1c1d086740977107a6f0..e3126bf06fd9870aa9f3f99c4753ce1c80138c85 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>@@ -219,7 +219,6 @@ __ZNK6webrtc18RtpSenderInterface19init_send_encodingsEv
> __ZNK6webrtc18RtpSenderInterface17GetFrameEncryptorEv
> __ZN6webrtc18RtpSenderInterface17SetFrameEncryptorEN3rtc13scoped_refptrINS_23FrameEncryptorInterfaceEEE
> __ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEE
>-__ZNK3rtc14NetworkManager16GetMDnsResponderEv
> __ZTVN6webrtc18RtpSenderInterfaceE
> __ZN3rtc23RTCCertificateGeneratorC1EPNS_6ThreadES2_
> __ZN3rtc9KeyParams3RSAEii
>@@ -234,3 +233,9 @@ __ZN6webrtc26PeerConnectionDependenciesC1EPNS_22PeerConnectionObserverE
> __ZN6webrtc30PeerConnectionFactoryInterface20CreatePeerConnectionERKNS_23PeerConnectionInterface16RTCConfigurationENSt3__110unique_ptrIN7cricket13PortAllocatorENS5_14default_deleteIS8_EEEENS6_IN3rtc32RTCCertificateGeneratorInterfaceENS9_ISD_EEEEPNS_22PeerConnectionObserverE
> __ZN6webrtc29SetSessionDescriptionObserver9OnFailureERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEE
> __ZN6webrtc32CreateSessionDescriptionObserver9OnFailureERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEE
>+__ZN6webrtc13CryptoOptionsC1ERKS0_
>+__ZN6webrtc13CryptoOptionsD1Ev
>+__ZNK6webrtc20AudioSourceInterface7optionsEv
>+__ZTVN6webrtc20AudioSourceInterfaceE
>+__ZN6webrtc23PeerConnectionInterface21peer_connection_stateEv
>+__ZNK3rtc14NetworkManager16GetMdnsResponderEv
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>index 2dcf039079fa19427d09d80882b8a7c3ca15d988..57599fcdfc944e1cf1686fe635d14897a2d3b096 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>@@ -32,8 +32,8 @@ GCC_PREPROCESSOR_DEFINITIONS[sdk=macosx*] = $(inherited) WEBRTC_USE_VTB_HARDWARE
> GCC_PREPROCESSOR_DEFINITIONS[sdk=iphoneos*] = $(inherited) WEBRTC_IOS;
> GCC_PREPROCESSOR_DEFINITIONS[sdk=iphonesimulator*] = $(inherited) WEBRTC_IOS;
> 
>-EXCLUDED_SOURCE_FILE_NAMES[sdk=iphoneos*] = *_sse.cc *_sse2.cc macutils.cc macwindowpicker.cc audio_device_mac.cc audio_mixer_manager_mac.cc;
>-EXCLUDED_SOURCE_FILE_NAMES[sdk=iphonesimulator*] = macutils.cc macwindowpicker.cc audio_device_mac.cc audio_mixer_manager_mac.cc;
>+EXCLUDED_SOURCE_FILE_NAMES[sdk=iphoneos*] = *_sse.cc *_sse2.cc macutils.cc macwindowpicker.cc audio_device_mac.cc audio_mixer_manager_mac.cc logging_mac.mm;
>+EXCLUDED_SOURCE_FILE_NAMES[sdk=iphonesimulator*] = macutils.cc macwindowpicker.cc audio_device_mac.cc audio_mixer_manager_mac.cc logging_mac.mm;
> EXCLUDED_SOURCE_FILE_NAMES[sdk=macosx*] = voice_processing_audio_unit.mm;
> 
> OTHER_LDFLAGS[sdk=macosx10.13*] = $(inherited);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/AbseilHelpers.cmake b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/AbseilHelpers.cmake
>index e4eafe49e47906515e448a2def59f7629b2ba393..0c9341747512ceb6173541c7f4a2667a15b57727 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/AbseilHelpers.cmake
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/AbseilHelpers.cmake
>@@ -123,7 +123,7 @@ endfunction()
> #
> # all tests will be register for execution with add_test()
> #
>-# test compilation and execution is disable when BUILD_TESTING=OFF
>+# test compilation and execution is disable when ABSL_RUN_TESTS=OFF
> #
> function(absl_test)
> 
>@@ -135,7 +135,7 @@ function(absl_test)
>   )
> 
> 
>-  if(BUILD_TESTING)
>+  if(ABSL_RUN_TESTS)
> 
>     set(_NAME ${ABSL_TEST_TARGET})
>     string(TOUPPER ${_NAME} _UPPER_NAME)
>@@ -153,7 +153,7 @@ function(absl_test)
>     set_property(TARGET ${_NAME}_bin PROPERTY FOLDER ${ABSL_IDE_FOLDER})
> 
>     add_test(${_NAME} ${_NAME}_bin)
>-  endif(BUILD_TESTING)
>+  endif(ABSL_RUN_TESTS)
> 
> endfunction()
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMakeLists.txt
>index 89a3386f7e258efc9739b3517566c591f02ade7e..9a7e1031b2351104e3ec832114a8d6b75d5504c1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMakeLists.txt
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMakeLists.txt
>@@ -32,7 +32,7 @@ if (MSVC)
>   # /wd4244  conversion from 'type1' to 'type2'
>   # /wd4267  conversion from 'size_t' to 'type2'
>   # /wd4800  force value to bool 'true' or 'false' (performance warning)
>-  add_compile_options(/W3 /WX /wd4005 /wd4068 /wd4244 /wd4267 /wd4800)
>+  add_compile_options(/W3 /wd4005 /wd4068 /wd4244 /wd4267 /wd4800)
>   add_definitions(/DNOMINMAX /DWIN32_LEAN_AND_MEAN=1 /D_CRT_SECURE_NO_WARNINGS /D_SCL_SECURE_NO_WARNINGS)
> else()
>   set(ABSL_STD_CXX_FLAG "-std=c++11" CACHE STRING "c++ std flag (default: c++11)")
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/OWNERS b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/OWNERS
>index ceca4a8ba2cb16bd8d1d7833bedf78e6f09c93a1..38ebaf2295d1a616056650403abb89a09280dd16 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/OWNERS
>@@ -1,2 +1,4 @@
>+danilchap@chromium.org
>+kwiberg@chromium.org
> mbonadei@chromium.org
> phoglund@chromium.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/README.chromium b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/README.chromium
>index 91d6d381a47704450b599a009f049c707b1a9232..6026a2f4e20613edb53e168bfb438f20fe4c1fd9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/README.chromium
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/README.chromium
>@@ -4,7 +4,7 @@ URL: https://github.com/abseil/abseil-cpp
> License: Apache 2.0
> License File: LICENSE
> Version: 0
>-Revision: bea85b52733022294eef108a2e42d77b616ddca2
>+Revision: fb462224c058487763f263b7995d70efd0242c17
> Security Critical: yes
> 
> Description:
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/WORKSPACE b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/WORKSPACE
>index e4a911978dc6825695a28413bdfa2355accda27d..72ef13980c29263832cbb22f98f79ae859821ffe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/WORKSPACE
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/WORKSPACE
>@@ -5,11 +5,11 @@ load("@bazel_tools//tools/build_defs/repo:http.bzl", "http_archive")
> http_archive(
>   name = "bazel_toolchains",
>   urls = [
>-    "https://mirror.bazel.build/github.com/bazelbuild/bazel-toolchains/archive/287b64e0a211fb7c23b74695f8d5f5205b61f4eb.tar.gz",
>-    "https://github.com/bazelbuild/bazel-toolchains/archive/287b64e0a211fb7c23b74695f8d5f5205b61f4eb.tar.gz",
>+    "https://mirror.bazel.build/github.com/bazelbuild/bazel-toolchains/archive/bc09b995c137df042bb80a395b73d7ce6f26afbe.tar.gz",
>+    "https://github.com/bazelbuild/bazel-toolchains/archive/bc09b995c137df042bb80a395b73d7ce6f26afbe.tar.gz",
>   ],
>-  strip_prefix = "bazel-toolchains-287b64e0a211fb7c23b74695f8d5f5205b61f4eb",
>-  sha256 = "aca8ac6afd7745027ee4a43032b51a725a61a75a30f02cc58681ee87e4dcdf4b",
>+  strip_prefix = "bazel-toolchains-bc09b995c137df042bb80a395b73d7ce6f26afbe",
>+  sha256 = "4329663fe6c523425ad4d3c989a8ac026b04e1acedeceb56aa4b190fa7f3973c",
> )
> 
> # GoogleTest/GoogleMock framework. Used by most unit-tests.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/CMakeLists.txt
>index 689f64e258baac99201cfa7894dbcb80542408df..1d09b1935d84735d6dc90742a645b28850aeec76 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/CMakeLists.txt
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/CMakeLists.txt
>@@ -20,6 +20,7 @@ add_subdirectory(base)
> add_subdirectory(algorithm)
> add_subdirectory(container)
> add_subdirectory(debugging)
>+add_subdirectory(hash)
> add_subdirectory(memory)
> add_subdirectory(meta)
> add_subdirectory(numeric)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container.h
>index 6af8c09799e56f223ad3eeca246d83ab07063a19..53ab15686c6e10e6217ed47b0b4d7a333756807c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container.h
>@@ -494,7 +494,7 @@ BidirectionalIterator c_copy_backward(const C& src,
> // Container-based version of the <algorithm> `std::move()` function to move
> // a container's elements into an iterator.
> template <typename C, typename OutputIterator>
>-OutputIterator c_move(C& src, OutputIterator dest) {
>+OutputIterator c_move(C&& src, OutputIterator dest) {
>   return std::move(container_algorithm_internal::c_begin(src),
>                    container_algorithm_internal::c_end(src), dest);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container_test.cc
>index de66f14680f1b8f22d2e965d441a05efc58a6c03..1502b17f8636b0d5edec9fa4535c6e8b80d35075 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container_test.cc
>@@ -636,6 +636,21 @@ TEST(MutatingTest, Move) {
>                                 Pointee(5)));
> }
> 
>+TEST(MutatingTest, MoveWithRvalue) {
>+  auto MakeRValueSrc = [] {
>+    std::vector<std::unique_ptr<int>> src;
>+    src.emplace_back(absl::make_unique<int>(1));
>+    src.emplace_back(absl::make_unique<int>(2));
>+    src.emplace_back(absl::make_unique<int>(3));
>+    return src;
>+  };
>+
>+  std::vector<std::unique_ptr<int>> dest = MakeRValueSrc();
>+  absl::c_move(MakeRValueSrc(), std::back_inserter(dest));
>+  EXPECT_THAT(dest, ElementsAre(Pointee(1), Pointee(2), Pointee(3), Pointee(1),
>+                                Pointee(2), Pointee(3)));
>+}
>+
> TEST(MutatingTest, SwapRanges) {
>   std::vector<int> odds = {2, 4, 6};
>   std::vector<int> evens = {1, 3, 5};
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.bazel
>index 06d092ebdfa83abf924bb3ee032694259b2e4b2d..f7d810135b8ccc5fb05de4ab725d6c1cd8df3a46 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.bazel
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.bazel
>@@ -19,6 +19,7 @@ load(
>     "ABSL_DEFAULT_COPTS",
>     "ABSL_TEST_COPTS",
>     "ABSL_EXCEPTIONS_FLAG",
>+    "ABSL_EXCEPTIONS_FLAG_LINKOPTS",
> )
> 
> package(default_visibility = ["//visibility:public"])
>@@ -29,6 +30,7 @@ cc_library(
>     name = "spinlock_wait",
>     srcs = [
>         "internal/spinlock_akaros.inc",
>+        "internal/spinlock_linux.inc",
>         "internal/spinlock_posix.inc",
>         "internal/spinlock_wait.cc",
>         "internal/spinlock_win32.inc",
>@@ -179,6 +181,7 @@ cc_library(
>     srcs = ["internal/throw_delegate.cc"],
>     hdrs = ["internal/throw_delegate.h"],
>     copts = ABSL_DEFAULT_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     visibility = [
>         "//absl:__subpackages__",
>     ],
>@@ -193,6 +196,7 @@ cc_test(
>     name = "throw_delegate_test",
>     srcs = ["throw_delegate_test.cc"],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":throw_delegate",
>         "@com_google_googletest//:gtest_main",
>@@ -225,6 +229,7 @@ cc_library(
>     srcs = ["internal/exception_safety_testing.cc"],
>     hdrs = ["internal/exception_safety_testing.h"],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":base",
>         ":config",
>@@ -241,6 +246,7 @@ cc_test(
>     name = "exception_safety_testing_test",
>     srcs = ["exception_safety_testing_test.cc"],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":exception_safety_testing",
>         "//absl/memory",
>@@ -299,6 +305,7 @@ cc_test(
>     size = "medium",
>     srcs = ["spinlock_test_common.cc"],
>     copts = ABSL_TEST_COPTS,
>+    tags = ["no_test_wasm"],
>     deps = [
>         ":base",
>         ":core_headers",
>@@ -337,6 +344,9 @@ cc_test(
>     name = "config_test",
>     srcs = ["config_test.cc"],
>     copts = ABSL_TEST_COPTS,
>+    tags = [
>+        "no_test_wasm",
>+    ],
>     deps = [
>         ":config",
>         "//absl/synchronization:thread_pool",
>@@ -348,6 +358,9 @@ cc_test(
>     name = "call_once_test",
>     srcs = ["call_once_test.cc"],
>     copts = ABSL_TEST_COPTS,
>+    tags = [
>+        "no_test_wasm",
>+    ],
>     deps = [
>         ":base",
>         ":core_headers",
>@@ -401,6 +414,9 @@ cc_test(
>         "//absl:windows": [],
>         "//conditions:default": ["-pthread"],
>     }),
>+    tags = [
>+        "no_test_wasm",
>+    ],
>     deps = [
>         ":base",
>         ":core_headers",
>@@ -421,3 +437,23 @@ cc_test(
>         "@com_github_google_benchmark//:benchmark_main",
>     ],
> )
>+
>+cc_library(
>+    name = "bits",
>+    hdrs = ["internal/bits.h"],
>+    visibility = [
>+        "//absl:__subpackages__",
>+    ],
>+    deps = [":core_headers"],
>+)
>+
>+cc_test(
>+    name = "bits_test",
>+    size = "small",
>+    srcs = ["internal/bits_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":bits",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.gn
>index a714656462290bcd40cd4999ca1fef20c8637e7d..6c540f3b2bb29ca96e45696a1cb335ccb4a0b1bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.gn
>@@ -23,6 +23,7 @@ source_set("spinlock_wait") {
>   public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>   sources = [
>     "internal/spinlock_akaros.inc",
>+    "internal/spinlock_linux.inc",
>     "internal/spinlock_posix.inc",
>     "internal/spinlock_wait.cc",
>     "internal/spinlock_win32.inc",
>@@ -71,6 +72,7 @@ source_set("dynamic_annotations") {
>   public = [
>     "dynamic_annotations.h",
>   ]
>+
>   # Abseil's dynamic annotations are only visible inside Abseil because
>   # their usage is deprecated in Chromium (see README.chromium for more info).
>   visibility = []
>@@ -296,3 +298,19 @@ source_set("endian") {
>     ":core_headers",
>   ]
> }
>+
>+source_set("bits") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public = [
>+    "internal/bits.h",
>+  ]
>+  deps = [
>+    ":core_headers",
>+  ]
>+  visibility = []
>+  visibility += [ "../*" ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/CMakeLists.txt
>index 01d2af085f58e15f3e2639c516cf66d46d121bed..04a6eb31955b602202ad597ba2b34ec04bc7e557 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/CMakeLists.txt
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/CMakeLists.txt
>@@ -31,6 +31,7 @@ list(APPEND BASE_PUBLIC_HEADERS
> 
> list(APPEND BASE_INTERNAL_HEADERS
>   "internal/atomic_hook.h"
>+  "internal/bits.h"
>   "internal/cycleclock.h"
>   "internal/direct_mmap.h"
>   "internal/endian.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/attributes.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/attributes.h
>index b1883b6d752cc15cd76e079240d0857ce18e8d0a..e8500224443bce5bf65ba5ce1fcd00d8c7044c8f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/attributes.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/attributes.h
>@@ -100,7 +100,7 @@
> // ABSL_PRINTF_ATTRIBUTE
> // ABSL_SCANF_ATTRIBUTE
> //
>-// Tells the compiler to perform `printf` format std::string checking if the
>+// Tells the compiler to perform `printf` format string checking if the
> // compiler supports it; see the 'format' attribute in
> // <http://gcc.gnu.org/onlinedocs/gcc-4.7.0/gcc/Function-Attributes.html>.
> //
>@@ -494,14 +494,27 @@
> #define ABSL_XRAY_LOG_ARGS(N)
> #endif
> 
>+// ABSL_ATTRIBUTE_REINITIALIZES
>+//
>+// Indicates that a member function reinitializes the entire object to a known
>+// state, independent of the previous state of the object.
>+//
>+// The clang-tidy check bugprone-use-after-move allows member functions marked
>+// with this attribute to be called on objects that have been moved from;
>+// without the attribute, this would result in a use-after-move warning.
>+#if ABSL_HAVE_CPP_ATTRIBUTE(clang::reinitializes)
>+#define ABSL_ATTRIBUTE_REINITIALIZES [[clang::reinitializes]]
>+#else
>+#define ABSL_ATTRIBUTE_REINITIALIZES
>+#endif
>+
> // -----------------------------------------------------------------------------
> // Variable Attributes
> // -----------------------------------------------------------------------------
> 
> // ABSL_ATTRIBUTE_UNUSED
> //
>-// Prevents the compiler from complaining about or optimizing away variables
>-// that appear unused.
>+// Prevents the compiler from complaining about variables that appear unused.
> #if ABSL_HAVE_ATTRIBUTE(unused) || (defined(__GNUC__) && !defined(__clang__))
> #undef ABSL_ATTRIBUTE_UNUSED
> #define ABSL_ATTRIBUTE_UNUSED __attribute__((__unused__))
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/config.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/config.h
>index 6890e313bf27af7e5e8a5c051325013e27215a0e..d4eb7d0cbd76b718f428979dce755d735afcb00d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/config.h
>@@ -199,7 +199,7 @@
> #define ABSL_HAVE_INTRINSIC_INT128 1
> #elif defined(__CUDACC__)
> // __CUDACC_VER__ is a full version number before CUDA 9, and is defined to a
>-// std::string explaining that it has been removed starting with CUDA 9. We use
>+// string explaining that it has been removed starting with CUDA 9. We use
> // nested #ifs because there is no short-circuiting in the preprocessor.
> // NOTE: `__CUDACC__` could be undefined while `__CUDACC_VER__` is defined.
> #if __CUDACC_VER__ >= 70000
>@@ -414,14 +414,13 @@
> // <string_view>, <variant> is implemented) or higher. Also, `__cplusplus` is
> // not correctly set by MSVC, so we use `_MSVC_LANG` to check the language
> // version.
>-// TODO(zhangxy): fix tests before enabling aliasing for `std::any`,
>-// `std::string_view`.
>+// TODO(zhangxy): fix tests before enabling aliasing for `std::any`.
> #if defined(_MSC_VER) && _MSC_VER >= 1910 && \
>     ((defined(_MSVC_LANG) && _MSVC_LANG > 201402) || __cplusplus > 201402)
> // #define ABSL_HAVE_STD_ANY 1
> #define ABSL_HAVE_STD_OPTIONAL 1
> #define ABSL_HAVE_STD_VARIANT 1
>-// #define ABSL_HAVE_STD_STRING_VIEW 1
>+#define ABSL_HAVE_STD_STRING_VIEW 1
> #endif
> 
> #endif  // ABSL_BASE_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/exception_safety_testing_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/exception_safety_testing_test.cc
>index 97c8d6f831894a880a984a1d0e090e04877282ee..106bc34b00f7ce27a7b43752ad710699fbc455f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/exception_safety_testing_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/exception_safety_testing_test.cc
>@@ -38,7 +38,7 @@ template <typename F>
> void ExpectNoThrow(const F& f) {
>   try {
>     f();
>-  } catch (TestException e) {
>+  } catch (const TestException& e) {
>     ADD_FAILURE() << "Unexpected exception thrown from " << e.what();
>   }
> }
>@@ -179,7 +179,7 @@ TEST(ThrowingValueTest, ThrowingStreamOps) {
> }
> 
> // Tests the operator<< of ThrowingValue by forcing ConstructorTracker to emit
>-// a nonfatal failure that contains the std::string representation of the Thrower
>+// a nonfatal failure that contains the string representation of the Thrower
> TEST(ThrowingValueTest, StreamOpsOutput) {
>   using ::testing::TypeSpec;
>   exceptions_internal::ConstructorTracker ct(exceptions_internal::countdown);
>@@ -548,21 +548,21 @@ TEST(ExceptionSafetyTesterTest, IncompleteTypesAreNotTestable) {
>   // Test that providing operation and inveriants still does not allow for the
>   // the invocation of .Test() and .Test(op) because it lacks a factory
>   auto without_fac =
>-      testing::MakeExceptionSafetyTester().WithOperation(op).WithInvariants(
>+      testing::MakeExceptionSafetyTester().WithOperation(op).WithContracts(
>           inv, testing::strong_guarantee);
>   EXPECT_FALSE(HasNullaryTest(without_fac));
>   EXPECT_FALSE(HasUnaryTest(without_fac));
> 
>-  // Test that providing invariants and factory allows the invocation of
>+  // Test that providing contracts and factory allows the invocation of
>   // .Test(op) but does not allow for .Test() because it lacks an operation
>   auto without_op = testing::MakeExceptionSafetyTester()
>-                        .WithInvariants(inv, testing::strong_guarantee)
>+                        .WithContracts(inv, testing::strong_guarantee)
>                         .WithFactory(fac);
>   EXPECT_FALSE(HasNullaryTest(without_op));
>   EXPECT_TRUE(HasUnaryTest(without_op));
> 
>   // Test that providing operation and factory still does not allow for the
>-  // the invocation of .Test() and .Test(op) because it lacks invariants
>+  // the invocation of .Test() and .Test(op) because it lacks contracts
>   auto without_inv =
>       testing::MakeExceptionSafetyTester().WithOperation(op).WithFactory(fac);
>   EXPECT_FALSE(HasNullaryTest(without_inv));
>@@ -577,7 +577,7 @@ std::unique_ptr<ExampleStruct> ExampleFunctionFactory() {
> 
> void ExampleFunctionOperation(ExampleStruct*) {}
> 
>-testing::AssertionResult ExampleFunctionInvariant(ExampleStruct*) {
>+testing::AssertionResult ExampleFunctionContract(ExampleStruct*) {
>   return testing::AssertionSuccess();
> }
> 
>@@ -593,16 +593,16 @@ struct {
> 
> struct {
>   testing::AssertionResult operator()(ExampleStruct* example_struct) const {
>-    return ExampleFunctionInvariant(example_struct);
>+    return ExampleFunctionContract(example_struct);
>   }
>-} example_struct_invariant;
>+} example_struct_contract;
> 
> auto example_lambda_factory = []() { return ExampleFunctionFactory(); };
> 
> auto example_lambda_operation = [](ExampleStruct*) {};
> 
>-auto example_lambda_invariant = [](ExampleStruct* example_struct) {
>-  return ExampleFunctionInvariant(example_struct);
>+auto example_lambda_contract = [](ExampleStruct* example_struct) {
>+  return ExampleFunctionContract(example_struct);
> };
> 
> // Testing that function references, pointers, structs with operator() and
>@@ -612,28 +612,28 @@ TEST(ExceptionSafetyTesterTest, MixedFunctionTypes) {
>   EXPECT_TRUE(testing::MakeExceptionSafetyTester()
>                   .WithFactory(ExampleFunctionFactory)
>                   .WithOperation(ExampleFunctionOperation)
>-                  .WithInvariants(ExampleFunctionInvariant)
>+                  .WithContracts(ExampleFunctionContract)
>                   .Test());
> 
>   // function pointer
>   EXPECT_TRUE(testing::MakeExceptionSafetyTester()
>                   .WithFactory(&ExampleFunctionFactory)
>                   .WithOperation(&ExampleFunctionOperation)
>-                  .WithInvariants(&ExampleFunctionInvariant)
>+                  .WithContracts(&ExampleFunctionContract)
>                   .Test());
> 
>   // struct
>   EXPECT_TRUE(testing::MakeExceptionSafetyTester()
>                   .WithFactory(example_struct_factory)
>                   .WithOperation(example_struct_operation)
>-                  .WithInvariants(example_struct_invariant)
>+                  .WithContracts(example_struct_contract)
>                   .Test());
> 
>   // lambda
>   EXPECT_TRUE(testing::MakeExceptionSafetyTester()
>                   .WithFactory(example_lambda_factory)
>                   .WithOperation(example_lambda_operation)
>-                  .WithInvariants(example_lambda_invariant)
>+                  .WithContracts(example_lambda_contract)
>                   .Test());
> }
> 
>@@ -658,9 +658,9 @@ struct {
> } invoker;
> 
> auto tester =
>-    testing::MakeExceptionSafetyTester().WithOperation(invoker).WithInvariants(
>+    testing::MakeExceptionSafetyTester().WithOperation(invoker).WithContracts(
>         CheckNonNegativeInvariants);
>-auto strong_tester = tester.WithInvariants(testing::strong_guarantee);
>+auto strong_tester = tester.WithContracts(testing::strong_guarantee);
> 
> struct FailsBasicGuarantee : public NonNegative {
>   void operator()() {
>@@ -690,7 +690,7 @@ TEST(ExceptionCheckTest, StrongGuaranteeFailure) {
>   EXPECT_FALSE(strong_tester.WithInitialValue(FollowsBasicGuarantee{}).Test());
> }
> 
>-struct BasicGuaranteeWithExtraInvariants : public NonNegative {
>+struct BasicGuaranteeWithExtraContracts : public NonNegative {
>   // After operator(), i is incremented.  If operator() throws, i is set to 9999
>   void operator()() {
>     int old_i = i;
>@@ -701,21 +701,21 @@ struct BasicGuaranteeWithExtraInvariants : public NonNegative {
> 
>   static constexpr int kExceptionSentinel = 9999;
> };
>-constexpr int BasicGuaranteeWithExtraInvariants::kExceptionSentinel;
>+constexpr int BasicGuaranteeWithExtraContracts::kExceptionSentinel;
> 
>-TEST(ExceptionCheckTest, BasicGuaranteeWithExtraInvariants) {
>+TEST(ExceptionCheckTest, BasicGuaranteeWithExtraContracts) {
>   auto tester_with_val =
>-      tester.WithInitialValue(BasicGuaranteeWithExtraInvariants{});
>+      tester.WithInitialValue(BasicGuaranteeWithExtraContracts{});
>   EXPECT_TRUE(tester_with_val.Test());
>   EXPECT_TRUE(
>       tester_with_val
>-          .WithInvariants([](BasicGuaranteeWithExtraInvariants* o) {
>-            if (o->i == BasicGuaranteeWithExtraInvariants::kExceptionSentinel) {
>+          .WithContracts([](BasicGuaranteeWithExtraContracts* o) {
>+            if (o->i == BasicGuaranteeWithExtraContracts::kExceptionSentinel) {
>               return testing::AssertionSuccess();
>             }
>             return testing::AssertionFailure()
>                    << "i should be "
>-                   << BasicGuaranteeWithExtraInvariants::kExceptionSentinel
>+                   << BasicGuaranteeWithExtraContracts::kExceptionSentinel
>                    << ", but is " << o->i;
>           })
>           .Test());
>@@ -740,7 +740,7 @@ struct HasReset : public NonNegative {
>   void reset() { i = 0; }
> };
> 
>-testing::AssertionResult CheckHasResetInvariants(HasReset* h) {
>+testing::AssertionResult CheckHasResetContracts(HasReset* h) {
>   h->reset();
>   return testing::AssertionResult(h->i == 0);
> }
>@@ -759,14 +759,14 @@ TEST(ExceptionCheckTest, ModifyingChecker) {
>   };
> 
>   EXPECT_FALSE(tester.WithInitialValue(FollowsBasicGuarantee{})
>-                   .WithInvariants(set_to_1000, is_1000)
>+                   .WithContracts(set_to_1000, is_1000)
>                    .Test());
>   EXPECT_TRUE(strong_tester.WithInitialValue(FollowsStrongGuarantee{})
>-                  .WithInvariants(increment)
>+                  .WithContracts(increment)
>                   .Test());
>   EXPECT_TRUE(testing::MakeExceptionSafetyTester()
>                   .WithInitialValue(HasReset{})
>-                  .WithInvariants(CheckHasResetInvariants)
>+                  .WithContracts(CheckHasResetContracts)
>                   .Test(invoker));
> }
> 
>@@ -799,7 +799,7 @@ TEST(ExceptionCheckTest, NonEqualityComparable) {
>     return testing::AssertionResult(nec->i == NonEqualityComparable().i);
>   };
>   auto strong_nec_tester = tester.WithInitialValue(NonEqualityComparable{})
>-                               .WithInvariants(nec_is_strong);
>+                               .WithContracts(nec_is_strong);
> 
>   EXPECT_TRUE(strong_nec_tester.Test());
>   EXPECT_FALSE(strong_nec_tester.Test(
>@@ -833,14 +833,14 @@ struct {
>   testing::AssertionResult operator()(ExhaustivenessTester<T>*) const {
>     return testing::AssertionSuccess();
>   }
>-} CheckExhaustivenessTesterInvariants;
>+} CheckExhaustivenessTesterContracts;
> 
> template <typename T>
> unsigned char ExhaustivenessTester<T>::successes = 0;
> 
> TEST(ExceptionCheckTest, Exhaustiveness) {
>   auto exhaust_tester = testing::MakeExceptionSafetyTester()
>-                            .WithInvariants(CheckExhaustivenessTesterInvariants)
>+                            .WithContracts(CheckExhaustivenessTesterContracts)
>                             .WithOperation(invoker);
> 
>   EXPECT_TRUE(
>@@ -849,7 +849,7 @@ TEST(ExceptionCheckTest, Exhaustiveness) {
> 
>   EXPECT_TRUE(
>       exhaust_tester.WithInitialValue(ExhaustivenessTester<ThrowingValue<>>{})
>-          .WithInvariants(testing::strong_guarantee)
>+          .WithContracts(testing::strong_guarantee)
>           .Test());
>   EXPECT_EQ(ExhaustivenessTester<ThrowingValue<>>::successes, 0xF);
> }
>@@ -931,8 +931,8 @@ TEST(ThrowingValueTraitsTest, RelationalOperators) {
> }
> 
> TEST(ThrowingAllocatorTraitsTest, Assignablility) {
>-  EXPECT_TRUE(std::is_move_assignable<ThrowingAllocator<int>>::value);
>-  EXPECT_TRUE(std::is_copy_assignable<ThrowingAllocator<int>>::value);
>+  EXPECT_TRUE(absl::is_move_assignable<ThrowingAllocator<int>>::value);
>+  EXPECT_TRUE(absl::is_copy_assignable<ThrowingAllocator<int>>::value);
>   EXPECT_TRUE(std::is_nothrow_move_assignable<ThrowingAllocator<int>>::value);
>   EXPECT_TRUE(std::is_nothrow_copy_assignable<ThrowingAllocator<int>>::value);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/bits.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/bits.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..bc7faaee3b257edc37cd9090b7777d7035d3c3b8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/bits.h
>@@ -0,0 +1,193 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_BASE_INTERNAL_BITS_H_
>+#define ABSL_BASE_INTERNAL_BITS_H_
>+
>+// This file contains bitwise ops which are implementation details of various
>+// absl libraries.
>+
>+#include <cstdint>
>+
>+// Clang on Windows has __builtin_clzll; otherwise we need to use the
>+// windows intrinsic functions.
>+#if defined(_MSC_VER)
>+#include <intrin.h>
>+#if defined(_M_X64)
>+#pragma intrinsic(_BitScanReverse64)
>+#pragma intrinsic(_BitScanForward64)
>+#endif
>+#pragma intrinsic(_BitScanReverse)
>+#pragma intrinsic(_BitScanForward)
>+#endif
>+
>+#include "absl/base/attributes.h"
>+
>+#if defined(_MSC_VER)
>+// We can achieve something similar to attribute((always_inline)) with MSVC by
>+// using the __forceinline keyword, however this is not perfect. MSVC is
>+// much less aggressive about inlining, and even with the __forceinline keyword.
>+#define ABSL_BASE_INTERNAL_FORCEINLINE __forceinline
>+#else
>+// Use default attribute inline.
>+#define ABSL_BASE_INTERNAL_FORCEINLINE inline ABSL_ATTRIBUTE_ALWAYS_INLINE
>+#endif
>+
>+
>+namespace absl {
>+namespace base_internal {
>+
>+ABSL_BASE_INTERNAL_FORCEINLINE int CountLeadingZeros64Slow(uint64_t n) {
>+  int zeroes = 60;
>+  if (n >> 32) zeroes -= 32, n >>= 32;
>+  if (n >> 16) zeroes -= 16, n >>= 16;
>+  if (n >> 8) zeroes -= 8, n >>= 8;
>+  if (n >> 4) zeroes -= 4, n >>= 4;
>+  return "\4\3\2\2\1\1\1\1\0\0\0\0\0\0\0"[n] + zeroes;
>+}
>+
>+ABSL_BASE_INTERNAL_FORCEINLINE int CountLeadingZeros64(uint64_t n) {
>+#if defined(_MSC_VER) && defined(_M_X64)
>+  // MSVC does not have __buitin_clzll. Use _BitScanReverse64.
>+  unsigned long result = 0;  // NOLINT(runtime/int)
>+  if (_BitScanReverse64(&result, n)) {
>+    return 63 - result;
>+  }
>+  return 64;
>+#elif defined(_MSC_VER)
>+  // MSVC does not have __buitin_clzll. Compose two calls to _BitScanReverse
>+  unsigned long result = 0;  // NOLINT(runtime/int)
>+  if ((n >> 32) && _BitScanReverse(&result, n >> 32)) {
>+    return 31 - result;
>+  }
>+  if (_BitScanReverse(&result, n)) {
>+    return 63 - result;
>+  }
>+  return 64;
>+#elif defined(__GNUC__)
>+  // Use __builtin_clzll, which uses the following instructions:
>+  //  x86: bsr
>+  //  ARM64: clz
>+  //  PPC: cntlzd
>+  static_assert(sizeof(unsigned long long) == sizeof(n),  // NOLINT(runtime/int)
>+                "__builtin_clzll does not take 64-bit arg");
>+
>+  // Handle 0 as a special case because __builtin_clzll(0) is undefined.
>+  if (n == 0) {
>+    return 64;
>+  }
>+  return __builtin_clzll(n);
>+#else
>+  return CountLeadingZeros64Slow(n);
>+#endif
>+}
>+
>+ABSL_BASE_INTERNAL_FORCEINLINE int CountLeadingZeros32Slow(uint64_t n) {
>+  int zeroes = 28;
>+  if (n >> 16) zeroes -= 16, n >>= 16;
>+  if (n >> 8) zeroes -= 8, n >>= 8;
>+  if (n >> 4) zeroes -= 4, n >>= 4;
>+  return "\4\3\2\2\1\1\1\1\0\0\0\0\0\0\0"[n] + zeroes;
>+}
>+
>+ABSL_BASE_INTERNAL_FORCEINLINE int CountLeadingZeros32(uint32_t n) {
>+#if defined(_MSC_VER)
>+  unsigned long result = 0;  // NOLINT(runtime/int)
>+  if (_BitScanReverse(&result, n)) {
>+    return 31 - result;
>+  }
>+  return 32;
>+#elif defined(__GNUC__)
>+  // Use __builtin_clz, which uses the following instructions:
>+  //  x86: bsr
>+  //  ARM64: clz
>+  //  PPC: cntlzd
>+  static_assert(sizeof(int) == sizeof(n),
>+                "__builtin_clz does not take 32-bit arg");
>+
>+  // Handle 0 as a special case because __builtin_clz(0) is undefined.
>+  if (n == 0) {
>+    return 32;
>+  }
>+  return __builtin_clz(n);
>+#else
>+  return CountLeadingZeros32Slow(n);
>+#endif
>+}
>+
>+ABSL_BASE_INTERNAL_FORCEINLINE int CountTrailingZerosNonZero64Slow(uint64_t n) {
>+  int c = 63;
>+  n &= ~n + 1;
>+  if (n & 0x00000000FFFFFFFF) c -= 32;
>+  if (n & 0x0000FFFF0000FFFF) c -= 16;
>+  if (n & 0x00FF00FF00FF00FF) c -= 8;
>+  if (n & 0x0F0F0F0F0F0F0F0F) c -= 4;
>+  if (n & 0x3333333333333333) c -= 2;
>+  if (n & 0x5555555555555555) c -= 1;
>+  return c;
>+}
>+
>+ABSL_BASE_INTERNAL_FORCEINLINE int CountTrailingZerosNonZero64(uint64_t n) {
>+#if defined(_MSC_VER) && defined(_M_X64)
>+  unsigned long result = 0;  // NOLINT(runtime/int)
>+  _BitScanForward64(&result, n);
>+  return result;
>+#elif defined(_MSC_VER)
>+  unsigned long result = 0;  // NOLINT(runtime/int)
>+  if (static_cast<uint32_t>(n) == 0) {
>+    _BitScanForward(&result, n >> 32);
>+    return result + 32;
>+  }
>+  _BitScanForward(&result, n);
>+  return result;
>+#elif defined(__GNUC__)
>+  static_assert(sizeof(unsigned long long) == sizeof(n),  // NOLINT(runtime/int)
>+                "__builtin_ctzll does not take 64-bit arg");
>+  return __builtin_ctzll(n);
>+#else
>+  return CountTrailingZerosNonZero64Slow(n);
>+#endif
>+}
>+
>+ABSL_BASE_INTERNAL_FORCEINLINE int CountTrailingZerosNonZero32Slow(uint32_t n) {
>+  int c = 31;
>+  n &= ~n + 1;
>+  if (n & 0x0000FFFF) c -= 16;
>+  if (n & 0x00FF00FF) c -= 8;
>+  if (n & 0x0F0F0F0F) c -= 4;
>+  if (n & 0x33333333) c -= 2;
>+  if (n & 0x55555555) c -= 1;
>+  return c;
>+}
>+
>+ABSL_BASE_INTERNAL_FORCEINLINE int CountTrailingZerosNonZero32(uint32_t n) {
>+#if defined(_MSC_VER)
>+  unsigned long result = 0;  // NOLINT(runtime/int)
>+  _BitScanForward(&result, n);
>+  return result;
>+#elif defined(__GNUC__)
>+  static_assert(sizeof(int) == sizeof(n),
>+                "__builtin_ctz does not take 32-bit arg");
>+  return __builtin_ctz(n);
>+#else
>+  return CountTrailingZerosNonZero32Slow(n);
>+#endif
>+}
>+
>+#undef ABSL_BASE_INTERNAL_FORCEINLINE
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_BITS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/bits_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/bits_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..e5d991d6725874aa9db803b15117817aa3086b51
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/bits_test.cc
>@@ -0,0 +1,97 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/bits.h"
>+
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+int CLZ64(uint64_t n) {
>+  int fast = absl::base_internal::CountLeadingZeros64(n);
>+  int slow = absl::base_internal::CountLeadingZeros64Slow(n);
>+  EXPECT_EQ(fast, slow) << n;
>+  return fast;
>+}
>+
>+TEST(BitsTest, CountLeadingZeros64) {
>+  EXPECT_EQ(64, CLZ64(uint64_t{}));
>+  EXPECT_EQ(0, CLZ64(~uint64_t{}));
>+
>+  for (int index = 0; index < 64; index++) {
>+    uint64_t x = static_cast<uint64_t>(1) << index;
>+    const auto cnt = 63 - index;
>+    ASSERT_EQ(cnt, CLZ64(x)) << index;
>+    ASSERT_EQ(cnt, CLZ64(x + x - 1)) << index;
>+  }
>+}
>+
>+int CLZ32(uint32_t n) {
>+  int fast = absl::base_internal::CountLeadingZeros32(n);
>+  int slow = absl::base_internal::CountLeadingZeros32Slow(n);
>+  EXPECT_EQ(fast, slow) << n;
>+  return fast;
>+}
>+
>+TEST(BitsTest, CountLeadingZeros32) {
>+  EXPECT_EQ(32, CLZ32(uint32_t{}));
>+  EXPECT_EQ(0, CLZ32(~uint32_t{}));
>+
>+  for (int index = 0; index < 32; index++) {
>+    uint32_t x = static_cast<uint32_t>(1) << index;
>+    const auto cnt = 31 - index;
>+    ASSERT_EQ(cnt, CLZ32(x)) << index;
>+    ASSERT_EQ(cnt, CLZ32(x + x - 1)) << index;
>+    ASSERT_EQ(CLZ64(x), CLZ32(x) + 32);
>+  }
>+}
>+
>+int CTZ64(uint64_t n) {
>+  int fast = absl::base_internal::CountTrailingZerosNonZero64(n);
>+  int slow = absl::base_internal::CountTrailingZerosNonZero64Slow(n);
>+  EXPECT_EQ(fast, slow) << n;
>+  return fast;
>+}
>+
>+TEST(BitsTest, CountTrailingZerosNonZero64) {
>+  EXPECT_EQ(0, CTZ64(~uint64_t{}));
>+
>+  for (int index = 0; index < 64; index++) {
>+    uint64_t x = static_cast<uint64_t>(1) << index;
>+    const auto cnt = index;
>+    ASSERT_EQ(cnt, CTZ64(x)) << index;
>+    ASSERT_EQ(cnt, CTZ64(~(x - 1))) << index;
>+  }
>+}
>+
>+int CTZ32(uint32_t n) {
>+  int fast = absl::base_internal::CountTrailingZerosNonZero32(n);
>+  int slow = absl::base_internal::CountTrailingZerosNonZero32Slow(n);
>+  EXPECT_EQ(fast, slow) << n;
>+  return fast;
>+}
>+
>+TEST(BitsTest, CountTrailingZerosNonZero32) {
>+  EXPECT_EQ(0, CTZ32(~uint32_t{}));
>+
>+  for (int index = 0; index < 32; index++) {
>+    uint32_t x = static_cast<uint32_t>(1) << index;
>+    const auto cnt = index;
>+    ASSERT_EQ(cnt, CTZ32(x)) << index;
>+    ASSERT_EQ(cnt, CTZ32(~(x - 1))) << index;
>+  }
>+}
>+
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_safety_testing.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_safety_testing.h
>index 8c2f5093fc4da828afa82342be44995d18a3bbe7..5665a1b67db2ec4ca2ad80ffa06fe6e9cfc491b9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_safety_testing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_safety_testing.h
>@@ -191,19 +191,19 @@ class TrackedObject {
>   ~TrackedObject() noexcept { ConstructorTracker::ObjectDestructed(this); }
> };
> 
>-template <typename Factory, typename Operation, typename Invariant>
>-absl::optional<testing::AssertionResult> TestSingleInvariantAtCountdownImpl(
>+template <typename Factory, typename Operation, typename Contract>
>+absl::optional<testing::AssertionResult> TestSingleContractAtCountdownImpl(
>     const Factory& factory, const Operation& operation, int count,
>-    const Invariant& invariant) {
>+    const Contract& contract) {
>   auto t_ptr = factory();
>   absl::optional<testing::AssertionResult> current_res;
>   SetCountdown(count);
>   try {
>     operation(t_ptr.get());
>   } catch (const exceptions_internal::TestException& e) {
>-    current_res.emplace(invariant(t_ptr.get()));
>+    current_res.emplace(contract(t_ptr.get()));
>     if (!current_res.value()) {
>-      *current_res << e.what() << " failed invariant check";
>+      *current_res << e.what() << " failed contract check";
>     }
>   }
>   UnsetCountdown();
>@@ -211,22 +211,22 @@ absl::optional<testing::AssertionResult> TestSingleInvariantAtCountdownImpl(
> }
> 
> template <typename Factory, typename Operation>
>-absl::optional<testing::AssertionResult> TestSingleInvariantAtCountdownImpl(
>+absl::optional<testing::AssertionResult> TestSingleContractAtCountdownImpl(
>     const Factory& factory, const Operation& operation, int count,
>     StrongGuaranteeTagType) {
>   using TPtr = typename decltype(factory())::pointer;
>   auto t_is_strong = [&](TPtr t) { return *t == *factory(); };
>-  return TestSingleInvariantAtCountdownImpl(factory, operation, count,
>-                                            t_is_strong);
>+  return TestSingleContractAtCountdownImpl(factory, operation, count,
>+                                           t_is_strong);
> }
> 
>-template <typename Factory, typename Operation, typename Invariant>
>-int TestSingleInvariantAtCountdown(
>+template <typename Factory, typename Operation, typename Contract>
>+int TestSingleContractAtCountdown(
>     const Factory& factory, const Operation& operation, int count,
>-    const Invariant& invariant,
>+    const Contract& contract,
>     absl::optional<testing::AssertionResult>* reduced_res) {
>   // If reduced_res is empty, it means the current call to
>-  // TestSingleInvariantAtCountdown(...) is the first test being run so we do
>+  // TestSingleContractAtCountdown(...) is the first test being run so we do
>   // want to run it. Alternatively, if it's not empty (meaning a previous test
>   // has run) we want to check if it passed. If the previous test did pass, we
>   // want to contine running tests so we do want to run the current one. If it
>@@ -234,22 +234,22 @@ int TestSingleInvariantAtCountdown(
>   // output. If that's the case, we do not run the current test and instead we
>   // simply return.
>   if (!reduced_res->has_value() || reduced_res->value()) {
>-    *reduced_res = TestSingleInvariantAtCountdownImpl(factory, operation, count,
>-                                                      invariant);
>+    *reduced_res =
>+        TestSingleContractAtCountdownImpl(factory, operation, count, contract);
>   }
>   return 0;
> }
> 
>-template <typename Factory, typename Operation, typename... Invariants>
>-inline absl::optional<testing::AssertionResult> TestAllInvariantsAtCountdown(
>+template <typename Factory, typename Operation, typename... Contracts>
>+inline absl::optional<testing::AssertionResult> TestAllContractsAtCountdown(
>     const Factory& factory, const Operation& operation, int count,
>-    const Invariants&... invariants) {
>+    const Contracts&... contracts) {
>   absl::optional<testing::AssertionResult> reduced_res;
> 
>   // Run each checker, short circuiting after the first failure
>   int dummy[] = {
>-      0, (TestSingleInvariantAtCountdown(factory, operation, count, invariants,
>-                                         &reduced_res))...};
>+      0, (TestSingleContractAtCountdown(factory, operation, count, contracts,
>+                                        &reduced_res))...};
>   static_cast<void>(dummy);
>   return reduced_res;
> }
>@@ -858,7 +858,7 @@ testing::AssertionResult TestNothrowOp(const Operation& operation) {
>   try {
>     operation();
>     return testing::AssertionSuccess();
>-  } catch (exceptions_internal::TestException) {
>+  } catch (const exceptions_internal::TestException&) {
>     return testing::AssertionFailure()
>            << "TestException thrown during call to operation() when nothrow "
>               "guarantee was expected.";
>@@ -884,15 +884,15 @@ class DefaultFactory {
>   T t_;
> };
> 
>-template <size_t LazyInvariantsCount, typename LazyFactory,
>+template <size_t LazyContractsCount, typename LazyFactory,
>           typename LazyOperation>
> using EnableIfTestable = typename absl::enable_if_t<
>-    LazyInvariantsCount != 0 &&
>+    LazyContractsCount != 0 &&
>     !std::is_same<LazyFactory, UninitializedT>::value &&
>     !std::is_same<LazyOperation, UninitializedT>::value>;
> 
> template <typename Factory = UninitializedT,
>-          typename Operation = UninitializedT, typename... Invariants>
>+          typename Operation = UninitializedT, typename... Contracts>
> class ExceptionSafetyTester;
> 
> }  // namespace exceptions_internal
>@@ -903,7 +903,7 @@ namespace exceptions_internal {
> 
> /*
>  * Builds a tester object that tests if performing a operation on a T follows
>- * exception safety guarantees. Verification is done via invariant assertion
>+ * exception safety guarantees. Verification is done via contract assertion
>  * callbacks applied to T instances post-throw.
>  *
>  * Template parameters for ExceptionSafetyTester:
>@@ -921,18 +921,18 @@ namespace exceptions_internal {
>  *   fresh T instance so it's free to modify and destroy the T instances as it
>  *   pleases.
>  *
>- * - Invariants...: The invariant assertion callback objects (passed in via
>- *   tester.WithInvariants(...)) must be invocable with the signature
>+ * - Contracts...: The contract assertion callback objects (passed in via
>+ *   tester.WithContracts(...)) must be invocable with the signature
>  *   `testing::AssertionResult operator()(T*) const` where T is the type being
>- *   tested. Invariant assertion callbacks are provided T instances post-throw.
>- *   They must return testing::AssertionSuccess when the type invariants of the
>- *   provided T instance hold. If the type invariants of the T instance do not
>+ *   tested. Contract assertion callbacks are provided T instances post-throw.
>+ *   They must return testing::AssertionSuccess when the type contracts of the
>+ *   provided T instance hold. If the type contracts of the T instance do not
>  *   hold, they must return testing::AssertionFailure. Execution order of
>- *   Invariants... is unspecified. They will each individually get a fresh T
>+ *   Contracts... is unspecified. They will each individually get a fresh T
>  *   instance so they are free to modify and destroy the T instances as they
>  *   please.
>  */
>-template <typename Factory, typename Operation, typename... Invariants>
>+template <typename Factory, typename Operation, typename... Contracts>
> class ExceptionSafetyTester {
>  public:
>   /*
>@@ -948,7 +948,7 @@ class ExceptionSafetyTester {
>    *   tester.WithFactory(...).
>    */
>   template <typename T>
>-  ExceptionSafetyTester<DefaultFactory<T>, Operation, Invariants...>
>+  ExceptionSafetyTester<DefaultFactory<T>, Operation, Contracts...>
>   WithInitialValue(const T& t) const {
>     return WithFactory(DefaultFactory<T>(t));
>   }
>@@ -961,9 +961,9 @@ class ExceptionSafetyTester {
>    * method tester.WithInitialValue(...).
>    */
>   template <typename NewFactory>
>-  ExceptionSafetyTester<absl::decay_t<NewFactory>, Operation, Invariants...>
>+  ExceptionSafetyTester<absl::decay_t<NewFactory>, Operation, Contracts...>
>   WithFactory(const NewFactory& new_factory) const {
>-    return {new_factory, operation_, invariants_};
>+    return {new_factory, operation_, contracts_};
>   }
> 
>   /*
>@@ -972,39 +972,39 @@ class ExceptionSafetyTester {
>    * tester.
>    */
>   template <typename NewOperation>
>-  ExceptionSafetyTester<Factory, absl::decay_t<NewOperation>, Invariants...>
>+  ExceptionSafetyTester<Factory, absl::decay_t<NewOperation>, Contracts...>
>   WithOperation(const NewOperation& new_operation) const {
>-    return {factory_, new_operation, invariants_};
>+    return {factory_, new_operation, contracts_};
>   }
> 
>   /*
>-   * Returns a new ExceptionSafetyTester with the provided MoreInvariants...
>-   * combined with the Invariants... that were already included in the instance
>-   * on which the method was called. Invariants... cannot be removed or replaced
>+   * Returns a new ExceptionSafetyTester with the provided MoreContracts...
>+   * combined with the Contracts... that were already included in the instance
>+   * on which the method was called. Contracts... cannot be removed or replaced
>    * once added to an ExceptionSafetyTester instance. A fresh object must be
>-   * created in order to get an empty Invariants... list.
>+   * created in order to get an empty Contracts... list.
>    *
>-   * In addition to passing in custom invariant assertion callbacks, this method
>+   * In addition to passing in custom contract assertion callbacks, this method
>    * accepts `testing::strong_guarantee` as an argument which checks T instances
>    * post-throw against freshly created T instances via operator== to verify
>    * that any state changes made during the execution of the operation were
>    * properly rolled back.
>    */
>-  template <typename... MoreInvariants>
>-  ExceptionSafetyTester<Factory, Operation, Invariants...,
>-                        absl::decay_t<MoreInvariants>...>
>-  WithInvariants(const MoreInvariants&... more_invariants) const {
>-    return {factory_, operation_,
>-            std::tuple_cat(invariants_,
>-                           std::tuple<absl::decay_t<MoreInvariants>...>(
>-                               more_invariants...))};
>+  template <typename... MoreContracts>
>+  ExceptionSafetyTester<Factory, Operation, Contracts...,
>+                        absl::decay_t<MoreContracts>...>
>+  WithContracts(const MoreContracts&... more_contracts) const {
>+    return {
>+        factory_, operation_,
>+        std::tuple_cat(contracts_, std::tuple<absl::decay_t<MoreContracts>...>(
>+                                       more_contracts...))};
>   }
> 
>   /*
>    * Returns a testing::AssertionResult that is the reduced result of the
>    * exception safety algorithm. The algorithm short circuits and returns
>-   * AssertionFailure after the first invariant callback returns an
>-   * AssertionFailure. Otherwise, if all invariant callbacks return an
>+   * AssertionFailure after the first contract callback returns an
>+   * AssertionFailure. Otherwise, if all contract callbacks return an
>    * AssertionSuccess, the reduced result is AssertionSuccess.
>    *
>    * The passed-in testable operation will not be saved in a new tester instance
>@@ -1013,33 +1013,33 @@ class ExceptionSafetyTester {
>    *
>    * Preconditions for tester.Test(const NewOperation& new_operation):
>    *
>-   * - May only be called after at least one invariant assertion callback and a
>+   * - May only be called after at least one contract assertion callback and a
>    *   factory or initial value have been provided.
>    */
>   template <
>       typename NewOperation,
>-      typename = EnableIfTestable<sizeof...(Invariants), Factory, NewOperation>>
>+      typename = EnableIfTestable<sizeof...(Contracts), Factory, NewOperation>>
>   testing::AssertionResult Test(const NewOperation& new_operation) const {
>-    return TestImpl(new_operation, absl::index_sequence_for<Invariants...>());
>+    return TestImpl(new_operation, absl::index_sequence_for<Contracts...>());
>   }
> 
>   /*
>    * Returns a testing::AssertionResult that is the reduced result of the
>    * exception safety algorithm. The algorithm short circuits and returns
>-   * AssertionFailure after the first invariant callback returns an
>-   * AssertionFailure. Otherwise, if all invariant callbacks return an
>+   * AssertionFailure after the first contract callback returns an
>+   * AssertionFailure. Otherwise, if all contract callbacks return an
>    * AssertionSuccess, the reduced result is AssertionSuccess.
>    *
>    * Preconditions for tester.Test():
>    *
>-   * - May only be called after at least one invariant assertion callback, a
>+   * - May only be called after at least one contract assertion callback, a
>    *   factory or initial value and a testable operation have been provided.
>    */
>-  template <typename LazyOperation = Operation,
>-            typename =
>-                EnableIfTestable<sizeof...(Invariants), Factory, LazyOperation>>
>+  template <
>+      typename LazyOperation = Operation,
>+      typename = EnableIfTestable<sizeof...(Contracts), Factory, LazyOperation>>
>   testing::AssertionResult Test() const {
>-    return TestImpl(operation_, absl::index_sequence_for<Invariants...>());
>+    return TestImpl(operation_, absl::index_sequence_for<Contracts...>());
>   }
> 
>  private:
>@@ -1051,8 +1051,8 @@ class ExceptionSafetyTester {
>   ExceptionSafetyTester() {}
> 
>   ExceptionSafetyTester(const Factory& f, const Operation& o,
>-                        const std::tuple<Invariants...>& i)
>-      : factory_(f), operation_(o), invariants_(i) {}
>+                        const std::tuple<Contracts...>& i)
>+      : factory_(f), operation_(o), contracts_(i) {}
> 
>   template <typename SelectedOperation, size_t... Indices>
>   testing::AssertionResult TestImpl(const SelectedOperation& selected_operation,
>@@ -1064,28 +1064,28 @@ class ExceptionSafetyTester {
> 
>       // Run the full exception safety test algorithm for the current countdown
>       auto reduced_res =
>-          TestAllInvariantsAtCountdown(factory_, selected_operation, count,
>-                                       std::get<Indices>(invariants_)...);
>-      // If there is no value in the optional, no invariants were run because no
>+          TestAllContractsAtCountdown(factory_, selected_operation, count,
>+                                      std::get<Indices>(contracts_)...);
>+      // If there is no value in the optional, no contracts were run because no
>       // exception was thrown. This means that the test is complete and the loop
>       // can exit successfully.
>       if (!reduced_res.has_value()) {
>         return testing::AssertionSuccess();
>       }
>-      // If the optional is not empty and the value is falsy, an invariant check
>+      // If the optional is not empty and the value is falsy, an contract check
>       // failed so the test must exit to propegate the failure.
>       if (!reduced_res.value()) {
>         return reduced_res.value();
>       }
>       // If the optional is not empty and the value is not falsy, it means
>-      // exceptions were thrown but the invariants passed so the test must
>+      // exceptions were thrown but the contracts passed so the test must
>       // continue to run.
>     }
>   }
> 
>   Factory factory_;
>   Operation operation_;
>-  std::tuple<Invariants...> invariants_;
>+  std::tuple<Contracts...> contracts_;
> };
> 
> }  // namespace exceptions_internal
>@@ -1096,7 +1096,7 @@ class ExceptionSafetyTester {
>  * instances of ExceptionSafetyTester.
>  *
>  * In order to test a T for exception safety, a factory for that T, a testable
>- * operation, and at least one invariant callback returning an assertion
>+ * operation, and at least one contract callback returning an assertion
>  * result must be applied using the respective methods.
>  */
> inline exceptions_internal::ExceptionSafetyTester<>
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.cc
>index 0626cd5478d97004eb1e9cde215a90e4862d29f1..159f945f5edd07078f8cafcea8b3730a3e8c718d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.cc
>@@ -401,16 +401,20 @@ bool LowLevelAlloc::DeleteArena(Arena *arena) {
>     ABSL_RAW_CHECK(munmap_result != 0,
>                    "LowLevelAlloc::DeleteArena: VitualFree failed");
> #else
>+#ifndef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>     if ((arena->flags & LowLevelAlloc::kAsyncSignalSafe) == 0) {
>       munmap_result = munmap(region, size);
>     } else {
>       munmap_result = base_internal::DirectMunmap(region, size);
>     }
>+#else
>+    munmap_result = munmap(region, size);
>+#endif  // ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>     if (munmap_result != 0) {
>       ABSL_RAW_LOG(FATAL, "LowLevelAlloc::DeleteArena: munmap failed: %d",
>                    errno);
>     }
>-#endif
>+#endif  // _WIN32
>   }
>   section.Leave();
>   arena->~Arena();
>@@ -545,6 +549,7 @@ static void *DoAllocWithArena(size_t request, LowLevelAlloc::Arena *arena) {
>                                MEM_RESERVE | MEM_COMMIT, PAGE_READWRITE);
>       ABSL_RAW_CHECK(new_pages != nullptr, "VirtualAlloc failed");
> #else
>+#ifndef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>       if ((arena->flags & LowLevelAlloc::kAsyncSignalSafe) != 0) {
>         new_pages = base_internal::DirectMmap(nullptr, new_pages_size,
>             PROT_WRITE|PROT_READ, MAP_ANONYMOUS|MAP_PRIVATE, -1, 0);
>@@ -552,10 +557,15 @@ static void *DoAllocWithArena(size_t request, LowLevelAlloc::Arena *arena) {
>         new_pages = mmap(nullptr, new_pages_size, PROT_WRITE | PROT_READ,
>                          MAP_ANONYMOUS | MAP_PRIVATE, -1, 0);
>       }
>+#else
>+      new_pages = mmap(nullptr, new_pages_size, PROT_WRITE | PROT_READ,
>+                       MAP_ANONYMOUS | MAP_PRIVATE, -1, 0);
>+#endif  // ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>       if (new_pages == MAP_FAILED) {
>         ABSL_RAW_LOG(FATAL, "mmap error: %d", errno);
>       }
>-#endif
>+
>+#endif  // _WIN32
>       arena->mu.Lock();
>       s = reinterpret_cast<AllocList *>(new_pages);
>       s->header.size = new_pages_size;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.h
>index 3c15605bed3578899389cc4ea296ceb5ba57a15e..fba9466a757427c01ed121464c3ab112de20c9a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.h
>@@ -39,10 +39,13 @@
> #define ABSL_LOW_LEVEL_ALLOC_MISSING 1
> #endif
> 
>-// Using LowLevelAlloc with kAsyncSignalSafe isn't supported on Windows.
>+// Using LowLevelAlloc with kAsyncSignalSafe isn't supported on Windows or
>+// asm.js / WebAssembly.
>+// See https://kripken.github.io/emscripten-site/docs/porting/pthreads.html
>+// for more information.
> #ifdef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
> #error ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING cannot be directly set
>-#elif defined(_WIN32)
>+#elif defined(_WIN32) || defined(__asmjs__) || defined(__wasm__)
> #define ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING 1
> #endif
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/raw_logging.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/raw_logging.h
>index 67abfd30798db8fff60b284aa6c92dfe23a45bef..79a7bb9b2f7a62bb1969ceeac01782ce4e9dec5a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/raw_logging.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/raw_logging.h
>@@ -114,7 +114,7 @@ void SafeWriteToStderr(const char *s, size_t len);
> 
> // compile-time function to get the "base" filename, that is, the part of
> // a filename after the last "/" or "\" path separator.  The search starts at
>-// the end of the std::string; the second parameter is the length of the std::string.
>+// the end of the string; the second parameter is the length of the string.
> constexpr const char* Basename(const char* fname, int offset) {
>   return offset == 0 || fname[offset - 1] == '/' || fname[offset - 1] == '\\'
>              ? fname + offset
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_linux.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_linux.inc
>new file mode 100644
>index 0000000000000000000000000000000000000000..94c861dc6ca23c28ed2082289ba23ef253860101
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_linux.inc
>@@ -0,0 +1,72 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This file is a Linux-specific part of spinlock_wait.cc
>+
>+#include <linux/futex.h>
>+#include <sys/syscall.h>
>+#include <unistd.h>
>+
>+#include <atomic>
>+#include <cerrno>
>+#include <climits>
>+#include <cstdint>
>+#include <ctime>
>+
>+#include "absl/base/attributes.h"
>+
>+// The SpinLock lockword is `std::atomic<uint32_t>`. Here we assert that
>+// `std::atomic<uint32_t>` is bitwise equivalent of the `int` expected
>+// by SYS_futex. We also assume that reads/writes done to the lockword
>+// by SYS_futex have rational semantics with regard to the
>+// std::atomic<> API. C++ provides no guarantees of these assumptions,
>+// but they are believed to hold in practice.
>+static_assert(sizeof(std::atomic<uint32_t>) == sizeof(int),
>+              "SpinLock lockword has the wrong size for a futex");
>+
>+// Some Android headers are missing these definitions even though they
>+// support these futex operations.
>+#ifdef __BIONIC__
>+#ifndef SYS_futex
>+#define SYS_futex __NR_futex
>+#endif
>+#ifndef FUTEX_PRIVATE_FLAG
>+#define FUTEX_PRIVATE_FLAG 128
>+#endif
>+#endif
>+
>+extern "C" {
>+
>+ABSL_ATTRIBUTE_WEAK void AbslInternalSpinLockDelay(
>+    std::atomic<uint32_t> *w, uint32_t value, int loop,
>+    absl::base_internal::SchedulingMode) {
>+  if (loop != 0) {
>+    int save_errno = errno;
>+    struct timespec tm;
>+    tm.tv_sec = 0;
>+    // Increase the delay; we expect (but do not rely on) explicit wakeups.
>+    // We don't rely on explicit wakeups because we intentionally allow for
>+    // a race on the kSpinLockSleeper bit.
>+    tm.tv_nsec = 16 * absl::base_internal::SpinLockSuggestedDelayNS(loop);
>+    syscall(SYS_futex, w, FUTEX_WAIT | FUTEX_PRIVATE_FLAG, value, &tm);
>+    errno = save_errno;
>+  }
>+}
>+
>+ABSL_ATTRIBUTE_WEAK void AbslInternalSpinLockWake(std::atomic<uint32_t> *w,
>+                                                  bool all) {
>+  syscall(SYS_futex, w, FUTEX_WAKE | FUTEX_PRIVATE_FLAG, all ? INT_MAX : 1, 0);
>+}
>+
>+}  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_wait.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_wait.cc
>index 9f6e9911e1096b2f187cc6a8430544e9fc166333..0fde9c04b73baf57d13499820cbd4471f35fdde2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_wait.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_wait.cc
>@@ -13,7 +13,7 @@
> // limitations under the License.
> 
> // The OS-specific header included below must provide two calls:
>-// base::subtle::SpinLockDelay() and base::subtle::SpinLockWake().
>+// AbslInternalSpinLockDelay() and AbslInternalSpinLockWake().
> // See spinlock_wait.h for the specs.
> 
> #include <atomic>
>@@ -23,6 +23,8 @@
> 
> #if defined(_WIN32)
> #include "absl/base/internal/spinlock_win32.inc"
>+#elif defined(__linux__)
>+#include "absl/base/internal/spinlock_linux.inc"
> #elif defined(__akaros__)
> #include "absl/base/internal/spinlock_akaros.inc"
> #else
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity.cc
>index 678e8568d7421e5148e9d19cd56e68084a003705..cff9c1b4f4dc39542b2aed944c8c7ed0663fa2cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity.cc
>@@ -68,6 +68,14 @@ void SetCurrentThreadIdentity(
>   // NOTE: Not async-safe.  But can be open-coded.
>   absl::call_once(init_thread_identity_key_once, AllocateThreadIdentityKey,
>                   reclaimer);
>+
>+#ifdef __EMSCRIPTEN__
>+  // Emscripten PThread implementation does not support signals.
>+  // See https://kripken.github.io/emscripten-site/docs/porting/pthreads.html
>+  // for more information.
>+  pthread_setspecific(thread_identity_pthread_key,
>+                      reinterpret_cast<void*>(identity));
>+#else
>   // We must mask signals around the call to setspecific as with current glibc,
>   // a concurrent getspecific (needed for GetCurrentThreadIdentityIfPresent())
>   // may zero our value.
>@@ -81,6 +89,8 @@ void SetCurrentThreadIdentity(
>   pthread_setspecific(thread_identity_pthread_key,
>                       reinterpret_cast<void*>(identity));
>   pthread_sigmask(SIG_SETMASK, &curr_signals, nullptr);
>+#endif  // !__EMSCRIPTEN__
>+
> #elif ABSL_THREAD_IDENTITY_MODE == ABSL_THREAD_IDENTITY_MODE_USE_TLS
>   // NOTE: Not async-safe.  But can be open-coded.
>   absl::call_once(init_thread_identity_key_once, AllocateThreadIdentityKey,
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unaligned_access.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unaligned_access.h
>index 5c7517abd71a6bf5250719d1d58cc7a05a906e1a..f9df3b7848c40a974fd9697def8cf7ffbfe029ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unaligned_access.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unaligned_access.h
>@@ -65,6 +65,7 @@ void __sanitizer_unaligned_store64(void *p, uint64_t v);
> }  // extern "C"
> 
> namespace absl {
>+namespace base_internal {
> 
> inline uint16_t UnalignedLoad16(const void *p) {
>   return __sanitizer_unaligned_load16(p);
>@@ -90,22 +91,27 @@ inline void UnalignedStore64(void *p, uint64_t v) {
>   __sanitizer_unaligned_store64(p, v);
> }
> 
>+}  // namespace base_internal
> }  // namespace absl
> 
>-#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) (absl::UnalignedLoad16(_p))
>-#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) (absl::UnalignedLoad32(_p))
>-#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) (absl::UnalignedLoad64(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) \
>+  (absl::base_internal::UnalignedLoad16(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) \
>+  (absl::base_internal::UnalignedLoad32(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) \
>+  (absl::base_internal::UnalignedLoad64(_p))
> 
> #define ABSL_INTERNAL_UNALIGNED_STORE16(_p, _val) \
>-  (absl::UnalignedStore16(_p, _val))
>+  (absl::base_internal::UnalignedStore16(_p, _val))
> #define ABSL_INTERNAL_UNALIGNED_STORE32(_p, _val) \
>-  (absl::UnalignedStore32(_p, _val))
>+  (absl::base_internal::UnalignedStore32(_p, _val))
> #define ABSL_INTERNAL_UNALIGNED_STORE64(_p, _val) \
>-  (absl::UnalignedStore64(_p, _val))
>+  (absl::base_internal::UnalignedStore64(_p, _val))
> 
> #elif defined(UNDEFINED_BEHAVIOR_SANITIZER)
> 
> namespace absl {
>+namespace base_internal {
> 
> inline uint16_t UnalignedLoad16(const void *p) {
>   uint16_t t;
>@@ -131,18 +137,22 @@ inline void UnalignedStore32(void *p, uint32_t v) { memcpy(p, &v, sizeof v); }
> 
> inline void UnalignedStore64(void *p, uint64_t v) { memcpy(p, &v, sizeof v); }
> 
>+}  // namespace base_internal
> }  // namespace absl
> 
>-#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) (absl::UnalignedLoad16(_p))
>-#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) (absl::UnalignedLoad32(_p))
>-#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) (absl::UnalignedLoad64(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) \
>+  (absl::base_internal::UnalignedLoad16(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) \
>+  (absl::base_internal::UnalignedLoad32(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) \
>+  (absl::base_internal::UnalignedLoad64(_p))
> 
> #define ABSL_INTERNAL_UNALIGNED_STORE16(_p, _val) \
>-  (absl::UnalignedStore16(_p, _val))
>+  (absl::base_internal::UnalignedStore16(_p, _val))
> #define ABSL_INTERNAL_UNALIGNED_STORE32(_p, _val) \
>-  (absl::UnalignedStore32(_p, _val))
>+  (absl::base_internal::UnalignedStore32(_p, _val))
> #define ABSL_INTERNAL_UNALIGNED_STORE64(_p, _val) \
>-  (absl::UnalignedStore64(_p, _val))
>+  (absl::base_internal::UnalignedStore64(_p, _val))
> 
> #elif defined(__x86_64__) || defined(_M_X64) || defined(__i386) || \
>     defined(_M_IX86) || defined(__ppc__) || defined(__PPC__) ||    \
>@@ -199,7 +209,7 @@ inline void UnalignedStore64(void *p, uint64_t v) { memcpy(p, &v, sizeof v); }
> // so we do that.
> 
> namespace absl {
>-namespace internal {
>+namespace base_internal {
> 
> struct Unaligned16Struct {
>   uint16_t value;
>@@ -211,22 +221,25 @@ struct Unaligned32Struct {
>   uint8_t dummy;  // To make the size non-power-of-two.
> } ABSL_ATTRIBUTE_PACKED;
> 
>-}  // namespace internal
>+}  // namespace base_internal
> }  // namespace absl
> 
>-#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) \
>-  ((reinterpret_cast<const ::absl::internal::Unaligned16Struct *>(_p))->value)
>-#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) \
>-  ((reinterpret_cast<const ::absl::internal::Unaligned32Struct *>(_p))->value)
>+#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p)                                  \
>+  ((reinterpret_cast<const ::absl::base_internal::Unaligned16Struct *>(_p)) \
>+       ->value)
>+#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p)                                  \
>+  ((reinterpret_cast<const ::absl::base_internal::Unaligned32Struct *>(_p)) \
>+       ->value)
> 
>-#define ABSL_INTERNAL_UNALIGNED_STORE16(_p, _val)                          \
>-  ((reinterpret_cast< ::absl::internal::Unaligned16Struct *>(_p))->value = \
>-       (_val))
>-#define ABSL_INTERNAL_UNALIGNED_STORE32(_p, _val)                          \
>-  ((reinterpret_cast< ::absl::internal::Unaligned32Struct *>(_p))->value = \
>-       (_val))
>+#define ABSL_INTERNAL_UNALIGNED_STORE16(_p, _val)                      \
>+  ((reinterpret_cast< ::absl::base_internal::Unaligned16Struct *>(_p)) \
>+       ->value = (_val))
>+#define ABSL_INTERNAL_UNALIGNED_STORE32(_p, _val)                      \
>+  ((reinterpret_cast< ::absl::base_internal::Unaligned32Struct *>(_p)) \
>+       ->value = (_val))
> 
> namespace absl {
>+namespace base_internal {
> 
> inline uint64_t UnalignedLoad64(const void *p) {
>   uint64_t t;
>@@ -236,11 +249,13 @@ inline uint64_t UnalignedLoad64(const void *p) {
> 
> inline void UnalignedStore64(void *p, uint64_t v) { memcpy(p, &v, sizeof v); }
> 
>+}  // namespace base_internal
> }  // namespace absl
> 
>-#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) (absl::UnalignedLoad64(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) \
>+  (absl::base_internal::UnalignedLoad64(_p))
> #define ABSL_INTERNAL_UNALIGNED_STORE64(_p, _val) \
>-  (absl::UnalignedStore64(_p, _val))
>+  (absl::base_internal::UnalignedStore64(_p, _val))
> 
> #else
> 
>@@ -252,6 +267,7 @@ inline void UnalignedStore64(void *p, uint64_t v) { memcpy(p, &v, sizeof v); }
> // unaligned loads and stores.
> 
> namespace absl {
>+namespace base_internal {
> 
> inline uint16_t UnalignedLoad16(const void *p) {
>   uint16_t t;
>@@ -277,18 +293,22 @@ inline void UnalignedStore32(void *p, uint32_t v) { memcpy(p, &v, sizeof v); }
> 
> inline void UnalignedStore64(void *p, uint64_t v) { memcpy(p, &v, sizeof v); }
> 
>+}  // namespace base_internal
> }  // namespace absl
> 
>-#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) (absl::UnalignedLoad16(_p))
>-#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) (absl::UnalignedLoad32(_p))
>-#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) (absl::UnalignedLoad64(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) \
>+  (absl::base_internal::UnalignedLoad16(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) \
>+  (absl::base_internal::UnalignedLoad32(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) \
>+  (absl::base_internal::UnalignedLoad64(_p))
> 
> #define ABSL_INTERNAL_UNALIGNED_STORE16(_p, _val) \
>-  (absl::UnalignedStore16(_p, _val))
>+  (absl::base_internal::UnalignedStore16(_p, _val))
> #define ABSL_INTERNAL_UNALIGNED_STORE32(_p, _val) \
>-  (absl::UnalignedStore32(_p, _val))
>+  (absl::base_internal::UnalignedStore32(_p, _val))
> #define ABSL_INTERNAL_UNALIGNED_STORE64(_p, _val) \
>-  (absl::UnalignedStore64(_p, _val))
>+  (absl::base_internal::UnalignedStore64(_p, _val))
> 
> #endif
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/log_severity.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/log_severity.h
>index e2931c34d1df54a7bbf7ed71de858503362c7337..5770d3629e7264689f6029cb6d64285545bf4fd1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/log_severity.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/log_severity.h
>@@ -39,7 +39,7 @@ constexpr std::array<absl::LogSeverity, 4> LogSeverities() {
>            absl::LogSeverity::kError, absl::LogSeverity::kFatal}};
> }
> 
>-// Returns the all-caps std::string representation (e.g. "INFO") of the specified
>+// Returns the all-caps string representation (e.g. "INFO") of the specified
> // severity level if it is one of the normal levels and "UNKNOWN" otherwise.
> constexpr const char* LogSeverityName(absl::LogSeverity s) {
>   return s == absl::LogSeverity::kInfo
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/raw_logging_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/raw_logging_test.cc
>index ebbc5db906727f6cb2c03a63e99a897f5ceb5aa4..b21cf651758e3c249262e69c1c8ba17c1c9a7aee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/raw_logging_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/raw_logging_test.cc
>@@ -40,7 +40,7 @@ TEST(RawLoggingCompilationTest, PassingCheck) {
> }
> 
> // Not all platforms support output from raw log, so we don't verify any
>-// particular output for RAW check failures (expecting the empty std::string
>+// particular output for RAW check failures (expecting the empty string
> // accomplishes this).  This test is primarily a compilation test, but we
> // are verifying process death when EXPECT_DEATH works for a platform.
> const char kExpectedDeathOutput[] = "";
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.bazel
>index 6d5c958f382ba6397f0b0df1cf996678a7a3c5a6..f2210e3c197983fab6034d818154779b71732f2f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.bazel
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.bazel
>@@ -19,6 +19,7 @@ load(
>     "ABSL_DEFAULT_COPTS",
>     "ABSL_TEST_COPTS",
>     "ABSL_EXCEPTIONS_FLAG",
>+    "ABSL_EXCEPTIONS_FLAG_LINKOPTS",
> )
> 
> package(default_visibility = ["//visibility:public"])
>@@ -62,9 +63,11 @@ cc_test(
>     name = "fixed_array_test",
>     srcs = ["fixed_array_test.cc"],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":fixed_array",
>         "//absl/base:exception_testing",
>+        "//absl/hash:hash_testing",
>         "//absl/memory",
>         "@com_google_googletest//:gtest_main",
>     ],
>@@ -77,6 +80,7 @@ cc_test(
>     deps = [
>         ":fixed_array",
>         "//absl/base:exception_testing",
>+        "//absl/hash:hash_testing",
>         "//absl/memory",
>         "@com_google_googletest//:gtest_main",
>     ],
>@@ -86,6 +90,7 @@ cc_test(
>     name = "fixed_array_exception_safety_test",
>     srcs = ["fixed_array_exception_safety_test.cc"],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":fixed_array",
>         "//absl/base:exception_safety_testing",
>@@ -120,12 +125,14 @@ cc_test(
>     name = "inlined_vector_test",
>     srcs = ["inlined_vector_test.cc"],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":inlined_vector",
>         ":test_instance_tracker",
>         "//absl/base",
>         "//absl/base:core_headers",
>         "//absl/base:exception_testing",
>+        "//absl/hash:hash_testing",
>         "//absl/memory",
>         "//absl/strings",
>         "@com_google_googletest//:gtest_main",
>@@ -142,6 +149,7 @@ cc_test(
>         "//absl/base",
>         "//absl/base:core_headers",
>         "//absl/base:exception_testing",
>+        "//absl/hash:hash_testing",
>         "//absl/memory",
>         "//absl/strings",
>         "@com_google_googletest//:gtest_main",
>@@ -181,3 +189,459 @@ cc_test(
>         "@com_google_googletest//:gtest_main",
>     ],
> )
>+
>+NOTEST_TAGS_NONMOBILE = [
>+    "no_test_darwin_x86_64",
>+    "no_test_loonix",
>+]
>+
>+NOTEST_TAGS_MOBILE = [
>+    "no_test_android_arm",
>+    "no_test_android_arm64",
>+    "no_test_android_x86",
>+    "no_test_ios_x86_64",
>+]
>+
>+NOTEST_TAGS = NOTEST_TAGS_MOBILE + NOTEST_TAGS_NONMOBILE
>+
>+cc_library(
>+    name = "flat_hash_map",
>+    hdrs = ["flat_hash_map.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":container_memory",
>+        ":hash_function_defaults",
>+        ":raw_hash_map",
>+        "//absl/memory",
>+    ],
>+)
>+
>+cc_test(
>+    name = "flat_hash_map_test",
>+    srcs = ["flat_hash_map_test.cc"],
>+    copts = ABSL_TEST_COPTS + ["-DUNORDERED_MAP_CXX17"],
>+    tags = NOTEST_TAGS_NONMOBILE,
>+    deps = [
>+        ":flat_hash_map",
>+        ":hash_generator_testing",
>+        ":unordered_map_constructor_test",
>+        ":unordered_map_lookup_test",
>+        ":unordered_map_modifiers_test",
>+        "//absl/types:any",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "flat_hash_set",
>+    hdrs = ["flat_hash_set.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":container_memory",
>+        ":hash_function_defaults",
>+        ":raw_hash_set",
>+        "//absl/base:core_headers",
>+        "//absl/memory",
>+    ],
>+)
>+
>+cc_test(
>+    name = "flat_hash_set_test",
>+    srcs = ["flat_hash_set_test.cc"],
>+    copts = ABSL_TEST_COPTS + ["-DUNORDERED_SET_CXX17"],
>+    tags = NOTEST_TAGS_NONMOBILE,
>+    deps = [
>+        ":flat_hash_set",
>+        ":hash_generator_testing",
>+        ":unordered_set_constructor_test",
>+        ":unordered_set_lookup_test",
>+        ":unordered_set_modifiers_test",
>+        "//absl/memory",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "node_hash_map",
>+    hdrs = ["node_hash_map.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":container_memory",
>+        ":hash_function_defaults",
>+        ":node_hash_policy",
>+        ":raw_hash_map",
>+        "//absl/memory",
>+    ],
>+)
>+
>+cc_test(
>+    name = "node_hash_map_test",
>+    srcs = ["node_hash_map_test.cc"],
>+    copts = ABSL_TEST_COPTS + ["-DUNORDERED_MAP_CXX17"],
>+    tags = NOTEST_TAGS_NONMOBILE,
>+    deps = [
>+        ":hash_generator_testing",
>+        ":node_hash_map",
>+        ":tracked",
>+        ":unordered_map_constructor_test",
>+        ":unordered_map_lookup_test",
>+        ":unordered_map_modifiers_test",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "node_hash_set",
>+    hdrs = ["node_hash_set.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":container_memory",
>+        ":hash_function_defaults",
>+        ":node_hash_policy",
>+        ":raw_hash_set",
>+        "//absl/memory",
>+    ],
>+)
>+
>+cc_test(
>+    name = "node_hash_set_test",
>+    srcs = ["node_hash_set_test.cc"],
>+    copts = ABSL_TEST_COPTS + ["-DUNORDERED_SET_CXX17"],
>+    tags = NOTEST_TAGS_NONMOBILE,
>+    deps = [
>+        ":hash_generator_testing",
>+        ":node_hash_set",
>+        ":unordered_set_constructor_test",
>+        ":unordered_set_lookup_test",
>+        ":unordered_set_modifiers_test",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "container_memory",
>+    hdrs = ["internal/container_memory.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/memory",
>+        "//absl/utility",
>+    ],
>+)
>+
>+cc_test(
>+    name = "container_memory_test",
>+    srcs = ["internal/container_memory_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = NOTEST_TAGS_NONMOBILE,
>+    deps = [
>+        ":container_memory",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "hash_function_defaults",
>+    hdrs = ["internal/hash_function_defaults.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base:config",
>+        "//absl/hash",
>+        "//absl/strings",
>+    ],
>+)
>+
>+cc_test(
>+    name = "hash_function_defaults_test",
>+    srcs = ["internal/hash_function_defaults_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = NOTEST_TAGS,
>+    deps = [
>+        ":hash_function_defaults",
>+        "//absl/hash",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "hash_generator_testing",
>+    testonly = 1,
>+    srcs = ["internal/hash_generator_testing.cc"],
>+    hdrs = ["internal/hash_generator_testing.h"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":hash_policy_testing",
>+        "//absl/meta:type_traits",
>+        "//absl/strings",
>+    ],
>+)
>+
>+cc_library(
>+    name = "hash_policy_testing",
>+    testonly = 1,
>+    hdrs = ["internal/hash_policy_testing.h"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        "//absl/hash",
>+        "//absl/strings",
>+    ],
>+)
>+
>+cc_test(
>+    name = "hash_policy_testing_test",
>+    srcs = ["internal/hash_policy_testing_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":hash_policy_testing",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "hash_policy_traits",
>+    hdrs = ["internal/hash_policy_traits.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = ["//absl/meta:type_traits"],
>+)
>+
>+cc_test(
>+    name = "hash_policy_traits_test",
>+    srcs = ["internal/hash_policy_traits_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":hash_policy_traits",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "hashtable_debug",
>+    hdrs = ["internal/hashtable_debug.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":hashtable_debug_hooks",
>+    ],
>+)
>+
>+cc_library(
>+    name = "hashtable_debug_hooks",
>+    hdrs = ["internal/hashtable_debug_hooks.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+)
>+
>+cc_library(
>+    name = "node_hash_policy",
>+    hdrs = ["internal/node_hash_policy.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+)
>+
>+cc_test(
>+    name = "node_hash_policy_test",
>+    srcs = ["internal/node_hash_policy_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":hash_policy_traits",
>+        ":node_hash_policy",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "raw_hash_map",
>+    hdrs = ["internal/raw_hash_map.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":container_memory",
>+        ":raw_hash_set",
>+    ],
>+)
>+
>+cc_library(
>+    name = "raw_hash_set",
>+    srcs = ["internal/raw_hash_set.cc"],
>+    hdrs = ["internal/raw_hash_set.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":compressed_tuple",
>+        ":container_memory",
>+        ":hash_policy_traits",
>+        ":hashtable_debug_hooks",
>+        ":layout",
>+        "//absl/base:bits",
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+        "//absl/base:endian",
>+        "//absl/memory",
>+        "//absl/meta:type_traits",
>+        "//absl/types:optional",
>+        "//absl/utility",
>+    ],
>+)
>+
>+cc_test(
>+    name = "raw_hash_set_test",
>+    srcs = ["internal/raw_hash_set_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    linkstatic = 1,
>+    tags = NOTEST_TAGS,
>+    deps = [
>+        ":container_memory",
>+        ":hash_function_defaults",
>+        ":hash_policy_testing",
>+        ":hashtable_debug",
>+        ":raw_hash_set",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "raw_hash_set_allocator_test",
>+    size = "small",
>+    srcs = ["internal/raw_hash_set_allocator_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":raw_hash_set",
>+        ":tracked",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "layout",
>+    hdrs = ["internal/layout.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base:core_headers",
>+        "//absl/meta:type_traits",
>+        "//absl/strings",
>+        "//absl/types:span",
>+        "//absl/utility",
>+    ],
>+)
>+
>+cc_test(
>+    name = "layout_test",
>+    size = "small",
>+    srcs = ["internal/layout_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = NOTEST_TAGS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":layout",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "//absl/types:span",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "tracked",
>+    testonly = 1,
>+    hdrs = ["internal/tracked.h"],
>+    copts = ABSL_TEST_COPTS,
>+)
>+
>+cc_library(
>+    name = "unordered_map_constructor_test",
>+    testonly = 1,
>+    hdrs = ["internal/unordered_map_constructor_test.h"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":hash_generator_testing",
>+        ":hash_policy_testing",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_library(
>+    name = "unordered_map_lookup_test",
>+    testonly = 1,
>+    hdrs = ["internal/unordered_map_lookup_test.h"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":hash_generator_testing",
>+        ":hash_policy_testing",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_library(
>+    name = "unordered_map_modifiers_test",
>+    testonly = 1,
>+    hdrs = ["internal/unordered_map_modifiers_test.h"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":hash_generator_testing",
>+        ":hash_policy_testing",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_library(
>+    name = "unordered_set_constructor_test",
>+    testonly = 1,
>+    hdrs = ["internal/unordered_set_constructor_test.h"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":hash_generator_testing",
>+        ":hash_policy_testing",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_library(
>+    name = "unordered_set_lookup_test",
>+    testonly = 1,
>+    hdrs = ["internal/unordered_set_lookup_test.h"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":hash_generator_testing",
>+        ":hash_policy_testing",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_library(
>+    name = "unordered_set_modifiers_test",
>+    testonly = 1,
>+    hdrs = ["internal/unordered_set_modifiers_test.h"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":hash_generator_testing",
>+        ":hash_policy_testing",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_test(
>+    name = "unordered_set_test",
>+    srcs = ["internal/unordered_set_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = NOTEST_TAGS_NONMOBILE,
>+    deps = [
>+        ":unordered_set_constructor_test",
>+        ":unordered_set_lookup_test",
>+        ":unordered_set_modifiers_test",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "unordered_map_test",
>+    srcs = ["internal/unordered_map_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = NOTEST_TAGS_NONMOBILE,
>+    deps = [
>+        ":unordered_map_constructor_test",
>+        ":unordered_map_lookup_test",
>+        ":unordered_map_modifiers_test",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.gn
>index 001a2a36d4846d57e851f8143f8aaa5c9915830a..a2fbd543316d456b5df1473133e31d0409b7270e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.gn
>@@ -84,3 +84,389 @@ source_set("test_instance_tracker") {
>   visibility = []
>   visibility += [ "../*" ]
> }
>+
>+source_set("flat_hash_map") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "flat_hash_map.h",
>+  ]
>+  deps = [
>+    ":container_memory",
>+    ":hash_function_defaults",
>+    ":raw_hash_map",
>+    "../memory",
>+  ]
>+}
>+
>+source_set("flat_hash_set") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "flat_hash_set.h",
>+  ]
>+  deps = [
>+    ":container_memory",
>+    ":hash_function_defaults",
>+    ":raw_hash_set",
>+    "../base:core_headers",
>+    "../memory",
>+  ]
>+}
>+
>+source_set("node_hash_map") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "node_hash_map.h",
>+  ]
>+  deps = [
>+    ":container_memory",
>+    ":hash_function_defaults",
>+    ":node_hash_policy",
>+    ":raw_hash_map",
>+    "../memory",
>+  ]
>+}
>+
>+source_set("node_hash_set") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "node_hash_set.h",
>+  ]
>+  deps = [
>+    ":container_memory",
>+    ":hash_function_defaults",
>+    ":node_hash_policy",
>+    ":raw_hash_set",
>+    "../memory",
>+  ]
>+}
>+
>+source_set("container_memory") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/container_memory.h",
>+  ]
>+  deps = [
>+    "../memory",
>+    "../utility",
>+  ]
>+}
>+
>+source_set("hash_function_defaults") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/hash_function_defaults.h",
>+  ]
>+  deps = [
>+    "../base:config",
>+    "../hash",
>+    "../strings",
>+  ]
>+}
>+
>+source_set("hash_generator_testing") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/hash_generator_testing.cc",
>+  ]
>+  public = [
>+    "internal/hash_generator_testing.h",
>+  ]
>+  deps = [
>+    ":hash_policy_testing",
>+    "../meta:type_traits",
>+    "../strings",
>+  ]
>+}
>+
>+source_set("hash_policy_testing") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/hash_policy_testing.h",
>+  ]
>+  deps = [
>+    "../hash",
>+    "../strings",
>+  ]
>+}
>+
>+source_set("hash_policy_traits") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/hash_policy_traits.h",
>+  ]
>+  deps = [
>+    "../meta:type_traits",
>+  ]
>+}
>+
>+source_set("hashtable_debug") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/hashtable_debug.h",
>+  ]
>+  deps = [
>+    ":hashtable_debug_hooks",
>+  ]
>+}
>+
>+source_set("hashtable_debug_hooks") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/hashtable_debug_hooks.h",
>+  ]
>+}
>+
>+source_set("node_hash_policy") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/node_hash_policy.h",
>+  ]
>+}
>+
>+source_set("raw_hash_map") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/raw_hash_map.h",
>+  ]
>+  deps = [
>+    ":container_memory",
>+    ":raw_hash_set",
>+  ]
>+}
>+
>+source_set("raw_hash_set") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/raw_hash_set.cc",
>+  ]
>+  public = [
>+    "internal/raw_hash_set.h",
>+  ]
>+  deps = [
>+    ":compressed_tuple",
>+    ":container_memory",
>+    ":hash_policy_traits",
>+    ":hashtable_debug_hooks",
>+    ":layout",
>+    "../base:bits",
>+    "../base:config",
>+    "../base:core_headers",
>+    "../base:endian",
>+    "../memory",
>+    "../meta:type_traits",
>+    "../types:optional",
>+    "../utility",
>+  ]
>+}
>+
>+source_set("layout") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/layout.h",
>+  ]
>+  deps = [
>+    "../base:core_headers",
>+    "../meta:type_traits",
>+    "../strings",
>+    "../types:span",
>+    "../utility",
>+  ]
>+}
>+
>+source_set("tracked") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/tracked.h",
>+  ]
>+}
>+
>+source_set("unordered_map_constructor_test") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/unordered_map_constructor_test.h",
>+  ]
>+  deps = [
>+    ":hash_generator_testing",
>+    ":hash_policy_testing",
>+    "//testing/gtest",
>+  ]
>+}
>+
>+source_set("unordered_map_lookup_test") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/unordered_map_lookup_test.h",
>+  ]
>+  deps = [
>+    ":hash_generator_testing",
>+    ":hash_policy_testing",
>+    "//testing/gtest",
>+  ]
>+}
>+
>+source_set("unordered_map_modifiers_test") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/unordered_map_modifiers_test.h",
>+  ]
>+  deps = [
>+    ":hash_generator_testing",
>+    ":hash_policy_testing",
>+    "//testing/gtest",
>+  ]
>+}
>+
>+source_set("unordered_set_constructor_test") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/unordered_set_constructor_test.h",
>+  ]
>+  deps = [
>+    ":hash_generator_testing",
>+    ":hash_policy_testing",
>+    "//testing/gtest",
>+  ]
>+}
>+
>+source_set("unordered_set_lookup_test") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/unordered_set_lookup_test.h",
>+  ]
>+  deps = [
>+    ":hash_generator_testing",
>+    ":hash_policy_testing",
>+    "//testing/gtest",
>+  ]
>+}
>+
>+source_set("unordered_set_modifiers_test") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/unordered_set_modifiers_test.h",
>+  ]
>+  deps = [
>+    ":hash_generator_testing",
>+    ":hash_policy_testing",
>+    "//testing/gtest",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/CMakeLists.txt
>index 123e4c4849aabe2eedea21f05efd2d0263830fb8..9e406902b0585deda3ba3072bee0ff277484ebc3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/CMakeLists.txt
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/CMakeLists.txt
>@@ -17,18 +17,42 @@
> 
> list(APPEND CONTAINER_PUBLIC_HEADERS
>   "fixed_array.h"
>+  "flat_hash_map.h"
>+  "flat_hash_set.h"
>   "inlined_vector.h"
>+  "node_hash_map.h"
>+  "node_hash_set.h"
> )
> 
> 
> list(APPEND CONTAINER_INTERNAL_HEADERS
>+  "internal/compressed_tuple.h"
>+  "internal/container_memory.h"
>+  "internal/hash_function_defaults.h"
>+  "internal/hash_generator_testing.h"
>+  "internal/hash_policy_testing.h"
>+  "internal/hash_policy_traits.h"
>+  "internal/hashtable_debug.h"
>+  "internal/layout.h"
>+  "internal/node_hash_policy.h"
>+  "internal/raw_hash_map.h"
>+  "internal/raw_hash_set.h"
>   "internal/test_instance_tracker.h"
>+  "internal/tracked.h"
>+  "internal/unordered_map_constructor_test.h"
>+  "internal/unordered_map_lookup_test.h"
>+  "internal/unordered_map_modifiers_test.h"
>+  "internal/unordered_set_constructor_test.h"
>+  "internal/unordered_set_lookup_test.h"
>+  "internal/unordered_set_modifiers_test.h"
> )
> 
> 
>-absl_header_library(
>+absl_library(
>   TARGET
>     absl_container
>+  SOURCES
>+    "internal/raw_hash_set.cc"
>   EXPORT_NAME
>     container
> )
>@@ -142,3 +166,11 @@ absl_test(
> )
> 
> 
>+absl_test(
>+  TARGET
>+    raw_hash_set_test
>+  SOURCES
>+    "internal/raw_hash_set_test.cc"
>+  PUBLIC_LIBRARIES
>+    absl::base absl::hash absl_throw_delegate test_instance_tracker_lib
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array.h
>index 182258f623e13aea80b7578e1906dad9478388b0..6d9fa5f745dcc9563c7a709160db1837cffbc2d7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array.h
>@@ -138,8 +138,8 @@ class FixedArray {
>   explicit FixedArray(size_type n, const allocator_type& a = allocator_type())
>       : storage_(n, a) {
>     if (DefaultConstructorIsNonTrivial()) {
>-      memory_internal::ConstructStorage(storage_.alloc(), storage_.begin(),
>-                                        storage_.end());
>+      memory_internal::ConstructRange(storage_.alloc(), storage_.begin(),
>+                                      storage_.end());
>     }
>   }
> 
>@@ -147,8 +147,8 @@ class FixedArray {
>   FixedArray(size_type n, const value_type& val,
>              const allocator_type& a = allocator_type())
>       : storage_(n, a) {
>-    memory_internal::ConstructStorage(storage_.alloc(), storage_.begin(),
>-                                      storage_.end(), val);
>+    memory_internal::ConstructRange(storage_.alloc(), storage_.begin(),
>+                                    storage_.end(), val);
>   }
> 
>   // Creates an array initialized with the size and contents of `init_list`.
>@@ -163,13 +163,12 @@ class FixedArray {
>   FixedArray(Iterator first, Iterator last,
>              const allocator_type& a = allocator_type())
>       : storage_(std::distance(first, last), a) {
>-    memory_internal::CopyToStorageFromRange(storage_.alloc(), storage_.begin(),
>-                                            first, last);
>+    memory_internal::CopyRange(storage_.alloc(), storage_.begin(), first, last);
>   }
> 
>   ~FixedArray() noexcept {
>     for (auto* cur = storage_.begin(); cur != storage_.end(); ++cur) {
>-      AllocatorTraits::destroy(*storage_.alloc(), cur);
>+      AllocatorTraits::destroy(storage_.alloc(), cur);
>     }
>   }
> 
>@@ -359,6 +358,13 @@ class FixedArray {
>   friend bool operator>=(const FixedArray& lhs, const FixedArray& rhs) {
>     return !(lhs < rhs);
>   }
>+
>+  template <typename H>
>+  friend H AbslHashValue(H h, const FixedArray& v) {
>+    return H::combine(H::combine_contiguous(std::move(h), v.data(), v.size()),
>+                      v.size());
>+  }
>+
>  private:
>   // StorageElement
>   //
>@@ -446,15 +452,15 @@ class FixedArray {
>       if (UsingInlinedStorage(size())) {
>         InlinedStorage::AnnotateDestruct(size());
>       } else {
>-        AllocatorTraits::deallocate(*alloc(), AsValueType(begin()), size());
>+        AllocatorTraits::deallocate(alloc(), AsValueType(begin()), size());
>       }
>     }
> 
>     size_type size() const { return size_alloc_.template get<0>(); }
>     StorageElement* begin() const { return data_; }
>     StorageElement* end() const { return begin() + size(); }
>-    allocator_type* alloc() {
>-      return std::addressof(size_alloc_.template get<1>());
>+    allocator_type& alloc() {
>+      return size_alloc_.template get<1>();
>     }
> 
>    private:
>@@ -468,7 +474,7 @@ class FixedArray {
>         return InlinedStorage::data();
>       } else {
>         return reinterpret_cast<StorageElement*>(
>-            AllocatorTraits::allocate(*alloc(), size()));
>+            AllocatorTraits::allocate(alloc(), size()));
>       }
>     }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_exception_safety_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_exception_safety_test.cc
>index c123c2a1c0d2d38484056c34f89101489e5c1424..da63dbfe38e71d34a33e800dfe42aa9f3ea317fa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_exception_safety_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_exception_safety_test.cc
>@@ -97,7 +97,7 @@ testing::AssertionResult ReadMemory(FixedArr* fixed_arr) {
> 
> TEST(FixedArrayExceptionSafety, Fill) {
>   auto test_fill = testing::MakeExceptionSafetyTester()
>-                       .WithInvariants(ReadMemory)
>+                       .WithContracts(ReadMemory)
>                        .WithOperation([&](FixedArr* fixed_arr_ptr) {
>                          auto thrower =
>                              Thrower(kUpdatedValue, testing::nothrow_ctor);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_test.cc
>index b07ebcb6d9ca6b0e426ca63ac34c06f8140983ce..205ff41fe1142bbe8e187a44f17a5a5a73b5c548 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_test.cc
>@@ -27,6 +27,7 @@
> #include "gmock/gmock.h"
> #include "gtest/gtest.h"
> #include "absl/base/internal/exception_testing.h"
>+#include "absl/hash/hash_testing.h"
> #include "absl/memory/memory.h"
> 
> using ::testing::ElementsAreArray;
>@@ -867,4 +868,5 @@ TEST(FixedArrayTest, AddressSanitizerAnnotations4) {
>   EXPECT_DEATH(raw[21] = ThreeInts(), "container-overflow");
> }
> #endif  // ADDRESS_SANITIZER
>+
> }  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_map.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_map.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..e5570d1004413166abdddb6c546bb57cabd4bf04
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_map.h
>@@ -0,0 +1,528 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: flat_hash_map.h
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::flat_hash_map<K, V>` is an unordered associative container of
>+// unique keys and associated values designed to be a more efficient replacement
>+// for `std::unordered_map`. Like `unordered_map`, search, insertion, and
>+// deletion of map elements can be done as an `O(1)` operation. However,
>+// `flat_hash_map` (and other unordered associative containers known as the
>+// collection of Abseil "Swiss tables") contain other optimizations that result
>+// in both memory and computation advantages.
>+//
>+// In most cases, your default choice for a hash map should be a map of type
>+// `flat_hash_map`.
>+
>+#ifndef ABSL_CONTAINER_FLAT_HASH_MAP_H_
>+#define ABSL_CONTAINER_FLAT_HASH_MAP_H_
>+
>+#include <cstddef>
>+#include <new>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/container/internal/container_memory.h"
>+#include "absl/container/internal/hash_function_defaults.h"  // IWYU pragma: export
>+#include "absl/container/internal/raw_hash_map.h"  // IWYU pragma: export
>+#include "absl/memory/memory.h"
>+
>+namespace absl {
>+namespace container_internal {
>+template <class K, class V>
>+struct FlatHashMapPolicy;
>+}  // namespace container_internal
>+
>+// -----------------------------------------------------------------------------
>+// absl::flat_hash_map
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::flat_hash_map<K, V>` is an unordered associative container which
>+// has been optimized for both speed and memory footprint in most common use
>+// cases. Its interface is similar to that of `std::unordered_map<K, V>` with
>+// the following notable differences:
>+//
>+// * Requires keys that are CopyConstructible
>+// * Requires values that are MoveConstructible
>+// * Supports heterogeneous lookup, through `find()`, `operator[]()` and
>+//   `insert()`, provided that the map is provided a compatible heterogeneous
>+//   hashing function and equality operator.
>+// * Invalidates any references and pointers to elements within the table after
>+//   `rehash()`.
>+// * Contains a `capacity()` member function indicating the number of element
>+//   slots (open, deleted, and empty) within the hash map.
>+// * Returns `void` from the `erase(iterator)` overload.
>+//
>+// By default, `flat_hash_map` uses the `absl::Hash` hashing framework.
>+// All fundamental and Abseil types that support the `absl::Hash` framework have
>+// a compatible equality operator for comparing insertions into `flat_hash_map`.
>+// If your type is not yet supported by the `asbl::Hash` framework, see
>+// absl/hash/hash.h for information on extending Abseil hashing to user-defined
>+// types.
>+//
>+// NOTE: A `flat_hash_map` stores its value types directly inside its
>+// implementation array to avoid memory indirection. Because a `flat_hash_map`
>+// is designed to move data when rehashed, map values will not retain pointer
>+// stability. If you require pointer stability, or your values are large,
>+// consider using `absl::flat_hash_map<Key, std::unique_ptr<Value>>` instead.
>+// If your types are not moveable or you require pointer stability for keys,
>+// consider `absl::node_hash_map`.
>+//
>+// Example:
>+//
>+//   // Create a flat hash map of three strings (that map to strings)
>+//   absl::flat_hash_map<std::string, std::string> ducks =
>+//     {{"a", "huey"}, {"b", "dewey"}, {"c", "louie"}};
>+//
>+//  // Insert a new element into the flat hash map
>+//  ducks.insert({"d", "donald"});
>+//
>+//  // Force a rehash of the flat hash map
>+//  ducks.rehash(0);
>+//
>+//  // Find the element with the key "b"
>+//  std::string search_key = "b";
>+//  auto result = ducks.find(search_key);
>+//  if (result != ducks.end()) {
>+//    std::cout << "Result: " << result->second << std::endl;
>+//  }
>+template <class K, class V,
>+          class Hash = absl::container_internal::hash_default_hash<K>,
>+          class Eq = absl::container_internal::hash_default_eq<K>,
>+          class Allocator = std::allocator<std::pair<const K, V>>>
>+class flat_hash_map : public absl::container_internal::raw_hash_map<
>+                          absl::container_internal::FlatHashMapPolicy<K, V>,
>+                          Hash, Eq, Allocator> {
>+  using Base = typename flat_hash_map::raw_hash_map;
>+
>+ public:
>+  flat_hash_map() {}
>+  using Base::Base;
>+
>+  // flat_hash_map::begin()
>+  //
>+  // Returns an iterator to the beginning of the `flat_hash_map`.
>+  using Base::begin;
>+
>+  // flat_hash_map::cbegin()
>+  //
>+  // Returns a const iterator to the beginning of the `flat_hash_map`.
>+  using Base::cbegin;
>+
>+  // flat_hash_map::cend()
>+  //
>+  // Returns a const iterator to the end of the `flat_hash_map`.
>+  using Base::cend;
>+
>+  // flat_hash_map::end()
>+  //
>+  // Returns an iterator to the end of the `flat_hash_map`.
>+  using Base::end;
>+
>+  // flat_hash_map::capacity()
>+  //
>+  // Returns the number of element slots (assigned, deleted, and empty)
>+  // available within the `flat_hash_map`.
>+  //
>+  // NOTE: this member function is particular to `absl::flat_hash_map` and is
>+  // not provided in the `std::unordered_map` API.
>+  using Base::capacity;
>+
>+  // flat_hash_map::empty()
>+  //
>+  // Returns whether or not the `flat_hash_map` is empty.
>+  using Base::empty;
>+
>+  // flat_hash_map::max_size()
>+  //
>+  // Returns the largest theoretical possible number of elements within a
>+  // `flat_hash_map` under current memory constraints. This value can be thought
>+  // of the largest value of `std::distance(begin(), end())` for a
>+  // `flat_hash_map<K, V>`.
>+  using Base::max_size;
>+
>+  // flat_hash_map::size()
>+  //
>+  // Returns the number of elements currently within the `flat_hash_map`.
>+  using Base::size;
>+
>+  // flat_hash_map::clear()
>+  //
>+  // Removes all elements from the `flat_hash_map`. Invalidates any references,
>+  // pointers, or iterators referring to contained elements.
>+  //
>+  // NOTE: this operation may shrink the underlying buffer. To avoid shrinking
>+  // the underlying buffer call `erase(begin(), end())`.
>+  using Base::clear;
>+
>+  // flat_hash_map::erase()
>+  //
>+  // Erases elements within the `flat_hash_map`. Erasing does not trigger a
>+  // rehash. Overloads are listed below.
>+  //
>+  // void erase(const_iterator pos):
>+  //
>+  //   Erases the element at `position` of the `flat_hash_map`, returning
>+  //   `void`.
>+  //
>+  //   NOTE: this return behavior is different than that of STL containers in
>+  //   general and `std::unordered_map` in particular.
>+  //
>+  // iterator erase(const_iterator first, const_iterator last):
>+  //
>+  //   Erases the elements in the open interval [`first`, `last`), returning an
>+  //   iterator pointing to `last`.
>+  //
>+  // size_type erase(const key_type& key):
>+  //
>+  //   Erases the element with the matching key, if it exists.
>+  using Base::erase;
>+
>+  // flat_hash_map::insert()
>+  //
>+  // Inserts an element of the specified value into the `flat_hash_map`,
>+  // returning an iterator pointing to the newly inserted element, provided that
>+  // an element with the given key does not already exist. If rehashing occurs
>+  // due to the insertion, all iterators are invalidated. Overloads are listed
>+  // below.
>+  //
>+  // std::pair<iterator,bool> insert(const init_type& value):
>+  //
>+  //   Inserts a value into the `flat_hash_map`. Returns a pair consisting of an
>+  //   iterator to the inserted element (or to the element that prevented the
>+  //   insertion) and a bool denoting whether the insertion took place.
>+  //
>+  // std::pair<iterator,bool> insert(T&& value):
>+  // std::pair<iterator,bool> insert(init_type&& value):
>+  //
>+  //   Inserts a moveable value into the `flat_hash_map`. Returns a pair
>+  //   consisting of an iterator to the inserted element (or to the element that
>+  //   prevented the insertion) and a bool denoting whether the insertion took
>+  //   place.
>+  //
>+  // iterator insert(const_iterator hint, const init_type& value):
>+  // iterator insert(const_iterator hint, T&& value):
>+  // iterator insert(const_iterator hint, init_type&& value);
>+  //
>+  //   Inserts a value, using the position of `hint` as a non-binding suggestion
>+  //   for where to begin the insertion search. Returns an iterator to the
>+  //   inserted element, or to the existing element that prevented the
>+  //   insertion.
>+  //
>+  // void insert(InputIterator first, InputIterator last):
>+  //
>+  //   Inserts a range of values [`first`, `last`).
>+  //
>+  //   NOTE: Although the STL does not specify which element may be inserted if
>+  //   multiple keys compare equivalently, for `flat_hash_map` we guarantee the
>+  //   first match is inserted.
>+  //
>+  // void insert(std::initializer_list<init_type> ilist):
>+  //
>+  //   Inserts the elements within the initializer list `ilist`.
>+  //
>+  //   NOTE: Although the STL does not specify which element may be inserted if
>+  //   multiple keys compare equivalently within the initializer list, for
>+  //   `flat_hash_map` we guarantee the first match is inserted.
>+  using Base::insert;
>+
>+  // flat_hash_map::insert_or_assign()
>+  //
>+  // Inserts an element of the specified value into the `flat_hash_map` provided
>+  // that a value with the given key does not already exist, or replaces it with
>+  // the element value if a key for that value already exists, returning an
>+  // iterator pointing to the newly inserted element.  If rehashing occurs due
>+  // to the insertion, all existing iterators are invalidated. Overloads are
>+  // listed below.
>+  //
>+  // pair<iterator, bool> insert_or_assign(const init_type& k, T&& obj):
>+  // pair<iterator, bool> insert_or_assign(init_type&& k, T&& obj):
>+  //
>+  //   Inserts/Assigns (or moves) the element of the specified key into the
>+  //   `flat_hash_map`.
>+  //
>+  // iterator insert_or_assign(const_iterator hint,
>+  //                           const init_type& k, T&& obj):
>+  // iterator insert_or_assign(const_iterator hint, init_type&& k, T&& obj):
>+  //
>+  //   Inserts/Assigns (or moves) the element of the specified key into the
>+  //   `flat_hash_map` using the position of `hint` as a non-binding suggestion
>+  //   for where to begin the insertion search.
>+  using Base::insert_or_assign;
>+
>+  // flat_hash_map::emplace()
>+  //
>+  // Inserts an element of the specified value by constructing it in-place
>+  // within the `flat_hash_map`, provided that no element with the given key
>+  // already exists.
>+  //
>+  // The element may be constructed even if there already is an element with the
>+  // key in the container, in which case the newly constructed element will be
>+  // destroyed immediately. Prefer `try_emplace()` unless your key is not
>+  // copyable or moveable.
>+  //
>+  // If rehashing occurs due to the insertion, all iterators are invalidated.
>+  using Base::emplace;
>+
>+  // flat_hash_map::emplace_hint()
>+  //
>+  // Inserts an element of the specified value by constructing it in-place
>+  // within the `flat_hash_map`, using the position of `hint` as a non-binding
>+  // suggestion for where to begin the insertion search, and only inserts
>+  // provided that no element with the given key already exists.
>+  //
>+  // The element may be constructed even if there already is an element with the
>+  // key in the container, in which case the newly constructed element will be
>+  // destroyed immediately. Prefer `try_emplace()` unless your key is not
>+  // copyable or moveable.
>+  //
>+  // If rehashing occurs due to the insertion, all iterators are invalidated.
>+  using Base::emplace_hint;
>+
>+  // flat_hash_map::try_emplace()
>+  //
>+  // Inserts an element of the specified value by constructing it in-place
>+  // within the `flat_hash_map`, provided that no element with the given key
>+  // already exists. Unlike `emplace()`, if an element with the given key
>+  // already exists, we guarantee that no element is constructed.
>+  //
>+  // If rehashing occurs due to the insertion, all iterators are invalidated.
>+  // Overloads are listed below.
>+  //
>+  //   pair<iterator, bool> try_emplace(const key_type& k, Args&&... args):
>+  //   pair<iterator, bool> try_emplace(key_type&& k, Args&&... args):
>+  //
>+  // Inserts (via copy or move) the element of the specified key into the
>+  // `flat_hash_map`.
>+  //
>+  //   iterator try_emplace(const_iterator hint,
>+  //                        const init_type& k, Args&&... args):
>+  //   iterator try_emplace(const_iterator hint, init_type&& k, Args&&... args):
>+  //
>+  // Inserts (via copy or move) the element of the specified key into the
>+  // `flat_hash_map` using the position of `hint` as a non-binding suggestion
>+  // for where to begin the insertion search.
>+  using Base::try_emplace;
>+
>+  // flat_hash_map::extract()
>+  //
>+  // Extracts the indicated element, erasing it in the process, and returns it
>+  // as a C++17-compatible node handle. Overloads are listed below.
>+  //
>+  // node_type extract(const_iterator position):
>+  //
>+  //   Extracts the key,value pair of the element at the indicated position and
>+  //   returns a node handle owning that extracted data.
>+  //
>+  // node_type extract(const key_type& x):
>+  //
>+  //   Extracts the key,value pair of the element with a key matching the passed
>+  //   key value and returns a node handle owning that extracted data. If the
>+  //   `flat_hash_map` does not contain an element with a matching key, this
>+  //   function returns an empty node handle.
>+  using Base::extract;
>+
>+  // flat_hash_map::merge()
>+  //
>+  // Extracts elements from a given `source` flat hash map into this
>+  // `flat_hash_map`. If the destination `flat_hash_map` already contains an
>+  // element with an equivalent key, that element is not extracted.
>+  using Base::merge;
>+
>+  // flat_hash_map::swap(flat_hash_map& other)
>+  //
>+  // Exchanges the contents of this `flat_hash_map` with those of the `other`
>+  // flat hash map, avoiding invocation of any move, copy, or swap operations on
>+  // individual elements.
>+  //
>+  // All iterators and references on the `flat_hash_map` remain valid, excepting
>+  // for the past-the-end iterator, which is invalidated.
>+  //
>+  // `swap()` requires that the flat hash map's hashing and key equivalence
>+  // functions be Swappable, and are exchaged using unqualified calls to
>+  // non-member `swap()`. If the map's allocator has
>+  // `std::allocator_traits<allocator_type>::propagate_on_container_swap::value`
>+  // set to `true`, the allocators are also exchanged using an unqualified call
>+  // to non-member `swap()`; otherwise, the allocators are not swapped.
>+  using Base::swap;
>+
>+  // flat_hash_map::rehash(count)
>+  //
>+  // Rehashes the `flat_hash_map`, setting the number of slots to be at least
>+  // the passed value. If the new number of slots increases the load factor more
>+  // than the current maximum load factor
>+  // (`count` < `size()` / `max_load_factor()`), then the new number of slots
>+  // will be at least `size()` / `max_load_factor()`.
>+  //
>+  // To force a rehash, pass rehash(0).
>+  //
>+  // NOTE: unlike behavior in `std::unordered_map`, references are also
>+  // invalidated upon a `rehash()`.
>+  using Base::rehash;
>+
>+  // flat_hash_map::reserve(count)
>+  //
>+  // Sets the number of slots in the `flat_hash_map` to the number needed to
>+  // accommodate at least `count` total elements without exceeding the current
>+  // maximum load factor, and may rehash the container if needed.
>+  using Base::reserve;
>+
>+  // flat_hash_map::at()
>+  //
>+  // Returns a reference to the mapped value of the element with key equivalent
>+  // to the passed key.
>+  using Base::at;
>+
>+  // flat_hash_map::contains()
>+  //
>+  // Determines whether an element with a key comparing equal to the given `key`
>+  // exists within the `flat_hash_map`, returning `true` if so or `false`
>+  // otherwise.
>+  using Base::contains;
>+
>+  // flat_hash_map::count(const Key& key) const
>+  //
>+  // Returns the number of elements with a key comparing equal to the given
>+  // `key` within the `flat_hash_map`. note that this function will return
>+  // either `1` or `0` since duplicate keys are not allowed within a
>+  // `flat_hash_map`.
>+  using Base::count;
>+
>+  // flat_hash_map::equal_range()
>+  //
>+  // Returns a closed range [first, last], defined by a `std::pair` of two
>+  // iterators, containing all elements with the passed key in the
>+  // `flat_hash_map`.
>+  using Base::equal_range;
>+
>+  // flat_hash_map::find()
>+  //
>+  // Finds an element with the passed `key` within the `flat_hash_map`.
>+  using Base::find;
>+
>+  // flat_hash_map::operator[]()
>+  //
>+  // Returns a reference to the value mapped to the passed key within the
>+  // `flat_hash_map`, performing an `insert()` if the key does not already
>+  // exist.
>+  //
>+  // If an insertion occurs and results in a rehashing of the container, all
>+  // iterators are invalidated. Otherwise iterators are not affected and
>+  // references are not invalidated. Overloads are listed below.
>+  //
>+  // T& operator[](const Key& key):
>+  //
>+  //   Inserts an init_type object constructed in-place if the element with the
>+  //   given key does not exist.
>+  //
>+  // T& operator[](Key&& key):
>+  //
>+  //   Inserts an init_type object constructed in-place provided that an element
>+  //   with the given key does not exist.
>+  using Base::operator[];
>+
>+  // flat_hash_map::bucket_count()
>+  //
>+  // Returns the number of "buckets" within the `flat_hash_map`. Note that
>+  // because a flat hash map contains all elements within its internal storage,
>+  // this value simply equals the current capacity of the `flat_hash_map`.
>+  using Base::bucket_count;
>+
>+  // flat_hash_map::load_factor()
>+  //
>+  // Returns the current load factor of the `flat_hash_map` (the average number
>+  // of slots occupied with a value within the hash map).
>+  using Base::load_factor;
>+
>+  // flat_hash_map::max_load_factor()
>+  //
>+  // Manages the maximum load factor of the `flat_hash_map`. Overloads are
>+  // listed below.
>+  //
>+  // float flat_hash_map::max_load_factor()
>+  //
>+  //   Returns the current maximum load factor of the `flat_hash_map`.
>+  //
>+  // void flat_hash_map::max_load_factor(float ml)
>+  //
>+  //   Sets the maximum load factor of the `flat_hash_map` to the passed value.
>+  //
>+  //   NOTE: This overload is provided only for API compatibility with the STL;
>+  //   `flat_hash_map` will ignore any set load factor and manage its rehashing
>+  //   internally as an implementation detail.
>+  using Base::max_load_factor;
>+
>+  // flat_hash_map::get_allocator()
>+  //
>+  // Returns the allocator function associated with this `flat_hash_map`.
>+  using Base::get_allocator;
>+
>+  // flat_hash_map::hash_function()
>+  //
>+  // Returns the hashing function used to hash the keys within this
>+  // `flat_hash_map`.
>+  using Base::hash_function;
>+
>+  // flat_hash_map::key_eq()
>+  //
>+  // Returns the function used for comparing keys equality.
>+  using Base::key_eq;
>+};
>+
>+namespace container_internal {
>+
>+template <class K, class V>
>+struct FlatHashMapPolicy {
>+  using slot_type = container_internal::slot_type<K, V>;
>+  using key_type = K;
>+  using mapped_type = V;
>+  using init_type = std::pair</*non const*/ key_type, mapped_type>;
>+
>+  template <class Allocator, class... Args>
>+  static void construct(Allocator* alloc, slot_type* slot, Args&&... args) {
>+    slot_type::construct(alloc, slot, std::forward<Args>(args)...);
>+  }
>+
>+  template <class Allocator>
>+  static void destroy(Allocator* alloc, slot_type* slot) {
>+    slot_type::destroy(alloc, slot);
>+  }
>+
>+  template <class Allocator>
>+  static void transfer(Allocator* alloc, slot_type* new_slot,
>+                       slot_type* old_slot) {
>+    slot_type::transfer(alloc, new_slot, old_slot);
>+  }
>+
>+  template <class F, class... Args>
>+  static decltype(absl::container_internal::DecomposePair(
>+      std::declval<F>(), std::declval<Args>()...))
>+  apply(F&& f, Args&&... args) {
>+    return absl::container_internal::DecomposePair(std::forward<F>(f),
>+                                                   std::forward<Args>(args)...);
>+  }
>+
>+  static size_t space_used(const slot_type*) { return 0; }
>+
>+  static std::pair<const K, V>& element(slot_type* slot) { return slot->value; }
>+
>+  static V& value(std::pair<const K, V>* kv) { return kv->second; }
>+  static const V& value(const std::pair<const K, V>* kv) { return kv->second; }
>+};
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+#endif  // ABSL_CONTAINER_FLAT_HASH_MAP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_map_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_map_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..10a781ffd6c7d9d149086e443174a7aa76ad22ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_map_test.cc
>@@ -0,0 +1,241 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/flat_hash_map.h"
>+
>+#include "absl/container/internal/hash_generator_testing.h"
>+#include "absl/container/internal/unordered_map_constructor_test.h"
>+#include "absl/container/internal/unordered_map_lookup_test.h"
>+#include "absl/container/internal/unordered_map_modifiers_test.h"
>+#include "absl/types/any.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+using ::absl::container_internal::hash_internal::Enum;
>+using ::absl::container_internal::hash_internal::EnumClass;
>+using ::testing::_;
>+using ::testing::Pair;
>+using ::testing::UnorderedElementsAre;
>+
>+template <class K, class V>
>+using Map =
>+    flat_hash_map<K, V, StatefulTestingHash, StatefulTestingEqual, Alloc<>>;
>+
>+static_assert(!std::is_standard_layout<NonStandardLayout>(), "");
>+
>+using MapTypes =
>+    ::testing::Types<Map<int, int>, Map<std::string, int>, Map<Enum, std::string>,
>+                     Map<EnumClass, int>, Map<int, NonStandardLayout>,
>+                     Map<NonStandardLayout, int>>;
>+
>+INSTANTIATE_TYPED_TEST_CASE_P(FlatHashMap, ConstructorTest, MapTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(FlatHashMap, LookupTest, MapTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(FlatHashMap, ModifiersTest, MapTypes);
>+
>+TEST(FlatHashMap, StandardLayout) {
>+  struct Int {
>+    explicit Int(size_t value) : value(value) {}
>+    Int() : value(0) { ADD_FAILURE(); }
>+    Int(const Int& other) : value(other.value) { ADD_FAILURE(); }
>+    Int(Int&&) = default;
>+    bool operator==(const Int& other) const { return value == other.value; }
>+    size_t value;
>+  };
>+  static_assert(std::is_standard_layout<Int>(), "");
>+
>+  struct Hash {
>+    size_t operator()(const Int& obj) const { return obj.value; }
>+  };
>+
>+  // Verify that neither the key nor the value get default-constructed or
>+  // copy-constructed.
>+  {
>+    flat_hash_map<Int, Int, Hash> m;
>+    m.try_emplace(Int(1), Int(2));
>+    m.try_emplace(Int(3), Int(4));
>+    m.erase(Int(1));
>+    m.rehash(2 * m.bucket_count());
>+  }
>+  {
>+    flat_hash_map<Int, Int, Hash> m;
>+    m.try_emplace(Int(1), Int(2));
>+    m.try_emplace(Int(3), Int(4));
>+    m.erase(Int(1));
>+    m.clear();
>+  }
>+}
>+
>+// gcc becomes unhappy if this is inside the method, so pull it out here.
>+struct balast {};
>+
>+TEST(FlatHashMap, IteratesMsan) {
>+  // Because SwissTable randomizes on pointer addresses, we keep old tables
>+  // around to ensure we don't reuse old memory.
>+  std::vector<absl::flat_hash_map<int, balast>> garbage;
>+  for (int i = 0; i < 100; ++i) {
>+    absl::flat_hash_map<int, balast> t;
>+    for (int j = 0; j < 100; ++j) {
>+      t[j];
>+      for (const auto& p : t) EXPECT_THAT(p, Pair(_, _));
>+    }
>+    garbage.push_back(std::move(t));
>+  }
>+}
>+
>+// Demonstration of the "Lazy Key" pattern.  This uses heterogenous insert to
>+// avoid creating expensive key elements when the item is already present in the
>+// map.
>+struct LazyInt {
>+  explicit LazyInt(size_t value, int* tracker)
>+      : value(value), tracker(tracker) {}
>+
>+  explicit operator size_t() const {
>+    ++*tracker;
>+    return value;
>+  }
>+
>+  size_t value;
>+  int* tracker;
>+};
>+
>+struct Hash {
>+  using is_transparent = void;
>+  int* tracker;
>+  size_t operator()(size_t obj) const {
>+    ++*tracker;
>+    return obj;
>+  }
>+  size_t operator()(const LazyInt& obj) const {
>+    ++*tracker;
>+    return obj.value;
>+  }
>+};
>+
>+struct Eq {
>+  using is_transparent = void;
>+  bool operator()(size_t lhs, size_t rhs) const {
>+    return lhs == rhs;
>+  }
>+  bool operator()(size_t lhs, const LazyInt& rhs) const {
>+    return lhs == rhs.value;
>+  }
>+};
>+
>+TEST(FlatHashMap, LazyKeyPattern) {
>+  // hashes are only guaranteed in opt mode, we use assertions to track internal
>+  // state that can cause extra calls to hash.
>+  int conversions = 0;
>+  int hashes = 0;
>+  flat_hash_map<size_t, size_t, Hash, Eq> m(0, Hash{&hashes});
>+
>+  m[LazyInt(1, &conversions)] = 1;
>+  EXPECT_THAT(m, UnorderedElementsAre(Pair(1, 1)));
>+  EXPECT_EQ(conversions, 1);
>+#ifdef NDEBUG
>+  EXPECT_EQ(hashes, 1);
>+#endif
>+
>+  m[LazyInt(1, &conversions)] = 2;
>+  EXPECT_THAT(m, UnorderedElementsAre(Pair(1, 2)));
>+  EXPECT_EQ(conversions, 1);
>+#ifdef NDEBUG
>+  EXPECT_EQ(hashes, 2);
>+#endif
>+
>+  m.try_emplace(LazyInt(2, &conversions), 3);
>+  EXPECT_THAT(m, UnorderedElementsAre(Pair(1, 2), Pair(2, 3)));
>+  EXPECT_EQ(conversions, 2);
>+#ifdef NDEBUG
>+  EXPECT_EQ(hashes, 3);
>+#endif
>+
>+  m.try_emplace(LazyInt(2, &conversions), 4);
>+  EXPECT_THAT(m, UnorderedElementsAre(Pair(1, 2), Pair(2, 3)));
>+  EXPECT_EQ(conversions, 2);
>+#ifdef NDEBUG
>+  EXPECT_EQ(hashes, 4);
>+#endif
>+}
>+
>+TEST(FlatHashMap, BitfieldArgument) {
>+  union {
>+    int n : 1;
>+  };
>+  n = 0;
>+  flat_hash_map<int, int> m;
>+  m.erase(n);
>+  m.count(n);
>+  m.prefetch(n);
>+  m.find(n);
>+  m.contains(n);
>+  m.equal_range(n);
>+  m.insert_or_assign(n, n);
>+  m.insert_or_assign(m.end(), n, n);
>+  m.try_emplace(n);
>+  m.try_emplace(m.end(), n);
>+  m.at(n);
>+  m[n];
>+}
>+
>+TEST(FlatHashMap, MergeExtractInsert) {
>+  // We can't test mutable keys, or non-copyable keys with flat_hash_map.
>+  // Test that the nodes have the proper API.
>+  absl::flat_hash_map<int, int> m = {{1, 7}, {2, 9}};
>+  auto node = m.extract(1);
>+  EXPECT_TRUE(node);
>+  EXPECT_EQ(node.key(), 1);
>+  EXPECT_EQ(node.mapped(), 7);
>+  EXPECT_THAT(m, UnorderedElementsAre(Pair(2, 9)));
>+
>+  node.mapped() = 17;
>+  m.insert(std::move(node));
>+  EXPECT_THAT(m, UnorderedElementsAre(Pair(1, 17), Pair(2, 9)));
>+}
>+#if !defined(__ANDROID__) && !defined(__APPLE__) && !defined(__EMSCRIPTEN__)
>+TEST(FlatHashMap, Any) {
>+  absl::flat_hash_map<int, absl::any> m;
>+  m.emplace(1, 7);
>+  auto it = m.find(1);
>+  ASSERT_NE(it, m.end());
>+  EXPECT_EQ(7, absl::any_cast<int>(it->second));
>+
>+  m.emplace(std::piecewise_construct, std::make_tuple(2), std::make_tuple(8));
>+  it = m.find(2);
>+  ASSERT_NE(it, m.end());
>+  EXPECT_EQ(8, absl::any_cast<int>(it->second));
>+
>+  m.emplace(std::piecewise_construct, std::make_tuple(3),
>+            std::make_tuple(absl::any(9)));
>+  it = m.find(3);
>+  ASSERT_NE(it, m.end());
>+  EXPECT_EQ(9, absl::any_cast<int>(it->second));
>+
>+  struct H {
>+    size_t operator()(const absl::any&) const { return 0; }
>+  };
>+  struct E {
>+    bool operator()(const absl::any&, const absl::any&) const { return true; }
>+  };
>+  absl::flat_hash_map<absl::any, int, H, E> m2;
>+  m2.emplace(1, 7);
>+  auto it2 = m2.find(1);
>+  ASSERT_NE(it2, m2.end());
>+  EXPECT_EQ(7, it2->second);
>+}
>+#endif  // __ANDROID__
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_set.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_set.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..98aead1a8ae209e2a45a956167be95e7270c544b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_set.h
>@@ -0,0 +1,439 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: flat_hash_set.h
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::flat_hash_set<T>` is an unordered associative container designed to
>+// be a more efficient replacement for `std::unordered_set`. Like
>+// `unordered_set`, search, insertion, and deletion of set elements can be done
>+// as an `O(1)` operation. However, `flat_hash_set` (and other unordered
>+// associative containers known as the collection of Abseil "Swiss tables")
>+// contain other optimizations that result in both memory and computation
>+// advantages.
>+//
>+// In most cases, your default choice for a hash set should be a set of type
>+// `flat_hash_set`.
>+#ifndef ABSL_CONTAINER_FLAT_HASH_SET_H_
>+#define ABSL_CONTAINER_FLAT_HASH_SET_H_
>+
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/base/macros.h"
>+#include "absl/container/internal/container_memory.h"
>+#include "absl/container/internal/hash_function_defaults.h"  // IWYU pragma: export
>+#include "absl/container/internal/raw_hash_set.h"  // IWYU pragma: export
>+#include "absl/memory/memory.h"
>+
>+namespace absl {
>+namespace container_internal {
>+template <typename T>
>+struct FlatHashSetPolicy;
>+}  // namespace container_internal
>+
>+// -----------------------------------------------------------------------------
>+// absl::flat_hash_set
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::flat_hash_set<T>` is an unordered associative container which has
>+// been optimized for both speed and memory footprint in most common use cases.
>+// Its interface is similar to that of `std::unordered_set<T>` with the
>+// following notable differences:
>+//
>+// * Requires keys that are CopyConstructible
>+// * Supports heterogeneous lookup, through `find()`, `operator[]()` and
>+//   `insert()`, provided that the set is provided a compatible heterogeneous
>+//   hashing function and equality operator.
>+// * Invalidates any references and pointers to elements within the table after
>+//   `rehash()`.
>+// * Contains a `capacity()` member function indicating the number of element
>+//   slots (open, deleted, and empty) within the hash set.
>+// * Returns `void` from the `erase(iterator)` overload.
>+//
>+// By default, `flat_hash_set` uses the `absl::Hash` hashing framework. All
>+// fundamental and Abseil types that support the `absl::Hash` framework have a
>+// compatible equality operator for comparing insertions into `flat_hash_map`.
>+// If your type is not yet supported by the `asbl::Hash` framework, see
>+// absl/hash/hash.h for information on extending Abseil hashing to user-defined
>+// types.
>+//
>+// NOTE: A `flat_hash_set` stores its keys directly inside its implementation
>+// array to avoid memory indirection. Because a `flat_hash_set` is designed to
>+// move data when rehashed, set keys will not retain pointer stability. If you
>+// require pointer stability, consider using
>+// `absl::flat_hash_set<std::unique_ptr<T>>`. If your type is not moveable and
>+// you require pointer stability, consider `absl::node_hash_set` instead.
>+//
>+// Example:
>+//
>+//   // Create a flat hash set of three strings
>+//   absl::flat_hash_set<std::string> ducks =
>+//     {"huey", "dewey", "louie"};
>+//
>+//  // Insert a new element into the flat hash set
>+//  ducks.insert("donald"};
>+//
>+//  // Force a rehash of the flat hash set
>+//  ducks.rehash(0);
>+//
>+//  // See if "dewey" is present
>+//  if (ducks.contains("dewey")) {
>+//    std::cout << "We found dewey!" << std::endl;
>+//  }
>+template <class T, class Hash = absl::container_internal::hash_default_hash<T>,
>+          class Eq = absl::container_internal::hash_default_eq<T>,
>+          class Allocator = std::allocator<T>>
>+class flat_hash_set
>+    : public absl::container_internal::raw_hash_set<
>+          absl::container_internal::FlatHashSetPolicy<T>, Hash, Eq, Allocator> {
>+  using Base = typename flat_hash_set::raw_hash_set;
>+
>+ public:
>+  flat_hash_set() {}
>+  using Base::Base;
>+
>+  // flat_hash_set::begin()
>+  //
>+  // Returns an iterator to the beginning of the `flat_hash_set`.
>+  using Base::begin;
>+
>+  // flat_hash_set::cbegin()
>+  //
>+  // Returns a const iterator to the beginning of the `flat_hash_set`.
>+  using Base::cbegin;
>+
>+  // flat_hash_set::cend()
>+  //
>+  // Returns a const iterator to the end of the `flat_hash_set`.
>+  using Base::cend;
>+
>+  // flat_hash_set::end()
>+  //
>+  // Returns an iterator to the end of the `flat_hash_set`.
>+  using Base::end;
>+
>+  // flat_hash_set::capacity()
>+  //
>+  // Returns the number of element slots (assigned, deleted, and empty)
>+  // available within the `flat_hash_set`.
>+  //
>+  // NOTE: this member function is particular to `absl::flat_hash_set` and is
>+  // not provided in the `std::unordered_map` API.
>+  using Base::capacity;
>+
>+  // flat_hash_set::empty()
>+  //
>+  // Returns whether or not the `flat_hash_set` is empty.
>+  using Base::empty;
>+
>+  // flat_hash_set::max_size()
>+  //
>+  // Returns the largest theoretical possible number of elements within a
>+  // `flat_hash_set` under current memory constraints. This value can be thought
>+  // of the largest value of `std::distance(begin(), end())` for a
>+  // `flat_hash_set<T>`.
>+  using Base::max_size;
>+
>+  // flat_hash_set::size()
>+  //
>+  // Returns the number of elements currently within the `flat_hash_set`.
>+  using Base::size;
>+
>+  // flat_hash_set::clear()
>+  //
>+  // Removes all elements from the `flat_hash_set`. Invalidates any references,
>+  // pointers, or iterators referring to contained elements.
>+  //
>+  // NOTE: this operation may shrink the underlying buffer. To avoid shrinking
>+  // the underlying buffer call `erase(begin(), end())`.
>+  using Base::clear;
>+
>+  // flat_hash_set::erase()
>+  //
>+  // Erases elements within the `flat_hash_set`. Erasing does not trigger a
>+  // rehash. Overloads are listed below.
>+  //
>+  // void erase(const_iterator pos):
>+  //
>+  //   Erases the element at `position` of the `flat_hash_set`, returning
>+  //   `void`.
>+  //
>+  //   NOTE: this return behavior is different than that of STL containers in
>+  //   general and `std::unordered_map` in particular.
>+  //
>+  // iterator erase(const_iterator first, const_iterator last):
>+  //
>+  //   Erases the elements in the open interval [`first`, `last`), returning an
>+  //   iterator pointing to `last`.
>+  //
>+  // size_type erase(const key_type& key):
>+  //
>+  //   Erases the element with the matching key, if it exists.
>+  using Base::erase;
>+
>+  // flat_hash_set::insert()
>+  //
>+  // Inserts an element of the specified value into the `flat_hash_set`,
>+  // returning an iterator pointing to the newly inserted element, provided that
>+  // an element with the given key does not already exist. If rehashing occurs
>+  // due to the insertion, all iterators are invalidated. Overloads are listed
>+  // below.
>+  //
>+  // std::pair<iterator,bool> insert(const T& value):
>+  //
>+  //   Inserts a value into the `flat_hash_set`. Returns a pair consisting of an
>+  //   iterator to the inserted element (or to the element that prevented the
>+  //   insertion) and a bool denoting whether the insertion took place.
>+  //
>+  // std::pair<iterator,bool> insert(T&& value):
>+  //
>+  //   Inserts a moveable value into the `flat_hash_set`. Returns a pair
>+  //   consisting of an iterator to the inserted element (or to the element that
>+  //   prevented the insertion) and a bool denoting whether the insertion took
>+  //   place.
>+  //
>+  // iterator insert(const_iterator hint, const T& value):
>+  // iterator insert(const_iterator hint, T&& value):
>+  //
>+  //   Inserts a value, using the position of `hint` as a non-binding suggestion
>+  //   for where to begin the insertion search. Returns an iterator to the
>+  //   inserted element, or to the existing element that prevented the
>+  //   insertion.
>+  //
>+  // void insert(InputIterator first, InputIterator last):
>+  //
>+  //   Inserts a range of values [`first`, `last`).
>+  //
>+  //   NOTE: Although the STL does not specify which element may be inserted if
>+  //   multiple keys compare equivalently, for `flat_hash_set` we guarantee the
>+  //   first match is inserted.
>+  //
>+  // void insert(std::initializer_list<T> ilist):
>+  //
>+  //   Inserts the elements within the initializer list `ilist`.
>+  //
>+  //   NOTE: Although the STL does not specify which element may be inserted if
>+  //   multiple keys compare equivalently within the initializer list, for
>+  //   `flat_hash_set` we guarantee the first match is inserted.
>+  using Base::insert;
>+
>+  // flat_hash_set::emplace()
>+  //
>+  // Inserts an element of the specified value by constructing it in-place
>+  // within the `flat_hash_set`, provided that no element with the given key
>+  // already exists.
>+  //
>+  // The element may be constructed even if there already is an element with the
>+  // key in the container, in which case the newly constructed element will be
>+  // destroyed immediately. Prefer `try_emplace()` unless your key is not
>+  // copyable or moveable.
>+  //
>+  // If rehashing occurs due to the insertion, all iterators are invalidated.
>+  using Base::emplace;
>+
>+  // flat_hash_set::emplace_hint()
>+  //
>+  // Inserts an element of the specified value by constructing it in-place
>+  // within the `flat_hash_set`, using the position of `hint` as a non-binding
>+  // suggestion for where to begin the insertion search, and only inserts
>+  // provided that no element with the given key already exists.
>+  //
>+  // The element may be constructed even if there already is an element with the
>+  // key in the container, in which case the newly constructed element will be
>+  // destroyed immediately. Prefer `try_emplace()` unless your key is not
>+  // copyable or moveable.
>+  //
>+  // If rehashing occurs due to the insertion, all iterators are invalidated.
>+  using Base::emplace_hint;
>+
>+  // flat_hash_set::extract()
>+  //
>+  // Extracts the indicated element, erasing it in the process, and returns it
>+  // as a C++17-compatible node handle. Overloads are listed below.
>+  //
>+  // node_type extract(const_iterator position):
>+  //
>+  //   Extracts the element at the indicated position and returns a node handle
>+  //   owning that extracted data.
>+  //
>+  // node_type extract(const key_type& x):
>+  //
>+  //   Extracts the element with the key matching the passed key value and
>+  //   returns a node handle owning that extracted data. If the `flat_hash_set`
>+  //   does not contain an element with a matching key, this function returns an
>+  //   empty node handle.
>+  using Base::extract;
>+
>+  // flat_hash_set::merge()
>+  //
>+  // Extracts elements from a given `source` flat hash map into this
>+  // `flat_hash_set`. If the destination `flat_hash_set` already contains an
>+  // element with an equivalent key, that element is not extracted.
>+  using Base::merge;
>+
>+  // flat_hash_set::swap(flat_hash_set& other)
>+  //
>+  // Exchanges the contents of this `flat_hash_set` with those of the `other`
>+  // flat hash map, avoiding invocation of any move, copy, or swap operations on
>+  // individual elements.
>+  //
>+  // All iterators and references on the `flat_hash_set` remain valid, excepting
>+  // for the past-the-end iterator, which is invalidated.
>+  //
>+  // `swap()` requires that the flat hash set's hashing and key equivalence
>+  // functions be Swappable, and are exchaged using unqualified calls to
>+  // non-member `swap()`. If the map's allocator has
>+  // `std::allocator_traits<allocator_type>::propagate_on_container_swap::value`
>+  // set to `true`, the allocators are also exchanged using an unqualified call
>+  // to non-member `swap()`; otherwise, the allocators are not swapped.
>+  using Base::swap;
>+
>+  // flat_hash_set::rehash(count)
>+  //
>+  // Rehashes the `flat_hash_set`, setting the number of slots to be at least
>+  // the passed value. If the new number of slots increases the load factor more
>+  // than the current maximum load factor
>+  // (`count` < `size()` / `max_load_factor()`), then the new number of slots
>+  // will be at least `size()` / `max_load_factor()`.
>+  //
>+  // To force a rehash, pass rehash(0).
>+  //
>+  // NOTE: unlike behavior in `std::unordered_set`, references are also
>+  // invalidated upon a `rehash()`.
>+  using Base::rehash;
>+
>+  // flat_hash_set::reserve(count)
>+  //
>+  // Sets the number of slots in the `flat_hash_set` to the number needed to
>+  // accommodate at least `count` total elements without exceeding the current
>+  // maximum load factor, and may rehash the container if needed.
>+  using Base::reserve;
>+
>+  // flat_hash_set::contains()
>+  //
>+  // Determines whether an element comparing equal to the given `key` exists
>+  // within the `flat_hash_set`, returning `true` if so or `false` otherwise.
>+  using Base::contains;
>+
>+  // flat_hash_set::count(const Key& key) const
>+  //
>+  // Returns the number of elements comparing equal to the given `key` within
>+  // the `flat_hash_set`. note that this function will return either `1` or `0`
>+  // since duplicate elements are not allowed within a `flat_hash_set`.
>+  using Base::count;
>+
>+  // flat_hash_set::equal_range()
>+  //
>+  // Returns a closed range [first, last], defined by a `std::pair` of two
>+  // iterators, containing all elements with the passed key in the
>+  // `flat_hash_set`.
>+  using Base::equal_range;
>+
>+  // flat_hash_set::find()
>+  //
>+  // Finds an element with the passed `key` within the `flat_hash_set`.
>+  using Base::find;
>+
>+  // flat_hash_set::bucket_count()
>+  //
>+  // Returns the number of "buckets" within the `flat_hash_set`. Note that
>+  // because a flat hash map contains all elements within its internal storage,
>+  // this value simply equals the current capacity of the `flat_hash_set`.
>+  using Base::bucket_count;
>+
>+  // flat_hash_set::load_factor()
>+  //
>+  // Returns the current load factor of the `flat_hash_set` (the average number
>+  // of slots occupied with a value within the hash map).
>+  using Base::load_factor;
>+
>+  // flat_hash_set::max_load_factor()
>+  //
>+  // Manages the maximum load factor of the `flat_hash_set`. Overloads are
>+  // listed below.
>+  //
>+  // float flat_hash_set::max_load_factor()
>+  //
>+  //   Returns the current maximum load factor of the `flat_hash_set`.
>+  //
>+  // void flat_hash_set::max_load_factor(float ml)
>+  //
>+  //   Sets the maximum load factor of the `flat_hash_set` to the passed value.
>+  //
>+  //   NOTE: This overload is provided only for API compatibility with the STL;
>+  //   `flat_hash_set` will ignore any set load factor and manage its rehashing
>+  //   internally as an implementation detail.
>+  using Base::max_load_factor;
>+
>+  // flat_hash_set::get_allocator()
>+  //
>+  // Returns the allocator function associated with this `flat_hash_set`.
>+  using Base::get_allocator;
>+
>+  // flat_hash_set::hash_function()
>+  //
>+  // Returns the hashing function used to hash the keys within this
>+  // `flat_hash_set`.
>+  using Base::hash_function;
>+
>+  // flat_hash_set::key_eq()
>+  //
>+  // Returns the function used for comparing keys equality.
>+  using Base::key_eq;
>+};
>+
>+namespace container_internal {
>+
>+template <class T>
>+struct FlatHashSetPolicy {
>+  using slot_type = T;
>+  using key_type = T;
>+  using init_type = T;
>+  using constant_iterators = std::true_type;
>+
>+  template <class Allocator, class... Args>
>+  static void construct(Allocator* alloc, slot_type* slot, Args&&... args) {
>+    absl::allocator_traits<Allocator>::construct(*alloc, slot,
>+                                                 std::forward<Args>(args)...);
>+  }
>+
>+  template <class Allocator>
>+  static void destroy(Allocator* alloc, slot_type* slot) {
>+    absl::allocator_traits<Allocator>::destroy(*alloc, slot);
>+  }
>+
>+  template <class Allocator>
>+  static void transfer(Allocator* alloc, slot_type* new_slot,
>+                       slot_type* old_slot) {
>+    construct(alloc, new_slot, std::move(*old_slot));
>+    destroy(alloc, old_slot);
>+  }
>+
>+  static T& element(slot_type* slot) { return *slot; }
>+
>+  template <class F, class... Args>
>+  static decltype(absl::container_internal::DecomposeValue(
>+      std::declval<F>(), std::declval<Args>()...))
>+  apply(F&& f, Args&&... args) {
>+    return absl::container_internal::DecomposeValue(
>+        std::forward<F>(f), std::forward<Args>(args)...);
>+  }
>+
>+  static size_t space_used(const T*) { return 0; }
>+};
>+}  // namespace container_internal
>+}  // namespace absl
>+#endif  // ABSL_CONTAINER_FLAT_HASH_SET_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_set_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_set_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..e52fd532cbeec75158be6205e561181cb174d4bb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/flat_hash_set_test.cc
>@@ -0,0 +1,126 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/flat_hash_set.h"
>+
>+#include <vector>
>+
>+#include "absl/container/internal/hash_generator_testing.h"
>+#include "absl/container/internal/unordered_set_constructor_test.h"
>+#include "absl/container/internal/unordered_set_lookup_test.h"
>+#include "absl/container/internal/unordered_set_modifiers_test.h"
>+#include "absl/memory/memory.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+using ::absl::container_internal::hash_internal::Enum;
>+using ::absl::container_internal::hash_internal::EnumClass;
>+using ::testing::Pointee;
>+using ::testing::UnorderedElementsAre;
>+using ::testing::UnorderedElementsAreArray;
>+
>+template <class T>
>+using Set =
>+    absl::flat_hash_set<T, StatefulTestingHash, StatefulTestingEqual, Alloc<T>>;
>+
>+using SetTypes =
>+    ::testing::Types<Set<int>, Set<std::string>, Set<Enum>, Set<EnumClass>>;
>+
>+INSTANTIATE_TYPED_TEST_CASE_P(FlatHashSet, ConstructorTest, SetTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(FlatHashSet, LookupTest, SetTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(FlatHashSet, ModifiersTest, SetTypes);
>+
>+TEST(FlatHashSet, EmplaceString) {
>+  std::vector<std::string> v = {"a", "b"};
>+  absl::flat_hash_set<absl::string_view> hs(v.begin(), v.end());
>+  EXPECT_THAT(hs, UnorderedElementsAreArray(v));
>+}
>+
>+TEST(FlatHashSet, BitfieldArgument) {
>+  union {
>+    int n : 1;
>+  };
>+  n = 0;
>+  absl::flat_hash_set<int> s = {n};
>+  s.insert(n);
>+  s.insert(s.end(), n);
>+  s.insert({n});
>+  s.erase(n);
>+  s.count(n);
>+  s.prefetch(n);
>+  s.find(n);
>+  s.contains(n);
>+  s.equal_range(n);
>+}
>+
>+TEST(FlatHashSet, MergeExtractInsert) {
>+  struct Hash {
>+    size_t operator()(const std::unique_ptr<int>& p) const { return *p; }
>+  };
>+  struct Eq {
>+    bool operator()(const std::unique_ptr<int>& a,
>+                    const std::unique_ptr<int>& b) const {
>+      return *a == *b;
>+    }
>+  };
>+  absl::flat_hash_set<std::unique_ptr<int>, Hash, Eq> set1, set2;
>+  set1.insert(absl::make_unique<int>(7));
>+  set1.insert(absl::make_unique<int>(17));
>+
>+  set2.insert(absl::make_unique<int>(7));
>+  set2.insert(absl::make_unique<int>(19));
>+
>+  EXPECT_THAT(set1, UnorderedElementsAre(Pointee(7), Pointee(17)));
>+  EXPECT_THAT(set2, UnorderedElementsAre(Pointee(7), Pointee(19)));
>+
>+  set1.merge(set2);
>+
>+  EXPECT_THAT(set1, UnorderedElementsAre(Pointee(7), Pointee(17), Pointee(19)));
>+  EXPECT_THAT(set2, UnorderedElementsAre(Pointee(7)));
>+
>+  auto node = set1.extract(absl::make_unique<int>(7));
>+  EXPECT_TRUE(node);
>+  EXPECT_THAT(node.value(), Pointee(7));
>+  EXPECT_THAT(set1, UnorderedElementsAre(Pointee(17), Pointee(19)));
>+
>+  auto insert_result = set2.insert(std::move(node));
>+  EXPECT_FALSE(node);
>+  EXPECT_FALSE(insert_result.inserted);
>+  EXPECT_TRUE(insert_result.node);
>+  EXPECT_THAT(insert_result.node.value(), Pointee(7));
>+  EXPECT_EQ(**insert_result.position, 7);
>+  EXPECT_NE(insert_result.position->get(), insert_result.node.value().get());
>+  EXPECT_THAT(set2, UnorderedElementsAre(Pointee(7)));
>+
>+  node = set1.extract(absl::make_unique<int>(17));
>+  EXPECT_TRUE(node);
>+  EXPECT_THAT(node.value(), Pointee(17));
>+  EXPECT_THAT(set1, UnorderedElementsAre(Pointee(19)));
>+
>+  node.value() = absl::make_unique<int>(23);
>+
>+  insert_result = set2.insert(std::move(node));
>+  EXPECT_FALSE(node);
>+  EXPECT_TRUE(insert_result.inserted);
>+  EXPECT_FALSE(insert_result.node);
>+  EXPECT_EQ(**insert_result.position, 23);
>+  EXPECT_THAT(set2, UnorderedElementsAre(Pointee(7), Pointee(23)));
>+}
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector.h
>index ca36fd3699b905c2ede9bb10725b4d9b66986877..12756bb820cf6ae956ca464a8ac7cd732cbc9e1f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector.h
>@@ -620,6 +620,12 @@ class InlinedVector {
>   // Returns the allocator of this inlined vector.
>   allocator_type get_allocator() const { return allocator(); }
> 
>+  template <typename H>
>+  friend H AbslHashValue(H h, const InlinedVector& v) {
>+    return H::combine(H::combine_contiguous(std::move(h), v.data(), v.size()),
>+                      v.size());
>+  }
>+
>  private:
>   static_assert(N > 0, "inlined vector with nonpositive size");
> 
>@@ -648,8 +654,7 @@ class InlinedVector {
>   // our instance of it for free.
>   class AllocatorAndTag : private allocator_type {
>    public:
>-    explicit AllocatorAndTag(const allocator_type& a, Tag t = Tag())
>-        : allocator_type(a), tag_(t) {}
>+    explicit AllocatorAndTag(const allocator_type& a) : allocator_type(a) {}
>     Tag& tag() { return tag_; }
>     const Tag& tag() const { return tag_; }
>     allocator_type& allocator() { return *this; }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_benchmark.cc
>index 24f2174928a38b01fef99ae27cb89e4198265ab3..a3ad0f8ae4f1a163c7f232932e46a38ee4ea670e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_benchmark.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_benchmark.cc
>@@ -66,7 +66,7 @@ BENCHMARK(BM_StdVectorFill)->Range(0, 1024);
> // The purpose of the next two benchmarks is to verify that
> // absl::InlinedVector is efficient when moving is more efficent than
> // copying. To do so, we use strings that are larger than the short
>-// std::string optimization.
>+// string optimization.
> bool StringRepresentedInline(std::string s) {
>   const char* chars = s.data();
>   std::string s1 = std::move(s);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_test.cc
>index 196a1bed976c4bbd060a0c81a30819aced4669b2..08dcd3ef66fb6a3302f862c52ad91f536260d400 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_test.cc
>@@ -31,6 +31,7 @@
> #include "absl/base/internal/raw_logging.h"
> #include "absl/base/macros.h"
> #include "absl/container/internal/test_instance_tracker.h"
>+#include "absl/hash/hash_testing.h"
> #include "absl/memory/memory.h"
> #include "absl/strings/str_cat.h"
> 
>@@ -1788,4 +1789,5 @@ TEST(AllocatorSupportTest, SizeAllocConstructor) {
>     EXPECT_THAT(v, AllOf(SizeIs(len), Each(0)));
>   }
> }
>+
> }  // anonymous namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/container_memory.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/container_memory.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..56c5d2df67314e829fc9f591b36e65ab9df3e0ca
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/container_memory.h
>@@ -0,0 +1,405 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_CONTAINER_MEMORY_H_
>+#define ABSL_CONTAINER_INTERNAL_CONTAINER_MEMORY_H_
>+
>+#ifdef ADDRESS_SANITIZER
>+#include <sanitizer/asan_interface.h>
>+#endif
>+
>+#ifdef MEMORY_SANITIZER
>+#include <sanitizer/msan_interface.h>
>+#endif
>+
>+#include <cassert>
>+#include <cstddef>
>+#include <memory>
>+#include <tuple>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "absl/utility/utility.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+// Allocates at least n bytes aligned to the specified alignment.
>+// Alignment must be a power of 2. It must be positive.
>+//
>+// Note that many allocators don't honor alignment requirements above certain
>+// threshold (usually either alignof(std::max_align_t) or alignof(void*)).
>+// Allocate() doesn't apply alignment corrections. If the underlying allocator
>+// returns insufficiently alignment pointer, that's what you are going to get.
>+template <size_t Alignment, class Alloc>
>+void* Allocate(Alloc* alloc, size_t n) {
>+  static_assert(Alignment > 0, "");
>+  assert(n && "n must be positive");
>+  struct alignas(Alignment) M {};
>+  using A = typename absl::allocator_traits<Alloc>::template rebind_alloc<M>;
>+  using AT = typename absl::allocator_traits<Alloc>::template rebind_traits<M>;
>+  A mem_alloc(*alloc);
>+  void* p = AT::allocate(mem_alloc, (n + sizeof(M) - 1) / sizeof(M));
>+  assert(reinterpret_cast<uintptr_t>(p) % Alignment == 0 &&
>+         "allocator does not respect alignment");
>+  return p;
>+}
>+
>+// The pointer must have been previously obtained by calling
>+// Allocate<Alignment>(alloc, n).
>+template <size_t Alignment, class Alloc>
>+void Deallocate(Alloc* alloc, void* p, size_t n) {
>+  static_assert(Alignment > 0, "");
>+  assert(n && "n must be positive");
>+  struct alignas(Alignment) M {};
>+  using A = typename absl::allocator_traits<Alloc>::template rebind_alloc<M>;
>+  using AT = typename absl::allocator_traits<Alloc>::template rebind_traits<M>;
>+  A mem_alloc(*alloc);
>+  AT::deallocate(mem_alloc, static_cast<M*>(p),
>+                 (n + sizeof(M) - 1) / sizeof(M));
>+}
>+
>+namespace memory_internal {
>+
>+// Constructs T into uninitialized storage pointed by `ptr` using the args
>+// specified in the tuple.
>+template <class Alloc, class T, class Tuple, size_t... I>
>+void ConstructFromTupleImpl(Alloc* alloc, T* ptr, Tuple&& t,
>+                            absl::index_sequence<I...>) {
>+  absl::allocator_traits<Alloc>::construct(
>+      *alloc, ptr, std::get<I>(std::forward<Tuple>(t))...);
>+}
>+
>+template <class T, class F>
>+struct WithConstructedImplF {
>+  template <class... Args>
>+  decltype(std::declval<F>()(std::declval<T>())) operator()(
>+      Args&&... args) const {
>+    return std::forward<F>(f)(T(std::forward<Args>(args)...));
>+  }
>+  F&& f;
>+};
>+
>+template <class T, class Tuple, size_t... Is, class F>
>+decltype(std::declval<F>()(std::declval<T>())) WithConstructedImpl(
>+    Tuple&& t, absl::index_sequence<Is...>, F&& f) {
>+  return WithConstructedImplF<T, F>{std::forward<F>(f)}(
>+      std::get<Is>(std::forward<Tuple>(t))...);
>+}
>+
>+template <class T, size_t... Is>
>+auto TupleRefImpl(T&& t, absl::index_sequence<Is...>)
>+    -> decltype(std::forward_as_tuple(std::get<Is>(std::forward<T>(t))...)) {
>+  return std::forward_as_tuple(std::get<Is>(std::forward<T>(t))...);
>+}
>+
>+// Returns a tuple of references to the elements of the input tuple. T must be a
>+// tuple.
>+template <class T>
>+auto TupleRef(T&& t) -> decltype(
>+    TupleRefImpl(std::forward<T>(t),
>+                 absl::make_index_sequence<
>+                     std::tuple_size<typename std::decay<T>::type>::value>())) {
>+  return TupleRefImpl(
>+      std::forward<T>(t),
>+      absl::make_index_sequence<
>+          std::tuple_size<typename std::decay<T>::type>::value>());
>+}
>+
>+template <class F, class K, class V>
>+decltype(std::declval<F>()(std::declval<const K&>(), std::piecewise_construct,
>+                           std::declval<std::tuple<K>>(), std::declval<V>()))
>+DecomposePairImpl(F&& f, std::pair<std::tuple<K>, V> p) {
>+  const auto& key = std::get<0>(p.first);
>+  return std::forward<F>(f)(key, std::piecewise_construct, std::move(p.first),
>+                            std::move(p.second));
>+}
>+
>+}  // namespace memory_internal
>+
>+// Constructs T into uninitialized storage pointed by `ptr` using the args
>+// specified in the tuple.
>+template <class Alloc, class T, class Tuple>
>+void ConstructFromTuple(Alloc* alloc, T* ptr, Tuple&& t) {
>+  memory_internal::ConstructFromTupleImpl(
>+      alloc, ptr, std::forward<Tuple>(t),
>+      absl::make_index_sequence<
>+          std::tuple_size<typename std::decay<Tuple>::type>::value>());
>+}
>+
>+// Constructs T using the args specified in the tuple and calls F with the
>+// constructed value.
>+template <class T, class Tuple, class F>
>+decltype(std::declval<F>()(std::declval<T>())) WithConstructed(
>+    Tuple&& t, F&& f) {
>+  return memory_internal::WithConstructedImpl<T>(
>+      std::forward<Tuple>(t),
>+      absl::make_index_sequence<
>+          std::tuple_size<typename std::decay<Tuple>::type>::value>(),
>+      std::forward<F>(f));
>+}
>+
>+// Given arguments of an std::pair's consructor, PairArgs() returns a pair of
>+// tuples with references to the passed arguments. The tuples contain
>+// constructor arguments for the first and the second elements of the pair.
>+//
>+// The following two snippets are equivalent.
>+//
>+// 1. std::pair<F, S> p(args...);
>+//
>+// 2. auto a = PairArgs(args...);
>+//    std::pair<F, S> p(std::piecewise_construct,
>+//                      std::move(p.first), std::move(p.second));
>+inline std::pair<std::tuple<>, std::tuple<>> PairArgs() { return {}; }
>+template <class F, class S>
>+std::pair<std::tuple<F&&>, std::tuple<S&&>> PairArgs(F&& f, S&& s) {
>+  return {std::piecewise_construct, std::forward_as_tuple(std::forward<F>(f)),
>+          std::forward_as_tuple(std::forward<S>(s))};
>+}
>+template <class F, class S>
>+std::pair<std::tuple<const F&>, std::tuple<const S&>> PairArgs(
>+    const std::pair<F, S>& p) {
>+  return PairArgs(p.first, p.second);
>+}
>+template <class F, class S>
>+std::pair<std::tuple<F&&>, std::tuple<S&&>> PairArgs(std::pair<F, S>&& p) {
>+  return PairArgs(std::forward<F>(p.first), std::forward<S>(p.second));
>+}
>+template <class F, class S>
>+auto PairArgs(std::piecewise_construct_t, F&& f, S&& s)
>+    -> decltype(std::make_pair(memory_internal::TupleRef(std::forward<F>(f)),
>+                               memory_internal::TupleRef(std::forward<S>(s)))) {
>+  return std::make_pair(memory_internal::TupleRef(std::forward<F>(f)),
>+                        memory_internal::TupleRef(std::forward<S>(s)));
>+}
>+
>+// A helper function for implementing apply() in map policies.
>+template <class F, class... Args>
>+auto DecomposePair(F&& f, Args&&... args)
>+    -> decltype(memory_internal::DecomposePairImpl(
>+        std::forward<F>(f), PairArgs(std::forward<Args>(args)...))) {
>+  return memory_internal::DecomposePairImpl(
>+      std::forward<F>(f), PairArgs(std::forward<Args>(args)...));
>+}
>+
>+// A helper function for implementing apply() in set policies.
>+template <class F, class Arg>
>+decltype(std::declval<F>()(std::declval<const Arg&>(), std::declval<Arg>()))
>+DecomposeValue(F&& f, Arg&& arg) {
>+  const auto& key = arg;
>+  return std::forward<F>(f)(key, std::forward<Arg>(arg));
>+}
>+
>+// Helper functions for asan and msan.
>+inline void SanitizerPoisonMemoryRegion(const void* m, size_t s) {
>+#ifdef ADDRESS_SANITIZER
>+  ASAN_POISON_MEMORY_REGION(m, s);
>+#endif
>+#ifdef MEMORY_SANITIZER
>+  __msan_poison(m, s);
>+#endif
>+  (void)m;
>+  (void)s;
>+}
>+
>+inline void SanitizerUnpoisonMemoryRegion(const void* m, size_t s) {
>+#ifdef ADDRESS_SANITIZER
>+  ASAN_UNPOISON_MEMORY_REGION(m, s);
>+#endif
>+#ifdef MEMORY_SANITIZER
>+  __msan_unpoison(m, s);
>+#endif
>+  (void)m;
>+  (void)s;
>+}
>+
>+template <typename T>
>+inline void SanitizerPoisonObject(const T* object) {
>+  SanitizerPoisonMemoryRegion(object, sizeof(T));
>+}
>+
>+template <typename T>
>+inline void SanitizerUnpoisonObject(const T* object) {
>+  SanitizerUnpoisonMemoryRegion(object, sizeof(T));
>+}
>+
>+namespace memory_internal {
>+
>+// If Pair is a standard-layout type, OffsetOf<Pair>::kFirst and
>+// OffsetOf<Pair>::kSecond are equivalent to offsetof(Pair, first) and
>+// offsetof(Pair, second) respectively. Otherwise they are -1.
>+//
>+// The purpose of OffsetOf is to avoid calling offsetof() on non-standard-layout
>+// type, which is non-portable.
>+template <class Pair, class = std::true_type>
>+struct OffsetOf {
>+  static constexpr size_t kFirst = -1;
>+  static constexpr size_t kSecond = -1;
>+};
>+
>+template <class Pair>
>+struct OffsetOf<Pair, typename std::is_standard_layout<Pair>::type> {
>+  static constexpr size_t kFirst = offsetof(Pair, first);
>+  static constexpr size_t kSecond = offsetof(Pair, second);
>+};
>+
>+template <class K, class V>
>+struct IsLayoutCompatible {
>+ private:
>+  struct Pair {
>+    K first;
>+    V second;
>+  };
>+
>+  // Is P layout-compatible with Pair?
>+  template <class P>
>+  static constexpr bool LayoutCompatible() {
>+    return std::is_standard_layout<P>() && sizeof(P) == sizeof(Pair) &&
>+           alignof(P) == alignof(Pair) &&
>+           memory_internal::OffsetOf<P>::kFirst ==
>+               memory_internal::OffsetOf<Pair>::kFirst &&
>+           memory_internal::OffsetOf<P>::kSecond ==
>+               memory_internal::OffsetOf<Pair>::kSecond;
>+  }
>+
>+ public:
>+  // Whether pair<const K, V> and pair<K, V> are layout-compatible. If they are,
>+  // then it is safe to store them in a union and read from either.
>+  static constexpr bool value = std::is_standard_layout<K>() &&
>+                                std::is_standard_layout<Pair>() &&
>+                                memory_internal::OffsetOf<Pair>::kFirst == 0 &&
>+                                LayoutCompatible<std::pair<K, V>>() &&
>+                                LayoutCompatible<std::pair<const K, V>>();
>+};
>+
>+}  // namespace memory_internal
>+
>+// If kMutableKeys is false, only the value member is accessed.
>+//
>+// If kMutableKeys is true, key is accessed through all slots while value and
>+// mutable_value are accessed only via INITIALIZED slots. Slots are created and
>+// destroyed via mutable_value so that the key can be moved later.
>+template <class K, class V>
>+union slot_type {
>+ private:
>+  static void emplace(slot_type* slot) {
>+    // The construction of union doesn't do anything at runtime but it allows us
>+    // to access its members without violating aliasing rules.
>+    new (slot) slot_type;
>+  }
>+  // If pair<const K, V> and pair<K, V> are layout-compatible, we can accept one
>+  // or the other via slot_type. We are also free to access the key via
>+  // slot_type::key in this case.
>+  using kMutableKeys =
>+      std::integral_constant<bool,
>+                             memory_internal::IsLayoutCompatible<K, V>::value>;
>+
>+ public:
>+  slot_type() {}
>+  ~slot_type() = delete;
>+  using value_type = std::pair<const K, V>;
>+  using mutable_value_type = std::pair<K, V>;
>+
>+  value_type value;
>+  mutable_value_type mutable_value;
>+  K key;
>+
>+  template <class Allocator, class... Args>
>+  static void construct(Allocator* alloc, slot_type* slot, Args&&... args) {
>+    emplace(slot);
>+    if (kMutableKeys::value) {
>+      absl::allocator_traits<Allocator>::construct(*alloc, &slot->mutable_value,
>+                                                   std::forward<Args>(args)...);
>+    } else {
>+      absl::allocator_traits<Allocator>::construct(*alloc, &slot->value,
>+                                                   std::forward<Args>(args)...);
>+    }
>+  }
>+
>+  // Construct this slot by moving from another slot.
>+  template <class Allocator>
>+  static void construct(Allocator* alloc, slot_type* slot, slot_type* other) {
>+    emplace(slot);
>+    if (kMutableKeys::value) {
>+      absl::allocator_traits<Allocator>::construct(
>+          *alloc, &slot->mutable_value, std::move(other->mutable_value));
>+    } else {
>+      absl::allocator_traits<Allocator>::construct(*alloc, &slot->value,
>+                                                   std::move(other->value));
>+    }
>+  }
>+
>+  template <class Allocator>
>+  static void destroy(Allocator* alloc, slot_type* slot) {
>+    if (kMutableKeys::value) {
>+      absl::allocator_traits<Allocator>::destroy(*alloc, &slot->mutable_value);
>+    } else {
>+      absl::allocator_traits<Allocator>::destroy(*alloc, &slot->value);
>+    }
>+  }
>+
>+  template <class Allocator>
>+  static void transfer(Allocator* alloc, slot_type* new_slot,
>+                       slot_type* old_slot) {
>+    emplace(new_slot);
>+    if (kMutableKeys::value) {
>+      absl::allocator_traits<Allocator>::construct(
>+          *alloc, &new_slot->mutable_value, std::move(old_slot->mutable_value));
>+    } else {
>+      absl::allocator_traits<Allocator>::construct(*alloc, &new_slot->value,
>+                                                   std::move(old_slot->value));
>+    }
>+    destroy(alloc, old_slot);
>+  }
>+
>+  template <class Allocator>
>+  static void swap(Allocator* alloc, slot_type* a, slot_type* b) {
>+    if (kMutableKeys::value) {
>+      using std::swap;
>+      swap(a->mutable_value, b->mutable_value);
>+    } else {
>+      value_type tmp = std::move(a->value);
>+      absl::allocator_traits<Allocator>::destroy(*alloc, &a->value);
>+      absl::allocator_traits<Allocator>::construct(*alloc, &a->value,
>+                                                   std::move(b->value));
>+      absl::allocator_traits<Allocator>::destroy(*alloc, &b->value);
>+      absl::allocator_traits<Allocator>::construct(*alloc, &b->value,
>+                                                   std::move(tmp));
>+    }
>+  }
>+
>+  template <class Allocator>
>+  static void move(Allocator* alloc, slot_type* src, slot_type* dest) {
>+    if (kMutableKeys::value) {
>+      dest->mutable_value = std::move(src->mutable_value);
>+    } else {
>+      absl::allocator_traits<Allocator>::destroy(*alloc, &dest->value);
>+      absl::allocator_traits<Allocator>::construct(*alloc, &dest->value,
>+                                                   std::move(src->value));
>+    }
>+  }
>+
>+  template <class Allocator>
>+  static void move(Allocator* alloc, slot_type* first, slot_type* last,
>+                   slot_type* result) {
>+    for (slot_type *src = first, *dest = result; src != last; ++src, ++dest)
>+      move(alloc, src, dest);
>+  }
>+};
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_CONTAINER_MEMORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/container_memory_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/container_memory_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..f1c4058298c157409fec9ac990850c0b1494971f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/container_memory_test.cc
>@@ -0,0 +1,188 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/container_memory.h"
>+
>+#include <cstdint>
>+#include <tuple>
>+#include <utility>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+using ::testing::Pair;
>+
>+TEST(Memory, AlignmentLargerThanBase) {
>+  std::allocator<int8_t> alloc;
>+  void* mem = Allocate<2>(&alloc, 3);
>+  EXPECT_EQ(0, reinterpret_cast<uintptr_t>(mem) % 2);
>+  memcpy(mem, "abc", 3);
>+  Deallocate<2>(&alloc, mem, 3);
>+}
>+
>+TEST(Memory, AlignmentSmallerThanBase) {
>+  std::allocator<int64_t> alloc;
>+  void* mem = Allocate<2>(&alloc, 3);
>+  EXPECT_EQ(0, reinterpret_cast<uintptr_t>(mem) % 2);
>+  memcpy(mem, "abc", 3);
>+  Deallocate<2>(&alloc, mem, 3);
>+}
>+
>+class Fixture : public ::testing::Test {
>+  using Alloc = std::allocator<std::string>;
>+
>+ public:
>+  Fixture() { ptr_ = std::allocator_traits<Alloc>::allocate(*alloc(), 1); }
>+  ~Fixture() override {
>+    std::allocator_traits<Alloc>::destroy(*alloc(), ptr_);
>+    std::allocator_traits<Alloc>::deallocate(*alloc(), ptr_, 1);
>+  }
>+  std::string* ptr() { return ptr_; }
>+  Alloc* alloc() { return &alloc_; }
>+
>+ private:
>+  Alloc alloc_;
>+  std::string* ptr_;
>+};
>+
>+TEST_F(Fixture, ConstructNoArgs) {
>+  ConstructFromTuple(alloc(), ptr(), std::forward_as_tuple());
>+  EXPECT_EQ(*ptr(), "");
>+}
>+
>+TEST_F(Fixture, ConstructOneArg) {
>+  ConstructFromTuple(alloc(), ptr(), std::forward_as_tuple("abcde"));
>+  EXPECT_EQ(*ptr(), "abcde");
>+}
>+
>+TEST_F(Fixture, ConstructTwoArg) {
>+  ConstructFromTuple(alloc(), ptr(), std::forward_as_tuple(5, 'a'));
>+  EXPECT_EQ(*ptr(), "aaaaa");
>+}
>+
>+TEST(PairArgs, NoArgs) {
>+  EXPECT_THAT(PairArgs(),
>+              Pair(std::forward_as_tuple(), std::forward_as_tuple()));
>+}
>+
>+TEST(PairArgs, TwoArgs) {
>+  EXPECT_EQ(
>+      std::make_pair(std::forward_as_tuple(1), std::forward_as_tuple('A')),
>+      PairArgs(1, 'A'));
>+}
>+
>+TEST(PairArgs, Pair) {
>+  EXPECT_EQ(
>+      std::make_pair(std::forward_as_tuple(1), std::forward_as_tuple('A')),
>+      PairArgs(std::make_pair(1, 'A')));
>+}
>+
>+TEST(PairArgs, Piecewise) {
>+  EXPECT_EQ(
>+      std::make_pair(std::forward_as_tuple(1), std::forward_as_tuple('A')),
>+      PairArgs(std::piecewise_construct, std::forward_as_tuple(1),
>+               std::forward_as_tuple('A')));
>+}
>+
>+TEST(WithConstructed, Simple) {
>+  EXPECT_EQ(1, WithConstructed<absl::string_view>(
>+                   std::make_tuple(std::string("a")),
>+                   [](absl::string_view str) { return str.size(); }));
>+}
>+
>+template <class F, class Arg>
>+decltype(DecomposeValue(std::declval<F>(), std::declval<Arg>()))
>+DecomposeValueImpl(int, F&& f, Arg&& arg) {
>+  return DecomposeValue(std::forward<F>(f), std::forward<Arg>(arg));
>+}
>+
>+template <class F, class Arg>
>+const char* DecomposeValueImpl(char, F&& f, Arg&& arg) {
>+  return "not decomposable";
>+}
>+
>+template <class F, class Arg>
>+decltype(DecomposeValueImpl(0, std::declval<F>(), std::declval<Arg>()))
>+TryDecomposeValue(F&& f, Arg&& arg) {
>+  return DecomposeValueImpl(0, std::forward<F>(f), std::forward<Arg>(arg));
>+}
>+
>+TEST(DecomposeValue, Decomposable) {
>+  auto f = [](const int& x, int&& y) {
>+    EXPECT_EQ(&x, &y);
>+    EXPECT_EQ(42, x);
>+    return 'A';
>+  };
>+  EXPECT_EQ('A', TryDecomposeValue(f, 42));
>+}
>+
>+TEST(DecomposeValue, NotDecomposable) {
>+  auto f = [](void*) {
>+    ADD_FAILURE() << "Must not be called";
>+    return 'A';
>+  };
>+  EXPECT_STREQ("not decomposable", TryDecomposeValue(f, 42));
>+}
>+
>+template <class F, class... Args>
>+decltype(DecomposePair(std::declval<F>(), std::declval<Args>()...))
>+DecomposePairImpl(int, F&& f, Args&&... args) {
>+  return DecomposePair(std::forward<F>(f), std::forward<Args>(args)...);
>+}
>+
>+template <class F, class... Args>
>+const char* DecomposePairImpl(char, F&& f, Args&&... args) {
>+  return "not decomposable";
>+}
>+
>+template <class F, class... Args>
>+decltype(DecomposePairImpl(0, std::declval<F>(), std::declval<Args>()...))
>+TryDecomposePair(F&& f, Args&&... args) {
>+  return DecomposePairImpl(0, std::forward<F>(f), std::forward<Args>(args)...);
>+}
>+
>+TEST(DecomposePair, Decomposable) {
>+  auto f = [](const int& x, std::piecewise_construct_t, std::tuple<int&&> k,
>+              std::tuple<double>&& v) {
>+    EXPECT_EQ(&x, &std::get<0>(k));
>+    EXPECT_EQ(42, x);
>+    EXPECT_EQ(0.5, std::get<0>(v));
>+    return 'A';
>+  };
>+  EXPECT_EQ('A', TryDecomposePair(f, 42, 0.5));
>+  EXPECT_EQ('A', TryDecomposePair(f, std::make_pair(42, 0.5)));
>+  EXPECT_EQ('A', TryDecomposePair(f, std::piecewise_construct,
>+                                  std::make_tuple(42), std::make_tuple(0.5)));
>+}
>+
>+TEST(DecomposePair, NotDecomposable) {
>+  auto f = [](...) {
>+    ADD_FAILURE() << "Must not be called";
>+    return 'A';
>+  };
>+  EXPECT_STREQ("not decomposable",
>+               TryDecomposePair(f));
>+  EXPECT_STREQ("not decomposable",
>+               TryDecomposePair(f, std::piecewise_construct, std::make_tuple(),
>+                                std::make_tuple(0.5)));
>+}
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_function_defaults.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_function_defaults.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..1f0d794d966388186dd7dfd69a8ae9a6adf709f6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_function_defaults.h
>@@ -0,0 +1,143 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Define the default Hash and Eq functions for SwissTable containers.
>+//
>+// std::hash<T> and std::equal_to<T> are not appropriate hash and equal
>+// functions for SwissTable containers. There are two reasons for this.
>+//
>+// SwissTable containers are power of 2 sized containers:
>+//
>+// This means they use the lower bits of the hash value to find the slot for
>+// each entry. The typical hash function for integral types is the identity.
>+// This is a very weak hash function for SwissTable and any power of 2 sized
>+// hashtable implementation which will lead to excessive collisions. For
>+// SwissTable we use murmur3 style mixing to reduce collisions to a minimum.
>+//
>+// SwissTable containers support heterogeneous lookup:
>+//
>+// In order to make heterogeneous lookup work, hash and equal functions must be
>+// polymorphic. At the same time they have to satisfy the same requirements the
>+// C++ standard imposes on hash functions and equality operators. That is:
>+//
>+//   if hash_default_eq<T>(a, b) returns true for any a and b of type T, then
>+//   hash_default_hash<T>(a) must equal hash_default_hash<T>(b)
>+//
>+// For SwissTable containers this requirement is relaxed to allow a and b of
>+// any and possibly different types. Note that like the standard the hash and
>+// equal functions are still bound to T. This is important because some type U
>+// can be hashed by/tested for equality differently depending on T. A notable
>+// example is `const char*`. `const char*` is treated as a c-style string when
>+// the hash function is hash<string> but as a pointer when the hash function is
>+// hash<void*>.
>+//
>+#ifndef ABSL_CONTAINER_INTERNAL_HASH_FUNCTION_DEFAULTS_H_
>+#define ABSL_CONTAINER_INTERNAL_HASH_FUNCTION_DEFAULTS_H_
>+
>+#include <stdint.h>
>+#include <cstddef>
>+#include <memory>
>+#include <string>
>+#include <type_traits>
>+
>+#include "absl/base/config.h"
>+#include "absl/hash/hash.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+// The hash of an object of type T is computed by using absl::Hash.
>+template <class T, class E = void>
>+struct HashEq {
>+  using Hash = absl::Hash<T>;
>+  using Eq = std::equal_to<T>;
>+};
>+
>+struct StringHash {
>+  using is_transparent = void;
>+
>+  size_t operator()(absl::string_view v) const {
>+    return absl::Hash<absl::string_view>{}(v);
>+  }
>+};
>+
>+// Supports heterogeneous lookup for string-like elements.
>+struct StringHashEq {
>+  using Hash = StringHash;
>+  struct Eq {
>+    using is_transparent = void;
>+    bool operator()(absl::string_view lhs, absl::string_view rhs) const {
>+      return lhs == rhs;
>+    }
>+  };
>+};
>+template <>
>+struct HashEq<std::string> : StringHashEq {};
>+template <>
>+struct HashEq<absl::string_view> : StringHashEq {};
>+
>+// Supports heterogeneous lookup for pointers and smart pointers.
>+template <class T>
>+struct HashEq<T*> {
>+  struct Hash {
>+    using is_transparent = void;
>+    template <class U>
>+    size_t operator()(const U& ptr) const {
>+      return absl::Hash<const T*>{}(HashEq::ToPtr(ptr));
>+    }
>+  };
>+  struct Eq {
>+    using is_transparent = void;
>+    template <class A, class B>
>+    bool operator()(const A& a, const B& b) const {
>+      return HashEq::ToPtr(a) == HashEq::ToPtr(b);
>+    }
>+  };
>+
>+ private:
>+  static const T* ToPtr(const T* ptr) { return ptr; }
>+  template <class U, class D>
>+  static const T* ToPtr(const std::unique_ptr<U, D>& ptr) {
>+    return ptr.get();
>+  }
>+  template <class U>
>+  static const T* ToPtr(const std::shared_ptr<U>& ptr) {
>+    return ptr.get();
>+  }
>+};
>+
>+template <class T, class D>
>+struct HashEq<std::unique_ptr<T, D>> : HashEq<T*> {};
>+template <class T>
>+struct HashEq<std::shared_ptr<T>> : HashEq<T*> {};
>+
>+// This header's visibility is restricted.  If you need to access the default
>+// hasher please use the container's ::hasher alias instead.
>+//
>+// Example: typename Hash = typename absl::flat_hash_map<K, V>::hasher
>+template <class T>
>+using hash_default_hash = typename container_internal::HashEq<T>::Hash;
>+
>+// This header's visibility is restricted.  If you need to access the default
>+// key equal please use the container's ::key_equal alias instead.
>+//
>+// Example: typename Eq = typename absl::flat_hash_map<K, V, Hash>::key_equal
>+template <class T>
>+using hash_default_eq = typename container_internal::HashEq<T>::Eq;
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_HASH_FUNCTION_DEFAULTS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_function_defaults_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_function_defaults_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..464baae02cddb0c41e612e59788d837d59f917fc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_function_defaults_test.cc
>@@ -0,0 +1,299 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/hash_function_defaults.h"
>+
>+#include <functional>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "gtest/gtest.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+using ::testing::Types;
>+
>+TEST(Eq, Int32) {
>+  hash_default_eq<int32_t> eq;
>+  EXPECT_TRUE(eq(1, 1u));
>+  EXPECT_TRUE(eq(1, char{1}));
>+  EXPECT_TRUE(eq(1, true));
>+  EXPECT_TRUE(eq(1, double{1.1}));
>+  EXPECT_FALSE(eq(1, char{2}));
>+  EXPECT_FALSE(eq(1, 2u));
>+  EXPECT_FALSE(eq(1, false));
>+  EXPECT_FALSE(eq(1, 2.));
>+}
>+
>+TEST(Hash, Int32) {
>+  hash_default_hash<int32_t> hash;
>+  auto h = hash(1);
>+  EXPECT_EQ(h, hash(1u));
>+  EXPECT_EQ(h, hash(char{1}));
>+  EXPECT_EQ(h, hash(true));
>+  EXPECT_EQ(h, hash(double{1.1}));
>+  EXPECT_NE(h, hash(2u));
>+  EXPECT_NE(h, hash(char{2}));
>+  EXPECT_NE(h, hash(false));
>+  EXPECT_NE(h, hash(2.));
>+}
>+
>+enum class MyEnum { A, B, C, D };
>+
>+TEST(Eq, Enum) {
>+  hash_default_eq<MyEnum> eq;
>+  EXPECT_TRUE(eq(MyEnum::A, MyEnum::A));
>+  EXPECT_FALSE(eq(MyEnum::A, MyEnum::B));
>+}
>+
>+TEST(Hash, Enum) {
>+  hash_default_hash<MyEnum> hash;
>+
>+  for (MyEnum e : {MyEnum::A, MyEnum::B, MyEnum::C}) {
>+    auto h = hash(e);
>+    EXPECT_EQ(h, hash_default_hash<int>{}(static_cast<int>(e)));
>+    EXPECT_NE(h, hash(MyEnum::D));
>+  }
>+}
>+
>+using StringTypes = ::testing::Types<std::string, absl::string_view>;
>+
>+template <class T>
>+struct EqString : ::testing::Test {
>+  hash_default_eq<T> key_eq;
>+};
>+
>+TYPED_TEST_CASE(EqString, StringTypes);
>+
>+template <class T>
>+struct HashString : ::testing::Test {
>+  hash_default_hash<T> hasher;
>+};
>+
>+TYPED_TEST_CASE(HashString, StringTypes);
>+
>+TYPED_TEST(EqString, Works) {
>+  auto eq = this->key_eq;
>+  EXPECT_TRUE(eq("a", "a"));
>+  EXPECT_TRUE(eq("a", absl::string_view("a")));
>+  EXPECT_TRUE(eq("a", std::string("a")));
>+  EXPECT_FALSE(eq("a", "b"));
>+  EXPECT_FALSE(eq("a", absl::string_view("b")));
>+  EXPECT_FALSE(eq("a", std::string("b")));
>+}
>+
>+TYPED_TEST(HashString, Works) {
>+  auto hash = this->hasher;
>+  auto h = hash("a");
>+  EXPECT_EQ(h, hash(absl::string_view("a")));
>+  EXPECT_EQ(h, hash(std::string("a")));
>+  EXPECT_NE(h, hash(absl::string_view("b")));
>+  EXPECT_NE(h, hash(std::string("b")));
>+}
>+
>+struct NoDeleter {
>+  template <class T>
>+  void operator()(const T* ptr) const {}
>+};
>+
>+using PointerTypes =
>+    ::testing::Types<const int*, int*, std::unique_ptr<const int>,
>+                     std::unique_ptr<const int, NoDeleter>,
>+                     std::unique_ptr<int>, std::unique_ptr<int, NoDeleter>,
>+                     std::shared_ptr<const int>, std::shared_ptr<int>>;
>+
>+template <class T>
>+struct EqPointer : ::testing::Test {
>+  hash_default_eq<T> key_eq;
>+};
>+
>+TYPED_TEST_CASE(EqPointer, PointerTypes);
>+
>+template <class T>
>+struct HashPointer : ::testing::Test {
>+  hash_default_hash<T> hasher;
>+};
>+
>+TYPED_TEST_CASE(HashPointer, PointerTypes);
>+
>+TYPED_TEST(EqPointer, Works) {
>+  int dummy;
>+  auto eq = this->key_eq;
>+  auto sptr = std::make_shared<int>();
>+  std::shared_ptr<const int> csptr = sptr;
>+  int* ptr = sptr.get();
>+  const int* cptr = ptr;
>+  std::unique_ptr<int, NoDeleter> uptr(ptr);
>+  std::unique_ptr<const int, NoDeleter> cuptr(ptr);
>+
>+  EXPECT_TRUE(eq(ptr, cptr));
>+  EXPECT_TRUE(eq(ptr, sptr));
>+  EXPECT_TRUE(eq(ptr, uptr));
>+  EXPECT_TRUE(eq(ptr, csptr));
>+  EXPECT_TRUE(eq(ptr, cuptr));
>+  EXPECT_FALSE(eq(&dummy, cptr));
>+  EXPECT_FALSE(eq(&dummy, sptr));
>+  EXPECT_FALSE(eq(&dummy, uptr));
>+  EXPECT_FALSE(eq(&dummy, csptr));
>+  EXPECT_FALSE(eq(&dummy, cuptr));
>+}
>+
>+TEST(Hash, DerivedAndBase) {
>+  struct Base {};
>+  struct Derived : Base {};
>+
>+  hash_default_hash<Base*> hasher;
>+
>+  Base base;
>+  Derived derived;
>+  EXPECT_NE(hasher(&base), hasher(&derived));
>+  EXPECT_EQ(hasher(static_cast<Base*>(&derived)), hasher(&derived));
>+
>+  auto dp = std::make_shared<Derived>();
>+  EXPECT_EQ(hasher(static_cast<Base*>(dp.get())), hasher(dp));
>+}
>+
>+TEST(Hash, FunctionPointer) {
>+  using Func = int (*)();
>+  hash_default_hash<Func> hasher;
>+  hash_default_eq<Func> eq;
>+
>+  Func p1 = [] { return 1; }, p2 = [] { return 2; };
>+  EXPECT_EQ(hasher(p1), hasher(p1));
>+  EXPECT_TRUE(eq(p1, p1));
>+
>+  EXPECT_NE(hasher(p1), hasher(p2));
>+  EXPECT_FALSE(eq(p1, p2));
>+}
>+
>+TYPED_TEST(HashPointer, Works) {
>+  int dummy;
>+  auto hash = this->hasher;
>+  auto sptr = std::make_shared<int>();
>+  std::shared_ptr<const int> csptr = sptr;
>+  int* ptr = sptr.get();
>+  const int* cptr = ptr;
>+  std::unique_ptr<int, NoDeleter> uptr(ptr);
>+  std::unique_ptr<const int, NoDeleter> cuptr(ptr);
>+
>+  EXPECT_EQ(hash(ptr), hash(cptr));
>+  EXPECT_EQ(hash(ptr), hash(sptr));
>+  EXPECT_EQ(hash(ptr), hash(uptr));
>+  EXPECT_EQ(hash(ptr), hash(csptr));
>+  EXPECT_EQ(hash(ptr), hash(cuptr));
>+  EXPECT_NE(hash(&dummy), hash(cptr));
>+  EXPECT_NE(hash(&dummy), hash(sptr));
>+  EXPECT_NE(hash(&dummy), hash(uptr));
>+  EXPECT_NE(hash(&dummy), hash(csptr));
>+  EXPECT_NE(hash(&dummy), hash(cuptr));
>+}
>+
>+// Cartesian product of (string, std::string, absl::string_view)
>+// with (string, std::string, absl::string_view, const char*).
>+using StringTypesCartesianProduct = Types<
>+    // clang-format off
>+
>+    std::pair<std::string, std::string>,
>+    std::pair<std::string, absl::string_view>,
>+    std::pair<std::string, const char*>,
>+
>+    std::pair<absl::string_view, std::string>,
>+    std::pair<absl::string_view, absl::string_view>,
>+    std::pair<absl::string_view, const char*>>;
>+// clang-format on
>+
>+constexpr char kFirstString[] = "abc123";
>+constexpr char kSecondString[] = "ijk456";
>+
>+template <typename T>
>+struct StringLikeTest : public ::testing::Test {
>+  typename T::first_type a1{kFirstString};
>+  typename T::second_type b1{kFirstString};
>+  typename T::first_type a2{kSecondString};
>+  typename T::second_type b2{kSecondString};
>+  hash_default_eq<typename T::first_type> eq;
>+  hash_default_hash<typename T::first_type> hash;
>+};
>+
>+TYPED_TEST_CASE_P(StringLikeTest);
>+
>+TYPED_TEST_P(StringLikeTest, Eq) {
>+  EXPECT_TRUE(this->eq(this->a1, this->b1));
>+  EXPECT_TRUE(this->eq(this->b1, this->a1));
>+}
>+
>+TYPED_TEST_P(StringLikeTest, NotEq) {
>+  EXPECT_FALSE(this->eq(this->a1, this->b2));
>+  EXPECT_FALSE(this->eq(this->b2, this->a1));
>+}
>+
>+TYPED_TEST_P(StringLikeTest, HashEq) {
>+  EXPECT_EQ(this->hash(this->a1), this->hash(this->b1));
>+  EXPECT_EQ(this->hash(this->a2), this->hash(this->b2));
>+  // It would be a poor hash function which collides on these strings.
>+  EXPECT_NE(this->hash(this->a1), this->hash(this->b2));
>+}
>+
>+TYPED_TEST_CASE(StringLikeTest, StringTypesCartesianProduct);
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+enum Hash : size_t {
>+  kStd = 0x2,       // std::hash
>+#ifdef _MSC_VER
>+  kExtension = kStd,  // In MSVC, std::hash == ::hash
>+#else                 // _MSC_VER
>+  kExtension = 0x4,  // ::hash (GCC extension)
>+#endif                // _MSC_VER
>+};
>+
>+// H is a bitmask of Hash enumerations.
>+// Hashable<H> is hashable via all means specified in H.
>+template <int H>
>+struct Hashable {
>+  static constexpr bool HashableBy(Hash h) { return h & H; }
>+};
>+
>+namespace std {
>+template <int H>
>+struct hash<Hashable<H>> {
>+  template <class E = Hashable<H>,
>+            class = typename std::enable_if<E::HashableBy(kStd)>::type>
>+  size_t operator()(E) const {
>+    return kStd;
>+  }
>+};
>+}  // namespace std
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+template <class T>
>+size_t Hash(const T& v) {
>+  return hash_default_hash<T>()(v);
>+}
>+
>+TEST(Delegate, HashDispatch) {
>+  EXPECT_EQ(Hash(kStd), Hash(Hashable<kStd>()));
>+}
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_generator_testing.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_generator_testing.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..0d6a9df16f8a336c143afaeec2069142361b889c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_generator_testing.cc
>@@ -0,0 +1,72 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/hash_generator_testing.h"
>+
>+#include <deque>
>+
>+namespace absl {
>+namespace container_internal {
>+namespace hash_internal {
>+namespace {
>+
>+class RandomDeviceSeedSeq {
>+ public:
>+  using result_type = typename std::random_device::result_type;
>+
>+  template <class Iterator>
>+  void generate(Iterator start, Iterator end) {
>+    while (start != end) {
>+      *start = gen_();
>+      ++start;
>+    }
>+  }
>+
>+ private:
>+  std::random_device gen_;
>+};
>+
>+}  // namespace
>+
>+std::mt19937_64* GetThreadLocalRng() {
>+  RandomDeviceSeedSeq seed_seq;
>+  thread_local auto* rng = new std::mt19937_64(seed_seq);
>+  return rng;
>+}
>+
>+std::string Generator<std::string>::operator()() const {
>+  // NOLINTNEXTLINE(runtime/int)
>+  std::uniform_int_distribution<short> chars(0x20, 0x7E);
>+  std::string res;
>+  res.resize(32);
>+  std::generate(res.begin(), res.end(),
>+                [&]() { return chars(*GetThreadLocalRng()); });
>+  return res;
>+}
>+
>+absl::string_view Generator<absl::string_view>::operator()() const {
>+  static auto* arena = new std::deque<std::string>();
>+  // NOLINTNEXTLINE(runtime/int)
>+  std::uniform_int_distribution<short> chars(0x20, 0x7E);
>+  arena->emplace_back();
>+  auto& res = arena->back();
>+  res.resize(32);
>+  std::generate(res.begin(), res.end(),
>+                [&]() { return chars(*GetThreadLocalRng()); });
>+  return res;
>+}
>+
>+}  // namespace hash_internal
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_generator_testing.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_generator_testing.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..50d771026c7b5005135d2863165a009cb4dbca98
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_generator_testing.h
>@@ -0,0 +1,150 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Generates random values for testing. Specialized only for the few types we
>+// care about.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_HASH_GENERATOR_TESTING_H_
>+#define ABSL_CONTAINER_INTERNAL_HASH_GENERATOR_TESTING_H_
>+
>+#include <stdint.h>
>+#include <algorithm>
>+#include <iosfwd>
>+#include <random>
>+#include <tuple>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/container/internal/hash_policy_testing.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace hash_internal {
>+namespace generator_internal {
>+
>+template <class Container, class = void>
>+struct IsMap : std::false_type {};
>+
>+template <class Map>
>+struct IsMap<Map, absl::void_t<typename Map::mapped_type>> : std::true_type {};
>+
>+}  // namespace generator_internal
>+
>+std::mt19937_64* GetThreadLocalRng();
>+
>+enum Enum {
>+  kEnumEmpty,
>+  kEnumDeleted,
>+};
>+
>+enum class EnumClass : uint64_t {
>+  kEmpty,
>+  kDeleted,
>+};
>+
>+inline std::ostream& operator<<(std::ostream& o, const EnumClass& ec) {
>+  return o << static_cast<uint64_t>(ec);
>+}
>+
>+template <class T, class E = void>
>+struct Generator;
>+
>+template <class T>
>+struct Generator<T, typename std::enable_if<std::is_integral<T>::value>::type> {
>+  T operator()() const {
>+    std::uniform_int_distribution<T> dist;
>+    return dist(*GetThreadLocalRng());
>+  }
>+};
>+
>+template <>
>+struct Generator<Enum> {
>+  Enum operator()() const {
>+    std::uniform_int_distribution<typename std::underlying_type<Enum>::type>
>+        dist;
>+    while (true) {
>+      auto variate = dist(*GetThreadLocalRng());
>+      if (variate != kEnumEmpty && variate != kEnumDeleted)
>+        return static_cast<Enum>(variate);
>+    }
>+  }
>+};
>+
>+template <>
>+struct Generator<EnumClass> {
>+  EnumClass operator()() const {
>+    std::uniform_int_distribution<
>+        typename std::underlying_type<EnumClass>::type>
>+        dist;
>+    while (true) {
>+      EnumClass variate = static_cast<EnumClass>(dist(*GetThreadLocalRng()));
>+      if (variate != EnumClass::kEmpty && variate != EnumClass::kDeleted)
>+        return static_cast<EnumClass>(variate);
>+    }
>+  }
>+};
>+
>+template <>
>+struct Generator<std::string> {
>+  std::string operator()() const;
>+};
>+
>+template <>
>+struct Generator<absl::string_view> {
>+  absl::string_view operator()() const;
>+};
>+
>+template <>
>+struct Generator<NonStandardLayout> {
>+  NonStandardLayout operator()() const {
>+    return NonStandardLayout(Generator<std::string>()());
>+  }
>+};
>+
>+template <class K, class V>
>+struct Generator<std::pair<K, V>> {
>+  std::pair<K, V> operator()() const {
>+    return std::pair<K, V>(Generator<typename std::decay<K>::type>()(),
>+                           Generator<typename std::decay<V>::type>()());
>+  }
>+};
>+
>+template <class... Ts>
>+struct Generator<std::tuple<Ts...>> {
>+  std::tuple<Ts...> operator()() const {
>+    return std::tuple<Ts...>(Generator<typename std::decay<Ts>::type>()()...);
>+  }
>+};
>+
>+template <class U>
>+struct Generator<U, absl::void_t<decltype(std::declval<U&>().key()),
>+                                decltype(std::declval<U&>().value())>>
>+    : Generator<std::pair<
>+          typename std::decay<decltype(std::declval<U&>().key())>::type,
>+          typename std::decay<decltype(std::declval<U&>().value())>::type>> {};
>+
>+template <class Container>
>+using GeneratedType = decltype(
>+    std::declval<const Generator<
>+        typename std::conditional<generator_internal::IsMap<Container>::value,
>+                                  typename Container::value_type,
>+                                  typename Container::key_type>::type>&>()());
>+
>+}  // namespace hash_internal
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_HASH_GENERATOR_TESTING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_testing.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_testing.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..ffc76ead7a6842e9fcbcf32042f12f7eac6e9f98
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_testing.h
>@@ -0,0 +1,178 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Utilities to help tests verify that hash tables properly handle stateful
>+// allocators and hash functions.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_HASH_POLICY_TESTING_H_
>+#define ABSL_CONTAINER_INTERNAL_HASH_POLICY_TESTING_H_
>+
>+#include <cstdlib>
>+#include <limits>
>+#include <memory>
>+#include <ostream>
>+#include <type_traits>
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/hash/hash.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace hash_testing_internal {
>+
>+template <class Derived>
>+struct WithId {
>+  WithId() : id_(next_id<Derived>()) {}
>+  WithId(const WithId& that) : id_(that.id_) {}
>+  WithId(WithId&& that) : id_(that.id_) { that.id_ = 0; }
>+  WithId& operator=(const WithId& that) {
>+    id_ = that.id_;
>+    return *this;
>+  }
>+  WithId& operator=(WithId&& that) {
>+    id_ = that.id_;
>+    that.id_ = 0;
>+    return *this;
>+  }
>+
>+  size_t id() const { return id_; }
>+
>+  friend bool operator==(const WithId& a, const WithId& b) {
>+    return a.id_ == b.id_;
>+  }
>+  friend bool operator!=(const WithId& a, const WithId& b) { return !(a == b); }
>+
>+ protected:
>+  explicit WithId(size_t id) : id_(id) {}
>+
>+ private:
>+  size_t id_;
>+
>+  template <class T>
>+  static size_t next_id() {
>+    // 0 is reserved for moved from state.
>+    static size_t gId = 1;
>+    return gId++;
>+  }
>+};
>+
>+}  // namespace hash_testing_internal
>+
>+struct NonStandardLayout {
>+  NonStandardLayout() {}
>+  explicit NonStandardLayout(std::string s) : value(std::move(s)) {}
>+  virtual ~NonStandardLayout() {}
>+
>+  friend bool operator==(const NonStandardLayout& a,
>+                         const NonStandardLayout& b) {
>+    return a.value == b.value;
>+  }
>+  friend bool operator!=(const NonStandardLayout& a,
>+                         const NonStandardLayout& b) {
>+    return a.value != b.value;
>+  }
>+
>+  template <typename H>
>+  friend H AbslHashValue(H h, const NonStandardLayout& v) {
>+    return H::combine(std::move(h), v.value);
>+  }
>+
>+  std::string value;
>+};
>+
>+struct StatefulTestingHash
>+    : absl::container_internal::hash_testing_internal::WithId<
>+          StatefulTestingHash> {
>+  template <class T>
>+  size_t operator()(const T& t) const {
>+    return absl::Hash<T>{}(t);
>+  }
>+};
>+
>+struct StatefulTestingEqual
>+    : absl::container_internal::hash_testing_internal::WithId<
>+          StatefulTestingEqual> {
>+  template <class T, class U>
>+  bool operator()(const T& t, const U& u) const {
>+    return t == u;
>+  }
>+};
>+
>+// It is expected that Alloc() == Alloc() for all allocators so we cannot use
>+// WithId base. We need to explicitly assign ids.
>+template <class T = int>
>+struct Alloc : std::allocator<T> {
>+  using propagate_on_container_swap = std::true_type;
>+
>+  // Using old paradigm for this to ensure compatibility.
>+  explicit Alloc(size_t id = 0) : id_(id) {}
>+
>+  Alloc(const Alloc&) = default;
>+  Alloc& operator=(const Alloc&) = default;
>+
>+  template <class U>
>+  Alloc(const Alloc<U>& that) : std::allocator<T>(that), id_(that.id()) {}
>+
>+  template <class U>
>+  struct rebind {
>+    using other = Alloc<U>;
>+  };
>+
>+  size_t id() const { return id_; }
>+
>+  friend bool operator==(const Alloc& a, const Alloc& b) {
>+    return a.id_ == b.id_;
>+  }
>+  friend bool operator!=(const Alloc& a, const Alloc& b) { return !(a == b); }
>+
>+ private:
>+  size_t id_ = std::numeric_limits<size_t>::max();
>+};
>+
>+template <class Map>
>+auto items(const Map& m) -> std::vector<
>+    std::pair<typename Map::key_type, typename Map::mapped_type>> {
>+  using std::get;
>+  std::vector<std::pair<typename Map::key_type, typename Map::mapped_type>> res;
>+  res.reserve(m.size());
>+  for (const auto& v : m) res.emplace_back(get<0>(v), get<1>(v));
>+  return res;
>+}
>+
>+template <class Set>
>+auto keys(const Set& s)
>+    -> std::vector<typename std::decay<typename Set::key_type>::type> {
>+  std::vector<typename std::decay<typename Set::key_type>::type> res;
>+  res.reserve(s.size());
>+  for (const auto& v : s) res.emplace_back(v);
>+  return res;
>+}
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+// ABSL_UNORDERED_SUPPORTS_ALLOC_CTORS is false for glibcxx versions
>+// where the unordered containers are missing certain constructors that
>+// take allocator arguments. This test is defined ad-hoc for the platforms
>+// we care about (notably Crosstool 17) because libstdcxx's useless
>+// versioning scheme precludes a more principled solution.
>+#if defined(__GLIBCXX__) && __GLIBCXX__ <= 20140425
>+#define ABSL_UNORDERED_SUPPORTS_ALLOC_CTORS 0
>+#else
>+#define ABSL_UNORDERED_SUPPORTS_ALLOC_CTORS 1
>+#endif
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_HASH_POLICY_TESTING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_testing_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_testing_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..c215c4237ae551686dd824d42da7cfb93b0ce915
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_testing_test.cc
>@@ -0,0 +1,43 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/hash_policy_testing.h"
>+
>+#include "gtest/gtest.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+TEST(_, Hash) {
>+  StatefulTestingHash h1;
>+  EXPECT_EQ(1, h1.id());
>+  StatefulTestingHash h2;
>+  EXPECT_EQ(2, h2.id());
>+  StatefulTestingHash h1c(h1);
>+  EXPECT_EQ(1, h1c.id());
>+  StatefulTestingHash h2m(std::move(h2));
>+  EXPECT_EQ(2, h2m.id());
>+  EXPECT_EQ(0, h2.id());
>+  StatefulTestingHash h3;
>+  EXPECT_EQ(3, h3.id());
>+  h3 = StatefulTestingHash();
>+  EXPECT_EQ(4, h3.id());
>+  h3 = std::move(h1);
>+  EXPECT_EQ(1, h3.id());
>+}
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_traits.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_traits.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..029e47e175c9dfae3524b7cb153931026de5bf97
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_traits.h
>@@ -0,0 +1,189 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_HASH_POLICY_TRAITS_H_
>+#define ABSL_CONTAINER_INTERNAL_HASH_POLICY_TRAITS_H_
>+
>+#include <cstddef>
>+#include <memory>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/meta/type_traits.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+// Defines how slots are initialized/destroyed/moved.
>+template <class Policy, class = void>
>+struct hash_policy_traits {
>+ private:
>+  struct ReturnKey {
>+    // We return `Key` here.
>+    // When Key=T&, we forward the lvalue reference.
>+    // When Key=T, we return by value to avoid a dangling reference.
>+    // eg, for string_hash_map.
>+    template <class Key, class... Args>
>+    Key operator()(Key&& k, const Args&...) const {
>+      return std::forward<Key>(k);
>+    }
>+  };
>+
>+  template <class P = Policy, class = void>
>+  struct ConstantIteratorsImpl : std::false_type {};
>+
>+  template <class P>
>+  struct ConstantIteratorsImpl<P, absl::void_t<typename P::constant_iterators>>
>+      : P::constant_iterators {};
>+
>+ public:
>+  // The actual object stored in the hash table.
>+  using slot_type = typename Policy::slot_type;
>+
>+  // The type of the keys stored in the hashtable.
>+  using key_type = typename Policy::key_type;
>+
>+  // The argument type for insertions into the hashtable. This is different
>+  // from value_type for increased performance. See initializer_list constructor
>+  // and insert() member functions for more details.
>+  using init_type = typename Policy::init_type;
>+
>+  using reference = decltype(Policy::element(std::declval<slot_type*>()));
>+  using pointer = typename std::remove_reference<reference>::type*;
>+  using value_type = typename std::remove_reference<reference>::type;
>+
>+  // Policies can set this variable to tell raw_hash_set that all iterators
>+  // should be constant, even `iterator`. This is useful for set-like
>+  // containers.
>+  // Defaults to false if not provided by the policy.
>+  using constant_iterators = ConstantIteratorsImpl<>;
>+
>+  // PRECONDITION: `slot` is UNINITIALIZED
>+  // POSTCONDITION: `slot` is INITIALIZED
>+  template <class Alloc, class... Args>
>+  static void construct(Alloc* alloc, slot_type* slot, Args&&... args) {
>+    Policy::construct(alloc, slot, std::forward<Args>(args)...);
>+  }
>+
>+  // PRECONDITION: `slot` is INITIALIZED
>+  // POSTCONDITION: `slot` is UNINITIALIZED
>+  template <class Alloc>
>+  static void destroy(Alloc* alloc, slot_type* slot) {
>+    Policy::destroy(alloc, slot);
>+  }
>+
>+  // Transfers the `old_slot` to `new_slot`. Any memory allocated by the
>+  // allocator inside `old_slot` to `new_slot` can be transfered.
>+  //
>+  // OPTIONAL: defaults to:
>+  //
>+  //     clone(new_slot, std::move(*old_slot));
>+  //     destroy(old_slot);
>+  //
>+  // PRECONDITION: `new_slot` is UNINITIALIZED and `old_slot` is INITIALIZED
>+  // POSTCONDITION: `new_slot` is INITIALIZED and `old_slot` is
>+  //                UNINITIALIZED
>+  template <class Alloc>
>+  static void transfer(Alloc* alloc, slot_type* new_slot, slot_type* old_slot) {
>+    transfer_impl(alloc, new_slot, old_slot, 0);
>+  }
>+
>+  // PRECONDITION: `slot` is INITIALIZED
>+  // POSTCONDITION: `slot` is INITIALIZED
>+  template <class P = Policy>
>+  static auto element(slot_type* slot) -> decltype(P::element(slot)) {
>+    return P::element(slot);
>+  }
>+
>+  // Returns the amount of memory owned by `slot`, exclusive of `sizeof(*slot)`.
>+  //
>+  // If `slot` is nullptr, returns the constant amount of memory owned by any
>+  // full slot or -1 if slots own variable amounts of memory.
>+  //
>+  // PRECONDITION: `slot` is INITIALIZED or nullptr
>+  template <class P = Policy>
>+  static size_t space_used(const slot_type* slot) {
>+    return P::space_used(slot);
>+  }
>+
>+  // Provides generalized access to the key for elements, both for elements in
>+  // the table and for elements that have not yet been inserted (or even
>+  // constructed).  We would like an API that allows us to say: `key(args...)`
>+  // but we cannot do that for all cases, so we use this more general API that
>+  // can be used for many things, including the following:
>+  //
>+  //   - Given an element in a table, get its key.
>+  //   - Given an element initializer, get its key.
>+  //   - Given `emplace()` arguments, get the element key.
>+  //
>+  // Implementations of this must adhere to a very strict technical
>+  // specification around aliasing and consuming arguments:
>+  //
>+  // Let `value_type` be the result type of `element()` without ref- and
>+  // cv-qualifiers. The first argument is a functor, the rest are constructor
>+  // arguments for `value_type`. Returns `std::forward<F>(f)(k, xs...)`, where
>+  // `k` is the element key, and `xs...` are the new constructor arguments for
>+  // `value_type`. It's allowed for `k` to alias `xs...`, and for both to alias
>+  // `ts...`. The key won't be touched once `xs...` are used to construct an
>+  // element; `ts...` won't be touched at all, which allows `apply()` to consume
>+  // any rvalues among them.
>+  //
>+  // If `value_type` is constructible from `Ts&&...`, `Policy::apply()` must not
>+  // trigger a hard compile error unless it originates from `f`. In other words,
>+  // `Policy::apply()` must be SFINAE-friendly. If `value_type` is not
>+  // constructible from `Ts&&...`, either SFINAE or a hard compile error is OK.
>+  //
>+  // If `Ts...` is `[cv] value_type[&]` or `[cv] init_type[&]`,
>+  // `Policy::apply()` must work. A compile error is not allowed, SFINAE or not.
>+  template <class F, class... Ts, class P = Policy>
>+  static auto apply(F&& f, Ts&&... ts)
>+      -> decltype(P::apply(std::forward<F>(f), std::forward<Ts>(ts)...)) {
>+    return P::apply(std::forward<F>(f), std::forward<Ts>(ts)...);
>+  }
>+
>+  // Returns the "key" portion of the slot.
>+  // Used for node handle manipulation.
>+  template <class P = Policy>
>+  static auto key(slot_type* slot)
>+      -> decltype(P::apply(ReturnKey(), element(slot))) {
>+    return P::apply(ReturnKey(), element(slot));
>+  }
>+
>+  // Returns the "value" (as opposed to the "key") portion of the element. Used
>+  // by maps to implement `operator[]`, `at()` and `insert_or_assign()`.
>+  template <class T, class P = Policy>
>+  static auto value(T* elem) -> decltype(P::value(elem)) {
>+    return P::value(elem);
>+  }
>+
>+ private:
>+  // Use auto -> decltype as an enabler.
>+  template <class Alloc, class P = Policy>
>+  static auto transfer_impl(Alloc* alloc, slot_type* new_slot,
>+                            slot_type* old_slot, int)
>+      -> decltype((void)P::transfer(alloc, new_slot, old_slot)) {
>+    P::transfer(alloc, new_slot, old_slot);
>+  }
>+  template <class Alloc>
>+  static void transfer_impl(Alloc* alloc, slot_type* new_slot,
>+                            slot_type* old_slot, char) {
>+    construct(alloc, new_slot, std::move(element(old_slot)));
>+    destroy(alloc, old_slot);
>+  }
>+};
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_HASH_POLICY_TRAITS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_traits_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_traits_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..423f1548e7ca138ee5aca09242afd3bc4f972b26
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hash_policy_traits_test.cc
>@@ -0,0 +1,142 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/hash_policy_traits.h"
>+
>+#include <functional>
>+#include <memory>
>+#include <new>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+using ::testing::MockFunction;
>+using ::testing::Return;
>+using ::testing::ReturnRef;
>+
>+using Alloc = std::allocator<int>;
>+using Slot = int;
>+
>+struct PolicyWithoutOptionalOps {
>+  using slot_type = Slot;
>+  using key_type = Slot;
>+  using init_type = Slot;
>+
>+  static std::function<void(void*, Slot*, Slot)> construct;
>+  static std::function<void(void*, Slot*)> destroy;
>+
>+  static std::function<Slot&(Slot*)> element;
>+  static int apply(int v) { return apply_impl(v); }
>+  static std::function<int(int)> apply_impl;
>+  static std::function<Slot&(Slot*)> value;
>+};
>+
>+std::function<void(void*, Slot*, Slot)> PolicyWithoutOptionalOps::construct;
>+std::function<void(void*, Slot*)> PolicyWithoutOptionalOps::destroy;
>+
>+std::function<Slot&(Slot*)> PolicyWithoutOptionalOps::element;
>+std::function<int(int)> PolicyWithoutOptionalOps::apply_impl;
>+std::function<Slot&(Slot*)> PolicyWithoutOptionalOps::value;
>+
>+struct PolicyWithOptionalOps : PolicyWithoutOptionalOps {
>+  static std::function<void(void*, Slot*, Slot*)> transfer;
>+};
>+
>+std::function<void(void*, Slot*, Slot*)> PolicyWithOptionalOps::transfer;
>+
>+struct Test : ::testing::Test {
>+  Test() {
>+    PolicyWithoutOptionalOps::construct = [&](void* a1, Slot* a2, Slot a3) {
>+      construct.Call(a1, a2, std::move(a3));
>+    };
>+    PolicyWithoutOptionalOps::destroy = [&](void* a1, Slot* a2) {
>+      destroy.Call(a1, a2);
>+    };
>+
>+    PolicyWithoutOptionalOps::element = [&](Slot* a1) -> Slot& {
>+      return element.Call(a1);
>+    };
>+    PolicyWithoutOptionalOps::apply_impl = [&](int a1) -> int {
>+      return apply.Call(a1);
>+    };
>+    PolicyWithoutOptionalOps::value = [&](Slot* a1) -> Slot& {
>+      return value.Call(a1);
>+    };
>+
>+    PolicyWithOptionalOps::transfer = [&](void* a1, Slot* a2, Slot* a3) {
>+      return transfer.Call(a1, a2, a3);
>+    };
>+  }
>+
>+  std::allocator<int> alloc;
>+  int a = 53;
>+
>+  MockFunction<void(void*, Slot*, Slot)> construct;
>+  MockFunction<void(void*, Slot*)> destroy;
>+
>+  MockFunction<Slot&(Slot*)> element;
>+  MockFunction<int(int)> apply;
>+  MockFunction<Slot&(Slot*)> value;
>+
>+  MockFunction<void(void*, Slot*, Slot*)> transfer;
>+};
>+
>+TEST_F(Test, construct) {
>+  EXPECT_CALL(construct, Call(&alloc, &a, 53));
>+  hash_policy_traits<PolicyWithoutOptionalOps>::construct(&alloc, &a, 53);
>+}
>+
>+TEST_F(Test, destroy) {
>+  EXPECT_CALL(destroy, Call(&alloc, &a));
>+  hash_policy_traits<PolicyWithoutOptionalOps>::destroy(&alloc, &a);
>+}
>+
>+TEST_F(Test, element) {
>+  int b = 0;
>+  EXPECT_CALL(element, Call(&a)).WillOnce(ReturnRef(b));
>+  EXPECT_EQ(&b, &hash_policy_traits<PolicyWithoutOptionalOps>::element(&a));
>+}
>+
>+TEST_F(Test, apply) {
>+  EXPECT_CALL(apply, Call(42)).WillOnce(Return(1337));
>+  EXPECT_EQ(1337, (hash_policy_traits<PolicyWithoutOptionalOps>::apply(42)));
>+}
>+
>+TEST_F(Test, value) {
>+  int b = 0;
>+  EXPECT_CALL(value, Call(&a)).WillOnce(ReturnRef(b));
>+  EXPECT_EQ(&b, &hash_policy_traits<PolicyWithoutOptionalOps>::value(&a));
>+}
>+
>+TEST_F(Test, without_transfer) {
>+  int b = 42;
>+  EXPECT_CALL(element, Call(&b)).WillOnce(::testing::ReturnRef(b));
>+  EXPECT_CALL(construct, Call(&alloc, &a, b));
>+  EXPECT_CALL(destroy, Call(&alloc, &b));
>+  hash_policy_traits<PolicyWithoutOptionalOps>::transfer(&alloc, &a, &b);
>+}
>+
>+TEST_F(Test, with_transfer) {
>+  int b = 42;
>+  EXPECT_CALL(transfer, Call(&alloc, &a, &b));
>+  hash_policy_traits<PolicyWithOptionalOps>::transfer(&alloc, &a, &b);
>+}
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hashtable_debug.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hashtable_debug.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..c3bd65c9c4ec52d5f4a796f76b18dbcda7292b34
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hashtable_debug.h
>@@ -0,0 +1,108 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This library provides APIs to debug the probing behavior of hash tables.
>+//
>+// In general, the probing behavior is a black box for users and only the
>+// side effects can be measured in the form of performance differences.
>+// These APIs give a glimpse on the actual behavior of the probing algorithms in
>+// these hashtables given a specified hash function and a set of elements.
>+//
>+// The probe count distribution can be used to assess the quality of the hash
>+// function for that particular hash table. Note that a hash function that
>+// performs well in one hash table implementation does not necessarily performs
>+// well in a different one.
>+//
>+// This library supports std::unordered_{set,map}, dense_hash_{set,map} and
>+// absl::{flat,node,string}_hash_{set,map}.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_HASHTABLE_DEBUG_H_
>+#define ABSL_CONTAINER_INTERNAL_HASHTABLE_DEBUG_H_
>+
>+#include <cstddef>
>+#include <algorithm>
>+#include <type_traits>
>+#include <vector>
>+
>+#include "absl/container/internal/hashtable_debug_hooks.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+// Returns the number of probes required to lookup `key`.  Returns 0 for a
>+// search with no collisions.  Higher values mean more hash collisions occurred;
>+// however, the exact meaning of this number varies according to the container
>+// type.
>+template <typename C>
>+size_t GetHashtableDebugNumProbes(
>+    const C& c, const typename C::key_type& key) {
>+  return absl::container_internal::hashtable_debug_internal::
>+      HashtableDebugAccess<C>::GetNumProbes(c, key);
>+}
>+
>+// Gets a histogram of the number of probes for each elements in the container.
>+// The sum of all the values in the vector is equal to container.size().
>+template <typename C>
>+std::vector<size_t> GetHashtableDebugNumProbesHistogram(const C& container) {
>+  std::vector<size_t> v;
>+  for (auto it = container.begin(); it != container.end(); ++it) {
>+    size_t num_probes = GetHashtableDebugNumProbes(
>+        container,
>+        absl::container_internal::hashtable_debug_internal::GetKey<C>(*it, 0));
>+    v.resize(std::max(v.size(), num_probes + 1));
>+    v[num_probes]++;
>+  }
>+  return v;
>+}
>+
>+struct HashtableDebugProbeSummary {
>+  size_t total_elements;
>+  size_t total_num_probes;
>+  double mean;
>+};
>+
>+// Gets a summary of the probe count distribution for the elements in the
>+// container.
>+template <typename C>
>+HashtableDebugProbeSummary GetHashtableDebugProbeSummary(const C& container) {
>+  auto probes = GetHashtableDebugNumProbesHistogram(container);
>+  HashtableDebugProbeSummary summary = {};
>+  for (size_t i = 0; i < probes.size(); ++i) {
>+    summary.total_elements += probes[i];
>+    summary.total_num_probes += probes[i] * i;
>+  }
>+  summary.mean = 1.0 * summary.total_num_probes / summary.total_elements;
>+  return summary;
>+}
>+
>+// Returns the number of bytes requested from the allocator by the container
>+// and not freed.
>+template <typename C>
>+size_t AllocatedByteSize(const C& c) {
>+  return absl::container_internal::hashtable_debug_internal::
>+      HashtableDebugAccess<C>::AllocatedByteSize(c);
>+}
>+
>+// Returns a tight lower bound for AllocatedByteSize(c) where `c` is of type `C`
>+// and `c.size()` is equal to `num_elements`.
>+template <typename C>
>+size_t LowerBoundAllocatedByteSize(size_t num_elements) {
>+  return absl::container_internal::hashtable_debug_internal::
>+      HashtableDebugAccess<C>::LowerBoundAllocatedByteSize(num_elements);
>+}
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_HASHTABLE_DEBUG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hashtable_debug_hooks.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hashtable_debug_hooks.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..8f219726bee01fe97a8f10e3306494868094e92b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/hashtable_debug_hooks.h
>@@ -0,0 +1,81 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Provides the internal API for hashtable_debug.h.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_HASHTABLE_DEBUG_HOOKS_H_
>+#define ABSL_CONTAINER_INTERNAL_HASHTABLE_DEBUG_HOOKS_H_
>+
>+#include <cstddef>
>+
>+#include <algorithm>
>+#include <type_traits>
>+#include <vector>
>+
>+namespace absl {
>+namespace container_internal {
>+namespace hashtable_debug_internal {
>+
>+// If it is a map, call get<0>().
>+using std::get;
>+template <typename T, typename = typename T::mapped_type>
>+auto GetKey(const typename T::value_type& pair, int) -> decltype(get<0>(pair)) {
>+  return get<0>(pair);
>+}
>+
>+// If it is not a map, return the value directly.
>+template <typename T>
>+const typename T::key_type& GetKey(const typename T::key_type& key, char) {
>+  return key;
>+}
>+
>+// Containers should specialize this to provide debug information for that
>+// container.
>+template <class Container, typename Enabler = void>
>+struct HashtableDebugAccess {
>+  // Returns the number of probes required to find `key` in `c`.  The "number of
>+  // probes" is a concept that can vary by container.  Implementations should
>+  // return 0 when `key` was found in the minimum number of operations and
>+  // should increment the result for each non-trivial operation required to find
>+  // `key`.
>+  //
>+  // The default implementation uses the bucket api from the standard and thus
>+  // works for `std::unordered_*` containers.
>+  static size_t GetNumProbes(const Container& c,
>+                             const typename Container::key_type& key) {
>+    if (!c.bucket_count()) return {};
>+    size_t num_probes = 0;
>+    size_t bucket = c.bucket(key);
>+    for (auto it = c.begin(bucket), e = c.end(bucket);; ++it, ++num_probes) {
>+      if (it == e) return num_probes;
>+      if (c.key_eq()(key, GetKey<Container>(*it, 0))) return num_probes;
>+    }
>+  }
>+
>+  // Returns the number of bytes requested from the allocator by the container
>+  // and not freed.
>+  //
>+  // static size_t AllocatedByteSize(const Container& c);
>+
>+  // Returns a tight lower bound for AllocatedByteSize(c) where `c` is of type
>+  // `Container` and `c.size()` is equal to `num_elements`.
>+  //
>+  // static size_t LowerBoundAllocatedByteSize(size_t num_elements);
>+};
>+
>+}  // namespace hashtable_debug_internal
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_HASHTABLE_DEBUG_HOOKS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/layout.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/layout.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..676c7d67ee68dcd566e17094d4958e3747547d99
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/layout.h
>@@ -0,0 +1,732 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+//                           MOTIVATION AND TUTORIAL
>+//
>+// If you want to put in a single heap allocation N doubles followed by M ints,
>+// it's easy if N and M are known at compile time.
>+//
>+//   struct S {
>+//     double a[N];
>+//     int b[M];
>+//   };
>+//
>+//   S* p = new S;
>+//
>+// But what if N and M are known only in run time? Class template Layout to the
>+// rescue! It's a portable generalization of the technique known as struct hack.
>+//
>+//   // This object will tell us everything we need to know about the memory
>+//   // layout of double[N] followed by int[M]. It's structurally identical to
>+//   // size_t[2] that stores N and M. It's very cheap to create.
>+//   const Layout<double, int> layout(N, M);
>+//
>+//   // Allocate enough memory for both arrays. `AllocSize()` tells us how much
>+//   // memory is needed. We are free to use any allocation function we want as
>+//   // long as it returns aligned memory.
>+//   std::unique_ptr<unsigned char[]> p(new unsigned char[layout.AllocSize()]);
>+//
>+//   // Obtain the pointer to the array of doubles.
>+//   // Equivalent to `reinterpret_cast<double*>(p.get())`.
>+//   //
>+//   // We could have written layout.Pointer<0>(p) instead. If all the types are
>+//   // unique you can use either form, but if some types are repeated you must
>+//   // use the index form.
>+//   double* a = layout.Pointer<double>(p.get());
>+//
>+//   // Obtain the pointer to the array of ints.
>+//   // Equivalent to `reinterpret_cast<int*>(p.get() + N * 8)`.
>+//   int* b = layout.Pointer<int>(p);
>+//
>+// If we are unable to specify sizes of all fields, we can pass as many sizes as
>+// we can to `Partial()`. In return, it'll allow us to access the fields whose
>+// locations and sizes can be computed from the provided information.
>+// `Partial()` comes in handy when the array sizes are embedded into the
>+// allocation.
>+//
>+//   // size_t[1] containing N, size_t[1] containing M, double[N], int[M].
>+//   using L = Layout<size_t, size_t, double, int>;
>+//
>+//   unsigned char* Allocate(size_t n, size_t m) {
>+//     const L layout(1, 1, n, m);
>+//     unsigned char* p = new unsigned char[layout.AllocSize()];
>+//     *layout.Pointer<0>(p) = n;
>+//     *layout.Pointer<1>(p) = m;
>+//     return p;
>+//   }
>+//
>+//   void Use(unsigned char* p) {
>+//     // First, extract N and M.
>+//     // Specify that the first array has only one element. Using `prefix` we
>+//     // can access the first two arrays but not more.
>+//     constexpr auto prefix = L::Partial(1);
>+//     size_t n = *prefix.Pointer<0>(p);
>+//     size_t m = *prefix.Pointer<1>(p);
>+//
>+//     // Now we can get pointers to the payload.
>+//     const L layout(1, 1, n, m);
>+//     double* a = layout.Pointer<double>(p);
>+//     int* b = layout.Pointer<int>(p);
>+//   }
>+//
>+// The layout we used above combines fixed-size with dynamically-sized fields.
>+// This is quite common. Layout is optimized for this use case and generates
>+// optimal code. All computations that can be performed at compile time are
>+// indeed performed at compile time.
>+//
>+// Efficiency tip: The order of fields matters. In `Layout<T1, ..., TN>` try to
>+// ensure that `alignof(T1) >= ... >= alignof(TN)`. This way you'll have no
>+// padding in between arrays.
>+//
>+// You can manually override the alignment of an array by wrapping the type in
>+// `Aligned<T, N>`. `Layout<..., Aligned<T, N>, ...>` has exactly the same API
>+// and behavior as `Layout<..., T, ...>` except that the first element of the
>+// array of `T` is aligned to `N` (the rest of the elements follow without
>+// padding). `N` cannot be less than `alignof(T)`.
>+//
>+// `AllocSize()` and `Pointer()` are the most basic methods for dealing with
>+// memory layouts. Check out the reference or code below to discover more.
>+//
>+//                            EXAMPLE
>+//
>+//   // Immutable move-only string with sizeof equal to sizeof(void*). The
>+//   // string size and the characters are kept in the same heap allocation.
>+//   class CompactString {
>+//    public:
>+//     CompactString(const char* s = "") {
>+//       const size_t size = strlen(s);
>+//       // size_t[1] followed by char[size + 1].
>+//       const L layout(1, size + 1);
>+//       p_.reset(new unsigned char[layout.AllocSize()]);
>+//       // If running under ASAN, mark the padding bytes, if any, to catch
>+//       // memory errors.
>+//       layout.PoisonPadding(p_.get());
>+//       // Store the size in the allocation.
>+//       *layout.Pointer<size_t>(p_.get()) = size;
>+//       // Store the characters in the allocation.
>+//       memcpy(layout.Pointer<char>(p_.get()), s, size + 1);
>+//     }
>+//
>+//     size_t size() const {
>+//       // Equivalent to reinterpret_cast<size_t&>(*p).
>+//       return *L::Partial().Pointer<size_t>(p_.get());
>+//     }
>+//
>+//     const char* c_str() const {
>+//       // Equivalent to reinterpret_cast<char*>(p.get() + sizeof(size_t)).
>+//       // The argument in Partial(1) specifies that we have size_t[1] in front
>+//       // of the characters.
>+//       return L::Partial(1).Pointer<char>(p_.get());
>+//     }
>+//
>+//    private:
>+//     // Our heap allocation contains a size_t followed by an array of chars.
>+//     using L = Layout<size_t, char>;
>+//     std::unique_ptr<unsigned char[]> p_;
>+//   };
>+//
>+//   int main() {
>+//     CompactString s = "hello";
>+//     assert(s.size() == 5);
>+//     assert(strcmp(s.c_str(), "hello") == 0);
>+//   }
>+//
>+//                               DOCUMENTATION
>+//
>+// The interface exported by this file consists of:
>+// - class `Layout<>` and its public members.
>+// - The public members of class `internal_layout::LayoutImpl<>`. That class
>+//   isn't intended to be used directly, and its name and template parameter
>+//   list are internal implementation details, but the class itself provides
>+//   most of the functionality in this file. See comments on its members for
>+//   detailed documentation.
>+//
>+// `Layout<T1,... Tn>::Partial(count1,..., countm)` (where `m` <= `n`) returns a
>+// `LayoutImpl<>` object. `Layout<T1,..., Tn> layout(count1,..., countn)`
>+// creates a `Layout` object, which exposes the same functionality by inheriting
>+// from `LayoutImpl<>`.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_LAYOUT_H_
>+#define ABSL_CONTAINER_INTERNAL_LAYOUT_H_
>+
>+#include <assert.h>
>+#include <stddef.h>
>+#include <stdint.h>
>+#include <ostream>
>+#include <string>
>+#include <tuple>
>+#include <type_traits>
>+#include <typeinfo>
>+#include <utility>
>+
>+#ifdef ADDRESS_SANITIZER
>+#include <sanitizer/asan_interface.h>
>+#endif
>+
>+#include "absl/meta/type_traits.h"
>+#include "absl/strings/str_cat.h"
>+#include "absl/types/span.h"
>+#include "absl/utility/utility.h"
>+
>+#if defined(__GXX_RTTI)
>+#define ABSL_INTERNAL_HAS_CXA_DEMANGLE
>+#endif
>+
>+#ifdef ABSL_INTERNAL_HAS_CXA_DEMANGLE
>+#include <cxxabi.h>
>+#endif
>+
>+namespace absl {
>+namespace container_internal {
>+
>+// A type wrapper that instructs `Layout` to use the specific alignment for the
>+// array. `Layout<..., Aligned<T, N>, ...>` has exactly the same API
>+// and behavior as `Layout<..., T, ...>` except that the first element of the
>+// array of `T` is aligned to `N` (the rest of the elements follow without
>+// padding).
>+//
>+// Requires: `N >= alignof(T)` and `N` is a power of 2.
>+template <class T, size_t N>
>+struct Aligned;
>+
>+namespace internal_layout {
>+
>+template <class T>
>+struct NotAligned {};
>+
>+template <class T, size_t N>
>+struct NotAligned<const Aligned<T, N>> {
>+  static_assert(sizeof(T) == 0, "Aligned<T, N> cannot be const-qualified");
>+};
>+
>+template <size_t>
>+using IntToSize = size_t;
>+
>+template <class>
>+using TypeToSize = size_t;
>+
>+template <class T>
>+struct Type : NotAligned<T> {
>+  using type = T;
>+};
>+
>+template <class T, size_t N>
>+struct Type<Aligned<T, N>> {
>+  using type = T;
>+};
>+
>+template <class T>
>+struct SizeOf : NotAligned<T>, std::integral_constant<size_t, sizeof(T)> {};
>+
>+template <class T, size_t N>
>+struct SizeOf<Aligned<T, N>> : std::integral_constant<size_t, sizeof(T)> {};
>+
>+template <class T>
>+struct AlignOf : NotAligned<T>, std::integral_constant<size_t, alignof(T)> {};
>+
>+template <class T, size_t N>
>+struct AlignOf<Aligned<T, N>> : std::integral_constant<size_t, N> {
>+  static_assert(N % alignof(T) == 0,
>+                "Custom alignment can't be lower than the type's alignment");
>+};
>+
>+// Does `Ts...` contain `T`?
>+template <class T, class... Ts>
>+using Contains = absl::disjunction<std::is_same<T, Ts>...>;
>+
>+template <class From, class To>
>+using CopyConst =
>+    typename std::conditional<std::is_const<From>::value, const To, To>::type;
>+
>+template <class T>
>+using SliceType = absl::Span<T>;
>+
>+// This namespace contains no types. It prevents functions defined in it from
>+// being found by ADL.
>+namespace adl_barrier {
>+
>+template <class Needle, class... Ts>
>+constexpr size_t Find(Needle, Needle, Ts...) {
>+  static_assert(!Contains<Needle, Ts...>(), "Duplicate element type");
>+  return 0;
>+}
>+
>+template <class Needle, class T, class... Ts>
>+constexpr size_t Find(Needle, T, Ts...) {
>+  return adl_barrier::Find(Needle(), Ts()...) + 1;
>+}
>+
>+constexpr bool IsPow2(size_t n) { return !(n & (n - 1)); }
>+
>+// Returns `q * m` for the smallest `q` such that `q * m >= n`.
>+// Requires: `m` is a power of two. It's enforced by IsLegalElementType below.
>+constexpr size_t Align(size_t n, size_t m) { return (n + m - 1) & ~(m - 1); }
>+
>+constexpr size_t Min(size_t a, size_t b) { return b < a ? b : a; }
>+
>+constexpr size_t Max(size_t a) { return a; }
>+
>+template <class... Ts>
>+constexpr size_t Max(size_t a, size_t b, Ts... rest) {
>+  return adl_barrier::Max(b < a ? a : b, rest...);
>+}
>+
>+template <class T>
>+std::string TypeName() {
>+  std::string out;
>+  int status = 0;
>+  char* demangled = nullptr;
>+#ifdef ABSL_INTERNAL_HAS_CXA_DEMANGLE
>+  demangled = abi::__cxa_demangle(typeid(T).name(), nullptr, nullptr, &status);
>+#endif
>+  if (status == 0 && demangled != nullptr) {  // Demangling succeeeded.
>+    absl::StrAppend(&out, "<", demangled, ">");
>+    free(demangled);
>+  } else {
>+#if defined(__GXX_RTTI) || defined(_CPPRTTI)
>+    absl::StrAppend(&out, "<", typeid(T).name(), ">");
>+#endif
>+  }
>+  return out;
>+}
>+
>+}  // namespace adl_barrier
>+
>+template <bool C>
>+using EnableIf = typename std::enable_if<C, int>::type;
>+
>+// Can `T` be a template argument of `Layout`?
>+template <class T>
>+using IsLegalElementType = std::integral_constant<
>+    bool, !std::is_reference<T>::value && !std::is_volatile<T>::value &&
>+              !std::is_reference<typename Type<T>::type>::value &&
>+              !std::is_volatile<typename Type<T>::type>::value &&
>+              adl_barrier::IsPow2(AlignOf<T>::value)>;
>+
>+template <class Elements, class SizeSeq, class OffsetSeq>
>+class LayoutImpl;
>+
>+// Public base class of `Layout` and the result type of `Layout::Partial()`.
>+//
>+// `Elements...` contains all template arguments of `Layout` that created this
>+// instance.
>+//
>+// `SizeSeq...` is `[0, NumSizes)` where `NumSizes` is the number of arguments
>+// passed to `Layout::Partial()` or `Layout::Layout()`.
>+//
>+// `OffsetSeq...` is `[0, NumOffsets)` where `NumOffsets` is
>+// `Min(sizeof...(Elements), NumSizes + 1)` (the number of arrays for which we
>+// can compute offsets).
>+template <class... Elements, size_t... SizeSeq, size_t... OffsetSeq>
>+class LayoutImpl<std::tuple<Elements...>, absl::index_sequence<SizeSeq...>,
>+                 absl::index_sequence<OffsetSeq...>> {
>+ private:
>+  static_assert(sizeof...(Elements) > 0, "At least one field is required");
>+  static_assert(absl::conjunction<IsLegalElementType<Elements>...>::value,
>+                "Invalid element type (see IsLegalElementType)");
>+
>+  enum {
>+    NumTypes = sizeof...(Elements),
>+    NumSizes = sizeof...(SizeSeq),
>+    NumOffsets = sizeof...(OffsetSeq),
>+  };
>+
>+  // These are guaranteed by `Layout`.
>+  static_assert(NumOffsets == adl_barrier::Min(NumTypes, NumSizes + 1),
>+                "Internal error");
>+  static_assert(NumTypes > 0, "Internal error");
>+
>+  // Returns the index of `T` in `Elements...`. Results in a compilation error
>+  // if `Elements...` doesn't contain exactly one instance of `T`.
>+  template <class T>
>+  static constexpr size_t ElementIndex() {
>+    static_assert(Contains<Type<T>, Type<typename Type<Elements>::type>...>(),
>+                  "Type not found");
>+    return adl_barrier::Find(Type<T>(),
>+                             Type<typename Type<Elements>::type>()...);
>+  }
>+
>+  template <size_t N>
>+  using ElementAlignment =
>+      AlignOf<typename std::tuple_element<N, std::tuple<Elements...>>::type>;
>+
>+ public:
>+  // Element types of all arrays packed in a tuple.
>+  using ElementTypes = std::tuple<typename Type<Elements>::type...>;
>+
>+  // Element type of the Nth array.
>+  template <size_t N>
>+  using ElementType = typename std::tuple_element<N, ElementTypes>::type;
>+
>+  constexpr explicit LayoutImpl(IntToSize<SizeSeq>... sizes)
>+      : size_{sizes...} {}
>+
>+  // Alignment of the layout, equal to the strictest alignment of all elements.
>+  // All pointers passed to the methods of layout must be aligned to this value.
>+  static constexpr size_t Alignment() {
>+    return adl_barrier::Max(AlignOf<Elements>::value...);
>+  }
>+
>+  // Offset in bytes of the Nth array.
>+  //
>+  //   // int[3], 4 bytes of padding, double[4].
>+  //   Layout<int, double> x(3, 4);
>+  //   assert(x.Offset<0>() == 0);   // The ints starts from 0.
>+  //   assert(x.Offset<1>() == 16);  // The doubles starts from 16.
>+  //
>+  // Requires: `N <= NumSizes && N < sizeof...(Ts)`.
>+  template <size_t N, EnableIf<N == 0> = 0>
>+  constexpr size_t Offset() const {
>+    return 0;
>+  }
>+
>+  template <size_t N, EnableIf<N != 0> = 0>
>+  constexpr size_t Offset() const {
>+    static_assert(N < NumOffsets, "Index out of bounds");
>+    return adl_barrier::Align(
>+        Offset<N - 1>() + SizeOf<ElementType<N - 1>>() * size_[N - 1],
>+        ElementAlignment<N>());
>+  }
>+
>+  // Offset in bytes of the array with the specified element type. There must
>+  // be exactly one such array and its zero-based index must be at most
>+  // `NumSizes`.
>+  //
>+  //   // int[3], 4 bytes of padding, double[4].
>+  //   Layout<int, double> x(3, 4);
>+  //   assert(x.Offset<int>() == 0);      // The ints starts from 0.
>+  //   assert(x.Offset<double>() == 16);  // The doubles starts from 16.
>+  template <class T>
>+  constexpr size_t Offset() const {
>+    return Offset<ElementIndex<T>()>();
>+  }
>+
>+  // Offsets in bytes of all arrays for which the offsets are known.
>+  constexpr std::array<size_t, NumOffsets> Offsets() const {
>+    return {{Offset<OffsetSeq>()...}};
>+  }
>+
>+  // The number of elements in the Nth array. This is the Nth argument of
>+  // `Layout::Partial()` or `Layout::Layout()` (zero-based).
>+  //
>+  //   // int[3], 4 bytes of padding, double[4].
>+  //   Layout<int, double> x(3, 4);
>+  //   assert(x.Size<0>() == 3);
>+  //   assert(x.Size<1>() == 4);
>+  //
>+  // Requires: `N < NumSizes`.
>+  template <size_t N>
>+  constexpr size_t Size() const {
>+    static_assert(N < NumSizes, "Index out of bounds");
>+    return size_[N];
>+  }
>+
>+  // The number of elements in the array with the specified element type.
>+  // There must be exactly one such array and its zero-based index must be
>+  // at most `NumSizes`.
>+  //
>+  //   // int[3], 4 bytes of padding, double[4].
>+  //   Layout<int, double> x(3, 4);
>+  //   assert(x.Size<int>() == 3);
>+  //   assert(x.Size<double>() == 4);
>+  template <class T>
>+  constexpr size_t Size() const {
>+    return Size<ElementIndex<T>()>();
>+  }
>+
>+    // The number of elements of all arrays for which they are known.
>+  constexpr std::array<size_t, NumSizes> Sizes() const {
>+    return {{Size<SizeSeq>()...}};
>+  }
>+
>+  // Pointer to the beginning of the Nth array.
>+  //
>+  // `Char` must be `[const] [signed|unsigned] char`.
>+  //
>+  //   // int[3], 4 bytes of padding, double[4].
>+  //   Layout<int, double> x(3, 4);
>+  //   unsigned char* p = new unsigned char[x.AllocSize()];
>+  //   int* ints = x.Pointer<0>(p);
>+  //   double* doubles = x.Pointer<1>(p);
>+  //
>+  // Requires: `N <= NumSizes && N < sizeof...(Ts)`.
>+  // Requires: `p` is aligned to `Alignment()`.
>+  template <size_t N, class Char>
>+  CopyConst<Char, ElementType<N>>* Pointer(Char* p) const {
>+    using C = typename std::remove_const<Char>::type;
>+    static_assert(
>+        std::is_same<C, char>() || std::is_same<C, unsigned char>() ||
>+            std::is_same<C, signed char>(),
>+        "The argument must be a pointer to [const] [signed|unsigned] char");
>+    constexpr size_t alignment = Alignment();
>+    (void)alignment;
>+    assert(reinterpret_cast<uintptr_t>(p) % alignment == 0);
>+    return reinterpret_cast<CopyConst<Char, ElementType<N>>*>(p + Offset<N>());
>+  }
>+
>+  // Pointer to the beginning of the array with the specified element type.
>+  // There must be exactly one such array and its zero-based index must be at
>+  // most `NumSizes`.
>+  //
>+  // `Char` must be `[const] [signed|unsigned] char`.
>+  //
>+  //   // int[3], 4 bytes of padding, double[4].
>+  //   Layout<int, double> x(3, 4);
>+  //   unsigned char* p = new unsigned char[x.AllocSize()];
>+  //   int* ints = x.Pointer<int>(p);
>+  //   double* doubles = x.Pointer<double>(p);
>+  //
>+  // Requires: `p` is aligned to `Alignment()`.
>+  template <class T, class Char>
>+  CopyConst<Char, T>* Pointer(Char* p) const {
>+    return Pointer<ElementIndex<T>()>(p);
>+  }
>+
>+  // Pointers to all arrays for which pointers are known.
>+  //
>+  // `Char` must be `[const] [signed|unsigned] char`.
>+  //
>+  //   // int[3], 4 bytes of padding, double[4].
>+  //   Layout<int, double> x(3, 4);
>+  //   unsigned char* p = new unsigned char[x.AllocSize()];
>+  //
>+  //   int* ints;
>+  //   double* doubles;
>+  //   std::tie(ints, doubles) = x.Pointers(p);
>+  //
>+  // Requires: `p` is aligned to `Alignment()`.
>+  //
>+  // Note: We're not using ElementType alias here because it does not compile
>+  // under MSVC.
>+  template <class Char>
>+  std::tuple<CopyConst<
>+      Char, typename std::tuple_element<OffsetSeq, ElementTypes>::type>*...>
>+  Pointers(Char* p) const {
>+    return std::tuple<CopyConst<Char, ElementType<OffsetSeq>>*...>(
>+        Pointer<OffsetSeq>(p)...);
>+  }
>+
>+  // The Nth array.
>+  //
>+  // `Char` must be `[const] [signed|unsigned] char`.
>+  //
>+  //   // int[3], 4 bytes of padding, double[4].
>+  //   Layout<int, double> x(3, 4);
>+  //   unsigned char* p = new unsigned char[x.AllocSize()];
>+  //   Span<int> ints = x.Slice<0>(p);
>+  //   Span<double> doubles = x.Slice<1>(p);
>+  //
>+  // Requires: `N < NumSizes`.
>+  // Requires: `p` is aligned to `Alignment()`.
>+  template <size_t N, class Char>
>+  SliceType<CopyConst<Char, ElementType<N>>> Slice(Char* p) const {
>+    return SliceType<CopyConst<Char, ElementType<N>>>(Pointer<N>(p), Size<N>());
>+  }
>+
>+  // The array with the specified element type. There must be exactly one
>+  // such array and its zero-based index must be less than `NumSizes`.
>+  //
>+  // `Char` must be `[const] [signed|unsigned] char`.
>+  //
>+  //   // int[3], 4 bytes of padding, double[4].
>+  //   Layout<int, double> x(3, 4);
>+  //   unsigned char* p = new unsigned char[x.AllocSize()];
>+  //   Span<int> ints = x.Slice<int>(p);
>+  //   Span<double> doubles = x.Slice<double>(p);
>+  //
>+  // Requires: `p` is aligned to `Alignment()`.
>+  template <class T, class Char>
>+  SliceType<CopyConst<Char, T>> Slice(Char* p) const {
>+    return Slice<ElementIndex<T>()>(p);
>+  }
>+
>+  // All arrays with known sizes.
>+  //
>+  // `Char` must be `[const] [signed|unsigned] char`.
>+  //
>+  //   // int[3], 4 bytes of padding, double[4].
>+  //   Layout<int, double> x(3, 4);
>+  //   unsigned char* p = new unsigned char[x.AllocSize()];
>+  //
>+  //   Span<int> ints;
>+  //   Span<double> doubles;
>+  //   std::tie(ints, doubles) = x.Slices(p);
>+  //
>+  // Requires: `p` is aligned to `Alignment()`.
>+  //
>+  // Note: We're not using ElementType alias here because it does not compile
>+  // under MSVC.
>+  template <class Char>
>+  std::tuple<SliceType<CopyConst<
>+      Char, typename std::tuple_element<SizeSeq, ElementTypes>::type>>...>
>+  Slices(Char* p) const {
>+    // Workaround for https://gcc.gnu.org/bugzilla/show_bug.cgi?id=63875 (fixed
>+    // in 6.1).
>+    (void)p;
>+    return std::tuple<SliceType<CopyConst<Char, ElementType<SizeSeq>>>...>(
>+        Slice<SizeSeq>(p)...);
>+  }
>+
>+  // The size of the allocation that fits all arrays.
>+  //
>+  //   // int[3], 4 bytes of padding, double[4].
>+  //   Layout<int, double> x(3, 4);
>+  //   unsigned char* p = new unsigned char[x.AllocSize()];  // 48 bytes
>+  //
>+  // Requires: `NumSizes == sizeof...(Ts)`.
>+  constexpr size_t AllocSize() const {
>+    static_assert(NumTypes == NumSizes, "You must specify sizes of all fields");
>+    return Offset<NumTypes - 1>() +
>+           SizeOf<ElementType<NumTypes - 1>>() * size_[NumTypes - 1];
>+  }
>+
>+  // If built with --config=asan, poisons padding bytes (if any) in the
>+  // allocation. The pointer must point to a memory block at least
>+  // `AllocSize()` bytes in length.
>+  //
>+  // `Char` must be `[const] [signed|unsigned] char`.
>+  //
>+  // Requires: `p` is aligned to `Alignment()`.
>+  template <class Char, size_t N = NumOffsets - 1, EnableIf<N == 0> = 0>
>+  void PoisonPadding(const Char* p) const {
>+    Pointer<0>(p);  // verify the requirements on `Char` and `p`
>+  }
>+
>+  template <class Char, size_t N = NumOffsets - 1, EnableIf<N != 0> = 0>
>+  void PoisonPadding(const Char* p) const {
>+    static_assert(N < NumOffsets, "Index out of bounds");
>+    (void)p;
>+#ifdef ADDRESS_SANITIZER
>+    PoisonPadding<Char, N - 1>(p);
>+    // The `if` is an optimization. It doesn't affect the observable behaviour.
>+    if (ElementAlignment<N - 1>() % ElementAlignment<N>()) {
>+      size_t start =
>+          Offset<N - 1>() + SizeOf<ElementType<N - 1>>() * size_[N - 1];
>+      ASAN_POISON_MEMORY_REGION(p + start, Offset<N>() - start);
>+    }
>+#endif
>+  }
>+
>+  // Human-readable description of the memory layout. Useful for debugging.
>+  // Slow.
>+  //
>+  //   // char[5], 3 bytes of padding, int[3], 4 bytes of padding, followed
>+  //   // by an unknown number of doubles.
>+  //   auto x = Layout<char, int, double>::Partial(5, 3);
>+  //   assert(x.DebugString() ==
>+  //          "@0<char>(1)[5]; @8<int>(4)[3]; @24<double>(8)");
>+  //
>+  // Each field is in the following format: @offset<type>(sizeof)[size] (<type>
>+  // may be missing depending on the target platform). For example,
>+  // @8<int>(4)[3] means that at offset 8 we have an array of ints, where each
>+  // int is 4 bytes, and we have 3 of those ints. The size of the last field may
>+  // be missing (as in the example above). Only fields with known offsets are
>+  // described. Type names may differ across platforms: one compiler might
>+  // produce "unsigned*" where another produces "unsigned int *".
>+  std::string DebugString() const {
>+    const auto offsets = Offsets();
>+    const size_t sizes[] = {SizeOf<ElementType<OffsetSeq>>()...};
>+    const std::string types[] = {adl_barrier::TypeName<ElementType<OffsetSeq>>()...};
>+    std::string res = absl::StrCat("@0", types[0], "(", sizes[0], ")");
>+    for (size_t i = 0; i != NumOffsets - 1; ++i) {
>+      absl::StrAppend(&res, "[", size_[i], "]; @", offsets[i + 1], types[i + 1],
>+                      "(", sizes[i + 1], ")");
>+    }
>+    // NumSizes is a constant that may be zero. Some compilers cannot see that
>+    // inside the if statement "size_[NumSizes - 1]" must be valid.
>+    int last = static_cast<int>(NumSizes) - 1;
>+    if (NumTypes == NumSizes && last >= 0) {
>+      absl::StrAppend(&res, "[", size_[last], "]");
>+    }
>+    return res;
>+  }
>+
>+ private:
>+  // Arguments of `Layout::Partial()` or `Layout::Layout()`.
>+  size_t size_[NumSizes > 0 ? NumSizes : 1];
>+};
>+
>+template <size_t NumSizes, class... Ts>
>+using LayoutType = LayoutImpl<
>+    std::tuple<Ts...>, absl::make_index_sequence<NumSizes>,
>+    absl::make_index_sequence<adl_barrier::Min(sizeof...(Ts), NumSizes + 1)>>;
>+
>+}  // namespace internal_layout
>+
>+// Descriptor of arrays of various types and sizes laid out in memory one after
>+// another. See the top of the file for documentation.
>+//
>+// Check out the public API of internal_layout::LayoutImpl above. The type is
>+// internal to the library but its methods are public, and they are inherited
>+// by `Layout`.
>+template <class... Ts>
>+class Layout : public internal_layout::LayoutType<sizeof...(Ts), Ts...> {
>+ public:
>+  static_assert(sizeof...(Ts) > 0, "At least one field is required");
>+  static_assert(
>+      absl::conjunction<internal_layout::IsLegalElementType<Ts>...>::value,
>+      "Invalid element type (see IsLegalElementType)");
>+
>+  // The result type of `Partial()` with `NumSizes` arguments.
>+  template <size_t NumSizes>
>+  using PartialType = internal_layout::LayoutType<NumSizes, Ts...>;
>+
>+  // `Layout` knows the element types of the arrays we want to lay out in
>+  // memory but not the number of elements in each array.
>+  // `Partial(size1, ..., sizeN)` allows us to specify the latter. The
>+  // resulting immutable object can be used to obtain pointers to the
>+  // individual arrays.
>+  //
>+  // It's allowed to pass fewer array sizes than the number of arrays. E.g.,
>+  // if all you need is to the offset of the second array, you only need to
>+  // pass one argument -- the number of elements in the first arrays.
>+  //
>+  //   // int[3] followed by 4 bytes of padding and an unknown number of
>+  //   // doubles.
>+  //   auto x = Layout<int, double>::Partial(3);
>+  //   // doubles start at byte 16.
>+  //   assert(x.Offset<1>() == 16);
>+  //
>+  // If you know the number of elements in all arrays, you can still call
>+  // `Partial()` but it's more convenient to use the constructor of `Layout`.
>+  //
>+  //   Layout<int, double> x(3, 5);
>+  //
>+  // Note: The sizes of the arrays must be specified in number of elements,
>+  // not in bytes.
>+  //
>+  // Requires: `sizeof...(Sizes) <= sizeof...(Ts)`.
>+  // Requires: all arguments are convertible to `size_t`.
>+  template <class... Sizes>
>+  static constexpr PartialType<sizeof...(Sizes)> Partial(Sizes&&... sizes) {
>+    static_assert(sizeof...(Sizes) <= sizeof...(Ts), "");
>+    return PartialType<sizeof...(Sizes)>(absl::forward<Sizes>(sizes)...);
>+  }
>+
>+  // Creates a layout with the sizes of all arrays specified. If you know
>+  // only the sizes of the first N arrays (where N can be zero), you can use
>+  // `Partial()` defined above. The constructor is essentially equivalent to
>+  // calling `Partial()` and passing in all array sizes; the constructor is
>+  // provided as a convenient abbreviation.
>+  //
>+  // Note: The sizes of the arrays must be specified in number of elements,
>+  // not in bytes.
>+  constexpr explicit Layout(internal_layout::TypeToSize<Ts>... sizes)
>+      : internal_layout::LayoutType<sizeof...(Ts), Ts...>(sizes...) {}
>+};
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_LAYOUT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/layout_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/layout_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..f35157a3bd85219886946fd062691c57053b2ede
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/layout_test.cc
>@@ -0,0 +1,1552 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/layout.h"
>+
>+// We need ::max_align_t because some libstdc++ versions don't provide
>+// std::max_align_t
>+#include <stddef.h>
>+#include <cstdint>
>+#include <memory>
>+#include <sstream>
>+#include <type_traits>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/types/span.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+using ::absl::Span;
>+using ::testing::ElementsAre;
>+
>+size_t Distance(const void* from, const void* to) {
>+  ABSL_RAW_CHECK(from <= to, "Distance must be non-negative");
>+  return static_cast<const char*>(to) - static_cast<const char*>(from);
>+}
>+
>+template <class Expected, class Actual>
>+Expected Type(Actual val) {
>+  static_assert(std::is_same<Expected, Actual>(), "");
>+  return val;
>+}
>+
>+using Int128 = int64_t[2];
>+
>+// Properties of types that this test relies on.
>+static_assert(sizeof(int8_t) == 1, "");
>+static_assert(alignof(int8_t) == 1, "");
>+static_assert(sizeof(int16_t) == 2, "");
>+static_assert(alignof(int16_t) == 2, "");
>+static_assert(sizeof(int32_t) == 4, "");
>+static_assert(alignof(int32_t) == 4, "");
>+static_assert(sizeof(Int128) == 16, "");
>+static_assert(alignof(Int128) == 8, "");
>+
>+template <class Expected, class Actual>
>+void SameType() {
>+  static_assert(std::is_same<Expected, Actual>(), "");
>+}
>+
>+TEST(Layout, ElementType) {
>+  {
>+    using L = Layout<int32_t>;
>+    SameType<int32_t, L::ElementType<0>>();
>+    SameType<int32_t, decltype(L::Partial())::ElementType<0>>();
>+    SameType<int32_t, decltype(L::Partial(0))::ElementType<0>>();
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    SameType<int32_t, L::ElementType<0>>();
>+    SameType<int32_t, L::ElementType<1>>();
>+    SameType<int32_t, decltype(L::Partial())::ElementType<0>>();
>+    SameType<int32_t, decltype(L::Partial())::ElementType<1>>();
>+    SameType<int32_t, decltype(L::Partial(0))::ElementType<0>>();
>+    SameType<int32_t, decltype(L::Partial(0))::ElementType<1>>();
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    SameType<int8_t, L::ElementType<0>>();
>+    SameType<int32_t, L::ElementType<1>>();
>+    SameType<Int128, L::ElementType<2>>();
>+    SameType<int8_t, decltype(L::Partial())::ElementType<0>>();
>+    SameType<int8_t, decltype(L::Partial(0))::ElementType<0>>();
>+    SameType<int32_t, decltype(L::Partial(0))::ElementType<1>>();
>+    SameType<int8_t, decltype(L::Partial(0, 0))::ElementType<0>>();
>+    SameType<int32_t, decltype(L::Partial(0, 0))::ElementType<1>>();
>+    SameType<Int128, decltype(L::Partial(0, 0))::ElementType<2>>();
>+    SameType<int8_t, decltype(L::Partial(0, 0, 0))::ElementType<0>>();
>+    SameType<int32_t, decltype(L::Partial(0, 0, 0))::ElementType<1>>();
>+    SameType<Int128, decltype(L::Partial(0, 0, 0))::ElementType<2>>();
>+  }
>+}
>+
>+TEST(Layout, ElementTypes) {
>+  {
>+    using L = Layout<int32_t>;
>+    SameType<std::tuple<int32_t>, L::ElementTypes>();
>+    SameType<std::tuple<int32_t>, decltype(L::Partial())::ElementTypes>();
>+    SameType<std::tuple<int32_t>, decltype(L::Partial(0))::ElementTypes>();
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    SameType<std::tuple<int32_t, int32_t>, L::ElementTypes>();
>+    SameType<std::tuple<int32_t, int32_t>, decltype(L::Partial())::ElementTypes>();
>+    SameType<std::tuple<int32_t, int32_t>, decltype(L::Partial(0))::ElementTypes>();
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    SameType<std::tuple<int8_t, int32_t, Int128>, L::ElementTypes>();
>+    SameType<std::tuple<int8_t, int32_t, Int128>,
>+             decltype(L::Partial())::ElementTypes>();
>+    SameType<std::tuple<int8_t, int32_t, Int128>,
>+             decltype(L::Partial(0))::ElementTypes>();
>+    SameType<std::tuple<int8_t, int32_t, Int128>,
>+             decltype(L::Partial(0, 0))::ElementTypes>();
>+    SameType<std::tuple<int8_t, int32_t, Int128>,
>+             decltype(L::Partial(0, 0, 0))::ElementTypes>();
>+  }
>+}
>+
>+TEST(Layout, OffsetByIndex) {
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, L::Partial().Offset<0>());
>+    EXPECT_EQ(0, L::Partial(3).Offset<0>());
>+    EXPECT_EQ(0, L(3).Offset<0>());
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    EXPECT_EQ(0, L::Partial().Offset<0>());
>+    EXPECT_EQ(0, L::Partial(3).Offset<0>());
>+    EXPECT_EQ(12, L::Partial(3).Offset<1>());
>+    EXPECT_EQ(0, L::Partial(3, 5).Offset<0>());
>+    EXPECT_EQ(12, L::Partial(3, 5).Offset<1>());
>+    EXPECT_EQ(0, L(3, 5).Offset<0>());
>+    EXPECT_EQ(12, L(3, 5).Offset<1>());
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(0, L::Partial().Offset<0>());
>+    EXPECT_EQ(0, L::Partial(0).Offset<0>());
>+    EXPECT_EQ(0, L::Partial(0).Offset<1>());
>+    EXPECT_EQ(0, L::Partial(1).Offset<0>());
>+    EXPECT_EQ(4, L::Partial(1).Offset<1>());
>+    EXPECT_EQ(0, L::Partial(5).Offset<0>());
>+    EXPECT_EQ(8, L::Partial(5).Offset<1>());
>+    EXPECT_EQ(0, L::Partial(0, 0).Offset<0>());
>+    EXPECT_EQ(0, L::Partial(0, 0).Offset<1>());
>+    EXPECT_EQ(0, L::Partial(0, 0).Offset<2>());
>+    EXPECT_EQ(0, L::Partial(1, 0).Offset<0>());
>+    EXPECT_EQ(4, L::Partial(1, 0).Offset<1>());
>+    EXPECT_EQ(8, L::Partial(1, 0).Offset<2>());
>+    EXPECT_EQ(0, L::Partial(5, 3).Offset<0>());
>+    EXPECT_EQ(8, L::Partial(5, 3).Offset<1>());
>+    EXPECT_EQ(24, L::Partial(5, 3).Offset<2>());
>+    EXPECT_EQ(0, L::Partial(0, 0, 0).Offset<0>());
>+    EXPECT_EQ(0, L::Partial(0, 0, 0).Offset<1>());
>+    EXPECT_EQ(0, L::Partial(0, 0, 0).Offset<2>());
>+    EXPECT_EQ(0, L::Partial(1, 0, 0).Offset<0>());
>+    EXPECT_EQ(4, L::Partial(1, 0, 0).Offset<1>());
>+    EXPECT_EQ(8, L::Partial(1, 0, 0).Offset<2>());
>+    EXPECT_EQ(0, L::Partial(5, 3, 1).Offset<0>());
>+    EXPECT_EQ(24, L::Partial(5, 3, 1).Offset<2>());
>+    EXPECT_EQ(8, L::Partial(5, 3, 1).Offset<1>());
>+    EXPECT_EQ(0, L(5, 3, 1).Offset<0>());
>+    EXPECT_EQ(24, L(5, 3, 1).Offset<2>());
>+    EXPECT_EQ(8, L(5, 3, 1).Offset<1>());
>+  }
>+}
>+
>+TEST(Layout, OffsetByType) {
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, L::Partial().Offset<int32_t>());
>+    EXPECT_EQ(0, L::Partial(3).Offset<int32_t>());
>+    EXPECT_EQ(0, L(3).Offset<int32_t>());
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(0, L::Partial().Offset<int8_t>());
>+    EXPECT_EQ(0, L::Partial(0).Offset<int8_t>());
>+    EXPECT_EQ(0, L::Partial(0).Offset<int32_t>());
>+    EXPECT_EQ(0, L::Partial(1).Offset<int8_t>());
>+    EXPECT_EQ(4, L::Partial(1).Offset<int32_t>());
>+    EXPECT_EQ(0, L::Partial(5).Offset<int8_t>());
>+    EXPECT_EQ(8, L::Partial(5).Offset<int32_t>());
>+    EXPECT_EQ(0, L::Partial(0, 0).Offset<int8_t>());
>+    EXPECT_EQ(0, L::Partial(0, 0).Offset<int32_t>());
>+    EXPECT_EQ(0, L::Partial(0, 0).Offset<Int128>());
>+    EXPECT_EQ(0, L::Partial(1, 0).Offset<int8_t>());
>+    EXPECT_EQ(4, L::Partial(1, 0).Offset<int32_t>());
>+    EXPECT_EQ(8, L::Partial(1, 0).Offset<Int128>());
>+    EXPECT_EQ(0, L::Partial(5, 3).Offset<int8_t>());
>+    EXPECT_EQ(8, L::Partial(5, 3).Offset<int32_t>());
>+    EXPECT_EQ(24, L::Partial(5, 3).Offset<Int128>());
>+    EXPECT_EQ(0, L::Partial(0, 0, 0).Offset<int8_t>());
>+    EXPECT_EQ(0, L::Partial(0, 0, 0).Offset<int32_t>());
>+    EXPECT_EQ(0, L::Partial(0, 0, 0).Offset<Int128>());
>+    EXPECT_EQ(0, L::Partial(1, 0, 0).Offset<int8_t>());
>+    EXPECT_EQ(4, L::Partial(1, 0, 0).Offset<int32_t>());
>+    EXPECT_EQ(8, L::Partial(1, 0, 0).Offset<Int128>());
>+    EXPECT_EQ(0, L::Partial(5, 3, 1).Offset<int8_t>());
>+    EXPECT_EQ(24, L::Partial(5, 3, 1).Offset<Int128>());
>+    EXPECT_EQ(8, L::Partial(5, 3, 1).Offset<int32_t>());
>+    EXPECT_EQ(0, L(5, 3, 1).Offset<int8_t>());
>+    EXPECT_EQ(24, L(5, 3, 1).Offset<Int128>());
>+    EXPECT_EQ(8, L(5, 3, 1).Offset<int32_t>());
>+  }
>+}
>+
>+TEST(Layout, Offsets) {
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_THAT(L::Partial().Offsets(), ElementsAre(0));
>+    EXPECT_THAT(L::Partial(3).Offsets(), ElementsAre(0));
>+    EXPECT_THAT(L(3).Offsets(), ElementsAre(0));
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    EXPECT_THAT(L::Partial().Offsets(), ElementsAre(0));
>+    EXPECT_THAT(L::Partial(3).Offsets(), ElementsAre(0, 12));
>+    EXPECT_THAT(L::Partial(3, 5).Offsets(), ElementsAre(0, 12));
>+    EXPECT_THAT(L(3, 5).Offsets(), ElementsAre(0, 12));
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_THAT(L::Partial().Offsets(), ElementsAre(0));
>+    EXPECT_THAT(L::Partial(1).Offsets(), ElementsAre(0, 4));
>+    EXPECT_THAT(L::Partial(5).Offsets(), ElementsAre(0, 8));
>+    EXPECT_THAT(L::Partial(0, 0).Offsets(), ElementsAre(0, 0, 0));
>+    EXPECT_THAT(L::Partial(1, 0).Offsets(), ElementsAre(0, 4, 8));
>+    EXPECT_THAT(L::Partial(5, 3).Offsets(), ElementsAre(0, 8, 24));
>+    EXPECT_THAT(L::Partial(0, 0, 0).Offsets(), ElementsAre(0, 0, 0));
>+    EXPECT_THAT(L::Partial(1, 0, 0).Offsets(), ElementsAre(0, 4, 8));
>+    EXPECT_THAT(L::Partial(5, 3, 1).Offsets(), ElementsAre(0, 8, 24));
>+    EXPECT_THAT(L(5, 3, 1).Offsets(), ElementsAre(0, 8, 24));
>+  }
>+}
>+
>+TEST(Layout, AllocSize) {
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, L::Partial(0).AllocSize());
>+    EXPECT_EQ(12, L::Partial(3).AllocSize());
>+    EXPECT_EQ(12, L(3).AllocSize());
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    EXPECT_EQ(32, L::Partial(3, 5).AllocSize());
>+    EXPECT_EQ(32, L(3, 5).AllocSize());
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(0, L::Partial(0, 0, 0).AllocSize());
>+    EXPECT_EQ(8, L::Partial(1, 0, 0).AllocSize());
>+    EXPECT_EQ(8, L::Partial(0, 1, 0).AllocSize());
>+    EXPECT_EQ(16, L::Partial(0, 0, 1).AllocSize());
>+    EXPECT_EQ(24, L::Partial(1, 1, 1).AllocSize());
>+    EXPECT_EQ(136, L::Partial(3, 5, 7).AllocSize());
>+    EXPECT_EQ(136, L(3, 5, 7).AllocSize());
>+  }
>+}
>+
>+TEST(Layout, SizeByIndex) {
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, L::Partial(0).Size<0>());
>+    EXPECT_EQ(3, L::Partial(3).Size<0>());
>+    EXPECT_EQ(3, L(3).Size<0>());
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    EXPECT_EQ(0, L::Partial(0).Size<0>());
>+    EXPECT_EQ(3, L::Partial(3).Size<0>());
>+    EXPECT_EQ(3, L::Partial(3, 5).Size<0>());
>+    EXPECT_EQ(5, L::Partial(3, 5).Size<1>());
>+    EXPECT_EQ(3, L(3, 5).Size<0>());
>+    EXPECT_EQ(5, L(3, 5).Size<1>());
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(3, L::Partial(3).Size<0>());
>+    EXPECT_EQ(3, L::Partial(3, 5).Size<0>());
>+    EXPECT_EQ(5, L::Partial(3, 5).Size<1>());
>+    EXPECT_EQ(3, L::Partial(3, 5, 7).Size<0>());
>+    EXPECT_EQ(5, L::Partial(3, 5, 7).Size<1>());
>+    EXPECT_EQ(7, L::Partial(3, 5, 7).Size<2>());
>+    EXPECT_EQ(3, L(3, 5, 7).Size<0>());
>+    EXPECT_EQ(5, L(3, 5, 7).Size<1>());
>+    EXPECT_EQ(7, L(3, 5, 7).Size<2>());
>+  }
>+}
>+
>+TEST(Layout, SizeByType) {
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, L::Partial(0).Size<int32_t>());
>+    EXPECT_EQ(3, L::Partial(3).Size<int32_t>());
>+    EXPECT_EQ(3, L(3).Size<int32_t>());
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(3, L::Partial(3).Size<int8_t>());
>+    EXPECT_EQ(3, L::Partial(3, 5).Size<int8_t>());
>+    EXPECT_EQ(5, L::Partial(3, 5).Size<int32_t>());
>+    EXPECT_EQ(3, L::Partial(3, 5, 7).Size<int8_t>());
>+    EXPECT_EQ(5, L::Partial(3, 5, 7).Size<int32_t>());
>+    EXPECT_EQ(7, L::Partial(3, 5, 7).Size<Int128>());
>+    EXPECT_EQ(3, L(3, 5, 7).Size<int8_t>());
>+    EXPECT_EQ(5, L(3, 5, 7).Size<int32_t>());
>+    EXPECT_EQ(7, L(3, 5, 7).Size<Int128>());
>+  }
>+}
>+
>+TEST(Layout, Sizes) {
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_THAT(L::Partial().Sizes(), ElementsAre());
>+    EXPECT_THAT(L::Partial(3).Sizes(), ElementsAre(3));
>+    EXPECT_THAT(L(3).Sizes(), ElementsAre(3));
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    EXPECT_THAT(L::Partial().Sizes(), ElementsAre());
>+    EXPECT_THAT(L::Partial(3).Sizes(), ElementsAre(3));
>+    EXPECT_THAT(L::Partial(3, 5).Sizes(), ElementsAre(3, 5));
>+    EXPECT_THAT(L(3, 5).Sizes(), ElementsAre(3, 5));
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_THAT(L::Partial().Sizes(), ElementsAre());
>+    EXPECT_THAT(L::Partial(3).Sizes(), ElementsAre(3));
>+    EXPECT_THAT(L::Partial(3, 5).Sizes(), ElementsAre(3, 5));
>+    EXPECT_THAT(L::Partial(3, 5, 7).Sizes(), ElementsAre(3, 5, 7));
>+    EXPECT_THAT(L(3, 5, 7).Sizes(), ElementsAre(3, 5, 7));
>+  }
>+}
>+
>+TEST(Layout, PointerByIndex) {
>+  alignas(max_align_t) const unsigned char p[100] = {};
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, Distance(p, Type<const int32_t*>(L::Partial().Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<const int32_t*>(L::Partial(3).Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<const int32_t*>(L(3).Pointer<0>(p))));
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    EXPECT_EQ(0, Distance(p, Type<const int32_t*>(L::Partial().Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<const int32_t*>(L::Partial(3).Pointer<0>(p))));
>+    EXPECT_EQ(12, Distance(p, Type<const int32_t*>(L::Partial(3).Pointer<1>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const int32_t*>(L::Partial(3, 5).Pointer<0>(p))));
>+    EXPECT_EQ(12,
>+              Distance(p, Type<const int32_t*>(L::Partial(3, 5).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<const int32_t*>(L(3, 5).Pointer<0>(p))));
>+    EXPECT_EQ(12, Distance(p, Type<const int32_t*>(L(3, 5).Pointer<1>(p))));
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(0, Distance(p, Type<const int8_t*>(L::Partial().Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<const int8_t*>(L::Partial(0).Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<const int32_t*>(L::Partial(0).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<const int8_t*>(L::Partial(1).Pointer<0>(p))));
>+    EXPECT_EQ(4, Distance(p, Type<const int32_t*>(L::Partial(1).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<const int8_t*>(L::Partial(5).Pointer<0>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<const int32_t*>(L::Partial(5).Pointer<1>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const int8_t*>(L::Partial(0, 0).Pointer<0>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const int32_t*>(L::Partial(0, 0).Pointer<1>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const Int128*>(L::Partial(0, 0).Pointer<2>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const int8_t*>(L::Partial(1, 0).Pointer<0>(p))));
>+    EXPECT_EQ(4,
>+              Distance(p, Type<const int32_t*>(L::Partial(1, 0).Pointer<1>(p))));
>+    EXPECT_EQ(8,
>+              Distance(p, Type<const Int128*>(L::Partial(1, 0).Pointer<2>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const int8_t*>(L::Partial(5, 3).Pointer<0>(p))));
>+    EXPECT_EQ(8,
>+              Distance(p, Type<const int32_t*>(L::Partial(5, 3).Pointer<1>(p))));
>+    EXPECT_EQ(24,
>+              Distance(p, Type<const Int128*>(L::Partial(5, 3).Pointer<2>(p))));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<const int8_t*>(L::Partial(0, 0, 0).Pointer<0>(p))));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<const int32_t*>(L::Partial(0, 0, 0).Pointer<1>(p))));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<const Int128*>(L::Partial(0, 0, 0).Pointer<2>(p))));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<const int8_t*>(L::Partial(1, 0, 0).Pointer<0>(p))));
>+    EXPECT_EQ(
>+        4, Distance(p, Type<const int32_t*>(L::Partial(1, 0, 0).Pointer<1>(p))));
>+    EXPECT_EQ(
>+        8, Distance(p, Type<const Int128*>(L::Partial(1, 0, 0).Pointer<2>(p))));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<const int8_t*>(L::Partial(5, 3, 1).Pointer<0>(p))));
>+    EXPECT_EQ(
>+        24,
>+        Distance(p, Type<const Int128*>(L::Partial(5, 3, 1).Pointer<2>(p))));
>+    EXPECT_EQ(
>+        8, Distance(p, Type<const int32_t*>(L::Partial(5, 3, 1).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<const int8_t*>(L(5, 3, 1).Pointer<0>(p))));
>+    EXPECT_EQ(24, Distance(p, Type<const Int128*>(L(5, 3, 1).Pointer<2>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<const int32_t*>(L(5, 3, 1).Pointer<1>(p))));
>+  }
>+}
>+
>+TEST(Layout, PointerByType) {
>+  alignas(max_align_t) const unsigned char p[100] = {};
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const int32_t*>(L::Partial().Pointer<int32_t>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const int32_t*>(L::Partial(3).Pointer<int32_t>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<const int32_t*>(L(3).Pointer<int32_t>(p))));
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(0, Distance(p, Type<const int8_t*>(L::Partial().Pointer<int8_t>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const int8_t*>(L::Partial(0).Pointer<int8_t>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const int32_t*>(L::Partial(0).Pointer<int32_t>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const int8_t*>(L::Partial(1).Pointer<int8_t>(p))));
>+    EXPECT_EQ(4,
>+              Distance(p, Type<const int32_t*>(L::Partial(1).Pointer<int32_t>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<const int8_t*>(L::Partial(5).Pointer<int8_t>(p))));
>+    EXPECT_EQ(8,
>+              Distance(p, Type<const int32_t*>(L::Partial(5).Pointer<int32_t>(p))));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<const int8_t*>(L::Partial(0, 0).Pointer<int8_t>(p))));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<const int32_t*>(L::Partial(0, 0).Pointer<int32_t>(p))));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<const Int128*>(L::Partial(0, 0).Pointer<Int128>(p))));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<const int8_t*>(L::Partial(1, 0).Pointer<int8_t>(p))));
>+    EXPECT_EQ(
>+        4, Distance(p, Type<const int32_t*>(L::Partial(1, 0).Pointer<int32_t>(p))));
>+    EXPECT_EQ(
>+        8,
>+        Distance(p, Type<const Int128*>(L::Partial(1, 0).Pointer<Int128>(p))));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<const int8_t*>(L::Partial(5, 3).Pointer<int8_t>(p))));
>+    EXPECT_EQ(
>+        8, Distance(p, Type<const int32_t*>(L::Partial(5, 3).Pointer<int32_t>(p))));
>+    EXPECT_EQ(
>+        24,
>+        Distance(p, Type<const Int128*>(L::Partial(5, 3).Pointer<Int128>(p))));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<const int8_t*>(L::Partial(0, 0, 0).Pointer<int8_t>(p))));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<const int32_t*>(L::Partial(0, 0, 0).Pointer<int32_t>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<const Int128*>(
>+                                 L::Partial(0, 0, 0).Pointer<Int128>(p))));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<const int8_t*>(L::Partial(1, 0, 0).Pointer<int8_t>(p))));
>+    EXPECT_EQ(
>+        4,
>+        Distance(p, Type<const int32_t*>(L::Partial(1, 0, 0).Pointer<int32_t>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<const Int128*>(
>+                                 L::Partial(1, 0, 0).Pointer<Int128>(p))));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<const int8_t*>(L::Partial(5, 3, 1).Pointer<int8_t>(p))));
>+    EXPECT_EQ(24, Distance(p, Type<const Int128*>(
>+                                  L::Partial(5, 3, 1).Pointer<Int128>(p))));
>+    EXPECT_EQ(
>+        8,
>+        Distance(p, Type<const int32_t*>(L::Partial(5, 3, 1).Pointer<int32_t>(p))));
>+    EXPECT_EQ(24,
>+              Distance(p, Type<const Int128*>(L(5, 3, 1).Pointer<Int128>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<const int32_t*>(L(5, 3, 1).Pointer<int32_t>(p))));
>+  }
>+}
>+
>+TEST(Layout, MutablePointerByIndex) {
>+  alignas(max_align_t) unsigned char p[100];
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial().Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial(3).Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L(3).Pointer<0>(p))));
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial().Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial(3).Pointer<0>(p))));
>+    EXPECT_EQ(12, Distance(p, Type<int32_t*>(L::Partial(3).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial(3, 5).Pointer<0>(p))));
>+    EXPECT_EQ(12, Distance(p, Type<int32_t*>(L::Partial(3, 5).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L(3, 5).Pointer<0>(p))));
>+    EXPECT_EQ(12, Distance(p, Type<int32_t*>(L(3, 5).Pointer<1>(p))));
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial().Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(0).Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial(0).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(1).Pointer<0>(p))));
>+    EXPECT_EQ(4, Distance(p, Type<int32_t*>(L::Partial(1).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(5).Pointer<0>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<int32_t*>(L::Partial(5).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(0, 0).Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial(0, 0).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<Int128*>(L::Partial(0, 0).Pointer<2>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(1, 0).Pointer<0>(p))));
>+    EXPECT_EQ(4, Distance(p, Type<int32_t*>(L::Partial(1, 0).Pointer<1>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<Int128*>(L::Partial(1, 0).Pointer<2>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(5, 3).Pointer<0>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<int32_t*>(L::Partial(5, 3).Pointer<1>(p))));
>+    EXPECT_EQ(24, Distance(p, Type<Int128*>(L::Partial(5, 3).Pointer<2>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(0, 0, 0).Pointer<0>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial(0, 0, 0).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<Int128*>(L::Partial(0, 0, 0).Pointer<2>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(1, 0, 0).Pointer<0>(p))));
>+    EXPECT_EQ(4, Distance(p, Type<int32_t*>(L::Partial(1, 0, 0).Pointer<1>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<Int128*>(L::Partial(1, 0, 0).Pointer<2>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(5, 3, 1).Pointer<0>(p))));
>+    EXPECT_EQ(24,
>+              Distance(p, Type<Int128*>(L::Partial(5, 3, 1).Pointer<2>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<int32_t*>(L::Partial(5, 3, 1).Pointer<1>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L(5, 3, 1).Pointer<0>(p))));
>+    EXPECT_EQ(24, Distance(p, Type<Int128*>(L(5, 3, 1).Pointer<2>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<int32_t*>(L(5, 3, 1).Pointer<1>(p))));
>+  }
>+}
>+
>+TEST(Layout, MutablePointerByType) {
>+  alignas(max_align_t) unsigned char p[100];
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial().Pointer<int32_t>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial(3).Pointer<int32_t>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L(3).Pointer<int32_t>(p))));
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial().Pointer<int8_t>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(0).Pointer<int8_t>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial(0).Pointer<int32_t>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(1).Pointer<int8_t>(p))));
>+    EXPECT_EQ(4, Distance(p, Type<int32_t*>(L::Partial(1).Pointer<int32_t>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(5).Pointer<int8_t>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<int32_t*>(L::Partial(5).Pointer<int32_t>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(0, 0).Pointer<int8_t>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int32_t*>(L::Partial(0, 0).Pointer<int32_t>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<Int128*>(L::Partial(0, 0).Pointer<Int128>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(1, 0).Pointer<int8_t>(p))));
>+    EXPECT_EQ(4, Distance(p, Type<int32_t*>(L::Partial(1, 0).Pointer<int32_t>(p))));
>+    EXPECT_EQ(8,
>+              Distance(p, Type<Int128*>(L::Partial(1, 0).Pointer<Int128>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L::Partial(5, 3).Pointer<int8_t>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<int32_t*>(L::Partial(5, 3).Pointer<int32_t>(p))));
>+    EXPECT_EQ(24,
>+              Distance(p, Type<Int128*>(L::Partial(5, 3).Pointer<Int128>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<int8_t*>(L::Partial(0, 0, 0).Pointer<int8_t>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<int32_t*>(L::Partial(0, 0, 0).Pointer<int32_t>(p))));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<Int128*>(L::Partial(0, 0, 0).Pointer<Int128>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<int8_t*>(L::Partial(1, 0, 0).Pointer<int8_t>(p))));
>+    EXPECT_EQ(4,
>+              Distance(p, Type<int32_t*>(L::Partial(1, 0, 0).Pointer<int32_t>(p))));
>+    EXPECT_EQ(
>+        8, Distance(p, Type<Int128*>(L::Partial(1, 0, 0).Pointer<Int128>(p))));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<int8_t*>(L::Partial(5, 3, 1).Pointer<int8_t>(p))));
>+    EXPECT_EQ(
>+        24, Distance(p, Type<Int128*>(L::Partial(5, 3, 1).Pointer<Int128>(p))));
>+    EXPECT_EQ(8,
>+              Distance(p, Type<int32_t*>(L::Partial(5, 3, 1).Pointer<int32_t>(p))));
>+    EXPECT_EQ(0, Distance(p, Type<int8_t*>(L(5, 3, 1).Pointer<int8_t>(p))));
>+    EXPECT_EQ(24, Distance(p, Type<Int128*>(L(5, 3, 1).Pointer<Int128>(p))));
>+    EXPECT_EQ(8, Distance(p, Type<int32_t*>(L(5, 3, 1).Pointer<int32_t>(p))));
>+  }
>+}
>+
>+TEST(Layout, Pointers) {
>+  alignas(max_align_t) const unsigned char p[100] = {};
>+  using L = Layout<int8_t, int8_t, Int128>;
>+  {
>+    const auto x = L::Partial();
>+    EXPECT_EQ(std::make_tuple(x.Pointer<0>(p)),
>+              Type<std::tuple<const int8_t*>>(x.Pointers(p)));
>+  }
>+  {
>+    const auto x = L::Partial(1);
>+    EXPECT_EQ(std::make_tuple(x.Pointer<0>(p), x.Pointer<1>(p)),
>+              (Type<std::tuple<const int8_t*, const int8_t*>>(x.Pointers(p))));
>+  }
>+  {
>+    const auto x = L::Partial(1, 2);
>+    EXPECT_EQ(
>+        std::make_tuple(x.Pointer<0>(p), x.Pointer<1>(p), x.Pointer<2>(p)),
>+        (Type<std::tuple<const int8_t*, const int8_t*, const Int128*>>(
>+            x.Pointers(p))));
>+  }
>+  {
>+    const auto x = L::Partial(1, 2, 3);
>+    EXPECT_EQ(
>+        std::make_tuple(x.Pointer<0>(p), x.Pointer<1>(p), x.Pointer<2>(p)),
>+        (Type<std::tuple<const int8_t*, const int8_t*, const Int128*>>(
>+            x.Pointers(p))));
>+  }
>+  {
>+    const L x(1, 2, 3);
>+    EXPECT_EQ(
>+        std::make_tuple(x.Pointer<0>(p), x.Pointer<1>(p), x.Pointer<2>(p)),
>+        (Type<std::tuple<const int8_t*, const int8_t*, const Int128*>>(
>+            x.Pointers(p))));
>+  }
>+}
>+
>+TEST(Layout, MutablePointers) {
>+  alignas(max_align_t) unsigned char p[100];
>+  using L = Layout<int8_t, int8_t, Int128>;
>+  {
>+    const auto x = L::Partial();
>+    EXPECT_EQ(std::make_tuple(x.Pointer<0>(p)),
>+              Type<std::tuple<int8_t*>>(x.Pointers(p)));
>+  }
>+  {
>+    const auto x = L::Partial(1);
>+    EXPECT_EQ(std::make_tuple(x.Pointer<0>(p), x.Pointer<1>(p)),
>+              (Type<std::tuple<int8_t*, int8_t*>>(x.Pointers(p))));
>+  }
>+  {
>+    const auto x = L::Partial(1, 2);
>+    EXPECT_EQ(
>+        std::make_tuple(x.Pointer<0>(p), x.Pointer<1>(p), x.Pointer<2>(p)),
>+        (Type<std::tuple<int8_t*, int8_t*, Int128*>>(x.Pointers(p))));
>+  }
>+  {
>+    const auto x = L::Partial(1, 2, 3);
>+    EXPECT_EQ(
>+        std::make_tuple(x.Pointer<0>(p), x.Pointer<1>(p), x.Pointer<2>(p)),
>+        (Type<std::tuple<int8_t*, int8_t*, Int128*>>(x.Pointers(p))));
>+  }
>+  {
>+    const L x(1, 2, 3);
>+    EXPECT_EQ(
>+        std::make_tuple(x.Pointer<0>(p), x.Pointer<1>(p), x.Pointer<2>(p)),
>+        (Type<std::tuple<int8_t*, int8_t*, Int128*>>(x.Pointers(p))));
>+  }
>+}
>+
>+TEST(Layout, SliceByIndexSize) {
>+  alignas(max_align_t) const unsigned char p[100] = {};
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, L::Partial(0).Slice<0>(p).size());
>+    EXPECT_EQ(3, L::Partial(3).Slice<0>(p).size());
>+    EXPECT_EQ(3, L(3).Slice<0>(p).size());
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    EXPECT_EQ(3, L::Partial(3).Slice<0>(p).size());
>+    EXPECT_EQ(5, L::Partial(3, 5).Slice<1>(p).size());
>+    EXPECT_EQ(5, L(3, 5).Slice<1>(p).size());
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(3, L::Partial(3).Slice<0>(p).size());
>+    EXPECT_EQ(3, L::Partial(3, 5).Slice<0>(p).size());
>+    EXPECT_EQ(5, L::Partial(3, 5).Slice<1>(p).size());
>+    EXPECT_EQ(3, L::Partial(3, 5, 7).Slice<0>(p).size());
>+    EXPECT_EQ(5, L::Partial(3, 5, 7).Slice<1>(p).size());
>+    EXPECT_EQ(7, L::Partial(3, 5, 7).Slice<2>(p).size());
>+    EXPECT_EQ(3, L(3, 5, 7).Slice<0>(p).size());
>+    EXPECT_EQ(5, L(3, 5, 7).Slice<1>(p).size());
>+    EXPECT_EQ(7, L(3, 5, 7).Slice<2>(p).size());
>+  }
>+}
>+
>+TEST(Layout, SliceByTypeSize) {
>+  alignas(max_align_t) const unsigned char p[100] = {};
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, L::Partial(0).Slice<int32_t>(p).size());
>+    EXPECT_EQ(3, L::Partial(3).Slice<int32_t>(p).size());
>+    EXPECT_EQ(3, L(3).Slice<int32_t>(p).size());
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(3, L::Partial(3).Slice<int8_t>(p).size());
>+    EXPECT_EQ(3, L::Partial(3, 5).Slice<int8_t>(p).size());
>+    EXPECT_EQ(5, L::Partial(3, 5).Slice<int32_t>(p).size());
>+    EXPECT_EQ(3, L::Partial(3, 5, 7).Slice<int8_t>(p).size());
>+    EXPECT_EQ(5, L::Partial(3, 5, 7).Slice<int32_t>(p).size());
>+    EXPECT_EQ(7, L::Partial(3, 5, 7).Slice<Int128>(p).size());
>+    EXPECT_EQ(3, L(3, 5, 7).Slice<int8_t>(p).size());
>+    EXPECT_EQ(5, L(3, 5, 7).Slice<int32_t>(p).size());
>+    EXPECT_EQ(7, L(3, 5, 7).Slice<Int128>(p).size());
>+  }
>+}
>+
>+TEST(Layout, MutableSliceByIndexSize) {
>+  alignas(max_align_t) unsigned char p[100];
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, L::Partial(0).Slice<0>(p).size());
>+    EXPECT_EQ(3, L::Partial(3).Slice<0>(p).size());
>+    EXPECT_EQ(3, L(3).Slice<0>(p).size());
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    EXPECT_EQ(3, L::Partial(3).Slice<0>(p).size());
>+    EXPECT_EQ(5, L::Partial(3, 5).Slice<1>(p).size());
>+    EXPECT_EQ(5, L(3, 5).Slice<1>(p).size());
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(3, L::Partial(3).Slice<0>(p).size());
>+    EXPECT_EQ(3, L::Partial(3, 5).Slice<0>(p).size());
>+    EXPECT_EQ(5, L::Partial(3, 5).Slice<1>(p).size());
>+    EXPECT_EQ(3, L::Partial(3, 5, 7).Slice<0>(p).size());
>+    EXPECT_EQ(5, L::Partial(3, 5, 7).Slice<1>(p).size());
>+    EXPECT_EQ(7, L::Partial(3, 5, 7).Slice<2>(p).size());
>+    EXPECT_EQ(3, L(3, 5, 7).Slice<0>(p).size());
>+    EXPECT_EQ(5, L(3, 5, 7).Slice<1>(p).size());
>+    EXPECT_EQ(7, L(3, 5, 7).Slice<2>(p).size());
>+  }
>+}
>+
>+TEST(Layout, MutableSliceByTypeSize) {
>+  alignas(max_align_t) unsigned char p[100];
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0, L::Partial(0).Slice<int32_t>(p).size());
>+    EXPECT_EQ(3, L::Partial(3).Slice<int32_t>(p).size());
>+    EXPECT_EQ(3, L(3).Slice<int32_t>(p).size());
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(3, L::Partial(3).Slice<int8_t>(p).size());
>+    EXPECT_EQ(3, L::Partial(3, 5).Slice<int8_t>(p).size());
>+    EXPECT_EQ(5, L::Partial(3, 5).Slice<int32_t>(p).size());
>+    EXPECT_EQ(3, L::Partial(3, 5, 7).Slice<int8_t>(p).size());
>+    EXPECT_EQ(5, L::Partial(3, 5, 7).Slice<int32_t>(p).size());
>+    EXPECT_EQ(7, L::Partial(3, 5, 7).Slice<Int128>(p).size());
>+    EXPECT_EQ(3, L(3, 5, 7).Slice<int8_t>(p).size());
>+    EXPECT_EQ(5, L(3, 5, 7).Slice<int32_t>(p).size());
>+    EXPECT_EQ(7, L(3, 5, 7).Slice<Int128>(p).size());
>+  }
>+}
>+
>+TEST(Layout, SliceByIndexData) {
>+  alignas(max_align_t) const unsigned char p[100] = {};
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<const int32_t>>(L::Partial(0).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<const int32_t>>(L::Partial(3).Slice<0>(p)).data()));
>+    EXPECT_EQ(0, Distance(p, Type<Span<const int32_t>>(L(3).Slice<0>(p)).data()));
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<const int32_t>>(L::Partial(3).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p,
>+                 Type<Span<const int32_t>>(L::Partial(3, 5).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        12,
>+        Distance(p,
>+                 Type<Span<const int32_t>>(L::Partial(3, 5).Slice<1>(p)).data()));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<Span<const int32_t>>(L(3, 5).Slice<0>(p)).data()));
>+    EXPECT_EQ(12,
>+              Distance(p, Type<Span<const int32_t>>(L(3, 5).Slice<1>(p)).data()));
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<const int8_t>>(L::Partial(0).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<const int8_t>>(L::Partial(1).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<const int8_t>>(L::Partial(5).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(
>+               p, Type<Span<const int8_t>>(L::Partial(0, 0).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p,
>+                 Type<Span<const int32_t>>(L::Partial(0, 0).Slice<1>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(
>+               p, Type<Span<const int8_t>>(L::Partial(1, 0).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        4,
>+        Distance(p,
>+                 Type<Span<const int32_t>>(L::Partial(1, 0).Slice<1>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(
>+               p, Type<Span<const int8_t>>(L::Partial(5, 3).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        8,
>+        Distance(p,
>+                 Type<Span<const int32_t>>(L::Partial(5, 3).Slice<1>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p, Type<Span<const int8_t>>(L::Partial(0, 0, 0).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p,
>+            Type<Span<const int32_t>>(L::Partial(0, 0, 0).Slice<1>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p,
>+            Type<Span<const Int128>>(L::Partial(0, 0, 0).Slice<2>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p, Type<Span<const int8_t>>(L::Partial(1, 0, 0).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        4,
>+        Distance(
>+            p,
>+            Type<Span<const int32_t>>(L::Partial(1, 0, 0).Slice<1>(p)).data()));
>+    EXPECT_EQ(
>+        8,
>+        Distance(
>+            p,
>+            Type<Span<const Int128>>(L::Partial(1, 0, 0).Slice<2>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p, Type<Span<const int8_t>>(L::Partial(5, 3, 1).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        24,
>+        Distance(
>+            p,
>+            Type<Span<const Int128>>(L::Partial(5, 3, 1).Slice<2>(p)).data()));
>+    EXPECT_EQ(
>+        8,
>+        Distance(
>+            p,
>+            Type<Span<const int32_t>>(L::Partial(5, 3, 1).Slice<1>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<Span<const int8_t>>(L(5, 3, 1).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        24,
>+        Distance(p, Type<Span<const Int128>>(L(5, 3, 1).Slice<2>(p)).data()));
>+    EXPECT_EQ(
>+        8, Distance(p, Type<Span<const int32_t>>(L(5, 3, 1).Slice<1>(p)).data()));
>+  }
>+}
>+
>+TEST(Layout, SliceByTypeData) {
>+  alignas(max_align_t) const unsigned char p[100] = {};
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p, Type<Span<const int32_t>>(L::Partial(0).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p, Type<Span<const int32_t>>(L::Partial(3).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<Span<const int32_t>>(L(3).Slice<int32_t>(p)).data()));
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(
>+        0, Distance(
>+               p, Type<Span<const int8_t>>(L::Partial(0).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(
>+               p, Type<Span<const int8_t>>(L::Partial(1).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(
>+               p, Type<Span<const int8_t>>(L::Partial(5).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p, Type<Span<const int8_t>>(L::Partial(0, 0).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p,
>+            Type<Span<const int32_t>>(L::Partial(0, 0).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p, Type<Span<const int8_t>>(L::Partial(1, 0).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        4,
>+        Distance(
>+            p,
>+            Type<Span<const int32_t>>(L::Partial(1, 0).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p, Type<Span<const int8_t>>(L::Partial(5, 3).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        8,
>+        Distance(
>+            p,
>+            Type<Span<const int32_t>>(L::Partial(5, 3).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p,
>+            Type<Span<const int8_t>>(L::Partial(0, 0, 0).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<const int32_t>>(L::Partial(0, 0, 0).Slice<int32_t>(p))
>+                        .data()));
>+    EXPECT_EQ(0, Distance(p, Type<Span<const Int128>>(
>+                                 L::Partial(0, 0, 0).Slice<Int128>(p))
>+                                 .data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p,
>+            Type<Span<const int8_t>>(L::Partial(1, 0, 0).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        4,
>+        Distance(p, Type<Span<const int32_t>>(L::Partial(1, 0, 0).Slice<int32_t>(p))
>+                        .data()));
>+    EXPECT_EQ(8, Distance(p, Type<Span<const Int128>>(
>+                                 L::Partial(1, 0, 0).Slice<Int128>(p))
>+                                 .data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p,
>+            Type<Span<const int8_t>>(L::Partial(5, 3, 1).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(24, Distance(p, Type<Span<const Int128>>(
>+                                  L::Partial(5, 3, 1).Slice<Int128>(p))
>+                                  .data()));
>+    EXPECT_EQ(
>+        8,
>+        Distance(p, Type<Span<const int32_t>>(L::Partial(5, 3, 1).Slice<int32_t>(p))
>+                        .data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<const int8_t>>(L(5, 3, 1).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        24,
>+        Distance(p,
>+                 Type<Span<const Int128>>(L(5, 3, 1).Slice<Int128>(p)).data()));
>+    EXPECT_EQ(
>+        8, Distance(
>+               p, Type<Span<const int32_t>>(L(5, 3, 1).Slice<int32_t>(p)).data()));
>+  }
>+}
>+
>+TEST(Layout, MutableSliceByIndexData) {
>+  alignas(max_align_t) unsigned char p[100];
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(0,
>+              Distance(p, Type<Span<int32_t>>(L::Partial(0).Slice<0>(p)).data()));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<Span<int32_t>>(L::Partial(3).Slice<0>(p)).data()));
>+    EXPECT_EQ(0, Distance(p, Type<Span<int32_t>>(L(3).Slice<0>(p)).data()));
>+  }
>+  {
>+    using L = Layout<int32_t, int32_t>;
>+    EXPECT_EQ(0,
>+              Distance(p, Type<Span<int32_t>>(L::Partial(3).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<Span<int32_t>>(L::Partial(3, 5).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        12,
>+        Distance(p, Type<Span<int32_t>>(L::Partial(3, 5).Slice<1>(p)).data()));
>+    EXPECT_EQ(0, Distance(p, Type<Span<int32_t>>(L(3, 5).Slice<0>(p)).data()));
>+    EXPECT_EQ(12, Distance(p, Type<Span<int32_t>>(L(3, 5).Slice<1>(p)).data()));
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(0,
>+              Distance(p, Type<Span<int8_t>>(L::Partial(0).Slice<0>(p)).data()));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<Span<int8_t>>(L::Partial(1).Slice<0>(p)).data()));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<Span<int8_t>>(L::Partial(5).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<Span<int8_t>>(L::Partial(0, 0).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<Span<int32_t>>(L::Partial(0, 0).Slice<1>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<Span<int8_t>>(L::Partial(1, 0).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        4, Distance(p, Type<Span<int32_t>>(L::Partial(1, 0).Slice<1>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<Span<int8_t>>(L::Partial(5, 3).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        8, Distance(p, Type<Span<int32_t>>(L::Partial(5, 3).Slice<1>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<int8_t>>(L::Partial(0, 0, 0).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<int32_t>>(L::Partial(0, 0, 0).Slice<1>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(
>+               p, Type<Span<Int128>>(L::Partial(0, 0, 0).Slice<2>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<int8_t>>(L::Partial(1, 0, 0).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        4,
>+        Distance(p, Type<Span<int32_t>>(L::Partial(1, 0, 0).Slice<1>(p)).data()));
>+    EXPECT_EQ(
>+        8, Distance(
>+               p, Type<Span<Int128>>(L::Partial(1, 0, 0).Slice<2>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<int8_t>>(L::Partial(5, 3, 1).Slice<0>(p)).data()));
>+    EXPECT_EQ(
>+        24, Distance(
>+                p, Type<Span<Int128>>(L::Partial(5, 3, 1).Slice<2>(p)).data()));
>+    EXPECT_EQ(
>+        8,
>+        Distance(p, Type<Span<int32_t>>(L::Partial(5, 3, 1).Slice<1>(p)).data()));
>+    EXPECT_EQ(0, Distance(p, Type<Span<int8_t>>(L(5, 3, 1).Slice<0>(p)).data()));
>+    EXPECT_EQ(24,
>+              Distance(p, Type<Span<Int128>>(L(5, 3, 1).Slice<2>(p)).data()));
>+    EXPECT_EQ(8, Distance(p, Type<Span<int32_t>>(L(5, 3, 1).Slice<1>(p)).data()));
>+  }
>+}
>+
>+TEST(Layout, MutableSliceByTypeData) {
>+  alignas(max_align_t) unsigned char p[100];
>+  {
>+    using L = Layout<int32_t>;
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<int32_t>>(L::Partial(0).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<int32_t>>(L::Partial(3).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(0, Distance(p, Type<Span<int32_t>>(L(3).Slice<int32_t>(p)).data()));
>+  }
>+  {
>+    using L = Layout<int8_t, int32_t, Int128>;
>+    EXPECT_EQ(
>+        0, Distance(p, Type<Span<int8_t>>(L::Partial(0).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<Span<int8_t>>(L::Partial(1).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(p, Type<Span<int8_t>>(L::Partial(5).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<int8_t>>(L::Partial(0, 0).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(
>+               p, Type<Span<int32_t>>(L::Partial(0, 0).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<int8_t>>(L::Partial(1, 0).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        4, Distance(
>+               p, Type<Span<int32_t>>(L::Partial(1, 0).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(p, Type<Span<int8_t>>(L::Partial(5, 3).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        8, Distance(
>+               p, Type<Span<int32_t>>(L::Partial(5, 3).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(
>+               p, Type<Span<int8_t>>(L::Partial(0, 0, 0).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p, Type<Span<int32_t>>(L::Partial(0, 0, 0).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(
>+        0,
>+        Distance(
>+            p,
>+            Type<Span<Int128>>(L::Partial(0, 0, 0).Slice<Int128>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(
>+               p, Type<Span<int8_t>>(L::Partial(1, 0, 0).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        4,
>+        Distance(
>+            p, Type<Span<int32_t>>(L::Partial(1, 0, 0).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(
>+        8,
>+        Distance(
>+            p,
>+            Type<Span<Int128>>(L::Partial(1, 0, 0).Slice<Int128>(p)).data()));
>+    EXPECT_EQ(
>+        0, Distance(
>+               p, Type<Span<int8_t>>(L::Partial(5, 3, 1).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        24,
>+        Distance(
>+            p,
>+            Type<Span<Int128>>(L::Partial(5, 3, 1).Slice<Int128>(p)).data()));
>+    EXPECT_EQ(
>+        8,
>+        Distance(
>+            p, Type<Span<int32_t>>(L::Partial(5, 3, 1).Slice<int32_t>(p)).data()));
>+    EXPECT_EQ(0,
>+              Distance(p, Type<Span<int8_t>>(L(5, 3, 1).Slice<int8_t>(p)).data()));
>+    EXPECT_EQ(
>+        24,
>+        Distance(p, Type<Span<Int128>>(L(5, 3, 1).Slice<Int128>(p)).data()));
>+    EXPECT_EQ(
>+        8, Distance(p, Type<Span<int32_t>>(L(5, 3, 1).Slice<int32_t>(p)).data()));
>+  }
>+}
>+
>+MATCHER_P(IsSameSlice, slice, "") {
>+  return arg.size() == slice.size() && arg.data() == slice.data();
>+}
>+
>+template <typename... M>
>+class TupleMatcher {
>+ public:
>+  explicit TupleMatcher(M... matchers) : matchers_(std::move(matchers)...) {}
>+
>+  template <typename Tuple>
>+  bool MatchAndExplain(const Tuple& p,
>+                       testing::MatchResultListener* /* listener */) const {
>+    static_assert(std::tuple_size<Tuple>::value == sizeof...(M), "");
>+    return MatchAndExplainImpl(
>+        p, absl::make_index_sequence<std::tuple_size<Tuple>::value>{});
>+  }
>+
>+  // For the matcher concept. Left empty as we don't really need the diagnostics
>+  // right now.
>+  void DescribeTo(::std::ostream* os) const {}
>+  void DescribeNegationTo(::std::ostream* os) const {}
>+
>+ private:
>+  template <typename Tuple, size_t... Is>
>+  bool MatchAndExplainImpl(const Tuple& p, absl::index_sequence<Is...>) const {
>+    // Using std::min as a simple variadic "and".
>+    return std::min(
>+        {true, testing::SafeMatcherCast<
>+                   const typename std::tuple_element<Is, Tuple>::type&>(
>+                   std::get<Is>(matchers_))
>+                   .Matches(std::get<Is>(p))...});
>+  }
>+
>+  std::tuple<M...> matchers_;
>+};
>+
>+template <typename... M>
>+testing::PolymorphicMatcher<TupleMatcher<M...>> Tuple(M... matchers) {
>+  return testing::MakePolymorphicMatcher(
>+      TupleMatcher<M...>(std::move(matchers)...));
>+}
>+
>+TEST(Layout, Slices) {
>+  alignas(max_align_t) const unsigned char p[100] = {};
>+  using L = Layout<int8_t, int8_t, Int128>;
>+  {
>+    const auto x = L::Partial();
>+    EXPECT_THAT(Type<std::tuple<>>(x.Slices(p)), Tuple());
>+  }
>+  {
>+    const auto x = L::Partial(1);
>+    EXPECT_THAT(Type<std::tuple<Span<const int8_t>>>(x.Slices(p)),
>+                Tuple(IsSameSlice(x.Slice<0>(p))));
>+  }
>+  {
>+    const auto x = L::Partial(1, 2);
>+    EXPECT_THAT(
>+        (Type<std::tuple<Span<const int8_t>, Span<const int8_t>>>(x.Slices(p))),
>+        Tuple(IsSameSlice(x.Slice<0>(p)), IsSameSlice(x.Slice<1>(p))));
>+  }
>+  {
>+    const auto x = L::Partial(1, 2, 3);
>+    EXPECT_THAT((Type<std::tuple<Span<const int8_t>, Span<const int8_t>,
>+                                 Span<const Int128>>>(x.Slices(p))),
>+                Tuple(IsSameSlice(x.Slice<0>(p)), IsSameSlice(x.Slice<1>(p)),
>+                      IsSameSlice(x.Slice<2>(p))));
>+  }
>+  {
>+    const L x(1, 2, 3);
>+    EXPECT_THAT((Type<std::tuple<Span<const int8_t>, Span<const int8_t>,
>+                                 Span<const Int128>>>(x.Slices(p))),
>+                Tuple(IsSameSlice(x.Slice<0>(p)), IsSameSlice(x.Slice<1>(p)),
>+                      IsSameSlice(x.Slice<2>(p))));
>+  }
>+}
>+
>+TEST(Layout, MutableSlices) {
>+  alignas(max_align_t) unsigned char p[100] = {};
>+  using L = Layout<int8_t, int8_t, Int128>;
>+  {
>+    const auto x = L::Partial();
>+    EXPECT_THAT(Type<std::tuple<>>(x.Slices(p)), Tuple());
>+  }
>+  {
>+    const auto x = L::Partial(1);
>+    EXPECT_THAT(Type<std::tuple<Span<int8_t>>>(x.Slices(p)),
>+                Tuple(IsSameSlice(x.Slice<0>(p))));
>+  }
>+  {
>+    const auto x = L::Partial(1, 2);
>+    EXPECT_THAT((Type<std::tuple<Span<int8_t>, Span<int8_t>>>(x.Slices(p))),
>+                Tuple(IsSameSlice(x.Slice<0>(p)), IsSameSlice(x.Slice<1>(p))));
>+  }
>+  {
>+    const auto x = L::Partial(1, 2, 3);
>+    EXPECT_THAT(
>+        (Type<std::tuple<Span<int8_t>, Span<int8_t>, Span<Int128>>>(x.Slices(p))),
>+        Tuple(IsSameSlice(x.Slice<0>(p)), IsSameSlice(x.Slice<1>(p)),
>+              IsSameSlice(x.Slice<2>(p))));
>+  }
>+  {
>+    const L x(1, 2, 3);
>+    EXPECT_THAT(
>+        (Type<std::tuple<Span<int8_t>, Span<int8_t>, Span<Int128>>>(x.Slices(p))),
>+        Tuple(IsSameSlice(x.Slice<0>(p)), IsSameSlice(x.Slice<1>(p)),
>+              IsSameSlice(x.Slice<2>(p))));
>+  }
>+}
>+
>+TEST(Layout, UnalignedTypes) {
>+  constexpr Layout<unsigned char, unsigned char, unsigned char> x(1, 2, 3);
>+  alignas(max_align_t) unsigned char p[x.AllocSize() + 1];
>+  EXPECT_THAT(x.Pointers(p + 1), Tuple(p + 1, p + 2, p + 4));
>+}
>+
>+TEST(Layout, CustomAlignment) {
>+  constexpr Layout<unsigned char, Aligned<unsigned char, 8>> x(1, 2);
>+  alignas(max_align_t) unsigned char p[x.AllocSize()];
>+  EXPECT_EQ(10, x.AllocSize());
>+  EXPECT_THAT(x.Pointers(p), Tuple(p + 0, p + 8));
>+}
>+
>+TEST(Layout, OverAligned) {
>+  constexpr size_t M = alignof(max_align_t);
>+  constexpr Layout<unsigned char, Aligned<unsigned char, 2 * M>> x(1, 3);
>+  alignas(2 * M) unsigned char p[x.AllocSize()];
>+  EXPECT_EQ(2 * M + 3, x.AllocSize());
>+  EXPECT_THAT(x.Pointers(p), Tuple(p + 0, p + 2 * M));
>+}
>+
>+TEST(Layout, Alignment) {
>+  static_assert(Layout<int8_t>::Alignment() == 1, "");
>+  static_assert(Layout<int32_t>::Alignment() == 4, "");
>+  static_assert(Layout<int64_t>::Alignment() == 8, "");
>+  static_assert(Layout<Aligned<int8_t, 64>>::Alignment() == 64, "");
>+  static_assert(Layout<int8_t, int32_t, int64_t>::Alignment() == 8, "");
>+  static_assert(Layout<int8_t, int64_t, int32_t>::Alignment() == 8, "");
>+  static_assert(Layout<int32_t, int8_t, int64_t>::Alignment() == 8, "");
>+  static_assert(Layout<int32_t, int64_t, int8_t>::Alignment() == 8, "");
>+  static_assert(Layout<int64_t, int8_t, int32_t>::Alignment() == 8, "");
>+  static_assert(Layout<int64_t, int32_t, int8_t>::Alignment() == 8, "");
>+}
>+
>+TEST(Layout, ConstexprPartial) {
>+  constexpr size_t M = alignof(max_align_t);
>+  constexpr Layout<unsigned char, Aligned<unsigned char, 2 * M>> x(1, 3);
>+  static_assert(x.Partial(1).template Offset<1>() == 2 * M, "");
>+}
>+// [from, to)
>+struct Region {
>+  size_t from;
>+  size_t to;
>+};
>+
>+void ExpectRegionPoisoned(const unsigned char* p, size_t n, bool poisoned) {
>+#ifdef ADDRESS_SANITIZER
>+  for (size_t i = 0; i != n; ++i) {
>+    EXPECT_EQ(poisoned, __asan_address_is_poisoned(p + i));
>+  }
>+#endif
>+}
>+
>+template <size_t N>
>+void ExpectPoisoned(const unsigned char (&buf)[N],
>+                    std::initializer_list<Region> reg) {
>+  size_t prev = 0;
>+  for (const Region& r : reg) {
>+    ExpectRegionPoisoned(buf + prev, r.from - prev, false);
>+    ExpectRegionPoisoned(buf + r.from, r.to - r.from, true);
>+    prev = r.to;
>+  }
>+  ExpectRegionPoisoned(buf + prev, N - prev, false);
>+}
>+
>+TEST(Layout, PoisonPadding) {
>+  using L = Layout<int8_t, int64_t, int32_t, Int128>;
>+
>+  constexpr size_t n = L::Partial(1, 2, 3, 4).AllocSize();
>+  {
>+    constexpr auto x = L::Partial();
>+    alignas(max_align_t) const unsigned char c[n] = {};
>+    x.PoisonPadding(c);
>+    EXPECT_EQ(x.Slices(c), x.Slices(c));
>+    ExpectPoisoned(c, {});
>+  }
>+  {
>+    constexpr auto x = L::Partial(1);
>+    alignas(max_align_t) const unsigned char c[n] = {};
>+    x.PoisonPadding(c);
>+    EXPECT_EQ(x.Slices(c), x.Slices(c));
>+    ExpectPoisoned(c, {{1, 8}});
>+  }
>+  {
>+    constexpr auto x = L::Partial(1, 2);
>+    alignas(max_align_t) const unsigned char c[n] = {};
>+    x.PoisonPadding(c);
>+    EXPECT_EQ(x.Slices(c), x.Slices(c));
>+    ExpectPoisoned(c, {{1, 8}});
>+  }
>+  {
>+    constexpr auto x = L::Partial(1, 2, 3);
>+    alignas(max_align_t) const unsigned char c[n] = {};
>+    x.PoisonPadding(c);
>+    EXPECT_EQ(x.Slices(c), x.Slices(c));
>+    ExpectPoisoned(c, {{1, 8}, {36, 40}});
>+  }
>+  {
>+    constexpr auto x = L::Partial(1, 2, 3, 4);
>+    alignas(max_align_t) const unsigned char c[n] = {};
>+    x.PoisonPadding(c);
>+    EXPECT_EQ(x.Slices(c), x.Slices(c));
>+    ExpectPoisoned(c, {{1, 8}, {36, 40}});
>+  }
>+  {
>+    constexpr L x(1, 2, 3, 4);
>+    alignas(max_align_t) const unsigned char c[n] = {};
>+    x.PoisonPadding(c);
>+    EXPECT_EQ(x.Slices(c), x.Slices(c));
>+    ExpectPoisoned(c, {{1, 8}, {36, 40}});
>+  }
>+}
>+
>+TEST(Layout, DebugString) {
>+  const std::string int64_type =
>+#ifdef _MSC_VER
>+  "__int64";
>+#else   // _MSC_VER
>+  std::is_same<int64_t, long long>::value ? "long long" : "long";  // NOLINT
>+#endif  // _MSC_VER
>+  {
>+    constexpr auto x = Layout<int8_t, int32_t, int8_t, Int128>::Partial();
>+    EXPECT_EQ("@0<signed char>(1)", x.DebugString());
>+  }
>+  {
>+    constexpr auto x = Layout<int8_t, int32_t, int8_t, Int128>::Partial(1);
>+    EXPECT_EQ("@0<signed char>(1)[1]; @4<int>(4)", x.DebugString());
>+  }
>+  {
>+    constexpr auto x = Layout<int8_t, int32_t, int8_t, Int128>::Partial(1, 2);
>+    EXPECT_EQ("@0<signed char>(1)[1]; @4<int>(4)[2]; @12<signed char>(1)",
>+              x.DebugString());
>+  }
>+  {
>+    constexpr auto x = Layout<int8_t, int32_t, int8_t, Int128>::Partial(1, 2, 3);
>+    EXPECT_EQ(
>+        "@0<signed char>(1)[1]; @4<int>(4)[2]; @12<signed char>(1)[3]; "
>+        "@16<" +
>+            int64_type + " [2]>(16)",
>+        x.DebugString());
>+  }
>+  {
>+    constexpr auto x = Layout<int8_t, int32_t, int8_t, Int128>::Partial(1, 2, 3, 4);
>+    EXPECT_EQ(
>+        "@0<signed char>(1)[1]; @4<int>(4)[2]; @12<signed char>(1)[3]; "
>+        "@16<" +
>+            int64_type + " [2]>(16)[4]",
>+        x.DebugString());
>+  }
>+  {
>+    constexpr Layout<int8_t, int32_t, int8_t, Int128> x(1, 2, 3, 4);
>+    EXPECT_EQ(
>+        "@0<signed char>(1)[1]; @4<int>(4)[2]; @12<signed char>(1)[3]; "
>+        "@16<" +
>+            int64_type + " [2]>(16)[4]",
>+        x.DebugString());
>+  }
>+}
>+
>+TEST(Layout, CharTypes) {
>+  constexpr Layout<int32_t> x(1);
>+  alignas(max_align_t) char c[x.AllocSize()] = {};
>+  alignas(max_align_t) unsigned char uc[x.AllocSize()] = {};
>+  alignas(max_align_t) signed char sc[x.AllocSize()] = {};
>+  alignas(max_align_t) const char cc[x.AllocSize()] = {};
>+  alignas(max_align_t) const unsigned char cuc[x.AllocSize()] = {};
>+  alignas(max_align_t) const signed char csc[x.AllocSize()] = {};
>+
>+  Type<int32_t*>(x.Pointer<0>(c));
>+  Type<int32_t*>(x.Pointer<0>(uc));
>+  Type<int32_t*>(x.Pointer<0>(sc));
>+  Type<const int32_t*>(x.Pointer<0>(cc));
>+  Type<const int32_t*>(x.Pointer<0>(cuc));
>+  Type<const int32_t*>(x.Pointer<0>(csc));
>+
>+  Type<int32_t*>(x.Pointer<int32_t>(c));
>+  Type<int32_t*>(x.Pointer<int32_t>(uc));
>+  Type<int32_t*>(x.Pointer<int32_t>(sc));
>+  Type<const int32_t*>(x.Pointer<int32_t>(cc));
>+  Type<const int32_t*>(x.Pointer<int32_t>(cuc));
>+  Type<const int32_t*>(x.Pointer<int32_t>(csc));
>+
>+  Type<std::tuple<int32_t*>>(x.Pointers(c));
>+  Type<std::tuple<int32_t*>>(x.Pointers(uc));
>+  Type<std::tuple<int32_t*>>(x.Pointers(sc));
>+  Type<std::tuple<const int32_t*>>(x.Pointers(cc));
>+  Type<std::tuple<const int32_t*>>(x.Pointers(cuc));
>+  Type<std::tuple<const int32_t*>>(x.Pointers(csc));
>+
>+  Type<Span<int32_t>>(x.Slice<0>(c));
>+  Type<Span<int32_t>>(x.Slice<0>(uc));
>+  Type<Span<int32_t>>(x.Slice<0>(sc));
>+  Type<Span<const int32_t>>(x.Slice<0>(cc));
>+  Type<Span<const int32_t>>(x.Slice<0>(cuc));
>+  Type<Span<const int32_t>>(x.Slice<0>(csc));
>+
>+  Type<std::tuple<Span<int32_t>>>(x.Slices(c));
>+  Type<std::tuple<Span<int32_t>>>(x.Slices(uc));
>+  Type<std::tuple<Span<int32_t>>>(x.Slices(sc));
>+  Type<std::tuple<Span<const int32_t>>>(x.Slices(cc));
>+  Type<std::tuple<Span<const int32_t>>>(x.Slices(cuc));
>+  Type<std::tuple<Span<const int32_t>>>(x.Slices(csc));
>+}
>+
>+TEST(Layout, ConstElementType) {
>+  constexpr Layout<const int32_t> x(1);
>+  alignas(int32_t) char c[x.AllocSize()] = {};
>+  const char* cc = c;
>+  const int32_t* p = reinterpret_cast<const int32_t*>(cc);
>+
>+  EXPECT_EQ(alignof(int32_t), x.Alignment());
>+
>+  EXPECT_EQ(0, x.Offset<0>());
>+  EXPECT_EQ(0, x.Offset<const int32_t>());
>+
>+  EXPECT_THAT(x.Offsets(), ElementsAre(0));
>+
>+  EXPECT_EQ(1, x.Size<0>());
>+  EXPECT_EQ(1, x.Size<const int32_t>());
>+
>+  EXPECT_THAT(x.Sizes(), ElementsAre(1));
>+
>+  EXPECT_EQ(sizeof(int32_t), x.AllocSize());
>+
>+  EXPECT_EQ(p, Type<const int32_t*>(x.Pointer<0>(c)));
>+  EXPECT_EQ(p, Type<const int32_t*>(x.Pointer<0>(cc)));
>+
>+  EXPECT_EQ(p, Type<const int32_t*>(x.Pointer<const int32_t>(c)));
>+  EXPECT_EQ(p, Type<const int32_t*>(x.Pointer<const int32_t>(cc)));
>+
>+  EXPECT_THAT(Type<std::tuple<const int32_t*>>(x.Pointers(c)), Tuple(p));
>+  EXPECT_THAT(Type<std::tuple<const int32_t*>>(x.Pointers(cc)), Tuple(p));
>+
>+  EXPECT_THAT(Type<Span<const int32_t>>(x.Slice<0>(c)),
>+              IsSameSlice(Span<const int32_t>(p, 1)));
>+  EXPECT_THAT(Type<Span<const int32_t>>(x.Slice<0>(cc)),
>+              IsSameSlice(Span<const int32_t>(p, 1)));
>+
>+  EXPECT_THAT(Type<Span<const int32_t>>(x.Slice<const int32_t>(c)),
>+              IsSameSlice(Span<const int32_t>(p, 1)));
>+  EXPECT_THAT(Type<Span<const int32_t>>(x.Slice<const int32_t>(cc)),
>+              IsSameSlice(Span<const int32_t>(p, 1)));
>+
>+  EXPECT_THAT(Type<std::tuple<Span<const int32_t>>>(x.Slices(c)),
>+              Tuple(IsSameSlice(Span<const int32_t>(p, 1))));
>+  EXPECT_THAT(Type<std::tuple<Span<const int32_t>>>(x.Slices(cc)),
>+              Tuple(IsSameSlice(Span<const int32_t>(p, 1))));
>+}
>+
>+namespace example {
>+
>+// Immutable move-only string with sizeof equal to sizeof(void*). The string
>+// size and the characters are kept in the same heap allocation.
>+class CompactString {
>+ public:
>+  CompactString(const char* s = "") {  // NOLINT
>+    const size_t size = strlen(s);
>+    // size_t[1], followed by char[size + 1].
>+    // This statement doesn't allocate memory.
>+    const L layout(1, size + 1);
>+    // AllocSize() tells us how much memory we need to allocate for all our
>+    // data.
>+    p_.reset(new unsigned char[layout.AllocSize()]);
>+    // If running under ASAN, mark the padding bytes, if any, to catch memory
>+    // errors.
>+    layout.PoisonPadding(p_.get());
>+    // Store the size in the allocation.
>+    // Pointer<size_t>() is a synonym for Pointer<0>().
>+    *layout.Pointer<size_t>(p_.get()) = size;
>+    // Store the characters in the allocation.
>+    memcpy(layout.Pointer<char>(p_.get()), s, size + 1);
>+  }
>+
>+  size_t size() const {
>+    // Equivalent to reinterpret_cast<size_t&>(*p).
>+    return *L::Partial().Pointer<size_t>(p_.get());
>+  }
>+
>+  const char* c_str() const {
>+    // Equivalent to reinterpret_cast<char*>(p.get() + sizeof(size_t)).
>+    // The argument in Partial(1) specifies that we have size_t[1] in front of
>+    // the
>+    // characters.
>+    return L::Partial(1).Pointer<char>(p_.get());
>+  }
>+
>+ private:
>+  // Our heap allocation contains a size_t followed by an array of chars.
>+  using L = Layout<size_t, char>;
>+  std::unique_ptr<unsigned char[]> p_;
>+};
>+
>+TEST(CompactString, Works) {
>+  CompactString s = "hello";
>+  EXPECT_EQ(5, s.size());
>+  EXPECT_STREQ("hello", s.c_str());
>+}
>+
>+}  // namespace example
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/node_hash_policy.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/node_hash_policy.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..065e7009e7e7e4c5b033a5d258cad93572cfda5e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/node_hash_policy.h
>@@ -0,0 +1,88 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Adapts a policy for nodes.
>+//
>+// The node policy should model:
>+//
>+// struct Policy {
>+//   // Returns a new node allocated and constructed using the allocator, using
>+//   // the specified arguments.
>+//   template <class Alloc, class... Args>
>+//   value_type* new_element(Alloc* alloc, Args&&... args) const;
>+//
>+//   // Destroys and deallocates node using the allocator.
>+//   template <class Alloc>
>+//   void delete_element(Alloc* alloc, value_type* node) const;
>+// };
>+//
>+// It may also optionally define `value()` and `apply()`. For documentation on
>+// these, see hash_policy_traits.h.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_NODE_HASH_POLICY_H_
>+#define ABSL_CONTAINER_INTERNAL_NODE_HASH_POLICY_H_
>+
>+#include <cassert>
>+#include <cstddef>
>+#include <memory>
>+#include <type_traits>
>+#include <utility>
>+
>+namespace absl {
>+namespace container_internal {
>+
>+template <class Reference, class Policy>
>+struct node_hash_policy {
>+  static_assert(std::is_lvalue_reference<Reference>::value, "");
>+
>+  using slot_type = typename std::remove_cv<
>+      typename std::remove_reference<Reference>::type>::type*;
>+
>+  template <class Alloc, class... Args>
>+  static void construct(Alloc* alloc, slot_type* slot, Args&&... args) {
>+    *slot = Policy::new_element(alloc, std::forward<Args>(args)...);
>+  }
>+
>+  template <class Alloc>
>+  static void destroy(Alloc* alloc, slot_type* slot) {
>+    Policy::delete_element(alloc, *slot);
>+  }
>+
>+  template <class Alloc>
>+  static void transfer(Alloc*, slot_type* new_slot, slot_type* old_slot) {
>+    *new_slot = *old_slot;
>+  }
>+
>+  static size_t space_used(const slot_type* slot) {
>+    if (slot == nullptr) return Policy::element_space_used(nullptr);
>+    return Policy::element_space_used(*slot);
>+  }
>+
>+  static Reference element(slot_type* slot) { return **slot; }
>+
>+  template <class T, class P = Policy>
>+  static auto value(T* elem) -> decltype(P::value(elem)) {
>+    return P::value(elem);
>+  }
>+
>+  template <class... Ts, class P = Policy>
>+  static auto apply(Ts&&... ts) -> decltype(P::apply(std::forward<Ts>(ts)...)) {
>+    return P::apply(std::forward<Ts>(ts)...);
>+  }
>+};
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_NODE_HASH_POLICY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/node_hash_policy_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/node_hash_policy_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..43d287e3c4a04ebd618c305a15b76305f72a7497
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/node_hash_policy_test.cc
>@@ -0,0 +1,67 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/node_hash_policy.h"
>+
>+#include <memory>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/container/internal/hash_policy_traits.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+using ::testing::Pointee;
>+
>+struct Policy : node_hash_policy<int&, Policy> {
>+  using key_type = int;
>+  using init_type = int;
>+
>+  template <class Alloc>
>+  static int* new_element(Alloc* alloc, int value) {
>+    return new int(value);
>+  }
>+
>+  template <class Alloc>
>+  static void delete_element(Alloc* alloc, int* elem) {
>+    delete elem;
>+  }
>+};
>+
>+using NodePolicy = hash_policy_traits<Policy>;
>+
>+struct NodeTest : ::testing::Test {
>+  std::allocator<int> alloc;
>+  int n = 53;
>+  int* a = &n;
>+};
>+
>+TEST_F(NodeTest, ConstructDestroy) {
>+  NodePolicy::construct(&alloc, &a, 42);
>+  EXPECT_THAT(a, Pointee(42));
>+  NodePolicy::destroy(&alloc, &a);
>+}
>+
>+TEST_F(NodeTest, transfer) {
>+  int s = 42;
>+  int* b = &s;
>+  NodePolicy::transfer(&alloc, &a, &b);
>+  EXPECT_EQ(&s, a);
>+}
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_map.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_map.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..1edc0071e7de829de2ddc79675846632f6604c46
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_map.h
>@@ -0,0 +1,182 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_RAW_HASH_MAP_H_
>+#define ABSL_CONTAINER_INTERNAL_RAW_HASH_MAP_H_
>+
>+#include <tuple>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/container/internal/container_memory.h"
>+#include "absl/container/internal/raw_hash_set.h"  // IWYU pragma: export
>+
>+namespace absl {
>+namespace container_internal {
>+
>+template <class Policy, class Hash, class Eq, class Alloc>
>+class raw_hash_map : public raw_hash_set<Policy, Hash, Eq, Alloc> {
>+  // P is Policy. It's passed as a template argument to support maps that have
>+  // incomplete types as values, as in unordered_map<K, IncompleteType>.
>+  // MappedReference<> may be a non-reference type.
>+  template <class P>
>+  using MappedReference = decltype(P::value(
>+      std::addressof(std::declval<typename raw_hash_map::reference>())));
>+
>+  // MappedConstReference<> may be a non-reference type.
>+  template <class P>
>+  using MappedConstReference = decltype(P::value(
>+      std::addressof(std::declval<typename raw_hash_map::const_reference>())));
>+
>+ public:
>+  using key_type = typename Policy::key_type;
>+  using mapped_type = typename Policy::mapped_type;
>+  template <typename K>
>+  using key_arg = typename raw_hash_map::raw_hash_set::template key_arg<K>;
>+
>+  static_assert(!std::is_reference<key_type>::value, "");
>+  // TODO(alkis): remove this assertion and verify that reference mapped_type is
>+  // supported.
>+  static_assert(!std::is_reference<mapped_type>::value, "");
>+
>+  using iterator = typename raw_hash_map::raw_hash_set::iterator;
>+  using const_iterator = typename raw_hash_map::raw_hash_set::const_iterator;
>+
>+  raw_hash_map() {}
>+  using raw_hash_map::raw_hash_set::raw_hash_set;
>+
>+  // The last two template parameters ensure that both arguments are rvalues
>+  // (lvalue arguments are handled by the overloads below). This is necessary
>+  // for supporting bitfield arguments.
>+  //
>+  //   union { int n : 1; };
>+  //   flat_hash_map<int, int> m;
>+  //   m.insert_or_assign(n, n);
>+  template <class K = key_type, class V = mapped_type, K* = nullptr,
>+            V* = nullptr>
>+  std::pair<iterator, bool> insert_or_assign(key_arg<K>&& k, V&& v) {
>+    return insert_or_assign_impl(std::forward<K>(k), std::forward<V>(v));
>+  }
>+
>+  template <class K = key_type, class V = mapped_type, K* = nullptr>
>+  std::pair<iterator, bool> insert_or_assign(key_arg<K>&& k, const V& v) {
>+    return insert_or_assign_impl(std::forward<K>(k), v);
>+  }
>+
>+  template <class K = key_type, class V = mapped_type, V* = nullptr>
>+  std::pair<iterator, bool> insert_or_assign(const key_arg<K>& k, V&& v) {
>+    return insert_or_assign_impl(k, std::forward<V>(v));
>+  }
>+
>+  template <class K = key_type, class V = mapped_type>
>+  std::pair<iterator, bool> insert_or_assign(const key_arg<K>& k, const V& v) {
>+    return insert_or_assign_impl(k, v);
>+  }
>+
>+  template <class K = key_type, class V = mapped_type, K* = nullptr,
>+            V* = nullptr>
>+  iterator insert_or_assign(const_iterator, key_arg<K>&& k, V&& v) {
>+    return insert_or_assign(std::forward<K>(k), std::forward<V>(v)).first;
>+  }
>+
>+  template <class K = key_type, class V = mapped_type, K* = nullptr>
>+  iterator insert_or_assign(const_iterator, key_arg<K>&& k, const V& v) {
>+    return insert_or_assign(std::forward<K>(k), v).first;
>+  }
>+
>+  template <class K = key_type, class V = mapped_type, V* = nullptr>
>+  iterator insert_or_assign(const_iterator, const key_arg<K>& k, V&& v) {
>+    return insert_or_assign(k, std::forward<V>(v)).first;
>+  }
>+
>+  template <class K = key_type, class V = mapped_type>
>+  iterator insert_or_assign(const_iterator, const key_arg<K>& k, const V& v) {
>+    return insert_or_assign(k, v).first;
>+  }
>+
>+  template <class K = key_type, class... Args,
>+            typename std::enable_if<
>+                !std::is_convertible<K, const_iterator>::value, int>::type = 0,
>+            K* = nullptr>
>+  std::pair<iterator, bool> try_emplace(key_arg<K>&& k, Args&&... args) {
>+    return try_emplace_impl(std::forward<K>(k), std::forward<Args>(args)...);
>+  }
>+
>+  template <class K = key_type, class... Args,
>+            typename std::enable_if<
>+                !std::is_convertible<K, const_iterator>::value, int>::type = 0>
>+  std::pair<iterator, bool> try_emplace(const key_arg<K>& k, Args&&... args) {
>+    return try_emplace_impl(k, std::forward<Args>(args)...);
>+  }
>+
>+  template <class K = key_type, class... Args, K* = nullptr>
>+  iterator try_emplace(const_iterator, key_arg<K>&& k, Args&&... args) {
>+    return try_emplace(std::forward<K>(k), std::forward<Args>(args)...).first;
>+  }
>+
>+  template <class K = key_type, class... Args>
>+  iterator try_emplace(const_iterator, const key_arg<K>& k, Args&&... args) {
>+    return try_emplace(k, std::forward<Args>(args)...).first;
>+  }
>+
>+  template <class K = key_type, class P = Policy>
>+  MappedReference<P> at(const key_arg<K>& key) {
>+    auto it = this->find(key);
>+    if (it == this->end()) std::abort();
>+    return Policy::value(&*it);
>+  }
>+
>+  template <class K = key_type, class P = Policy>
>+  MappedConstReference<P> at(const key_arg<K>& key) const {
>+    auto it = this->find(key);
>+    if (it == this->end()) std::abort();
>+    return Policy::value(&*it);
>+  }
>+
>+  template <class K = key_type, class P = Policy, K* = nullptr>
>+  MappedReference<P> operator[](key_arg<K>&& key) {
>+    return Policy::value(&*try_emplace(std::forward<K>(key)).first);
>+  }
>+
>+  template <class K = key_type, class P = Policy>
>+  MappedReference<P> operator[](const key_arg<K>& key) {
>+    return Policy::value(&*try_emplace(key).first);
>+  }
>+
>+ private:
>+  template <class K, class V>
>+  std::pair<iterator, bool> insert_or_assign_impl(K&& k, V&& v) {
>+    auto res = this->find_or_prepare_insert(k);
>+    if (res.second)
>+      this->emplace_at(res.first, std::forward<K>(k), std::forward<V>(v));
>+    else
>+      Policy::value(&*this->iterator_at(res.first)) = std::forward<V>(v);
>+    return {this->iterator_at(res.first), res.second};
>+  }
>+
>+  template <class K = key_type, class... Args>
>+  std::pair<iterator, bool> try_emplace_impl(K&& k, Args&&... args) {
>+    auto res = this->find_or_prepare_insert(k);
>+    if (res.second)
>+      this->emplace_at(res.first, std::piecewise_construct,
>+                       std::forward_as_tuple(std::forward<K>(k)),
>+                       std::forward_as_tuple(std::forward<Args>(args)...));
>+    return {this->iterator_at(res.first), res.second};
>+  }
>+};
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_RAW_HASH_MAP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..10153129fd1e6ebe7513009ea8423fdec45e3634
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set.cc
>@@ -0,0 +1,45 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/raw_hash_set.h"
>+
>+#include <cstddef>
>+
>+#include "absl/base/config.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+constexpr size_t Group::kWidth;
>+
>+// Returns "random" seed.
>+inline size_t RandomSeed() {
>+#if ABSL_HAVE_THREAD_LOCAL
>+  static thread_local size_t counter = 0;
>+  size_t value = ++counter;
>+#else   // ABSL_HAVE_THREAD_LOCAL
>+  static std::atomic<size_t> counter;
>+  size_t value = counter.fetch_add(1, std::memory_order_relaxed);
>+#endif  // ABSL_HAVE_THREAD_LOCAL
>+  return value ^ static_cast<size_t>(reinterpret_cast<uintptr_t>(&counter));
>+}
>+
>+bool ShouldInsertBackwards(size_t hash, ctrl_t* ctrl) {
>+  // To avoid problems with weak hashes and single bit tests, we use % 13.
>+  // TODO(kfm,sbenza): revisit after we do unconditional mixing
>+  return (H1(hash, ctrl) ^ RandomSeed()) % 13 > 6;
>+}
>+
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..70da90f79a4fc63a9676582727cfecc7ed8c2be5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set.h
>@@ -0,0 +1,1915 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// An open-addressing
>+// hashtable with quadratic probing.
>+//
>+// This is a low level hashtable on top of which different interfaces can be
>+// implemented, like flat_hash_set, node_hash_set, string_hash_set, etc.
>+//
>+// The table interface is similar to that of std::unordered_set. Notable
>+// differences are that most member functions support heterogeneous keys when
>+// BOTH the hash and eq functions are marked as transparent. They do so by
>+// providing a typedef called `is_transparent`.
>+//
>+// When heterogeneous lookup is enabled, functions that take key_type act as if
>+// they have an overload set like:
>+//
>+//   iterator find(const key_type& key);
>+//   template <class K>
>+//   iterator find(const K& key);
>+//
>+//   size_type erase(const key_type& key);
>+//   template <class K>
>+//   size_type erase(const K& key);
>+//
>+//   std::pair<iterator, iterator> equal_range(const key_type& key);
>+//   template <class K>
>+//   std::pair<iterator, iterator> equal_range(const K& key);
>+//
>+// When heterogeneous lookup is disabled, only the explicit `key_type` overloads
>+// exist.
>+//
>+// find() also supports passing the hash explicitly:
>+//
>+//   iterator find(const key_type& key, size_t hash);
>+//   template <class U>
>+//   iterator find(const U& key, size_t hash);
>+//
>+// In addition the pointer to element and iterator stability guarantees are
>+// weaker: all iterators and pointers are invalidated after a new element is
>+// inserted.
>+//
>+// IMPLEMENTATION DETAILS
>+//
>+// The table stores elements inline in a slot array. In addition to the slot
>+// array the table maintains some control state per slot. The extra state is one
>+// byte per slot and stores empty or deleted marks, or alternatively 7 bits from
>+// the hash of an occupied slot. The table is split into logical groups of
>+// slots, like so:
>+//
>+//      Group 1         Group 2        Group 3
>+// +---------------+---------------+---------------+
>+// | | | | | | | | | | | | | | | | | | | | | | | | |
>+// +---------------+---------------+---------------+
>+//
>+// On lookup the hash is split into two parts:
>+// - H2: 7 bits (those stored in the control bytes)
>+// - H1: the rest of the bits
>+// The groups are probed using H1. For each group the slots are matched to H2 in
>+// parallel. Because H2 is 7 bits (128 states) and the number of slots per group
>+// is low (8 or 16) in almost all cases a match in H2 is also a lookup hit.
>+//
>+// On insert, once the right group is found (as in lookup), its slots are
>+// filled in order.
>+//
>+// On erase a slot is cleared. In case the group did not have any empty slots
>+// before the erase, the erased slot is marked as deleted.
>+//
>+// Groups without empty slots (but maybe with deleted slots) extend the probe
>+// sequence. The probing algorithm is quadratic. Given N the number of groups,
>+// the probing function for the i'th probe is:
>+//
>+//   P(0) = H1 % N
>+//
>+//   P(i) = (P(i - 1) + i) % N
>+//
>+// This probing function guarantees that after N probes, all the groups of the
>+// table will be probed exactly once.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_RAW_HASH_SET_H_
>+#define ABSL_CONTAINER_INTERNAL_RAW_HASH_SET_H_
>+
>+#ifndef SWISSTABLE_HAVE_SSE2
>+#ifdef __SSE2__
>+#define SWISSTABLE_HAVE_SSE2 1
>+#else
>+#define SWISSTABLE_HAVE_SSE2 0
>+#endif
>+#endif
>+
>+#ifndef SWISSTABLE_HAVE_SSSE3
>+#ifdef __SSSE3__
>+#define SWISSTABLE_HAVE_SSSE3 1
>+#else
>+#define SWISSTABLE_HAVE_SSSE3 0
>+#endif
>+#endif
>+
>+#if SWISSTABLE_HAVE_SSSE3 && !SWISSTABLE_HAVE_SSE2
>+#error "Bad configuration!"
>+#endif
>+
>+#if SWISSTABLE_HAVE_SSE2
>+#include <x86intrin.h>
>+#endif
>+
>+#include <algorithm>
>+#include <cmath>
>+#include <cstdint>
>+#include <cstring>
>+#include <iterator>
>+#include <limits>
>+#include <memory>
>+#include <tuple>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/base/internal/bits.h"
>+#include "absl/base/internal/endian.h"
>+#include "absl/base/port.h"
>+#include "absl/container/internal/compressed_tuple.h"
>+#include "absl/container/internal/container_memory.h"
>+#include "absl/container/internal/hash_policy_traits.h"
>+#include "absl/container/internal/hashtable_debug_hooks.h"
>+#include "absl/container/internal/layout.h"
>+#include "absl/memory/memory.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/types/optional.h"
>+#include "absl/utility/utility.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+template <size_t Width>
>+class probe_seq {
>+ public:
>+  probe_seq(size_t hash, size_t mask) {
>+    assert(((mask + 1) & mask) == 0 && "not a mask");
>+    mask_ = mask;
>+    offset_ = hash & mask_;
>+  }
>+  size_t offset() const { return offset_; }
>+  size_t offset(size_t i) const { return (offset_ + i) & mask_; }
>+
>+  void next() {
>+    index_ += Width;
>+    offset_ += index_;
>+    offset_ &= mask_;
>+  }
>+  // 0-based probe index. The i-th probe in the probe sequence.
>+  size_t index() const { return index_; }
>+
>+ private:
>+  size_t mask_;
>+  size_t offset_;
>+  size_t index_ = 0;
>+};
>+
>+template <class ContainerKey, class Hash, class Eq>
>+struct RequireUsableKey {
>+  template <class PassedKey, class... Args>
>+  std::pair<
>+      decltype(std::declval<const Hash&>()(std::declval<const PassedKey&>())),
>+      decltype(std::declval<const Eq&>()(std::declval<const ContainerKey&>(),
>+                                         std::declval<const PassedKey&>()))>*
>+  operator()(const PassedKey&, const Args&...) const;
>+};
>+
>+template <class E, class Policy, class Hash, class Eq, class... Ts>
>+struct IsDecomposable : std::false_type {};
>+
>+template <class Policy, class Hash, class Eq, class... Ts>
>+struct IsDecomposable<
>+    absl::void_t<decltype(
>+        Policy::apply(RequireUsableKey<typename Policy::key_type, Hash, Eq>(),
>+                      std::declval<Ts>()...))>,
>+    Policy, Hash, Eq, Ts...> : std::true_type {};
>+
>+template <class, class = void>
>+struct IsTransparent : std::false_type {};
>+template <class T>
>+struct IsTransparent<T, absl::void_t<typename T::is_transparent>>
>+    : std::true_type {};
>+
>+// TODO(alkis): Switch to std::is_nothrow_swappable when gcc/clang supports it.
>+template <class T>
>+constexpr bool IsNoThrowSwappable() {
>+  using std::swap;
>+  return noexcept(swap(std::declval<T&>(), std::declval<T&>()));
>+}
>+
>+template <typename T>
>+int TrailingZeros(T x) {
>+  return sizeof(T) == 8 ? base_internal::CountTrailingZerosNonZero64(x)
>+                        : base_internal::CountTrailingZerosNonZero32(x);
>+}
>+
>+template <typename T>
>+int LeadingZeros(T x) {
>+  return sizeof(T) == 8 ? base_internal::CountLeadingZeros64(x)
>+                        : base_internal::CountLeadingZeros32(x);
>+}
>+
>+// An abstraction over a bitmask. It provides an easy way to iterate through the
>+// indexes of the set bits of a bitmask.  When Shift=0 (platforms with SSE),
>+// this is a true bitmask.  On non-SSE, platforms the arithematic used to
>+// emulate the SSE behavior works in bytes (Shift=3) and leaves each bytes as
>+// either 0x00 or 0x80.
>+//
>+// For example:
>+//   for (int i : BitMask<uint32_t, 16>(0x5)) -> yields 0, 2
>+//   for (int i : BitMask<uint64_t, 8, 3>(0x0000000080800000)) -> yields 2, 3
>+template <class T, int SignificantBits, int Shift = 0>
>+class BitMask {
>+  static_assert(std::is_unsigned<T>::value, "");
>+  static_assert(Shift == 0 || Shift == 3, "");
>+
>+ public:
>+  // These are useful for unit tests (gunit).
>+  using value_type = int;
>+  using iterator = BitMask;
>+  using const_iterator = BitMask;
>+
>+  explicit BitMask(T mask) : mask_(mask) {}
>+  BitMask& operator++() {
>+    mask_ &= (mask_ - 1);
>+    return *this;
>+  }
>+  explicit operator bool() const { return mask_ != 0; }
>+  int operator*() const { return LowestBitSet(); }
>+  int LowestBitSet() const {
>+    return container_internal::TrailingZeros(mask_) >> Shift;
>+  }
>+  int HighestBitSet() const {
>+    return (sizeof(T) * CHAR_BIT - container_internal::LeadingZeros(mask_) -
>+            1) >>
>+           Shift;
>+  }
>+
>+  BitMask begin() const { return *this; }
>+  BitMask end() const { return BitMask(0); }
>+
>+  int TrailingZeros() const {
>+    return container_internal::TrailingZeros(mask_) >> Shift;
>+  }
>+
>+  int LeadingZeros() const {
>+    constexpr int total_significant_bits = SignificantBits << Shift;
>+    constexpr int extra_bits = sizeof(T) * 8 - total_significant_bits;
>+    return container_internal::LeadingZeros(mask_ << extra_bits) >> Shift;
>+  }
>+
>+ private:
>+  friend bool operator==(const BitMask& a, const BitMask& b) {
>+    return a.mask_ == b.mask_;
>+  }
>+  friend bool operator!=(const BitMask& a, const BitMask& b) {
>+    return a.mask_ != b.mask_;
>+  }
>+
>+  T mask_;
>+};
>+
>+using ctrl_t = signed char;
>+using h2_t = uint8_t;
>+
>+// The values here are selected for maximum performance. See the static asserts
>+// below for details.
>+enum Ctrl : ctrl_t {
>+  kEmpty = -128,   // 0b10000000
>+  kDeleted = -2,   // 0b11111110
>+  kSentinel = -1,  // 0b11111111
>+};
>+static_assert(
>+    kEmpty & kDeleted & kSentinel & 0x80,
>+    "Special markers need to have the MSB to make checking for them efficient");
>+static_assert(kEmpty < kSentinel && kDeleted < kSentinel,
>+              "kEmpty and kDeleted must be smaller than kSentinel to make the "
>+              "SIMD test of IsEmptyOrDeleted() efficient");
>+static_assert(kSentinel == -1,
>+              "kSentinel must be -1 to elide loading it from memory into SIMD "
>+              "registers (pcmpeqd xmm, xmm)");
>+static_assert(kEmpty == -128,
>+              "kEmpty must be -128 to make the SIMD check for its "
>+              "existence efficient (psignb xmm, xmm)");
>+static_assert(~kEmpty & ~kDeleted & kSentinel & 0x7F,
>+              "kEmpty and kDeleted must share an unset bit that is not shared "
>+              "by kSentinel to make the scalar test for MatchEmptyOrDeleted() "
>+              "efficient");
>+static_assert(kDeleted == -2,
>+              "kDeleted must be -2 to make the implementation of "
>+              "ConvertSpecialToEmptyAndFullToDeleted efficient");
>+
>+// A single block of empty control bytes for tables without any slots allocated.
>+// This enables removing a branch in the hot path of find().
>+inline ctrl_t* EmptyGroup() {
>+  alignas(16) static constexpr ctrl_t empty_group[] = {
>+      kSentinel, kEmpty, kEmpty, kEmpty, kEmpty, kEmpty, kEmpty, kEmpty,
>+      kEmpty,    kEmpty, kEmpty, kEmpty, kEmpty, kEmpty, kEmpty, kEmpty};
>+  return const_cast<ctrl_t*>(empty_group);
>+}
>+
>+// Mixes a randomly generated per-process seed with `hash` and `ctrl` to
>+// randomize insertion order within groups.
>+bool ShouldInsertBackwards(size_t hash, ctrl_t* ctrl);
>+
>+// Returns a hash seed.
>+//
>+// The seed consists of the ctrl_ pointer, which adds enough entropy to ensure
>+// non-determinism of iteration order in most cases.
>+inline size_t HashSeed(const ctrl_t* ctrl) {
>+  // The low bits of the pointer have little or no entropy because of
>+  // alignment. We shift the pointer to try to use higher entropy bits. A
>+  // good number seems to be 12 bits, because that aligns with page size.
>+  return reinterpret_cast<uintptr_t>(ctrl) >> 12;
>+}
>+
>+inline size_t H1(size_t hash, const ctrl_t* ctrl) {
>+  return (hash >> 7) ^ HashSeed(ctrl);
>+}
>+inline ctrl_t H2(size_t hash) { return hash & 0x7F; }
>+
>+inline bool IsEmpty(ctrl_t c) { return c == kEmpty; }
>+inline bool IsFull(ctrl_t c) { return c >= 0; }
>+inline bool IsDeleted(ctrl_t c) { return c == kDeleted; }
>+inline bool IsEmptyOrDeleted(ctrl_t c) { return c < kSentinel; }
>+
>+#if SWISSTABLE_HAVE_SSE2
>+struct Group {
>+  static constexpr size_t kWidth = 16;  // the number of slots per group
>+
>+  explicit Group(const ctrl_t* pos) {
>+    ctrl = _mm_loadu_si128(reinterpret_cast<const __m128i*>(pos));
>+  }
>+
>+  // Returns a bitmask representing the positions of slots that match hash.
>+  BitMask<uint32_t, kWidth> Match(h2_t hash) const {
>+    auto match = _mm_set1_epi8(hash);
>+    return BitMask<uint32_t, kWidth>(
>+        _mm_movemask_epi8(_mm_cmpeq_epi8(match, ctrl)));
>+  }
>+
>+  // Returns a bitmask representing the positions of empty slots.
>+  BitMask<uint32_t, kWidth> MatchEmpty() const {
>+#if SWISSTABLE_HAVE_SSSE3
>+    // This only works because kEmpty is -128.
>+    return BitMask<uint32_t, kWidth>(
>+        _mm_movemask_epi8(_mm_sign_epi8(ctrl, ctrl)));
>+#else
>+    return Match(kEmpty);
>+#endif
>+  }
>+
>+  // Returns a bitmask representing the positions of empty or deleted slots.
>+  BitMask<uint32_t, kWidth> MatchEmptyOrDeleted() const {
>+    auto special = _mm_set1_epi8(kSentinel);
>+    return BitMask<uint32_t, kWidth>(
>+        _mm_movemask_epi8(_mm_cmpgt_epi8(special, ctrl)));
>+  }
>+
>+  // Returns the number of trailing empty or deleted elements in the group.
>+  uint32_t CountLeadingEmptyOrDeleted() const {
>+    auto special = _mm_set1_epi8(kSentinel);
>+    return TrailingZeros(_mm_movemask_epi8(_mm_cmpgt_epi8(special, ctrl)) + 1);
>+  }
>+
>+  void ConvertSpecialToEmptyAndFullToDeleted(ctrl_t* dst) const {
>+    auto msbs = _mm_set1_epi8(0x80);
>+    auto x126 = _mm_set1_epi8(126);
>+#if SWISSTABLE_HAVE_SSSE3
>+    auto res = _mm_or_si128(_mm_shuffle_epi8(x126, ctrl), msbs);
>+#else
>+    auto zero = _mm_setzero_si128();
>+    auto special_mask = _mm_cmpgt_epi8(zero, ctrl);
>+    auto res = _mm_or_si128(msbs, _mm_andnot_si128(special_mask, x126));
>+#endif
>+    _mm_storeu_si128(reinterpret_cast<__m128i*>(dst), res);
>+  }
>+
>+  __m128i ctrl;
>+};
>+#else
>+struct Group {
>+  static constexpr size_t kWidth = 8;
>+
>+  explicit Group(const ctrl_t* pos) : ctrl(little_endian::Load64(pos)) {}
>+
>+  BitMask<uint64_t, kWidth, 3> Match(h2_t hash) const {
>+    // For the technique, see:
>+    // http://graphics.stanford.edu/~seander/bithacks.html##ValueInWord
>+    // (Determine if a word has a byte equal to n).
>+    //
>+    // Caveat: there are false positives but:
>+    // - they only occur if there is a real match
>+    // - they never occur on kEmpty, kDeleted, kSentinel
>+    // - they will be handled gracefully by subsequent checks in code
>+    //
>+    // Example:
>+    //   v = 0x1716151413121110
>+    //   hash = 0x12
>+    //   retval = (v - lsbs) & ~v & msbs = 0x0000000080800000
>+    constexpr uint64_t msbs = 0x8080808080808080ULL;
>+    constexpr uint64_t lsbs = 0x0101010101010101ULL;
>+    auto x = ctrl ^ (lsbs * hash);
>+    return BitMask<uint64_t, kWidth, 3>((x - lsbs) & ~x & msbs);
>+  }
>+
>+  BitMask<uint64_t, kWidth, 3> MatchEmpty() const {
>+    constexpr uint64_t msbs = 0x8080808080808080ULL;
>+    return BitMask<uint64_t, kWidth, 3>((ctrl & (~ctrl << 6)) & msbs);
>+  }
>+
>+  BitMask<uint64_t, kWidth, 3> MatchEmptyOrDeleted() const {
>+    constexpr uint64_t msbs = 0x8080808080808080ULL;
>+    return BitMask<uint64_t, kWidth, 3>((ctrl & (~ctrl << 7)) & msbs);
>+  }
>+
>+  uint32_t CountLeadingEmptyOrDeleted() const {
>+    constexpr uint64_t gaps = 0x00FEFEFEFEFEFEFEULL;
>+    return (TrailingZeros(((~ctrl & (ctrl >> 7)) | gaps) + 1) + 7) >> 3;
>+  }
>+
>+  void ConvertSpecialToEmptyAndFullToDeleted(ctrl_t* dst) const {
>+    constexpr uint64_t msbs = 0x8080808080808080ULL;
>+    constexpr uint64_t lsbs = 0x0101010101010101ULL;
>+    auto x = ctrl & msbs;
>+    auto res = (~x + (x >> 7)) & ~lsbs;
>+    little_endian::Store64(dst, res);
>+  }
>+
>+  uint64_t ctrl;
>+};
>+#endif  // SWISSTABLE_HAVE_SSE2
>+
>+template <class Policy, class Hash, class Eq, class Alloc>
>+class raw_hash_set;
>+
>+
>+inline bool IsValidCapacity(size_t n) {
>+  return ((n + 1) & n) == 0 && n >= Group::kWidth - 1;
>+}
>+
>+// PRECONDITION:
>+//   IsValidCapacity(capacity)
>+//   ctrl[capacity] == kSentinel
>+//   ctrl[i] != kSentinel for all i < capacity
>+// Applies mapping for every byte in ctrl:
>+//   DELETED -> EMPTY
>+//   EMPTY -> EMPTY
>+//   FULL -> DELETED
>+inline void ConvertDeletedToEmptyAndFullToDeleted(
>+    ctrl_t* ctrl, size_t capacity) {
>+  assert(ctrl[capacity] == kSentinel);
>+  assert(IsValidCapacity(capacity));
>+  for (ctrl_t* pos = ctrl; pos != ctrl + capacity + 1; pos += Group::kWidth) {
>+    Group{pos}.ConvertSpecialToEmptyAndFullToDeleted(pos);
>+  }
>+  // Copy the cloned ctrl bytes.
>+  std::memcpy(ctrl + capacity + 1, ctrl, Group::kWidth);
>+  ctrl[capacity] = kSentinel;
>+}
>+
>+// Rounds up the capacity to the next power of 2 minus 1 and ensures it is
>+// greater or equal to Group::kWidth - 1.
>+inline size_t NormalizeCapacity(size_t n) {
>+  constexpr size_t kMinCapacity = Group::kWidth - 1;
>+  return n <= kMinCapacity
>+             ? kMinCapacity
>+             : std::numeric_limits<size_t>::max() >> LeadingZeros(n);
>+}
>+
>+// The node_handle concept from C++17.
>+// We specialize node_handle for sets and maps. node_handle_base holds the
>+// common API of both.
>+template <typename Policy, typename Alloc>
>+class node_handle_base {
>+ protected:
>+  using PolicyTraits = hash_policy_traits<Policy>;
>+  using slot_type = typename PolicyTraits::slot_type;
>+
>+ public:
>+  using allocator_type = Alloc;
>+
>+  constexpr node_handle_base() {}
>+  node_handle_base(node_handle_base&& other) noexcept {
>+    *this = std::move(other);
>+  }
>+  ~node_handle_base() { destroy(); }
>+  node_handle_base& operator=(node_handle_base&& other) {
>+    destroy();
>+    if (!other.empty()) {
>+      alloc_ = other.alloc_;
>+      PolicyTraits::transfer(alloc(), slot(), other.slot());
>+      other.reset();
>+    }
>+    return *this;
>+  }
>+
>+  bool empty() const noexcept { return !alloc_; }
>+  explicit operator bool() const noexcept { return !empty(); }
>+  allocator_type get_allocator() const { return *alloc_; }
>+
>+ protected:
>+  template <typename, typename, typename, typename>
>+  friend class raw_hash_set;
>+
>+  node_handle_base(const allocator_type& a, slot_type* s) : alloc_(a) {
>+    PolicyTraits::transfer(alloc(), slot(), s);
>+  }
>+
>+  void destroy() {
>+    if (!empty()) {
>+      PolicyTraits::destroy(alloc(), slot());
>+      reset();
>+    }
>+  }
>+
>+  void reset() {
>+    assert(alloc_.has_value());
>+    alloc_ = absl::nullopt;
>+  }
>+
>+  slot_type* slot() const {
>+    assert(!empty());
>+    return reinterpret_cast<slot_type*>(std::addressof(slot_space_));
>+  }
>+  allocator_type* alloc() { return std::addressof(*alloc_); }
>+
>+ private:
>+  absl::optional<allocator_type> alloc_;
>+  mutable absl::aligned_storage_t<sizeof(slot_type), alignof(slot_type)>
>+      slot_space_;
>+};
>+
>+// For sets.
>+template <typename Policy, typename Alloc, typename = void>
>+class node_handle : public node_handle_base<Policy, Alloc> {
>+  using Base = typename node_handle::node_handle_base;
>+
>+ public:
>+  using value_type = typename Base::PolicyTraits::value_type;
>+
>+  constexpr node_handle() {}
>+
>+  value_type& value() const {
>+    return Base::PolicyTraits::element(this->slot());
>+  }
>+
>+ private:
>+  template <typename, typename, typename, typename>
>+  friend class raw_hash_set;
>+
>+  node_handle(const Alloc& a, typename Base::slot_type* s) : Base(a, s) {}
>+};
>+
>+// For maps.
>+template <typename Policy, typename Alloc>
>+class node_handle<Policy, Alloc, absl::void_t<typename Policy::mapped_type>>
>+    : public node_handle_base<Policy, Alloc> {
>+  using Base = typename node_handle::node_handle_base;
>+
>+ public:
>+  using key_type = typename Policy::key_type;
>+  using mapped_type = typename Policy::mapped_type;
>+
>+  constexpr node_handle() {}
>+
>+  auto key() const -> decltype(Base::PolicyTraits::key(this->slot())) {
>+    return Base::PolicyTraits::key(this->slot());
>+  }
>+
>+  mapped_type& mapped() const {
>+    return Base::PolicyTraits::value(
>+        &Base::PolicyTraits::element(this->slot()));
>+  }
>+
>+ private:
>+  template <typename, typename, typename, typename>
>+  friend class raw_hash_set;
>+
>+  node_handle(const Alloc& a, typename Base::slot_type* s) : Base(a, s) {}
>+};
>+
>+// Implement the insert_return_type<> concept of C++17.
>+template <class Iterator, class NodeType>
>+struct insert_return_type {
>+  Iterator position;
>+  bool inserted;
>+  NodeType node;
>+};
>+
>+// Helper trait to allow or disallow arbitrary keys when the hash and
>+// eq functions are transparent.
>+// It is very important that the inner template is an alias and that the type it
>+// produces is not a dependent type. Otherwise, type deduction would fail.
>+template <bool is_transparent>
>+struct KeyArg {
>+  // Transparent. Forward `K`.
>+  template <typename K, typename key_type>
>+  using type = K;
>+};
>+
>+template <>
>+struct KeyArg<false> {
>+  // Not transparent. Always use `key_type`.
>+  template <typename K, typename key_type>
>+  using type = key_type;
>+};
>+
>+// Policy: a policy defines how to perform different operations on
>+// the slots of the hashtable (see hash_policy_traits.h for the full interface
>+// of policy).
>+//
>+// Hash: a (possibly polymorphic) functor that hashes keys of the hashtable. The
>+// functor should accept a key and return size_t as hash. For best performance
>+// it is important that the hash function provides high entropy across all bits
>+// of the hash.
>+//
>+// Eq: a (possibly polymorphic) functor that compares two keys for equality. It
>+// should accept two (of possibly different type) keys and return a bool: true
>+// if they are equal, false if they are not. If two keys compare equal, then
>+// their hash values as defined by Hash MUST be equal.
>+//
>+// Allocator: an Allocator [http://devdocs.io/cpp/concept/allocator] with which
>+// the storage of the hashtable will be allocated and the elements will be
>+// constructed and destroyed.
>+template <class Policy, class Hash, class Eq, class Alloc>
>+class raw_hash_set {
>+  using PolicyTraits = hash_policy_traits<Policy>;
>+  using KeyArgImpl = container_internal::KeyArg<IsTransparent<Eq>::value &&
>+                                                IsTransparent<Hash>::value>;
>+
>+ public:
>+  using init_type = typename PolicyTraits::init_type;
>+  using key_type = typename PolicyTraits::key_type;
>+  // TODO(sbenza): Hide slot_type as it is an implementation detail. Needs user
>+  // code fixes!
>+  using slot_type = typename PolicyTraits::slot_type;
>+  using allocator_type = Alloc;
>+  using size_type = size_t;
>+  using difference_type = ptrdiff_t;
>+  using hasher = Hash;
>+  using key_equal = Eq;
>+  using policy_type = Policy;
>+  using value_type = typename PolicyTraits::value_type;
>+  using reference = value_type&;
>+  using const_reference = const value_type&;
>+  using pointer = typename absl::allocator_traits<
>+      allocator_type>::template rebind_traits<value_type>::pointer;
>+  using const_pointer = typename absl::allocator_traits<
>+      allocator_type>::template rebind_traits<value_type>::const_pointer;
>+
>+  // Alias used for heterogeneous lookup functions.
>+  // `key_arg<K>` evaluates to `K` when the functors are tranparent and to
>+  // `key_type` otherwise. It permits template argument deduction on `K` for the
>+  // transparent case.
>+  template <class K>
>+  using key_arg = typename KeyArgImpl::template type<K, key_type>;
>+
>+ private:
>+  // Give an early error when key_type is not hashable/eq.
>+  auto KeyTypeCanBeHashed(const Hash& h, const key_type& k) -> decltype(h(k));
>+  auto KeyTypeCanBeEq(const Eq& eq, const key_type& k) -> decltype(eq(k, k));
>+
>+  using Layout = absl::container_internal::Layout<ctrl_t, slot_type>;
>+
>+  static Layout MakeLayout(size_t capacity) {
>+    assert(IsValidCapacity(capacity));
>+    return Layout(capacity + Group::kWidth + 1, capacity);
>+  }
>+
>+  using AllocTraits = absl::allocator_traits<allocator_type>;
>+  using SlotAlloc = typename absl::allocator_traits<
>+      allocator_type>::template rebind_alloc<slot_type>;
>+  using SlotAllocTraits = typename absl::allocator_traits<
>+      allocator_type>::template rebind_traits<slot_type>;
>+
>+  static_assert(std::is_lvalue_reference<reference>::value,
>+                "Policy::element() must return a reference");
>+
>+  template <typename T>
>+  struct SameAsElementReference
>+      : std::is_same<typename std::remove_cv<
>+                         typename std::remove_reference<reference>::type>::type,
>+                     typename std::remove_cv<
>+                         typename std::remove_reference<T>::type>::type> {};
>+
>+  // An enabler for insert(T&&): T must be convertible to init_type or be the
>+  // same as [cv] value_type [ref].
>+  // Note: we separate SameAsElementReference into its own type to avoid using
>+  // reference unless we need to. MSVC doesn't seem to like it in some
>+  // cases.
>+  template <class T>
>+  using RequiresInsertable = typename std::enable_if<
>+      absl::disjunction<std::is_convertible<T, init_type>,
>+                        SameAsElementReference<T>>::value,
>+      int>::type;
>+
>+  // RequiresNotInit is a workaround for gcc prior to 7.1.
>+  // See https://godbolt.org/g/Y4xsUh.
>+  template <class T>
>+  using RequiresNotInit =
>+      typename std::enable_if<!std::is_same<T, init_type>::value, int>::type;
>+
>+  template <class... Ts>
>+  using IsDecomposable = IsDecomposable<void, PolicyTraits, Hash, Eq, Ts...>;
>+
>+ public:
>+  static_assert(std::is_same<pointer, value_type*>::value,
>+                "Allocators with custom pointer types are not supported");
>+  static_assert(std::is_same<const_pointer, const value_type*>::value,
>+                "Allocators with custom pointer types are not supported");
>+
>+  class iterator {
>+    friend class raw_hash_set;
>+
>+   public:
>+    using iterator_category = std::forward_iterator_tag;
>+    using value_type = typename raw_hash_set::value_type;
>+    using reference =
>+        absl::conditional_t<PolicyTraits::constant_iterators::value,
>+                            const value_type&, value_type&>;
>+    using pointer = absl::remove_reference_t<reference>*;
>+    using difference_type = typename raw_hash_set::difference_type;
>+
>+    iterator() {}
>+
>+    // PRECONDITION: not an end() iterator.
>+    reference operator*() const { return PolicyTraits::element(slot_); }
>+
>+    // PRECONDITION: not an end() iterator.
>+    pointer operator->() const { return &operator*(); }
>+
>+    // PRECONDITION: not an end() iterator.
>+    iterator& operator++() {
>+      ++ctrl_;
>+      ++slot_;
>+      skip_empty_or_deleted();
>+      return *this;
>+    }
>+    // PRECONDITION: not an end() iterator.
>+    iterator operator++(int) {
>+      auto tmp = *this;
>+      ++*this;
>+      return tmp;
>+    }
>+
>+    friend bool operator==(const iterator& a, const iterator& b) {
>+      return a.ctrl_ == b.ctrl_;
>+    }
>+    friend bool operator!=(const iterator& a, const iterator& b) {
>+      return !(a == b);
>+    }
>+
>+   private:
>+    iterator(ctrl_t* ctrl) : ctrl_(ctrl) {}  // for end()
>+    iterator(ctrl_t* ctrl, slot_type* slot) : ctrl_(ctrl), slot_(slot) {}
>+
>+    void skip_empty_or_deleted() {
>+      while (IsEmptyOrDeleted(*ctrl_)) {
>+        // ctrl is not necessarily aligned to Group::kWidth. It is also likely
>+        // to read past the space for ctrl bytes and into slots. This is ok
>+        // because ctrl has sizeof() == 1 and slot has sizeof() >= 1 so there
>+        // is no way to read outside the combined slot array.
>+        uint32_t shift = Group{ctrl_}.CountLeadingEmptyOrDeleted();
>+        ctrl_ += shift;
>+        slot_ += shift;
>+      }
>+    }
>+
>+    ctrl_t* ctrl_ = nullptr;
>+    slot_type* slot_;
>+  };
>+
>+  class const_iterator {
>+    friend class raw_hash_set;
>+
>+   public:
>+    using iterator_category = typename iterator::iterator_category;
>+    using value_type = typename raw_hash_set::value_type;
>+    using reference = typename raw_hash_set::const_reference;
>+    using pointer = typename raw_hash_set::const_pointer;
>+    using difference_type = typename raw_hash_set::difference_type;
>+
>+    const_iterator() {}
>+    // Implicit construction from iterator.
>+    const_iterator(iterator i) : inner_(std::move(i)) {}
>+
>+    reference operator*() const { return *inner_; }
>+    pointer operator->() const { return inner_.operator->(); }
>+
>+    const_iterator& operator++() {
>+      ++inner_;
>+      return *this;
>+    }
>+    const_iterator operator++(int) { return inner_++; }
>+
>+    friend bool operator==(const const_iterator& a, const const_iterator& b) {
>+      return a.inner_ == b.inner_;
>+    }
>+    friend bool operator!=(const const_iterator& a, const const_iterator& b) {
>+      return !(a == b);
>+    }
>+
>+   private:
>+    const_iterator(const ctrl_t* ctrl, const slot_type* slot)
>+        : inner_(const_cast<ctrl_t*>(ctrl), const_cast<slot_type*>(slot)) {}
>+
>+    iterator inner_;
>+  };
>+
>+  using node_type = container_internal::node_handle<Policy, Alloc>;
>+
>+  raw_hash_set() noexcept(
>+      std::is_nothrow_default_constructible<hasher>::value&&
>+          std::is_nothrow_default_constructible<key_equal>::value&&
>+              std::is_nothrow_default_constructible<allocator_type>::value) {}
>+
>+  explicit raw_hash_set(size_t bucket_count, const hasher& hash = hasher(),
>+                        const key_equal& eq = key_equal(),
>+                        const allocator_type& alloc = allocator_type())
>+      : ctrl_(EmptyGroup()), settings_(0, hash, eq, alloc) {
>+    if (bucket_count) {
>+      capacity_ = NormalizeCapacity(bucket_count);
>+      growth_left() = static_cast<size_t>(capacity_ * kMaxLoadFactor);
>+      initialize_slots();
>+    }
>+  }
>+
>+  raw_hash_set(size_t bucket_count, const hasher& hash,
>+               const allocator_type& alloc)
>+      : raw_hash_set(bucket_count, hash, key_equal(), alloc) {}
>+
>+  raw_hash_set(size_t bucket_count, const allocator_type& alloc)
>+      : raw_hash_set(bucket_count, hasher(), key_equal(), alloc) {}
>+
>+  explicit raw_hash_set(const allocator_type& alloc)
>+      : raw_hash_set(0, hasher(), key_equal(), alloc) {}
>+
>+  template <class InputIter>
>+  raw_hash_set(InputIter first, InputIter last, size_t bucket_count = 0,
>+               const hasher& hash = hasher(), const key_equal& eq = key_equal(),
>+               const allocator_type& alloc = allocator_type())
>+      : raw_hash_set(bucket_count, hash, eq, alloc) {
>+    insert(first, last);
>+  }
>+
>+  template <class InputIter>
>+  raw_hash_set(InputIter first, InputIter last, size_t bucket_count,
>+               const hasher& hash, const allocator_type& alloc)
>+      : raw_hash_set(first, last, bucket_count, hash, key_equal(), alloc) {}
>+
>+  template <class InputIter>
>+  raw_hash_set(InputIter first, InputIter last, size_t bucket_count,
>+               const allocator_type& alloc)
>+      : raw_hash_set(first, last, bucket_count, hasher(), key_equal(), alloc) {}
>+
>+  template <class InputIter>
>+  raw_hash_set(InputIter first, InputIter last, const allocator_type& alloc)
>+      : raw_hash_set(first, last, 0, hasher(), key_equal(), alloc) {}
>+
>+  // Instead of accepting std::initializer_list<value_type> as the first
>+  // argument like std::unordered_set<value_type> does, we have two overloads
>+  // that accept std::initializer_list<T> and std::initializer_list<init_type>.
>+  // This is advantageous for performance.
>+  //
>+  //   // Turns {"abc", "def"} into std::initializer_list<std::string>, then copies
>+  //   // the strings into the set.
>+  //   std::unordered_set<std::string> s = {"abc", "def"};
>+  //
>+  //   // Turns {"abc", "def"} into std::initializer_list<const char*>, then
>+  //   // copies the strings into the set.
>+  //   absl::flat_hash_set<std::string> s = {"abc", "def"};
>+  //
>+  // The same trick is used in insert().
>+  //
>+  // The enabler is necessary to prevent this constructor from triggering where
>+  // the copy constructor is meant to be called.
>+  //
>+  //   absl::flat_hash_set<int> a, b{a};
>+  //
>+  // RequiresNotInit<T> is a workaround for gcc prior to 7.1.
>+  template <class T, RequiresNotInit<T> = 0, RequiresInsertable<T> = 0>
>+  raw_hash_set(std::initializer_list<T> init, size_t bucket_count = 0,
>+               const hasher& hash = hasher(), const key_equal& eq = key_equal(),
>+               const allocator_type& alloc = allocator_type())
>+      : raw_hash_set(init.begin(), init.end(), bucket_count, hash, eq, alloc) {}
>+
>+  raw_hash_set(std::initializer_list<init_type> init, size_t bucket_count = 0,
>+               const hasher& hash = hasher(), const key_equal& eq = key_equal(),
>+               const allocator_type& alloc = allocator_type())
>+      : raw_hash_set(init.begin(), init.end(), bucket_count, hash, eq, alloc) {}
>+
>+  template <class T, RequiresNotInit<T> = 0, RequiresInsertable<T> = 0>
>+  raw_hash_set(std::initializer_list<T> init, size_t bucket_count,
>+               const hasher& hash, const allocator_type& alloc)
>+      : raw_hash_set(init, bucket_count, hash, key_equal(), alloc) {}
>+
>+  raw_hash_set(std::initializer_list<init_type> init, size_t bucket_count,
>+               const hasher& hash, const allocator_type& alloc)
>+      : raw_hash_set(init, bucket_count, hash, key_equal(), alloc) {}
>+
>+  template <class T, RequiresNotInit<T> = 0, RequiresInsertable<T> = 0>
>+  raw_hash_set(std::initializer_list<T> init, size_t bucket_count,
>+               const allocator_type& alloc)
>+      : raw_hash_set(init, bucket_count, hasher(), key_equal(), alloc) {}
>+
>+  raw_hash_set(std::initializer_list<init_type> init, size_t bucket_count,
>+               const allocator_type& alloc)
>+      : raw_hash_set(init, bucket_count, hasher(), key_equal(), alloc) {}
>+
>+  template <class T, RequiresNotInit<T> = 0, RequiresInsertable<T> = 0>
>+  raw_hash_set(std::initializer_list<T> init, const allocator_type& alloc)
>+      : raw_hash_set(init, 0, hasher(), key_equal(), alloc) {}
>+
>+  raw_hash_set(std::initializer_list<init_type> init,
>+               const allocator_type& alloc)
>+      : raw_hash_set(init, 0, hasher(), key_equal(), alloc) {}
>+
>+  raw_hash_set(const raw_hash_set& that)
>+      : raw_hash_set(that, AllocTraits::select_on_container_copy_construction(
>+                               that.alloc_ref())) {}
>+
>+  raw_hash_set(const raw_hash_set& that, const allocator_type& a)
>+      : raw_hash_set(0, that.hash_ref(), that.eq_ref(), a) {
>+    reserve(that.size());
>+    // Because the table is guaranteed to be empty, we can do something faster
>+    // than a full `insert`.
>+    for (const auto& v : that) {
>+      const size_t hash = PolicyTraits::apply(HashElement{hash_ref()}, v);
>+      const size_t i = find_first_non_full(hash);
>+      set_ctrl(i, H2(hash));
>+      emplace_at(i, v);
>+    }
>+    size_ = that.size();
>+    growth_left() -= that.size();
>+  }
>+
>+  raw_hash_set(raw_hash_set&& that) noexcept(
>+      std::is_nothrow_copy_constructible<hasher>::value&&
>+          std::is_nothrow_copy_constructible<key_equal>::value&&
>+              std::is_nothrow_copy_constructible<allocator_type>::value)
>+      : ctrl_(absl::exchange(that.ctrl_, EmptyGroup())),
>+        slots_(absl::exchange(that.slots_, nullptr)),
>+        size_(absl::exchange(that.size_, 0)),
>+        capacity_(absl::exchange(that.capacity_, 0)),
>+        // Hash, equality and allocator are copied instead of moved because
>+        // `that` must be left valid. If Hash is std::function<Key>, moving it
>+        // would create a nullptr functor that cannot be called.
>+        settings_(that.settings_) {
>+    // growth_left was copied above, reset the one from `that`.
>+    that.growth_left() = 0;
>+  }
>+
>+  raw_hash_set(raw_hash_set&& that, const allocator_type& a)
>+      : ctrl_(EmptyGroup()),
>+        slots_(nullptr),
>+        size_(0),
>+        capacity_(0),
>+        settings_(0, that.hash_ref(), that.eq_ref(), a) {
>+    if (a == that.alloc_ref()) {
>+      std::swap(ctrl_, that.ctrl_);
>+      std::swap(slots_, that.slots_);
>+      std::swap(size_, that.size_);
>+      std::swap(capacity_, that.capacity_);
>+      std::swap(growth_left(), that.growth_left());
>+    } else {
>+      reserve(that.size());
>+      // Note: this will copy elements of dense_set and unordered_set instead of
>+      // moving them. This can be fixed if it ever becomes an issue.
>+      for (auto& elem : that) insert(std::move(elem));
>+    }
>+  }
>+
>+  raw_hash_set& operator=(const raw_hash_set& that) {
>+    raw_hash_set tmp(that,
>+                     AllocTraits::propagate_on_container_copy_assignment::value
>+                         ? that.alloc_ref()
>+                         : alloc_ref());
>+    swap(tmp);
>+    return *this;
>+  }
>+
>+  raw_hash_set& operator=(raw_hash_set&& that) noexcept(
>+      absl::allocator_traits<allocator_type>::is_always_equal::value&&
>+          std::is_nothrow_move_assignable<hasher>::value&&
>+              std::is_nothrow_move_assignable<key_equal>::value) {
>+    // TODO(sbenza): We should only use the operations from the noexcept clause
>+    // to make sure we actually adhere to that contract.
>+    return move_assign(
>+        std::move(that),
>+        typename AllocTraits::propagate_on_container_move_assignment());
>+  }
>+
>+  ~raw_hash_set() { destroy_slots(); }
>+
>+  iterator begin() {
>+    auto it = iterator_at(0);
>+    it.skip_empty_or_deleted();
>+    return it;
>+  }
>+  iterator end() { return {ctrl_ + capacity_}; }
>+
>+  const_iterator begin() const {
>+    return const_cast<raw_hash_set*>(this)->begin();
>+  }
>+  const_iterator end() const { return const_cast<raw_hash_set*>(this)->end(); }
>+  const_iterator cbegin() const { return begin(); }
>+  const_iterator cend() const { return end(); }
>+
>+  bool empty() const { return !size(); }
>+  size_t size() const { return size_; }
>+  size_t capacity() const { return capacity_; }
>+  size_t max_size() const { return std::numeric_limits<size_t>::max(); }
>+
>+  void clear() {
>+    // Iterating over this container is O(bucket_count()). When bucket_count()
>+    // is much greater than size(), iteration becomes prohibitively expensive.
>+    // For clear() it is more important to reuse the allocated array when the
>+    // container is small because allocation takes comparatively long time
>+    // compared to destruction of the elements of the container. So we pick the
>+    // largest bucket_count() threshold for which iteration is still fast and
>+    // past that we simply deallocate the array.
>+    if (capacity_ > 127) {
>+      destroy_slots();
>+    } else if (capacity_) {
>+      for (size_t i = 0; i != capacity_; ++i) {
>+        if (IsFull(ctrl_[i])) {
>+          PolicyTraits::destroy(&alloc_ref(), slots_ + i);
>+        }
>+      }
>+      size_ = 0;
>+      reset_ctrl();
>+      growth_left() = static_cast<size_t>(capacity_ * kMaxLoadFactor);
>+    }
>+    assert(empty());
>+  }
>+
>+  // This overload kicks in when the argument is an rvalue of insertable and
>+  // decomposable type other than init_type.
>+  //
>+  //   flat_hash_map<std::string, int> m;
>+  //   m.insert(std::make_pair("abc", 42));
>+  template <class T, RequiresInsertable<T> = 0,
>+            typename std::enable_if<IsDecomposable<T>::value, int>::type = 0,
>+            T* = nullptr>
>+  std::pair<iterator, bool> insert(T&& value) {
>+    return emplace(std::forward<T>(value));
>+  }
>+
>+  // This overload kicks in when the argument is a bitfield or an lvalue of
>+  // insertable and decomposable type.
>+  //
>+  //   union { int n : 1; };
>+  //   flat_hash_set<int> s;
>+  //   s.insert(n);
>+  //
>+  //   flat_hash_set<std::string> s;
>+  //   const char* p = "hello";
>+  //   s.insert(p);
>+  //
>+  // TODO(romanp): Once we stop supporting gcc 5.1 and below, replace
>+  // RequiresInsertable<T> with RequiresInsertable<const T&>.
>+  // We are hitting this bug: https://godbolt.org/g/1Vht4f.
>+  template <
>+      class T, RequiresInsertable<T> = 0,
>+      typename std::enable_if<IsDecomposable<const T&>::value, int>::type = 0>
>+  std::pair<iterator, bool> insert(const T& value) {
>+    return emplace(value);
>+  }
>+
>+  // This overload kicks in when the argument is an rvalue of init_type. Its
>+  // purpose is to handle brace-init-list arguments.
>+  //
>+  //   flat_hash_set<std::string, int> s;
>+  //   s.insert({"abc", 42});
>+  std::pair<iterator, bool> insert(init_type&& value) {
>+    return emplace(std::move(value));
>+  }
>+
>+  template <class T, RequiresInsertable<T> = 0,
>+            typename std::enable_if<IsDecomposable<T>::value, int>::type = 0,
>+            T* = nullptr>
>+  iterator insert(const_iterator, T&& value) {
>+    return insert(std::forward<T>(value)).first;
>+  }
>+
>+  // TODO(romanp): Once we stop supporting gcc 5.1 and below, replace
>+  // RequiresInsertable<T> with RequiresInsertable<const T&>.
>+  // We are hitting this bug: https://godbolt.org/g/1Vht4f.
>+  template <
>+      class T, RequiresInsertable<T> = 0,
>+      typename std::enable_if<IsDecomposable<const T&>::value, int>::type = 0>
>+  iterator insert(const_iterator, const T& value) {
>+    return insert(value).first;
>+  }
>+
>+  iterator insert(const_iterator, init_type&& value) {
>+    return insert(std::move(value)).first;
>+  }
>+
>+  template <class InputIt>
>+  void insert(InputIt first, InputIt last) {
>+    for (; first != last; ++first) insert(*first);
>+  }
>+
>+  template <class T, RequiresNotInit<T> = 0, RequiresInsertable<const T&> = 0>
>+  void insert(std::initializer_list<T> ilist) {
>+    insert(ilist.begin(), ilist.end());
>+  }
>+
>+  void insert(std::initializer_list<init_type> ilist) {
>+    insert(ilist.begin(), ilist.end());
>+  }
>+
>+  insert_return_type<iterator, node_type> insert(node_type&& node) {
>+    if (!node) return {end(), false, node_type()};
>+    const auto& elem = PolicyTraits::element(node.slot());
>+    auto res = PolicyTraits::apply(
>+        InsertSlot<false>{*this, std::move(*node.slot())}, elem);
>+    if (res.second) {
>+      node.reset();
>+      return {res.first, true, node_type()};
>+    } else {
>+      return {res.first, false, std::move(node)};
>+    }
>+  }
>+
>+  iterator insert(const_iterator, node_type&& node) {
>+    return insert(std::move(node)).first;
>+  }
>+
>+  // This overload kicks in if we can deduce the key from args. This enables us
>+  // to avoid constructing value_type if an entry with the same key already
>+  // exists.
>+  //
>+  // For example:
>+  //
>+  //   flat_hash_map<std::string, std::string> m = {{"abc", "def"}};
>+  //   // Creates no std::string copies and makes no heap allocations.
>+  //   m.emplace("abc", "xyz");
>+  template <class... Args, typename std::enable_if<
>+                               IsDecomposable<Args...>::value, int>::type = 0>
>+  std::pair<iterator, bool> emplace(Args&&... args) {
>+    return PolicyTraits::apply(EmplaceDecomposable{*this},
>+                               std::forward<Args>(args)...);
>+  }
>+
>+  // This overload kicks in if we cannot deduce the key from args. It constructs
>+  // value_type unconditionally and then either moves it into the table or
>+  // destroys.
>+  template <class... Args, typename std::enable_if<
>+                               !IsDecomposable<Args...>::value, int>::type = 0>
>+  std::pair<iterator, bool> emplace(Args&&... args) {
>+    typename std::aligned_storage<sizeof(slot_type), alignof(slot_type)>::type
>+        raw;
>+    slot_type* slot = reinterpret_cast<slot_type*>(&raw);
>+
>+    PolicyTraits::construct(&alloc_ref(), slot, std::forward<Args>(args)...);
>+    const auto& elem = PolicyTraits::element(slot);
>+    return PolicyTraits::apply(InsertSlot<true>{*this, std::move(*slot)}, elem);
>+  }
>+
>+  template <class... Args>
>+  iterator emplace_hint(const_iterator, Args&&... args) {
>+    return emplace(std::forward<Args>(args)...).first;
>+  }
>+
>+  // Extension API: support for lazy emplace.
>+  //
>+  // Looks up key in the table. If found, returns the iterator to the element.
>+  // Otherwise calls f with one argument of type raw_hash_set::constructor. f
>+  // MUST call raw_hash_set::constructor with arguments as if a
>+  // raw_hash_set::value_type is constructed, otherwise the behavior is
>+  // undefined.
>+  //
>+  // For example:
>+  //
>+  //   std::unordered_set<ArenaString> s;
>+  //   // Makes ArenaStr even if "abc" is in the map.
>+  //   s.insert(ArenaString(&arena, "abc"));
>+  //
>+  //   flat_hash_set<ArenaStr> s;
>+  //   // Makes ArenaStr only if "abc" is not in the map.
>+  //   s.lazy_emplace("abc", [&](const constructor& ctor) {
>+  //     ctor(&arena, "abc");
>+  //   });
>+  //
>+  // WARNING: This API is currently experimental. If there is a way to implement
>+  // the same thing with the rest of the API, prefer that.
>+  class constructor {
>+    friend class raw_hash_set;
>+
>+   public:
>+    template <class... Args>
>+    void operator()(Args&&... args) const {
>+      assert(*slot_);
>+      PolicyTraits::construct(alloc_, *slot_, std::forward<Args>(args)...);
>+      *slot_ = nullptr;
>+    }
>+
>+   private:
>+    constructor(allocator_type* a, slot_type** slot) : alloc_(a), slot_(slot) {}
>+
>+    allocator_type* alloc_;
>+    slot_type** slot_;
>+  };
>+
>+  template <class K = key_type, class F>
>+  iterator lazy_emplace(const key_arg<K>& key, F&& f) {
>+    auto res = find_or_prepare_insert(key);
>+    if (res.second) {
>+      slot_type* slot = slots_ + res.first;
>+      std::forward<F>(f)(constructor(&alloc_ref(), &slot));
>+      assert(!slot);
>+    }
>+    return iterator_at(res.first);
>+  }
>+
>+  // Extension API: support for heterogeneous keys.
>+  //
>+  //   std::unordered_set<std::string> s;
>+  //   // Turns "abc" into std::string.
>+  //   s.erase("abc");
>+  //
>+  //   flat_hash_set<std::string> s;
>+  //   // Uses "abc" directly without copying it into std::string.
>+  //   s.erase("abc");
>+  template <class K = key_type>
>+  size_type erase(const key_arg<K>& key) {
>+    auto it = find(key);
>+    if (it == end()) return 0;
>+    erase(it);
>+    return 1;
>+  }
>+
>+  // Erases the element pointed to by `it`.  Unlike `std::unordered_set::erase`,
>+  // this method returns void to reduce algorithmic complexity to O(1).  In
>+  // order to erase while iterating across a map, use the following idiom (which
>+  // also works for standard containers):
>+  //
>+  // for (auto it = m.begin(), end = m.end(); it != end;) {
>+  //   if (<pred>) {
>+  //     m.erase(it++);
>+  //   } else {
>+  //     ++it;
>+  //   }
>+  // }
>+  void erase(const_iterator cit) { erase(cit.inner_); }
>+
>+  // This overload is necessary because otherwise erase<K>(const K&) would be
>+  // a better match if non-const iterator is passed as an argument.
>+  void erase(iterator it) {
>+    assert(it != end());
>+    PolicyTraits::destroy(&alloc_ref(), it.slot_);
>+    erase_meta_only(it);
>+  }
>+
>+  iterator erase(const_iterator first, const_iterator last) {
>+    while (first != last) {
>+      erase(first++);
>+    }
>+    return last.inner_;
>+  }
>+
>+  // Moves elements from `src` into `this`.
>+  // If the element already exists in `this`, it is left unmodified in `src`.
>+  template <typename H, typename E>
>+  void merge(raw_hash_set<Policy, H, E, Alloc>& src) {  // NOLINT
>+    assert(this != &src);
>+    for (auto it = src.begin(), e = src.end(); it != e; ++it) {
>+      if (PolicyTraits::apply(InsertSlot<false>{*this, std::move(*it.slot_)},
>+                              PolicyTraits::element(it.slot_))
>+              .second) {
>+        src.erase_meta_only(it);
>+      }
>+    }
>+  }
>+
>+  template <typename H, typename E>
>+  void merge(raw_hash_set<Policy, H, E, Alloc>&& src) {
>+    merge(src);
>+  }
>+
>+  node_type extract(const_iterator position) {
>+    node_type node(alloc_ref(), position.inner_.slot_);
>+    erase_meta_only(position);
>+    return node;
>+  }
>+
>+  template <
>+      class K = key_type,
>+      typename std::enable_if<!std::is_same<K, iterator>::value, int>::type = 0>
>+  node_type extract(const key_arg<K>& key) {
>+    auto it = find(key);
>+    return it == end() ? node_type() : extract(const_iterator{it});
>+  }
>+
>+  void swap(raw_hash_set& that) noexcept(
>+      IsNoThrowSwappable<hasher>() && IsNoThrowSwappable<key_equal>() &&
>+      (!AllocTraits::propagate_on_container_swap::value ||
>+       IsNoThrowSwappable<allocator_type>())) {
>+    using std::swap;
>+    swap(ctrl_, that.ctrl_);
>+    swap(slots_, that.slots_);
>+    swap(size_, that.size_);
>+    swap(capacity_, that.capacity_);
>+    swap(growth_left(), that.growth_left());
>+    swap(hash_ref(), that.hash_ref());
>+    swap(eq_ref(), that.eq_ref());
>+    if (AllocTraits::propagate_on_container_swap::value) {
>+      swap(alloc_ref(), that.alloc_ref());
>+    } else {
>+      // If the allocators do not compare equal it is officially undefined
>+      // behavior. We choose to do nothing.
>+    }
>+  }
>+
>+  void rehash(size_t n) {
>+    if (n == 0 && capacity_ == 0) return;
>+    if (n == 0 && size_ == 0) return destroy_slots();
>+    auto m = NormalizeCapacity(std::max(n, NumSlotsFast(size())));
>+    // n == 0 unconditionally rehashes as per the standard.
>+    if (n == 0 || m > capacity_) {
>+      resize(m);
>+    }
>+  }
>+
>+  void reserve(size_t n) {
>+    rehash(NumSlotsFast(n));
>+  }
>+
>+  // Extension API: support for heterogeneous keys.
>+  //
>+  //   std::unordered_set<std::string> s;
>+  //   // Turns "abc" into std::string.
>+  //   s.count("abc");
>+  //
>+  //   ch_set<std::string> s;
>+  //   // Uses "abc" directly without copying it into std::string.
>+  //   s.count("abc");
>+  template <class K = key_type>
>+  size_t count(const key_arg<K>& key) const {
>+    return find(key) == end() ? 0 : 1;
>+  }
>+
>+  // Issues CPU prefetch instructions for the memory needed to find or insert
>+  // a key.  Like all lookup functions, this support heterogeneous keys.
>+  //
>+  // NOTE: This is a very low level operation and should not be used without
>+  // specific benchmarks indicating its importance.
>+  template <class K = key_type>
>+  void prefetch(const key_arg<K>& key) const {
>+    (void)key;
>+#if defined(__GNUC__)
>+    auto seq = probe(hash_ref()(key));
>+    __builtin_prefetch(static_cast<const void*>(ctrl_ + seq.offset()));
>+    __builtin_prefetch(static_cast<const void*>(slots_ + seq.offset()));
>+#endif  // __GNUC__
>+  }
>+
>+  // The API of find() has two extensions.
>+  //
>+  // 1. The hash can be passed by the user. It must be equal to the hash of the
>+  // key.
>+  //
>+  // 2. The type of the key argument doesn't have to be key_type. This is so
>+  // called heterogeneous key support.
>+  template <class K = key_type>
>+  iterator find(const key_arg<K>& key, size_t hash) {
>+    auto seq = probe(hash);
>+    while (true) {
>+      Group g{ctrl_ + seq.offset()};
>+      for (int i : g.Match(H2(hash))) {
>+        if (ABSL_PREDICT_TRUE(PolicyTraits::apply(
>+                EqualElement<K>{key, eq_ref()},
>+                PolicyTraits::element(slots_ + seq.offset(i)))))
>+          return iterator_at(seq.offset(i));
>+      }
>+      if (ABSL_PREDICT_TRUE(g.MatchEmpty())) return end();
>+      seq.next();
>+    }
>+  }
>+  template <class K = key_type>
>+  iterator find(const key_arg<K>& key) {
>+    return find(key, hash_ref()(key));
>+  }
>+
>+  template <class K = key_type>
>+  const_iterator find(const key_arg<K>& key, size_t hash) const {
>+    return const_cast<raw_hash_set*>(this)->find(key, hash);
>+  }
>+  template <class K = key_type>
>+  const_iterator find(const key_arg<K>& key) const {
>+    return find(key, hash_ref()(key));
>+  }
>+
>+  template <class K = key_type>
>+  bool contains(const key_arg<K>& key) const {
>+    return find(key) != end();
>+  }
>+
>+  template <class K = key_type>
>+  std::pair<iterator, iterator> equal_range(const key_arg<K>& key) {
>+    auto it = find(key);
>+    if (it != end()) return {it, std::next(it)};
>+    return {it, it};
>+  }
>+  template <class K = key_type>
>+  std::pair<const_iterator, const_iterator> equal_range(
>+      const key_arg<K>& key) const {
>+    auto it = find(key);
>+    if (it != end()) return {it, std::next(it)};
>+    return {it, it};
>+  }
>+
>+  size_t bucket_count() const { return capacity_; }
>+  float load_factor() const {
>+    return capacity_ ? static_cast<double>(size()) / capacity_ : 0.0;
>+  }
>+  float max_load_factor() const { return 1.0f; }
>+  void max_load_factor(float) {
>+    // Does nothing.
>+  }
>+
>+  hasher hash_function() const { return hash_ref(); }
>+  key_equal key_eq() const { return eq_ref(); }
>+  allocator_type get_allocator() const { return alloc_ref(); }
>+
>+  friend bool operator==(const raw_hash_set& a, const raw_hash_set& b) {
>+    if (a.size() != b.size()) return false;
>+    const raw_hash_set* outer = &a;
>+    const raw_hash_set* inner = &b;
>+    if (outer->capacity() > inner->capacity()) std::swap(outer, inner);
>+    for (const value_type& elem : *outer)
>+      if (!inner->has_element(elem)) return false;
>+    return true;
>+  }
>+
>+  friend bool operator!=(const raw_hash_set& a, const raw_hash_set& b) {
>+    return !(a == b);
>+  }
>+
>+  friend void swap(raw_hash_set& a,
>+                   raw_hash_set& b) noexcept(noexcept(a.swap(b))) {
>+    a.swap(b);
>+  }
>+
>+ private:
>+  template <class Container, typename Enabler>
>+  friend struct absl::container_internal::hashtable_debug_internal::
>+      HashtableDebugAccess;
>+
>+  struct FindElement {
>+    template <class K, class... Args>
>+    const_iterator operator()(const K& key, Args&&...) const {
>+      return s.find(key);
>+    }
>+    const raw_hash_set& s;
>+  };
>+
>+  struct HashElement {
>+    template <class K, class... Args>
>+    size_t operator()(const K& key, Args&&...) const {
>+      return h(key);
>+    }
>+    const hasher& h;
>+  };
>+
>+  template <class K1>
>+  struct EqualElement {
>+    template <class K2, class... Args>
>+    bool operator()(const K2& lhs, Args&&...) const {
>+      return eq(lhs, rhs);
>+    }
>+    const K1& rhs;
>+    const key_equal& eq;
>+  };
>+
>+  struct EmplaceDecomposable {
>+    template <class K, class... Args>
>+    std::pair<iterator, bool> operator()(const K& key, Args&&... args) const {
>+      auto res = s.find_or_prepare_insert(key);
>+      if (res.second) {
>+        s.emplace_at(res.first, std::forward<Args>(args)...);
>+      }
>+      return {s.iterator_at(res.first), res.second};
>+    }
>+    raw_hash_set& s;
>+  };
>+
>+  template <bool do_destroy>
>+  struct InsertSlot {
>+    template <class K, class... Args>
>+    std::pair<iterator, bool> operator()(const K& key, Args&&...) && {
>+      auto res = s.find_or_prepare_insert(key);
>+      if (res.second) {
>+        PolicyTraits::transfer(&s.alloc_ref(), s.slots_ + res.first, &slot);
>+      } else if (do_destroy) {
>+        PolicyTraits::destroy(&s.alloc_ref(), &slot);
>+      }
>+      return {s.iterator_at(res.first), res.second};
>+    }
>+    raw_hash_set& s;
>+    // Constructed slot. Either moved into place or destroyed.
>+    slot_type&& slot;
>+  };
>+
>+  // Computes std::ceil(n / kMaxLoadFactor). Faster than calling std::ceil.
>+  static inline size_t NumSlotsFast(size_t n) {
>+    return static_cast<size_t>(
>+        (n * kMaxLoadFactorDenominator + (kMaxLoadFactorNumerator - 1)) /
>+        kMaxLoadFactorNumerator);
>+  }
>+
>+  // "erases" the object from the container, except that it doesn't actually
>+  // destroy the object. It only updates all the metadata of the class.
>+  // This can be used in conjunction with Policy::transfer to move the object to
>+  // another place.
>+  void erase_meta_only(const_iterator it) {
>+    assert(IsFull(*it.inner_.ctrl_) && "erasing a dangling iterator");
>+    --size_;
>+    const size_t index = it.inner_.ctrl_ - ctrl_;
>+    const size_t index_before = (index - Group::kWidth) & capacity_;
>+    const auto empty_after = Group(it.inner_.ctrl_).MatchEmpty();
>+    const auto empty_before = Group(ctrl_ + index_before).MatchEmpty();
>+
>+    // We count how many consecutive non empties we have to the right and to the
>+    // left of `it`. If the sum is >= kWidth then there is at least one probe
>+    // window that might have seen a full group.
>+    bool was_never_full =
>+        empty_before && empty_after &&
>+        static_cast<size_t>(empty_after.TrailingZeros() +
>+                            empty_before.LeadingZeros()) < Group::kWidth;
>+
>+    set_ctrl(index, was_never_full ? kEmpty : kDeleted);
>+    growth_left() += was_never_full;
>+  }
>+
>+  void initialize_slots() {
>+    assert(capacity_);
>+    auto layout = MakeLayout(capacity_);
>+    char* mem = static_cast<char*>(
>+        Allocate<Layout::Alignment()>(&alloc_ref(), layout.AllocSize()));
>+    ctrl_ = reinterpret_cast<ctrl_t*>(layout.template Pointer<0>(mem));
>+    slots_ = layout.template Pointer<1>(mem);
>+    reset_ctrl();
>+    growth_left() = static_cast<size_t>(capacity_ * kMaxLoadFactor) - size_;
>+  }
>+
>+  void destroy_slots() {
>+    if (!capacity_) return;
>+    for (size_t i = 0; i != capacity_; ++i) {
>+      if (IsFull(ctrl_[i])) {
>+        PolicyTraits::destroy(&alloc_ref(), slots_ + i);
>+      }
>+    }
>+    auto layout = MakeLayout(capacity_);
>+    // Unpoison before returning the memory to the allocator.
>+    SanitizerUnpoisonMemoryRegion(slots_, sizeof(slot_type) * capacity_);
>+    Deallocate<Layout::Alignment()>(&alloc_ref(), ctrl_, layout.AllocSize());
>+    ctrl_ = EmptyGroup();
>+    slots_ = nullptr;
>+    size_ = 0;
>+    capacity_ = 0;
>+    growth_left() = 0;
>+  }
>+
>+  void resize(size_t new_capacity) {
>+    assert(IsValidCapacity(new_capacity));
>+    auto* old_ctrl = ctrl_;
>+    auto* old_slots = slots_;
>+    const size_t old_capacity = capacity_;
>+    capacity_ = new_capacity;
>+    initialize_slots();
>+
>+    for (size_t i = 0; i != old_capacity; ++i) {
>+      if (IsFull(old_ctrl[i])) {
>+        size_t hash = PolicyTraits::apply(HashElement{hash_ref()},
>+                                          PolicyTraits::element(old_slots + i));
>+        size_t new_i = find_first_non_full(hash);
>+        set_ctrl(new_i, H2(hash));
>+        PolicyTraits::transfer(&alloc_ref(), slots_ + new_i, old_slots + i);
>+      }
>+    }
>+    if (old_capacity) {
>+      SanitizerUnpoisonMemoryRegion(old_slots,
>+                                    sizeof(slot_type) * old_capacity);
>+      auto layout = MakeLayout(old_capacity);
>+      Deallocate<Layout::Alignment()>(&alloc_ref(), old_ctrl,
>+                                      layout.AllocSize());
>+    }
>+  }
>+
>+  void drop_deletes_without_resize() ABSL_ATTRIBUTE_NOINLINE {
>+    assert(IsValidCapacity(capacity_));
>+    // Algorithm:
>+    // - mark all DELETED slots as EMPTY
>+    // - mark all FULL slots as DELETED
>+    // - for each slot marked as DELETED
>+    //     hash = Hash(element)
>+    //     target = find_first_non_full(hash)
>+    //     if target is in the same group
>+    //       mark slot as FULL
>+    //     else if target is EMPTY
>+    //       transfer element to target
>+    //       mark slot as EMPTY
>+    //       mark target as FULL
>+    //     else if target is DELETED
>+    //       swap current element with target element
>+    //       mark target as FULL
>+    //       repeat procedure for current slot with moved from element (target)
>+    ConvertDeletedToEmptyAndFullToDeleted(ctrl_, capacity_);
>+    typename std::aligned_storage<sizeof(slot_type), alignof(slot_type)>::type
>+        raw;
>+    slot_type* slot = reinterpret_cast<slot_type*>(&raw);
>+    for (size_t i = 0; i != capacity_; ++i) {
>+      if (!IsDeleted(ctrl_[i])) continue;
>+      size_t hash = PolicyTraits::apply(HashElement{hash_ref()},
>+                                        PolicyTraits::element(slots_ + i));
>+      size_t new_i = find_first_non_full(hash);
>+
>+      // Verify if the old and new i fall within the same group wrt the hash.
>+      // If they do, we don't need to move the object as it falls already in the
>+      // best probe we can.
>+      const auto probe_index = [&](size_t pos) {
>+        return ((pos - probe(hash).offset()) & capacity_) / Group::kWidth;
>+      };
>+
>+      // Element doesn't move.
>+      if (ABSL_PREDICT_TRUE(probe_index(new_i) == probe_index(i))) {
>+        set_ctrl(i, H2(hash));
>+        continue;
>+      }
>+      if (IsEmpty(ctrl_[new_i])) {
>+        // Transfer element to the empty spot.
>+        // set_ctrl poisons/unpoisons the slots so we have to call it at the
>+        // right time.
>+        set_ctrl(new_i, H2(hash));
>+        PolicyTraits::transfer(&alloc_ref(), slots_ + new_i, slots_ + i);
>+        set_ctrl(i, kEmpty);
>+      } else {
>+        assert(IsDeleted(ctrl_[new_i]));
>+        set_ctrl(new_i, H2(hash));
>+        // Until we are done rehashing, DELETED marks previously FULL slots.
>+        // Swap i and new_i elements.
>+        PolicyTraits::transfer(&alloc_ref(), slot, slots_ + i);
>+        PolicyTraits::transfer(&alloc_ref(), slots_ + i, slots_ + new_i);
>+        PolicyTraits::transfer(&alloc_ref(), slots_ + new_i, slot);
>+        --i;  // repeat
>+      }
>+    }
>+    growth_left() = static_cast<size_t>(capacity_ * kMaxLoadFactor) - size_;
>+  }
>+
>+  void rehash_and_grow_if_necessary() {
>+    if (capacity_ == 0) {
>+      resize(Group::kWidth - 1);
>+    } else if (size() <= kMaxLoadFactor / 2 * capacity_) {
>+      // Squash DELETED without growing if there is enough capacity.
>+      drop_deletes_without_resize();
>+    } else {
>+      // Otherwise grow the container.
>+      resize(capacity_ * 2 + 1);
>+    }
>+  }
>+
>+  bool has_element(const value_type& elem) const {
>+    size_t hash = PolicyTraits::apply(HashElement{hash_ref()}, elem);
>+    auto seq = probe(hash);
>+    while (true) {
>+      Group g{ctrl_ + seq.offset()};
>+      for (int i : g.Match(H2(hash))) {
>+        if (ABSL_PREDICT_TRUE(PolicyTraits::element(slots_ + seq.offset(i)) ==
>+                              elem))
>+          return true;
>+      }
>+      if (ABSL_PREDICT_TRUE(g.MatchEmpty())) return false;
>+      seq.next();
>+      assert(seq.index() < capacity_ && "full table!");
>+    }
>+    return false;
>+  }
>+
>+  // Probes the raw_hash_set with the probe sequence for hash and returns the
>+  // pointer to the first empty or deleted slot.
>+  // NOTE: this function must work with tables having both kEmpty and kDelete
>+  // in one group. Such tables appears during drop_deletes_without_resize.
>+  //
>+  // This function is very useful when insertions happen and:
>+  // - the input is already a set
>+  // - there are enough slots
>+  // - the element with the hash is not in the table
>+  size_t find_first_non_full(size_t hash) {
>+    auto seq = probe(hash);
>+    while (true) {
>+      Group g{ctrl_ + seq.offset()};
>+      auto mask = g.MatchEmptyOrDeleted();
>+      if (mask) {
>+#if !defined(NDEBUG)
>+        // We want to force small tables to have random entries too, so
>+        // in debug build we will randomly insert in either the front or back of
>+        // the group.
>+        // TODO(kfm,sbenza): revisit after we do unconditional mixing
>+        if (ShouldInsertBackwards(hash, ctrl_))
>+          return seq.offset(mask.HighestBitSet());
>+        else
>+          return seq.offset(mask.LowestBitSet());
>+#else
>+        return seq.offset(mask.LowestBitSet());
>+#endif
>+      }
>+      assert(seq.index() < capacity_ && "full table!");
>+      seq.next();
>+    }
>+  }
>+
>+  // TODO(alkis): Optimize this assuming *this and that don't overlap.
>+  raw_hash_set& move_assign(raw_hash_set&& that, std::true_type) {
>+    raw_hash_set tmp(std::move(that));
>+    swap(tmp);
>+    return *this;
>+  }
>+  raw_hash_set& move_assign(raw_hash_set&& that, std::false_type) {
>+    raw_hash_set tmp(std::move(that), alloc_ref());
>+    swap(tmp);
>+    return *this;
>+  }
>+
>+ protected:
>+  template <class K>
>+  std::pair<size_t, bool> find_or_prepare_insert(const K& key) {
>+    auto hash = hash_ref()(key);
>+    auto seq = probe(hash);
>+    while (true) {
>+      Group g{ctrl_ + seq.offset()};
>+      for (int i : g.Match(H2(hash))) {
>+        if (ABSL_PREDICT_TRUE(PolicyTraits::apply(
>+                EqualElement<K>{key, eq_ref()},
>+                PolicyTraits::element(slots_ + seq.offset(i)))))
>+          return {seq.offset(i), false};
>+      }
>+      if (ABSL_PREDICT_TRUE(g.MatchEmpty())) break;
>+      seq.next();
>+    }
>+    return {prepare_insert(hash), true};
>+  }
>+
>+  size_t prepare_insert(size_t hash) ABSL_ATTRIBUTE_NOINLINE {
>+    size_t target = find_first_non_full(hash);
>+    if (ABSL_PREDICT_FALSE(growth_left() == 0 && !IsDeleted(ctrl_[target]))) {
>+      rehash_and_grow_if_necessary();
>+      target = find_first_non_full(hash);
>+    }
>+    ++size_;
>+    growth_left() -= IsEmpty(ctrl_[target]);
>+    set_ctrl(target, H2(hash));
>+    return target;
>+  }
>+
>+  // Constructs the value in the space pointed by the iterator. This only works
>+  // after an unsuccessful find_or_prepare_insert() and before any other
>+  // modifications happen in the raw_hash_set.
>+  //
>+  // PRECONDITION: i is an index returned from find_or_prepare_insert(k), where
>+  // k is the key decomposed from `forward<Args>(args)...`, and the bool
>+  // returned by find_or_prepare_insert(k) was true.
>+  // POSTCONDITION: *m.iterator_at(i) == value_type(forward<Args>(args)...).
>+  template <class... Args>
>+  void emplace_at(size_t i, Args&&... args) {
>+    PolicyTraits::construct(&alloc_ref(), slots_ + i,
>+                            std::forward<Args>(args)...);
>+
>+    assert(PolicyTraits::apply(FindElement{*this}, *iterator_at(i)) ==
>+               iterator_at(i) &&
>+           "constructed value does not match the lookup key");
>+  }
>+
>+  iterator iterator_at(size_t i) { return {ctrl_ + i, slots_ + i}; }
>+  const_iterator iterator_at(size_t i) const { return {ctrl_ + i, slots_ + i}; }
>+
>+ private:
>+  friend struct RawHashSetTestOnlyAccess;
>+
>+  probe_seq<Group::kWidth> probe(size_t hash) const {
>+    return probe_seq<Group::kWidth>(H1(hash, ctrl_), capacity_);
>+  }
>+
>+  // Reset all ctrl bytes back to kEmpty, except the sentinel.
>+  void reset_ctrl() {
>+    std::memset(ctrl_, kEmpty, capacity_ + Group::kWidth);
>+    ctrl_[capacity_] = kSentinel;
>+    SanitizerPoisonMemoryRegion(slots_, sizeof(slot_type) * capacity_);
>+  }
>+
>+  // Sets the control byte, and if `i < Group::kWidth`, set the cloned byte at
>+  // the end too.
>+  void set_ctrl(size_t i, ctrl_t h) {
>+    assert(i < capacity_);
>+
>+    if (IsFull(h)) {
>+      SanitizerUnpoisonObject(slots_ + i);
>+    } else {
>+      SanitizerPoisonObject(slots_ + i);
>+    }
>+
>+    ctrl_[i] = h;
>+    ctrl_[((i - Group::kWidth) & capacity_) + Group::kWidth] = h;
>+  }
>+
>+  size_t& growth_left() { return settings_.template get<0>(); }
>+
>+  hasher& hash_ref() { return settings_.template get<1>(); }
>+  const hasher& hash_ref() const { return settings_.template get<1>(); }
>+  key_equal& eq_ref() { return settings_.template get<2>(); }
>+  const key_equal& eq_ref() const { return settings_.template get<2>(); }
>+  allocator_type& alloc_ref() { return settings_.template get<3>(); }
>+  const allocator_type& alloc_ref() const {
>+    return settings_.template get<3>();
>+  }
>+
>+  // On average each group has 2 empty slot (for the vectorized case).
>+  static constexpr int64_t kMaxLoadFactorNumerator = 14;
>+  static constexpr int64_t kMaxLoadFactorDenominator = 16;
>+  static constexpr float kMaxLoadFactor =
>+      1.0 * kMaxLoadFactorNumerator / kMaxLoadFactorDenominator;
>+
>+  // TODO(alkis): Investigate removing some of these fields:
>+  // - ctrl/slots can be derived from each other
>+  // - size can be moved into the slot array
>+  ctrl_t* ctrl_ = EmptyGroup();    // [(capacity + 1) * ctrl_t]
>+  slot_type* slots_ = nullptr;     // [capacity * slot_type]
>+  size_t size_ = 0;                // number of full slots
>+  size_t capacity_ = 0;            // total number of slots
>+  absl::container_internal::CompressedTuple<size_t /* growth_left */, hasher,
>+                                            key_equal, allocator_type>
>+      settings_{0, hasher{}, key_equal{}, allocator_type{}};
>+};
>+
>+namespace hashtable_debug_internal {
>+template <typename Set>
>+struct HashtableDebugAccess<Set, absl::void_t<typename Set::raw_hash_set>> {
>+  using Traits = typename Set::PolicyTraits;
>+  using Slot = typename Traits::slot_type;
>+
>+  static size_t GetNumProbes(const Set& set,
>+                             const typename Set::key_type& key) {
>+    size_t num_probes = 0;
>+    size_t hash = set.hash_ref()(key);
>+    auto seq = set.probe(hash);
>+    while (true) {
>+      container_internal::Group g{set.ctrl_ + seq.offset()};
>+      for (int i : g.Match(container_internal::H2(hash))) {
>+        if (Traits::apply(
>+                typename Set::template EqualElement<typename Set::key_type>{
>+                    key, set.eq_ref()},
>+                Traits::element(set.slots_ + seq.offset(i))))
>+          return num_probes;
>+        ++num_probes;
>+      }
>+      if (g.MatchEmpty()) return num_probes;
>+      seq.next();
>+      ++num_probes;
>+    }
>+  }
>+
>+  static size_t AllocatedByteSize(const Set& c) {
>+    size_t capacity = c.capacity_;
>+    if (capacity == 0) return 0;
>+    auto layout = Set::MakeLayout(capacity);
>+    size_t m = layout.AllocSize();
>+
>+    size_t per_slot = Traits::space_used(static_cast<const Slot*>(nullptr));
>+    if (per_slot != ~size_t{}) {
>+      m += per_slot * c.size();
>+    } else {
>+      for (size_t i = 0; i != capacity; ++i) {
>+        if (container_internal::IsFull(c.ctrl_[i])) {
>+          m += Traits::space_used(c.slots_ + i);
>+        }
>+      }
>+    }
>+    return m;
>+  }
>+
>+  static size_t LowerBoundAllocatedByteSize(size_t size) {
>+    size_t capacity = container_internal::NormalizeCapacity(
>+        std::ceil(size / Set::kMaxLoadFactor));
>+    if (capacity == 0) return 0;
>+    auto layout = Set::MakeLayout(capacity);
>+    size_t m = layout.AllocSize();
>+    size_t per_slot = Traits::space_used(static_cast<const Slot*>(nullptr));
>+    if (per_slot != ~size_t{}) {
>+      m += per_slot * size;
>+    }
>+    return m;
>+  }
>+};
>+
>+}  // namespace hashtable_debug_internal
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_RAW_HASH_SET_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set_allocator_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set_allocator_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..891fa450fe08c223e202b58fc38a02b8bf77822b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set_allocator_test.cc
>@@ -0,0 +1,428 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <limits>
>+#include <scoped_allocator>
>+
>+#include "gtest/gtest.h"
>+#include "absl/container/internal/raw_hash_set.h"
>+#include "absl/container/internal/tracked.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+enum AllocSpec {
>+  kPropagateOnCopy = 1,
>+  kPropagateOnMove = 2,
>+  kPropagateOnSwap = 4,
>+};
>+
>+struct AllocState {
>+  size_t num_allocs = 0;
>+  std::set<void*> owned;
>+};
>+
>+template <class T,
>+          int Spec = kPropagateOnCopy | kPropagateOnMove | kPropagateOnSwap>
>+class CheckedAlloc {
>+ public:
>+  template <class, int>
>+  friend class CheckedAlloc;
>+
>+  using value_type = T;
>+
>+  CheckedAlloc() {}
>+  explicit CheckedAlloc(size_t id) : id_(id) {}
>+  CheckedAlloc(const CheckedAlloc&) = default;
>+  CheckedAlloc& operator=(const CheckedAlloc&) = default;
>+
>+  template <class U>
>+  CheckedAlloc(const CheckedAlloc<U, Spec>& that)
>+      : id_(that.id_), state_(that.state_) {}
>+
>+  template <class U>
>+  struct rebind {
>+    using other = CheckedAlloc<U, Spec>;
>+  };
>+
>+  using propagate_on_container_copy_assignment =
>+      std::integral_constant<bool, (Spec & kPropagateOnCopy) != 0>;
>+
>+  using propagate_on_container_move_assignment =
>+      std::integral_constant<bool, (Spec & kPropagateOnMove) != 0>;
>+
>+  using propagate_on_container_swap =
>+      std::integral_constant<bool, (Spec & kPropagateOnSwap) != 0>;
>+
>+  CheckedAlloc select_on_container_copy_construction() const {
>+    if (Spec & kPropagateOnCopy) return *this;
>+    return {};
>+  }
>+
>+  T* allocate(size_t n) {
>+    T* ptr = std::allocator<T>().allocate(n);
>+    track_alloc(ptr);
>+    return ptr;
>+  }
>+  void deallocate(T* ptr, size_t n) {
>+    memset(ptr, 0, n * sizeof(T));  // The freed memory must be unpoisoned.
>+    track_dealloc(ptr);
>+    return std::allocator<T>().deallocate(ptr, n);
>+  }
>+
>+  friend bool operator==(const CheckedAlloc& a, const CheckedAlloc& b) {
>+    return a.id_ == b.id_;
>+  }
>+  friend bool operator!=(const CheckedAlloc& a, const CheckedAlloc& b) {
>+    return !(a == b);
>+  }
>+
>+  size_t num_allocs() const { return state_->num_allocs; }
>+
>+  void swap(CheckedAlloc& that) {
>+    using std::swap;
>+    swap(id_, that.id_);
>+    swap(state_, that.state_);
>+  }
>+
>+  friend void swap(CheckedAlloc& a, CheckedAlloc& b) { a.swap(b); }
>+
>+  friend std::ostream& operator<<(std::ostream& o, const CheckedAlloc& a) {
>+    return o << "alloc(" << a.id_ << ")";
>+  }
>+
>+ private:
>+  void track_alloc(void* ptr) {
>+    AllocState* state = state_.get();
>+    ++state->num_allocs;
>+    if (!state->owned.insert(ptr).second)
>+      ADD_FAILURE() << *this << " got previously allocated memory: " << ptr;
>+  }
>+  void track_dealloc(void* ptr) {
>+    if (state_->owned.erase(ptr) != 1)
>+      ADD_FAILURE() << *this
>+                    << " deleting memory owned by another allocator: " << ptr;
>+  }
>+
>+  size_t id_ = std::numeric_limits<size_t>::max();
>+
>+  std::shared_ptr<AllocState> state_ = std::make_shared<AllocState>();
>+};
>+
>+struct Identity {
>+  int32_t operator()(int32_t v) const { return v; }
>+};
>+
>+struct Policy {
>+  using slot_type = Tracked<int32_t>;
>+  using init_type = Tracked<int32_t>;
>+  using key_type = int32_t;
>+
>+  template <class allocator_type, class... Args>
>+  static void construct(allocator_type* alloc, slot_type* slot,
>+                        Args&&... args) {
>+    std::allocator_traits<allocator_type>::construct(
>+        *alloc, slot, std::forward<Args>(args)...);
>+  }
>+
>+  template <class allocator_type>
>+  static void destroy(allocator_type* alloc, slot_type* slot) {
>+    std::allocator_traits<allocator_type>::destroy(*alloc, slot);
>+  }
>+
>+  template <class allocator_type>
>+  static void transfer(allocator_type* alloc, slot_type* new_slot,
>+                       slot_type* old_slot) {
>+    construct(alloc, new_slot, std::move(*old_slot));
>+    destroy(alloc, old_slot);
>+  }
>+
>+  template <class F>
>+  static auto apply(F&& f, int32_t v) -> decltype(std::forward<F>(f)(v, v)) {
>+    return std::forward<F>(f)(v, v);
>+  }
>+
>+  template <class F>
>+  static auto apply(F&& f, const slot_type& v)
>+      -> decltype(std::forward<F>(f)(v.val(), v)) {
>+    return std::forward<F>(f)(v.val(), v);
>+  }
>+
>+  template <class F>
>+  static auto apply(F&& f, slot_type&& v)
>+      -> decltype(std::forward<F>(f)(v.val(), std::move(v))) {
>+    return std::forward<F>(f)(v.val(), std::move(v));
>+  }
>+
>+  static slot_type& element(slot_type* slot) { return *slot; }
>+};
>+
>+template <int Spec>
>+struct PropagateTest : public ::testing::Test {
>+  using Alloc = CheckedAlloc<Tracked<int32_t>, Spec>;
>+
>+  using Table = raw_hash_set<Policy, Identity, std::equal_to<int32_t>, Alloc>;
>+
>+  PropagateTest() {
>+    EXPECT_EQ(a1, t1.get_allocator());
>+    EXPECT_NE(a2, t1.get_allocator());
>+  }
>+
>+  Alloc a1 = Alloc(1);
>+  Table t1 = Table(0, a1);
>+  Alloc a2 = Alloc(2);
>+};
>+
>+using PropagateOnAll =
>+    PropagateTest<kPropagateOnCopy | kPropagateOnMove | kPropagateOnSwap>;
>+using NoPropagateOnCopy = PropagateTest<kPropagateOnMove | kPropagateOnSwap>;
>+using NoPropagateOnMove = PropagateTest<kPropagateOnCopy | kPropagateOnSwap>;
>+
>+TEST_F(PropagateOnAll, Empty) { EXPECT_EQ(0, a1.num_allocs()); }
>+
>+TEST_F(PropagateOnAll, InsertAllocates) {
>+  auto it = t1.insert(0).first;
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, InsertDecomposes) {
>+  auto it = t1.insert(0).first;
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+
>+  EXPECT_FALSE(t1.insert(0).second);
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, RehashMoves) {
>+  auto it = t1.insert(0).first;
>+  EXPECT_EQ(0, it->num_moves());
>+  t1.rehash(2 * t1.capacity());
>+  EXPECT_EQ(2, a1.num_allocs());
>+  it = t1.find(0);
>+  EXPECT_EQ(1, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, CopyConstructor) {
>+  auto it = t1.insert(0).first;
>+  Table u(t1);
>+  EXPECT_EQ(2, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(1, it->num_copies());
>+}
>+
>+TEST_F(NoPropagateOnCopy, CopyConstructor) {
>+  auto it = t1.insert(0).first;
>+  Table u(t1);
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(1, u.get_allocator().num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(1, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, CopyConstructorWithSameAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(t1, a1);
>+  EXPECT_EQ(2, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(1, it->num_copies());
>+}
>+
>+TEST_F(NoPropagateOnCopy, CopyConstructorWithSameAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(t1, a1);
>+  EXPECT_EQ(2, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(1, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, CopyConstructorWithDifferentAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(t1, a2);
>+  EXPECT_EQ(a2, u.get_allocator());
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(1, a2.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(1, it->num_copies());
>+}
>+
>+TEST_F(NoPropagateOnCopy, CopyConstructorWithDifferentAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(t1, a2);
>+  EXPECT_EQ(a2, u.get_allocator());
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(1, a2.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(1, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, MoveConstructor) {
>+  auto it = t1.insert(0).first;
>+  Table u(std::move(t1));
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(NoPropagateOnMove, MoveConstructor) {
>+  auto it = t1.insert(0).first;
>+  Table u(std::move(t1));
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, MoveConstructorWithSameAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(std::move(t1), a1);
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(NoPropagateOnMove, MoveConstructorWithSameAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(std::move(t1), a1);
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, MoveConstructorWithDifferentAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(std::move(t1), a2);
>+  it = u.find(0);
>+  EXPECT_EQ(a2, u.get_allocator());
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(1, a2.num_allocs());
>+  EXPECT_EQ(1, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(NoPropagateOnMove, MoveConstructorWithDifferentAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(std::move(t1), a2);
>+  it = u.find(0);
>+  EXPECT_EQ(a2, u.get_allocator());
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(1, a2.num_allocs());
>+  EXPECT_EQ(1, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, CopyAssignmentWithSameAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(0, a1);
>+  u = t1;
>+  EXPECT_EQ(2, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(1, it->num_copies());
>+}
>+
>+TEST_F(NoPropagateOnCopy, CopyAssignmentWithSameAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(0, a1);
>+  u = t1;
>+  EXPECT_EQ(2, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(1, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, CopyAssignmentWithDifferentAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(0, a2);
>+  u = t1;
>+  EXPECT_EQ(a1, u.get_allocator());
>+  EXPECT_EQ(2, a1.num_allocs());
>+  EXPECT_EQ(0, a2.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(1, it->num_copies());
>+}
>+
>+TEST_F(NoPropagateOnCopy, CopyAssignmentWithDifferentAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(0, a2);
>+  u = t1;
>+  EXPECT_EQ(a2, u.get_allocator());
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(1, a2.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(1, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, MoveAssignmentWithSameAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(0, a1);
>+  u = std::move(t1);
>+  EXPECT_EQ(a1, u.get_allocator());
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(NoPropagateOnMove, MoveAssignmentWithSameAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(0, a1);
>+  u = std::move(t1);
>+  EXPECT_EQ(a1, u.get_allocator());
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, MoveAssignmentWithDifferentAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(0, a2);
>+  u = std::move(t1);
>+  EXPECT_EQ(a1, u.get_allocator());
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(0, a2.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(NoPropagateOnMove, MoveAssignmentWithDifferentAlloc) {
>+  auto it = t1.insert(0).first;
>+  Table u(0, a2);
>+  u = std::move(t1);
>+  it = u.find(0);
>+  EXPECT_EQ(a2, u.get_allocator());
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(1, a2.num_allocs());
>+  EXPECT_EQ(1, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+TEST_F(PropagateOnAll, Swap) {
>+  auto it = t1.insert(0).first;
>+  Table u(0, a2);
>+  u.swap(t1);
>+  EXPECT_EQ(a1, u.get_allocator());
>+  EXPECT_EQ(a2, t1.get_allocator());
>+  EXPECT_EQ(1, a1.num_allocs());
>+  EXPECT_EQ(0, a2.num_allocs());
>+  EXPECT_EQ(0, it->num_moves());
>+  EXPECT_EQ(0, it->num_copies());
>+}
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..cd33a3acb9178888855ff91de94146934c837556
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/raw_hash_set_test.cc
>@@ -0,0 +1,1961 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/raw_hash_set.h"
>+
>+#include <array>
>+#include <cmath>
>+#include <cstdint>
>+#include <deque>
>+#include <functional>
>+#include <memory>
>+#include <numeric>
>+#include <random>
>+#include <string>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/cycleclock.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/container/internal/container_memory.h"
>+#include "absl/container/internal/hash_function_defaults.h"
>+#include "absl/container/internal/hash_policy_testing.h"
>+#include "absl/container/internal/hashtable_debug.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+struct RawHashSetTestOnlyAccess {
>+  template <typename C>
>+  static auto GetSlots(const C& c) -> decltype(c.slots_) {
>+    return c.slots_;
>+  }
>+};
>+
>+namespace {
>+
>+using ::testing::DoubleNear;
>+using ::testing::ElementsAre;
>+using ::testing::Optional;
>+using ::testing::Pair;
>+using ::testing::UnorderedElementsAre;
>+
>+TEST(Util, NormalizeCapacity) {
>+  constexpr size_t kMinCapacity = Group::kWidth - 1;
>+  EXPECT_EQ(kMinCapacity, NormalizeCapacity(0));
>+  EXPECT_EQ(kMinCapacity, NormalizeCapacity(1));
>+  EXPECT_EQ(kMinCapacity, NormalizeCapacity(2));
>+  EXPECT_EQ(kMinCapacity, NormalizeCapacity(kMinCapacity));
>+  EXPECT_EQ(kMinCapacity * 2 + 1, NormalizeCapacity(kMinCapacity + 1));
>+  EXPECT_EQ(kMinCapacity * 2 + 1, NormalizeCapacity(kMinCapacity + 2));
>+}
>+
>+TEST(Util, probe_seq) {
>+  probe_seq<16> seq(0, 127);
>+  auto gen = [&]() {
>+    size_t res = seq.offset();
>+    seq.next();
>+    return res;
>+  };
>+  std::vector<size_t> offsets(8);
>+  std::generate_n(offsets.begin(), 8, gen);
>+  EXPECT_THAT(offsets, ElementsAre(0, 16, 48, 96, 32, 112, 80, 64));
>+  seq = probe_seq<16>(128, 127);
>+  std::generate_n(offsets.begin(), 8, gen);
>+  EXPECT_THAT(offsets, ElementsAre(0, 16, 48, 96, 32, 112, 80, 64));
>+}
>+
>+TEST(BitMask, Smoke) {
>+  EXPECT_FALSE((BitMask<uint8_t, 8>(0)));
>+  EXPECT_TRUE((BitMask<uint8_t, 8>(5)));
>+
>+  EXPECT_THAT((BitMask<uint8_t, 8>(0)), ElementsAre());
>+  EXPECT_THAT((BitMask<uint8_t, 8>(0x1)), ElementsAre(0));
>+  EXPECT_THAT((BitMask<uint8_t, 8>(0x2)), ElementsAre(1));
>+  EXPECT_THAT((BitMask<uint8_t, 8>(0x3)), ElementsAre(0, 1));
>+  EXPECT_THAT((BitMask<uint8_t, 8>(0x4)), ElementsAre(2));
>+  EXPECT_THAT((BitMask<uint8_t, 8>(0x5)), ElementsAre(0, 2));
>+  EXPECT_THAT((BitMask<uint8_t, 8>(0x55)), ElementsAre(0, 2, 4, 6));
>+  EXPECT_THAT((BitMask<uint8_t, 8>(0xAA)), ElementsAre(1, 3, 5, 7));
>+}
>+
>+TEST(BitMask, WithShift) {
>+  // See the non-SSE version of Group for details on what this math is for.
>+  uint64_t ctrl = 0x1716151413121110;
>+  uint64_t hash = 0x12;
>+  constexpr uint64_t msbs = 0x8080808080808080ULL;
>+  constexpr uint64_t lsbs = 0x0101010101010101ULL;
>+  auto x = ctrl ^ (lsbs * hash);
>+  uint64_t mask = (x - lsbs) & ~x & msbs;
>+  EXPECT_EQ(0x0000000080800000, mask);
>+
>+  BitMask<uint64_t, 8, 3> b(mask);
>+  EXPECT_EQ(*b, 2);
>+}
>+
>+TEST(BitMask, LeadingTrailing) {
>+  EXPECT_EQ((BitMask<uint32_t, 16>(0b0001101001000000).LeadingZeros()), 3);
>+  EXPECT_EQ((BitMask<uint32_t, 16>(0b0001101001000000).TrailingZeros()), 6);
>+
>+  EXPECT_EQ((BitMask<uint32_t, 16>(0b0000000000000001).LeadingZeros()), 15);
>+  EXPECT_EQ((BitMask<uint32_t, 16>(0b0000000000000001).TrailingZeros()), 0);
>+
>+  EXPECT_EQ((BitMask<uint32_t, 16>(0b1000000000000000).LeadingZeros()), 0);
>+  EXPECT_EQ((BitMask<uint32_t, 16>(0b1000000000000000).TrailingZeros()), 15);
>+
>+  EXPECT_EQ((BitMask<uint64_t, 8, 3>(0x0000008080808000).LeadingZeros()), 3);
>+  EXPECT_EQ((BitMask<uint64_t, 8, 3>(0x0000008080808000).TrailingZeros()), 1);
>+
>+  EXPECT_EQ((BitMask<uint64_t, 8, 3>(0x0000000000000080).LeadingZeros()), 7);
>+  EXPECT_EQ((BitMask<uint64_t, 8, 3>(0x0000000000000080).TrailingZeros()), 0);
>+
>+  EXPECT_EQ((BitMask<uint64_t, 8, 3>(0x8000000000000000).LeadingZeros()), 0);
>+  EXPECT_EQ((BitMask<uint64_t, 8, 3>(0x8000000000000000).TrailingZeros()), 7);
>+}
>+
>+TEST(Group, EmptyGroup) {
>+  for (h2_t h = 0; h != 128; ++h) EXPECT_FALSE(Group{EmptyGroup()}.Match(h));
>+}
>+
>+#if SWISSTABLE_HAVE_SSE2
>+TEST(Group, Match) {
>+  ctrl_t group[] = {kEmpty, 1, kDeleted, 3, kEmpty, 5, kSentinel, 7,
>+                    7,      5, 3,        1, 1,      1, 1,         1};
>+  EXPECT_THAT(Group{group}.Match(0), ElementsAre());
>+  EXPECT_THAT(Group{group}.Match(1), ElementsAre(1, 11, 12, 13, 14, 15));
>+  EXPECT_THAT(Group{group}.Match(3), ElementsAre(3, 10));
>+  EXPECT_THAT(Group{group}.Match(5), ElementsAre(5, 9));
>+  EXPECT_THAT(Group{group}.Match(7), ElementsAre(7, 8));
>+}
>+
>+TEST(Group, MatchEmpty) {
>+  ctrl_t group[] = {kEmpty, 1, kDeleted, 3, kEmpty, 5, kSentinel, 7,
>+                    7,      5, 3,        1, 1,      1, 1,         1};
>+  EXPECT_THAT(Group{group}.MatchEmpty(), ElementsAre(0, 4));
>+}
>+
>+TEST(Group, MatchEmptyOrDeleted) {
>+  ctrl_t group[] = {kEmpty, 1, kDeleted, 3, kEmpty, 5, kSentinel, 7,
>+                    7,      5, 3,        1, 1,      1, 1,         1};
>+  EXPECT_THAT(Group{group}.MatchEmptyOrDeleted(), ElementsAre(0, 2, 4));
>+}
>+#else
>+TEST(Group, Match) {
>+  ctrl_t group[] = {kEmpty, 1, 2, kDeleted, 2, 1, kSentinel, 1};
>+  EXPECT_THAT(Group{group}.Match(0), ElementsAre());
>+  EXPECT_THAT(Group{group}.Match(1), ElementsAre(1, 5, 7));
>+  EXPECT_THAT(Group{group}.Match(2), ElementsAre(2, 4));
>+}
>+TEST(Group, MatchEmpty) {
>+  ctrl_t group[] = {kEmpty, 1, 2, kDeleted, 2, 1, kSentinel, 1};
>+  EXPECT_THAT(Group{group}.MatchEmpty(), ElementsAre(0));
>+}
>+
>+TEST(Group, MatchEmptyOrDeleted) {
>+  ctrl_t group[] = {kEmpty, 1, 2, kDeleted, 2, 1, kSentinel, 1};
>+  EXPECT_THAT(Group{group}.MatchEmptyOrDeleted(), ElementsAre(0, 3));
>+}
>+#endif
>+
>+TEST(Batch, DropDeletes) {
>+  constexpr size_t kCapacity = 63;
>+  constexpr size_t kGroupWidth = container_internal::Group::kWidth;
>+  std::vector<ctrl_t> ctrl(kCapacity + 1 + kGroupWidth);
>+  ctrl[kCapacity] = kSentinel;
>+  std::vector<ctrl_t> pattern = {kEmpty, 2, kDeleted, 2, kEmpty, 1, kDeleted};
>+  for (size_t i = 0; i != kCapacity; ++i) {
>+    ctrl[i] = pattern[i % pattern.size()];
>+    if (i < kGroupWidth - 1)
>+      ctrl[i + kCapacity + 1] = pattern[i % pattern.size()];
>+  }
>+  ConvertDeletedToEmptyAndFullToDeleted(ctrl.data(), kCapacity);
>+  ASSERT_EQ(ctrl[kCapacity], kSentinel);
>+  for (size_t i = 0; i < kCapacity + 1 + kGroupWidth; ++i) {
>+    ctrl_t expected = pattern[i % (kCapacity + 1) % pattern.size()];
>+    if (i == kCapacity) expected = kSentinel;
>+    if (expected == kDeleted) expected = kEmpty;
>+    if (IsFull(expected)) expected = kDeleted;
>+    EXPECT_EQ(ctrl[i], expected)
>+        << i << " " << int{pattern[i % pattern.size()]};
>+  }
>+}
>+
>+TEST(Group, CountLeadingEmptyOrDeleted) {
>+  const std::vector<ctrl_t> empty_examples = {kEmpty, kDeleted};
>+  const std::vector<ctrl_t> full_examples = {0, 1, 2, 3, 5, 9, 127, kSentinel};
>+
>+  for (ctrl_t empty : empty_examples) {
>+    std::vector<ctrl_t> e(Group::kWidth, empty);
>+    EXPECT_EQ(Group::kWidth, Group{e.data()}.CountLeadingEmptyOrDeleted());
>+    for (ctrl_t full : full_examples) {
>+      for (size_t i = 0; i != Group::kWidth; ++i) {
>+        std::vector<ctrl_t> f(Group::kWidth, empty);
>+        f[i] = full;
>+        EXPECT_EQ(i, Group{f.data()}.CountLeadingEmptyOrDeleted());
>+      }
>+      std::vector<ctrl_t> f(Group::kWidth, empty);
>+      f[Group::kWidth * 2 / 3] = full;
>+      f[Group::kWidth / 2] = full;
>+      EXPECT_EQ(
>+          Group::kWidth / 2, Group{f.data()}.CountLeadingEmptyOrDeleted());
>+    }
>+  }
>+}
>+
>+struct IntPolicy {
>+  using slot_type = int64_t;
>+  using key_type = int64_t;
>+  using init_type = int64_t;
>+
>+  static void construct(void*, int64_t* slot, int64_t v) { *slot = v; }
>+  static void destroy(void*, int64_t*) {}
>+  static void transfer(void*, int64_t* new_slot, int64_t* old_slot) {
>+    *new_slot = *old_slot;
>+  }
>+
>+  static int64_t& element(slot_type* slot) { return *slot; }
>+
>+  template <class F>
>+  static auto apply(F&& f, int64_t x) -> decltype(std::forward<F>(f)(x, x)) {
>+    return std::forward<F>(f)(x, x);
>+  }
>+};
>+
>+class StringPolicy {
>+  template <class F, class K, class V,
>+            class = typename std::enable_if<
>+                std::is_convertible<const K&, absl::string_view>::value>::type>
>+  decltype(std::declval<F>()(
>+      std::declval<const absl::string_view&>(), std::piecewise_construct,
>+      std::declval<std::tuple<K>>(),
>+      std::declval<V>())) static apply_impl(F&& f,
>+                                            std::pair<std::tuple<K>, V> p) {
>+    const absl::string_view& key = std::get<0>(p.first);
>+    return std::forward<F>(f)(key, std::piecewise_construct, std::move(p.first),
>+                              std::move(p.second));
>+  }
>+
>+ public:
>+  struct slot_type {
>+    struct ctor {};
>+
>+    template <class... Ts>
>+    slot_type(ctor, Ts&&... ts) : pair(std::forward<Ts>(ts)...) {}
>+
>+    std::pair<std::string, std::string> pair;
>+  };
>+
>+  using key_type = std::string;
>+  using init_type = std::pair<std::string, std::string>;
>+
>+  template <class allocator_type, class... Args>
>+  static void construct(allocator_type* alloc, slot_type* slot, Args... args) {
>+    std::allocator_traits<allocator_type>::construct(
>+        *alloc, slot, typename slot_type::ctor(), std::forward<Args>(args)...);
>+  }
>+
>+  template <class allocator_type>
>+  static void destroy(allocator_type* alloc, slot_type* slot) {
>+    std::allocator_traits<allocator_type>::destroy(*alloc, slot);
>+  }
>+
>+  template <class allocator_type>
>+  static void transfer(allocator_type* alloc, slot_type* new_slot,
>+                       slot_type* old_slot) {
>+    construct(alloc, new_slot, std::move(old_slot->pair));
>+    destroy(alloc, old_slot);
>+  }
>+
>+  static std::pair<std::string, std::string>& element(slot_type* slot) {
>+    return slot->pair;
>+  }
>+
>+  template <class F, class... Args>
>+  static auto apply(F&& f, Args&&... args)
>+      -> decltype(apply_impl(std::forward<F>(f),
>+                             PairArgs(std::forward<Args>(args)...))) {
>+    return apply_impl(std::forward<F>(f),
>+                      PairArgs(std::forward<Args>(args)...));
>+  }
>+};
>+
>+struct StringHash : absl::Hash<absl::string_view> {
>+  using is_transparent = void;
>+};
>+struct StringEq : std::equal_to<absl::string_view> {
>+  using is_transparent = void;
>+};
>+
>+struct StringTable
>+    : raw_hash_set<StringPolicy, StringHash, StringEq, std::allocator<int>> {
>+  using Base = typename StringTable::raw_hash_set;
>+  StringTable() {}
>+  using Base::Base;
>+};
>+
>+struct IntTable
>+    : raw_hash_set<IntPolicy, container_internal::hash_default_hash<int64_t>,
>+                   std::equal_to<int64_t>, std::allocator<int64_t>> {
>+  using Base = typename IntTable::raw_hash_set;
>+  IntTable() {}
>+  using Base::Base;
>+};
>+
>+struct BadFastHash {
>+  template <class T>
>+  size_t operator()(const T&) const {
>+    return 0;
>+  }
>+};
>+
>+struct BadTable : raw_hash_set<IntPolicy, BadFastHash, std::equal_to<int>,
>+                               std::allocator<int>> {
>+  using Base = typename BadTable::raw_hash_set;
>+  BadTable() {}
>+  using Base::Base;
>+};
>+
>+TEST(Table, EmptyFunctorOptimization) {
>+  static_assert(std::is_empty<std::equal_to<absl::string_view>>::value, "");
>+  static_assert(std::is_empty<std::allocator<int>>::value, "");
>+
>+  struct MockTable {
>+    void* ctrl;
>+    void* slots;
>+    size_t size;
>+    size_t capacity;
>+    size_t growth_left;
>+  };
>+  struct StatelessHash {
>+    size_t operator()(absl::string_view) const { return 0; }
>+  };
>+  struct StatefulHash : StatelessHash {
>+    size_t dummy;
>+  };
>+
>+  EXPECT_EQ(
>+      sizeof(MockTable),
>+      sizeof(
>+          raw_hash_set<StringPolicy, StatelessHash,
>+                       std::equal_to<absl::string_view>, std::allocator<int>>));
>+
>+  EXPECT_EQ(
>+      sizeof(MockTable) + sizeof(StatefulHash),
>+      sizeof(
>+          raw_hash_set<StringPolicy, StatefulHash,
>+                       std::equal_to<absl::string_view>, std::allocator<int>>));
>+}
>+
>+TEST(Table, Empty) {
>+  IntTable t;
>+  EXPECT_EQ(0, t.size());
>+  EXPECT_TRUE(t.empty());
>+}
>+
>+#ifdef __GNUC__
>+template <class T>
>+ABSL_ATTRIBUTE_ALWAYS_INLINE inline void DoNotOptimize(const T& v) {
>+  asm volatile("" : : "r,m"(v) : "memory");
>+}
>+#endif
>+
>+TEST(Table, Prefetch) {
>+  IntTable t;
>+  t.emplace(1);
>+  // Works for both present and absent keys.
>+  t.prefetch(1);
>+  t.prefetch(2);
>+
>+  // Do not run in debug mode, when prefetch is not implemented, or when
>+  // sanitizers are enabled.
>+#if defined(NDEBUG) && defined(__GNUC__) && !defined(ADDRESS_SANITIZER) && \
>+    !defined(MEMORY_SANITIZER) && !defined(THREAD_SANITIZER) &&            \
>+    !defined(UNDEFINED_BEHAVIOR_SANITIZER)
>+  const auto now = [] { return absl::base_internal::CycleClock::Now(); };
>+
>+  static constexpr int size = 1000000;
>+  for (int i = 0; i < size; ++i) t.insert(i);
>+
>+  int64_t no_prefetch = 0, prefetch = 0;
>+  for (int iter = 0; iter < 10; ++iter) {
>+    int64_t time = now();
>+    for (int i = 0; i < size; ++i) {
>+      DoNotOptimize(t.find(i));
>+    }
>+    no_prefetch += now() - time;
>+
>+    time = now();
>+    for (int i = 0; i < size; ++i) {
>+      t.prefetch(i + 20);
>+      DoNotOptimize(t.find(i));
>+    }
>+    prefetch += now() - time;
>+  }
>+
>+  // no_prefetch is at least 30% slower.
>+  EXPECT_GE(1.0 * no_prefetch / prefetch, 1.3);
>+#endif
>+}
>+
>+TEST(Table, LookupEmpty) {
>+  IntTable t;
>+  auto it = t.find(0);
>+  EXPECT_TRUE(it == t.end());
>+}
>+
>+TEST(Table, Insert1) {
>+  IntTable t;
>+  EXPECT_TRUE(t.find(0) == t.end());
>+  auto res = t.emplace(0);
>+  EXPECT_TRUE(res.second);
>+  EXPECT_THAT(*res.first, 0);
>+  EXPECT_EQ(1, t.size());
>+  EXPECT_THAT(*t.find(0), 0);
>+}
>+
>+TEST(Table, Insert2) {
>+  IntTable t;
>+  EXPECT_TRUE(t.find(0) == t.end());
>+  auto res = t.emplace(0);
>+  EXPECT_TRUE(res.second);
>+  EXPECT_THAT(*res.first, 0);
>+  EXPECT_EQ(1, t.size());
>+  EXPECT_TRUE(t.find(1) == t.end());
>+  res = t.emplace(1);
>+  EXPECT_TRUE(res.second);
>+  EXPECT_THAT(*res.first, 1);
>+  EXPECT_EQ(2, t.size());
>+  EXPECT_THAT(*t.find(0), 0);
>+  EXPECT_THAT(*t.find(1), 1);
>+}
>+
>+TEST(Table, InsertCollision) {
>+  BadTable t;
>+  EXPECT_TRUE(t.find(1) == t.end());
>+  auto res = t.emplace(1);
>+  EXPECT_TRUE(res.second);
>+  EXPECT_THAT(*res.first, 1);
>+  EXPECT_EQ(1, t.size());
>+
>+  EXPECT_TRUE(t.find(2) == t.end());
>+  res = t.emplace(2);
>+  EXPECT_THAT(*res.first, 2);
>+  EXPECT_TRUE(res.second);
>+  EXPECT_EQ(2, t.size());
>+
>+  EXPECT_THAT(*t.find(1), 1);
>+  EXPECT_THAT(*t.find(2), 2);
>+}
>+
>+// Test that we do not add existent element in case we need to search through
>+// many groups with deleted elements
>+TEST(Table, InsertCollisionAndFindAfterDelete) {
>+  BadTable t;  // all elements go to the same group.
>+  // Have at least 2 groups with Group::kWidth collisions
>+  // plus some extra collisions in the last group.
>+  constexpr size_t kNumInserts = Group::kWidth * 2 + 5;
>+  for (size_t i = 0; i < kNumInserts; ++i) {
>+    auto res = t.emplace(i);
>+    EXPECT_TRUE(res.second);
>+    EXPECT_THAT(*res.first, i);
>+    EXPECT_EQ(i + 1, t.size());
>+  }
>+
>+  // Remove elements one by one and check
>+  // that we still can find all other elements.
>+  for (size_t i = 0; i < kNumInserts; ++i) {
>+    EXPECT_EQ(1, t.erase(i)) << i;
>+    for (size_t j = i + 1; j < kNumInserts; ++j) {
>+      EXPECT_THAT(*t.find(j), j);
>+      auto res = t.emplace(j);
>+      EXPECT_FALSE(res.second) << i << " " << j;
>+      EXPECT_THAT(*res.first, j);
>+      EXPECT_EQ(kNumInserts - i - 1, t.size());
>+    }
>+  }
>+  EXPECT_TRUE(t.empty());
>+}
>+
>+TEST(Table, LazyEmplace) {
>+  StringTable t;
>+  bool called = false;
>+  auto it = t.lazy_emplace("abc", [&](const StringTable::constructor& f) {
>+    called = true;
>+    f("abc", "ABC");
>+  });
>+  EXPECT_TRUE(called);
>+  EXPECT_THAT(*it, Pair("abc", "ABC"));
>+  called = false;
>+  it = t.lazy_emplace("abc", [&](const StringTable::constructor& f) {
>+    called = true;
>+    f("abc", "DEF");
>+  });
>+  EXPECT_FALSE(called);
>+  EXPECT_THAT(*it, Pair("abc", "ABC"));
>+}
>+
>+TEST(Table, ContainsEmpty) {
>+  IntTable t;
>+
>+  EXPECT_FALSE(t.contains(0));
>+}
>+
>+TEST(Table, Contains1) {
>+  IntTable t;
>+
>+  EXPECT_TRUE(t.insert(0).second);
>+  EXPECT_TRUE(t.contains(0));
>+  EXPECT_FALSE(t.contains(1));
>+
>+  EXPECT_EQ(1, t.erase(0));
>+  EXPECT_FALSE(t.contains(0));
>+}
>+
>+TEST(Table, Contains2) {
>+  IntTable t;
>+
>+  EXPECT_TRUE(t.insert(0).second);
>+  EXPECT_TRUE(t.contains(0));
>+  EXPECT_FALSE(t.contains(1));
>+
>+  t.clear();
>+  EXPECT_FALSE(t.contains(0));
>+}
>+
>+int decompose_constructed;
>+struct DecomposeType {
>+  DecomposeType(int i) : i(i) {  // NOLINT
>+    ++decompose_constructed;
>+  }
>+
>+  explicit DecomposeType(const char* d) : DecomposeType(*d) {}
>+
>+  int i;
>+};
>+
>+struct DecomposeHash {
>+  using is_transparent = void;
>+  size_t operator()(DecomposeType a) const { return a.i; }
>+  size_t operator()(int a) const { return a; }
>+  size_t operator()(const char* a) const { return *a; }
>+};
>+
>+struct DecomposeEq {
>+  using is_transparent = void;
>+  bool operator()(DecomposeType a, DecomposeType b) const { return a.i == b.i; }
>+  bool operator()(DecomposeType a, int b) const { return a.i == b; }
>+  bool operator()(DecomposeType a, const char* b) const { return a.i == *b; }
>+};
>+
>+struct DecomposePolicy {
>+  using slot_type = DecomposeType;
>+  using key_type = DecomposeType;
>+  using init_type = DecomposeType;
>+
>+  template <typename T>
>+  static void construct(void*, DecomposeType* slot, T&& v) {
>+    *slot = DecomposeType(std::forward<T>(v));
>+  }
>+  static void destroy(void*, DecomposeType*) {}
>+  static DecomposeType& element(slot_type* slot) { return *slot; }
>+
>+  template <class F, class T>
>+  static auto apply(F&& f, const T& x) -> decltype(std::forward<F>(f)(x, x)) {
>+    return std::forward<F>(f)(x, x);
>+  }
>+};
>+
>+template <typename Hash, typename Eq>
>+void TestDecompose(bool construct_three) {
>+  DecomposeType elem{0};
>+  const int one = 1;
>+  const char* three_p = "3";
>+  const auto& three = three_p;
>+
>+  raw_hash_set<DecomposePolicy, Hash, Eq, std::allocator<int>> set1;
>+
>+  decompose_constructed = 0;
>+  int expected_constructed = 0;
>+  EXPECT_EQ(expected_constructed, decompose_constructed);
>+  set1.insert(elem);
>+  EXPECT_EQ(expected_constructed, decompose_constructed);
>+  set1.insert(1);
>+  EXPECT_EQ(++expected_constructed, decompose_constructed);
>+  set1.emplace("3");
>+  EXPECT_EQ(++expected_constructed, decompose_constructed);
>+  EXPECT_EQ(expected_constructed, decompose_constructed);
>+
>+  {  // insert(T&&)
>+    set1.insert(1);
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+  }
>+
>+  {  // insert(const T&)
>+    set1.insert(one);
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+  }
>+
>+  {  // insert(hint, T&&)
>+    set1.insert(set1.begin(), 1);
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+  }
>+
>+  {  // insert(hint, const T&)
>+    set1.insert(set1.begin(), one);
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+  }
>+
>+  {  // emplace(...)
>+    set1.emplace(1);
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+    set1.emplace("3");
>+    expected_constructed += construct_three;
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+    set1.emplace(one);
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+    set1.emplace(three);
>+    expected_constructed += construct_three;
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+  }
>+
>+  {  // emplace_hint(...)
>+    set1.emplace_hint(set1.begin(), 1);
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+    set1.emplace_hint(set1.begin(), "3");
>+    expected_constructed += construct_three;
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+    set1.emplace_hint(set1.begin(), one);
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+    set1.emplace_hint(set1.begin(), three);
>+    expected_constructed += construct_three;
>+    EXPECT_EQ(expected_constructed, decompose_constructed);
>+  }
>+}
>+
>+TEST(Table, Decompose) {
>+  TestDecompose<DecomposeHash, DecomposeEq>(false);
>+
>+  struct TransparentHashIntOverload {
>+    size_t operator()(DecomposeType a) const { return a.i; }
>+    size_t operator()(int a) const { return a; }
>+  };
>+  struct TransparentEqIntOverload {
>+    bool operator()(DecomposeType a, DecomposeType b) const {
>+      return a.i == b.i;
>+    }
>+    bool operator()(DecomposeType a, int b) const { return a.i == b; }
>+  };
>+  TestDecompose<TransparentHashIntOverload, DecomposeEq>(true);
>+  TestDecompose<TransparentHashIntOverload, TransparentEqIntOverload>(true);
>+  TestDecompose<DecomposeHash, TransparentEqIntOverload>(true);
>+}
>+
>+// Returns the largest m such that a table with m elements has the same number
>+// of buckets as a table with n elements.
>+size_t MaxDensitySize(size_t n) {
>+  IntTable t;
>+  t.reserve(n);
>+  for (size_t i = 0; i != n; ++i) t.emplace(i);
>+  const size_t c = t.bucket_count();
>+  while (c == t.bucket_count()) t.emplace(n++);
>+  return t.size() - 1;
>+}
>+
>+struct Modulo1000Hash {
>+  size_t operator()(int x) const { return x % 1000; }
>+};
>+
>+struct Modulo1000HashTable
>+    : public raw_hash_set<IntPolicy, Modulo1000Hash, std::equal_to<int>,
>+                          std::allocator<int>> {};
>+
>+// Test that rehash with no resize happen in case of many deleted slots.
>+TEST(Table, RehashWithNoResize) {
>+  Modulo1000HashTable t;
>+  // Adding the same length (and the same hash) strings
>+  // to have at least kMinFullGroups groups
>+  // with Group::kWidth collisions. Then feel upto MaxDensitySize;
>+  const size_t kMinFullGroups = 7;
>+  std::vector<int> keys;
>+  for (size_t i = 0; i < MaxDensitySize(Group::kWidth * kMinFullGroups); ++i) {
>+    int k = i * 1000;
>+    t.emplace(k);
>+    keys.push_back(k);
>+  }
>+  const size_t capacity = t.capacity();
>+
>+  // Remove elements from all groups except the first and the last one.
>+  // All elements removed from full groups will be marked as kDeleted.
>+  const size_t erase_begin = Group::kWidth / 2;
>+  const size_t erase_end = (t.size() / Group::kWidth - 1) * Group::kWidth;
>+  for (size_t i = erase_begin; i < erase_end; ++i) {
>+    EXPECT_EQ(1, t.erase(keys[i])) << i;
>+  }
>+  keys.erase(keys.begin() + erase_begin, keys.begin() + erase_end);
>+
>+  auto last_key = keys.back();
>+  size_t last_key_num_probes = GetHashtableDebugNumProbes(t, last_key);
>+
>+  // Make sure that we have to make a lot of probes for last key.
>+  ASSERT_GT(last_key_num_probes, kMinFullGroups);
>+
>+  int x = 1;
>+  // Insert and erase one element, before inplace rehash happen.
>+  while (last_key_num_probes == GetHashtableDebugNumProbes(t, last_key)) {
>+    t.emplace(x);
>+    ASSERT_EQ(capacity, t.capacity());
>+    // All elements should be there.
>+    ASSERT_TRUE(t.find(x) != t.end()) << x;
>+    for (const auto& k : keys) {
>+      ASSERT_TRUE(t.find(k) != t.end()) << k;
>+    }
>+    t.erase(x);
>+    ++x;
>+  }
>+}
>+
>+TEST(Table, InsertEraseStressTest) {
>+  IntTable t;
>+  const size_t kMinElementCount = 250;
>+  std::deque<int> keys;
>+  size_t i = 0;
>+  for (; i < MaxDensitySize(kMinElementCount); ++i) {
>+    t.emplace(i);
>+    keys.push_back(i);
>+  }
>+  const size_t kNumIterations = 1000000;
>+  for (; i < kNumIterations; ++i) {
>+    ASSERT_EQ(1, t.erase(keys.front()));
>+    keys.pop_front();
>+    t.emplace(i);
>+    keys.push_back(i);
>+  }
>+}
>+
>+TEST(Table, InsertOverloads) {
>+  StringTable t;
>+  // These should all trigger the insert(init_type) overload.
>+  t.insert({{}, {}});
>+  t.insert({"ABC", {}});
>+  t.insert({"DEF", "!!!"});
>+
>+  EXPECT_THAT(t, UnorderedElementsAre(Pair("", ""), Pair("ABC", ""),
>+                                      Pair("DEF", "!!!")));
>+}
>+
>+TEST(Table, LargeTable) {
>+  IntTable t;
>+  for (int64_t i = 0; i != 100000; ++i) t.emplace(i << 40);
>+  for (int64_t i = 0; i != 100000; ++i) ASSERT_EQ(i << 40, *t.find(i << 40));
>+}
>+
>+// Timeout if copy is quadratic as it was in Rust.
>+TEST(Table, EnsureNonQuadraticAsInRust) {
>+  static const size_t kLargeSize = 1 << 15;
>+
>+  IntTable t;
>+  for (size_t i = 0; i != kLargeSize; ++i) {
>+    t.insert(i);
>+  }
>+
>+  // If this is quadratic, the test will timeout.
>+  IntTable t2;
>+  for (const auto& entry : t) t2.insert(entry);
>+}
>+
>+TEST(Table, ClearBug) {
>+  IntTable t;
>+  constexpr size_t capacity = container_internal::Group::kWidth - 1;
>+  constexpr size_t max_size = capacity / 2;
>+  for (size_t i = 0; i < max_size; ++i) {
>+    t.insert(i);
>+  }
>+  ASSERT_EQ(capacity, t.capacity());
>+  intptr_t original = reinterpret_cast<intptr_t>(&*t.find(2));
>+  t.clear();
>+  ASSERT_EQ(capacity, t.capacity());
>+  for (size_t i = 0; i < max_size; ++i) {
>+    t.insert(i);
>+  }
>+  ASSERT_EQ(capacity, t.capacity());
>+  intptr_t second = reinterpret_cast<intptr_t>(&*t.find(2));
>+  // We are checking that original and second are close enough to each other
>+  // that they are probably still in the same group.  This is not strictly
>+  // guaranteed.
>+  EXPECT_LT(std::abs(original - second),
>+            capacity * sizeof(IntTable::value_type));
>+}
>+
>+TEST(Table, Erase) {
>+  IntTable t;
>+  EXPECT_TRUE(t.find(0) == t.end());
>+  auto res = t.emplace(0);
>+  EXPECT_TRUE(res.second);
>+  EXPECT_EQ(1, t.size());
>+  t.erase(res.first);
>+  EXPECT_EQ(0, t.size());
>+  EXPECT_TRUE(t.find(0) == t.end());
>+}
>+
>+// Collect N bad keys by following algorithm:
>+// 1. Create an empty table and reserve it to 2 * N.
>+// 2. Insert N random elements.
>+// 3. Take first Group::kWidth - 1 to bad_keys array.
>+// 4. Clear the table without resize.
>+// 5. Go to point 2 while N keys not collected
>+std::vector<int64_t> CollectBadMergeKeys(size_t N) {
>+  static constexpr int kGroupSize = Group::kWidth - 1;
>+
>+  auto topk_range = [](size_t b, size_t e, IntTable* t) -> std::vector<int64_t> {
>+    for (size_t i = b; i != e; ++i) {
>+      t->emplace(i);
>+    }
>+    std::vector<int64_t> res;
>+    res.reserve(kGroupSize);
>+    auto it = t->begin();
>+    for (size_t i = b; i != e && i != b + kGroupSize; ++i, ++it) {
>+      res.push_back(*it);
>+    }
>+    return res;
>+  };
>+
>+  std::vector<int64_t> bad_keys;
>+  bad_keys.reserve(N);
>+  IntTable t;
>+  t.reserve(N * 2);
>+
>+  for (size_t b = 0; bad_keys.size() < N; b += N) {
>+    auto keys = topk_range(b, b + N, &t);
>+    bad_keys.insert(bad_keys.end(), keys.begin(), keys.end());
>+    t.erase(t.begin(), t.end());
>+    EXPECT_TRUE(t.empty());
>+  }
>+  return bad_keys;
>+}
>+
>+struct ProbeStats {
>+  // Number of elements with specific probe length over all tested tables.
>+  std::vector<size_t> all_probes_histogram;
>+  // Ratios total_probe_length/size for every tested table.
>+  std::vector<double> single_table_ratios;
>+
>+  friend ProbeStats operator+(const ProbeStats& a, const ProbeStats& b) {
>+    ProbeStats res = a;
>+    res.all_probes_histogram.resize(std::max(res.all_probes_histogram.size(),
>+                                             b.all_probes_histogram.size()));
>+    std::transform(b.all_probes_histogram.begin(), b.all_probes_histogram.end(),
>+                   res.all_probes_histogram.begin(),
>+                   res.all_probes_histogram.begin(), std::plus<size_t>());
>+    res.single_table_ratios.insert(res.single_table_ratios.end(),
>+                                   b.single_table_ratios.begin(),
>+                                   b.single_table_ratios.end());
>+    return res;
>+  }
>+
>+  // Average ratio total_probe_length/size over tables.
>+  double AvgRatio() const {
>+    return std::accumulate(single_table_ratios.begin(),
>+                           single_table_ratios.end(), 0.0) /
>+           single_table_ratios.size();
>+  }
>+
>+  // Maximum ratio total_probe_length/size over tables.
>+  double MaxRatio() const {
>+    return *std::max_element(single_table_ratios.begin(),
>+                             single_table_ratios.end());
>+  }
>+
>+  // Percentile ratio total_probe_length/size over tables.
>+  double PercentileRatio(double Percentile = 0.95) const {
>+    auto r = single_table_ratios;
>+    auto mid = r.begin() + static_cast<size_t>(r.size() * Percentile);
>+    if (mid != r.end()) {
>+      std::nth_element(r.begin(), mid, r.end());
>+      return *mid;
>+    } else {
>+      return MaxRatio();
>+    }
>+  }
>+
>+  // Maximum probe length over all elements and all tables.
>+  size_t MaxProbe() const { return all_probes_histogram.size(); }
>+
>+  // Fraction of elements with specified probe length.
>+  std::vector<double> ProbeNormalizedHistogram() const {
>+    double total_elements = std::accumulate(all_probes_histogram.begin(),
>+                                            all_probes_histogram.end(), 0ull);
>+    std::vector<double> res;
>+    for (size_t p : all_probes_histogram) {
>+      res.push_back(p / total_elements);
>+    }
>+    return res;
>+  }
>+
>+  size_t PercentileProbe(double Percentile = 0.99) const {
>+    size_t idx = 0;
>+    for (double p : ProbeNormalizedHistogram()) {
>+      if (Percentile > p) {
>+        Percentile -= p;
>+        ++idx;
>+      } else {
>+        return idx;
>+      }
>+    }
>+    return idx;
>+  }
>+
>+  friend std::ostream& operator<<(std::ostream& out, const ProbeStats& s) {
>+    out << "{AvgRatio:" << s.AvgRatio() << ", MaxRatio:" << s.MaxRatio()
>+        << ", PercentileRatio:" << s.PercentileRatio()
>+        << ", MaxProbe:" << s.MaxProbe() << ", Probes=[";
>+    for (double p : s.ProbeNormalizedHistogram()) {
>+      out << p << ",";
>+    }
>+    out << "]}";
>+
>+    return out;
>+  }
>+};
>+
>+struct ExpectedStats {
>+  double avg_ratio;
>+  double max_ratio;
>+  std::vector<std::pair<double, double>> pecentile_ratios;
>+  std::vector<std::pair<double, double>> pecentile_probes;
>+
>+  friend std::ostream& operator<<(std::ostream& out, const ExpectedStats& s) {
>+    out << "{AvgRatio:" << s.avg_ratio << ", MaxRatio:" << s.max_ratio
>+        << ", PercentileRatios: [";
>+    for (auto el : s.pecentile_ratios) {
>+      out << el.first << ":" << el.second << ", ";
>+    }
>+    out << "], PercentileProbes: [";
>+    for (auto el : s.pecentile_probes) {
>+      out << el.first << ":" << el.second << ", ";
>+    }
>+    out << "]}";
>+
>+    return out;
>+  }
>+};
>+
>+void VerifyStats(size_t size, const ExpectedStats& exp,
>+                 const ProbeStats& stats) {
>+  EXPECT_LT(stats.AvgRatio(), exp.avg_ratio) << size << " " << stats;
>+  EXPECT_LT(stats.MaxRatio(), exp.max_ratio) << size << " " << stats;
>+  for (auto pr : exp.pecentile_ratios) {
>+    EXPECT_LE(stats.PercentileRatio(pr.first), pr.second)
>+        << size << " " << pr.first << " " << stats;
>+  }
>+
>+  for (auto pr : exp.pecentile_probes) {
>+    EXPECT_LE(stats.PercentileProbe(pr.first), pr.second)
>+        << size << " " << pr.first << " " << stats;
>+  }
>+}
>+
>+using ProbeStatsPerSize = std::map<size_t, ProbeStats>;
>+
>+// Collect total ProbeStats on num_iters iterations of the following algorithm:
>+// 1. Create new table and reserve it to keys.size() * 2
>+// 2. Insert all keys xored with seed
>+// 3. Collect ProbeStats from final table.
>+ProbeStats CollectProbeStatsOnKeysXoredWithSeed(const std::vector<int64_t>& keys,
>+                                                size_t num_iters) {
>+  const size_t reserve_size = keys.size() * 2;
>+
>+  ProbeStats stats;
>+
>+  int64_t seed = 0x71b1a19b907d6e33;
>+  while (num_iters--) {
>+    seed = static_cast<int64_t>(static_cast<uint64_t>(seed) * 17 + 13);
>+    IntTable t1;
>+    t1.reserve(reserve_size);
>+    for (const auto& key : keys) {
>+      t1.emplace(key ^ seed);
>+    }
>+
>+    auto probe_histogram = GetHashtableDebugNumProbesHistogram(t1);
>+    stats.all_probes_histogram.resize(
>+        std::max(stats.all_probes_histogram.size(), probe_histogram.size()));
>+    std::transform(probe_histogram.begin(), probe_histogram.end(),
>+                   stats.all_probes_histogram.begin(),
>+                   stats.all_probes_histogram.begin(), std::plus<size_t>());
>+
>+    size_t total_probe_seq_length = 0;
>+    for (size_t i = 0; i < probe_histogram.size(); ++i) {
>+      total_probe_seq_length += i * probe_histogram[i];
>+    }
>+    stats.single_table_ratios.push_back(total_probe_seq_length * 1.0 /
>+                                        keys.size());
>+    t1.erase(t1.begin(), t1.end());
>+  }
>+  return stats;
>+}
>+
>+ExpectedStats XorSeedExpectedStats() {
>+  constexpr bool kRandomizesInserts =
>+#if NDEBUG
>+      false;
>+#else   // NDEBUG
>+      true;
>+#endif  // NDEBUG
>+
>+  // The effective load factor is larger in non-opt mode because we insert
>+  // elements out of order.
>+  switch (container_internal::Group::kWidth) {
>+    case 8:
>+      if (kRandomizesInserts) {
>+  return {0.05,
>+          1.0,
>+          {{0.95, 0.5}},
>+          {{0.95, 0}, {0.99, 2}, {0.999, 4}, {0.9999, 10}}};
>+      } else {
>+  return {0.05,
>+          2.0,
>+          {{0.95, 0.1}},
>+          {{0.95, 0}, {0.99, 2}, {0.999, 4}, {0.9999, 10}}};
>+      }
>+      break;
>+    case 16:
>+      if (kRandomizesInserts) {
>+        return {0.1,
>+                1.0,
>+                {{0.95, 0.1}},
>+                {{0.95, 0}, {0.99, 1}, {0.999, 8}, {0.9999, 15}}};
>+      } else {
>+        return {0.05,
>+                1.0,
>+                {{0.95, 0.05}},
>+                {{0.95, 0}, {0.99, 1}, {0.999, 4}, {0.9999, 10}}};
>+      }
>+      break;
>+    default:
>+      ABSL_RAW_LOG(FATAL, "%s", "Unknown Group width");
>+  }
>+  return {};
>+}
>+TEST(Table, DISABLED_EnsureNonQuadraticTopNXorSeedByProbeSeqLength) {
>+  ProbeStatsPerSize stats;
>+  std::vector<size_t> sizes = {Group::kWidth << 5, Group::kWidth << 10};
>+  for (size_t size : sizes) {
>+    stats[size] =
>+        CollectProbeStatsOnKeysXoredWithSeed(CollectBadMergeKeys(size), 200);
>+  }
>+  auto expected = XorSeedExpectedStats();
>+  for (size_t size : sizes) {
>+    auto& stat = stats[size];
>+    VerifyStats(size, expected, stat);
>+  }
>+}
>+
>+// Collect total ProbeStats on num_iters iterations of the following algorithm:
>+// 1. Create new table
>+// 2. Select 10% of keys and insert 10 elements key * 17 + j * 13
>+// 3. Collect ProbeStats from final table
>+ProbeStats CollectProbeStatsOnLinearlyTransformedKeys(
>+    const std::vector<int64_t>& keys, size_t num_iters) {
>+  ProbeStats stats;
>+
>+  std::random_device rd;
>+  std::mt19937 rng(rd());
>+  auto linear_transform = [](size_t x, size_t y) { return x * 17 + y * 13; };
>+  std::uniform_int_distribution<size_t> dist(0, keys.size()-1);
>+  while (num_iters--) {
>+    IntTable t1;
>+    size_t num_keys = keys.size() / 10;
>+    size_t start = dist(rng);
>+    for (size_t i = 0; i != num_keys; ++i) {
>+      for (size_t j = 0; j != 10; ++j) {
>+        t1.emplace(linear_transform(keys[(i + start) % keys.size()], j));
>+      }
>+    }
>+
>+    auto probe_histogram = GetHashtableDebugNumProbesHistogram(t1);
>+    stats.all_probes_histogram.resize(
>+        std::max(stats.all_probes_histogram.size(), probe_histogram.size()));
>+    std::transform(probe_histogram.begin(), probe_histogram.end(),
>+                   stats.all_probes_histogram.begin(),
>+                   stats.all_probes_histogram.begin(), std::plus<size_t>());
>+
>+    size_t total_probe_seq_length = 0;
>+    for (size_t i = 0; i < probe_histogram.size(); ++i) {
>+      total_probe_seq_length += i * probe_histogram[i];
>+    }
>+    stats.single_table_ratios.push_back(total_probe_seq_length * 1.0 /
>+                                        t1.size());
>+    t1.erase(t1.begin(), t1.end());
>+  }
>+  return stats;
>+}
>+
>+ExpectedStats LinearTransformExpectedStats() {
>+  constexpr bool kRandomizesInserts =
>+#if NDEBUG
>+      false;
>+#else   // NDEBUG
>+      true;
>+#endif  // NDEBUG
>+
>+  // The effective load factor is larger in non-opt mode because we insert
>+  // elements out of order.
>+  switch (container_internal::Group::kWidth) {
>+    case 8:
>+      if (kRandomizesInserts) {
>+        return {0.1,
>+                0.5,
>+                {{0.95, 0.3}},
>+                {{0.95, 0}, {0.99, 1}, {0.999, 8}, {0.9999, 15}}};
>+      } else {
>+        return {0.15,
>+                0.5,
>+                {{0.95, 0.3}},
>+                {{0.95, 0}, {0.99, 3}, {0.999, 15}, {0.9999, 25}}};
>+      }
>+      break;
>+    case 16:
>+      if (kRandomizesInserts) {
>+        return {0.1,
>+                0.4,
>+                {{0.95, 0.3}},
>+                {{0.95, 0}, {0.99, 1}, {0.999, 8}, {0.9999, 15}}};
>+      } else {
>+        return {0.05,
>+                0.2,
>+                {{0.95, 0.1}},
>+                {{0.95, 0}, {0.99, 1}, {0.999, 6}, {0.9999, 10}}};
>+      }
>+      break;
>+    default:
>+      ABSL_RAW_LOG(FATAL, "%s", "Unknown Group width");
>+  }
>+  return {};
>+}
>+TEST(Table, DISABLED_EnsureNonQuadraticTopNLinearTransformByProbeSeqLength) {
>+  ProbeStatsPerSize stats;
>+  std::vector<size_t> sizes = {Group::kWidth << 5, Group::kWidth << 10};
>+  for (size_t size : sizes) {
>+    stats[size] = CollectProbeStatsOnLinearlyTransformedKeys(
>+        CollectBadMergeKeys(size), 300);
>+  }
>+  auto expected = LinearTransformExpectedStats();
>+  for (size_t size : sizes) {
>+    auto& stat = stats[size];
>+    VerifyStats(size, expected, stat);
>+  }
>+}
>+
>+TEST(Table, EraseCollision) {
>+  BadTable t;
>+
>+  // 1 2 3
>+  t.emplace(1);
>+  t.emplace(2);
>+  t.emplace(3);
>+  EXPECT_THAT(*t.find(1), 1);
>+  EXPECT_THAT(*t.find(2), 2);
>+  EXPECT_THAT(*t.find(3), 3);
>+  EXPECT_EQ(3, t.size());
>+
>+  // 1 DELETED 3
>+  t.erase(t.find(2));
>+  EXPECT_THAT(*t.find(1), 1);
>+  EXPECT_TRUE(t.find(2) == t.end());
>+  EXPECT_THAT(*t.find(3), 3);
>+  EXPECT_EQ(2, t.size());
>+
>+  // DELETED DELETED 3
>+  t.erase(t.find(1));
>+  EXPECT_TRUE(t.find(1) == t.end());
>+  EXPECT_TRUE(t.find(2) == t.end());
>+  EXPECT_THAT(*t.find(3), 3);
>+  EXPECT_EQ(1, t.size());
>+
>+  // DELETED DELETED DELETED
>+  t.erase(t.find(3));
>+  EXPECT_TRUE(t.find(1) == t.end());
>+  EXPECT_TRUE(t.find(2) == t.end());
>+  EXPECT_TRUE(t.find(3) == t.end());
>+  EXPECT_EQ(0, t.size());
>+}
>+
>+TEST(Table, EraseInsertProbing) {
>+  BadTable t(100);
>+
>+  // 1 2 3 4
>+  t.emplace(1);
>+  t.emplace(2);
>+  t.emplace(3);
>+  t.emplace(4);
>+
>+  // 1 DELETED 3 DELETED
>+  t.erase(t.find(2));
>+  t.erase(t.find(4));
>+
>+  // 1 10 3 11 12
>+  t.emplace(10);
>+  t.emplace(11);
>+  t.emplace(12);
>+
>+  EXPECT_EQ(5, t.size());
>+  EXPECT_THAT(t, UnorderedElementsAre(1, 10, 3, 11, 12));
>+}
>+
>+TEST(Table, Clear) {
>+  IntTable t;
>+  EXPECT_TRUE(t.find(0) == t.end());
>+  t.clear();
>+  EXPECT_TRUE(t.find(0) == t.end());
>+  auto res = t.emplace(0);
>+  EXPECT_TRUE(res.second);
>+  EXPECT_EQ(1, t.size());
>+  t.clear();
>+  EXPECT_EQ(0, t.size());
>+  EXPECT_TRUE(t.find(0) == t.end());
>+}
>+
>+TEST(Table, Swap) {
>+  IntTable t;
>+  EXPECT_TRUE(t.find(0) == t.end());
>+  auto res = t.emplace(0);
>+  EXPECT_TRUE(res.second);
>+  EXPECT_EQ(1, t.size());
>+  IntTable u;
>+  t.swap(u);
>+  EXPECT_EQ(0, t.size());
>+  EXPECT_EQ(1, u.size());
>+  EXPECT_TRUE(t.find(0) == t.end());
>+  EXPECT_THAT(*u.find(0), 0);
>+}
>+
>+TEST(Table, Rehash) {
>+  IntTable t;
>+  EXPECT_TRUE(t.find(0) == t.end());
>+  t.emplace(0);
>+  t.emplace(1);
>+  EXPECT_EQ(2, t.size());
>+  t.rehash(128);
>+  EXPECT_EQ(2, t.size());
>+  EXPECT_THAT(*t.find(0), 0);
>+  EXPECT_THAT(*t.find(1), 1);
>+}
>+
>+TEST(Table, RehashDoesNotRehashWhenNotNecessary) {
>+  IntTable t;
>+  t.emplace(0);
>+  t.emplace(1);
>+  auto* p = &*t.find(0);
>+  t.rehash(1);
>+  EXPECT_EQ(p, &*t.find(0));
>+}
>+
>+TEST(Table, RehashZeroDoesNotAllocateOnEmptyTable) {
>+  IntTable t;
>+  t.rehash(0);
>+  EXPECT_EQ(0, t.bucket_count());
>+}
>+
>+TEST(Table, RehashZeroDeallocatesEmptyTable) {
>+  IntTable t;
>+  t.emplace(0);
>+  t.clear();
>+  EXPECT_NE(0, t.bucket_count());
>+  t.rehash(0);
>+  EXPECT_EQ(0, t.bucket_count());
>+}
>+
>+TEST(Table, RehashZeroForcesRehash) {
>+  IntTable t;
>+  t.emplace(0);
>+  t.emplace(1);
>+  auto* p = &*t.find(0);
>+  t.rehash(0);
>+  EXPECT_NE(p, &*t.find(0));
>+}
>+
>+TEST(Table, ConstructFromInitList) {
>+  using P = std::pair<std::string, std::string>;
>+  struct Q {
>+    operator P() const { return {}; }
>+  };
>+  StringTable t = {P(), Q(), {}, {{}, {}}};
>+}
>+
>+TEST(Table, CopyConstruct) {
>+  IntTable t;
>+  t.max_load_factor(.321f);
>+  t.emplace(0);
>+  EXPECT_EQ(1, t.size());
>+  {
>+    IntTable u(t);
>+    EXPECT_EQ(1, u.size());
>+    EXPECT_EQ(t.max_load_factor(), u.max_load_factor());
>+    EXPECT_THAT(*u.find(0), 0);
>+  }
>+  {
>+    IntTable u{t};
>+    EXPECT_EQ(1, u.size());
>+    EXPECT_EQ(t.max_load_factor(), u.max_load_factor());
>+    EXPECT_THAT(*u.find(0), 0);
>+  }
>+  {
>+    IntTable u = t;
>+    EXPECT_EQ(1, u.size());
>+    EXPECT_EQ(t.max_load_factor(), u.max_load_factor());
>+    EXPECT_THAT(*u.find(0), 0);
>+  }
>+}
>+
>+TEST(Table, CopyConstructWithAlloc) {
>+  StringTable t;
>+  t.max_load_factor(.321f);
>+  t.emplace("a", "b");
>+  EXPECT_EQ(1, t.size());
>+  StringTable u(t, Alloc<std::pair<std::string, std::string>>());
>+  EXPECT_EQ(1, u.size());
>+  EXPECT_EQ(t.max_load_factor(), u.max_load_factor());
>+  EXPECT_THAT(*u.find("a"), Pair("a", "b"));
>+}
>+
>+struct ExplicitAllocIntTable
>+    : raw_hash_set<IntPolicy, container_internal::hash_default_hash<int64_t>,
>+                   std::equal_to<int64_t>, Alloc<int64_t>> {
>+  ExplicitAllocIntTable() {}
>+};
>+
>+TEST(Table, AllocWithExplicitCtor) {
>+  ExplicitAllocIntTable t;
>+  EXPECT_EQ(0, t.size());
>+}
>+
>+TEST(Table, MoveConstruct) {
>+  {
>+    StringTable t;
>+    t.max_load_factor(.321f);
>+    const float lf = t.max_load_factor();
>+    t.emplace("a", "b");
>+    EXPECT_EQ(1, t.size());
>+
>+    StringTable u(std::move(t));
>+    EXPECT_EQ(1, u.size());
>+    EXPECT_EQ(lf, u.max_load_factor());
>+    EXPECT_THAT(*u.find("a"), Pair("a", "b"));
>+  }
>+  {
>+    StringTable t;
>+    t.max_load_factor(.321f);
>+    const float lf = t.max_load_factor();
>+    t.emplace("a", "b");
>+    EXPECT_EQ(1, t.size());
>+
>+    StringTable u{std::move(t)};
>+    EXPECT_EQ(1, u.size());
>+    EXPECT_EQ(lf, u.max_load_factor());
>+    EXPECT_THAT(*u.find("a"), Pair("a", "b"));
>+  }
>+  {
>+    StringTable t;
>+    t.max_load_factor(.321f);
>+    const float lf = t.max_load_factor();
>+    t.emplace("a", "b");
>+    EXPECT_EQ(1, t.size());
>+
>+    StringTable u = std::move(t);
>+    EXPECT_EQ(1, u.size());
>+    EXPECT_EQ(lf, u.max_load_factor());
>+    EXPECT_THAT(*u.find("a"), Pair("a", "b"));
>+  }
>+}
>+
>+TEST(Table, MoveConstructWithAlloc) {
>+  StringTable t;
>+  t.max_load_factor(.321f);
>+  const float lf = t.max_load_factor();
>+  t.emplace("a", "b");
>+  EXPECT_EQ(1, t.size());
>+  StringTable u(std::move(t), Alloc<std::pair<std::string, std::string>>());
>+  EXPECT_EQ(1, u.size());
>+  EXPECT_EQ(lf, u.max_load_factor());
>+  EXPECT_THAT(*u.find("a"), Pair("a", "b"));
>+}
>+
>+TEST(Table, CopyAssign) {
>+  StringTable t;
>+  t.max_load_factor(.321f);
>+  t.emplace("a", "b");
>+  EXPECT_EQ(1, t.size());
>+  StringTable u;
>+  u = t;
>+  EXPECT_EQ(1, u.size());
>+  EXPECT_EQ(t.max_load_factor(), u.max_load_factor());
>+  EXPECT_THAT(*u.find("a"), Pair("a", "b"));
>+}
>+
>+TEST(Table, CopySelfAssign) {
>+  StringTable t;
>+  t.max_load_factor(.321f);
>+  const float lf = t.max_load_factor();
>+  t.emplace("a", "b");
>+  EXPECT_EQ(1, t.size());
>+  t = *&t;
>+  EXPECT_EQ(1, t.size());
>+  EXPECT_EQ(lf, t.max_load_factor());
>+  EXPECT_THAT(*t.find("a"), Pair("a", "b"));
>+}
>+
>+TEST(Table, MoveAssign) {
>+  StringTable t;
>+  t.max_load_factor(.321f);
>+  const float lf = t.max_load_factor();
>+  t.emplace("a", "b");
>+  EXPECT_EQ(1, t.size());
>+  StringTable u;
>+  u = std::move(t);
>+  EXPECT_EQ(1, u.size());
>+  EXPECT_EQ(lf, u.max_load_factor());
>+  EXPECT_THAT(*u.find("a"), Pair("a", "b"));
>+}
>+
>+TEST(Table, Equality) {
>+  StringTable t;
>+  std::vector<std::pair<std::string, std::string>> v = {{"a", "b"}, {"aa", "bb"}};
>+  t.insert(std::begin(v), std::end(v));
>+  StringTable u = t;
>+  EXPECT_EQ(u, t);
>+}
>+
>+TEST(Table, Equality2) {
>+  StringTable t;
>+  std::vector<std::pair<std::string, std::string>> v1 = {{"a", "b"}, {"aa", "bb"}};
>+  t.insert(std::begin(v1), std::end(v1));
>+  StringTable u;
>+  std::vector<std::pair<std::string, std::string>> v2 = {{"a", "a"}, {"aa", "aa"}};
>+  u.insert(std::begin(v2), std::end(v2));
>+  EXPECT_NE(u, t);
>+}
>+
>+TEST(Table, Equality3) {
>+  StringTable t;
>+  std::vector<std::pair<std::string, std::string>> v1 = {{"b", "b"}, {"bb", "bb"}};
>+  t.insert(std::begin(v1), std::end(v1));
>+  StringTable u;
>+  std::vector<std::pair<std::string, std::string>> v2 = {{"a", "a"}, {"aa", "aa"}};
>+  u.insert(std::begin(v2), std::end(v2));
>+  EXPECT_NE(u, t);
>+}
>+
>+TEST(Table, NumDeletedRegression) {
>+  IntTable t;
>+  t.emplace(0);
>+  t.erase(t.find(0));
>+  // construct over a deleted slot.
>+  t.emplace(0);
>+  t.clear();
>+}
>+
>+TEST(Table, FindFullDeletedRegression) {
>+  IntTable t;
>+  for (int i = 0; i < 1000; ++i) {
>+    t.emplace(i);
>+    t.erase(t.find(i));
>+  }
>+  EXPECT_EQ(0, t.size());
>+}
>+
>+TEST(Table, ReplacingDeletedSlotDoesNotRehash) {
>+  size_t n;
>+  {
>+    // Compute n such that n is the maximum number of elements before rehash.
>+    IntTable t;
>+    t.emplace(0);
>+    size_t c = t.bucket_count();
>+    for (n = 1; c == t.bucket_count(); ++n) t.emplace(n);
>+    --n;
>+  }
>+  IntTable t;
>+  t.rehash(n);
>+  const size_t c = t.bucket_count();
>+  for (size_t i = 0; i != n; ++i) t.emplace(i);
>+  EXPECT_EQ(c, t.bucket_count()) << "rehashing threshold = " << n;
>+  t.erase(0);
>+  t.emplace(0);
>+  EXPECT_EQ(c, t.bucket_count()) << "rehashing threshold = " << n;
>+}
>+
>+TEST(Table, NoThrowMoveConstruct) {
>+  ASSERT_TRUE(
>+      std::is_nothrow_copy_constructible<absl::Hash<absl::string_view>>::value);
>+  ASSERT_TRUE(std::is_nothrow_copy_constructible<
>+              std::equal_to<absl::string_view>>::value);
>+  ASSERT_TRUE(std::is_nothrow_copy_constructible<std::allocator<int>>::value);
>+  EXPECT_TRUE(std::is_nothrow_move_constructible<StringTable>::value);
>+}
>+
>+TEST(Table, NoThrowMoveAssign) {
>+  ASSERT_TRUE(
>+      std::is_nothrow_move_assignable<absl::Hash<absl::string_view>>::value);
>+  ASSERT_TRUE(
>+      std::is_nothrow_move_assignable<std::equal_to<absl::string_view>>::value);
>+  ASSERT_TRUE(std::is_nothrow_move_assignable<std::allocator<int>>::value);
>+  ASSERT_TRUE(
>+      absl::allocator_traits<std::allocator<int>>::is_always_equal::value);
>+  EXPECT_TRUE(std::is_nothrow_move_assignable<StringTable>::value);
>+}
>+
>+TEST(Table, NoThrowSwappable) {
>+  ASSERT_TRUE(
>+      container_internal::IsNoThrowSwappable<absl::Hash<absl::string_view>>());
>+  ASSERT_TRUE(container_internal::IsNoThrowSwappable<
>+              std::equal_to<absl::string_view>>());
>+  ASSERT_TRUE(container_internal::IsNoThrowSwappable<std::allocator<int>>());
>+  EXPECT_TRUE(container_internal::IsNoThrowSwappable<StringTable>());
>+}
>+
>+TEST(Table, HeterogeneousLookup) {
>+  struct Hash {
>+    size_t operator()(int64_t i) const { return i; }
>+    size_t operator()(double i) const {
>+      ADD_FAILURE();
>+      return i;
>+    }
>+  };
>+  struct Eq {
>+    bool operator()(int64_t a, int64_t b) const { return a == b; }
>+    bool operator()(double a, int64_t b) const {
>+      ADD_FAILURE();
>+      return a == b;
>+    }
>+    bool operator()(int64_t a, double b) const {
>+      ADD_FAILURE();
>+      return a == b;
>+    }
>+    bool operator()(double a, double b) const {
>+      ADD_FAILURE();
>+      return a == b;
>+    }
>+  };
>+
>+  struct THash {
>+    using is_transparent = void;
>+    size_t operator()(int64_t i) const { return i; }
>+    size_t operator()(double i) const { return i; }
>+  };
>+  struct TEq {
>+    using is_transparent = void;
>+    bool operator()(int64_t a, int64_t b) const { return a == b; }
>+    bool operator()(double a, int64_t b) const { return a == b; }
>+    bool operator()(int64_t a, double b) const { return a == b; }
>+    bool operator()(double a, double b) const { return a == b; }
>+  };
>+
>+  raw_hash_set<IntPolicy, Hash, Eq, Alloc<int64_t>> s{0, 1, 2};
>+  // It will convert to int64_t before the query.
>+  EXPECT_EQ(1, *s.find(double{1.1}));
>+
>+  raw_hash_set<IntPolicy, THash, TEq, Alloc<int64_t>> ts{0, 1, 2};
>+  // It will try to use the double, and fail to find the object.
>+  EXPECT_TRUE(ts.find(1.1) == ts.end());
>+}
>+
>+template <class Table>
>+using CallFind = decltype(std::declval<Table&>().find(17));
>+
>+template <class Table>
>+using CallErase = decltype(std::declval<Table&>().erase(17));
>+
>+template <class Table>
>+using CallExtract = decltype(std::declval<Table&>().extract(17));
>+
>+template <class Table>
>+using CallPrefetch = decltype(std::declval<Table&>().prefetch(17));
>+
>+template <class Table>
>+using CallCount = decltype(std::declval<Table&>().count(17));
>+
>+template <template <typename> class C, class Table, class = void>
>+struct VerifyResultOf : std::false_type {};
>+
>+template <template <typename> class C, class Table>
>+struct VerifyResultOf<C, Table, absl::void_t<C<Table>>> : std::true_type {};
>+
>+TEST(Table, HeterogeneousLookupOverloads) {
>+  using NonTransparentTable =
>+      raw_hash_set<StringPolicy, absl::Hash<absl::string_view>,
>+                   std::equal_to<absl::string_view>, std::allocator<int>>;
>+
>+  EXPECT_FALSE((VerifyResultOf<CallFind, NonTransparentTable>()));
>+  EXPECT_FALSE((VerifyResultOf<CallErase, NonTransparentTable>()));
>+  EXPECT_FALSE((VerifyResultOf<CallExtract, NonTransparentTable>()));
>+  EXPECT_FALSE((VerifyResultOf<CallPrefetch, NonTransparentTable>()));
>+  EXPECT_FALSE((VerifyResultOf<CallCount, NonTransparentTable>()));
>+
>+  using TransparentTable = raw_hash_set<
>+      StringPolicy,
>+      absl::container_internal::hash_default_hash<absl::string_view>,
>+      absl::container_internal::hash_default_eq<absl::string_view>,
>+      std::allocator<int>>;
>+
>+  EXPECT_TRUE((VerifyResultOf<CallFind, TransparentTable>()));
>+  EXPECT_TRUE((VerifyResultOf<CallErase, TransparentTable>()));
>+  EXPECT_TRUE((VerifyResultOf<CallExtract, TransparentTable>()));
>+  EXPECT_TRUE((VerifyResultOf<CallPrefetch, TransparentTable>()));
>+  EXPECT_TRUE((VerifyResultOf<CallCount, TransparentTable>()));
>+}
>+
>+// TODO(alkis): Expand iterator tests.
>+TEST(Iterator, IsDefaultConstructible) {
>+  StringTable::iterator i;
>+  EXPECT_TRUE(i == StringTable::iterator());
>+}
>+
>+TEST(ConstIterator, IsDefaultConstructible) {
>+  StringTable::const_iterator i;
>+  EXPECT_TRUE(i == StringTable::const_iterator());
>+}
>+
>+TEST(Iterator, ConvertsToConstIterator) {
>+  StringTable::iterator i;
>+  EXPECT_TRUE(i == StringTable::const_iterator());
>+}
>+
>+TEST(Iterator, Iterates) {
>+  IntTable t;
>+  for (size_t i = 3; i != 6; ++i) EXPECT_TRUE(t.emplace(i).second);
>+  EXPECT_THAT(t, UnorderedElementsAre(3, 4, 5));
>+}
>+
>+TEST(Table, Merge) {
>+  StringTable t1, t2;
>+  t1.emplace("0", "-0");
>+  t1.emplace("1", "-1");
>+  t2.emplace("0", "~0");
>+  t2.emplace("2", "~2");
>+
>+  EXPECT_THAT(t1, UnorderedElementsAre(Pair("0", "-0"), Pair("1", "-1")));
>+  EXPECT_THAT(t2, UnorderedElementsAre(Pair("0", "~0"), Pair("2", "~2")));
>+
>+  t1.merge(t2);
>+  EXPECT_THAT(t1, UnorderedElementsAre(Pair("0", "-0"), Pair("1", "-1"),
>+                                       Pair("2", "~2")));
>+  EXPECT_THAT(t2, UnorderedElementsAre(Pair("0", "~0")));
>+}
>+
>+TEST(Nodes, EmptyNodeType) {
>+  using node_type = StringTable::node_type;
>+  node_type n;
>+  EXPECT_FALSE(n);
>+  EXPECT_TRUE(n.empty());
>+
>+  EXPECT_TRUE((std::is_same<node_type::allocator_type,
>+                            StringTable::allocator_type>::value));
>+}
>+
>+TEST(Nodes, ExtractInsert) {
>+  constexpr char k0[] = "Very long std::string zero.";
>+  constexpr char k1[] = "Very long std::string one.";
>+  constexpr char k2[] = "Very long std::string two.";
>+  StringTable t = {{k0, ""}, {k1, ""}, {k2, ""}};
>+  EXPECT_THAT(t,
>+              UnorderedElementsAre(Pair(k0, ""), Pair(k1, ""), Pair(k2, "")));
>+
>+  auto node = t.extract(k0);
>+  EXPECT_THAT(t, UnorderedElementsAre(Pair(k1, ""), Pair(k2, "")));
>+  EXPECT_TRUE(node);
>+  EXPECT_FALSE(node.empty());
>+
>+  StringTable t2;
>+  auto res = t2.insert(std::move(node));
>+  EXPECT_TRUE(res.inserted);
>+  EXPECT_THAT(*res.position, Pair(k0, ""));
>+  EXPECT_FALSE(res.node);
>+  EXPECT_THAT(t2, UnorderedElementsAre(Pair(k0, "")));
>+
>+  // Not there.
>+  EXPECT_THAT(t, UnorderedElementsAre(Pair(k1, ""), Pair(k2, "")));
>+  node = t.extract("Not there!");
>+  EXPECT_THAT(t, UnorderedElementsAre(Pair(k1, ""), Pair(k2, "")));
>+  EXPECT_FALSE(node);
>+
>+  // Inserting nothing.
>+  res = t2.insert(std::move(node));
>+  EXPECT_FALSE(res.inserted);
>+  EXPECT_EQ(res.position, t2.end());
>+  EXPECT_FALSE(res.node);
>+  EXPECT_THAT(t2, UnorderedElementsAre(Pair(k0, "")));
>+
>+  t.emplace(k0, "1");
>+  node = t.extract(k0);
>+
>+  // Insert duplicate.
>+  res = t2.insert(std::move(node));
>+  EXPECT_FALSE(res.inserted);
>+  EXPECT_THAT(*res.position, Pair(k0, ""));
>+  EXPECT_TRUE(res.node);
>+  EXPECT_FALSE(node);
>+}
>+
>+StringTable MakeSimpleTable(size_t size) {
>+  StringTable t;
>+  for (size_t i = 0; i < size; ++i) t.emplace(std::string(1, 'A' + i), "");
>+  return t;
>+}
>+
>+std::string OrderOfIteration(const StringTable& t) {
>+  std::string order;
>+  for (auto& p : t) order += p.first;
>+  return order;
>+}
>+
>+TEST(Table, IterationOrderChangesByInstance) {
>+  // Needs to be more than kWidth elements to be able to affect order.
>+  const StringTable reference = MakeSimpleTable(20);
>+
>+  // Since order is non-deterministic we can't just try once and verify.
>+  // We'll try until we find that order changed. It should not take many tries
>+  // for that.
>+  // Important: we have to keep the old tables around. Otherwise tcmalloc will
>+  // just give us the same blocks and we would be doing the same order again.
>+  std::vector<StringTable> garbage;
>+  for (int i = 0; i < 10; ++i) {
>+    auto trial = MakeSimpleTable(20);
>+    if (OrderOfIteration(trial) != OrderOfIteration(reference)) {
>+      // We are done.
>+      return;
>+    }
>+    garbage.push_back(std::move(trial));
>+  }
>+  FAIL();
>+}
>+
>+TEST(Table, IterationOrderChangesOnRehash) {
>+  // Since order is non-deterministic we can't just try once and verify.
>+  // We'll try until we find that order changed. It should not take many tries
>+  // for that.
>+  // Important: we have to keep the old tables around. Otherwise tcmalloc will
>+  // just give us the same blocks and we would be doing the same order again.
>+  std::vector<StringTable> garbage;
>+  for (int i = 0; i < 10; ++i) {
>+    // Needs to be more than kWidth elements to be able to affect order.
>+    StringTable t = MakeSimpleTable(20);
>+    const std::string reference = OrderOfIteration(t);
>+    // Force rehash to the same size.
>+    t.rehash(0);
>+    std::string trial = OrderOfIteration(t);
>+    if (trial != reference) {
>+      // We are done.
>+      return;
>+    }
>+    garbage.push_back(std::move(t));
>+  }
>+  FAIL();
>+}
>+
>+TEST(Table, IterationOrderChangesForSmallTables) {
>+  // Since order is non-deterministic we can't just try once and verify.
>+  // We'll try until we find that order changed.
>+  // Important: we have to keep the old tables around. Otherwise tcmalloc will
>+  // just give us the same blocks and we would be doing the same order again.
>+  StringTable reference_table = MakeSimpleTable(5);
>+  const std::string reference = OrderOfIteration(reference_table);
>+  std::vector<StringTable> garbage;
>+  for (int i = 0; i < 50; ++i) {
>+    StringTable t = MakeSimpleTable(5);
>+    std::string trial = OrderOfIteration(t);
>+    if (trial != reference) {
>+      // We are done.
>+      return;
>+    }
>+    garbage.push_back(std::move(t));
>+  }
>+  FAIL() << "Iteration order remained the same across many attempts.";
>+}
>+
>+// Fill the table to 3 different load factors (min, median, max) and evaluate
>+// the percentage of perfect hits using the debug API.
>+template <class Table, class AddFn>
>+std::vector<double> CollectPerfectRatios(Table t, AddFn add) {
>+  using Key = typename Table::key_type;
>+
>+  // First, fill enough to have a good distribution.
>+  constexpr size_t kMinSize = 10000;
>+  std::vector<Key> keys;
>+  while (t.size() < kMinSize) keys.push_back(add(t));
>+  // Then, insert until we reach min load factor.
>+  double lf = t.load_factor();
>+  while (lf <= t.load_factor()) keys.push_back(add(t));
>+
>+  // We are now at min load factor. Take a snapshot.
>+  size_t perfect = 0;
>+  auto update_perfect = [&](Key k) {
>+    perfect += GetHashtableDebugNumProbes(t, k) == 0;
>+  };
>+  for (const auto& k : keys) update_perfect(k);
>+
>+  std::vector<double> perfect_ratios;
>+  // Keep going until we hit max load factor.
>+  while (t.load_factor() < .6) {
>+    perfect_ratios.push_back(1.0 * perfect / t.size());
>+    update_perfect(add(t));
>+  }
>+  while (t.load_factor() > .5) {
>+    perfect_ratios.push_back(1.0 * perfect / t.size());
>+    update_perfect(add(t));
>+  }
>+  return perfect_ratios;
>+}
>+
>+std::vector<std::pair<double, double>> StringTablePefectRatios() {
>+  constexpr bool kRandomizesInserts =
>+#if NDEBUG
>+      false;
>+#else   // NDEBUG
>+      true;
>+#endif  // NDEBUG
>+
>+  // The effective load factor is larger in non-opt mode because we insert
>+  // elements out of order.
>+  switch (container_internal::Group::kWidth) {
>+    case 8:
>+      if (kRandomizesInserts) {
>+        return {{0.986, 0.02}, {0.95, 0.02}, {0.89, 0.02}};
>+      } else {
>+        return {{0.995, 0.01}, {0.97, 0.01}, {0.89, 0.02}};
>+      }
>+      break;
>+    case 16:
>+      if (kRandomizesInserts) {
>+        return {{0.973, 0.01}, {0.965, 0.01}, {0.92, 0.02}};
>+      } else {
>+        return {{0.995, 0.005}, {0.99, 0.005}, {0.94, 0.01}};
>+      }
>+      break;
>+    default:
>+      // Ignore anything else.
>+      return {};
>+  }
>+}
>+
>+// This is almost a change detector, but it allows us to know how we are
>+// affecting the probe distribution.
>+TEST(Table, EffectiveLoadFactorStrings) {
>+  std::vector<double> perfect_ratios =
>+      CollectPerfectRatios(StringTable(), [](StringTable& t) {
>+        return t.emplace(std::to_string(t.size()), "").first->first;
>+      });
>+
>+  auto ratios = StringTablePefectRatios();
>+  if (ratios.empty()) return;
>+
>+  EXPECT_THAT(perfect_ratios.front(),
>+              DoubleNear(ratios[0].first, ratios[0].second));
>+  EXPECT_THAT(perfect_ratios[perfect_ratios.size() / 2],
>+              DoubleNear(ratios[1].first, ratios[1].second));
>+  EXPECT_THAT(perfect_ratios.back(),
>+              DoubleNear(ratios[2].first, ratios[2].second));
>+}
>+
>+std::vector<std::pair<double, double>> IntTablePefectRatios() {
>+  constexpr bool kRandomizesInserts =
>+#ifdef NDEBUG
>+      false;
>+#else   // NDEBUG
>+      true;
>+#endif  // NDEBUG
>+
>+  // The effective load factor is larger in non-opt mode because we insert
>+  // elements out of order.
>+  switch (container_internal::Group::kWidth) {
>+    case 8:
>+      if (kRandomizesInserts) {
>+        return {{0.99, 0.02}, {0.985, 0.02}, {0.95, 0.05}};
>+      } else {
>+        return {{0.99, 0.01}, {0.99, 0.01}, {0.95, 0.02}};
>+      }
>+      break;
>+    case 16:
>+      if (kRandomizesInserts) {
>+        return {{0.98, 0.02}, {0.978, 0.02}, {0.96, 0.02}};
>+      } else {
>+        return {{0.998, 0.003}, {0.995, 0.01}, {0.975, 0.02}};
>+      }
>+      break;
>+    default:
>+      // Ignore anything else.
>+      return {};
>+  }
>+}
>+
>+// This is almost a change detector, but it allows us to know how we are
>+// affecting the probe distribution.
>+TEST(Table, EffectiveLoadFactorInts) {
>+  std::vector<double> perfect_ratios = CollectPerfectRatios(
>+      IntTable(), [](IntTable& t) { return *t.emplace(t.size()).first; });
>+
>+  auto ratios = IntTablePefectRatios();
>+  if (ratios.empty()) return;
>+
>+  EXPECT_THAT(perfect_ratios.front(),
>+              DoubleNear(ratios[0].first, ratios[0].second));
>+  EXPECT_THAT(perfect_ratios[perfect_ratios.size() / 2],
>+              DoubleNear(ratios[1].first, ratios[1].second));
>+  EXPECT_THAT(perfect_ratios.back(),
>+              DoubleNear(ratios[2].first, ratios[2].second));
>+}
>+
>+// Confirm that we assert if we try to erase() end().
>+TEST(TableDeathTest, EraseOfEndAsserts) {
>+  // Use an assert with side-effects to figure out if they are actually enabled.
>+  bool assert_enabled = false;
>+  assert([&]() {
>+    assert_enabled = true;
>+    return true;
>+  }());
>+  if (!assert_enabled) return;
>+
>+  IntTable t;
>+  // Extra simple "regexp" as regexp support is highly varied across platforms.
>+  constexpr char kDeathMsg[] = "it != end";
>+  EXPECT_DEATH_IF_SUPPORTED(t.erase(t.end()), kDeathMsg);
>+}
>+
>+#ifdef ADDRESS_SANITIZER
>+TEST(Sanitizer, PoisoningUnused) {
>+  IntTable t;
>+  // Insert something to force an allocation.
>+  int64_t& v1 = *t.insert(0).first;
>+
>+  // Make sure there is something to test.
>+  ASSERT_GT(t.capacity(), 1);
>+
>+  int64_t* slots = RawHashSetTestOnlyAccess::GetSlots(t);
>+  for (size_t i = 0; i < t.capacity(); ++i) {
>+    EXPECT_EQ(slots + i != &v1, __asan_address_is_poisoned(slots + i));
>+  }
>+}
>+
>+TEST(Sanitizer, PoisoningOnErase) {
>+  IntTable t;
>+  int64_t& v = *t.insert(0).first;
>+
>+  EXPECT_FALSE(__asan_address_is_poisoned(&v));
>+  t.erase(0);
>+  EXPECT_TRUE(__asan_address_is_poisoned(&v));
>+}
>+#endif  // ADDRESS_SANITIZER
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.cc
>index fe00aca8fb98f120f235202369b0b5be876d4747..b18e0bb7a1ea5fffdc9618f1e793b8ff92ed9038 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.cc
>@@ -21,6 +21,7 @@ int BaseCountedInstance::num_live_instances_ = 0;
> int BaseCountedInstance::num_moves_ = 0;
> int BaseCountedInstance::num_copies_ = 0;
> int BaseCountedInstance::num_swaps_ = 0;
>+int BaseCountedInstance::num_comparisons_ = 0;
> 
> }  // namespace test_internal
> }  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.h
>index cf8f3a531e6212fcaa13d6535170af86bffa698a..ec45f574037fc8e0b6937bcd8fc1f42756e08108 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.h
>@@ -22,8 +22,8 @@ namespace absl {
> namespace test_internal {
> 
> // A type that counts number of occurences of the type, the live occurrences of
>-// the type, as well as the number of copies, moves, and swaps that have
>-// occurred on the type. This is used as a base class for the copyable,
>+// the type, as well as the number of copies, moves, swaps, and comparisons that
>+// have occurred on the type. This is used as a base class for the copyable,
> // copyable+movable, and movable types below that are used in actual tests. Use
> // InstanceTracker in tests to track the number of instances.
> class BaseCountedInstance {
>@@ -66,6 +66,36 @@ class BaseCountedInstance {
>     return *this;
>   }
> 
>+  bool operator==(const BaseCountedInstance& x) const {
>+    ++num_comparisons_;
>+    return value_ == x.value_;
>+  }
>+
>+  bool operator!=(const BaseCountedInstance& x) const {
>+    ++num_comparisons_;
>+    return value_ != x.value_;
>+  }
>+
>+  bool operator<(const BaseCountedInstance& x) const {
>+    ++num_comparisons_;
>+    return value_ < x.value_;
>+  }
>+
>+  bool operator>(const BaseCountedInstance& x) const {
>+    ++num_comparisons_;
>+    return value_ > x.value_;
>+  }
>+
>+  bool operator<=(const BaseCountedInstance& x) const {
>+    ++num_comparisons_;
>+    return value_ <= x.value_;
>+  }
>+
>+  bool operator>=(const BaseCountedInstance& x) const {
>+    ++num_comparisons_;
>+    return value_ >= x.value_;
>+  }
>+
>   int value() const {
>     if (!is_live_) std::abort();
>     return value_;
>@@ -108,6 +138,9 @@ class BaseCountedInstance {
> 
>   // Number of times that BaseCountedInstance objects were swapped.
>   static int num_swaps_;
>+
>+  // Number of times that BaseCountedInstance objects were compared.
>+  static int num_comparisons_;
> };
> 
> // Helper to track the BaseCountedInstance instance counters. Expects that the
>@@ -152,13 +185,21 @@ class InstanceTracker {
>   // construction or the last call to ResetCopiesMovesSwaps().
>   int swaps() const { return BaseCountedInstance::num_swaps_ - start_swaps_; }
> 
>-  // Resets the base values for moves, copies and swaps to the current values,
>-  // so that subsequent Get*() calls for moves, copies and swaps will compare to
>-  // the situation at the point of this call.
>+  // Returns the number of comparisons on BaseCountedInstance objects since
>+  // construction or the last call to ResetCopiesMovesSwaps().
>+  int comparisons() const {
>+    return BaseCountedInstance::num_comparisons_ - start_comparisons_;
>+  }
>+
>+  // Resets the base values for moves, copies, comparisons, and swaps to the
>+  // current values, so that subsequent Get*() calls for moves, copies,
>+  // comparisons, and swaps will compare to the situation at the point of this
>+  // call.
>   void ResetCopiesMovesSwaps() {
>     start_moves_ = BaseCountedInstance::num_moves_;
>     start_copies_ = BaseCountedInstance::num_copies_;
>     start_swaps_ = BaseCountedInstance::num_swaps_;
>+    start_comparisons_ = BaseCountedInstance::num_comparisons_;
>   }
> 
>  private:
>@@ -167,6 +208,7 @@ class InstanceTracker {
>   int start_moves_;
>   int start_copies_;
>   int start_swaps_;
>+  int start_comparisons_;
> };
> 
> // Copyable, not movable.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker_test.cc
>index 9efb6771cf086db16fef39460c591921ecd5f691..0ae57636df21338979210395bb3996bbd8c97b8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker_test.cc
>@@ -157,4 +157,26 @@ TEST(TestInstanceTracker, ExistingInstances) {
>   EXPECT_EQ(1, tracker.moves());
> }
> 
>+TEST(TestInstanceTracker, Comparisons) {
>+  InstanceTracker tracker;
>+  MovableOnlyInstance one(1), two(2);
>+
>+  EXPECT_EQ(0, tracker.comparisons());
>+  EXPECT_FALSE(one == two);
>+  EXPECT_EQ(1, tracker.comparisons());
>+  EXPECT_TRUE(one != two);
>+  EXPECT_EQ(2, tracker.comparisons());
>+  EXPECT_TRUE(one < two);
>+  EXPECT_EQ(3, tracker.comparisons());
>+  EXPECT_FALSE(one > two);
>+  EXPECT_EQ(4, tracker.comparisons());
>+  EXPECT_TRUE(one <= two);
>+  EXPECT_EQ(5, tracker.comparisons());
>+  EXPECT_FALSE(one >= two);
>+  EXPECT_EQ(6, tracker.comparisons());
>+
>+  tracker.ResetCopiesMovesSwaps();
>+  EXPECT_EQ(0, tracker.comparisons());
>+}
>+
> }  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/tracked.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/tracked.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..7d14af03138d3efd66237b797f3ef73c003c8815
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/tracked.h
>@@ -0,0 +1,78 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_TRACKED_H_
>+#define ABSL_CONTAINER_INTERNAL_TRACKED_H_
>+
>+#include <stddef.h>
>+#include <memory>
>+#include <utility>
>+
>+namespace absl {
>+namespace container_internal {
>+
>+// A class that tracks its copies and moves so that it can be queried in tests.
>+template <class T>
>+class Tracked {
>+ public:
>+  Tracked() {}
>+  // NOLINTNEXTLINE(runtime/explicit)
>+  Tracked(const T& val) : val_(val) {}
>+  Tracked(const Tracked& that)
>+      : val_(that.val_),
>+        num_moves_(that.num_moves_),
>+        num_copies_(that.num_copies_) {
>+    ++(*num_copies_);
>+  }
>+  Tracked(Tracked&& that)
>+      : val_(std::move(that.val_)),
>+        num_moves_(std::move(that.num_moves_)),
>+        num_copies_(std::move(that.num_copies_)) {
>+    ++(*num_moves_);
>+  }
>+  Tracked& operator=(const Tracked& that) {
>+    val_ = that.val_;
>+    num_moves_ = that.num_moves_;
>+    num_copies_ = that.num_copies_;
>+    ++(*num_copies_);
>+  }
>+  Tracked& operator=(Tracked&& that) {
>+    val_ = std::move(that.val_);
>+    num_moves_ = std::move(that.num_moves_);
>+    num_copies_ = std::move(that.num_copies_);
>+    ++(*num_moves_);
>+  }
>+
>+  const T& val() const { return val_; }
>+
>+  friend bool operator==(const Tracked& a, const Tracked& b) {
>+    return a.val_ == b.val_;
>+  }
>+  friend bool operator!=(const Tracked& a, const Tracked& b) {
>+    return !(a == b);
>+  }
>+
>+  size_t num_copies() { return *num_copies_; }
>+  size_t num_moves() { return *num_moves_; }
>+
>+ private:
>+  T val_;
>+  std::shared_ptr<size_t> num_moves_ = std::make_shared<size_t>(0);
>+  std::shared_ptr<size_t> num_copies_ = std::make_shared<size_t>(0);
>+};
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_TRACKED_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_constructor_test.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_constructor_test.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..2ffb646cb264a65efa4beccdef2ca28e14e9a5d7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_constructor_test.h
>@@ -0,0 +1,404 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_UNORDERED_MAP_CONSTRUCTOR_TEST_H_
>+#define ABSL_CONTAINER_INTERNAL_UNORDERED_MAP_CONSTRUCTOR_TEST_H_
>+
>+#include <algorithm>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/container/internal/hash_generator_testing.h"
>+#include "absl/container/internal/hash_policy_testing.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+template <class UnordMap>
>+class ConstructorTest : public ::testing::Test {};
>+
>+TYPED_TEST_CASE_P(ConstructorTest);
>+
>+TYPED_TEST_P(ConstructorTest, NoArgs) {
>+  TypeParam m;
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(m, ::testing::UnorderedElementsAre());
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCount) {
>+  TypeParam m(123);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(m, ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCountHash) {
>+  using H = typename TypeParam::hasher;
>+  H hasher;
>+  TypeParam m(123, hasher);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(m, ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCountHashEqual) {
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  H hasher;
>+  E equal;
>+  TypeParam m(123, hasher, equal);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.key_eq(), equal);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(m, ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCountHashEqualAlloc) {
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(123, hasher, equal, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.key_eq(), equal);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(m, ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCountAlloc) {
>+#if defined(UNORDERED_MAP_CXX14) || defined(UNORDERED_MAP_CXX17)
>+  using A = typename TypeParam::allocator_type;
>+  A alloc(0);
>+  TypeParam m(123, alloc);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(m, ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCountHashAlloc) {
>+#if defined(UNORDERED_MAP_CXX14) || defined(UNORDERED_MAP_CXX17)
>+  using H = typename TypeParam::hasher;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  A alloc(0);
>+  TypeParam m(123, hasher, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(m, ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketAlloc) {
>+#if ABSL_UNORDERED_SUPPORTS_ALLOC_CTORS
>+  using A = typename TypeParam::allocator_type;
>+  A alloc(0);
>+  TypeParam m(alloc);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(m, ::testing::UnorderedElementsAre());
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, InputIteratorBucketHashEqualAlloc) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end(), 123, hasher, equal, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.key_eq(), equal);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, InputIteratorBucketAlloc) {
>+#if defined(UNORDERED_MAP_CXX14) || defined(UNORDERED_MAP_CXX17)
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using A = typename TypeParam::allocator_type;
>+  A alloc(0);
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end(), 123, alloc);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, InputIteratorBucketHashAlloc) {
>+#if defined(UNORDERED_MAP_CXX14) || defined(UNORDERED_MAP_CXX17)
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  A alloc(0);
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end(), 123, hasher, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, CopyConstructor) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(123, hasher, equal, alloc);
>+  for (size_t i = 0; i != 10; ++i) m.insert(hash_internal::Generator<T>()());
>+  TypeParam n(m);
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_EQ(m.get_allocator(), n.get_allocator());
>+  EXPECT_EQ(m, n);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, CopyConstructorAlloc) {
>+#if ABSL_UNORDERED_SUPPORTS_ALLOC_CTORS
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(123, hasher, equal, alloc);
>+  for (size_t i = 0; i != 10; ++i) m.insert(hash_internal::Generator<T>()());
>+  TypeParam n(m, A(11));
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_NE(m.get_allocator(), n.get_allocator());
>+  EXPECT_EQ(m, n);
>+#endif
>+}
>+
>+// TODO(alkis): Test non-propagating allocators on copy constructors.
>+
>+TYPED_TEST_P(ConstructorTest, MoveConstructor) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(123, hasher, equal, alloc);
>+  for (size_t i = 0; i != 10; ++i) m.insert(hash_internal::Generator<T>()());
>+  TypeParam t(m);
>+  TypeParam n(std::move(t));
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_EQ(m.get_allocator(), n.get_allocator());
>+  EXPECT_EQ(m, n);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, MoveConstructorAlloc) {
>+#if ABSL_UNORDERED_SUPPORTS_ALLOC_CTORS
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(123, hasher, equal, alloc);
>+  for (size_t i = 0; i != 10; ++i) m.insert(hash_internal::Generator<T>()());
>+  TypeParam t(m);
>+  TypeParam n(std::move(t), A(1));
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_NE(m.get_allocator(), n.get_allocator());
>+  EXPECT_EQ(m, n);
>+#endif
>+}
>+
>+// TODO(alkis): Test non-propagating allocators on move constructors.
>+
>+TYPED_TEST_P(ConstructorTest, InitializerListBucketHashEqualAlloc) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(values, 123, hasher, equal, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.key_eq(), equal);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, InitializerListBucketAlloc) {
>+#if defined(UNORDERED_MAP_CXX14) || defined(UNORDERED_MAP_CXX17)
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using A = typename TypeParam::allocator_type;
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  A alloc(0);
>+  TypeParam m(values, 123, alloc);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, InitializerListBucketHashAlloc) {
>+#if defined(UNORDERED_MAP_CXX14) || defined(UNORDERED_MAP_CXX17)
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  A alloc(0);
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  TypeParam m(values, 123, hasher, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, Assignment) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  hash_internal::Generator<T> gen;
>+  TypeParam m({gen(), gen(), gen()}, 123, hasher, equal, alloc);
>+  TypeParam n;
>+  n = m;
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_EQ(m, n);
>+}
>+
>+// TODO(alkis): Test [non-]propagating allocators on move/copy assignments
>+// (it depends on traits).
>+
>+TYPED_TEST_P(ConstructorTest, MoveAssignment) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  hash_internal::Generator<T> gen;
>+  TypeParam m({gen(), gen(), gen()}, 123, hasher, equal, alloc);
>+  TypeParam t(m);
>+  TypeParam n;
>+  n = std::move(t);
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_EQ(m, n);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, AssignmentFromInitializerList) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  TypeParam m;
>+  m = values;
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+}
>+
>+TYPED_TEST_P(ConstructorTest, AssignmentOverwritesExisting) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  TypeParam m({gen(), gen(), gen()});
>+  TypeParam n({gen()});
>+  n = m;
>+  EXPECT_EQ(m, n);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, MoveAssignmentOverwritesExisting) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  TypeParam m({gen(), gen(), gen()});
>+  TypeParam t(m);
>+  TypeParam n({gen()});
>+  n = std::move(t);
>+  EXPECT_EQ(m, n);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, AssignmentFromInitializerListOverwritesExisting) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  TypeParam m;
>+  m = values;
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+}
>+
>+TYPED_TEST_P(ConstructorTest, AssignmentOnSelf) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  TypeParam m(values);
>+  m = *&m;  // Avoid -Wself-assign
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+}
>+
>+// We cannot test self move as standard states that it leaves standard
>+// containers in unspecified state (and in practice in causes memory-leak
>+// according to heap-checker!).
>+
>+REGISTER_TYPED_TEST_CASE_P(
>+    ConstructorTest, NoArgs, BucketCount, BucketCountHash, BucketCountHashEqual,
>+    BucketCountHashEqualAlloc, BucketCountAlloc, BucketCountHashAlloc,
>+    BucketAlloc, InputIteratorBucketHashEqualAlloc, InputIteratorBucketAlloc,
>+    InputIteratorBucketHashAlloc, CopyConstructor, CopyConstructorAlloc,
>+    MoveConstructor, MoveConstructorAlloc, InitializerListBucketHashEqualAlloc,
>+    InitializerListBucketAlloc, InitializerListBucketHashAlloc, Assignment,
>+    MoveAssignment, AssignmentFromInitializerList,
>+    AssignmentOverwritesExisting, MoveAssignmentOverwritesExisting,
>+    AssignmentFromInitializerListOverwritesExisting, AssignmentOnSelf);
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+#endif  // ABSL_CONTAINER_INTERNAL_UNORDERED_MAP_CONSTRUCTOR_TEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_lookup_test.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_lookup_test.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..1f1b6b489b301eb9fadd1bf84919cad637c9ab8b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_lookup_test.h
>@@ -0,0 +1,114 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_UNORDERED_MAP_LOOKUP_TEST_H_
>+#define ABSL_CONTAINER_INTERNAL_UNORDERED_MAP_LOOKUP_TEST_H_
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/container/internal/hash_generator_testing.h"
>+#include "absl/container/internal/hash_policy_testing.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+template <class UnordMap>
>+class LookupTest : public ::testing::Test {};
>+
>+TYPED_TEST_CASE_P(LookupTest);
>+
>+TYPED_TEST_P(LookupTest, At) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end());
>+  for (const auto& p : values) {
>+    const auto& val = m.at(p.first);
>+    EXPECT_EQ(p.second, val) << ::testing::PrintToString(p.first);
>+  }
>+}
>+
>+TYPED_TEST_P(LookupTest, OperatorBracket) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using V = typename TypeParam::mapped_type;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m;
>+  for (const auto& p : values) {
>+    auto& val = m[p.first];
>+    EXPECT_EQ(V(), val) << ::testing::PrintToString(p.first);
>+    val = p.second;
>+  }
>+  for (const auto& p : values)
>+    EXPECT_EQ(p.second, m[p.first]) << ::testing::PrintToString(p.first);
>+}
>+
>+TYPED_TEST_P(LookupTest, Count) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m;
>+  for (const auto& p : values)
>+    EXPECT_EQ(0, m.count(p.first)) << ::testing::PrintToString(p.first);
>+  m.insert(values.begin(), values.end());
>+  for (const auto& p : values)
>+    EXPECT_EQ(1, m.count(p.first)) << ::testing::PrintToString(p.first);
>+}
>+
>+TYPED_TEST_P(LookupTest, Find) {
>+  using std::get;
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m;
>+  for (const auto& p : values)
>+    EXPECT_TRUE(m.end() == m.find(p.first))
>+        << ::testing::PrintToString(p.first);
>+  m.insert(values.begin(), values.end());
>+  for (const auto& p : values) {
>+    auto it = m.find(p.first);
>+    EXPECT_TRUE(m.end() != it) << ::testing::PrintToString(p.first);
>+    EXPECT_EQ(p.second, get<1>(*it)) << ::testing::PrintToString(p.first);
>+  }
>+}
>+
>+TYPED_TEST_P(LookupTest, EqualRange) {
>+  using std::get;
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m;
>+  for (const auto& p : values) {
>+    auto r = m.equal_range(p.first);
>+    ASSERT_EQ(0, std::distance(r.first, r.second));
>+  }
>+  m.insert(values.begin(), values.end());
>+  for (const auto& p : values) {
>+    auto r = m.equal_range(p.first);
>+    ASSERT_EQ(1, std::distance(r.first, r.second));
>+    EXPECT_EQ(p.second, get<1>(*r.first)) << ::testing::PrintToString(p.first);
>+  }
>+}
>+
>+REGISTER_TYPED_TEST_CASE_P(LookupTest, At, OperatorBracket, Count, Find,
>+                           EqualRange);
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+#endif  // ABSL_CONTAINER_INTERNAL_UNORDERED_MAP_LOOKUP_TEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_modifiers_test.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_modifiers_test.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..b6c633ae27357c145e6e1f206ba2d2cc24c8854a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_modifiers_test.h
>@@ -0,0 +1,272 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_UNORDERED_MAP_MODIFIERS_TEST_H_
>+#define ABSL_CONTAINER_INTERNAL_UNORDERED_MAP_MODIFIERS_TEST_H_
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/container/internal/hash_generator_testing.h"
>+#include "absl/container/internal/hash_policy_testing.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+template <class UnordMap>
>+class ModifiersTest : public ::testing::Test {};
>+
>+TYPED_TEST_CASE_P(ModifiersTest);
>+
>+TYPED_TEST_P(ModifiersTest, Clear) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end());
>+  ASSERT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+  m.clear();
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAre());
>+  EXPECT_TRUE(m.empty());
>+}
>+
>+TYPED_TEST_P(ModifiersTest, Insert) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using V = typename TypeParam::mapped_type;
>+  T val = hash_internal::Generator<T>()();
>+  TypeParam m;
>+  auto p = m.insert(val);
>+  EXPECT_TRUE(p.second);
>+  EXPECT_EQ(val, *p.first);
>+  T val2 = {val.first, hash_internal::Generator<V>()()};
>+  p = m.insert(val2);
>+  EXPECT_FALSE(p.second);
>+  EXPECT_EQ(val, *p.first);
>+}
>+
>+TYPED_TEST_P(ModifiersTest, InsertHint) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using V = typename TypeParam::mapped_type;
>+  T val = hash_internal::Generator<T>()();
>+  TypeParam m;
>+  auto it = m.insert(m.end(), val);
>+  EXPECT_TRUE(it != m.end());
>+  EXPECT_EQ(val, *it);
>+  T val2 = {val.first, hash_internal::Generator<V>()()};
>+  it = m.insert(it, val2);
>+  EXPECT_TRUE(it != m.end());
>+  EXPECT_EQ(val, *it);
>+}
>+
>+TYPED_TEST_P(ModifiersTest, InsertRange) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m;
>+  m.insert(values.begin(), values.end());
>+  ASSERT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+}
>+
>+TYPED_TEST_P(ModifiersTest, InsertOrAssign) {
>+#ifdef UNORDERED_MAP_CXX17
>+  using std::get;
>+  using K = typename TypeParam::key_type;
>+  using V = typename TypeParam::mapped_type;
>+  K k = hash_internal::Generator<K>()();
>+  V val = hash_internal::Generator<V>()();
>+  TypeParam m;
>+  auto p = m.insert_or_assign(k, val);
>+  EXPECT_TRUE(p.second);
>+  EXPECT_EQ(k, get<0>(*p.first));
>+  EXPECT_EQ(val, get<1>(*p.first));
>+  V val2 = hash_internal::Generator<V>()();
>+  p = m.insert_or_assign(k, val2);
>+  EXPECT_FALSE(p.second);
>+  EXPECT_EQ(k, get<0>(*p.first));
>+  EXPECT_EQ(val2, get<1>(*p.first));
>+#endif
>+}
>+
>+TYPED_TEST_P(ModifiersTest, InsertOrAssignHint) {
>+#ifdef UNORDERED_MAP_CXX17
>+  using std::get;
>+  using K = typename TypeParam::key_type;
>+  using V = typename TypeParam::mapped_type;
>+  K k = hash_internal::Generator<K>()();
>+  V val = hash_internal::Generator<V>()();
>+  TypeParam m;
>+  auto it = m.insert_or_assign(m.end(), k, val);
>+  EXPECT_TRUE(it != m.end());
>+  EXPECT_EQ(k, get<0>(*it));
>+  EXPECT_EQ(val, get<1>(*it));
>+  V val2 = hash_internal::Generator<V>()();
>+  it = m.insert_or_assign(it, k, val2);
>+  EXPECT_EQ(k, get<0>(*it));
>+  EXPECT_EQ(val2, get<1>(*it));
>+#endif
>+}
>+
>+TYPED_TEST_P(ModifiersTest, Emplace) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using V = typename TypeParam::mapped_type;
>+  T val = hash_internal::Generator<T>()();
>+  TypeParam m;
>+  // TODO(alkis): We need a way to run emplace in a more meaningful way. Perhaps
>+  // with test traits/policy.
>+  auto p = m.emplace(val);
>+  EXPECT_TRUE(p.second);
>+  EXPECT_EQ(val, *p.first);
>+  T val2 = {val.first, hash_internal::Generator<V>()()};
>+  p = m.emplace(val2);
>+  EXPECT_FALSE(p.second);
>+  EXPECT_EQ(val, *p.first);
>+}
>+
>+TYPED_TEST_P(ModifiersTest, EmplaceHint) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using V = typename TypeParam::mapped_type;
>+  T val = hash_internal::Generator<T>()();
>+  TypeParam m;
>+  // TODO(alkis): We need a way to run emplace in a more meaningful way. Perhaps
>+  // with test traits/policy.
>+  auto it = m.emplace_hint(m.end(), val);
>+  EXPECT_EQ(val, *it);
>+  T val2 = {val.first, hash_internal::Generator<V>()()};
>+  it = m.emplace_hint(it, val2);
>+  EXPECT_EQ(val, *it);
>+}
>+
>+TYPED_TEST_P(ModifiersTest, TryEmplace) {
>+#ifdef UNORDERED_MAP_CXX17
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using V = typename TypeParam::mapped_type;
>+  T val = hash_internal::Generator<T>()();
>+  TypeParam m;
>+  // TODO(alkis): We need a way to run emplace in a more meaningful way. Perhaps
>+  // with test traits/policy.
>+  auto p = m.try_emplace(val.first, val.second);
>+  EXPECT_TRUE(p.second);
>+  EXPECT_EQ(val, *p.first);
>+  T val2 = {val.first, hash_internal::Generator<V>()()};
>+  p = m.try_emplace(val2.first, val2.second);
>+  EXPECT_FALSE(p.second);
>+  EXPECT_EQ(val, *p.first);
>+#endif
>+}
>+
>+TYPED_TEST_P(ModifiersTest, TryEmplaceHint) {
>+#ifdef UNORDERED_MAP_CXX17
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using V = typename TypeParam::mapped_type;
>+  T val = hash_internal::Generator<T>()();
>+  TypeParam m;
>+  // TODO(alkis): We need a way to run emplace in a more meaningful way. Perhaps
>+  // with test traits/policy.
>+  auto it = m.try_emplace(m.end(), val.first, val.second);
>+  EXPECT_EQ(val, *it);
>+  T val2 = {val.first, hash_internal::Generator<V>()()};
>+  it = m.try_emplace(it, val2.first, val2.second);
>+  EXPECT_EQ(val, *it);
>+#endif
>+}
>+
>+template <class V>
>+using IfNotVoid = typename std::enable_if<!std::is_void<V>::value, V>::type;
>+
>+// In openmap we chose not to return the iterator from erase because that's
>+// more expensive. As such we adapt erase to return an iterator here.
>+struct EraseFirst {
>+  template <class Map>
>+  auto operator()(Map* m, int) const
>+      -> IfNotVoid<decltype(m->erase(m->begin()))> {
>+    return m->erase(m->begin());
>+  }
>+  template <class Map>
>+  typename Map::iterator operator()(Map* m, ...) const {
>+    auto it = m->begin();
>+    m->erase(it++);
>+    return it;
>+  }
>+};
>+
>+TYPED_TEST_P(ModifiersTest, Erase) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using std::get;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end());
>+  ASSERT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+  auto& first = *m.begin();
>+  std::vector<T> values2;
>+  for (const auto& val : values)
>+    if (get<0>(val) != get<0>(first)) values2.push_back(val);
>+  auto it = EraseFirst()(&m, 0);
>+  ASSERT_TRUE(it != m.end());
>+  EXPECT_EQ(1, std::count(values2.begin(), values2.end(), *it));
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAreArray(values2.begin(),
>+                                                             values2.end()));
>+}
>+
>+TYPED_TEST_P(ModifiersTest, EraseRange) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end());
>+  ASSERT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+  auto it = m.erase(m.begin(), m.end());
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAre());
>+  EXPECT_TRUE(it == m.end());
>+}
>+
>+TYPED_TEST_P(ModifiersTest, EraseKey) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end());
>+  ASSERT_THAT(items(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_EQ(1, m.erase(values[0].first));
>+  EXPECT_EQ(0, std::count(m.begin(), m.end(), values[0]));
>+  EXPECT_THAT(items(m), ::testing::UnorderedElementsAreArray(values.begin() + 1,
>+                                                             values.end()));
>+}
>+
>+TYPED_TEST_P(ModifiersTest, Swap) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> v1;
>+  std::vector<T> v2;
>+  std::generate_n(std::back_inserter(v1), 5, hash_internal::Generator<T>());
>+  std::generate_n(std::back_inserter(v2), 5, hash_internal::Generator<T>());
>+  TypeParam m1(v1.begin(), v1.end());
>+  TypeParam m2(v2.begin(), v2.end());
>+  EXPECT_THAT(items(m1), ::testing::UnorderedElementsAreArray(v1));
>+  EXPECT_THAT(items(m2), ::testing::UnorderedElementsAreArray(v2));
>+  m1.swap(m2);
>+  EXPECT_THAT(items(m1), ::testing::UnorderedElementsAreArray(v2));
>+  EXPECT_THAT(items(m2), ::testing::UnorderedElementsAreArray(v1));
>+}
>+
>+// TODO(alkis): Write tests for extract.
>+// TODO(alkis): Write tests for merge.
>+
>+REGISTER_TYPED_TEST_CASE_P(ModifiersTest, Clear, Insert, InsertHint,
>+                           InsertRange, InsertOrAssign, InsertOrAssignHint,
>+                           Emplace, EmplaceHint, TryEmplace, TryEmplaceHint,
>+                           Erase, EraseRange, EraseKey, Swap);
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+#endif  // ABSL_CONTAINER_INTERNAL_UNORDERED_MAP_MODIFIERS_TEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..40e799cda89c3c90553b38db7ab32a2c14ed0280
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_map_test.cc
>@@ -0,0 +1,38 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <unordered_map>
>+
>+#include "absl/container/internal/unordered_map_constructor_test.h"
>+#include "absl/container/internal/unordered_map_lookup_test.h"
>+#include "absl/container/internal/unordered_map_modifiers_test.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+using MapTypes = ::testing::Types<
>+    std::unordered_map<int, int, StatefulTestingHash, StatefulTestingEqual,
>+                       Alloc<std::pair<const int, int>>>,
>+    std::unordered_map<std::string, std::string, StatefulTestingHash,
>+                       StatefulTestingEqual,
>+                       Alloc<std::pair<const std::string, std::string>>>>;
>+
>+INSTANTIATE_TYPED_TEST_CASE_P(UnorderedMap, ConstructorTest, MapTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(UnorderedMap, LookupTest, MapTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(UnorderedMap, ModifiersTest, MapTypes);
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_constructor_test.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_constructor_test.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..cb593704685c47faba1303e2c5397bc155244e22
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_constructor_test.h
>@@ -0,0 +1,408 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_UNORDERED_SET_CONSTRUCTOR_TEST_H_
>+#define ABSL_CONTAINER_INTERNAL_UNORDERED_SET_CONSTRUCTOR_TEST_H_
>+
>+#include <algorithm>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/container/internal/hash_generator_testing.h"
>+#include "absl/container/internal/hash_policy_testing.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+template <class UnordMap>
>+class ConstructorTest : public ::testing::Test {};
>+
>+TYPED_TEST_CASE_P(ConstructorTest);
>+
>+TYPED_TEST_P(ConstructorTest, NoArgs) {
>+  TypeParam m;
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAre());
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCount) {
>+  TypeParam m(123);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCountHash) {
>+  using H = typename TypeParam::hasher;
>+  H hasher;
>+  TypeParam m(123, hasher);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCountHashEqual) {
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  H hasher;
>+  E equal;
>+  TypeParam m(123, hasher, equal);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.key_eq(), equal);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCountHashEqualAlloc) {
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(123, hasher, equal, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.key_eq(), equal);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+
>+  const auto& cm = m;
>+  EXPECT_EQ(cm.hash_function(), hasher);
>+  EXPECT_EQ(cm.key_eq(), equal);
>+  EXPECT_EQ(cm.get_allocator(), alloc);
>+  EXPECT_TRUE(cm.empty());
>+  EXPECT_THAT(keys(cm), ::testing::UnorderedElementsAre());
>+  EXPECT_GE(cm.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCountAlloc) {
>+#if defined(UNORDERED_SET_CXX14) || defined(UNORDERED_SET_CXX17)
>+  using A = typename TypeParam::allocator_type;
>+  A alloc(0);
>+  TypeParam m(123, alloc);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketCountHashAlloc) {
>+#if defined(UNORDERED_SET_CXX14) || defined(UNORDERED_SET_CXX17)
>+  using H = typename TypeParam::hasher;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  A alloc(0);
>+  TypeParam m(123, hasher, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAre());
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, BucketAlloc) {
>+#if ABSL_UNORDERED_SUPPORTS_ALLOC_CTORS
>+  using A = typename TypeParam::allocator_type;
>+  A alloc(0);
>+  TypeParam m(alloc);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_TRUE(m.empty());
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAre());
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, InputIteratorBucketHashEqualAlloc) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  std::vector<T> values;
>+  for (size_t i = 0; i != 10; ++i)
>+    values.push_back(hash_internal::Generator<T>()());
>+  TypeParam m(values.begin(), values.end(), 123, hasher, equal, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.key_eq(), equal);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, InputIteratorBucketAlloc) {
>+#if defined(UNORDERED_SET_CXX14) || defined(UNORDERED_SET_CXX17)
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using A = typename TypeParam::allocator_type;
>+  A alloc(0);
>+  std::vector<T> values;
>+  for (size_t i = 0; i != 10; ++i)
>+    values.push_back(hash_internal::Generator<T>()());
>+  TypeParam m(values.begin(), values.end(), 123, alloc);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, InputIteratorBucketHashAlloc) {
>+#if defined(UNORDERED_SET_CXX14) || defined(UNORDERED_SET_CXX17)
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  A alloc(0);
>+  std::vector<T> values;
>+  for (size_t i = 0; i != 10; ++i)
>+    values.push_back(hash_internal::Generator<T>()());
>+  TypeParam m(values.begin(), values.end(), 123, hasher, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, CopyConstructor) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(123, hasher, equal, alloc);
>+  for (size_t i = 0; i != 10; ++i) m.insert(hash_internal::Generator<T>()());
>+  TypeParam n(m);
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_EQ(m.get_allocator(), n.get_allocator());
>+  EXPECT_EQ(m, n);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, CopyConstructorAlloc) {
>+#if ABSL_UNORDERED_SUPPORTS_ALLOC_CTORS
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(123, hasher, equal, alloc);
>+  for (size_t i = 0; i != 10; ++i) m.insert(hash_internal::Generator<T>()());
>+  TypeParam n(m, A(11));
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_NE(m.get_allocator(), n.get_allocator());
>+  EXPECT_EQ(m, n);
>+#endif
>+}
>+
>+// TODO(alkis): Test non-propagating allocators on copy constructors.
>+
>+TYPED_TEST_P(ConstructorTest, MoveConstructor) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(123, hasher, equal, alloc);
>+  for (size_t i = 0; i != 10; ++i) m.insert(hash_internal::Generator<T>()());
>+  TypeParam t(m);
>+  TypeParam n(std::move(t));
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_EQ(m.get_allocator(), n.get_allocator());
>+  EXPECT_EQ(m, n);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, MoveConstructorAlloc) {
>+#if ABSL_UNORDERED_SUPPORTS_ALLOC_CTORS
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(123, hasher, equal, alloc);
>+  for (size_t i = 0; i != 10; ++i) m.insert(hash_internal::Generator<T>()());
>+  TypeParam t(m);
>+  TypeParam n(std::move(t), A(1));
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_NE(m.get_allocator(), n.get_allocator());
>+  EXPECT_EQ(m, n);
>+#endif
>+}
>+
>+// TODO(alkis): Test non-propagating allocators on move constructors.
>+
>+TYPED_TEST_P(ConstructorTest, InitializerListBucketHashEqualAlloc) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  TypeParam m(values, 123, hasher, equal, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.key_eq(), equal);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, InitializerListBucketAlloc) {
>+#if defined(UNORDERED_SET_CXX14) || defined(UNORDERED_SET_CXX17)
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using A = typename TypeParam::allocator_type;
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  A alloc(0);
>+  TypeParam m(values, 123, alloc);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, InitializerListBucketHashAlloc) {
>+#if defined(UNORDERED_SET_CXX14) || defined(UNORDERED_SET_CXX17)
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  A alloc(0);
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  TypeParam m(values, 123, hasher, alloc);
>+  EXPECT_EQ(m.hash_function(), hasher);
>+  EXPECT_EQ(m.get_allocator(), alloc);
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_GE(m.bucket_count(), 123);
>+#endif
>+}
>+
>+TYPED_TEST_P(ConstructorTest, Assignment) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  hash_internal::Generator<T> gen;
>+  TypeParam m({gen(), gen(), gen()}, 123, hasher, equal, alloc);
>+  TypeParam n;
>+  n = m;
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_EQ(m, n);
>+}
>+
>+// TODO(alkis): Test [non-]propagating allocators on move/copy assignments
>+// (it depends on traits).
>+
>+TYPED_TEST_P(ConstructorTest, MoveAssignment) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  using H = typename TypeParam::hasher;
>+  using E = typename TypeParam::key_equal;
>+  using A = typename TypeParam::allocator_type;
>+  H hasher;
>+  E equal;
>+  A alloc(0);
>+  hash_internal::Generator<T> gen;
>+  TypeParam m({gen(), gen(), gen()}, 123, hasher, equal, alloc);
>+  TypeParam t(m);
>+  TypeParam n;
>+  n = std::move(t);
>+  EXPECT_EQ(m.hash_function(), n.hash_function());
>+  EXPECT_EQ(m.key_eq(), n.key_eq());
>+  EXPECT_EQ(m, n);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, AssignmentFromInitializerList) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  TypeParam m;
>+  m = values;
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+}
>+
>+TYPED_TEST_P(ConstructorTest, AssignmentOverwritesExisting) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  TypeParam m({gen(), gen(), gen()});
>+  TypeParam n({gen()});
>+  n = m;
>+  EXPECT_EQ(m, n);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, MoveAssignmentOverwritesExisting) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  TypeParam m({gen(), gen(), gen()});
>+  TypeParam t(m);
>+  TypeParam n({gen()});
>+  n = std::move(t);
>+  EXPECT_EQ(m, n);
>+}
>+
>+TYPED_TEST_P(ConstructorTest, AssignmentFromInitializerListOverwritesExisting) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  TypeParam m;
>+  m = values;
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+}
>+
>+TYPED_TEST_P(ConstructorTest, AssignmentOnSelf) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  hash_internal::Generator<T> gen;
>+  std::initializer_list<T> values = {gen(), gen(), gen(), gen(), gen()};
>+  TypeParam m(values);
>+  m = *&m;  // Avoid -Wself-assign.
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+}
>+
>+REGISTER_TYPED_TEST_CASE_P(
>+    ConstructorTest, NoArgs, BucketCount, BucketCountHash, BucketCountHashEqual,
>+    BucketCountHashEqualAlloc, BucketCountAlloc, BucketCountHashAlloc,
>+    BucketAlloc, InputIteratorBucketHashEqualAlloc, InputIteratorBucketAlloc,
>+    InputIteratorBucketHashAlloc, CopyConstructor, CopyConstructorAlloc,
>+    MoveConstructor, MoveConstructorAlloc, InitializerListBucketHashEqualAlloc,
>+    InitializerListBucketAlloc, InitializerListBucketHashAlloc, Assignment,
>+    MoveAssignment, AssignmentFromInitializerList,
>+    AssignmentOverwritesExisting, MoveAssignmentOverwritesExisting,
>+    AssignmentFromInitializerListOverwritesExisting, AssignmentOnSelf);
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+#endif  // ABSL_CONTAINER_INTERNAL_UNORDERED_SET_CONSTRUCTOR_TEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_lookup_test.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_lookup_test.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..aca9c6a5df7b725c175993f47069da4e921729d3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_lookup_test.h
>@@ -0,0 +1,88 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_UNORDERED_SET_LOOKUP_TEST_H_
>+#define ABSL_CONTAINER_INTERNAL_UNORDERED_SET_LOOKUP_TEST_H_
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/container/internal/hash_generator_testing.h"
>+#include "absl/container/internal/hash_policy_testing.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+template <class UnordSet>
>+class LookupTest : public ::testing::Test {};
>+
>+TYPED_TEST_CASE_P(LookupTest);
>+
>+TYPED_TEST_P(LookupTest, Count) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m;
>+  for (const auto& v : values)
>+    EXPECT_EQ(0, m.count(v)) << ::testing::PrintToString(v);
>+  m.insert(values.begin(), values.end());
>+  for (const auto& v : values)
>+    EXPECT_EQ(1, m.count(v)) << ::testing::PrintToString(v);
>+}
>+
>+TYPED_TEST_P(LookupTest, Find) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m;
>+  for (const auto& v : values)
>+    EXPECT_TRUE(m.end() == m.find(v)) << ::testing::PrintToString(v);
>+  m.insert(values.begin(), values.end());
>+  for (const auto& v : values) {
>+    typename TypeParam::iterator it = m.find(v);
>+    static_assert(std::is_same<const typename TypeParam::value_type&,
>+                               decltype(*it)>::value,
>+                  "");
>+    static_assert(std::is_same<const typename TypeParam::value_type*,
>+                               decltype(it.operator->())>::value,
>+                  "");
>+    EXPECT_TRUE(m.end() != it) << ::testing::PrintToString(v);
>+    EXPECT_EQ(v, *it) << ::testing::PrintToString(v);
>+  }
>+}
>+
>+TYPED_TEST_P(LookupTest, EqualRange) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m;
>+  for (const auto& v : values) {
>+    auto r = m.equal_range(v);
>+    ASSERT_EQ(0, std::distance(r.first, r.second));
>+  }
>+  m.insert(values.begin(), values.end());
>+  for (const auto& v : values) {
>+    auto r = m.equal_range(v);
>+    ASSERT_EQ(1, std::distance(r.first, r.second));
>+    EXPECT_EQ(v, *r.first);
>+  }
>+}
>+
>+REGISTER_TYPED_TEST_CASE_P(LookupTest, Count, Find, EqualRange);
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+#endif  // ABSL_CONTAINER_INTERNAL_UNORDERED_SET_LOOKUP_TEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_modifiers_test.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_modifiers_test.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..9beacf3316973dd7bcdcb9382a136c18d0a8c76b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_modifiers_test.h
>@@ -0,0 +1,187 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_UNORDERED_SET_MODIFIERS_TEST_H_
>+#define ABSL_CONTAINER_INTERNAL_UNORDERED_SET_MODIFIERS_TEST_H_
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/container/internal/hash_generator_testing.h"
>+#include "absl/container/internal/hash_policy_testing.h"
>+
>+namespace absl {
>+namespace container_internal {
>+
>+template <class UnordSet>
>+class ModifiersTest : public ::testing::Test {};
>+
>+TYPED_TEST_CASE_P(ModifiersTest);
>+
>+TYPED_TEST_P(ModifiersTest, Clear) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end());
>+  ASSERT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+  m.clear();
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAre());
>+  EXPECT_TRUE(m.empty());
>+}
>+
>+TYPED_TEST_P(ModifiersTest, Insert) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  T val = hash_internal::Generator<T>()();
>+  TypeParam m;
>+  auto p = m.insert(val);
>+  EXPECT_TRUE(p.second);
>+  EXPECT_EQ(val, *p.first);
>+  p = m.insert(val);
>+  EXPECT_FALSE(p.second);
>+}
>+
>+TYPED_TEST_P(ModifiersTest, InsertHint) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  T val = hash_internal::Generator<T>()();
>+  TypeParam m;
>+  auto it = m.insert(m.end(), val);
>+  EXPECT_TRUE(it != m.end());
>+  EXPECT_EQ(val, *it);
>+  it = m.insert(it, val);
>+  EXPECT_TRUE(it != m.end());
>+  EXPECT_EQ(val, *it);
>+}
>+
>+TYPED_TEST_P(ModifiersTest, InsertRange) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m;
>+  m.insert(values.begin(), values.end());
>+  ASSERT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+}
>+
>+TYPED_TEST_P(ModifiersTest, Emplace) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  T val = hash_internal::Generator<T>()();
>+  TypeParam m;
>+  // TODO(alkis): We need a way to run emplace in a more meaningful way. Perhaps
>+  // with test traits/policy.
>+  auto p = m.emplace(val);
>+  EXPECT_TRUE(p.second);
>+  EXPECT_EQ(val, *p.first);
>+  p = m.emplace(val);
>+  EXPECT_FALSE(p.second);
>+  EXPECT_EQ(val, *p.first);
>+}
>+
>+TYPED_TEST_P(ModifiersTest, EmplaceHint) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  T val = hash_internal::Generator<T>()();
>+  TypeParam m;
>+  // TODO(alkis): We need a way to run emplace in a more meaningful way. Perhaps
>+  // with test traits/policy.
>+  auto it = m.emplace_hint(m.end(), val);
>+  EXPECT_EQ(val, *it);
>+  it = m.emplace_hint(it, val);
>+  EXPECT_EQ(val, *it);
>+}
>+
>+template <class V>
>+using IfNotVoid = typename std::enable_if<!std::is_void<V>::value, V>::type;
>+
>+// In openmap we chose not to return the iterator from erase because that's
>+// more expensive. As such we adapt erase to return an iterator here.
>+struct EraseFirst {
>+  template <class Map>
>+  auto operator()(Map* m, int) const
>+      -> IfNotVoid<decltype(m->erase(m->begin()))> {
>+    return m->erase(m->begin());
>+  }
>+  template <class Map>
>+  typename Map::iterator operator()(Map* m, ...) const {
>+    auto it = m->begin();
>+    m->erase(it++);
>+    return it;
>+  }
>+};
>+
>+TYPED_TEST_P(ModifiersTest, Erase) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end());
>+  ASSERT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+  std::vector<T> values2;
>+  for (const auto& val : values)
>+    if (val != *m.begin()) values2.push_back(val);
>+  auto it = EraseFirst()(&m, 0);
>+  ASSERT_TRUE(it != m.end());
>+  EXPECT_EQ(1, std::count(values2.begin(), values2.end(), *it));
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values2.begin(),
>+                                                            values2.end()));
>+}
>+
>+TYPED_TEST_P(ModifiersTest, EraseRange) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end());
>+  ASSERT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+  auto it = m.erase(m.begin(), m.end());
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAre());
>+  EXPECT_TRUE(it == m.end());
>+}
>+
>+TYPED_TEST_P(ModifiersTest, EraseKey) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> values;
>+  std::generate_n(std::back_inserter(values), 10,
>+                  hash_internal::Generator<T>());
>+  TypeParam m(values.begin(), values.end());
>+  ASSERT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values));
>+  EXPECT_EQ(1, m.erase(values[0]));
>+  EXPECT_EQ(0, std::count(m.begin(), m.end(), values[0]));
>+  EXPECT_THAT(keys(m), ::testing::UnorderedElementsAreArray(values.begin() + 1,
>+                                                            values.end()));
>+}
>+
>+TYPED_TEST_P(ModifiersTest, Swap) {
>+  using T = hash_internal::GeneratedType<TypeParam>;
>+  std::vector<T> v1;
>+  std::vector<T> v2;
>+  std::generate_n(std::back_inserter(v1), 5, hash_internal::Generator<T>());
>+  std::generate_n(std::back_inserter(v2), 5, hash_internal::Generator<T>());
>+  TypeParam m1(v1.begin(), v1.end());
>+  TypeParam m2(v2.begin(), v2.end());
>+  EXPECT_THAT(keys(m1), ::testing::UnorderedElementsAreArray(v1));
>+  EXPECT_THAT(keys(m2), ::testing::UnorderedElementsAreArray(v2));
>+  m1.swap(m2);
>+  EXPECT_THAT(keys(m1), ::testing::UnorderedElementsAreArray(v2));
>+  EXPECT_THAT(keys(m2), ::testing::UnorderedElementsAreArray(v1));
>+}
>+
>+// TODO(alkis): Write tests for extract.
>+// TODO(alkis): Write tests for merge.
>+
>+REGISTER_TYPED_TEST_CASE_P(ModifiersTest, Clear, Insert, InsertHint,
>+                           InsertRange, Emplace, EmplaceHint, Erase, EraseRange,
>+                           EraseKey, Swap);
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+#endif  // ABSL_CONTAINER_INTERNAL_UNORDERED_SET_MODIFIERS_TEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..1281ce53d5e2e40728fb6ca8dd90c0c9498aefa3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/unordered_set_test.cc
>@@ -0,0 +1,37 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <unordered_set>
>+
>+#include "absl/container/internal/unordered_set_constructor_test.h"
>+#include "absl/container/internal/unordered_set_lookup_test.h"
>+#include "absl/container/internal/unordered_set_modifiers_test.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+using SetTypes =
>+    ::testing::Types<std::unordered_set<int, StatefulTestingHash,
>+                                        StatefulTestingEqual, Alloc<int>>,
>+                     std::unordered_set<std::string, StatefulTestingHash,
>+                                        StatefulTestingEqual, Alloc<std::string>>>;
>+
>+INSTANTIATE_TYPED_TEST_CASE_P(UnorderedSet, ConstructorTest, SetTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(UnorderedSet, LookupTest, SetTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(UnorderedSet, ModifiersTest, SetTypes);
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_map.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_map.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..6369ec3ad6431a47e64316588e46714317e61d9a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_map.h
>@@ -0,0 +1,530 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: node_hash_map.h
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::node_hash_map<K, V>` is an unordered associative container of
>+// unique keys and associated values designed to be a more efficient replacement
>+// for `std::unordered_map`. Like `unordered_map`, search, insertion, and
>+// deletion of map elements can be done as an `O(1)` operation. However,
>+// `node_hash_map` (and other unordered associative containers known as the
>+// collection of Abseil "Swiss tables") contain other optimizations that result
>+// in both memory and computation advantages.
>+//
>+// In most cases, your default choice for a hash map should be a map of type
>+// `flat_hash_map`. However, if you need pointer stability and cannot store
>+// a `flat_hash_map` with `unique_ptr` elements, a `node_hash_map` may be a
>+// valid alternative. As well, if you are migrating your code from using
>+// `std::unordered_map`, a `node_hash_map` provides a more straightforward
>+// migration, because it guarantees pointer stability. Consider migrating to
>+// `node_hash_map` and perhaps converting to a more efficient `flat_hash_map`
>+// upon further review.
>+
>+#ifndef ABSL_CONTAINER_NODE_HASH_MAP_H_
>+#define ABSL_CONTAINER_NODE_HASH_MAP_H_
>+
>+#include <tuple>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/container/internal/container_memory.h"
>+#include "absl/container/internal/hash_function_defaults.h"  // IWYU pragma: export
>+#include "absl/container/internal/node_hash_policy.h"
>+#include "absl/container/internal/raw_hash_map.h"  // IWYU pragma: export
>+#include "absl/memory/memory.h"
>+
>+namespace absl {
>+namespace container_internal {
>+template <class Key, class Value>
>+class NodeHashMapPolicy;
>+}  // namespace container_internal
>+
>+// -----------------------------------------------------------------------------
>+// absl::node_hash_map
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::node_hash_map<K, V>` is an unordered associative container which
>+// has been optimized for both speed and memory footprint in most common use
>+// cases. Its interface is similar to that of `std::unordered_map<K, V>` with
>+// the following notable differences:
>+//
>+// * Supports heterogeneous lookup, through `find()`, `operator[]()` and
>+//   `insert()`, provided that the map is provided a compatible heterogeneous
>+//   hashing function and equality operator.
>+// * Contains a `capacity()` member function indicating the number of element
>+//   slots (open, deleted, and empty) within the hash map.
>+// * Returns `void` from the `erase(iterator)` overload.
>+//
>+// By default, `node_hash_map` uses the `absl::Hash` hashing framework.
>+// All fundamental and Abseil types that support the `absl::Hash` framework have
>+// a compatible equality operator for comparing insertions into `node_hash_map`.
>+// If your type is not yet supported by the `asbl::Hash` framework, see
>+// absl/hash/hash.h for information on extending Abseil hashing to user-defined
>+// types.
>+//
>+// Example:
>+//
>+//   // Create a node hash map of three strings (that map to strings)
>+//   absl::node_hash_map<std::string, std::string> ducks =
>+//     {{"a", "huey"}, {"b", "dewey"}, {"c", "louie"}};
>+//
>+//  // Insert a new element into the node hash map
>+//  ducks.insert({"d", "donald"}};
>+//
>+//  // Force a rehash of the node hash map
>+//  ducks.rehash(0);
>+//
>+//  // Find the element with the key "b"
>+//  std::string search_key = "b";
>+//  auto result = ducks.find(search_key);
>+//  if (result != ducks.end()) {
>+//    std::cout << "Result: " << search_key->second << std::endl;
>+//  }
>+template <class Key, class Value,
>+          class Hash = absl::container_internal::hash_default_hash<Key>,
>+          class Eq = absl::container_internal::hash_default_eq<Key>,
>+          class Alloc = std::allocator<std::pair<const Key, Value>>>
>+class node_hash_map
>+    : public absl::container_internal::raw_hash_map<
>+          absl::container_internal::NodeHashMapPolicy<Key, Value>, Hash, Eq,
>+          Alloc> {
>+  using Base = typename node_hash_map::raw_hash_map;
>+
>+ public:
>+  node_hash_map() {}
>+  using Base::Base;
>+
>+  // node_hash_map::begin()
>+  //
>+  // Returns an iterator to the beginning of the `node_hash_map`.
>+  using Base::begin;
>+
>+  // node_hash_map::cbegin()
>+  //
>+  // Returns a const iterator to the beginning of the `node_hash_map`.
>+  using Base::cbegin;
>+
>+  // node_hash_map::cend()
>+  //
>+  // Returns a const iterator to the end of the `node_hash_map`.
>+  using Base::cend;
>+
>+  // node_hash_map::end()
>+  //
>+  // Returns an iterator to the end of the `node_hash_map`.
>+  using Base::end;
>+
>+  // node_hash_map::capacity()
>+  //
>+  // Returns the number of element slots (assigned, deleted, and empty)
>+  // available within the `node_hash_map`.
>+  //
>+  // NOTE: this member function is particular to `absl::node_hash_map` and is
>+  // not provided in the `std::unordered_map` API.
>+  using Base::capacity;
>+
>+  // node_hash_map::empty()
>+  //
>+  // Returns whether or not the `node_hash_map` is empty.
>+  using Base::empty;
>+
>+  // node_hash_map::max_size()
>+  //
>+  // Returns the largest theoretical possible number of elements within a
>+  // `node_hash_map` under current memory constraints. This value can be thought
>+  // of as the largest value of `std::distance(begin(), end())` for a
>+  // `node_hash_map<K, V>`.
>+  using Base::max_size;
>+
>+  // node_hash_map::size()
>+  //
>+  // Returns the number of elements currently within the `node_hash_map`.
>+  using Base::size;
>+
>+  // node_hash_map::clear()
>+  //
>+  // Removes all elements from the `node_hash_map`. Invalidates any references,
>+  // pointers, or iterators referring to contained elements.
>+  //
>+  // NOTE: this operation may shrink the underlying buffer. To avoid shrinking
>+  // the underlying buffer call `erase(begin(), end())`.
>+  using Base::clear;
>+
>+  // node_hash_map::erase()
>+  //
>+  // Erases elements within the `node_hash_map`. Erasing does not trigger a
>+  // rehash. Overloads are listed below.
>+  //
>+  // void erase(const_iterator pos):
>+  //
>+  //   Erases the element at `position` of the `node_hash_map`, returning
>+  //   `void`.
>+  //
>+  //   NOTE: this return behavior is different than that of STL containers in
>+  //   general and `std::unordered_map` in particular.
>+  //
>+  // iterator erase(const_iterator first, const_iterator last):
>+  //
>+  //   Erases the elements in the open interval [`first`, `last`), returning an
>+  //   iterator pointing to `last`.
>+  //
>+  // size_type erase(const key_type& key):
>+  //
>+  //   Erases the element with the matching key, if it exists.
>+  using Base::erase;
>+
>+  // node_hash_map::insert()
>+  //
>+  // Inserts an element of the specified value into the `node_hash_map`,
>+  // returning an iterator pointing to the newly inserted element, provided that
>+  // an element with the given key does not already exist. If rehashing occurs
>+  // due to the insertion, all iterators are invalidated. Overloads are listed
>+  // below.
>+  //
>+  // std::pair<iterator,bool> insert(const init_type& value):
>+  //
>+  //   Inserts a value into the `node_hash_map`. Returns a pair consisting of an
>+  //   iterator to the inserted element (or to the element that prevented the
>+  //   insertion) and a `bool` denoting whether the insertion took place.
>+  //
>+  // std::pair<iterator,bool> insert(T&& value):
>+  // std::pair<iterator,bool> insert(init_type&& value):
>+  //
>+  //   Inserts a moveable value into the `node_hash_map`. Returns a `std::pair`
>+  //   consisting of an iterator to the inserted element (or to the element that
>+  //   prevented the insertion) and a `bool` denoting whether the insertion took
>+  //   place.
>+  //
>+  // iterator insert(const_iterator hint, const init_type& value):
>+  // iterator insert(const_iterator hint, T&& value):
>+  // iterator insert(const_iterator hint, init_type&& value);
>+  //
>+  //   Inserts a value, using the position of `hint` as a non-binding suggestion
>+  //   for where to begin the insertion search. Returns an iterator to the
>+  //   inserted element, or to the existing element that prevented the
>+  //   insertion.
>+  //
>+  // void insert(InputIterator first, InputIterator last):
>+  //
>+  //   Inserts a range of values [`first`, `last`).
>+  //
>+  //   NOTE: Although the STL does not specify which element may be inserted if
>+  //   multiple keys compare equivalently, for `node_hash_map` we guarantee the
>+  //   first match is inserted.
>+  //
>+  // void insert(std::initializer_list<init_type> ilist):
>+  //
>+  //   Inserts the elements within the initializer list `ilist`.
>+  //
>+  //   NOTE: Although the STL does not specify which element may be inserted if
>+  //   multiple keys compare equivalently within the initializer list, for
>+  //   `node_hash_map` we guarantee the first match is inserted.
>+  using Base::insert;
>+
>+  // node_hash_map::insert_or_assign()
>+  //
>+  // Inserts an element of the specified value into the `node_hash_map` provided
>+  // that a value with the given key does not already exist, or replaces it with
>+  // the element value if a key for that value already exists, returning an
>+  // iterator pointing to the newly inserted element. If rehashing occurs due to
>+  // the insertion, all iterators are invalidated. Overloads are listed
>+  // below.
>+  //
>+  // std::pair<iterator, bool> insert_or_assign(const init_type& k, T&& obj):
>+  // std::pair<iterator, bool> insert_or_assign(init_type&& k, T&& obj):
>+  //
>+  //   Inserts/Assigns (or moves) the element of the specified key into the
>+  //   `node_hash_map`.
>+  //
>+  // iterator insert_or_assign(const_iterator hint,
>+  //                           const init_type& k, T&& obj):
>+  // iterator insert_or_assign(const_iterator hint, init_type&& k, T&& obj):
>+  //
>+  //   Inserts/Assigns (or moves) the element of the specified key into the
>+  //   `node_hash_map` using the position of `hint` as a non-binding suggestion
>+  //   for where to begin the insertion search.
>+  using Base::insert_or_assign;
>+
>+  // node_hash_map::emplace()
>+  //
>+  // Inserts an element of the specified value by constructing it in-place
>+  // within the `node_hash_map`, provided that no element with the given key
>+  // already exists.
>+  //
>+  // The element may be constructed even if there already is an element with the
>+  // key in the container, in which case the newly constructed element will be
>+  // destroyed immediately. Prefer `try_emplace()` unless your key is not
>+  // copyable or moveable.
>+  //
>+  // If rehashing occurs due to the insertion, all iterators are invalidated.
>+  using Base::emplace;
>+
>+  // node_hash_map::emplace_hint()
>+  //
>+  // Inserts an element of the specified value by constructing it in-place
>+  // within the `node_hash_map`, using the position of `hint` as a non-binding
>+  // suggestion for where to begin the insertion search, and only inserts
>+  // provided that no element with the given key already exists.
>+  //
>+  // The element may be constructed even if there already is an element with the
>+  // key in the container, in which case the newly constructed element will be
>+  // destroyed immediately. Prefer `try_emplace()` unless your key is not
>+  // copyable or moveable.
>+  //
>+  // If rehashing occurs due to the insertion, all iterators are invalidated.
>+  using Base::emplace_hint;
>+
>+  // node_hash_map::try_emplace()
>+  //
>+  // Inserts an element of the specified value by constructing it in-place
>+  // within the `node_hash_map`, provided that no element with the given key
>+  // already exists. Unlike `emplace()`, if an element with the given key
>+  // already exists, we guarantee that no element is constructed.
>+  //
>+  // If rehashing occurs due to the insertion, all iterators are invalidated.
>+  // Overloads are listed below.
>+  //
>+  //   std::pair<iterator, bool> try_emplace(const key_type& k, Args&&... args):
>+  //   std::pair<iterator, bool> try_emplace(key_type&& k, Args&&... args):
>+  //
>+  // Inserts (via copy or move) the element of the specified key into the
>+  // `node_hash_map`.
>+  //
>+  //   iterator try_emplace(const_iterator hint,
>+  //                        const init_type& k, Args&&... args):
>+  //   iterator try_emplace(const_iterator hint, init_type&& k, Args&&... args):
>+  //
>+  // Inserts (via copy or move) the element of the specified key into the
>+  // `node_hash_map` using the position of `hint` as a non-binding suggestion
>+  // for where to begin the insertion search.
>+  using Base::try_emplace;
>+
>+  // node_hash_map::extract()
>+  //
>+  // Extracts the indicated element, erasing it in the process, and returns it
>+  // as a C++17-compatible node handle. Overloads are listed below.
>+  //
>+  // node_type extract(const_iterator position):
>+  //
>+  //   Extracts the key,value pair of the element at the indicated position and
>+  //   returns a node handle owning that extracted data.
>+  //
>+  // node_type extract(const key_type& x):
>+  //
>+  //   Extracts the key,value pair of the element with a key matching the passed
>+  //   key value and returns a node handle owning that extracted data. If the
>+  //   `node_hash_map` does not contain an element with a matching key, this
>+  //   function returns an empty node handle.
>+  using Base::extract;
>+
>+  // node_hash_map::merge()
>+  //
>+  // Extracts elements from a given `source` node hash map into this
>+  // `node_hash_map`. If the destination `node_hash_map` already contains an
>+  // element with an equivalent key, that element is not extracted.
>+  using Base::merge;
>+
>+  // node_hash_map::swap(node_hash_map& other)
>+  //
>+  // Exchanges the contents of this `node_hash_map` with those of the `other`
>+  // node hash map, avoiding invocation of any move, copy, or swap operations on
>+  // individual elements.
>+  //
>+  // All iterators and references on the `node_hash_map` remain valid, excepting
>+  // for the past-the-end iterator, which is invalidated.
>+  //
>+  // `swap()` requires that the node hash map's hashing and key equivalence
>+  // functions be Swappable, and are exchaged using unqualified calls to
>+  // non-member `swap()`. If the map's allocator has
>+  // `std::allocator_traits<allocator_type>::propagate_on_container_swap::value`
>+  // set to `true`, the allocators are also exchanged using an unqualified call
>+  // to non-member `swap()`; otherwise, the allocators are not swapped.
>+  using Base::swap;
>+
>+  // node_hash_map::rehash(count)
>+  //
>+  // Rehashes the `node_hash_map`, setting the number of slots to be at least
>+  // the passed value. If the new number of slots increases the load factor more
>+  // than the current maximum load factor
>+  // (`count` < `size()` / `max_load_factor()`), then the new number of slots
>+  // will be at least `size()` / `max_load_factor()`.
>+  //
>+  // To force a rehash, pass rehash(0).
>+  using Base::rehash;
>+
>+  // node_hash_map::reserve(count)
>+  //
>+  // Sets the number of slots in the `node_hash_map` to the number needed to
>+  // accommodate at least `count` total elements without exceeding the current
>+  // maximum load factor, and may rehash the container if needed.
>+  using Base::reserve;
>+
>+  // node_hash_map::at()
>+  //
>+  // Returns a reference to the mapped value of the element with key equivalent
>+  // to the passed key.
>+  using Base::at;
>+
>+  // node_hash_map::contains()
>+  //
>+  // Determines whether an element with a key comparing equal to the given `key`
>+  // exists within the `node_hash_map`, returning `true` if so or `false`
>+  // otherwise.
>+  using Base::contains;
>+
>+  // node_hash_map::count(const Key& key) const
>+  //
>+  // Returns the number of elements with a key comparing equal to the given
>+  // `key` within the `node_hash_map`. note that this function will return
>+  // either `1` or `0` since duplicate keys are not allowed within a
>+  // `node_hash_map`.
>+  using Base::count;
>+
>+  // node_hash_map::equal_range()
>+  //
>+  // Returns a closed range [first, last], defined by a `std::pair` of two
>+  // iterators, containing all elements with the passed key in the
>+  // `node_hash_map`.
>+  using Base::equal_range;
>+
>+  // node_hash_map::find()
>+  //
>+  // Finds an element with the passed `key` within the `node_hash_map`.
>+  using Base::find;
>+
>+  // node_hash_map::operator[]()
>+  //
>+  // Returns a reference to the value mapped to the passed key within the
>+  // `node_hash_map`, performing an `insert()` if the key does not already
>+  // exist. If an insertion occurs and results in a rehashing of the container,
>+  // all iterators are invalidated. Otherwise iterators are not affected and
>+  // references are not invalidated. Overloads are listed below.
>+  //
>+  // T& operator[](const Key& key):
>+  //
>+  //   Inserts an init_type object constructed in-place if the element with the
>+  //   given key does not exist.
>+  //
>+  // T& operator[](Key&& key):
>+  //
>+  //   Inserts an init_type object constructed in-place provided that an element
>+  //   with the given key does not exist.
>+  using Base::operator[];
>+
>+  // node_hash_map::bucket_count()
>+  //
>+  // Returns the number of "buckets" within the `node_hash_map`.
>+  using Base::bucket_count;
>+
>+  // node_hash_map::load_factor()
>+  //
>+  // Returns the current load factor of the `node_hash_map` (the average number
>+  // of slots occupied with a value within the hash map).
>+  using Base::load_factor;
>+
>+  // node_hash_map::max_load_factor()
>+  //
>+  // Manages the maximum load factor of the `node_hash_map`. Overloads are
>+  // listed below.
>+  //
>+  // float node_hash_map::max_load_factor()
>+  //
>+  //   Returns the current maximum load factor of the `node_hash_map`.
>+  //
>+  // void node_hash_map::max_load_factor(float ml)
>+  //
>+  //   Sets the maximum load factor of the `node_hash_map` to the passed value.
>+  //
>+  //   NOTE: This overload is provided only for API compatibility with the STL;
>+  //   `node_hash_map` will ignore any set load factor and manage its rehashing
>+  //   internally as an implementation detail.
>+  using Base::max_load_factor;
>+
>+  // node_hash_map::get_allocator()
>+  //
>+  // Returns the allocator function associated with this `node_hash_map`.
>+  using Base::get_allocator;
>+
>+  // node_hash_map::hash_function()
>+  //
>+  // Returns the hashing function used to hash the keys within this
>+  // `node_hash_map`.
>+  using Base::hash_function;
>+
>+  // node_hash_map::key_eq()
>+  //
>+  // Returns the function used for comparing keys equality.
>+  using Base::key_eq;
>+
>+  ABSL_DEPRECATED("Call `hash_function()` instead.")
>+  typename Base::hasher hash_funct() { return this->hash_function(); }
>+
>+  ABSL_DEPRECATED("Call `rehash()` instead.")
>+  void resize(typename Base::size_type hint) { this->rehash(hint); }
>+};
>+
>+namespace container_internal {
>+
>+template <class Key, class Value>
>+class NodeHashMapPolicy
>+    : public absl::container_internal::node_hash_policy<
>+          std::pair<const Key, Value>&, NodeHashMapPolicy<Key, Value>> {
>+  using value_type = std::pair<const Key, Value>;
>+
>+ public:
>+  using key_type = Key;
>+  using mapped_type = Value;
>+  using init_type = std::pair</*non const*/ key_type, mapped_type>;
>+
>+  template <class Allocator, class... Args>
>+  static value_type* new_element(Allocator* alloc, Args&&... args) {
>+    using PairAlloc = typename absl::allocator_traits<
>+        Allocator>::template rebind_alloc<value_type>;
>+    PairAlloc pair_alloc(*alloc);
>+    value_type* res =
>+        absl::allocator_traits<PairAlloc>::allocate(pair_alloc, 1);
>+    absl::allocator_traits<PairAlloc>::construct(pair_alloc, res,
>+                                                 std::forward<Args>(args)...);
>+    return res;
>+  }
>+
>+  template <class Allocator>
>+  static void delete_element(Allocator* alloc, value_type* pair) {
>+    using PairAlloc = typename absl::allocator_traits<
>+        Allocator>::template rebind_alloc<value_type>;
>+    PairAlloc pair_alloc(*alloc);
>+    absl::allocator_traits<PairAlloc>::destroy(pair_alloc, pair);
>+    absl::allocator_traits<PairAlloc>::deallocate(pair_alloc, pair, 1);
>+  }
>+
>+  template <class F, class... Args>
>+  static decltype(absl::container_internal::DecomposePair(
>+      std::declval<F>(), std::declval<Args>()...))
>+  apply(F&& f, Args&&... args) {
>+    return absl::container_internal::DecomposePair(std::forward<F>(f),
>+                                                   std::forward<Args>(args)...);
>+  }
>+
>+  static size_t element_space_used(const value_type*) {
>+    return sizeof(value_type);
>+  }
>+
>+  static Value& value(value_type* elem) { return elem->second; }
>+  static const Value& value(const value_type* elem) { return elem->second; }
>+};
>+}  // namespace container_internal
>+}  // namespace absl
>+#endif  // ABSL_CONTAINER_NODE_HASH_MAP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_map_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_map_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..bd789644c24e8c81741493e55bf0b61e2f356660
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_map_test.cc
>@@ -0,0 +1,218 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/node_hash_map.h"
>+
>+#include "absl/container/internal/tracked.h"
>+#include "absl/container/internal/unordered_map_constructor_test.h"
>+#include "absl/container/internal/unordered_map_lookup_test.h"
>+#include "absl/container/internal/unordered_map_modifiers_test.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+using ::testing::Field;
>+using ::testing::Pair;
>+using ::testing::UnorderedElementsAre;
>+
>+using MapTypes = ::testing::Types<
>+    absl::node_hash_map<int, int, StatefulTestingHash, StatefulTestingEqual,
>+                        Alloc<std::pair<const int, int>>>,
>+    absl::node_hash_map<std::string, std::string, StatefulTestingHash,
>+                        StatefulTestingEqual,
>+                        Alloc<std::pair<const std::string, std::string>>>>;
>+
>+INSTANTIATE_TYPED_TEST_CASE_P(NodeHashMap, ConstructorTest, MapTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(NodeHashMap, LookupTest, MapTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(NodeHashMap, ModifiersTest, MapTypes);
>+
>+using M = absl::node_hash_map<std::string, Tracked<int>>;
>+
>+TEST(NodeHashMap, Emplace) {
>+  M m;
>+  Tracked<int> t(53);
>+  m.emplace("a", t);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(1, t.num_copies());
>+
>+  m.emplace(std::string("a"), t);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(1, t.num_copies());
>+
>+  std::string a("a");
>+  m.emplace(a, t);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(1, t.num_copies());
>+
>+  const std::string ca("a");
>+  m.emplace(a, t);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(1, t.num_copies());
>+
>+  m.emplace(std::make_pair("a", t));
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(2, t.num_copies());
>+
>+  m.emplace(std::make_pair(std::string("a"), t));
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(3, t.num_copies());
>+
>+  std::pair<std::string, Tracked<int>> p("a", t);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(4, t.num_copies());
>+  m.emplace(p);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(4, t.num_copies());
>+
>+  const std::pair<std::string, Tracked<int>> cp("a", t);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(5, t.num_copies());
>+  m.emplace(cp);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(5, t.num_copies());
>+
>+  std::pair<const std::string, Tracked<int>> pc("a", t);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(6, t.num_copies());
>+  m.emplace(pc);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(6, t.num_copies());
>+
>+  const std::pair<const std::string, Tracked<int>> cpc("a", t);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(7, t.num_copies());
>+  m.emplace(cpc);
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(7, t.num_copies());
>+
>+  m.emplace(std::piecewise_construct, std::forward_as_tuple("a"),
>+            std::forward_as_tuple(t));
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(7, t.num_copies());
>+
>+  m.emplace(std::piecewise_construct, std::forward_as_tuple(std::string("a")),
>+            std::forward_as_tuple(t));
>+  ASSERT_EQ(0, t.num_moves());
>+  ASSERT_EQ(7, t.num_copies());
>+}
>+
>+TEST(NodeHashMap, AssignRecursive) {
>+  struct Tree {
>+    // Verify that unordered_map<K, IncompleteType> can be instantiated.
>+    absl::node_hash_map<int, Tree> children;
>+  };
>+  Tree root;
>+  const Tree& child = root.children.emplace().first->second;
>+  // Verify that `lhs = rhs` doesn't read rhs after clearing lhs.
>+  root = child;
>+}
>+
>+TEST(FlatHashMap, MoveOnlyKey) {
>+  struct Key {
>+    Key() = default;
>+    Key(Key&&) = default;
>+    Key& operator=(Key&&) = default;
>+  };
>+  struct Eq {
>+    bool operator()(const Key&, const Key&) const { return true; }
>+  };
>+  struct Hash {
>+    size_t operator()(const Key&) const { return 0; }
>+  };
>+  absl::node_hash_map<Key, int, Hash, Eq> m;
>+  m[Key()];
>+}
>+
>+struct NonMovableKey {
>+  explicit NonMovableKey(int i) : i(i) {}
>+  NonMovableKey(NonMovableKey&&) = delete;
>+  int i;
>+};
>+struct NonMovableKeyHash {
>+  using is_transparent = void;
>+  size_t operator()(const NonMovableKey& k) const { return k.i; }
>+  size_t operator()(int k) const { return k; }
>+};
>+struct NonMovableKeyEq {
>+  using is_transparent = void;
>+  bool operator()(const NonMovableKey& a, const NonMovableKey& b) const {
>+    return a.i == b.i;
>+  }
>+  bool operator()(const NonMovableKey& a, int b) const { return a.i == b; }
>+};
>+
>+TEST(NodeHashMap, MergeExtractInsert) {
>+  absl::node_hash_map<NonMovableKey, int, NonMovableKeyHash, NonMovableKeyEq>
>+      set1, set2;
>+  set1.emplace(std::piecewise_construct, std::make_tuple(7),
>+               std::make_tuple(-7));
>+  set1.emplace(std::piecewise_construct, std::make_tuple(17),
>+               std::make_tuple(-17));
>+
>+  set2.emplace(std::piecewise_construct, std::make_tuple(7),
>+               std::make_tuple(-70));
>+  set2.emplace(std::piecewise_construct, std::make_tuple(19),
>+               std::make_tuple(-190));
>+
>+  auto Elem = [](int key, int value) {
>+    return Pair(Field(&NonMovableKey::i, key), value);
>+  };
>+
>+  EXPECT_THAT(set1, UnorderedElementsAre(Elem(7, -7), Elem(17, -17)));
>+  EXPECT_THAT(set2, UnorderedElementsAre(Elem(7, -70), Elem(19, -190)));
>+
>+  // NonMovableKey is neither copyable nor movable. We should still be able to
>+  // move nodes around.
>+  static_assert(!std::is_move_constructible<NonMovableKey>::value, "");
>+  set1.merge(set2);
>+
>+  EXPECT_THAT(set1,
>+              UnorderedElementsAre(Elem(7, -7), Elem(17, -17), Elem(19, -190)));
>+  EXPECT_THAT(set2, UnorderedElementsAre(Elem(7, -70)));
>+
>+  auto node = set1.extract(7);
>+  EXPECT_TRUE(node);
>+  EXPECT_EQ(node.key().i, 7);
>+  EXPECT_EQ(node.mapped(), -7);
>+  EXPECT_THAT(set1, UnorderedElementsAre(Elem(17, -17), Elem(19, -190)));
>+
>+  auto insert_result = set2.insert(std::move(node));
>+  EXPECT_FALSE(node);
>+  EXPECT_FALSE(insert_result.inserted);
>+  EXPECT_TRUE(insert_result.node);
>+  EXPECT_EQ(insert_result.node.key().i, 7);
>+  EXPECT_EQ(insert_result.node.mapped(), -7);
>+  EXPECT_THAT(*insert_result.position, Elem(7, -70));
>+  EXPECT_THAT(set2, UnorderedElementsAre(Elem(7, -70)));
>+
>+  node = set1.extract(17);
>+  EXPECT_TRUE(node);
>+  EXPECT_EQ(node.key().i, 17);
>+  EXPECT_EQ(node.mapped(), -17);
>+  EXPECT_THAT(set1, UnorderedElementsAre(Elem(19, -190)));
>+
>+  node.mapped() = 23;
>+
>+  insert_result = set2.insert(std::move(node));
>+  EXPECT_FALSE(node);
>+  EXPECT_TRUE(insert_result.inserted);
>+  EXPECT_FALSE(insert_result.node);
>+  EXPECT_THAT(*insert_result.position, Elem(17, 23));
>+  EXPECT_THAT(set2, UnorderedElementsAre(Elem(7, -70), Elem(17, 23)));
>+}
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_set.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_set.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..90d4ce0c1631d78cfecfcd8fc29a1c484ae9f637
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_set.h
>@@ -0,0 +1,439 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: node_hash_set.h
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::node_hash_set<T>` is an unordered associative container designed to
>+// be a more efficient replacement for `std::unordered_set`. Like
>+// `unordered_set`, search, insertion, and deletion of map elements can be done
>+// as an `O(1)` operation. However, `node_hash_set` (and other unordered
>+// associative containers known as the collection of Abseil "Swiss tables")
>+// contain other optimizations that result in both memory and computation
>+// advantages.
>+//
>+// In most cases, your default choice for a hash table should be a map of type
>+// `flat_hash_map` or a set of type `flat_hash_set`. However, if you need
>+// pointer stability, a `node_hash_set` should be your preferred choice. As
>+// well, if you are migrating your code from using `std::unordered_set`, a
>+// `node_hash_set` should be an easy migration. Consider migrating to
>+// `node_hash_set` and perhaps converting to a more efficient `flat_hash_set`
>+// upon further review.
>+
>+#ifndef ABSL_CONTAINER_NODE_HASH_SET_H_
>+#define ABSL_CONTAINER_NODE_HASH_SET_H_
>+
>+#include <type_traits>
>+
>+#include "absl/container/internal/hash_function_defaults.h"  // IWYU pragma: export
>+#include "absl/container/internal/node_hash_policy.h"
>+#include "absl/container/internal/raw_hash_set.h"  // IWYU pragma: export
>+#include "absl/memory/memory.h"
>+
>+namespace absl {
>+namespace container_internal {
>+template <typename T>
>+struct NodeHashSetPolicy;
>+}  // namespace container_internal
>+
>+// -----------------------------------------------------------------------------
>+// absl::node_hash_set
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::node_hash_set<T>` is an unordered associative container which
>+// has been optimized for both speed and memory footprint in most common use
>+// cases. Its interface is similar to that of `std::unordered_set<T>` with the
>+// following notable differences:
>+//
>+// * Supports heterogeneous lookup, through `find()`, `operator[]()` and
>+//   `insert()`, provided that the map is provided a compatible heterogeneous
>+//   hashing function and equality operator.
>+// * Contains a `capacity()` member function indicating the number of element
>+//   slots (open, deleted, and empty) within the hash set.
>+// * Returns `void` from the `erase(iterator)` overload.
>+//
>+// By default, `node_hash_set` uses the `absl::Hash` hashing framework.
>+// All fundamental and Abseil types that support the `absl::Hash` framework have
>+// a compatible equality operator for comparing insertions into `node_hash_set`.
>+// If your type is not yet supported by the `asbl::Hash` framework, see
>+// absl/hash/hash.h for information on extending Abseil hashing to user-defined
>+// types.
>+//
>+// Example:
>+//
>+//   // Create a node hash set of three strings
>+//   absl::node_hash_map<std::string, std::string> ducks =
>+//     {"huey", "dewey"}, "louie"};
>+//
>+//  // Insert a new element into the node hash map
>+//  ducks.insert("donald"};
>+//
>+//  // Force a rehash of the node hash map
>+//  ducks.rehash(0);
>+//
>+//  // See if "dewey" is present
>+//  if (ducks.contains("dewey")) {
>+//    std::cout << "We found dewey!" << std::endl;
>+//  }
>+template <class T, class Hash = absl::container_internal::hash_default_hash<T>,
>+          class Eq = absl::container_internal::hash_default_eq<T>,
>+          class Alloc = std::allocator<T>>
>+class node_hash_set
>+    : public absl::container_internal::raw_hash_set<
>+          absl::container_internal::NodeHashSetPolicy<T>, Hash, Eq, Alloc> {
>+  using Base = typename node_hash_set::raw_hash_set;
>+
>+ public:
>+  node_hash_set() {}
>+  using Base::Base;
>+
>+  // node_hash_set::begin()
>+  //
>+  // Returns an iterator to the beginning of the `node_hash_set`.
>+  using Base::begin;
>+
>+  // node_hash_set::cbegin()
>+  //
>+  // Returns a const iterator to the beginning of the `node_hash_set`.
>+  using Base::cbegin;
>+
>+  // node_hash_set::cend()
>+  //
>+  // Returns a const iterator to the end of the `node_hash_set`.
>+  using Base::cend;
>+
>+  // node_hash_set::end()
>+  //
>+  // Returns an iterator to the end of the `node_hash_set`.
>+  using Base::end;
>+
>+  // node_hash_set::capacity()
>+  //
>+  // Returns the number of element slots (assigned, deleted, and empty)
>+  // available within the `node_hash_set`.
>+  //
>+  // NOTE: this member function is particular to `absl::node_hash_set` and is
>+  // not provided in the `std::unordered_map` API.
>+  using Base::capacity;
>+
>+  // node_hash_set::empty()
>+  //
>+  // Returns whether or not the `node_hash_set` is empty.
>+  using Base::empty;
>+
>+  // node_hash_set::max_size()
>+  //
>+  // Returns the largest theoretical possible number of elements within a
>+  // `node_hash_set` under current memory constraints. This value can be thought
>+  // of the largest value of `std::distance(begin(), end())` for a
>+  // `node_hash_set<T>`.
>+  using Base::max_size;
>+
>+  // node_hash_set::size()
>+  //
>+  // Returns the number of elements currently within the `node_hash_set`.
>+  using Base::size;
>+
>+  // node_hash_set::clear()
>+  //
>+  // Removes all elements from the `node_hash_set`. Invalidates any references,
>+  // pointers, or iterators referring to contained elements.
>+  //
>+  // NOTE: this operation may shrink the underlying buffer. To avoid shrinking
>+  // the underlying buffer call `erase(begin(), end())`.
>+  using Base::clear;
>+
>+  // node_hash_set::erase()
>+  //
>+  // Erases elements within the `node_hash_set`. Erasing does not trigger a
>+  // rehash. Overloads are listed below.
>+  //
>+  // void erase(const_iterator pos):
>+  //
>+  //   Erases the element at `position` of the `node_hash_set`, returning
>+  //   `void`.
>+  //
>+  //   NOTE: this return behavior is different than that of STL containers in
>+  //   general and `std::unordered_map` in particular.
>+  //
>+  // iterator erase(const_iterator first, const_iterator last):
>+  //
>+  //   Erases the elements in the open interval [`first`, `last`), returning an
>+  //   iterator pointing to `last`.
>+  //
>+  // size_type erase(const key_type& key):
>+  //
>+  //   Erases the element with the matching key, if it exists.
>+  using Base::erase;
>+
>+  // node_hash_set::insert()
>+  //
>+  // Inserts an element of the specified value into the `node_hash_set`,
>+  // returning an iterator pointing to the newly inserted element, provided that
>+  // an element with the given key does not already exist. If rehashing occurs
>+  // due to the insertion, all iterators are invalidated. Overloads are listed
>+  // below.
>+  //
>+  // std::pair<iterator,bool> insert(const T& value):
>+  //
>+  //   Inserts a value into the `node_hash_set`. Returns a pair consisting of an
>+  //   iterator to the inserted element (or to the element that prevented the
>+  //   insertion) and a bool denoting whether the insertion took place.
>+  //
>+  // std::pair<iterator,bool> insert(T&& value):
>+  //
>+  //   Inserts a moveable value into the `node_hash_set`. Returns a pair
>+  //   consisting of an iterator to the inserted element (or to the element that
>+  //   prevented the insertion) and a bool denoting whether the insertion took
>+  //   place.
>+  //
>+  // iterator insert(const_iterator hint, const T& value):
>+  // iterator insert(const_iterator hint, T&& value):
>+  //
>+  //   Inserts a value, using the position of `hint` as a non-binding suggestion
>+  //   for where to begin the insertion search. Returns an iterator to the
>+  //   inserted element, or to the existing element that prevented the
>+  //   insertion.
>+  //
>+  // void insert(InputIterator first, InputIterator last):
>+  //
>+  //   Inserts a range of values [`first`, `last`).
>+  //
>+  //   NOTE: Although the STL does not specify which element may be inserted if
>+  //   multiple keys compare equivalently, for `node_hash_set` we guarantee the
>+  //   first match is inserted.
>+  //
>+  // void insert(std::initializer_list<T> ilist):
>+  //
>+  //   Inserts the elements within the initializer list `ilist`.
>+  //
>+  //   NOTE: Although the STL does not specify which element may be inserted if
>+  //   multiple keys compare equivalently within the initializer list, for
>+  //   `node_hash_set` we guarantee the first match is inserted.
>+  using Base::insert;
>+
>+  // node_hash_set::emplace()
>+  //
>+  // Inserts an element of the specified value by constructing it in-place
>+  // within the `node_hash_set`, provided that no element with the given key
>+  // already exists.
>+  //
>+  // The element may be constructed even if there already is an element with the
>+  // key in the container, in which case the newly constructed element will be
>+  // destroyed immediately. Prefer `try_emplace()` unless your key is not
>+  // copyable or moveable.
>+  //
>+  // If rehashing occurs due to the insertion, all iterators are invalidated.
>+  using Base::emplace;
>+
>+  // node_hash_set::emplace_hint()
>+  //
>+  // Inserts an element of the specified value by constructing it in-place
>+  // within the `node_hash_set`, using the position of `hint` as a non-binding
>+  // suggestion for where to begin the insertion search, and only inserts
>+  // provided that no element with the given key already exists.
>+  //
>+  // The element may be constructed even if there already is an element with the
>+  // key in the container, in which case the newly constructed element will be
>+  // destroyed immediately. Prefer `try_emplace()` unless your key is not
>+  // copyable or moveable.
>+  //
>+  // If rehashing occurs due to the insertion, all iterators are invalidated.
>+  using Base::emplace_hint;
>+
>+  // node_hash_set::extract()
>+  //
>+  // Extracts the indicated element, erasing it in the process, and returns it
>+  // as a C++17-compatible node handle. Overloads are listed below.
>+  //
>+  // node_type extract(const_iterator position):
>+  //
>+  //   Extracts the element at the indicated position and returns a node handle
>+  //   owning that extracted data.
>+  //
>+  // node_type extract(const key_type& x):
>+  //
>+  //   Extracts the element with the key matching the passed key value and
>+  //   returns a node handle owning that extracted data. If the `node_hash_set`
>+  //   does not contain an element with a matching key, this function returns an
>+  // empty node handle.
>+  using Base::extract;
>+
>+  // node_hash_set::merge()
>+  //
>+  // Extracts elements from a given `source` flat hash map into this
>+  // `node_hash_set`. If the destination `node_hash_set` already contains an
>+  // element with an equivalent key, that element is not extracted.
>+  using Base::merge;
>+
>+  // node_hash_set::swap(node_hash_set& other)
>+  //
>+  // Exchanges the contents of this `node_hash_set` with those of the `other`
>+  // flat hash map, avoiding invocation of any move, copy, or swap operations on
>+  // individual elements.
>+  //
>+  // All iterators and references on the `node_hash_set` remain valid, excepting
>+  // for the past-the-end iterator, which is invalidated.
>+  //
>+  // `swap()` requires that the flat hash set's hashing and key equivalence
>+  // functions be Swappable, and are exchaged using unqualified calls to
>+  // non-member `swap()`. If the map's allocator has
>+  // `std::allocator_traits<allocator_type>::propagate_on_container_swap::value`
>+  // set to `true`, the allocators are also exchanged using an unqualified call
>+  // to non-member `swap()`; otherwise, the allocators are not swapped.
>+  using Base::swap;
>+
>+  // node_hash_set::rehash(count)
>+  //
>+  // Rehashes the `node_hash_set`, setting the number of slots to be at least
>+  // the passed value. If the new number of slots increases the load factor more
>+  // than the current maximum load factor
>+  // (`count` < `size()` / `max_load_factor()`), then the new number of slots
>+  // will be at least `size()` / `max_load_factor()`.
>+  //
>+  // To force a rehash, pass rehash(0).
>+  //
>+  // NOTE: unlike behavior in `std::unordered_set`, references are also
>+  // invalidated upon a `rehash()`.
>+  using Base::rehash;
>+
>+  // node_hash_set::reserve(count)
>+  //
>+  // Sets the number of slots in the `node_hash_set` to the number needed to
>+  // accommodate at least `count` total elements without exceeding the current
>+  // maximum load factor, and may rehash the container if needed.
>+  using Base::reserve;
>+
>+  // node_hash_set::contains()
>+  //
>+  // Determines whether an element comparing equal to the given `key` exists
>+  // within the `node_hash_set`, returning `true` if so or `false` otherwise.
>+  using Base::contains;
>+
>+  // node_hash_set::count(const Key& key) const
>+  //
>+  // Returns the number of elements comparing equal to the given `key` within
>+  // the `node_hash_set`. note that this function will return either `1` or `0`
>+  // since duplicate elements are not allowed within a `node_hash_set`.
>+  using Base::count;
>+
>+  // node_hash_set::equal_range()
>+  //
>+  // Returns a closed range [first, last], defined by a `std::pair` of two
>+  // iterators, containing all elements with the passed key in the
>+  // `node_hash_set`.
>+  using Base::equal_range;
>+
>+  // node_hash_set::find()
>+  //
>+  // Finds an element with the passed `key` within the `node_hash_set`.
>+  using Base::find;
>+
>+  // node_hash_set::bucket_count()
>+  //
>+  // Returns the number of "buckets" within the `node_hash_set`. Note that
>+  // because a flat hash map contains all elements within its internal storage,
>+  // this value simply equals the current capacity of the `node_hash_set`.
>+  using Base::bucket_count;
>+
>+  // node_hash_set::load_factor()
>+  //
>+  // Returns the current load factor of the `node_hash_set` (the average number
>+  // of slots occupied with a value within the hash map).
>+  using Base::load_factor;
>+
>+  // node_hash_set::max_load_factor()
>+  //
>+  // Manages the maximum load factor of the `node_hash_set`. Overloads are
>+  // listed below.
>+  //
>+  // float node_hash_set::max_load_factor()
>+  //
>+  //   Returns the current maximum load factor of the `node_hash_set`.
>+  //
>+  // void node_hash_set::max_load_factor(float ml)
>+  //
>+  //   Sets the maximum load factor of the `node_hash_set` to the passed value.
>+  //
>+  //   NOTE: This overload is provided only for API compatibility with the STL;
>+  //   `node_hash_set` will ignore any set load factor and manage its rehashing
>+  //   internally as an implementation detail.
>+  using Base::max_load_factor;
>+
>+  // node_hash_set::get_allocator()
>+  //
>+  // Returns the allocator function associated with this `node_hash_set`.
>+  using Base::get_allocator;
>+
>+  // node_hash_set::hash_function()
>+  //
>+  // Returns the hashing function used to hash the keys within this
>+  // `node_hash_set`.
>+  using Base::hash_function;
>+
>+  // node_hash_set::key_eq()
>+  //
>+  // Returns the function used for comparing keys equality.
>+  using Base::key_eq;
>+
>+  ABSL_DEPRECATED("Call `hash_function()` instead.")
>+  typename Base::hasher hash_funct() { return this->hash_function(); }
>+
>+  ABSL_DEPRECATED("Call `rehash()` instead.")
>+  void resize(typename Base::size_type hint) { this->rehash(hint); }
>+};
>+
>+namespace container_internal {
>+
>+template <class T>
>+struct NodeHashSetPolicy
>+    : absl::container_internal::node_hash_policy<T&, NodeHashSetPolicy<T>> {
>+  using key_type = T;
>+  using init_type = T;
>+  using constant_iterators = std::true_type;
>+
>+  template <class Allocator, class... Args>
>+  static T* new_element(Allocator* alloc, Args&&... args) {
>+    using ValueAlloc =
>+        typename absl::allocator_traits<Allocator>::template rebind_alloc<T>;
>+    ValueAlloc value_alloc(*alloc);
>+    T* res = absl::allocator_traits<ValueAlloc>::allocate(value_alloc, 1);
>+    absl::allocator_traits<ValueAlloc>::construct(value_alloc, res,
>+                                                  std::forward<Args>(args)...);
>+    return res;
>+  }
>+
>+  template <class Allocator>
>+  static void delete_element(Allocator* alloc, T* elem) {
>+    using ValueAlloc =
>+        typename absl::allocator_traits<Allocator>::template rebind_alloc<T>;
>+    ValueAlloc value_alloc(*alloc);
>+    absl::allocator_traits<ValueAlloc>::destroy(value_alloc, elem);
>+    absl::allocator_traits<ValueAlloc>::deallocate(value_alloc, elem, 1);
>+  }
>+
>+  template <class F, class... Args>
>+  static decltype(absl::container_internal::DecomposeValue(
>+      std::declval<F>(), std::declval<Args>()...))
>+  apply(F&& f, Args&&... args) {
>+    return absl::container_internal::DecomposeValue(
>+        std::forward<F>(f), std::forward<Args>(args)...);
>+  }
>+
>+  static size_t element_space_used(const T*) { return sizeof(T); }
>+};
>+}  // namespace container_internal
>+}  // namespace absl
>+#endif  // ABSL_CONTAINER_NODE_HASH_SET_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_set_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_set_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..7e498f0f779b4d6e764ecafca62b1cb08ccfd5ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/node_hash_set_test.cc
>@@ -0,0 +1,103 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/node_hash_set.h"
>+
>+#include "absl/container/internal/unordered_set_constructor_test.h"
>+#include "absl/container/internal/unordered_set_lookup_test.h"
>+#include "absl/container/internal/unordered_set_modifiers_test.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+using ::absl::container_internal::hash_internal::Enum;
>+using ::absl::container_internal::hash_internal::EnumClass;
>+using ::testing::Pointee;
>+using ::testing::UnorderedElementsAre;
>+
>+using SetTypes = ::testing::Types<
>+    node_hash_set<int, StatefulTestingHash, StatefulTestingEqual, Alloc<int>>,
>+    node_hash_set<std::string, StatefulTestingHash, StatefulTestingEqual,
>+                  Alloc<int>>,
>+    node_hash_set<Enum, StatefulTestingHash, StatefulTestingEqual, Alloc<Enum>>,
>+    node_hash_set<EnumClass, StatefulTestingHash, StatefulTestingEqual,
>+                  Alloc<EnumClass>>>;
>+
>+INSTANTIATE_TYPED_TEST_CASE_P(NodeHashSet, ConstructorTest, SetTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(NodeHashSet, LookupTest, SetTypes);
>+INSTANTIATE_TYPED_TEST_CASE_P(NodeHashSet, ModifiersTest, SetTypes);
>+
>+TEST(NodeHashSet, MoveableNotCopyableCompiles) {
>+  node_hash_set<std::unique_ptr<void*>> t;
>+  node_hash_set<std::unique_ptr<void*>> u;
>+  u = std::move(t);
>+}
>+
>+TEST(NodeHashSet, MergeExtractInsert) {
>+  struct Hash {
>+    size_t operator()(const std::unique_ptr<int>& p) const { return *p; }
>+  };
>+  struct Eq {
>+    bool operator()(const std::unique_ptr<int>& a,
>+                    const std::unique_ptr<int>& b) const {
>+      return *a == *b;
>+    }
>+  };
>+  absl::node_hash_set<std::unique_ptr<int>, Hash, Eq> set1, set2;
>+  set1.insert(absl::make_unique<int>(7));
>+  set1.insert(absl::make_unique<int>(17));
>+
>+  set2.insert(absl::make_unique<int>(7));
>+  set2.insert(absl::make_unique<int>(19));
>+
>+  EXPECT_THAT(set1, UnorderedElementsAre(Pointee(7), Pointee(17)));
>+  EXPECT_THAT(set2, UnorderedElementsAre(Pointee(7), Pointee(19)));
>+
>+  set1.merge(set2);
>+
>+  EXPECT_THAT(set1, UnorderedElementsAre(Pointee(7), Pointee(17), Pointee(19)));
>+  EXPECT_THAT(set2, UnorderedElementsAre(Pointee(7)));
>+
>+  auto node = set1.extract(absl::make_unique<int>(7));
>+  EXPECT_TRUE(node);
>+  EXPECT_THAT(node.value(), Pointee(7));
>+  EXPECT_THAT(set1, UnorderedElementsAre(Pointee(17), Pointee(19)));
>+
>+  auto insert_result = set2.insert(std::move(node));
>+  EXPECT_FALSE(node);
>+  EXPECT_FALSE(insert_result.inserted);
>+  EXPECT_TRUE(insert_result.node);
>+  EXPECT_THAT(insert_result.node.value(), Pointee(7));
>+  EXPECT_EQ(**insert_result.position, 7);
>+  EXPECT_NE(insert_result.position->get(), insert_result.node.value().get());
>+  EXPECT_THAT(set2, UnorderedElementsAre(Pointee(7)));
>+
>+  node = set1.extract(absl::make_unique<int>(17));
>+  EXPECT_TRUE(node);
>+  EXPECT_THAT(node.value(), Pointee(17));
>+  EXPECT_THAT(set1, UnorderedElementsAre(Pointee(19)));
>+
>+  node.value() = absl::make_unique<int>(23);
>+
>+  insert_result = set2.insert(std::move(node));
>+  EXPECT_FALSE(node);
>+  EXPECT_TRUE(insert_result.inserted);
>+  EXPECT_FALSE(insert_result.node);
>+  EXPECT_EQ(**insert_result.position, 23);
>+  EXPECT_THAT(set2, UnorderedElementsAre(Pointee(7), Pointee(23)));
>+}
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/copts.bzl b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/copts.bzl
>index 0168ac5abddc39ff7c4781ff5e3e2e9be0a2057e..5c508f17df9747618ca52c5e4b933f818ee9b9c8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/copts.bzl
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/copts.bzl
>@@ -35,7 +35,6 @@ GCC_TEST_FLAGS = [
> # Docs on groups of flags is preceded by ###.
> 
> LLVM_FLAGS = [
>-    # All warnings are treated as errors by implicit -Werror flag
>     "-Wall",
>     "-Wextra",
>     "-Weverything",
>@@ -54,6 +53,10 @@ LLVM_FLAGS = [
>     "-Wno-packed",
>     "-Wno-padded",
>     ###
>+    # Google style does not use unsigned integers, though STL containers
>+    # have unsigned types.
>+    "-Wno-sign-compare",
>+    ###
>     "-Wno-float-conversion",
>     "-Wno-float-equal",
>     "-Wno-format-nonliteral",
>@@ -101,6 +104,7 @@ LLVM_TEST_FLAGS = [
>     "-Wno-c99-extensions",
>     "-Wno-missing-noreturn",
>     "-Wno-missing-prototypes",
>+    "-Wno-missing-variable-declarations",
>     "-Wno-null-conversion",
>     "-Wno-shadow",
>     "-Wno-shift-sign-overflow",
>@@ -112,13 +116,15 @@ LLVM_TEST_FLAGS = [
>     "-Wno-unused-template",
>     "-Wno-used-but-marked-unused",
>     "-Wno-zero-as-null-pointer-constant",
>+    # gtest depends on this GNU extension being offered.
>+    "-Wno-gnu-zero-variadic-macro-arguments",
> ]
> 
> MSVC_FLAGS = [
>     "/W3",
>-    "/WX",
>     "/wd4005",  # macro-redefinition
>     "/wd4068",  # unknown pragma
>+    "/wd4180",  # qualifier applied to function type has no meaning; ignored
>     "/wd4244",  # conversion from 'type1' to 'type2', possible loss of data
>     "/wd4267",  # conversion from 'size_t' to 'type', possible loss of data
>     "/wd4800",  # forcing value to bool 'true' or 'false' (performance warning)
>@@ -152,3 +158,7 @@ ABSL_EXCEPTIONS_FLAG = select({
>     "//absl:windows": ["/U_HAS_EXCEPTIONS", "/D_HAS_EXCEPTIONS=1", "/EHsc"],
>     "//conditions:default": ["-fexceptions"],
> })
>+
>+ABSL_EXCEPTIONS_FLAG_LINKOPTS = select({
>+    "//conditions:default": [],
>+})
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/BUILD.gn
>index b50b0a6ddc8321968eb1fa176321bc8e1f974254..d3582d6a426f775716eb11762d57a1991b04cb32 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/BUILD.gn
>@@ -92,10 +92,10 @@ source_set("failure_signal_handler") {
>   ]
>   public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>   sources = [
>-    "failure_signal_handler.cc"
>+    "failure_signal_handler.cc",
>   ]
>   public = [
>-    "failure_signal_handler.h"
>+    "failure_signal_handler.h",
>   ]
>   deps = [
>     ":examine_stack",
>@@ -126,6 +126,7 @@ source_set("debugging_internal") {
>   ]
>   deps = [
>     "../base",
>+    "../base:core_headers",
>     "../base:dynamic_annotations",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle.cc
>index c9ca2f3bdb170e77d28dec15795ba58829fb868a..48354459bc85f86dfa0d319e9741188b3e9d6746 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle.cc
>@@ -340,7 +340,7 @@ static bool ZeroOrMore(ParseFunc parse_func, State *state) {
> }
> 
> // Append "str" at "out_cur_idx".  If there is an overflow, out_cur_idx is
>-// set to out_end_idx+1.  The output std::string is ensured to
>+// set to out_end_idx+1.  The output string is ensured to
> // always terminate with '\0' as long as there is no overflow.
> static void Append(State *state, const char *const str, const int length) {
>   for (int i = 0; i < length; ++i) {
>@@ -840,7 +840,7 @@ static bool ParseNumber(State *state, int *number_out) {
> }
> 
> // Floating-point literals are encoded using a fixed-length lowercase
>-// hexadecimal std::string.
>+// hexadecimal string.
> static bool ParseFloatNumber(State *state) {
>   ComplexityGuard guard(state);
>   if (guard.IsTooComplex()) return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/stacktrace.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/stacktrace.cc
>index 61fee6190f545842adc362c8fcfcec5c0e75658a..463fad269c16afba3e383eed12906c548849f38f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/stacktrace.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/stacktrace.cc
>@@ -80,11 +80,13 @@ ABSL_ATTRIBUTE_ALWAYS_INLINE inline int Unwind(void** result, int* sizes,
> 
> }  // anonymous namespace
> 
>+ABSL_ATTRIBUTE_NOINLINE
> int GetStackFrames(void** result, int* sizes, int max_depth, int skip_count) {
>   return Unwind<true, false>(result, sizes, max_depth, skip_count, nullptr,
>                              nullptr);
> }
> 
>+ABSL_ATTRIBUTE_NOINLINE
> int GetStackFramesWithContext(void** result, int* sizes, int max_depth,
>                               int skip_count, const void* uc,
>                               int* min_dropped_frames) {
>@@ -92,11 +94,13 @@ int GetStackFramesWithContext(void** result, int* sizes, int max_depth,
>                             min_dropped_frames);
> }
> 
>+ABSL_ATTRIBUTE_NOINLINE
> int GetStackTrace(void** result, int max_depth, int skip_count) {
>   return Unwind<false, false>(result, nullptr, max_depth, skip_count, nullptr,
>                               nullptr);
> }
> 
>+ABSL_ATTRIBUTE_NOINLINE
> int GetStackTraceWithContext(void** result, int max_depth, int skip_count,
>                              const void* uc, int* min_dropped_frames) {
>   return Unwind<false, true>(result, nullptr, max_depth, skip_count, uc,
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_test.cc
>index 5f2af47ee45d654b422d0e864fbf060cb5c51075..8029fbe9d2e6fb720816ef0435e07099cc387292 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_test.cc
>@@ -321,7 +321,7 @@ TEST(Symbolize, SymbolizeWithMultipleMaps) {
>   }
> }
> 
>-// Appends std::string(*args->arg) to args->symbol_buf.
>+// Appends string(*args->arg) to args->symbol_buf.
> static void DummySymbolDecorator(
>     const absl::debugging_internal::SymbolDecoratorArgs *args) {
>   std::string *message = static_cast<std::string *>(args->arg);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/BUILD.bazel
>new file mode 100644
>index 0000000000000000000000000000000000000000..50aa5506c11c87ed0b4a0642b950ba160f8a2f79
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/BUILD.bazel
>@@ -0,0 +1,114 @@
>+#
>+# Copyright 2018 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+)
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "hash",
>+    srcs = [
>+        "internal/hash.cc",
>+        "internal/hash.h",
>+    ],
>+    hdrs = ["hash.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":city",
>+        "//absl/base:core_headers",
>+        "//absl/base:endian",
>+        "//absl/container:fixed_array",
>+        "//absl/meta:type_traits",
>+        "//absl/numeric:int128",
>+        "//absl/strings",
>+        "//absl/types:optional",
>+        "//absl/types:variant",
>+        "//absl/utility",
>+    ],
>+)
>+
>+cc_library(
>+    name = "hash_testing",
>+    testonly = 1,
>+    hdrs = ["hash_testing.h"],
>+    deps = [
>+        ":spy_hash_state",
>+        "//absl/meta:type_traits",
>+        "//absl/strings",
>+        "//absl/types:variant",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_test(
>+    name = "hash_test",
>+    srcs = ["hash_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":hash",
>+        ":hash_testing",
>+        "//absl/base:core_headers",
>+        "//absl/container:flat_hash_set",
>+        "//absl/hash:spy_hash_state",
>+        "//absl/meta:type_traits",
>+        "//absl/numeric:int128",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "spy_hash_state",
>+    testonly = 1,
>+    hdrs = ["internal/spy_hash_state.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":hash",
>+        "//absl/strings",
>+        "//absl/strings:str_format",
>+    ],
>+)
>+
>+cc_library(
>+    name = "city",
>+    srcs = ["internal/city.cc"],
>+    hdrs = [
>+        "internal/city.h",
>+        "internal/city_crc.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+        "//absl/base:endian",
>+    ],
>+)
>+
>+cc_test(
>+    name = "city_test",
>+    srcs = ["internal/city_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":city",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/BUILD.gn
>new file mode 100644
>index 0000000000000000000000000000000000000000..40b6b3a22b37132bb6ef92abccfba1f669cfe47f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/BUILD.gn
>@@ -0,0 +1,104 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("hash") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/hash.cc",
>+    "internal/hash.h",
>+  ]
>+  public = [
>+    "hash.h",
>+  ]
>+  deps = [
>+    ":city",
>+    "../base:core_headers",
>+    "../base:endian",
>+    "../container:fixed_array",
>+    "../meta:type_traits",
>+    "../numeric:int128",
>+    "../strings",
>+    "../types:optional",
>+    "../types:variant",
>+    "../utility",
>+  ]
>+}
>+
>+source_set("hash_testing") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "hash_testing.h",
>+  ]
>+  deps = [
>+    ":spy_hash_state",
>+    "../meta:type_traits",
>+    "../strings",
>+    "../types:variant",
>+    "//testing/gtest",
>+  ]
>+}
>+
>+source_set("spy_hash_state") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/spy_hash_state.h",
>+  ]
>+  deps = [
>+    ":hash",
>+    "../strings",
>+    "../strings:str_format",
>+  ]
>+  visibility = []
>+  visibility += [ "../*" ]
>+}
>+
>+source_set("city") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/city.h",
>+    "internal/city_crc.h",
>+  ]
>+  sources = [
>+    "internal/city.cc",
>+  ]
>+  deps = [
>+    "../base:config",
>+    "../base:core_headers",
>+    "../base:endian",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/CMakeLists.txt
>new file mode 100644
>index 0000000000000000000000000000000000000000..35081e3774f8f322a9d75da948ef27a2213d227e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/CMakeLists.txt
>@@ -0,0 +1,80 @@
>+#
>+# Copyright 2018 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+list(APPEND HASH_PUBLIC_HEADERS
>+  "hash.h"
>+)
>+
>+list(APPEND HASH_INTERNAL_HEADERS
>+  "internal/city.h"
>+  "internal/city_crc.h"
>+  "internal/hash.h"
>+)
>+
>+# absl_hash library
>+list(APPEND HASH_SRC
>+  "internal/city.cc"
>+  "internal/hash.cc"
>+  ${HASH_PUBLIC_HEADERS}
>+  ${HASH_INTERNAL_HEADERS}
>+)
>+
>+set(HASH_PUBLIC_LIBRARIES absl::hash absl::container absl::strings absl::str_format absl::utility)
>+
>+absl_library(
>+  TARGET
>+    absl_hash
>+  SOURCES
>+    ${HASH_SRC}
>+  PUBLIC_LIBRARIES
>+    ${HASH_PUBLIC_LIBRARIES}
>+  EXPORT_NAME
>+    hash
>+)
>+
>+#
>+## TESTS
>+#
>+
>+# testing support
>+set(HASH_TEST_HEADERS hash_testing.h internal/spy_hash_state.h)
>+set(HASH_TEST_PUBLIC_LIBRARIES absl::hash absl::container absl::numeric absl::strings absl::str_format)
>+
>+# hash_test
>+set(HASH_TEST_SRC "hash_test.cc" ${HASH_TEST_HEADERS})
>+
>+absl_test(
>+  TARGET
>+    hash_test
>+  SOURCES
>+    ${HASH_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${HASH_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+# hash_test
>+set(CITY_TEST_SRC "internal/city_test.cc")
>+
>+absl_test(
>+  TARGET
>+    city_test
>+  SOURCES
>+    ${CITY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${HASH_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/hash.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/hash.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..c7ba4c2b7a5c198f74ce273910bac881c4559f89
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/hash.h
>@@ -0,0 +1,312 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: hash.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines the Abseil `hash` library and the Abseil hashing
>+// framework. This framework consists of the following:
>+//
>+//   * The `absl::Hash` functor, which is used to invoke the hasher within the
>+//     Abseil hashing framework. `absl::Hash<T>` supports most basic types and
>+//     a number of Abseil types out of the box.
>+//   * `AbslHashValue`, an extension point that allows you to extend types to
>+//     support Abseil hashing without requiring you to define a hashing
>+//     algorithm.
>+//   * `HashState`, a type-erased class which implement the manipulation of the
>+//     hash state (H) itself. containing member functions `combine()` and
>+//     `combine_contiguous()`, which you can use to contribute to an existing
>+//     hash state when hashing your types.
>+//
>+// Unlike `std::hash` or other hashing frameworks, the Abseil hashing framework
>+// provides most of its utility by abstracting away the hash algorithm (and its
>+// implementation) entirely. Instead, a type invokes the Abseil hashing
>+// framework by simply combining its state with the state of known, hashable
>+// types. Hashing of that combined state is separately done by `absl::Hash`.
>+//
>+// Example:
>+//
>+//   // Suppose we have a class `Circle` for which we want to add hashing
>+//   class Circle {
>+//     public:
>+//       ...
>+//     private:
>+//       std::pair<int, int> center_;
>+//       int radius_;
>+//     };
>+//
>+//   // To add hashing support to `Circle`, we simply need to add an ordinary
>+//   // function `AbslHashValue()`, and return the combined hash state of the
>+//   // existing hash state and the class state:
>+//
>+//     template <typename H>
>+//     friend H AbslHashValue(H h, const Circle& c) {
>+//       return H::combine(std::move(h), c.center_, c.radius_);
>+//     }
>+//
>+// For more information, see Adding Type Support to `absl::Hash` below.
>+//
>+#ifndef ABSL_HASH_HASH_H_
>+#define ABSL_HASH_HASH_H_
>+
>+#include "absl/hash/internal/hash.h"
>+
>+namespace absl {
>+
>+// -----------------------------------------------------------------------------
>+// `absl::Hash`
>+// -----------------------------------------------------------------------------
>+//
>+// `absl::Hash<T>` is a convenient general-purpose hash functor for a type `T`
>+// satisfying any of the following conditions (in order):
>+//
>+//  * T is an arithmetic or pointer type
>+//  * T defines an overload for `AbslHashValue(H, const T&)` for an arbitrary
>+//    hash state `H`.
>+//  - T defines a specialization of `HASH_NAMESPACE::hash<T>`
>+//  - T defines a specialization of `std::hash<T>`
>+//
>+// `absl::Hash` intrinsically supports the following types:
>+//
>+//   * All integral types (including bool)
>+//   * All enum types
>+//   * All floating-point types (although hashing them is discouraged)
>+//   * All pointer types, including nullptr_t
>+//   * std::pair<T1, T2>, if T1 and T2 are hashable
>+//   * std::tuple<Ts...>, if all the Ts... are hashable
>+//   * std::unique_ptr and std::shared_ptr
>+//   * All string-like types including:
>+//     * std::string
>+//     * std::string_view (as well as any instance of std::basic_string that
>+//       uses char and std::char_traits)
>+//  * All the standard sequence containers (provided the elements are hashable)
>+//  * All the standard ordered associative containers (provided the elements are
>+//    hashable)
>+//  * absl types such as the following:
>+//    * absl::string_view
>+//    * absl::InlinedVector
>+//    * absl::FixedArray
>+//    * absl::unit128
>+//    * absl::Time, absl::Duration, and absl::TimeZone
>+//
>+// Note: the list above is not meant to be exhaustive. Additional type support
>+// may be added, in which case the above list will be updated.
>+//
>+// -----------------------------------------------------------------------------
>+// absl::Hash Invocation Evaluation
>+// -----------------------------------------------------------------------------
>+//
>+// When invoked, `absl::Hash<T>` searches for supplied hash functions in the
>+// following order:
>+//
>+//   * Natively supported types out of the box (see above)
>+//   * Types for which an `AbslHashValue()` overload is provided (such as
>+//     user-defined types). See "Adding Type Support to `absl::Hash`" below.
>+//   * Types which define a `HASH_NAMESPACE::hash<T>` specialization (aka
>+//     `__gnu_cxx::hash<T>` for gcc/Clang or `stdext::hash<T>` for MSVC)
>+//   * Types which define a `std::hash<T>` specialization
>+//
>+// The fallback to legacy hash functions exists mainly for backwards
>+// compatibility. If you have a choice, prefer defining an `AbslHashValue`
>+// overload instead of specializing any legacy hash functors.
>+//
>+// -----------------------------------------------------------------------------
>+// The Hash State Concept, and using `HashState` for Type Erasure
>+// -----------------------------------------------------------------------------
>+//
>+// The `absl::Hash` framework relies on the Concept of a "hash state." Such a
>+// hash state is used in several places:
>+//
>+// * Within existing implementations of `absl::Hash<T>` to store the hashed
>+//   state of an object. Note that it is up to the implementation how it stores
>+//   such state. A hash table, for example, may mix the state to produce an
>+//   integer value; a testing framework may simply hold a vector of that state.
>+// * Within implementations of `AbslHashValue()` used to extend user-defined
>+//   types. (See "Adding Type Support to absl::Hash" below.)
>+// * Inside a `HashState`, providing type erasure for the concept of a hash
>+//   state, which you can use to extend the `absl::Hash` framework for types
>+//   that are otherwise difficult to extend using `AbslHashValue()`. (See the
>+//   `HashState` class below.)
>+//
>+// The "hash state" concept contains two member functions for mixing hash state:
>+//
>+// * `H::combine()`
>+//
>+//   Combines an arbitrary number of values into a hash state, returning the
>+//   updated state. Note that the existing hash state is move-only and must be
>+//   passed by value.
>+//
>+//   Each of the value types T must be hashable by H.
>+//
>+//   NOTE:
>+//
>+//     state = H::combine(std::move(state), value1, value2, value3);
>+//
>+//   must be guaranteed to produce the same hash expansion as
>+//
>+//     state = H::combine(std::move(state), value1);
>+//     state = H::combine(std::move(state), value2);
>+//     state = H::combine(std::move(state), value3);
>+//
>+// * `H::combine_contiguous()`
>+//
>+//    Combines a contiguous array of `size` elements into a hash state,
>+//    returning the updated state. Note that the existing hash state is
>+//    move-only and must be passed by value.
>+//
>+//    NOTE:
>+//
>+//      state = H::combine_contiguous(std::move(state), data, size);
>+//
>+//    need NOT be guaranteed to produce the same hash expansion as a loop
>+//    (it may perform internal optimizations). If you need this guarantee, use a
>+//    loop instead.
>+//
>+// -----------------------------------------------------------------------------
>+// Adding Type Support to `absl::Hash`
>+// -----------------------------------------------------------------------------
>+//
>+// To add support for your user-defined type, add a proper `AbslHashValue()`
>+// overload as a free (non-member) function. The overload will take an
>+// existing hash state and should combine that state with state from the type.
>+//
>+// Example:
>+//
>+//   template <typename H>
>+//   H AbslHashValue(H state, const MyType& v) {
>+//     return H::combine(std::move(state), v.field1, ..., v.fieldN);
>+//   }
>+//
>+// where `(field1, ..., fieldN)` are the members you would use on your
>+// `operator==` to define equality.
>+//
>+// Notice that `AbslHashValue` is not a class member, but an ordinary function.
>+// An `AbslHashValue` overload for a type should only be declared in the same
>+// file and namespace as said type. The proper `AbslHashValue` implementation
>+// for a given type will be discovered via ADL.
>+//
>+// Note: unlike `std::hash', `absl::Hash` should never be specialized. It must
>+// only be extended by adding `AbslHashValue()` overloads.
>+//
>+template <typename T>
>+using Hash = absl::hash_internal::Hash<T>;
>+
>+// HashState
>+//
>+// A type erased version of the hash state concept, for use in user-defined
>+// `AbslHashValue` implementations that can't use templates (such as PImpl
>+// classes, virtual functions, etc.). The type erasure adds overhead so it
>+// should be avoided unless necessary.
>+//
>+// Note: This wrapper will only erase calls to:
>+//     combine_contiguous(H, const unsigned char*, size_t)
>+//
>+// All other calls will be handled internally and will not invoke overloads
>+// provided by the wrapped class.
>+//
>+// Users of this class should still define a template `AbslHashValue` function,
>+// but can use `absl::HashState::Create(&state)` to erase the type of the hash
>+// state and dispatch to their private hashing logic.
>+//
>+// This state can be used like any other hash state. In particular, you can call
>+// `HashState::combine()` and `HashState::combine_contiguous()` on it.
>+//
>+// Example:
>+//
>+//   class Interface {
>+//    public:
>+//     template <typename H>
>+//     friend H AbslHashValue(H state, const Interface& value) {
>+//       state = H::combine(std::move(state), std::type_index(typeid(*this)));
>+//       value.HashValue(absl::HashState::Create(&state));
>+//       return state;
>+//     }
>+//    private:
>+//     virtual void HashValue(absl::HashState state) const = 0;
>+//  };
>+//
>+//   class Impl : Interface {
>+//    private:
>+//     void HashValue(absl::HashState state) const override {
>+//       absl::HashState::combine(std::move(state), v1_, v2_);
>+//     }
>+//     int v1_;
>+//     string v2_;
>+//   };
>+class HashState : public hash_internal::HashStateBase<HashState> {
>+ public:
>+  // HashState::Create()
>+  //
>+  // Create a new `HashState` instance that wraps `state`. All calls to
>+  // `combine()` and `combine_contiguous()` on the new instance will be
>+  // redirected to the original `state` object. The `state` object must outlive
>+  // the `HashState` instance.
>+  template <typename T>
>+  static HashState Create(T* state) {
>+    HashState s;
>+    s.Init(state);
>+    return s;
>+  }
>+
>+  HashState(const HashState&) = delete;
>+  HashState& operator=(const HashState&) = delete;
>+  HashState(HashState&&) = default;
>+  HashState& operator=(HashState&&) = default;
>+
>+  // HashState::combine()
>+  //
>+  // Combines an arbitrary number of values into a hash state, returning the
>+  // updated state.
>+  using HashState::HashStateBase::combine;
>+
>+  // HashState::combine_contiguous()
>+  //
>+  // Combines a contiguous array of `size` elements into a hash state, returning
>+  // the updated state.
>+  static HashState combine_contiguous(HashState hash_state,
>+                                      const unsigned char* first, size_t size) {
>+    hash_state.combine_contiguous_(hash_state.state_, first, size);
>+    return hash_state;
>+  }
>+  using HashState::HashStateBase::combine_contiguous;
>+
>+ private:
>+  HashState() = default;
>+
>+  template <typename T>
>+  static void CombineContiguousImpl(void* p, const unsigned char* first,
>+                                    size_t size) {
>+    T& state = *static_cast<T*>(p);
>+    state = T::combine_contiguous(std::move(state), first, size);
>+  }
>+
>+  template <typename T>
>+  void Init(T* state) {
>+    state_ = state;
>+    combine_contiguous_ = &CombineContiguousImpl<T>;
>+  }
>+
>+  // Do not erase an already erased state.
>+  void Init(HashState* state) {
>+    state_ = state->state_;
>+    combine_contiguous_ = state->combine_contiguous_;
>+  }
>+
>+  void* state_;
>+  void (*combine_contiguous_)(void*, const unsigned char*, size_t);
>+};
>+
>+}  // namespace absl
>+#endif  // ABSL_HASH_HASH_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/hash_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/hash_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..7b6fb2e7ac1cb0931c5e6b9833def64eace2ac77
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/hash_test.cc
>@@ -0,0 +1,425 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/hash/hash.h"
>+
>+#include <array>
>+#include <cstring>
>+#include <deque>
>+#include <forward_list>
>+#include <functional>
>+#include <iterator>
>+#include <limits>
>+#include <list>
>+#include <map>
>+#include <memory>
>+#include <numeric>
>+#include <random>
>+#include <set>
>+#include <string>
>+#include <tuple>
>+#include <type_traits>
>+#include <unordered_map>
>+#include <utility>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/container/flat_hash_set.h"
>+#include "absl/hash/hash_testing.h"
>+#include "absl/hash/internal/spy_hash_state.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/numeric/int128.h"
>+
>+namespace {
>+
>+using absl::Hash;
>+using absl::hash_internal::SpyHashState;
>+
>+template <typename T>
>+class HashValueIntTest : public testing::Test {
>+};
>+TYPED_TEST_CASE_P(HashValueIntTest);
>+
>+template <typename T>
>+SpyHashState SpyHash(const T& value) {
>+  return SpyHashState::combine(SpyHashState(), value);
>+}
>+
>+// Helper trait to verify if T is hashable. We use absl::Hash's poison status to
>+// detect it.
>+template <typename T>
>+using is_hashable = std::is_default_constructible<absl::Hash<T>>;
>+
>+TYPED_TEST_P(HashValueIntTest, BasicUsage) {
>+  EXPECT_TRUE((is_hashable<TypeParam>::value));
>+
>+  TypeParam n = 42;
>+  EXPECT_EQ(SpyHash(n), SpyHash(TypeParam{42}));
>+  EXPECT_NE(SpyHash(n), SpyHash(TypeParam{0}));
>+  EXPECT_NE(SpyHash(std::numeric_limits<TypeParam>::max()),
>+            SpyHash(std::numeric_limits<TypeParam>::min()));
>+}
>+
>+TYPED_TEST_P(HashValueIntTest, FastPath) {
>+  // Test the fast-path to make sure the values are the same.
>+  TypeParam n = 42;
>+  EXPECT_EQ(absl::Hash<TypeParam>{}(n),
>+            absl::Hash<std::tuple<TypeParam>>{}(std::tuple<TypeParam>(n)));
>+}
>+
>+REGISTER_TYPED_TEST_CASE_P(HashValueIntTest, BasicUsage, FastPath);
>+using IntTypes = testing::Types<unsigned char, char, int, int32_t, int64_t, uint32_t,
>+                                uint64_t, size_t>;
>+INSTANTIATE_TYPED_TEST_CASE_P(My, HashValueIntTest, IntTypes);
>+
>+template <typename T, typename = void>
>+struct IsHashCallble : std::false_type {};
>+
>+template <typename T>
>+struct IsHashCallble<T, absl::void_t<decltype(std::declval<absl::Hash<T>>()(
>+                            std::declval<const T&>()))>> : std::true_type {};
>+
>+template <typename T, typename = void>
>+struct IsAggregateInitializable : std::false_type {};
>+
>+template <typename T>
>+struct IsAggregateInitializable<T, absl::void_t<decltype(T{})>>
>+    : std::true_type {};
>+
>+TEST(IsHashableTest, ValidHash) {
>+  EXPECT_TRUE((is_hashable<int>::value));
>+  EXPECT_TRUE(std::is_default_constructible<absl::Hash<int>>::value);
>+  EXPECT_TRUE(std::is_copy_constructible<absl::Hash<int>>::value);
>+  EXPECT_TRUE(std::is_move_constructible<absl::Hash<int>>::value);
>+  EXPECT_TRUE(absl::is_copy_assignable<absl::Hash<int>>::value);
>+  EXPECT_TRUE(absl::is_move_assignable<absl::Hash<int>>::value);
>+  EXPECT_TRUE(IsHashCallble<int>::value);
>+  EXPECT_TRUE(IsAggregateInitializable<absl::Hash<int>>::value);
>+}
>+#if ABSL_HASH_INTERNAL_CAN_POISON_ && !defined(__APPLE__)
>+TEST(IsHashableTest, PoisonHash) {
>+  struct X {};
>+  EXPECT_FALSE((is_hashable<X>::value));
>+  EXPECT_FALSE(std::is_default_constructible<absl::Hash<X>>::value);
>+  EXPECT_FALSE(std::is_copy_constructible<absl::Hash<X>>::value);
>+  EXPECT_FALSE(std::is_move_constructible<absl::Hash<X>>::value);
>+  EXPECT_FALSE(absl::is_copy_assignable<absl::Hash<X>>::value);
>+  EXPECT_FALSE(absl::is_move_assignable<absl::Hash<X>>::value);
>+  EXPECT_FALSE(IsHashCallble<X>::value);
>+  EXPECT_FALSE(IsAggregateInitializable<absl::Hash<X>>::value);
>+}
>+#endif  // ABSL_HASH_INTERNAL_CAN_POISON_
>+
>+// Hashable types
>+//
>+// These types exist simply to exercise various AbslHashValue behaviors, so
>+// they are named by what their AbslHashValue overload does.
>+struct NoOp {
>+  template <typename HashCode>
>+  friend HashCode AbslHashValue(HashCode h, NoOp n) {
>+    return std::move(h);
>+  }
>+};
>+
>+struct EmptyCombine {
>+  template <typename HashCode>
>+  friend HashCode AbslHashValue(HashCode h, EmptyCombine e) {
>+    return HashCode::combine(std::move(h));
>+  }
>+};
>+
>+template <typename Int>
>+struct CombineIterative {
>+  template <typename HashCode>
>+  friend HashCode AbslHashValue(HashCode h, CombineIterative c) {
>+    for (int i = 0; i < 5; ++i) {
>+      h = HashCode::combine(std::move(h), Int(i));
>+    }
>+    return h;
>+  }
>+};
>+
>+template <typename Int>
>+struct CombineVariadic {
>+  template <typename HashCode>
>+  friend HashCode AbslHashValue(HashCode h, CombineVariadic c) {
>+    return HashCode::combine(std::move(h), Int(0), Int(1), Int(2), Int(3),
>+                             Int(4));
>+  }
>+};
>+
>+using InvokeTag = absl::hash_internal::InvokeHashTag;
>+template <InvokeTag T>
>+using InvokeTagConstant = std::integral_constant<InvokeTag, T>;
>+
>+template <InvokeTag... Tags>
>+struct MinTag;
>+
>+template <InvokeTag a, InvokeTag b, InvokeTag... Tags>
>+struct MinTag<a, b, Tags...> : MinTag<(a < b ? a : b), Tags...> {};
>+
>+template <InvokeTag a>
>+struct MinTag<a> : InvokeTagConstant<a> {};
>+
>+template <InvokeTag... Tags>
>+struct CustomHashType {
>+  size_t value;
>+};
>+
>+template <InvokeTag allowed, InvokeTag... tags>
>+struct EnableIfContained
>+    : std::enable_if<absl::disjunction<
>+          std::integral_constant<bool, allowed == tags>...>::value> {};
>+
>+template <
>+    typename H, InvokeTag... Tags,
>+    typename = typename EnableIfContained<InvokeTag::kHashValue, Tags...>::type>
>+H AbslHashValue(H state, CustomHashType<Tags...> t) {
>+  static_assert(MinTag<Tags...>::value == InvokeTag::kHashValue, "");
>+  return H::combine(std::move(state),
>+                    t.value + static_cast<int>(InvokeTag::kHashValue));
>+}
>+
>+}  // namespace
>+
>+namespace absl {
>+namespace hash_internal {
>+template <InvokeTag... Tags>
>+struct is_uniquely_represented<
>+    CustomHashType<Tags...>,
>+    typename EnableIfContained<InvokeTag::kUniquelyRepresented, Tags...>::type>
>+    : std::true_type {};
>+}  // namespace hash_internal
>+}  // namespace absl
>+
>+#if ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_
>+namespace ABSL_INTERNAL_LEGACY_HASH_NAMESPACE {
>+template <InvokeTag... Tags>
>+struct hash<CustomHashType<Tags...>> {
>+  template <InvokeTag... TagsIn, typename = typename EnableIfContained<
>+                                     InvokeTag::kLegacyHash, TagsIn...>::type>
>+  size_t operator()(CustomHashType<TagsIn...> t) const {
>+    static_assert(MinTag<Tags...>::value == InvokeTag::kLegacyHash, "");
>+    return t.value + static_cast<int>(InvokeTag::kLegacyHash);
>+  }
>+};
>+}  // namespace ABSL_INTERNAL_LEGACY_HASH_NAMESPACE
>+#endif  // ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_
>+
>+namespace std {
>+template <InvokeTag... Tags>  // NOLINT
>+struct hash<CustomHashType<Tags...>> {
>+  template <InvokeTag... TagsIn, typename = typename EnableIfContained<
>+                                     InvokeTag::kStdHash, TagsIn...>::type>
>+  size_t operator()(CustomHashType<TagsIn...> t) const {
>+    static_assert(MinTag<Tags...>::value == InvokeTag::kStdHash, "");
>+    return t.value + static_cast<int>(InvokeTag::kStdHash);
>+  }
>+};
>+}  // namespace std
>+
>+namespace {
>+
>+template <typename... T>
>+void TestCustomHashType(InvokeTagConstant<InvokeTag::kNone>, T...) {
>+  using type = CustomHashType<T::value...>;
>+  SCOPED_TRACE(testing::PrintToString(std::vector<InvokeTag>{T::value...}));
>+  EXPECT_TRUE(is_hashable<type>());
>+  EXPECT_TRUE(is_hashable<const type>());
>+  EXPECT_TRUE(is_hashable<const type&>());
>+
>+  const size_t offset = static_cast<int>(std::min({T::value...}));
>+  EXPECT_EQ(SpyHash(type{7}), SpyHash(size_t{7 + offset}));
>+}
>+
>+void TestCustomHashType(InvokeTagConstant<InvokeTag::kNone>) {
>+#if ABSL_HASH_INTERNAL_CAN_POISON_
>+  // is_hashable is false if we don't support any of the hooks.
>+  using type = CustomHashType<>;
>+  EXPECT_FALSE(is_hashable<type>());
>+  EXPECT_FALSE(is_hashable<const type>());
>+  EXPECT_FALSE(is_hashable<const type&>());
>+#endif  // ABSL_HASH_INTERNAL_CAN_POISON_
>+}
>+
>+template <InvokeTag Tag, typename... T>
>+void TestCustomHashType(InvokeTagConstant<Tag> tag, T... t) {
>+  constexpr auto next = static_cast<InvokeTag>(static_cast<int>(Tag) + 1);
>+  TestCustomHashType(InvokeTagConstant<next>(), tag, t...);
>+  TestCustomHashType(InvokeTagConstant<next>(), t...);
>+}
>+
>+TEST(HashTest, CustomHashType) {
>+  TestCustomHashType(InvokeTagConstant<InvokeTag{}>());
>+}
>+
>+TEST(HashTest, NoOpsAreEquivalent) {
>+  EXPECT_EQ(Hash<NoOp>()({}), Hash<NoOp>()({}));
>+  EXPECT_EQ(Hash<NoOp>()({}), Hash<EmptyCombine>()({}));
>+}
>+
>+template <typename T>
>+class HashIntTest : public testing::Test {
>+};
>+TYPED_TEST_CASE_P(HashIntTest);
>+
>+TYPED_TEST_P(HashIntTest, BasicUsage) {
>+  EXPECT_NE(Hash<NoOp>()({}), Hash<TypeParam>()(0));
>+  EXPECT_NE(Hash<NoOp>()({}),
>+            Hash<TypeParam>()(std::numeric_limits<TypeParam>::max()));
>+  if (std::numeric_limits<TypeParam>::min() != 0) {
>+    EXPECT_NE(Hash<NoOp>()({}),
>+              Hash<TypeParam>()(std::numeric_limits<TypeParam>::min()));
>+  }
>+
>+  EXPECT_EQ(Hash<CombineIterative<TypeParam>>()({}),
>+            Hash<CombineVariadic<TypeParam>>()({}));
>+}
>+
>+REGISTER_TYPED_TEST_CASE_P(HashIntTest, BasicUsage);
>+using IntTypes = testing::Types<unsigned char, char, int, int32_t, int64_t, uint32_t,
>+                                uint64_t, size_t>;
>+INSTANTIATE_TYPED_TEST_CASE_P(My, HashIntTest, IntTypes);
>+
>+struct StructWithPadding {
>+  char c;
>+  int i;
>+
>+  template <typename H>
>+  friend H AbslHashValue(H hash_state, const StructWithPadding& s) {
>+    return H::combine(std::move(hash_state), s.c, s.i);
>+  }
>+};
>+
>+static_assert(sizeof(StructWithPadding) > sizeof(char) + sizeof(int),
>+              "StructWithPadding doesn't have padding");
>+static_assert(std::is_standard_layout<StructWithPadding>::value, "");
>+
>+// This check has to be disabled because libstdc++ doesn't support it.
>+// static_assert(std::is_trivially_constructible<StructWithPadding>::value, "");
>+
>+template <typename T>
>+struct ArraySlice {
>+  T* begin;
>+  T* end;
>+
>+  template <typename H>
>+  friend H AbslHashValue(H hash_state, const ArraySlice& slice) {
>+    for (auto t = slice.begin; t != slice.end; ++t) {
>+      hash_state = H::combine(std::move(hash_state), *t);
>+    }
>+    return hash_state;
>+  }
>+};
>+
>+TEST(HashTest, HashNonUniquelyRepresentedType) {
>+  // Create equal StructWithPadding objects that are known to have non-equal
>+  // padding bytes.
>+  static const size_t kNumStructs = 10;
>+  unsigned char buffer1[kNumStructs * sizeof(StructWithPadding)];
>+  std::memset(buffer1, 0, sizeof(buffer1));
>+  auto* s1 = reinterpret_cast<StructWithPadding*>(buffer1);
>+
>+  unsigned char buffer2[kNumStructs * sizeof(StructWithPadding)];
>+  std::memset(buffer2, 255, sizeof(buffer2));
>+  auto* s2 = reinterpret_cast<StructWithPadding*>(buffer2);
>+  for (int i = 0; i < kNumStructs; ++i) {
>+    SCOPED_TRACE(i);
>+    s1[i].c = s2[i].c = '0' + i;
>+    s1[i].i = s2[i].i = i;
>+    ASSERT_FALSE(memcmp(buffer1 + i * sizeof(StructWithPadding),
>+                        buffer2 + i * sizeof(StructWithPadding),
>+                        sizeof(StructWithPadding)) == 0)
>+        << "Bug in test code: objects do not have unequal"
>+        << " object representations";
>+  }
>+
>+  EXPECT_EQ(Hash<StructWithPadding>()(s1[0]), Hash<StructWithPadding>()(s2[0]));
>+  EXPECT_EQ(Hash<ArraySlice<StructWithPadding>>()({s1, s1 + kNumStructs}),
>+            Hash<ArraySlice<StructWithPadding>>()({s2, s2 + kNumStructs}));
>+}
>+
>+TEST(HashTest, StandardHashContainerUsage) {
>+  std::unordered_map<int, std::string, Hash<int>> map = {{0, "foo"}, { 42, "bar" }};
>+
>+  EXPECT_NE(map.find(0), map.end());
>+  EXPECT_EQ(map.find(1), map.end());
>+  EXPECT_NE(map.find(0u), map.end());
>+}
>+
>+struct ConvertibleFromNoOp {
>+  ConvertibleFromNoOp(NoOp) {}  // NOLINT(runtime/explicit)
>+
>+  template <typename H>
>+  friend H AbslHashValue(H hash_state, ConvertibleFromNoOp) {
>+    return H::combine(std::move(hash_state), 1);
>+  }
>+};
>+
>+TEST(HashTest, HeterogeneousCall) {
>+  EXPECT_NE(Hash<ConvertibleFromNoOp>()(NoOp()),
>+            Hash<NoOp>()(NoOp()));
>+}
>+
>+TEST(IsUniquelyRepresentedTest, SanityTest) {
>+  using absl::hash_internal::is_uniquely_represented;
>+
>+  EXPECT_TRUE(is_uniquely_represented<unsigned char>::value);
>+  EXPECT_TRUE(is_uniquely_represented<int>::value);
>+  EXPECT_FALSE(is_uniquely_represented<bool>::value);
>+  EXPECT_FALSE(is_uniquely_represented<int*>::value);
>+}
>+
>+struct IntAndString {
>+  int i;
>+  std::string s;
>+
>+  template <typename H>
>+  friend H AbslHashValue(H hash_state, IntAndString int_and_string) {
>+    return H::combine(std::move(hash_state), int_and_string.s,
>+                      int_and_string.i);
>+  }
>+};
>+
>+TEST(HashTest, SmallValueOn64ByteBoundary) {
>+  Hash<IntAndString>()(IntAndString{0, std::string(63, '0')});
>+}
>+
>+struct TypeErased {
>+  size_t n;
>+
>+  template <typename H>
>+  friend H AbslHashValue(H hash_state, const TypeErased& v) {
>+    v.HashValue(absl::HashState::Create(&hash_state));
>+    return hash_state;
>+  }
>+
>+  void HashValue(absl::HashState state) const {
>+    absl::HashState::combine(std::move(state), n);
>+  }
>+};
>+
>+TEST(HashTest, TypeErased) {
>+  EXPECT_TRUE((is_hashable<TypeErased>::value));
>+  EXPECT_TRUE((is_hashable<std::pair<TypeErased, int>>::value));
>+
>+  EXPECT_EQ(SpyHash(TypeErased{7}), SpyHash(size_t{7}));
>+  EXPECT_NE(SpyHash(TypeErased{7}), SpyHash(size_t{13}));
>+
>+  EXPECT_EQ(SpyHash(std::make_pair(TypeErased{7}, 17)),
>+            SpyHash(std::make_pair(size_t{7}, 17)));
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/hash_testing.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/hash_testing.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..1e3cda64467dfab0ee1661c075f63c06bc21800d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/hash_testing.h
>@@ -0,0 +1,372 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_HASH_HASH_TESTING_H_
>+#define ABSL_HASH_HASH_TESTING_H_
>+
>+#include <initializer_list>
>+#include <tuple>
>+#include <type_traits>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/hash/internal/spy_hash_state.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/strings/str_cat.h"
>+#include "absl/types/variant.h"
>+
>+namespace absl {
>+
>+// Run the absl::Hash algorithm over all the elements passed in and verify that
>+// their hash expansion is congruent with their `==` operator.
>+//
>+// It is used in conjunction with EXPECT_TRUE. Failures will output information
>+// on what requirement failed and on which objects.
>+//
>+// Users should pass a collection of types as either an initializer list or a
>+// container of cases.
>+//
>+//   EXPECT_TRUE(absl::VerifyTypeImplementsAbslHashCorrectly(
>+//       {v1, v2, ..., vN}));
>+//
>+//   std::vector<MyType> cases;
>+//   // Fill cases...
>+//   EXPECT_TRUE(absl::VerifyTypeImplementsAbslHashCorrectly(cases));
>+//
>+// Users can pass a variety of types for testing heterogeneous lookup with
>+// `std::make_tuple`:
>+//
>+//   EXPECT_TRUE(absl::VerifyTypeImplementsAbslHashCorrectly(
>+//       std::make_tuple(v1, v2, ..., vN)));
>+//
>+//
>+// Ideally, the values passed should provide enough coverage of the `==`
>+// operator and the AbslHashValue implementations.
>+// For dynamically sized types, the empty state should usually be included in
>+// the values.
>+//
>+// The function accepts an optional comparator function, in case that `==` is
>+// not enough for the values provided.
>+//
>+// Usage:
>+//
>+//   EXPECT_TRUE(absl::VerifyTypeImplementsAbslHashCorrectly(
>+//       std::make_tuple(v1, v2, ..., vN), MyCustomEq{}));
>+//
>+// It checks the following requirements:
>+//   1. The expansion for a value is deterministic.
>+//   2. For any two objects `a` and `b` in the sequence, if `a == b` evaluates
>+//      to true, then their hash expansion must be equal.
>+//   3. If `a == b` evaluates to false their hash expansion must be unequal.
>+//   4. If `a == b` evaluates to false neither hash expansion can be a
>+//      suffix of the other.
>+//   5. AbslHashValue overloads should not be called by the user. They are only
>+//      meant to be called by the framework. Users should call H::combine() and
>+//      H::combine_contiguous().
>+//   6. No moved-from instance of the hash state is used in the implementation
>+//      of AbslHashValue.
>+//
>+// The values do not have to have the same type. This can be useful for
>+// equivalent types that support heterogeneous lookup.
>+//
>+// A possible reason for breaking (2) is combining state in the hash expansion
>+// that was not used in `==`.
>+// For example:
>+//
>+// struct Bad2 {
>+//   int a, b;
>+//   template <typename H>
>+//   friend H AbslHashValue(H state, Bad2 x) {
>+//     // Uses a and b.
>+//     return H::combine(x.a, x.b);
>+//   }
>+//   friend bool operator==(Bad2 x, Bad2 y) {
>+//     // Only uses a.
>+//     return x.a == y.a;
>+//   }
>+// };
>+//
>+// As for (3), breaking this usually means that there is state being passed to
>+// the `==` operator that is not used in the hash expansion.
>+// For example:
>+//
>+// struct Bad3 {
>+//   int a, b;
>+//   template <typename H>
>+//   friend H AbslHashValue(H state, Bad3 x) {
>+//     // Only uses a.
>+//     return H::combine(x.a);
>+//   }
>+//   friend bool operator==(Bad3 x, Bad3 y) {
>+//     // Uses a and b.
>+//     return x.a == y.a && x.b == y.b;
>+//   }
>+// };
>+//
>+// Finally, a common way to break 4 is by combining dynamic ranges without
>+// combining the size of the range.
>+// For example:
>+//
>+// struct Bad4 {
>+//   int *p, size;
>+//   template <typename H>
>+//   friend H AbslHashValue(H state, Bad4 x) {
>+//     return H::combine_range(x.p, x.p + x.size);
>+//   }
>+//   friend bool operator==(Bad4 x, Bad4 y) {
>+//     return std::equal(x.p, x.p + x.size, y.p, y.p + y.size);
>+//   }
>+// };
>+//
>+// An easy solution to this is to combine the size after combining the range,
>+// like so:
>+//   template <typename H>
>+//   friend H AbslHashValue(H state, Bad4 x) {
>+//     return H::combine(H::combine_range(x.p, x.p + x.size), x.size);
>+//   }
>+//
>+template <int&... ExplicitBarrier, typename Container>
>+ABSL_MUST_USE_RESULT testing::AssertionResult
>+VerifyTypeImplementsAbslHashCorrectly(const Container& values);
>+
>+template <int&... ExplicitBarrier, typename Container, typename Eq>
>+ABSL_MUST_USE_RESULT testing::AssertionResult
>+VerifyTypeImplementsAbslHashCorrectly(const Container& values, Eq equals);
>+
>+template <int&..., typename T>
>+ABSL_MUST_USE_RESULT testing::AssertionResult
>+VerifyTypeImplementsAbslHashCorrectly(std::initializer_list<T> values);
>+
>+template <int&..., typename T, typename Eq>
>+ABSL_MUST_USE_RESULT testing::AssertionResult
>+VerifyTypeImplementsAbslHashCorrectly(std::initializer_list<T> values,
>+                                      Eq equals);
>+
>+namespace hash_internal {
>+
>+struct PrintVisitor {
>+  size_t index;
>+  template <typename T>
>+  std::string operator()(const T* value) const {
>+    return absl::StrCat("#", index, "(", testing::PrintToString(*value), ")");
>+  }
>+};
>+
>+template <typename Eq>
>+struct EqVisitor {
>+  Eq eq;
>+  template <typename T, typename U>
>+  bool operator()(const T* t, const U* u) const {
>+    return eq(*t, *u);
>+  }
>+};
>+
>+struct ExpandVisitor {
>+  template <typename T>
>+  SpyHashState operator()(const T* value) const {
>+    return SpyHashState::combine(SpyHashState(), *value);
>+  }
>+};
>+
>+template <typename Container, typename Eq>
>+ABSL_MUST_USE_RESULT testing::AssertionResult
>+VerifyTypeImplementsAbslHashCorrectly(const Container& values, Eq equals) {
>+  using V = typename Container::value_type;
>+
>+  struct Info {
>+    const V& value;
>+    size_t index;
>+    std::string ToString() const { return absl::visit(PrintVisitor{index}, value); }
>+    SpyHashState expand() const { return absl::visit(ExpandVisitor{}, value); }
>+  };
>+
>+  using EqClass = std::vector<Info>;
>+  std::vector<EqClass> classes;
>+
>+  // Gather the values in equivalence classes.
>+  size_t i = 0;
>+  for (const auto& value : values) {
>+    EqClass* c = nullptr;
>+    for (auto& eqclass : classes) {
>+      if (absl::visit(EqVisitor<Eq>{equals}, value, eqclass[0].value)) {
>+        c = &eqclass;
>+        break;
>+      }
>+    }
>+    if (c == nullptr) {
>+      classes.emplace_back();
>+      c = &classes.back();
>+    }
>+    c->push_back({value, i});
>+    ++i;
>+
>+    // Verify potential errors captured by SpyHashState.
>+    if (auto error = c->back().expand().error()) {
>+      return testing::AssertionFailure() << *error;
>+    }
>+  }
>+
>+  if (classes.size() < 2) {
>+    return testing::AssertionFailure()
>+           << "At least two equivalence classes are expected.";
>+  }
>+
>+  // We assume that equality is correctly implemented.
>+  // Now we verify that AbslHashValue is also correctly implemented.
>+
>+  for (const auto& c : classes) {
>+    // All elements of the equivalence class must have the same hash expansion.
>+    const SpyHashState expected = c[0].expand();
>+    for (const Info& v : c) {
>+      if (v.expand() != v.expand()) {
>+        return testing::AssertionFailure()
>+               << "Hash expansion for " << v.ToString()
>+               << " is non-deterministic.";
>+      }
>+      if (v.expand() != expected) {
>+        return testing::AssertionFailure()
>+               << "Values " << c[0].ToString() << " and " << v.ToString()
>+               << " evaluate as equal but have an unequal hash expansion.";
>+      }
>+    }
>+
>+    // Elements from other classes must have different hash expansion.
>+    for (const auto& c2 : classes) {
>+      if (&c == &c2) continue;
>+      const SpyHashState c2_hash = c2[0].expand();
>+      switch (SpyHashState::Compare(expected, c2_hash)) {
>+        case SpyHashState::CompareResult::kEqual:
>+          return testing::AssertionFailure()
>+                 << "Values " << c[0].ToString() << " and " << c2[0].ToString()
>+                 << " evaluate as unequal but have an equal hash expansion.";
>+        case SpyHashState::CompareResult::kBSuffixA:
>+          return testing::AssertionFailure()
>+                 << "Hash expansion of " << c2[0].ToString()
>+                 << " is a suffix of the hash expansion of " << c[0].ToString()
>+                 << ".";
>+        case SpyHashState::CompareResult::kASuffixB:
>+          return testing::AssertionFailure()
>+                 << "Hash expansion of " << c[0].ToString()
>+                 << " is a suffix of the hash expansion of " << c2[0].ToString()
>+                 << ".";
>+        case SpyHashState::CompareResult::kUnequal:
>+          break;
>+      }
>+    }
>+  }
>+  return testing::AssertionSuccess();
>+}
>+
>+template <typename... T>
>+struct TypeSet {
>+  template <typename U, bool = disjunction<std::is_same<T, U>...>::value>
>+  struct Insert {
>+    using type = TypeSet<U, T...>;
>+  };
>+  template <typename U>
>+  struct Insert<U, true> {
>+    using type = TypeSet;
>+  };
>+
>+  template <template <typename...> class C>
>+  using apply = C<T...>;
>+};
>+
>+template <typename... T>
>+struct MakeTypeSet : TypeSet<>{};
>+template <typename T, typename... Ts>
>+struct MakeTypeSet<T, Ts...> : MakeTypeSet<Ts...>::template Insert<T>::type {};
>+
>+template <typename... T>
>+using VariantForTypes = typename MakeTypeSet<
>+    const typename std::decay<T>::type*...>::template apply<absl::variant>;
>+
>+template <typename Container>
>+struct ContainerAsVector {
>+  using V = absl::variant<const typename Container::value_type*>;
>+  using Out = std::vector<V>;
>+
>+  static Out Do(const Container& values) {
>+    Out out;
>+    for (const auto& v : values) out.push_back(&v);
>+    return out;
>+  }
>+};
>+
>+template <typename... T>
>+struct ContainerAsVector<std::tuple<T...>> {
>+  using V = VariantForTypes<T...>;
>+  using Out = std::vector<V>;
>+
>+  template <size_t... I>
>+  static Out DoImpl(const std::tuple<T...>& tuple, absl::index_sequence<I...>) {
>+    return Out{&std::get<I>(tuple)...};
>+  }
>+
>+  static Out Do(const std::tuple<T...>& values) {
>+    return DoImpl(values, absl::index_sequence_for<T...>());
>+  }
>+};
>+
>+template <>
>+struct ContainerAsVector<std::tuple<>> {
>+  static std::vector<VariantForTypes<int>> Do(std::tuple<>) { return {}; }
>+};
>+
>+struct DefaultEquals {
>+  template <typename T, typename U>
>+  bool operator()(const T& t, const U& u) const {
>+    return t == u;
>+  }
>+};
>+
>+}  // namespace hash_internal
>+
>+template <int&..., typename Container>
>+ABSL_MUST_USE_RESULT testing::AssertionResult
>+VerifyTypeImplementsAbslHashCorrectly(const Container& values) {
>+  return hash_internal::VerifyTypeImplementsAbslHashCorrectly(
>+      hash_internal::ContainerAsVector<Container>::Do(values),
>+      hash_internal::DefaultEquals{});
>+}
>+
>+template <int&..., typename Container, typename Eq>
>+ABSL_MUST_USE_RESULT testing::AssertionResult
>+VerifyTypeImplementsAbslHashCorrectly(const Container& values, Eq equals) {
>+  return hash_internal::VerifyTypeImplementsAbslHashCorrectly(
>+      hash_internal::ContainerAsVector<Container>::Do(values),
>+      equals);
>+}
>+
>+template <int&..., typename T>
>+ABSL_MUST_USE_RESULT testing::AssertionResult
>+VerifyTypeImplementsAbslHashCorrectly(std::initializer_list<T> values) {
>+  return hash_internal::VerifyTypeImplementsAbslHashCorrectly(
>+      hash_internal::ContainerAsVector<std::initializer_list<T>>::Do(values),
>+      hash_internal::DefaultEquals{});
>+}
>+
>+template <int&..., typename T, typename Eq>
>+ABSL_MUST_USE_RESULT testing::AssertionResult
>+VerifyTypeImplementsAbslHashCorrectly(std::initializer_list<T> values,
>+                                      Eq equals) {
>+  return hash_internal::VerifyTypeImplementsAbslHashCorrectly(
>+      hash_internal::ContainerAsVector<std::initializer_list<T>>::Do(values),
>+      equals);
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_HASH_HASH_TESTING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..8f72dd1b7d0c9f538d91d627dff0662e4e0c9b2c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city.cc
>@@ -0,0 +1,590 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This file provides CityHash64() and related functions.
>+//
>+// It's probably possible to create even faster hash functions by
>+// writing a program that systematically explores some of the space of
>+// possible hash functions, by using SIMD instructions, or by
>+// compromising on hash quality.
>+
>+#include "absl/hash/internal/city.h"
>+
>+#include <string.h>  // for memcpy and memset
>+#include <algorithm>
>+
>+#include "absl/base/config.h"
>+#include "absl/base/internal/endian.h"
>+#include "absl/base/internal/unaligned_access.h"
>+#include "absl/base/optimization.h"
>+
>+namespace absl {
>+namespace hash_internal {
>+
>+#ifdef ABSL_IS_BIG_ENDIAN
>+#define uint32_in_expected_order(x) (absl::gbswap_32(x))
>+#define uint64_in_expected_order(x) (absl::gbswap_64(x))
>+#else
>+#define uint32_in_expected_order(x) (x)
>+#define uint64_in_expected_order(x) (x)
>+#endif
>+
>+static uint64_t Fetch64(const char *p) {
>+  return uint64_in_expected_order(ABSL_INTERNAL_UNALIGNED_LOAD64(p));
>+}
>+
>+static uint32_t Fetch32(const char *p) {
>+  return uint32_in_expected_order(ABSL_INTERNAL_UNALIGNED_LOAD32(p));
>+}
>+
>+// Some primes between 2^63 and 2^64 for various uses.
>+static const uint64_t k0 = 0xc3a5c85c97cb3127ULL;
>+static const uint64_t k1 = 0xb492b66fbe98f273ULL;
>+static const uint64_t k2 = 0x9ae16a3b2f90404fULL;
>+
>+// Magic numbers for 32-bit hashing.  Copied from Murmur3.
>+static const uint32_t c1 = 0xcc9e2d51;
>+static const uint32_t c2 = 0x1b873593;
>+
>+// A 32-bit to 32-bit integer hash copied from Murmur3.
>+static uint32_t fmix(uint32_t h) {
>+  h ^= h >> 16;
>+  h *= 0x85ebca6b;
>+  h ^= h >> 13;
>+  h *= 0xc2b2ae35;
>+  h ^= h >> 16;
>+  return h;
>+}
>+
>+static uint32_t Rotate32(uint32_t val, int shift) {
>+  // Avoid shifting by 32: doing so yields an undefined result.
>+  return shift == 0 ? val : ((val >> shift) | (val << (32 - shift)));
>+}
>+
>+#undef PERMUTE3
>+#define PERMUTE3(a, b, c) \
>+  do {                    \
>+    std::swap(a, b);      \
>+    std::swap(a, c);      \
>+  } while (0)
>+
>+static uint32_t Mur(uint32_t a, uint32_t h) {
>+  // Helper from Murmur3 for combining two 32-bit values.
>+  a *= c1;
>+  a = Rotate32(a, 17);
>+  a *= c2;
>+  h ^= a;
>+  h = Rotate32(h, 19);
>+  return h * 5 + 0xe6546b64;
>+}
>+
>+static uint32_t Hash32Len13to24(const char *s, size_t len) {
>+  uint32_t a = Fetch32(s - 4 + (len >> 1));
>+  uint32_t b = Fetch32(s + 4);
>+  uint32_t c = Fetch32(s + len - 8);
>+  uint32_t d = Fetch32(s + (len >> 1));
>+  uint32_t e = Fetch32(s);
>+  uint32_t f = Fetch32(s + len - 4);
>+  uint32_t h = len;
>+
>+  return fmix(Mur(f, Mur(e, Mur(d, Mur(c, Mur(b, Mur(a, h)))))));
>+}
>+
>+static uint32_t Hash32Len0to4(const char *s, size_t len) {
>+  uint32_t b = 0;
>+  uint32_t c = 9;
>+  for (size_t i = 0; i < len; i++) {
>+    signed char v = s[i];
>+    b = b * c1 + v;
>+    c ^= b;
>+  }
>+  return fmix(Mur(b, Mur(len, c)));
>+}
>+
>+static uint32_t Hash32Len5to12(const char *s, size_t len) {
>+  uint32_t a = len, b = len * 5, c = 9, d = b;
>+  a += Fetch32(s);
>+  b += Fetch32(s + len - 4);
>+  c += Fetch32(s + ((len >> 1) & 4));
>+  return fmix(Mur(c, Mur(b, Mur(a, d))));
>+}
>+
>+uint32_t CityHash32(const char *s, size_t len) {
>+  if (len <= 24) {
>+    return len <= 12
>+               ? (len <= 4 ? Hash32Len0to4(s, len) : Hash32Len5to12(s, len))
>+               : Hash32Len13to24(s, len);
>+  }
>+
>+  // len > 24
>+  uint32_t h = len, g = c1 * len, f = g;
>+
>+  uint32_t a0 = Rotate32(Fetch32(s + len - 4) * c1, 17) * c2;
>+  uint32_t a1 = Rotate32(Fetch32(s + len - 8) * c1, 17) * c2;
>+  uint32_t a2 = Rotate32(Fetch32(s + len - 16) * c1, 17) * c2;
>+  uint32_t a3 = Rotate32(Fetch32(s + len - 12) * c1, 17) * c2;
>+  uint32_t a4 = Rotate32(Fetch32(s + len - 20) * c1, 17) * c2;
>+  h ^= a0;
>+  h = Rotate32(h, 19);
>+  h = h * 5 + 0xe6546b64;
>+  h ^= a2;
>+  h = Rotate32(h, 19);
>+  h = h * 5 + 0xe6546b64;
>+  g ^= a1;
>+  g = Rotate32(g, 19);
>+  g = g * 5 + 0xe6546b64;
>+  g ^= a3;
>+  g = Rotate32(g, 19);
>+  g = g * 5 + 0xe6546b64;
>+  f += a4;
>+  f = Rotate32(f, 19);
>+  f = f * 5 + 0xe6546b64;
>+  size_t iters = (len - 1) / 20;
>+  do {
>+    uint32_t b0 = Rotate32(Fetch32(s) * c1, 17) * c2;
>+    uint32_t b1 = Fetch32(s + 4);
>+    uint32_t b2 = Rotate32(Fetch32(s + 8) * c1, 17) * c2;
>+    uint32_t b3 = Rotate32(Fetch32(s + 12) * c1, 17) * c2;
>+    uint32_t b4 = Fetch32(s + 16);
>+    h ^= b0;
>+    h = Rotate32(h, 18);
>+    h = h * 5 + 0xe6546b64;
>+    f += b1;
>+    f = Rotate32(f, 19);
>+    f = f * c1;
>+    g += b2;
>+    g = Rotate32(g, 18);
>+    g = g * 5 + 0xe6546b64;
>+    h ^= b3 + b1;
>+    h = Rotate32(h, 19);
>+    h = h * 5 + 0xe6546b64;
>+    g ^= b4;
>+    g = absl::gbswap_32(g) * 5;
>+    h += b4 * 5;
>+    h = absl::gbswap_32(h);
>+    f += b0;
>+    PERMUTE3(f, h, g);
>+    s += 20;
>+  } while (--iters != 0);
>+  g = Rotate32(g, 11) * c1;
>+  g = Rotate32(g, 17) * c1;
>+  f = Rotate32(f, 11) * c1;
>+  f = Rotate32(f, 17) * c1;
>+  h = Rotate32(h + g, 19);
>+  h = h * 5 + 0xe6546b64;
>+  h = Rotate32(h, 17) * c1;
>+  h = Rotate32(h + f, 19);
>+  h = h * 5 + 0xe6546b64;
>+  h = Rotate32(h, 17) * c1;
>+  return h;
>+}
>+
>+// Bitwise right rotate.  Normally this will compile to a single
>+// instruction, especially if the shift is a manifest constant.
>+static uint64_t Rotate(uint64_t val, int shift) {
>+  // Avoid shifting by 64: doing so yields an undefined result.
>+  return shift == 0 ? val : ((val >> shift) | (val << (64 - shift)));
>+}
>+
>+static uint64_t ShiftMix(uint64_t val) { return val ^ (val >> 47); }
>+
>+static uint64_t HashLen16(uint64_t u, uint64_t v) {
>+  return Hash128to64(uint128(u, v));
>+}
>+
>+static uint64_t HashLen16(uint64_t u, uint64_t v, uint64_t mul) {
>+  // Murmur-inspired hashing.
>+  uint64_t a = (u ^ v) * mul;
>+  a ^= (a >> 47);
>+  uint64_t b = (v ^ a) * mul;
>+  b ^= (b >> 47);
>+  b *= mul;
>+  return b;
>+}
>+
>+static uint64_t HashLen0to16(const char *s, size_t len) {
>+  if (len >= 8) {
>+    uint64_t mul = k2 + len * 2;
>+    uint64_t a = Fetch64(s) + k2;
>+    uint64_t b = Fetch64(s + len - 8);
>+    uint64_t c = Rotate(b, 37) * mul + a;
>+    uint64_t d = (Rotate(a, 25) + b) * mul;
>+    return HashLen16(c, d, mul);
>+  }
>+  if (len >= 4) {
>+    uint64_t mul = k2 + len * 2;
>+    uint64_t a = Fetch32(s);
>+    return HashLen16(len + (a << 3), Fetch32(s + len - 4), mul);
>+  }
>+  if (len > 0) {
>+    uint8_t a = s[0];
>+    uint8_t b = s[len >> 1];
>+    uint8_t c = s[len - 1];
>+    uint32_t y = static_cast<uint32_t>(a) + (static_cast<uint32_t>(b) << 8);
>+    uint32_t z = len + (static_cast<uint32_t>(c) << 2);
>+    return ShiftMix(y * k2 ^ z * k0) * k2;
>+  }
>+  return k2;
>+}
>+
>+// This probably works well for 16-byte strings as well, but it may be overkill
>+// in that case.
>+static uint64_t HashLen17to32(const char *s, size_t len) {
>+  uint64_t mul = k2 + len * 2;
>+  uint64_t a = Fetch64(s) * k1;
>+  uint64_t b = Fetch64(s + 8);
>+  uint64_t c = Fetch64(s + len - 8) * mul;
>+  uint64_t d = Fetch64(s + len - 16) * k2;
>+  return HashLen16(Rotate(a + b, 43) + Rotate(c, 30) + d,
>+                   a + Rotate(b + k2, 18) + c, mul);
>+}
>+
>+// Return a 16-byte hash for 48 bytes.  Quick and dirty.
>+// Callers do best to use "random-looking" values for a and b.
>+static std::pair<uint64_t, uint64_t> WeakHashLen32WithSeeds(uint64_t w, uint64_t x,
>+                                                        uint64_t y, uint64_t z,
>+                                                        uint64_t a, uint64_t b) {
>+  a += w;
>+  b = Rotate(b + a + z, 21);
>+  uint64_t c = a;
>+  a += x;
>+  a += y;
>+  b += Rotate(a, 44);
>+  return std::make_pair(a + z, b + c);
>+}
>+
>+// Return a 16-byte hash for s[0] ... s[31], a, and b.  Quick and dirty.
>+static std::pair<uint64_t, uint64_t> WeakHashLen32WithSeeds(const char *s, uint64_t a,
>+                                                        uint64_t b) {
>+  return WeakHashLen32WithSeeds(Fetch64(s), Fetch64(s + 8), Fetch64(s + 16),
>+                                Fetch64(s + 24), a, b);
>+}
>+
>+// Return an 8-byte hash for 33 to 64 bytes.
>+static uint64_t HashLen33to64(const char *s, size_t len) {
>+  uint64_t mul = k2 + len * 2;
>+  uint64_t a = Fetch64(s) * k2;
>+  uint64_t b = Fetch64(s + 8);
>+  uint64_t c = Fetch64(s + len - 24);
>+  uint64_t d = Fetch64(s + len - 32);
>+  uint64_t e = Fetch64(s + 16) * k2;
>+  uint64_t f = Fetch64(s + 24) * 9;
>+  uint64_t g = Fetch64(s + len - 8);
>+  uint64_t h = Fetch64(s + len - 16) * mul;
>+  uint64_t u = Rotate(a + g, 43) + (Rotate(b, 30) + c) * 9;
>+  uint64_t v = ((a + g) ^ d) + f + 1;
>+  uint64_t w = absl::gbswap_64((u + v) * mul) + h;
>+  uint64_t x = Rotate(e + f, 42) + c;
>+  uint64_t y = (absl::gbswap_64((v + w) * mul) + g) * mul;
>+  uint64_t z = e + f + c;
>+  a = absl::gbswap_64((x + z) * mul + y) + b;
>+  b = ShiftMix((z + a) * mul + d + h) * mul;
>+  return b + x;
>+}
>+
>+uint64_t CityHash64(const char *s, size_t len) {
>+  if (len <= 32) {
>+    if (len <= 16) {
>+      return HashLen0to16(s, len);
>+    } else {
>+      return HashLen17to32(s, len);
>+    }
>+  } else if (len <= 64) {
>+    return HashLen33to64(s, len);
>+  }
>+
>+  // For strings over 64 bytes we hash the end first, and then as we
>+  // loop we keep 56 bytes of state: v, w, x, y, and z.
>+  uint64_t x = Fetch64(s + len - 40);
>+  uint64_t y = Fetch64(s + len - 16) + Fetch64(s + len - 56);
>+  uint64_t z = HashLen16(Fetch64(s + len - 48) + len, Fetch64(s + len - 24));
>+  std::pair<uint64_t, uint64_t> v = WeakHashLen32WithSeeds(s + len - 64, len, z);
>+  std::pair<uint64_t, uint64_t> w = WeakHashLen32WithSeeds(s + len - 32, y + k1, x);
>+  x = x * k1 + Fetch64(s);
>+
>+  // Decrease len to the nearest multiple of 64, and operate on 64-byte chunks.
>+  len = (len - 1) & ~static_cast<size_t>(63);
>+  do {
>+    x = Rotate(x + y + v.first + Fetch64(s + 8), 37) * k1;
>+    y = Rotate(y + v.second + Fetch64(s + 48), 42) * k1;
>+    x ^= w.second;
>+    y += v.first + Fetch64(s + 40);
>+    z = Rotate(z + w.first, 33) * k1;
>+    v = WeakHashLen32WithSeeds(s, v.second * k1, x + w.first);
>+    w = WeakHashLen32WithSeeds(s + 32, z + w.second, y + Fetch64(s + 16));
>+    std::swap(z, x);
>+    s += 64;
>+    len -= 64;
>+  } while (len != 0);
>+  return HashLen16(HashLen16(v.first, w.first) + ShiftMix(y) * k1 + z,
>+                   HashLen16(v.second, w.second) + x);
>+}
>+
>+uint64_t CityHash64WithSeed(const char *s, size_t len, uint64_t seed) {
>+  return CityHash64WithSeeds(s, len, k2, seed);
>+}
>+
>+uint64_t CityHash64WithSeeds(const char *s, size_t len, uint64_t seed0,
>+                           uint64_t seed1) {
>+  return HashLen16(CityHash64(s, len) - seed0, seed1);
>+}
>+
>+// A subroutine for CityHash128().  Returns a decent 128-bit hash for strings
>+// of any length representable in signed long.  Based on City and Murmur.
>+static uint128 CityMurmur(const char *s, size_t len, uint128 seed) {
>+  uint64_t a = Uint128Low64(seed);
>+  uint64_t b = Uint128High64(seed);
>+  uint64_t c = 0;
>+  uint64_t d = 0;
>+  int64_t l = len - 16;
>+  if (l <= 0) {  // len <= 16
>+    a = ShiftMix(a * k1) * k1;
>+    c = b * k1 + HashLen0to16(s, len);
>+    d = ShiftMix(a + (len >= 8 ? Fetch64(s) : c));
>+  } else {  // len > 16
>+    c = HashLen16(Fetch64(s + len - 8) + k1, a);
>+    d = HashLen16(b + len, c + Fetch64(s + len - 16));
>+    a += d;
>+    do {
>+      a ^= ShiftMix(Fetch64(s) * k1) * k1;
>+      a *= k1;
>+      b ^= a;
>+      c ^= ShiftMix(Fetch64(s + 8) * k1) * k1;
>+      c *= k1;
>+      d ^= c;
>+      s += 16;
>+      l -= 16;
>+    } while (l > 0);
>+  }
>+  a = HashLen16(a, c);
>+  b = HashLen16(d, b);
>+  return uint128(a ^ b, HashLen16(b, a));
>+}
>+
>+uint128 CityHash128WithSeed(const char *s, size_t len, uint128 seed) {
>+  if (len < 128) {
>+    return CityMurmur(s, len, seed);
>+  }
>+
>+  // We expect len >= 128 to be the common case.  Keep 56 bytes of state:
>+  // v, w, x, y, and z.
>+  std::pair<uint64_t, uint64_t> v, w;
>+  uint64_t x = Uint128Low64(seed);
>+  uint64_t y = Uint128High64(seed);
>+  uint64_t z = len * k1;
>+  v.first = Rotate(y ^ k1, 49) * k1 + Fetch64(s);
>+  v.second = Rotate(v.first, 42) * k1 + Fetch64(s + 8);
>+  w.first = Rotate(y + z, 35) * k1 + x;
>+  w.second = Rotate(x + Fetch64(s + 88), 53) * k1;
>+
>+  // This is the same inner loop as CityHash64(), manually unrolled.
>+  do {
>+    x = Rotate(x + y + v.first + Fetch64(s + 8), 37) * k1;
>+    y = Rotate(y + v.second + Fetch64(s + 48), 42) * k1;
>+    x ^= w.second;
>+    y += v.first + Fetch64(s + 40);
>+    z = Rotate(z + w.first, 33) * k1;
>+    v = WeakHashLen32WithSeeds(s, v.second * k1, x + w.first);
>+    w = WeakHashLen32WithSeeds(s + 32, z + w.second, y + Fetch64(s + 16));
>+    std::swap(z, x);
>+    s += 64;
>+    x = Rotate(x + y + v.first + Fetch64(s + 8), 37) * k1;
>+    y = Rotate(y + v.second + Fetch64(s + 48), 42) * k1;
>+    x ^= w.second;
>+    y += v.first + Fetch64(s + 40);
>+    z = Rotate(z + w.first, 33) * k1;
>+    v = WeakHashLen32WithSeeds(s, v.second * k1, x + w.first);
>+    w = WeakHashLen32WithSeeds(s + 32, z + w.second, y + Fetch64(s + 16));
>+    std::swap(z, x);
>+    s += 64;
>+    len -= 128;
>+  } while (ABSL_PREDICT_TRUE(len >= 128));
>+  x += Rotate(v.first + z, 49) * k0;
>+  y = y * k0 + Rotate(w.second, 37);
>+  z = z * k0 + Rotate(w.first, 27);
>+  w.first *= 9;
>+  v.first *= k0;
>+  // If 0 < len < 128, hash up to 4 chunks of 32 bytes each from the end of s.
>+  for (size_t tail_done = 0; tail_done < len;) {
>+    tail_done += 32;
>+    y = Rotate(x + y, 42) * k0 + v.second;
>+    w.first += Fetch64(s + len - tail_done + 16);
>+    x = x * k0 + w.first;
>+    z += w.second + Fetch64(s + len - tail_done);
>+    w.second += v.first;
>+    v = WeakHashLen32WithSeeds(s + len - tail_done, v.first + z, v.second);
>+    v.first *= k0;
>+  }
>+  // At this point our 56 bytes of state should contain more than
>+  // enough information for a strong 128-bit hash.  We use two
>+  // different 56-byte-to-8-byte hashes to get a 16-byte final result.
>+  x = HashLen16(x, v.first);
>+  y = HashLen16(y + z, w.first);
>+  return uint128(HashLen16(x + v.second, w.second) + y,
>+                 HashLen16(x + w.second, y + v.second));
>+}
>+
>+uint128 CityHash128(const char *s, size_t len) {
>+  return len >= 16
>+             ? CityHash128WithSeed(s + 16, len - 16,
>+                                   uint128(Fetch64(s), Fetch64(s + 8) + k0))
>+             : CityHash128WithSeed(s, len, uint128(k0, k1));
>+}
>+}  // namespace hash_internal
>+}  // namespace absl
>+
>+#ifdef __SSE4_2__
>+#include <nmmintrin.h>
>+#include "absl/hash/internal/city_crc.h"
>+
>+namespace absl {
>+namespace hash_internal {
>+
>+// Requires len >= 240.
>+static void CityHashCrc256Long(const char *s, size_t len, uint32_t seed,
>+                               uint64_t *result) {
>+  uint64_t a = Fetch64(s + 56) + k0;
>+  uint64_t b = Fetch64(s + 96) + k0;
>+  uint64_t c = result[0] = HashLen16(b, len);
>+  uint64_t d = result[1] = Fetch64(s + 120) * k0 + len;
>+  uint64_t e = Fetch64(s + 184) + seed;
>+  uint64_t f = 0;
>+  uint64_t g = 0;
>+  uint64_t h = c + d;
>+  uint64_t x = seed;
>+  uint64_t y = 0;
>+  uint64_t z = 0;
>+
>+  // 240 bytes of input per iter.
>+  size_t iters = len / 240;
>+  len -= iters * 240;
>+  do {
>+#undef CHUNK
>+#define CHUNK(r)               \
>+  PERMUTE3(x, z, y);           \
>+  b += Fetch64(s);             \
>+  c += Fetch64(s + 8);         \
>+  d += Fetch64(s + 16);        \
>+  e += Fetch64(s + 24);        \
>+  f += Fetch64(s + 32);        \
>+  a += b;                      \
>+  h += f;                      \
>+  b += c;                      \
>+  f += d;                      \
>+  g += e;                      \
>+  e += z;                      \
>+  g += x;                      \
>+  z = _mm_crc32_u64(z, b + g); \
>+  y = _mm_crc32_u64(y, e + h); \
>+  x = _mm_crc32_u64(x, f + a); \
>+  e = Rotate(e, r);            \
>+  c += e;                      \
>+  s += 40
>+
>+    CHUNK(0);
>+    PERMUTE3(a, h, c);
>+    CHUNK(33);
>+    PERMUTE3(a, h, f);
>+    CHUNK(0);
>+    PERMUTE3(b, h, f);
>+    CHUNK(42);
>+    PERMUTE3(b, h, d);
>+    CHUNK(0);
>+    PERMUTE3(b, h, e);
>+    CHUNK(33);
>+    PERMUTE3(a, h, e);
>+  } while (--iters > 0);
>+
>+  while (len >= 40) {
>+    CHUNK(29);
>+    e ^= Rotate(a, 20);
>+    h += Rotate(b, 30);
>+    g ^= Rotate(c, 40);
>+    f += Rotate(d, 34);
>+    PERMUTE3(c, h, g);
>+    len -= 40;
>+  }
>+  if (len > 0) {
>+    s = s + len - 40;
>+    CHUNK(33);
>+    e ^= Rotate(a, 43);
>+    h += Rotate(b, 42);
>+    g ^= Rotate(c, 41);
>+    f += Rotate(d, 40);
>+  }
>+  result[0] ^= h;
>+  result[1] ^= g;
>+  g += h;
>+  a = HashLen16(a, g + z);
>+  x += y << 32;
>+  b += x;
>+  c = HashLen16(c, z) + h;
>+  d = HashLen16(d, e + result[0]);
>+  g += e;
>+  h += HashLen16(x, f);
>+  e = HashLen16(a, d) + g;
>+  z = HashLen16(b, c) + a;
>+  y = HashLen16(g, h) + c;
>+  result[0] = e + z + y + x;
>+  a = ShiftMix((a + y) * k0) * k0 + b;
>+  result[1] += a + result[0];
>+  a = ShiftMix(a * k0) * k0 + c;
>+  result[2] = a + result[1];
>+  a = ShiftMix((a + e) * k0) * k0;
>+  result[3] = a + result[2];
>+}
>+
>+// Requires len < 240.
>+static void CityHashCrc256Short(const char *s, size_t len, uint64_t *result) {
>+  char buf[240];
>+  memcpy(buf, s, len);
>+  memset(buf + len, 0, 240 - len);
>+  CityHashCrc256Long(buf, 240, ~static_cast<uint32_t>(len), result);
>+}
>+
>+void CityHashCrc256(const char *s, size_t len, uint64_t *result) {
>+  if (ABSL_PREDICT_TRUE(len >= 240)) {
>+    CityHashCrc256Long(s, len, 0, result);
>+  } else {
>+    CityHashCrc256Short(s, len, result);
>+  }
>+}
>+
>+uint128 CityHashCrc128WithSeed(const char *s, size_t len, uint128 seed) {
>+  if (len <= 900) {
>+    return CityHash128WithSeed(s, len, seed);
>+  } else {
>+    uint64_t result[4];
>+    CityHashCrc256(s, len, result);
>+    uint64_t u = Uint128High64(seed) + result[0];
>+    uint64_t v = Uint128Low64(seed) + result[1];
>+    return uint128(HashLen16(u, v + result[2]),
>+                   HashLen16(Rotate(v, 32), u * k0 + result[3]));
>+  }
>+}
>+
>+uint128 CityHashCrc128(const char *s, size_t len) {
>+  if (len <= 900) {
>+    return CityHash128(s, len);
>+  } else {
>+    uint64_t result[4];
>+    CityHashCrc256(s, len, result);
>+    return uint128(result[2], result[3]);
>+  }
>+}
>+
>+}  // namespace hash_internal
>+}  // namespace absl
>+
>+#endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..55b37b875f0b3fd8467da99dbed4297ac89f1819
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city.h
>@@ -0,0 +1,108 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// http://code.google.com/p/cityhash/
>+//
>+// This file provides a few functions for hashing strings.  All of them are
>+// high-quality functions in the sense that they pass standard tests such
>+// as Austin Appleby's SMHasher.  They are also fast.
>+//
>+// For 64-bit x86 code, on short strings, we don't know of anything faster than
>+// CityHash64 that is of comparable quality.  We believe our nearest competitor
>+// is Murmur3.  For 64-bit x86 code, CityHash64 is an excellent choice for hash
>+// tables and most other hashing (excluding cryptography).
>+//
>+// For 64-bit x86 code, on long strings, the picture is more complicated.
>+// On many recent Intel CPUs, such as Nehalem, Westmere, Sandy Bridge, etc.,
>+// CityHashCrc128 appears to be faster than all competitors of comparable
>+// quality.  CityHash128 is also good but not quite as fast.  We believe our
>+// nearest competitor is Bob Jenkins' Spooky.  We don't have great data for
>+// other 64-bit CPUs, but for long strings we know that Spooky is slightly
>+// faster than CityHash on some relatively recent AMD x86-64 CPUs, for example.
>+// Note that CityHashCrc128 is declared in citycrc.h.
>+//
>+// For 32-bit x86 code, we don't know of anything faster than CityHash32 that
>+// is of comparable quality.  We believe our nearest competitor is Murmur3A.
>+// (On 64-bit CPUs, it is typically faster to use the other CityHash variants.)
>+//
>+// Functions in the CityHash family are not suitable for cryptography.
>+//
>+// Please see CityHash's README file for more details on our performance
>+// measurements and so on.
>+//
>+// WARNING: This code has been only lightly tested on big-endian platforms!
>+// It is known to work well on little-endian platforms that have a small penalty
>+// for unaligned reads, such as current Intel and AMD moderate-to-high-end CPUs.
>+// It should work on all 32-bit and 64-bit platforms that allow unaligned reads;
>+// bug reports are welcome.
>+//
>+// By the way, for some hash functions, given strings a and b, the hash
>+// of a+b is easily derived from the hashes of a and b.  This property
>+// doesn't hold for any hash functions in this file.
>+
>+#ifndef ABSL_HASH_INTERNAL_CITY_H_
>+#define ABSL_HASH_INTERNAL_CITY_H_
>+
>+#include <stdint.h>
>+#include <stdlib.h>  // for size_t.
>+#include <utility>
>+
>+
>+namespace absl {
>+namespace hash_internal {
>+
>+typedef std::pair<uint64_t, uint64_t> uint128;
>+
>+inline uint64_t Uint128Low64(const uint128 &x) { return x.first; }
>+inline uint64_t Uint128High64(const uint128 &x) { return x.second; }
>+
>+// Hash function for a byte array.
>+uint64_t CityHash64(const char *s, size_t len);
>+
>+// Hash function for a byte array.  For convenience, a 64-bit seed is also
>+// hashed into the result.
>+uint64_t CityHash64WithSeed(const char *s, size_t len, uint64_t seed);
>+
>+// Hash function for a byte array.  For convenience, two seeds are also
>+// hashed into the result.
>+uint64_t CityHash64WithSeeds(const char *s, size_t len, uint64_t seed0,
>+                           uint64_t seed1);
>+
>+// Hash function for a byte array.
>+uint128 CityHash128(const char *s, size_t len);
>+
>+// Hash function for a byte array.  For convenience, a 128-bit seed is also
>+// hashed into the result.
>+uint128 CityHash128WithSeed(const char *s, size_t len, uint128 seed);
>+
>+// Hash function for a byte array.  Most useful in 32-bit binaries.
>+uint32_t CityHash32(const char *s, size_t len);
>+
>+// Hash 128 input bits down to 64 bits of output.
>+// This is intended to be a reasonably good hash function.
>+inline uint64_t Hash128to64(const uint128 &x) {
>+  // Murmur-inspired hashing.
>+  const uint64_t kMul = 0x9ddfea08eb382d69ULL;
>+  uint64_t a = (Uint128Low64(x) ^ Uint128High64(x)) * kMul;
>+  a ^= (a >> 47);
>+  uint64_t b = (Uint128High64(x) ^ a) * kMul;
>+  b ^= (b >> 47);
>+  b *= kMul;
>+  return b;
>+}
>+
>+}  // namespace hash_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HASH_INTERNAL_CITY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city_crc.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city_crc.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..6be6557d213f2b40a8c474c6a0c279f0a6bcaf27
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city_crc.h
>@@ -0,0 +1,41 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This file declares the subset of the CityHash functions that require
>+// _mm_crc32_u64().  See the CityHash README for details.
>+//
>+// Functions in the CityHash family are not suitable for cryptography.
>+
>+#ifndef ABSL_HASH_INTERNAL_CITY_CRC_H_
>+#define ABSL_HASH_INTERNAL_CITY_CRC_H_
>+
>+#include "absl/hash/internal/city.h"
>+
>+namespace absl {
>+namespace hash_internal {
>+
>+// Hash function for a byte array.
>+uint128 CityHashCrc128(const char *s, size_t len);
>+
>+// Hash function for a byte array.  For convenience, a 128-bit seed is also
>+// hashed into the result.
>+uint128 CityHashCrc128WithSeed(const char *s, size_t len, uint128 seed);
>+
>+// Hash function for a byte array.  Sets result[0] ... result[3].
>+void CityHashCrc256(const char *s, size_t len, uint64_t *result);
>+
>+}  // namespace hash_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HASH_INTERNAL_CITY_CRC_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..678da53d34d04d6011fe87b06a7e26613bdbef99
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/city_test.cc
>@@ -0,0 +1,1812 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/hash/internal/city.h"
>+
>+#include <string.h>
>+#include <cstdio>
>+#include <iostream>
>+#include "gtest/gtest.h"
>+#ifdef __SSE4_2__
>+#include "absl/hash/internal/city_crc.h"
>+#endif
>+
>+namespace absl {
>+namespace hash_internal {
>+
>+static const uint64_t k0 = 0xc3a5c85c97cb3127ULL;
>+static const uint64_t kSeed0 = 1234567;
>+static const uint64_t kSeed1 = k0;
>+static const uint128 kSeed128(kSeed0, kSeed1);
>+static const int kDataSize = 1 << 20;
>+static const int kTestSize = 300;
>+
>+static char data[kDataSize];
>+
>+// Initialize data to pseudorandom values.
>+void setup() {
>+  uint64_t a = 9;
>+  uint64_t b = 777;
>+  for (int i = 0; i < kDataSize; i++) {
>+    a += b;
>+    b += a;
>+    a = (a ^ (a >> 41)) * k0;
>+    b = (b ^ (b >> 41)) * k0 + i;
>+    uint8_t u = b >> 37;
>+    memcpy(data + i, &u, 1);  // uint8_t -> char
>+  }
>+}
>+
>+#define C(x) 0x##x##ULL
>+static const uint64_t testdata[kTestSize][16] = {
>+    {C(9ae16a3b2f90404f), C(75106db890237a4a), C(3feac5f636039766),
>+     C(3df09dfc64c09a2b), C(3cb540c392e51e29), C(6b56343feac0663),
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>+     C(bedaa1a4a0250c28), C(9d50546624ff9a57), C(4abaf523d1c090f6), C(b994a88)},
>+    {C(754435bae3496fc), C(5707fc006f094dcf), C(8951c86ab19d8e40),
>+     C(57c5208e8f021a77), C(f7653fbb69cd9276), C(a484410af21d75cb),
>+     C(f19b6844b3d627e8), C(57c5208e8f021a77), C(f7653fbb69cd9276),
>+     C(a484410af21d75cb), C(f19b6844b3d627e8), C(f37400fc3ffd9514),
>+     C(36ae0d821734edfd), C(5f37820af1f1f306), C(be637d40e6a5ad0), C(a05d43c0)},
>+    {C(fda9877ea8e3805f), C(31e868b6ffd521b7), C(b08c90681fb6a0fd),
>+     C(68110a7f83f5d3ff), C(6d77e045901b85a8), C(84ef681113036d8b),
>+     C(3b9f8e3928f56160), C(68110a7f83f5d3ff), C(6d77e045901b85a8),
>+     C(84ef681113036d8b), C(3b9f8e3928f56160), C(fc8b7f56c130835),
>+     C(a11f3e800638e841), C(d9572267f5cf28c1), C(7897c8149803f2aa),
>+     C(c79f73a8)},
>+    {C(2e36f523ca8f5eb5), C(8b22932f89b27513), C(331cd6ecbfadc1bb),
>+     C(d1bfe4df12b04cbf), C(f58c17243fd63842), C(3a453cdba80a60af),
>+     C(5737b2ca7470ea95), C(d1bfe4df12b04cbf), C(f58c17243fd63842),
>+     C(3a453cdba80a60af), C(5737b2ca7470ea95), C(54d44a3f4477030c),
>+     C(8168e02d4869aa7f), C(77f383a17778559d), C(95e1737d77a268fc),
>+     C(a490aff5)},
>+    {C(21a378ef76828208), C(a5c13037fa841da2), C(506d22a53fbe9812),
>+     C(61c9c95d91017da5), C(16f7c83ba68f5279), C(9c0619b0808d05f7),
>+     C(83c117ce4e6b70a3), C(61c9c95d91017da5), C(16f7c83ba68f5279),
>+     C(9c0619b0808d05f7), C(83c117ce4e6b70a3), C(cfb4c8af7fd01413),
>+     C(fdef04e602e72296), C(ed6124d337889b1), C(4919c86707b830da), C(dfad65b4)},
>+    {C(ccdd5600054b16ca), C(f78846e84204cb7b), C(1f9faec82c24eac9),
>+     C(58634004c7b2d19a), C(24bb5f51ed3b9073), C(46409de018033d00),
>+     C(4a9805eed5ac802e), C(58634004c7b2d19a), C(24bb5f51ed3b9073),
>+     C(46409de018033d00), C(4a9805eed5ac802e), C(e18de8db306baf82),
>+     C(46bbf75f1fa025ff), C(5faf2fb09be09487), C(3fbc62bd4e558fb3), C(1d07dfb)},
>+    {C(7854468f4e0cabd0), C(3a3f6b4f098d0692), C(ae2423ec7799d30d),
>+     C(29c3529eb165eeba), C(443de3703b657c35), C(66acbce31ae1bc8d),
>+     C(1acc99effe1d547e), C(29c3529eb165eeba), C(443de3703b657c35),
>+     C(66acbce31ae1bc8d), C(1acc99effe1d547e), C(cf07f8a57906573d),
>+     C(31bafb0bbb9a86e7), C(40c69492702a9346), C(7df61fdaa0b858af),
>+     C(416df9a0)},
>+    {C(7f88db5346d8f997), C(88eac9aacc653798), C(68a4d0295f8eefa1),
>+     C(ae59ca86f4c3323d), C(25906c09906d5c4c), C(8dd2aa0c0a6584ae),
>+     C(232a7d96b38f40e9), C(ae59ca86f4c3323d), C(25906c09906d5c4c),
>+     C(8dd2aa0c0a6584ae), C(232a7d96b38f40e9), C(8986ee00a2ed0042),
>+     C(c49ae7e428c8a7d1), C(b7dd8280713ac9c2), C(e018720aed1ebc28),
>+     C(1f8fb9cc)},
>+    {C(bb3fb5fb01d60fcf), C(1b7cc0847a215eb6), C(1246c994437990a1),
>+     C(d4edc954c07cd8f3), C(224f47e7c00a30ab), C(d5ad7ad7f41ef0c6),
>+     C(59e089281d869fd7), C(d4edc954c07cd8f3), C(224f47e7c00a30ab),
>+     C(d5ad7ad7f41ef0c6), C(59e089281d869fd7), C(f29340d07a14b6f1),
>+     C(c87c5ef76d9c4ef3), C(463118794193a9a), C(2922dcb0540f0dbc), C(7abf48e3)},
>+    {C(2e783e1761acd84d), C(39158042bac975a0), C(1cd21c5a8071188d),
>+     C(b1b7ec44f9302176), C(5cb476450dc0c297), C(dc5ef652521ef6a2),
>+     C(3cc79a9e334e1f84), C(b1b7ec44f9302176), C(5cb476450dc0c297),
>+     C(dc5ef652521ef6a2), C(3cc79a9e334e1f84), C(769e2a283dbcc651),
>+     C(9f24b105c8511d3f), C(c31c15575de2f27e), C(ecfecf32c3ae2d66),
>+     C(dea4e3dd)},
>+    {C(392058251cf22acc), C(944ec4475ead4620), C(b330a10b5cb94166),
>+     C(54bc9bee7cbe1767), C(485820bdbe442431), C(54d6120ea2972e90),
>+     C(f437a0341f29b72a), C(54bc9bee7cbe1767), C(485820bdbe442431),
>+     C(54d6120ea2972e90), C(f437a0341f29b72a), C(8f30885c784d5704),
>+     C(aa95376b16c7906a), C(e826928cfaf93dc3), C(20e8f54d1c16d7d8),
>+     C(c6064f22)},
>+    {C(adf5c1e5d6419947), C(2a9747bc659d28aa), C(95c5b8cb1f5d62c),
>+     C(80973ea532b0f310), C(a471829aa9c17dd9), C(c2ff3479394804ab),
>+     C(6bf44f8606753636), C(80973ea532b0f310), C(a471829aa9c17dd9),
>+     C(c2ff3479394804ab), C(6bf44f8606753636), C(5184d2973e6dd827),
>+     C(121b96369a332d9a), C(5c25d3475ab69e50), C(26d2961d62884168),
>+     C(743bed9c)},
>+    {C(6bc1db2c2bee5aba), C(e63b0ed635307398), C(7b2eca111f30dbbc),
>+     C(230d2b3e47f09830), C(ec8624a821c1caf4), C(ea6ec411cdbf1cb1),
>+     C(5f38ae82af364e27), C(230d2b3e47f09830), C(ec8624a821c1caf4),
>+     C(ea6ec411cdbf1cb1), C(5f38ae82af364e27), C(a519ef515ea7187c),
>+     C(6bad5efa7ebae05f), C(748abacb11a74a63), C(a28eef963d1396eb),
>+     C(fce254d5)},
>+    {C(b00f898229efa508), C(83b7590ad7f6985c), C(2780e70a0592e41d),
>+     C(7122413bdbc94035), C(e7f90fae33bf7763), C(4b6bd0fb30b12387),
>+     C(557359c0c44f48ca), C(7122413bdbc94035), C(e7f90fae33bf7763),
>+     C(4b6bd0fb30b12387), C(557359c0c44f48ca), C(d5656c3d6bc5f0d),
>+     C(983ff8e5e784da99), C(628479671b445bf), C(e179a1e27ce68f5d), C(e47ec9d1)},
>+    {C(b56eb769ce0d9a8c), C(ce196117bfbcaf04), C(b26c3c3797d66165),
>+     C(5ed12338f630ab76), C(fab19fcb319116d), C(167f5f42b521724b),
>+     C(c4aa56c409568d74), C(5ed12338f630ab76), C(fab19fcb319116d),
>+     C(167f5f42b521724b), C(c4aa56c409568d74), C(75fff4b42f8e9778),
>+     C(94218f94710c1ea3), C(b7b05efb738b06a6), C(83fff2deabf9cd3), C(334a145c)},
>+    {C(70c0637675b94150), C(259e1669305b0a15), C(46e1dd9fd387a58d),
>+     C(fca4e5bc9292788e), C(cd509dc1facce41c), C(bbba575a59d82fe),
>+     C(4e2e71c15b45d4d3), C(fca4e5bc9292788e), C(cd509dc1facce41c),
>+     C(bbba575a59d82fe), C(4e2e71c15b45d4d3), C(5dc54582ead999c),
>+     C(72612d1571963c6f), C(30318a9d2d3d1829), C(785dd00f4cc9c9a0),
>+     C(adec1e3c)},
>+    {C(74c0b8a6821faafe), C(abac39d7491370e7), C(faf0b2a48a4e6aed),
>+     C(967e970df9673d2a), C(d465247cffa415c0), C(33a1df0ca1107722),
>+     C(49fc2a10adce4a32), C(967e970df9673d2a), C(d465247cffa415c0),
>+     C(33a1df0ca1107722), C(49fc2a10adce4a32), C(c5707e079a284308),
>+     C(573028266635dda6), C(f786f5eee6127fa0), C(b30d79cebfb51266),
>+     C(f6a9fbf8)},
>+    {C(5fb5e48ac7b7fa4f), C(a96170f08f5acbc7), C(bbf5c63d4f52a1e5),
>+     C(6cc09e60700563e9), C(d18f23221e964791), C(ffc23eeef7af26eb),
>+     C(693a954a3622a315), C(815308a32a9b0daf), C(efb2ab27bf6fd0bd),
>+     C(9f1ffc0986111118), C(f9a3aa1778ea3985), C(698fe54b2b93933b),
>+     C(dacc2b28404d0f10), C(815308a32a9b0daf), C(efb2ab27bf6fd0bd),
>+     C(5398210c)},
>+};
>+
>+void TestUnchanging(const uint64_t* expected, int offset, int len) {
>+  const uint128 u = CityHash128(data + offset, len);
>+  const uint128 v = CityHash128WithSeed(data + offset, len, kSeed128);
>+  EXPECT_EQ(expected[0], CityHash64(data + offset, len));
>+  EXPECT_EQ(expected[15], CityHash32(data + offset, len));
>+  EXPECT_EQ(expected[1], CityHash64WithSeed(data + offset, len, kSeed0));
>+  EXPECT_EQ(expected[2],
>+            CityHash64WithSeeds(data + offset, len, kSeed0, kSeed1));
>+  EXPECT_EQ(expected[3], Uint128Low64(u));
>+  EXPECT_EQ(expected[4], Uint128High64(u));
>+  EXPECT_EQ(expected[5], Uint128Low64(v));
>+  EXPECT_EQ(expected[6], Uint128High64(v));
>+#ifdef __SSE4_2__
>+  const uint128 y = CityHashCrc128(data + offset, len);
>+  const uint128 z = CityHashCrc128WithSeed(data + offset, len, kSeed128);
>+  uint64_t crc256_results[4];
>+  CityHashCrc256(data + offset, len, crc256_results);
>+  EXPECT_EQ(expected[7], Uint128Low64(y));
>+  EXPECT_EQ(expected[8], Uint128High64(y));
>+  EXPECT_EQ(expected[9], Uint128Low64(z));
>+  EXPECT_EQ(expected[10], Uint128High64(z));
>+  for (int i = 0; i < 4; i++) {
>+    EXPECT_EQ(expected[11 + i], crc256_results[i]);
>+  }
>+#endif
>+}
>+
>+TEST(CityHashTest, Unchanging) {
>+  setup();
>+  int i = 0;
>+  for (; i < kTestSize - 1; i++) {
>+    TestUnchanging(testdata[i], i * i, i);
>+  }
>+  TestUnchanging(testdata[i], 0, kDataSize);
>+}
>+
>+}  // namespace hash_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/hash.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/hash.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..4bf64096e7eaa3b346ed2f2a22df2cd2a8e71574
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/hash.cc
>@@ -0,0 +1,23 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/hash/internal/hash.h"
>+
>+namespace absl {
>+namespace hash_internal {
>+
>+ABSL_CONST_INIT const void* const CityHashState::kSeed = &kSeed;
>+
>+}  // namespace hash_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/hash.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/hash.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..78217d2fe760003cb83e8a317a59185e82a730f7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/hash.h
>@@ -0,0 +1,887 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: hash.h
>+// -----------------------------------------------------------------------------
>+//
>+#ifndef ABSL_HASH_INTERNAL_HASH_H_
>+#define ABSL_HASH_INTERNAL_HASH_H_
>+
>+#include <algorithm>
>+#include <array>
>+#include <cmath>
>+#include <cstring>
>+#include <deque>
>+#include <forward_list>
>+#include <functional>
>+#include <iterator>
>+#include <limits>
>+#include <list>
>+#include <map>
>+#include <memory>
>+#include <set>
>+#include <string>
>+#include <tuple>
>+#include <type_traits>
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/base/internal/endian.h"
>+#include "absl/base/port.h"
>+#include "absl/container/fixed_array.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/numeric/int128.h"
>+#include "absl/strings/string_view.h"
>+#include "absl/types/optional.h"
>+#include "absl/types/variant.h"
>+#include "absl/utility/utility.h"
>+#include "absl/hash/internal/city.h"
>+
>+namespace absl {
>+namespace hash_internal {
>+
>+// HashStateBase
>+//
>+// A hash state object represents an intermediate state in the computation
>+// of an unspecified hash algorithm. `HashStateBase` provides a CRTP style
>+// base class for hash state implementations. Developers adding type support
>+// for `absl::Hash` should not rely on any parts of the state object other than
>+// the following member functions:
>+//
>+//   * HashStateBase::combine()
>+//   * HashStateBase::combine_contiguous()
>+//
>+// A derived hash state class of type `H` must provide a static member function
>+// with a signature similar to the following:
>+//
>+//    `static H combine_contiguous(H state, const unsigned char*, size_t)`.
>+//
>+// `HashStateBase` will provide a complete implementations for a hash state
>+// object in terms of this method.
>+//
>+// Example:
>+//
>+//   // Use CRTP to define your derived class.
>+//   struct MyHashState : HashStateBase<MyHashState> {
>+//       static H combine_contiguous(H state, const unsigned char*, size_t);
>+//       using MyHashState::HashStateBase::combine;
>+//       using MyHashState::HashStateBase::combine_contiguous;
>+//   };
>+template <typename H>
>+class HashStateBase {
>+ public:
>+  // HashStateBase::combine()
>+  //
>+  // Combines an arbitrary number of values into a hash state, returning the
>+  // updated state.
>+  //
>+  // Each of the value types `T` must be separately hashable by the Abseil
>+  // hashing framework.
>+  //
>+  // NOTE:
>+  //
>+  //   state = H::combine(std::move(state), value1, value2, value3);
>+  //
>+  // is guaranteed to produce the same hash expansion as:
>+  //
>+  //   state = H::combine(std::move(state), value1);
>+  //   state = H::combine(std::move(state), value2);
>+  //   state = H::combine(std::move(state), value3);
>+  template <typename T, typename... Ts>
>+  static H combine(H state, const T& value, const Ts&... values);
>+  static H combine(H state) { return state; }
>+
>+  // HashStateBase::combine_contiguous()
>+  //
>+  // Combines a contiguous array of `size` elements into a hash state, returning
>+  // the updated state.
>+  //
>+  // NOTE:
>+  //
>+  //   state = H::combine_contiguous(std::move(state), data, size);
>+  //
>+  // is NOT guaranteed to produce the same hash expansion as a for-loop (it may
>+  // perform internal optimizations).  If you need this guarantee, use the
>+  // for-loop instead.
>+  template <typename T>
>+  static H combine_contiguous(H state, const T* data, size_t size);
>+};
>+
>+// is_uniquely_represented
>+//
>+// `is_uniquely_represented<T>` is a trait class that indicates whether `T`
>+// is uniquely represented.
>+//
>+// A type is "uniquely represented" if two equal values of that type are
>+// guaranteed to have the same bytes in their underlying storage. In other
>+// words, if `a == b`, then `memcmp(&a, &b, sizeof(T))` is guaranteed to be
>+// zero. This property cannot be detected automatically, so this trait is false
>+// by default, but can be specialized by types that wish to assert that they are
>+// uniquely represented. This makes them eligible for certain optimizations.
>+//
>+// If you have any doubt whatsoever, do not specialize this template.
>+// The default is completely safe, and merely disables some optimizations
>+// that will not matter for most types. Specializing this template,
>+// on the other hand, can be very hazardous.
>+//
>+// To be uniquely represented, a type must not have multiple ways of
>+// representing the same value; for example, float and double are not
>+// uniquely represented, because they have distinct representations for
>+// +0 and -0. Furthermore, the type's byte representation must consist
>+// solely of user-controlled data, with no padding bits and no compiler-
>+// controlled data such as vptrs or sanitizer metadata. This is usually
>+// very difficult to guarantee, because in most cases the compiler can
>+// insert data and padding bits at its own discretion.
>+//
>+// If you specialize this template for a type `T`, you must do so in the file
>+// that defines that type (or in this file). If you define that specialization
>+// anywhere else, `is_uniquely_represented<T>` could have different meanings
>+// in different places.
>+//
>+// The Enable parameter is meaningless; it is provided as a convenience,
>+// to support certain SFINAE techniques when defining specializations.
>+template <typename T, typename Enable = void>
>+struct is_uniquely_represented : std::false_type {};
>+
>+// is_uniquely_represented<unsigned char>
>+//
>+// unsigned char is a synonym for "byte", so it is guaranteed to be
>+// uniquely represented.
>+template <>
>+struct is_uniquely_represented<unsigned char> : std::true_type {};
>+
>+// is_uniquely_represented for non-standard integral types
>+//
>+// Integral types other than bool should be uniquely represented on any
>+// platform that this will plausibly be ported to.
>+template <typename Integral>
>+struct is_uniquely_represented<
>+    Integral, typename std::enable_if<std::is_integral<Integral>::value>::type>
>+    : std::true_type {};
>+
>+// is_uniquely_represented<bool>
>+//
>+//
>+template <>
>+struct is_uniquely_represented<bool> : std::false_type {};
>+
>+// hash_bytes()
>+//
>+// Convenience function that combines `hash_state` with the byte representation
>+// of `value`.
>+template <typename H, typename T>
>+H hash_bytes(H hash_state, const T& value) {
>+  const unsigned char* start = reinterpret_cast<const unsigned char*>(&value);
>+  return H::combine_contiguous(std::move(hash_state), start, sizeof(value));
>+}
>+
>+// -----------------------------------------------------------------------------
>+// AbslHashValue for Basic Types
>+// -----------------------------------------------------------------------------
>+
>+// Note: Default `AbslHashValue` implementations live in `hash_internal`. This
>+// allows us to block lexical scope lookup when doing an unqualified call to
>+// `AbslHashValue` below. User-defined implementations of `AbslHashValue` can
>+// only be found via ADL.
>+
>+// AbslHashValue() for hashing bool values
>+//
>+// We use SFINAE to ensure that this overload only accepts bool, not types that
>+// are convertible to bool.
>+template <typename H, typename B>
>+typename std::enable_if<std::is_same<B, bool>::value, H>::type AbslHashValue(
>+    H hash_state, B value) {
>+  return H::combine(std::move(hash_state),
>+                    static_cast<unsigned char>(value ? 1 : 0));
>+}
>+
>+// AbslHashValue() for hashing enum values
>+template <typename H, typename Enum>
>+typename std::enable_if<std::is_enum<Enum>::value, H>::type AbslHashValue(
>+    H hash_state, Enum e) {
>+  // In practice, we could almost certainly just invoke hash_bytes directly,
>+  // but it's possible that a sanitizer might one day want to
>+  // store data in the unused bits of an enum. To avoid that risk, we
>+  // convert to the underlying type before hashing. Hopefully this will get
>+  // optimized away; if not, we can reopen discussion with c-toolchain-team.
>+  return H::combine(std::move(hash_state),
>+                    static_cast<typename std::underlying_type<Enum>::type>(e));
>+}
>+// AbslHashValue() for hashing floating-point values
>+template <typename H, typename Float>
>+typename std::enable_if<std::is_floating_point<Float>::value, H>::type
>+AbslHashValue(H hash_state, Float value) {
>+  return hash_internal::hash_bytes(std::move(hash_state),
>+                                   value == 0 ? 0 : value);
>+}
>+
>+// Long double has the property that it might have extra unused bytes in it.
>+// For example, in x86 sizeof(long double)==16 but it only really uses 80-bits
>+// of it. This means we can't use hash_bytes on a long double and have to
>+// convert it to something else first.
>+template <typename H>
>+H AbslHashValue(H hash_state, long double value) {
>+  const int category = std::fpclassify(value);
>+  switch (category) {
>+    case FP_INFINITE:
>+      // Add the sign bit to differentiate between +Inf and -Inf
>+      hash_state = H::combine(std::move(hash_state), std::signbit(value));
>+      break;
>+
>+    case FP_NAN:
>+    case FP_ZERO:
>+    default:
>+      // Category is enough for these.
>+      break;
>+
>+    case FP_NORMAL:
>+    case FP_SUBNORMAL:
>+      // We can't convert `value` directly to double because this would have
>+      // undefined behavior if the value is out of range.
>+      // std::frexp gives us a value in the range (-1, -.5] or [.5, 1) that is
>+      // guaranteed to be in range for `double`. The truncation is
>+      // implementation defined, but that works as long as it is deterministic.
>+      int exp;
>+      auto mantissa = static_cast<double>(std::frexp(value, &exp));
>+      hash_state = H::combine(std::move(hash_state), mantissa, exp);
>+  }
>+
>+  return H::combine(std::move(hash_state), category);
>+}
>+
>+// AbslHashValue() for hashing pointers
>+template <typename H, typename T>
>+H AbslHashValue(H hash_state, T* ptr) {
>+  return hash_internal::hash_bytes(std::move(hash_state), ptr);
>+}
>+
>+// AbslHashValue() for hashing nullptr_t
>+template <typename H>
>+H AbslHashValue(H hash_state, std::nullptr_t) {
>+  return H::combine(std::move(hash_state), static_cast<void*>(nullptr));
>+}
>+
>+// -----------------------------------------------------------------------------
>+// AbslHashValue for Composite Types
>+// -----------------------------------------------------------------------------
>+
>+// is_hashable()
>+//
>+// Trait class which returns true if T is hashable by the absl::Hash framework.
>+// Used for the AbslHashValue implementations for composite types below.
>+template <typename T>
>+struct is_hashable;
>+
>+// AbslHashValue() for hashing pairs
>+template <typename H, typename T1, typename T2>
>+typename std::enable_if<is_hashable<T1>::value && is_hashable<T2>::value,
>+                        H>::type
>+AbslHashValue(H hash_state, const std::pair<T1, T2>& p) {
>+  return H::combine(std::move(hash_state), p.first, p.second);
>+}
>+
>+// hash_tuple()
>+//
>+// Helper function for hashing a tuple. The third argument should
>+// be an index_sequence running from 0 to tuple_size<Tuple> - 1.
>+template <typename H, typename Tuple, size_t... Is>
>+H hash_tuple(H hash_state, const Tuple& t, absl::index_sequence<Is...>) {
>+  return H::combine(std::move(hash_state), std::get<Is>(t)...);
>+}
>+
>+// AbslHashValue for hashing tuples
>+template <typename H, typename... Ts>
>+#if defined(_MSC_VER)
>+// This SFINAE gets MSVC confused under some conditions. Let's just disable it
>+// for now.
>+H
>+#else  // _MSC_VER
>+typename std::enable_if<absl::conjunction<is_hashable<Ts>...>::value, H>::type
>+#endif  // _MSC_VER
>+AbslHashValue(H hash_state, const std::tuple<Ts...>& t) {
>+  return hash_internal::hash_tuple(std::move(hash_state), t,
>+                                   absl::make_index_sequence<sizeof...(Ts)>());
>+}
>+
>+// -----------------------------------------------------------------------------
>+// AbslHashValue for Pointers
>+// -----------------------------------------------------------------------------
>+
>+// AbslHashValue for hashing unique_ptr
>+template <typename H, typename T, typename D>
>+H AbslHashValue(H hash_state, const std::unique_ptr<T, D>& ptr) {
>+  return H::combine(std::move(hash_state), ptr.get());
>+}
>+
>+// AbslHashValue for hashing shared_ptr
>+template <typename H, typename T>
>+H AbslHashValue(H hash_state, const std::shared_ptr<T>& ptr) {
>+  return H::combine(std::move(hash_state), ptr.get());
>+}
>+
>+// -----------------------------------------------------------------------------
>+// AbslHashValue for String-Like Types
>+// -----------------------------------------------------------------------------
>+
>+// AbslHashValue for hashing strings
>+//
>+// All the string-like types supported here provide the same hash expansion for
>+// the same character sequence. These types are:
>+//
>+//  - `std::string` (and std::basic_string<char, std::char_traits<char>, A> for
>+//      any allocator A)
>+//  - `absl::string_view` and `std::string_view`
>+//
>+// For simplicity, we currently support only `char` strings. This support may
>+// be broadened, if necessary, but with some caution - this overload would
>+// misbehave in cases where the traits' `eq()` member isn't equivalent to `==`
>+// on the underlying character type.
>+template <typename H>
>+H AbslHashValue(H hash_state, absl::string_view str) {
>+  return H::combine(
>+      H::combine_contiguous(std::move(hash_state), str.data(), str.size()),
>+      str.size());
>+}
>+
>+// -----------------------------------------------------------------------------
>+// AbslHashValue for Sequence Containers
>+// -----------------------------------------------------------------------------
>+
>+// AbslHashValue for hashing std::array
>+template <typename H, typename T, size_t N>
>+typename std::enable_if<is_hashable<T>::value, H>::type AbslHashValue(
>+    H hash_state, const std::array<T, N>& array) {
>+  return H::combine_contiguous(std::move(hash_state), array.data(),
>+                               array.size());
>+}
>+
>+// AbslHashValue for hashing std::deque
>+template <typename H, typename T, typename Allocator>
>+typename std::enable_if<is_hashable<T>::value, H>::type AbslHashValue(
>+    H hash_state, const std::deque<T, Allocator>& deque) {
>+  // TODO(gromer): investigate a more efficient implementation taking
>+  // advantage of the chunk structure.
>+  for (const auto& t : deque) {
>+    hash_state = H::combine(std::move(hash_state), t);
>+  }
>+  return H::combine(std::move(hash_state), deque.size());
>+}
>+
>+// AbslHashValue for hashing std::forward_list
>+template <typename H, typename T, typename Allocator>
>+typename std::enable_if<is_hashable<T>::value, H>::type AbslHashValue(
>+    H hash_state, const std::forward_list<T, Allocator>& list) {
>+  size_t size = 0;
>+  for (const T& t : list) {
>+    hash_state = H::combine(std::move(hash_state), t);
>+    ++size;
>+  }
>+  return H::combine(std::move(hash_state), size);
>+}
>+
>+// AbslHashValue for hashing std::list
>+template <typename H, typename T, typename Allocator>
>+typename std::enable_if<is_hashable<T>::value, H>::type AbslHashValue(
>+    H hash_state, const std::list<T, Allocator>& list) {
>+  for (const auto& t : list) {
>+    hash_state = H::combine(std::move(hash_state), t);
>+  }
>+  return H::combine(std::move(hash_state), list.size());
>+}
>+
>+// AbslHashValue for hashing std::vector
>+//
>+// Do not use this for vector<bool>. It does not have a .data(), and a fallback
>+// for std::hash<> is most likely faster.
>+template <typename H, typename T, typename Allocator>
>+typename std::enable_if<is_hashable<T>::value && !std::is_same<T, bool>::value,
>+                        H>::type
>+AbslHashValue(H hash_state, const std::vector<T, Allocator>& vector) {
>+  return H::combine(H::combine_contiguous(std::move(hash_state), vector.data(),
>+                                          vector.size()),
>+                    vector.size());
>+}
>+
>+// -----------------------------------------------------------------------------
>+// AbslHashValue for Ordered Associative Containers
>+// -----------------------------------------------------------------------------
>+
>+// AbslHashValue for hashing std::map
>+template <typename H, typename Key, typename T, typename Compare,
>+          typename Allocator>
>+typename std::enable_if<is_hashable<Key>::value && is_hashable<T>::value,
>+                        H>::type
>+AbslHashValue(H hash_state, const std::map<Key, T, Compare, Allocator>& map) {
>+  for (const auto& t : map) {
>+    hash_state = H::combine(std::move(hash_state), t);
>+  }
>+  return H::combine(std::move(hash_state), map.size());
>+}
>+
>+// AbslHashValue for hashing std::multimap
>+template <typename H, typename Key, typename T, typename Compare,
>+          typename Allocator>
>+typename std::enable_if<is_hashable<Key>::value && is_hashable<T>::value,
>+                        H>::type
>+AbslHashValue(H hash_state,
>+              const std::multimap<Key, T, Compare, Allocator>& map) {
>+  for (const auto& t : map) {
>+    hash_state = H::combine(std::move(hash_state), t);
>+  }
>+  return H::combine(std::move(hash_state), map.size());
>+}
>+
>+// AbslHashValue for hashing std::set
>+template <typename H, typename Key, typename Compare, typename Allocator>
>+typename std::enable_if<is_hashable<Key>::value, H>::type AbslHashValue(
>+    H hash_state, const std::set<Key, Compare, Allocator>& set) {
>+  for (const auto& t : set) {
>+    hash_state = H::combine(std::move(hash_state), t);
>+  }
>+  return H::combine(std::move(hash_state), set.size());
>+}
>+
>+// AbslHashValue for hashing std::multiset
>+template <typename H, typename Key, typename Compare, typename Allocator>
>+typename std::enable_if<is_hashable<Key>::value, H>::type AbslHashValue(
>+    H hash_state, const std::multiset<Key, Compare, Allocator>& set) {
>+  for (const auto& t : set) {
>+    hash_state = H::combine(std::move(hash_state), t);
>+  }
>+  return H::combine(std::move(hash_state), set.size());
>+}
>+
>+// -----------------------------------------------------------------------------
>+// AbslHashValue for Wrapper Types
>+// -----------------------------------------------------------------------------
>+
>+// AbslHashValue for hashing absl::optional
>+template <typename H, typename T>
>+typename std::enable_if<is_hashable<T>::value, H>::type AbslHashValue(
>+    H hash_state, const absl::optional<T>& opt) {
>+  if (opt) hash_state = H::combine(std::move(hash_state), *opt);
>+  return H::combine(std::move(hash_state), opt.has_value());
>+}
>+
>+// VariantVisitor
>+template <typename H>
>+struct VariantVisitor {
>+  H&& hash_state;
>+  template <typename T>
>+  H operator()(const T& t) const {
>+    return H::combine(std::move(hash_state), t);
>+  }
>+};
>+
>+// AbslHashValue for hashing absl::variant
>+template <typename H, typename... T>
>+typename std::enable_if<conjunction<is_hashable<T>...>::value, H>::type
>+AbslHashValue(H hash_state, const absl::variant<T...>& v) {
>+  if (!v.valueless_by_exception()) {
>+    hash_state = absl::visit(VariantVisitor<H>{std::move(hash_state)}, v);
>+  }
>+  return H::combine(std::move(hash_state), v.index());
>+}
>+// -----------------------------------------------------------------------------
>+
>+// hash_range_or_bytes()
>+//
>+// Mixes all values in the range [data, data+size) into the hash state.
>+// This overload accepts only uniquely-represented types, and hashes them by
>+// hashing the entire range of bytes.
>+template <typename H, typename T>
>+typename std::enable_if<is_uniquely_represented<T>::value, H>::type
>+hash_range_or_bytes(H hash_state, const T* data, size_t size) {
>+  const auto* bytes = reinterpret_cast<const unsigned char*>(data);
>+  return H::combine_contiguous(std::move(hash_state), bytes, sizeof(T) * size);
>+}
>+
>+// hash_range_or_bytes()
>+template <typename H, typename T>
>+typename std::enable_if<!is_uniquely_represented<T>::value, H>::type
>+hash_range_or_bytes(H hash_state, const T* data, size_t size) {
>+  for (const auto end = data + size; data < end; ++data) {
>+    hash_state = H::combine(std::move(hash_state), *data);
>+  }
>+  return hash_state;
>+}
>+
>+// InvokeHashTag
>+//
>+// InvokeHash(H, const T&) invokes the appropriate hash implementation for a
>+// hasher of type `H` and a value of type `T`. If `T` is not hashable, there
>+// will be no matching overload of InvokeHash().
>+// Note: Some platforms (eg MSVC) do not support the detect idiom on
>+// std::hash. In those platforms the last fallback will be std::hash and
>+// InvokeHash() will always have a valid overload even if std::hash<T> is not
>+// valid.
>+//
>+// We try the following options in order:
>+//   * If is_uniquely_represented, hash bytes directly.
>+//   * ADL AbslHashValue(H, const T&) call.
>+//   * std::hash<T>
>+
>+// In MSVC we can't probe std::hash or stdext::hash because it triggers a
>+// static_assert instead of failing substitution.
>+#if defined(_MSC_VER)
>+#define ABSL_HASH_INTERNAL_CAN_POISON_ 0
>+#else   // _MSC_VER
>+#define ABSL_HASH_INTERNAL_CAN_POISON_ 1
>+#endif  // _MSC_VER
>+
>+#if defined(ABSL_INTERNAL_LEGACY_HASH_NAMESPACE) && \
>+    ABSL_HASH_INTERNAL_CAN_POISON_
>+#define ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_ 1
>+#else
>+#define ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_ 0
>+#endif
>+
>+enum class InvokeHashTag {
>+  kUniquelyRepresented,
>+  kHashValue,
>+#if ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_
>+  kLegacyHash,
>+#endif  // ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_
>+  kStdHash,
>+  kNone
>+};
>+
>+// HashSelect
>+//
>+// Type trait to select the appropriate hash implementation to use.
>+// HashSelect<T>::value is an instance of InvokeHashTag that indicates the best
>+// available hashing mechanism.
>+// See `Note` above about MSVC.
>+template <typename T>
>+struct HashSelect {
>+ private:
>+  struct State : HashStateBase<State> {
>+    static State combine_contiguous(State hash_state, const unsigned char*,
>+                                    size_t);
>+    using State::HashStateBase::combine_contiguous;
>+  };
>+
>+  // `Probe<V, Tag>::value` evaluates to `V<T>::value` if it is a valid
>+  // expression, and `false` otherwise.
>+  // `Probe<V, Tag>::tag` always evaluates to `Tag`.
>+  template <template <typename> class V, InvokeHashTag Tag>
>+  struct Probe {
>+   private:
>+    template <typename U, typename std::enable_if<V<U>::value, int>::type = 0>
>+    static std::true_type Test(int);
>+    template <typename U>
>+    static std::false_type Test(char);
>+
>+   public:
>+    static constexpr InvokeHashTag kTag = Tag;
>+    static constexpr bool value = decltype(
>+        Test<absl::remove_const_t<absl::remove_reference_t<T>>>(0))::value;
>+  };
>+
>+  template <typename U>
>+  using ProbeUniquelyRepresented = is_uniquely_represented<U>;
>+
>+  template <typename U>
>+  using ProbeHashValue =
>+      std::is_same<State, decltype(AbslHashValue(std::declval<State>(),
>+                                                 std::declval<const U&>()))>;
>+
>+#if ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_
>+  template <typename U>
>+  using ProbeLegacyHash =
>+      std::is_convertible<decltype(ABSL_INTERNAL_LEGACY_HASH_NAMESPACE::hash<
>+                                   U>()(std::declval<const U&>())),
>+                          size_t>;
>+#endif  // ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_
>+
>+  template <typename U>
>+  using ProbeStdHash =
>+#if ABSL_HASH_INTERNAL_CAN_POISON_
>+      std::is_convertible<decltype(std::hash<U>()(std::declval<const U&>())),
>+                          size_t>;
>+#else   // ABSL_HASH_INTERNAL_CAN_POISON_
>+      std::true_type;
>+#endif  // ABSL_HASH_INTERNAL_CAN_POISON_
>+
>+  template <typename U>
>+  using ProbeNone = std::true_type;
>+
>+ public:
>+  // Probe each implementation in order.
>+  // disjunction provides short circuting wrt instantiation.
>+  static constexpr InvokeHashTag value = absl::disjunction<
>+      Probe<ProbeUniquelyRepresented, InvokeHashTag::kUniquelyRepresented>,
>+      Probe<ProbeHashValue, InvokeHashTag::kHashValue>,
>+#if ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_
>+      Probe<ProbeLegacyHash, InvokeHashTag::kLegacyHash>,
>+#endif  // ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_
>+      Probe<ProbeStdHash, InvokeHashTag::kStdHash>,
>+      Probe<ProbeNone, InvokeHashTag::kNone>>::kTag;
>+};
>+
>+template <typename T>
>+struct is_hashable : std::integral_constant<bool, HashSelect<T>::value !=
>+                                                      InvokeHashTag::kNone> {};
>+
>+template <typename H, typename T>
>+absl::enable_if_t<HashSelect<T>::value == InvokeHashTag::kUniquelyRepresented,
>+                  H>
>+InvokeHash(H state, const T& value) {
>+  return hash_internal::hash_bytes(std::move(state), value);
>+}
>+
>+template <typename H, typename T>
>+absl::enable_if_t<HashSelect<T>::value == InvokeHashTag::kHashValue, H>
>+InvokeHash(H state, const T& value) {
>+  return AbslHashValue(std::move(state), value);
>+}
>+
>+#if ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_
>+template <typename H, typename T>
>+absl::enable_if_t<HashSelect<T>::value == InvokeHashTag::kLegacyHash, H>
>+InvokeHash(H state, const T& value) {
>+  return hash_internal::hash_bytes(
>+      std::move(state), ABSL_INTERNAL_LEGACY_HASH_NAMESPACE::hash<T>{}(value));
>+}
>+#endif  // ABSL_HASH_INTERNAL_SUPPORT_LEGACY_HASH_
>+
>+template <typename H, typename T>
>+absl::enable_if_t<HashSelect<T>::value == InvokeHashTag::kStdHash, H>
>+InvokeHash(H state, const T& value) {
>+  return hash_internal::hash_bytes(std::move(state), std::hash<T>{}(value));
>+}
>+
>+// CityHashState
>+class CityHashState : public HashStateBase<CityHashState> {
>+  // absl::uint128 is not an alias or a thin wrapper around the intrinsic.
>+  // We use the intrinsic when available to improve performance.
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+  using uint128 = __uint128_t;
>+#else   // ABSL_HAVE_INTRINSIC_INT128
>+  using uint128 = absl::uint128;
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+
>+  static constexpr uint64_t kMul =
>+      sizeof(size_t) == 4 ? uint64_t{0xcc9e2d51} : uint64_t{0x9ddfea08eb382d69};
>+
>+  template <typename T>
>+  using IntegralFastPath =
>+      conjunction<std::is_integral<T>, is_uniquely_represented<T>>;
>+
>+ public:
>+  // Move only
>+  CityHashState(CityHashState&&) = default;
>+  CityHashState& operator=(CityHashState&&) = default;
>+
>+  // CityHashState::combine_contiguous()
>+  //
>+  // Fundamental base case for hash recursion: mixes the given range of bytes
>+  // into the hash state.
>+  static CityHashState combine_contiguous(CityHashState hash_state,
>+                                          const unsigned char* first,
>+                                          size_t size) {
>+    return CityHashState(
>+        CombineContiguousImpl(hash_state.state_, first, size,
>+                              std::integral_constant<int, sizeof(size_t)>{}));
>+  }
>+  using CityHashState::HashStateBase::combine_contiguous;
>+
>+  // CityHashState::hash()
>+  //
>+  // For performance reasons in non-opt mode, we specialize this for
>+  // integral types.
>+  // Otherwise we would be instantiating and calling dozens of functions for
>+  // something that is just one multiplication and a couple xor's.
>+  // The result should be the same as running the whole algorithm, but faster.
>+  template <typename T, absl::enable_if_t<IntegralFastPath<T>::value, int> = 0>
>+  static size_t hash(T value) {
>+    return static_cast<size_t>(Mix(Seed(), static_cast<uint64_t>(value)));
>+  }
>+
>+  // Overload of CityHashState::hash()
>+  template <typename T, absl::enable_if_t<!IntegralFastPath<T>::value, int> = 0>
>+  static size_t hash(const T& value) {
>+    return static_cast<size_t>(combine(CityHashState{}, value).state_);
>+  }
>+
>+ private:
>+  // Invoked only once for a given argument; that plus the fact that this is
>+  // move-only ensures that there is only one non-moved-from object.
>+  CityHashState() : state_(Seed()) {}
>+
>+  // Workaround for MSVC bug.
>+  // We make the type copyable to fix the calling convention, even though we
>+  // never actually copy it. Keep it private to not affect the public API of the
>+  // type.
>+  CityHashState(const CityHashState&) = default;
>+
>+  explicit CityHashState(uint64_t state) : state_(state) {}
>+
>+  // Implementation of the base case for combine_contiguous where we actually
>+  // mix the bytes into the state.
>+  // Dispatch to different implementations of the combine_contiguous depending
>+  // on the value of `sizeof(size_t)`.
>+  static uint64_t CombineContiguousImpl(uint64_t state,
>+                                        const unsigned char* first, size_t len,
>+                                        std::integral_constant<int, 4>
>+                                        /* sizeof_size_t */);
>+  static uint64_t CombineContiguousImpl(uint64_t state,
>+                                        const unsigned char* first, size_t len,
>+                                        std::integral_constant<int, 8>
>+                                        /* sizeof_size_t*/);
>+
>+  // Reads 9 to 16 bytes from p.
>+  // The first 8 bytes are in .first, the rest (zero padded) bytes are in
>+  // .second.
>+  static std::pair<uint64_t, uint64_t> Read9To16(const unsigned char* p,
>+                                                 size_t len) {
>+    uint64_t high = little_endian::Load64(p + len - 8);
>+    return {little_endian::Load64(p), high >> (128 - len * 8)};
>+  }
>+
>+  // Reads 4 to 8 bytes from p. Zero pads to fill uint64_t.
>+  static uint64_t Read4To8(const unsigned char* p, size_t len) {
>+    return (static_cast<uint64_t>(little_endian::Load32(p + len - 4))
>+            << (len - 4) * 8) |
>+           little_endian::Load32(p);
>+  }
>+
>+  // Reads 1 to 3 bytes from p. Zero pads to fill uint32_t.
>+  static uint32_t Read1To3(const unsigned char* p, size_t len) {
>+    return static_cast<uint32_t>((p[0]) |                         //
>+                                 (p[len / 2] << (len / 2 * 8)) |  //
>+                                 (p[len - 1] << ((len - 1) * 8)));
>+  }
>+
>+  ABSL_ATTRIBUTE_ALWAYS_INLINE static uint64_t Mix(uint64_t state, uint64_t v) {
>+    using MultType =
>+        absl::conditional_t<sizeof(size_t) == 4, uint64_t, uint128>;
>+    // We do the addition in 64-bit space to make sure the 128-bit
>+    // multiplication is fast. If we were to do it as MultType the compiler has
>+    // to assume that the high word is non-zero and needs to perform 2
>+    // multiplications instead of one.
>+    MultType m = state + v;
>+    m *= kMul;
>+    return static_cast<uint64_t>(m ^ (m >> (sizeof(m) * 8 / 2)));
>+  }
>+
>+  // Seed()
>+  //
>+  // A non-deterministic seed.
>+  //
>+  // The current purpose of this seed is to generate non-deterministic results
>+  // and prevent having users depend on the particular hash values.
>+  // It is not meant as a security feature right now, but it leaves the door
>+  // open to upgrade it to a true per-process random seed. A true random seed
>+  // costs more and we don't need to pay for that right now.
>+  //
>+  // On platforms with ASLR, we take advantage of it to make a per-process
>+  // random value.
>+  // See https://en.wikipedia.org/wiki/Address_space_layout_randomization
>+  //
>+  // On other platforms this is still going to be non-deterministic but most
>+  // probably per-build and not per-process.
>+  ABSL_ATTRIBUTE_ALWAYS_INLINE static uint64_t Seed() {
>+    return static_cast<uint64_t>(reinterpret_cast<uintptr_t>(kSeed));
>+  }
>+  static const void* const kSeed;
>+
>+  uint64_t state_;
>+};
>+
>+// CityHashState::CombineContiguousImpl()
>+inline uint64_t CityHashState::CombineContiguousImpl(
>+    uint64_t state, const unsigned char* first, size_t len,
>+    std::integral_constant<int, 4> /* sizeof_size_t */) {
>+  // For large values we use CityHash, for small ones we just use a
>+  // multiplicative hash.
>+  uint64_t v;
>+  if (len > 8) {
>+    v = absl::hash_internal::CityHash32(reinterpret_cast<const char*>(first), len);
>+  } else if (len >= 4) {
>+    v = Read4To8(first, len);
>+  } else if (len > 0) {
>+    v = Read1To3(first, len);
>+  } else {
>+    // Empty ranges have no effect.
>+    return state;
>+  }
>+  return Mix(state, v);
>+}
>+
>+// Overload of CityHashState::CombineContiguousImpl()
>+inline uint64_t CityHashState::CombineContiguousImpl(
>+    uint64_t state, const unsigned char* first, size_t len,
>+    std::integral_constant<int, 8> /* sizeof_size_t */) {
>+  // For large values we use CityHash, for small ones we just use a
>+  // multiplicative hash.
>+  uint64_t v;
>+  if (len > 16) {
>+    v = absl::hash_internal::CityHash64(reinterpret_cast<const char*>(first), len);
>+  } else if (len > 8) {
>+    auto p = Read9To16(first, len);
>+    state = Mix(state, p.first);
>+    v = p.second;
>+  } else if (len >= 4) {
>+    v = Read4To8(first, len);
>+  } else if (len > 0) {
>+    v = Read1To3(first, len);
>+  } else {
>+    // Empty ranges have no effect.
>+    return state;
>+  }
>+  return Mix(state, v);
>+}
>+
>+
>+struct AggregateBarrier {};
>+
>+// HashImpl
>+
>+// Add a private base class to make sure this type is not an aggregate.
>+// Aggregates can be aggregate initialized even if the default constructor is
>+// deleted.
>+struct PoisonedHash : private AggregateBarrier {
>+  PoisonedHash() = delete;
>+  PoisonedHash(const PoisonedHash&) = delete;
>+  PoisonedHash& operator=(const PoisonedHash&) = delete;
>+};
>+
>+template <typename T>
>+struct HashImpl {
>+  size_t operator()(const T& value) const { return CityHashState::hash(value); }
>+};
>+
>+template <typename T>
>+struct Hash
>+    : absl::conditional_t<is_hashable<T>::value, HashImpl<T>, PoisonedHash> {};
>+
>+template <typename H>
>+template <typename T, typename... Ts>
>+H HashStateBase<H>::combine(H state, const T& value, const Ts&... values) {
>+  return H::combine(hash_internal::InvokeHash(std::move(state), value),
>+                    values...);
>+}
>+
>+// HashStateBase::combine_contiguous()
>+template <typename H>
>+template <typename T>
>+H HashStateBase<H>::combine_contiguous(H state, const T* data, size_t size) {
>+  return hash_internal::hash_range_or_bytes(std::move(state), data, size);
>+}
>+}  // namespace hash_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HASH_INTERNAL_HASH_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/print_hash_of.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/print_hash_of.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..b6df31cc5e85b9046e0a54ebcbe839bc60402175
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/print_hash_of.cc
>@@ -0,0 +1,23 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <cstdlib>
>+
>+#include "absl/hash/hash.h"
>+
>+// Prints the hash of argv[1].
>+int main(int argc, char** argv) {
>+  if (argc < 2) return 1;
>+  printf("%zu\n", absl::Hash<int>{}(std::atoi(argv[1])));  // NOLINT
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/spy_hash_state.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/spy_hash_state.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..03d795b09001e3dc303013738f6246aa2de0ac5f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/hash/internal/spy_hash_state.h
>@@ -0,0 +1,218 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_HASH_INTERNAL_SPY_HASH_STATE_H_
>+#define ABSL_HASH_INTERNAL_SPY_HASH_STATE_H_
>+
>+#include <ostream>
>+#include <string>
>+#include <vector>
>+
>+#include "absl/hash/hash.h"
>+#include "absl/strings/match.h"
>+#include "absl/strings/str_format.h"
>+#include "absl/strings/str_join.h"
>+
>+namespace absl {
>+namespace hash_internal {
>+
>+// SpyHashState is an implementation of the HashState API that simply
>+// accumulates all input bytes in an internal buffer. This makes it useful
>+// for testing AbslHashValue overloads (so long as they are templated on the
>+// HashState parameter), since it can report the exact hash representation
>+// that the AbslHashValue overload produces.
>+//
>+// Sample usage:
>+// EXPECT_EQ(SpyHashState::combine(SpyHashState(), foo),
>+//           SpyHashState::combine(SpyHashState(), bar));
>+template <typename T>
>+class SpyHashStateImpl : public HashStateBase<SpyHashStateImpl<T>> {
>+ public:
>+  SpyHashStateImpl()
>+      : error_(std::make_shared<absl::optional<std::string>>()) {
>+    static_assert(std::is_void<T>::value, "");
>+  }
>+
>+  // Move-only
>+  SpyHashStateImpl(const SpyHashStateImpl&) = delete;
>+  SpyHashStateImpl& operator=(const SpyHashStateImpl&) = delete;
>+
>+  SpyHashStateImpl(SpyHashStateImpl&& other) noexcept {
>+    *this = std::move(other);
>+  }
>+
>+  SpyHashStateImpl& operator=(SpyHashStateImpl&& other) noexcept {
>+    hash_representation_ = std::move(other.hash_representation_);
>+    error_ = other.error_;
>+    moved_from_ = other.moved_from_;
>+    other.moved_from_ = true;
>+    return *this;
>+  }
>+
>+  template <typename U>
>+  SpyHashStateImpl(SpyHashStateImpl<U>&& other) {  // NOLINT
>+    hash_representation_ = std::move(other.hash_representation_);
>+    error_ = other.error_;
>+    moved_from_ = other.moved_from_;
>+    other.moved_from_ = true;
>+  }
>+
>+  template <typename A, typename... Args>
>+  static SpyHashStateImpl combine(SpyHashStateImpl s, const A& a,
>+                                  const Args&... args) {
>+    // Pass an instance of SpyHashStateImpl<A> when trying to combine `A`. This
>+    // allows us to test that the user only uses this instance for combine calls
>+    // and does not call AbslHashValue directly.
>+    // See AbslHashValue implementation at the bottom.
>+    s = SpyHashStateImpl<A>::HashStateBase::combine(std::move(s), a);
>+    return SpyHashStateImpl::combine(std::move(s), args...);
>+  }
>+  static SpyHashStateImpl combine(SpyHashStateImpl s) {
>+    if (direct_absl_hash_value_error_) {
>+      *s.error_ = "AbslHashValue should not be invoked directly.";
>+    } else if (s.moved_from_) {
>+      *s.error_ = "Used moved-from instance of the hash state object.";
>+    }
>+    return s;
>+  }
>+
>+  static void SetDirectAbslHashValueError() {
>+    direct_absl_hash_value_error_ = true;
>+  }
>+
>+  // Two SpyHashStateImpl objects are equal if they hold equal hash
>+  // representations.
>+  friend bool operator==(const SpyHashStateImpl& lhs,
>+                         const SpyHashStateImpl& rhs) {
>+    return lhs.hash_representation_ == rhs.hash_representation_;
>+  }
>+
>+  friend bool operator!=(const SpyHashStateImpl& lhs,
>+                         const SpyHashStateImpl& rhs) {
>+    return !(lhs == rhs);
>+  }
>+
>+  enum class CompareResult {
>+    kEqual,
>+    kASuffixB,
>+    kBSuffixA,
>+    kUnequal,
>+  };
>+
>+  static CompareResult Compare(const SpyHashStateImpl& a,
>+                               const SpyHashStateImpl& b) {
>+    const std::string a_flat = absl::StrJoin(a.hash_representation_, "");
>+    const std::string b_flat = absl::StrJoin(b.hash_representation_, "");
>+    if (a_flat == b_flat) return CompareResult::kEqual;
>+    if (absl::EndsWith(a_flat, b_flat)) return CompareResult::kBSuffixA;
>+    if (absl::EndsWith(b_flat, a_flat)) return CompareResult::kASuffixB;
>+    return CompareResult::kUnequal;
>+  }
>+
>+  // operator<< prints the hash representation as a hex and ASCII dump, to
>+  // facilitate debugging.
>+  friend std::ostream& operator<<(std::ostream& out,
>+                                  const SpyHashStateImpl& hash_state) {
>+    out << "[\n";
>+    for (auto& s : hash_state.hash_representation_) {
>+      size_t offset = 0;
>+      for (char c : s) {
>+        if (offset % 16 == 0) {
>+          out << absl::StreamFormat("\n0x%04x: ", offset);
>+        }
>+        if (offset % 2 == 0) {
>+          out << " ";
>+        }
>+        out << absl::StreamFormat("%02x", c);
>+        ++offset;
>+      }
>+      out << "\n";
>+    }
>+    return out << "]";
>+  }
>+
>+  // The base case of the combine recursion, which writes raw bytes into the
>+  // internal buffer.
>+  static SpyHashStateImpl combine_contiguous(SpyHashStateImpl hash_state,
>+                                             const unsigned char* begin,
>+                                             size_t size) {
>+    hash_state.hash_representation_.emplace_back(
>+        reinterpret_cast<const char*>(begin), size);
>+    return hash_state;
>+  }
>+
>+  using SpyHashStateImpl::HashStateBase::combine_contiguous;
>+
>+  absl::optional<std::string> error() const {
>+    if (moved_from_) {
>+      return "Returned a moved-from instance of the hash state object.";
>+    }
>+    return *error_;
>+  }
>+
>+ private:
>+  template <typename U>
>+  friend class SpyHashStateImpl;
>+
>+  // This is true if SpyHashStateImpl<T> has been passed to a call of
>+  // AbslHashValue with the wrong type. This detects that the user called
>+  // AbslHashValue directly (because the hash state type does not match).
>+  static bool direct_absl_hash_value_error_;
>+
>+
>+  std::vector<std::string> hash_representation_;
>+  // This is a shared_ptr because we want all instances of the particular
>+  // SpyHashState run to share the field. This way we can set the error for
>+  // use-after-move and all the copies will see it.
>+  std::shared_ptr<absl::optional<std::string>> error_;
>+  bool moved_from_ = false;
>+};
>+
>+template <typename T>
>+bool SpyHashStateImpl<T>::direct_absl_hash_value_error_;
>+
>+template <bool& B>
>+struct OdrUse {
>+  constexpr OdrUse() {}
>+  bool& b = B;
>+};
>+
>+template <void (*)()>
>+struct RunOnStartup {
>+  static bool run;
>+  static constexpr OdrUse<run> kOdrUse{};
>+};
>+
>+template <void (*f)()>
>+bool RunOnStartup<f>::run = (f(), true);
>+
>+template <
>+    typename T, typename U,
>+    // Only trigger for when (T != U),
>+    absl::enable_if_t<!std::is_same<T, U>::value, int> = 0,
>+    // This statement works in two ways:
>+    //  - First, it instantiates RunOnStartup and forces the initialization of
>+    //    `run`, which set the global variable.
>+    //  - Second, it triggers a SFINAE error disabling the overload to prevent
>+    //    compile time errors. If we didn't disable the overload we would get
>+    //    ambiguous overload errors, which we don't want.
>+    int = RunOnStartup<SpyHashStateImpl<T>::SetDirectAbslHashValueError>::run>
>+void AbslHashValue(SpyHashStateImpl<T>, const U&);
>+
>+using SpyHashState = SpyHashStateImpl<void>;
>+
>+}  // namespace hash_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HASH_INTERNAL_SPY_HASH_STATE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/BUILD.bazel
>index 46f47b12bb2adb2f672fb050fb40d77ecf85bc21..89a312eac4b19bf88ba1f2100ca045d1cc32785e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/BUILD.bazel
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/BUILD.bazel
>@@ -19,6 +19,7 @@ load(
>     "ABSL_DEFAULT_COPTS",
>     "ABSL_TEST_COPTS",
>     "ABSL_EXCEPTIONS_FLAG",
>+    "ABSL_EXCEPTIONS_FLAG_LINKOPTS",
> )
> 
> package(default_visibility = ["//visibility:public"])
>@@ -53,6 +54,7 @@ cc_test(
>         "memory_exception_safety_test.cc",
>     ],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":memory",
>         "//absl/base:exception_safety_testing",
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory.h
>index c7caf8b94dc91b2838d13eb48904d172d2a30f73..1eaec0f40aead57d692e4b858e64214045645f81 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory.h
>@@ -39,16 +39,30 @@ namespace absl {
> // Function Template: WrapUnique()
> // -----------------------------------------------------------------------------
> //
>-//  Adopts ownership from a raw pointer and transfers it to the returned
>-//  `std::unique_ptr`, whose type is deduced. DO NOT specify the template type T
>-//  when calling WrapUnique.
>+// Adopts ownership from a raw pointer and transfers it to the returned
>+// `std::unique_ptr`, whose type is deduced. Because of this deduction, *do not*
>+// specify the template type `T` when calling `WrapUnique`.
> //
> // Example:
> //   X* NewX(int, int);
> //   auto x = WrapUnique(NewX(1, 2));  // 'x' is std::unique_ptr<X>.
> //
>-// `absl::WrapUnique` is useful for capturing the output of a raw pointer
>-// factory. However, prefer 'absl::make_unique<T>(args...) over
>+// The purpose of WrapUnique is to automatically deduce the pointer type. If you
>+// wish to make the type explicit, for readability reasons or because you prefer
>+// to use a base-class pointer rather than a derived one, just use
>+// `std::unique_ptr` directly.
>+//
>+// Example:
>+//   X* Factory(int, int);
>+//   auto x = std::unique_ptr<X>(Factory(1, 2));
>+//                  - or -
>+//   std::unique_ptr<X> x(Factory(1, 2));
>+//
>+// This has the added advantage of working whether Factory returns a raw
>+// pointer or a `std::unique_ptr`.
>+//
>+// While `absl::WrapUnique` is useful for capturing the output of a raw
>+// pointer factory, prefer 'absl::make_unique<T>(args...)' over
> // 'absl::WrapUnique(new T(args...))'.
> //
> //   auto x = WrapUnique(new X(1, 2));  // works, but nonideal.
>@@ -641,55 +655,59 @@ struct default_allocator_is_nothrow : std::false_type {};
> #endif
> 
> namespace memory_internal {
>-#ifdef ABSL_HAVE_EXCEPTIONS
>-template <typename Allocator, typename StorageElement, typename... Args>
>-void ConstructStorage(Allocator* alloc, StorageElement* first,
>-                      StorageElement* last, const Args&... args) {
>-  for (StorageElement* cur = first; cur != last; ++cur) {
>+#ifdef ABSL_HAVE_EXCEPTIONS  // ConstructRange
>+template <typename Allocator, typename Iterator, typename... Args>
>+void ConstructRange(Allocator& alloc, Iterator first, Iterator last,
>+                    const Args&... args) {
>+  for (Iterator cur = first; cur != last; ++cur) {
>     try {
>-      std::allocator_traits<Allocator>::construct(*alloc, cur, args...);
>+      std::allocator_traits<Allocator>::construct(alloc, cur, args...);
>     } catch (...) {
>       while (cur != first) {
>         --cur;
>-        std::allocator_traits<Allocator>::destroy(*alloc, cur);
>+        std::allocator_traits<Allocator>::destroy(alloc, cur);
>       }
>       throw;
>     }
>   }
> }
>-template <typename Allocator, typename StorageElement, typename Iterator>
>-void CopyToStorageFromRange(Allocator* alloc, StorageElement* destination,
>-                            Iterator first, Iterator last) {
>-  for (StorageElement* cur = destination; first != last;
>+#else   // ABSL_HAVE_EXCEPTIONS  // ConstructRange
>+template <typename Allocator, typename Iterator, typename... Args>
>+void ConstructRange(Allocator& alloc, Iterator first, Iterator last,
>+                    const Args&... args) {
>+  for (; first != last; ++first) {
>+    std::allocator_traits<Allocator>::construct(alloc, first, args...);
>+  }
>+}
>+#endif  // ABSL_HAVE_EXCEPTIONS  // ConstructRange
>+
>+#ifdef ABSL_HAVE_EXCEPTIONS  // CopyRange
>+template <typename Allocator, typename Iterator, typename InputIterator>
>+void CopyRange(Allocator& alloc, Iterator destination, InputIterator first,
>+               InputIterator last) {
>+  for (Iterator cur = destination; first != last;
>        static_cast<void>(++cur), static_cast<void>(++first)) {
>     try {
>-      std::allocator_traits<Allocator>::construct(*alloc, cur, *first);
>+      std::allocator_traits<Allocator>::construct(alloc, cur, *first);
>     } catch (...) {
>       while (cur != destination) {
>         --cur;
>-        std::allocator_traits<Allocator>::destroy(*alloc, cur);
>+        std::allocator_traits<Allocator>::destroy(alloc, cur);
>       }
>       throw;
>     }
>   }
> }
>-#else   // ABSL_HAVE_EXCEPTIONS
>-template <typename Allocator, typename StorageElement, typename... Args>
>-void ConstructStorage(Allocator* alloc, StorageElement* first,
>-                      StorageElement* last, const Args&... args) {
>-  for (; first != last; ++first) {
>-    std::allocator_traits<Allocator>::construct(*alloc, first, args...);
>-  }
>-}
>-template <typename Allocator, typename StorageElement, typename Iterator>
>-void CopyToStorageFromRange(Allocator* alloc, StorageElement* destination,
>-                            Iterator first, Iterator last) {
>+#else   // ABSL_HAVE_EXCEPTIONS  // CopyRange
>+template <typename Allocator, typename Iterator, typename InputIterator>
>+void CopyRange(Allocator& alloc, Iterator destination, InputIterator first,
>+               InputIterator last) {
>   for (; first != last;
>        static_cast<void>(++destination), static_cast<void>(++first)) {
>-    std::allocator_traits<Allocator>::construct(*alloc, destination, *first);
>+    std::allocator_traits<Allocator>::construct(alloc, destination, *first);
>   }
> }
>-#endif  // ABSL_HAVE_EXCEPTIONS
>+#endif  // ABSL_HAVE_EXCEPTIONS  // CopyRange
> }  // namespace memory_internal
> }  // namespace absl
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_exception_safety_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_exception_safety_test.cc
>index d1f6e84f10f8a668ac25fb6d90ccc5a783202219..00d2b192cfc32df624363ca863ec79f90f6185b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_exception_safety_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_exception_safety_test.cc
>@@ -32,7 +32,7 @@ TEST(MakeUnique, CheckForLeaks) {
>                     .WithInitialValue(Thrower(kValue))
>                     // Ensures make_unique does not modify the input. The real
>                     // test, though, is ConstructorTracker checking for leaks.
>-                    .WithInvariants(testing::strong_guarantee);
>+                    .WithContracts(testing::strong_guarantee);
> 
>   EXPECT_TRUE(tester.Test([](Thrower* thrower) {
>     static_cast<void>(absl::make_unique<Thrower>(*thrower));
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_test.cc
>index dee9b486a30d30d255705ac3814f19f781994053..2e453e22ba5ba39cfd690fc591e4afe487cb78a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_test.cc
>@@ -149,7 +149,6 @@ TEST(Make_UniqueTest, NotAmbiguousWithStdMakeUnique) {
> }
> 
> #if 0
>-// TODO(billydonahue): Make a proper NC test.
> // These tests shouldn't compile.
> TEST(MakeUniqueTestNC, AcceptMoveOnlyLvalue) {
>   auto m = MoveOnly();
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits.h
>index 457b890841a64fb4ed36cde8621d7822190b798d..23ebd6ed0990fd3d9510d4d91066f074cfe11101 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits.h
>@@ -105,8 +105,25 @@ template <class To, template <class...> class Op, class... Args>
> struct is_detected_convertible
>     : is_detected_convertible_impl<void, To, Op, Args...>::type {};
> 
>+template <typename T>
>+using IsCopyAssignableImpl =
>+    decltype(std::declval<T&>() = std::declval<const T&>());
>+
>+template <typename T>
>+using IsMoveAssignableImpl = decltype(std::declval<T&>() = std::declval<T&&>());
>+
> }  // namespace type_traits_internal
> 
>+template <typename T>
>+struct is_copy_assignable : type_traits_internal::is_detected<
>+                                type_traits_internal::IsCopyAssignableImpl, T> {
>+};
>+
>+template <typename T>
>+struct is_move_assignable : type_traits_internal::is_detected<
>+                                type_traits_internal::IsMoveAssignableImpl, T> {
>+};
>+
> // void_t()
> //
> // Ignores the type of any its arguments and returns `void`. In general, this
>@@ -309,7 +326,7 @@ template <typename T>
> struct is_trivially_copy_assignable
>     : std::integral_constant<
>           bool, __has_trivial_assign(typename std::remove_reference<T>::type) &&
>-                    std::is_copy_assignable<T>::value> {
>+                    absl::is_copy_assignable<T>::value> {
> #ifdef ABSL_HAVE_STD_IS_TRIVIALLY_ASSIGNABLE
>  private:
>   static constexpr bool compliant =
>@@ -409,11 +426,11 @@ struct IsHashEnabled
>     : absl::conjunction<std::is_default_constructible<std::hash<Key>>,
>                         std::is_copy_constructible<std::hash<Key>>,
>                         std::is_destructible<std::hash<Key>>,
>-                        std::is_copy_assignable<std::hash<Key>>,
>+                        absl::is_copy_assignable<std::hash<Key>>,
>                         IsHashable<Key>> {};
>+
> }  // namespace type_traits_internal
> 
> }  // namespace absl
> 
>-
> #endif  // ABSL_META_TYPE_TRAITS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits_test.cc
>index 81b4bd323977e5e3cd27660c50cced86ab225017..f51f5dedee435d683b4b2f5eb3012bd61b5e2aa3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits_test.cc
>@@ -877,4 +877,80 @@ TEST(TypeTraitsTest, TestResultOf) {
>   EXPECT_EQ(TypeEnum::D, GetTypeExt(Wrap<TypeD>()));
> }
> 
>+template <typename T>
>+bool TestCopyAssign() {
>+  return absl::is_copy_assignable<T>::value ==
>+         std::is_copy_assignable<T>::value;
>+}
>+
>+TEST(TypeTraitsTest, IsCopyAssignable) {
>+  EXPECT_TRUE(TestCopyAssign<int>());
>+  EXPECT_TRUE(TestCopyAssign<int&>());
>+  EXPECT_TRUE(TestCopyAssign<int&&>());
>+
>+  struct S {};
>+  EXPECT_TRUE(TestCopyAssign<S>());
>+  EXPECT_TRUE(TestCopyAssign<S&>());
>+  EXPECT_TRUE(TestCopyAssign<S&&>());
>+
>+  class C {
>+   public:
>+    explicit C(C* c) : c_(c) {}
>+    ~C() { delete c_; }
>+
>+   private:
>+    C* c_;
>+  };
>+  EXPECT_TRUE(TestCopyAssign<C>());
>+  EXPECT_TRUE(TestCopyAssign<C&>());
>+  EXPECT_TRUE(TestCopyAssign<C&&>());
>+
>+  // Reason for ifndef: add_lvalue_reference<T> in libc++ breaks for these cases
>+#ifndef _LIBCPP_VERSION
>+  EXPECT_TRUE(TestCopyAssign<int()>());
>+  EXPECT_TRUE(TestCopyAssign<int(int) const>());
>+  EXPECT_TRUE(TestCopyAssign<int(...) volatile&>());
>+  EXPECT_TRUE(TestCopyAssign<int(int, ...) const volatile&&>());
>+#endif  // _LIBCPP_VERSION
>+}
>+
>+template <typename T>
>+bool TestMoveAssign() {
>+  return absl::is_move_assignable<T>::value ==
>+         std::is_move_assignable<T>::value;
>+}
>+
>+TEST(TypeTraitsTest, IsMoveAssignable) {
>+  EXPECT_TRUE(TestMoveAssign<int>());
>+  EXPECT_TRUE(TestMoveAssign<int&>());
>+  EXPECT_TRUE(TestMoveAssign<int&&>());
>+
>+  struct S {};
>+  EXPECT_TRUE(TestMoveAssign<S>());
>+  EXPECT_TRUE(TestMoveAssign<S&>());
>+  EXPECT_TRUE(TestMoveAssign<S&&>());
>+
>+  class C {
>+   public:
>+    explicit C(C* c) : c_(c) {}
>+    ~C() { delete c_; }
>+    void operator=(const C&) = delete;
>+    void operator=(C&&) = delete;
>+
>+   private:
>+    C* c_;
>+  };
>+  EXPECT_TRUE(TestMoveAssign<C>());
>+  EXPECT_TRUE(TestMoveAssign<C&>());
>+  EXPECT_TRUE(TestMoveAssign<C&&>());
>+
>+  // Reason for ifndef: add_lvalue_reference<T> in libc++ breaks for these cases
>+#ifndef _LIBCPP_VERSION
>+  EXPECT_TRUE(TestMoveAssign<int()>());
>+  EXPECT_TRUE(TestMoveAssign<int(int) const>());
>+  EXPECT_TRUE(TestMoveAssign<int(...) volatile&>());
>+  EXPECT_TRUE(TestMoveAssign<int(int, ...) const volatile&&>());
>+#endif  // _LIBCPP_VERSION
>+}
>+
> }  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/BUILD.bazel
>index f49571ebb3e1042434151a2237127a99df1ad301..324ce6695e402c8973fc5b5df257aa5f265203dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/BUILD.bazel
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/BUILD.bazel
>@@ -49,6 +49,7 @@ cc_test(
>         ":int128",
>         "//absl/base",
>         "//absl/base:core_headers",
>+        "//absl/hash:hash_testing",
>         "//absl/meta:type_traits",
>         "@com_google_googletest//:gtest_main",
>     ],
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128.h
>index 2d131b8bda1c348d91743e7c42d38a57968c330b..79b62a7586695d5b2f431d38af52a0ab2274ae2c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128.h
>@@ -192,6 +192,12 @@ class
>   // Returns the highest value for a 128-bit unsigned integer.
>   friend constexpr uint128 Uint128Max();
> 
>+  // Support for absl::Hash.
>+  template <typename H>
>+  friend H AbslHashValue(H h, uint128 v) {
>+    return H::combine(std::move(h), Uint128High64(v), Uint128Low64(v));
>+  }
>+
>  private:
>   constexpr uint128(uint64_t high, uint64_t low);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_test.cc
>index 1eb3e0ec8961c1c6c88756ef1f201f0203aee0a2..dfe3475aca0627b29e633cc2f85ce846b61bc3f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_test.cc
>@@ -23,6 +23,7 @@
> 
> #include "gtest/gtest.h"
> #include "absl/base/internal/cycleclock.h"
>+#include "absl/hash/hash_testing.h"
> #include "absl/meta/type_traits.h"
> 
> #if defined(_MSC_VER) && _MSC_VER == 1900
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.bazel
>index 3a5f1332cda0e1f7040ef0dbdae557b018213854..6d7c2619a530e000f74ea3115dfd797d702054ff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.bazel
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.bazel
>@@ -19,6 +19,7 @@ load(
>     "ABSL_DEFAULT_COPTS",
>     "ABSL_TEST_COPTS",
>     "ABSL_EXCEPTIONS_FLAG",
>+    "ABSL_EXCEPTIONS_FLAG_LINKOPTS",
> )
> 
> package(
>@@ -69,6 +70,7 @@ cc_library(
>     deps = [
>         ":internal",
>         "//absl/base",
>+        "//absl/base:bits",
>         "//absl/base:config",
>         "//absl/base:core_headers",
>         "//absl/base:endian",
>@@ -86,7 +88,6 @@ cc_library(
>         "internal/utf8.cc",
>     ],
>     hdrs = [
>-        "internal/bits.h",
>         "internal/char_map.h",
>         "internal/ostringstream.h",
>         "internal/resize_uninitialized.h",
>@@ -212,7 +213,6 @@ cc_test(
>     visibility = ["//visibility:private"],
>     deps = [
>         ":internal",
>-        ":strings",
>         "//absl/base:core_headers",
>         "@com_google_googletest//:gtest_main",
>     ],
>@@ -237,6 +237,7 @@ cc_test(
>     size = "small",
>     srcs = ["string_view_test.cc"],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     visibility = ["//visibility:private"],
>     deps = [
>         ":strings",
>@@ -373,7 +374,6 @@ cc_test(
>     visibility = ["//visibility:private"],
>     deps = [
>         ":strings",
>-        "//absl/memory",
>         "@com_github_google_benchmark//:benchmark_main",
>     ],
> )
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.gn
>index 9a8a10d0e7108e79e42c9c5b36355cf879016b95..20af83c0e518b650eb2e899d3e7b3c9efe7c108b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.gn
>@@ -59,6 +59,7 @@ source_set("strings") {
>   deps = [
>     ":internal",
>     "../base",
>+    "../base:bits",
>     "../base:config",
>     "../base:core_headers",
>     "../base:endian",
>@@ -81,7 +82,6 @@ source_set("internal") {
>     "internal/utf8.cc",
>   ]
>   public = [
>-    "internal/bits.h",
>     "internal/char_map.h",
>     "internal/ostringstream.h",
>     "internal/resize_uninitialized.h",
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/CMakeLists.txt
>index cd122134729fa686e98610269f18acb18f8d3007..f3e4162387f15c832d75062adcc6b7ed5d5da17a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/CMakeLists.txt
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/CMakeLists.txt
>@@ -32,7 +32,6 @@ list(APPEND STRINGS_PUBLIC_HEADERS
> 
> 
> list(APPEND STRINGS_INTERNAL_HEADERS
>-  "internal/bits.h"
>   "internal/char_map.h"
>   "internal/charconv_bigint.h"
>   "internal/charconv_parse.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii.h
>index 96a64541c321a710d29efc8d9263862f1b9960e7..48a9da22e769a85128a69cf2dfc332980afed445 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii.h
>@@ -165,7 +165,7 @@ inline char ascii_tolower(unsigned char c) {
> // Converts the characters in `s` to lowercase, changing the contents of `s`.
> void AsciiStrToLower(std::string* s);
> 
>-// Creates a lowercase std::string from a given absl::string_view.
>+// Creates a lowercase string from a given absl::string_view.
> ABSL_MUST_USE_RESULT inline std::string AsciiStrToLower(absl::string_view s) {
>   std::string result(s);
>   absl::AsciiStrToLower(&result);
>@@ -183,7 +183,7 @@ inline char ascii_toupper(unsigned char c) {
> // Converts the characters in `s` to uppercase, changing the contents of `s`.
> void AsciiStrToUpper(std::string* s);
> 
>-// Creates an uppercase std::string from a given absl::string_view.
>+// Creates an uppercase string from a given absl::string_view.
> ABSL_MUST_USE_RESULT inline std::string AsciiStrToUpper(absl::string_view s) {
>   std::string result(s);
>   absl::AsciiStrToUpper(&result);
>@@ -195,10 +195,10 @@ ABSL_MUST_USE_RESULT inline std::string AsciiStrToUpper(absl::string_view s) {
> ABSL_MUST_USE_RESULT inline absl::string_view StripLeadingAsciiWhitespace(
>     absl::string_view str) {
>   auto it = std::find_if_not(str.begin(), str.end(), absl::ascii_isspace);
>-  return absl::string_view(it, str.end() - it);
>+  return str.substr(it - str.begin());
> }
> 
>-// Strips in place whitespace from the beginning of the given std::string.
>+// Strips in place whitespace from the beginning of the given string.
> inline void StripLeadingAsciiWhitespace(std::string* str) {
>   auto it = std::find_if_not(str->begin(), str->end(), absl::ascii_isspace);
>   str->erase(str->begin(), it);
>@@ -209,10 +209,10 @@ inline void StripLeadingAsciiWhitespace(std::string* str) {
> ABSL_MUST_USE_RESULT inline absl::string_view StripTrailingAsciiWhitespace(
>     absl::string_view str) {
>   auto it = std::find_if_not(str.rbegin(), str.rend(), absl::ascii_isspace);
>-  return absl::string_view(str.begin(), str.rend() - it);
>+  return str.substr(0, str.rend() - it);
> }
> 
>-// Strips in place whitespace from the end of the given std::string
>+// Strips in place whitespace from the end of the given string
> inline void StripTrailingAsciiWhitespace(std::string* str) {
>   auto it = std::find_if_not(str->rbegin(), str->rend(), absl::ascii_isspace);
>   str->erase(str->rend() - it);
>@@ -225,7 +225,7 @@ ABSL_MUST_USE_RESULT inline absl::string_view StripAsciiWhitespace(
>   return StripTrailingAsciiWhitespace(StripLeadingAsciiWhitespace(str));
> }
> 
>-// Strips in place whitespace from both ends of the given std::string
>+// Strips in place whitespace from both ends of the given string
> inline void StripAsciiWhitespace(std::string* str) {
>   StripTrailingAsciiWhitespace(str);
>   StripLeadingAsciiWhitespace(str);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.cc
>index 08c3947eccd475dee33310d247bf8740039980c8..c7b8c98b1b4459c56ea851e9281e35a92613c60f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.cc
>@@ -20,8 +20,8 @@
> #include <cstring>
> 
> #include "absl/base/casts.h"
>+#include "absl/base/internal/bits.h"
> #include "absl/numeric/int128.h"
>-#include "absl/strings/internal/bits.h"
> #include "absl/strings/internal/charconv_bigint.h"
> #include "absl/strings/internal/charconv_parse.h"
> 
>@@ -243,9 +243,9 @@ struct CalculatedFloat {
> // minus the number of leading zero bits.)
> int BitWidth(uint128 value) {
>   if (Uint128High64(value) == 0) {
>-    return 64 - strings_internal::CountLeadingZeros64(Uint128Low64(value));
>+    return 64 - base_internal::CountLeadingZeros64(Uint128Low64(value));
>   }
>-  return 128 - strings_internal::CountLeadingZeros64(Uint128High64(value));
>+  return 128 - base_internal::CountLeadingZeros64(Uint128High64(value));
> }
> 
> // Calculates how far to the right a mantissa needs to be shifted to create a
>@@ -518,7 +518,7 @@ CalculatedFloat CalculateFromParsedHexadecimal(
>     const strings_internal::ParsedFloat& parsed_hex) {
>   uint64_t mantissa = parsed_hex.mantissa;
>   int exponent = parsed_hex.exponent;
>-  int mantissa_width = 64 - strings_internal::CountLeadingZeros64(mantissa);
>+  int mantissa_width = 64 - base_internal::CountLeadingZeros64(mantissa);
>   const int shift = NormalizedShiftSize<FloatType>(mantissa_width, exponent);
>   bool result_exact;
>   exponent += shift;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.h
>index 3e313679c9618aa88dc9e8b55e88e96a0f7f696b..0735382962356cff7bfb5f3ebd3a0393d2d2810a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.h
>@@ -22,7 +22,7 @@ namespace absl {
> // Workalike compatibilty version of std::chars_format from C++17.
> //
> // This is an bitfield enumerator which can be passed to absl::from_chars to
>-// configure the std::string-to-float conversion.
>+// configure the string-to-float conversion.
> enum class chars_format {
>   scientific = 1,
>   fixed = 2,
>@@ -30,7 +30,7 @@ enum class chars_format {
>   general = fixed | scientific,
> };
> 
>-// The return result of a std::string-to-number conversion.
>+// The return result of a string-to-number conversion.
> //
> // `ec` will be set to `invalid_argument` if a well-formed number was not found
> // at the start of the input range, `result_out_of_range` if a well-formed
>@@ -67,7 +67,7 @@ struct from_chars_result {
> // If `fmt` is set, it must be one of the enumerator values of the chars_format.
> // (This is despite the fact that chars_format is a bitmask type.)  If set to
> // `scientific`, a matching number must contain an exponent.  If set to `fixed`,
>-// then an exponent will never match.  (For example, the std::string "1e5" will be
>+// then an exponent will never match.  (For example, the string "1e5" will be
> // parsed as "1".)  If set to `hex`, then a hexadecimal float is parsed in the
> // format that strtod() accepts, except that a "0x" prefix is NOT matched.
> // (In particular, in `hex` mode, the input "0xff" results in the largest
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv_test.cc
>index f8d71cc6c448c7a3e7f0caf98098fc7b4d2d5642..89418fe948ffedc4b75ac2fa810fd4416605c3bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv_test.cc
>@@ -33,7 +33,7 @@ namespace {
> 
> #if ABSL_COMPILER_DOES_EXACT_ROUNDING
> 
>-// Tests that the given std::string is accepted by absl::from_chars, and that it
>+// Tests that the given string is accepted by absl::from_chars, and that it
> // converts exactly equal to the given number.
> void TestDoubleParse(absl::string_view str, double expected_number) {
>   SCOPED_TRACE(str);
>@@ -250,7 +250,7 @@ TEST(FromChars, NearRoundingCasesExplicit) {
>   EXPECT_EQ(ToFloat("459926601011.e15"), ldexpf(12466336, 65));
> }
> 
>-// Common test logic for converting a std::string which lies exactly halfway between
>+// Common test logic for converting a string which lies exactly halfway between
> // two target floats.
> //
> // mantissa and exponent represent the precise value between two floating point
>@@ -655,7 +655,7 @@ int NextStep(int step) {
> // is correct for in-bounds values, and that overflow and underflow are done
> // correctly for out-of-bounds values.
> //
>-// input_generator maps from an integer index to a std::string to test.
>+// input_generator maps from an integer index to a string to test.
> // expected_generator maps from an integer index to an expected Float value.
> // from_chars conversion of input_generator(i) should result in
> // expected_generator(i).
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.cc
>index fbc9f756315fa7fe9a7bb025d48c3fb284b9e8fb..8d8b00b2001ce8307b48788007cc5655a75f54fa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.cc
>@@ -89,7 +89,7 @@ inline bool IsSurrogate(char32_t c, absl::string_view src, std::string* error) {
> //
> //    Unescapes C escape sequences and is the reverse of CEscape().
> //
>-//    If 'source' is valid, stores the unescaped std::string and its size in
>+//    If 'source' is valid, stores the unescaped string and its size in
> //    'dest' and 'dest_len' respectively, and returns true. Otherwise
> //    returns false and optionally stores the error description in
> //    'error'. Set 'error' to nullptr to disable error reporting.
>@@ -104,7 +104,7 @@ bool CUnescapeInternal(absl::string_view source, bool leave_nulls_escaped,
>                        char* dest, ptrdiff_t* dest_len, std::string* error) {
>   char* d = dest;
>   const char* p = source.data();
>-  const char* end = source.end();
>+  const char* end = p + source.size();
>   const char* last_byte = end - 1;
> 
>   // Small optimization for case where source = dest and there's no escaping
>@@ -294,7 +294,7 @@ bool CUnescapeInternal(absl::string_view source, bool leave_nulls_escaped,
> // ----------------------------------------------------------------------
> // CUnescapeInternal()
> //
>-//    Same as above but uses a C++ std::string for output. 'source' and 'dest'
>+//    Same as above but uses a C++ string for output. 'source' and 'dest'
> //    may be the same.
> // ----------------------------------------------------------------------
> bool CUnescapeInternal(absl::string_view source, bool leave_nulls_escaped,
>@@ -304,7 +304,7 @@ bool CUnescapeInternal(absl::string_view source, bool leave_nulls_escaped,
>   ptrdiff_t dest_size;
>   if (!CUnescapeInternal(source,
>                          leave_nulls_escaped,
>-                         const_cast<char*>(dest->data()),
>+                         &(*dest)[0],
>                          &dest_size,
>                          error)) {
>     return false;
>@@ -684,7 +684,7 @@ bool Base64UnescapeInternal(const char* src_param, size_t szsrc, char* dest,
> // The arrays below were generated by the following code
> // #include <sys/time.h>
> // #include <stdlib.h>
>-// #include <std::string.h>
>+// #include <string.h>
> // main()
> // {
> //   static const char Base64[] =
>@@ -939,7 +939,8 @@ constexpr char kBase64Chars[] =
> constexpr char kWebSafeBase64Chars[] =
>     "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789-_";
> 
>-void Base64EscapeInternal(const unsigned char* src, size_t szsrc, std::string* dest,
>+template <typename String>
>+void Base64EscapeInternal(const unsigned char* src, size_t szsrc, String* dest,
>                           bool do_padding, const char* base64_chars) {
>   const size_t calc_escaped_size =
>       CalculateBase64EscapedLenInternal(szsrc, do_padding);
>@@ -951,7 +952,8 @@ void Base64EscapeInternal(const unsigned char* src, size_t szsrc, std::string* d
>   dest->erase(escaped_len);
> }
> 
>-bool Base64UnescapeInternal(const char* src, size_t slen, std::string* dest,
>+template <typename String>
>+bool Base64UnescapeInternal(const char* src, size_t slen, String* dest,
>                             const signed char* unbase64) {
>   // Determine the size of the output std::string.  Base64 encodes every 3 bytes into
>   // 4 characters.  any leftover chars are added directly for good measure.
>@@ -999,7 +1001,7 @@ constexpr char kHexValue[256] = {
> /* clang-format on */
> 
> // This is a templated function so that T can be either a char*
>-// or a std::string.  This works because we use the [] operator to access
>+// or a string.  This works because we use the [] operator to access
> // individual characters at a time.
> template <typename T>
> void HexStringToBytesInternal(const char* from, T to, ptrdiff_t num) {
>@@ -1009,7 +1011,7 @@ void HexStringToBytesInternal(const char* from, T to, ptrdiff_t num) {
>   }
> }
> 
>-// This is a templated function so that T can be either a char* or a std::string.
>+// This is a templated function so that T can be either a char* or a string.
> template <typename T>
> void BytesToHexStringInternal(const unsigned char* src, T dest, ptrdiff_t num) {
>   auto dest_ptr = &dest[0];
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.h
>index 7f1ab96d8ff15d56042b4659d66148214705f0a5..296597304e6ce3f5865dc03a967f34318e69b91c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.h
>@@ -17,7 +17,7 @@
> // File: escaping.h
> // -----------------------------------------------------------------------------
> //
>-// This header file contains std::string utilities involved in escaping and
>+// This header file contains string utilities involved in escaping and
> // unescaping strings in various ways.
> //
> 
>@@ -37,7 +37,7 @@ namespace absl {
> 
> // CUnescape()
> //
>-// Unescapes a `source` std::string and copies it into `dest`, rewriting C-style
>+// Unescapes a `source` string and copies it into `dest`, rewriting C-style
> // escape sequences (http://en.cppreference.com/w/cpp/language/escape) into
> // their proper code point equivalents, returning `true` if successful.
> //
>@@ -57,9 +57,10 @@ namespace absl {
> //     0x99).
> //
> //
>-// If any errors are encountered, this function returns `false` and stores the
>-// first encountered error in `error`. To disable error reporting, set `error`
>-// to `nullptr` or use the overload with no error reporting below.
>+// If any errors are encountered, this function returns `false`, leaving the
>+// `dest` output parameter in an unspecified state, and stores the first
>+// encountered error in `error`. To disable error reporting, set `error` to
>+// `nullptr` or use the overload with no error reporting below.
> //
> // Example:
> //
>@@ -78,7 +79,7 @@ inline bool CUnescape(absl::string_view source, std::string* dest) {
> 
> // CEscape()
> //
>-// Escapes a 'src' std::string using C-style escapes sequences
>+// Escapes a 'src' string using C-style escapes sequences
> // (http://en.cppreference.com/w/cpp/language/escape), escaping other
> // non-printable/non-whitespace bytes as octal sequences (e.g. "\377").
> //
>@@ -91,7 +92,7 @@ std::string CEscape(absl::string_view src);
> 
> // CHexEscape()
> //
>-// Escapes a 'src' std::string using C-style escape sequences, escaping
>+// Escapes a 'src' string using C-style escape sequences, escaping
> // other non-printable/non-whitespace bytes as hexadecimal sequences (e.g.
> // "\xFF").
> //
>@@ -104,7 +105,7 @@ std::string CHexEscape(absl::string_view src);
> 
> // Utf8SafeCEscape()
> //
>-// Escapes a 'src' std::string using C-style escape sequences, escaping bytes as
>+// Escapes a 'src' string using C-style escape sequences, escaping bytes as
> // octal sequences, and passing through UTF-8 characters without conversion.
> // I.e., when encountering any bytes with their high bit set, this function
> // will not escape those values, whether or not they are valid UTF-8.
>@@ -112,47 +113,47 @@ std::string Utf8SafeCEscape(absl::string_view src);
> 
> // Utf8SafeCHexEscape()
> //
>-// Escapes a 'src' std::string using C-style escape sequences, escaping bytes as
>+// Escapes a 'src' string using C-style escape sequences, escaping bytes as
> // hexadecimal sequences, and passing through UTF-8 characters without
> // conversion.
> std::string Utf8SafeCHexEscape(absl::string_view src);
> 
> // Base64Unescape()
> //
>-// Converts a `src` std::string encoded in Base64 to its binary equivalent, writing
>+// Converts a `src` string encoded in Base64 to its binary equivalent, writing
> // it to a `dest` buffer, returning `true` on success. If `src` contains invalid
> // characters, `dest` is cleared and returns `false`.
> bool Base64Unescape(absl::string_view src, std::string* dest);
> 
>-// WebSafeBase64Unescape(absl::string_view, std::string*)
>+// WebSafeBase64Unescape()
> //
>-// Converts a `src` std::string encoded in Base64 to its binary equivalent, writing
>+// Converts a `src` string encoded in Base64 to its binary equivalent, writing
> // it to a `dest` buffer, but using '-' instead of '+', and '_' instead of '/'.
> // If `src` contains invalid characters, `dest` is cleared and returns `false`.
> bool WebSafeBase64Unescape(absl::string_view src, std::string* dest);
> 
> // Base64Escape()
> //
>-// Encodes a `src` std::string into a `dest` buffer using base64 encoding, with
>+// Encodes a `src` string into a `dest` buffer using base64 encoding, with
> // padding characters. This function conforms with RFC 4648 section 4 (base64).
> void Base64Escape(absl::string_view src, std::string* dest);
> 
> // WebSafeBase64Escape()
> //
>-// Encodes a `src` std::string into a `dest` buffer using '-' instead of '+' and
>+// Encodes a `src` string into a `dest` buffer using '-' instead of '+' and
> // '_' instead of '/', and without padding. This function conforms with RFC 4648
> // section 5 (base64url).
> void WebSafeBase64Escape(absl::string_view src, std::string* dest);
> 
> // HexStringToBytes()
> //
>-// Converts an ASCII hex std::string into bytes, returning binary data of length
>+// Converts an ASCII hex string into bytes, returning binary data of length
> // `from.size()/2`.
> std::string HexStringToBytes(absl::string_view from);
> 
> // BytesToHexString()
> //
>-// Converts binary data into an ASCII text std::string, returning a std::string of size
>+// Converts binary data into an ASCII text string, returning a string of size
> // `2*from.size()`.
> std::string BytesToHexString(absl::string_view from);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping_test.cc
>index 3f65ec107f4555c950ff777e89e93d23e43378e5..9dc27f3f8fa768a2a91cdaf95037d74cff01aebc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping_test.cc
>@@ -543,18 +543,19 @@ static struct {
>     {"abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4eXo="},
> };
> 
>-TEST(Base64, EscapeAndUnescape) {
>+template <typename StringType>
>+void TestEscapeAndUnescape() {
>   // Check the short strings; this tests the math (and boundaries)
>   for (const auto& tc : base64_tests) {
>-    std::string encoded("this junk should be ignored");
>+    StringType encoded("this junk should be ignored");
>     absl::Base64Escape(tc.plaintext, &encoded);
>     EXPECT_EQ(encoded, tc.cyphertext);
> 
>-    std::string decoded("this junk should be ignored");
>+    StringType decoded("this junk should be ignored");
>     EXPECT_TRUE(absl::Base64Unescape(encoded, &decoded));
>     EXPECT_EQ(decoded, tc.plaintext);
> 
>-    std::string websafe(tc.cyphertext);
>+    StringType websafe(tc.cyphertext);
>     for (int c = 0; c < websafe.size(); ++c) {
>       if ('+' == websafe[c]) websafe[c] = '-';
>       if ('/' == websafe[c]) websafe[c] = '_';
>@@ -576,7 +577,7 @@ TEST(Base64, EscapeAndUnescape) {
> 
>   // Now try the long strings, this tests the streaming
>   for (const auto& tc : absl::strings_internal::base64_strings()) {
>-    std::string buffer;
>+    StringType buffer;
>     absl::WebSafeBase64Escape(tc.plaintext, &buffer);
>     EXPECT_EQ(tc.cyphertext, buffer);
>   }
>@@ -586,7 +587,7 @@ TEST(Base64, EscapeAndUnescape) {
>     absl::string_view data_set[] = {"ab-/", absl::string_view("\0bcd", 4),
>                                     absl::string_view("abc.\0", 5)};
>     for (absl::string_view bad_data : data_set) {
>-      std::string buf;
>+      StringType buf;
>       EXPECT_FALSE(absl::Base64Unescape(bad_data, &buf));
>       EXPECT_FALSE(absl::WebSafeBase64Unescape(bad_data, &buf));
>       EXPECT_TRUE(buf.empty());
>@@ -594,6 +595,10 @@ TEST(Base64, EscapeAndUnescape) {
>   }
> }
> 
>+TEST(Base64, EscapeAndUnescape) {
>+  TestEscapeAndUnescape<std::string>();
>+}
>+
> TEST(Base64, DISABLED_HugeData) {
>   const size_t kSize = size_t(3) * 1000 * 1000 * 1000;
>   static_assert(kSize % 3 == 0, "kSize must be divisible by 3");
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/bits.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/bits.h
>deleted file mode 100644
>index 901082ccddfca78457c746fbe22ee5709933bfaf..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/bits.h
>+++ /dev/null
>@@ -1,53 +0,0 @@
>-// Copyright 2018 The Abseil Authors.
>-//
>-// Licensed under the Apache License, Version 2.0 (the "License");
>-// you may not use this file except in compliance with the License.
>-// You may obtain a copy of the License at
>-//
>-//      http://www.apache.org/licenses/LICENSE-2.0
>-//
>-// Unless required by applicable law or agreed to in writing, software
>-// distributed under the License is distributed on an "AS IS" BASIS,
>-// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>-// See the License for the specific language governing permissions and
>-// limitations under the License.
>-
>-#ifndef ABSL_STRINGS_INTERNAL_BITS_H_
>-#define ABSL_STRINGS_INTERNAL_BITS_H_
>-
>-#include <cstdint>
>-
>-#if defined(_MSC_VER) && defined(_M_X64)
>-#include <intrin.h>
>-#pragma intrinsic(_BitScanReverse64)
>-#endif
>-
>-namespace absl {
>-namespace strings_internal {
>-
>-// Returns the number of leading 0 bits in a 64-bit value.
>-inline int CountLeadingZeros64(uint64_t n) {
>-#if defined(__GNUC__)
>-  static_assert(sizeof(unsigned long long) == sizeof(n),  // NOLINT(runtime/int)
>-                "__builtin_clzll does not take 64bit arg");
>-  return n == 0 ? 64 : __builtin_clzll(n);
>-#elif defined(_MSC_VER) && defined(_M_X64)
>-  unsigned long result;  // NOLINT(runtime/int)
>-  if (_BitScanReverse64(&result, n)) {
>-    return 63 - result;
>-  }
>-  return 64;
>-#else
>-  int zeroes = 60;
>-  if (n >> 32) zeroes -= 32, n >>= 32;
>-  if (n >> 16) zeroes -= 16, n >>= 16;
>-  if (n >> 8) zeroes -= 8, n >>= 8;
>-  if (n >> 4) zeroes -= 4, n >>= 4;
>-  return "\4\3\2\2\1\1\1\1\0\0\0\0\0\0\0\0"[n] + zeroes;
>-#endif
>-}
>-
>-}  // namespace strings_internal
>-}  // namespace absl
>-
>-#endif  // ABSL_STRINGS_INTERNAL_BITS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse.cc
>index a04cc67669a7f75f936a767c5f223cb3592d9ebb..7e4dabc2262eff7726b72f54803154a042aaa718 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse.cc
>@@ -91,7 +91,7 @@ static_assert(std::numeric_limits<int>::digits10 >= kDecimalExponentDigitsMax,
> 
> // To avoid incredibly large inputs causing integer overflow for our exponent,
> // we impose an arbitrary but very large limit on the number of significant
>-// digits we will accept.  The implementation refuses to match a std::string with
>+// digits we will accept.  The implementation refuses to match a string with
> // more consecutive significant mantissa digits than this.
> constexpr int kDecimalDigitLimit = 50000000;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse_test.cc
>index 1ff86004973a82fcd70e301c808a94349ed3bbb1..f48b9aee1a5e84db3b1cc296aab363f2c33db372 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse_test.cc
>@@ -29,16 +29,16 @@ using absl::strings_internal::ParseFloat;
> 
> namespace {
> 
>-// Check that a given std::string input is parsed to the expected mantissa and
>+// Check that a given string input is parsed to the expected mantissa and
> // exponent.
> //
>-// Input std::string `s` must contain a '$' character.  It marks the end of the
>+// Input string `s` must contain a '$' character.  It marks the end of the
> // characters that should be consumed by the match.  It is stripped from the
> // input to ParseFloat.
> //
>-// If input std::string `s` contains '[' and ']' characters, these mark the region
>+// If input string `s` contains '[' and ']' characters, these mark the region
> // of characters that should be marked as the "subrange".  For NaNs, this is
>-// the location of the extended NaN std::string.  For numbers, this is the location
>+// the location of the extended NaN string.  For numbers, this is the location
> // of the full, over-large mantissa.
> template <int base>
> void ExpectParsedFloat(std::string s, absl::chars_format format_flags,
>@@ -92,10 +92,10 @@ void ExpectParsedFloat(std::string s, absl::chars_format format_flags,
>   EXPECT_EQ(characters_matched, expected_characters_matched);
> }
> 
>-// Check that a given std::string input is parsed to the expected mantissa and
>+// Check that a given string input is parsed to the expected mantissa and
> // exponent.
> //
>-// Input std::string `s` must contain a '$' character.  It marks the end of the
>+// Input string `s` must contain a '$' character.  It marks the end of the
> // characters that were consumed by the match.
> template <int base>
> void ExpectNumber(std::string s, absl::chars_format format_flags,
>@@ -106,7 +106,7 @@ void ExpectNumber(std::string s, absl::chars_format format_flags,
>                           expected_literal_exponent);
> }
> 
>-// Check that a given std::string input is parsed to the given special value.
>+// Check that a given string input is parsed to the given special value.
> //
> // This tests against both number bases, since infinities and NaNs have
> // identical representations in both modes.
>@@ -116,7 +116,7 @@ void ExpectSpecial(const std::string& s, absl::chars_format format_flags,
>   ExpectParsedFloat<16>(s, format_flags, type, 0, 0);
> }
> 
>-// Check that a given input std::string is not matched by Float.
>+// Check that a given input string is not matched by Float.
> template <int base>
> void ExpectFailedParse(absl::string_view s, absl::chars_format format_flags) {
>   ParsedFloat parsed =
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil.h
>index a6f1c69138e3872f127e70aeb26e159a82925a0c..7de383b19ec3fc0f6c9331a41d94173c551aecbd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil.h
>@@ -14,7 +14,7 @@
> // limitations under the License.
> //
> 
>-// These routines provide mem versions of standard C std::string routines,
>+// These routines provide mem versions of standard C string routines,
> // such as strpbrk.  They function exactly the same as the str versions,
> // so if you wonder what they are, replace the word "mem" by
> // "str" and check out the man page.  I could return void*, as the
>@@ -22,14 +22,14 @@
> // since this is by far the most common way these functions are called.
> //
> // The difference between the mem and str versions is the mem version
>-// takes a pointer and a length, rather than a '\0'-terminated std::string.
>+// takes a pointer and a length, rather than a '\0'-terminated string.
> // The memcase* routines defined here assume the locale is "C"
> // (they use absl::ascii_tolower instead of tolower).
> //
> // These routines are based on the BSD library.
> //
>-// Here's a list of routines from std::string.h, and their mem analogues.
>-// Functions in lowercase are defined in std::string.h; those in UPPERCASE
>+// Here's a list of routines from string.h, and their mem analogues.
>+// Functions in lowercase are defined in string.h; those in UPPERCASE
> // are defined here:
> //
> // strlen                  --
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream.h
>index 6e1325b9140f8a0c3eedb6f5a13e21d8d6e0aaf6..e81a89affea25be26bbcd3120cc7094502a2d806 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream.h
>@@ -25,18 +25,18 @@
> namespace absl {
> namespace strings_internal {
> 
>-// The same as std::ostringstream but appends to a user-specified std::string,
>+// The same as std::ostringstream but appends to a user-specified string,
> // and is faster. It is ~70% faster to create, ~50% faster to write to, and
>-// completely free to extract the result std::string.
>+// completely free to extract the result string.
> //
>-//   std::string s;
>+//   string s;
> //   OStringStream strm(&s);
> //   strm << 42 << ' ' << 3.14;  // appends to `s`
> //
> // The stream object doesn't have to be named. Starting from C++11 operator<<
> // works with rvalues of std::ostream.
> //
>-//   std::string s;
>+//   string s;
> //   OStringStream(&s) << 42 << ' ' << 3.14;  // appends to `s`
> //
> // OStringStream is faster to create than std::ostringstream but it's still
>@@ -45,14 +45,14 @@ namespace strings_internal {
> //
> // Creates unnecessary instances of OStringStream: slow.
> //
>-//   std::string s;
>+//   string s;
> //   OStringStream(&s) << 42;
> //   OStringStream(&s) << ' ';
> //   OStringStream(&s) << 3.14;
> //
> // Creates a single instance of OStringStream and reuses it: fast.
> //
>-//   std::string s;
>+//   string s;
> //   OStringStream strm(&s);
> //   strm << 42;
> //   strm << ' ';
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/resize_uninitialized.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/resize_uninitialized.h
>index 0157ca0245f350bece6dcbe1987bdb80b4a54551..a94e0547b50f154234374e8e608042a6c7033bc9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/resize_uninitialized.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/resize_uninitialized.h
>@@ -44,8 +44,8 @@ void ResizeUninit(string_type* s, size_t new_size, std::false_type) {
>   s->resize(new_size);
> }
> 
>-// Returns true if the std::string implementation supports a resize where
>-// the new characters added to the std::string are left untouched.
>+// Returns true if the string implementation supports a resize where
>+// the new characters added to the string are left untouched.
> //
> // (A better name might be "STLStringSupportsUninitializedResize", alluding to
> // the previous function.)
>@@ -57,7 +57,7 @@ inline constexpr bool STLStringSupportsNontrashingResize(string_type*) {
> // Like str->resize(new_size), except any new characters added to "*str" as a
> // result of resizing may be left uninitialized, rather than being filled with
> // '0' bytes. Typically used when code is then going to overwrite the backing
>-// store of the std::string with known data. Uses a Google extension to std::string.
>+// store of the string with known data. Uses a Google extension to ::string.
> template <typename string_type, typename = void>
> inline void STLStringResizeUninitialized(string_type* s, size_t new_size) {
>   ResizeUninit(s, new_size, HasResizeUninitialized<string_type>());
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.cc
>index eafb068fe2867446270f228a6449e0db5a4ccab5..b40be8ff3824e9c66b4f7c1ecc0856e4178fdd3d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.cc
>@@ -112,7 +112,7 @@ class ConvertedIntInfo {
> // Note: 'o' conversions do not have a base indicator, it's just that
> // the '#' flag is specified to modify the precision for 'o' conversions.
> string_view BaseIndicator(const ConvertedIntInfo &info,
>-                          const ConversionSpec &conv) {
>+                          const ConversionSpec conv) {
>   bool alt = conv.flags().alt;
>   int radix = conv.conv().radix();
>   if (conv.conv().id() == ConversionChar::p)
>@@ -127,7 +127,7 @@ string_view BaseIndicator(const ConvertedIntInfo &info,
>   return {};
> }
> 
>-string_view SignColumn(bool neg, const ConversionSpec &conv) {
>+string_view SignColumn(bool neg, const ConversionSpec conv) {
>   if (conv.conv().is_signed()) {
>     if (neg) return "-";
>     if (conv.flags().show_pos) return "+";
>@@ -136,7 +136,7 @@ string_view SignColumn(bool neg, const ConversionSpec &conv) {
>   return {};
> }
> 
>-bool ConvertCharImpl(unsigned char v, const ConversionSpec &conv,
>+bool ConvertCharImpl(unsigned char v, const ConversionSpec conv,
>                      FormatSinkImpl *sink) {
>   size_t fill = 0;
>   if (conv.width() >= 0) fill = conv.width();
>@@ -148,7 +148,7 @@ bool ConvertCharImpl(unsigned char v, const ConversionSpec &conv,
> }
> 
> bool ConvertIntImplInner(const ConvertedIntInfo &info,
>-                         const ConversionSpec &conv, FormatSinkImpl *sink) {
>+                         const ConversionSpec conv, FormatSinkImpl *sink) {
>   // Print as a sequence of Substrings:
>   //   [left_spaces][sign][base_indicator][zeroes][formatted][right_spaces]
>   size_t fill = 0;
>@@ -202,8 +202,7 @@ bool ConvertIntImplInner(const ConvertedIntInfo &info,
> }
> 
> template <typename T>
>-bool ConvertIntImplInner(T v, const ConversionSpec &conv,
>-                         FormatSinkImpl *sink) {
>+bool ConvertIntImplInner(T v, const ConversionSpec conv, FormatSinkImpl *sink) {
>   ConvertedIntInfo info(v, conv.conv());
>   if (conv.flags().basic && conv.conv().id() != ConversionChar::p) {
>     if (info.is_neg()) sink->Append(1, '-');
>@@ -218,7 +217,7 @@ bool ConvertIntImplInner(T v, const ConversionSpec &conv,
> }
> 
> template <typename T>
>-bool ConvertIntArg(T v, const ConversionSpec &conv, FormatSinkImpl *sink) {
>+bool ConvertIntArg(T v, const ConversionSpec conv, FormatSinkImpl *sink) {
>   if (conv.conv().is_float()) {
>     return FormatConvertImpl(static_cast<double>(v), conv, sink).value;
>   }
>@@ -234,11 +233,11 @@ bool ConvertIntArg(T v, const ConversionSpec &conv, FormatSinkImpl *sink) {
> }
> 
> template <typename T>
>-bool ConvertFloatArg(T v, const ConversionSpec &conv, FormatSinkImpl *sink) {
>+bool ConvertFloatArg(T v, const ConversionSpec conv, FormatSinkImpl *sink) {
>   return conv.conv().is_float() && ConvertFloatImpl(v, conv, sink);
> }
> 
>-inline bool ConvertStringArg(string_view v, const ConversionSpec &conv,
>+inline bool ConvertStringArg(string_view v, const ConversionSpec conv,
>                              FormatSinkImpl *sink) {
>   if (conv.conv().id() != ConversionChar::s)
>     return false;
>@@ -254,19 +253,19 @@ inline bool ConvertStringArg(string_view v, const ConversionSpec &conv,
> 
> // ==================== Strings ====================
> ConvertResult<Conv::s> FormatConvertImpl(const std::string &v,
>-                                         const ConversionSpec &conv,
>+                                         const ConversionSpec conv,
>                                          FormatSinkImpl *sink) {
>   return {ConvertStringArg(v, conv, sink)};
> }
> 
> ConvertResult<Conv::s> FormatConvertImpl(string_view v,
>-                                         const ConversionSpec &conv,
>+                                         const ConversionSpec conv,
>                                          FormatSinkImpl *sink) {
>   return {ConvertStringArg(v, conv, sink)};
> }
> 
> ConvertResult<Conv::s | Conv::p> FormatConvertImpl(const char *v,
>-                                                   const ConversionSpec &conv,
>+                                                   const ConversionSpec conv,
>                                                    FormatSinkImpl *sink) {
>   if (conv.conv().id() == ConversionChar::p)
>     return {FormatConvertImpl(VoidPtr(v), conv, sink).value};
>@@ -283,7 +282,7 @@ ConvertResult<Conv::s | Conv::p> FormatConvertImpl(const char *v,
> }
> 
> // ==================== Raw pointers ====================
>-ConvertResult<Conv::p> FormatConvertImpl(VoidPtr v, const ConversionSpec &conv,
>+ConvertResult<Conv::p> FormatConvertImpl(VoidPtr v, const ConversionSpec conv,
>                                          FormatSinkImpl *sink) {
>   if (conv.conv().id() != ConversionChar::p)
>     return {false};
>@@ -295,104 +294,83 @@ ConvertResult<Conv::p> FormatConvertImpl(VoidPtr v, const ConversionSpec &conv,
> }
> 
> // ==================== Floats ====================
>-FloatingConvertResult FormatConvertImpl(float v, const ConversionSpec &conv,
>+FloatingConvertResult FormatConvertImpl(float v, const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertFloatArg(v, conv, sink)};
> }
>-FloatingConvertResult FormatConvertImpl(double v, const ConversionSpec &conv,
>+FloatingConvertResult FormatConvertImpl(double v, const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertFloatArg(v, conv, sink)};
> }
> FloatingConvertResult FormatConvertImpl(long double v,
>-                                        const ConversionSpec &conv,
>+                                        const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertFloatArg(v, conv, sink)};
> }
> 
> // ==================== Chars ====================
>-IntegralConvertResult FormatConvertImpl(char v, const ConversionSpec &conv,
>+IntegralConvertResult FormatConvertImpl(char v, const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
> IntegralConvertResult FormatConvertImpl(signed char v,
>-                                        const ConversionSpec &conv,
>+                                        const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
> IntegralConvertResult FormatConvertImpl(unsigned char v,
>-                                        const ConversionSpec &conv,
>+                                        const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
> 
> // ==================== Ints ====================
> IntegralConvertResult FormatConvertImpl(short v,  // NOLINT
>-                                        const ConversionSpec &conv,
>+                                        const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
> IntegralConvertResult FormatConvertImpl(unsigned short v,  // NOLINT
>-                                        const ConversionSpec &conv,
>+                                        const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
>-IntegralConvertResult FormatConvertImpl(int v, const ConversionSpec &conv,
>+IntegralConvertResult FormatConvertImpl(int v, const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
>-IntegralConvertResult FormatConvertImpl(unsigned v, const ConversionSpec &conv,
>+IntegralConvertResult FormatConvertImpl(unsigned v, const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
> IntegralConvertResult FormatConvertImpl(long v,  // NOLINT
>-                                        const ConversionSpec &conv,
>+                                        const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
> IntegralConvertResult FormatConvertImpl(unsigned long v,  // NOLINT
>-                                        const ConversionSpec &conv,
>+                                        const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
> IntegralConvertResult FormatConvertImpl(long long v,  // NOLINT
>-                                        const ConversionSpec &conv,
>+                                        const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
> IntegralConvertResult FormatConvertImpl(unsigned long long v,  // NOLINT
>-                                        const ConversionSpec &conv,
>+                                        const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
> IntegralConvertResult FormatConvertImpl(absl::uint128 v,
>-                                        const ConversionSpec &conv,
>+                                        const ConversionSpec conv,
>                                         FormatSinkImpl *sink) {
>   return {ConvertIntArg(v, conv, sink)};
> }
> 
>-template struct FormatArgImpl::TypedVTable<str_format_internal::VoidPtr>;
>-
>-template struct FormatArgImpl::TypedVTable<bool>;
>-template struct FormatArgImpl::TypedVTable<char>;
>-template struct FormatArgImpl::TypedVTable<signed char>;
>-template struct FormatArgImpl::TypedVTable<unsigned char>;
>-template struct FormatArgImpl::TypedVTable<short>;           // NOLINT
>-template struct FormatArgImpl::TypedVTable<unsigned short>;  // NOLINT
>-template struct FormatArgImpl::TypedVTable<int>;
>-template struct FormatArgImpl::TypedVTable<unsigned>;
>-template struct FormatArgImpl::TypedVTable<long>;                // NOLINT
>-template struct FormatArgImpl::TypedVTable<unsigned long>;       // NOLINT
>-template struct FormatArgImpl::TypedVTable<long long>;           // NOLINT
>-template struct FormatArgImpl::TypedVTable<unsigned long long>;  // NOLINT
>-template struct FormatArgImpl::TypedVTable<absl::uint128>;
>-
>-template struct FormatArgImpl::TypedVTable<float>;
>-template struct FormatArgImpl::TypedVTable<double>;
>-template struct FormatArgImpl::TypedVTable<long double>;
>-
>-template struct FormatArgImpl::TypedVTable<const char *>;
>-template struct FormatArgImpl::TypedVTable<std::string>;
>-template struct FormatArgImpl::TypedVTable<string_view>;
>+ABSL_INTERNAL_FORMAT_DISPATCH_OVERLOADS_EXPAND_();
>+
> 
> }  // namespace str_format_internal
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.h
>index a9562188ea91973a629828e0dec82bd470c9c07a..3376d48afda20ff233a1d1b7f41b30078ae1cce6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.h
>@@ -33,7 +33,7 @@ struct HasUserDefinedConvert : std::false_type {};
> template <typename T>
> struct HasUserDefinedConvert<
>     T, void_t<decltype(AbslFormatConvert(
>-           std::declval<const T&>(), std::declval<const ConversionSpec&>(),
>+           std::declval<const T&>(), std::declval<ConversionSpec>(),
>            std::declval<FormatSink*>()))>> : std::true_type {};
> template <typename T>
> class StreamedWrapper;
>@@ -50,25 +50,23 @@ struct VoidPtr {
>       : value(ptr ? reinterpret_cast<uintptr_t>(ptr) : 0) {}
>   uintptr_t value;
> };
>-ConvertResult<Conv::p> FormatConvertImpl(VoidPtr v, const ConversionSpec& conv,
>+ConvertResult<Conv::p> FormatConvertImpl(VoidPtr v, ConversionSpec conv,
>                                          FormatSinkImpl* sink);
> 
> // Strings.
>-ConvertResult<Conv::s> FormatConvertImpl(const std::string& v,
>-                                         const ConversionSpec& conv,
>+ConvertResult<Conv::s> FormatConvertImpl(const std::string& v, ConversionSpec conv,
>                                          FormatSinkImpl* sink);
>-ConvertResult<Conv::s> FormatConvertImpl(string_view v,
>-                                         const ConversionSpec& conv,
>+ConvertResult<Conv::s> FormatConvertImpl(string_view v, ConversionSpec conv,
>                                          FormatSinkImpl* sink);
> ConvertResult<Conv::s | Conv::p> FormatConvertImpl(const char* v,
>-                                                   const ConversionSpec& conv,
>+                                                   ConversionSpec conv,
>                                                    FormatSinkImpl* sink);
> template <class AbslCord,
>           typename std::enable_if<
>               std::is_same<AbslCord, ::Cord>::value>::type* = nullptr,
>           class AbslCordReader = ::CordReader>
> ConvertResult<Conv::s> FormatConvertImpl(const AbslCord& value,
>-                                         const ConversionSpec& conv,
>+                                         ConversionSpec conv,
>                                          FormatSinkImpl* sink) {
>   if (conv.conv().id() != ConversionChar::s) return {false};
> 
>@@ -104,51 +102,48 @@ using IntegralConvertResult =
> using FloatingConvertResult = ConvertResult<Conv::floating>;
> 
> // Floats.
>-FloatingConvertResult FormatConvertImpl(float v, const ConversionSpec& conv,
>+FloatingConvertResult FormatConvertImpl(float v, ConversionSpec conv,
>                                         FormatSinkImpl* sink);
>-FloatingConvertResult FormatConvertImpl(double v, const ConversionSpec& conv,
>+FloatingConvertResult FormatConvertImpl(double v, ConversionSpec conv,
>                                         FormatSinkImpl* sink);
>-FloatingConvertResult FormatConvertImpl(long double v,
>-                                        const ConversionSpec& conv,
>+FloatingConvertResult FormatConvertImpl(long double v, ConversionSpec conv,
>                                         FormatSinkImpl* sink);
> 
> // Chars.
>-IntegralConvertResult FormatConvertImpl(char v, const ConversionSpec& conv,
>+IntegralConvertResult FormatConvertImpl(char v, ConversionSpec conv,
>                                         FormatSinkImpl* sink);
>-IntegralConvertResult FormatConvertImpl(signed char v,
>-                                        const ConversionSpec& conv,
>+IntegralConvertResult FormatConvertImpl(signed char v, ConversionSpec conv,
>                                         FormatSinkImpl* sink);
>-IntegralConvertResult FormatConvertImpl(unsigned char v,
>-                                        const ConversionSpec& conv,
>+IntegralConvertResult FormatConvertImpl(unsigned char v, ConversionSpec conv,
>                                         FormatSinkImpl* sink);
> 
> // Ints.
> IntegralConvertResult FormatConvertImpl(short v,  // NOLINT
>-                                        const ConversionSpec& conv,
>+                                        ConversionSpec conv,
>                                         FormatSinkImpl* sink);
> IntegralConvertResult FormatConvertImpl(unsigned short v,  // NOLINT
>-                                        const ConversionSpec& conv,
>+                                        ConversionSpec conv,
>                                         FormatSinkImpl* sink);
>-IntegralConvertResult FormatConvertImpl(int v, const ConversionSpec& conv,
>+IntegralConvertResult FormatConvertImpl(int v, ConversionSpec conv,
>                                         FormatSinkImpl* sink);
>-IntegralConvertResult FormatConvertImpl(unsigned v, const ConversionSpec& conv,
>+IntegralConvertResult FormatConvertImpl(unsigned v, ConversionSpec conv,
>                                         FormatSinkImpl* sink);
> IntegralConvertResult FormatConvertImpl(long v,  // NOLINT
>-                                        const ConversionSpec& conv,
>+                                        ConversionSpec conv,
>                                         FormatSinkImpl* sink);
> IntegralConvertResult FormatConvertImpl(unsigned long v,  // NOLINT
>-                                        const ConversionSpec& conv,
>+                                        ConversionSpec conv,
>                                         FormatSinkImpl* sink);
> IntegralConvertResult FormatConvertImpl(long long v,  // NOLINT
>-                                        const ConversionSpec& conv,
>+                                        ConversionSpec conv,
>                                         FormatSinkImpl* sink);
> IntegralConvertResult FormatConvertImpl(unsigned long long v,  // NOLINT
>-                                        const ConversionSpec& conv,
>+                                        ConversionSpec conv,
>                                         FormatSinkImpl* sink);
>-IntegralConvertResult FormatConvertImpl(uint128 v, const ConversionSpec& conv,
>+IntegralConvertResult FormatConvertImpl(uint128 v, ConversionSpec conv,
>                                         FormatSinkImpl* sink);
> template <typename T, enable_if_t<std::is_same<T, bool>::value, int> = 0>
>-IntegralConvertResult FormatConvertImpl(T v, const ConversionSpec& conv,
>+IntegralConvertResult FormatConvertImpl(T v, ConversionSpec conv,
>                                         FormatSinkImpl* sink) {
>   return FormatConvertImpl(static_cast<int>(v), conv, sink);
> }
>@@ -159,11 +154,11 @@ template <typename T>
> typename std::enable_if<std::is_enum<T>::value &&
>                             !HasUserDefinedConvert<T>::value,
>                         IntegralConvertResult>::type
>-FormatConvertImpl(T v, const ConversionSpec& conv, FormatSinkImpl* sink);
>+FormatConvertImpl(T v, ConversionSpec conv, FormatSinkImpl* sink);
> 
> template <typename T>
> ConvertResult<Conv::s> FormatConvertImpl(const StreamedWrapper<T>& v,
>-                                         const ConversionSpec& conv,
>+                                         ConversionSpec conv,
>                                          FormatSinkImpl* out) {
>   std::ostringstream oss;
>   oss << v.v_;
>@@ -176,7 +171,7 @@ ConvertResult<Conv::s> FormatConvertImpl(const StreamedWrapper<T>& v,
> struct FormatCountCaptureHelper {
>   template <class T = int>
>   static ConvertResult<Conv::n> ConvertHelper(const FormatCountCapture& v,
>-                                              const ConversionSpec& conv,
>+                                              ConversionSpec conv,
>                                               FormatSinkImpl* sink) {
>     const absl::enable_if_t<sizeof(T) != 0, FormatCountCapture>& v2 = v;
> 
>@@ -189,7 +184,7 @@ struct FormatCountCaptureHelper {
> 
> template <class T = int>
> ConvertResult<Conv::n> FormatConvertImpl(const FormatCountCapture& v,
>-                                         const ConversionSpec& conv,
>+                                         ConversionSpec conv,
>                                          FormatSinkImpl* sink) {
>   return FormatCountCaptureHelper::ConvertHelper(v, conv, sink);
> }
>@@ -199,20 +194,20 @@ ConvertResult<Conv::n> FormatConvertImpl(const FormatCountCapture& v,
> struct FormatArgImplFriend {
>   template <typename Arg>
>   static bool ToInt(Arg arg, int* out) {
>-    if (!arg.vtbl_->to_int) return false;
>-    *out = arg.vtbl_->to_int(arg.data_);
>-    return true;
>+    // A value initialized ConversionSpec has a `none` conv, which tells the
>+    // dispatcher to run the `int` conversion.
>+    return arg.dispatcher_(arg.data_, {}, out);
>   }
> 
>   template <typename Arg>
>-  static bool Convert(Arg arg, const str_format_internal::ConversionSpec& conv,
>+  static bool Convert(Arg arg, str_format_internal::ConversionSpec conv,
>                       FormatSinkImpl* out) {
>-    return arg.vtbl_->convert(arg.data_, conv, out);
>+    return arg.dispatcher_(arg.data_, conv, out);
>   }
> 
>   template <typename Arg>
>-  static const void* GetVTablePtrForTest(Arg arg) {
>-    return arg.vtbl_;
>+  static typename Arg::Dispatcher GetVTablePtrForTest(Arg arg) {
>+    return arg.dispatcher_;
>   }
> };
> 
>@@ -229,11 +224,7 @@ class FormatArgImpl {
>     char buf[kInlinedSpace];
>   };
> 
>-  struct VTable {
>-    bool (*convert)(Data, const str_format_internal::ConversionSpec& conv,
>-                    FormatSinkImpl* out);
>-    int (*to_int)(Data);
>-  };
>+  using Dispatcher = bool (*)(Data, ConversionSpec, void* out);
> 
>   template <typename T>
>   struct store_by_value
>@@ -253,10 +244,6 @@ class FormatArgImpl {
>                                                                 : ByPointer))> {
>   };
> 
>-  // An instance of an FormatArgImpl::VTable suitable for 'T'.
>-  template <typename T>
>-  struct TypedVTable;
>-
>   // To reduce the number of vtables we will decay values before hand.
>   // Anything with a user-defined Convert will get its own vtable.
>   // For everything else:
>@@ -338,7 +325,10 @@ class FormatArgImpl {
>   };
> 
>   template <typename T>
>-  void Init(const T& value);
>+  void Init(const T& value) {
>+    data_ = Manager<T>::SetValue(value);
>+    dispatcher_ = &Dispatch<T>;
>+  }
> 
>   template <typename T>
>   static int ToIntVal(const T& val) {
>@@ -355,79 +345,75 @@ class FormatArgImpl {
>     return static_cast<int>(val);
>   }
> 
>-  Data data_;
>-  const VTable* vtbl_;
>-};
>-
>-template <typename T>
>-struct FormatArgImpl::TypedVTable {
>- private:
>-  static bool ConvertImpl(Data arg,
>-                          const str_format_internal::ConversionSpec& conv,
>-                          FormatSinkImpl* out) {
>-    return str_format_internal::FormatConvertImpl(Manager<T>::Value(arg), conv,
>-                                                  out)
>-        .value;
>+  template <typename T>
>+  static bool ToInt(Data arg, int* out, std::true_type /* is_integral */,
>+                    std::false_type) {
>+    *out = ToIntVal(Manager<T>::Value(arg));
>+    return true;
>   }
> 
>-  template <typename U = T, typename = void>
>-  struct ToIntImpl {
>-    static constexpr int (*value)(Data) = nullptr;
>-  };
>+  template <typename T>
>+  static bool ToInt(Data arg, int* out, std::false_type,
>+                    std::true_type /* is_enum */) {
>+    *out = ToIntVal(static_cast<typename std::underlying_type<T>::type>(
>+        Manager<T>::Value(arg)));
>+    return true;
>+  }
> 
>-  template <typename U>
>-  struct ToIntImpl<U,
>-                   typename std::enable_if<std::is_integral<U>::value>::type> {
>-    static int Invoke(Data arg) { return ToIntVal(Manager<T>::Value(arg)); }
>-    static constexpr int (*value)(Data) = &Invoke;
>-  };
>+  template <typename T>
>+  static bool ToInt(Data, int*, std::false_type, std::false_type) {
>+    return false;
>+  }
> 
>-  template <typename U>
>-  struct ToIntImpl<U, typename std::enable_if<std::is_enum<U>::value>::type> {
>-    static int Invoke(Data arg) {
>-      return ToIntVal(static_cast<typename std::underlying_type<T>::type>(
>-          Manager<T>::Value(arg)));
>+  template <typename T>
>+  static bool Dispatch(Data arg, ConversionSpec spec, void* out) {
>+    // A `none` conv indicates that we want the `int` conversion.
>+    if (ABSL_PREDICT_FALSE(spec.conv().id() == ConversionChar::none)) {
>+      return ToInt<T>(arg, static_cast<int*>(out), std::is_integral<T>(),
>+                      std::is_enum<T>());
>     }
>-    static constexpr int (*value)(Data) = &Invoke;
>-  };
> 
>- public:
>-  static constexpr VTable value{&ConvertImpl, ToIntImpl<>::value};
>-};
>+    return str_format_internal::FormatConvertImpl(
>+               Manager<T>::Value(arg), spec, static_cast<FormatSinkImpl*>(out))
>+        .value;
>+  }
> 
>-template <typename T>
>-constexpr FormatArgImpl::VTable FormatArgImpl::TypedVTable<T>::value;
>+  Data data_;
>+  Dispatcher dispatcher_;
>+};
> 
>-template <typename T>
>-void FormatArgImpl::Init(const T& value) {
>-  data_ = Manager<T>::SetValue(value);
>-  vtbl_ = &TypedVTable<T>::value;
>-}
>+#define ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(T, E) \
>+  E template bool FormatArgImpl::Dispatch<T>(Data, ConversionSpec, void*)
>+
>+#define ABSL_INTERNAL_FORMAT_DISPATCH_OVERLOADS_EXPAND_(...)                   \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(str_format_internal::VoidPtr,     \
>+                                             __VA_ARGS__);                     \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(bool, __VA_ARGS__);               \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(char, __VA_ARGS__);               \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(signed char, __VA_ARGS__);        \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(unsigned char, __VA_ARGS__);      \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(short, __VA_ARGS__); /* NOLINT */ \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(unsigned short,      /* NOLINT */ \
>+                                             __VA_ARGS__);                     \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(int, __VA_ARGS__);                \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(unsigned int, __VA_ARGS__);       \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(long, __VA_ARGS__); /* NOLINT */  \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(unsigned long,      /* NOLINT */  \
>+                                             __VA_ARGS__);                     \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(long long, /* NOLINT */           \
>+                                             __VA_ARGS__);                     \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(unsigned long long, /* NOLINT */  \
>+                                             __VA_ARGS__);                     \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(uint128, __VA_ARGS__);            \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(float, __VA_ARGS__);              \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(double, __VA_ARGS__);             \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(long double, __VA_ARGS__);        \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(const char*, __VA_ARGS__);        \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(std::string, __VA_ARGS__);             \
>+  ABSL_INTERNAL_FORMAT_DISPATCH_INSTANTIATE_(string_view, __VA_ARGS__)
>+
>+ABSL_INTERNAL_FORMAT_DISPATCH_OVERLOADS_EXPAND_(extern);
> 
>-extern template struct FormatArgImpl::TypedVTable<str_format_internal::VoidPtr>;
>-
>-extern template struct FormatArgImpl::TypedVTable<bool>;
>-extern template struct FormatArgImpl::TypedVTable<char>;
>-extern template struct FormatArgImpl::TypedVTable<signed char>;
>-extern template struct FormatArgImpl::TypedVTable<unsigned char>;
>-extern template struct FormatArgImpl::TypedVTable<short>;           // NOLINT
>-extern template struct FormatArgImpl::TypedVTable<unsigned short>;  // NOLINT
>-extern template struct FormatArgImpl::TypedVTable<int>;
>-extern template struct FormatArgImpl::TypedVTable<unsigned>;
>-extern template struct FormatArgImpl::TypedVTable<long>;           // NOLINT
>-extern template struct FormatArgImpl::TypedVTable<unsigned long>;  // NOLINT
>-extern template struct FormatArgImpl::TypedVTable<long long>;      // NOLINT
>-extern template struct FormatArgImpl::TypedVTable<
>-    unsigned long long>;  // NOLINT
>-extern template struct FormatArgImpl::TypedVTable<uint128>;
>-
>-extern template struct FormatArgImpl::TypedVTable<float>;
>-extern template struct FormatArgImpl::TypedVTable<double>;
>-extern template struct FormatArgImpl::TypedVTable<long double>;
>-
>-extern template struct FormatArgImpl::TypedVTable<const char*>;
>-extern template struct FormatArgImpl::TypedVTable<std::string>;
>-extern template struct FormatArgImpl::TypedVTable<string_view>;
> }  // namespace str_format_internal
> }  // namespace absl
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.cc
>index 33e8641558ab86f3e892addeecd9a6def8b40a97..c4eddd17ddbe56a76c5ec1a5184079bf107166bc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.cc
>@@ -103,15 +103,15 @@ class ConverterConsumer {
> };
> 
> template <typename Converter>
>-bool ConvertAll(const UntypedFormatSpecImpl& format,
>-                absl::Span<const FormatArgImpl> args,
>-                const Converter& converter) {
>-  const ParsedFormatBase* pc = format.parsed_conversion();
>-  if (pc)
>-    return pc->ProcessFormat(ConverterConsumer<Converter>(converter, args));
>-
>-  return ParseFormatString(format.str(),
>-                           ConverterConsumer<Converter>(converter, args));
>+bool ConvertAll(const UntypedFormatSpecImpl format,
>+                absl::Span<const FormatArgImpl> args, Converter converter) {
>+  if (format.has_parsed_conversion()) {
>+    return format.parsed_conversion()->ProcessFormat(
>+        ConverterConsumer<Converter>(converter, args));
>+  } else {
>+    return ParseFormatString(format.str(),
>+                             ConverterConsumer<Converter>(converter, args));
>+  }
> }
> 
> class DefaultConverter {
>@@ -158,7 +158,7 @@ bool BindWithPack(const UnboundConversion* props,
>   return ArgContext(pack).Bind(props, bound);
> }
> 
>-std::string Summarize(const UntypedFormatSpecImpl& format,
>+std::string Summarize(const UntypedFormatSpecImpl format,
>                  absl::Span<const FormatArgImpl> args) {
>   typedef SummarizingConverter Converter;
>   std::string out;
>@@ -167,23 +167,18 @@ std::string Summarize(const UntypedFormatSpecImpl& format,
>     // flush.
>     FormatSinkImpl sink(&out);
>     if (!ConvertAll(format, args, Converter(&sink))) {
>-      sink.Flush();
>-      out.clear();
>+      return "";
>     }
>   }
>   return out;
> }
> 
> bool FormatUntyped(FormatRawSinkImpl raw_sink,
>-                   const UntypedFormatSpecImpl& format,
>+                   const UntypedFormatSpecImpl format,
>                    absl::Span<const FormatArgImpl> args) {
>   FormatSinkImpl sink(raw_sink);
>   using Converter = DefaultConverter;
>-  if (!ConvertAll(format, args, Converter(&sink))) {
>-    sink.Flush();
>-    return false;
>-  }
>-  return true;
>+  return ConvertAll(format, args, Converter(&sink));
> }
> 
> std::ostream& Streamable::Print(std::ostream& os) const {
>@@ -191,14 +186,16 @@ std::ostream& Streamable::Print(std::ostream& os) const {
>   return os;
> }
> 
>-std::string& AppendPack(std::string* out, const UntypedFormatSpecImpl& format,
>+std::string& AppendPack(std::string* out, const UntypedFormatSpecImpl format,
>                    absl::Span<const FormatArgImpl> args) {
>   size_t orig = out->size();
>-  if (!FormatUntyped(out, format, args)) out->resize(orig);
>+  if (ABSL_PREDICT_FALSE(!FormatUntyped(out, format, args))) {
>+    out->erase(orig);
>+  }
>   return *out;
> }
> 
>-int FprintF(std::FILE* output, const UntypedFormatSpecImpl& format,
>+int FprintF(std::FILE* output, const UntypedFormatSpecImpl format,
>             absl::Span<const FormatArgImpl> args) {
>   FILERawSink sink(output);
>   if (!FormatUntyped(&sink, format, args)) {
>@@ -216,7 +213,7 @@ int FprintF(std::FILE* output, const UntypedFormatSpecImpl& format,
>   return static_cast<int>(sink.count());
> }
> 
>-int SnprintF(char* output, size_t size, const UntypedFormatSpecImpl& format,
>+int SnprintF(char* output, size_t size, const UntypedFormatSpecImpl format,
>              absl::Span<const FormatArgImpl> args) {
>   BufferRawSink sink(output, size ? size - 1 : 0);
>   if (!FormatUntyped(&sink, format, args)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.h
>index 4008611211cfe6053c51af0648a051354bdf81a2..1b52df9c7f5655f1d78dadc50825c4d23e4260f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.h
>@@ -33,13 +33,21 @@ class UntypedFormatSpecImpl {
>  public:
>   UntypedFormatSpecImpl() = delete;
> 
>-  explicit UntypedFormatSpecImpl(string_view s) : str_(s), pc_() {}
>+  explicit UntypedFormatSpecImpl(string_view s)
>+      : data_(s.data()), size_(s.size()) {}
>   explicit UntypedFormatSpecImpl(
>       const str_format_internal::ParsedFormatBase* pc)
>-      : pc_(pc) {}
>-  string_view str() const { return str_; }
>+      : data_(pc), size_(~size_t{}) {}
>+
>+  bool has_parsed_conversion() const { return size_ == ~size_t{}; }
>+
>+  string_view str() const {
>+    assert(!has_parsed_conversion());
>+    return string_view(static_cast<const char*>(data_), size_);
>+  }
>   const str_format_internal::ParsedFormatBase* parsed_conversion() const {
>-    return pc_;
>+    assert(has_parsed_conversion());
>+    return static_cast<const str_format_internal::ParsedFormatBase*>(data_);
>   }
> 
>   template <typename T>
>@@ -48,8 +56,8 @@ class UntypedFormatSpecImpl {
>   }
> 
>  private:
>-  string_view str_;
>-  const str_format_internal::ParsedFormatBase* pc_;
>+  const void* data_;
>+  size_t size_;
> };
> 
> template <typename T, typename...>
>@@ -144,31 +152,31 @@ class Streamable {
> };
> 
> // for testing
>-std::string Summarize(const UntypedFormatSpecImpl& format,
>+std::string Summarize(UntypedFormatSpecImpl format,
>                  absl::Span<const FormatArgImpl> args);
> bool BindWithPack(const UnboundConversion* props,
>                   absl::Span<const FormatArgImpl> pack, BoundConversion* bound);
> 
> bool FormatUntyped(FormatRawSinkImpl raw_sink,
>-                   const UntypedFormatSpecImpl& format,
>+                   UntypedFormatSpecImpl format,
>                    absl::Span<const FormatArgImpl> args);
> 
>-std::string& AppendPack(std::string* out, const UntypedFormatSpecImpl& format,
>+std::string& AppendPack(std::string* out, UntypedFormatSpecImpl format,
>                    absl::Span<const FormatArgImpl> args);
> 
>-inline std::string FormatPack(const UntypedFormatSpecImpl& format,
>+inline std::string FormatPack(const UntypedFormatSpecImpl format,
>                          absl::Span<const FormatArgImpl> args) {
>   std::string out;
>   AppendPack(&out, format, args);
>   return out;
> }
> 
>-int FprintF(std::FILE* output, const UntypedFormatSpecImpl& format,
>+int FprintF(std::FILE* output, UntypedFormatSpecImpl format,
>             absl::Span<const FormatArgImpl> args);
>-int SnprintF(char* output, size_t size, const UntypedFormatSpecImpl& format,
>+int SnprintF(char* output, size_t size, UntypedFormatSpecImpl format,
>              absl::Span<const FormatArgImpl> args);
> 
>-// Returned by Streamed(v). Converts via '%s' to the std::string created
>+// Returned by Streamed(v). Converts via '%s' to the string created
> // by std::ostream << v.
> template <typename T>
> class StreamedWrapper {
>@@ -178,7 +186,7 @@ class StreamedWrapper {
>  private:
>   template <typename S>
>   friend ConvertResult<Conv::s> FormatConvertImpl(const StreamedWrapper<S>& v,
>-                                                  const ConversionSpec& conv,
>+                                                  ConversionSpec conv,
>                                                   FormatSinkImpl* out);
>   const T& v_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension.h
>index 810330b9d71bd188c26eae215abd61d48ed4bd78..11b996aefd37d157a3f08daa5a22dc01c0a2d023 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension.h
>@@ -18,6 +18,7 @@
> #define ABSL_STRINGS_INTERNAL_STR_FORMAT_EXTENSION_H_
> 
> #include <limits.h>
>+#include <cstddef>
> #include <cstring>
> #include <ostream>
> 
>@@ -57,7 +58,7 @@ class FormatRawSinkImpl {
>   void (*write_)(void*, string_view);
> };
> 
>-// An abstraction to which conversions write their std::string data.
>+// An abstraction to which conversions write their string data.
> class FormatSinkImpl {
>  public:
>   explicit FormatSinkImpl(FormatRawSinkImpl raw) : raw_(raw) {}
>@@ -307,7 +308,12 @@ class ConversionSpec {
>  public:
>   Flags flags() const { return flags_; }
>   LengthMod length_mod() const { return length_mod_; }
>-  ConversionChar conv() const { return conv_; }
>+  ConversionChar conv() const {
>+    // Keep this field first in the struct . It generates better code when
>+    // accessing it when ConversionSpec is passed by value in registers.
>+    static_assert(offsetof(ConversionSpec, conv_) == 0, "");
>+    return conv_;
>+  }
> 
>   // Returns the specified width. If width is unspecfied, it returns a negative
>   // value.
>@@ -324,9 +330,9 @@ class ConversionSpec {
>   void set_left(bool b) { flags_.left = b; }
> 
>  private:
>+  ConversionChar conv_;
>   Flags flags_;
>   LengthMod length_mod_;
>-  ConversionChar conv_;
>   int width_;
>   int precision_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output.h
>index 3b0aa5e7157ecbdaf8f243e44a13a751638b575d..12ecd99e680d1ebd679d3c882f7f26f3e9fb6e46 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output.h
>@@ -69,10 +69,10 @@ class FILERawSink {
> 
> // Provide RawSink integration with common types from the STL.
> inline void AbslFormatFlush(std::string* out, string_view s) {
>-  out->append(s.begin(), s.size());
>+  out->append(s.data(), s.size());
> }
> inline void AbslFormatFlush(std::ostream* out, string_view s) {
>-  out->write(s.begin(), s.size());
>+  out->write(s.data(), s.size());
> }
> 
> template <class AbslCord, typename = typename std::enable_if<
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.cc
>index 10114f489c0189f0586149685378e13a8df94684..5e3d0d0547d1dbad47b3a503b685076374373320 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.cc
>@@ -81,19 +81,27 @@ static constexpr std::int8_t kIds[] = {
> template <bool is_positional>
> bool ConsumeConversion(string_view *src, UnboundConversion *conv,
>                        int *next_arg) {
>-  const char *pos = src->begin();
>-  const char *const end = src->end();
>+  const char *pos = src->data();
>+  const char *const end = pos + src->size();
>   char c;
>-  // Read the next char into `c` and update `pos`. Reads '\0' if at end.
>-  const auto get_char = [&] { c = pos == end ? '\0' : *pos++; };
>+  // Read the next char into `c` and update `pos`. Returns false if there are
>+  // no more chars to read.
>+#define ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR()        \
>+  do {                                                \
>+    if (ABSL_PREDICT_FALSE(pos == end)) return false; \
>+    c = *pos++;                                       \
>+  } while (0)
> 
>   const auto parse_digits = [&] {
>     int digits = c - '0';
>-    // We do not want to overflow `digits` so we consume at most digits10-1
>+    // We do not want to overflow `digits` so we consume at most digits10
>     // digits. If there are more digits the parsing will fail later on when the
>     // digit doesn't match the expected characters.
>-    int num_digits = std::numeric_limits<int>::digits10 - 2;
>-    for (get_char(); num_digits && std::isdigit(c); get_char()) {
>+    int num_digits = std::numeric_limits<int>::digits10;
>+    for (;;) {
>+      if (ABSL_PREDICT_FALSE(pos == end || !num_digits)) break;
>+      c = *pos++;
>+      if (!std::isdigit(c)) break;
>       --num_digits;
>       digits = 10 * digits + c - '0';
>     }
>@@ -101,14 +109,14 @@ bool ConsumeConversion(string_view *src, UnboundConversion *conv,
>   };
> 
>   if (is_positional) {
>-    get_char();
>-    if (c < '1' || c > '9') return false;
>+    ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR();
>+    if (ABSL_PREDICT_FALSE(c < '1' || c > '9')) return false;
>     conv->arg_position = parse_digits();
>     assert(conv->arg_position > 0);
>-    if (c != '$') return false;
>+    if (ABSL_PREDICT_FALSE(c != '$')) return false;
>   }
> 
>-  get_char();
>+  ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR();
> 
>   // We should start with the basic flag on.
>   assert(conv->flags.basic);
>@@ -119,32 +127,39 @@ bool ConsumeConversion(string_view *src, UnboundConversion *conv,
>   if (c < 'A') {
>     conv->flags.basic = false;
> 
>-    for (; c <= '0'; get_char()) {
>+    for (; c <= '0';) {
>+      // FIXME: We might be able to speed this up reusing the kIds lookup table
>+      // from above.
>+      // It might require changing Flags to be a plain integer where we can |= a
>+      // value.
>       switch (c) {
>         case '-':
>           conv->flags.left = true;
>-          continue;
>+          break;
>         case '+':
>           conv->flags.show_pos = true;
>-          continue;
>+          break;
>         case ' ':
>           conv->flags.sign_col = true;
>-          continue;
>+          break;
>         case '#':
>           conv->flags.alt = true;
>-          continue;
>+          break;
>         case '0':
>           conv->flags.zero = true;
>-          continue;
>+          break;
>+        default:
>+          goto flags_done;
>       }
>-      break;
>+      ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR();
>     }
>+flags_done:
> 
>     if (c <= '9') {
>       if (c >= '0') {
>         int maybe_width = parse_digits();
>         if (!is_positional && c == '$') {
>-          if (*next_arg != 0) return false;
>+          if (ABSL_PREDICT_FALSE(*next_arg != 0)) return false;
>           // Positional conversion.
>           *next_arg = -1;
>           conv->flags = Flags();
>@@ -153,12 +168,12 @@ bool ConsumeConversion(string_view *src, UnboundConversion *conv,
>         }
>         conv->width.set_value(maybe_width);
>       } else if (c == '*') {
>-        get_char();
>+        ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR();
>         if (is_positional) {
>-          if (c < '1' || c > '9') return false;
>+          if (ABSL_PREDICT_FALSE(c < '1' || c > '9')) return false;
>           conv->width.set_from_arg(parse_digits());
>-          if (c != '$') return false;
>-          get_char();
>+          if (ABSL_PREDICT_FALSE(c != '$')) return false;
>+          ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR();
>         } else {
>           conv->width.set_from_arg(++*next_arg);
>         }
>@@ -166,16 +181,16 @@ bool ConsumeConversion(string_view *src, UnboundConversion *conv,
>     }
> 
>     if (c == '.') {
>-      get_char();
>+      ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR();
>       if (std::isdigit(c)) {
>         conv->precision.set_value(parse_digits());
>       } else if (c == '*') {
>-        get_char();
>+        ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR();
>         if (is_positional) {
>-          if (c < '1' || c > '9') return false;
>+          if (ABSL_PREDICT_FALSE(c < '1' || c > '9')) return false;
>           conv->precision.set_from_arg(parse_digits());
>           if (c != '$') return false;
>-          get_char();
>+          ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR();
>         } else {
>           conv->precision.set_from_arg(++*next_arg);
>         }
>@@ -188,23 +203,23 @@ bool ConsumeConversion(string_view *src, UnboundConversion *conv,
>   std::int8_t id = kIds[static_cast<unsigned char>(c)];
> 
>   if (id < 0) {
>-    if (id == none) return false;
>+    if (ABSL_PREDICT_FALSE(id == none)) return false;
> 
>     // It is a length modifier.
>     using str_format_internal::LengthMod;
>     LengthMod length_mod = LengthMod::FromId(static_cast<LM>(~id));
>-    get_char();
>+    ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR();
>     if (c == 'h' && length_mod.id() == LengthMod::h) {
>       conv->length_mod = LengthMod::FromId(LengthMod::hh);
>-      get_char();
>+      ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR();
>     } else if (c == 'l' && length_mod.id() == LengthMod::l) {
>       conv->length_mod = LengthMod::FromId(LengthMod::ll);
>-      get_char();
>+      ABSL_FORMAT_PARSER_INTERNAL_GET_CHAR();
>     } else {
>       conv->length_mod = length_mod;
>     }
>     id = kIds[static_cast<unsigned char>(c)];
>-    if (id < 0) return false;
>+    if (ABSL_PREDICT_FALSE(id < 0)) return false;
>   }
> 
>   assert(CheckFastPathSetting(*conv));
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.h
>index 5bebc95540e6c28d39d0847305366742ced9edfa..1022f06297a74a584e82e41ca4b824fbbf63391c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.h
>@@ -75,14 +75,14 @@ struct UnboundConversion {
> bool ConsumeUnboundConversion(string_view* src, UnboundConversion* conv,
>                               int* next_arg);
> 
>-// Parse the format std::string provided in 'src' and pass the identified items into
>+// Parse the format string provided in 'src' and pass the identified items into
> // 'consumer'.
> // Text runs will be passed by calling
> //   Consumer::Append(string_view);
> // ConversionItems will be passed by calling
> //   Consumer::ConvertOne(UnboundConversion, string_view);
> // In the case of ConvertOne, the string_view that is passed is the
>-// portion of the format std::string corresponding to the conversion, not including
>+// portion of the format string corresponding to the conversion, not including
> // the leading %. On success, it returns true. On failure, it stops and returns
> // false.
> template <typename Consumer>
>@@ -90,7 +90,7 @@ bool ParseFormatString(string_view src, Consumer consumer) {
>   int next_arg = 0;
>   while (!src.empty()) {
>     const char* percent =
>-        static_cast<const char*>(memchr(src.begin(), '%', src.size()));
>+        static_cast<const char*>(memchr(src.data(), '%', src.size()));
>     if (!percent) {
>       // We found the last substring.
>       return consumer.Append(src);
>@@ -98,7 +98,7 @@ bool ParseFormatString(string_view src, Consumer consumer) {
>     // We found a percent, so push the text run then process the percent.
>     size_t percent_loc = percent - src.data();
>     if (!consumer.Append(string_view(src.data(), percent_loc))) return false;
>-    if (percent + 1 >= src.end()) return false;
>+    if (percent + 1 >= src.data() + src.size()) return false;
> 
>     UnboundConversion conv;
> 
>@@ -178,7 +178,8 @@ class ParsedFormatBase {
>     const char* const base = data_.get();
>     string_view text(base, 0);
>     for (const auto& item : items_) {
>-      text = string_view(text.end(), (base + item.text_end) - text.end());
>+      const char* const end = text.data() + text.size();
>+      text = string_view(end, (base + item.text_end) - end);
>       if (item.is_conversion) {
>         if (!consumer.ConvertOne(item.conv, text)) return false;
>       } else {
>@@ -237,7 +238,7 @@ class ParsedFormatBase {
> // This class also supports runtime format checking with the ::New() and
> // ::NewAllowIgnored() factory functions.
> // This is the only API that allows the user to pass a runtime specified format
>-// std::string. These factory functions will return NULL if the format does not match
>+// string. These factory functions will return NULL if the format does not match
> // the conversions requested by the user.
> template <str_format_internal::Conv... C>
> class ExtendedParsedFormat : public str_format_internal::ParsedFormatBase {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser_test.cc
>index e698020b1aba38452c7b8899fd3ed2769b0c3e51..ae40203191b97f9c3c4f0c4be281a22921db6e6b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser_test.cc
>@@ -66,10 +66,10 @@ class ConsumeUnboundConversionTest : public ::testing::Test {
>   typedef UnboundConversion Props;
>   string_view Consume(string_view* src) {
>     int next = 0;
>-    const char* prev_begin = src->begin();
>+    const char* prev_begin = src->data();
>     o = UnboundConversion();  // refresh
>     ConsumeUnboundConversion(src, &o, &next);
>-    return {prev_begin, static_cast<size_t>(src->begin() - prev_begin)};
>+    return {prev_begin, static_cast<size_t>(src->data() - prev_begin)};
>   }
> 
>   bool Run(const char *fmt, bool force_positional = false) {
>@@ -84,9 +84,9 @@ class ConsumeUnboundConversionTest : public ::testing::Test {
> TEST_F(ConsumeUnboundConversionTest, ConsumeSpecification) {
>   struct Expectation {
>     int line;
>-    const char *src;
>-    const char *out;
>-    const char *src_post;
>+    string_view src;
>+    string_view out;
>+    string_view src_post;
>   };
>   const Expectation kExpect[] = {
>     {__LINE__, "",     "",     ""  },
>@@ -236,6 +236,16 @@ TEST_F(ConsumeUnboundConversionTest, WidthAndPrecision) {
>   EXPECT_EQ(9, o.precision.get_from_arg());
> 
>   EXPECT_FALSE(Run(".*0$d")) << "no arg 0";
>+
>+  // Large values
>+  EXPECT_TRUE(Run("999999999.999999999d"));
>+  EXPECT_FALSE(o.width.is_from_arg());
>+  EXPECT_EQ(999999999, o.width.value());
>+  EXPECT_FALSE(o.precision.is_from_arg());
>+  EXPECT_EQ(999999999, o.precision.value());
>+
>+  EXPECT_FALSE(Run("1000000000.999999999d"));
>+  EXPECT_FALSE(Run("999999999.1000000000d"));
> }
> 
> TEST_F(ConsumeUnboundConversionTest, Flags) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_join_internal.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_join_internal.h
>index a734758c21ee30951f77a2c69c96ec75d71ee96a..0058fc8aa4d5a81210a98f2d1bb11bc0c65a965c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_join_internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_join_internal.h
>@@ -189,7 +189,7 @@ struct DefaultFormatter<std::unique_ptr<ValueType>>
> //
> 
> // The main joining algorithm. This simply joins the elements in the given
>-// iterator range, each separated by the given separator, into an output std::string,
>+// iterator range, each separated by the given separator, into an output string,
> // and formats each element using the provided Formatter object.
> template <typename Iterator, typename Formatter>
> std::string JoinAlgorithm(Iterator start, Iterator end, absl::string_view s,
>@@ -205,20 +205,20 @@ std::string JoinAlgorithm(Iterator start, Iterator end, absl::string_view s,
> }
> 
> // A joining algorithm that's optimized for a forward iterator range of
>-// std::string-like objects that do not need any additional formatting. This is to
>-// optimize the common case of joining, say, a std::vector<std::string> or a
>+// string-like objects that do not need any additional formatting. This is to
>+// optimize the common case of joining, say, a std::vector<string> or a
> // std::vector<absl::string_view>.
> //
> // This is an overload of the previous JoinAlgorithm() function. Here the
> // Formatter argument is of type NoFormatter. Since NoFormatter is an internal
> // type, this overload is only invoked when strings::Join() is called with a
>-// range of std::string-like objects (e.g., std::string, absl::string_view), and an
>+// range of string-like objects (e.g., string, absl::string_view), and an
> // explicit Formatter argument was NOT specified.
> //
> // The optimization is that the needed space will be reserved in the output
>-// std::string to avoid the need to resize while appending. To do this, the iterator
>+// string to avoid the need to resize while appending. To do this, the iterator
> // range will be traversed twice: once to calculate the total needed size, and
>-// then again to copy the elements and delimiters to the output std::string.
>+// then again to copy the elements and delimiters to the output string.
> template <typename Iterator,
>           typename = typename std::enable_if<std::is_convertible<
>               typename std::iterator_traits<Iterator>::iterator_category,
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_split_internal.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_split_internal.h
>index 9cf0833f49028ab039c7c422504a996d99e943dc..81e8d55544680d671e56eb0254995b42bd548d40 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_split_internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_split_internal.h
>@@ -51,7 +51,7 @@ namespace strings_internal {
> 
> // This class is implicitly constructible from everything that absl::string_view
> // is implicitly constructible from. If it's constructed from a temporary
>-// std::string, the data is moved into a data member so its lifetime matches that of
>+// string, the data is moved into a data member so its lifetime matches that of
> // the ConvertibleToStringView instance.
> class ConvertibleToStringView {
>  public:
>@@ -102,7 +102,7 @@ ConvertibleToStringView(std::string&& s)  // NOLINT(runtime/explicit)
>   absl::string_view value_;
> };
> 
>-// An iterator that enumerates the parts of a std::string from a Splitter. The text
>+// An iterator that enumerates the parts of a string from a Splitter. The text
> // to be split, the Delimiter, and the Predicate are all taken from the given
> // Splitter object. Iterators may only be compared if they refer to the same
> // Splitter instance.
>@@ -159,7 +159,7 @@ class SplitIterator {
>       }
>       const absl::string_view text = splitter_->text();
>       const absl::string_view d = delimiter_.Find(text, pos_);
>-      if (d.data() == text.end()) state_ = kLastState;
>+      if (d.data() == text.data() + text.size()) state_ = kLastState;
>       curr_ = text.substr(pos_, d.data() - (text.data() + pos_));
>       pos_ += curr_.size() + d.size();
>     } while (!predicate_(curr_));
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.cc
>index 25bd7f0b828bfdec72ca587d4871bffb692862e3..3d10c57784ebe92549855bcfb0776799966f1717 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.cc
>@@ -27,6 +27,13 @@ bool CaseEqual(absl::string_view piece1, absl::string_view piece2) {
> }
> }  // namespace
> 
>+bool EqualsIgnoreCase(absl::string_view piece1, absl::string_view piece2) {
>+  return (piece1.size() == piece2.size() &&
>+          0 == absl::strings_internal::memcasecmp(piece1.data(), piece2.data(),
>+                                                  piece1.size()));
>+  // memcasecmp uses absl::ascii_tolower().
>+}
>+
> bool StartsWithIgnoreCase(absl::string_view text, absl::string_view prefix) {
>   return (text.size() >= prefix.size()) &&
>          CaseEqual(text.substr(0, prefix.size()), prefix);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.h
>index 108b6048b048ec265f518aa3a6ee04d1019f3436..6e8ed10fc50feb6f7196db419c4a38a4240b4f7b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.h
>@@ -17,7 +17,7 @@
> // File: match.h
> // -----------------------------------------------------------------------------
> //
>-// This file contains simple utilities for performing std::string matching checks.
>+// This file contains simple utilities for performing string matching checks.
> // All of these function parameters are specified as `absl::string_view`,
> // meaning that these functions can accept `std::string`, `absl::string_view` or
> // nul-terminated C-style strings.
>@@ -41,14 +41,14 @@ namespace absl {
> 
> // StrContains()
> //
>-// Returns whether a given std::string `haystack` contains the substring `needle`.
>+// Returns whether a given string `haystack` contains the substring `needle`.
> inline bool StrContains(absl::string_view haystack, absl::string_view needle) {
>   return haystack.find(needle, 0) != haystack.npos;
> }
> 
> // StartsWith()
> //
>-// Returns whether a given std::string `text` begins with `prefix`.
>+// Returns whether a given string `text` begins with `prefix`.
> inline bool StartsWith(absl::string_view text, absl::string_view prefix) {
>   return prefix.empty() ||
>          (text.size() >= prefix.size() &&
>@@ -57,7 +57,7 @@ inline bool StartsWith(absl::string_view text, absl::string_view prefix) {
> 
> // EndsWith()
> //
>-// Returns whether a given std::string `text` ends with `suffix`.
>+// Returns whether a given string `text` ends with `suffix`.
> inline bool EndsWith(absl::string_view text, absl::string_view suffix) {
>   return suffix.empty() ||
>          (text.size() >= suffix.size() &&
>@@ -66,16 +66,22 @@ inline bool EndsWith(absl::string_view text, absl::string_view suffix) {
>          );
> }
> 
>-// StartsWithIgnoreCase()
>+// EqualsIgnoreCase()
> //
>-// Returns whether a given std::string `text` starts with `starts_with`, ignoring
>+// Returns whether given ASCII strings `piece1` and `piece2` are equal, ignoring
> // case in the comparison.
>+bool EqualsIgnoreCase(absl::string_view piece1, absl::string_view piece2);
>+
>+// StartsWithIgnoreCase()
>+//
>+// Returns whether a given ASCII string `text` starts with `starts_with`,
>+// ignoring case in the comparison.
> bool StartsWithIgnoreCase(absl::string_view text, absl::string_view prefix);
> 
> // EndsWithIgnoreCase()
> //
>-// Returns whether a given std::string `text` ends with `ends_with`, ignoring case
>-// in the comparison.
>+// Returns whether a given ASCII string `text` ends with `ends_with`, ignoring
>+// case in the comparison.
> bool EndsWithIgnoreCase(absl::string_view text, absl::string_view suffix);
> 
> }  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match_test.cc
>index d194f0e689bf6e001656c52e357b5d778a67b55d..c21e00bf807c94f7f7c5a88f265f20694cdf1143 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match_test.cc
>@@ -80,6 +80,17 @@ TEST(MatchTest, ContainsNull) {
>   EXPECT_FALSE(absl::StrContains(cs, sv2));
> }
> 
>+TEST(MatchTest, EqualsIgnoreCase) {
>+  std::string text = "the";
>+  absl::string_view data(text);
>+
>+  EXPECT_TRUE(absl::EqualsIgnoreCase(data, "The"));
>+  EXPECT_TRUE(absl::EqualsIgnoreCase(data, "THE"));
>+  EXPECT_TRUE(absl::EqualsIgnoreCase(data, "the"));
>+  EXPECT_FALSE(absl::EqualsIgnoreCase(data, "Quick"));
>+  EXPECT_FALSE(absl::EqualsIgnoreCase(data, "then"));
>+}
>+
> TEST(MatchTest, StartsWithIgnoreCase) {
>   EXPECT_TRUE(absl::StartsWithIgnoreCase("foo", "foo"));
>   EXPECT_TRUE(absl::StartsWithIgnoreCase("foo", "Fo"));
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.cc
>index f842ed85e9f595efb8da1ec050275dd368a26e97..9309f9da9dcc5c907e6835eca6d047c90f2c6f32 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.cc
>@@ -12,7 +12,7 @@
> // See the License for the specific language governing permissions and
> // limitations under the License.
> 
>-// This file contains std::string processing functions related to
>+// This file contains string processing functions related to
> // numeric values.
> 
> #include "absl/strings/numbers.h"
>@@ -30,10 +30,10 @@
> #include <memory>
> #include <utility>
> 
>+#include "absl/base/internal/bits.h"
> #include "absl/base/internal/raw_logging.h"
> #include "absl/strings/ascii.h"
> #include "absl/strings/charconv.h"
>-#include "absl/strings/internal/bits.h"
> #include "absl/strings/internal/memutil.h"
> #include "absl/strings/str_cat.h"
> 
>@@ -161,8 +161,8 @@ bool SimpleAtob(absl::string_view str, bool* value) {
> // their output to the beginning of the buffer.  The caller is responsible
> // for ensuring that the buffer has enough space to hold the output.
> //
>-// Returns a pointer to the end of the std::string (i.e. the null character
>-// terminating the std::string).
>+// Returns a pointer to the end of the string (i.e. the null character
>+// terminating the string).
> // ----------------------------------------------------------------------
> 
> namespace {
>@@ -344,7 +344,7 @@ static std::pair<uint64_t, uint64_t> Mul32(std::pair<uint64_t, uint64_t> num,
>   uint64_t bits128_up = (bits96_127 >> 32) + (bits64_127 < bits64_95);
>   if (bits128_up == 0) return {bits64_127, bits0_63};
> 
>-  int shift = 64 - strings_internal::CountLeadingZeros64(bits128_up);
>+  int shift = 64 - base_internal::CountLeadingZeros64(bits128_up);
>   uint64_t lo = (bits0_63 >> shift) + (bits64_127 << (64 - shift));
>   uint64_t hi = (bits64_127 >> shift) + (bits128_up << (64 - shift));
>   return {hi, lo};
>@@ -375,7 +375,7 @@ static std::pair<uint64_t, uint64_t> PowFive(uint64_t num, int expfive) {
>       5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5,
>       5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5};
>   result = Mul32(result, powers_of_five[expfive & 15]);
>-  int shift = strings_internal::CountLeadingZeros64(result.first);
>+  int shift = base_internal::CountLeadingZeros64(result.first);
>   if (shift != 0) {
>     result.first = (result.first << shift) + (result.second >> (64 - shift));
>     result.second = (result.second << shift);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.h
>index cf3c597266cf71474d405dd0cfdea5cdeb58badc..f9b2ccef94d8bd55b1bd82de582e589cbd212f51 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.h
>@@ -41,32 +41,32 @@ namespace absl {
> 
> // SimpleAtoi()
> //
>-// Converts the given std::string into an integer value, returning `true` if
>-// successful. The std::string must reflect a base-10 integer (optionally followed or
>+// Converts the given string into an integer value, returning `true` if
>+// successful. The string must reflect a base-10 integer (optionally followed or
> // preceded by ASCII whitespace) whose value falls within the range of the
>-// integer type,
>+// integer type.
> template <typename int_type>
> ABSL_MUST_USE_RESULT bool SimpleAtoi(absl::string_view s, int_type* out);
> 
> // SimpleAtof()
> //
>-// Converts the given std::string (optionally followed or preceded by ASCII
>+// Converts the given string (optionally followed or preceded by ASCII
> // whitespace) into a float, which may be rounded on overflow or underflow.
>-// See http://en.cppreference.com/w/c/std::string/byte/strtof for details about the
>+// See http://en.cppreference.com/w/c/string/byte/strtof for details about the
> // allowed formats for `str`.
> ABSL_MUST_USE_RESULT bool SimpleAtof(absl::string_view str, float* value);
> 
> // SimpleAtod()
> //
>-// Converts the given std::string (optionally followed or preceded by ASCII
>+// Converts the given string (optionally followed or preceded by ASCII
> // whitespace) into a double, which may be rounded on overflow or underflow.
>-// See http://en.cppreference.com/w/c/std::string/byte/strtof for details about the
>+// See http://en.cppreference.com/w/c/string/byte/strtof for details about the
> // allowed formats for `str`.
> ABSL_MUST_USE_RESULT bool SimpleAtod(absl::string_view str, double* value);
> 
> // SimpleAtob()
> //
>-// Converts the given std::string into a boolean, returning `true` if successful.
>+// Converts the given string into a boolean, returning `true` if successful.
> // The following case-insensitive strings are interpreted as boolean `true`:
> // "true", "t", "yes", "y", "1". The following case-insensitive strings
> // are interpreted as boolean `false`: "false", "f", "no", "n", "0".
>@@ -169,9 +169,9 @@ ABSL_MUST_USE_RESULT bool safe_strtoi_base(absl::string_view s, int_type* out,
> 
> // SimpleAtoi()
> //
>-// Converts a std::string to an integer, using `safe_strto?()` functions for actual
>+// Converts a string to an integer, using `safe_strto?()` functions for actual
> // parsing, returning `true` if successful. The `safe_strto?()` functions apply
>-// strict checking; the std::string must be a base-10 integer, optionally followed or
>+// strict checking; the string must be a base-10 integer, optionally followed or
> // preceded by ASCII whitespace, with a value in the range of the corresponding
> // integer type.
> template <typename int_type>
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_benchmark.cc
>index 8ef650b9a3eccdd79664dc79d2d6f261a6686a32..0570b758633ee0406d1217e9e70012b28c180c71 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_benchmark.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_benchmark.cc
>@@ -158,7 +158,7 @@ BENCHMARK(BM_safe_strtou64_string)
>     ->ArgPair(16, 10)
>     ->ArgPair(16, 16);
> 
>-// Returns a vector of `num_strings` strings. Each std::string represents a
>+// Returns a vector of `num_strings` strings. Each string represents a
> // floating point number with `num_digits` digits before the decimal point and
> // another `num_digits` digits after.
> std::vector<std::string> MakeFloatStrings(int num_strings, int num_digits) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_test.cc
>index 27cc0479e3412c838b4785dfafb8895efe096f63..36fc0d64fd5130507b1e811b38dfe57e32493b4d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_test.cc
>@@ -12,7 +12,7 @@
> // See the License for the specific language governing permissions and
> // limitations under the License.
> 
>-// This file tests std::string processing functions related to numeric values.
>+// This file tests string processing functions related to numeric values.
> 
> #include "absl/strings/numbers.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.cc
>index 3fe8c95eca9ef979491693638900858b08725d0b..efa4fd73dbac77048747ecc4a7f1fcb0e67d7706 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.cc
>@@ -79,7 +79,7 @@ AlphaNum::AlphaNum(Dec dec) {
> // ----------------------------------------------------------------------
> // StrCat()
> //    This merges the given strings or integers, with no delimiter.  This
>-//    is designed to be the fastest possible way to construct a std::string out
>+//    is designed to be the fastest possible way to construct a string out
> //    of a mix of raw C strings, StringPieces, strings, and integer values.
> // ----------------------------------------------------------------------
> 
>@@ -154,10 +154,10 @@ std::string CatPieces(std::initializer_list<absl::string_view> pieces) {
> }
> 
> // It's possible to call StrAppend with an absl::string_view that is itself a
>-// fragment of the std::string we're appending to.  However the results of this are
>+// fragment of the string we're appending to.  However the results of this are
> // random. Therefore, check for this in debug mode.  Use unsigned math so we
> // only have to do one comparison. Note, there's an exception case: appending an
>-// empty std::string is always allowed.
>+// empty string is always allowed.
> #define ASSERT_NO_OVERLAP(dest, src) \
>   assert(((src).size() == 0) ||      \
>          (uintptr_t((src).data() - (dest).data()) > uintptr_t((dest).size())))
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.h
>index e5501a5012ee8d9cf79cbe6b50b04756d191bf9e..da9ed9a269e990b87922ff2d19aa9717e5e0d721 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.h
>@@ -23,7 +23,7 @@
> // designed to be used as a parameter type that efficiently manages conversion
> // to strings and avoids copies in the above operations.
> //
>-// Any routine accepting either a std::string or a number may accept `AlphaNum`.
>+// Any routine accepting either a string or a number may accept `AlphaNum`.
> // The basic idea is that by accepting a `const AlphaNum &` as an argument
> // to your function, your callers will automagically convert bools, integers,
> // and floating point values to strings for you.
>@@ -65,7 +65,7 @@
> namespace absl {
> 
> namespace strings_internal {
>-// AlphaNumBuffer allows a way to pass a std::string to StrCat without having to do
>+// AlphaNumBuffer allows a way to pass a string to StrCat without having to do
> // memory allocation.  It is simply a pair of a fixed-size character array, and
> // a size.  Please don't use outside of absl, yet.
> template <size_t max_size>
>@@ -119,8 +119,8 @@ enum PadSpec : uint8_t {
> // Hex
> // -----------------------------------------------------------------------------
> //
>-// `Hex` stores a set of hexadecimal std::string conversion parameters for use
>-// within `AlphaNum` std::string conversions.
>+// `Hex` stores a set of hexadecimal string conversion parameters for use
>+// within `AlphaNum` string conversions.
> struct Hex {
>   uint64_t value;
>   uint8_t width;
>@@ -168,8 +168,8 @@ struct Hex {
> // Dec
> // -----------------------------------------------------------------------------
> //
>-// `Dec` stores a set of decimal std::string conversion parameters for use
>-// within `AlphaNum` std::string conversions.  Dec is slower than the default
>+// `Dec` stores a set of decimal string conversion parameters for use
>+// within `AlphaNum` string conversions.  Dec is slower than the default
> // integer conversion, so use it only if you need padding.
> struct Dec {
>   uint64_t value;
>@@ -271,7 +271,7 @@ class AlphaNum {
> //
> // Merges given strings or numbers, using no delimiter(s).
> //
>-// `StrCat()` is designed to be the fastest possible way to construct a std::string
>+// `StrCat()` is designed to be the fastest possible way to construct a string
> // out of a mix of raw C strings, string_views, strings, bool values,
> // and numeric values.
> //
>@@ -279,7 +279,7 @@ class AlphaNum {
> // works poorly on strings built up out of fragments.
> //
> // For clarity and performance, don't use `StrCat()` when appending to a
>-// std::string. Use `StrAppend()` instead. In particular, avoid using any of these
>+// string. Use `StrAppend()` instead. In particular, avoid using any of these
> // (anti-)patterns:
> //
> //   str.append(StrCat(...))
>@@ -328,26 +328,26 @@ ABSL_MUST_USE_RESULT inline std::string StrCat(const AlphaNum& a, const AlphaNum
> // StrAppend()
> // -----------------------------------------------------------------------------
> //
>-// Appends a std::string or set of strings to an existing std::string, in a similar
>+// Appends a string or set of strings to an existing string, in a similar
> // fashion to `StrCat()`.
> //
> // WARNING: `StrAppend(&str, a, b, c, ...)` requires that none of the
> // a, b, c, parameters be a reference into str. For speed, `StrAppend()` does
> // not try to check each of its input arguments to be sure that they are not
>-// a subset of the std::string being appended to. That is, while this will work:
>+// a subset of the string being appended to. That is, while this will work:
> //
>-//   std::string s = "foo";
>+//   string s = "foo";
> //   s += s;
> //
> // This output is undefined:
> //
>-//   std::string s = "foo";
>+//   string s = "foo";
> //   StrAppend(&s, s);
> //
> // This output is undefined as well, since `absl::string_view` does not own its
> // data:
> //
>-//   std::string s = "foobar";
>+//   string s = "foobar";
> //   absl::string_view p = s;
> //   StrAppend(&s, p);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat_test.cc
>index e9d67690638271920d26234d9739053c30b27489..555d8db3eb85438be5f8cc7f59dd5507d6d5ba54 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat_test.cc
>@@ -28,7 +28,8 @@
> #define ABSL_EXPECT_DEBUG_DEATH(statement, regex) \
>   EXPECT_DEBUG_DEATH(statement, ".*")
> #else
>-#define ABSL_EXPECT_DEBUG_DEATH EXPECT_DEBUG_DEATH
>+#define ABSL_EXPECT_DEBUG_DEATH(statement, regex) \
>+  EXPECT_DEBUG_DEATH(statement, regex)
> #endif
> 
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format.h
>index 70a811b75e2ff8863c9e3687c6c282f90c2317f9..2d07725de21fcf7906950f02d20615b5ad1098ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format.h
>@@ -18,20 +18,20 @@
> // -----------------------------------------------------------------------------
> //
> // The `str_format` library is a typesafe replacement for the family of
>-// `printf()` std::string formatting routines within the `<cstdio>` standard library
>+// `printf()` string formatting routines within the `<cstdio>` standard library
> // header. Like the `printf` family, the `str_format` uses a "format string" to
> // perform argument substitutions based on types.
> //
> // Example:
> //
>-//   std::string s = absl::StrFormat("%s %s You have $%d!", "Hello", name, dollars);
>+//   string s = absl::StrFormat("%s %s You have $%d!", "Hello", name, dollars);
> //
> // The library consists of the following basic utilities:
> //
> //   * `absl::StrFormat()`, a type-safe replacement for `std::sprintf()`, to
>-//     write a format std::string to a `string` value.
>-//   * `absl::StrAppendFormat()` to append a format std::string to a `string`
>-//   * `absl::StreamFormat()` to more efficiently write a format std::string to a
>+//     write a format string to a `string` value.
>+//   * `absl::StrAppendFormat()` to append a format string to a `string`
>+//   * `absl::StreamFormat()` to more efficiently write a format string to a
> //     stream, such as`std::cout`.
> //   * `absl::PrintF()`, `absl::FPrintF()` and `absl::SNPrintF()` as
> //     replacements for `std::printf()`, `std::fprintf()` and `std::snprintf()`.
>@@ -39,15 +39,15 @@
> //     Note: a version of `std::sprintf()` is not supported as it is
> //     generally unsafe due to buffer overflows.
> //
>-// Additionally, you can provide a format std::string (and its associated arguments)
>+// Additionally, you can provide a format string (and its associated arguments)
> // using one of the following abstractions:
> //
>-//   * A `FormatSpec` class template fully encapsulates a format std::string and its
>+//   * A `FormatSpec` class template fully encapsulates a format string and its
> //     type arguments and is usually provided to `str_format` functions as a
> //     variadic argument of type `FormatSpec<Arg...>`. The `FormatSpec<Args...>`
> //     template is evaluated at compile-time, providing type safety.
> //   * A `ParsedFormat` instance, which encapsulates a specific, pre-compiled
>-//     format std::string for a specific set of type(s), and which can be passed
>+//     format string for a specific set of type(s), and which can be passed
> //     between API boundaries. (The `FormatSpec` type should not be used
> //     directly.)
> //
>@@ -60,7 +60,7 @@
> //
> //   * A `FormatUntyped()` function that is similar to `Format()` except it is
> //     loosely typed. `FormatUntyped()` is not a template and does not perform
>-//     any compile-time checking of the format std::string; instead, it returns a
>+//     any compile-time checking of the format string; instead, it returns a
> //     boolean from a runtime check.
> //
> // In addition, the `str_format` library provides extension points for
>@@ -89,7 +89,7 @@ namespace absl {
> // Example:
> //
> //   absl::UntypedFormatSpec format("%d");
>-//   std::string out;
>+//   string out;
> //   CHECK(absl::FormatUntyped(&out, format, {absl::FormatArg(1)}));
> class UntypedFormatSpec {
>  public:
>@@ -135,7 +135,7 @@ str_format_internal::StreamedWrapper<T> FormatStreamed(const T& v) {
> // Example:
> //
> //   int n = 0;
>-//   std::string s = absl::StrFormat("%s%d%n", "hello", 123,
>+//   string s = absl::StrFormat("%s%d%n", "hello", 123,
> //                   absl::FormatCountCapture(&n));
> //   EXPECT_EQ(8, n);
> class FormatCountCapture {
>@@ -155,10 +155,10 @@ class FormatCountCapture {
> 
> // FormatSpec
> //
>-// The `FormatSpec` type defines the makeup of a format std::string within the
>+// The `FormatSpec` type defines the makeup of a format string within the
> // `str_format` library. You should not need to use or manipulate this type
> // directly. A `FormatSpec` is a variadic class template that is evaluated at
>-// compile-time, according to the format std::string and arguments that are passed
>+// compile-time, according to the format string and arguments that are passed
> // to it.
> //
> // For a `FormatSpec` to be valid at compile-time, it must be provided as
>@@ -166,12 +166,12 @@ class FormatCountCapture {
> //
> // * A `constexpr` literal or `absl::string_view`, which is how it most often
> //   used.
>-// * A `ParsedFormat` instantiation, which ensures the format std::string is
>+// * A `ParsedFormat` instantiation, which ensures the format string is
> //   valid before use. (See below.)
> //
> // Example:
> //
>-//   // Provided as a std::string literal.
>+//   // Provided as a string literal.
> //   absl::StrFormat("Welcome to %s, Number %d!", "The Village", 6);
> //
> //   // Provided as a constexpr absl::string_view.
>@@ -183,7 +183,7 @@ class FormatCountCapture {
> //   absl::ParsedFormat<'s', 'd'> formatString("Welcome to %s, Number %d!");
> //   absl::StrFormat(formatString, "TheVillage", 6);
> //
>-// A format std::string generally follows the POSIX syntax as used within the POSIX
>+// A format string generally follows the POSIX syntax as used within the POSIX
> // `printf` specification.
> //
> // (See http://pubs.opengroup.org/onlinepubs/9699919799/utilities/printf.html.)
>@@ -223,7 +223,7 @@ class FormatCountCapture {
> //     "%p", *int               -> "0x7ffdeb6ad2a4"
> //
> //     int n = 0;
>-//     std::string s = absl::StrFormat(
>+//     string s = absl::StrFormat(
> //         "%s%d%n", "hello", 123, absl::FormatCountCapture(&n));
> //     EXPECT_EQ(8, n);
> //
>@@ -236,7 +236,7 @@ class FormatCountCapture {
> //
> // However, in the `str_format` library, a format conversion specifies a broader
> // C++ conceptual category instead of an exact type. For example, `%s` binds to
>-// any std::string-like argument, so `std::string`, `absl::string_view`, and
>+// any string-like argument, so `std::string`, `absl::string_view`, and
> // `const char*` are all accepted. Likewise, `%d` accepts any integer-like
> // argument, etc.
> 
>@@ -248,7 +248,7 @@ using FormatSpec =
> //
> // A `ParsedFormat` is a class template representing a preparsed `FormatSpec`,
> // with template arguments specifying the conversion characters used within the
>-// format std::string. Such characters must be valid format type specifiers, and
>+// format string. Such characters must be valid format type specifiers, and
> // these type specifiers are checked at compile-time.
> //
> // Instances of `ParsedFormat` can be created, copied, and reused to speed up
>@@ -275,26 +275,26 @@ using ParsedFormat = str_format_internal::ExtendedParsedFormat<
> 
> // StrFormat()
> //
>-// Returns a `string` given a `printf()`-style format std::string and zero or more
>+// Returns a `string` given a `printf()`-style format string and zero or more
> // additional arguments. Use it as you would `sprintf()`. `StrFormat()` is the
> // primary formatting function within the `str_format` library, and should be
> // used in most cases where you need type-safe conversion of types into
> // formatted strings.
> //
>-// The format std::string generally consists of ordinary character data along with
>+// The format string generally consists of ordinary character data along with
> // one or more format conversion specifiers (denoted by the `%` character).
>-// Ordinary character data is returned unchanged into the result std::string, while
>+// Ordinary character data is returned unchanged into the result string, while
> // each conversion specification performs a type substitution from
> // `StrFormat()`'s other arguments. See the comments for `FormatSpec` for full
>-// information on the makeup of this format std::string.
>+// information on the makeup of this format string.
> //
> // Example:
> //
>-//   std::string s = absl::StrFormat(
>+//   string s = absl::StrFormat(
> //       "Welcome to %s, Number %d!", "The Village", 6);
> //   EXPECT_EQ("Welcome to The Village, Number 6!", s);
> //
>-// Returns an empty std::string in case of error.
>+// Returns an empty string in case of error.
> template <typename... Args>
> ABSL_MUST_USE_RESULT std::string StrFormat(const FormatSpec<Args...>& format,
>                                       const Args&... args) {
>@@ -305,13 +305,13 @@ ABSL_MUST_USE_RESULT std::string StrFormat(const FormatSpec<Args...>& format,
> 
> // StrAppendFormat()
> //
>-// Appends to a `dst` std::string given a format std::string, and zero or more additional
>+// Appends to a `dst` string given a format string, and zero or more additional
> // arguments, returning `*dst` as a convenience for chaining purposes. Appends
> // nothing in case of error (but possibly alters its capacity).
> //
> // Example:
> //
>-//   std::string orig("For example PI is approximately ");
>+//   string orig("For example PI is approximately ");
> //   std::cout << StrAppendFormat(&orig, "%12.6f", 3.14);
> template <typename... Args>
> std::string& StrAppendFormat(std::string* dst, const FormatSpec<Args...>& format,
>@@ -323,7 +323,7 @@ std::string& StrAppendFormat(std::string* dst, const FormatSpec<Args...>& format
> 
> // StreamFormat()
> //
>-// Writes to an output stream given a format std::string and zero or more arguments,
>+// Writes to an output stream given a format string and zero or more arguments,
> // generally in a manner that is more efficient than streaming the result of
> // `absl:: StrFormat()`. The returned object must be streamed before the full
> // expression ends.
>@@ -341,7 +341,7 @@ ABSL_MUST_USE_RESULT str_format_internal::Streamable StreamFormat(
> 
> // PrintF()
> //
>-// Writes to stdout given a format std::string and zero or more arguments. This
>+// Writes to stdout given a format string and zero or more arguments. This
> // function is functionally equivalent to `std::printf()` (and type-safe);
> // prefer `absl::PrintF()` over `std::printf()`.
> //
>@@ -361,14 +361,14 @@ int PrintF(const FormatSpec<Args...>& format, const Args&... args) {
> 
> // FPrintF()
> //
>-// Writes to a file given a format std::string and zero or more arguments. This
>+// Writes to a file given a format string and zero or more arguments. This
> // function is functionally equivalent to `std::fprintf()` (and type-safe);
> // prefer `absl::FPrintF()` over `std::fprintf()`.
> //
> // Example:
> //
> //   std::string_view s = "Ulaanbaatar";
>-//   absl::FPrintF("The capital of Mongolia is %s", s);
>+//   absl::FPrintF(stdout, "The capital of Mongolia is %s", s);
> //
> //   Outputs: "The capital of Mongolia is Ulaanbaatar"
> //
>@@ -382,7 +382,7 @@ int FPrintF(std::FILE* output, const FormatSpec<Args...>& format,
> 
> // SNPrintF()
> //
>-// Writes to a sized buffer given a format std::string and zero or more arguments.
>+// Writes to a sized buffer given a format string and zero or more arguments.
> // This function is functionally equivalent to `std::snprintf()` (and
> // type-safe); prefer `absl::SNPrintF()` over `std::snprintf()`.
> //
>@@ -430,14 +430,14 @@ class FormatRawSink {
> 
> // Format()
> //
>-// Writes a formatted std::string to an arbitrary sink object (implementing the
>-// `absl::FormatRawSink` interface), using a format std::string and zero or more
>+// Writes a formatted string to an arbitrary sink object (implementing the
>+// `absl::FormatRawSink` interface), using a format string and zero or more
> // additional arguments.
> //
> // By default, `string` and `std::ostream` are supported as destination objects.
> //
> // `absl::Format()` is a generic version of `absl::StrFormat(), for custom
>-// sinks. The format std::string, like format strings for `StrFormat()`, is checked
>+// sinks. The format string, like format strings for `StrFormat()`, is checked
> // at compile-time.
> //
> // On failure, this function returns `false` and the state of the sink is
>@@ -463,13 +463,13 @@ using FormatArg = str_format_internal::FormatArgImpl;
> 
> // FormatUntyped()
> //
>-// Writes a formatted std::string to an arbitrary sink object (implementing the
>+// Writes a formatted string to an arbitrary sink object (implementing the
> // `absl::FormatRawSink` interface), using an `UntypedFormatSpec` and zero or
> // more additional arguments.
> //
> // This function acts as the most generic formatting function in the
> // `str_format` library. The caller provides a raw sink, an unchecked format
>-// std::string, and (usually) a runtime specified list of arguments; no compile-time
>+// string, and (usually) a runtime specified list of arguments; no compile-time
> // checking of formatting is performed within this function. As a result, a
> // caller should check the return value to verify that no error occurred.
> // On failure, this function returns `false` and the state of the sink is
>@@ -483,9 +483,9 @@ using FormatArg = str_format_internal::FormatArgImpl;
> //
> // Example:
> //
>-//   std::optional<std::string> FormatDynamic(const std::string& in_format,
>-//                                       const vector<std::string>& in_args) {
>-//     std::string out;
>+//   std::optional<string> FormatDynamic(const string& in_format,
>+//                                       const vector<string>& in_args) {
>+//     string out;
> //     std::vector<absl::FormatArg> args;
> //     for (const auto& v : in_args) {
> //       // It is important that 'v' is a reference to the objects in in_args.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format_test.cc
>index fed75fafb5c65c11b92f8edbca538a4acdd1177c..aa14e211709ab7393796a320830c755916acc883 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format_test.cc
>@@ -391,7 +391,7 @@ TEST(StrFormat, BehavesAsDocumented) {
> #endif
> 
>   // Other conversion
>-  int64_t value = 0x7ffdeb6;
>+  int64_t value = 0x7ffdeb4;
>   auto ptr_value = static_cast<uintptr_t>(value);
>   const int& something = *reinterpret_cast<const int*>(ptr_value);
>   EXPECT_EQ(StrFormat("%p", &something), StrFormat("0x%x", ptr_value));
>@@ -601,3 +601,39 @@ TEST_F(ParsedFormatTest, RegressionMixPositional) {
> 
> }  // namespace
> }  // namespace absl
>+
>+// Some codegen thunks that we can use to easily dump the generated assembly for
>+// different StrFormat calls.
>+
>+inline std::string CodegenAbslStrFormatInt(int i) {
>+  return absl::StrFormat("%d", i);
>+}
>+
>+inline std::string CodegenAbslStrFormatIntStringInt64(int i, const std::string& s,
>+                                                 int64_t i64) {
>+  return absl::StrFormat("%d %s %d", i, s, i64);
>+}
>+
>+inline void CodegenAbslStrAppendFormatInt(std::string* out, int i) {
>+  absl::StrAppendFormat(out, "%d", i);
>+}
>+
>+inline void CodegenAbslStrAppendFormatIntStringInt64(std::string* out, int i,
>+                                                     const std::string& s,
>+                                                     int64_t i64) {
>+  absl::StrAppendFormat(out, "%d %s %d", i, s, i64);
>+}
>+
>+auto absl_internal_str_format_force_codegen_funcs = std::make_tuple(
>+    CodegenAbslStrFormatInt, CodegenAbslStrFormatIntStringInt64,
>+    CodegenAbslStrAppendFormatInt, CodegenAbslStrAppendFormatIntStringInt64);
>+
>+bool absl_internal_str_format_force_codegen_always_false;
>+// Force the compiler to generate the functions by making it look like we
>+// escape the function pointers.
>+// It can't statically know that
>+// absl_internal_str_format_force_codegen_always_false is not changed by someone
>+// else.
>+bool absl_internal_str_format_force_codegen =
>+    absl_internal_str_format_force_codegen_always_false &&
>+    printf("%p", &absl_internal_str_format_force_codegen_funcs) == 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join.h
>index bd4d0e1d9324cc729214449935b57337a6716fb7..f9611ad3b47d72d0d992bd2bef9075c8608fd25e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join.h
>@@ -18,10 +18,10 @@
> // -----------------------------------------------------------------------------
> //
> // This header file contains functions for joining a range of elements and
>-// returning the result as a std::string. StrJoin operations are specified by passing
>-// a range, a separator std::string to use between the elements joined, and an
>+// returning the result as a string. StrJoin operations are specified by passing
>+// a range, a separator string to use between the elements joined, and an
> // optional Formatter responsible for converting each argument in the range to a
>-// std::string. If omitted, a default `AlphaNumFormatter()` is called on the elements
>+// string. If omitted, a default `AlphaNumFormatter()` is called on the elements
> // to be joined, using the same formatting that `absl::StrCat()` uses. This
> // package defines a number of default formatters, and you can define your own
> // implementations.
>@@ -29,7 +29,7 @@
> // Ranges are specified by passing a container with `std::begin()` and
> // `std::end()` iterators, container-specific `begin()` and `end()` iterators, a
> // brace-initialized `std::initializer_list`, or a `std::tuple` of heterogeneous
>-// objects. The separator std::string is specified as an `absl::string_view`.
>+// objects. The separator string is specified as an `absl::string_view`.
> //
> // Because the default formatter uses the `absl::AlphaNum` class,
> // `absl::StrJoin()`, like `absl::StrCat()`, will work out-of-the-box on
>@@ -37,8 +37,8 @@
> //
> // Example:
> //
>-//   std::vector<std::string> v = {"foo", "bar", "baz"};
>-//   std::string s = absl::StrJoin(v, "-");
>+//   std::vector<string> v = {"foo", "bar", "baz"};
>+//   string s = absl::StrJoin(v, "-");
> //   EXPECT_EQ("foo-bar-baz", s);
> //
> // See comments on the `absl::StrJoin()` function for more examples.
>@@ -52,6 +52,7 @@
> #include <iterator>
> #include <string>
> #include <tuple>
>+#include <type_traits>
> #include <utility>
> 
> #include "absl/base/macros.h"
>@@ -65,16 +66,16 @@ namespace absl {
> // -----------------------------------------------------------------------------
> //
> // A Formatter is a function object that is responsible for formatting its
>-// argument as a std::string and appending it to a given output std::string. Formatters
>+// argument as a string and appending it to a given output string. Formatters
> // may be implemented as function objects, lambdas, or normal functions. You may
> // provide your own Formatter to enable `absl::StrJoin()` to work with arbitrary
> // types.
> //
> // The following is an example of a custom Formatter that simply uses
>-// `std::to_string()` to format an integer as a std::string.
>+// `std::to_string()` to format an integer as a string.
> //
> //   struct MyFormatter {
>-//     void operator()(std::string* out, int i) const {
>+//     void operator()(string* out, int i) const {
> //       out->append(std::to_string(i));
> //     }
> //   };
>@@ -83,7 +84,7 @@ namespace absl {
> // argument to `absl::StrJoin()`:
> //
> //   std::vector<int> v = {1, 2, 3, 4};
>-//   std::string s = absl::StrJoin(v, "-", MyFormatter());
>+//   string s = absl::StrJoin(v, "-", MyFormatter());
> //   EXPECT_EQ("1-2-3-4", s);
> //
> // The following standard formatters are provided within this file:
>@@ -155,10 +156,10 @@ DereferenceFormatter() {
> // StrJoin()
> // -----------------------------------------------------------------------------
> //
>-// Joins a range of elements and returns the result as a std::string.
>-// `absl::StrJoin()` takes a range, a separator std::string to use between the
>+// Joins a range of elements and returns the result as a string.
>+// `absl::StrJoin()` takes a range, a separator string to use between the
> // elements joined, and an optional Formatter responsible for converting each
>-// argument in the range to a std::string.
>+// argument in the range to a string.
> //
> // If omitted, the default `AlphaNumFormatter()` is called on the elements to be
> // joined.
>@@ -166,22 +167,22 @@ DereferenceFormatter() {
> // Example 1:
> //   // Joins a collection of strings. This pattern also works with a collection
> //   // of `absl::string_view` or even `const char*`.
>-//   std::vector<std::string> v = {"foo", "bar", "baz"};
>-//   std::string s = absl::StrJoin(v, "-");
>+//   std::vector<string> v = {"foo", "bar", "baz"};
>+//   string s = absl::StrJoin(v, "-");
> //   EXPECT_EQ("foo-bar-baz", s);
> //
> // Example 2:
> //   // Joins the values in the given `std::initializer_list<>` specified using
> //   // brace initialization. This pattern also works with an initializer_list
> //   // of ints or `absl::string_view` -- any `AlphaNum`-compatible type.
>-//   std::string s = absl::StrJoin({"foo", "bar", "baz"}, "-");
>+//   string s = absl::StrJoin({"foo", "bar", "baz"}, "-");
> //   EXPECT_EQ("foo-bar-baz", s);
> //
> // Example 3:
> //   // Joins a collection of ints. This pattern also works with floats,
> //   // doubles, int64s -- any `StrCat()`-compatible type.
> //   std::vector<int> v = {1, 2, 3, -4};
>-//   std::string s = absl::StrJoin(v, "-");
>+//   string s = absl::StrJoin(v, "-");
> //   EXPECT_EQ("1-2-3--4", s);
> //
> // Example 4:
>@@ -192,7 +193,7 @@ DereferenceFormatter() {
> //   // `std::vector<int*>`.
> //   int x = 1, y = 2, z = 3;
> //   std::vector<int*> v = {&x, &y, &z};
>-//   std::string s = absl::StrJoin(v, "-");
>+//   string s = absl::StrJoin(v, "-");
> //   EXPECT_EQ("1-2-3", s);
> //
> // Example 5:
>@@ -201,42 +202,42 @@ DereferenceFormatter() {
> //   v.emplace_back(new int(1));
> //   v.emplace_back(new int(2));
> //   v.emplace_back(new int(3));
>-//   std::string s = absl::StrJoin(v, "-");
>+//   string s = absl::StrJoin(v, "-");
> //   EXPECT_EQ("1-2-3", s);
> //
> // Example 6:
> //   // Joins a `std::map`, with each key-value pair separated by an equals
> //   // sign. This pattern would also work with, say, a
> //   // `std::vector<std::pair<>>`.
>-//   std::map<std::string, int> m = {
>+//   std::map<string, int> m = {
> //       std::make_pair("a", 1),
> //       std::make_pair("b", 2),
> //       std::make_pair("c", 3)};
>-//   std::string s = absl::StrJoin(m, ",", absl::PairFormatter("="));
>+//   string s = absl::StrJoin(m, ",", absl::PairFormatter("="));
> //   EXPECT_EQ("a=1,b=2,c=3", s);
> //
> // Example 7:
> //   // These examples show how `absl::StrJoin()` handles a few common edge
> //   // cases:
>-//   std::vector<std::string> v_empty;
>+//   std::vector<string> v_empty;
> //   EXPECT_EQ("", absl::StrJoin(v_empty, "-"));
> //
>-//   std::vector<std::string> v_one_item = {"foo"};
>+//   std::vector<string> v_one_item = {"foo"};
> //   EXPECT_EQ("foo", absl::StrJoin(v_one_item, "-"));
> //
>-//   std::vector<std::string> v_empty_string = {""};
>+//   std::vector<string> v_empty_string = {""};
> //   EXPECT_EQ("", absl::StrJoin(v_empty_string, "-"));
> //
>-//   std::vector<std::string> v_one_item_empty_string = {"a", ""};
>+//   std::vector<string> v_one_item_empty_string = {"a", ""};
> //   EXPECT_EQ("a-", absl::StrJoin(v_one_item_empty_string, "-"));
> //
>-//   std::vector<std::string> v_two_empty_string = {"", ""};
>+//   std::vector<string> v_two_empty_string = {"", ""};
> //   EXPECT_EQ("-", absl::StrJoin(v_two_empty_string, "-"));
> //
> // Example 8:
> //   // Joins a `std::tuple<T...>` of heterogeneous types, converting each to
>-//   // a std::string using the `absl::AlphaNum` class.
>-//   std::string s = absl::StrJoin(std::make_tuple(123, "abc", 0.456), "-");
>+//   // a string using the `absl::AlphaNum` class.
>+//   string s = absl::StrJoin(std::make_tuple(123, "abc", 0.456), "-");
> //   EXPECT_EQ("123-abc-0.456", s);
> 
> template <typename Iterator, typename Formatter>
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace.h
>index f4d9bb9545d689762180a1a430e64b5065bcf27c..3bfe4c61ebd1504d3c8eb8e50ba870fa8fa68939 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace.h
>@@ -17,19 +17,19 @@
> // File: str_replace.h
> // -----------------------------------------------------------------------------
> //
>-// This file defines `absl::StrReplaceAll()`, a general-purpose std::string
>+// This file defines `absl::StrReplaceAll()`, a general-purpose string
> // replacement function designed for large, arbitrary text substitutions,
> // especially on strings which you are receiving from some other system for
> // further processing (e.g. processing regular expressions, escaping HTML
>-// entities, etc. `StrReplaceAll` is designed to be efficient even when only
>+// entities, etc.). `StrReplaceAll` is designed to be efficient even when only
> // one substitution is being performed, or when substitution is rare.
> //
>-// If the std::string being modified is known at compile-time, and the substitutions
>+// If the string being modified is known at compile-time, and the substitutions
> // vary, `absl::Substitute()` may be a better choice.
> //
> // Example:
> //
>-// std::string html_escaped = absl::StrReplaceAll(user_input, {
>+// string html_escaped = absl::StrReplaceAll(user_input, {
> //                                           {"&", "&amp;"},
> //                                           {"<", "&lt;"},
> //                                           {">", "&gt;"},
>@@ -49,16 +49,16 @@ namespace absl {
> 
> // StrReplaceAll()
> //
>-// Replaces character sequences within a given std::string with replacements provided
>+// Replaces character sequences within a given string with replacements provided
> // within an initializer list of key/value pairs. Candidate replacements are
>-// considered in order as they occur within the std::string, with earlier matches
>+// considered in order as they occur within the string, with earlier matches
> // taking precedence, and longer matches taking precedence for candidates
>-// starting at the same position in the std::string. Once a substitution is made, the
>+// starting at the same position in the string. Once a substitution is made, the
> // replaced text is not considered for any further substitutions.
> //
> // Example:
> //
>-//   std::string s = absl::StrReplaceAll("$who bought $count #Noun. Thanks $who!",
>+//   string s = absl::StrReplaceAll("$who bought $count #Noun. Thanks $who!",
> //                                  {{"$count", absl::StrCat(5)},
> //                                   {"$who", "Bob"},
> //                                   {"#Noun", "Apples"}});
>@@ -78,28 +78,28 @@ ABSL_MUST_USE_RESULT std::string StrReplaceAll(
> //   replacements["$who"] = "Bob";
> //   replacements["$count"] = "5";
> //   replacements["#Noun"] = "Apples";
>-//   std::string s = absl::StrReplaceAll("$who bought $count #Noun. Thanks $who!",
>+//   string s = absl::StrReplaceAll("$who bought $count #Noun. Thanks $who!",
> //                                  replacements);
> //   EXPECT_EQ("Bob bought 5 Apples. Thanks Bob!", s);
> //
> //   // A std::vector of std::pair elements can be more efficient.
>-//   std::vector<std::pair<const absl::string_view, std::string>> replacements;
>+//   std::vector<std::pair<const absl::string_view, string>> replacements;
> //   replacements.push_back({"&", "&amp;"});
> //   replacements.push_back({"<", "&lt;"});
> //   replacements.push_back({">", "&gt;"});
>-//   std::string s = absl::StrReplaceAll("if (ptr < &foo)",
>+//   string s = absl::StrReplaceAll("if (ptr < &foo)",
> //                                  replacements);
> //   EXPECT_EQ("if (ptr &lt; &amp;foo)", s);
> template <typename StrToStrMapping>
> std::string StrReplaceAll(absl::string_view s, const StrToStrMapping& replacements);
> 
> // Overload of `StrReplaceAll()` to replace character sequences within a given
>-// output std::string *in place* with replacements provided within an initializer
>+// output string *in place* with replacements provided within an initializer
> // list of key/value pairs, returning the number of substitutions that occurred.
> //
> // Example:
> //
>-//   std::string s = std::string("$who bought $count #Noun. Thanks $who!");
>+//   string s = std::string("$who bought $count #Noun. Thanks $who!");
> //   int count;
> //   count = absl::StrReplaceAll({{"$count", absl::StrCat(5)},
> //                               {"$who", "Bob"},
>@@ -112,12 +112,12 @@ int StrReplaceAll(
>     std::string* target);
> 
> // Overload of `StrReplaceAll()` to replace patterns within a given output
>-// std::string *in place* with replacements provided within a container of key/value
>+// string *in place* with replacements provided within a container of key/value
> // pairs.
> //
> // Example:
> //
>-//   std::string s = std::string("if (ptr < &foo)");
>+//   string s = std::string("if (ptr < &foo)");
> //   int count = absl::StrReplaceAll({{"&", "&amp;"},
> //                                    {"<", "&lt;"},
> //                                    {">", "&gt;"}}, &s);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace_benchmark.cc
>index e608de8d19e41ba73c3d54ffa37643c72be802db..8386f2e6bf10ece46ba479ae253d879667c5e1f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace_benchmark.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace_benchmark.cc
>@@ -38,16 +38,16 @@ struct Replacement {
>     {"liquor", "shakes"},    //
> };
> 
>-// Here, we set up a std::string for use in global-replace benchmarks.
>+// Here, we set up a string for use in global-replace benchmarks.
> // We started with a million blanks, and then deterministically insert
>-// 10,000 copies each of two pangrams.  The result is a std::string that is
>+// 10,000 copies each of two pangrams.  The result is a string that is
> // 40% blank space and 60% these words.  'the' occurs 18,247 times and
> // all the substitutions together occur 49,004 times.
> //
>-// We then create "after_replacing_the" to be a std::string that is a result of
>+// We then create "after_replacing_the" to be a string that is a result of
> // replacing "the" with "box" in big_string.
> //
>-// And then we create "after_replacing_many" to be a std::string that is result
>+// And then we create "after_replacing_many" to be a string that is result
> // of preferring several substitutions.
> void SetUpStrings() {
>   if (big_string == nullptr) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.cc
>index 0207213c26baf913d9981497943d7867a1cdeeea..0a68c52d14f19b88d3add2e37227c513c59d1246 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.cc
>@@ -43,10 +43,11 @@ absl::string_view GenericFind(absl::string_view text,
>   if (delimiter.empty() && text.length() > 0) {
>     // Special case for empty std::string delimiters: always return a zero-length
>     // absl::string_view referring to the item at position 1 past pos.
>-    return absl::string_view(text.begin() + pos + 1, 0);
>+    return absl::string_view(text.data() + pos + 1, 0);
>   }
>   size_t found_pos = absl::string_view::npos;
>-  absl::string_view found(text.end(), 0);  // By default, not found
>+  absl::string_view found(text.data() + text.size(),
>+                          0);  // By default, not found
>   found_pos = find_policy.Find(text, delimiter, pos);
>   if (found_pos != absl::string_view::npos) {
>     found = absl::string_view(text.data() + found_pos,
>@@ -87,7 +88,7 @@ absl::string_view ByString::Find(absl::string_view text, size_t pos) const {
>     // absl::string_view.
>     size_t found_pos = text.find(delimiter_[0], pos);
>     if (found_pos == absl::string_view::npos)
>-      return absl::string_view(text.end(), 0);
>+      return absl::string_view(text.data() + text.size(), 0);
>     return text.substr(found_pos, 1);
>   }
>   return GenericFind(text, delimiter_, pos, LiteralPolicy());
>@@ -100,7 +101,7 @@ absl::string_view ByString::Find(absl::string_view text, size_t pos) const {
> absl::string_view ByChar::Find(absl::string_view text, size_t pos) const {
>   size_t found_pos = text.find(c_, pos);
>   if (found_pos == absl::string_view::npos)
>-    return absl::string_view(text.end(), 0);
>+    return absl::string_view(text.data() + text.size(), 0);
>   return text.substr(found_pos, 1);
> }
> 
>@@ -128,9 +129,9 @@ absl::string_view ByLength::Find(absl::string_view text,
>   // If the std::string is shorter than the chunk size we say we
>   // "can't find the delimiter" so this will be the last chunk.
>   if (substr.length() <= static_cast<size_t>(length_))
>-    return absl::string_view(text.end(), 0);
>+    return absl::string_view(text.data() + text.size(), 0);
> 
>-  return absl::string_view(substr.begin() + length_, 0);
>+  return absl::string_view(substr.data() + length_, 0);
> }
> 
> }  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.h
>index 9a7be2b0534425872c25e986aebc21db7e7fb756..c7eb280c41130ab86e343369c7e5fc90cd69de26 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.h
>@@ -19,13 +19,13 @@
> //
> // This file contains functions for splitting strings. It defines the main
> // `StrSplit()` function, several delimiters for determining the boundaries on
>-// which to split the std::string, and predicates for filtering delimited results.
>+// which to split the string, and predicates for filtering delimited results.
> // `StrSplit()` adapts the returned collection to the type specified by the
> // caller.
> //
> // Example:
> //
>-//   // Splits the given std::string on commas. Returns the results in a
>+//   // Splits the given string on commas. Returns the results in a
> //   // vector of strings.
> //   std::vector<std::string> v = absl::StrSplit("a,b,c", ',');
> //   // Can also use ","
>@@ -55,7 +55,7 @@ namespace absl {
> //------------------------------------------------------------------------------
> //
> // `StrSplit()` uses delimiters to define the boundaries between elements in the
>-// provided input. Several `Delimiter` types are defined below. If a std::string
>+// provided input. Several `Delimiter` types are defined below. If a string
> // (`const char*`, `std::string`, or `absl::string_view`) is passed in place of
> // an explicit `Delimiter` object, `StrSplit()` treats it the same way as if it
> // were passed a `ByString` delimiter.
>@@ -65,7 +65,7 @@ namespace absl {
> //
> // The following `Delimiter` types are available for use within `StrSplit()`:
> //
>-//   - `ByString` (default for std::string arguments)
>+//   - `ByString` (default for string arguments)
> //   - `ByChar` (default for a char argument)
> //   - `ByAnyChar`
> //   - `ByLength`
>@@ -76,15 +76,15 @@ namespace absl {
> // be split and the position to begin searching for the next delimiter in the
> // input text. The returned absl::string_view should refer to the next
> // occurrence (after pos) of the represented delimiter; this returned
>-// absl::string_view represents the next location where the input std::string should
>+// absl::string_view represents the next location where the input string should
> // be broken. The returned absl::string_view may be zero-length if the Delimiter
>-// does not represent a part of the std::string (e.g., a fixed-length delimiter). If
>+// does not represent a part of the string (e.g., a fixed-length delimiter). If
> // no delimiter is found in the given text, a zero-length absl::string_view
> // referring to text.end() should be returned (e.g.,
> // absl::string_view(text.end(), 0)). It is important that the returned
> // absl::string_view always be within the bounds of input text given as an
>-// argument--it must not refer to a std::string that is physically located outside of
>-// the given std::string.
>+// argument--it must not refer to a string that is physically located outside of
>+// the given string.
> //
> // The following example is a simple Delimiter object that is created with a
> // single char and will look for that char in the text passed to the Find()
>@@ -104,13 +104,13 @@ namespace absl {
> 
> // ByString
> //
>-// A sub-std::string delimiter. If `StrSplit()` is passed a std::string in place of a
>-// `Delimiter` object, the std::string will be implicitly converted into a
>+// A sub-string delimiter. If `StrSplit()` is passed a string in place of a
>+// `Delimiter` object, the string will be implicitly converted into a
> // `ByString` delimiter.
> //
> // Example:
> //
>-//   // Because a std::string literal is converted to an `absl::ByString`,
>+//   // Because a string literal is converted to an `absl::ByString`,
> //   // the following two splits are equivalent.
> //
> //   std::vector<std::string> v1 = absl::StrSplit("a, b, c", ", ");
>@@ -131,7 +131,7 @@ class ByString {
> // ByChar
> //
> // A single character delimiter. `ByChar` is functionally equivalent to a
>-// 1-char std::string within a `ByString` delimiter, but slightly more
>+// 1-char string within a `ByString` delimiter, but slightly more
> // efficient.
> //
> // Example:
>@@ -164,9 +164,9 @@ class ByChar {
> // ByAnyChar
> //
> // A delimiter that will match any of the given byte-sized characters within
>-// its provided std::string.
>+// its provided string.
> //
>-// Note: this delimiter works with single-byte std::string data, but does not work
>+// Note: this delimiter works with single-byte string data, but does not work
> // with variable-width encodings, such as UTF-8.
> //
> // Example:
>@@ -175,8 +175,8 @@ class ByChar {
> //   std::vector<std::string> v = absl::StrSplit("a,b=c", ByAnyChar(",="));
> //   // v[0] == "a", v[1] == "b", v[2] == "c"
> //
>-// If `ByAnyChar` is given the empty std::string, it behaves exactly like
>-// `ByString` and matches each individual character in the input std::string.
>+// If `ByAnyChar` is given the empty string, it behaves exactly like
>+// `ByString` and matches each individual character in the input string.
> //
> class ByAnyChar {
>  public:
>@@ -192,7 +192,7 @@ class ByAnyChar {
> // A delimiter for splitting into equal-length strings. The length argument to
> // the constructor must be greater than 0.
> //
>-// Note: this delimiter works with single-byte std::string data, but does not work
>+// Note: this delimiter works with single-byte string data, but does not work
> // with variable-width encodings, such as UTF-8.
> //
> // Example:
>@@ -202,7 +202,7 @@ class ByAnyChar {
> 
> //   // v[0] == "123", v[1] == "456", v[2] == "789"
> //
>-// Note that the std::string does not have to be a multiple of the fixed split
>+// Note that the string does not have to be a multiple of the fixed split
> // length. In such a case, the last substring will be shorter.
> //
> //   using absl::ByLength;
>@@ -223,9 +223,9 @@ namespace strings_internal {
> // A traits-like metafunction for selecting the default Delimiter object type
> // for a particular Delimiter type. The base case simply exposes type Delimiter
> // itself as the delimiter's Type. However, there are specializations for
>-// std::string-like objects that map them to the ByString delimiter object.
>+// string-like objects that map them to the ByString delimiter object.
> // This allows functions like absl::StrSplit() and absl::MaxSplits() to accept
>-// std::string-like objects (e.g., ',') as delimiter arguments but they will be
>+// string-like objects (e.g., ',') as delimiter arguments but they will be
> // treated as if a ByString delimiter was given.
> template <typename Delimiter>
> struct SelectDelimiter {
>@@ -261,7 +261,8 @@ class MaxSplitsImpl {
>       : delimiter_(delimiter), limit_(limit), count_(0) {}
>   absl::string_view Find(absl::string_view text, size_t pos) {
>     if (count_++ == limit_) {
>-      return absl::string_view(text.end(), 0);  // No more matches.
>+      return absl::string_view(text.data() + text.size(),
>+                               0);  // No more matches.
>     }
>     return delimiter_.Find(text, pos);
>   }
>@@ -331,7 +332,7 @@ struct AllowEmpty {
> // SkipEmpty()
> //
> // Returns `false` if the given `absl::string_view` is empty, indicating that
>-// `StrSplit()` should omit the empty std::string.
>+// `StrSplit()` should omit the empty string.
> //
> // Example:
> //
>@@ -339,7 +340,7 @@ struct AllowEmpty {
> //
> //   // v[0] == "a", v[1] == "b"
> //
>-// Note: `SkipEmpty()` does not consider a std::string containing only whitespace
>+// Note: `SkipEmpty()` does not consider a string containing only whitespace
> // to be empty. To skip such whitespace as well, use the `SkipWhitespace()`
> // predicate.
> struct SkipEmpty {
>@@ -349,7 +350,7 @@ struct SkipEmpty {
> // SkipWhitespace()
> //
> // Returns `false` if the given `absl::string_view` is empty *or* contains only
>-// whitespace, indicating that `StrSplit()` should omit the std::string.
>+// whitespace, indicating that `StrSplit()` should omit the string.
> //
> // Example:
> //
>@@ -373,7 +374,7 @@ struct SkipWhitespace {
> 
> // StrSplit()
> //
>-// Splits a given std::string based on the provided `Delimiter` object, returning the
>+// Splits a given string based on the provided `Delimiter` object, returning the
> // elements within the type specified by the caller. Optionally, you may pass a
> // `Predicate` to `StrSplit()` indicating whether to include or exclude the
> // resulting element within the final result set. (See the overviews for
>@@ -412,7 +413,7 @@ struct SkipWhitespace {
> //
> // The `StrSplit()` function adapts the returned collection to the collection
> // specified by the caller (e.g. `std::vector` above). The returned collections
>-// may contain `string`, `absl::string_view` (in which case the original std::string
>+// may contain `string`, `absl::string_view` (in which case the original string
> // being split must ensure that it outlives the collection), or any object that
> // can be explicitly created from an `absl::string_view`. This behavior works
> // for:
>@@ -424,7 +425,7 @@ struct SkipWhitespace {
> // Example:
> //
> //   // The results are returned as `absl::string_view` objects. Note that we
>-//   // have to ensure that the input std::string outlives any results.
>+//   // have to ensure that the input string outlives any results.
> //   std::vector<absl::string_view> v = absl::StrSplit("a,b,c", ',');
> //
> //   // Stores results in a std::set<std::string>, which also performs
>@@ -444,7 +445,7 @@ struct SkipWhitespace {
> //   // is provided as a series of key/value pairs. For example, the 0th element
> //   // resulting from the split will be stored as a key to the 1st element. If
> //   // an odd number of elements are resolved, the last element is paired with
>-//   // a default-constructed value (e.g., empty std::string).
>+//   // a default-constructed value (e.g., empty string).
> //   std::map<std::string, std::string> m = absl::StrSplit("a,b,c", ',');
> //   // m["a"] == "b", m["c"] == ""     // last component value equals ""
> //
>@@ -452,14 +453,14 @@ struct SkipWhitespace {
> // elements and is not a collection type. When splitting to a `std::pair` the
> // first two split strings become the `std::pair` `.first` and `.second`
> // members, respectively. The remaining split substrings are discarded. If there
>-// are less than two split substrings, the empty std::string is used for the
>+// are less than two split substrings, the empty string is used for the
> // corresponding
> // `std::pair` member.
> //
> // Example:
> //
> //   // Stores first two split strings as the members in a std::pair.
>-//   std::pair<std::string, std::string> p = absl::StrSplit("a,b,c", ',');
>+//   std::pair<string, string> p = absl::StrSplit("a,b,c", ',');
> //   // p.first == "a", p.second == "b"       // "c" is omitted.
> //
> // The `StrSplit()` function can be used multiple times to perform more
>@@ -467,9 +468,9 @@ struct SkipWhitespace {
> //
> // Example:
> //
>-//   // The input std::string "a=b=c,d=e,f=,g" becomes
>+//   // The input string "a=b=c,d=e,f=,g" becomes
> //   // { "a" => "b=c", "d" => "e", "f" => "", "g" => "" }
>-//   std::map<std::string, std::string> m;
>+//   std::map<string, string> m;
> //   for (absl::string_view sp : absl::StrSplit("a=b=c,d=e,f=,g", ',')) {
> //     m.insert(absl::StrSplit(sp, absl::MaxSplits('=', 1)));
> //   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split_test.cc
>index 29a68047c99aef31bc7d3673a32b7228542c79a3..413ad318bf9b29337b7e88788a550a9db3e50eae 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split_test.cc
>@@ -276,7 +276,7 @@ TEST(SplitIterator, Basics) {
>   EXPECT_EQ(it, end);
> }
> 
>-// Simple Predicate to skip a particular std::string.
>+// Simple Predicate to skip a particular string.
> class Skip {
>  public:
>   explicit Skip(const std::string& s) : s_(s) {}
>@@ -763,12 +763,12 @@ template <typename Delimiter>
> static bool IsFoundAtStartingPos(absl::string_view text, Delimiter d,
>                                  size_t starting_pos, int expected_pos) {
>   absl::string_view found = d.Find(text, starting_pos);
>-  return found.data() != text.end() &&
>+  return found.data() != text.data() + text.size() &&
>          expected_pos == found.data() - text.data();
> }
> 
> // Helper function for testing Delimiter objects. Returns true if the given
>-// Delimiter is found in the given std::string at the given position. This function
>+// Delimiter is found in the given string at the given position. This function
> // tests two cases:
> //   1. The actual text given, staring at position 0
> //   2. The text given with leading padding that should be ignored
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view.h
>index 1b8f8a374745cb6466b1cd622197fdc2f6dcd9d5..ec9f62be15733e3cde8256df1cfc520c22e11f0d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view.h
>@@ -19,7 +19,7 @@
> //
> // This file contains the definition of the `absl::string_view` class. A
> // `string_view` points to a contiguous span of characters, often part or all of
>-// another `std::string`, double-quoted std::string literal, character array, or even
>+// another `std::string`, double-quoted string literal, character array, or even
> // another `string_view`.
> //
> // This `absl::string_view` abstraction is designed to be a drop-in
>@@ -56,15 +56,15 @@ namespace absl {
> 
> // absl::string_view
> //
>-// A `string_view` provides a lightweight view into the std::string data provided by
>-// a `std::string`, double-quoted std::string literal, character array, or even
>-// another `string_view`. A `string_view` does *not* own the std::string to which it
>+// A `string_view` provides a lightweight view into the string data provided by
>+// a `std::string`, double-quoted string literal, character array, or even
>+// another `string_view`. A `string_view` does *not* own the string to which it
> // points, and that data cannot be modified through the view.
> //
> // You can use `string_view` as a function or method parameter anywhere a
>-// parameter can receive a double-quoted std::string literal, `const char*`,
>+// parameter can receive a double-quoted string literal, `const char*`,
> // `std::string`, or another `absl::string_view` argument with no need to copy
>-// the std::string data. Systematic use of `string_view` within function arguments
>+// the string data. Systematic use of `string_view` within function arguments
> // reduces data copies and `strlen()` calls.
> //
> // Because of its small size, prefer passing `string_view` by value:
>@@ -97,8 +97,8 @@ namespace absl {
> // `string_view` this way, it is your responsibility to ensure that the object
> // pointed to by the `string_view` outlives the `string_view`.
> //
>-// A `string_view` may represent a whole std::string or just part of a std::string. For
>-// example, when splitting a std::string, `std::vector<absl::string_view>` is a
>+// A `string_view` may represent a whole string or just part of a string. For
>+// example, when splitting a string, `std::vector<absl::string_view>` is a
> // natural data type for the output.
> //
> //
>@@ -141,7 +141,7 @@ namespace absl {
> // All empty `string_view` objects whether null or not, are equal:
> //
> //   absl::string_view() == absl::string_view("", 0)
>-//   absl::string_view(nullptr, 0) == absl:: string_view("abcdef"+6, 0)
>+//   absl::string_view(nullptr, 0) == absl::string_view("abcdef"+6, 0)
> class string_view {
>  public:
>   using traits_type = std::char_traits<char>;
>@@ -173,8 +173,19 @@ class string_view {
>   // Implicit constructor of a `string_view` from nul-terminated `str`. When
>   // accepting possibly null strings, use `absl::NullSafeStringView(str)`
>   // instead (see below).
>+#if ABSL_HAVE_BUILTIN(__builtin_strlen) || \
>+    (defined(__GNUC__) && !defined(__clang__))
>+  // GCC has __builtin_strlen according to
>+  // https://gcc.gnu.org/onlinedocs/gcc-4.7.0/gcc/Other-Builtins.html, but
>+  // ABSL_HAVE_BUILTIN doesn't detect that, so we use the extra checks above.
>+  // __builtin_strlen is constexpr.
>+  constexpr string_view(const char* str)  // NOLINT(runtime/explicit)
>+      : ptr_(str),
>+        length_(CheckLengthInternal(str ? __builtin_strlen(str) : 0)) {}
>+#else
>   constexpr string_view(const char* str)  // NOLINT(runtime/explicit)
>-      : ptr_(str), length_(CheckLengthInternal(StrLenInternal(str))) {}
>+      : ptr_(str), length_(CheckLengthInternal(str ? strlen(str) : 0)) {}
>+#endif
> 
>   // Implicit constructor of a `string_view` from a `const char*` and length.
>   constexpr string_view(const char* data, size_type len)
>@@ -340,7 +351,7 @@ class string_view {
>   //
>   // Returns a "substring" of the `string_view` (at offset `pos` and length
>   // `n`) as another string_view. This function throws `std::out_of_bounds` if
>-  // `pos > size'.
>+  // `pos > size`.
>   string_view substr(size_type pos, size_type n = npos) const {
>     if (ABSL_PREDICT_FALSE(pos > length_))
>       base_internal::ThrowStdOutOfRange("absl::string_view::substr");
>@@ -351,7 +362,7 @@ class string_view {
>   // string_view::compare()
>   //
>   // Performs a lexicographical comparison between the `string_view` and
>-  // another `absl::string_view), returning -1 if `this` is less than, 0 if
>+  // another `absl::string_view`, returning -1 if `this` is less than, 0 if
>   // `this` is equal to, and 1 if `this` is greater than the passed std::string
>   // view. Note that in the case of data equality, a further comparison is made
>   // on the respective sizes of the two `string_view`s to determine which is
>@@ -481,22 +492,6 @@ class string_view {
>   static constexpr size_type kMaxSize =
>       std::numeric_limits<difference_type>::max();
> 
>-  // check whether __builtin_strlen is provided by the compiler.
>-  // GCC doesn't have __has_builtin()
>-  // (https://gcc.gnu.org/bugzilla/show_bug.cgi?id=66970),
>-  // but has __builtin_strlen according to
>-  // https://gcc.gnu.org/onlinedocs/gcc-4.7.0/gcc/Other-Builtins.html.
>-#if ABSL_HAVE_BUILTIN(__builtin_strlen) || \
>-    (defined(__GNUC__) && !defined(__clang__))
>-  static constexpr size_type StrLenInternal(const char* str) {
>-    return str ? __builtin_strlen(str) : 0;
>-  }
>-#else
>-  static constexpr size_type StrLenInternal(const char* str) {
>-    return str ? strlen(str) : 0;
>-  }
>-#endif
>-
>   static constexpr size_type CheckLengthInternal(size_type len) {
>     return len;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view_test.cc
>index b19d07c7fd61d821223bfc0bf97b2a2a6234d38c..a94e82254ee4521dfc11b05aa89d0e741ca5b7ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view_test.cc
>@@ -35,7 +35,8 @@
> #define ABSL_EXPECT_DEATH_IF_SUPPORTED(statement, regex) \
>   EXPECT_DEATH_IF_SUPPORTED(statement, ".*")
> #else
>-#define ABSL_EXPECT_DEATH_IF_SUPPORTED EXPECT_DEATH_IF_SUPPORTED
>+#define ABSL_EXPECT_DEATH_IF_SUPPORTED(statement, regex) \
>+  EXPECT_DEATH_IF_SUPPORTED(statement, regex)
> #endif
> 
> namespace {
>@@ -283,7 +284,7 @@ TEST(StringViewTest, ComparisonOperatorsByCharacterPosition) {
> }
> #undef COMPARE
> 
>-// Sadly, our users often confuse std::string::npos with absl::string_view::npos;
>+// Sadly, our users often confuse string::npos with absl::string_view::npos;
> // So much so that we test here that they are the same.  They need to
> // both be unsigned, and both be the maximum-valued integer of their type.
> 
>@@ -811,15 +812,18 @@ TEST(StringViewTest, FrontBackSingleChar) {
> }
> 
> // `std::string_view::string_view(const char*)` calls
>-// `std::char_traits<char>::length(const char*)` to get the std::string length. In
>+// `std::char_traits<char>::length(const char*)` to get the string length. In
> // libc++, it doesn't allow `nullptr` in the constexpr context, with the error
> // "read of dereferenced null pointer is not allowed in a constant expression".
> // At run time, the behavior of `std::char_traits::length()` on `nullptr` is
>-// undefined by the standard and usually results in crash with libc++. This
>-// conforms to the standard, but `absl::string_view` implements a different
>+// undefined by the standard and usually results in crash with libc++.
>+// In MSVC, creating a constexpr string_view from nullptr also triggers an
>+// "unevaluable pointer value" error. This compiler implementation conforms
>+// to the standard, but `absl::string_view` implements a different
> // behavior for historical reasons. We work around tests that construct
> // `string_view` from `nullptr` when using libc++.
>-#if !defined(ABSL_HAVE_STD_STRING_VIEW) || !defined(_LIBCPP_VERSION)
>+#if !defined(ABSL_HAVE_STD_STRING_VIEW) || \
>+    (!defined(_LIBCPP_VERSION) && !defined(_MSC_VER))
> #define ABSL_HAVE_STRING_VIEW_FROM_NULLPTR 1
> #endif  // !defined(ABSL_HAVE_STD_STRING_VIEW) || !defined(_LIBCPP_VERSION)
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip.h
>index 2f8d21f7deb9fca02938ca7a9240a0e17756f211..8d0d7c6bfc1230f0c39a151218e1cc927b8836bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip.h
>@@ -17,7 +17,7 @@
> // File: strip.h
> // -----------------------------------------------------------------------------
> //
>-// This file contains various functions for stripping substrings from a std::string.
>+// This file contains various functions for stripping substrings from a string.
> #ifndef ABSL_STRINGS_STRIP_H_
> #define ABSL_STRINGS_STRIP_H_
> 
>@@ -33,7 +33,7 @@ namespace absl {
> 
> // ConsumePrefix()
> //
>-// Strips the `expected` prefix from the start of the given std::string, returning
>+// Strips the `expected` prefix from the start of the given string, returning
> // `true` if the strip operation succeeded or false otherwise.
> //
> // Example:
>@@ -48,7 +48,7 @@ inline bool ConsumePrefix(absl::string_view* str, absl::string_view expected) {
> }
> // ConsumeSuffix()
> //
>-// Strips the `expected` suffix from the end of the given std::string, returning
>+// Strips the `expected` suffix from the end of the given string, returning
> // `true` if the strip operation succeeded or false otherwise.
> //
> // Example:
>@@ -64,9 +64,9 @@ inline bool ConsumeSuffix(absl::string_view* str, absl::string_view expected) {
> 
> // StripPrefix()
> //
>-// Returns a view into the input std::string 'str' with the given 'prefix' removed,
>-// but leaving the original std::string intact. If the prefix does not match at the
>-// start of the std::string, returns the original std::string instead.
>+// Returns a view into the input string 'str' with the given 'prefix' removed,
>+// but leaving the original string intact. If the prefix does not match at the
>+// start of the string, returns the original string instead.
> ABSL_MUST_USE_RESULT inline absl::string_view StripPrefix(
>     absl::string_view str, absl::string_view prefix) {
>   if (absl::StartsWith(str, prefix)) str.remove_prefix(prefix.size());
>@@ -75,9 +75,9 @@ ABSL_MUST_USE_RESULT inline absl::string_view StripPrefix(
> 
> // StripSuffix()
> //
>-// Returns a view into the input std::string 'str' with the given 'suffix' removed,
>-// but leaving the original std::string intact. If the suffix does not match at the
>-// end of the std::string, returns the original std::string instead.
>+// Returns a view into the input string 'str' with the given 'suffix' removed,
>+// but leaving the original string intact. If the suffix does not match at the
>+// end of the string, returns the original string instead.
> ABSL_MUST_USE_RESULT inline absl::string_view StripSuffix(
>     absl::string_view str, absl::string_view suffix) {
>   if (absl::EndsWith(str, suffix)) str.remove_suffix(suffix.size());
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip_test.cc
>index 205c160c1929263796a59323ac59509e8ab6589e..40c4c6071211208104c7ddb2fdcf376519924347 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip_test.cc
>@@ -12,8 +12,8 @@
> // See the License for the specific language governing permissions and
> // limitations under the License.
> 
>-// This file contains functions that remove a defined part from the std::string,
>-// i.e., strip the std::string.
>+// This file contains functions that remove a defined part from the string,
>+// i.e., strip the string.
> 
> #include "absl/strings/strip.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute.h
>index c4b25ba70952c19d64ebc849b7e3c09cafbc52cf..4de7b4e75ff23b41341fd635cf221da669bd51d4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute.h
>@@ -17,46 +17,46 @@
> // File: substitute.h
> // -----------------------------------------------------------------------------
> //
>-// This package contains functions for efficiently performing std::string
>-// substitutions using a format std::string with positional notation:
>+// This package contains functions for efficiently performing string
>+// substitutions using a format string with positional notation:
> // `Substitute()` and `SubstituteAndAppend()`.
> //
> // Unlike printf-style format specifiers, `Substitute()` functions do not need
> // to specify the type of the substitution arguments. Supported arguments
>-// following the format std::string, such as strings, string_views, ints,
>+// following the format string, such as strings, string_views, ints,
> // floats, and bools, are automatically converted to strings during the
> // substitution process. (See below for a full list of supported types.)
> //
> // `Substitute()` does not allow you to specify *how* to format a value, beyond
>-// the default conversion to std::string. For example, you cannot format an integer
>+// the default conversion to string. For example, you cannot format an integer
> // in hex.
> //
>-// The format std::string uses positional identifiers indicated by a dollar sign ($)
>+// The format string uses positional identifiers indicated by a dollar sign ($)
> // and single digit positional ids to indicate which substitution arguments to
>-// use at that location within the format std::string.
>+// use at that location within the format string.
> //
> // Example 1:
>-//   std::string s = Substitute("$1 purchased $0 $2. Thanks $1!",
>+//   string s = Substitute("$1 purchased $0 $2. Thanks $1!",
> //                         5, "Bob", "Apples");
> //   EXPECT_EQ("Bob purchased 5 Apples. Thanks Bob!", s);
> //
> // Example 2:
>-//   std::string s = "Hi. ";
>+//   string s = "Hi. ";
> //   SubstituteAndAppend(&s, "My name is $0 and I am $1 years old.", "Bob", 5);
> //   EXPECT_EQ("Hi. My name is Bob and I am 5 years old.", s);
> //
> //
> // Supported types:
>-//   * absl::string_view, std::string, const char* (null is equivalent to "")
>+//   * absl::string_view, string, const char* (null is equivalent to "")
> //   * int32_t, int64_t, uint32_t, uint64
> //   * float, double
> //   * bool (Printed as "true" or "false")
>-//   * pointer types other than char* (Printed as "0x<lower case hex std::string>",
>+//   * pointer types other than char* (Printed as "0x<lower case hex string>",
> //     except that null is printed as "NULL")
> //
>-// If an invalid format std::string is provided, Substitute returns an empty std::string
>-// and SubstituteAndAppend does not change the provided output std::string.
>-// A format std::string is invalid if it:
>+// If an invalid format string is provided, Substitute returns an empty string
>+// and SubstituteAndAppend does not change the provided output string.
>+// A format string is invalid if it:
> //   * ends in an unescaped $ character,
> //     e.g. "Hello $", or
> //   * calls for a position argument which is not provided,
>@@ -88,7 +88,7 @@ namespace substitute_internal {
> //
> // This class provides an argument type for `absl::Substitute()` and
> // `absl::SubstituteAndAppend()`. `Arg` handles implicit conversion of various
>-// types to a std::string. (`Arg` is very similar to the `AlphaNum` class in
>+// types to a string. (`Arg` is very similar to the `AlphaNum` class in
> // `StrCat()`.)
> //
> // This class has implicit constructors.
>@@ -197,8 +197,8 @@ constexpr int PlaceholderBitmask(const char* format) {
> 
> // SubstituteAndAppend()
> //
>-// Substitutes variables into a given format std::string and appends to a given
>-// output std::string. See file comments above for usage.
>+// Substitutes variables into a given format string and appends to a given
>+// output string. See file comments above for usage.
> //
> // The declarations of `SubstituteAndAppend()` below consist of overloads
> // for passing 0 to 10 arguments, respectively.
>@@ -444,7 +444,7 @@ void SubstituteAndAppend(
> 
> // Substitute()
> //
>-// Substitutes variables into a given format std::string. See file comments above
>+// Substitutes variables into a given format string. See file comments above
> // for usage.
> //
> // The declarations of `Substitute()` below consist of overloads for passing 0
>@@ -456,7 +456,7 @@ void SubstituteAndAppend(
> // Example:
> //  template <typename... Args>
> //  void VarMsg(absl::string_view format, const Args&... args) {
>-//    std::string s = absl::Substitute(format, args...);
>+//    string s = absl::Substitute(format, args...);
> 
> ABSL_MUST_USE_RESULT inline std::string Substitute(absl::string_view format) {
>   std::string result;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.bazel
>index 8d302e01223b45bd92a21670a11f5592f9b15c64..f52e9d41644a8d931bf39fae7ec84327beaf41e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.bazel
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.bazel
>@@ -88,6 +88,9 @@ cc_test(
>     size = "small",
>     srcs = ["barrier_test.cc"],
>     copts = ABSL_TEST_COPTS,
>+    tags = [
>+        "no_test_wasm",
>+    ],
>     deps = [
>         ":synchronization",
>         "//absl/time",
>@@ -100,6 +103,9 @@ cc_test(
>     size = "small",
>     srcs = ["blocking_counter_test.cc"],
>     copts = ABSL_TEST_COPTS,
>+    tags = [
>+        "no_test_wasm",
>+    ],
>     deps = [
>         ":synchronization",
>         "//absl/time",
>@@ -138,6 +144,9 @@ cc_library(
>     name = "thread_pool",
>     testonly = 1,
>     hdrs = ["internal/thread_pool.h"],
>+    visibility = [
>+        "//absl:__subpackages__",
>+    ],
>     deps = [
>         ":synchronization",
>         "//absl/base:core_headers",
>@@ -149,6 +158,7 @@ cc_test(
>     size = "large",
>     srcs = ["mutex_test.cc"],
>     copts = ABSL_TEST_COPTS,
>+    shard_count = 25,
>     deps = [
>         ":synchronization",
>         ":thread_pool",
>@@ -205,6 +215,7 @@ cc_test(
>     name = "per_thread_sem_test",
>     size = "medium",
>     copts = ABSL_TEST_COPTS,
>+    tags = ["no_test_wasm"],
>     deps = [
>         ":per_thread_sem_test_common",
>         ":synchronization",
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.gn
>index 3664aa13274f4040f9369b95f40deef2b590db8f..e6f763a060ba9ce0aaee9c141810fff6b4e78c11 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.gn
>@@ -50,8 +50,8 @@ source_set("synchronization") {
>     "internal/create_thread_identity.cc",
>     "internal/per_thread_sem.cc",
>     "internal/waiter.cc",
>+    "mutex.cc",
>     "notification.cc",
>-    "mutex.cc"
>   ]
>   public = [
>     "barrier.h",
>@@ -93,6 +93,8 @@ source_set("thread_pool") {
>     ":synchronization",
>     "../base:core_headers",
>   ]
>+  visibility = []
>+  visibility += [ "../*" ]
> }
> 
> source_set("per_thread_sem_test_common") {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles.cc
>index ab1f3f84e74c1d7edef6379e02684b00b53a7e34..d3878de2b68f21929f00b8634bcb9794a0e4f086 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles.cc
>@@ -204,8 +204,7 @@ class NodeSet {
>   }
> 
>  private:
>-  static const int32_t kEmpty;
>-  static const int32_t kDel;
>+  enum : int32_t { kEmpty = -1, kDel = -2 };
>   Vec<int32_t> table_;
>   uint32_t occupied_;     // Count of non-empty slots (includes deleted slots)
> 
>@@ -255,9 +254,6 @@ class NodeSet {
>   NodeSet& operator=(const NodeSet&) = delete;
> };
> 
>-const int32_t NodeSet::kEmpty = -1;
>-const int32_t NodeSet::kDel = -2;
>-
> // We encode a node index and a node version in GraphId.  The version
> // number is incremented when the GraphId is freed which automatically
> // invalidates all copies of the GraphId.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem_test.cc
>index 2b52ea76ab0e942601eb50f160fe845b9b0685cf..c29d8403df43fbc3d8d2c5c7ccdaf3aeed2f2ad4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem_test.cc
>@@ -153,12 +153,15 @@ TEST_F(PerThreadSemTest, WithTimeout) {
> 
> TEST_F(PerThreadSemTest, Timeouts) {
>   absl::Time timeout = absl::Now() + absl::Milliseconds(50);
>+  // Allow for a slight early return, to account for quality of implementation
>+  // issues on various platforms.
>+  const absl::Duration slop = absl::Microseconds(200);
>   EXPECT_FALSE(Wait(timeout));
>-  EXPECT_LE(timeout, absl::Now());
>+  EXPECT_LE(timeout, absl::Now() + slop);
> 
>   absl::Time negative_timeout = absl::UnixEpoch() - absl::Milliseconds(100);
>   EXPECT_FALSE(Wait(negative_timeout));
>-  EXPECT_LE(negative_timeout, absl::Now());  // trivially true :)
>+  EXPECT_LE(negative_timeout, absl::Now() + slop);  // trivially true :)
> 
>   Post(GetOrCreateCurrentThreadIdentity());
>   // The wait here has an expired timeout, but we have a wake to consume,
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.cc
>index 80f34f035fc6b679eeb0b14d89cfaeaa1f80f051..9d59a79d92d9f7928f3dfe2a730434e154acec47 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.cc
>@@ -298,7 +298,7 @@ static struct SynchEvent {     // this is a trivial hash table for the events
> // set "bits" in the word there (waiting until lockbit is clear before doing
> // so), and return a refcounted reference that will remain valid until
> // UnrefSynchEvent() is called.  If a new SynchEvent is allocated,
>-// the std::string name is copied into it.
>+// the string name is copied into it.
> // When used with a mutex, the caller should also ensure that kMuEvent
> // is set in the mutex word, and similarly for condition variables and kCVEvent.
> static SynchEvent *EnsureSynchEvent(std::atomic<intptr_t> *addr,
>@@ -1827,8 +1827,8 @@ bool Mutex::LockSlowWithDeadline(MuHow how, const Condition *cond,
>          cond == nullptr || EvalConditionAnnotated(cond, this, true, how);
> }
> 
>-// RAW_CHECK_FMT() takes a condition, a printf-style format std::string, and
>-// the printf-style argument list.   The format std::string must be a literal.
>+// RAW_CHECK_FMT() takes a condition, a printf-style format string, and
>+// the printf-style argument list.   The format string must be a literal.
> // Arguments after the first are not evaluated unless the condition is true.
> #define RAW_CHECK_FMT(cond, ...)                                   \
>   do {                                                             \
>@@ -1975,7 +1975,7 @@ void Mutex::LockSlowLoop(SynchWaitParams *waitp, int flags) {
> // Unlock this mutex, which is held by the current thread.
> // If waitp is non-zero, it must be the wait parameters for the current thread
> // which holds the lock but is not runnable because its condition is false
>-// or it n the process of blocking on a condition variable; it must requeue
>+// or it is in the process of blocking on a condition variable; it must requeue
> // itself on the mutex/condvar to wait for its condition to become true.
> void Mutex::UnlockSlow(SynchWaitParams *waitp) {
>   intptr_t v = mu_.load(std::memory_order_relaxed);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.h
>index 83c214867ac53a9122dd4e9753fc8bf72e128be5..a378190538e2d236eaf399315007d00872e20869 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.h
>@@ -962,7 +962,7 @@ void RegisterMutexTracer(void (*fn)(const char *msg, const void *obj,
> //
> // The function pointer registered here will be called here on various CondVar
> // events.  The callback is given an opaque handle to the CondVar object and
>-// a std::string identifying the event.  This is thread-safe, but only a single
>+// a string identifying the event.  This is thread-safe, but only a single
> // tracer can be registered.
> //
> // Events that can be sent are "Wait", "Unwait", "Signal wakeup", and
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification_test.cc
>index 9b3b6a5a9e84ec1a3655a2c8f43eb8fbc5832ef6..d8708d551a1575746ea2a85426972fd981911016 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification_test.cc
>@@ -71,10 +71,13 @@ static void BasicTests(bool notify_before_waiting, Notification* notification) {
>       notification->WaitForNotificationWithTimeout(absl::Milliseconds(0)));
>   EXPECT_FALSE(notification->WaitForNotificationWithDeadline(absl::Now()));
> 
>+  const absl::Duration delay = absl::Milliseconds(50);
>+  // Allow for a slight early return, to account for quality of implementation
>+  // issues on various platforms.
>+  const absl::Duration slop = absl::Microseconds(200);
>   absl::Time start = absl::Now();
>-  EXPECT_FALSE(
>-      notification->WaitForNotificationWithTimeout(absl::Milliseconds(50)));
>-  EXPECT_LE(start + absl::Milliseconds(50), absl::Now());
>+  EXPECT_FALSE(notification->WaitForNotificationWithTimeout(delay));
>+  EXPECT_LE(start + delay, absl::Now() + slop);
> 
>   ThreadSafeCounter ready_counter;
>   ThreadSafeCounter done_counter;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/BUILD.gn
>index 9927af8343cb16099e022cd3780f486e7afa411d..5758e9dc1e3c3dc45f28a525ffe6013a83314749 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/BUILD.gn
>@@ -62,8 +62,9 @@ source_set("test_util") {
>     ":time",
>     "../base",
>     "../time/internal/cctz:time_zone",
>-    "//testing/gtest",
>     "//testing/gmock",
>+    "//testing/gtest",
>+    "//third_party/googletest/:gmock",
>   ]
>   visibility = []
>   visibility += [ "../time:*" ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration.cc
>index f402137b0a650d2b7ed605262fd954062aab29a0..2950c7cdc6320b4d3f8434b1e2b01ad56d72db40 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration.cc
>@@ -666,7 +666,7 @@ std::chrono::hours ToChronoHours(Duration d) {
> }
> 
> //
>-// To/From std::string formatting.
>+// To/From string formatting.
> //
> 
> namespace {
>@@ -744,7 +744,7 @@ void AppendNumberUnit(std::string* out, double n, DisplayUnit unit) {
> }  // namespace
> 
> // From Go's doc at http://golang.org/pkg/time/#Duration.String
>-//   [FormatDuration] returns a std::string representing the duration in the
>+//   [FormatDuration] returns a string representing the duration in the
> //   form "72h3m0.5s".  Leading zero units are omitted.  As a special
> //   case, durations less than one second format use a smaller unit
> //   (milli-, micro-, or nanoseconds) to ensure that the leading digit
>@@ -787,8 +787,8 @@ std::string FormatDuration(Duration d) {
> namespace {
> 
> // A helper for ParseDuration() that parses a leading number from the given
>-// std::string and stores the result in *int_part/*frac_part/*frac_scale.  The
>-// given std::string pointer is modified to point to the first unconsumed char.
>+// string and stores the result in *int_part/*frac_part/*frac_scale.  The
>+// given string pointer is modified to point to the first unconsumed char.
> bool ConsumeDurationNumber(const char** dpp, int64_t* int_part,
>                            int64_t* frac_part, int64_t* frac_scale) {
>   *int_part = 0;
>@@ -816,8 +816,8 @@ bool ConsumeDurationNumber(const char** dpp, int64_t* int_part,
> }
> 
> // A helper for ParseDuration() that parses a leading unit designator (e.g.,
>-// ns, us, ms, s, m, h) from the given std::string and stores the resulting unit
>-// in "*unit".  The given std::string pointer is modified to point to the first
>+// ns, us, ms, s, m, h) from the given string and stores the resulting unit
>+// in "*unit".  The given string pointer is modified to point to the first
> // unconsumed char.
> bool ConsumeDurationUnit(const char** start, Duration* unit) {
>   const char *s = *start;
>@@ -850,7 +850,7 @@ bool ConsumeDurationUnit(const char** start, Duration* unit) {
> }  // namespace
> 
> // From Go's doc at http://golang.org/pkg/time/#ParseDuration
>-//   [ParseDuration] parses a duration std::string.  A duration std::string is
>+//   [ParseDuration] parses a duration string.  A duration string is
> //   a possibly signed sequence of decimal numbers, each with optional
> //   fraction and a unit suffix, such as "300ms", "-1.5h" or "2h45m".
> //   Valid time units are "ns", "us" "ms", "s", "m", "h".
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format.cc
>index e98e60a372bfc3a4472c8a17ac7a5383782b887a..ee597e407a9ddd226a8f880659f2d50427f9192b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format.cc
>@@ -88,7 +88,7 @@ bool ParseTime(const std::string& format, const std::string& input, absl::Time*
>   return absl::ParseTime(format, input, absl::UTCTimeZone(), time, err);
> }
> 
>-// If the input std::string does not contain an explicit UTC offset, interpret
>+// If the input string does not contain an explicit UTC offset, interpret
> // the fields with respect to the given TimeZone.
> bool ParseTime(const std::string& format, const std::string& input, absl::TimeZone tz,
>                absl::Time* time, std::string* err) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time.h
>index 898222b4c7af29830ea20507d9c294f2d0ef969c..0842fa4a468952fbb71fe9e98b883e2fa47d96e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time.h
>@@ -59,7 +59,7 @@ namespace cctz {
> // inferior fields to their minimum valid value (as described above). The
> // following are examples of how each of the six types would align the fields
> // representing November 22, 2015 at 12:34:56 in the afternoon. (Note: the
>-// std::string format used here is not important; it's just a shorthand way of
>+// string format used here is not important; it's just a shorthand way of
> // showing the six YMDHMS fields.)
> //
> //   civil_second  2015-11-22 12:34:56
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time_detail.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time_detail.h
>index 2362a4f4fbc958fa30dcfe7a14187688df462adf..d7f72717ece254088740e7375828596c963fb013 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time_detail.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time_detail.h
>@@ -355,11 +355,11 @@ class civil_time {
>       : civil_time(ct.f_) {}
> 
>   // Factories for the maximum/minimum representable civil_time.
>-  static civil_time max() {
>+  static CONSTEXPR_F civil_time max() {
>     const auto max_year = std::numeric_limits<std::int_least64_t>::max();
>     return civil_time(max_year, 12, 31, 23, 59, 59);
>   }
>-  static civil_time min() {
>+  static CONSTEXPR_F civil_time min() {
>     const auto min_year = std::numeric_limits<std::int_least64_t>::min();
>     return civil_time(min_year, 1, 1, 0, 0, 0);
>   }
>@@ -416,6 +416,12 @@ class civil_time {
>     return difference(T{}, lhs.f_, rhs.f_);
>   }
> 
>+  template <typename H>
>+  friend H AbslHashValue(H h, civil_time a) {
>+    return H::combine(std::move(h), a.f_.y, a.f_.m, a.f_.d,
>+                                    a.f_.hh, a.f_.mm, a.f_.ss);
>+  }
>+
>  private:
>   // All instantiations of this template are allowed to call the following
>   // private constructor and access the private fields member.
>@@ -508,12 +514,8 @@ CONSTEXPR_F weekday get_weekday(const civil_day& cd) noexcept {
>   CONSTEXPR_D int k_weekday_offsets[1 + 12] = {
>       -1, 0, 3, 2, 5, 0, 3, 5, 1, 4, 6, 2, 4,
>   };
>-  year_t wd = cd.year() - (cd.month() < 3);
>-  if (wd >= 0) {
>-    wd += wd / 4 - wd / 100 + wd / 400;
>-  } else {
>-    wd += (wd - 3) / 4 - (wd - 99) / 100 + (wd - 399) / 400;
>-  }
>+  year_t wd = 2400 + (cd.year() % 400) - (cd.month() < 3);
>+  wd += wd / 4 - wd / 100 + wd / 400;
>   wd += k_weekday_offsets[cd.month()] + cd.day();
>   return k_weekday_by_sun_off[(wd % 7 + 7) % 7];
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/time_zone.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/time_zone.h
>index 0b9764ea72a9caa3e5f2f7fee0427357aed28808..f28dad17550659b07f0c20038bc47f25a02f62a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/time_zone.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/time_zone.h
>@@ -224,6 +224,11 @@ class time_zone {
>     return !(lhs == rhs);
>   }
> 
>+  template <typename H>
>+  friend H AbslHashValue(H h, time_zone tz) {
>+    return H::combine(std::move(h), &tz.effective_impl());
>+  }
>+
>   class Impl;
> 
>  private:
>@@ -279,7 +284,7 @@ bool parse(const std::string&, const std::string&, const time_zone&,
> }  // namespace detail
> 
> // Formats the given time_point in the given cctz::time_zone according to
>-// the provided format std::string. Uses strftime()-like formatting options,
>+// the provided format string. Uses strftime()-like formatting options,
> // with the following extensions:
> //
> //   - %Ez  - RFC3339-compatible numeric UTC offset (+hh:mm or -hh:mm)
>@@ -298,7 +303,7 @@ bool parse(const std::string&, const std::string&, const time_zone&,
> // more than four characters, just like %Y.
> //
> // Tip: Format strings should include the UTC offset (e.g., %z, %Ez, or %E*z)
>-// so that the resulting std::string uniquely identifies an absolute time.
>+// so that the resulting string uniquely identifies an absolute time.
> //
> // Example:
> //   cctz::time_zone lax;
>@@ -314,7 +319,7 @@ inline std::string format(const std::string& fmt, const time_point<D>& tp,
>   return detail::format(fmt, p.first, n, tz);
> }
> 
>-// Parses an input std::string according to the provided format std::string and
>+// Parses an input string according to the provided format string and
> // returns the corresponding time_point. Uses strftime()-like formatting
> // options, with the same extensions as cctz::format(), but with the
> // exceptions that %E#S is interpreted as %E*S, and %E#f as %E*f. %Ez
>@@ -328,7 +333,7 @@ inline std::string format(const std::string& fmt, const time_point<D>& tp,
> //
> //   "1970-01-01 00:00:00.0 +0000"
> //
>-// For example, parsing a std::string of "15:45" (%H:%M) will return a time_point
>+// For example, parsing a string of "15:45" (%H:%M) will return a time_point
> // that represents "1970-01-01 15:45:00.0 +0000".
> //
> // Note that parse() returns time instants, so it makes most sense to parse
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/cctz_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/cctz_benchmark.cc
>index c97df78c09c83945e68f5511b7b92e4af31770c4..4498d7d0b6b7e0c4a6fdb92149c78796d3d08b6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/cctz_benchmark.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/cctz_benchmark.cc
>@@ -778,13 +778,13 @@ void BM_Zone_UTCTimeZone(benchmark::State& state) {
> }
> BENCHMARK(BM_Zone_UTCTimeZone);
> 
>-// In each "ToDateTime" benchmark we switch between two instants
>-// separated by at least one transition in order to defeat any
>-// internal caching of previous results (e.g., see local_time_hint_).
>+// In each "ToCivil" benchmark we switch between two instants separated
>+// by at least one transition in order to defeat any internal caching of
>+// previous results (e.g., see local_time_hint_).
> //
> // The "UTC" variants use UTC instead of the Google/local time zone.
> 
>-void BM_Time_ToDateTime_CCTZ(benchmark::State& state) {
>+void BM_Time_ToCivil_CCTZ(benchmark::State& state) {
>   const cctz::time_zone tz = TestTimeZone();
>   std::chrono::system_clock::time_point tp =
>       std::chrono::system_clock::from_time_t(1384569027);
>@@ -796,9 +796,9 @@ void BM_Time_ToDateTime_CCTZ(benchmark::State& state) {
>     benchmark::DoNotOptimize(cctz::convert(tp, tz));
>   }
> }
>-BENCHMARK(BM_Time_ToDateTime_CCTZ);
>+BENCHMARK(BM_Time_ToCivil_CCTZ);
> 
>-void BM_Time_ToDateTime_Libc(benchmark::State& state) {
>+void BM_Time_ToCivil_Libc(benchmark::State& state) {
>   // No timezone support, so just use localtime.
>   time_t t = 1384569027;
>   time_t t2 = 1418962578;
>@@ -813,9 +813,9 @@ void BM_Time_ToDateTime_Libc(benchmark::State& state) {
> #endif
>   }
> }
>-BENCHMARK(BM_Time_ToDateTime_Libc);
>+BENCHMARK(BM_Time_ToCivil_Libc);
> 
>-void BM_Time_ToDateTimeUTC_CCTZ(benchmark::State& state) {
>+void BM_Time_ToCivilUTC_CCTZ(benchmark::State& state) {
>   const cctz::time_zone tz = cctz::utc_time_zone();
>   std::chrono::system_clock::time_point tp =
>       std::chrono::system_clock::from_time_t(1384569027);
>@@ -824,9 +824,9 @@ void BM_Time_ToDateTimeUTC_CCTZ(benchmark::State& state) {
>     benchmark::DoNotOptimize(cctz::convert(tp, tz));
>   }
> }
>-BENCHMARK(BM_Time_ToDateTimeUTC_CCTZ);
>+BENCHMARK(BM_Time_ToCivilUTC_CCTZ);
> 
>-void BM_Time_ToDateTimeUTC_Libc(benchmark::State& state) {
>+void BM_Time_ToCivilUTC_Libc(benchmark::State& state) {
>   time_t t = 1384569027;
>   struct tm tm;
>   while (state.KeepRunning()) {
>@@ -838,16 +838,16 @@ void BM_Time_ToDateTimeUTC_Libc(benchmark::State& state) {
> #endif
>   }
> }
>-BENCHMARK(BM_Time_ToDateTimeUTC_Libc);
>+BENCHMARK(BM_Time_ToCivilUTC_Libc);
> 
>-// In each "FromDateTime" benchmark we switch between two YMDhms
>-// values separated by at least one transition in order to defeat any
>-// internal caching of previous results (e.g., see time_local_hint_).
>+// In each "FromCivil" benchmark we switch between two YMDhms values
>+// separated by at least one transition in order to defeat any internal
>+// caching of previous results (e.g., see time_local_hint_).
> //
> // The "UTC" variants use UTC instead of the Google/local time zone.
> // The "Day0" variants require normalization of the day of month.
> 
>-void BM_Time_FromDateTime_CCTZ(benchmark::State& state) {
>+void BM_Time_FromCivil_CCTZ(benchmark::State& state) {
>   const cctz::time_zone tz = TestTimeZone();
>   int i = 0;
>   while (state.KeepRunning()) {
>@@ -860,9 +860,9 @@ void BM_Time_FromDateTime_CCTZ(benchmark::State& state) {
>     }
>   }
> }
>-BENCHMARK(BM_Time_FromDateTime_CCTZ);
>+BENCHMARK(BM_Time_FromCivil_CCTZ);
> 
>-void BM_Time_FromDateTime_Libc(benchmark::State& state) {
>+void BM_Time_FromCivil_Libc(benchmark::State& state) {
>   // No timezone support, so just use localtime.
>   int i = 0;
>   while (state.KeepRunning()) {
>@@ -886,20 +886,20 @@ void BM_Time_FromDateTime_Libc(benchmark::State& state) {
>     benchmark::DoNotOptimize(mktime(&tm));
>   }
> }
>-BENCHMARK(BM_Time_FromDateTime_Libc);
>+BENCHMARK(BM_Time_FromCivil_Libc);
> 
>-void BM_Time_FromDateTimeUTC_CCTZ(benchmark::State& state) {
>+void BM_Time_FromCivilUTC_CCTZ(benchmark::State& state) {
>   const cctz::time_zone tz = cctz::utc_time_zone();
>   while (state.KeepRunning()) {
>     benchmark::DoNotOptimize(
>         cctz::convert(cctz::civil_second(2014, 12, 18, 20, 16, 18), tz));
>   }
> }
>-BENCHMARK(BM_Time_FromDateTimeUTC_CCTZ);
>+BENCHMARK(BM_Time_FromCivilUTC_CCTZ);
> 
>-// There is no BM_Time_FromDateTimeUTC_Libc.
>+// There is no BM_Time_FromCivilUTC_Libc.
> 
>-void BM_Time_FromDateTimeDay0_CCTZ(benchmark::State& state) {
>+void BM_Time_FromCivilDay0_CCTZ(benchmark::State& state) {
>   const cctz::time_zone tz = TestTimeZone();
>   int i = 0;
>   while (state.KeepRunning()) {
>@@ -912,9 +912,9 @@ void BM_Time_FromDateTimeDay0_CCTZ(benchmark::State& state) {
>     }
>   }
> }
>-BENCHMARK(BM_Time_FromDateTimeDay0_CCTZ);
>+BENCHMARK(BM_Time_FromCivilDay0_CCTZ);
> 
>-void BM_Time_FromDateTimeDay0_Libc(benchmark::State& state) {
>+void BM_Time_FromCivilDay0_Libc(benchmark::State& state) {
>   // No timezone support, so just use localtime.
>   int i = 0;
>   while (state.KeepRunning()) {
>@@ -938,7 +938,7 @@ void BM_Time_FromDateTimeDay0_Libc(benchmark::State& state) {
>     benchmark::DoNotOptimize(mktime(&tm));
>   }
> }
>-BENCHMARK(BM_Time_FromDateTimeDay0_Libc);
>+BENCHMARK(BM_Time_FromCivilDay0_Libc);
> 
> const char* const kFormats[] = {
>     RFC1123_full,         // 0
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/civil_time_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/civil_time_test.cc
>index f6648c8f1f21f3207aba1b5e4bb066061b064b65..faffde470af4f45c3b8a4baa0c0d2bcfa318804a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/civil_time_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/civil_time_test.cc
>@@ -620,7 +620,7 @@ TEST(CivilTime, Relational) {
>   TEST_RELATIONAL(civil_second(2014, 1, 1, 1, 1, 0),
>                   civil_second(2014, 1, 1, 1, 1, 1));
> 
>-  // Tests the relational operators of two different CivilTime types.
>+  // Tests the relational operators of two different civil-time types.
>   TEST_RELATIONAL(civil_day(2014, 1, 1), civil_minute(2014, 1, 1, 1, 1));
>   TEST_RELATIONAL(civil_day(2014, 1, 1), civil_month(2014, 2));
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_fixed.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_fixed.cc
>index 598b08fde42236cf137e0e82d09d9f2c1f1b22f0..db9a475a9b4904cb2c4da6b5cf280ca8ca437270 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_fixed.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_fixed.cc
>@@ -15,8 +15,8 @@
> #include "time_zone_fixed.h"
> 
> #include <algorithm>
>+#include <cassert>
> #include <chrono>
>-#include <cstdio>
> #include <cstring>
> #include <string>
> 
>@@ -29,8 +29,15 @@ namespace {
> // The prefix used for the internal names of fixed-offset zones.
> const char kFixedOffsetPrefix[] = "Fixed/UTC";
> 
>+const char kDigits[] = "0123456789";
>+
>+char* Format02d(char* p, int v) {
>+  *p++ = kDigits[(v / 10) % 10];
>+  *p++ = kDigits[v % 10];
>+  return p;
>+}
>+
> int Parse02d(const char* p) {
>-  static const char kDigits[] = "0123456789";
>   if (const char* ap = std::strchr(kDigits, *p)) {
>     int v = static_cast<int>(ap - kDigits);
>     if (const char* bp = std::strchr(kDigits, *++p)) {
>@@ -95,9 +102,17 @@ std::string FixedOffsetToName(const seconds& offset) {
>   }
>   int hours = minutes / 60;
>   minutes %= 60;
>-  char buf[sizeof(kFixedOffsetPrefix) + sizeof("-24:00:00")];
>-  snprintf(buf, sizeof(buf), "%s%c%02d:%02d:%02d",
>-           kFixedOffsetPrefix, sign, hours, minutes, seconds);
>+  char buf[sizeof(kFixedOffsetPrefix) - 1 + sizeof("-24:00:00")];
>+  std::strcpy(buf, kFixedOffsetPrefix);
>+  char* ep = buf + sizeof(kFixedOffsetPrefix) - 1;
>+  *ep++ = sign;
>+  ep = Format02d(ep, hours);
>+  *ep++ = ':';
>+  ep = Format02d(ep, minutes);
>+  *ep++ = ':';
>+  ep = Format02d(ep, seconds);
>+  *ep++ = '\0';
>+  assert(ep == buf + sizeof(buf));
>   return buf;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format.cc
>index 1b023848efa18e81562d57b3955ac120df855474..a02b1e341ac0ccefd60e0fa859014e8d0ad05b09 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format.cc
>@@ -533,7 +533,7 @@ const char* ParseSubSeconds(const char* dp, detail::femtoseconds* subseconds) {
>   return dp;
> }
> 
>-// Parses a std::string into a std::tm using strptime(3).
>+// Parses a string into a std::tm using strptime(3).
> const char* ParseTM(const char* dp, const char* fmt, std::tm* tm) {
>   if (dp != nullptr) {
>     dp = strptime(dp, fmt, tm);
>@@ -743,7 +743,7 @@ bool parse(const std::string& format, const std::string& input,
>     data = ParseTM(data, spec.c_str(), &tm);
> 
>     // If we successfully parsed %p we need to remember whether the result
>-    // was AM or PM so that we can adjust tm_hour before ConvertDateTime().
>+    // was AM or PM so that we can adjust tm_hour before time_zone::lookup().
>     // So reparse the input with a known AM hour, and check if it is shifted
>     // to a PM hour.
>     if (spec == "%p" && data != nullptr) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format_test.cc
>index a90dda7603a8a10e37b3b58d9665968b583962be..6b9928ed32f431027c4b92a49c32e9a08a18d891 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format_test.cc
>@@ -669,13 +669,13 @@ TEST(Parse, WithTimeZone) {
>                     utc_time_zone(), &tp));
>   ExpectTime(tp, tz, 2013, 6, 28, 19 - 8 - 7, 8, 9, -7 * 60 * 60, true, "PDT");
> 
>-  // Check a skipped time (a Spring DST transition).  parse() returns
>-  // the preferred-offset result, as defined for ConvertDateTime().
>+  // Check a skipped time (a Spring DST transition). parse() uses the
>+  // pre-transition offset.
>   EXPECT_TRUE(parse("%Y-%m-%d %H:%M:%S", "2011-03-13 02:15:00", tz, &tp));
>   ExpectTime(tp, tz, 2011, 3, 13, 3, 15, 0, -7 * 60 * 60, true, "PDT");
> 
>-  // Check a repeated time (a Fall DST transition).  parse() returns
>-  // the preferred-offset result, as defined for ConvertDateTime().
>+  // Check a repeated time (a Fall DST transition).  parse() uses the
>+  // pre-transition offset.
>   EXPECT_TRUE(parse("%Y-%m-%d %H:%M:%S", "2011-11-06 01:15:00", tz, &tp));
>   ExpectTime(tp, tz, 2011, 11, 6, 1, 15, 0, -7 * 60 * 60, true, "PDT");
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_info.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_info.cc
>index bf73635d4c6a8c033b6e487925c41a2eab805184..2cb358d048e2d82191de8bcf202829c7ea2cb883 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_info.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_info.cc
>@@ -286,7 +286,7 @@ bool TimeZoneInfo::EquivTransitions(std::uint_fast8_t tt1_index,
>   return true;
> }
> 
>-// Use the POSIX-TZ-environment-variable-style std::string to handle times
>+// Use the POSIX-TZ-environment-variable-style string to handle times
> // in years after the last transition stored in the zoneinfo data.
> void TimeZoneInfo::ExtendTransitions(const std::string& name,
>                                      const Header& hdr) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_posix.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_posix.h
>index 6619f27edcf63895abeb23f3ddb48eff4f9f2ac8..9ccd4a8b68bd5761f56e314b9d8a250171b48d69 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_posix.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_posix.h
>@@ -89,7 +89,7 @@ struct PosixTransition {
>   } time;
> };
> 
>-// The entirety of a POSIX-std::string specified time-zone rule. The standard
>+// The entirety of a POSIX-string specified time-zone rule. The standard
> // abbreviation and offset are always given. If the time zone includes
> // daylight saving, then the daylight abbrevation is non-empty and the
> // remaining fields are also valid. Note that the start/end transitions
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time.h
>index c41cb89c5eff1d4daf8ecb16c5b15639702f6462..50bf971df0a29cb4750aef70c11e705447646a22 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time.h
>@@ -25,11 +25,12 @@
> //  * `absl::TimeZone` defines geopolitical time zone regions (as collected
> //     within the IANA Time Zone database (https://www.iana.org/time-zones)).
> //
>+//
> // Example:
> //
> //   absl::TimeZone nyc;
> //
>-//   // LoadTimeZone may fail so it's always better to check for success.
>+//   // LoadTimeZone() may fail so it's always better to check for success.
> //   if (!absl::LoadTimeZone("America/New_York", &nyc)) {
> //      // handle error case
> //   }
>@@ -162,6 +163,11 @@ class Duration {
>   Duration& operator*=(float r) { return *this *= static_cast<double>(r); }
>   Duration& operator/=(float r) { return *this /= static_cast<double>(r); }
> 
>+  template <typename H>
>+  friend H AbslHashValue(H h, Duration d) {
>+    return H::combine(std::move(h), d.rep_hi_, d.rep_lo_);
>+  }
>+
>  private:
>   friend constexpr int64_t time_internal::GetRepHi(Duration d);
>   friend constexpr uint32_t time_internal::GetRepLo(Duration d);
>@@ -321,6 +327,9 @@ Duration Ceil(Duration d, Duration unit);
> //   0 == d / inf
> //   INT64_MAX == inf / d
> //
>+//   d < inf
>+//   -inf < d
>+//
> //   // Division by zero returns infinity, or INT64_MIN/MAX where appropriate.
> //   inf == d / 0
> //   INT64_MAX == d / absl::ZeroDuration()
>@@ -481,7 +490,7 @@ std::chrono::hours ToChronoHours(Duration d);
> 
> // FormatDuration()
> //
>-// Returns a std::string representing the duration in the form "72h3m0.5s".
>+// Returns a string representing the duration in the form "72h3m0.5s".
> // Returns "inf" or "-inf" for +/- `InfiniteDuration()`.
> std::string FormatDuration(Duration d);
> 
>@@ -492,11 +501,11 @@ inline std::ostream& operator<<(std::ostream& os, Duration d) {
> 
> // ParseDuration()
> //
>-// Parses a duration std::string consisting of a possibly signed sequence of
>+// Parses a duration string consisting of a possibly signed sequence of
> // decimal numbers, each with an optional fractional part and a unit
> // suffix.  The valid suffixes are "ns", "us" "ms", "s", "m", and "h".
> // Simple examples include "300ms", "-1.5h", and "2h45m".  Parses "0" as
>-// `ZeroDuration()`.  Parses "inf" and "-inf" as +/- `InfiniteDuration()`.
>+// `ZeroDuration()`. Parses "inf" and "-inf" as +/- `InfiniteDuration()`.
> bool ParseDuration(const std::string& dur_string, Duration* d);
> 
> // Support for flag values of type Duration. Duration flags must be specified
>@@ -607,6 +616,11 @@ class Time {
>   // Returns the breakdown of this instant in the given TimeZone.
>   Breakdown In(TimeZone tz) const;
> 
>+  template <typename H>
>+  friend H AbslHashValue(H h, Time t) {
>+    return H::combine(std::move(h), t.rep_);
>+  }
>+
>  private:
>   friend constexpr Time time_internal::FromUnixDuration(Duration d);
>   friend constexpr Duration time_internal::ToUnixDuration(Time t);
>@@ -689,7 +703,9 @@ constexpr Time InfinitePast() {
> // Examples:
> //
> //   absl::TimeZone lax;
>-//   if (!absl::LoadTimeZone("America/Los_Angeles", &lax)) { ... }
>+//   if (!absl::LoadTimeZone("America/Los_Angeles", &lax)) {
>+//     // handle error case
>+//   }
> //
> //   // A unique civil time
> //   absl::TimeConversion jan01 =
>@@ -759,7 +775,9 @@ TimeConversion ConvertDateTime(int64_t year, int mon, int day, int hour,
> // Example:
> //
> //   absl::TimeZone seattle;
>-//   if (!absl::LoadTimeZone("America/Los_Angeles", &seattle)) { ... }
>+//   if (!absl::LoadTimeZone("America/Los_Angeles", &seattle)) {
>+//     // handle error case
>+//   }
> //   absl::Time t =  absl::FromDateTime(2017, 9, 26, 9, 30, 0, seattle);
> Time FromDateTime(int64_t year, int mon, int day, int hour, int min, int sec,
>                   TimeZone tz);
>@@ -871,8 +889,10 @@ std::chrono::system_clock::time_point ToChronoTime(Time);
> // FormatTime()/ParseTime() format specifiers for RFC3339 date/time strings,
> // with trailing zeros trimmed or with fractional seconds omitted altogether.
> //
>-// Note that RFC3339_sec[] matches an ISO 8601 extended format for date
>-// and time with UTC offset.
>+// Note that RFC3339_sec[] matches an ISO 8601 extended format for date and
>+// time with UTC offset.  Also note the use of "%Y": RFC3339 mandates that
>+// years have exactly four digits, but we allow them to take their natural
>+// width.
> extern const char RFC3339_full[];  // %Y-%m-%dT%H:%M:%E*S%Ez
> extern const char RFC3339_sec[];   // %Y-%m-%dT%H:%M:%S%Ez
> 
>@@ -886,7 +906,7 @@ extern const char RFC1123_no_wday[];  // %d %b %E4Y %H:%M:%S %z
> // FormatTime()
> //
> // Formats the given `absl::Time` in the `absl::TimeZone` according to the
>-// provided format std::string. Uses strftime()-like formatting options, with
>+// provided format string. Uses strftime()-like formatting options, with
> // the following extensions:
> //
> //   - %Ez  - RFC3339-compatible numeric UTC offset (+hh:mm or -hh:mm)
>@@ -910,16 +930,18 @@ extern const char RFC1123_no_wday[];  // %d %b %E4Y %H:%M:%S %z
> // Example:
> //
> //   absl::TimeZone lax;
>-//   if (!absl::LoadTimeZone("America/Los_Angeles", &lax)) { ... }
>+//   if (!absl::LoadTimeZone("America/Los_Angeles", &lax)) {
>+//     // handle error case
>+//   }
> //   absl::Time t = absl::FromDateTime(2013, 1, 2, 3, 4, 5, lax);
> //
>-//   std::string f = absl::FormatTime("%H:%M:%S", t, lax);  // "03:04:05"
>+//   string f = absl::FormatTime("%H:%M:%S", t, lax);  // "03:04:05"
> //   f = absl::FormatTime("%H:%M:%E3S", t, lax);  // "03:04:05.000"
> //
> // Note: If the given `absl::Time` is `absl::InfiniteFuture()`, the returned
>-// std::string will be exactly "infinite-future". If the given `absl::Time` is
>-// `absl::InfinitePast()`, the returned std::string will be exactly "infinite-past".
>-// In both cases the given format std::string and `absl::TimeZone` are ignored.
>+// string will be exactly "infinite-future". If the given `absl::Time` is
>+// `absl::InfinitePast()`, the returned string will be exactly "infinite-past".
>+// In both cases the given format string and `absl::TimeZone` are ignored.
> //
> std::string FormatTime(const std::string& format, Time t, TimeZone tz);
> 
>@@ -936,7 +958,7 @@ inline std::ostream& operator<<(std::ostream& os, Time t) {
> 
> // ParseTime()
> //
>-// Parses an input std::string according to the provided format std::string and
>+// Parses an input string according to the provided format string and
> // returns the corresponding `absl::Time`. Uses strftime()-like formatting
> // options, with the same extensions as FormatTime(), but with the
> // exceptions that %E#S is interpreted as %E*S, and %E#f as %E*f.  %Ez
>@@ -950,7 +972,7 @@ inline std::ostream& operator<<(std::ostream& os, Time t) {
> //
> //   "1970-01-01 00:00:00.0 +0000"
> //
>-// For example, parsing a std::string of "15:45" (%H:%M) will return an absl::Time
>+// For example, parsing a string of "15:45" (%H:%M) will return an absl::Time
> // that represents "1970-01-01 15:45:00.0 +0000".
> //
> // Note that since ParseTime() returns time instants, it makes the most sense
>@@ -977,15 +999,15 @@ inline std::ostream& operator<<(std::ostream& os, Time t) {
> // Errors are indicated by returning false and assigning an error message
> // to the "err" out param if it is non-null.
> //
>-// Note: If the input std::string is exactly "infinite-future", the returned
>+// Note: If the input string is exactly "infinite-future", the returned
> // `absl::Time` will be `absl::InfiniteFuture()` and `true` will be returned.
>-// If the input std::string is "infinite-past", the returned `absl::Time` will be
>+// If the input string is "infinite-past", the returned `absl::Time` will be
> // `absl::InfinitePast()` and `true` will be returned.
> //
> bool ParseTime(const std::string& format, const std::string& input, Time* time,
>                std::string* err);
> 
>-// Like ParseTime() above, but if the format std::string does not contain a UTC
>+// Like ParseTime() above, but if the format string does not contain a UTC
> // offset specification (%z/%Ez/%E*z) then the input is interpreted in the
> // given TimeZone.  This means that the input, by itself, does not identify a
> // unique instant.  Being time-zone dependent, it also admits the possibility
>@@ -1028,7 +1050,9 @@ std::string UnparseFlag(Time t);
> //   absl::TimeZone pst = absl::FixedTimeZone(-8 * 60 * 60);
> //   absl::TimeZone loc = absl::LocalTimeZone();
> //   absl::TimeZone lax;
>-//   if (!absl::LoadTimeZone("America/Los_Angeles", &lax)) { ... }
>+//   if (!absl::LoadTimeZone("America/Los_Angeles", &lax)) {
>+//     // handle error case
>+//   }
> //
> // See also:
> // - https://github.com/google/cctz
>@@ -1045,6 +1069,11 @@ class TimeZone {
> 
>   std::string name() const { return cz_.name(); }
> 
>+  template <typename H>
>+  friend H AbslHashValue(H h, TimeZone tz) {
>+    return H::combine(std::move(h), tz.cz_);
>+  }
>+
>  private:
>   friend bool operator==(TimeZone a, TimeZone b) { return a.cz_ == b.cz_; }
>   friend bool operator!=(TimeZone a, TimeZone b) { return a.cz_ != b.cz_; }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.bazel
>index 096c119e638cbd35040550bed10b192e7b467401..32f690c94c1d347c1437283b8088d2b195f51cbb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.bazel
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.bazel
>@@ -19,6 +19,7 @@ load(
>     "ABSL_DEFAULT_COPTS",
>     "ABSL_TEST_COPTS",
>     "ABSL_EXCEPTIONS_FLAG",
>+    "ABSL_EXCEPTIONS_FLAG_LINKOPTS",
> )
> 
> package(default_visibility = ["//visibility:public"])
>@@ -55,6 +56,7 @@ cc_library(
>         "bad_any_cast.h",
>     ],
>     copts = ABSL_EXCEPTIONS_FLAG + ABSL_DEFAULT_COPTS,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     visibility = ["//visibility:private"],
>     deps = [
>         "//absl/base",
>@@ -69,6 +71,7 @@ cc_test(
>         "any_test.cc",
>     ],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":any",
>         "//absl/base",
>@@ -100,6 +103,7 @@ cc_test(
>     name = "any_exception_safety_test",
>     srcs = ["any_exception_safety_test.cc"],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":any",
>         "//absl/base:exception_safety_testing",
>@@ -124,6 +128,7 @@ cc_test(
>     size = "small",
>     srcs = ["span_test.cc"],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":span",
>         "//absl/base:config",
>@@ -131,6 +136,7 @@ cc_test(
>         "//absl/base:exception_testing",
>         "//absl/container:fixed_array",
>         "//absl/container:inlined_vector",
>+        "//absl/hash:hash_testing",
>         "//absl/strings",
>         "@com_google_googletest//:gtest_main",
>     ],
>@@ -148,6 +154,7 @@ cc_test(
>         "//absl/base:exception_testing",
>         "//absl/container:fixed_array",
>         "//absl/container:inlined_vector",
>+        "//absl/hash:hash_testing",
>         "//absl/strings",
>         "@com_google_googletest//:gtest_main",
>     ],
>@@ -172,6 +179,7 @@ cc_library(
>     srcs = ["bad_optional_access.cc"],
>     hdrs = ["bad_optional_access.h"],
>     copts = ABSL_DEFAULT_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         "//absl/base",
>         "//absl/base:config",
>@@ -183,6 +191,7 @@ cc_library(
>     srcs = ["bad_variant_access.cc"],
>     hdrs = ["bad_variant_access.h"],
>     copts = ABSL_EXCEPTIONS_FLAG + ABSL_DEFAULT_COPTS,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         "//absl/base",
>         "//absl/base:config",
>@@ -196,6 +205,7 @@ cc_test(
>         "optional_test.cc",
>     ],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":optional",
>         "//absl/base",
>@@ -212,6 +222,7 @@ cc_test(
>         "optional_exception_safety_test.cc",
>     ],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":optional",
>         "//absl/base:exception_safety_testing",
>@@ -239,6 +250,7 @@ cc_test(
>     size = "small",
>     srcs = ["variant_test.cc"],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":variant",
>         "//absl/base:config",
>@@ -271,6 +283,7 @@ cc_test(
>         "variant_exception_safety_test.cc",
>     ],
>     copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    linkopts = ABSL_EXCEPTIONS_FLAG_LINKOPTS,
>     deps = [
>         ":variant",
>         "//absl/base:exception_safety_testing",
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.gn
>index fd7e89dee34c3b32905a7fa78fbfbfe2abb01e55..bfdafb317e3b4bf1d3fec252cd0cd1d193ec1929 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.gn
>@@ -104,6 +104,7 @@ source_set("optional") {
>   deps = [
>     ":bad_optional_access",
>     "../base:config",
>+    "../base:core_headers",
>     "../memory",
>     "../meta:type_traits",
>     "../utility",
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any_exception_safety_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any_exception_safety_test.cc
>index 36955f6c5c376d32fb11a55895ba4ba06fe5a78e..f9dd8c482c25543b9142e42ba175a8c0149c3566 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any_exception_safety_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any_exception_safety_test.cc
>@@ -62,7 +62,7 @@ testing::AssertionResult AnyInvariants(absl::any* a) {
>     static_cast<void>(unused);
>     return AssertionFailure()
>            << "A reset `any` should not be able to be any_cast";
>-  } catch (absl::bad_any_cast) {
>+  } catch (const absl::bad_any_cast&) {
>   } catch (...) {
>     return AssertionFailure()
>            << "Unexpected exception thrown from absl::any_cast";
>@@ -107,7 +107,7 @@ TEST(AnyExceptionSafety, Assignment) {
>   };
>   auto any_strong_tester = testing::MakeExceptionSafetyTester()
>                                .WithInitialValue(original)
>-                               .WithInvariants(AnyInvariants, any_is_strong);
>+                               .WithContracts(AnyInvariants, any_is_strong);
> 
>   Thrower val(2);
>   absl::any any_val(val);
>@@ -129,7 +129,7 @@ TEST(AnyExceptionSafety, Assignment) {
>   auto strong_empty_any_tester =
>       testing::MakeExceptionSafetyTester()
>           .WithInitialValue(absl::any{})
>-          .WithInvariants(AnyInvariants, empty_any_is_strong);
>+          .WithContracts(AnyInvariants, empty_any_is_strong);
> 
>   EXPECT_TRUE(strong_empty_any_tester.Test(assign_any));
>   EXPECT_TRUE(strong_empty_any_tester.Test(assign_val));
>@@ -142,7 +142,7 @@ TEST(AnyExceptionSafety, Emplace) {
>       absl::any{absl::in_place_type_t<Thrower>(), 1, testing::nothrow_ctor};
>   auto one_tester = testing::MakeExceptionSafetyTester()
>                         .WithInitialValue(initial_val)
>-                        .WithInvariants(AnyInvariants, AnyIsEmpty);
>+                        .WithContracts(AnyInvariants, AnyIsEmpty);
> 
>   auto emp_thrower = [](absl::any* ap) { ap->emplace<Thrower>(2); };
>   auto emp_throwervec = [](absl::any* ap) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/internal/variant.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/internal/variant.h
>index 7708e67cb14f416ade16719e2369aaf648bb6283..eff4fefed409d92d507dba8669fe1231ed4cb15e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/internal/variant.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/internal/variant.h
>@@ -1,4 +1,4 @@
>-// Copyright 2017 The Abseil Authors.
>+// Copyright 2018 The Abseil Authors.
> //
> // Licensed under the Apache License, Version 2.0 (the "License");
> // you may not use this file except in compliance with the License.
>@@ -37,6 +37,8 @@
> #include "absl/types/bad_variant_access.h"
> #include "absl/utility/utility.h"
> 
>+#if !defined(ABSL_HAVE_STD_VARIANT)
>+
> namespace absl {
> 
> template <class... Types>
>@@ -908,6 +910,11 @@ struct PerformVisitation {
>   template <std::size_t... TupIs, std::size_t... Is>
>   constexpr ReturnType Run(std::false_type /*has_valueless*/,
>                            index_sequence<TupIs...>, SizeT<Is>...) const {
>+    static_assert(
>+        std::is_same<ReturnType,
>+                     absl::result_of_t<Op(VariantAccessResult<
>+                                          Is, QualifiedVariants>...)>>::value,
>+        "All visitation overloads must have the same return type.");
>     return absl::base_internal::Invoke(
>         absl::forward<Op>(op),
>         VariantCoreAccess::Access<Is>(
>@@ -1227,23 +1234,23 @@ using VariantCopyBase = absl::conditional_t<
> // Base that is dependent on whether or not the move-assign can be trivial.
> template <class... T>
> using VariantMoveAssignBase = absl::conditional_t<
>-    absl::disjunction<absl::conjunction<std::is_move_assignable<Union<T...>>,
>+    absl::disjunction<absl::conjunction<absl::is_move_assignable<Union<T...>>,
>                                         std::is_move_constructible<Union<T...>>,
>                                         std::is_destructible<Union<T...>>>,
>                       absl::negation<absl::conjunction<
>                           std::is_move_constructible<T>...,
>-                          std::is_move_assignable<T>...>>>::value,
>+                          absl::is_move_assignable<T>...>>>::value,
>     VariantCopyBase<T...>, VariantMoveAssignBaseNontrivial<T...>>;
> 
> // Base that is dependent on whether or not the copy-assign can be trivial.
> template <class... T>
> using VariantCopyAssignBase = absl::conditional_t<
>-    absl::disjunction<absl::conjunction<std::is_copy_assignable<Union<T...>>,
>+    absl::disjunction<absl::conjunction<absl::is_copy_assignable<Union<T...>>,
>                                         std::is_copy_constructible<Union<T...>>,
>                                         std::is_destructible<Union<T...>>>,
>                       absl::negation<absl::conjunction<
>                           std::is_copy_constructible<T>...,
>-                          std::is_copy_assignable<T>...>>>::value,
>+                          absl::is_copy_assignable<T>...>>>::value,
>     VariantMoveAssignBase<T...>, VariantCopyAssignBaseNontrivial<T...>>;
> 
> template <class... T>
>@@ -1612,4 +1619,5 @@ struct VariantHashBase<Variant,
> }  // namespace variant_internal
> }  // namespace absl
> 
>+#endif  // !defined(ABSL_HAVE_STD_VARIANT)
> #endif  // ABSL_TYPES_variant_internal_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional.h
>index c837cddeef48942647e7f8292d0163750946b2b6..14210018815c4c661207fe40f452b0135c9a0af3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional.h
>@@ -59,6 +59,7 @@ using std::nullopt;
> #include <type_traits>
> #include <utility>
> 
>+#include "absl/base/attributes.h"
> #include "absl/memory/memory.h"
> #include "absl/meta/type_traits.h"
> #include "absl/types/bad_optional_access.h"
>@@ -411,10 +412,10 @@ constexpr copy_traits get_ctor_copy_traits() {
> 
> template <typename T>
> constexpr copy_traits get_assign_copy_traits() {
>-  return std::is_copy_assignable<T>::value &&
>+  return absl::is_copy_assignable<T>::value &&
>                  std::is_copy_constructible<T>::value
>              ? copy_traits::copyable
>-             : std::is_move_assignable<T>::value &&
>+             : absl::is_move_assignable<T>::value &&
>                        std::is_move_constructible<T>::value
>                    ? copy_traits::movable
>                    : copy_traits::non_movable;
>@@ -700,7 +701,7 @@ class optional : private optional_internal::optional_data<T>,
>   // optional::reset()
>   //
>   // Destroys the inner `T` value of an `absl::optional` if one is present.
>-  void reset() noexcept { this->destruct(); }
>+  ABSL_ATTRIBUTE_REINITIALIZES void reset() noexcept { this->destruct(); }
> 
>   // optional::emplace()
>   //
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_exception_safety_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_exception_safety_test.cc
>index d2ef04b8d1b083c755c2306855460aa8f0de81dd..d117ee5184919fe6b12d1233de16e9aee80e803e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_exception_safety_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_exception_safety_test.cc
>@@ -38,12 +38,12 @@ constexpr int kUpdatedInteger = 10;
> template <typename OptionalT>
> bool ValueThrowsBadOptionalAccess(const OptionalT& optional) try {
>   return (static_cast<void>(optional.value()), false);
>-} catch (absl::bad_optional_access) {
>+} catch (const absl::bad_optional_access&) {
>   return true;
> }
> 
> template <typename OptionalT>
>-AssertionResult CheckInvariants(OptionalT* optional_ptr) {
>+AssertionResult OptionalInvariants(OptionalT* optional_ptr) {
>   // Check the current state post-throw for validity
>   auto& optional = *optional_ptr;
> 
>@@ -123,8 +123,8 @@ TEST(OptionalExceptionSafety, NothrowConstructors) {
> TEST(OptionalExceptionSafety, Emplace) {
>   // Test the basic guarantee plus test the result of optional::has_value()
>   // is false in all cases
>-  auto disengaged_test = MakeExceptionSafetyTester().WithInvariants(
>-      CheckInvariants<Optional>, CheckDisengaged<Optional>);
>+  auto disengaged_test = MakeExceptionSafetyTester().WithContracts(
>+      OptionalInvariants<Optional>, CheckDisengaged<Optional>);
>   auto disengaged_test_empty = disengaged_test.WithInitialValue(Optional());
>   auto disengaged_test_nonempty =
>       disengaged_test.WithInitialValue(Optional(kInitialInteger));
>@@ -147,11 +147,11 @@ TEST(OptionalExceptionSafety, EverythingThrowsSwap) {
>   // Test the basic guarantee plus test the result of optional::has_value()
>   // remains the same
>   auto test =
>-      MakeExceptionSafetyTester().WithInvariants(CheckInvariants<Optional>);
>+      MakeExceptionSafetyTester().WithContracts(OptionalInvariants<Optional>);
>   auto disengaged_test_empty = test.WithInitialValue(Optional())
>-                                   .WithInvariants(CheckDisengaged<Optional>);
>+                                   .WithContracts(CheckDisengaged<Optional>);
>   auto engaged_test_nonempty = test.WithInitialValue(Optional(kInitialInteger))
>-                                   .WithInvariants(CheckEngaged<Optional>);
>+                                   .WithContracts(CheckEngaged<Optional>);
> 
>   auto swap_empty = [](Optional* optional_ptr) {
>     auto empty = Optional();
>@@ -192,11 +192,11 @@ TEST(OptionalExceptionSafety, CopyAssign) {
>   // Test the basic guarantee plus test the result of optional::has_value()
>   // remains the same
>   auto test =
>-      MakeExceptionSafetyTester().WithInvariants(CheckInvariants<Optional>);
>+      MakeExceptionSafetyTester().WithContracts(OptionalInvariants<Optional>);
>   auto disengaged_test_empty = test.WithInitialValue(Optional())
>-                                   .WithInvariants(CheckDisengaged<Optional>);
>+                                   .WithContracts(CheckDisengaged<Optional>);
>   auto engaged_test_nonempty = test.WithInitialValue(Optional(kInitialInteger))
>-                                   .WithInvariants(CheckEngaged<Optional>);
>+                                   .WithContracts(CheckEngaged<Optional>);
> 
>   auto copyassign_nonempty = [](Optional* optional_ptr) {
>     auto nonempty =
>@@ -218,11 +218,11 @@ TEST(OptionalExceptionSafety, MoveAssign) {
>   // Test the basic guarantee plus test the result of optional::has_value()
>   // remains the same
>   auto test =
>-      MakeExceptionSafetyTester().WithInvariants(CheckInvariants<Optional>);
>+      MakeExceptionSafetyTester().WithContracts(OptionalInvariants<Optional>);
>   auto disengaged_test_empty = test.WithInitialValue(Optional())
>-                                   .WithInvariants(CheckDisengaged<Optional>);
>+                                   .WithContracts(CheckDisengaged<Optional>);
>   auto engaged_test_nonempty = test.WithInitialValue(Optional(kInitialInteger))
>-                                   .WithInvariants(CheckEngaged<Optional>);
>+                                   .WithContracts(CheckEngaged<Optional>);
> 
>   auto moveassign_empty = [](Optional* optional_ptr) {
>     auto empty = Optional();
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_test.cc
>index 179bfd66d2fee0eb9321b0559b2b9dc18910dbb3..d90db9f821b52d4a935557792e212e14199e7809 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_test.cc
>@@ -607,11 +607,12 @@ TEST(optionalTest, CopyAssignment) {
>   opt2_to_empty = empty;
>   EXPECT_FALSE(opt2_to_empty);
> 
>-  EXPECT_FALSE(std::is_copy_assignable<absl::optional<const int>>::value);
>-  EXPECT_TRUE(std::is_copy_assignable<absl::optional<Copyable>>::value);
>-  EXPECT_FALSE(std::is_copy_assignable<absl::optional<MoveableThrow>>::value);
>-  EXPECT_FALSE(std::is_copy_assignable<absl::optional<MoveableNoThrow>>::value);
>-  EXPECT_FALSE(std::is_copy_assignable<absl::optional<NonMovable>>::value);
>+  EXPECT_FALSE(absl::is_copy_assignable<absl::optional<const int>>::value);
>+  EXPECT_TRUE(absl::is_copy_assignable<absl::optional<Copyable>>::value);
>+  EXPECT_FALSE(absl::is_copy_assignable<absl::optional<MoveableThrow>>::value);
>+  EXPECT_FALSE(
>+      absl::is_copy_assignable<absl::optional<MoveableNoThrow>>::value);
>+  EXPECT_FALSE(absl::is_copy_assignable<absl::optional<NonMovable>>::value);
> 
>   EXPECT_TRUE(absl::is_trivially_copy_assignable<int>::value);
>   EXPECT_TRUE(absl::is_trivially_copy_assignable<volatile int>::value);
>@@ -625,9 +626,9 @@ TEST(optionalTest, CopyAssignment) {
>   };
> 
>   EXPECT_TRUE(absl::is_trivially_copy_assignable<Trivial>::value);
>-  EXPECT_FALSE(std::is_copy_assignable<const Trivial>::value);
>-  EXPECT_FALSE(std::is_copy_assignable<volatile Trivial>::value);
>-  EXPECT_TRUE(std::is_copy_assignable<NonTrivial>::value);
>+  EXPECT_FALSE(absl::is_copy_assignable<const Trivial>::value);
>+  EXPECT_FALSE(absl::is_copy_assignable<volatile Trivial>::value);
>+  EXPECT_TRUE(absl::is_copy_assignable<NonTrivial>::value);
>   EXPECT_FALSE(absl::is_trivially_copy_assignable<NonTrivial>::value);
> 
>   // std::optional doesn't support volatile nontrivial types.
>@@ -695,11 +696,11 @@ TEST(optionalTest, MoveAssignment) {
>     EXPECT_EQ(1, listener.volatile_move_assign);
>   }
> #endif  // ABSL_HAVE_STD_OPTIONAL
>-  EXPECT_FALSE(std::is_move_assignable<absl::optional<const int>>::value);
>-  EXPECT_TRUE(std::is_move_assignable<absl::optional<Copyable>>::value);
>-  EXPECT_TRUE(std::is_move_assignable<absl::optional<MoveableThrow>>::value);
>-  EXPECT_TRUE(std::is_move_assignable<absl::optional<MoveableNoThrow>>::value);
>-  EXPECT_FALSE(std::is_move_assignable<absl::optional<NonMovable>>::value);
>+  EXPECT_FALSE(absl::is_move_assignable<absl::optional<const int>>::value);
>+  EXPECT_TRUE(absl::is_move_assignable<absl::optional<Copyable>>::value);
>+  EXPECT_TRUE(absl::is_move_assignable<absl::optional<MoveableThrow>>::value);
>+  EXPECT_TRUE(absl::is_move_assignable<absl::optional<MoveableNoThrow>>::value);
>+  EXPECT_FALSE(absl::is_move_assignable<absl::optional<NonMovable>>::value);
> 
>   EXPECT_FALSE(
>       std::is_nothrow_move_assignable<absl::optional<MoveableThrow>>::value);
>@@ -1619,7 +1620,7 @@ TEST(optionalTest, AssignmentConstraints) {
>   EXPECT_TRUE(
>       (std::is_assignable<absl::optional<AnyLike>&, const AnyLike&>::value));
>   EXPECT_TRUE(std::is_move_assignable<absl::optional<AnyLike>>::value);
>-  EXPECT_TRUE(std::is_copy_assignable<absl::optional<AnyLike>>::value);
>+  EXPECT_TRUE(absl::is_copy_assignable<absl::optional<AnyLike>>::value);
> }
> 
> }  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span.h
>index 76be819ecca2186e5dd49dba0b2fd4a6d27ff9e8..911af0c57a61a6785557a91fbb692d7d90191458 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span.h
>@@ -87,7 +87,7 @@ constexpr auto GetDataImpl(C& c, char) noexcept  // NOLINT(runtime/references)
>   return c.data();
> }
> 
>-// Before C++17, std::string::data returns a const char* in all cases.
>+// Before C++17, string::data returns a const char* in all cases.
> inline char* GetDataImpl(std::string& s,  // NOLINT(runtime/references)
>                          int) noexcept {
>   return &s[0];
>@@ -379,64 +379,70 @@ class Span {
>   //
>   // Returns a reference to the i'th element of this span.
>   constexpr reference at(size_type i) const {
>-    return ABSL_PREDICT_TRUE(i < size())
>-               ? ptr_[i]
>+    return ABSL_PREDICT_TRUE(i < size())  //
>+               ? *(data() + i)
>                : (base_internal::ThrowStdOutOfRange(
>                       "Span::at failed bounds check"),
>-                  ptr_[i]);
>+                  *(data() + i));
>   }
> 
>   // Span::front()
>   //
>   // Returns a reference to the first element of this span.
>-  reference front() const noexcept { return ABSL_ASSERT(size() > 0), ptr_[0]; }
>+  constexpr reference front() const noexcept {
>+    return ABSL_ASSERT(size() > 0), *data();
>+  }
> 
>   // Span::back()
>   //
>   // Returns a reference to the last element of this span.
>-  reference back() const noexcept {
>-    return ABSL_ASSERT(size() > 0), ptr_[size() - 1];
>+  constexpr reference back() const noexcept {
>+    return ABSL_ASSERT(size() > 0), *(data() + size() - 1);
>   }
> 
>   // Span::begin()
>   //
>   // Returns an iterator to the first element of this span.
>-  constexpr iterator begin() const noexcept { return ptr_; }
>+  constexpr iterator begin() const noexcept { return data(); }
> 
>   // Span::cbegin()
>   //
>   // Returns a const iterator to the first element of this span.
>-  constexpr const_iterator cbegin() const noexcept { return ptr_; }
>+  constexpr const_iterator cbegin() const noexcept { return begin(); }
> 
>   // Span::end()
>   //
>   // Returns an iterator to the last element of this span.
>-  iterator end() const noexcept { return ptr_ + len_; }
>+  constexpr iterator end() const noexcept { return data() + size(); }
> 
>   // Span::cend()
>   //
>   // Returns a const iterator to the last element of this span.
>-  const_iterator cend() const noexcept { return end(); }
>+  constexpr const_iterator cend() const noexcept { return end(); }
> 
>   // Span::rbegin()
>   //
>   // Returns a reverse iterator starting at the last element of this span.
>-  reverse_iterator rbegin() const noexcept { return reverse_iterator(end()); }
>+  constexpr reverse_iterator rbegin() const noexcept {
>+    return reverse_iterator(end());
>+  }
> 
>   // Span::crbegin()
>   //
>   // Returns a reverse const iterator starting at the last element of this span.
>-  const_reverse_iterator crbegin() const noexcept { return rbegin(); }
>+  constexpr const_reverse_iterator crbegin() const noexcept { return rbegin(); }
> 
>   // Span::rend()
>   //
>   // Returns a reverse iterator starting at the first element of this span.
>-  reverse_iterator rend() const noexcept { return reverse_iterator(begin()); }
>+  constexpr reverse_iterator rend() const noexcept {
>+    return reverse_iterator(begin());
>+  }
> 
>   // Span::crend()
>   //
>   // Returns a reverse iterator starting at the first element of this span.
>-  const_reverse_iterator crend() const noexcept { return rend(); }
>+  constexpr const_reverse_iterator crend() const noexcept { return rend(); }
> 
>   // Span mutations
> 
>@@ -444,7 +450,7 @@ class Span {
>   //
>   // Removes the first `n` elements from the span.
>   void remove_prefix(size_type n) noexcept {
>-    assert(len_ >= n);
>+    assert(size() >= n);
>     ptr_ += n;
>     len_ -= n;
>   }
>@@ -453,7 +459,7 @@ class Span {
>   //
>   // Removes the last `n` elements from the span.
>   void remove_suffix(size_type n) noexcept {
>-    assert(len_ >= n);
>+    assert(size() >= n);
>     len_ -= n;
>   }
> 
>@@ -474,11 +480,18 @@ class Span {
>   //   absl::MakeSpan(vec).subspan(4);     // {}
>   //   absl::MakeSpan(vec).subspan(5);     // throws std::out_of_range
>   constexpr Span subspan(size_type pos = 0, size_type len = npos) const {
>-    return (pos <= len_)
>-               ? Span(ptr_ + pos, span_internal::Min(len_ - pos, len))
>+    return (pos <= size())
>+               ? Span(data() + pos, span_internal::Min(size() - pos, len))
>                : (base_internal::ThrowStdOutOfRange("pos > size()"), Span());
>   }
> 
>+  // Support for absl::Hash.
>+  template <typename H>
>+  friend H AbslHashValue(H h, Span v) {
>+    return H::combine(H::combine_contiguous(std::move(h), v.data(), v.size()),
>+                      v.size());
>+  }
>+
>  private:
>   pointer ptr_;
>   size_type len_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span_test.cc
>index fbce7e87479841cf89138026993238c399862168..bd739ff2a070a6fcec1a64ea133d1b81f00cfc7a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span_test.cc
>@@ -29,6 +29,7 @@
> #include "absl/base/internal/exception_testing.h"
> #include "absl/container/fixed_array.h"
> #include "absl/container/inlined_vector.h"
>+#include "absl/hash/hash_testing.h"
> #include "absl/strings/str_cat.h"
> 
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant.h
>index 17e0634de03746433674e99ec077b6f773e8b2d9..2f78722f8b7c956d6d176e902d4e009ca9694a2d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant.h
>@@ -414,9 +414,9 @@ constexpr absl::add_pointer_t<const T> get_if(
> //   };
> //
> //   // Declare our variant, and call `absl::visit()` on it.
>-//   std::variant<int, std::string> foo = std::string("foo");
>+//   absl::variant<int, std::string> foo = std::string("foo");
> //   GetVariant visitor;
>-//   std::visit(visitor, foo);  // Prints `The variant's value is: foo'
>+//   absl::visit(visitor, foo);  // Prints `The variant's value is: foo'
> template <typename Visitor, typename... Variants>
> variant_internal::VisitResult<Visitor, Variants...> visit(Visitor&& vis,
>                                                           Variants&&... vars) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_exception_safety_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_exception_safety_test.cc
>index 27c0b96ca6f340850b0251bc6e916d37cdf11645..58436f07b843748e7a63d11c829e16f5551a6db8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_exception_safety_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_exception_safety_test.cc
>@@ -53,13 +53,13 @@ void ToValuelessByException(ThrowingVariant& v) {  // NOLINT
>   try {
>     v.emplace<Thrower>();
>     v.emplace<Thrower>(ExceptionOnConversion<Thrower>());
>-  } catch (ConversionException& /*e*/) {
>+  } catch (const ConversionException&) {
>     // This space intentionally left blank.
>   }
> }
> 
> // Check that variant is still in a usable state after an exception is thrown.
>-testing::AssertionResult CheckInvariants(ThrowingVariant* v) {
>+testing::AssertionResult VariantInvariants(ThrowingVariant* v) {
>   using testing::AssertionFailure;
>   using testing::AssertionSuccess;
> 
>@@ -100,7 +100,7 @@ testing::AssertionResult CheckInvariants(ThrowingVariant* v) {
>     auto unused = absl::get<Thrower>(*v);
>     static_cast<void>(unused);
>     return AssertionFailure() << "Variant should not contain Thrower";
>-  } catch (absl::bad_variant_access) {
>+  } catch (const absl::bad_variant_access&) {
>   } catch (...) {
>     return AssertionFailure() << "Unexpected exception throw from absl::get";
>   }
>@@ -213,8 +213,8 @@ TEST(VariantExceptionSafetyTest, CopyAssign) {
>         MakeExceptionSafetyTester()
>             .WithInitialValue(WithThrower())
>             .WithOperation([&rhs](ThrowingVariant* lhs) { *lhs = rhs; });
>-    EXPECT_TRUE(tester.WithInvariants(CheckInvariants).Test());
>-    EXPECT_FALSE(tester.WithInvariants(strong_guarantee).Test());
>+    EXPECT_TRUE(tester.WithContracts(VariantInvariants).Test());
>+    EXPECT_FALSE(tester.WithContracts(strong_guarantee).Test());
>   }
>   {
>     const ThrowingVariant rhs(ExpectedThrowerVec());
>@@ -222,8 +222,8 @@ TEST(VariantExceptionSafetyTest, CopyAssign) {
>         MakeExceptionSafetyTester()
>             .WithInitialValue(WithThrowerVec())
>             .WithOperation([&rhs](ThrowingVariant* lhs) { *lhs = rhs; });
>-    EXPECT_TRUE(tester.WithInvariants(CheckInvariants).Test());
>-    EXPECT_FALSE(tester.WithInvariants(strong_guarantee).Test());
>+    EXPECT_TRUE(tester.WithContracts(VariantInvariants).Test());
>+    EXPECT_FALSE(tester.WithContracts(strong_guarantee).Test());
>   }
>   // libstdc++ std::variant has bugs on copy assignment regarding exception
>   // safety.
>@@ -251,12 +251,12 @@ TEST(VariantExceptionSafetyTest, CopyAssign) {
>             .WithInitialValue(WithCopyNoThrow())
>             .WithOperation([&rhs](ThrowingVariant* lhs) { *lhs = rhs; });
>     EXPECT_TRUE(tester
>-                    .WithInvariants(CheckInvariants,
>-                                    [](ThrowingVariant* lhs) {
>-                                      return lhs->valueless_by_exception();
>-                                    })
>+                    .WithContracts(VariantInvariants,
>+                                   [](ThrowingVariant* lhs) {
>+                                     return lhs->valueless_by_exception();
>+                                   })
>                     .Test());
>-    EXPECT_FALSE(tester.WithInvariants(strong_guarantee).Test());
>+    EXPECT_FALSE(tester.WithContracts(strong_guarantee).Test());
>   }
> #endif  // !(defined(ABSL_HAVE_STD_VARIANT) && defined(__GLIBCXX__))
>   {
>@@ -268,7 +268,7 @@ TEST(VariantExceptionSafetyTest, CopyAssign) {
>     const ThrowingVariant rhs(MoveNothrow{});
>     EXPECT_TRUE(MakeExceptionSafetyTester()
>                     .WithInitialValue(WithThrower())
>-                    .WithInvariants(CheckInvariants, strong_guarantee)
>+                    .WithContracts(VariantInvariants, strong_guarantee)
>                     .Test([&rhs](ThrowingVariant* lhs) { *lhs = rhs; }));
>   }
> }
>@@ -304,11 +304,11 @@ TEST(VariantExceptionSafetyTest, MoveAssign) {
>                         *lhs = std::move(copy);
>                       });
>     EXPECT_TRUE(tester
>-                    .WithInvariants(
>-                        CheckInvariants,
>+                    .WithContracts(
>+                        VariantInvariants,
>                         [&](ThrowingVariant* lhs) { return lhs->index() == j; })
>                     .Test());
>-    EXPECT_FALSE(tester.WithInvariants(strong_guarantee).Test());
>+    EXPECT_FALSE(tester.WithContracts(strong_guarantee).Test());
>   }
>   {
>     // - otherwise (index() != j), equivalent to
>@@ -318,10 +318,10 @@ TEST(VariantExceptionSafetyTest, MoveAssign) {
>     ThrowingVariant rhs(CopyNothrow{});
>     EXPECT_TRUE(MakeExceptionSafetyTester()
>                     .WithInitialValue(WithThrower())
>-                    .WithInvariants(CheckInvariants,
>-                                    [](ThrowingVariant* lhs) {
>-                                      return lhs->valueless_by_exception();
>-                                    })
>+                    .WithContracts(VariantInvariants,
>+                                   [](ThrowingVariant* lhs) {
>+                                     return lhs->valueless_by_exception();
>+                                   })
>                     .Test([&](ThrowingVariant* lhs) {
>                       auto copy = rhs;
>                       *lhs = std::move(copy);
>@@ -347,12 +347,12 @@ TEST(VariantExceptionSafetyTest, ValueAssign) {
>             .WithInitialValue(WithThrower())
>             .WithOperation([rhs](ThrowingVariant* lhs) { *lhs = rhs; });
>     EXPECT_TRUE(copy_tester
>-                    .WithInvariants(CheckInvariants,
>-                                    [](ThrowingVariant* lhs) {
>-                                      return !lhs->valueless_by_exception();
>-                                    })
>+                    .WithContracts(VariantInvariants,
>+                                   [](ThrowingVariant* lhs) {
>+                                     return !lhs->valueless_by_exception();
>+                                   })
>                     .Test());
>-    EXPECT_FALSE(copy_tester.WithInvariants(strong_guarantee).Test());
>+    EXPECT_FALSE(copy_tester.WithContracts(strong_guarantee).Test());
>     // move assign
>     auto move_tester = MakeExceptionSafetyTester()
>                            .WithInitialValue(WithThrower())
>@@ -361,13 +361,13 @@ TEST(VariantExceptionSafetyTest, ValueAssign) {
>                              *lhs = std::move(copy);
>                            });
>     EXPECT_TRUE(move_tester
>-                    .WithInvariants(CheckInvariants,
>-                                    [](ThrowingVariant* lhs) {
>-                                      return !lhs->valueless_by_exception();
>-                                    })
>+                    .WithContracts(VariantInvariants,
>+                                   [](ThrowingVariant* lhs) {
>+                                     return !lhs->valueless_by_exception();
>+                                   })
>                     .Test());
> 
>-    EXPECT_FALSE(move_tester.WithInvariants(strong_guarantee).Test());
>+    EXPECT_FALSE(move_tester.WithContracts(strong_guarantee).Test());
>   }
>   // Otherwise (*this holds something else), if is_nothrow_constructible_v<Tj,
>   // T> || !is_nothrow_move_constructible_v<Tj> is true, equivalent to
>@@ -400,12 +400,12 @@ TEST(VariantExceptionSafetyTest, ValueAssign) {
>             .WithInitialValue(WithCopyNoThrow())
>             .WithOperation([&rhs](ThrowingVariant* lhs) { *lhs = rhs; });
>     EXPECT_TRUE(copy_tester
>-                    .WithInvariants(CheckInvariants,
>-                                    [](ThrowingVariant* lhs) {
>-                                      return lhs->valueless_by_exception();
>-                                    })
>+                    .WithContracts(VariantInvariants,
>+                                   [](ThrowingVariant* lhs) {
>+                                     return lhs->valueless_by_exception();
>+                                   })
>                     .Test());
>-    EXPECT_FALSE(copy_tester.WithInvariants(strong_guarantee).Test());
>+    EXPECT_FALSE(copy_tester.WithContracts(strong_guarantee).Test());
>     // move
>     auto move_tester = MakeExceptionSafetyTester()
>                            .WithInitialValue(WithCopyNoThrow())
>@@ -413,12 +413,12 @@ TEST(VariantExceptionSafetyTest, ValueAssign) {
>                              *lhs = ExpectedThrower(testing::nothrow_ctor);
>                            });
>     EXPECT_TRUE(move_tester
>-                    .WithInvariants(CheckInvariants,
>-                                    [](ThrowingVariant* lhs) {
>-                                      return lhs->valueless_by_exception();
>-                                    })
>+                    .WithContracts(VariantInvariants,
>+                                   [](ThrowingVariant* lhs) {
>+                                     return lhs->valueless_by_exception();
>+                                   })
>                     .Test());
>-    EXPECT_FALSE(move_tester.WithInvariants(strong_guarantee).Test());
>+    EXPECT_FALSE(move_tester.WithContracts(strong_guarantee).Test());
>   }
>   // Otherwise (if is_nothrow_constructible_v<Tj, T> == false &&
>   // is_nothrow_move_constructible<Tj> == true),
>@@ -432,7 +432,7 @@ TEST(VariantExceptionSafetyTest, ValueAssign) {
>     MoveNothrow rhs;
>     EXPECT_TRUE(MakeExceptionSafetyTester()
>                     .WithInitialValue(WithThrower())
>-                    .WithInvariants(CheckInvariants, strong_guarantee)
>+                    .WithContracts(VariantInvariants, strong_guarantee)
>                     .Test([&rhs](ThrowingVariant* lhs) { *lhs = rhs; }));
>   }
> #endif  // !(defined(ABSL_HAVE_STD_VARIANT) && defined(__GLIBCXX__))
>@@ -450,12 +450,12 @@ TEST(VariantExceptionSafetyTest, Emplace) {
>                         v->emplace<Thrower>(args);
>                       });
>     EXPECT_TRUE(tester
>-                    .WithInvariants(CheckInvariants,
>-                                    [](ThrowingVariant* v) {
>-                                      return v->valueless_by_exception();
>-                                    })
>+                    .WithContracts(VariantInvariants,
>+                                   [](ThrowingVariant* v) {
>+                                     return v->valueless_by_exception();
>+                                   })
>                     .Test());
>-    EXPECT_FALSE(tester.WithInvariants(strong_guarantee).Test());
>+    EXPECT_FALSE(tester.WithContracts(strong_guarantee).Test());
>   }
> }
> 
>@@ -472,7 +472,7 @@ TEST(VariantExceptionSafetyTest, Swap) {
>     ThrowingVariant rhs = ExpectedThrower();
>     EXPECT_TRUE(MakeExceptionSafetyTester()
>                     .WithInitialValue(WithThrower())
>-                    .WithInvariants(CheckInvariants)
>+                    .WithContracts(VariantInvariants)
>                     .Test([&](ThrowingVariant* lhs) {
>                       auto copy = rhs;
>                       lhs->swap(copy);
>@@ -486,7 +486,7 @@ TEST(VariantExceptionSafetyTest, Swap) {
>     ThrowingVariant rhs = ExpectedThrower();
>     EXPECT_TRUE(MakeExceptionSafetyTester()
>                     .WithInitialValue(WithCopyNoThrow())
>-                    .WithInvariants(CheckInvariants)
>+                    .WithContracts(VariantInvariants)
>                     .Test([&](ThrowingVariant* lhs) {
>                       auto copy = rhs;
>                       lhs->swap(copy);
>@@ -496,7 +496,7 @@ TEST(VariantExceptionSafetyTest, Swap) {
>     ThrowingVariant rhs = ExpectedThrower();
>     EXPECT_TRUE(MakeExceptionSafetyTester()
>                     .WithInitialValue(WithCopyNoThrow())
>-                    .WithInvariants(CheckInvariants)
>+                    .WithContracts(VariantInvariants)
>                     .Test([&](ThrowingVariant* lhs) {
>                       auto copy = rhs;
>                       copy.swap(*lhs);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_test.cc
>index 262bd9446c1841a570715745c3434ee1b8e19f95..bfb8bd7a0223f3cd9834e4764dc5a73bc1664def 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_test.cc
>@@ -403,7 +403,7 @@ struct is_trivially_move_constructible
> 
> template <class T>
> struct is_trivially_move_assignable
>-    : std::is_move_assignable<SingleUnion<T>>::type {};
>+    : absl::is_move_assignable<SingleUnion<T>>::type {};
> 
> TEST(VariantTest, NothrowMoveConstructible) {
>   // Verify that variant is nothrow move constructible iff its template
>@@ -2439,14 +2439,14 @@ TEST(VariantTest, TestMoveConversionViaConvertVariantTo) {
> 
> TEST(VariantTest, TestCopyAndMoveTypeTraits) {
>   EXPECT_TRUE(std::is_copy_constructible<variant<std::string>>::value);
>-  EXPECT_TRUE(std::is_copy_assignable<variant<std::string>>::value);
>+  EXPECT_TRUE(absl::is_copy_assignable<variant<std::string>>::value);
>   EXPECT_TRUE(std::is_move_constructible<variant<std::string>>::value);
>-  EXPECT_TRUE(std::is_move_assignable<variant<std::string>>::value);
>+  EXPECT_TRUE(absl::is_move_assignable<variant<std::string>>::value);
>   EXPECT_TRUE(std::is_move_constructible<variant<std::unique_ptr<int>>>::value);
>-  EXPECT_TRUE(std::is_move_assignable<variant<std::unique_ptr<int>>>::value);
>+  EXPECT_TRUE(absl::is_move_assignable<variant<std::unique_ptr<int>>>::value);
>   EXPECT_FALSE(
>       std::is_copy_constructible<variant<std::unique_ptr<int>>>::value);
>-  EXPECT_FALSE(std::is_copy_assignable<variant<std::unique_ptr<int>>>::value);
>+  EXPECT_FALSE(absl::is_copy_assignable<variant<std::unique_ptr<int>>>::value);
> 
>   EXPECT_FALSE(
>       absl::is_trivially_copy_constructible<variant<std::string>>::value);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/utility.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/utility.h
>index d73602c47d3b6da166357fbcdc5d41519d623561..aef4baa02dcd67aceb254f0a9e4bff00da4ce5b0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/utility.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/utility.h
>@@ -235,13 +235,13 @@ auto apply_helper(Functor&& functor, Tuple&& t, index_sequence<Indexes...>)
> // Example:
> //
> //   class Foo{void Bar(int);};
>-//   void user_function(int, std::string);
>+//   void user_function(int, string);
> //   void user_function(std::unique_ptr<Foo>);
> //
> //   int main()
> //   {
>-//       std::tuple<int, std::string> tuple1(42, "bar");
>-//       // Invokes the user function overload on int, std::string.
>+//       std::tuple<int, string> tuple1(42, "bar");
>+//       // Invokes the user function overload on int, string.
> //       absl::apply(&user_function, tuple1);
> //
> //       auto foo = absl::make_unique<Foo>();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/AUTHORS b/Source/ThirdParty/libwebrtc/Source/webrtc/AUTHORS
>index 6b0afd861810fabf5b6d53f335b88579e137d647..1216b48d95905562cd6e1ffe797f0c63ed23d8a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/AUTHORS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/AUTHORS
>@@ -16,6 +16,7 @@ Cody Barnes <conceptgenesis@gmail.com>
> Colin Plumb
> David Porter <david@porter.me>
> Dax Booysen <dax@younow.com>
>+Danail Kirov <dkirovbroadsoft@gmail.com>
> Dmitry Lizin <sdkdimon@gmail.com>
> Eric Rescorla, RTFM Inc. <ekr@rtfm.com>
> Frederik Riedel, Frogg GmbH <frederik.riedel@frogg.io>
>@@ -47,6 +48,7 @@ Peng Yu <yupeng323@gmail.com>
> Rafael Lopez Diez <rafalopezdiez@gmail.com>
> Ralph Giles <giles@ghostscript.com>
> Riku Voipio <riku.voipio@linaro.org>
>+Robert Bares <robert@bares.me>
> Robert Nagy <robert.nagy@gmail.com>
> Ryan Yoakum <ryoakum@skobalt.com>
> Satender Saroha <ssaroha@yahoo.com>
>@@ -56,6 +58,7 @@ Silviu Caragea <silviu.cpp@gmail.com>
> Stefan Gula <steweg@gmail.com>
> Steve Reid <sreid@sea-to-sky.net>
> Tarun Chawla <trnkumarchawla@gmail.com>
>+Uladzislau Susha <landby@gmail.com>
> Vladimir Beloborodov <VladimirTechMan@gmail.com>
> Vicken Simonian <vsimon@gmail.com>
> Victor Costan <costan@gmail.com>
>@@ -70,6 +73,10 @@ Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
> Maxim Pavlov <pavllovmax@gmail.com>
> Yusuke Suzuki <utatane.tea@gmail.com>
> Piasy Xu <xz4215@gmail.com>
>+Tomas Popela <tomas.popela@gmail.com>
>+Jan Grulich <grulja@gmail.com>
>+Eike Rathke <erathke@redhat.com>
>+Michel Promonet <michel.promonet.1@gmail.com>
> 
> &yet LLC <*@andyet.com>
> Agora IO <*@agora.io>
>@@ -80,6 +87,7 @@ Google Inc. <*@google.com>
> HyperConnect Inc. <*@hpcnt.com>
> Life On Air Inc. <*@lifeonair.com>
> Intel Corporation <*@intel.com>
>+Microsoft Corporation <*@microsoft.com>
> MIPS Technologies <*@mips.com>
> Mozilla Foundation <*@mozilla.com>
> Opera Software ASA <*@opera.com>
>@@ -97,4 +105,5 @@ Wire Swiss GmbH <*@wire.com>
> Miguel Paris <mparisdiaz@gmail.com>
> Vewd Software AS <*@vewd.com>
> Highfive, Inc. <*@highfive.com>
>-CoSMo Software Consulting, Pte Ltd <*@cosmosoftware.io>
>\ No newline at end of file
>+CoSMo Software Consulting, Pte Ltd <*@cosmosoftware.io>
>+Tuple, LLC <*@tuple.app>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/BUILD.gn
>index 5a1446d9563063214847f419f10e140c0ff32673..249094bdd6ed8810810bb6a0f375e48a03d75386 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/BUILD.gn
>@@ -40,12 +40,14 @@ if (!build_with_chromium) {
>     if (rtc_include_tests) {
>       deps += [
>         ":rtc_unittests",
>+        ":slow_tests",
>         ":video_engine_tests",
>         ":webrtc_nonparallel_tests",
>         ":webrtc_perf_tests",
>         "call:fake_network_unittests",
>         "common_audio:common_audio_unittests",
>         "common_video:common_video_unittests",
>+        "examples:examples_unittests",
>         "media:rtc_media_unittests",
>         "modules:modules_tests",
>         "modules:modules_unittests",
>@@ -54,10 +56,8 @@ if (!build_with_chromium) {
>         "modules/remote_bitrate_estimator:bwe_simulations_tests",
>         "modules/rtp_rtcp:test_packet_masks_metrics",
>         "modules/video_capture:video_capture_internal_impl",
>-        "ortc:ortc_unittests",
>         "pc:peerconnection_unittests",
>         "pc:rtc_pc_unittests",
>-        "rtc_base:rtc_base_tests_utils",
>         "stats:rtc_stats_unittests",
>         "system_wrappers:system_wrappers_unittests",
>         "test",
>@@ -98,6 +98,10 @@ config("common_inherited_config") {
>   cflags = []
>   ldflags = []
> 
>+  if (rtc_enable_symbol_export) {
>+    defines = [ "WEBRTC_ENABLE_SYMBOL_EXPORT" ]
>+  }
>+
>   if (build_with_mozilla) {
>     defines += [ "WEBRTC_MOZILLA_BUILD" ]
>   }
>@@ -218,8 +222,8 @@ config("common_config") {
>     defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
>   }
> 
>-  if (!rtc_libvpx_build_vp9) {
>-    defines += [ "RTC_DISABLE_VP9" ]
>+  if (rtc_libvpx_build_vp9) {
>+    defines += [ "RTC_ENABLE_VP9" ]
>   }
> 
>   if (rtc_enable_sctp) {
>@@ -374,29 +378,33 @@ if (!build_with_chromium) {
> 
>     deps = [
>       ":webrtc_common",
>+      "api:libjingle_peerconnection_api",
>       "api:transport_api",
>       "audio",
>       "call",
>       "common_audio",
>       "common_video",
>+      "logging:rtc_event_log_api",
>+      "logging:rtc_event_log_impl_base",
>       "media",
>       "modules",
>       "modules/video_capture:video_capture_internal_impl",
>-      "ortc",
>+      "p2p:rtc_p2p",
>+      "pc:libjingle_peerconnection",
>+      "pc:peerconnection",
>+      "pc:rtc_pc",
>+      "pc:rtc_pc_base",
>       "rtc_base",
>       "sdk",
>       "video",
>     ]
> 
>-    # Additional factory functions to be exposed.
>-    # Rational: These factories are small enough (89 KiB / 32+ MiB)
>-    #           to be unconditionaly included for user convenience.
>-    # That begin said:
>-    # TODO(yvesg) Consider making all non-core APIs optional, so that users
>-    #             can build a customized library tailored to their needs.
>+    # Include audio and video codecs by default.
>     deps += [
>       "api/audio_codecs:builtin_audio_decoder_factory",
>       "api/audio_codecs:builtin_audio_encoder_factory",
>+      "api/video_codecs:builtin_video_decoder_factory",
>+      "api/video_codecs:builtin_video_encoder_factory",
>     ]
> 
>     if (build_with_mozilla) {
>@@ -430,9 +438,9 @@ rtc_source_set("webrtc_common") {
>   deps = [
>     "api:array_view",
>     "api/video:video_bitrate_allocation",
>+    "api/video:video_frame",
>     "rtc_base:checks",
>-    "rtc_base:deprecation",
>-    "rtc_base:stringutils",
>+    "//third_party/abseil-cpp/absl/strings",
>   ]
> 
>   if (!build_with_chromium && is_clang) {
>@@ -467,7 +475,6 @@ if (rtc_include_tests) {
>       "p2p:rtc_p2p_unittests",
>       "rtc_base:rtc_base_approved_unittests",
>       "rtc_base:rtc_base_tests_main",
>-      "rtc_base:rtc_base_tests_utils",
>       "rtc_base:rtc_base_unittests",
>       "rtc_base:rtc_json_unittests",
>       "rtc_base:rtc_numerics_unittests",
>@@ -499,6 +506,17 @@ if (rtc_include_tests) {
>     }
>   }
> 
>+  # This runs tests that must run in real time and therefore can take some
>+  # time to execute. They are in a separate executable to avoid making the
>+  # regular unittest suite too slow to run frequently.
>+  rtc_test("slow_tests") {
>+    testonly = true
>+    deps = [
>+      "modules/congestion_controller/goog_cc:goog_cc_slow_tests",
>+      "test:test_main",
>+    ]
>+  }
>+
>   # TODO(pbos): Rename test suite, this is no longer "just" for video targets.
>   video_engine_tests_resources = [
>     "resources/foreman_cif_short.yuv",
>@@ -524,7 +542,6 @@ if (rtc_include_tests) {
>       # should move them to a more appropriate test suite.
>       "call:call_tests",
>       "modules/video_capture",
>-      "rtc_base:rtc_base_tests_utils",
>       "test:test_common",
>       "test:test_main",
>       "test:video_test_common",
>@@ -617,6 +634,7 @@ if (rtc_include_tests) {
>         "sdk/android/tests/src/org/webrtc/GlGenericDrawerTest.java",
>         "sdk/android/tests/src/org/webrtc/HardwareVideoEncoderTest.java",
>         "sdk/android/tests/src/org/webrtc/ScalingSettingsTest.java",
>+        "sdk/android/tests/src/org/webrtc/CryptoOptionsTest.java",
>       ]
> 
>       deps = [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/DEPS
>index 67aa4a803c20919c26355ca3f09e8c25bde1d51d..555c27f10e7e199730006d4db97d360d2c158376 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/DEPS
>@@ -7,16 +7,16 @@ vars = {
>   'checkout_configuration': 'default',
>   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
>   'webrtc_git': 'https://webrtc.googlesource.com',
>-  'chromium_revision': 'ccb83d4a5512802f1193a506b5431e85e0138043',
>+  'chromium_revision': 'b04e513f825d6ce22e147601cbfc75ea23eb9073',
>   'boringssl_git': 'https://boringssl.googlesource.com',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling swarming_client
>   # and whatever else without interference from each other.
>-  'swarming_revision': '486c9b53c4d54dd4b95bb6ce0e31160e600dfc11',
>+  'swarming_revision': '157bec8a25cc4ebd6a16052510d08b05b6102aad',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling BoringSSL
>   # and whatever else without interference from each other.
>-  'boringssl_revision': 'ce00828c89df3d4c40de7d715b1a032eb03c525c',
>+  'boringssl_revision': '6965d25602754bc419c5f757d008ba1f4da49ae4',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling lss
>   # and whatever else without interference from each other.
>@@ -24,29 +24,29 @@ vars = {
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling catapult
>   # and whatever else without interference from each other.
>-  'catapult_revision': '4fc4281d2152a36762a0bf47ba76bf8d4a6ee234',
>+  'catapult_revision': '17079a5cc1f27a2dc57558a0897666cac4a5bc64',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling libFuzzer
>   # and whatever else without interference from each other.
>-  'libfuzzer_revision': 'a305a5eb85ed42edc5c965c14f308f576cb245ca',
>+  'libfuzzer_revision': '2a53098584c48af50aec3fb51febe5e651489774',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling freetype
>   # and whatever else without interference from each other.
>-  'freetype_revision': 'abd997aa7cf2bc9219136782c7363d14d325199c',
>+  'freetype_revision': 'fb0d66d04c4dd8d7f9604af1a6001b2737cb5098',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling HarfBuzz
>   # and whatever else without interference from each other.
>-  'harfbuzz_revision': '54d332dd9b0263821376161cdffb60ffb3c7847f',
>+  'harfbuzz_revision': '574d888c8a409295a952361a39c8e83a52a0fc3d',
> }
> deps = {
>   # TODO(kjellander): Move this to be Android-only once the libevent dependency
>   # in base/third_party/libevent is solved.
>   'src/base':
>-    Var('chromium_git') + '/chromium/src/base' + '@' + '0db61379d8f76807e6f78a35ccff9227205b6195',
>+    Var('chromium_git') + '/chromium/src/base' + '@' + 'b9901f9bc5b65a1169d5db696ef4428a65cec622',
>   'src/build':
>-    Var('chromium_git') + '/chromium/src/build' + '@' + '8bb9fef862bd7570c5b465860c427701c81c1531',
>+    Var('chromium_git') + '/chromium/src/build' + '@' + 'bbd67a350d745200e5798cace11a28edfd9fc3b2',
>   'src/buildtools':
>-    Var('chromium_git') + '/chromium/buildtools.git' + '@' + '2dff9c9c74e9d732e6fe57c84ef7fd044cc45d96',
>+    Var('chromium_git') + '/chromium/buildtools.git' + '@' + '04161ec8d7c781e4498c699254c69ba0dd959fde',
>   # Gradle 4.3-rc4. Used for testing Android Studio project generation for WebRTC.
>   'src/examples/androidtests/third_party/gradle': {
>     'url': Var('chromium_git') + '/external/github.com/gradle/gradle.git' + '@' +
>@@ -54,21 +54,44 @@ deps = {
>     'condition': 'checkout_android',
>   },
>   'src/ios': {
>-    'url': Var('chromium_git') + '/chromium/src/ios' + '@' + 'd4e010516156c2ef2ccd974d4aa0abd9ef48c698',
>+    'url': Var('chromium_git') + '/chromium/src/ios' + '@' + 'eb9e8e09dc4b28e691cfd70242ba50ceecb6ab20',
>     'condition': 'checkout_ios',
>   },
>   'src/testing':
>-    Var('chromium_git') + '/chromium/src/testing' + '@' + '677e1ef2627ca32226f5e801ebec39613035873b',
>+    Var('chromium_git') + '/chromium/src/testing' + '@' + '5a776bca050aedfb0269cdd8408ff0559acde2aa',
>   'src/third_party':
>-    Var('chromium_git') + '/chromium/src/third_party' + '@' + 'dcfac1c5eda73c8fbd0451c8e40454b284c67110',
>+    Var('chromium_git') + '/chromium/src/third_party' + '@' + '77f6c16720eff7721b0c0b2438ce9126f59e1f0b',
>   'src/third_party/android_ndk': {
>       'url': Var('chromium_git') + '/android_ndk.git' + '@' + '4e2cea441bfd43f0863d14f57b1e1844260b9884',
>       'condition': 'checkout_android',
>   },
>   'src/third_party/android_tools': {
>-    'url': Var('chromium_git') + '/android_tools.git' + '@' + '130499e25286f4d56acafa252fee09f3cc595c49',
>+    'url': Var('chromium_git') + '/android_tools.git' + '@' + '6fecaa542f73dd5aeed170d9a4cf340159b42976',
>     'condition': 'checkout_android',
>   },
>+
>+  'src/third_party/android_build_tools/aapt2': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_build_tools/aapt2',
>+              'version': 'version:3.3.0-beta01-5013011-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/android_build_tools/bundletool': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_tools_bundletool',
>+              'version': 'version:0.7.1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>   'src/third_party/auto/src': {
>     'url': Var('chromium_git') + '/external/github.com/google/auto.git' + '@' + '8a81a858ae7b78a1aef71ac3905fade0bbd64e82',
>     'condition': 'checkout_android',
>@@ -84,7 +107,7 @@ deps = {
>   'src/third_party/colorama/src':
>     Var('chromium_git') + '/external/colorama.git' + '@' + '799604a1041e9b3bc5d2789ecbd7e8db2e18e6b8',
>   'src/third_party/depot_tools':
>-    Var('chromium_git') + '/chromium/tools/depot_tools.git' + '@' + '71e3be7a50c21faeee91ed99a8d5addfb7594e7c',
>+    Var('chromium_git') + '/chromium/tools/depot_tools.git' + '@' + '44d4b29082f0d8bacacd623f91c4d29637b4b901',
>   'src/third_party/errorprone/lib': {
>       'url': Var('chromium_git') + '/chromium/third_party/errorprone.git' + '@' + '980d49e839aa4984015efed34b0134d4b2c9b6d7',
>       'condition': 'checkout_android',
>@@ -101,7 +124,7 @@ deps = {
>     Var('chromium_git') + '/external/github.com/harfbuzz/harfbuzz.git' + '@' + Var('harfbuzz_revision'),
>   # WebRTC-only dependency (not present in Chromium).
>   'src/third_party/gtest-parallel':
>-    Var('chromium_git') + '/external/github.com/google/gtest-parallel' + '@' + 'fe7f791f14769390d0b124ef8231cde4d575eb12',
>+    Var('chromium_git') + '/external/github.com/google/gtest-parallel' + '@' + 'e472187d1129e508890aa20ac914adeac2e7f7b6',
>   'src/third_party/google-truth': {
>       'packages': [
>           {
>@@ -113,9 +136,9 @@ deps = {
>       'dep_type': 'cipd',
>   },
>   'src/third_party/googletest/src':
>-    Var('chromium_git') + '/external/github.com/google/googletest.git' + '@' + '2e68926a9d4929e9289373cd49e40ddcb9a628f7',
>+    Var('chromium_git') + '/external/github.com/google/googletest.git' + '@' + '879ac092fde0a19e1b3a61b2546b2a422b1528bc',
>   'src/third_party/icu': {
>-    'url': Var('chromium_git') + '/chromium/deps/icu.git' + '@' + 'c52a2a250d6c5f5cbdd015dff36af7c5d0ae1150',
>+    'url': Var('chromium_git') + '/chromium/deps/icu.git' + '@' + '407b39301e71006b68bd38e770f35d32398a7b14',
>   },
>   'src/third_party/jsr-305/src': {
>     'url': Var('chromium_git') + '/external/jsr-305.git' + '@' + '642c508235471f7220af6d5df2d3210e3bfc0919',
>@@ -135,9 +158,9 @@ deps = {
>   'src/third_party/libsrtp':
>     Var('chromium_git') + '/chromium/deps/libsrtp.git' + '@' + '650611720ecc23e0e6b32b0e3100f8b4df91696c',
>   'src/third_party/libvpx/source/libvpx':
>-    Var('chromium_git') + '/webm/libvpx.git' + '@' +  '2beb5c9f91e7166c2c9d01c94bf84767815121e4',
>+    Var('chromium_git') + '/webm/libvpx.git' + '@' +  'ac3eccdc24bccece5f73ee67b88154f3bf4a4e9a',
>   'src/third_party/libyuv':
>-    Var('chromium_git') + '/libyuv/libyuv.git' + '@' + '9a07219dc8fbf2b77e390d16bd24809444838a91',
>+    Var('chromium_git') + '/libyuv/libyuv.git' + '@' + 'b36c86fdfe746d7be904c3a565b047b24d58087e',
>   'src/third_party/lss': {
>     'url': Var('chromium_git') + '/linux-syscall-support.git' + '@' + Var('lss_revision'),
>     'condition': 'checkout_android or checkout_linux',
>@@ -146,13 +169,20 @@ deps = {
>     'url': Var('chromium_git') + '/external/mockito/mockito.git' + '@' + '04a2a289a4222f80ad20717c25144981210d2eac',
>     'condition': 'checkout_android',
>   },
>+
>+  # Used by boringssl.
>+  'src/third_party/nasm': {
>+      'url': Var('chromium_git') + '/chromium/deps/nasm.git' + '@' +
>+      'a0a6951e259bd347c133969740348bb5ebb468c4'
>+  },
>+
>   'src/third_party/openh264/src':
>     Var('chromium_git') + '/external/github.com/cisco/openh264' + '@' + '3b51f16a4a41df729f8d647f03e48c5f272911ff',
>   'src/third_party/r8': {
>       'packages': [
>           {
>               'package': 'chromium/third_party/r8',
>-              'version': 'version:1.2.48',
>+              'version': 'version:1.4.4-cr0',
>           },
>       ],
>       'condition': 'checkout_android',
>@@ -195,7 +225,7 @@ deps = {
>   'src/third_party/yasm/source/patched-yasm':
>     Var('chromium_git') + '/chromium/deps/yasm/patched-yasm.git' + '@' + '720b70524a4424b15fc57e82263568c8ba0496ad',
>   'src/tools':
>-    Var('chromium_git') + '/chromium/src/tools' + '@' + '731f63f3126cd6f5ceb2aafceade80857900a409',
>+    Var('chromium_git') + '/chromium/src/tools' + '@' + 'b47c6315878eabd73bf71c1aa0cbd9b816716789',
>   'src/tools/swarming_client':
>     Var('chromium_git') + '/infra/luci/client-py.git' + '@' +  Var('swarming_revision'),
> 
>@@ -895,17 +925,6 @@ deps = {
>       'dep_type': 'cipd',
>   },
> 
>-  'src/third_party/android_deps/libs/com_google_android_play_core': {
>-      'packages': [
>-          {
>-              'package': 'chromium/third_party/android_deps/libs/com_google_android_play_core',
>-              'version': 'version:1.3.0-cr0',
>-          },
>-      ],
>-      'condition': 'checkout_android',
>-      'dep_type': 'cipd',
>-  },
>-
>   'src/third_party/android_deps/libs/com_google_code_findbugs_jsr305': {
>       'packages': [
>           {
>@@ -1016,6 +1035,17 @@ deps = {
>       'dep_type': 'cipd',
>   },
> 
>+  'src/third_party/android_deps/libs/com_google_protobuf_protobuf_lite': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_google_protobuf_protobuf_lite',
>+              'version': 'version:3.0.1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>   'src/third_party/android_deps/libs/com_squareup_javapoet': {
>       'packages': [
>           {
>@@ -1393,8 +1423,6 @@ hooks = [
> recursedeps = [
>   # buildtools provides clang_format, libc++, and libc++abi.
>   'src/buildtools',
>-  # android_tools manages the NDK.
>-  'src/third_party/android_tools',
> ]
> 
> # Define rules for which include paths are allowed in our source.
>@@ -1416,9 +1444,12 @@ include_rules = [
>   "+test",
>   "+rtc_tools",
> 
>-  # Abseil whitelist. Keep this in sync with abseil-in-webrtc-md.
>+  # Abseil whitelist. Keep this in sync with abseil-in-webrtc.md.
>   "+absl/container/inlined_vector.h",
>   "+absl/memory/memory.h",
>+  "+absl/meta/type_traits.h",
>+  "+absl/strings/ascii.h",
>+  "+absl/strings/match.h",
>   "+absl/strings/string_view.h",
>   "+absl/types/optional.h",
>   "+absl/types/variant.h",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/OWNERS
>index bfa7801bd827b825e431fa359afddecc90f34234..289b5a3c7ea1839318bb52fee9635516e8615628 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/OWNERS
>@@ -4,19 +4,14 @@ mflodman@webrtc.org
> niklas.enbom@webrtc.org
> phoglund@webrtc.org
> stefan@webrtc.org
>-tina.legrand@webrtc.org
> tommi@webrtc.org
> per-file .gitignore=*
> per-file .gn=mbonadei@webrtc.org
> per-file *.gn=mbonadei@webrtc.org
> per-file *.gni=mbonadei@webrtc.org
>-per-file *.py=phoglund@webrtc.org
> per-file AUTHORS=*
>-per-file BUILD.gn=phoglund@webrtc.org
> per-file DEPS=*
> per-file pylintrc=phoglund@webrtc.org
>-per-file THIRD_PARTY_DEPS=phoglund@webrtc.org
>-per-file THIRD_PARTY_DEPS=titovartem@webrtc.org
> per-file WATCHLISTS=*
> per-file abseil-in-webrtc.md=danilchap@webrtc.org
> per-file abseil-in-webrtc.md=kwiberg@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/PRESUBMIT.py b/Source/ThirdParty/libwebrtc/Source/webrtc/PRESUBMIT.py
>index a7b5cc14a48fa0b80702462aa9c9101d8a027571..34b5adf777838390aa588e986513f520a2e305cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/PRESUBMIT.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/PRESUBMIT.py
>@@ -481,7 +481,8 @@ def CheckNoStreamUsageIsAdded(input_api, output_api,
>   file_filter = lambda x: (input_api.FilterSourceFile(x)
>                            and source_file_filter(x))
>   for f in input_api.AffectedSourceFiles(file_filter):
>-    if f.LocalPath() == 'PRESUBMIT.py':
>+    # Usage of stringstream is allowed under examples/.
>+    if f.LocalPath() == 'PRESUBMIT.py' or f.LocalPath().startswith('examples'):
>       continue
>     for line_num, line in f.ChangedContents():
>       if ((include_re.search(line) or usage_re.search(line))
>@@ -862,8 +863,89 @@ def CommonChecks(input_api, output_api):
>   results.extend(CheckNoStreamUsageIsAdded(
>       input_api, output_api, non_third_party_sources))
>   results.extend(CheckAddedDepsHaveTargetApprovals(input_api, output_api))
>+  results.extend(CheckApiDepsFileIsUpToDate(input_api, output_api))
>+  results.extend(CheckAbslMemoryInclude(
>+      input_api, output_api, non_third_party_sources))
>+  return results
>+
>+
>+def CheckApiDepsFileIsUpToDate(input_api, output_api):
>+  """Check that 'include_rules' in api/DEPS is up to date.
>+
>+  The file api/DEPS must be kept up to date in order to avoid to avoid to
>+  include internal header from WebRTC's api/ headers.
>+
>+  This check is focused on ensuring that 'include_rules' contains a deny
>+  rule for each root level directory. More focused allow rules can be
>+  added to 'specific_include_rules'.
>+  """
>+  results = []
>+  api_deps = os.path.join(input_api.PresubmitLocalPath(), 'api', 'DEPS')
>+  with open(api_deps) as f:
>+    deps_content = _ParseDeps(f.read())
>+
>+  include_rules = deps_content.get('include_rules', [])
>+
>+  # Only check top level directories affected by the current CL.
>+  dirs_to_check = set()
>+  for f in input_api.AffectedFiles():
>+    path_tokens = [t for t in f.LocalPath().split(os.sep) if t]
>+    if len(path_tokens) > 1:
>+      if (path_tokens[0] != 'api' and
>+          os.path.isdir(os.path.join(input_api.PresubmitLocalPath(),
>+                                     path_tokens[0]))):
>+        dirs_to_check.add(path_tokens[0])
>+
>+  missing_include_rules = set()
>+  for p in dirs_to_check:
>+    rule = '-%s' % p
>+    if rule not in include_rules:
>+      missing_include_rules.add(rule)
>+
>+  if missing_include_rules:
>+    error_msg = [
>+      'include_rules = [\n',
>+      '  ...\n',
>+    ]
>+
>+    for r in sorted(missing_include_rules):
>+      error_msg.append('  "%s",\n' % str(r))
>+
>+    error_msg.append('  ...\n')
>+    error_msg.append(']\n')
>+
>+    results.append(output_api.PresubmitError(
>+        'New root level directory detected! WebRTC api/ headers should '
>+        'not #include headers from \n'
>+        'the new directory, so please update "include_rules" in file\n'
>+        '"%s". Example:\n%s\n' % (api_deps, ''.join(error_msg))))
>+
>   return results
> 
>+def CheckAbslMemoryInclude(input_api, output_api, source_file_filter):
>+  pattern = input_api.re.compile(
>+      r'^#include\s*"absl/memory/memory.h"', input_api.re.MULTILINE)
>+  file_filter = lambda f: (f.LocalPath().endswith(('.cc', '.h'))
>+                           and source_file_filter(f))
>+
>+  files = []
>+  for f in input_api.AffectedFiles(
>+      include_deletes=False, file_filter=file_filter):
>+    contents = input_api.ReadFile(f)
>+    if pattern.search(contents):
>+      continue
>+    for _, line in f.ChangedContents():
>+      if 'absl::make_unique' in line:
>+        files.append(f)
>+        break
>+
>+  if len(files):
>+    return [output_api.PresubmitError(
>+        'Please include "absl/memory/memory.h" header for'
>+        ' absl::make_unique.\nThis header may or may not be included'
>+        ' transitively depends on the C++ standard version.',
>+        files)]
>+  return []
> 
> def CheckChangeOnUpload(input_api, output_api):
>   results = []
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/WATCHLISTS b/Source/ThirdParty/libwebrtc/Source/webrtc/WATCHLISTS
>index dc8d48c51f95bc8c43cbf0bdb2e6bb0da18f62e6..4260bf91bbc599730e04cd5324e475419d841927 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/WATCHLISTS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/WATCHLISTS
>@@ -79,6 +79,9 @@
>     'bitrate_controller': {
>       'filepath': 'modules/bitrate_controller/.*'
>     },
>+    'congestion_controller': {
>+      'filepath': 'modules/congestion_controller/.*'
>+    },
>     'remote_bitrate_estimator': {
>       'filepath': 'modules/remote_bitrate_estimator/.*'
>     },
>@@ -97,8 +100,8 @@
>   },
> 
>   'WATCHLISTS': {
>-    'this_file': [''],
>-    'all_webrtc': ['tterriberry@mozilla.com'],
>+    'this_file': [],
>+    'all_webrtc': [],
>     'root_files': ['niklas.enbom@webrtc.org',
>                    'peah@webrtc.org',
>                    'qiang.lu@intel.com',
>@@ -106,16 +109,14 @@
>     'build_files': ['mbonadei@webrtc.org'],
>     'common_audio': ['alessiob@webrtc.org',
>                      'aluebs@webrtc.org',
>-                     'andrew@webrtc.org',
>                      'audio-team@agora.io',
>-                     'bjornv@webrtc.org',
>                      'minyue@webrtc.org',
>                      'peah@webrtc.org'],
>-    'audio': ['solenberg@webrtc.org',
>-              'tina.legrand@webrtc.org'],
>+    'audio': ['solenberg@webrtc.org'],
>     'api': ['kwiberg@webrtc.org',
>             'solenberg@webrtc.org'],
>-    'base': ['kwiberg@webrtc.org'],
>+    'base': ['kwiberg@webrtc.org',
>+             'benwright@webrtc.org'],
>     'call': ['mflodman@webrtc.org',
>              'solenberg@webrtc.org',
>              'stefan@webrtc.org'],
>@@ -137,8 +138,7 @@
>                      'henrik.lundin@webrtc.org',
>                      'kwiberg@webrtc.org',
>                      'minyue@webrtc.org',
>-                     'peah@webrtc.org',
>-                     'tina.legrand@webrtc.org'],
>+                     'peah@webrtc.org'],
>     'neteq': ['alessiob@webrtc.org',
>               'audio-team@agora.io',
>               'henrik.lundin@webrtc.org',
>@@ -149,9 +149,7 @@
>     'audio_processing': ['aleloi@webrtc.org',
>                          'alessiob@webrtc.org',
>                          'aluebs@webrtc.org',
>-                         'andrew@webrtc.org',
>                          'audio-team@agora.io',
>-                         'bjornv@webrtc.org',
>                          'henrik.lundin@webrtc.org',
>                          'kwiberg@webrtc.org',
>                          'minyue@webrtc.org',
>@@ -166,11 +164,14 @@
>                          'zhengzhonghou@agora.io'],
>     'bitrate_controller': ['mflodman@webrtc.org',
>                            'stefan@webrtc.org',
>+                           'srte@webrtc.org',
>                            'zhuangzesen@agora.io'],
>+    'congestion_controller': ['srte@webrtc.org'],
>     'remote_bitrate_estimator': ['mflodman@webrtc.org',
>                                  'stefan@webrtc.org',
>                                  'zhuangzesen@agora.io'],
>     'pacing': ['mflodman@webrtc.org',
>+               'srte@webrtc.org',
>                'stefan@webrtc.org',
>                'zhuangzesen@agora.io'],
>     'rtp_rtcp': ['mflodman@webrtc.org',
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/abseil-in-webrtc.md b/Source/ThirdParty/libwebrtc/Source/webrtc/abseil-in-webrtc.md
>index 04ae8d28de12b74ccba5cd51a6e521f384329d47..1075aa9b7962015e2e973c3550c5e69b55da25da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/abseil-in-webrtc.md
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/abseil-in-webrtc.md
>@@ -15,6 +15,10 @@ adds the first use.
> * `absl::make_unique` and `absl::WrapUnique`
> * `absl::optional` and related stuff from `absl/types/optional.h`.
> * `absl::string_view`
>+* The functions in `absl/strings/ascii.h` and `absl/strings/match.h`
>+* `absl::is_trivially_copy_constructible`,
>+  `absl::is_trivially_copy_assignable`, and
>+  `absl::is_trivially_destructible` from `absl/meta/type_traits.h`.
> * `absl::variant` and related stuff from `absl/types/variant.h`.
> 
> ## **Disallowed**
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/BUILD.gn
>index 5b21334fe9859e4eb4d04affb346253f89b7c7be..fe27c7b8880b8f2eba155f39ef7fb49a3629bc8d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/BUILD.gn
>@@ -44,6 +44,33 @@ rtc_source_set("callfactory_api") {
>   ]
> }
> 
>+rtc_static_library("create_peerconnection_factory") {
>+  visibility = [ "*" ]
>+  allow_poison = [ "software_video_codecs" ]
>+  sources = [
>+    "create_peerconnection_factory.cc",
>+    "create_peerconnection_factory.h",
>+  ]
>+  deps = [
>+    ":callfactory_api",
>+    ":fec_controller_api",
>+    ":libjingle_peerconnection_api",
>+    "../logging:rtc_event_log_api",
>+    "../logging:rtc_event_log_impl_base",
>+    "../media:rtc_audio_video",
>+    "../modules/audio_device:audio_device_api",
>+    "../modules/audio_processing:api",
>+    "../pc:peerconnection",
>+    "../rtc_base:ptr_util",
>+    "../rtc_base:rtc_base",
>+    "../rtc_base:rtc_base_approved",
>+    "audio:audio_mixer_api",
>+    "audio_codecs:audio_codecs_api",
>+    "transport:network_control",
>+    "video_codecs:video_codecs_api",
>+  ]
>+}
>+
> rtc_static_library("libjingle_peerconnection_api") {
>   visibility = [ "*" ]
>   cflags = []
>@@ -52,6 +79,8 @@ rtc_static_library("libjingle_peerconnection_api") {
>     "bitrate_constraints.h",
>     "candidate.cc",
>     "candidate.h",
>+    "crypto/cryptooptions.cc",
>+    "crypto/cryptooptions.h",
>     "crypto/framedecryptorinterface.h",
>     "crypto/frameencryptorinterface.h",
>     "cryptoparams.h",
>@@ -63,6 +92,7 @@ rtc_static_library("libjingle_peerconnection_api") {
>     "jsepicecandidate.cc",
>     "jsepicecandidate.h",
>     "jsepsessiondescription.h",
>+    "media_transport_interface.cc",
>     "media_transport_interface.h",
>     "mediaconstraintsinterface.cc",
>     "mediaconstraintsinterface.h",
>@@ -109,6 +139,7 @@ rtc_static_library("libjingle_peerconnection_api") {
>     "audio_codecs:audio_codecs_api",
>     "transport:bitrate_settings",
>     "transport:network_control",
>+    "video:encoded_image",
>     "video:video_frame",
>     "//third_party/abseil-cpp/absl/types:optional",
> 
>@@ -124,6 +155,7 @@ rtc_static_library("libjingle_peerconnection_api") {
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:stringutils",
>+    "../rtc_base/system:rtc_export",
>   ]
> 
>   if (is_nacl) {
>@@ -132,6 +164,13 @@ rtc_static_library("libjingle_peerconnection_api") {
>   }
> }
> 
>+rtc_source_set("scoped_refptr") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "scoped_refptr.h",
>+  ]
>+}
>+
> rtc_source_set("video_quality_test_fixture_api") {
>   visibility = [ "*" ]
>   testonly = true
>@@ -164,6 +203,7 @@ rtc_source_set("test_dependency_factory") {
>   deps = [
>     ":video_quality_test_fixture_api",
>     "../rtc_base:thread_checker",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -203,34 +243,15 @@ rtc_source_set("libjingle_logging_api") {
> rtc_source_set("ortc_api") {
>   visibility = [ "*" ]
>   sources = [
>-    "ortc/mediadescription.cc",
>-    "ortc/mediadescription.h",
>-    "ortc/ortcfactoryinterface.h",
>-    "ortc/ortcrtpreceiverinterface.h",
>-    "ortc/ortcrtpsenderinterface.h",
>     "ortc/packettransportinterface.h",
>-    "ortc/rtptransportcontrollerinterface.h",
>     "ortc/rtptransportinterface.h",
>-    "ortc/sessiondescription.cc",
>-    "ortc/sessiondescription.h",
>     "ortc/srtptransportinterface.h",
>-    "ortc/udptransportinterface.h",
>   ]
> 
>-  # For mediastreaminterface.h, etc.
>-  # TODO(deadbeef): Create a separate target for the common things ORTC and
>-  # PeerConnection code shares, so that ortc_api can depend on that instead of
>-  # libjingle_peerconnection_api.
>   deps = [
>     ":libjingle_peerconnection_api",
>-    "..:webrtc_common",
>-    "../rtc_base:rtc_base",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
> }
> 
> rtc_source_set("rtc_stats_api") {
>@@ -246,6 +267,7 @@ rtc_source_set("rtc_stats_api") {
>   deps = [
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/system:rtc_export",
>   ]
> }
> 
>@@ -270,6 +292,17 @@ rtc_source_set("transport_api") {
>   ]
> }
> 
>+rtc_source_set("bitrate_allocation") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "call/bitrate_allocation.h",
>+  ]
>+  deps = [
>+    "units:data_rate",
>+    "units:time_delta",
>+  ]
>+}
>+
> rtc_source_set("simulated_network_api") {
>   visibility = [ "*" ]
>   sources = [
>@@ -347,6 +380,7 @@ if (rtc_include_tests) {
>       ]
> 
>       deps = [
>+        "../modules/audio_processing:api",
>         "../modules/audio_processing:audio_processing",
>         "../modules/audio_processing:audioproc_f_impl",
>       ]
>@@ -446,11 +480,33 @@ if (rtc_include_tests) {
>     ]
>   }
> 
>-  rtc_source_set("fake_frame_crypto") {
>+  rtc_source_set("mock_frame_encryptor") {
>+    testonly = true
>+    sources = [
>+      "test/mock_frame_encryptor.cc",
>+      "test/mock_frame_encryptor.h",
>+    ]
>+    deps = [
>+      ":libjingle_peerconnection_api",
>+      "../test:test_support",
>+    ]
>+  }
>+
>+  rtc_source_set("mock_frame_decryptor") {
>+    testonly = true
>+    sources = [
>+      "test/mock_frame_decryptor.cc",
>+      "test/mock_frame_decryptor.h",
>+    ]
>+    deps = [
>+      ":libjingle_peerconnection_api",
>+      "../test:test_support",
>+    ]
>+  }
>+
>+  rtc_source_set("fake_frame_encryptor") {
>     testonly = true
>     sources = [
>-      "test/fake_frame_decryptor.cc",
>-      "test/fake_frame_decryptor.h",
>       "test/fake_frame_encryptor.cc",
>       "test/fake_frame_encryptor.h",
>     ]
>@@ -463,6 +519,21 @@ if (rtc_include_tests) {
>     ]
>   }
> 
>+  rtc_source_set("fake_frame_decryptor") {
>+    testonly = true
>+    sources = [
>+      "test/fake_frame_decryptor.cc",
>+      "test/fake_frame_decryptor.h",
>+    ]
>+    deps = [
>+      ":array_view",
>+      ":libjingle_peerconnection_api",
>+      "..:webrtc_common",
>+      "../rtc_base:checks",
>+      "../rtc_base:rtc_base_approved",
>+    ]
>+  }
>+
>   rtc_source_set("mock_peerconnectioninterface") {
>     testonly = true
>     sources = [
>@@ -500,6 +571,18 @@ if (rtc_include_tests) {
>     ]
>   }
> 
>+  rtc_source_set("mock_video_bitrate_allocator_factory") {
>+    testonly = true
>+    sources = [
>+      "test/mock_video_bitrate_allocator_factory.h",
>+    ]
>+
>+    deps = [
>+      "../api/video:video_bitrate_allocator_factory",
>+      "../test:test_support",
>+    ]
>+  }
>+
>   rtc_source_set("mock_video_codec_factory") {
>     testonly = true
>     sources = [
>@@ -513,15 +596,72 @@ if (rtc_include_tests) {
>     ]
>   }
> 
>+  rtc_source_set("mock_video_decoder") {
>+    visibility = [ "*" ]
>+
>+    testonly = true
>+    sources = [
>+      "test/mock_video_decoder.cc",
>+      "test/mock_video_decoder.h",
>+    ]
>+
>+    deps = [
>+      "../api/video_codecs:video_codecs_api",
>+      "../test:test_support",
>+    ]
>+  }
>+
>+  rtc_source_set("mock_video_encoder") {
>+    visibility = [ "*" ]
>+
>+    testonly = true
>+    sources = [
>+      "test/mock_video_encoder.cc",
>+      "test/mock_video_encoder.h",
>+    ]
>+
>+    deps = [
>+      "../api/video_codecs:video_codecs_api",
>+      "../test:test_support",
>+    ]
>+  }
>+
>+  rtc_source_set("fake_media_transport") {
>+    testonly = true
>+
>+    sources = [
>+      "test/fake_media_transport.h",
>+    ]
>+
>+    deps = [
>+      ":libjingle_peerconnection_api",
>+      "../rtc_base:checks",
>+      "//third_party/abseil-cpp/absl/memory:memory",
>+    ]
>+  }
>+
>+  rtc_source_set("loopback_media_transport") {
>+    testonly = true
>+
>+    sources = [
>+      "test/loopback_media_transport.h",
>+    ]
>+
>+    deps = [
>+      ":libjingle_peerconnection_api",
>+      "../rtc_base:checks",
>+      "../rtc_base:rtc_base",
>+    ]
>+  }
>+
>   rtc_source_set("rtc_api_unittests") {
>     testonly = true
> 
>     sources = [
>       "array_view_unittest.cc",
>-      "ortc/mediadescription_unittest.cc",
>-      "ortc/sessiondescription_unittest.cc",
>       "rtcerror_unittest.cc",
>       "rtpparameters_unittest.cc",
>+      "test/loopback_media_transport_unittest.cc",
>     ]
> 
>     if (!build_with_chromium && is_clang) {
>@@ -532,10 +672,10 @@ if (rtc_include_tests) {
>     deps = [
>       ":array_view",
>       ":libjingle_peerconnection_api",
>-      ":ortc_api",
>+      ":loopback_media_transport",
>       "../rtc_base:checks",
>+      "../rtc_base:gunit_helpers",
>       "../rtc_base:rtc_base_approved",
>-      "../rtc_base:rtc_base_tests_utils",
>       "../test:test_support",
>       "units:units_unittests",
>     ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/api/DEPS
>index 847cce259f9ae63a020e0d62a24201bca72457a2..286d8d1e1e6c778b40458f77d065e882212a8a30 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/DEPS
>@@ -1,15 +1,55 @@
>+# This is supposed to be a complete list of top-level directories,
>+# excepting only api/ itself.
> include_rules = [
>-  "+third_party/libyuv",
>-  "+common_video",
>-  "+media",
>-  "+p2p",
>-  "+pc",
>-  "+logging/rtc_event_log/rtc_event_log_factory_interface.h",
>-  "+modules/audio_processing/include",
>-  "+system_wrappers/include",
>+  "-audio",
>+  "-base",
>+  "-build",
>+  "-buildtools",
>+  "-build_overrides",
>+  "-call",
>+  "-common_audio",
>+  "-common_video",
>+  "-data",
>+  "-examples",
>+  "-ios",
>+  "-infra",
>+  "-logging",
>+  "-media",
>+  "-modules",
>+  "-out",
>+  "-p2p",
>+  "-pc",
>+  "-resources",
>+  "-rtc_base",
>+  "-rtc_tools",
>+  "-sdk",
>+  "-stats",
>+  "-style-guide",
>+  "-system_wrappers",
>+  "-test",
>+  "-testing",
>+  "-third_party",
>+  "-tools",
>+  "-tools_webrtc",
>+  "-video",
>+  "-external/webrtc/webrtc",  # Android platform build.
>+  "-libyuv",
>+  "-common_types.h",
>+  "-WebRTC",
> ]
> 
> specific_include_rules = {
>+  # Some internal headers are allowed even in API headers:
>+  ".*\.h": [
>+    "+rtc_base/checks.h",
>+    "+rtc_base/system/rtc_export.h",
>+    "+rtc_base/units/unit_base.h",
>+  ],
>+
>+  "array_view\.h": [
>+    "+rtc_base/type_traits.h",
>+  ],
>+
>   # Needed because AudioEncoderOpus is in the wrong place for
>   # backwards compatibilty reasons. See
>   # https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
>@@ -17,13 +57,252 @@ specific_include_rules = {
>     "+modules/audio_coding/codecs/opus/audio_encoder_opus.h",
>   ],
> 
>-  # We allow .cc files in webrtc/api/ to #include a bunch of stuff
>-  # that's off-limits for the .h files. That's because .h files leak
>-  # their #includes to whoever's #including them, but .cc files do not
>-  # since no one #includes them.
>+  "asyncresolverfactory\.h": [
>+    "+rtc_base/asyncresolverinterface.h",
>+  ],
>+
>+  "candidate\.h": [
>+    "+rtc_base/network_constants.h",
>+    "+rtc_base/socketaddress.h",
>+  ],
>+
>+  "create_peerconnection_factory\.h": [
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  "datachannelinterface\.h": [
>+    "+rtc_base/copyonwritebuffer.h",
>+    "+rtc_base/refcount.h",
>+  ],
>+
>+  "dtmfsenderinterface\.h": [
>+    "+rtc_base/refcount.h",
>+  ],
>+
>+  "fec_controller\.h": [
>+    "+modules/include/module_fec_types.h",
>+  ],
>+
>+  "jsep\.h": [
>+    "+rtc_base/refcount.h",
>+  ],
>+
>+  "jsepicecandidate\.h": [
>+    "+rtc_base/constructormagic.h",
>+  ],
>+
>+  "jsepsessiondescription\.h": [
>+    "+rtc_base/constructormagic.h",
>+  ],
>+
>+  "mediastreaminterface\.h": [
>+    "+modules/audio_processing/include/audio_processing_statistics.h",
>+    "+rtc_base/refcount.h",
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  "media_transport_interface\.h": [
>+    "+rtc_base/copyonwritebuffer.h",  # As used by datachannelinterface.h
>+    "+rtc_base/networkroute.h"
>+  ],
>+
>+  "peerconnectionfactoryproxy\.h": [
>+    "+rtc_base/bind.h",
>+  ],
>+
>+  "refcountedbase\.h": [
>+    "+rtc_base/constructormagic.h",
>+    "+rtc_base/refcount.h",
>+    "+rtc_base/refcounter.h",
>+  ],
>+
>+  "rtcerror\.h": [
>+    "+rtc_base/logging.h",
>+  ],
>+
>+  "rtpreceiverinterface\.h": [
>+    "+rtc_base/refcount.h",
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  "rtpsenderinterface\.h": [
>+    "+rtc_base/refcount.h",
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  "rtptransceiverinterface\.h": [
>+    "+rtc_base/refcount.h",
>+  ],
>+
>+  "setremotedescriptionobserverinterface\.h": [
>+    "+rtc_base/refcount.h",
>+  ],
>+
>+  "statstypes\.h": [
>+    "+rtc_base/constructormagic.h",
>+    "+rtc_base/refcount.h",
>+    "+rtc_base/scoped_ref_ptr.h",
>+    "+rtc_base/stringencode.h",
>+    "+rtc_base/thread_checker.h",
>+  ],
>+
>+  "umametrics\.h": [
>+    "+rtc_base/refcount.h",
>+  ],
>+
>+  "audio_frame\.h": [
>+    "+rtc_base/constructormagic.h",
>+  ],
>+
>+  "audio_mixer\.h": [
>+    "+rtc_base/refcount.h",
>+  ],
>+
>+  "audio_decoder\.h": [
>+    "+rtc_base/buffer.h",
>+    "+rtc_base/constructormagic.h",
>+  ],
>+
>+  "audio_decoder_factory\.h": [
>+    "+rtc_base/refcount.h",
>+  ],
>+
>+  "audio_decoder_factory_template\.h": [
>+    "+rtc_base/refcountedobject.h",
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  "audio_encoder\.h": [
>+    "+rtc_base/buffer.h",
>+    "+rtc_base/deprecation.h",
>+  ],
>+
>+  "audio_encoder_factory\.h": [
>+    "+rtc_base/refcount.h",
>+  ],
>+
>+  "audio_encoder_factory_template\.h": [
>+    "+rtc_base/refcountedobject.h",
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  "builtin_audio_decoder_factory\.h": [
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  "builtin_audio_encoder_factory\.h": [
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  "framedecryptorinterface\.h": [
>+    "+rtc_base/refcount.h",
>+  ],
>+
>+  "frameencryptorinterface\.h": [
>+    "+rtc_base/refcount.h",
>+  ],
>+
>+  "ortcfactoryinterface\.h": [
>+    "+rtc_base/network.h",
>+    "+rtc_base/scoped_ref_ptr.h",
>+    "+rtc_base/thread.h",
>+  ],
>+
>+  "udptransportinterface\.h": [
>+    "+rtc_base/socketaddress.h",
>+  ],
>+
>+  "rtcstatscollectorcallback\.h": [
>+    "+rtc_base/refcount.h",
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  "rtcstatsreport\.h": [
>+    "+rtc_base/refcount.h",
>+    "+rtc_base/refcountedobject.h",
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  "audioproc_float\.h": [
>+    "+modules/audio_processing/include/audio_processing.h",
>+  ],
>+
>+  "fake_frame_decryptor\.h": [
>+    "+rtc_base/refcountedobject.h",
>+  ],
>+
>+  "fake_frame_encryptor\.h": [
>+    "+rtc_base/refcountedobject.h",
>+  ],
>+
>+  "fakeconstraints\.h": [
>+    "+rtc_base/stringencode.h",
>+  ],
>+
>+  "mock.*\.h": [
>+    "+test/gmock.h",
>+  ],
>+
>+  "simulated_network\.h": [
>+    "+rtc_base/criticalsection.h",
>+    "+rtc_base/random.h",
>+    "+rtc_base/thread_annotations.h",
>+  ],
>+
>+  "test_dependency_factory\.h": [
>+    "+rtc_base/thread_checker.h",
>+  ],
>+
>+  "videocodec_test_fixture\.h": [
>+    "+modules/video_coding/include/video_codec_interface.h"
>+  ],
>+
>+  "i010_buffer\.h": [
>+    "+rtc_base/memory/aligned_malloc.h"
>+  ],
>+
>+  "i420_buffer\.h": [
>+    "+rtc_base/memory/aligned_malloc.h",
>+  ],
>+
>+  "video_frame_buffer\.h": [
>+    "+rtc_base/refcount.h",
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  "video_timing\.h": [
>+    "+rtc_base/numerics/safe_conversions.h",
>+  ],
>+
>+  "video_encoder_config\.h": [
>+    "+rtc_base/refcount.h",
>+    "+rtc_base/scoped_ref_ptr.h",
>+  ],
>+
>+  # .cc files in api/ should not be restricted in what they can #include,
>+  # so we re-add all the top-level directories here. (That's because .h
>+  # files leak their #includes to whoever's #including them, but .cc files
>+  # do not since no one #includes them.)
>   ".*\.cc": [
>-    "+modules/audio_coding",
>-    "+modules/audio_processing",
>-    "+modules/video_coding",
>+    "+audio",
>+    "+call",
>+    "+common_audio",
>+    "+common_video",
>+    "+examples",
>+    "+logging",
>+    "+media",
>+    "+modules",
>+    "+p2p",
>+    "+pc",
>+    "+rtc_base",
>+    "+rtc_tools",
>+    "+sdk",
>+    "+stats",
>+    "+system_wrappers",
>+    "+test",
>+    "+tools",
>+    "+tools_webrtc",
>+    "+video",
>+    "+third_party",
>   ],
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/api/OWNERS
>index 593de230d37879a2598f5859df2d432aacfb32b2..1ae5c646e2317abae35464ba83fa3ffe0be12a0b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/OWNERS
>@@ -13,3 +13,10 @@ per-file peerconnection*=hbos@webrtc.org
> 
> per-file *.gn=phoglund@webrtc.org
> per-file *.gni=phoglund@webrtc.org
>+
>+per-file DEPS=mbonadei@webrtc.org
>+per-file DEPS=kwiberg@webrtc.org
>+
>+per-file media_transport*=sukhanov@webrtc.org
>+per-file media_transport*=psla@webrtc.org
>+per-file media_transport*=mellem@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/BUILD.gn
>index 286a5a6f1c6cf523f420fe4ff7c927b439bc5015..b66be763431f72b33f000f541dcc0c1da18389af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/BUILD.gn
>@@ -39,6 +39,27 @@ rtc_source_set("aec3_config") {
>     "echo_canceller3_config.cc",
>     "echo_canceller3_config.h",
>   ]
>+  deps = [
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:safe_minmax",
>+    "../../rtc_base/system:rtc_export",
>+  ]
>+}
>+
>+rtc_source_set("aec3_config_json") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "echo_canceller3_config_json.cc",
>+    "echo_canceller3_config_json.h",
>+  ]
>+  deps = [
>+    ":aec3_config",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:rtc_json",
>+    "../../rtc_base/system:rtc_export",
>+    "//third_party/abseil-cpp/absl/strings:strings",
>+  ]
> }
> 
> rtc_source_set("aec3_factory") {
>@@ -54,6 +75,7 @@ rtc_source_set("aec3_factory") {
>     ":echo_control",
>     "../../modules/audio_processing/aec3",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.cc
>index 3b03d13c1ca51d913e9ee8429859a5e31529c110..3eb2a8d2a009b8fe1a4ef870e76dce4709189274 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.cc
>@@ -9,7 +9,36 @@
>  */
> #include "api/audio/echo_canceller3_config.h"
> 
>+#include <algorithm>
>+#include <cmath>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+
> namespace webrtc {
>+namespace {
>+bool Limit(float* value, float min, float max) {
>+  float clamped = rtc::SafeClamp(*value, min, max);
>+  clamped = std::isfinite(clamped) ? clamped : min;
>+  bool res = *value == clamped;
>+  *value = clamped;
>+  return res;
>+}
>+
>+bool Limit(size_t* value, size_t min, size_t max) {
>+  size_t clamped = rtc::SafeClamp(*value, min, max);
>+  bool res = *value == clamped;
>+  *value = clamped;
>+  return res;
>+}
>+
>+bool Limit(int* value, int min, int max) {
>+  int clamped = rtc::SafeClamp(*value, min, max);
>+  bool res = *value == clamped;
>+  *value = clamped;
>+  return res;
>+}
>+}  // namespace
> 
> EchoCanceller3Config::EchoCanceller3Config() = default;
> EchoCanceller3Config::EchoCanceller3Config(const EchoCanceller3Config& e) =
>@@ -18,9 +47,6 @@ EchoCanceller3Config::Delay::Delay() = default;
> EchoCanceller3Config::Delay::Delay(const EchoCanceller3Config::Delay& e) =
>     default;
> 
>-EchoCanceller3Config::Mask::Mask() = default;
>-EchoCanceller3Config::Mask::Mask(const EchoCanceller3Config::Mask& m) = default;
>-
> EchoCanceller3Config::EchoModel::EchoModel() = default;
> EchoCanceller3Config::EchoModel::EchoModel(
>     const EchoCanceller3Config::EchoModel& e) = default;
>@@ -51,4 +77,179 @@ EchoCanceller3Config::Suppressor::Tuning::Tuning(MaskingThresholds mask_lf,
> EchoCanceller3Config::Suppressor::Tuning::Tuning(
>     const EchoCanceller3Config::Suppressor::Tuning& e) = default;
> 
>+bool EchoCanceller3Config::Validate(EchoCanceller3Config* config) {
>+  RTC_DCHECK(config);
>+  EchoCanceller3Config* c = config;
>+  bool res = true;
>+
>+  if (c->delay.down_sampling_factor != 4 &&
>+      c->delay.down_sampling_factor != 8) {
>+    c->delay.down_sampling_factor = 4;
>+    res = false;
>+  }
>+  if (c->delay.delay_headroom_blocks <= 1 &&
>+      c->delay.hysteresis_limit_1_blocks == 1) {
>+    c->delay.hysteresis_limit_1_blocks = 0;
>+    res = false;
>+  }
>+  res = res & Limit(&c->delay.default_delay, 0, 5000);
>+  res = res & Limit(&c->delay.num_filters, 0, 5000);
>+  res = res & Limit(&c->delay.api_call_jitter_blocks, 1, 5000);
>+  res = res & Limit(&c->delay.min_echo_path_delay_blocks, 0, 5000);
>+  res = res & Limit(&c->delay.delay_headroom_blocks, 0, 5000);
>+  res = res & Limit(&c->delay.hysteresis_limit_1_blocks, 0, 5000);
>+  res = res & Limit(&c->delay.hysteresis_limit_2_blocks, 0, 5000);
>+  res = res & Limit(&c->delay.skew_hysteresis_blocks, 0, 5000);
>+  res = res & Limit(&c->delay.fixed_capture_delay_samples, 0, 5000);
>+  res = res & Limit(&c->delay.delay_estimate_smoothing, 0.f, 1.f);
>+  res = res & Limit(&c->delay.delay_candidate_detection_threshold, 0.f, 1.f);
>+  res = res & Limit(&c->delay.delay_selection_thresholds.initial, 1, 250);
>+  res = res & Limit(&c->delay.delay_selection_thresholds.converged, 1, 250);
>+
>+  res = res & Limit(&c->filter.main.length_blocks, 1, 50);
>+  res = res & Limit(&c->filter.main.leakage_converged, 0.f, 1000.f);
>+  res = res & Limit(&c->filter.main.leakage_diverged, 0.f, 1000.f);
>+  res = res & Limit(&c->filter.main.error_floor, 0.f, 1000.f);
>+  res = res & Limit(&c->filter.main.error_ceil, 0.f, 100000000.f);
>+  res = res & Limit(&c->filter.main.noise_gate, 0.f, 100000000.f);
>+
>+  res = res & Limit(&c->filter.main_initial.length_blocks, 1, 50);
>+  res = res & Limit(&c->filter.main_initial.leakage_converged, 0.f, 1000.f);
>+  res = res & Limit(&c->filter.main_initial.leakage_diverged, 0.f, 1000.f);
>+  res = res & Limit(&c->filter.main_initial.error_floor, 0.f, 1000.f);
>+  res = res & Limit(&c->filter.main_initial.error_ceil, 0.f, 100000000.f);
>+  res = res & Limit(&c->filter.main_initial.noise_gate, 0.f, 100000000.f);
>+
>+  if (c->filter.main.length_blocks < c->filter.main_initial.length_blocks) {
>+    c->filter.main_initial.length_blocks = c->filter.main.length_blocks;
>+    res = false;
>+  }
>+
>+  res = res & Limit(&c->filter.shadow.length_blocks, 1, 50);
>+  res = res & Limit(&c->filter.shadow.rate, 0.f, 1.f);
>+  res = res & Limit(&c->filter.shadow.noise_gate, 0.f, 100000000.f);
>+
>+  res = res & Limit(&c->filter.shadow_initial.length_blocks, 1, 50);
>+  res = res & Limit(&c->filter.shadow_initial.rate, 0.f, 1.f);
>+  res = res & Limit(&c->filter.shadow_initial.noise_gate, 0.f, 100000000.f);
>+
>+  if (c->filter.shadow.length_blocks < c->filter.shadow_initial.length_blocks) {
>+    c->filter.shadow_initial.length_blocks = c->filter.shadow.length_blocks;
>+    res = false;
>+  }
>+
>+  res = res & Limit(&c->filter.config_change_duration_blocks, 0, 100000);
>+  res = res & Limit(&c->filter.initial_state_seconds, 0.f, 100.f);
>+
>+  res = res & Limit(&c->erle.min, 1.f, 100000.f);
>+  res = res & Limit(&c->erle.max_l, 1.f, 100000.f);
>+  res = res & Limit(&c->erle.max_h, 1.f, 100000.f);
>+  if (c->erle.min > c->erle.max_l || c->erle.min > c->erle.max_h) {
>+    c->erle.min = std::min(c->erle.max_l, c->erle.max_h);
>+    res = false;
>+  }
>+  res = res & Limit(&c->erle.num_sections, 1, c->filter.main.length_blocks);
>+
>+  res = res & Limit(&c->ep_strength.lf, 0.f, 1000000.f);
>+  res = res & Limit(&c->ep_strength.mf, 0.f, 1000000.f);
>+  res = res & Limit(&c->ep_strength.hf, 0.f, 1000000.f);
>+  res = res & Limit(&c->ep_strength.default_len, -1.f, 1.f);
>+
>+  res =
>+      res & Limit(&c->echo_audibility.low_render_limit, 0.f, 32768.f * 32768.f);
>+  res = res &
>+        Limit(&c->echo_audibility.normal_render_limit, 0.f, 32768.f * 32768.f);
>+  res = res & Limit(&c->echo_audibility.floor_power, 0.f, 32768.f * 32768.f);
>+  res = res & Limit(&c->echo_audibility.audibility_threshold_lf, 0.f,
>+                    32768.f * 32768.f);
>+  res = res & Limit(&c->echo_audibility.audibility_threshold_mf, 0.f,
>+                    32768.f * 32768.f);
>+  res = res & Limit(&c->echo_audibility.audibility_threshold_hf, 0.f,
>+                    32768.f * 32768.f);
>+
>+  res = res &
>+        Limit(&c->render_levels.active_render_limit, 0.f, 32768.f * 32768.f);
>+  res = res & Limit(&c->render_levels.poor_excitation_render_limit, 0.f,
>+                    32768.f * 32768.f);
>+  res = res & Limit(&c->render_levels.poor_excitation_render_limit_ds8, 0.f,
>+                    32768.f * 32768.f);
>+
>+  res =
>+      res & Limit(&c->echo_removal_control.gain_rampup.initial_gain, 0.f, 1.f);
>+  res = res & Limit(&c->echo_removal_control.gain_rampup.first_non_zero_gain,
>+                    0.f, 1.f);
>+  res = res & Limit(&c->echo_removal_control.gain_rampup.non_zero_gain_blocks,
>+                    0, 100000);
>+  res = res &
>+        Limit(&c->echo_removal_control.gain_rampup.full_gain_blocks, 0, 100000);
>+
>+  res = res & Limit(&c->echo_model.noise_floor_hold, 0, 1000);
>+  res = res & Limit(&c->echo_model.min_noise_floor_power, 0, 2000000.f);
>+  res = res & Limit(&c->echo_model.stationary_gate_slope, 0, 1000000.f);
>+  res = res & Limit(&c->echo_model.noise_gate_power, 0, 1000000.f);
>+  res = res & Limit(&c->echo_model.noise_gate_slope, 0, 1000000.f);
>+  res = res & Limit(&c->echo_model.render_pre_window_size, 0, 100);
>+  res = res & Limit(&c->echo_model.render_post_window_size, 0, 100);
>+  res = res & Limit(&c->echo_model.render_pre_window_size_init, 0, 100);
>+  res = res & Limit(&c->echo_model.render_post_window_size_init, 0, 100);
>+  res = res & Limit(&c->echo_model.nonlinear_hold, 0, 100);
>+  res = res & Limit(&c->echo_model.nonlinear_release, 0, 1.f);
>+
>+  res = res & Limit(&c->suppressor.nearend_average_blocks, 1, 5000);
>+
>+  res = res &
>+        Limit(&c->suppressor.normal_tuning.mask_lf.enr_transparent, 0.f, 100.f);
>+  res = res &
>+        Limit(&c->suppressor.normal_tuning.mask_lf.enr_suppress, 0.f, 100.f);
>+  res = res &
>+        Limit(&c->suppressor.normal_tuning.mask_lf.emr_transparent, 0.f, 100.f);
>+  res = res &
>+        Limit(&c->suppressor.normal_tuning.mask_hf.enr_transparent, 0.f, 100.f);
>+  res = res &
>+        Limit(&c->suppressor.normal_tuning.mask_hf.enr_suppress, 0.f, 100.f);
>+  res = res &
>+        Limit(&c->suppressor.normal_tuning.mask_hf.emr_transparent, 0.f, 100.f);
>+  res = res & Limit(&c->suppressor.normal_tuning.max_inc_factor, 0.f, 100.f);
>+  res = res & Limit(&c->suppressor.normal_tuning.max_dec_factor_lf, 0.f, 100.f);
>+
>+  res = res & Limit(&c->suppressor.nearend_tuning.mask_lf.enr_transparent, 0.f,
>+                    100.f);
>+  res = res &
>+        Limit(&c->suppressor.nearend_tuning.mask_lf.enr_suppress, 0.f, 100.f);
>+  res = res & Limit(&c->suppressor.nearend_tuning.mask_lf.emr_transparent, 0.f,
>+                    100.f);
>+  res = res & Limit(&c->suppressor.nearend_tuning.mask_hf.enr_transparent, 0.f,
>+                    100.f);
>+  res = res &
>+        Limit(&c->suppressor.nearend_tuning.mask_hf.enr_suppress, 0.f, 100.f);
>+  res = res & Limit(&c->suppressor.nearend_tuning.mask_hf.emr_transparent, 0.f,
>+                    100.f);
>+  res = res & Limit(&c->suppressor.nearend_tuning.max_inc_factor, 0.f, 100.f);
>+  res =
>+      res & Limit(&c->suppressor.nearend_tuning.max_dec_factor_lf, 0.f, 100.f);
>+
>+  res = res & Limit(&c->suppressor.dominant_nearend_detection.enr_threshold,
>+                    0.f, 1000000.f);
>+  res = res & Limit(&c->suppressor.dominant_nearend_detection.snr_threshold,
>+                    0.f, 1000000.f);
>+  res = res & Limit(&c->suppressor.dominant_nearend_detection.hold_duration, 0,
>+                    10000);
>+  res = res & Limit(&c->suppressor.dominant_nearend_detection.trigger_threshold,
>+                    0, 10000);
>+
>+  res = res & Limit(&c->suppressor.high_bands_suppression.enr_threshold, 0.f,
>+                    1000000.f);
>+  res = res & Limit(&c->suppressor.high_bands_suppression.max_gain_during_echo,
>+                    0.f, 1.f);
>+
>+  res = res & Limit(&c->suppressor.floor_first_increase, 0.f, 1000000.f);
>+
>+  if (c->delay.delay_headroom_blocks >
>+      c->filter.main_initial.length_blocks - 1) {
>+    c->delay.delay_headroom_blocks = c->filter.main_initial.length_blocks - 1;
>+    res = false;
>+  }
>+
>+  return res;
>+}
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.h
>index e7376ed7bcd3c4eb9c826c4344172051ad6ebdd5..ea6e51baf9c53d1236f0759556b26e2d921ac769 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.h
>@@ -13,18 +13,31 @@
> 
> #include <stddef.h>  // size_t
> 
>+#include "rtc_base/system/rtc_export.h"
>+
> namespace webrtc {
> 
> // Configuration struct for EchoCanceller3
>-struct EchoCanceller3Config {
>+struct RTC_EXPORT EchoCanceller3Config {
>+  // Checks and updates the config parameters to lie within (mostly) reasonable
>+  // ranges. Returns true if and only of the config did not need to be changed.
>+  static bool Validate(EchoCanceller3Config* config);
>+
>   EchoCanceller3Config();
>   EchoCanceller3Config(const EchoCanceller3Config& e);
>+
>+  struct Buffering {
>+    bool use_new_render_buffering = true;
>+    size_t excess_render_detection_interval_blocks = 250;
>+    size_t max_allowed_excess_render_blocks = 8;
>+  } buffering;
>+
>   struct Delay {
>     Delay();
>     Delay(const Delay& e);
>     size_t default_delay = 5;
>     size_t down_sampling_factor = 4;
>-    size_t num_filters = 6;
>+    size_t num_filters = 5;
>     size_t api_call_jitter_blocks = 26;
>     size_t min_echo_path_delay_blocks = 0;
>     size_t delay_headroom_blocks = 2;
>@@ -37,7 +50,7 @@ struct EchoCanceller3Config {
>     struct DelaySelectionThresholds {
>       int initial;
>       int converged;
>-    } delay_selection_thresholds = {25, 25};
>+    } delay_selection_thresholds = {5, 20};
>   } delay;
> 
>   struct Filter {
>@@ -46,6 +59,7 @@ struct EchoCanceller3Config {
>       float leakage_converged;
>       float leakage_diverged;
>       float error_floor;
>+      float error_ceil;
>       float noise_gate;
>     };
> 
>@@ -55,10 +69,11 @@ struct EchoCanceller3Config {
>       float noise_gate;
>     };
> 
>-    MainConfiguration main = {13, 0.00005f, 0.01f, 0.1f, 20075344.f};
>+    MainConfiguration main = {13, 0.00005f, 0.05f, 0.001f, 2.f, 20075344.f};
>     ShadowConfiguration shadow = {13, 0.7f, 20075344.f};
> 
>-    MainConfiguration main_initial = {12, 0.005f, 0.5f, 0.001f, 20075344.f};
>+    MainConfiguration main_initial = {12,     0.005f, 0.5f,
>+                                      0.001f, 2.f,    20075344.f};
>     ShadowConfiguration shadow_initial = {12, 0.9f, 20075344.f};
> 
>     size_t config_change_duration_blocks = 250;
>@@ -72,38 +87,19 @@ struct EchoCanceller3Config {
>     float max_l = 4.f;
>     float max_h = 1.5f;
>     bool onset_detection = true;
>+    size_t num_sections = 1;
>   } erle;
> 
>   struct EpStrength {
>     float lf = 1.f;
>     float mf = 1.f;
>     float hf = 1.f;
>-    float default_len = 0.88f;
>+    float default_len = 0.83f;
>     bool reverb_based_on_render = true;
>     bool echo_can_saturate = true;
>     bool bounded_erl = false;
>   } ep_strength;
> 
>-  struct Mask {
>-    Mask();
>-    Mask(const Mask& m);
>-    float m0 = 0.1f;
>-    float m1 = 0.01f;
>-    float m2 = 0.0001f;
>-    float m3 = 0.01f;
>-    float m5 = 0.01f;
>-    float m6 = 0.0001f;
>-    float m7 = 0.01f;
>-    float m8 = 0.0001f;
>-    float m9 = 0.1f;
>-
>-    float gain_curve_offset = 1.45f;
>-    float gain_curve_slope = 5.f;
>-    float temporal_masking_lf = 0.9f;
>-    float temporal_masking_hf = 0.6f;
>-    size_t temporal_masking_lf_bands = 3;
>-  } gain_mask;
>-
>   struct EchoAudibility {
>     float low_render_limit = 4 * 64.f;
>     float normal_render_limit = 64.f;
>@@ -111,7 +107,7 @@ struct EchoCanceller3Config {
>     float audibility_threshold_lf = 10;
>     float audibility_threshold_mf = 10;
>     float audibility_threshold_hf = 10;
>-    bool use_stationary_properties = true;
>+    bool use_stationary_properties = false;
>     bool use_stationarity_properties_at_init = false;
>   } echo_audibility;
> 
>@@ -176,20 +172,22 @@ struct EchoCanceller3Config {
>       float max_dec_factor_lf;
>     };
> 
>-    Tuning normal_tuning = Tuning(MaskingThresholds(.2f, .3f, .3f),
>+    Tuning normal_tuning = Tuning(MaskingThresholds(.3f, .4f, .3f),
>                                   MaskingThresholds(.07f, .1f, .3f),
>                                   2.0f,
>                                   0.25f);
>-    Tuning nearend_tuning = Tuning(MaskingThresholds(.2f, .3f, .3f),
>-                                   MaskingThresholds(.07f, .1f, .3f),
>+    Tuning nearend_tuning = Tuning(MaskingThresholds(1.09f, 1.1f, .3f),
>+                                   MaskingThresholds(.1f, .3f, .3f),
>                                    2.0f,
>                                    0.25f);
> 
>     struct DominantNearendDetection {
>-      float enr_threshold = 10.f;
>-      float snr_threshold = 10.f;
>-      int hold_duration = 25;
>-      int trigger_threshold = 15;
>+      float enr_threshold = 4.f;
>+      float enr_exit_threshold = .1f;
>+      float snr_threshold = 30.f;
>+      int hold_duration = 50;
>+      int trigger_threshold = 12;
>+      bool use_during_initial_phase = true;
>     } dominant_nearend_detection;
> 
>     struct HighBandsSuppression {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config_json.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config_json.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..01a831cb4c996082cca08255df5db84effb1edf2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config_json.cc
>@@ -0,0 +1,583 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "api/audio/echo_canceller3_config_json.h"
>+
>+#include <string>
>+#include <vector>
>+
>+#include "rtc_base/logging.h"
>+#include "rtc_base/strings/json.h"
>+#include "rtc_base/strings/string_builder.h"
>+
>+namespace webrtc {
>+namespace {
>+void ReadParam(const Json::Value& root, std::string param_name, bool* param) {
>+  RTC_DCHECK(param);
>+  bool v;
>+  if (rtc::GetBoolFromJsonObject(root, param_name, &v)) {
>+    *param = v;
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root, std::string param_name, size_t* param) {
>+  RTC_DCHECK(param);
>+  int v;
>+  if (rtc::GetIntFromJsonObject(root, param_name, &v)) {
>+    RTC_DCHECK_GE(v, 0);
>+    *param = v;
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root, std::string param_name, int* param) {
>+  RTC_DCHECK(param);
>+  int v;
>+  if (rtc::GetIntFromJsonObject(root, param_name, &v)) {
>+    *param = v;
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root, std::string param_name, float* param) {
>+  RTC_DCHECK(param);
>+  double v;
>+  if (rtc::GetDoubleFromJsonObject(root, param_name, &v)) {
>+    *param = static_cast<float>(v);
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root,
>+               std::string param_name,
>+               EchoCanceller3Config::Filter::MainConfiguration* param) {
>+  RTC_DCHECK(param);
>+  Json::Value json_array;
>+  if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
>+    std::vector<double> v;
>+    rtc::JsonArrayToDoubleVector(json_array, &v);
>+    if (v.size() != 6) {
>+      RTC_LOG(LS_ERROR) << "Incorrect array size for " << param_name;
>+      return;
>+    }
>+    param->length_blocks = static_cast<size_t>(v[0]);
>+    param->leakage_converged = static_cast<float>(v[1]);
>+    param->leakage_diverged = static_cast<float>(v[2]);
>+    param->error_floor = static_cast<float>(v[3]);
>+    param->error_ceil = static_cast<float>(v[4]);
>+    param->noise_gate = static_cast<float>(v[5]);
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root,
>+               std::string param_name,
>+               EchoCanceller3Config::Filter::ShadowConfiguration* param) {
>+  RTC_DCHECK(param);
>+  Json::Value json_array;
>+  if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
>+    std::vector<double> v;
>+    rtc::JsonArrayToDoubleVector(json_array, &v);
>+    if (v.size() != 3) {
>+      RTC_LOG(LS_ERROR) << "Incorrect array size for " << param_name;
>+      return;
>+    }
>+    param->length_blocks = static_cast<size_t>(v[0]);
>+    param->rate = static_cast<float>(v[1]);
>+    param->noise_gate = static_cast<float>(v[2]);
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root,
>+               std::string param_name,
>+               EchoCanceller3Config::Suppressor::MaskingThresholds* param) {
>+  RTC_DCHECK(param);
>+  Json::Value json_array;
>+  if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
>+    std::vector<double> v;
>+    rtc::JsonArrayToDoubleVector(json_array, &v);
>+    if (v.size() != 3) {
>+      RTC_LOG(LS_ERROR) << "Incorrect array size for " << param_name;
>+      return;
>+    }
>+    param->enr_transparent = static_cast<float>(v[0]);
>+    param->enr_suppress = static_cast<float>(v[1]);
>+    param->emr_transparent = static_cast<float>(v[2]);
>+  }
>+}
>+}  // namespace
>+
>+void Aec3ConfigFromJsonString(absl::string_view json_string,
>+                              EchoCanceller3Config* config,
>+                              bool* parsing_successful) {
>+  RTC_DCHECK(config);
>+  RTC_DCHECK(parsing_successful);
>+  EchoCanceller3Config& cfg = *config;
>+  cfg = EchoCanceller3Config();
>+  *parsing_successful = true;
>+
>+  Json::Value root;
>+  bool success = Json::Reader().parse(std::string(json_string), root);
>+  if (!success) {
>+    RTC_LOG(LS_ERROR) << "Incorrect JSON format: " << json_string;
>+    *parsing_successful = false;
>+    return;
>+  }
>+
>+  Json::Value aec3_root;
>+  success = rtc::GetValueFromJsonObject(root, "aec3", &aec3_root);
>+  if (!success) {
>+    RTC_LOG(LS_ERROR) << "Missing AEC3 config field: " << json_string;
>+    *parsing_successful = false;
>+    return;
>+  }
>+
>+  Json::Value section;
>+  if (rtc::GetValueFromJsonObject(root, "buffering", &section)) {
>+    ReadParam(section, "use_new_render_buffering",
>+              &cfg.buffering.use_new_render_buffering);
>+    ReadParam(section, "excess_render_detection_interval_blocks",
>+              &cfg.buffering.excess_render_detection_interval_blocks);
>+    ReadParam(section, "max_allowed_excess_render_blocks",
>+              &cfg.buffering.max_allowed_excess_render_blocks);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(aec3_root, "delay", &section)) {
>+    ReadParam(section, "default_delay", &cfg.delay.default_delay);
>+    ReadParam(section, "down_sampling_factor", &cfg.delay.down_sampling_factor);
>+    ReadParam(section, "num_filters", &cfg.delay.num_filters);
>+    ReadParam(section, "api_call_jitter_blocks",
>+              &cfg.delay.api_call_jitter_blocks);
>+    ReadParam(section, "min_echo_path_delay_blocks",
>+              &cfg.delay.min_echo_path_delay_blocks);
>+    ReadParam(section, "delay_headroom_blocks",
>+              &cfg.delay.delay_headroom_blocks);
>+    ReadParam(section, "hysteresis_limit_1_blocks",
>+              &cfg.delay.hysteresis_limit_1_blocks);
>+    ReadParam(section, "hysteresis_limit_2_blocks",
>+              &cfg.delay.hysteresis_limit_2_blocks);
>+    ReadParam(section, "skew_hysteresis_blocks",
>+              &cfg.delay.skew_hysteresis_blocks);
>+    ReadParam(section, "fixed_capture_delay_samples",
>+              &cfg.delay.fixed_capture_delay_samples);
>+    ReadParam(section, "delay_estimate_smoothing",
>+              &cfg.delay.delay_estimate_smoothing);
>+    ReadParam(section, "delay_candidate_detection_threshold",
>+              &cfg.delay.delay_candidate_detection_threshold);
>+
>+    Json::Value subsection;
>+    if (rtc::GetValueFromJsonObject(section, "delay_selection_thresholds",
>+                                    &subsection)) {
>+      ReadParam(subsection, "initial",
>+                &cfg.delay.delay_selection_thresholds.initial);
>+      ReadParam(subsection, "converged",
>+                &cfg.delay.delay_selection_thresholds.converged);
>+    }
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(aec3_root, "filter", &section)) {
>+    ReadParam(section, "main", &cfg.filter.main);
>+    ReadParam(section, "shadow", &cfg.filter.shadow);
>+    ReadParam(section, "main_initial", &cfg.filter.main_initial);
>+    ReadParam(section, "shadow_initial", &cfg.filter.shadow_initial);
>+    ReadParam(section, "config_change_duration_blocks",
>+              &cfg.filter.config_change_duration_blocks);
>+    ReadParam(section, "initial_state_seconds",
>+              &cfg.filter.initial_state_seconds);
>+    ReadParam(section, "conservative_initial_phase",
>+              &cfg.filter.conservative_initial_phase);
>+    ReadParam(section, "enable_shadow_filter_output_usage",
>+              &cfg.filter.enable_shadow_filter_output_usage);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(aec3_root, "erle", &section)) {
>+    ReadParam(section, "min", &cfg.erle.min);
>+    ReadParam(section, "max_l", &cfg.erle.max_l);
>+    ReadParam(section, "max_h", &cfg.erle.max_h);
>+    ReadParam(section, "onset_detection", &cfg.erle.onset_detection);
>+    ReadParam(section, "num_sections", &cfg.erle.num_sections);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(aec3_root, "ep_strength", &section)) {
>+    ReadParam(section, "lf", &cfg.ep_strength.lf);
>+    ReadParam(section, "mf", &cfg.ep_strength.mf);
>+    ReadParam(section, "hf", &cfg.ep_strength.hf);
>+    ReadParam(section, "default_len", &cfg.ep_strength.default_len);
>+    ReadParam(section, "reverb_based_on_render",
>+              &cfg.ep_strength.reverb_based_on_render);
>+    ReadParam(section, "echo_can_saturate", &cfg.ep_strength.echo_can_saturate);
>+    ReadParam(section, "bounded_erl", &cfg.ep_strength.bounded_erl);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(aec3_root, "echo_audibility", &section)) {
>+    ReadParam(section, "low_render_limit",
>+              &cfg.echo_audibility.low_render_limit);
>+    ReadParam(section, "normal_render_limit",
>+              &cfg.echo_audibility.normal_render_limit);
>+
>+    ReadParam(section, "floor_power", &cfg.echo_audibility.floor_power);
>+    ReadParam(section, "audibility_threshold_lf",
>+              &cfg.echo_audibility.audibility_threshold_lf);
>+    ReadParam(section, "audibility_threshold_mf",
>+              &cfg.echo_audibility.audibility_threshold_mf);
>+    ReadParam(section, "audibility_threshold_hf",
>+              &cfg.echo_audibility.audibility_threshold_hf);
>+    ReadParam(section, "use_stationary_properties",
>+              &cfg.echo_audibility.use_stationary_properties);
>+    ReadParam(section, "use_stationarity_properties_at_init",
>+              &cfg.echo_audibility.use_stationarity_properties_at_init);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(aec3_root, "render_levels", &section)) {
>+    ReadParam(section, "active_render_limit",
>+              &cfg.render_levels.active_render_limit);
>+    ReadParam(section, "poor_excitation_render_limit",
>+              &cfg.render_levels.poor_excitation_render_limit);
>+    ReadParam(section, "poor_excitation_render_limit_ds8",
>+              &cfg.render_levels.poor_excitation_render_limit_ds8);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(aec3_root, "echo_removal_control",
>+                                  &section)) {
>+    Json::Value subsection;
>+    if (rtc::GetValueFromJsonObject(section, "gain_rampup", &subsection)) {
>+      ReadParam(subsection, "initial_gain",
>+                &cfg.echo_removal_control.gain_rampup.initial_gain);
>+      ReadParam(subsection, "first_non_zero_gain",
>+                &cfg.echo_removal_control.gain_rampup.first_non_zero_gain);
>+      ReadParam(subsection, "non_zero_gain_blocks",
>+                &cfg.echo_removal_control.gain_rampup.non_zero_gain_blocks);
>+      ReadParam(subsection, "full_gain_blocks",
>+                &cfg.echo_removal_control.gain_rampup.full_gain_blocks);
>+    }
>+    ReadParam(section, "has_clock_drift",
>+              &cfg.echo_removal_control.has_clock_drift);
>+    ReadParam(section, "linear_and_stable_echo_path",
>+              &cfg.echo_removal_control.linear_and_stable_echo_path);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(aec3_root, "echo_model", &section)) {
>+    Json::Value subsection;
>+    ReadParam(section, "noise_floor_hold", &cfg.echo_model.noise_floor_hold);
>+    ReadParam(section, "min_noise_floor_power",
>+              &cfg.echo_model.min_noise_floor_power);
>+    ReadParam(section, "stationary_gate_slope",
>+              &cfg.echo_model.stationary_gate_slope);
>+    ReadParam(section, "noise_gate_power", &cfg.echo_model.noise_gate_power);
>+    ReadParam(section, "noise_gate_slope", &cfg.echo_model.noise_gate_slope);
>+    ReadParam(section, "render_pre_window_size",
>+              &cfg.echo_model.render_pre_window_size);
>+    ReadParam(section, "render_post_window_size",
>+              &cfg.echo_model.render_post_window_size);
>+    ReadParam(section, "render_pre_window_size_init",
>+              &cfg.echo_model.render_pre_window_size_init);
>+    ReadParam(section, "render_post_window_size_init",
>+              &cfg.echo_model.render_post_window_size_init);
>+    ReadParam(section, "nonlinear_hold", &cfg.echo_model.nonlinear_hold);
>+    ReadParam(section, "nonlinear_release", &cfg.echo_model.nonlinear_release);
>+  }
>+
>+  Json::Value subsection;
>+  if (rtc::GetValueFromJsonObject(aec3_root, "suppressor", &section)) {
>+    ReadParam(section, "nearend_average_blocks",
>+              &cfg.suppressor.nearend_average_blocks);
>+
>+    if (rtc::GetValueFromJsonObject(section, "normal_tuning", &subsection)) {
>+      ReadParam(subsection, "mask_lf", &cfg.suppressor.normal_tuning.mask_lf);
>+      ReadParam(subsection, "mask_hf", &cfg.suppressor.normal_tuning.mask_hf);
>+      ReadParam(subsection, "max_inc_factor",
>+                &cfg.suppressor.normal_tuning.max_inc_factor);
>+      ReadParam(subsection, "max_dec_factor_lf",
>+                &cfg.suppressor.normal_tuning.max_dec_factor_lf);
>+    }
>+
>+    if (rtc::GetValueFromJsonObject(section, "nearend_tuning", &subsection)) {
>+      ReadParam(subsection, "mask_lf", &cfg.suppressor.nearend_tuning.mask_lf);
>+      ReadParam(subsection, "mask_hf", &cfg.suppressor.nearend_tuning.mask_hf);
>+      ReadParam(subsection, "max_inc_factor",
>+                &cfg.suppressor.nearend_tuning.max_inc_factor);
>+      ReadParam(subsection, "max_dec_factor_lf",
>+                &cfg.suppressor.nearend_tuning.max_dec_factor_lf);
>+    }
>+
>+    if (rtc::GetValueFromJsonObject(section, "dominant_nearend_detection",
>+                                    &subsection)) {
>+      ReadParam(subsection, "enr_threshold",
>+                &cfg.suppressor.dominant_nearend_detection.enr_threshold);
>+      ReadParam(subsection, "enr_exit_threshold",
>+                &cfg.suppressor.dominant_nearend_detection.enr_exit_threshold);
>+      ReadParam(subsection, "snr_threshold",
>+                &cfg.suppressor.dominant_nearend_detection.snr_threshold);
>+      ReadParam(subsection, "hold_duration",
>+                &cfg.suppressor.dominant_nearend_detection.hold_duration);
>+      ReadParam(subsection, "trigger_threshold",
>+                &cfg.suppressor.dominant_nearend_detection.trigger_threshold);
>+      ReadParam(
>+          subsection, "use_during_initial_phase",
>+          &cfg.suppressor.dominant_nearend_detection.use_during_initial_phase);
>+    }
>+
>+    if (rtc::GetValueFromJsonObject(section, "high_bands_suppression",
>+                                    &subsection)) {
>+      ReadParam(subsection, "enr_threshold",
>+                &cfg.suppressor.high_bands_suppression.enr_threshold);
>+      ReadParam(subsection, "max_gain_during_echo",
>+                &cfg.suppressor.high_bands_suppression.max_gain_during_echo);
>+    }
>+
>+    ReadParam(section, "floor_first_increase",
>+              &cfg.suppressor.floor_first_increase);
>+    ReadParam(section, "enforce_transparent",
>+              &cfg.suppressor.enforce_transparent);
>+    ReadParam(section, "enforce_empty_higher_bands",
>+              &cfg.suppressor.enforce_empty_higher_bands);
>+  }
>+}
>+
>+EchoCanceller3Config Aec3ConfigFromJsonString(absl::string_view json_string) {
>+  EchoCanceller3Config cfg;
>+  bool not_used;
>+  Aec3ConfigFromJsonString(json_string, &cfg, &not_used);
>+  return cfg;
>+}
>+
>+std::string Aec3ConfigToJsonString(const EchoCanceller3Config& config) {
>+  rtc::StringBuilder ost;
>+  ost << "{";
>+  ost << "\"aec3\": {";
>+  ost << "\"delay\": {";
>+  ost << "\"default_delay\": " << config.delay.default_delay << ",";
>+  ost << "\"down_sampling_factor\": " << config.delay.down_sampling_factor
>+      << ",";
>+  ost << "\"num_filters\": " << config.delay.num_filters << ",";
>+  ost << "\"api_call_jitter_blocks\": " << config.delay.api_call_jitter_blocks
>+      << ",";
>+  ost << "\"min_echo_path_delay_blocks\": "
>+      << config.delay.min_echo_path_delay_blocks << ",";
>+  ost << "\"delay_headroom_blocks\": " << config.delay.delay_headroom_blocks
>+      << ",";
>+  ost << "\"hysteresis_limit_1_blocks\": "
>+      << config.delay.hysteresis_limit_1_blocks << ",";
>+  ost << "\"hysteresis_limit_2_blocks\": "
>+      << config.delay.hysteresis_limit_2_blocks << ",";
>+  ost << "\"skew_hysteresis_blocks\": " << config.delay.skew_hysteresis_blocks
>+      << ",";
>+  ost << "\"fixed_capture_delay_samples\": "
>+      << config.delay.fixed_capture_delay_samples << ",";
>+  ost << "\"delay_estimate_smoothing\": "
>+      << config.delay.delay_estimate_smoothing << ",";
>+  ost << "\"delay_candidate_detection_threshold\": "
>+      << config.delay.delay_candidate_detection_threshold << ",";
>+
>+  ost << "\"delay_selection_thresholds\": {";
>+  ost << "\"initial\": " << config.delay.delay_selection_thresholds.initial
>+      << ",";
>+  ost << "\"converged\": " << config.delay.delay_selection_thresholds.converged;
>+  ost << "}";
>+
>+  ost << "},";
>+
>+  ost << "\"filter\": {";
>+  ost << "\"main\": [";
>+  ost << config.filter.main.length_blocks << ",";
>+  ost << config.filter.main.leakage_converged << ",";
>+  ost << config.filter.main.leakage_diverged << ",";
>+  ost << config.filter.main.error_floor << ",";
>+  ost << config.filter.main.error_ceil << ",";
>+  ost << config.filter.main.noise_gate;
>+  ost << "],";
>+
>+  ost << "\"shadow\": [";
>+  ost << config.filter.shadow.length_blocks << ",";
>+  ost << config.filter.shadow.rate << ",";
>+  ost << config.filter.shadow.noise_gate;
>+  ost << "],";
>+
>+  ost << "\"main_initial\": [";
>+  ost << config.filter.main_initial.length_blocks << ",";
>+  ost << config.filter.main_initial.leakage_converged << ",";
>+  ost << config.filter.main_initial.leakage_diverged << ",";
>+  ost << config.filter.main_initial.error_floor << ",";
>+  ost << config.filter.main_initial.error_ceil << ",";
>+  ost << config.filter.main_initial.noise_gate;
>+  ost << "],";
>+
>+  ost << "\"shadow_initial\": [";
>+  ost << config.filter.shadow_initial.length_blocks << ",";
>+  ost << config.filter.shadow_initial.rate << ",";
>+  ost << config.filter.shadow_initial.noise_gate;
>+  ost << "],";
>+
>+  ost << "\"config_change_duration_blocks\": "
>+      << config.filter.config_change_duration_blocks << ",";
>+  ost << "\"initial_state_seconds\": " << config.filter.initial_state_seconds
>+      << ",";
>+  ost << "\"conservative_initial_phase\": "
>+      << (config.filter.conservative_initial_phase ? "true" : "false") << ",";
>+  ost << "\"enable_shadow_filter_output_usage\": "
>+      << (config.filter.enable_shadow_filter_output_usage ? "true" : "false");
>+
>+  ost << "},";
>+
>+  ost << "\"erle\": {";
>+  ost << "\"min\": " << config.erle.min << ",";
>+  ost << "\"max_l\": " << config.erle.max_l << ",";
>+  ost << "\"max_h\": " << config.erle.max_h << ",";
>+  ost << "\"onset_detection\": "
>+      << (config.erle.onset_detection ? "true" : "false") << ",";
>+  ost << "\"num_sections\": " << config.erle.num_sections;
>+  ost << "},";
>+
>+  ost << "\"ep_strength\": {";
>+  ost << "\"lf\": " << config.ep_strength.lf << ",";
>+  ost << "\"mf\": " << config.ep_strength.mf << ",";
>+  ost << "\"hf\": " << config.ep_strength.hf << ",";
>+  ost << "\"default_len\": " << config.ep_strength.default_len << ",";
>+  ost << "\"reverb_based_on_render\": "
>+      << (config.ep_strength.reverb_based_on_render ? "true" : "false") << ",";
>+  ost << "\"echo_can_saturate\": "
>+      << (config.ep_strength.echo_can_saturate ? "true" : "false") << ",";
>+  ost << "\"bounded_erl\": "
>+      << (config.ep_strength.bounded_erl ? "true" : "false");
>+
>+  ost << "},";
>+
>+  ost << "\"echo_audibility\": {";
>+  ost << "\"low_render_limit\": " << config.echo_audibility.low_render_limit
>+      << ",";
>+  ost << "\"normal_render_limit\": "
>+      << config.echo_audibility.normal_render_limit << ",";
>+  ost << "\"floor_power\": " << config.echo_audibility.floor_power << ",";
>+  ost << "\"audibility_threshold_lf\": "
>+      << config.echo_audibility.audibility_threshold_lf << ",";
>+  ost << "\"audibility_threshold_mf\": "
>+      << config.echo_audibility.audibility_threshold_mf << ",";
>+  ost << "\"audibility_threshold_hf\": "
>+      << config.echo_audibility.audibility_threshold_hf << ",";
>+  ost << "\"use_stationary_properties\": "
>+      << (config.echo_audibility.use_stationary_properties ? "true" : "false")
>+      << ",";
>+  ost << "\"use_stationarity_properties_at_init\": "
>+      << (config.echo_audibility.use_stationarity_properties_at_init ? "true"
>+                                                                     : "false");
>+  ost << "},";
>+
>+  ost << "\"render_levels\": {";
>+  ost << "\"active_render_limit\": " << config.render_levels.active_render_limit
>+      << ",";
>+  ost << "\"poor_excitation_render_limit\": "
>+      << config.render_levels.poor_excitation_render_limit << ",";
>+  ost << "\"poor_excitation_render_limit_ds8\": "
>+      << config.render_levels.poor_excitation_render_limit_ds8;
>+  ost << "},";
>+
>+  ost << "\"echo_removal_control\": {";
>+  ost << "\"gain_rampup\": {";
>+  ost << "\"initial_gain\": "
>+      << config.echo_removal_control.gain_rampup.initial_gain << ",";
>+  ost << "\"first_non_zero_gain\": "
>+      << config.echo_removal_control.gain_rampup.first_non_zero_gain << ",";
>+  ost << "\"non_zero_gain_blocks\": "
>+      << config.echo_removal_control.gain_rampup.non_zero_gain_blocks << ",";
>+  ost << "\"full_gain_blocks\": "
>+      << config.echo_removal_control.gain_rampup.full_gain_blocks;
>+  ost << "},";
>+  ost << "\"has_clock_drift\": "
>+      << (config.echo_removal_control.has_clock_drift ? "true" : "false")
>+      << ",";
>+  ost << "\"linear_and_stable_echo_path\": "
>+      << (config.echo_removal_control.linear_and_stable_echo_path ? "true"
>+                                                                  : "false");
>+
>+  ost << "},";
>+
>+  ost << "\"echo_model\": {";
>+  ost << "\"noise_floor_hold\": " << config.echo_model.noise_floor_hold << ",";
>+  ost << "\"min_noise_floor_power\": "
>+      << config.echo_model.min_noise_floor_power << ",";
>+  ost << "\"stationary_gate_slope\": "
>+      << config.echo_model.stationary_gate_slope << ",";
>+  ost << "\"noise_gate_power\": " << config.echo_model.noise_gate_power << ",";
>+  ost << "\"noise_gate_slope\": " << config.echo_model.noise_gate_slope << ",";
>+  ost << "\"render_pre_window_size\": "
>+      << config.echo_model.render_pre_window_size << ",";
>+  ost << "\"render_post_window_size\": "
>+      << config.echo_model.render_post_window_size << ",";
>+  ost << "\"render_pre_window_size_init\": "
>+      << config.echo_model.render_pre_window_size_init << ",";
>+  ost << "\"render_post_window_size_init\": "
>+      << config.echo_model.render_post_window_size_init << ",";
>+  ost << "\"nonlinear_hold\": " << config.echo_model.nonlinear_hold << ",";
>+  ost << "\"nonlinear_release\": " << config.echo_model.nonlinear_release;
>+  ost << "},";
>+
>+  ost << "\"suppressor\": {";
>+  ost << "\"nearend_average_blocks\": "
>+      << config.suppressor.nearend_average_blocks << ",";
>+  ost << "\"normal_tuning\": {";
>+  ost << "\"mask_lf\": [";
>+  ost << config.suppressor.normal_tuning.mask_lf.enr_transparent << ",";
>+  ost << config.suppressor.normal_tuning.mask_lf.enr_suppress << ",";
>+  ost << config.suppressor.normal_tuning.mask_lf.emr_transparent;
>+  ost << "],";
>+  ost << "\"mask_hf\": [";
>+  ost << config.suppressor.normal_tuning.mask_hf.enr_transparent << ",";
>+  ost << config.suppressor.normal_tuning.mask_hf.enr_suppress << ",";
>+  ost << config.suppressor.normal_tuning.mask_hf.emr_transparent;
>+  ost << "],";
>+  ost << "\"max_inc_factor\": "
>+      << config.suppressor.normal_tuning.max_inc_factor << ",";
>+  ost << "\"max_dec_factor_lf\": "
>+      << config.suppressor.normal_tuning.max_dec_factor_lf;
>+  ost << "},";
>+  ost << "\"nearend_tuning\": {";
>+  ost << "\"mask_lf\": [";
>+  ost << config.suppressor.nearend_tuning.mask_lf.enr_transparent << ",";
>+  ost << config.suppressor.nearend_tuning.mask_lf.enr_suppress << ",";
>+  ost << config.suppressor.nearend_tuning.mask_lf.emr_transparent;
>+  ost << "],";
>+  ost << "\"mask_hf\": [";
>+  ost << config.suppressor.nearend_tuning.mask_hf.enr_transparent << ",";
>+  ost << config.suppressor.nearend_tuning.mask_hf.enr_suppress << ",";
>+  ost << config.suppressor.nearend_tuning.mask_hf.emr_transparent;
>+  ost << "],";
>+  ost << "\"max_inc_factor\": "
>+      << config.suppressor.nearend_tuning.max_inc_factor << ",";
>+  ost << "\"max_dec_factor_lf\": "
>+      << config.suppressor.nearend_tuning.max_dec_factor_lf;
>+  ost << "},";
>+  ost << "\"dominant_nearend_detection\": {";
>+  ost << "\"enr_threshold\": "
>+      << config.suppressor.dominant_nearend_detection.enr_threshold << ",";
>+  ost << "\"enr_exit_threshold\": "
>+      << config.suppressor.dominant_nearend_detection.enr_exit_threshold << ",";
>+  ost << "\"snr_threshold\": "
>+      << config.suppressor.dominant_nearend_detection.snr_threshold << ",";
>+  ost << "\"hold_duration\": "
>+      << config.suppressor.dominant_nearend_detection.hold_duration << ",";
>+  ost << "\"trigger_threshold\": "
>+      << config.suppressor.dominant_nearend_detection.trigger_threshold << ",";
>+  ost << "\"use_during_initial_phase\": "
>+      << config.suppressor.dominant_nearend_detection.use_during_initial_phase;
>+  ost << "},";
>+  ost << "\"high_bands_suppression\": {";
>+  ost << "\"enr_threshold\": "
>+      << config.suppressor.high_bands_suppression.enr_threshold << ",";
>+  ost << "\"max_gain_during_echo\": "
>+      << config.suppressor.high_bands_suppression.max_gain_during_echo;
>+  ost << "},";
>+  ost << "\"floor_first_increase\": " << config.suppressor.floor_first_increase
>+      << ",";
>+  ost << "\"enforce_transparent\": "
>+      << (config.suppressor.enforce_transparent ? "true" : "false") << ",";
>+  ost << "\"enforce_empty_higher_bands\": "
>+      << (config.suppressor.enforce_empty_higher_bands ? "true" : "false");
>+  ost << "}";
>+  ost << "}";
>+  ost << "}";
>+
>+  return ost.Release();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config_json.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config_json.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..8973650f85c1a61753cbc591d4f1621286a981d4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config_json.h
>@@ -0,0 +1,44 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_AUDIO_ECHO_CANCELLER3_CONFIG_JSON_H_
>+#define API_AUDIO_ECHO_CANCELLER3_CONFIG_JSON_H_
>+
>+#include <string>
>+
>+#include "absl/strings/string_view.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "rtc_base/system/rtc_export.h"
>+
>+namespace webrtc {
>+// Parses a JSON-encoded string into an Aec3 config. Fields corresponds to
>+// substruct names, with the addition that there must be a top-level node
>+// "aec3". Produces default config values for anything that cannot be parsed
>+// from the string. If any error was found in the parsing, parsing_successful is
>+// set to false.
>+RTC_EXPORT void Aec3ConfigFromJsonString(absl::string_view json_string,
>+                                         EchoCanceller3Config* config,
>+                                         bool* parsing_successful);
>+
>+// To be deprecated.
>+// Parses a JSON-encoded string into an Aec3 config. Fields corresponds to
>+// substruct names, with the addition that there must be a top-level node
>+// "aec3". Returns default config values for anything that cannot be parsed from
>+// the string.
>+RTC_EXPORT EchoCanceller3Config
>+Aec3ConfigFromJsonString(absl::string_view json_string);
>+
>+// Encodes an Aec3 config in JSON format. Fields corresponds to substruct names,
>+// with the addition that the top-level node is named "aec3".
>+std::string Aec3ConfigToJsonString(const EchoCanceller3Config& config);
>+
>+}  // namespace webrtc
>+
>+#endif  // API_AUDIO_ECHO_CANCELLER3_CONFIG_JSON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.h
>index f6db11685037a1a2c32d84a6f765a287fc475b7f..9052d99bb1eb891bdc3ceb9f9ffa6a920ac32622 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.h
>@@ -15,10 +15,11 @@
> 
> #include "api/audio/echo_canceller3_config.h"
> #include "api/audio/echo_control.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>-class EchoCanceller3Factory : public EchoControlFactory {
>+class RTC_EXPORT EchoCanceller3Factory : public EchoControlFactory {
>  public:
>   // Factory producing EchoCanceller3 instances with the default configuration.
>   EchoCanceller3Factory();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/BUILD.gn
>index 9a797788f68dcea72de92123c1b5524a51447bad..4e04a8a64a1048af0332431b46c157fda6ddded1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/BUILD.gn
>@@ -17,8 +17,12 @@ if (rtc_include_tests) {
>     testonly = true
>     sources = [
>       "audio_frame_unittest.cc",
>+      "echo_canceller3_config_json_unittest.cc",
>+      "echo_canceller3_config_unittest.cc",
>     ]
>     deps = [
>+      "..:aec3_config",
>+      "..:aec3_config_json",
>       "..:audio_frame_api",
>       "../../../rtc_base:rtc_base_approved",
>       "../../../test:test_support",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/echo_canceller3_config_json_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/echo_canceller3_config_json_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..7c3a445c60a3412d03a0a2cef2947de14d156b5d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/echo_canceller3_config_json_unittest.cc
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/audio/echo_canceller3_config_json.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+TEST(EchoCanceller3JsonHelpers, ToStringAndParseJson) {
>+  EchoCanceller3Config cfg;
>+  cfg.delay.down_sampling_factor = 1u;
>+  cfg.filter.shadow_initial.length_blocks = 7u;
>+  cfg.suppressor.normal_tuning.mask_hf.enr_suppress = .5f;
>+  std::string json_string = Aec3ConfigToJsonString(cfg);
>+  EchoCanceller3Config cfg_transformed = Aec3ConfigFromJsonString(json_string);
>+
>+  // Expect unchanged values to remain default.
>+  EXPECT_EQ(cfg.filter.main.error_floor,
>+            cfg_transformed.filter.main.error_floor);
>+  EXPECT_EQ(cfg.ep_strength.default_len,
>+            cfg_transformed.ep_strength.default_len);
>+  EXPECT_EQ(cfg.suppressor.normal_tuning.mask_lf.enr_suppress,
>+            cfg_transformed.suppressor.normal_tuning.mask_lf.enr_suppress);
>+
>+  // Expect changed values to carry through the transformation.
>+  EXPECT_EQ(cfg.delay.down_sampling_factor,
>+            cfg_transformed.delay.down_sampling_factor);
>+  EXPECT_EQ(cfg.filter.shadow_initial.length_blocks,
>+            cfg_transformed.filter.shadow_initial.length_blocks);
>+  EXPECT_EQ(cfg.suppressor.normal_tuning.mask_hf.enr_suppress,
>+            cfg_transformed.suppressor.normal_tuning.mask_hf.enr_suppress);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/echo_canceller3_config_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/echo_canceller3_config_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..ffe72ac958df59c554e910297707b5d7ca14c30e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/echo_canceller3_config_unittest.cc
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <string>
>+
>+#include "api/audio/echo_canceller3_config.h"
>+#include "api/audio/echo_canceller3_config_json.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+TEST(EchoCanceller3Config, ValidConfigIsNotModified) {
>+  EchoCanceller3Config config;
>+  EXPECT_TRUE(EchoCanceller3Config::Validate(&config));
>+  EchoCanceller3Config default_config;
>+  EXPECT_EQ(Aec3ConfigToJsonString(config),
>+            Aec3ConfigToJsonString(default_config));
>+}
>+
>+TEST(EchoCanceller3Config, InvalidConfigIsCorrected) {
>+  // Change a parameter and validate.
>+  EchoCanceller3Config config;
>+  config.echo_model.min_noise_floor_power = -1600000.f;
>+  EXPECT_FALSE(EchoCanceller3Config::Validate(&config));
>+  EXPECT_GE(config.echo_model.min_noise_floor_power, 0.f);
>+  // Verify remaining parameters are unchanged.
>+  EchoCanceller3Config default_config;
>+  config.echo_model.min_noise_floor_power =
>+      default_config.echo_model.min_noise_floor_power;
>+  EXPECT_EQ(Aec3ConfigToJsonString(config),
>+            Aec3ConfigToJsonString(default_config));
>+}
>+
>+TEST(EchoCanceller3Config, ValidatedConfigsAreValid) {
>+  EchoCanceller3Config config;
>+  config.delay.down_sampling_factor = 983;
>+  EXPECT_FALSE(EchoCanceller3Config::Validate(&config));
>+  EXPECT_TRUE(EchoCanceller3Config::Validate(&config));
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/BUILD.gn
>index a7060a229adf0171814724333a698343e2ccaad5..31fe2bec58b03be53e1428494a2702a0e6254389 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/BUILD.gn
>@@ -30,11 +30,13 @@ rtc_source_set("audio_codecs_api") {
>   ]
>   deps = [
>     "..:array_view",
>+    "..:bitrate_allocation",
>     "../..:webrtc_common",
>     "../../rtc_base:checks",
>     "../../rtc_base:deprecation",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:sanitizer",
>+    "../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/BUILD.gn
>index 972480500125fa9dce27307b4753d6211661fe04..34ec2e4208ef34856b5854ba9c0cdef4b99372cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/BUILD.gn
>@@ -21,11 +21,12 @@ rtc_static_library("audio_encoder_L16") {
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:pcm16b",
>     "../../../rtc_base:rtc_base_approved",
>     "../../../rtc_base:safe_minmax",
>+    "../../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -39,10 +40,11 @@ rtc_static_library("audio_decoder_L16") {
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:pcm16b",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc
>index a71e3087007df3fd76fa232fc3e3468fa3821e99..be0c6b56b25b569eae5e65bc7785987c4c586217 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc
>@@ -11,7 +11,7 @@
> #include "api/audio_codecs/L16/audio_decoder_L16.h"
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
> #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
> #include "rtc_base/numerics/safe_conversions.h"
>@@ -23,7 +23,7 @@ absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
>   Config config;
>   config.sample_rate_hz = format.clockrate_hz;
>   config.num_channels = rtc::checked_cast<int>(format.num_channels);
>-  return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
>+  return absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()
>              ? absl::optional<Config>(config)
>              : absl::nullopt;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.h
>index 184ec24ed92f1e22d30c9b2e6be7f3d7f77e15c8..f0be03659c548c1c478b3ae0e9564e5595303135 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.h
>@@ -18,14 +18,13 @@
> #include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // L16 decoder API for use as a template parameter to
> // CreateAudioDecoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
>-struct AudioDecoderL16 {
>+struct RTC_EXPORT AudioDecoderL16 {
>   struct Config {
>     bool IsOk() const {
>       return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
>index b516f62863da24fce35ef52d46a0e04c2d5c797f..1bb552bc5c1e6e3257b18572f97b3345fdae3b7f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
>@@ -11,7 +11,7 @@
> #include "api/audio_codecs/L16/audio_encoder_L16.h"
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
> #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
> #include "rtc_base/numerics/safe_conversions.h"
>@@ -35,7 +35,7 @@ absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
>       config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
>     }
>   }
>-  return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
>+  return absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()
>              ? absl::optional<Config>(config)
>              : absl::nullopt;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.h
>index 340e3af47fbaf77caaeb844311a74b89506edfca..b410286802fe7689acc4f9889778b88cc8ee8a3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.h
>@@ -18,14 +18,13 @@
> #include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // L16 encoder API for use as a template parameter to
> // CreateAudioEncoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
>-struct AudioEncoderL16 {
>+struct RTC_EXPORT AudioEncoderL16 {
>   struct Config {
>     bool IsOk() const {
>       return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_codec_pair_id.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_codec_pair_id.cc
>index ac84107eda108249d94d31a8717763e9f04af178..6cb51ed6b72c7e4f4652dc728aa33ae3e3952614 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_codec_pair_id.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_codec_pair_id.cc
>@@ -11,6 +11,7 @@
> #include "api/audio_codecs/audio_codec_pair_id.h"
> 
> #include <atomic>
>+#include <cstdint>
> 
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.h
>index b01a66a064a44ebf89e5e87891c9bd13bdda9894..889e2c62d0ed89fa04879f8b7a8971ebb3a5fef4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.h
>@@ -11,6 +11,8 @@
> #ifndef API_AUDIO_CODECS_AUDIO_DECODER_H_
> #define API_AUDIO_CODECS_AUDIO_DECODER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> #include <vector>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory.h
>index 90f93f0cd5fc908e99183196fad06fddf78ee4fb..55e197a14a92708226d3dfabc357f23c0aa75f1f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory.h
>@@ -23,7 +23,6 @@
> namespace webrtc {
> 
> // A factory that creates AudioDecoders.
>-// NOTE: This class is still under development and may change without notice.
> class AudioDecoderFactory : public rtc::RefCountInterface {
>  public:
>   virtual std::vector<AudioCodecSpec> GetSupportedDecoders() = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory_template.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory_template.h
>index cdbe8bde7a842e8e8d5a2a863ca8ae1e8cecb287..22bb7de472136e8c4eeac2b1d3e864144436311b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory_template.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory_template.h
>@@ -112,8 +112,6 @@ class AudioDecoderFactoryT : public AudioDecoderFactory {
> // decoder types in the order they were specified in the template argument
> // list, stopping at the first one that claims to be able to do the job.
> //
>-// NOTE: This function is still under development and may change without notice.
>-//
> // TODO(kwiberg): Point at CreateBuiltinAudioDecoderFactory() for an example of
> // how it is used.
> template <typename... Ts>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.cc
>index 595c11131e7c41066f34ee241e7f19f32906274a..1d885f9cecf84d8098f4e4c241271f07d7efe58f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.cc
>@@ -92,6 +92,11 @@ void AudioEncoder::OnReceivedUplinkBandwidth(
>     int target_audio_bitrate_bps,
>     absl::optional<int64_t> bwe_period_ms) {}
> 
>+void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) {
>+  OnReceivedUplinkBandwidth(update.target_bitrate.bps(),
>+                            update.bwe_period.ms());
>+}
>+
> void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
> 
> void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.h
>index 6a1f6534b21a8a3ffb6d209d95571c89ae208a16..c908518063d4da7c94aa0e9346db7ee3acbef1e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.h
>@@ -11,13 +11,13 @@
> #ifndef API_AUDIO_CODECS_AUDIO_ENCODER_H_
> #define API_AUDIO_CODECS_AUDIO_ENCODER_H_
> 
>-#include <algorithm>
> #include <memory>
> #include <string>
> #include <vector>
> 
> #include "absl/types/optional.h"
> #include "api/array_view.h"
>+#include "api/call/bitrate_allocation.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/deprecation.h"
> 
>@@ -222,6 +222,10 @@ class AudioEncoder {
>   virtual void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
>                                          absl::optional<int64_t> bwe_period_ms);
> 
>+  // Provides target audio bitrate and corresponding probing interval of
>+  // the bandwidth estimator to this encoder to allow it to adapt.
>+  virtual void OnReceivedUplinkAllocation(BitrateAllocationUpdate update);
>+
>   // Provides RTT to this encoder to allow it to adapt.
>   virtual void OnReceivedRtt(int rtt_ms);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory.h
>index fb4e23ffe78b5b22c67e22888a3272c67ad3b9b9..b290967c7803fd747d8b47b3465f449e74beae09 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory.h
>@@ -23,7 +23,6 @@
> namespace webrtc {
> 
> // A factory that creates AudioEncoders.
>-// NOTE: This class is still under development and may change without notice.
> class AudioEncoderFactory : public rtc::RefCountInterface {
>  public:
>   // Returns a prioritized list of audio codecs, to use for signaling etc.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory_template.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory_template.h
>index 376b39e4c6f9fc8eaa3c7ef37adcdee61c428383..20886190196262ba2777ce6cbba78fede669e760 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory_template.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory_template.h
>@@ -131,8 +131,6 @@ class AudioEncoderFactoryT : public AudioEncoderFactory {
> // encoders in the order they were specified in the template argument list,
> // stopping at the first one that claims to be able to do the job.
> //
>-// NOTE: This function is still under development and may change without notice.
>-//
> // TODO(kwiberg): Point at CreateBuiltinAudioEncoderFactory() for an example of
> // how it is used.
> template <typename... Ts>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.cc
>index 81ed299ddb66aaf836967d4f446f73e908bc7211..11788b91e829198bf277bc34dffe80b9ff539f6e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.cc
>@@ -10,7 +10,7 @@
> 
> #include "api/audio_codecs/audio_format.h"
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> 
> namespace webrtc {
> 
>@@ -32,7 +32,7 @@ SdpAudioFormat::SdpAudioFormat(absl::string_view name,
>       parameters(param) {}
> 
> bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const {
>-  return STR_CASE_CMP(name.c_str(), o.name.c_str()) == 0 &&
>+  return absl::EqualsIgnoreCase(name, o.name) &&
>          clockrate_hz == o.clockrate_hz && num_channels == o.num_channels;
> }
> 
>@@ -41,19 +41,11 @@ SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
> SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
> 
> bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
>-  return STR_CASE_CMP(a.name.c_str(), b.name.c_str()) == 0 &&
>+  return absl::EqualsIgnoreCase(a.name, b.name) &&
>          a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
>          a.parameters == b.parameters;
> }
> 
>-void swap(SdpAudioFormat& a, SdpAudioFormat& b) {
>-  using std::swap;
>-  swap(a.name, b.name);
>-  swap(a.clockrate_hz, b.clockrate_hz);
>-  swap(a.num_channels, b.num_channels);
>-  swap(a.parameters, b.parameters);
>-}
>-
> AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
>                                size_t num_channels,
>                                int bitrate_bps)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.h
>index aa5dbb13b65d72ed0f4d6d6ee686d47202fd479c..b8b042178ba1a5e24275e55957d76a52d8a480a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.h
>@@ -11,19 +11,18 @@
> #ifndef API_AUDIO_CODECS_AUDIO_FORMAT_H_
> #define API_AUDIO_CODECS_AUDIO_FORMAT_H_
> 
>+#include <stddef.h>
> #include <map>
> #include <string>
>-#include <utility>
> 
> #include "absl/strings/string_view.h"
>-#include "absl/types/optional.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // SDP specification for a single audio codec.
>-// NOTE: This class is still under development and may change without notice.
>-struct SdpAudioFormat {
>+struct RTC_EXPORT SdpAudioFormat {
>   using Parameters = std::map<std::string, std::string>;
> 
>   SdpAudioFormat(const SdpAudioFormat&);
>@@ -54,8 +53,6 @@ struct SdpAudioFormat {
>   Parameters parameters;
> };
> 
>-void swap(SdpAudioFormat& a, SdpAudioFormat& b);
>-
> // Information about how an audio format is treated by the codec implementation.
> // Contains basic information, such as sample rate and number of channels, which
> // isn't uniformly presented by SDP. Also contains flags indicating support for
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_decoder_factory.h
>index 3127403e243fb901863105a76d94dc66b937dd68..3accd4a3020d2429918149a4b6b9bf3e685ec1f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_decoder_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_decoder_factory.h
>@@ -17,7 +17,6 @@
> namespace webrtc {
> 
> // Creates a new factory that can create the built-in types of audio decoders.
>-// NOTE: This function is still under development and may change without notice.
> rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory();
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_encoder_factory.h
>index d37ff257e6e32e39cc777845fe8d3972eb3e13cc..3c67dd30e66b2351ed40aceac154b1b428fbd459 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_encoder_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_encoder_factory.h
>@@ -17,7 +17,6 @@
> namespace webrtc {
> 
> // Creates a new factory that can create the built-in types of audio encoders.
>-// NOTE: This function is still under development and may change without notice.
> rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory();
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/BUILD.gn
>index 169172a66829ab793f51831ad4c9ba888507666b..3b8f23c1bc6e7430755ac5f72ee928ffa4342b8d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/BUILD.gn
>@@ -21,11 +21,12 @@ rtc_static_library("audio_encoder_g711") {
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:g711",
>     "../../../rtc_base:rtc_base_approved",
>     "../../../rtc_base:safe_minmax",
>+    "../../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -39,10 +40,11 @@ rtc_static_library("audio_decoder_g711") {
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:g711",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.cc
>index cb16584cdee5402eb205b43b47b939531a77a609..91599c416adf93c459895c3003cdfdf32789b6a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.cc
>@@ -14,7 +14,7 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
> #include "rtc_base/numerics/safe_conversions.h"
> 
>@@ -22,8 +22,8 @@ namespace webrtc {
> 
> absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
>     const SdpAudioFormat& format) {
>-  const bool is_pcmu = STR_CASE_CMP(format.name.c_str(), "PCMU") == 0;
>-  const bool is_pcma = STR_CASE_CMP(format.name.c_str(), "PCMA") == 0;
>+  const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
>+  const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
>   if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
>       (is_pcmu || is_pcma)) {
>     Config config;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.h
>index 8275a8c7a14bff4df4d14c4c8b38593673789c2e..ccd1ee0480052733b716c4683efd7924d2b519dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.h
>@@ -18,14 +18,13 @@
> #include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // G711 decoder API for use as a template parameter to
> // CreateAudioDecoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
>-struct AudioDecoderG711 {
>+struct RTC_EXPORT AudioDecoderG711 {
>   struct Config {
>     enum class Type { kPcmU, kPcmA };
>     bool IsOk() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.cc
>index 1d5e541276b9aedd86c68d01f5c7c707cc28fe3c..0cc8dc4ddd29ef5b55ee071927a6943e60334d89 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.cc
>@@ -14,7 +14,7 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/numerics/safe_minmax.h"
>@@ -24,8 +24,8 @@ namespace webrtc {
> 
> absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
>     const SdpAudioFormat& format) {
>-  const bool is_pcmu = STR_CASE_CMP(format.name.c_str(), "PCMU") == 0;
>-  const bool is_pcma = STR_CASE_CMP(format.name.c_str(), "PCMA") == 0;
>+  const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
>+  const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
>   if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
>       (is_pcmu || is_pcma)) {
>     Config config;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.h
>index 6b6eb5fce0ce54fcbd60c64fea67d9bcda1b1686..23ae18b5e3f032b839cf7e909e08445f3b68e0eb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.h
>@@ -18,14 +18,13 @@
> #include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // G711 encoder API for use as a template parameter to
> // CreateAudioEncoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
>-struct AudioEncoderG711 {
>+struct RTC_EXPORT AudioEncoderG711 {
>   struct Config {
>     enum class Type { kPcmU, kPcmA };
>     bool IsOk() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/BUILD.gn
>index 50b1396d2915ebff2c6d11ce35dcc7891c3b1c30..e4321d299317af8f01ad3e562a043096ee1ef214 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/BUILD.gn
>@@ -29,11 +29,12 @@ rtc_static_library("audio_encoder_g722") {
>   deps = [
>     ":audio_encoder_g722_config",
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:g722",
>     "../../../rtc_base:rtc_base_approved",
>     "../../../rtc_base:safe_minmax",
>+    "../../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -47,10 +48,11 @@ rtc_static_library("audio_decoder_g722") {
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:g722",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.cc
>index f1e2afbab50d0ff7d038645e84cf7f256b198e1a..2cc16c37d94d650b4eeabb26cdc4d29b8045583a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.cc
>@@ -14,7 +14,7 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
> #include "rtc_base/numerics/safe_conversions.h"
> 
>@@ -22,7 +22,7 @@ namespace webrtc {
> 
> absl::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
>     const SdpAudioFormat& format) {
>-  return STR_CASE_CMP(format.name.c_str(), "G722") == 0 &&
>+  return absl::EqualsIgnoreCase(format.name, "G722") &&
>                  format.clockrate_hz == 8000 &&
>                  (format.num_channels == 1 || format.num_channels == 2)
>              ? absl::optional<Config>(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.h
>index b7bb0893f3303e3a4f98a57e687bee620504ec9e..2a674926dbccf1c76c090595deeb1390c3a13f81 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.h
>@@ -18,14 +18,13 @@
> #include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // G722 decoder API for use as a template parameter to
> // CreateAudioDecoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
>-struct AudioDecoderG722 {
>+struct RTC_EXPORT AudioDecoderG722 {
>   struct Config {
>     bool IsOk() const { return num_channels == 1 || num_channels == 2; }
>     int num_channels;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc
>index 0cf71639bf32f937d2fd452470a41be87f89ab4f..6374ae802faa87748ccba46e172752bb7668c376 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc
>@@ -14,7 +14,7 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/numerics/safe_minmax.h"
>@@ -24,7 +24,7 @@ namespace webrtc {
> 
> absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
>     const SdpAudioFormat& format) {
>-  if (STR_CASE_CMP(format.name.c_str(), "g722") != 0 ||
>+  if (!absl::EqualsIgnoreCase(format.name, "g722") ||
>       format.clockrate_hz != 8000) {
>     return absl::nullopt;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.h
>index b97fe1b147890d0854276d1311208e99b10ccf9a..327c0af04aa7ce0455841f981c662abf862b6dbd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.h
>@@ -19,14 +19,13 @@
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
> #include "api/audio_codecs/g722/audio_encoder_g722_config.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // G722 encoder API for use as a template parameter to
> // CreateAudioEncoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
>-struct AudioEncoderG722 {
>+struct RTC_EXPORT AudioEncoderG722 {
>   using Config = AudioEncoderG722Config;
>   static absl::optional<AudioEncoderG722Config> SdpToConfig(
>       const SdpAudioFormat& audio_format);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h
>index 773e430ce3c0c8c06ad2e5c55a0453914038274d..287898589fcaaa02fbbd895a1cfbfc97d3891c99 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h
>@@ -13,7 +13,6 @@
> 
> namespace webrtc {
> 
>-// NOTE: This struct is still under development and may change without notice.
> struct AudioEncoderG722Config {
>   bool IsOk() const {
>     return frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/BUILD.gn
>index d4f504fd4f9137cbcffc42818e8c04aa9c21dfc7..d766fa4a97a138c62aa44206ee14944d30da4dce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/BUILD.gn
>@@ -29,11 +29,11 @@ rtc_static_library("audio_encoder_ilbc") {
>   deps = [
>     ":audio_encoder_ilbc_config",
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:ilbc",
>     "../../../rtc_base:rtc_base_approved",
>     "../../../rtc_base:safe_minmax",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -47,10 +47,10 @@ rtc_static_library("audio_decoder_ilbc") {
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:ilbc",
>     "../../../rtc_base:rtc_base_approved",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
>index 1f4c475c406428b771634cdefc3541e6937dfc72..4a00f8dbf0054c6a3a6a169a1ea1faa6ead19ebf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
>@@ -14,14 +14,14 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
> 
> namespace webrtc {
> 
> absl::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
>     const SdpAudioFormat& format) {
>-  return STR_CASE_CMP(format.name.c_str(), "ILBC") == 0 &&
>+  return absl::EqualsIgnoreCase(format.name, "ILBC") &&
>                  format.clockrate_hz == 8000 && format.num_channels == 1
>              ? absl::optional<Config>(Config())
>              : absl::nullopt;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h
>index 20f6ffd287f9c8f0976b850872e164f6a7ed8255..9ab847977d81127169903c065bab3d9f5b34ae86 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h
>@@ -23,8 +23,6 @@ namespace webrtc {
> 
> // ILBC decoder API for use as a template parameter to
> // CreateAudioDecoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
> struct AudioDecoderIlbc {
>   struct Config {};  // Empty---no config values needed!
>   static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>index 2ae75474ccd8dea9d503de7f80af541d0d8050e1..4d028c9aaf4c0884c38b1034ea07300fe72125d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>@@ -14,7 +14,7 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/numerics/safe_minmax.h"
>@@ -40,7 +40,7 @@ int GetIlbcBitrate(int ptime) {
> 
> absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
>     const SdpAudioFormat& format) {
>-  if (STR_CASE_CMP(format.name.c_str(), "ILBC") != 0 ||
>+  if (!absl::EqualsIgnoreCase(format.name.c_str(), "ILBC") ||
>       format.clockrate_hz != 8000 || format.num_channels != 1) {
>     return absl::nullopt;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h
>index 0a86b162e81c628ccfbfdd907a01dd8a895e533b..e4aeca70de840e174621398541e9561b34285027 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h
>@@ -24,8 +24,6 @@ namespace webrtc {
> 
> // ILBC encoder API for use as a template parameter to
> // CreateAudioEncoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
> struct AudioEncoderIlbc {
>   using Config = AudioEncoderIlbcConfig;
>   static absl::optional<AudioEncoderIlbcConfig> SdpToConfig(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h
>index 22909a957bb257b683983c129c757187c0e86664..4d82f9901cdfe193a4da543d5830ebad20f9947a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h
>@@ -13,7 +13,6 @@
> 
> namespace webrtc {
> 
>-// NOTE: This struct is still under development and may change without notice.
> struct AudioEncoderIlbcConfig {
>   bool IsOk() const {
>     return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 ||
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/BUILD.gn
>index ed9d9622577cf6aa1cdc46f68d0d808147a01d5c..c7d6e43271529113cf1fd56fd8dcd80fcdaf0a10 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/BUILD.gn
>@@ -77,10 +77,10 @@ rtc_static_library("audio_encoder_isac_fix") {
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:isac_fix",
>     "../../../rtc_base:rtc_base_approved",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -94,10 +94,10 @@ rtc_static_library("audio_decoder_isac_fix") {
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:isac_fix",
>     "../../../rtc_base:rtc_base_approved",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -111,10 +111,11 @@ rtc_static_library("audio_encoder_isac_float") {
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:isac",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -128,10 +129,11 @@ rtc_static_library("audio_decoder_isac_float") {
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:isac",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.cc
>index 446640ff70c06b4808528bbe720f3ebbc967a3e7..51ae572fa982e5e49ba972e973e1709e58d1087b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.cc
>@@ -11,14 +11,14 @@
> #include "api/audio_codecs/isac/audio_decoder_isac_fix.h"
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
> 
> namespace webrtc {
> 
> absl::optional<AudioDecoderIsacFix::Config> AudioDecoderIsacFix::SdpToConfig(
>     const SdpAudioFormat& format) {
>-  return STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
>+  return absl::EqualsIgnoreCase(format.name, "ISAC") &&
>                  format.clockrate_hz == 16000 && format.num_channels == 1
>              ? absl::optional<Config>(Config())
>              : absl::nullopt;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h
>index a4ce6858385852d1c3b6adeeca5413b6c4f6bec6..b7a5cef6dc5cd73629949e12f08668e992b60b2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h
>@@ -23,8 +23,6 @@ namespace webrtc {
> 
> // iSAC decoder API (fixed-point implementation) for use as a template
> // parameter to CreateAudioDecoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
> struct AudioDecoderIsacFix {
>   struct Config {};  // Empty---no config values needed!
>   static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.cc
>index 1c1926f1a1ebfbc0ae8c3647ab4f2cad198cdfa1..d9de3a08b30121849f99c644c320e7049a25e00d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.cc
>@@ -11,14 +11,14 @@
> #include "api/audio_codecs/isac/audio_decoder_isac_float.h"
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
> 
> namespace webrtc {
> 
> absl::optional<AudioDecoderIsacFloat::Config>
> AudioDecoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
>-  if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
>+  if (absl::EqualsIgnoreCase(format.name, "ISAC") &&
>       (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
>       format.num_channels == 1) {
>     Config config;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h
>index cc139634196ddf5ee3d2a43f6726834d66388ec8..e78f8b81ee01fd634a07711a36fb45a64a6d3da9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h
>@@ -18,14 +18,13 @@
> #include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // iSAC decoder API (floating-point implementation) for use as a template
> // parameter to CreateAudioDecoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
>-struct AudioDecoderIsacFloat {
>+struct RTC_EXPORT AudioDecoderIsacFloat {
>   struct Config {
>     bool IsOk() const {
>       return sample_rate_hz == 16000 || sample_rate_hz == 32000;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc
>index cd8975398b4178c9e985495cbc4ad7c055d5c66c..a10d1ee61ae8f896cd2c4cad4b368aec4d0a8650 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc
>@@ -11,7 +11,7 @@
> #include "api/audio_codecs/isac/audio_encoder_isac_fix.h"
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
> #include "rtc_base/string_to_number.h"
> 
>@@ -19,7 +19,7 @@ namespace webrtc {
> 
> absl::optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
>     const SdpAudioFormat& format) {
>-  if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
>+  if (absl::EqualsIgnoreCase(format.name, "ISAC") &&
>       format.clockrate_hz == 16000 && format.num_channels == 1) {
>     Config config;
>     const auto ptime_iter = format.parameters.find("ptime");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h
>index 731e48d0c27668aa77c3b08a56b1610edb6bc5a6..c3c3672b583d1959cd0c01c2f48f21681d9e1e30 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h
>@@ -23,8 +23,6 @@ namespace webrtc {
> 
> // iSAC encoder API (fixed-point implementation) for use as a template
> // parameter to CreateAudioEncoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
> struct AudioEncoderIsacFix {
>   struct Config {
>     bool IsOk() const { return frame_size_ms == 30 || frame_size_ms == 60; }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
>index 83d1faff2fc24ff20bff42c51dd226364ba30103..37982b1f15872f73490578aa4a35bf114684706d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
>@@ -11,7 +11,7 @@
> #include "api/audio_codecs/isac/audio_encoder_isac_float.h"
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
> #include "rtc_base/string_to_number.h"
> 
>@@ -19,7 +19,7 @@ namespace webrtc {
> 
> absl::optional<AudioEncoderIsacFloat::Config>
> AudioEncoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
>-  if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
>+  if (absl::EqualsIgnoreCase(format.name, "ISAC") &&
>       (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
>       format.num_channels == 1) {
>     Config config;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
>index 5df2dd3d6f4b065d74d065fd2e06bd35fa85b320..0cb9c17d7105ed3865c82dab909a110416ec2589 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
>@@ -18,14 +18,13 @@
> #include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // iSAC encoder API (floating-point implementation) for use as a template
> // parameter to CreateAudioEncoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
>-struct AudioEncoderIsacFloat {
>+struct RTC_EXPORT AudioEncoderIsacFloat {
>   struct Config {
>     bool IsOk() const {
>       switch (sample_rate_hz) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/BUILD.gn
>index 800abbe279bfe3a65d16bb637bcdff053f092e7a..5552c21fa51079676d89705a0992b13e94ff7d83 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/BUILD.gn
>@@ -20,6 +20,7 @@ rtc_static_library("audio_encoder_opus_config") {
>   ]
>   deps = [
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
>   defines = []
>@@ -44,6 +45,8 @@ rtc_source_set("audio_encoder_opus") {
>     "..:audio_codecs_api",
>     "../../../modules/audio_coding:webrtc_opus",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/system:rtc_export",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -57,10 +60,11 @@ rtc_static_library("audio_decoder_opus") {
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../../..:webrtc_common",
>     "../../../modules/audio_coding:webrtc_opus",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc
>index 41397f04d2024893e26922cde0edb61542b21f9f..2f1668b825bc020908d983981a512bf7f3760dd4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc
>@@ -15,7 +15,7 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
> 
> namespace webrtc {
>@@ -35,7 +35,7 @@ absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
>     }
>     return 1;  // Default to mono.
>   }();
>-  if (STR_CASE_CMP(format.name.c_str(), "opus") == 0 &&
>+  if (absl::EqualsIgnoreCase(format.name, "opus") &&
>       format.clockrate_hz == 48000 && format.num_channels == 2 &&
>       num_channels) {
>     return Config{*num_channels};
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.h
>index de26026b7833979b3fed7ad9b5740e3ea720607a..6fbbcb598a14d77ae6dbf246bb387e4f8b8b6d49 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.h
>@@ -18,14 +18,13 @@
> #include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // Opus decoder API for use as a template parameter to
> // CreateAudioDecoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
>-struct AudioDecoderOpus {
>+struct RTC_EXPORT AudioDecoderOpus {
>   struct Config {
>     int num_channels;
>   };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.h
>index 20aaaf726f93149490320d082e02fbb0df33ddbc..03cb0d6b38c8334569fb668872ae4d8e76a4d640 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.h
>@@ -19,14 +19,13 @@
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
> #include "api/audio_codecs/opus/audio_encoder_opus_config.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // Opus encoder API for use as a template parameter to
> // CreateAudioEncoderFactory<...>().
>-//
>-// NOTE: This struct is still under development and may change without notice.
>-struct AudioEncoderOpus {
>+struct RTC_EXPORT AudioEncoderOpus {
>   using Config = AudioEncoderOpusConfig;
>   static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
>       const SdpAudioFormat& audio_format);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
>index c7067bb1c324186f2b6ddc56731c5ebbe2ad3567..0a3ee2c8dfb0d40d8a2c12080c0513f5dd5061cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
>@@ -16,11 +16,11 @@
> #include <vector>
> 
> #include "absl/types/optional.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>-// NOTE: This struct is still under development and may change without notice.
>-struct AudioEncoderOpusConfig {
>+struct RTC_EXPORT AudioEncoderOpusConfig {
>   static constexpr int kDefaultFrameSizeMs = 20;
> 
>   // Opus API allows a min bitrate of 500bps, but Opus documentation suggests
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.cc
>index a4a37d219a43c9ea25765ea59cc3a3716a9593fa..e33214bad04d6ce9ba4f482fcc450d8ecbba4f43 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.cc
>@@ -49,8 +49,9 @@ void AudioOptions::SetAll(const AudioOptions& change) {
>           change.audio_jitter_buffer_max_packets);
>   SetFrom(&audio_jitter_buffer_fast_accelerate,
>           change.audio_jitter_buffer_fast_accelerate);
>+  SetFrom(&audio_jitter_buffer_min_delay_ms,
>+          change.audio_jitter_buffer_min_delay_ms);
>   SetFrom(&typing_detection, change.typing_detection);
>-  SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
>   SetFrom(&experimental_agc, change.experimental_agc);
>   SetFrom(&extended_filter_aec, change.extended_filter_aec);
>   SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
>@@ -77,8 +78,9 @@ bool AudioOptions::operator==(const AudioOptions& o) const {
>          audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
>          audio_jitter_buffer_fast_accelerate ==
>              o.audio_jitter_buffer_fast_accelerate &&
>+         audio_jitter_buffer_min_delay_ms ==
>+             o.audio_jitter_buffer_min_delay_ms &&
>          typing_detection == o.typing_detection &&
>-         aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
>          experimental_agc == o.experimental_agc &&
>          extended_filter_aec == o.extended_filter_aec &&
>          delay_agnostic_aec == o.delay_agnostic_aec &&
>@@ -109,8 +111,9 @@ std::string AudioOptions::ToString() const {
>                 audio_jitter_buffer_max_packets);
>   ToStringIfSet(&result, "audio_jitter_buffer_fast_accelerate",
>                 audio_jitter_buffer_fast_accelerate);
>+  ToStringIfSet(&result, "audio_jitter_buffer_min_delay_ms",
>+                audio_jitter_buffer_min_delay_ms);
>   ToStringIfSet(&result, "typing", typing_detection);
>-  ToStringIfSet(&result, "comfort_noise", aecm_generate_comfort_noise);
>   ToStringIfSet(&result, "experimental_agc", experimental_agc);
>   ToStringIfSet(&result, "extended_filter_aec", extended_filter_aec);
>   ToStringIfSet(&result, "delay_agnostic_aec", delay_agnostic_aec);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.h
>index aecff414298cd248ee42c106f2d530e53e632551..c2d1f4487c36874d5389d6d54a9b1ea4218b5842 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.h
>@@ -11,6 +11,7 @@
> #ifndef API_AUDIO_OPTIONS_H_
> #define API_AUDIO_OPTIONS_H_
> 
>+#include <stdint.h>
> #include <string>
> 
> #include "absl/types/optional.h"
>@@ -53,9 +54,10 @@ struct AudioOptions {
>   absl::optional<int> audio_jitter_buffer_max_packets;
>   // Audio receiver jitter buffer (NetEq) fast accelerate mode.
>   absl::optional<bool> audio_jitter_buffer_fast_accelerate;
>+  // Audio receiver jitter buffer (NetEq) minimum target delay in milliseconds.
>+  absl::optional<int> audio_jitter_buffer_min_delay_ms;
>   // Audio processing to detect typing.
>   absl::optional<bool> typing_detection;
>-  absl::optional<bool> aecm_generate_comfort_noise;
>   absl::optional<bool> experimental_agc;
>   absl::optional<bool> extended_filter_aec;
>   absl::optional<bool> delay_agnostic_aec;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/bitrate_allocation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/bitrate_allocation.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..2d7f21bc1e941ca49bf85d0c7d36ff3630a16517
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/bitrate_allocation.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef API_CALL_BITRATE_ALLOCATION_H_
>+#define API_CALL_BITRATE_ALLOCATION_H_
>+
>+#include "api/units/data_rate.h"
>+#include "api/units/time_delta.h"
>+
>+namespace webrtc {
>+
>+// BitrateAllocationUpdate provides information to allocated streams about their
>+// bitrate allocation. It originates from the BitrateAllocater class and is
>+// propagated from there.
>+struct BitrateAllocationUpdate {
>+  // The allocated target bitrate. Media streams should produce this amount of
>+  // data. (Note that this may include packet overhead depending on
>+  // configuration.)
>+  DataRate target_bitrate = DataRate::Zero();
>+  // The allocated part of the estimated link capacity. This is more stable than
>+  // the target as it is based on the underlying link capacity estimate. This
>+  // should be used to change encoder configuration when the cost of change is
>+  // high.
>+  DataRate link_capacity = DataRate::Zero();
>+  // Predicted packet loss ratio.
>+  double packet_loss_ratio = 0;
>+  // Predicted round trip time.
>+  TimeDelta round_trip_time = TimeDelta::PlusInfinity();
>+  // |bwe_period| is deprecated, use the link capacity allocation instead.
>+  TimeDelta bwe_period = TimeDelta::PlusInfinity();
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_CALL_BITRATE_ALLOCATION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.cc
>index 0a9dd5bcc7e15851b448fc27811eb4b5e1707dbb..bcadc762de4d1934f4083dac18f25ab8dda237b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.cc
>@@ -10,6 +10,8 @@
> 
> #include "api/call/transport.h"
> 
>+#include <cstdint>
>+
> namespace webrtc {
> 
> PacketOptions::PacketOptions() = default;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.h
>index 18d22705cd6df915c1c6234efa84cf3b59e99796..32e5ddf874fb1c4cc10b5f1bfd913ede31151872 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.h
>@@ -32,6 +32,8 @@ struct PacketOptions {
>   std::vector<uint8_t> application_data;
>   // Whether this is a retransmission of an earlier packet.
>   bool is_retransmit = false;
>+  bool included_in_feedback = false;
>+  bool included_in_allocation = false;
> };
> 
> class Transport {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.cc
>index c2c6c530172cce1b5e19e50cffa601ab807285f3..275b17366c21cdcd0e7d4e5ea3c7808ee5473ca2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.cc
>@@ -10,6 +10,8 @@
> 
> #include "api/candidate.h"
> 
>+#include "rtc_base/helpers.h"
>+#include "rtc_base/ipaddress.h"
> #include "rtc_base/strings/string_builder.h"
> 
> namespace cricket {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.h
>index 6e0547b5e2fd84e8df013887f50adfd4170440ff..4c650d9bdbd52d3c34de0f35406d17b802cfac08 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.h
>@@ -18,16 +18,16 @@
> #include <string>
> 
> #include "rtc_base/checks.h"
>-#include "rtc_base/helpers.h"
> #include "rtc_base/network_constants.h"
> #include "rtc_base/socketaddress.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace cricket {
> 
> // Candidate for ICE based connection discovery.
> // TODO(phoglund): remove things in here that are not needed in the public API.
> 
>-class Candidate {
>+class RTC_EXPORT Candidate {
>  public:
>   Candidate();
>   // TODO(pthatcher): Match the ordering and param list as per RFC 5245
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/create_peerconnection_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/create_peerconnection_factory.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..1a6d086f6600d17c15c069da86220c29578d70f6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/create_peerconnection_factory.cc
>@@ -0,0 +1,182 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/create_peerconnection_factory.h"
>+
>+#include <memory>
>+#include <utility>
>+
>+#include "api/call/callfactoryinterface.h"
>+#include "api/peerconnectioninterface.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+#include "logging/rtc_event_log/rtc_event_log_factory.h"
>+#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
>+#include "media/engine/webrtcmediaengine.h"
>+#include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "rtc_base/bind.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/thread.h"
>+
>+namespace webrtc {
>+
>+#if defined(USE_BUILTIN_SW_CODECS)
>+rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory) {
>+  return CreatePeerConnectionFactoryWithAudioMixer(
>+      nullptr /*network_thread*/, nullptr /*worker_thread*/,
>+      nullptr /*signaling_thread*/, nullptr /*default_adm*/,
>+      audio_encoder_factory, audio_decoder_factory,
>+      nullptr /*video_encoder_factory*/, nullptr /*video_decoder_factory*/,
>+      nullptr /*audio_mixer*/);
>+}
>+
>+// Note: all the other CreatePeerConnectionFactory variants just end up calling
>+// this, ultimately.
>+rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    AudioDeviceModule* default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>+    rtc::scoped_refptr<AudioMixer> audio_mixer,
>+    rtc::scoped_refptr<AudioProcessing> audio_processing) {
>+  rtc::scoped_refptr<AudioProcessing> audio_processing_use = audio_processing;
>+  if (!audio_processing_use) {
>+    audio_processing_use = AudioProcessingBuilder().Create();
>+  }
>+
>+  std::unique_ptr<cricket::MediaEngineInterface> media_engine(
>+      cricket::WebRtcMediaEngineFactory::Create(
>+          default_adm, audio_encoder_factory, audio_decoder_factory,
>+          video_encoder_factory, video_decoder_factory, audio_mixer,
>+          audio_processing_use));
>+
>+  std::unique_ptr<CallFactoryInterface> call_factory = CreateCallFactory();
>+
>+  std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory =
>+      CreateRtcEventLogFactory();
>+
>+  return CreateModularPeerConnectionFactory(
>+      network_thread, worker_thread, signaling_thread, std::move(media_engine),
>+      std::move(call_factory), std::move(event_log_factory));
>+}
>+
>+rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    AudioDeviceModule* default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>+    rtc::scoped_refptr<AudioMixer> audio_mixer,
>+    rtc::scoped_refptr<AudioProcessing> audio_processing,
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
>+    std::unique_ptr<NetworkControllerFactoryInterface>
>+        network_controller_factory) {
>+  rtc::scoped_refptr<AudioProcessing> audio_processing_use = audio_processing;
>+  if (!audio_processing_use) {
>+    audio_processing_use = AudioProcessingBuilder().Create();
>+  }
>+
>+  std::unique_ptr<cricket::MediaEngineInterface> media_engine(
>+      cricket::WebRtcMediaEngineFactory::Create(
>+          default_adm, audio_encoder_factory, audio_decoder_factory,
>+          video_encoder_factory, video_decoder_factory, audio_mixer,
>+          audio_processing_use));
>+
>+  std::unique_ptr<CallFactoryInterface> call_factory = CreateCallFactory();
>+
>+  std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory =
>+      CreateRtcEventLogFactory();
>+
>+  return CreateModularPeerConnectionFactory(
>+      network_thread, worker_thread, signaling_thread, std::move(media_engine),
>+      std::move(call_factory), std::move(event_log_factory),
>+      std::move(fec_controller_factory), std::move(network_controller_factory));
>+}
>+#endif
>+
>+rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    rtc::scoped_refptr<AudioDeviceModule> default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    std::unique_ptr<VideoEncoderFactory> video_encoder_factory,
>+    std::unique_ptr<VideoDecoderFactory> video_decoder_factory,
>+    rtc::scoped_refptr<AudioMixer> audio_mixer,
>+    rtc::scoped_refptr<AudioProcessing> audio_processing) {
>+  if (!audio_processing)
>+    audio_processing = AudioProcessingBuilder().Create();
>+
>+  std::unique_ptr<cricket::MediaEngineInterface> media_engine =
>+      cricket::WebRtcMediaEngineFactory::Create(
>+          default_adm, audio_encoder_factory, audio_decoder_factory,
>+          std::move(video_encoder_factory), std::move(video_decoder_factory),
>+          audio_mixer, audio_processing);
>+
>+  std::unique_ptr<CallFactoryInterface> call_factory = CreateCallFactory();
>+
>+  std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory =
>+      CreateRtcEventLogFactory();
>+  PeerConnectionFactoryDependencies dependencies;
>+  dependencies.network_thread = network_thread;
>+  dependencies.worker_thread = worker_thread;
>+  dependencies.signaling_thread = signaling_thread;
>+  dependencies.media_engine = std::move(media_engine);
>+  dependencies.call_factory = std::move(call_factory);
>+  dependencies.event_log_factory = std::move(event_log_factory);
>+  return CreateModularPeerConnectionFactory(std::move(dependencies));
>+}
>+
>+#if defined(USE_BUILTIN_SW_CODECS)
>+rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreatePeerConnectionFactoryWithAudioMixer(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    AudioDeviceModule* default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>+    rtc::scoped_refptr<AudioMixer> audio_mixer) {
>+  return CreatePeerConnectionFactory(
>+      network_thread, worker_thread, signaling_thread, default_adm,
>+      audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
>+      video_decoder_factory, audio_mixer, nullptr);
>+}
>+
>+rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    AudioDeviceModule* default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory) {
>+  return CreatePeerConnectionFactoryWithAudioMixer(
>+      network_thread, worker_thread, signaling_thread, default_adm,
>+      audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
>+      video_decoder_factory, nullptr);
>+}
>+#endif
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/create_peerconnection_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/create_peerconnection_factory.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..baa50c7bd8063d562395e01f7f2330212efaf4e1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/create_peerconnection_factory.h
>@@ -0,0 +1,179 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_CREATE_PEERCONNECTION_FACTORY_H_
>+#define API_CREATE_PEERCONNECTION_FACTORY_H_
>+
>+#include <memory>
>+
>+#include "api/audio/audio_mixer.h"
>+#include "api/audio_codecs/audio_decoder_factory.h"
>+#include "api/audio_codecs/audio_encoder_factory.h"
>+#include "api/fec_controller.h"
>+#include "api/peerconnectioninterface.h"
>+#include "api/transport/network_control.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+
>+namespace rtc {
>+// TODO(bugs.webrtc.org/9987): Move rtc::Thread to api/ or expose a better
>+// type. At the moment, rtc::Thread is not part of api/ so it cannot be
>+// included in order to avoid to leak internal types.
>+class Thread;
>+}  // namespace rtc
>+
>+namespace cricket {
>+class WebRtcVideoDecoderFactory;
>+class WebRtcVideoEncoderFactory;
>+}  // namespace cricket
>+
>+namespace webrtc {
>+
>+class AudioDeviceModule;
>+class AudioProcessing;
>+
>+#if defined(USE_BUILTIN_SW_CODECS)
>+// Create a new instance of PeerConnectionFactoryInterface.
>+//
>+// This method relies on the thread it's called on as the "signaling thread"
>+// for the PeerConnectionFactory it creates.
>+//
>+// As such, if the current thread is not already running an rtc::Thread message
>+// loop, an application using this method must eventually either call
>+// rtc::Thread::Current()->Run(), or call
>+// rtc::Thread::Current()->ProcessMessages() within the application's own
>+// message loop.
>+RTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreatePeerConnectionFactory(
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory);
>+
>+// Create a new instance of PeerConnectionFactoryInterface.
>+//
>+// |network_thread|, |worker_thread| and |signaling_thread| are
>+// the only mandatory parameters.
>+//
>+// If non-null, a reference is added to |default_adm|, and ownership of
>+// |video_encoder_factory| and |video_decoder_factory| is transferred to the
>+// returned factory.
>+// TODO(deadbeef): Use rtc::scoped_refptr<> and std::unique_ptr<> to make this
>+// ownership transfer and ref counting more obvious.
>+RTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreatePeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    AudioDeviceModule* default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
>+
>+// Create a new instance of PeerConnectionFactoryInterface with optional
>+// external audio mixed and audio processing modules.
>+//
>+// If |audio_mixer| is null, an internal audio mixer will be created and used.
>+// If |audio_processing| is null, an internal audio processing module will be
>+// created and used.
>+RTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreatePeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    AudioDeviceModule* default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>+    rtc::scoped_refptr<AudioMixer> audio_mixer,
>+    rtc::scoped_refptr<AudioProcessing> audio_processing);
>+
>+// Create a new instance of PeerConnectionFactoryInterface with optional
>+// external audio mixer, audio processing, and fec controller modules.
>+//
>+// If |audio_mixer| is null, an internal audio mixer will be created and used.
>+// If |audio_processing| is null, an internal audio processing module will be
>+// created and used.
>+// If |fec_controller_factory| is null, an internal fec controller module will
>+// be created and used.
>+// If |network_controller_factory| is provided, it will be used if enabled via
>+// field trial.
>+RTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreatePeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    AudioDeviceModule* default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>+    rtc::scoped_refptr<AudioMixer> audio_mixer,
>+    rtc::scoped_refptr<AudioProcessing> audio_processing,
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
>+    std::unique_ptr<NetworkControllerFactoryInterface>
>+        network_controller_factory = nullptr);
>+#endif  // defined(USE_BUILTIN_SW_CODECS)
>+
>+// Create a new instance of PeerConnectionFactoryInterface with optional video
>+// codec factories. These video factories represents all video codecs, i.e. no
>+// extra internal video codecs will be added.
>+// When building WebRTC with rtc_use_builtin_sw_codecs = false, this is the
>+// only available CreatePeerConnectionFactory overload.
>+RTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreatePeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    rtc::scoped_refptr<AudioDeviceModule> default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    std::unique_ptr<VideoEncoderFactory> video_encoder_factory,
>+    std::unique_ptr<VideoDecoderFactory> video_decoder_factory,
>+    rtc::scoped_refptr<AudioMixer> audio_mixer,
>+    rtc::scoped_refptr<AudioProcessing> audio_processing);
>+
>+#if defined(USE_BUILTIN_SW_CODECS)
>+// Create a new instance of PeerConnectionFactoryInterface with external audio
>+// mixer.
>+//
>+// If |audio_mixer| is null, an internal audio mixer will be created and used.
>+RTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreatePeerConnectionFactoryWithAudioMixer(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    AudioDeviceModule* default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>+    rtc::scoped_refptr<AudioMixer> audio_mixer);
>+
>+// Create a new instance of PeerConnectionFactoryInterface.
>+// Same thread is used as worker and network thread.
>+RTC_EXPORT inline rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreatePeerConnectionFactory(
>+    rtc::Thread* worker_and_network_thread,
>+    rtc::Thread* signaling_thread,
>+    AudioDeviceModule* default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory) {
>+  return CreatePeerConnectionFactory(
>+      worker_and_network_thread, worker_and_network_thread, signaling_thread,
>+      default_adm, audio_encoder_factory, audio_decoder_factory,
>+      video_encoder_factory, video_decoder_factory);
>+}
>+#endif  // defined(USE_BUILTIN_SW_CODECS)
>+
>+}  // namespace webrtc
>+
>+#endif  // API_CREATE_PEERCONNECTION_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/crypto/cryptooptions.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/crypto/cryptooptions.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..7f34f19b5385722dc2abaefc9b52f1d06ffb4350
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/crypto/cryptooptions.cc
>@@ -0,0 +1,78 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/crypto/cryptooptions.h"
>+#include "rtc_base/sslstreamadapter.h"
>+
>+namespace webrtc {
>+
>+CryptoOptions::CryptoOptions() {}
>+
>+CryptoOptions::CryptoOptions(const CryptoOptions& other) {
>+  srtp = other.srtp;
>+  sframe = other.sframe;
>+}
>+
>+CryptoOptions::~CryptoOptions() {}
>+
>+// static
>+CryptoOptions CryptoOptions::NoGcm() {
>+  CryptoOptions options;
>+  options.srtp.enable_gcm_crypto_suites = false;
>+  return options;
>+}
>+
>+std::vector<int> CryptoOptions::GetSupportedDtlsSrtpCryptoSuites() const {
>+  std::vector<int> crypto_suites;
>+  if (srtp.enable_gcm_crypto_suites) {
>+    crypto_suites.push_back(rtc::SRTP_AEAD_AES_256_GCM);
>+    crypto_suites.push_back(rtc::SRTP_AEAD_AES_128_GCM);
>+  }
>+  // Note: SRTP_AES128_CM_SHA1_80 is what is required to be supported (by
>+  // draft-ietf-rtcweb-security-arch), but SRTP_AES128_CM_SHA1_32 is allowed as
>+  // well, and saves a few bytes per packet if it ends up selected.
>+  // As the cipher suite is potentially insecure, it will only be used if
>+  // enabled by both peers.
>+  if (srtp.enable_aes128_sha1_32_crypto_cipher) {
>+    crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_32);
>+  }
>+  crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_80);
>+  return crypto_suites;
>+}
>+
>+bool CryptoOptions::operator==(const CryptoOptions& other) const {
>+  struct data_being_tested_for_equality {
>+    struct Srtp {
>+      bool enable_gcm_crypto_suites;
>+      bool enable_aes128_sha1_32_crypto_cipher;
>+      bool enable_encrypted_rtp_header_extensions;
>+    } srtp;
>+    struct SFrame {
>+      bool require_frame_encryption;
>+    } sframe;
>+  };
>+  static_assert(sizeof(data_being_tested_for_equality) == sizeof(*this),
>+                "Did you add something to CryptoOptions and forget to "
>+                "update operator==?");
>+
>+  return srtp.enable_gcm_crypto_suites == other.srtp.enable_gcm_crypto_suites &&
>+         srtp.enable_aes128_sha1_32_crypto_cipher ==
>+             other.srtp.enable_aes128_sha1_32_crypto_cipher &&
>+         srtp.enable_encrypted_rtp_header_extensions ==
>+             other.srtp.enable_encrypted_rtp_header_extensions &&
>+         sframe.require_frame_encryption ==
>+             other.sframe.require_frame_encryption;
>+}
>+
>+bool CryptoOptions::operator!=(const CryptoOptions& other) const {
>+  return !(*this == other);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/crypto/cryptooptions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/crypto/cryptooptions.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..e730ab208959800702857d9f469c94d659a8fb9d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/crypto/cryptooptions.h
>@@ -0,0 +1,66 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_CRYPTO_CRYPTOOPTIONS_H_
>+#define API_CRYPTO_CRYPTOOPTIONS_H_
>+
>+#include <vector>
>+
>+namespace webrtc {
>+
>+// CryptoOptions defines advanced cryptographic settings for native WebRTC.
>+// These settings must be passed into PeerConnectionFactoryInterface::Options
>+// and are only applicable to native use cases of WebRTC.
>+struct CryptoOptions {
>+  CryptoOptions();
>+  CryptoOptions(const CryptoOptions& other);
>+  ~CryptoOptions();
>+
>+  // Helper method to return an instance of the CryptoOptions with GCM crypto
>+  // suites disabled. This method should be used instead of depending on current
>+  // default values set by the constructor.
>+  static CryptoOptions NoGcm();
>+
>+  // Returns a list of the supported DTLS-SRTP Crypto suites based on this set
>+  // of crypto options.
>+  std::vector<int> GetSupportedDtlsSrtpCryptoSuites() const;
>+
>+  bool operator==(const CryptoOptions& other) const;
>+  bool operator!=(const CryptoOptions& other) const;
>+
>+  // SRTP Related Peer Connection options.
>+  struct Srtp {
>+    // Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
>+    // if both sides enable it.
>+    bool enable_gcm_crypto_suites = false;
>+
>+    // If set to true, the (potentially insecure) crypto cipher
>+    // SRTP_AES128_CM_SHA1_32 will be included in the list of supported ciphers
>+    // during negotiation. It will only be used if both peers support it and no
>+    // other ciphers get preferred.
>+    bool enable_aes128_sha1_32_crypto_cipher = false;
>+
>+    // If set to true, encrypted RTP header extensions as defined in RFC 6904
>+    // will be negotiated. They will only be used if both peers support them.
>+    bool enable_encrypted_rtp_header_extensions = false;
>+  } srtp;
>+
>+  // Options to be used when the FrameEncryptor / FrameDecryptor APIs are used.
>+  struct SFrame {
>+    // If set all RtpSenders must have an FrameEncryptor attached to them before
>+    // they are allowed to send packets. All RtpReceivers must have a
>+    // FrameDecryptor attached to them before they are able to receive packets.
>+    bool require_frame_encryption = false;
>+  } sframe;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_CRYPTO_CRYPTOOPTIONS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/cryptoparams.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/cryptoparams.h
>index 2350528358f26e012b30fa160576c21294e325a3..abe905546229c8327e2d7a9c685f8ee44698db20 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/cryptoparams.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/cryptoparams.h
>@@ -16,6 +16,8 @@
> namespace cricket {
> 
> // Parameters for SRTP negotiation, as described in RFC 4568.
>+// TODO(benwright) - Rename to SrtpCryptoParams as these only apply to SRTP and
>+// not generic crypto parameters for WebRTC.
> struct CryptoParams {
>   CryptoParams() : tag(0) {}
>   CryptoParams(int t,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>index a0d2b3b51a59858b1ba84fdeae8b69133d19cce9..7cb55822337f8045b3f0998546213f1ae1033dc8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>@@ -14,6 +14,8 @@
> #ifndef API_DATACHANNELINTERFACE_H_
> #define API_DATACHANNELINTERFACE_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <string>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.cc
>index 52a60f95e32e90ad2196b671d837c2a54718e08a..01f5720563d908347f5dcb5f4381b9b21c4bb98b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.cc
>@@ -38,4 +38,32 @@ void SetSessionDescriptionObserver::OnFailure(const std::string& error) {
>   OnFailure(RTCError(RTCErrorType::INTERNAL_ERROR, std::string(error)));
> }
> 
>+const char SessionDescriptionInterface::kOffer[] = "offer";
>+const char SessionDescriptionInterface::kPrAnswer[] = "pranswer";
>+const char SessionDescriptionInterface::kAnswer[] = "answer";
>+
>+const char* SdpTypeToString(SdpType type) {
>+  switch (type) {
>+    case SdpType::kOffer:
>+      return SessionDescriptionInterface::kOffer;
>+    case SdpType::kPrAnswer:
>+      return SessionDescriptionInterface::kPrAnswer;
>+    case SdpType::kAnswer:
>+      return SessionDescriptionInterface::kAnswer;
>+  }
>+  return "";
>+}
>+
>+absl::optional<SdpType> SdpTypeFromString(const std::string& type_str) {
>+  if (type_str == SessionDescriptionInterface::kOffer) {
>+    return SdpType::kOffer;
>+  } else if (type_str == SessionDescriptionInterface::kPrAnswer) {
>+    return SdpType::kPrAnswer;
>+  } else if (type_str == SessionDescriptionInterface::kAnswer) {
>+    return SdpType::kAnswer;
>+  } else {
>+    return absl::nullopt;
>+  }
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>index 4d4bcc0bfb62ed034b978689bddac7c3b4826266..1c50455d04df9764dfbfa77728b205e5de768043 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>@@ -29,6 +29,7 @@
> #include "absl/types/optional.h"
> #include "api/rtcerror.h"
> #include "rtc_base/refcount.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace cricket {
> class Candidate;
>@@ -73,13 +74,13 @@ class IceCandidateInterface {
> // Creates a IceCandidateInterface based on SDP string.
> // Returns null if the sdp string can't be parsed.
> // |error| may be null.
>-IceCandidateInterface* CreateIceCandidate(const std::string& sdp_mid,
>-                                          int sdp_mline_index,
>-                                          const std::string& sdp,
>-                                          SdpParseError* error);
>+RTC_EXPORT IceCandidateInterface* CreateIceCandidate(const std::string& sdp_mid,
>+                                                     int sdp_mline_index,
>+                                                     const std::string& sdp,
>+                                                     SdpParseError* error);
> 
> // Creates an IceCandidateInterface based on a parsed candidate structure.
>-std::unique_ptr<IceCandidateInterface> CreateIceCandidate(
>+RTC_EXPORT std::unique_ptr<IceCandidateInterface> CreateIceCandidate(
>     const std::string& sdp_mid,
>     int sdp_mline_index,
>     const cricket::Candidate& candidate);
>@@ -121,7 +122,7 @@ absl::optional<SdpType> SdpTypeFromString(const std::string& type_str);
> // and is therefore not expected to be thread safe.
> //
> // An instance can be created by CreateSessionDescription.
>-class SessionDescriptionInterface {
>+class RTC_EXPORT SessionDescriptionInterface {
>  public:
>   // String representations of the supported SDP types.
>   static const char kOffer[];
>@@ -181,21 +182,21 @@ class SessionDescriptionInterface {
> // |error| may be null.
> // TODO(steveanton): This function is deprecated. Please use the functions below
> // which take an SdpType enum instead. Remove this once it is no longer used.
>-SessionDescriptionInterface* CreateSessionDescription(const std::string& type,
>-                                                      const std::string& sdp,
>-                                                      SdpParseError* error);
>+RTC_EXPORT SessionDescriptionInterface* CreateSessionDescription(
>+    const std::string& type,
>+    const std::string& sdp,
>+    SdpParseError* error);
> 
> // Creates a SessionDescriptionInterface based on the SDP string and the type.
> // Returns null if the SDP string cannot be parsed.
> // If using the signature with |error_out|, details of the parsing error may be
> // written to |error_out| if it is not null.
>-std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
>-    SdpType type,
>-    const std::string& sdp);
>-std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
>-    SdpType type,
>-    const std::string& sdp,
>-    SdpParseError* error_out);
>+RTC_EXPORT std::unique_ptr<SessionDescriptionInterface>
>+CreateSessionDescription(SdpType type, const std::string& sdp);
>+RTC_EXPORT std::unique_ptr<SessionDescriptionInterface>
>+CreateSessionDescription(SdpType type,
>+                         const std::string& sdp,
>+                         SdpParseError* error_out);
> 
> // Creates a SessionDescriptionInterface based on a parsed SDP structure and the
> // given type, ID and version.
>@@ -206,7 +207,8 @@ std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
>     std::unique_ptr<cricket::SessionDescription> description);
> 
> // CreateOffer and CreateAnswer callback interface.
>-class CreateSessionDescriptionObserver : public rtc::RefCountInterface {
>+class RTC_EXPORT CreateSessionDescriptionObserver
>+    : public rtc::RefCountInterface {
>  public:
>   // This callback transfers the ownership of the |desc|.
>   // TODO(deadbeef): Make this take an std::unique_ptr<> to avoid confusion
>@@ -227,7 +229,7 @@ class CreateSessionDescriptionObserver : public rtc::RefCountInterface {
> };
> 
> // SetLocalDescription and SetRemoteDescription callback interface.
>-class SetSessionDescriptionObserver : public rtc::RefCountInterface {
>+class RTC_EXPORT SetSessionDescriptionObserver : public rtc::RefCountInterface {
>  public:
>   virtual void OnSuccess() = 0;
>   // See description in CreateSessionDescriptionObserver for OnFailure.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.cc
>index b9ba2fea29aaaad62c0bcf832ba8683fc702ec04..ed2f7927f272b34de36d3dc03f0c630041edaa3e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.cc
>@@ -10,6 +10,9 @@
> 
> #include "api/jsepicecandidate.h"
> 
>+#include <algorithm>
>+#include <utility>
>+
> namespace webrtc {
> 
> std::string JsepIceCandidate::sdp_mid() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.h
>index 50520fe7279b2a6696c05d3bfcfea001df027951..9cc7443f3fbbdc784fd7b51bfdba9c1871b438f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.h
>@@ -14,8 +14,8 @@
> #ifndef API_JSEPICECANDIDATE_H_
> #define API_JSEPICECANDIDATE_H_
> 
>+#include <stddef.h>
> #include <string>
>-#include <utility>
> #include <vector>
> 
> #include "api/candidate.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/media_transport_interface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/media_transport_interface.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..ef223aa788022e7a79f8c9870b7ba8db51a82088
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/media_transport_interface.cc
>@@ -0,0 +1,124 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// This is EXPERIMENTAL interface for media transport.
>+//
>+// The goal is to refactor WebRTC code so that audio and video frames
>+// are sent / received through the media transport interface. This will
>+// enable different media transport implementations, including QUIC-based
>+// media transport.
>+
>+#include "api/media_transport_interface.h"
>+
>+#include <cstdint>
>+#include <utility>
>+
>+namespace webrtc {
>+
>+MediaTransportSettings::MediaTransportSettings() = default;
>+MediaTransportSettings::MediaTransportSettings(const MediaTransportSettings&) =
>+    default;
>+MediaTransportSettings& MediaTransportSettings::operator=(
>+    const MediaTransportSettings&) = default;
>+MediaTransportSettings::~MediaTransportSettings() = default;
>+
>+MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}
>+
>+MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
>+    int sampling_rate_hz,
>+    int starting_sample_index,
>+    int samples_per_channel,
>+    int sequence_number,
>+    FrameType frame_type,
>+    uint8_t payload_type,
>+    std::vector<uint8_t> encoded_data)
>+    : sampling_rate_hz_(sampling_rate_hz),
>+      starting_sample_index_(starting_sample_index),
>+      samples_per_channel_(samples_per_channel),
>+      sequence_number_(sequence_number),
>+      frame_type_(frame_type),
>+      payload_type_(payload_type),
>+      encoded_data_(std::move(encoded_data)) {}
>+
>+MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
>+    const MediaTransportEncodedAudioFrame&) = default;
>+
>+MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
>+    MediaTransportEncodedAudioFrame&&) = default;
>+
>+MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
>+    const MediaTransportEncodedAudioFrame&) = default;
>+
>+MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
>+    MediaTransportEncodedAudioFrame&&) = default;
>+
>+MediaTransportEncodedVideoFrame::~MediaTransportEncodedVideoFrame() {}
>+
>+MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
>+    int64_t frame_id,
>+    std::vector<int64_t> referenced_frame_ids,
>+    VideoCodecType codec_type,
>+    const webrtc::EncodedImage& encoded_image)
>+    : codec_type_(codec_type),
>+      encoded_image_(encoded_image),
>+      frame_id_(frame_id),
>+      referenced_frame_ids_(std::move(referenced_frame_ids)) {}
>+
>+MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
>+    const MediaTransportEncodedVideoFrame&) = default;
>+
>+MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
>+    MediaTransportEncodedVideoFrame&&) = default;
>+
>+MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
>+    const MediaTransportEncodedVideoFrame&) = default;
>+
>+MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
>+    MediaTransportEncodedVideoFrame&&) = default;
>+
>+SendDataParams::SendDataParams() = default;
>+
>+RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
>+MediaTransportFactory::CreateMediaTransport(
>+    rtc::PacketTransportInternal* packet_transport,
>+    rtc::Thread* network_thread,
>+    bool is_caller) {
>+  MediaTransportSettings settings;
>+  settings.is_caller = is_caller;
>+  return CreateMediaTransport(packet_transport, network_thread, settings);
>+}
>+
>+RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
>+MediaTransportFactory::CreateMediaTransport(
>+    rtc::PacketTransportInternal* packet_transport,
>+    rtc::Thread* network_thread,
>+    const MediaTransportSettings& settings) {
>+  return std::unique_ptr<MediaTransportInterface>(nullptr);
>+}
>+
>+absl::optional<TargetTransferRate>
>+MediaTransportInterface::GetLatestTargetTransferRate() {
>+  return absl::nullopt;
>+}
>+
>+void MediaTransportInterface::SetNetworkChangeCallback(
>+    MediaTransportNetworkChangeCallback* callback) {}
>+
>+void MediaTransportInterface::RemoveTargetTransferRateObserver(
>+    webrtc::TargetTransferRateObserver* observer) {}
>+
>+void MediaTransportInterface::AddTargetTransferRateObserver(
>+    webrtc::TargetTransferRateObserver* observer) {}
>+
>+size_t MediaTransportInterface::GetAudioPacketOverhead() const {
>+  return 0;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/media_transport_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/media_transport_interface.h
>index 2c88cd4ce6a1502a472e955dbc439db521dd2bf2..b10dd63a465c45a982f8589565dee98d23abea5c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/media_transport_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/media_transport_interface.h
>@@ -1,5 +1,4 @@
>-/*
>- *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+/* Copyright 2018 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -18,12 +17,18 @@
> #ifndef API_MEDIA_TRANSPORT_INTERFACE_H_
> #define API_MEDIA_TRANSPORT_INTERFACE_H_
> 
>+#include <api/transport/network_control.h>
> #include <memory>
>+#include <string>
> #include <utility>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
> #include "api/rtcerror.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "api/video/encoded_image.h"
>+#include "rtc_base/copyonwritebuffer.h"
>+#include "rtc_base/networkroute.h"
> 
> namespace rtc {
> class PacketTransportInternal;
>@@ -32,16 +37,36 @@ class Thread;
> 
> namespace webrtc {
> 
>+// A collection of settings for creation of media transport.
>+struct MediaTransportSettings final {
>+  MediaTransportSettings();
>+  MediaTransportSettings(const MediaTransportSettings&);
>+  MediaTransportSettings& operator=(const MediaTransportSettings&);
>+  ~MediaTransportSettings();
>+
>+  // Group calls are not currently supported, in 1:1 call one side must set
>+  // is_caller = true and another is_caller = false.
>+  bool is_caller;
>+
>+  // Must be set if a pre-shared key is used for the call.
>+  // TODO(bugs.webrtc.org/9944): This should become zero buffer in the distant
>+  // future.
>+  absl::optional<std::string> pre_shared_key;
>+};
>+
> // Represents encoded audio frame in any encoding (type of encoding is opaque).
> // To avoid copying of encoded data use move semantics when passing by value.
>-class MediaTransportEncodedAudioFrame {
>+class MediaTransportEncodedAudioFrame final {
>  public:
>   enum class FrameType {
>     // Normal audio frame (equivalent to webrtc::kAudioFrameSpeech).
>     kSpeech,
> 
>     // DTX frame (equivalent to webrtc::kAudioFrameCN).
>-    kDiscountinuousTransmission,
>+    // DTX frame (equivalent to webrtc::kAudioFrameCN).
>+    kDiscontinuousTransmission,
>+    // TODO(nisse): Mis-spelled version, update users, then delete.
>+    kDiscountinuousTransmission = kDiscontinuousTransmission,
>   };
> 
>   MediaTransportEncodedAudioFrame(
>@@ -75,14 +100,15 @@ class MediaTransportEncodedAudioFrame {
>       uint8_t payload_type,
> 
>       // Vector with opaque encoded data.
>-      std::vector<uint8_t> encoded_data)
>-      : sampling_rate_hz_(sampling_rate_hz),
>-        starting_sample_index_(starting_sample_index),
>-        samples_per_channel_(samples_per_channel),
>-        sequence_number_(sequence_number),
>-        frame_type_(frame_type),
>-        payload_type_(payload_type),
>-        encoded_data_(std::move(encoded_data)) {}
>+      std::vector<uint8_t> encoded_data);
>+
>+  ~MediaTransportEncodedAudioFrame();
>+  MediaTransportEncodedAudioFrame(const MediaTransportEncodedAudioFrame&);
>+  MediaTransportEncodedAudioFrame& operator=(
>+      const MediaTransportEncodedAudioFrame& other);
>+  MediaTransportEncodedAudioFrame& operator=(
>+      MediaTransportEncodedAudioFrame&& other);
>+  MediaTransportEncodedAudioFrame(MediaTransportEncodedAudioFrame&&);
> 
>   // Getters.
>   int sampling_rate_hz() const { return sampling_rate_hz_; }
>@@ -113,6 +139,16 @@ class MediaTransportEncodedAudioFrame {
>   std::vector<uint8_t> encoded_data_;
> };
> 
>+// Callback to notify about network route changes.
>+class MediaTransportNetworkChangeCallback {
>+ public:
>+  virtual ~MediaTransportNetworkChangeCallback() = default;
>+
>+  // Called when the network route is changed, with the new network route.
>+  virtual void OnNetworkRouteChanged(
>+      const rtc::NetworkRoute& new_network_route) = 0;
>+};
>+
> // Interface for receiving encoded audio frames from MediaTransportInterface
> // implementations.
> class MediaTransportAudioSinkInterface {
>@@ -125,16 +161,19 @@ class MediaTransportAudioSinkInterface {
> };
> 
> // Represents encoded video frame, along with the codec information.
>-class MediaTransportEncodedVideoFrame {
>+class MediaTransportEncodedVideoFrame final {
>  public:
>   MediaTransportEncodedVideoFrame(int64_t frame_id,
>                                   std::vector<int64_t> referenced_frame_ids,
>                                   VideoCodecType codec_type,
>-                                  const webrtc::EncodedImage& encoded_image)
>-      : codec_type_(codec_type),
>-        encoded_image_(encoded_image),
>-        frame_id_(frame_id),
>-        referenced_frame_ids_(std::move(referenced_frame_ids)) {}
>+                                  const webrtc::EncodedImage& encoded_image);
>+  ~MediaTransportEncodedVideoFrame();
>+  MediaTransportEncodedVideoFrame(const MediaTransportEncodedVideoFrame&);
>+  MediaTransportEncodedVideoFrame& operator=(
>+      const MediaTransportEncodedVideoFrame& other);
>+  MediaTransportEncodedVideoFrame& operator=(
>+      MediaTransportEncodedVideoFrame&& other);
>+  MediaTransportEncodedVideoFrame(MediaTransportEncodedVideoFrame&&);
> 
>   VideoCodecType codec_type() const { return codec_type_; }
>   const webrtc::EncodedImage& encoded_image() const { return encoded_image_; }
>@@ -183,6 +222,85 @@ class MediaTransportVideoSinkInterface {
>   virtual void OnKeyFrameRequested(uint64_t channel_id) = 0;
> };
> 
>+// State of the media transport.  Media transport begins in the pending state.
>+// It transitions to writable when it is ready to send media.  It may transition
>+// back to pending if the connection is blocked.  It may transition to closed at
>+// any time.  Closed is terminal: a transport will never re-open once closed.
>+enum class MediaTransportState {
>+  kPending,
>+  kWritable,
>+  kClosed,
>+};
>+
>+// Callback invoked whenever the state of the media transport changes.
>+class MediaTransportStateCallback {
>+ public:
>+  virtual ~MediaTransportStateCallback() = default;
>+
>+  // Invoked whenever the state of the media transport changes.
>+  virtual void OnStateChanged(MediaTransportState state) = 0;
>+};
>+
>+// Supported types of application data messages.
>+enum class DataMessageType {
>+  // Application data buffer with the binary bit unset.
>+  kText,
>+
>+  // Application data buffer with the binary bit set.
>+  kBinary,
>+
>+  // Transport-agnostic control messages, such as open or open-ack messages.
>+  kControl,
>+};
>+
>+// Parameters for sending data.  The parameters may change from message to
>+// message, even within a single channel.  For example, control messages may be
>+// sent reliably and in-order, even if the data channel is configured for
>+// unreliable delivery.
>+struct SendDataParams {
>+  SendDataParams();
>+
>+  DataMessageType type = DataMessageType::kText;
>+
>+  // Whether to deliver the message in order with respect to other ordered
>+  // messages with the same channel_id.
>+  bool ordered = false;
>+
>+  // If set, the maximum number of times this message may be
>+  // retransmitted by the transport before it is dropped.
>+  // Setting this value to zero disables retransmission.
>+  // Must be non-negative. |max_rtx_count| and |max_rtx_ms| may not be set
>+  // simultaneously.
>+  absl::optional<int> max_rtx_count;
>+
>+  // If set, the maximum number of milliseconds for which the transport
>+  // may retransmit this message before it is dropped.
>+  // Setting this value to zero disables retransmission.
>+  // Must be non-negative. |max_rtx_count| and |max_rtx_ms| may not be set
>+  // simultaneously.
>+  absl::optional<int> max_rtx_ms;
>+};
>+
>+// Sink for callbacks related to a data channel.
>+class DataChannelSink {
>+ public:
>+  virtual ~DataChannelSink() = default;
>+
>+  // Callback issued when data is received by the transport.
>+  virtual void OnDataReceived(int channel_id,
>+                              DataMessageType type,
>+                              const rtc::CopyOnWriteBuffer& buffer) = 0;
>+
>+  // Callback issued when a remote data channel begins the closing procedure.
>+  // Messages sent after the closing procedure begins will not be transmitted.
>+  virtual void OnChannelClosing(int channel_id) = 0;
>+
>+  // Callback issued when a (remote or local) data channel completes the closing
>+  // procedure.  Closing channels become closed after all pending data has been
>+  // transmitted.
>+  virtual void OnChannelClosed(int channel_id) = 0;
>+};
>+
> // Media transport interface for sending / receiving encoded audio/video frames
> // and receiving bandwidth estimate update from congestion control.
> class MediaTransportInterface {
>@@ -217,8 +335,62 @@ class MediaTransportInterface {
>   // pass a nullptr.
>   virtual void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) = 0;
> 
>+  // Adds a target bitrate observer. Before media transport is destructed
>+  // the observer must be unregistered (by calling
>+  // RemoveTargetTransferRateObserver).
>+  // A newly registered observer will be called back with the latest recorded
>+  // target rate, if available.
>+  virtual void AddTargetTransferRateObserver(
>+      webrtc::TargetTransferRateObserver* observer);
>+
>+  // Removes an existing |observer| from observers. If observer was never
>+  // registered, an error is logged and method does nothing.
>+  virtual void RemoveTargetTransferRateObserver(
>+      webrtc::TargetTransferRateObserver* observer);
>+
>+  // Returns the last known target transfer rate as reported to the above
>+  // observers.
>+  virtual absl::optional<TargetTransferRate> GetLatestTargetTransferRate();
>+
>+  // Gets the audio packet overhead in bytes. Returned overhead does not include
>+  // transport overhead (ipv4/6, turn channeldata, tcp/udp, etc.).
>+  // If the transport is capable of fusing packets together, this overhead
>+  // might not be a very accurate number.
>+  virtual size_t GetAudioPacketOverhead() const;
>+
>+  // Sets an observer for network change events. If the network route is already
>+  // established when the callback is set, |callback| will be called immediately
>+  // with the current network route.
>+  // Before media transport is destroyed, the callback must be unregistered by
>+  // setting it to nullptr.
>+  virtual void SetNetworkChangeCallback(
>+      MediaTransportNetworkChangeCallback* callback);
>+
>+  // Sets a state observer callback. Before media transport is destroyed, the
>+  // callback must be unregistered by setting it to nullptr.
>+  // A newly registered callback will be called with the current state.
>+  // Media transport does not invoke this callback concurrently.
>+  virtual void SetMediaTransportStateCallback(
>+      MediaTransportStateCallback* callback) = 0;
>+
>+  // Sends a data buffer to the remote endpoint using the given send parameters.
>+  // |buffer| may not be larger than 256 KiB. Returns an error if the send
>+  // fails.
>+  virtual RTCError SendData(int channel_id,
>+                            const SendDataParams& params,
>+                            const rtc::CopyOnWriteBuffer& buffer) = 0;
>+
>+  // Closes |channel_id| gracefully.  Returns an error if |channel_id| is not
>+  // open.  Data sent after the closing procedure begins will not be
>+  // transmitted. The channel becomes closed after pending data is transmitted.
>+  virtual RTCError CloseChannel(int channel_id) = 0;
>+
>+  // Sets a sink for data messages and channel state callbacks. Before media
>+  // transport is destroyed, the sink must be unregistered by setting it to
>+  // nullptr.
>+  virtual void SetDataSink(DataChannelSink* sink) = 0;
>+
>   // TODO(sukhanov): RtcEventLogs.
>-  // TODO(sukhanov): Bandwidth updates.
> };
> 
> // If media transport factory is set in peer connection factory, it will be
>@@ -236,10 +408,21 @@ class MediaTransportFactory {
>   // - Does not take ownership of packet_transport or network_thread.
>   // - Does not support group calls, in 1:1 call one side must set
>   //   is_caller = true and another is_caller = false.
>+  // TODO(bugs.webrtc.org/9938) This constructor will be removed and replaced
>+  // with the one below.
>+  virtual RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
>+  CreateMediaTransport(rtc::PacketTransportInternal* packet_transport,
>+                       rtc::Thread* network_thread,
>+                       bool is_caller);
>+
>+  // Creates media transport.
>+  // - Does not take ownership of packet_transport or network_thread.
>+  // TODO(bugs.webrtc.org/9938): remove default implementation once all children
>+  // override it.
>   virtual RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
>   CreateMediaTransport(rtc::PacketTransportInternal* packet_transport,
>                        rtc::Thread* network_thread,
>-                       bool is_caller) = 0;
>+                       const MediaTransportSettings& settings);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.cc
>index 55677869bedd7c3212400b87cfae3277acc34d30..b66ba0503ac7f58f7170d675918de9a24c6a6992 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.cc
>@@ -10,7 +10,9 @@
> 
> #include "api/mediaconstraintsinterface.h"
> 
>+#include "absl/types/optional.h"
> #include "api/peerconnectioninterface.h"
>+#include "media/base/mediaconfig.h"
> #include "rtc_base/stringencode.h"
> 
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.h
>index c6a914aa564f9831b128ba3086baef7a580e397c..c9a6e1b18858df599a13b2cd2103565c05970d3d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.h
>@@ -21,10 +21,11 @@
> #ifndef API_MEDIACONSTRAINTSINTERFACE_H_
> #define API_MEDIACONSTRAINTSINTERFACE_H_
> 
>+#include <stddef.h>
> #include <string>
> #include <vector>
> 
>-#include "absl/types/optional.h"
>+#include "api/audio_options.h"
> #include "api/peerconnectioninterface.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.cc
>index 6f08a0caf1c1b1234ac51779f13d5cada99c474c..955e7e4c608ee505fc0e0c90cda3415f1b0d1d74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.cc
>@@ -17,34 +17,6 @@ namespace webrtc {
> const char MediaStreamTrackInterface::kVideoKind[] = "video";
> const char MediaStreamTrackInterface::kAudioKind[] = "audio";
> 
>-void AudioProcessorInterface::GetStats(AudioProcessorStats* /*stats*/) {
>-  RTC_NOTREACHED() << "Old-style GetStats() is called but it has no "
>-                   << "implementation.";
>-  RTC_LOG(LS_ERROR) << "Old-style GetStats() is called but it has no "
>-                    << "implementation.";
>-}
>-
>-// TODO(ivoc): Remove this when the function becomes pure virtual.
>-AudioProcessorInterface::AudioProcessorStatistics
>-AudioProcessorInterface::GetStats(bool /*has_remote_tracks*/) {
>-  AudioProcessorStats stats;
>-  GetStats(&stats);
>-  AudioProcessorStatistics new_stats;
>-  new_stats.apm_statistics.divergent_filter_fraction =
>-      stats.aec_divergent_filter_fraction;
>-  new_stats.apm_statistics.delay_median_ms = stats.echo_delay_median_ms;
>-  new_stats.apm_statistics.delay_standard_deviation_ms =
>-      stats.echo_delay_std_ms;
>-  new_stats.apm_statistics.echo_return_loss = stats.echo_return_loss;
>-  new_stats.apm_statistics.echo_return_loss_enhancement =
>-      stats.echo_return_loss_enhancement;
>-  new_stats.apm_statistics.residual_echo_likelihood =
>-      stats.residual_echo_likelihood;
>-  new_stats.apm_statistics.residual_echo_likelihood_recent_max =
>-      stats.residual_echo_likelihood_recent_max;
>-  return new_stats;
>-}
>-
> VideoTrackInterface::ContentHint VideoTrackInterface::content_hint() const {
>   return ContentHint::kNone;
> }
>@@ -58,4 +30,8 @@ AudioTrackInterface::GetAudioProcessor() {
>   return nullptr;
> }
> 
>+const cricket::AudioOptions AudioSourceInterface::options() const {
>+  return {};
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>index b661351e9fc985ed6d15621afe274de8ce1c47e7..6d967660c573adb8c598a3534f67f5fc8f274a77 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>@@ -23,18 +23,13 @@
> #include <vector>
> 
> #include "absl/types/optional.h"
>+#include "api/audio_options.h"
> #include "api/video/video_frame.h"
>-// TODO(zhihuang): Remove unrelated headers once downstream applications stop
>-// relying on them; they were previously transitively included by
>-// mediachannel.h, which is no longer a dependency of this file.
> #include "api/video/video_sink_interface.h"
> #include "api/video/video_source_interface.h"
> #include "modules/audio_processing/include/audio_processing_statistics.h"
>-#include "rtc_base/ratetracker.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "rtc_base/thread.h"
>-#include "rtc_base/timeutils.h"
> 
> namespace webrtc {
> 
>@@ -213,53 +208,28 @@ class AudioSourceInterface : public MediaSourceInterface {
>   // TODO(tommi): Make pure virtual.
>   virtual void AddSink(AudioTrackSinkInterface* sink) {}
>   virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
>+
>+  // Returns options for the AudioSource.
>+  // (for some of the settings this approach is broken, e.g. setting
>+  // audio network adaptation on the source is the wrong layer of abstraction).
>+  virtual const cricket::AudioOptions options() const;
> };
> 
> // Interface of the audio processor used by the audio track to collect
> // statistics.
> class AudioProcessorInterface : public rtc::RefCountInterface {
>  public:
>-  // Deprecated, use AudioProcessorStatistics instead.
>-  // TODO(ivoc): Remove this when all implementations have switched to the new
>-  //             GetStats function. See b/67926135.
>-  struct AudioProcessorStats {
>-    AudioProcessorStats()
>-        : typing_noise_detected(false),
>-          echo_return_loss(0),
>-          echo_return_loss_enhancement(0),
>-          echo_delay_median_ms(0),
>-          echo_delay_std_ms(0),
>-          residual_echo_likelihood(0.0f),
>-          residual_echo_likelihood_recent_max(0.0f),
>-          aec_divergent_filter_fraction(0.0) {}
>-    ~AudioProcessorStats() {}
>-
>-    bool typing_noise_detected;
>-    int echo_return_loss;
>-    int echo_return_loss_enhancement;
>-    int echo_delay_median_ms;
>-    int echo_delay_std_ms;
>-    float residual_echo_likelihood;
>-    float residual_echo_likelihood_recent_max;
>-    float aec_divergent_filter_fraction;
>-  };
>-  // This struct maintains the optionality of the stats, and will replace the
>-  // regular stats struct when all users have been updated.
>   struct AudioProcessorStatistics {
>     bool typing_noise_detected = false;
>     AudioProcessingStats apm_statistics;
>   };
> 
>-  // Get audio processor statistics.
>-  virtual void GetStats(AudioProcessorStats* stats);
>-
>   // Get audio processor statistics. The |has_remote_tracks| argument should be
>   // set if there are active remote tracks (this would usually be true during
>   // a call). If there are no remote tracks some of the stats will not be set by
>   // the AudioProcessor, because they only make sense if there is at least one
>   // remote track.
>-  // TODO(ivoc): Make pure virtual when all implementions are updated.
>-  virtual AudioProcessorStatistics GetStats(bool has_remote_tracks);
>+  virtual AudioProcessorStatistics GetStats(bool has_remote_tracks) = 0;
> 
>  protected:
>   ~AudioProcessorInterface() override = default;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription.cc
>deleted file mode 100644
>index d5155f22fe9a35885cdd5370af282753861f337d..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription.cc
>+++ /dev/null
>@@ -1,13 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/ortc/mediadescription.h"
>-
>-namespace webrtc {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription.h
>deleted file mode 100644
>index 5cf1d1a67be0187702c5819d6bff6219efae027f..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription.h
>+++ /dev/null
>@@ -1,53 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef API_ORTC_MEDIADESCRIPTION_H_
>-#define API_ORTC_MEDIADESCRIPTION_H_
>-
>-#include <string>
>-#include <utility>
>-#include <vector>
>-
>-#include "absl/types/optional.h"
>-#include "api/cryptoparams.h"
>-
>-namespace webrtc {
>-
>-// A structured representation of a media description within an SDP session
>-// description.
>-class MediaDescription {
>- public:
>-  explicit MediaDescription(std::string mid) : mid_(std::move(mid)) {}
>-
>-  ~MediaDescription() {}
>-
>-  // The mid(media stream identification) is used for identifying media streams
>-  // within a session description.
>-  // https://tools.ietf.org/html/rfc5888#section-6
>-  absl::optional<std::string> mid() const { return mid_; }
>-  void set_mid(std::string mid) { mid_.emplace(std::move(mid)); }
>-
>-  // Security keys and parameters for this media stream. Can be used to
>-  // negotiate parameters for SRTP.
>-  // https://tools.ietf.org/html/rfc4568#page-5
>-  std::vector<cricket::CryptoParams>& sdes_params() { return sdes_params_; }
>-  const std::vector<cricket::CryptoParams>& sdes_params() const {
>-    return sdes_params_;
>-  }
>-
>- private:
>-  absl::optional<std::string> mid_;
>-
>-  std::vector<cricket::CryptoParams> sdes_params_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // API_ORTC_MEDIADESCRIPTION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription_unittest.cc
>deleted file mode 100644
>index 9ff943af6f405fa8cbfca229e7b02336b40b33c7..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription_unittest.cc
>+++ /dev/null
>@@ -1,30 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/ortc/mediadescription.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-
>-class MediaDescriptionTest : public testing::Test {};
>-
>-TEST_F(MediaDescriptionTest, CreateMediaDescription) {
>-  MediaDescription m("a");
>-  EXPECT_EQ("a", m.mid());
>-}
>-
>-TEST_F(MediaDescriptionTest, AddSdesParam) {
>-  MediaDescription m("a");
>-  m.sdes_params().push_back(cricket::CryptoParams());
>-  const std::vector<cricket::CryptoParams>& params = m.sdes_params();
>-  EXPECT_EQ(1u, params.size());
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcfactoryinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcfactoryinterface.h
>deleted file mode 100644
>index 99373528318cd7b24683898757314b496d7c3317..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcfactoryinterface.h
>+++ /dev/null
>@@ -1,232 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef API_ORTC_ORTCFACTORYINTERFACE_H_
>-#define API_ORTC_ORTCFACTORYINTERFACE_H_
>-
>-#include <memory>
>-#include <string>
>-#include <utility>  // For std::move.
>-
>-#include "api/mediastreaminterface.h"
>-#include "api/mediatypes.h"
>-#include "api/ortc/ortcrtpreceiverinterface.h"
>-#include "api/ortc/ortcrtpsenderinterface.h"
>-#include "api/ortc/packettransportinterface.h"
>-#include "api/ortc/rtptransportcontrollerinterface.h"
>-#include "api/ortc/rtptransportinterface.h"
>-#include "api/ortc/srtptransportinterface.h"
>-#include "api/ortc/udptransportinterface.h"
>-#include "api/peerconnectioninterface.h"
>-#include "api/rtcerror.h"
>-#include "api/rtpparameters.h"
>-#include "rtc_base/network.h"
>-#include "rtc_base/scoped_ref_ptr.h"
>-#include "rtc_base/thread.h"
>-
>-namespace webrtc {
>-
>-// TODO(deadbeef): This should be part of /api/, but currently it's not and
>-// including its header violates checkdeps rules.
>-class AudioDeviceModule;
>-
>-// WARNING: This is experimental/under development, so use at your own risk; no
>-// guarantee about API stability is guaranteed here yet.
>-//
>-// This class is the ORTC analog of PeerConnectionFactory. It acts as a factory
>-// for ORTC objects that can be connected to each other.
>-//
>-// Some of these objects may not be represented by the ORTC specification, but
>-// follow the same general principles.
>-//
>-// If one of the factory methods takes another object as an argument, it MUST
>-// have been created by the same OrtcFactory.
>-//
>-// On object lifetimes: objects should be destroyed in this order:
>-// 1. Objects created by the factory.
>-// 2. The factory itself.
>-// 3. Objects passed into OrtcFactoryInterface::Create.
>-class OrtcFactoryInterface {
>- public:
>-  // |network_thread| is the thread on which packets are sent and received.
>-  // If null, a new rtc::Thread with a default socket server is created.
>-  //
>-  // |signaling_thread| is used for callbacks to the consumer of the API. If
>-  // null, the current thread will be used, which assumes that the API consumer
>-  // is running a message loop on this thread (either using an existing
>-  // rtc::Thread, or by calling rtc::Thread::Current()->ProcessMessages).
>-  //
>-  // |network_manager| is used to determine which network interfaces are
>-  // available. This is used for ICE, for example. If null, a default
>-  // implementation will be used. Only accessed on |network_thread|.
>-  //
>-  // |socket_factory| is used (on the network thread) for creating sockets. If
>-  // it's null, a default implementation will be used, which assumes
>-  // |network_thread| is a normal rtc::Thread.
>-  //
>-  // |adm| is optional, and allows a different audio device implementation to
>-  // be injected; otherwise a platform-specific module will be used that will
>-  // use the default audio input.
>-  //
>-  // |audio_encoder_factory| and |audio_decoder_factory| are used to
>-  // instantiate audio codecs; they determine what codecs are supported.
>-  //
>-  // Note that the OrtcFactoryInterface does not take ownership of any of the
>-  // objects passed in by raw pointer, and as previously stated, these objects
>-  // can't be destroyed before the factory is.
>-  static RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> Create(
>-      rtc::Thread* network_thread,
>-      rtc::Thread* signaling_thread,
>-      rtc::NetworkManager* network_manager,
>-      rtc::PacketSocketFactory* socket_factory,
>-      AudioDeviceModule* adm,
>-      rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-      rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory);
>-
>-  // Constructor for convenience which uses default implementations where
>-  // possible (though does still require that the current thread runs a message
>-  // loop; see above).
>-  static RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> Create(
>-      rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-      rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory) {
>-    return Create(nullptr, nullptr, nullptr, nullptr, nullptr,
>-                  audio_encoder_factory, audio_decoder_factory);
>-  }
>-
>-  virtual ~OrtcFactoryInterface() {}
>-
>-  // Creates an RTP transport controller, which is used in calls to
>-  // CreateRtpTransport methods. If your application has some notion of a
>-  // "call", you should create one transport controller per call.
>-  //
>-  // However, if you only are using one RtpTransport object, this doesn't need
>-  // to be called explicitly; CreateRtpTransport will create one automatically
>-  // if |rtp_transport_controller| is null. See below.
>-  //
>-  // TODO(deadbeef): Add MediaConfig and RtcEventLog arguments?
>-  virtual RTCErrorOr<std::unique_ptr<RtpTransportControllerInterface>>
>-  CreateRtpTransportController() = 0;
>-
>-  // Creates an RTP transport using the provided packet transports and
>-  // transport controller.
>-  //
>-  // |rtp| will be used for sending RTP packets, and |rtcp| for RTCP packets.
>-  //
>-  // |rtp| can't be null. |rtcp| must be non-null if and only if
>-  // |rtp_parameters.rtcp.mux| is false, indicating that RTCP muxing isn't used.
>-  // Note that if RTCP muxing isn't enabled initially, it can still enabled
>-  // later through SetParameters.
>-  //
>-  // If |transport_controller| is null, one will automatically be created, and
>-  // its lifetime managed by the returned RtpTransport. This should only be
>-  // done if a single RtpTransport is being used to communicate with the remote
>-  // endpoint.
>-  virtual RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateRtpTransport(
>-      const RtpTransportParameters& rtp_parameters,
>-      PacketTransportInterface* rtp,
>-      PacketTransportInterface* rtcp,
>-      RtpTransportControllerInterface* transport_controller) = 0;
>-
>-  // Creates an SrtpTransport which is an RTP transport that uses SRTP.
>-  virtual RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
>-  CreateSrtpTransport(
>-      const RtpTransportParameters& rtp_parameters,
>-      PacketTransportInterface* rtp,
>-      PacketTransportInterface* rtcp,
>-      RtpTransportControllerInterface* transport_controller) = 0;
>-
>-  // Returns the capabilities of an RTP sender of type |kind|. These
>-  // capabilities can be used to determine what RtpParameters to use to create
>-  // an RtpSender.
>-  //
>-  // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
>-  virtual RtpCapabilities GetRtpSenderCapabilities(
>-      cricket::MediaType kind) const = 0;
>-
>-  // Creates an RTP sender with |track|. Will not start sending until Send is
>-  // called. This is provided as a convenience; it's equivalent to calling
>-  // CreateRtpSender with a kind (see below), followed by SetTrack.
>-  //
>-  // |track| and |transport| must not be null.
>-  virtual RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateRtpSender(
>-      rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-      RtpTransportInterface* transport) = 0;
>-
>-  // Overload of CreateRtpSender allows creating the sender without a track.
>-  //
>-  // |kind| must be MEDIA_TYPE_AUDIO or MEDIA_TYPE_VIDEO.
>-  virtual RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateRtpSender(
>-      cricket::MediaType kind,
>-      RtpTransportInterface* transport) = 0;
>-
>-  // Returns the capabilities of an RTP receiver of type |kind|. These
>-  // capabilities can be used to determine what RtpParameters to use to create
>-  // an RtpReceiver.
>-  //
>-  // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
>-  virtual RtpCapabilities GetRtpReceiverCapabilities(
>-      cricket::MediaType kind) const = 0;
>-
>-  // Creates an RTP receiver of type |kind|. Will not start receiving media
>-  // until Receive is called.
>-  //
>-  // |kind| must be MEDIA_TYPE_AUDIO or MEDIA_TYPE_VIDEO.
>-  //
>-  // |transport| must not be null.
>-  virtual RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>>
>-  CreateRtpReceiver(cricket::MediaType kind,
>-                    RtpTransportInterface* transport) = 0;
>-
>-  // Create a UDP transport with IP address family |family|, using a port
>-  // within the specified range.
>-  //
>-  // |family| must be AF_INET or AF_INET6.
>-  //
>-  // |min_port|/|max_port| values of 0 indicate no range restriction.
>-  //
>-  // Returns an error if the transport wasn't successfully created.
>-  virtual RTCErrorOr<std::unique_ptr<UdpTransportInterface>>
>-  CreateUdpTransport(int family, uint16_t min_port, uint16_t max_port) = 0;
>-
>-  // Method for convenience that has no port range restrictions.
>-  RTCErrorOr<std::unique_ptr<UdpTransportInterface>> CreateUdpTransport(
>-      int family) {
>-    return CreateUdpTransport(family, 0, 0);
>-  }
>-
>-  // NOTE: The methods below to create tracks/sources return scoped_refptrs
>-  // rather than unique_ptrs, because these interfaces are also used with
>-  // PeerConnection, where everything is ref-counted.
>-
>-  // Creates a audio source representing the default microphone input.
>-  // |options| decides audio processing settings.
>-  virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
>-      const cricket::AudioOptions& options) = 0;
>-
>-  // Version of the above method that uses default options.
>-  rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource() {
>-    return CreateAudioSource(cricket::AudioOptions());
>-  }
>-
>-  // Creates a new local video track wrapping |source|. The same |source| can
>-  // be used in several tracks.
>-  virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
>-      const std::string& id,
>-      VideoTrackSourceInterface* source) = 0;
>-
>-  // Creates an new local audio track wrapping |source|.
>-  virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
>-      const std::string& id,
>-      AudioSourceInterface* source) = 0;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // API_ORTC_ORTCFACTORYINTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcrtpreceiverinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcrtpreceiverinterface.h
>deleted file mode 100644
>index 59ff97762135372a9f1b250c39b5d477cb8e20a4..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcrtpreceiverinterface.h
>+++ /dev/null
>@@ -1,84 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// This file contains interfaces for RtpReceivers:
>-// http://publications.ortc.org/2016/20161202/#rtcrtpreceiver*
>-//
>-// However, underneath the RtpReceiver is an RtpTransport, rather than a
>-// DtlsTransport. This is to allow different types of RTP transports (besides
>-// DTLS-SRTP) to be used.
>-
>-#ifndef API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
>-#define API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
>-
>-#include "api/mediastreaminterface.h"
>-#include "api/mediatypes.h"
>-#include "api/ortc/rtptransportinterface.h"
>-#include "api/rtcerror.h"
>-#include "api/rtpparameters.h"
>-
>-namespace webrtc {
>-
>-// Note: Since receiver capabilities may depend on how the OrtcFactory was
>-// created, instead of a static "GetCapabilities" method on this interface,
>-// there is a "GetRtpReceiverCapabilities" method on the OrtcFactory.
>-class OrtcRtpReceiverInterface {
>- public:
>-  virtual ~OrtcRtpReceiverInterface() {}
>-
>-  // Returns a track representing the media received by this receiver.
>-  //
>-  // Currently, this will return null until Receive has been successfully
>-  // called. Also, a new track will be created every time the primary SSRC
>-  // changes.
>-  //
>-  // If encodings are removed, GetTrack will return null. Though deactivating
>-  // an encoding (setting |active| to false) will not do this.
>-  //
>-  // In the future, these limitations will be fixed, and GetTrack will return
>-  // the same track for the lifetime of the RtpReceiver. So it's not
>-  // recommended to write code that depends on this non-standard behavior.
>-  virtual rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const = 0;
>-
>-  // Once supported, will switch to receiving media on a new transport.
>-  // However, this is not currently supported and will always return an error.
>-  virtual RTCError SetTransport(RtpTransportInterface* transport) = 0;
>-  // Returns previously set (or constructed-with) transport.
>-  virtual RtpTransportInterface* GetTransport() const = 0;
>-
>-  // Start receiving media with |parameters| (if |parameters| contains an
>-  // active encoding).
>-  //
>-  // There are no limitations to how the parameters can be changed after the
>-  // initial call to Receive, as long as they're valid (for example, they can't
>-  // use the same payload type for two codecs).
>-  virtual RTCError Receive(const RtpParameters& parameters) = 0;
>-  // Returns parameters that were last successfully passed into Receive, or
>-  // empty parameters if that hasn't yet occurred.
>-  //
>-  // Note that for parameters that are described as having an "implementation
>-  // default" value chosen, GetParameters() will return those chosen defaults,
>-  // with the exception of SSRCs which have special behavior. See
>-  // rtpparameters.h for more details.
>-  virtual RtpParameters GetParameters() const = 0;
>-
>-  // Audio or video receiver?
>-  //
>-  // Once GetTrack() starts always returning a track, this method will be
>-  // redundant, as one can call "GetTrack()->kind()". However, it's still a
>-  // nice convenience, and is symmetric with OrtcRtpSenderInterface::GetKind.
>-  virtual cricket::MediaType GetKind() const = 0;
>-
>-  // TODO(deadbeef): GetContributingSources
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcrtpsenderinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcrtpsenderinterface.h
>deleted file mode 100644
>index fd4dfaa7907ffc28c7716c6bed0726eeb4d8fec4..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcrtpsenderinterface.h
>+++ /dev/null
>@@ -1,77 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// This file contains interfaces for RtpSenders:
>-// http://publications.ortc.org/2016/20161202/#rtcrtpsender*
>-//
>-// However, underneath the RtpSender is an RtpTransport, rather than a
>-// DtlsTransport. This is to allow different types of RTP transports (besides
>-// DTLS-SRTP) to be used.
>-
>-#ifndef API_ORTC_ORTCRTPSENDERINTERFACE_H_
>-#define API_ORTC_ORTCRTPSENDERINTERFACE_H_
>-
>-#include "api/mediastreaminterface.h"
>-#include "api/mediatypes.h"
>-#include "api/ortc/rtptransportinterface.h"
>-#include "api/rtcerror.h"
>-#include "api/rtpparameters.h"
>-
>-namespace webrtc {
>-
>-// Note: Since sender capabilities may depend on how the OrtcFactory was
>-// created, instead of a static "GetCapabilities" method on this interface,
>-// there is a "GetRtpSenderCapabilities" method on the OrtcFactory.
>-class OrtcRtpSenderInterface {
>- public:
>-  virtual ~OrtcRtpSenderInterface() {}
>-
>-  // Sets the source of media that will be sent by this sender.
>-  //
>-  // If Send has already been called, will immediately switch to sending this
>-  // track. If |track| is null, will stop sending media.
>-  //
>-  // Returns INVALID_PARAMETER error if an audio track is set on a video
>-  // RtpSender, or vice-versa.
>-  virtual RTCError SetTrack(MediaStreamTrackInterface* track) = 0;
>-  // Returns previously set (or constructed-with) track.
>-  virtual rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const = 0;
>-
>-  // Once supported, will switch to sending media on a new transport. However,
>-  // this is not currently supported and will always return an error.
>-  virtual RTCError SetTransport(RtpTransportInterface* transport) = 0;
>-  // Returns previously set (or constructed-with) transport.
>-  virtual RtpTransportInterface* GetTransport() const = 0;
>-
>-  // Start sending media with |parameters| (if |parameters| contains an active
>-  // encoding).
>-  //
>-  // There are no limitations to how the parameters can be changed after the
>-  // initial call to Send, as long as they're valid (for example, they can't
>-  // use the same payload type for two codecs).
>-  virtual RTCError Send(const RtpParameters& parameters) = 0;
>-  // Returns parameters that were last successfully passed into Send, or empty
>-  // parameters if that hasn't yet occurred.
>-  //
>-  // Note that for parameters that are described as having an "implementation
>-  // default" value chosen, GetParameters() will return those chosen defaults,
>-  // with the exception of SSRCs which have special behavior. See
>-  // rtpparameters.h for more details.
>-  virtual RtpParameters GetParameters() const = 0;
>-
>-  // Audio or video sender?
>-  virtual cricket::MediaType GetKind() const = 0;
>-
>-  // TODO(deadbeef): SSRC conflict signal.
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // API_ORTC_ORTCRTPSENDERINTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/rtptransportcontrollerinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/rtptransportcontrollerinterface.h
>deleted file mode 100644
>index 85f37fa7a0d5d208dd769e3d9bcc0377b9975cab..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/rtptransportcontrollerinterface.h
>+++ /dev/null
>@@ -1,57 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
>-#define API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
>-
>-#include <vector>
>-
>-#include "api/ortc/rtptransportinterface.h"
>-
>-namespace webrtc {
>-
>-class RtpTransportControllerAdapter;
>-
>-// Used to group RTP transports between a local endpoint and the same remote
>-// endpoint, for the purpose of sharing bandwidth estimation and other things.
>-//
>-// Comparing this to the PeerConnection model, non-budled audio/video would use
>-// two RtpTransports with a single RtpTransportController, whereas bundled
>-// media would use a single RtpTransport, and two PeerConnections would use
>-// independent RtpTransportControllers.
>-//
>-// RtpTransports are associated with this controller when they're created, by
>-// passing the controller into OrtcFactory's relevant "CreateRtpTransport"
>-// method. When a transport is destroyed, it's automatically disassociated.
>-// GetTransports returns all currently associated transports.
>-//
>-// This is the RTP equivalent of "IceTransportController" in ORTC; RtpTransport
>-// is to RtpTransportController as IceTransport is to IceTransportController.
>-class RtpTransportControllerInterface {
>- public:
>-  virtual ~RtpTransportControllerInterface() {}
>-
>-  // Returns all transports associated with this controller (see explanation
>-  // above). No ordering is guaranteed.
>-  virtual std::vector<RtpTransportInterface*> GetTransports() const = 0;
>-
>- protected:
>-  // Only for internal use. Returns a pointer to an internal interface, for use
>-  // by the implementation.
>-  virtual RtpTransportControllerAdapter* GetInternal() = 0;
>-
>-  // Classes that can use this internal interface.
>-  friend class OrtcFactory;
>-  friend class RtpTransportAdapter;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/rtptransportinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/rtptransportinterface.h
>index b0d30e8560e27a528ba67a5354afb32da0a2db64..4b9d16158f65a0481dcf7653f6b42f4de9d82ede 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/rtptransportinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/rtptransportinterface.h
>@@ -18,7 +18,6 @@
> #include "api/rtcerror.h"
> #include "api/rtp_headers.h"
> #include "api/rtpparameters.h"
>-#include "common_types.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription.cc
>deleted file mode 100644
>index 1078884ecb45b207896534a89028a115753f9e77..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription.cc
>+++ /dev/null
>@@ -1,13 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/ortc/sessiondescription.h"
>-
>-namespace webrtc {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription.h
>deleted file mode 100644
>index ebbaa27d6ffc0011450326e51c17c8b17df9f952..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription.h
>+++ /dev/null
>@@ -1,45 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef API_ORTC_SESSIONDESCRIPTION_H_
>-#define API_ORTC_SESSIONDESCRIPTION_H_
>-
>-#include <string>
>-#include <utility>
>-
>-namespace webrtc {
>-
>-// A structured representation of an SDP session description.
>-class SessionDescription {
>- public:
>-  SessionDescription(int64_t session_id, std::string session_version)
>-      : session_id_(session_id), session_version_(std::move(session_version)) {}
>-
>-  // https://tools.ietf.org/html/rfc4566#section-5.2
>-  // o=<username> <sess-id> <sess-version> <nettype> <addrtype>
>-  //   <unicast-address>
>-  // session_id_ is the "sess-id" field.
>-  // session_version_ is the "sess-version" field.
>-  int64_t session_id() { return session_id_; }
>-  void set_session_id(int64_t session_id) { session_id_ = session_id; }
>-
>-  const std::string& session_version() const { return session_version_; }
>-  void set_session_version(std::string session_version) {
>-    session_version_ = std::move(session_version);
>-  }
>-
>- private:
>-  int64_t session_id_;
>-  std::string session_version_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // API_ORTC_SESSIONDESCRIPTION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription_unittest.cc
>deleted file mode 100644
>index e4611c6c6b2bf7a0be4f5c92166f40934b05f706..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription_unittest.cc
>+++ /dev/null
>@@ -1,23 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/ortc/sessiondescription.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-
>-class SessionDescriptionTest : public testing::Test {};
>-
>-TEST_F(SessionDescriptionTest, CreateSessionDescription) {
>-  SessionDescription s(-1, "0");
>-  EXPECT_EQ(-1, s.session_id());
>-  EXPECT_EQ("0", s.session_version());
>-}
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/udptransportinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/udptransportinterface.h
>deleted file mode 100644
>index f246a25e9ddd0f5ecafe0069bc863d0308ef2b81..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/udptransportinterface.h
>+++ /dev/null
>@@ -1,49 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef API_ORTC_UDPTRANSPORTINTERFACE_H_
>-#define API_ORTC_UDPTRANSPORTINTERFACE_H_
>-
>-#include "api/ortc/packettransportinterface.h"
>-#include "api/proxy.h"
>-#include "rtc_base/socketaddress.h"
>-
>-namespace webrtc {
>-
>-// Interface for a raw UDP transport (not using ICE), meaning a combination of
>-// a local/remote IP address/port.
>-//
>-// An instance can be instantiated using OrtcFactory.
>-//
>-// Each instance reserves a UDP port, which will be freed when the
>-// UdpTransportInterface destructor is called.
>-//
>-// Calling SetRemoteAddress sets the destination of outgoing packets; without a
>-// destination, packets can't be sent, but they can be received.
>-class UdpTransportInterface : public virtual PacketTransportInterface {
>- public:
>-  // Get the address of the socket allocated for this transport.
>-  virtual rtc::SocketAddress GetLocalAddress() const = 0;
>-
>-  // Sets the address to which packets will be delivered.
>-  //
>-  // Calling with a "nil" (default-constructed) address is legal, and unsets
>-  // any previously set destination.
>-  //
>-  // However, calling with an incomplete address (port or IP not set) will
>-  // fail.
>-  virtual bool SetRemoteAddress(const rtc::SocketAddress& dest) = 0;
>-  // Simple getter. If never set, returns nil address.
>-  virtual rtc::SocketAddress GetRemoteAddress() const = 0;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // API_ORTC_UDPTRANSPORTINTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.cc
>index b4148d7480aabe443cb99e74df742e0d584a1507..f2953b5b0c28c611fb10efbf8306de245e673f2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.cc
>@@ -159,6 +159,11 @@ RTCError PeerConnectionInterface::SetBitrate(
>   return SetBitrate(bitrate);
> }
> 
>+PeerConnectionInterface::PeerConnectionState
>+PeerConnectionInterface::peer_connection_state() {
>+  return PeerConnectionInterface::PeerConnectionState::kNew;
>+}
>+
> bool PeerConnectionInterface::StartRtcEventLog(rtc::PlatformFile file,
>                                                int64_t max_size_bytes) {
>   return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>index 1c32b69692d00e6efab782c5bdba1cd7a830e2c8..54161b8da59b0226083a6abf98296ae8339df6b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>@@ -69,7 +69,6 @@
> 
> #include <memory>
> #include <string>
>-#include <utility>
> #include <vector>
> 
> #include "api/asyncresolverfactory.h"
>@@ -78,10 +77,11 @@
> #include "api/audio_codecs/audio_encoder_factory.h"
> #include "api/audio_options.h"
> #include "api/call/callfactoryinterface.h"
>+#include "api/crypto/cryptooptions.h"
> #include "api/datachannelinterface.h"
>-#include "api/dtmfsenderinterface.h"
> #include "api/fec_controller.h"
> #include "api/jsep.h"
>+#include "api/media_transport_interface.h"
> #include "api/mediastreaminterface.h"
> #include "api/rtcerror.h"
> #include "api/rtceventlogoutput.h"
>@@ -94,7 +94,6 @@
> #include "api/transport/bitrate_settings.h"
> #include "api/transport/network_control.h"
> #include "api/turncustomizer.h"
>-#include "api/umametrics.h"
> #include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
> #include "media/base/mediaconfig.h"
> // TODO(bugs.webrtc.org/6353): cricket::VideoCapturer is deprecated and should
>@@ -114,6 +113,7 @@
> #include "rtc_base/socketaddress.h"
> #include "rtc_base/sslcertificate.h"
> #include "rtc_base/sslstreamadapter.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace rtc {
> class SSLIdentity;
>@@ -160,7 +160,7 @@ enum class SdpSemantics { kPlanB, kUnifiedPlan };
> 
> class PeerConnectionInterface : public rtc::RefCountInterface {
>  public:
>-  // See https://w3c.github.io/webrtc-pc/#state-definitions
>+  // See https://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
>   enum SignalingState {
>     kStable,
>     kHaveLocalOffer,
>@@ -170,12 +170,24 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     kClosed,
>   };
> 
>+  // See https://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate
>   enum IceGatheringState {
>     kIceGatheringNew,
>     kIceGatheringGathering,
>     kIceGatheringComplete
>   };
> 
>+  // See https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate
>+  enum class PeerConnectionState {
>+    kNew,
>+    kConnecting,
>+    kConnected,
>+    kDisconnected,
>+    kFailed,
>+    kClosed,
>+  };
>+
>+  // See https://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate
>   enum IceConnectionState {
>     kIceConnectionNew,
>     kIceConnectionChecking,
>@@ -282,7 +294,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   // organization of the implementation, which isn't stable. So we
>   // need getters and setters at least for fields which applications
>   // are interested in.
>-  struct RTCConfiguration {
>+  struct RTC_EXPORT RTCConfiguration {
>     // This struct is subject to reorganization, both for naming
>     // consistency, and to group settings to match where they are used
>     // in the implementation. To do that, we need getter and setter
>@@ -328,6 +340,22 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>       media_config.video.experiment_cpu_load_estimator = enable;
>     }
> 
>+    int audio_rtcp_report_interval_ms() const {
>+      return media_config.audio.rtcp_report_interval_ms;
>+    }
>+    void set_audio_rtcp_report_interval_ms(int audio_rtcp_report_interval_ms) {
>+      media_config.audio.rtcp_report_interval_ms =
>+          audio_rtcp_report_interval_ms;
>+    }
>+
>+    int video_rtcp_report_interval_ms() const {
>+      return media_config.video.rtcp_report_interval_ms;
>+    }
>+    void set_video_rtcp_report_interval_ms(int video_rtcp_report_interval_ms) {
>+      media_config.video.rtcp_report_interval_ms =
>+          video_rtcp_report_interval_ms;
>+    }
>+
>     static const int kUndefined = -1;
>     // Default maximum number of packets in the audio jitter buffer.
>     static const int kAudioJitterBufferMaxPackets = 50;
>@@ -394,6 +422,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // Use new combined audio/video bandwidth estimation?
>     absl::optional<bool> combined_audio_video_bwe;
> 
>+    // TODO(bugs.webrtc.org/9891) - Move to crypto_options
>     // Can be used to disable DTLS-SRTP. This should never be done, but can be
>     // useful for testing purposes, for example in setting up a loopback call
>     // with a single PeerConnection.
>@@ -421,6 +450,9 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // if it falls behind.
>     bool audio_jitter_buffer_fast_accelerate = false;
> 
>+    // The minimum delay in milliseconds for the audio jitter buffer.
>+    int audio_jitter_buffer_min_delay_ms = 0;
>+
>     // Timeout in milliseconds before an ICE candidate pair is considered to be
>     // "not receiving", after which a lower priority candidate pair may be
>     // selected.
>@@ -556,6 +588,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // For all other users, specify kUnifiedPlan.
>     SdpSemantics sdp_semantics = SdpSemantics::kPlanB;
> 
>+    // TODO(bugs.webrtc.org/9891) - Move to crypto_options or remove.
>     // Actively reset the SRTP parameters whenever the DTLS transports
>     // underneath are reset for every offer/answer negotiation.
>     // This is only intended to be a workaround for crbug.com/835958
>@@ -563,6 +596,34 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // correctly. This flag will be deprecated soon. Do not rely on it.
>     bool active_reset_srtp_params = false;
> 
>+    // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
>+    // informs PeerConnection that it should use the MediaTransportInterface.
>+    // It's invalid to set it to |true| if the MediaTransportFactory wasn't
>+    // provided.
>+    bool use_media_transport = false;
>+
>+    // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
>+    // informs PeerConnection that it should use the MediaTransportInterface for
>+    // data channels.  It's invalid to set it to |true| if the
>+    // MediaTransportFactory wasn't provided.  Data channels over media
>+    // transport are not compatible with RTP or SCTP data channels.  Setting
>+    // both |use_media_transport_for_data_channels| and
>+    // |enable_rtp_data_channel| is invalid.
>+    bool use_media_transport_for_data_channels = false;
>+
>+    // Defines advanced optional cryptographic settings related to SRTP and
>+    // frame encryption for native WebRTC. Setting this will overwrite any
>+    // settings set in PeerConnectionFactory (which is deprecated).
>+    absl::optional<CryptoOptions> crypto_options;
>+
>+    // Configure if we should include the SDP attribute extmap-allow-mixed in
>+    // our offer. Although we currently do support this, it's not included in
>+    // our offer by default due to a previous bug that caused the SDP parser to
>+    // abort parsing if this attribute was present. This is fixed in Chrome 71.
>+    // TODO(webrtc:9985): Change default to true once sufficient time has
>+    // passed.
>+    bool offer_extmap_allow_mixed = false;
>+
>     //
>     // Don't forget to update operator== if adding something.
>     //
>@@ -972,11 +1033,13 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   virtual SignalingState signaling_state() = 0;
> 
>   // Returns the aggregate state of all ICE *and* DTLS transports.
>-  // TODO(deadbeef): Implement "PeerConnectionState" according to the standard,
>-  // to aggregate ICE+DTLS state, and change the scope of IceConnectionState to
>-  // be just the ICE layer. See: crbug.com/webrtc/6145
>+  // TODO(jonasolsson): Replace with standardized_ice_connection_state once it
>+  // is ready, see crbug.com/webrtc/6145
>   virtual IceConnectionState ice_connection_state() = 0;
> 
>+  // Returns the aggregated state of all ICE and DTLS transports.
>+  virtual PeerConnectionState peer_connection_state();
>+
>   virtual IceGatheringState ice_gathering_state() = 0;
> 
>   // Starts RtcEventLog using existing file. Takes ownership of |file| and
>@@ -1045,6 +1108,10 @@ class PeerConnectionObserver {
>   virtual void OnIceConnectionChange(
>       PeerConnectionInterface::IceConnectionState new_state) = 0;
> 
>+  // Called any time the PeerConnectionState changes.
>+  virtual void OnConnectionChange(
>+      PeerConnectionInterface::PeerConnectionState new_state) {}
>+
>   // Called any time the IceGatheringState changes.
>   virtual void OnIceGatheringChange(
>       PeerConnectionInterface::IceGatheringState new_state) = 0;
>@@ -1156,6 +1223,7 @@ struct PeerConnectionFactoryDependencies final {
>   std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory;
>   std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory;
>   std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
>+  std::unique_ptr<MediaTransportFactory> media_transport_factory;
> };
> 
> // PeerConnectionFactoryInterface is the factory interface used for creating
>@@ -1175,7 +1243,7 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>  public:
>   class Options {
>    public:
>-    Options() : crypto_options(rtc::CryptoOptions::NoGcm()) {}
>+    Options() {}
> 
>     // If set to true, created PeerConnections won't enforce any SRTP
>     // requirement, allowing unsecured media. Should only be used for
>@@ -1204,7 +1272,7 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>     rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
> 
>     // Sets crypto related options, e.g. enabled cipher suites.
>-    rtc::CryptoOptions crypto_options;
>+    CryptoOptions crypto_options = CryptoOptions::NoGcm();
>   };
> 
>   // Set the options to be used for subsequently created PeerConnections.
>@@ -1307,137 +1375,6 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>   ~PeerConnectionFactoryInterface() override = default;
> };
> 
>-#if defined(USE_BUILTIN_SW_CODECS)
>-// Create a new instance of PeerConnectionFactoryInterface.
>-//
>-// This method relies on the thread it's called on as the "signaling thread"
>-// for the PeerConnectionFactory it creates.
>-//
>-// As such, if the current thread is not already running an rtc::Thread message
>-// loop, an application using this method must eventually either call
>-// rtc::Thread::Current()->Run(), or call
>-// rtc::Thread::Current()->ProcessMessages() within the application's own
>-// message loop.
>-rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory);
>-
>-// Create a new instance of PeerConnectionFactoryInterface.
>-//
>-// |network_thread|, |worker_thread| and |signaling_thread| are
>-// the only mandatory parameters.
>-//
>-// If non-null, a reference is added to |default_adm|, and ownership of
>-// |video_encoder_factory| and |video_decoder_factory| is transferred to the
>-// returned factory.
>-// TODO(deadbeef): Use rtc::scoped_refptr<> and std::unique_ptr<> to make this
>-// ownership transfer and ref counting more obvious.
>-rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* signaling_thread,
>-    AudioDeviceModule* default_adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
>-
>-// Create a new instance of PeerConnectionFactoryInterface with optional
>-// external audio mixed and audio processing modules.
>-//
>-// If |audio_mixer| is null, an internal audio mixer will be created and used.
>-// If |audio_processing| is null, an internal audio processing module will be
>-// created and used.
>-rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* signaling_thread,
>-    AudioDeviceModule* default_adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>-    rtc::scoped_refptr<AudioMixer> audio_mixer,
>-    rtc::scoped_refptr<AudioProcessing> audio_processing);
>-
>-// Create a new instance of PeerConnectionFactoryInterface with optional
>-// external audio mixer, audio processing, and fec controller modules.
>-//
>-// If |audio_mixer| is null, an internal audio mixer will be created and used.
>-// If |audio_processing| is null, an internal audio processing module will be
>-// created and used.
>-// If |fec_controller_factory| is null, an internal fec controller module will
>-// be created and used.
>-// If |network_controller_factory| is provided, it will be used if enabled via
>-// field trial.
>-rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* signaling_thread,
>-    AudioDeviceModule* default_adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>-    rtc::scoped_refptr<AudioMixer> audio_mixer,
>-    rtc::scoped_refptr<AudioProcessing> audio_processing,
>-    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
>-    std::unique_ptr<NetworkControllerFactoryInterface>
>-        network_controller_factory = nullptr);
>-#endif
>-
>-// Create a new instance of PeerConnectionFactoryInterface with optional video
>-// codec factories. These video factories represents all video codecs, i.e. no
>-// extra internal video codecs will be added.
>-// When building WebRTC with rtc_use_builtin_sw_codecs = false, this is the
>-// only available CreatePeerConnectionFactory overload.
>-rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* signaling_thread,
>-    rtc::scoped_refptr<AudioDeviceModule> default_adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>-    std::unique_ptr<VideoEncoderFactory> video_encoder_factory,
>-    std::unique_ptr<VideoDecoderFactory> video_decoder_factory,
>-    rtc::scoped_refptr<AudioMixer> audio_mixer,
>-    rtc::scoped_refptr<AudioProcessing> audio_processing);
>-
>-#if defined(USE_BUILTIN_SW_CODECS)
>-// Create a new instance of PeerConnectionFactoryInterface with external audio
>-// mixer.
>-//
>-// If |audio_mixer| is null, an internal audio mixer will be created and used.
>-rtc::scoped_refptr<PeerConnectionFactoryInterface>
>-CreatePeerConnectionFactoryWithAudioMixer(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* signaling_thread,
>-    AudioDeviceModule* default_adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>-    rtc::scoped_refptr<AudioMixer> audio_mixer);
>-
>-// Create a new instance of PeerConnectionFactoryInterface.
>-// Same thread is used as worker and network thread.
>-inline rtc::scoped_refptr<PeerConnectionFactoryInterface>
>-CreatePeerConnectionFactory(
>-    rtc::Thread* worker_and_network_thread,
>-    rtc::Thread* signaling_thread,
>-    AudioDeviceModule* default_adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory) {
>-  return CreatePeerConnectionFactory(
>-      worker_and_network_thread, worker_and_network_thread, signaling_thread,
>-      default_adm, audio_encoder_factory, audio_decoder_factory,
>-      video_encoder_factory, video_decoder_factory);
>-}
>-#endif
>-
> // This is a lower-level version of the CreatePeerConnectionFactory functions
> // above. It's implemented in the "peerconnection" build target, whereas the
> // above methods are only implemented in the broader "libjingle_peerconnection"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.cc
>index c86bddfa8b1a946d010c3ef92e7efb40df8877f0..e668285ba29ba61d3a7275990dea890f76d0e27f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.cc
>@@ -14,7 +14,7 @@ namespace webrtc {
> namespace internal {
> 
> SynchronousMethodCall::SynchronousMethodCall(rtc::MessageHandler* proxy)
>-    : e_(), proxy_(proxy) {}
>+    : proxy_(proxy) {}
> 
> SynchronousMethodCall::~SynchronousMethodCall() = default;
> 
>@@ -23,15 +23,14 @@ void SynchronousMethodCall::Invoke(const rtc::Location& posted_from,
>   if (t->IsCurrent()) {
>     proxy_->OnMessage(nullptr);
>   } else {
>-    e_.reset(new rtc::Event(false, false));
>     t->Post(posted_from, this, 0);
>-    e_->Wait(rtc::Event::kForever);
>+    e_.Wait(rtc::Event::kForever);
>   }
> }
> 
> void SynchronousMethodCall::OnMessage(rtc::Message*) {
>   proxy_->OnMessage(nullptr);
>-  e_->Set();
>+  e_.Set();
> }
> 
> }  // namespace internal
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.h
>index c8962efed7c8d2c903eebe70ec0f5509927f197e..99160517deebd79cd33d371febe21c4a7710caa1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.h
>@@ -143,7 +143,7 @@ class SynchronousMethodCall : public rtc::MessageData,
>  private:
>   void OnMessage(rtc::Message*) override;
> 
>-  std::unique_ptr<rtc::Event> e_;
>+  rtc::Event e_;
>   rtc::MessageHandler* proxy_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.cc
>index da7f1ea9fd4002f77234239058116e9d867bfdba..bf973b6fe5771d42361adb8369b30cd9b2b372c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.cc
>@@ -10,14 +10,6 @@
> 
> #include "api/rtp_headers.h"
> 
>-#include <string.h>
>-#include <algorithm>
>-#include <limits>
>-#include <type_traits>
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/stringutils.h"
>-
> namespace webrtc {
> 
> RTPHeaderExtension::RTPHeaderExtension()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.h
>index 41f39888b8081a6bfb935907f3b2cd4a4d783dba..c766899b9fd88f4a2c2ee9090f7f8cb69e9462bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.h
>@@ -12,20 +12,17 @@
> #define API_RTP_HEADERS_H_
> 
> #include <stddef.h>
>+#include <stdint.h>
> #include <string.h>
>-#include <string>
>-#include <vector>
> 
> #include "absl/types/optional.h"
> #include "api/array_view.h"
>+#include "api/video/color_space.h"
> #include "api/video/video_content_type.h"
> #include "api/video/video_frame_marking.h"
> #include "api/video/video_rotation.h"
> #include "api/video/video_timing.h"
>-
> #include "common_types.h"  // NOLINT(build/include)
>-#include "rtc_base/checks.h"
>-#include "rtc_base/deprecation.h"
> 
> namespace webrtc {
> 
>@@ -131,6 +128,8 @@ struct RTPHeaderExtension {
>   // For identifying the media section used to interpret this RTP packet. See
>   // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-38
>   Mid mid;
>+
>+  absl::optional<ColorSpace> color_space;
> };
> 
> struct RTPHeader {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.cc
>index 98ced9b3ead35bed8f7c9b37d4d210c5aea6bfb0..063d106a00a5901492ab17aa8a0fad01571eff15 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.cc
>@@ -12,7 +12,7 @@
> #include <algorithm>
> #include <string>
> 
>-#include "rtc_base/checks.h"
>+#include "api/array_view.h"
> #include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.h
>index 678ac02c4b7da7dc5a3bcaa5011e39452e7ee197..badda074599a93bb5da325bbefdd783b07ff1b18 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.h
>@@ -11,12 +11,14 @@
> #ifndef API_RTPPARAMETERS_H_
> #define API_RTPPARAMETERS_H_
> 
>+#include <stdint.h>
> #include <string>
> #include <unordered_map>
> #include <vector>
> 
> #include "absl/types/optional.h"
> #include "api/mediatypes.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -404,6 +406,14 @@ struct RtpEncodingParameters {
>   // bitrate priority.
>   double bitrate_priority = kDefaultBitratePriority;
> 
>+  // The relative DiffServ Code Point priority for this encoding, allowing
>+  // packets to be marked relatively higher or lower without affecting
>+  // bandwidth allocations. See https://w3c.github.io/webrtc-dscp-exp/ . NB
>+  // we follow chromium's translation of the allowed string enum values for
>+  // this field to 1.0, 0.5, et cetera, similar to bitrate_priority above.
>+  // TODO(http://crbug.com/webrtc/8630): Implement this per encoding parameter.
>+  double network_priority = kDefaultBitratePriority;
>+
>   // Indicates the preferred duration of media represented by a packet in
>   // milliseconds for this encoding. If set, this will take precedence over the
>   // ptime set in the RtpCodecParameters. This could happen if SDP negotiation
>@@ -471,7 +481,8 @@ struct RtpEncodingParameters {
>   bool operator==(const RtpEncodingParameters& o) const {
>     return ssrc == o.ssrc && codec_payload_type == o.codec_payload_type &&
>            fec == o.fec && rtx == o.rtx && dtx == o.dtx &&
>-           bitrate_priority == o.bitrate_priority && ptime == o.ptime &&
>+           bitrate_priority == o.bitrate_priority &&
>+           network_priority == o.network_priority && ptime == o.ptime &&
>            max_bitrate_bps == o.max_bitrate_bps &&
>            min_bitrate_bps == o.min_bitrate_bps &&
>            max_framerate == o.max_framerate &&
>@@ -607,7 +618,7 @@ struct RtcpParameters final {
>   bool operator!=(const RtcpParameters& o) const { return !(*this == o); }
> };
> 
>-struct RtpParameters {
>+struct RTC_EXPORT RtpParameters {
>   RtpParameters();
>   RtpParameters(const RtpParameters&);
>   ~RtpParameters();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsenderinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsenderinterface.h
>index 9554c1c4da79735547caebd28401816997420d8c..7c94c216e64efbeb5c4628bebe6925a6f5023e15 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsenderinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsenderinterface.h
>@@ -24,7 +24,6 @@
> #include "api/proxy.h"
> #include "api/rtcerror.h"
> #include "api/rtpparameters.h"
>-#include "rtc_base/deprecation.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.cc
>index d833339fa6646850faa9be22e99806abd81c3ebe..e62b014c6564335f70694115df988660aac43513 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.cc
>@@ -10,6 +10,8 @@
> 
> #include "api/rtptransceiverinterface.h"
> 
>+#include "rtc_base/checks.h"
>+
> namespace webrtc {
> 
> RtpTransceiverInit::RtpTransceiverInit() = default;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.h
>index 301a3809f8b4def3e76d569508064b403d258a6b..c01fdafb8323cf91bbb557fd2bf272a45ee61157 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.h
>@@ -16,9 +16,12 @@
> 
> #include "absl/types/optional.h"
> #include "api/array_view.h"
>+#include "api/mediatypes.h"
>+#include "api/rtpparameters.h"
> #include "api/rtpreceiverinterface.h"
> #include "api/rtpsenderinterface.h"
> #include "rtc_base/refcount.h"
>+#include "rtc_base/scoped_ref_ptr.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/scoped_refptr.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/scoped_refptr.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..0993e032cc7249ae060bbf83292135d2e7fc296b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/scoped_refptr.h
>@@ -0,0 +1,162 @@
>+/*
>+ *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// Originally these classes are from Chromium.
>+// http://src.chromium.org/viewvc/chrome/trunk/src/base/memory/ref_counted.h?view=markup
>+
>+//
>+// A smart pointer class for reference counted objects.  Use this class instead
>+// of calling AddRef and Release manually on a reference counted object to
>+// avoid common memory leaks caused by forgetting to Release an object
>+// reference.  Sample usage:
>+//
>+//   class MyFoo : public RefCounted<MyFoo> {
>+//    ...
>+//   };
>+//
>+//   void some_function() {
>+//     scoped_refptr<MyFoo> foo = new MyFoo();
>+//     foo->Method(param);
>+//     // |foo| is released when this function returns
>+//   }
>+//
>+//   void some_other_function() {
>+//     scoped_refptr<MyFoo> foo = new MyFoo();
>+//     ...
>+//     foo = nullptr;  // explicitly releases |foo|
>+//     ...
>+//     if (foo)
>+//       foo->Method(param);
>+//   }
>+//
>+// The above examples show how scoped_refptr<T> acts like a pointer to T.
>+// Given two scoped_refptr<T> classes, it is also possible to exchange
>+// references between the two objects, like so:
>+//
>+//   {
>+//     scoped_refptr<MyFoo> a = new MyFoo();
>+//     scoped_refptr<MyFoo> b;
>+//
>+//     b.swap(a);
>+//     // now, |b| references the MyFoo object, and |a| references null.
>+//   }
>+//
>+// To make both |a| and |b| in the above example reference the same MyFoo
>+// object, simply use the assignment operator:
>+//
>+//   {
>+//     scoped_refptr<MyFoo> a = new MyFoo();
>+//     scoped_refptr<MyFoo> b;
>+//
>+//     b = a;
>+//     // now, |a| and |b| each own a reference to the same MyFoo object.
>+//   }
>+//
>+
>+#ifndef API_SCOPED_REFPTR_H_
>+#define API_SCOPED_REFPTR_H_
>+
>+#include <memory>
>+#include <utility>
>+
>+namespace rtc {
>+
>+template <class T>
>+class scoped_refptr {
>+ public:
>+  scoped_refptr() : ptr_(nullptr) {}
>+
>+  scoped_refptr(T* p) : ptr_(p) {  // NOLINT(runtime/explicit)
>+    if (ptr_)
>+      ptr_->AddRef();
>+  }
>+
>+  scoped_refptr(const scoped_refptr<T>& r) : ptr_(r.ptr_) {
>+    if (ptr_)
>+      ptr_->AddRef();
>+  }
>+
>+  template <typename U>
>+  scoped_refptr(const scoped_refptr<U>& r) : ptr_(r.get()) {
>+    if (ptr_)
>+      ptr_->AddRef();
>+  }
>+
>+  // Move constructors.
>+  scoped_refptr(scoped_refptr<T>&& r) : ptr_(r.release()) {}
>+
>+  template <typename U>
>+  scoped_refptr(scoped_refptr<U>&& r) : ptr_(r.release()) {}
>+
>+  ~scoped_refptr() {
>+    if (ptr_)
>+      ptr_->Release();
>+  }
>+
>+  T* get() const { return ptr_; }
>+  operator T*() const { return ptr_; }
>+  T* operator->() const { return ptr_; }
>+
>+  // Returns the (possibly null) raw pointer, and makes the scoped_refptr hold a
>+  // null pointer, all without touching the reference count of the underlying
>+  // pointed-to object. The object is still reference counted, and the caller of
>+  // release() is now the proud owner of one reference, so it is responsible for
>+  // calling Release() once on the object when no longer using it.
>+  T* release() {
>+    T* retVal = ptr_;
>+    ptr_ = nullptr;
>+    return retVal;
>+  }
>+
>+  scoped_refptr<T>& operator=(T* p) {
>+    // AddRef first so that self assignment should work
>+    if (p)
>+      p->AddRef();
>+    if (ptr_)
>+      ptr_->Release();
>+    ptr_ = p;
>+    return *this;
>+  }
>+
>+  scoped_refptr<T>& operator=(const scoped_refptr<T>& r) {
>+    return *this = r.ptr_;
>+  }
>+
>+  template <typename U>
>+  scoped_refptr<T>& operator=(const scoped_refptr<U>& r) {
>+    return *this = r.get();
>+  }
>+
>+  scoped_refptr<T>& operator=(scoped_refptr<T>&& r) {
>+    scoped_refptr<T>(std::move(r)).swap(*this);
>+    return *this;
>+  }
>+
>+  template <typename U>
>+  scoped_refptr<T>& operator=(scoped_refptr<U>&& r) {
>+    scoped_refptr<T>(std::move(r)).swap(*this);
>+    return *this;
>+  }
>+
>+  void swap(T** pp) {
>+    T* p = ptr_;
>+    ptr_ = *pp;
>+    *pp = p;
>+  }
>+
>+  void swap(scoped_refptr<T>& r) { swap(&r.ptr_); }
>+
>+ protected:
>+  T* ptr_;
>+};
>+
>+}  // namespace rtc
>+
>+#endif  // API_SCOPED_REFPTR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats.h
>index 1705f6ab58ee79ddf69a5a576f65f72d2eaa0f6e..7fd0d84e0bb145581b78c5b2059f7d980a65970d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats.h
>@@ -18,6 +18,7 @@
> #include <vector>
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -47,7 +48,7 @@ class RTCStatsMemberInterface;
> // for (const RTCStatsMemberInterface* member : foo.Members()) {
> //   printf("%s = %s\n", member->name(), member->ValueToString().c_str());
> // }
>-class RTCStats {
>+class RTC_EXPORT RTCStats {
>  public:
>   RTCStats(const std::string& id, int64_t timestamp_us)
>       : id_(id), timestamp_us_(timestamp_us) {}
>@@ -247,7 +248,7 @@ class RTCStatsMemberInterface {
> // (undefined reference to |kType|). The supported types are the ones described
> // by |RTCStatsMemberInterface::Type|.
> template <typename T>
>-class RTCStatsMember : public RTCStatsMemberInterface {
>+class RTC_EXPORT RTCStatsMember : public RTCStatsMemberInterface {
>  public:
>   static const Type kType;
> 
>@@ -294,15 +295,6 @@ class RTCStatsMember : public RTCStatsMemberInterface {
>     is_defined_ = true;
>     return value_;
>   }
>-  T& operator=(const RTCStatsMember<T>& other) {
>-    RTC_DCHECK(other.is_defined_);
>-    // Shouldn't be attempting to assign an RTCNonStandardStatsMember to an
>-    // RTCStatsMember or vice versa.
>-    RTC_DCHECK(is_standardized() == other.is_standardized());
>-    value_ = other.value_;
>-    is_defined_ = true;
>-    return value_;
>-  }
> 
>   // Value getters.
>   T& operator*() {
>@@ -347,6 +339,11 @@ class RTCNonStandardStatsMember : public RTCStatsMember<T> {
>       : RTCStatsMember<T>(std::move(other)) {}
> 
>   bool is_standardized() const override { return false; }
>+
>+  T& operator=(const T& value) { return RTCStatsMember<T>::operator=(value); }
>+  T& operator=(const T&& value) {
>+    return RTCStatsMember<T>::operator=(std::move(value));
>+  }
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats_objects.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats_objects.h
>index f377809ba3a71da62505987a1b5697797c4f5804..8392231218953331da532ed8d2914fb554e8217f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats_objects.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats_objects.h
>@@ -16,6 +16,7 @@
> #include <vector>
> 
> #include "api/stats/rtcstats.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -73,7 +74,7 @@ struct RTCNetworkType {
> };
> 
> // https://w3c.github.io/webrtc-stats/#certificatestats-dict*
>-class RTCCertificateStats final : public RTCStats {
>+class RTC_EXPORT RTCCertificateStats final : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>@@ -89,7 +90,7 @@ class RTCCertificateStats final : public RTCStats {
> };
> 
> // https://w3c.github.io/webrtc-stats/#codec-dict*
>-class RTCCodecStats final : public RTCStats {
>+class RTC_EXPORT RTCCodecStats final : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>@@ -110,7 +111,7 @@ class RTCCodecStats final : public RTCStats {
> };
> 
> // https://w3c.github.io/webrtc-stats/#dcstats-dict*
>-class RTCDataChannelStats final : public RTCStats {
>+class RTC_EXPORT RTCDataChannelStats final : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>@@ -132,7 +133,7 @@ class RTCDataChannelStats final : public RTCStats {
> 
> // https://w3c.github.io/webrtc-stats/#candidatepair-dict*
> // TODO(hbos): Tracking bug https://bugs.webrtc.org/7062
>-class RTCIceCandidatePairStats final : public RTCStats {
>+class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>@@ -186,7 +187,7 @@ class RTCIceCandidatePairStats final : public RTCStats {
> // crbug.com/632723
> // TODO(qingsi): Add the stats of STUN binding requests (keepalives) and collect
> // them in the new PeerConnection::GetStats.
>-class RTCIceCandidateStats : public RTCStats {
>+class RTC_EXPORT RTCIceCandidateStats : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>@@ -221,7 +222,7 @@ class RTCIceCandidateStats : public RTCStats {
> // |kType| need to be different ("RTCStatsType type") in the local/remote case.
> // https://w3c.github.io/webrtc-stats/#rtcstatstype-str*
> // This forces us to have to override copy() and type().
>-class RTCLocalIceCandidateStats final : public RTCIceCandidateStats {
>+class RTC_EXPORT RTCLocalIceCandidateStats final : public RTCIceCandidateStats {
>  public:
>   static const char kType[];
>   RTCLocalIceCandidateStats(const std::string& id, int64_t timestamp_us);
>@@ -230,7 +231,8 @@ class RTCLocalIceCandidateStats final : public RTCIceCandidateStats {
>   const char* type() const override;
> };
> 
>-class RTCRemoteIceCandidateStats final : public RTCIceCandidateStats {
>+class RTC_EXPORT RTCRemoteIceCandidateStats final
>+    : public RTCIceCandidateStats {
>  public:
>   static const char kType[];
>   RTCRemoteIceCandidateStats(const std::string& id, int64_t timestamp_us);
>@@ -241,7 +243,7 @@ class RTCRemoteIceCandidateStats final : public RTCIceCandidateStats {
> 
> // https://w3c.github.io/webrtc-stats/#msstats-dict*
> // TODO(hbos): Tracking bug crbug.com/660827
>-class RTCMediaStreamStats final : public RTCStats {
>+class RTC_EXPORT RTCMediaStreamStats final : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>@@ -256,7 +258,7 @@ class RTCMediaStreamStats final : public RTCStats {
> 
> // https://w3c.github.io/webrtc-stats/#mststats-dict*
> // TODO(hbos): Tracking bug crbug.com/659137
>-class RTCMediaStreamTrackStats final : public RTCStats {
>+class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>@@ -306,10 +308,14 @@ class RTCMediaStreamTrackStats final : public RTCStats {
>   RTCStatsMember<double> total_samples_duration;
>   RTCStatsMember<uint64_t> concealed_samples;
>   RTCStatsMember<uint64_t> concealment_events;
>+  // Non-standard audio-only member
>+  // TODO(kuddai): Add description to standard. crbug.com/webrtc/10042
>+  RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
>+  RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
> };
> 
> // https://w3c.github.io/webrtc-stats/#pcstats-dict*
>-class RTCPeerConnectionStats final : public RTCStats {
>+class RTC_EXPORT RTCPeerConnectionStats final : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>@@ -324,7 +330,7 @@ class RTCPeerConnectionStats final : public RTCStats {
> 
> // https://w3c.github.io/webrtc-stats/#streamstats-dict*
> // TODO(hbos): Tracking bug crbug.com/657854
>-class RTCRTPStreamStats : public RTCStats {
>+class RTC_EXPORT RTCRTPStreamStats : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>@@ -362,7 +368,7 @@ class RTCRTPStreamStats : public RTCStats {
> // https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
> // TODO(hbos): Support the remote case |is_remote = true|.
> // https://bugs.webrtc.org/7065
>-class RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
>+class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>@@ -406,7 +412,7 @@ class RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
> // https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
> // TODO(hbos): Support the remote case |is_remote = true|.
> // https://bugs.webrtc.org/7066
>-class RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
>+class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>@@ -423,7 +429,7 @@ class RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
> };
> 
> // https://w3c.github.io/webrtc-stats/#transportstats-dict*
>-class RTCTransportStats final : public RTCStats {
>+class RTC_EXPORT RTCTransportStats final : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatsreport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatsreport.h
>index f7410b39f79affa2376366ebcf93be614aec302f..8bf1cb57d9d55ae50432b256355dd21f284fb436 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatsreport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatsreport.h
>@@ -20,12 +20,13 @@
> #include "rtc_base/refcount.h"
> #include "rtc_base/refcountedobject.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // A collection of stats.
> // This is accessible as a map from |RTCStats::id| to |RTCStats|.
>-class RTCStatsReport : public rtc::RefCountInterface {
>+class RTC_EXPORT RTCStatsReport : public rtc::RefCountInterface {
>  public:
>   typedef std::map<std::string, std::unique_ptr<const RTCStats>> StatsMap;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/DEPS
>index 98b1ad391fca2b644c61a3f50328dd5efede2edf..9c293b323e703fc6cd49d5d9830cab7f8f54f98d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/DEPS
>@@ -5,4 +5,10 @@ specific_include_rules = {
>   ".*": [
>     "+video"
>   ],
>+  "loopback_media_transport\.h": [
>+    "+rtc_base/asyncinvoker.h",
>+    "+rtc_base/criticalsection.h",
>+    "+rtc_base/thread.h",
>+    "+rtc_base/thread_checker.h",
>+  ],
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_decryptor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_decryptor.cc
>index 432664a03063c8b58ed243694e3cdb02b5cb93da..b77017fdb43684003fe552f3b8446bc09361a908 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_decryptor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_decryptor.cc
>@@ -25,19 +25,20 @@ int FakeFrameDecryptor::Decrypt(cricket::MediaType media_type,
>                                 rtc::ArrayView<uint8_t> frame,
>                                 size_t* bytes_written) {
>   if (fail_decryption_) {
>-    return 1;
>+    return static_cast<int>(FakeDecryptStatus::FORCED_FAILURE);
>   }
> 
>   RTC_CHECK_EQ(frame.size() + 1, encrypted_frame.size());
>   for (size_t i = 0; i < frame.size(); i++) {
>-    frame[i] ^= fake_key_;
>+    frame[i] = encrypted_frame[i] ^ fake_key_;
>   }
> 
>   if (encrypted_frame[frame.size()] != expected_postfix_byte_) {
>-    return 1;
>+    return static_cast<int>(FakeDecryptStatus::INVALID_POSTFIX);
>   }
> 
>-  return 0;
>+  *bytes_written = frame.size();
>+  return static_cast<int>(FakeDecryptStatus::OK);
> }
> 
> size_t FakeFrameDecryptor::GetMaxPlaintextByteSize(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_decryptor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_decryptor.h
>index b945def59a8b0a5b74745f52b74208bf165de5c3..bcb8080fd4c84c66b335242c093148d2983b709a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_decryptor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_decryptor.h
>@@ -22,34 +22,40 @@ namespace webrtc {
> // FrameDecryptorInterface. It is constructed with a simple single digit key and
> // a fixed postfix byte. This is just to validate that the core code works
> // as expected.
>-class FakeFrameDecryptor
>+class FakeFrameDecryptor final
>     : public rtc::RefCountedObject<FrameDecryptorInterface> {
>  public:
>   // Provide a key (0,255) and some postfix byte (0,255) this should match the
>   // byte you expect from the FakeFrameEncryptor.
>-  explicit FakeFrameDecryptor(uint8_t fake_key = 1,
>+  explicit FakeFrameDecryptor(uint8_t fake_key = 0xAA,
>                               uint8_t expected_postfix_byte = 255);
>-
>-  // FrameDecryptorInterface implementation
>+  // Fake decryption that just xors the payload with the 1 byte key and checks
>+  // the postfix byte. This will always fail if fail_decryption_ is set to true.
>   int Decrypt(cricket::MediaType media_type,
>               const std::vector<uint32_t>& csrcs,
>               rtc::ArrayView<const uint8_t> additional_data,
>               rtc::ArrayView<const uint8_t> encrypted_frame,
>               rtc::ArrayView<uint8_t> frame,
>               size_t* bytes_written) override;
>-
>+  // Always returns 1 less than the size of the encrypted frame.
>   size_t GetMaxPlaintextByteSize(cricket::MediaType media_type,
>                                  size_t encrypted_frame_size) override;
>-
>+  // Sets the fake key to use for encryption.
>   void SetFakeKey(uint8_t fake_key);
>-
>+  // Returns the fake key used for encryption.
>   uint8_t GetFakeKey() const;
>-
>+  // Set the Postfix byte that is expected in the encrypted payload.
>   void SetExpectedPostfixByte(uint8_t expected_postfix_byte);
>-
>+  // Returns the postfix byte that will be checked for in the encrypted payload.
>   uint8_t GetExpectedPostfixByte() const;
>-
>+  // If set to true will force all encryption to fail.
>   void SetFailDecryption(bool fail_decryption);
>+  // Simple error codes for tests to validate against.
>+  enum class FakeDecryptStatus : int {
>+    OK = 0,
>+    FORCED_FAILURE = 1,
>+    INVALID_POSTFIX = 2
>+  };
> 
>  private:
>   uint8_t fake_key_ = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_encryptor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_encryptor.cc
>index 013058f96aa6dfe9b0613938e57daef5da38c36c..edf3cc2136ad00f785d76f7a490b28554ad2d53d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_encryptor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_encryptor.cc
>@@ -22,18 +22,18 @@ int FakeFrameEncryptor::Encrypt(cricket::MediaType media_type,
>                                 rtc::ArrayView<const uint8_t> frame,
>                                 rtc::ArrayView<uint8_t> encrypted_frame,
>                                 size_t* bytes_written) {
>-  // Useful if you want to test failure cases.
>   if (fail_encryption_) {
>-    return 1;
>+    return static_cast<int>(FakeEncryptionStatus::FORCED_FAILURE);
>   }
> 
>   RTC_CHECK_EQ(frame.size() + 1, encrypted_frame.size());
>   for (size_t i = 0; i < frame.size(); i++) {
>-    encrypted_frame[i] ^= fake_key_;
>+    encrypted_frame[i] = frame[i] ^ fake_key_;
>   }
>+
>   encrypted_frame[frame.size()] = postfix_byte_;
>   *bytes_written = encrypted_frame.size();
>-  return 0;
>+  return static_cast<int>(FakeEncryptionStatus::OK);
> }
> 
> size_t FakeFrameEncryptor::GetMaxCiphertextByteSize(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_encryptor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_encryptor.h
>index 61ae9383f6a54390997bb14c3b096822fee0d4e4..0bec967dc229646eaafdb1bc6ebcf9b6660a7afb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_encryptor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_frame_encryptor.h
>@@ -24,29 +24,35 @@ class FakeFrameEncryptor
>     : public rtc::RefCountedObject<FrameEncryptorInterface> {
>  public:
>   // Provide a key (0,255) and some postfix byte (0,255).
>-  explicit FakeFrameEncryptor(uint8_t fake_key = 1, uint8_t postfix_byte = 255);
>-
>-  // FrameEncryptorInterface implementation
>+  explicit FakeFrameEncryptor(uint8_t fake_key = 0xAA,
>+                              uint8_t postfix_byte = 255);
>+  // Simply xors each payload with the provided fake key and adds the postfix
>+  // bit to the end. This will always fail if fail_encryption_ is set to true.
>   int Encrypt(cricket::MediaType media_type,
>               uint32_t ssrc,
>               rtc::ArrayView<const uint8_t> additional_data,
>               rtc::ArrayView<const uint8_t> frame,
>               rtc::ArrayView<uint8_t> encrypted_frame,
>               size_t* bytes_written) override;
>-
>+  // Always returns 1 more than the size of the frame.
>   size_t GetMaxCiphertextByteSize(cricket::MediaType media_type,
>                                   size_t frame_size) override;
>-
>+  // Sets the fake key to use during encryption.
>   void SetFakeKey(uint8_t fake_key);
>-
>+  // Returns the fake key used during encryption.
>   uint8_t GetFakeKey() const;
>-
>+  // Set the postfix byte to use.
>   void SetPostfixByte(uint8_t expected_postfix_byte);
>-
>+  // Return a postfix byte added to each outgoing payload.
>   uint8_t GetPostfixByte() const;
>-
>+  // Force all encryptions to fail.
>   void SetFailEncryption(bool fail_encryption);
> 
>+  enum class FakeEncryptionStatus : int {
>+    OK = 0,
>+    FORCED_FAILURE = 1,
>+  };
>+
>  private:
>   uint8_t fake_key_ = 0;
>   uint8_t postfix_byte_ = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_media_transport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_media_transport.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..eb6ec93ae2b7922b8223d9891a1d358d8a225534
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fake_media_transport.h
>@@ -0,0 +1,131 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_FAKE_MEDIA_TRANSPORT_H_
>+#define API_TEST_FAKE_MEDIA_TRANSPORT_H_
>+
>+#include <memory>
>+#include <string>
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/memory/memory.h"
>+#include "api/media_transport_interface.h"
>+
>+namespace webrtc {
>+
>+// TODO(sukhanov): For now fake media transport does nothing and is used only
>+// in jsepcontroller unittests. In the future we should implement fake media
>+// transport, which forwards frames to another fake media transport, so we
>+// could unit test audio / video integration.
>+class FakeMediaTransport : public MediaTransportInterface {
>+ public:
>+  explicit FakeMediaTransport(const MediaTransportSettings& settings)
>+      : settings_(settings) {}
>+  ~FakeMediaTransport() = default;
>+
>+  RTCError SendAudioFrame(uint64_t channel_id,
>+                          MediaTransportEncodedAudioFrame frame) override {
>+    return RTCError::OK();
>+  }
>+
>+  RTCError SendVideoFrame(
>+      uint64_t channel_id,
>+      const MediaTransportEncodedVideoFrame& frame) override {
>+    return RTCError::OK();
>+  }
>+
>+  RTCError RequestKeyFrame(uint64_t channel_id) override {
>+    return RTCError::OK();
>+  };
>+
>+  void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) override {}
>+  void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) override {}
>+
>+  // Returns true if fake media transport was created as a caller.
>+  bool is_caller() const { return settings_.is_caller; }
>+  absl::optional<std::string> pre_shared_key() const {
>+    return settings_.pre_shared_key;
>+  }
>+
>+  RTCError SendData(int channel_id,
>+                    const SendDataParams& params,
>+                    const rtc::CopyOnWriteBuffer& buffer) override {
>+    return RTCError::OK();
>+  }
>+
>+  RTCError CloseChannel(int channel_id) override { return RTCError::OK(); }
>+
>+  void SetDataSink(DataChannelSink* sink) override {}
>+
>+  void SetMediaTransportStateCallback(
>+      MediaTransportStateCallback* callback) override {
>+    state_callback_ = callback;
>+  }
>+
>+  void SetState(webrtc::MediaTransportState state) {
>+    if (state_callback_) {
>+      state_callback_->OnStateChanged(state);
>+    }
>+  }
>+
>+  void AddTargetTransferRateObserver(
>+      webrtc::TargetTransferRateObserver* observer) override {
>+    RTC_CHECK(std::find(target_rate_observers_.begin(),
>+                        target_rate_observers_.end(),
>+                        observer) == target_rate_observers_.end());
>+    target_rate_observers_.push_back(observer);
>+  }
>+
>+  void RemoveTargetTransferRateObserver(
>+      webrtc::TargetTransferRateObserver* observer) override {
>+    auto it = std::find(target_rate_observers_.begin(),
>+                        target_rate_observers_.end(), observer);
>+    if (it != target_rate_observers_.end()) {
>+      target_rate_observers_.erase(it);
>+    }
>+  }
>+
>+  int target_rate_observers_size() { return target_rate_observers_.size(); }
>+
>+ private:
>+  const MediaTransportSettings settings_;
>+  MediaTransportStateCallback* state_callback_;
>+  std::vector<webrtc::TargetTransferRateObserver*> target_rate_observers_;
>+};
>+
>+// Fake media transport factory creates fake media transport.
>+class FakeMediaTransportFactory : public MediaTransportFactory {
>+ public:
>+  FakeMediaTransportFactory() = default;
>+  ~FakeMediaTransportFactory() = default;
>+
>+  RTCErrorOr<std::unique_ptr<MediaTransportInterface>> CreateMediaTransport(
>+      rtc::PacketTransportInternal* packet_transport,
>+      rtc::Thread* network_thread,
>+      bool is_caller) override {
>+    MediaTransportSettings settings;
>+    settings.is_caller = is_caller;
>+    return CreateMediaTransport(packet_transport, network_thread, settings);
>+  }
>+
>+  RTCErrorOr<std::unique_ptr<MediaTransportInterface>> CreateMediaTransport(
>+      rtc::PacketTransportInternal* packet_transport,
>+      rtc::Thread* network_thread,
>+      const MediaTransportSettings& settings) override {
>+    std::unique_ptr<MediaTransportInterface> media_transport =
>+        absl::make_unique<FakeMediaTransport>(settings);
>+    return std::move(media_transport);
>+  }
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_FAKE_MEDIA_TRANSPORT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/loopback_media_transport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/loopback_media_transport.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..48255b135e7508cab00496630a537fdc379b473c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/loopback_media_transport.h
>@@ -0,0 +1,291 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
>+#define API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
>+
>+#include <memory>
>+#include <utility>
>+
>+#include "api/media_transport_interface.h"
>+#include "rtc_base/asyncinvoker.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/thread.h"
>+#include "rtc_base/thread_checker.h"
>+
>+namespace webrtc {
>+
>+// Wrapper used to hand out unique_ptrs to loopback media transports without
>+// ownership changes.
>+class WrapperMediaTransport : public MediaTransportInterface {
>+ public:
>+  explicit WrapperMediaTransport(MediaTransportInterface* wrapped)
>+      : wrapped_(wrapped) {}
>+
>+  RTCError SendAudioFrame(uint64_t channel_id,
>+                          MediaTransportEncodedAudioFrame frame) override {
>+    return wrapped_->SendAudioFrame(channel_id, std::move(frame));
>+  }
>+
>+  RTCError SendVideoFrame(
>+      uint64_t channel_id,
>+      const MediaTransportEncodedVideoFrame& frame) override {
>+    return wrapped_->SendVideoFrame(channel_id, frame);
>+  }
>+
>+  RTCError RequestKeyFrame(uint64_t channel_id) override {
>+    return wrapped_->RequestKeyFrame(channel_id);
>+  }
>+
>+  void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) override {
>+    wrapped_->SetReceiveAudioSink(sink);
>+  }
>+
>+  void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) override {
>+    wrapped_->SetReceiveVideoSink(sink);
>+  }
>+
>+  void SetMediaTransportStateCallback(
>+      MediaTransportStateCallback* callback) override {
>+    wrapped_->SetMediaTransportStateCallback(callback);
>+  }
>+
>+  RTCError SendData(int channel_id,
>+                    const SendDataParams& params,
>+                    const rtc::CopyOnWriteBuffer& buffer) override {
>+    return wrapped_->SendData(channel_id, params, buffer);
>+  }
>+
>+  RTCError CloseChannel(int channel_id) override {
>+    return wrapped_->CloseChannel(channel_id);
>+  }
>+
>+  void SetDataSink(DataChannelSink* sink) override {
>+    wrapped_->SetDataSink(sink);
>+  }
>+
>+ private:
>+  MediaTransportInterface* wrapped_;
>+};
>+
>+class WrapperMediaTransportFactory : public MediaTransportFactory {
>+ public:
>+  explicit WrapperMediaTransportFactory(MediaTransportInterface* wrapped)
>+      : wrapped_(wrapped) {}
>+
>+  RTCErrorOr<std::unique_ptr<MediaTransportInterface>> CreateMediaTransport(
>+      rtc::PacketTransportInternal* packet_transport,
>+      rtc::Thread* network_thread,
>+      const MediaTransportSettings& settings) override {
>+    return {absl::make_unique<WrapperMediaTransport>(wrapped_)};
>+  }
>+
>+ private:
>+  MediaTransportInterface* wrapped_;
>+};
>+
>+// Contains two MediaTransportsInterfaces that are connected to each other.
>+// Currently supports audio only.
>+class MediaTransportPair {
>+ public:
>+  struct Stats {
>+    int sent_audio_frames = 0;
>+    int received_audio_frames = 0;
>+  };
>+
>+  explicit MediaTransportPair(rtc::Thread* thread)
>+      : first_(thread, &second_), second_(thread, &first_) {}
>+
>+  // Ownership stays with MediaTransportPair
>+  MediaTransportInterface* first() { return &first_; }
>+  MediaTransportInterface* second() { return &second_; }
>+
>+  std::unique_ptr<MediaTransportFactory> first_factory() {
>+    return absl::make_unique<WrapperMediaTransportFactory>(&first_);
>+  }
>+
>+  std::unique_ptr<MediaTransportFactory> second_factory() {
>+    return absl::make_unique<WrapperMediaTransportFactory>(&second_);
>+  }
>+
>+  void SetState(MediaTransportState state) {
>+    first_.SetState(state);
>+    second_.SetState(state);
>+  }
>+
>+  void FlushAsyncInvokes() {
>+    first_.FlushAsyncInvokes();
>+    second_.FlushAsyncInvokes();
>+  }
>+
>+  Stats FirstStats() { return first_.GetStats(); }
>+  Stats SecondStats() { return second_.GetStats(); }
>+
>+ private:
>+  class LoopbackMediaTransport : public MediaTransportInterface {
>+   public:
>+    LoopbackMediaTransport(rtc::Thread* thread, LoopbackMediaTransport* other)
>+        : thread_(thread), other_(other) {}
>+
>+    ~LoopbackMediaTransport() {
>+      rtc::CritScope lock(&sink_lock_);
>+      RTC_CHECK(sink_ == nullptr);
>+      RTC_CHECK(data_sink_ == nullptr);
>+    }
>+
>+    RTCError SendAudioFrame(uint64_t channel_id,
>+                            MediaTransportEncodedAudioFrame frame) override {
>+      {
>+        rtc::CritScope lock(&stats_lock_);
>+        ++stats_.sent_audio_frames;
>+      }
>+      invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_,
>+                                 [this, channel_id, frame] {
>+                                   other_->OnData(channel_id, std::move(frame));
>+                                 });
>+      return RTCError::OK();
>+    };
>+
>+    RTCError SendVideoFrame(
>+        uint64_t channel_id,
>+        const MediaTransportEncodedVideoFrame& frame) override {
>+      return RTCError::OK();
>+    }
>+
>+    RTCError RequestKeyFrame(uint64_t channel_id) override {
>+      return RTCError::OK();
>+    }
>+
>+    void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) override {
>+      rtc::CritScope lock(&sink_lock_);
>+      if (sink) {
>+        RTC_CHECK(sink_ == nullptr);
>+      }
>+      sink_ = sink;
>+    }
>+
>+    void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) override {}
>+
>+    void SetMediaTransportStateCallback(
>+        MediaTransportStateCallback* callback) override {
>+      rtc::CritScope lock(&sink_lock_);
>+      state_callback_ = callback;
>+      invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this] {
>+        RTC_DCHECK_RUN_ON(thread_);
>+        OnStateChanged();
>+      });
>+    }
>+
>+    RTCError SendData(int channel_id,
>+                      const SendDataParams& params,
>+                      const rtc::CopyOnWriteBuffer& buffer) override {
>+      invoker_.AsyncInvoke<void>(
>+          RTC_FROM_HERE, thread_, [this, channel_id, params, buffer] {
>+            other_->OnData(channel_id, params.type, buffer);
>+          });
>+      return RTCError::OK();
>+    }
>+
>+    RTCError CloseChannel(int channel_id) override {
>+      invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this, channel_id] {
>+        other_->OnRemoteCloseChannel(channel_id);
>+        rtc::CritScope lock(&sink_lock_);
>+        if (data_sink_) {
>+          data_sink_->OnChannelClosed(channel_id);
>+        }
>+      });
>+      return RTCError::OK();
>+    }
>+
>+    void SetDataSink(DataChannelSink* sink) override {
>+      rtc::CritScope lock(&sink_lock_);
>+      data_sink_ = sink;
>+    }
>+
>+    void SetState(MediaTransportState state) {
>+      invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this, state] {
>+        RTC_DCHECK_RUN_ON(thread_);
>+        state_ = state;
>+        OnStateChanged();
>+      });
>+    }
>+
>+    void FlushAsyncInvokes() { invoker_.Flush(thread_); }
>+
>+    Stats GetStats() {
>+      rtc::CritScope lock(&stats_lock_);
>+      return stats_;
>+    }
>+
>+   private:
>+    void OnData(uint64_t channel_id, MediaTransportEncodedAudioFrame frame) {
>+      {
>+        rtc::CritScope lock(&sink_lock_);
>+        if (sink_) {
>+          sink_->OnData(channel_id, frame);
>+        }
>+      }
>+      {
>+        rtc::CritScope lock(&stats_lock_);
>+        ++stats_.received_audio_frames;
>+      }
>+    }
>+
>+    void OnData(int channel_id,
>+                DataMessageType type,
>+                const rtc::CopyOnWriteBuffer& buffer) {
>+      rtc::CritScope lock(&sink_lock_);
>+      if (data_sink_) {
>+        data_sink_->OnDataReceived(channel_id, type, buffer);
>+      }
>+    }
>+
>+    void OnRemoteCloseChannel(int channel_id) {
>+      rtc::CritScope lock(&sink_lock_);
>+      if (data_sink_) {
>+        data_sink_->OnChannelClosing(channel_id);
>+        data_sink_->OnChannelClosed(channel_id);
>+      }
>+    }
>+
>+    void OnStateChanged() RTC_RUN_ON(thread_) {
>+      rtc::CritScope lock(&sink_lock_);
>+      if (state_callback_) {
>+        state_callback_->OnStateChanged(state_);
>+      }
>+    }
>+
>+    rtc::Thread* const thread_;
>+    rtc::CriticalSection sink_lock_;
>+    rtc::CriticalSection stats_lock_;
>+
>+    MediaTransportAudioSinkInterface* sink_ RTC_GUARDED_BY(sink_lock_) =
>+        nullptr;
>+    DataChannelSink* data_sink_ RTC_GUARDED_BY(sink_lock_) = nullptr;
>+    MediaTransportStateCallback* state_callback_ RTC_GUARDED_BY(sink_lock_) =
>+        nullptr;
>+
>+    MediaTransportState state_ RTC_GUARDED_BY(thread_) =
>+        MediaTransportState::kPending;
>+
>+    LoopbackMediaTransport* const other_;
>+
>+    Stats stats_ RTC_GUARDED_BY(stats_lock_);
>+
>+    rtc::AsyncInvoker invoker_;
>+  };
>+
>+  LoopbackMediaTransport first_;
>+  LoopbackMediaTransport second_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/loopback_media_transport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/loopback_media_transport_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..ba741a05ca79862d68340e825d7ccd50deaf27b7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/loopback_media_transport_unittest.cc
>@@ -0,0 +1,171 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "api/test/loopback_media_transport.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+class MockMediaTransportAudioSinkInterface
>+    : public MediaTransportAudioSinkInterface {
>+ public:
>+  MOCK_METHOD2(OnData, void(uint64_t, MediaTransportEncodedAudioFrame));
>+};
>+
>+class MockDataChannelSink : public DataChannelSink {
>+ public:
>+  MOCK_METHOD3(OnDataReceived,
>+               void(int, DataMessageType, const rtc::CopyOnWriteBuffer&));
>+  MOCK_METHOD1(OnChannelClosing, void(int));
>+  MOCK_METHOD1(OnChannelClosed, void(int));
>+};
>+
>+class MockStateCallback : public MediaTransportStateCallback {
>+ public:
>+  MOCK_METHOD1(OnStateChanged, void(MediaTransportState));
>+};
>+
>+// Test only uses the sequence number.
>+MediaTransportEncodedAudioFrame CreateAudioFrame(int sequence_number) {
>+  static constexpr int kSamplingRateHz = 48000;
>+  static constexpr int kStartingSampleIndex = 0;
>+  static constexpr int kSamplesPerChannel = 480;
>+  static constexpr uint8_t kPayloadType = 17;
>+
>+  return MediaTransportEncodedAudioFrame(
>+      kSamplingRateHz, kStartingSampleIndex, kSamplesPerChannel,
>+      sequence_number, MediaTransportEncodedAudioFrame::FrameType::kSpeech,
>+      kPayloadType, std::vector<uint8_t>(kSamplesPerChannel));
>+}
>+
>+}  // namespace
>+
>+TEST(LoopbackMediaTransport, AudioWithNoSinkSilentlyIgnored) {
>+  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
>+  thread->Start();
>+  MediaTransportPair transport_pair(thread.get());
>+  transport_pair.first()->SendAudioFrame(1, CreateAudioFrame(0));
>+  transport_pair.second()->SendAudioFrame(2, CreateAudioFrame(0));
>+  transport_pair.FlushAsyncInvokes();
>+}
>+
>+TEST(LoopbackMediaTransport, AudioDeliveredToSink) {
>+  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
>+  thread->Start();
>+  MediaTransportPair transport_pair(thread.get());
>+  testing::StrictMock<MockMediaTransportAudioSinkInterface> sink;
>+  EXPECT_CALL(sink,
>+              OnData(1, testing::Property(
>+                            &MediaTransportEncodedAudioFrame::sequence_number,
>+                            testing::Eq(10))));
>+  transport_pair.second()->SetReceiveAudioSink(&sink);
>+  transport_pair.first()->SendAudioFrame(1, CreateAudioFrame(10));
>+
>+  transport_pair.FlushAsyncInvokes();
>+  transport_pair.second()->SetReceiveAudioSink(nullptr);
>+}
>+
>+TEST(LoopbackMediaTransport, DataDeliveredToSink) {
>+  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
>+  thread->Start();
>+  MediaTransportPair transport_pair(thread.get());
>+
>+  MockDataChannelSink sink;
>+  transport_pair.first()->SetDataSink(&sink);
>+
>+  const int channel_id = 1;
>+  EXPECT_CALL(sink,
>+              OnDataReceived(
>+                  channel_id, DataMessageType::kText,
>+                  testing::Property<rtc::CopyOnWriteBuffer, const char*>(
>+                      &rtc::CopyOnWriteBuffer::cdata, testing::StrEq("foo"))));
>+
>+  SendDataParams params;
>+  params.type = DataMessageType::kText;
>+  rtc::CopyOnWriteBuffer buffer("foo");
>+  transport_pair.second()->SendData(channel_id, params, buffer);
>+
>+  transport_pair.FlushAsyncInvokes();
>+  transport_pair.first()->SetDataSink(nullptr);
>+}
>+
>+TEST(LoopbackMediaTransport, CloseDeliveredToSink) {
>+  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
>+  thread->Start();
>+  MediaTransportPair transport_pair(thread.get());
>+
>+  MockDataChannelSink first_sink;
>+  transport_pair.first()->SetDataSink(&first_sink);
>+
>+  MockDataChannelSink second_sink;
>+  transport_pair.second()->SetDataSink(&second_sink);
>+
>+  const int channel_id = 1;
>+  {
>+    testing::InSequence s;
>+    EXPECT_CALL(second_sink, OnChannelClosing(channel_id));
>+    EXPECT_CALL(second_sink, OnChannelClosed(channel_id));
>+    EXPECT_CALL(first_sink, OnChannelClosed(channel_id));
>+  }
>+
>+  transport_pair.first()->CloseChannel(channel_id);
>+
>+  transport_pair.FlushAsyncInvokes();
>+  transport_pair.first()->SetDataSink(nullptr);
>+  transport_pair.second()->SetDataSink(nullptr);
>+}
>+
>+TEST(LoopbackMediaTransport, InitialStateDeliveredWhenCallbackSet) {
>+  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
>+  thread->Start();
>+  MediaTransportPair transport_pair(thread.get());
>+
>+  MockStateCallback state_callback;
>+
>+  EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kPending));
>+  transport_pair.first()->SetMediaTransportStateCallback(&state_callback);
>+  transport_pair.FlushAsyncInvokes();
>+}
>+
>+TEST(LoopbackMediaTransport, ChangedStateDeliveredWhenCallbackSet) {
>+  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
>+  thread->Start();
>+  MediaTransportPair transport_pair(thread.get());
>+
>+  transport_pair.SetState(MediaTransportState::kWritable);
>+  transport_pair.FlushAsyncInvokes();
>+
>+  MockStateCallback state_callback;
>+
>+  EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kWritable));
>+  transport_pair.first()->SetMediaTransportStateCallback(&state_callback);
>+  transport_pair.FlushAsyncInvokes();
>+}
>+
>+TEST(LoopbackMediaTransport, StateChangeDeliveredToCallback) {
>+  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
>+  thread->Start();
>+  MediaTransportPair transport_pair(thread.get());
>+
>+  MockStateCallback state_callback;
>+
>+  EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kPending));
>+  EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kWritable));
>+  transport_pair.first()->SetMediaTransportStateCallback(&state_callback);
>+  transport_pair.SetState(MediaTransportState::kWritable);
>+  transport_pair.FlushAsyncInvokes();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_decryptor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_decryptor.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..f4b54f966cd8c06158566a4b50573529549f0aec
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_decryptor.cc
>@@ -0,0 +1,18 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/test/mock_frame_decryptor.h"
>+
>+namespace webrtc {
>+
>+MockFrameDecryptor::MockFrameDecryptor() = default;
>+MockFrameDecryptor::~MockFrameDecryptor() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_decryptor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_decryptor.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..e80396554f4e05bf72c8f7ccdd67862984ad1dc1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_decryptor.h
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_MOCK_FRAME_DECRYPTOR_H_
>+#define API_TEST_MOCK_FRAME_DECRYPTOR_H_
>+
>+#include <vector>
>+
>+#include "api/crypto/framedecryptorinterface.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+class MockFrameDecryptor : public FrameDecryptorInterface {
>+ public:
>+  MockFrameDecryptor();
>+  ~MockFrameDecryptor() override;
>+
>+  MOCK_METHOD6(Decrypt,
>+               int(cricket::MediaType,
>+                   const std::vector<uint32_t>&,
>+                   rtc::ArrayView<const uint8_t>,
>+                   rtc::ArrayView<const uint8_t>,
>+                   rtc::ArrayView<uint8_t>,
>+                   size_t*));
>+
>+  MOCK_METHOD2(GetMaxPlaintextByteSize,
>+               size_t(cricket::MediaType, size_t encrypted_frame_size));
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_MOCK_FRAME_DECRYPTOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_encryptor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_encryptor.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..0972259f057ec72b471df07312a04a4918dee26c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_encryptor.cc
>@@ -0,0 +1,19 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/test/mock_frame_encryptor.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+MockFrameEncryptor::MockFrameEncryptor() = default;
>+MockFrameEncryptor::~MockFrameEncryptor() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_encryptor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_encryptor.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..c4fb1fde87b825a17a73fdf4e6060dda6f01dbc0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_frame_encryptor.h
>@@ -0,0 +1,38 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_MOCK_FRAME_ENCRYPTOR_H_
>+#define API_TEST_MOCK_FRAME_ENCRYPTOR_H_
>+
>+#include "api/crypto/frameencryptorinterface.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+class MockFrameEncryptor : public FrameEncryptorInterface {
>+ public:
>+  MockFrameEncryptor();
>+  ~MockFrameEncryptor() override;
>+
>+  MOCK_METHOD6(Encrypt,
>+               int(cricket::MediaType,
>+                   uint32_t,
>+                   rtc::ArrayView<const uint8_t>,
>+                   rtc::ArrayView<const uint8_t>,
>+                   rtc::ArrayView<uint8_t>,
>+                   size_t*));
>+
>+  MOCK_METHOD2(GetMaxCiphertextByteSize,
>+               size_t(cricket::MediaType media_type, size_t frame_size));
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_MOCK_FRAME_ENCRYPTOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_bitrate_allocator_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_bitrate_allocator_factory.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..0cae061ab75dcd667570dbb7cf333cf5506f1f84
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_bitrate_allocator_factory.h
>@@ -0,0 +1,37 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_MOCK_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>+#define API_TEST_MOCK_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>+
>+#include <memory>
>+
>+#include "api/video/video_bitrate_allocator_factory.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+class MockVideoBitrateAllocatorFactory
>+    : public webrtc::VideoBitrateAllocatorFactory {
>+ public:
>+  virtual std::unique_ptr<VideoBitrateAllocator> CreateVideoBitrateAllocator(
>+      const VideoCodec& codec) {
>+    return std::unique_ptr<VideoBitrateAllocator>(
>+        CreateVideoBitrateAllocatorProxy(codec));
>+  }
>+  ~MockVideoBitrateAllocatorFactory() { Die(); }
>+  MOCK_METHOD1(CreateVideoBitrateAllocatorProxy,
>+               VideoBitrateAllocator*(const VideoCodec&));
>+  MOCK_METHOD0(Die, void());
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_MOCK_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_decoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_decoder.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..85ed0e16782aa9039a4dea663c9561fe3e7158d3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_decoder.cc
>@@ -0,0 +1,20 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/test/mock_video_decoder.h"
>+
>+namespace webrtc {
>+
>+MockDecodedImageCallback::MockDecodedImageCallback() = default;
>+MockDecodedImageCallback::~MockDecodedImageCallback() = default;
>+MockVideoDecoder::MockVideoDecoder() = default;
>+MockVideoDecoder::~MockVideoDecoder() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_decoder.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..56ff5463fde1407dbd821dfa4e35ea43c31ad0f0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_decoder.h
>@@ -0,0 +1,57 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_MOCK_VIDEO_DECODER_H_
>+#define API_TEST_MOCK_VIDEO_DECODER_H_
>+
>+#include "api/video_codecs/video_decoder.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+class MockDecodedImageCallback : public DecodedImageCallback {
>+ public:
>+  MockDecodedImageCallback();
>+  ~MockDecodedImageCallback() override;
>+
>+  MOCK_METHOD1(Decoded, int32_t(VideoFrame& decodedImage));  // NOLINT
>+  MOCK_METHOD2(Decoded,
>+               int32_t(VideoFrame& decodedImage,  // NOLINT
>+                       int64_t decode_time_ms));
>+  MOCK_METHOD3(Decoded,
>+               void(VideoFrame& decodedImage,  // NOLINT
>+                    absl::optional<int32_t> decode_time_ms,
>+                    absl::optional<uint8_t> qp));
>+  MOCK_METHOD1(ReceivedDecodedReferenceFrame,
>+               int32_t(const uint64_t pictureId));
>+  MOCK_METHOD1(ReceivedDecodedFrame, int32_t(const uint64_t pictureId));
>+};
>+
>+class MockVideoDecoder : public VideoDecoder {
>+ public:
>+  MockVideoDecoder();
>+  ~MockVideoDecoder() override;
>+
>+  MOCK_METHOD2(InitDecode,
>+               int32_t(const VideoCodec* codecSettings, int32_t numberOfCores));
>+  MOCK_METHOD4(Decode,
>+               int32_t(const EncodedImage& inputImage,
>+                       bool missingFrames,
>+                       const CodecSpecificInfo* codecSpecificInfo,
>+                       int64_t renderTimeMs));
>+  MOCK_METHOD1(RegisterDecodeCompleteCallback,
>+               int32_t(DecodedImageCallback* callback));
>+  MOCK_METHOD0(Release, int32_t());
>+  MOCK_METHOD0(Copy, VideoDecoder*());
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_MOCK_VIDEO_DECODER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_encoder.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..a0d82b1c33b1fc27ebaee948a6b9bc8e3cf39ea7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_encoder.cc
>@@ -0,0 +1,20 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/test/mock_video_encoder.h"
>+
>+namespace webrtc {
>+
>+MockEncodedImageCallback::MockEncodedImageCallback() = default;
>+MockEncodedImageCallback::~MockEncodedImageCallback() = default;
>+MockVideoEncoder::MockVideoEncoder() = default;
>+MockVideoEncoder::~MockVideoEncoder() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_encoder.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..62f17bab884255be8890f5648e1d9a77d78a9fdc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_encoder.h
>@@ -0,0 +1,57 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_MOCK_VIDEO_ENCODER_H_
>+#define API_TEST_MOCK_VIDEO_ENCODER_H_
>+
>+#include <vector>
>+
>+#include "api/video_codecs/video_encoder.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+class MockEncodedImageCallback : public EncodedImageCallback {
>+ public:
>+  MockEncodedImageCallback();
>+  ~MockEncodedImageCallback();
>+  MOCK_METHOD3(OnEncodedImage,
>+               Result(const EncodedImage& encodedImage,
>+                      const CodecSpecificInfo* codecSpecificInfo,
>+                      const RTPFragmentationHeader* fragmentation));
>+};
>+
>+class MockVideoEncoder : public VideoEncoder {
>+ public:
>+  MockVideoEncoder();
>+  ~MockVideoEncoder();
>+  MOCK_CONST_METHOD2(Version, int32_t(int8_t* version, int32_t length));
>+  MOCK_METHOD3(InitEncode,
>+               int32_t(const VideoCodec* codecSettings,
>+                       int32_t numberOfCores,
>+                       size_t maxPayloadSize));
>+  MOCK_METHOD3(Encode,
>+               int32_t(const VideoFrame& inputImage,
>+                       const CodecSpecificInfo* codecSpecificInfo,
>+                       const std::vector<FrameType>* frame_types));
>+  MOCK_METHOD1(RegisterEncodeCompleteCallback,
>+               int32_t(EncodedImageCallback* callback));
>+  MOCK_METHOD0(Release, int32_t());
>+  MOCK_METHOD0(Reset, int32_t());
>+  MOCK_METHOD2(SetRates, int32_t(uint32_t newBitRate, uint32_t frameRate));
>+  MOCK_METHOD2(SetRateAllocation,
>+               int32_t(const VideoBitrateAllocation& newBitRate,
>+                       uint32_t frameRate));
>+  MOCK_CONST_METHOD0(GetEncoderInfo, EncoderInfo(void));
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_MOCK_VIDEO_ENCODER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulated_network.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulated_network.h
>index aec300caa2b3e4fd8eb6eb29569b52cf81ec6574..c03b4ca4cbf28b5adb1b9d56153ef9cbac1521f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulated_network.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulated_network.h
>@@ -42,11 +42,11 @@ struct PacketDeliveryInfo {
>   uint64_t packet_id;
> };
> 
>-// DefaultNetworkSimulationConfig is a default network behavior configuration
>-// for default network behavior that will be used by WebRTC if no custom
>+// BuiltInNetworkBehaviorConfig is a built-in network behavior configuration
>+// for built-in network behavior that will be used by WebRTC if no custom
> // NetworkBehaviorInterface is provided.
>-struct DefaultNetworkSimulationConfig {
>-  DefaultNetworkSimulationConfig() {}
>+struct BuiltInNetworkBehaviorConfig {
>+  BuiltInNetworkBehaviorConfig() {}
>   // Queue length in number of packets.
>   size_t queue_length_packets = 0;
>   // Delay in addition to capacity induced delay.
>@@ -75,9 +75,6 @@ class NetworkBehaviorInterface {
>   virtual ~NetworkBehaviorInterface() = default;
> };
> 
>-// Deprecated. DO NOT USE. Use NetworkBehaviorInterface instead.
>-using NetworkSimulationInterface = NetworkBehaviorInterface;
>-
> }  // namespace webrtc
> 
> #endif  // API_TEST_SIMULATED_NETWORK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulcast_test_fixture.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulcast_test_fixture.h
>index e7eab241348da180ca5c83d565a31191cf889472..5270d133064a0d14928b318ed0b8260e08f4296d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulcast_test_fixture.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulcast_test_fixture.h
>@@ -33,6 +33,7 @@ class SimulcastTestFixture {
>   virtual void TestSpatioTemporalLayers333PatternEncoder() = 0;
>   virtual void TestSpatioTemporalLayers321PatternEncoder() = 0;
>   virtual void TestStrideEncodeDecode() = 0;
>+  virtual void TestDecodeWidthHeightSet() = 0;
> };
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/test_dependency_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/test_dependency_factory.cc
>index 3e59bc095c87474970b54d86172770e54792de04..179c623940f2bf4ff0bb091dd79399d9bc5dad76 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/test_dependency_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/test_dependency_factory.cc
>@@ -11,6 +11,7 @@
> #include <memory>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "api/test/test_dependency_factory.h"
> #include "rtc_base/thread_checker.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video/BUILD.gn
>new file mode 100644
>index 0000000000000000000000000000000000000000..9b63a4acb3da2f4a59ea92d70fe7ea2f8042b44e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video/BUILD.gn
>@@ -0,0 +1,24 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("function_video_factory") {
>+  visibility = [ "*" ]
>+  testonly = true
>+  public = [
>+    "function_video_decoder_factory.h",
>+    "function_video_encoder_factory.h",
>+  ]
>+
>+  deps = [
>+    "../../../rtc_base:checks",
>+    "../../video_codecs:video_codecs_api",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video/function_video_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video/function_video_decoder_factory.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..03a43239976a01139ef709e294e038f1990492f9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video/function_video_decoder_factory.h
>@@ -0,0 +1,56 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_VIDEO_FUNCTION_VIDEO_DECODER_FACTORY_H_
>+#define API_TEST_VIDEO_FUNCTION_VIDEO_DECODER_FACTORY_H_
>+
>+#include <functional>
>+#include <memory>
>+#include <utility>
>+#include <vector>
>+
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// A decoder factory producing decoders by calling a supplied create function.
>+class FunctionVideoDecoderFactory final : public VideoDecoderFactory {
>+ public:
>+  explicit FunctionVideoDecoderFactory(
>+      std::function<std::unique_ptr<VideoDecoder>()> create)
>+      : create_([create](const SdpVideoFormat&) { return create(); }) {}
>+  explicit FunctionVideoDecoderFactory(
>+      std::function<std::unique_ptr<VideoDecoder>(const SdpVideoFormat&)>
>+          create)
>+      : create_(std::move(create)) {}
>+
>+  // Unused by tests.
>+  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
>+    RTC_NOTREACHED();
>+    return {};
>+  }
>+
>+  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
>+      const SdpVideoFormat& format) override {
>+    return create_(format);
>+  }
>+
>+ private:
>+  const std::function<std::unique_ptr<VideoDecoder>(const SdpVideoFormat&)>
>+      create_;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_VIDEO_FUNCTION_VIDEO_DECODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video/function_video_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video/function_video_encoder_factory.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..85f848cd1fdc85328a2402db2bad81ad0178cc29
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video/function_video_encoder_factory.h
>@@ -0,0 +1,65 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_VIDEO_FUNCTION_VIDEO_ENCODER_FACTORY_H_
>+#define API_TEST_VIDEO_FUNCTION_VIDEO_ENCODER_FACTORY_H_
>+
>+#include <functional>
>+#include <memory>
>+#include <utility>
>+#include <vector>
>+
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// An encoder factory producing encoders by calling a supplied create
>+// function.
>+class FunctionVideoEncoderFactory final : public VideoEncoderFactory {
>+ public:
>+  explicit FunctionVideoEncoderFactory(
>+      std::function<std::unique_ptr<VideoEncoder>()> create)
>+      : create_([create](const SdpVideoFormat&) { return create(); }) {}
>+  explicit FunctionVideoEncoderFactory(
>+      std::function<std::unique_ptr<VideoEncoder>(const SdpVideoFormat&)>
>+          create)
>+      : create_(std::move(create)) {}
>+
>+  // Unused by tests.
>+  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
>+    RTC_NOTREACHED();
>+    return {};
>+  }
>+
>+  CodecInfo QueryVideoEncoder(
>+      const SdpVideoFormat& /* format */) const override {
>+    CodecInfo codec_info;
>+    codec_info.is_hardware_accelerated = false;
>+    codec_info.has_internal_source = false;
>+    return codec_info;
>+  }
>+
>+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>+      const SdpVideoFormat& format) override {
>+    return create_(format);
>+  }
>+
>+ private:
>+  const std::function<std::unique_ptr<VideoEncoder>(const SdpVideoFormat&)>
>+      create_;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_VIDEO_FUNCTION_VIDEO_ENCODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video_quality_test_fixture.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video_quality_test_fixture.h
>index ff1472e9a58e61354e805192528f3ef875cec923..069a24353de7ebccde392c068a5d4e05cc627cf3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video_quality_test_fixture.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video_quality_test_fixture.h
>@@ -82,15 +82,11 @@ class VideoQualityTestFixtureInterface {
>       std::string graph_data_output_filename;
>       std::string graph_title;
>     } analyzer;
>-    // Deprecated. DO NOT USE. Use config instead. This is not pipe actually,
>-    // it is just configuration, that will be passed to default implementation
>-    // of simulation layer.
>-    DefaultNetworkSimulationConfig pipe;
>     // Config for default simulation implementation. Must be nullopt if
>     // `sender_network` and `receiver_network` in InjectionComponents are
>     // non-null. May be nullopt even if `sender_network` and `receiver_network`
>     // are null; in that case, a default config will be used.
>-    absl::optional<DefaultNetworkSimulationConfig> config;
>+    absl::optional<BuiltInNetworkBehaviorConfig> config;
>     struct SS {                          // Spatial scalability.
>       std::vector<VideoStream> streams;  // If empty, one stream is assumed.
>       size_t selected_stream;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_fixture.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_fixture.h
>index a0b8083d7236c37a068beb7012de36d6637bbf56..925c60c3c92c9f5ebd149d53f188335731a1e424 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_fixture.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_fixture.h
>@@ -123,10 +123,6 @@ class VideoCodecTestFixture {
>           webrtc::H264PacketizationMode::NonInterleaved;
>     } h264_codec_settings;
> 
>-    // Should hardware accelerated codecs be used?
>-    bool hw_encoder = false;
>-    bool hw_decoder = false;
>-
>     // Custom checker that will be called for each frame.
>     const EncodedFrameChecker* encoded_frame_checker = nullptr;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/BUILD.gn
>index f473179dd14adfa4ff61f1c224b0ef97a2c4a2c8..357ff0054bd0db6de234ee73ecb978ff68865bc8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/BUILD.gn
>@@ -19,7 +19,15 @@ rtc_source_set("bitrate_settings") {
>   ]
> }
> 
>+rtc_source_set("enums") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "enums.h",
>+  ]
>+}
>+
> rtc_static_library("network_control") {
>+  visibility = [ "*" ]
>   sources = [
>     "network_control.h",
>     "network_types.cc",
>@@ -35,6 +43,19 @@ rtc_static_library("network_control") {
>   ]
> }
> 
>+rtc_static_library("goog_cc") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "goog_cc_factory.cc",
>+    "goog_cc_factory.h",
>+  ]
>+  deps = [
>+    ":network_control",
>+    "../../modules/congestion_controller/goog_cc",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
> if (rtc_include_tests) {
>   rtc_source_set("network_control_test") {
>     testonly = true
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/OWNERS
>new file mode 100644
>index 0000000000000000000000000000000000000000..5991f6fc56a079d56a9c9310ba6f7828db34e11f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/OWNERS
>@@ -0,0 +1,2 @@
>+srte@webrtc.org
>+terelius@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/enums.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/enums.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..b1d5770cb962aa7987f890481931b0648300dec7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/enums.h
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TRANSPORT_ENUMS_H_
>+#define API_TRANSPORT_ENUMS_H_
>+
>+namespace webrtc {
>+
>+// See https://w3c.github.io/webrtc-pc/#rtcicetransportstate
>+// Note that kFailed is currently not a terminal state, and a transport might
>+// incorrectly be marked as failed while gathering candidates, see
>+// bugs.webrtc.org/8833
>+enum class IceTransportState {
>+  kNew,
>+  kChecking,
>+  kConnected,
>+  kCompleted,
>+  kFailed,
>+  kDisconnected,
>+  kClosed,
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TRANSPORT_ENUMS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/goog_cc_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/goog_cc_factory.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..119e2dcf84abcb8d3ffdc6963919bb7d2471e088
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/goog_cc_factory.cc
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/transport/goog_cc_factory.h"
>+
>+#include "absl/memory/memory.h"
>+#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
>+namespace webrtc {
>+GoogCcNetworkControllerFactory::GoogCcNetworkControllerFactory(
>+    RtcEventLog* event_log)
>+    : event_log_(event_log) {}
>+
>+std::unique_ptr<NetworkControllerInterface>
>+GoogCcNetworkControllerFactory::Create(NetworkControllerConfig config) {
>+  return absl::make_unique<GoogCcNetworkController>(event_log_, config, false);
>+}
>+
>+TimeDelta GoogCcNetworkControllerFactory::GetProcessInterval() const {
>+  const int64_t kUpdateIntervalMs = 25;
>+  return TimeDelta::ms(kUpdateIntervalMs);
>+}
>+
>+GoogCcFeedbackNetworkControllerFactory::GoogCcFeedbackNetworkControllerFactory(
>+    RtcEventLog* event_log)
>+    : event_log_(event_log) {}
>+
>+std::unique_ptr<NetworkControllerInterface>
>+GoogCcFeedbackNetworkControllerFactory::Create(NetworkControllerConfig config) {
>+  return absl::make_unique<GoogCcNetworkController>(event_log_, config, true);
>+}
>+
>+TimeDelta GoogCcFeedbackNetworkControllerFactory::GetProcessInterval() const {
>+  const int64_t kUpdateIntervalMs = 25;
>+  return TimeDelta::ms(kUpdateIntervalMs);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/goog_cc_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/goog_cc_factory.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..2e3b3172d02fd6ffa0b1f3ffe338ce57e2a574d4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/goog_cc_factory.h
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TRANSPORT_GOOG_CC_FACTORY_H_
>+#define API_TRANSPORT_GOOG_CC_FACTORY_H_
>+#include <memory>
>+
>+#include "api/transport/network_control.h"
>+
>+namespace webrtc {
>+class RtcEventLog;
>+
>+class GoogCcNetworkControllerFactory
>+    : public NetworkControllerFactoryInterface {
>+ public:
>+  explicit GoogCcNetworkControllerFactory(RtcEventLog*);
>+  std::unique_ptr<NetworkControllerInterface> Create(
>+      NetworkControllerConfig config) override;
>+  TimeDelta GetProcessInterval() const override;
>+
>+ private:
>+  RtcEventLog* const event_log_;
>+};
>+
>+// Factory to create packet feedback only GoogCC, this can be used for
>+// connections providing packet receive time feedback but no other reports.
>+class GoogCcFeedbackNetworkControllerFactory
>+    : public NetworkControllerFactoryInterface {
>+ public:
>+  explicit GoogCcFeedbackNetworkControllerFactory(RtcEventLog*);
>+  std::unique_ptr<NetworkControllerInterface> Create(
>+      NetworkControllerConfig config) override;
>+  TimeDelta GetProcessInterval() const override;
>+
>+ private:
>+  RtcEventLog* const event_log_;
>+};
>+}  // namespace webrtc
>+
>+#endif  // API_TRANSPORT_GOOG_CC_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.cc
>index 48bdcab162bbe2a08f2af22db025af93b71f8c3d..80214de1a2381247875ba13f1c1aaad6e30b1846 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.cc
>@@ -38,7 +38,7 @@ std::vector<PacketResult> TransportPacketsFeedback::ReceivedWithSendInfo()
>     const {
>   std::vector<PacketResult> res;
>   for (const PacketResult& fb : packet_feedbacks) {
>-    if (fb.receive_time.IsFinite() && fb.sent_packet.has_value()) {
>+    if (fb.receive_time.IsFinite()) {
>       res.push_back(fb);
>     }
>   }
>@@ -48,7 +48,7 @@ std::vector<PacketResult> TransportPacketsFeedback::ReceivedWithSendInfo()
> std::vector<PacketResult> TransportPacketsFeedback::LostWithSendInfo() const {
>   std::vector<PacketResult> res;
>   for (const PacketResult& fb : packet_feedbacks) {
>-    if (fb.receive_time.IsPlusInfinity() && fb.sent_packet.has_value()) {
>+    if (fb.receive_time.IsPlusInfinity()) {
>       res.push_back(fb);
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.h
>index ad20abc53d710f820d884af62dcf915440b6b98a..0f1d7ab68334c3d40ef09a29bd376f727fc6fe7a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.h
>@@ -88,12 +88,13 @@ struct PacedPacketInfo {
> struct SentPacket {
>   Timestamp send_time = Timestamp::PlusInfinity();
>   DataSize size = DataSize::Zero();
>+  DataSize prior_unacked_data = DataSize::Zero();
>   PacedPacketInfo pacing_info;
>   // Transport independent sequence number, any tracked packet should have a
>   // sequence number that is unique over the whole call and increasing by 1 for
>   // each packet.
>   int64_t sequence_number;
>-  // Data in flight when the packet was sent, including the packet.
>+  // Tracked data in flight when the packet was sent, excluding unacked data.
>   DataSize data_in_flight = DataSize::Zero();
> };
> 
>@@ -125,7 +126,7 @@ struct PacketResult {
>   PacketResult(const PacketResult&);
>   ~PacketResult();
> 
>-  absl::optional<SentPacket> sent_packet;
>+  SentPacket sent_packet;
>   Timestamp receive_time = Timestamp::PlusInfinity();
> };
> 
>@@ -139,6 +140,9 @@ struct TransportPacketsFeedback {
>   DataSize prior_in_flight = DataSize::Zero();
>   std::vector<PacketResult> packet_feedbacks;
> 
>+  // Arrival times for messages without send time information.
>+  std::vector<Timestamp> sendless_arrival_times;
>+
>   std::vector<PacketResult> ReceivedWithSendInfo() const;
>   std::vector<PacketResult> LostWithSendInfo() const;
>   std::vector<PacketResult> PacketsWithFeedback() const;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/network_control_tester.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/network_control_tester.cc
>index 2edabd0717e5949a0be7744ac6c043dd9e9ee029..c590ae88bc4130a78c16d5d083216b7c95b2b033 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/network_control_tester.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/network_control_tester.cc
>@@ -87,7 +87,7 @@ void NetworkControllerTester::RunSimulation(TimeDelta duration,
>     if (state_.congestion_window && state_.congestion_window->IsFinite()) {
>       DataSize data_in_flight = DataSize::Zero();
>       for (PacketResult& packet : outstanding_packets_)
>-        data_in_flight += packet.sent_packet->size;
>+        data_in_flight += packet.sent_packet.size;
>       if (data_in_flight > *state_.congestion_window)
>         send_packet = false;
>     }
>@@ -98,7 +98,7 @@ void NetworkControllerTester::RunSimulation(TimeDelta duration,
>       sent_packet.sequence_number = packet_sequence_number_++;
>       sent_packet.data_in_flight = sent_packet.size;
>       for (PacketResult& packet : outstanding_packets_)
>-        sent_packet.data_in_flight += packet.sent_packet->size;
>+        sent_packet.data_in_flight += packet.sent_packet.size;
>       Update(&state_, controller_->OnSentPacket(sent_packet));
> 
>       TimeDelta time_in_flight = sent_packet.size / actual_bandwidth;
>@@ -120,7 +120,7 @@ void NetworkControllerTester::RunSimulation(TimeDelta duration,
>       TransportPacketsFeedback feedback;
>       feedback.prior_in_flight = DataSize::Zero();
>       for (PacketResult& packet : outstanding_packets_)
>-        feedback.prior_in_flight += packet.sent_packet->size;
>+        feedback.prior_in_flight += packet.sent_packet.size;
>       while (!outstanding_packets_.empty() &&
>              current_time_ >= outstanding_packets_.front().receive_time +
>                                   propagation_delay) {
>@@ -131,7 +131,7 @@ void NetworkControllerTester::RunSimulation(TimeDelta duration,
>           feedback.packet_feedbacks.back().receive_time + propagation_delay;
>       feedback.data_in_flight = DataSize::Zero();
>       for (PacketResult& packet : outstanding_packets_)
>-        feedback.data_in_flight += packet.sent_packet->size;
>+        feedback.data_in_flight += packet.sent_packet.size;
>       Update(&state_, controller_->OnTransportPacketsFeedback(feedback));
>     }
>     current_time_ += packet_interval;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/BUILD.gn
>index 7dbddf4e622f4ed7fbc1f73fcc952de3428fbe6b..7e150f176679b23343571e1d148b2b30929bce02 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/BUILD.gn
>@@ -9,6 +9,7 @@
> import("../../webrtc.gni")
> 
> rtc_source_set("data_rate") {
>+  visibility = [ "*" ]
>   sources = [
>     "data_rate.cc",
>     "data_rate.h",
>@@ -18,11 +19,13 @@ rtc_source_set("data_rate") {
>     ":data_size",
>     ":time_delta",
>     "../../rtc_base:checks",
>-    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:stringutils",
>+    "../../rtc_base/units:unit_base",
>   ]
> }
> 
> rtc_source_set("data_size") {
>+  visibility = [ "*" ]
>   sources = [
>     "data_size.cc",
>     "data_size.h",
>@@ -30,10 +33,13 @@ rtc_source_set("data_size") {
> 
>   deps = [
>     "../../rtc_base:checks",
>-    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:stringutils",
>+    "../../rtc_base/units:unit_base",
>   ]
> }
>+
> rtc_source_set("time_delta") {
>+  visibility = [ "*" ]
>   sources = [
>     "time_delta.cc",
>     "time_delta.h",
>@@ -41,11 +47,13 @@ rtc_source_set("time_delta") {
> 
>   deps = [
>     "../../rtc_base:checks",
>-    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:stringutils",
>+    "../../rtc_base/units:unit_base",
>   ]
> }
> 
> rtc_source_set("timestamp") {
>+  visibility = [ "*" ]
>   sources = [
>     "timestamp.cc",
>     "timestamp.h",
>@@ -54,7 +62,8 @@ rtc_source_set("timestamp") {
>   deps = [
>     ":time_delta",
>     "../../rtc_base:checks",
>-    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:stringutils",
>+    "../../rtc_base/units:unit_base",
>   ]
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.cc
>index 9170627715849809fc0dfcd6c3855e078775eea2..d72d958c028f976b2c150f2d67df0a3cbdc0f60b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.cc
>@@ -14,7 +14,7 @@
> 
> namespace webrtc {
> 
>-std::string ToString(const DataRate& value) {
>+std::string ToString(DataRate value) {
>   char buf[64];
>   rtc::SimpleStringBuilder sb(buf);
>   if (value.IsInfinite()) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.h
>index 4bb988b65888bb934addaadfd0629f6196262e92..7119284874e92b7c54e8d2cda5555892a5a3bf64 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.h
>@@ -15,22 +15,17 @@
> #include <ostream>  // no-presubmit-check TODO(webrtc:8982)
> #endif              // UNIT_TEST
> 
>-#include <stdint.h>
>-#include <algorithm>
>-#include <cmath>
> #include <limits>
> #include <string>
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/numerics/safe_conversions.h"
>+#include <type_traits>
> 
> #include "api/units/data_size.h"
> #include "api/units/time_delta.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/units/unit_base.h"
> 
> namespace webrtc {
> namespace data_rate_impl {
>-constexpr int64_t kPlusInfinityVal = std::numeric_limits<int64_t>::max();
>-
> inline int64_t Microbits(const DataSize& size) {
>   constexpr int64_t kMaxBeforeConversion =
>       std::numeric_limits<int64_t>::max() / 8000000;
>@@ -43,184 +38,64 @@ inline int64_t Microbits(const DataSize& size) {
> // DataRate is a class that represents a given data rate. This can be used to
> // represent bandwidth, encoding bitrate, etc. The internal storage is bits per
> // second (bps).
>-class DataRate {
>+class DataRate final : public rtc_units_impl::RelativeUnit<DataRate> {
>  public:
>   DataRate() = delete;
>-  static constexpr DataRate Zero() { return DataRate(0); }
>-  static constexpr DataRate Infinity() {
>-    return DataRate(data_rate_impl::kPlusInfinityVal);
>-  }
>+  static constexpr DataRate Infinity() { return PlusInfinity(); }
>   template <int64_t bps>
>   static constexpr DataRate BitsPerSec() {
>-    static_assert(bps >= 0, "");
>-    static_assert(bps < data_rate_impl::kPlusInfinityVal, "");
>-    return DataRate(bps);
>+    return FromStaticValue<bps>();
>   }
>   template <int64_t kbps>
>   static constexpr DataRate KilobitsPerSec() {
>-    static_assert(kbps >= 0, "");
>-    static_assert(kbps < data_rate_impl::kPlusInfinityVal / 1000, "");
>-    return DataRate(kbps * 1000);
>+    return FromStaticFraction<kbps, 1000>();
>   }
>-
>-  template <
>-      typename T,
>-      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>-  static DataRate bps(T bits_per_second) {
>-    RTC_DCHECK_GE(bits_per_second, 0);
>-    RTC_DCHECK_LT(bits_per_second, data_rate_impl::kPlusInfinityVal);
>-    return DataRate(rtc::dchecked_cast<int64_t>(bits_per_second));
>-  }
>-  template <
>-      typename T,
>-      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>-  static DataRate kbps(T kilobits_per_sec) {
>-    RTC_DCHECK_GE(kilobits_per_sec, 0);
>-    RTC_DCHECK_LT(kilobits_per_sec, data_rate_impl::kPlusInfinityVal / 1000);
>-    return DataRate::bps(rtc::dchecked_cast<int64_t>(kilobits_per_sec) * 1000);
>-  }
>-
>-  template <typename T,
>-            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>-                nullptr>
>-  static DataRate bps(T bits_per_second) {
>-    if (bits_per_second == std::numeric_limits<T>::infinity()) {
>-      return Infinity();
>-    } else {
>-      RTC_DCHECK(!std::isnan(bits_per_second));
>-      RTC_DCHECK_GE(bits_per_second, 0);
>-      RTC_DCHECK_LT(bits_per_second, data_rate_impl::kPlusInfinityVal);
>-      return DataRate(rtc::dchecked_cast<int64_t>(bits_per_second));
>-    }
>+  template <typename T>
>+  static constexpr DataRate bps(T bits_per_second) {
>+    return FromValue(bits_per_second);
>   }
>-  template <typename T,
>-            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>-                nullptr>
>-  static DataRate kbps(T kilobits_per_sec) {
>-    return DataRate::bps(kilobits_per_sec * 1e3);
>+  template <typename T>
>+  static constexpr DataRate kbps(T kilobits_per_sec) {
>+    return FromFraction<1000>(kilobits_per_sec);
>   }
>-
>   template <typename T = int64_t>
>-  typename std::enable_if<std::is_integral<T>::value, T>::type bps() const {
>-    RTC_DCHECK(IsFinite());
>-    return rtc::dchecked_cast<T>(bits_per_sec_);
>+  constexpr T bps() const {
>+    return ToValue<T>();
>   }
>   template <typename T = int64_t>
>-  typename std::enable_if<std::is_integral<T>::value, T>::type kbps() const {
>-    RTC_DCHECK(IsFinite());
>-    return rtc::dchecked_cast<T>(UnsafeKilobitsPerSec());
>-  }
>-
>-  template <typename T>
>-  typename std::enable_if<std::is_floating_point<T>::value,
>-                          T>::type constexpr bps() const {
>-    return IsInfinite() ? std::numeric_limits<T>::infinity() : bits_per_sec_;
>-  }
>-  template <typename T>
>-  typename std::enable_if<std::is_floating_point<T>::value,
>-                          T>::type constexpr kbps() const {
>-    return bps<T>() * 1e-3;
>+  T kbps() const {
>+    return ToFraction<1000, T>();
>   }
>-
>   constexpr int64_t bps_or(int64_t fallback_value) const {
>-    return IsFinite() ? bits_per_sec_ : fallback_value;
>+    return ToValueOr(fallback_value);
>   }
>   constexpr int64_t kbps_or(int64_t fallback_value) const {
>-    return IsFinite() ? UnsafeKilobitsPerSec() : fallback_value;
>-  }
>-
>-  constexpr bool IsZero() const { return bits_per_sec_ == 0; }
>-  constexpr bool IsInfinite() const {
>-    return bits_per_sec_ == data_rate_impl::kPlusInfinityVal;
>-  }
>-  constexpr bool IsFinite() const { return !IsInfinite(); }
>-  DataRate Clamped(DataRate min_rate, DataRate max_rate) const {
>-    return std::max(min_rate, std::min(*this, max_rate));
>-  }
>-  void Clamp(DataRate min_rate, DataRate max_rate) {
>-    *this = Clamped(min_rate, max_rate);
>-  }
>-  DataRate operator-(const DataRate& other) const {
>-    return DataRate::bps(bps() - other.bps());
>-  }
>-  DataRate operator+(const DataRate& other) const {
>-    return DataRate::bps(bps() + other.bps());
>-  }
>-  DataRate& operator-=(const DataRate& other) {
>-    *this = *this - other;
>-    return *this;
>-  }
>-  DataRate& operator+=(const DataRate& other) {
>-    *this = *this + other;
>-    return *this;
>-  }
>-  constexpr double operator/(const DataRate& other) const {
>-    return bps<double>() / other.bps<double>();
>-  }
>-  constexpr bool operator==(const DataRate& other) const {
>-    return bits_per_sec_ == other.bits_per_sec_;
>-  }
>-  constexpr bool operator!=(const DataRate& other) const {
>-    return bits_per_sec_ != other.bits_per_sec_;
>-  }
>-  constexpr bool operator<=(const DataRate& other) const {
>-    return bits_per_sec_ <= other.bits_per_sec_;
>-  }
>-  constexpr bool operator>=(const DataRate& other) const {
>-    return bits_per_sec_ >= other.bits_per_sec_;
>-  }
>-  constexpr bool operator>(const DataRate& other) const {
>-    return bits_per_sec_ > other.bits_per_sec_;
>-  }
>-  constexpr bool operator<(const DataRate& other) const {
>-    return bits_per_sec_ < other.bits_per_sec_;
>+    return ToFractionOr<1000>(fallback_value);
>   }
> 
>  private:
>   // Bits per second used internally to simplify debugging by making the value
>   // more recognizable.
>-  explicit constexpr DataRate(int64_t bits_per_second)
>-      : bits_per_sec_(bits_per_second) {}
>-  constexpr int64_t UnsafeKilobitsPerSec() const {
>-    return (bits_per_sec_ + 500) / 1000;
>-  }
>-  int64_t bits_per_sec_;
>+  friend class rtc_units_impl::UnitBase<DataRate>;
>+  using RelativeUnit::RelativeUnit;
>+  static constexpr bool one_sided = true;
> };
> 
>-inline DataRate operator*(const DataRate& rate, const double& scalar) {
>-  return DataRate::bps(std::round(rate.bps() * scalar));
>-}
>-inline DataRate operator*(const double& scalar, const DataRate& rate) {
>-  return rate * scalar;
>-}
>-inline DataRate operator*(const DataRate& rate, const int64_t& scalar) {
>-  return DataRate::bps(rate.bps() * scalar);
>-}
>-inline DataRate operator*(const int64_t& scalar, const DataRate& rate) {
>-  return rate * scalar;
>-}
>-inline DataRate operator*(const DataRate& rate, const int32_t& scalar) {
>-  return DataRate::bps(rate.bps() * scalar);
>-}
>-inline DataRate operator*(const int32_t& scalar, const DataRate& rate) {
>-  return rate * scalar;
>-}
>-
>-inline DataRate operator/(const DataSize& size, const TimeDelta& duration) {
>+inline DataRate operator/(const DataSize size, const TimeDelta duration) {
>   return DataRate::bps(data_rate_impl::Microbits(size) / duration.us());
> }
>-inline TimeDelta operator/(const DataSize& size, const DataRate& rate) {
>+inline TimeDelta operator/(const DataSize size, const DataRate rate) {
>   return TimeDelta::us(data_rate_impl::Microbits(size) / rate.bps());
> }
>-inline DataSize operator*(const DataRate& rate, const TimeDelta& duration) {
>+inline DataSize operator*(const DataRate rate, const TimeDelta duration) {
>   int64_t microbits = rate.bps() * duration.us();
>   return DataSize::bytes((microbits + 4000000) / 8000000);
> }
>-inline DataSize operator*(const TimeDelta& duration, const DataRate& rate) {
>+inline DataSize operator*(const TimeDelta duration, const DataRate rate) {
>   return rate * duration;
> }
> 
>-std::string ToString(const DataRate& value);
>+std::string ToString(DataRate value);
> 
> #ifdef UNIT_TEST
> inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate_unittest.cc
>index 8e5b6605948fd88fda51533c15ee531e8c9b3d9b..996298c1fcee8bff67f0cf78dd35241b16721aec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate_unittest.cc
>@@ -130,6 +130,9 @@ TEST(DataRateTest, MathOperations) {
> 
>   EXPECT_EQ(rate_a / rate_b, static_cast<double>(kValueA) / kValueB);
> 
>+  EXPECT_EQ((rate_a / 10).bps(), kValueA / 10);
>+  EXPECT_NEAR((rate_a / 0.5).bps(), kValueA * 2, 1);
>+
>   DataRate mutable_rate = DataRate::bps(kValueA);
>   mutable_rate += rate_b;
>   EXPECT_EQ(mutable_rate.bps(), kValueA + kValueB);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.cc
>index 4440f89d03c2ff84a1b50cfabae43f1e979a9d4e..8a87786c77a3e97eab433b94ab65acca117588a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.cc
>@@ -14,7 +14,7 @@
> 
> namespace webrtc {
> 
>-std::string ToString(const DataSize& value) {
>+std::string ToString(DataSize value) {
>   char buf[64];
>   rtc::SimpleStringBuilder sb(buf);
>   if (value.IsInfinite()) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.h
>index 8958b2469730cf200efa4824a88e84c6fb4ffc65..b4cbb652f3c907c097196eb5f722e9643a19f319 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.h
>@@ -15,143 +15,44 @@
> #include <ostream>  // no-presubmit-check TODO(webrtc:8982)
> #endif              // UNIT_TEST
> 
>-#include <stdint.h>
>-#include <cmath>
>-#include <limits>
> #include <string>
> #include <type_traits>
> 
>-#include "rtc_base/checks.h"
>-#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/units/unit_base.h"
> 
> namespace webrtc {
>-namespace data_size_impl {
>-constexpr int64_t kPlusInfinityVal = std::numeric_limits<int64_t>::max();
>-}  // namespace data_size_impl
>-
> // DataSize is a class represeting a count of bytes.
>-class DataSize {
>+class DataSize final : public rtc_units_impl::RelativeUnit<DataSize> {
>  public:
>   DataSize() = delete;
>-  static constexpr DataSize Zero() { return DataSize(0); }
>-  static constexpr DataSize Infinity() {
>-    return DataSize(data_size_impl::kPlusInfinityVal);
>-  }
>+  static constexpr DataSize Infinity() { return PlusInfinity(); }
>   template <int64_t bytes>
>   static constexpr DataSize Bytes() {
>-    static_assert(bytes >= 0, "");
>-    static_assert(bytes < data_size_impl::kPlusInfinityVal, "");
>-    return DataSize(bytes);
>+    return FromStaticValue<bytes>();
>   }
> 
>   template <
>       typename T,
>-      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>-  static DataSize bytes(T bytes) {
>-    RTC_DCHECK_GE(bytes, 0);
>-    RTC_DCHECK_LT(bytes, data_size_impl::kPlusInfinityVal);
>-    return DataSize(rtc::dchecked_cast<int64_t>(bytes));
>-  }
>-
>-  template <typename T,
>-            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>-                nullptr>
>+      typename std::enable_if<std::is_arithmetic<T>::value>::type* = nullptr>
>   static DataSize bytes(T bytes) {
>-    if (bytes == std::numeric_limits<T>::infinity()) {
>-      return Infinity();
>-    } else {
>-      RTC_DCHECK(!std::isnan(bytes));
>-      RTC_DCHECK_GE(bytes, 0);
>-      RTC_DCHECK_LT(bytes, data_size_impl::kPlusInfinityVal);
>-      return DataSize(rtc::dchecked_cast<int64_t>(bytes));
>-    }
>+    return FromValue(bytes);
>   }
>-
>   template <typename T = int64_t>
>-  typename std::enable_if<std::is_integral<T>::value, T>::type bytes() const {
>-    RTC_DCHECK(IsFinite());
>-    return rtc::dchecked_cast<T>(bytes_);
>-  }
>-
>-  template <typename T>
>-  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>-  bytes() const {
>-    return IsInfinite() ? std::numeric_limits<T>::infinity() : bytes_;
>+  typename std::enable_if<std::is_arithmetic<T>::value, T>::type bytes() const {
>+    return ToValue<T>();
>   }
> 
>   constexpr int64_t bytes_or(int64_t fallback_value) const {
>-    return IsFinite() ? bytes_ : fallback_value;
>-  }
>-
>-  constexpr bool IsZero() const { return bytes_ == 0; }
>-  constexpr bool IsInfinite() const {
>-    return bytes_ == data_size_impl::kPlusInfinityVal;
>-  }
>-  constexpr bool IsFinite() const { return !IsInfinite(); }
>-  DataSize operator-(const DataSize& other) const {
>-    return DataSize::bytes(bytes() - other.bytes());
>-  }
>-  DataSize operator+(const DataSize& other) const {
>-    return DataSize::bytes(bytes() + other.bytes());
>-  }
>-  DataSize& operator-=(const DataSize& other) {
>-    *this = *this - other;
>-    return *this;
>-  }
>-  DataSize& operator+=(const DataSize& other) {
>-    *this = *this + other;
>-    return *this;
>-  }
>-  constexpr double operator/(const DataSize& other) const {
>-    return bytes<double>() / other.bytes<double>();
>-  }
>-  constexpr bool operator==(const DataSize& other) const {
>-    return bytes_ == other.bytes_;
>-  }
>-  constexpr bool operator!=(const DataSize& other) const {
>-    return bytes_ != other.bytes_;
>-  }
>-  constexpr bool operator<=(const DataSize& other) const {
>-    return bytes_ <= other.bytes_;
>-  }
>-  constexpr bool operator>=(const DataSize& other) const {
>-    return bytes_ >= other.bytes_;
>-  }
>-  constexpr bool operator>(const DataSize& other) const {
>-    return bytes_ > other.bytes_;
>-  }
>-  constexpr bool operator<(const DataSize& other) const {
>-    return bytes_ < other.bytes_;
>+    return ToValueOr(fallback_value);
>   }
> 
>  private:
>-  explicit constexpr DataSize(int64_t bytes) : bytes_(bytes) {}
>-  int64_t bytes_;
>+  friend class rtc_units_impl::UnitBase<DataSize>;
>+  using RelativeUnit::RelativeUnit;
>+  static constexpr bool one_sided = true;
> };
> 
>-inline DataSize operator*(const DataSize& size, const double& scalar) {
>-  return DataSize::bytes(std::round(size.bytes() * scalar));
>-}
>-inline DataSize operator*(const double& scalar, const DataSize& size) {
>-  return size * scalar;
>-}
>-inline DataSize operator*(const DataSize& size, const int64_t& scalar) {
>-  return DataSize::bytes(size.bytes() * scalar);
>-}
>-inline DataSize operator*(const int64_t& scalar, const DataSize& size) {
>-  return size * scalar;
>-}
>-inline DataSize operator*(const DataSize& size, const int32_t& scalar) {
>-  return DataSize::bytes(size.bytes() * scalar);
>-}
>-inline DataSize operator*(const int32_t& scalar, const DataSize& size) {
>-  return size * scalar;
>-}
>-inline DataSize operator/(const DataSize& size, const int64_t& scalar) {
>-  return DataSize::bytes(size.bytes() / scalar);
>-}
>-
>-std::string ToString(const DataSize& value);
>+std::string ToString(DataSize value);
> 
> #ifdef UNIT_TEST
> inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.cc
>index d38387a5669aee4b952adccca6d884bfee7bc13c..f90451b429a7a0b1d99d3ba968c3d6615cbad91b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.cc
>@@ -14,7 +14,7 @@
> 
> namespace webrtc {
> 
>-std::string ToString(const TimeDelta& value) {
>+std::string ToString(TimeDelta value) {
>   char buf[64];
>   rtc::SimpleStringBuilder sb(buf);
>   if (value.IsPlusInfinity()) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.h
>index ec364172aa18df3a84fccdcafa1ffaaf92cd7dfd..64583698937120f94e2f6113dd6852cbc924db2d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.h
>@@ -15,21 +15,13 @@
> #include <ostream>  // no-presubmit-check TODO(webrtc:8982)
> #endif              // UNIT_TEST
> 
>-#include <stdint.h>
>-#include <cmath>
> #include <cstdlib>
>-#include <limits>
> #include <string>
>+#include <type_traits>
> 
>-#include "rtc_base/checks.h"
>-#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/units/unit_base.h"
> 
> namespace webrtc {
>-namespace timedelta_impl {
>-constexpr int64_t kPlusInfinityVal = std::numeric_limits<int64_t>::max();
>-constexpr int64_t kMinusInfinityVal = std::numeric_limits<int64_t>::min();
>-}  // namespace timedelta_impl
>-
> // TimeDelta represents the difference between two timestamps. Commonly this can
> // be a duration. However since two Timestamps are not guaranteed to have the
> // same epoch (they might come from different computers, making exact
>@@ -37,245 +29,69 @@ constexpr int64_t kMinusInfinityVal = std::numeric_limits<int64_t>::min();
> // undefined. To simplify usage, it can be constructed and converted to
> // different units, specifically seconds (s), milliseconds (ms) and
> // microseconds (us).
>-class TimeDelta {
>+class TimeDelta final : public rtc_units_impl::RelativeUnit<TimeDelta> {
>  public:
>   TimeDelta() = delete;
>-  static constexpr TimeDelta Zero() { return TimeDelta(0); }
>-  static constexpr TimeDelta PlusInfinity() {
>-    return TimeDelta(timedelta_impl::kPlusInfinityVal);
>-  }
>-  static constexpr TimeDelta MinusInfinity() {
>-    return TimeDelta(timedelta_impl::kMinusInfinityVal);
>-  }
>   template <int64_t seconds>
>   static constexpr TimeDelta Seconds() {
>-    static_assert(seconds > timedelta_impl::kMinusInfinityVal / 1000000, "");
>-    static_assert(seconds < timedelta_impl::kPlusInfinityVal / 1000000, "");
>-    return TimeDelta(seconds * 1000000);
>+    return FromStaticFraction<seconds, 1000000>();
>   }
>   template <int64_t ms>
>   static constexpr TimeDelta Millis() {
>-    static_assert(ms > timedelta_impl::kMinusInfinityVal / 1000, "");
>-    static_assert(ms < timedelta_impl::kPlusInfinityVal / 1000, "");
>-    return TimeDelta(ms * 1000);
>+    return FromStaticFraction<ms, 1000>();
>   }
>   template <int64_t us>
>   static constexpr TimeDelta Micros() {
>-    static_assert(us > timedelta_impl::kMinusInfinityVal, "");
>-    static_assert(us < timedelta_impl::kPlusInfinityVal, "");
>-    return TimeDelta(us);
>-  }
>-
>-  template <
>-      typename T,
>-      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>-  static TimeDelta seconds(T seconds) {
>-    RTC_DCHECK_GT(seconds, timedelta_impl::kMinusInfinityVal / 1000000);
>-    RTC_DCHECK_LT(seconds, timedelta_impl::kPlusInfinityVal / 1000000);
>-    return TimeDelta(rtc::dchecked_cast<int64_t>(seconds) * 1000000);
>+    return FromStaticValue<us>();
>   }
>-  template <
>-      typename T,
>-      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>-  static TimeDelta ms(T milliseconds) {
>-    RTC_DCHECK_GT(milliseconds, timedelta_impl::kMinusInfinityVal / 1000);
>-    RTC_DCHECK_LT(milliseconds, timedelta_impl::kPlusInfinityVal / 1000);
>-    return TimeDelta(rtc::dchecked_cast<int64_t>(milliseconds) * 1000);
>-  }
>-  template <
>-      typename T,
>-      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>-  static TimeDelta us(T microseconds) {
>-    RTC_DCHECK_GT(microseconds, timedelta_impl::kMinusInfinityVal);
>-    RTC_DCHECK_LT(microseconds, timedelta_impl::kPlusInfinityVal);
>-    return TimeDelta(rtc::dchecked_cast<int64_t>(microseconds));
>-  }
>-
>-  template <typename T,
>-            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>-                nullptr>
>+  template <typename T>
>   static TimeDelta seconds(T seconds) {
>-    return TimeDelta::us(seconds * 1e6);
>+    return FromFraction<1000000>(seconds);
>   }
>-  template <typename T,
>-            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>-                nullptr>
>+  template <typename T>
>   static TimeDelta ms(T milliseconds) {
>-    return TimeDelta::us(milliseconds * 1e3);
>+    return FromFraction<1000>(milliseconds);
>   }
>-  template <typename T,
>-            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>-                nullptr>
>+  template <typename T>
>   static TimeDelta us(T microseconds) {
>-    if (microseconds == std::numeric_limits<T>::infinity()) {
>-      return PlusInfinity();
>-    } else if (microseconds == -std::numeric_limits<T>::infinity()) {
>-      return MinusInfinity();
>-    } else {
>-      RTC_DCHECK(!std::isnan(microseconds));
>-      RTC_DCHECK_GT(microseconds, timedelta_impl::kMinusInfinityVal);
>-      RTC_DCHECK_LT(microseconds, timedelta_impl::kPlusInfinityVal);
>-      return TimeDelta(rtc::dchecked_cast<int64_t>(microseconds));
>-    }
>+    return FromValue(microseconds);
>   }
>-
>   template <typename T = int64_t>
>-  typename std::enable_if<std::is_integral<T>::value, T>::type seconds() const {
>-    RTC_DCHECK(IsFinite());
>-    return rtc::dchecked_cast<T>(UnsafeSeconds());
>+  T seconds() const {
>+    return ToFraction<1000000, T>();
>   }
>   template <typename T = int64_t>
>-  typename std::enable_if<std::is_integral<T>::value, T>::type ms() const {
>-    RTC_DCHECK(IsFinite());
>-    return rtc::dchecked_cast<T>(UnsafeMillis());
>+  T ms() const {
>+    return ToFraction<1000, T>();
>   }
>   template <typename T = int64_t>
>-  typename std::enable_if<std::is_integral<T>::value, T>::type us() const {
>-    RTC_DCHECK(IsFinite());
>-    return rtc::dchecked_cast<T>(microseconds_);
>+  T us() const {
>+    return ToValue<T>();
>   }
>   template <typename T = int64_t>
>-  typename std::enable_if<std::is_integral<T>::value, T>::type ns() const {
>-    RTC_DCHECK_GE(us(), std::numeric_limits<T>::min() / 1000);
>-    RTC_DCHECK_LE(us(), std::numeric_limits<T>::max() / 1000);
>-    return rtc::dchecked_cast<T>(us() * 1000);
>-  }
>-
>-  template <typename T>
>-  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>-  seconds() const {
>-    return us<T>() * 1e-6;
>-  }
>-  template <typename T>
>-  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>-  ms() const {
>-    return us<T>() * 1e-3;
>-  }
>-  template <typename T>
>-  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>-  us() const {
>-    return IsPlusInfinity()
>-               ? std::numeric_limits<T>::infinity()
>-               : IsMinusInfinity() ? -std::numeric_limits<T>::infinity()
>-                                   : microseconds_;
>-  }
>-  template <typename T>
>-  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>-  ns() const {
>-    return us<T>() * 1e3;
>+  T ns() const {
>+    return ToMultiple<1000, T>();
>   }
> 
>   constexpr int64_t seconds_or(int64_t fallback_value) const {
>-    return IsFinite() ? UnsafeSeconds() : fallback_value;
>+    return ToFractionOr<1000000>(fallback_value);
>   }
>   constexpr int64_t ms_or(int64_t fallback_value) const {
>-    return IsFinite() ? UnsafeMillis() : fallback_value;
>+    return ToFractionOr<1000>(fallback_value);
>   }
>   constexpr int64_t us_or(int64_t fallback_value) const {
>-    return IsFinite() ? microseconds_ : fallback_value;
>+    return ToValueOr(fallback_value);
>   }
> 
>   TimeDelta Abs() const { return TimeDelta::us(std::abs(us())); }
>-  constexpr bool IsZero() const { return microseconds_ == 0; }
>-  constexpr bool IsFinite() const { return !IsInfinite(); }
>-  constexpr bool IsInfinite() const {
>-    return microseconds_ == timedelta_impl::kPlusInfinityVal ||
>-           microseconds_ == timedelta_impl::kMinusInfinityVal;
>-  }
>-  constexpr bool IsPlusInfinity() const {
>-    return microseconds_ == timedelta_impl::kPlusInfinityVal;
>-  }
>-  constexpr bool IsMinusInfinity() const {
>-    return microseconds_ == timedelta_impl::kMinusInfinityVal;
>-  }
>-  TimeDelta operator+(const TimeDelta& other) const {
>-    if (IsPlusInfinity() || other.IsPlusInfinity()) {
>-      RTC_DCHECK(!IsMinusInfinity());
>-      RTC_DCHECK(!other.IsMinusInfinity());
>-      return PlusInfinity();
>-    } else if (IsMinusInfinity() || other.IsMinusInfinity()) {
>-      RTC_DCHECK(!IsPlusInfinity());
>-      RTC_DCHECK(!other.IsPlusInfinity());
>-      return MinusInfinity();
>-    }
>-    return TimeDelta::us(us() + other.us());
>-  }
>-  TimeDelta operator-(const TimeDelta& other) const {
>-    if (IsPlusInfinity() || other.IsMinusInfinity()) {
>-      RTC_DCHECK(!IsMinusInfinity());
>-      RTC_DCHECK(!other.IsPlusInfinity());
>-      return PlusInfinity();
>-    } else if (IsMinusInfinity() || other.IsPlusInfinity()) {
>-      RTC_DCHECK(!IsPlusInfinity());
>-      RTC_DCHECK(!other.IsMinusInfinity());
>-      return MinusInfinity();
>-    }
>-    return TimeDelta::us(us() - other.us());
>-  }
>-  TimeDelta& operator-=(const TimeDelta& other) {
>-    *this = *this - other;
>-    return *this;
>-  }
>-  TimeDelta& operator+=(const TimeDelta& other) {
>-    *this = *this + other;
>-    return *this;
>-  }
>-  constexpr double operator/(const TimeDelta& other) const {
>-    return us<double>() / other.us<double>();
>-  }
>-  constexpr bool operator==(const TimeDelta& other) const {
>-    return microseconds_ == other.microseconds_;
>-  }
>-  constexpr bool operator!=(const TimeDelta& other) const {
>-    return microseconds_ != other.microseconds_;
>-  }
>-  constexpr bool operator<=(const TimeDelta& other) const {
>-    return microseconds_ <= other.microseconds_;
>-  }
>-  constexpr bool operator>=(const TimeDelta& other) const {
>-    return microseconds_ >= other.microseconds_;
>-  }
>-  constexpr bool operator>(const TimeDelta& other) const {
>-    return microseconds_ > other.microseconds_;
>-  }
>-  constexpr bool operator<(const TimeDelta& other) const {
>-    return microseconds_ < other.microseconds_;
>-  }
> 
>  private:
>-  explicit constexpr TimeDelta(int64_t us) : microseconds_(us) {}
>-  constexpr int64_t UnsafeSeconds() const {
>-    return (microseconds_ + (microseconds_ >= 0 ? 500000 : -500000)) / 1000000;
>-  }
>-  constexpr int64_t UnsafeMillis() const {
>-    return (microseconds_ + (microseconds_ >= 0 ? 500 : -500)) / 1000;
>-  }
>-  int64_t microseconds_;
>+  friend class rtc_units_impl::UnitBase<TimeDelta>;
>+  using RelativeUnit::RelativeUnit;
>+  static constexpr bool one_sided = false;
> };
> 
>-inline TimeDelta operator*(const TimeDelta& delta, const double& scalar) {
>-  return TimeDelta::us(std::round(delta.us() * scalar));
>-}
>-inline TimeDelta operator*(const double& scalar, const TimeDelta& delta) {
>-  return delta * scalar;
>-}
>-inline TimeDelta operator*(const TimeDelta& delta, const int64_t& scalar) {
>-  return TimeDelta::us(delta.us() * scalar);
>-}
>-inline TimeDelta operator*(const int64_t& scalar, const TimeDelta& delta) {
>-  return delta * scalar;
>-}
>-inline TimeDelta operator*(const TimeDelta& delta, const int32_t& scalar) {
>-  return TimeDelta::us(delta.us() * scalar);
>-}
>-inline TimeDelta operator*(const int32_t& scalar, const TimeDelta& delta) {
>-  return delta * scalar;
>-}
>-
>-inline TimeDelta operator/(const TimeDelta& delta, const int64_t& scalar) {
>-  return TimeDelta::us(delta.us() / scalar);
>-}
>-std::string ToString(const TimeDelta& value);
>+std::string ToString(TimeDelta value);
> 
> #ifdef UNIT_TEST
> inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta_unittest.cc
>index bf8bbce69befed616228d1b4ce44c163bc447e3f..a46ba835cb18f4e5bb62a9a39a873cded37dff13 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta_unittest.cc
>@@ -10,6 +10,8 @@
> 
> #include "api/units/time_delta.h"
> 
>+#include <limits>
>+
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -106,6 +108,27 @@ TEST(TimeDeltaTest, ComparisonOperators) {
>   EXPECT_LT(TimeDelta::MinusInfinity(), TimeDelta::Zero());
> }
> 
>+TEST(TimeDeltaTest, Clamping) {
>+  const TimeDelta upper = TimeDelta::ms(800);
>+  const TimeDelta lower = TimeDelta::ms(100);
>+  const TimeDelta under = TimeDelta::ms(100);
>+  const TimeDelta inside = TimeDelta::ms(500);
>+  const TimeDelta over = TimeDelta::ms(1000);
>+  EXPECT_EQ(under.Clamped(lower, upper), lower);
>+  EXPECT_EQ(inside.Clamped(lower, upper), inside);
>+  EXPECT_EQ(over.Clamped(lower, upper), upper);
>+
>+  TimeDelta mutable_delta = lower;
>+  mutable_delta.Clamp(lower, upper);
>+  EXPECT_EQ(mutable_delta, lower);
>+  mutable_delta = inside;
>+  mutable_delta.Clamp(lower, upper);
>+  EXPECT_EQ(mutable_delta, inside);
>+  mutable_delta = over;
>+  mutable_delta.Clamp(lower, upper);
>+  EXPECT_EQ(mutable_delta, upper);
>+}
>+
> TEST(TimeDeltaTest, CanBeInititializedFromLargeInt) {
>   const int kMaxInt = std::numeric_limits<int>::max();
>   EXPECT_EQ(TimeDelta::seconds(kMaxInt).us(),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.cc
>index feb144789ca2b59f024dd9b735010b47d4f5d1d2..d3417cfb05b2f2673e44779f9941e39553d9b387 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.cc
>@@ -13,7 +13,7 @@
> #include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
>-std::string ToString(const Timestamp& value) {
>+std::string ToString(Timestamp value) {
>   char buf[64];
>   rtc::SimpleStringBuilder sb(buf);
>   if (value.IsInfinite()) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.h
>index 0298f5da979342bdbc29980876622b2ea040acf1..a6e450f6850f559332c3d56a9ea3ee9402c55d4c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.h
>@@ -15,189 +15,94 @@
> #include <ostream>  // no-presubmit-check TODO(webrtc:8982)
> #endif              // UNIT_TEST
> 
>-#include <stdint.h>
>-#include <limits>
> #include <string>
>+#include <type_traits>
> 
> #include "api/units/time_delta.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/numerics/safe_conversions.h"
> 
> namespace webrtc {
>-namespace timestamp_impl {
>-constexpr int64_t kPlusInfinityVal = std::numeric_limits<int64_t>::max();
>-constexpr int64_t kMinusInfinityVal = std::numeric_limits<int64_t>::min();
>-}  // namespace timestamp_impl
>-
> // Timestamp represents the time that has passed since some unspecified epoch.
> // The epoch is assumed to be before any represented timestamps, this means that
> // negative values are not valid. The most notable feature is that the
> // difference of two Timestamps results in a TimeDelta.
>-class Timestamp {
>+class Timestamp final : public rtc_units_impl::UnitBase<Timestamp> {
>  public:
>   Timestamp() = delete;
>-  static constexpr Timestamp PlusInfinity() {
>-    return Timestamp(timestamp_impl::kPlusInfinityVal);
>-  }
>-  static constexpr Timestamp MinusInfinity() {
>-    return Timestamp(timestamp_impl::kMinusInfinityVal);
>-  }
>+
>   template <int64_t seconds>
>   static constexpr Timestamp Seconds() {
>-    static_assert(seconds >= 0, "");
>-    static_assert(seconds < timestamp_impl::kPlusInfinityVal / 1000000, "");
>-    return Timestamp(seconds * 1000000);
>+    return FromStaticFraction<seconds, 1000000>();
>   }
>   template <int64_t ms>
>   static constexpr Timestamp Millis() {
>-    static_assert(ms >= 0, "");
>-    static_assert(ms < timestamp_impl::kPlusInfinityVal / 1000, "");
>-    return Timestamp(ms * 1000);
>+    return FromStaticFraction<ms, 1000>();
>   }
>   template <int64_t us>
>   static constexpr Timestamp Micros() {
>-    static_assert(us >= 0, "");
>-    static_assert(us < timestamp_impl::kPlusInfinityVal, "");
>-    return Timestamp(us);
>-  }
>-
>-  template <
>-      typename T,
>-      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>-  static Timestamp seconds(T seconds) {
>-    RTC_DCHECK_GE(seconds, 0);
>-    RTC_DCHECK_LT(seconds, timestamp_impl::kPlusInfinityVal / 1000000);
>-    return Timestamp(rtc::dchecked_cast<int64_t>(seconds) * 1000000);
>-  }
>-
>-  template <
>-      typename T,
>-      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>-  static Timestamp ms(T milliseconds) {
>-    RTC_DCHECK_GE(milliseconds, 0);
>-    RTC_DCHECK_LT(milliseconds, timestamp_impl::kPlusInfinityVal / 1000);
>-    return Timestamp(rtc::dchecked_cast<int64_t>(milliseconds) * 1000);
>+    return FromStaticValue<us>();
>   }
> 
>-  template <
>-      typename T,
>-      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>-  static Timestamp us(T microseconds) {
>-    RTC_DCHECK_GE(microseconds, 0);
>-    RTC_DCHECK_LT(microseconds, timestamp_impl::kPlusInfinityVal);
>-    return Timestamp(rtc::dchecked_cast<int64_t>(microseconds));
>-  }
>-
>-  template <typename T,
>-            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>-                nullptr>
>+  template <typename T>
>   static Timestamp seconds(T seconds) {
>-    return Timestamp::us(seconds * 1e6);
>+    return FromFraction<1000000>(seconds);
>   }
>-
>-  template <typename T,
>-            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>-                nullptr>
>+  template <typename T>
>   static Timestamp ms(T milliseconds) {
>-    return Timestamp::us(milliseconds * 1e3);
>+    return FromFraction<1000>(milliseconds);
>   }
>-  template <typename T,
>-            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>-                nullptr>
>+  template <typename T>
>   static Timestamp us(T microseconds) {
>-    if (microseconds == std::numeric_limits<double>::infinity()) {
>-      return PlusInfinity();
>-    } else if (microseconds == -std::numeric_limits<double>::infinity()) {
>-      return MinusInfinity();
>-    } else {
>-      RTC_DCHECK(!std::isnan(microseconds));
>-      RTC_DCHECK_GE(microseconds, 0);
>-      RTC_DCHECK_LT(microseconds, timestamp_impl::kPlusInfinityVal);
>-      return Timestamp(rtc::dchecked_cast<int64_t>(microseconds));
>-    }
>+    return FromValue(microseconds);
>   }
>-
>   template <typename T = int64_t>
>-  typename std::enable_if<std::is_integral<T>::value, T>::type seconds() const {
>-    RTC_DCHECK(IsFinite());
>-    return rtc::dchecked_cast<T>(UnsafeSeconds());
>+  T seconds() const {
>+    return ToFraction<1000000, T>();
>   }
>   template <typename T = int64_t>
>-  typename std::enable_if<std::is_integral<T>::value, T>::type ms() const {
>-    RTC_DCHECK(IsFinite());
>-    return rtc::dchecked_cast<T>(UnsafeMillis());
>+  T ms() const {
>+    return ToFraction<1000, T>();
>   }
>   template <typename T = int64_t>
>-  typename std::enable_if<std::is_integral<T>::value, T>::type us() const {
>-    RTC_DCHECK(IsFinite());
>-    return rtc::dchecked_cast<T>(microseconds_);
>-  }
>-
>-  template <typename T>
>-  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>-  seconds() const {
>-    return us<T>() * 1e-6;
>-  }
>-  template <typename T>
>-  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>-  ms() const {
>-    return us<T>() * 1e-3;
>-  }
>-  template <typename T>
>-  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>-  us() const {
>-    return IsPlusInfinity()
>-               ? std::numeric_limits<T>::infinity()
>-               : IsMinusInfinity() ? -std::numeric_limits<T>::infinity()
>-                                   : microseconds_;
>+  T us() const {
>+    return ToValue<T>();
>   }
> 
>   constexpr int64_t seconds_or(int64_t fallback_value) const {
>-    return IsFinite() ? UnsafeSeconds() : fallback_value;
>+    return ToFractionOr<1000000>(fallback_value);
>   }
>   constexpr int64_t ms_or(int64_t fallback_value) const {
>-    return IsFinite() ? UnsafeMillis() : fallback_value;
>+    return ToFractionOr<1000>(fallback_value);
>   }
>   constexpr int64_t us_or(int64_t fallback_value) const {
>-    return IsFinite() ? microseconds_ : fallback_value;
>+    return ToValueOr(fallback_value);
>   }
> 
>-  constexpr bool IsFinite() const { return !IsInfinite(); }
>-  constexpr bool IsInfinite() const {
>-    return microseconds_ == timedelta_impl::kPlusInfinityVal ||
>-           microseconds_ == timedelta_impl::kMinusInfinityVal;
>-  }
>-  constexpr bool IsPlusInfinity() const {
>-    return microseconds_ == timedelta_impl::kPlusInfinityVal;
>-  }
>-  constexpr bool IsMinusInfinity() const {
>-    return microseconds_ == timedelta_impl::kMinusInfinityVal;
>-  }
>-  Timestamp operator+(const TimeDelta& other) const {
>-    if (IsPlusInfinity() || other.IsPlusInfinity()) {
>+  Timestamp operator+(const TimeDelta delta) const {
>+    if (IsPlusInfinity() || delta.IsPlusInfinity()) {
>       RTC_DCHECK(!IsMinusInfinity());
>-      RTC_DCHECK(!other.IsMinusInfinity());
>+      RTC_DCHECK(!delta.IsMinusInfinity());
>       return PlusInfinity();
>-    } else if (IsMinusInfinity() || other.IsMinusInfinity()) {
>+    } else if (IsMinusInfinity() || delta.IsMinusInfinity()) {
>       RTC_DCHECK(!IsPlusInfinity());
>-      RTC_DCHECK(!other.IsPlusInfinity());
>+      RTC_DCHECK(!delta.IsPlusInfinity());
>       return MinusInfinity();
>     }
>-    return Timestamp::us(us() + other.us());
>+    return Timestamp::us(us() + delta.us());
>   }
>-  Timestamp operator-(const TimeDelta& other) const {
>-    if (IsPlusInfinity() || other.IsMinusInfinity()) {
>+  Timestamp operator-(const TimeDelta delta) const {
>+    if (IsPlusInfinity() || delta.IsMinusInfinity()) {
>       RTC_DCHECK(!IsMinusInfinity());
>-      RTC_DCHECK(!other.IsPlusInfinity());
>+      RTC_DCHECK(!delta.IsPlusInfinity());
>       return PlusInfinity();
>-    } else if (IsMinusInfinity() || other.IsPlusInfinity()) {
>+    } else if (IsMinusInfinity() || delta.IsPlusInfinity()) {
>       RTC_DCHECK(!IsPlusInfinity());
>-      RTC_DCHECK(!other.IsMinusInfinity());
>+      RTC_DCHECK(!delta.IsMinusInfinity());
>       return MinusInfinity();
>     }
>-    return Timestamp::us(us() - other.us());
>+    return Timestamp::us(us() - delta.us());
>   }
>-  TimeDelta operator-(const Timestamp& other) const {
>+  TimeDelta operator-(const Timestamp other) const {
>     if (IsPlusInfinity() || other.IsMinusInfinity()) {
>       RTC_DCHECK(!IsMinusInfinity());
>       RTC_DCHECK(!other.IsPlusInfinity());
>@@ -209,45 +114,22 @@ class Timestamp {
>     }
>     return TimeDelta::us(us() - other.us());
>   }
>-  Timestamp& operator-=(const TimeDelta& other) {
>-    *this = *this - other;
>+  Timestamp& operator-=(const TimeDelta delta) {
>+    *this = *this - delta;
>     return *this;
>   }
>-  Timestamp& operator+=(const TimeDelta& other) {
>-    *this = *this + other;
>+  Timestamp& operator+=(const TimeDelta delta) {
>+    *this = *this + delta;
>     return *this;
>   }
>-  constexpr bool operator==(const Timestamp& other) const {
>-    return microseconds_ == other.microseconds_;
>-  }
>-  constexpr bool operator!=(const Timestamp& other) const {
>-    return microseconds_ != other.microseconds_;
>-  }
>-  constexpr bool operator<=(const Timestamp& other) const {
>-    return microseconds_ <= other.microseconds_;
>-  }
>-  constexpr bool operator>=(const Timestamp& other) const {
>-    return microseconds_ >= other.microseconds_;
>-  }
>-  constexpr bool operator>(const Timestamp& other) const {
>-    return microseconds_ > other.microseconds_;
>-  }
>-  constexpr bool operator<(const Timestamp& other) const {
>-    return microseconds_ < other.microseconds_;
>-  }
> 
>  private:
>-  explicit constexpr Timestamp(int64_t us) : microseconds_(us) {}
>-  constexpr int64_t UnsafeSeconds() const {
>-    return (microseconds_ + 500000) / 1000000;
>-  }
>-  constexpr int64_t UnsafeMillis() const {
>-    return (microseconds_ + 500) / 1000;
>-  }
>-  int64_t microseconds_;
>+  friend class rtc_units_impl::UnitBase<Timestamp>;
>+  using UnitBase::UnitBase;
>+  static constexpr bool one_sided = true;
> };
> 
>-std::string ToString(const Timestamp& value);
>+std::string ToString(Timestamp value);
> 
> #ifdef UNIT_TEST
> inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/BUILD.gn
>index d04126021da5a6c1a292cc5b5289c502eaa5f800..5ed70898f7b7081df0d27b0fdb179a2ce8c91ddd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/BUILD.gn
>@@ -13,6 +13,9 @@ rtc_source_set("video_frame") {
>   sources = [
>     "color_space.cc",
>     "color_space.h",
>+    "hdr_metadata.cc",
>+    "hdr_metadata.h",
>+    "video_codec_type.h",
>     "video_content_type.cc",
>     "video_content_type.h",
>     "video_frame.cc",
>@@ -31,6 +34,7 @@ rtc_source_set("video_frame") {
>   deps = [
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -46,6 +50,7 @@ rtc_source_set("video_frame_i420") {
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base",
>     "../../rtc_base/memory:aligned_malloc",
>+    "../../rtc_base/system:rtc_export",
>     "//third_party/libyuv",
>   ]
> }
>@@ -66,6 +71,24 @@ rtc_source_set("video_frame_i010") {
>   ]
> }
> 
>+rtc_source_set("encoded_image") {
>+  sources = [
>+    "encoded_image.cc",
>+    "encoded_image.h",
>+  ]
>+  deps = [
>+    ":video_frame",
>+    "../..:webrtc_common",
>+
>+    # TODO(bugs.webrtc.org/8311) Cleanup when kMaxSpatialLayers is moved to proper include.
>+    "../../api/video:video_bitrate_allocation",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base/system:rtc_export",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
> rtc_source_set("encoded_frame") {
>   visibility = [ "*" ]
>   sources = [
>@@ -152,6 +175,7 @@ rtc_source_set("video_stream_encoder") {
>   ]
> 
>   deps = [
>+    ":video_bitrate_allocator_factory",
>     ":video_frame",
> 
>     # For rtpparameters.h
>@@ -182,3 +206,22 @@ rtc_source_set("video_stream_encoder_create") {
>     "//third_party/abseil-cpp/absl/memory",
>   ]
> }
>+
>+rtc_static_library("builtin_video_bitrate_allocator_factory") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "builtin_video_bitrate_allocator_factory.cc",
>+    "builtin_video_bitrate_allocator_factory.h",
>+  ]
>+
>+  deps = [
>+    ":video_bitrate_allocation",
>+    ":video_bitrate_allocator_factory",
>+    "../../media:rtc_media_base",
>+    "../../modules/video_coding:video_coding_utility",
>+    "../../modules/video_coding:webrtc_vp9_helpers",
>+    "../../rtc_base:ptr_util",
>+    "../../rtc_base/system:fallthrough",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/builtin_video_bitrate_allocator_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/builtin_video_bitrate_allocator_factory.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..70f6ad04cbe2520d9342d4979f95a7402a22d652
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/builtin_video_bitrate_allocator_factory.cc
>@@ -0,0 +1,56 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
>+
>+#include "absl/memory/memory.h"
>+#include "media/base/codec.h"
>+#include "modules/video_coding/codecs/vp9/svc_rate_allocator.h"
>+#include "modules/video_coding/utility/default_video_bitrate_allocator.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
>+#include "rtc_base/system/fallthrough.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+class BuiltinVideoBitrateAllocatorFactory
>+    : public VideoBitrateAllocatorFactory {
>+ public:
>+  BuiltinVideoBitrateAllocatorFactory() = default;
>+  ~BuiltinVideoBitrateAllocatorFactory() override = default;
>+
>+  std::unique_ptr<VideoBitrateAllocator> CreateVideoBitrateAllocator(
>+      const VideoCodec& codec) override {
>+    std::unique_ptr<VideoBitrateAllocator> rate_allocator;
>+    switch (codec.codecType) {
>+      case kVideoCodecVP8:
>+        RTC_FALLTHROUGH();
>+      case kVideoCodecH264:
>+        rate_allocator.reset(new SimulcastRateAllocator(codec));
>+        break;
>+      case kVideoCodecVP9:
>+        rate_allocator.reset(new SvcRateAllocator(codec));
>+        break;
>+      default:
>+        rate_allocator.reset(new DefaultVideoBitrateAllocator(codec));
>+    }
>+    return rate_allocator;
>+  }
>+};
>+
>+}  // namespace
>+
>+std::unique_ptr<VideoBitrateAllocatorFactory>
>+CreateBuiltinVideoBitrateAllocatorFactory() {
>+  return absl::make_unique<BuiltinVideoBitrateAllocatorFactory>();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/builtin_video_bitrate_allocator_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/builtin_video_bitrate_allocator_factory.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..ac880a0863e8ce42c14637de7d7b0a4b398dcdb7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/builtin_video_bitrate_allocator_factory.h
>@@ -0,0 +1,25 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_BUILTIN_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>+#define API_VIDEO_BUILTIN_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>+
>+#include <memory>
>+
>+#include "api/video/video_bitrate_allocator_factory.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<VideoBitrateAllocatorFactory>
>+CreateBuiltinVideoBitrateAllocatorFactory();
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_BUILTIN_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.cc
>index a8be5cd8280202d5fcf93ac7c5983b96de12dc64..ad138abe8093d0d6befaf56d2442f899651e4c64 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.cc
>@@ -10,18 +10,72 @@
> 
> #include "api/video/color_space.h"
> 
>+namespace {
>+// Try to convert |enum_value| into the enum class T. |enum_bitmask| is created
>+// by the funciton below. Returns true if conversion was successful, false
>+// otherwise.
>+template <typename T>
>+bool SetFromUint8(uint8_t enum_value, uint64_t enum_bitmask, T* out) {
>+  if ((enum_value < 64) && ((enum_bitmask >> enum_value) & 1)) {
>+    *out = static_cast<T>(enum_value);
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// This function serves as an assert for the constexpr function below. It's on
>+// purpose not declared as constexpr so that it causes a build problem if enum
>+// values of 64 or above are used. The bitmask and the code generating it would
>+// have to be extended if the standard is updated to include enum values >= 64.
>+int EnumMustBeLessThan64() {
>+  return -1;
>+}
>+
>+template <typename T, size_t N>
>+constexpr int MakeMask(const int index, const int length, T (&values)[N]) {
>+  return length > 1
>+             ? (MakeMask(index, 1, values) +
>+                MakeMask(index + 1, length - 1, values))
>+             : (static_cast<uint8_t>(values[index]) < 64
>+                    ? (uint64_t{1} << static_cast<uint8_t>(values[index]))
>+                    : EnumMustBeLessThan64());
>+}
>+
>+// Create a bitmask where each bit corresponds to one potential enum value.
>+// |values| should be an array listing all possible enum values. The bit is set
>+// to one if the corresponding enum exists. Only works for enums with values
>+// less than 64.
>+template <typename T, size_t N>
>+constexpr uint64_t CreateEnumBitmask(T (&values)[N]) {
>+  return MakeMask(0, N, values);
>+}
>+
>+}  // namespace
>+
> namespace webrtc {
> 
> ColorSpace::ColorSpace() = default;
>+ColorSpace::ColorSpace(const ColorSpace& other) = default;
>+ColorSpace::ColorSpace(ColorSpace&& other) = default;
>+ColorSpace& ColorSpace::operator=(const ColorSpace& other) = default;
> 
> ColorSpace::ColorSpace(PrimaryID primaries,
>                        TransferID transfer,
>                        MatrixID matrix,
>                        RangeID range)
>+    : ColorSpace(primaries, transfer, matrix, range, nullptr) {}
>+
>+ColorSpace::ColorSpace(PrimaryID primaries,
>+                       TransferID transfer,
>+                       MatrixID matrix,
>+                       RangeID range,
>+                       const HdrMetadata* hdr_metadata)
>     : primaries_(primaries),
>       transfer_(transfer),
>       matrix_(matrix),
>-      range_(range) {}
>+      range_(range),
>+      hdr_metadata_(hdr_metadata ? absl::make_optional(*hdr_metadata)
>+                                 : absl::nullopt) {}
> 
> ColorSpace::PrimaryID ColorSpace::primaries() const {
>   return primaries_;
>@@ -39,4 +93,61 @@ ColorSpace::RangeID ColorSpace::range() const {
>   return range_;
> }
> 
>+const HdrMetadata* ColorSpace::hdr_metadata() const {
>+  return hdr_metadata_ ? &*hdr_metadata_ : nullptr;
>+}
>+
>+bool ColorSpace::set_primaries_from_uint8(uint8_t enum_value) {
>+  constexpr PrimaryID kPrimaryIds[] = {
>+      PrimaryID::kInvalid,    PrimaryID::kBT709,      PrimaryID::kUNSPECIFIED,
>+      PrimaryID::kBT470M,     PrimaryID::kBT470BG,    PrimaryID::kSMPTE170M,
>+      PrimaryID::kSMPTE240M,  PrimaryID::kFILM,       PrimaryID::kBT2020,
>+      PrimaryID::kSMPTEST428, PrimaryID::kSMPTEST431, PrimaryID::kSMPTEST432,
>+      PrimaryID::kJEDECP22};
>+  constexpr uint64_t enum_bitmask = CreateEnumBitmask(kPrimaryIds);
>+
>+  return SetFromUint8(enum_value, enum_bitmask, &primaries_);
>+}
>+
>+bool ColorSpace::set_transfer_from_uint8(uint8_t enum_value) {
>+  constexpr TransferID kTransferIds[] = {
>+      TransferID::kInvalid,      TransferID::kBT709,
>+      TransferID::kUNSPECIFIED,  TransferID::kGAMMA22,
>+      TransferID::kGAMMA28,      TransferID::kSMPTE170M,
>+      TransferID::kSMPTE240M,    TransferID::kLINEAR,
>+      TransferID::kLOG,          TransferID::kLOG_SQRT,
>+      TransferID::kIEC61966_2_4, TransferID::kBT1361_ECG,
>+      TransferID::kIEC61966_2_1, TransferID::kBT2020_10,
>+      TransferID::kBT2020_12,    TransferID::kSMPTEST2084,
>+      TransferID::kSMPTEST428,   TransferID::kARIB_STD_B67};
>+  constexpr uint64_t enum_bitmask = CreateEnumBitmask(kTransferIds);
>+
>+  return SetFromUint8(enum_value, enum_bitmask, &transfer_);
>+}
>+
>+bool ColorSpace::set_matrix_from_uint8(uint8_t enum_value) {
>+  constexpr MatrixID kMatrixIds[] = {
>+      MatrixID::kRGB,       MatrixID::kBT709,        MatrixID::kUNSPECIFIED,
>+      MatrixID::kFCC,       MatrixID::kBT470BG,      MatrixID::kSMPTE170M,
>+      MatrixID::kSMPTE240M, MatrixID::kYCOCG,        MatrixID::kBT2020_NCL,
>+      MatrixID::kBT2020_CL, MatrixID::kSMPTE2085,    MatrixID::kCDNCLS,
>+      MatrixID::kCDCLS,     MatrixID::kBT2100_ICTCP, MatrixID::kInvalid};
>+  constexpr uint64_t enum_bitmask = CreateEnumBitmask(kMatrixIds);
>+
>+  return SetFromUint8(enum_value, enum_bitmask, &matrix_);
>+}
>+
>+bool ColorSpace::set_range_from_uint8(uint8_t enum_value) {
>+  constexpr RangeID kRangeIds[] = {RangeID::kInvalid, RangeID::kLimited,
>+                                   RangeID::kFull, RangeID::kDerived};
>+  constexpr uint64_t enum_bitmask = CreateEnumBitmask(kRangeIds);
>+
>+  return SetFromUint8(enum_value, enum_bitmask, &range_);
>+}
>+
>+void ColorSpace::set_hdr_metadata(const HdrMetadata* hdr_metadata) {
>+  hdr_metadata_ =
>+      hdr_metadata ? absl::make_optional(*hdr_metadata) : absl::nullopt;
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.h
>index 8102647b6ff8847a049004cbb59858e08b4d8471..79a15f59b393287bb89b98658ac64b44e186a069 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.h
>@@ -11,95 +11,145 @@
> #ifndef API_VIDEO_COLOR_SPACE_H_
> #define API_VIDEO_COLOR_SPACE_H_
> 
>+#include <stdint.h>
>+
>+#include "absl/types/optional.h"
>+#include "api/video/hdr_metadata.h"
>+
> namespace webrtc {
> 
>-// Used to represent a color space for the purpose of color conversion. This
>-// class only represents color information that can be transferred through the
>-// bitstream of WebRTC's internal supported codecs:
>+// This class represents color information as specified in T-REC H.273,
>+// available from https://www.itu.int/rec/T-REC-H.273.
>+//
>+// WebRTC's supported codecs:
> // - VP9 supports color profiles, see VP9 Bitstream & Decoding Process
> // Specification Version 0.6 Section 7.2.2 "Color config semantics" available
> // from https://www.webmproject.org.
>-// TODO(emircan): Extract these values from decode and add to the existing ones.
> // - VP8 only supports BT.601, see
> // https://tools.ietf.org/html/rfc6386#section-9.2
>-// - H264 supports different color primaries, transfer characteristics, matrix
>-// coefficients and range. See T-REC-H.264 E.2.1, "VUI parameters semantics",
>-// available from https://www.itu.int/rec/T-REC-H.264.
>+// - H264 uses the exact same representation as T-REC H.273. See T-REC-H.264
>+// E.2.1, "VUI parameters semantics", available from
>+// https://www.itu.int/rec/T-REC-H.264.
>+
> class ColorSpace {
>  public:
>-  enum class PrimaryID {
>-    kInvalid,
>-    kBT709,
>-    kBT470M,
>-    kBT470BG,
>-    kSMPTE170M,  // Identical to BT601
>-    kSMPTE240M,
>-    kFILM,
>-    kBT2020,
>-    kSMPTEST428,
>-    kSMPTEST431,
>-    kSMPTEST432,
>-    kJEDECP22,
>+  enum class PrimaryID : uint8_t {
>+    // The indices are equal to the values specified in T-REC H.273 Table 2.
>+    kInvalid = 0,
>+    kBT709 = 1,
>+    kUNSPECIFIED = 2,
>+    kBT470M = 4,
>+    kBT470BG = 5,
>+    kSMPTE170M = 6,  // Identical to BT601
>+    kSMPTE240M = 7,
>+    kFILM = 8,
>+    kBT2020 = 9,
>+    kSMPTEST428 = 10,
>+    kSMPTEST431 = 11,
>+    kSMPTEST432 = 12,
>+    kJEDECP22 = 22,  // Identical to EBU3213-E
>+    // When adding/removing entries here, please make sure to do the
>+    // corresponding change to kPrimaryIds.
>   };
> 
>-  enum class TransferID {
>-    kInvalid,
>-    kBT709,
>-    kGAMMA22,
>-    kGAMMA28,
>-    kSMPTE170M,
>-    kSMPTE240M,
>-    kLINEAR,
>-    kLOG,
>-    kLOG_SQRT,
>-    kIEC61966_2_4,
>-    kBT1361_ECG,
>-    kIEC61966_2_1,
>-    kBT2020_10,
>-    kBT2020_12,
>-    kSMPTEST2084,
>-    kSMPTEST428,
>-    kARIB_STD_B67,
>+  enum class TransferID : uint8_t {
>+    // The indices are equal to the values specified in T-REC H.273 Table 3.
>+    kInvalid = 0,
>+    kBT709 = 1,
>+    kUNSPECIFIED = 2,
>+    kGAMMA22 = 4,
>+    kGAMMA28 = 5,
>+    kSMPTE170M = 6,
>+    kSMPTE240M = 7,
>+    kLINEAR = 8,
>+    kLOG = 9,
>+    kLOG_SQRT = 10,
>+    kIEC61966_2_4 = 11,
>+    kBT1361_ECG = 12,
>+    kIEC61966_2_1 = 13,
>+    kBT2020_10 = 14,
>+    kBT2020_12 = 15,
>+    kSMPTEST2084 = 16,
>+    kSMPTEST428 = 17,
>+    kARIB_STD_B67 = 18,
>+    // When adding/removing entries here, please make sure to do the
>+    // corresponding change to kTransferIds.
>   };
> 
>-  enum class MatrixID {
>-    kInvalid,
>-    kRGB,
>-    kBT709,
>-    kFCC,
>-    kBT470BG,
>-    kSMPTE170M,
>-    kSMPTE240M,
>-    kYCOCG,
>-    kBT2020_NCL,
>-    kBT2020_CL,
>-    kSMPTE2085,
>+  enum class MatrixID : uint8_t {
>+    // The indices are equal to the values specified in T-REC H.273 Table 4.
>+    kRGB = 0,
>+    kBT709 = 1,
>+    kUNSPECIFIED = 2,
>+    kFCC = 4,
>+    kBT470BG = 5,
>+    kSMPTE170M = 6,
>+    kSMPTE240M = 7,
>+    kYCOCG = 8,
>+    kBT2020_NCL = 9,
>+    kBT2020_CL = 10,
>+    kSMPTE2085 = 11,
>+    kCDNCLS = 12,
>+    kCDCLS = 13,
>+    kBT2100_ICTCP = 14,
>+    kInvalid = 63,
>+    // When adding/removing entries here, please make sure to do the
>+    // corresponding change to kMatrixIds.
>   };
> 
>   enum class RangeID {
>-    kInvalid,
>+    // The indices are equal to the values specified at
>+    // https://www.webmproject.org/docs/container/#colour for the element Range.
>+    kInvalid = 0,
>     // Limited Rec. 709 color range with RGB values ranging from 16 to 235.
>-    kLimited,
>+    kLimited = 1,
>     // Full RGB color range with RGB valees from 0 to 255.
>-    kFull,
>+    kFull = 2,
>+    // Range is defined by MatrixCoefficients/TransferCharacteristics.
>+    kDerived = 3,
>+    // When adding/removing entries here, please make sure to do the
>+    // corresponding change to kRangeIds.
>   };
> 
>   ColorSpace();
>+  ColorSpace(const ColorSpace& other);
>+  ColorSpace(ColorSpace&& other);
>+  ColorSpace& operator=(const ColorSpace& other);
>   ColorSpace(PrimaryID primaries,
>              TransferID transfer,
>              MatrixID matrix,
>              RangeID full_range);
>+  ColorSpace(PrimaryID primaries,
>+             TransferID transfer,
>+             MatrixID matrix,
>+             RangeID range,
>+             const HdrMetadata* hdr_metadata);
>+  bool operator==(const ColorSpace& other) const {
>+    return primaries_ == other.primaries() && transfer_ == other.transfer() &&
>+           matrix_ == other.matrix() && range_ == other.range() &&
>+           ((hdr_metadata_.has_value() && other.hdr_metadata() &&
>+             *hdr_metadata_ == *other.hdr_metadata()) ||
>+            (!hdr_metadata_.has_value() && other.hdr_metadata() == nullptr));
>+  }
> 
>   PrimaryID primaries() const;
>   TransferID transfer() const;
>   MatrixID matrix() const;
>   RangeID range() const;
>+  const HdrMetadata* hdr_metadata() const;
>+
>+  bool set_primaries_from_uint8(uint8_t enum_value);
>+  bool set_transfer_from_uint8(uint8_t enum_value);
>+  bool set_matrix_from_uint8(uint8_t enum_value);
>+  bool set_range_from_uint8(uint8_t enum_value);
>+  void set_hdr_metadata(const HdrMetadata* hdr_metadata);
> 
>  private:
>   PrimaryID primaries_ = PrimaryID::kInvalid;
>   TransferID transfer_ = TransferID::kInvalid;
>   MatrixID matrix_ = MatrixID::kInvalid;
>   RangeID range_ = RangeID::kInvalid;
>+  absl::optional<HdrMetadata> hdr_metadata_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_image.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_image.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..e7c6fadee703c89acce249813c1d2224aef2238c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_image.cc
>@@ -0,0 +1,42 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video/encoded_image.h"
>+
>+#include <string.h>
>+
>+namespace webrtc {
>+
>+// FFmpeg's decoder, used by H264DecoderImpl, requires up to 8 bytes padding due
>+// to optimized bitstream readers. See avcodec_decode_video2.
>+const size_t EncodedImage::kBufferPaddingBytesH264 = 8;
>+
>+size_t EncodedImage::GetBufferPaddingBytes(VideoCodecType codec_type) {
>+  switch (codec_type) {
>+    case kVideoCodecH264:
>+      return kBufferPaddingBytesH264;
>+    default:
>+      return 0;
>+  }
>+}
>+
>+EncodedImage::EncodedImage() : EncodedImage(nullptr, 0, 0) {}
>+
>+EncodedImage::EncodedImage(const EncodedImage&) = default;
>+
>+EncodedImage::EncodedImage(uint8_t* buffer, size_t length, size_t size)
>+    : _buffer(buffer), _length(length), _size(size) {}
>+
>+void EncodedImage::SetEncodeTime(int64_t encode_start_ms,
>+                                 int64_t encode_finish_ms) {
>+  timing_.encode_start_ms = encode_start_ms;
>+  timing_.encode_finish_ms = encode_finish_ms;
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_image.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_image.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..a7c719ce3a5678d51ac24fc732039eebd1b90df0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_image.h
>@@ -0,0 +1,109 @@
>+/*
>+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_ENCODED_IMAGE_H_
>+#define API_VIDEO_ENCODED_IMAGE_H_
>+
>+#include <stdint.h>
>+
>+#include "absl/types/optional.h"
>+#include "api/video/color_space.h"
>+#include "api/video/video_bitrate_allocation.h"
>+#include "api/video/video_codec_type.h"
>+#include "api/video/video_content_type.h"
>+#include "api/video/video_rotation.h"
>+#include "api/video/video_timing.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "rtc_base/checks.h"
>+#include "rtc_base/system/rtc_export.h"
>+
>+namespace webrtc {
>+
>+// TODO(bug.webrtc.org/9378): This is a legacy api class, which is slowly being
>+// cleaned up. Direct use of its members is strongly discouraged.
>+class RTC_EXPORT EncodedImage {
>+ public:
>+  static const size_t kBufferPaddingBytesH264;
>+
>+  // Some decoders require encoded image buffers to be padded with a small
>+  // number of additional bytes (due to over-reading byte readers).
>+  static size_t GetBufferPaddingBytes(VideoCodecType codec_type);
>+
>+  EncodedImage();
>+  EncodedImage(const EncodedImage&);
>+  EncodedImage(uint8_t* buffer, size_t length, size_t size);
>+
>+  // TODO(nisse): Change style to timestamp(), set_timestamp(), for consistency
>+  // with the VideoFrame class.
>+  // Set frame timestamp (90kHz).
>+  void SetTimestamp(uint32_t timestamp) { timestamp_rtp_ = timestamp; }
>+
>+  // Get frame timestamp (90kHz).
>+  uint32_t Timestamp() const { return timestamp_rtp_; }
>+
>+  void SetEncodeTime(int64_t encode_start_ms, int64_t encode_finish_ms);
>+
>+  absl::optional<int> SpatialIndex() const {
>+    return spatial_index_;
>+  }
>+  void SetSpatialIndex(absl::optional<int> spatial_index) {
>+    RTC_DCHECK_GE(spatial_index.value_or(0), 0);
>+    RTC_DCHECK_LT(spatial_index.value_or(0), kMaxSpatialLayers);
>+    spatial_index_ = spatial_index;
>+  }
>+
>+  const webrtc::ColorSpace* ColorSpace() const {
>+    return color_space_ ? &*color_space_ : nullptr;
>+  }
>+  void SetColorSpace(const webrtc::ColorSpace* color_space) {
>+    color_space_ =
>+        color_space ? absl::make_optional(*color_space) : absl::nullopt;
>+  }
>+
>+  uint32_t _encodedWidth = 0;
>+  uint32_t _encodedHeight = 0;
>+  // NTP time of the capture time in local timebase in milliseconds.
>+  int64_t ntp_time_ms_ = 0;
>+  int64_t capture_time_ms_ = 0;
>+  FrameType _frameType = kVideoFrameDelta;
>+  uint8_t* _buffer;
>+  size_t _length;
>+  size_t _size;
>+  VideoRotation rotation_ = kVideoRotation_0;
>+  VideoContentType content_type_ = VideoContentType::UNSPECIFIED;
>+  bool _completeFrame = false;
>+  int qp_ = -1;  // Quantizer value.
>+
>+  // When an application indicates non-zero values here, it is taken as an
>+  // indication that all future frames will be constrained with those limits
>+  // until the application indicates a change again.
>+  PlayoutDelay playout_delay_ = {-1, -1};
>+
>+  struct Timing {
>+    uint8_t flags = VideoSendTiming::kInvalid;
>+    int64_t encode_start_ms = 0;
>+    int64_t encode_finish_ms = 0;
>+    int64_t packetization_finish_ms = 0;
>+    int64_t pacer_exit_ms = 0;
>+    int64_t network_timestamp_ms = 0;
>+    int64_t network2_timestamp_ms = 0;
>+    int64_t receive_start_ms = 0;
>+    int64_t receive_finish_ms = 0;
>+  } timing_;
>+
>+ private:
>+  uint32_t timestamp_rtp_ = 0;
>+  absl::optional<int> spatial_index_;
>+  absl::optional<webrtc::ColorSpace> color_space_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_ENCODED_IMAGE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/hdr_metadata.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/hdr_metadata.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..e2a669c98a3c0f079af5c176945ec29828bd2ddd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/hdr_metadata.cc
>@@ -0,0 +1,21 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video/hdr_metadata.h"
>+
>+namespace webrtc {
>+
>+HdrMasteringMetadata::Chromaticity::Chromaticity() = default;
>+
>+HdrMasteringMetadata::HdrMasteringMetadata() = default;
>+
>+HdrMetadata::HdrMetadata() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/hdr_metadata.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/hdr_metadata.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..676a900c99486421430f04dbed2b288713cee3bd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/hdr_metadata.h
>@@ -0,0 +1,87 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_HDR_METADATA_H_
>+#define API_VIDEO_HDR_METADATA_H_
>+
>+#include <stdint.h>
>+
>+namespace webrtc {
>+
>+// SMPTE ST 2086 mastering metadata,
>+// see https://ieeexplore.ieee.org/document/8353899.
>+struct HdrMasteringMetadata {
>+  struct Chromaticity {
>+    // xy chromaticity coordinates must be calculated as specified in ISO
>+    // 11664-3:2012 Section 7, and must be specified with four decimal places.
>+    // The x coordinate must be in the range [0.0001, 0.7400] and the y
>+    // coordinate must be in the range [0.0001, 0.8400].
>+    float x = 0.0f;
>+    float y = 0.0f;
>+    bool operator==(const Chromaticity& rhs) const {
>+      return x == rhs.x && y == rhs.y;
>+    }
>+
>+    Chromaticity();
>+  };
>+
>+  // The nominal primaries of the mastering display.
>+  Chromaticity primary_r;
>+  Chromaticity primary_g;
>+  Chromaticity primary_b;
>+
>+  // The nominal chromaticity of the white point of the mastering display.
>+  Chromaticity white_point;
>+
>+  // The nominal maximum display luminance of the mastering display. Specified
>+  // in the unit candela/m2. The value must be in the range [5, 10000] with zero
>+  // decimal places.
>+  float luminance_max = 0.0f;
>+
>+  // The nominal minimum display luminance of the mastering display. Specified
>+  // in the unit candela/m2. The value must be in the range [0.0001, 5.0000]
>+  // with four decimal places.
>+  float luminance_min = 0.0f;
>+
>+  HdrMasteringMetadata();
>+
>+  bool operator==(const HdrMasteringMetadata& rhs) const {
>+    return ((primary_r == rhs.primary_r) && (primary_g == rhs.primary_g) &&
>+            (primary_b == rhs.primary_b) && (white_point == rhs.white_point) &&
>+            (luminance_max == rhs.luminance_max) &&
>+            (luminance_min == rhs.luminance_min));
>+  }
>+};
>+
>+// High dynamic range (HDR) metadata common for HDR10 and WebM/VP9-based HDR
>+// formats. This struct replicates the HDRMetadata struct defined in
>+// https://cs.chromium.org/chromium/src/media/base/hdr_metadata.h
>+struct HdrMetadata {
>+  HdrMasteringMetadata mastering_metadata;
>+  // Max content light level (CLL), i.e. maximum brightness level present in the
>+  // stream, in nits. 1 nit = 1 candela/m2.
>+  uint32_t max_content_light_level = 0;
>+  // Max frame-average light level (FALL), i.e. maximum average brightness of
>+  // the brightest frame in the stream, in nits.
>+  uint32_t max_frame_average_light_level = 0;
>+
>+  HdrMetadata();
>+
>+  bool operator==(const HdrMetadata& rhs) const {
>+    return (
>+        (max_content_light_level == rhs.max_content_light_level) &&
>+        (max_frame_average_light_level == rhs.max_frame_average_light_level) &&
>+        (mastering_metadata == rhs.mastering_metadata));
>+  }
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_HDR_METADATA_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.h
>index 282b2426ca36e159228ee5b3ee71835c8bf77119..631e394e534f0b27479c1c6f6df61a051cc55f5b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.h
>@@ -11,16 +11,19 @@
> #ifndef API_VIDEO_I420_BUFFER_H_
> #define API_VIDEO_I420_BUFFER_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "api/video/video_frame_buffer.h"
> #include "api/video/video_rotation.h"
> #include "rtc_base/memory/aligned_malloc.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // Plain I420 buffer in standard memory.
>-class I420Buffer : public I420BufferInterface {
>+class RTC_EXPORT I420Buffer : public I420BufferInterface {
>  public:
>   static rtc::scoped_refptr<I420Buffer> Create(int width, int height);
>   static rtc::scoped_refptr<I420Buffer> Create(int width,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/BUILD.gn
>index cba42458d2215d13c5f50fe894ac8fca4c78fad2..c8269efdc079bbe37ede277aaa7f39b51205c89b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/BUILD.gn
>@@ -11,10 +11,12 @@ import("../../../webrtc.gni")
> rtc_source_set("rtc_api_video_unittests") {
>   testonly = true
>   sources = [
>+    "color_space_unittest.cc",
>     "video_bitrate_allocation_unittest.cc",
>   ]
>   deps = [
>     "..:video_bitrate_allocation",
>+    "..:video_frame",
>     "../../../test:test_support",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/color_space_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/color_space_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..b71bcef86a6bc52c1d16ed9b39e647607bb0b5f5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/color_space_unittest.cc
>@@ -0,0 +1,55 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <stdint.h>
>+
>+#include "api/video/color_space.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+TEST(ColorSpace, TestSettingPrimariesFromUint8) {
>+  ColorSpace color_space;
>+  EXPECT_TRUE(color_space.set_primaries_from_uint8(
>+      static_cast<uint8_t>(ColorSpace::PrimaryID::kBT470BG)));
>+  EXPECT_EQ(ColorSpace::PrimaryID::kBT470BG, color_space.primaries());
>+  EXPECT_FALSE(color_space.set_primaries_from_uint8(3));
>+  EXPECT_FALSE(color_space.set_primaries_from_uint8(23));
>+  EXPECT_FALSE(color_space.set_primaries_from_uint8(64));
>+}
>+
>+TEST(ColorSpace, TestSettingTransferFromUint8) {
>+  ColorSpace color_space;
>+  EXPECT_TRUE(color_space.set_transfer_from_uint8(
>+      static_cast<uint8_t>(ColorSpace::TransferID::kBT2020_10)));
>+  EXPECT_EQ(ColorSpace::TransferID::kBT2020_10, color_space.transfer());
>+  EXPECT_FALSE(color_space.set_transfer_from_uint8(3));
>+  EXPECT_FALSE(color_space.set_transfer_from_uint8(19));
>+  EXPECT_FALSE(color_space.set_transfer_from_uint8(128));
>+}
>+
>+TEST(ColorSpace, TestSettingMatrixFromUint8) {
>+  ColorSpace color_space;
>+  EXPECT_TRUE(color_space.set_matrix_from_uint8(
>+      static_cast<uint8_t>(ColorSpace::MatrixID::kCDNCLS)));
>+  EXPECT_EQ(ColorSpace::MatrixID::kCDNCLS, color_space.matrix());
>+  EXPECT_FALSE(color_space.set_matrix_from_uint8(3));
>+  EXPECT_FALSE(color_space.set_matrix_from_uint8(15));
>+  EXPECT_FALSE(color_space.set_matrix_from_uint8(255));
>+}
>+
>+TEST(ColorSpace, TestSettingRangeFromUint8) {
>+  ColorSpace color_space;
>+  EXPECT_TRUE(color_space.set_range_from_uint8(
>+      static_cast<uint8_t>(ColorSpace::RangeID::kFull)));
>+  EXPECT_EQ(ColorSpace::RangeID::kFull, color_space.range());
>+  EXPECT_FALSE(color_space.set_range_from_uint8(4));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.cc
>index 6c5ad1eae5fac150d1340682c31369f132bfc449..1b3569047e0e7809c34d3eb0d8be51418e399f5c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.cc
>@@ -10,12 +10,11 @@
> 
> #include "api/video/video_bitrate_allocation.h"
> 
>-#include <limits>
>+#include <cstdint>
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/strings/string_builder.h"
>-#include "rtc_base/stringutils.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.h
>index 9e2501d85a09aacb919e9cb8afaf70dae4e49c9b..d1771b44a435cb83a29ea94c631d2890307a9cff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.h
>@@ -11,6 +11,8 @@
> #ifndef API_VIDEO_VIDEO_BITRATE_ALLOCATION_H_
> #define API_VIDEO_VIDEO_BITRATE_ALLOCATION_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <limits>
> #include <string>
> #include <vector>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_codec_type.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_codec_type.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..447723cf088cdab2832bd1b51fc28810b120935c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_codec_type.h
>@@ -0,0 +1,30 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_VIDEO_CODEC_TYPE_H_
>+#define API_VIDEO_VIDEO_CODEC_TYPE_H_
>+
>+namespace webrtc {
>+
>+// Video codec types
>+enum VideoCodecType {
>+  // There are various memset(..., 0, ...) calls in the code that rely on
>+  // kVideoCodecGeneric being zero.
>+  kVideoCodecGeneric = 0,
>+  kVideoCodecVP8,
>+  kVideoCodecVP9,
>+  kVideoCodecH264,
>+  kVideoCodecI420,
>+  kVideoCodecMultiplex,
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_VIDEO_CODEC_TYPE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_content_type.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_content_type.h
>index 8c6460288d821cb66ef7140d9895e3c569893426..2d38a62366c577576a82bfd2d0bd4894dbc23731 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_content_type.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_content_type.h
>@@ -13,8 +13,6 @@
> 
> #include <stdint.h>
> 
>-#include <string>
>-
> namespace webrtc {
> 
> enum class VideoContentType : uint8_t {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.cc
>index 5521ad8a587441223743c2f9a8c91d43590050d9..eaae33b2b3ffd84f5f42e5553e053c1d2aafad7f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.cc
>@@ -64,6 +64,13 @@ VideoFrame::Builder& VideoFrame::Builder::set_color_space(
>   return *this;
> }
> 
>+VideoFrame::Builder& VideoFrame::Builder::set_color_space(
>+    const ColorSpace* color_space) {
>+  color_space_ =
>+      color_space ? absl::make_optional(*color_space) : absl::nullopt;
>+  return *this;
>+}
>+
> VideoFrame::VideoFrame(const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
>                        webrtc::VideoRotation rotation,
>                        int64_t timestamp_us)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.h
>index dcb533e29f0c09f2bae8797313ba18d52e716ffc..2c5d081ba621dc76dba6cef8fcc1296218e39992 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.h
>@@ -15,12 +15,15 @@
> 
> #include "absl/types/optional.h"
> #include "api/video/color_space.h"
>+#include "api/video/hdr_metadata.h"
> #include "api/video/video_frame_buffer.h"
> #include "api/video/video_rotation.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>-class VideoFrame {
>+class RTC_EXPORT VideoFrame {
>  public:
>   // Preferred way of building VideoFrame objects.
>   class Builder {
>@@ -37,6 +40,7 @@ class VideoFrame {
>     Builder& set_ntp_time_ms(int64_t ntp_time_ms);
>     Builder& set_rotation(VideoRotation rotation);
>     Builder& set_color_space(const ColorSpace& color_space);
>+    Builder& set_color_space(const ColorSpace* color_space);
> 
>    private:
>     rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer_;
>@@ -110,8 +114,10 @@ class VideoFrame {
>   VideoRotation rotation() const { return rotation_; }
>   void set_rotation(VideoRotation rotation) { rotation_ = rotation; }
> 
>-  // Set Color space when available.
>-  absl::optional<ColorSpace> color_space() const { return color_space_; }
>+  // Get color space when available.
>+  const ColorSpace* color_space() const {
>+    return color_space_ ? &*color_space_ : nullptr;
>+  }
> 
>   // Get render time in milliseconds.
>   // TODO(nisse): Deprecated. Migrate all users to timestamp_us().
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_source_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_source_interface.h
>index 4ee471945529df133c3d4d1833f8069e70d8e887..2bf73704bfaf1ab13c657b75e7f6544112a55413 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_source_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_source_interface.h
>@@ -15,12 +15,13 @@
> 
> #include "absl/types/optional.h"
> #include "api/video/video_sink_interface.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace rtc {
> 
> // VideoSinkWants is used for notifying the source of properties a video frame
> // should have when it is delivered to a certain sink.
>-struct VideoSinkWants {
>+struct RTC_EXPORT VideoSinkWants {
>   VideoSinkWants();
>   VideoSinkWants(const VideoSinkWants&);
>   ~VideoSinkWants();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_observer.h
>index b940efdf1150031f104e2a3971d8e73460eb57d3..98b5cfcb2cd30d9dfee04b3a183991f9361a53e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_observer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_observer.h
>@@ -11,6 +11,7 @@
> #ifndef API_VIDEO_VIDEO_STREAM_ENCODER_OBSERVER_H_
> #define API_VIDEO_VIDEO_STREAM_ENCODER_OBSERVER_H_
> 
>+#include <string>
> #include <vector>
> 
> #include "absl/types/optional.h"
>@@ -68,6 +69,9 @@ class VideoStreamEncoderObserver : public CpuOveruseMetricsObserver {
>   virtual void OnSendEncodedImage(const EncodedImage& encoded_image,
>                                   const CodecSpecificInfo* codec_info) = 0;
> 
>+  virtual void OnEncoderImplementationChanged(
>+      const std::string& implementation_name) = 0;
>+
>   virtual void OnFrameDropped(DropReason reason) = 0;
> 
>   // Used to indicate change in content type, which may require a change in
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_settings.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_settings.h
>index b67f33c982dbbecc08b18774b548bf0d89fc5831..37c1de7015f21b3a85d69bf20b0330cbaccfd9c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_settings.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_settings.h
>@@ -11,6 +11,7 @@
> #ifndef API_VIDEO_VIDEO_STREAM_ENCODER_SETTINGS_H_
> #define API_VIDEO_VIDEO_STREAM_ENCODER_SETTINGS_H_
> 
>+#include "api/video/video_bitrate_allocator_factory.h"
> #include "api/video_codecs/video_encoder_factory.h"
> 
> namespace webrtc {
>@@ -24,6 +25,9 @@ struct VideoStreamEncoderSettings {
> 
>   // Ownership stays with WebrtcVideoEngine (delegated from PeerConnection).
>   VideoEncoderFactory* encoder_factory = nullptr;
>+
>+  // Ownership stays with WebrtcVideoEngine (delegated from PeerConnection).
>+  VideoBitrateAllocatorFactory* bitrate_allocator_factory = nullptr;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/BUILD.gn
>index 55674d09010a98eda9a236b66b8c42ee4dcf2db6..d0b1b9ba8ea48b32fc0d1d41078701374089d97e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/BUILD.gn
>@@ -28,15 +28,18 @@ rtc_source_set("video_codecs_api") {
>     "video_encoder_config.cc",
>     "video_encoder_config.h",
>     "video_encoder_factory.h",
>+    "vp8_temporal_layers.h",
>   ]
> 
>   deps = [
>     "../..:webrtc_common",
>-    "../../common_video",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base/system:rtc_export",
>+    "../video:encoded_image",
>     "../video:video_bitrate_allocation",
>     "../video:video_frame",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -56,6 +59,7 @@ rtc_static_library("builtin_video_decoder_factory") {
>     ":video_codecs_api",
>     "../../media:rtc_internal_video_codecs",
>     "../../rtc_base:ptr_util",
>+    "../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
> }
>@@ -75,8 +79,29 @@ rtc_static_library("builtin_video_encoder_factory") {
>     ":video_codecs_api",
>     "../../media:rtc_internal_video_codecs",
>     "../../media:rtc_media_base",
>+    "../../media:rtc_vp8_encoder_simulcast_proxy",
>     "../../rtc_base:ptr_util",
>+    "../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>+  ]
>+}
>+
>+rtc_static_library("create_vp8_temporal_layers") {
>+  visibility = [ "*" ]
>+  allow_poison = [ "software_video_codecs" ]
>+  sources = [
>+    "create_vp8_temporal_layers.cc",
>+    "create_vp8_temporal_layers.h",
>+  ]
>+
>+  deps = [
>+    ":video_codecs_api",
>+    "../..:webrtc_common",
>+    "../../modules/video_coding:video_codec_interface",
>+    "../../modules/video_coding:webrtc_vp8_temporal_layers",
>+    "../../system_wrappers:system_wrappers",
>+    "//third_party/abseil-cpp/absl/memory:memory",
>   ]
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/OWNERS
>new file mode 100644
>index 0000000000000000000000000000000000000000..0a70bddd34d3b8375fa11b4b36043f5189ab75a1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/OWNERS
>@@ -0,0 +1,4 @@
>+magjed@webrtc.org
>+nisse@webrtc.org
>+sprang@webrtc.org
>+brandtr@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_decoder_factory.h
>index 1f8e75c9ae7d52ada95f33ba0ec572612d1b658b..d516077d9991bb18af8b02cd7e6fee9e2d7008d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_decoder_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_decoder_factory.h
>@@ -14,11 +14,13 @@
> #include <memory>
> 
> #include "api/video_codecs/video_decoder_factory.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // Creates a new factory that can create the built-in types of video decoders.
>-std::unique_ptr<VideoDecoderFactory> CreateBuiltinVideoDecoderFactory();
>+RTC_EXPORT std::unique_ptr<VideoDecoderFactory>
>+CreateBuiltinVideoDecoderFactory();
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.cc
>index ca389b99e3b5612d91cc7a852060dc86fe1dcb4c..1d0827cbc0c4d996f71499149080bae53a4dc0ba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.cc
>@@ -13,6 +13,7 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>+#include "absl/strings/match.h"
> #include "api/video_codecs/sdp_video_format.h"
> #include "media/base/codec.h"
> #include "media/base/mediaconstants.h"
>@@ -59,7 +60,7 @@ class BuiltinVideoEncoderFactory : public VideoEncoderFactory {
>     if (IsFormatSupported(internal_encoder_factory_->GetSupportedFormats(),
>                           format)) {
>       internal_encoder =
>-          cricket::CodecNamesEq(format.name.c_str(), cricket::kVp8CodecName)
>+          absl::EqualsIgnoreCase(format.name, cricket::kVp8CodecName)
>               ? absl::make_unique<VP8EncoderSimulcastProxy>(
>                     internal_encoder_factory_.get(), format)
>               : internal_encoder_factory_->CreateVideoEncoder(format);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.h
>index 6a6618fb32724e4a138bda51d51bf99cf41c9e6d..2c4537205c59418f52c100f305bad4368c4642f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.h
>@@ -14,12 +14,14 @@
> #include <memory>
> 
> #include "api/video_codecs/video_encoder_factory.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // Creates a new factory that can create the built-in types of video encoders.
> // The factory has simulcast support for VP8.
>-std::unique_ptr<VideoEncoderFactory> CreateBuiltinVideoEncoderFactory();
>+RTC_EXPORT std::unique_ptr<VideoEncoderFactory>
>+CreateBuiltinVideoEncoderFactory();
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/create_vp8_temporal_layers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/create_vp8_temporal_layers.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..dd290f39fdd477eba05d4498a05aef8cc045825d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/create_vp8_temporal_layers.cc
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video_codecs/create_vp8_temporal_layers.h"
>+#include "absl/memory/memory.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
>+#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
>+#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>+#include "system_wrappers/include/clock.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<Vp8TemporalLayers> CreateVp8TemporalLayers(
>+    Vp8TemporalLayersType type,
>+    int num_temporal_layers) {
>+  switch (type) {
>+    case Vp8TemporalLayersType::kFixedPattern:
>+      return absl::make_unique<DefaultTemporalLayers>(num_temporal_layers);
>+    case Vp8TemporalLayersType::kBitrateDynamic:
>+      // Conference mode temporal layering for screen content in base stream.
>+      return absl::make_unique<ScreenshareLayers>(num_temporal_layers,
>+                                                  Clock::GetRealTimeClock());
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/create_vp8_temporal_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/create_vp8_temporal_layers.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..52f14c8eda39f34cea5ccd3f2ddfea93a9ddeb35
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/create_vp8_temporal_layers.h
>@@ -0,0 +1,26 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_CODECS_CREATE_VP8_TEMPORAL_LAYERS_H_
>+#define API_VIDEO_CODECS_CREATE_VP8_TEMPORAL_LAYERS_H_
>+
>+#include <memory>
>+
>+#include "api/video_codecs/vp8_temporal_layers.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<Vp8TemporalLayers> CreateVp8TemporalLayers(
>+    Vp8TemporalLayersType type,
>+    int num_temporal_layers);
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_CODECS_CREATE_VP8_TEMPORAL_LAYERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/sdp_video_format.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/sdp_video_format.h
>index c25b857fde7f705c2741435cf370dd99c3ee17c8..edb78191c75318fec5ec3b6c0e61bcca6a45bac4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/sdp_video_format.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/sdp_video_format.h
>@@ -14,11 +14,13 @@
> #include <map>
> #include <string>
> 
>+#include "rtc_base/system/rtc_export.h"
>+
> namespace webrtc {
> 
> // SDP specification for a single video codec.
> // NOTE: This class is still under development and may change without notice.
>-struct SdpVideoFormat {
>+struct RTC_EXPORT SdpVideoFormat {
>   using Parameters = std::map<std::string, std::string>;
> 
>   explicit SdpVideoFormat(const std::string& name);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/BUILD.gn
>index fa191625a1a4c021a5442ff924ff5e8ead031836..6f9dcaa2d8a1dee1a5b7b58202e51f7c55709da1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/BUILD.gn
>@@ -21,6 +21,7 @@ if (rtc_include_tests) {
>       "..:builtin_video_encoder_factory",
>       "..:rtc_software_fallback_wrappers",
>       "..:video_codecs_api",
>+      "../..:mock_video_encoder",
>       "../../../modules/video_coding:video_codec_interface",
>       "../../../modules/video_coding:video_coding_utility",
>       "../../../modules/video_coding:webrtc_vp8",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/builtin_video_encoder_factory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/builtin_video_encoder_factory_unittest.cc
>index 73957c999f04d35b6102e007de031413a9619787..30754d9bdbd1a18b98e3ae19f7d6bbc8bd339a8a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/builtin_video_encoder_factory_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/builtin_video_encoder_factory_unittest.cc
>@@ -27,11 +27,11 @@ TEST(BuiltinVideoEncoderFactoryTest, AnnouncesVp9AccordingToBuildFlags) {
>       break;
>     }
>   }
>-#if defined(RTC_DISABLE_VP9)
>-  EXPECT_FALSE(claims_vp9_support);
>-#else
>+#if defined(RTC_ENABLE_VP9)
>   EXPECT_TRUE(claims_vp9_support);
>-#endif  // defined(RTC_DISABLE_VP9)
>+#else
>+  EXPECT_FALSE(claims_vp9_support);
>+#endif  // defined(RTC_ENABLE_VP9)
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
>index 98ebed230511d8075ee256ba80ed97f5d1d02d5b..b374f3d1e1638da7f24906f06fa94250f9def9bb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
>@@ -8,23 +8,26 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "api/video_codecs/video_encoder_software_fallback_wrapper.h"
>-
> #include <utility>
> 
>+#include "api/test/mock_video_encoder.h"
> #include "api/video/i420_buffer.h"
> #include "api/video/video_bitrate_allocation.h"
>+#include "api/video_codecs/video_encoder_software_fallback_wrapper.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/include/video_error_codes.h"
> #include "modules/video_coding/utility/simulcast_rate_allocator.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/fakeclock.h"
> #include "test/field_trial.h"
>+#include "test/gmock.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>+using ::testing::Return;
>+
> namespace {
> const int kWidth = 320;
> const int kHeight = 240;
>@@ -34,6 +37,12 @@ const size_t kMaxPayloadSize = 800;
> const int kDefaultMinPixelsPerFrame = 320 * 180;
> const int kLowThreshold = 10;
> const int kHighThreshold = 20;
>+
>+VideoEncoder::EncoderInfo GetEncoderInfo(bool trusted_rate_controller) {
>+  VideoEncoder::EncoderInfo info;
>+  info.has_trusted_rate_controller = trusted_rate_controller;
>+  return info;
>+}
> }  // namespace
> 
> class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
>@@ -62,9 +71,7 @@ class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
>       ++encode_count_;
>       if (encode_complete_callback_ &&
>           encode_return_code_ == WEBRTC_VIDEO_CODEC_OK) {
>-        CodecSpecificInfo info;
>-        info.codec_name = ImplementationName();
>-        encode_complete_callback_->OnEncodedImage(EncodedImage(), &info,
>+        encode_complete_callback_->OnEncodedImage(EncodedImage(), nullptr,
>                                                   nullptr);
>       }
>       return encode_return_code_;
>@@ -81,26 +88,19 @@ class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
>       return WEBRTC_VIDEO_CODEC_OK;
>     }
> 
>-    int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override {
>-      ++set_channel_parameters_count_;
>-      return WEBRTC_VIDEO_CODEC_OK;
>-    }
>-
>     int32_t SetRateAllocation(const VideoBitrateAllocation& bitrate_allocation,
>                               uint32_t framerate) override {
>       ++set_rates_count_;
>       return WEBRTC_VIDEO_CODEC_OK;
>     }
> 
>-    bool SupportsNativeHandle() const override {
>+    EncoderInfo GetEncoderInfo() const override {
>       ++supports_native_handle_count_;
>-      return supports_native_handle_;
>-    }
>-
>-    const char* ImplementationName() const override { return "fake-encoder"; }
>-
>-    VideoEncoder::ScalingSettings GetScalingSettings() const override {
>-      return VideoEncoder::ScalingSettings(kLowThreshold, kHighThreshold);
>+      EncoderInfo info;
>+      info.scaling_settings = ScalingSettings(kLowThreshold, kHighThreshold);
>+      info.supports_native_handle = supports_native_handle_;
>+      info.implementation_name = "fake-encoder";
>+      return info;
>     }
> 
>     int init_encode_count_ = 0;
>@@ -109,7 +109,6 @@ class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
>     int encode_count_ = 0;
>     EncodedImageCallback* encode_complete_callback_ = nullptr;
>     int release_count_ = 0;
>-    int set_channel_parameters_count_ = 0;
>     int set_rates_count_ = 0;
>     mutable int supports_native_handle_count_ = 0;
>     bool supports_native_handle_ = false;
>@@ -122,11 +121,9 @@ class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
>         const CodecSpecificInfo* codec_specific_info,
>         const RTPFragmentationHeader* fragmentation) override {
>       ++callback_count_;
>-      last_codec_name_ = codec_specific_info->codec_name;
>       return Result(Result::OK, callback_count_);
>     }
>     int callback_count_ = 0;
>-    std::string last_codec_name_;
>   };
> 
>   void UtilizeFallbackEncoder();
>@@ -134,7 +131,8 @@ class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
>   void EncodeFrame();
>   void EncodeFrame(int expected_ret);
>   void CheckLastEncoderName(const char* expected_name) {
>-    EXPECT_STREQ(expected_name, callback_.last_codec_name_.c_str());
>+    EXPECT_EQ(expected_name,
>+              fallback_wrapper_->GetEncoderInfo().implementation_name);
>   }
> 
>   test::ScopedFieldTrials override_field_trials_;
>@@ -271,15 +269,6 @@ TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->Release());
> }
> 
>-TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>-       SetChannelParametersForwardedDuringFallback) {
>-  UtilizeFallbackEncoder();
>-  EXPECT_EQ(0, fake_encoder_->set_channel_parameters_count_);
>-  fallback_wrapper_->SetChannelParameters(1, 1);
>-  EXPECT_EQ(1, fake_encoder_->set_channel_parameters_count_);
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->Release());
>-}
>-
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>        SetRatesForwardedDuringFallback) {
>   UtilizeFallbackEncoder();
>@@ -291,15 +280,19 @@ TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
> 
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>        SupportsNativeHandleForwardedWithoutFallback) {
>-  fallback_wrapper_->SupportsNativeHandle();
>+  fallback_wrapper_->GetEncoderInfo();
>   EXPECT_EQ(1, fake_encoder_->supports_native_handle_count_);
> }
> 
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>        SupportsNativeHandleNotForwardedDuringFallback) {
>+  // Fake encoder signals support for native handle, default (libvpx) does not.
>+  fake_encoder_->supports_native_handle_ = true;
>+  EXPECT_TRUE(fallback_wrapper_->GetEncoderInfo().supports_native_handle);
>   UtilizeFallbackEncoder();
>-  fallback_wrapper_->SupportsNativeHandle();
>-  EXPECT_EQ(0, fake_encoder_->supports_native_handle_count_);
>+  EXPECT_FALSE(fallback_wrapper_->GetEncoderInfo().supports_native_handle);
>+  // Both times, both encoders are queried.
>+  EXPECT_EQ(2, fake_encoder_->supports_native_handle_count_);
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->Release());
> }
> 
>@@ -489,7 +482,7 @@ TEST_F(ForcedFallbackTestDisabled, GetScaleSettings) {
>   EncodeFrameAndVerifyLastName("fake-encoder");
> 
>   // Default min pixels per frame should be used.
>-  const auto settings = fallback_wrapper_->GetScalingSettings();
>+  const auto settings = fallback_wrapper_->GetEncoderInfo().scaling_settings;
>   EXPECT_TRUE(settings.thresholds.has_value());
>   EXPECT_EQ(kDefaultMinPixelsPerFrame, settings.min_pixels_per_frame);
> }
>@@ -500,7 +493,7 @@ TEST_F(ForcedFallbackTestEnabled, GetScaleSettingsWithNoFallback) {
>   EncodeFrameAndVerifyLastName("fake-encoder");
> 
>   // Configured min pixels per frame should be used.
>-  const auto settings = fallback_wrapper_->GetScalingSettings();
>+  const auto settings = fallback_wrapper_->GetEncoderInfo().scaling_settings;
>   EXPECT_EQ(kMinPixelsPerFrame, settings.min_pixels_per_frame);
>   ASSERT_TRUE(settings.thresholds);
>   EXPECT_EQ(kLowThreshold, settings.thresholds->low);
>@@ -513,7 +506,7 @@ TEST_F(ForcedFallbackTestEnabled, GetScaleSettingsWithFallback) {
>   EncodeFrameAndVerifyLastName("libvpx");
> 
>   // Configured min pixels per frame should be used.
>-  const auto settings = fallback_wrapper_->GetScalingSettings();
>+  const auto settings = fallback_wrapper_->GetEncoderInfo().scaling_settings;
>   EXPECT_TRUE(settings.thresholds.has_value());
>   EXPECT_EQ(kMinPixelsPerFrame, settings.min_pixels_per_frame);
> }
>@@ -525,8 +518,77 @@ TEST_F(ForcedFallbackTestEnabled, ScalingDisabledIfResizeOff) {
>   EncodeFrameAndVerifyLastName("libvpx");
> 
>   // Should be disabled for automatic resize off.
>-  const auto settings = fallback_wrapper_->GetScalingSettings();
>+  const auto settings = fallback_wrapper_->GetEncoderInfo().scaling_settings;
>   EXPECT_FALSE(settings.thresholds.has_value());
> }
> 
>+TEST(SoftwareFallbackEncoderTest, BothRateControllersNotTrusted) {
>+  auto* sw_encoder = new testing::NiceMock<MockVideoEncoder>();
>+  auto* hw_encoder = new testing::NiceMock<MockVideoEncoder>();
>+
>+  EXPECT_CALL(*sw_encoder, GetEncoderInfo())
>+      .WillRepeatedly(Return(GetEncoderInfo(false)));
>+  EXPECT_CALL(*hw_encoder, GetEncoderInfo())
>+      .WillRepeatedly(Return(GetEncoderInfo(false)));
>+
>+  std::unique_ptr<VideoEncoder> wrapper =
>+      CreateVideoEncoderSoftwareFallbackWrapper(
>+          std::unique_ptr<VideoEncoder>(sw_encoder),
>+          std::unique_ptr<VideoEncoder>(hw_encoder));
>+  EXPECT_FALSE(wrapper->GetEncoderInfo().has_trusted_rate_controller);
>+}
>+
>+TEST(SoftwareFallbackEncoderTest, SwRateControllerTrusted) {
>+  auto* sw_encoder = new testing::NiceMock<MockVideoEncoder>();
>+  auto* hw_encoder = new testing::NiceMock<MockVideoEncoder>();
>+  EXPECT_CALL(*sw_encoder, GetEncoderInfo())
>+      .WillRepeatedly(Return(GetEncoderInfo(true)));
>+  EXPECT_CALL(*hw_encoder, GetEncoderInfo())
>+      .WillRepeatedly(Return(GetEncoderInfo(false)));
>+
>+  std::unique_ptr<VideoEncoder> wrapper =
>+      CreateVideoEncoderSoftwareFallbackWrapper(
>+          std::unique_ptr<VideoEncoder>(sw_encoder),
>+          std::unique_ptr<VideoEncoder>(hw_encoder));
>+  EXPECT_FALSE(wrapper->GetEncoderInfo().has_trusted_rate_controller);
>+}
>+
>+TEST(SoftwareFallbackEncoderTest, HwRateControllerTrusted) {
>+  auto* sw_encoder = new testing::NiceMock<MockVideoEncoder>();
>+  auto* hw_encoder = new testing::NiceMock<MockVideoEncoder>();
>+  EXPECT_CALL(*sw_encoder, GetEncoderInfo())
>+      .WillRepeatedly(Return(GetEncoderInfo(false)));
>+  EXPECT_CALL(*hw_encoder, GetEncoderInfo())
>+      .WillRepeatedly(Return(GetEncoderInfo(true)));
>+
>+  std::unique_ptr<VideoEncoder> wrapper =
>+      CreateVideoEncoderSoftwareFallbackWrapper(
>+          std::unique_ptr<VideoEncoder>(sw_encoder),
>+          std::unique_ptr<VideoEncoder>(hw_encoder));
>+  EXPECT_TRUE(wrapper->GetEncoderInfo().has_trusted_rate_controller);
>+
>+  // Trigger fallback to software.
>+  EXPECT_CALL(*hw_encoder, Encode)
>+      .WillOnce(Return(WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE));
>+  VideoFrame frame = VideoFrame::Builder().build();
>+  wrapper->Encode(frame, nullptr, nullptr);
>+
>+  EXPECT_FALSE(wrapper->GetEncoderInfo().has_trusted_rate_controller);
>+}
>+
>+TEST(SoftwareFallbackEncoderTest, BothRateControllersTrusted) {
>+  auto* sw_encoder = new testing::NiceMock<MockVideoEncoder>();
>+  auto* hw_encoder = new testing::NiceMock<MockVideoEncoder>();
>+  EXPECT_CALL(*sw_encoder, GetEncoderInfo())
>+      .WillRepeatedly(Return(GetEncoderInfo(true)));
>+  EXPECT_CALL(*hw_encoder, GetEncoderInfo())
>+      .WillRepeatedly(Return(GetEncoderInfo(true)));
>+
>+  std::unique_ptr<VideoEncoder> wrapper =
>+      CreateVideoEncoderSoftwareFallbackWrapper(
>+          std::unique_ptr<VideoEncoder>(sw_encoder),
>+          std::unique_ptr<VideoEncoder>(hw_encoder));
>+  EXPECT_TRUE(wrapper->GetEncoderInfo().has_trusted_rate_controller);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.cc
>index d36535b31008d36dec340cb5553f75d9f8989d2e..0819c822a06de799c13d7a5d2d8206f470ed3708 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.cc
>@@ -11,13 +11,10 @@
> #include "api/video_codecs/video_codec.h"
> 
> #include <string.h>
>-#include <algorithm>
>-#include <limits>
> #include <string>
>-#include <type_traits>
> 
>+#include "absl/strings/match.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/strings/string_builder.h"
> #include "rtc_base/stringutils.h"
> 
> namespace webrtc {
>@@ -118,10 +115,6 @@ static const char* kPayloadNameI420 = "I420";
> static const char* kPayloadNameGeneric = "Generic";
> static const char* kPayloadNameMultiplex = "Multiplex";
> 
>-static bool CodecNamesEq(const char* name1, const char* name2) {
>-  return _stricmp(name1, name2) == 0;
>-}
>-
> const char* CodecTypeToPayloadString(VideoCodecType type) {
>   switch (type) {
>     case kVideoCodecVP8:
>@@ -139,15 +132,15 @@ const char* CodecTypeToPayloadString(VideoCodecType type) {
> }
> 
> VideoCodecType PayloadStringToCodecType(const std::string& name) {
>-  if (CodecNamesEq(name.c_str(), kPayloadNameVp8))
>+  if (absl::EqualsIgnoreCase(name, kPayloadNameVp8))
>     return kVideoCodecVP8;
>-  if (CodecNamesEq(name.c_str(), kPayloadNameVp9))
>+  if (absl::EqualsIgnoreCase(name, kPayloadNameVp9))
>     return kVideoCodecVP9;
>-  if (CodecNamesEq(name.c_str(), kPayloadNameH264))
>+  if (absl::EqualsIgnoreCase(name, kPayloadNameH264))
>     return kVideoCodecH264;
>-  if (CodecNamesEq(name.c_str(), kPayloadNameI420))
>+  if (absl::EqualsIgnoreCase(name, kPayloadNameI420))
>     return kVideoCodecI420;
>-  if (CodecNamesEq(name.c_str(), kPayloadNameMultiplex))
>+  if (absl::EqualsIgnoreCase(name, kPayloadNameMultiplex))
>     return kVideoCodecMultiplex;
>   return kVideoCodecGeneric;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.h
>index 5b4ebb25c074604bbeece813bdd4f6b6621c7660..9fba60c4e872710fcc8579ef24094febdbb47541 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.h
>@@ -11,9 +11,14 @@
> #ifndef API_VIDEO_CODECS_VIDEO_CODEC_H_
> #define API_VIDEO_CODECS_VIDEO_CODEC_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <string>
> 
>+#include "api/video/video_bitrate_allocation.h"
>+#include "api/video/video_codec_type.h"
> #include "common_types.h"  // NOLINT(build/include)
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -86,8 +91,8 @@ struct VideoCodecH264 {
> };
> 
> // Translates from name of codec to codec type and vice versa.
>-const char* CodecTypeToPayloadString(VideoCodecType type);
>-VideoCodecType PayloadStringToCodecType(const std::string& name);
>+RTC_EXPORT const char* CodecTypeToPayloadString(VideoCodecType type);
>+RTC_EXPORT VideoCodecType PayloadStringToCodecType(const std::string& name);
> 
> union VideoCodecUnion {
>   VideoCodecVP8 VP8;
>@@ -98,7 +103,7 @@ union VideoCodecUnion {
> enum class VideoCodecMode { kRealtimeVideo, kScreensharing };
> 
> // Common video codec properties
>-class VideoCodec {
>+class RTC_EXPORT VideoCodec {
>  public:
>   VideoCodec();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h
>index 7995fccb0563da6e10e3ce969a59d2ecee667991..36cf80fe84c763db25ef2ae7398b240421634806 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h
>@@ -15,9 +15,10 @@
> #include <string>
> #include <vector>
> 
>+#include "api/video/encoded_image.h"
> #include "api/video/video_frame.h"
> #include "api/video_codecs/video_codec.h"
>-#include "common_video/include/video_frame.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -25,7 +26,7 @@ namespace webrtc {
> struct CodecSpecificInfo;
> class VideoCodec;
> 
>-class DecodedImageCallback {
>+class RTC_EXPORT DecodedImageCallback {
>  public:
>   virtual ~DecodedImageCallback() {}
> 
>@@ -47,7 +48,7 @@ class DecodedImageCallback {
>   virtual int32_t ReceivedDecodedFrame(const uint64_t pictureId);
> };
> 
>-class VideoDecoder {
>+class RTC_EXPORT VideoDecoder {
>  public:
>   virtual ~VideoDecoder() {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.cc
>index 008780e38c4c19e2007df344441d1b7ab4f15c64..135687c970b7b6db891bd26f884c1a9682eb2473 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.cc
>@@ -10,6 +10,10 @@
> 
> #include "api/video_codecs/video_encoder.h"
> 
>+#include <string.h>
>+
>+#include "rtc_base/checks.h"
>+
> namespace webrtc {
> 
> // TODO(mflodman): Add default complexity for VP9 and VP9.
>@@ -79,6 +83,14 @@ VideoEncoder::ScalingSettings::~ScalingSettings() {}
> constexpr VideoEncoder::ScalingSettings::KOff
>     VideoEncoder::ScalingSettings::kOff;
> 
>+VideoEncoder::EncoderInfo::EncoderInfo()
>+    : scaling_settings(VideoEncoder::ScalingSettings::kOff),
>+      supports_native_handle(false),
>+      implementation_name("unknown"),
>+      has_trusted_rate_controller(false) {}
>+
>+VideoEncoder::EncoderInfo::~EncoderInfo() = default;
>+
> int32_t VideoEncoder::SetRates(uint32_t bitrate, uint32_t framerate) {
>   RTC_NOTREACHED() << "SetRate(uint32_t, uint32_t) is deprecated.";
>   return -1;
>@@ -101,4 +113,14 @@ bool VideoEncoder::SupportsNativeHandle() const {
> const char* VideoEncoder::ImplementationName() const {
>   return "unknown";
> }
>+
>+// TODO(webrtc:9722): Remove and make pure virtual when the three legacy
>+// methods called here are gone.
>+VideoEncoder::EncoderInfo VideoEncoder::GetEncoderInfo() const {
>+  EncoderInfo info;
>+  info.scaling_settings = GetScalingSettings();
>+  info.supports_native_handle = SupportsNativeHandle();
>+  info.implementation_name = ImplementationName();
>+  return info;
>+}
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h
>index 68d9b445580f388253d256915f9b8f96964a1dbd..e7deb7dbb9cdb2670e0424c2ba2d8a20d21292b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h
>@@ -16,11 +16,12 @@
> #include <vector>
> 
> #include "absl/types/optional.h"
>+#include "api/video/encoded_image.h"
> #include "api/video/video_bitrate_allocation.h"
> #include "api/video/video_frame.h"
> #include "api/video_codecs/video_codec.h"
>-#include "common_video/include/video_frame.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -73,7 +74,7 @@ class EncodedImageCallback {
>   virtual void OnDroppedFrame(DropReason reason) {}
> };
> 
>-class VideoEncoder {
>+class RTC_EXPORT VideoEncoder {
>  public:
>   struct QpThresholds {
>     QpThresholds(int l, int h) : low(l), high(h) {}
>@@ -102,13 +103,13 @@ class VideoEncoder {
>     ScalingSettings(KOff);  // NOLINT(runtime/explicit)
>     ~ScalingSettings();
> 
>-    const absl::optional<QpThresholds> thresholds;
>+    absl::optional<QpThresholds> thresholds;
> 
>     // We will never ask for a resolution lower than this.
>     // TODO(kthelgason): Lower this limit when better testing
>     // on MediaCodec and fallback implementations are in place.
>     // See https://bugs.chromium.org/p/webrtc/issues/detail?id=7206
>-    const int min_pixels_per_frame = 320 * 180;
>+    int min_pixels_per_frame = 320 * 180;
> 
>    private:
>     // Private constructor; to get an object without thresholds, use
>@@ -116,6 +117,36 @@ class VideoEncoder {
>     ScalingSettings();
>   };
> 
>+  // Struct containing metadata about the encoder implementing this interface.
>+  struct EncoderInfo {
>+    EncoderInfo();
>+    ~EncoderInfo();
>+
>+    // Any encoder implementation wishing to use the WebRTC provided
>+    // quality scaler must populate this field.
>+    ScalingSettings scaling_settings;
>+
>+    // If true, encoder supports working with a native handle (e.g. texture
>+    // handle for hw codecs) rather than requiring a raw I420 buffer.
>+    bool supports_native_handle;
>+
>+    // The name of this particular encoder implementation, e.g. "libvpx".
>+    std::string implementation_name;
>+
>+    // If this field is true, the encoder rate controller must perform
>+    // well even in difficult situations, and produce close to the specified
>+    // target bitrate seen over a reasonable time window, drop frames if
>+    // necessary in order to keep the rate correct, and react quickly to
>+    // changing bitrate targets. If this method returns true, we disable the
>+    // frame dropper in the media optimization module and rely entirely on the
>+    // encoder to produce media at a bitrate that closely matches the target.
>+    // Any overshooting may result in delay buildup. If this method returns
>+    // false (default behavior), the media opt frame dropper will drop input
>+    // frames if it suspect encoder misbehavior. Misbehavior is common,
>+    // especially in hardware codecs. Disable media opt at your own risk.
>+    bool has_trusted_rate_controller;
>+  };
>+
>   static VideoCodecVP8 GetDefaultVp8Settings();
>   static VideoCodecVP9 GetDefaultVp9Settings();
>   static VideoCodecH264 GetDefaultH264Settings();
>@@ -171,17 +202,6 @@ class VideoEncoder {
>                          const CodecSpecificInfo* codec_specific_info,
>                          const std::vector<FrameType>* frame_types) = 0;
> 
>-  // Inform the encoder of the new packet loss rate and the round-trip time of
>-  // the network.
>-  //
>-  // Input:
>-  //          - packet_loss : Fraction lost
>-  //                          (loss rate in percent = 100 * packetLoss / 255)
>-  //          - rtt         : Round-trip time in milliseconds
>-  // Return value           : WEBRTC_VIDEO_CODEC_OK if OK
>-  //                          <0 - Errors: WEBRTC_VIDEO_CODEC_ERROR
>-  virtual int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) = 0;
>-
>   // Inform the encoder about the new target bit rate.
>   //
>   // Input:
>@@ -196,12 +216,17 @@ class VideoEncoder {
>   virtual int32_t SetRateAllocation(const VideoBitrateAllocation& allocation,
>                                     uint32_t framerate);
> 
>-  // Any encoder implementation wishing to use the WebRTC provided
>-  // quality scaler must implement this method.
>+  // GetScalingSettings(), SupportsNativeHandle(), ImplementationName() are
>+  // deprecated, use GetEncoderInfo() instead.
>   virtual ScalingSettings GetScalingSettings() const;
>-
>   virtual bool SupportsNativeHandle() const;
>   virtual const char* ImplementationName() const;
>+
>+  // Returns meta-data about the encoder, such as implementation name.
>+  // The output of this method may change during runtime. For instance if a
>+  // hardware encoder fails, it may fall back to doing software encoding using
>+  // an implementation with different characteristics.
>+  virtual EncoderInfo GetEncoderInfo() const;
> };
> }  // namespace webrtc
> #endif  // API_VIDEO_CODECS_VIDEO_ENCODER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.cc
>index 74977eaea3fdd59fc573c1b183044209d71a89e9..66ff7c94eb43b756a33623c598d2b063a6e6f50a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.cc
>@@ -9,7 +9,6 @@
>  */
> #include "api/video_codecs/video_encoder_config.h"
> 
>-#include <algorithm>
> #include <string>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.h
>index e10f08190c68a332c31cc2cdaad49e2caebf8e32..9dd8e4fa5d02c4495cdbd9835c63a0cb5630b9c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.h
>@@ -11,6 +11,7 @@
> #ifndef API_VIDEO_CODECS_VIDEO_ENCODER_CONFIG_H_
> #define API_VIDEO_CODECS_VIDEO_ENCODER_CONFIG_H_
> 
>+#include <stddef.h>
> #include <string>
> #include <vector>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
>index 99f29707e441c029250b7edd9b05d983c1d9219c..fd0678543abefaa3d730852323d9d425087f6cc3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
>@@ -87,12 +87,9 @@ class VideoEncoderSoftwareFallbackWrapper final : public VideoEncoder {
>   int32_t Encode(const VideoFrame& frame,
>                  const CodecSpecificInfo* codec_specific_info,
>                  const std::vector<FrameType>* frame_types) override;
>-  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>   int32_t SetRateAllocation(const VideoBitrateAllocation& bitrate_allocation,
>                             uint32_t framerate) override;
>-  bool SupportsNativeHandle() const override;
>-  ScalingSettings GetScalingSettings() const override;
>-  const char* ImplementationName() const override;
>+  EncoderInfo GetEncoderInfo() const override;
> 
>  private:
>   bool InitFallbackEncoder();
>@@ -162,7 +159,7 @@ VideoEncoderSoftwareFallbackWrapper::VideoEncoderSoftwareFallbackWrapper(
>   if (forced_fallback_possible_) {
>     GetForcedFallbackParamsFromFieldTrialGroup(
>         &forced_fallback_.min_pixels_, &forced_fallback_.max_pixels_,
>-        encoder_->GetScalingSettings().min_pixels_per_frame -
>+        encoder_->GetEncoderInfo().scaling_settings.min_pixels_per_frame -
>             1);  // No HW below.
>   }
> }
>@@ -185,8 +182,6 @@ bool VideoEncoderSoftwareFallbackWrapper::InitFallbackEncoder() {
>     fallback_encoder_->RegisterEncodeCompleteCallback(callback_);
>   if (rates_set_)
>     fallback_encoder_->SetRateAllocation(bitrate_allocation_, framerate_);
>-  if (channel_parameters_set_)
>-    fallback_encoder_->SetChannelParameters(packet_loss_, rtt_);
> 
>   // Since we're switching to the fallback encoder, Release the real encoder. It
>   // may be re-initialized via InitEncode later, and it will continue to get
>@@ -206,7 +201,6 @@ int32_t VideoEncoderSoftwareFallbackWrapper::InitEncode(
>   max_payload_size_ = max_payload_size;
>   // Clear stored rate/channel parameters.
>   rates_set_ = false;
>-  channel_parameters_set_ = false;
>   ValidateSettingsForForcedFallback();
> 
>   // Try to reinit forced software codec if it is in use.
>@@ -270,18 +264,6 @@ int32_t VideoEncoderSoftwareFallbackWrapper::Encode(
>   return ret;
> }
> 
>-int32_t VideoEncoderSoftwareFallbackWrapper::SetChannelParameters(
>-    uint32_t packet_loss,
>-    int64_t rtt) {
>-  channel_parameters_set_ = true;
>-  packet_loss_ = packet_loss;
>-  rtt_ = rtt;
>-  int32_t ret = encoder_->SetChannelParameters(packet_loss, rtt);
>-  if (use_fallback_encoder_)
>-    return fallback_encoder_->SetChannelParameters(packet_loss, rtt);
>-  return ret;
>-}
>-
> int32_t VideoEncoderSoftwareFallbackWrapper::SetRateAllocation(
>     const VideoBitrateAllocation& bitrate_allocation,
>     uint32_t framerate) {
>@@ -294,29 +276,29 @@ int32_t VideoEncoderSoftwareFallbackWrapper::SetRateAllocation(
>   return ret;
> }
> 
>-bool VideoEncoderSoftwareFallbackWrapper::SupportsNativeHandle() const {
>-  return use_fallback_encoder_ ? fallback_encoder_->SupportsNativeHandle()
>-                               : encoder_->SupportsNativeHandle();
>-}
>+VideoEncoder::EncoderInfo VideoEncoderSoftwareFallbackWrapper::GetEncoderInfo()
>+    const {
>+  EncoderInfo fallback_encoder_info = fallback_encoder_->GetEncoderInfo();
>+  EncoderInfo default_encoder_info = encoder_->GetEncoderInfo();
>+
>+  EncoderInfo info =
>+      use_fallback_encoder_ ? fallback_encoder_info : default_encoder_info;
> 
>-VideoEncoder::ScalingSettings
>-VideoEncoderSoftwareFallbackWrapper::GetScalingSettings() const {
>   if (forced_fallback_possible_) {
>     const auto settings = forced_fallback_.active_
>-                              ? fallback_encoder_->GetScalingSettings()
>-                              : encoder_->GetScalingSettings();
>-    return settings.thresholds
>-               ? VideoEncoder::ScalingSettings(settings.thresholds->low,
>-                                               settings.thresholds->high,
>-                                               forced_fallback_.min_pixels_)
>-               : VideoEncoder::ScalingSettings::kOff;
>+                              ? fallback_encoder_info.scaling_settings
>+                              : default_encoder_info.scaling_settings;
>+    info.scaling_settings =
>+        settings.thresholds
>+            ? VideoEncoder::ScalingSettings(settings.thresholds->low,
>+                                            settings.thresholds->high,
>+                                            forced_fallback_.min_pixels_)
>+            : VideoEncoder::ScalingSettings::kOff;
>+  } else {
>+    info.scaling_settings = default_encoder_info.scaling_settings;
>   }
>-  return encoder_->GetScalingSettings();
>-}
> 
>-const char* VideoEncoderSoftwareFallbackWrapper::ImplementationName() const {
>-  return use_fallback_encoder_ ? fallback_encoder_->ImplementationName()
>-                               : encoder_->ImplementationName();
>+  return info;
> }
> 
> bool VideoEncoderSoftwareFallbackWrapper::IsForcedFallbackActive() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/vp8_temporal_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/vp8_temporal_layers.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..70223a42096545ac85727e8cce31ee7850783047
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/vp8_temporal_layers.h
>@@ -0,0 +1,195 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_CODECS_VP8_TEMPORAL_LAYERS_H_
>+#define API_VIDEO_CODECS_VP8_TEMPORAL_LAYERS_H_
>+
>+#include <memory>
>+#include <vector>
>+
>+namespace webrtc {
>+
>+// Some notes on the prerequisites of the TemporalLayers interface.
>+// * Vp8TemporalLayers is not thread safe, synchronization is the caller's
>+//   responsibility.
>+// * The encoder is assumed to encode all frames in order, and callbacks to
>+//   PopulateCodecSpecific() / FrameEncoded() must happen in the same order.
>+//
>+// This means that in the case of pipelining encoders, it is OK to have a chain
>+// of calls such as this:
>+// - UpdateLayerConfig(timestampA)
>+// - UpdateLayerConfig(timestampB)
>+// - PopulateCodecSpecific(timestampA, ...)
>+// - UpdateLayerConfig(timestampC)
>+// - OnEncodeDone(timestampA, 1234, ...)
>+// - UpdateLayerConfig(timestampC)
>+// - OnEncodeDone(timestampB, 0, ...)
>+// - OnEncodeDone(timestampC, 1234, ...)
>+// Note that UpdateLayerConfig() for a new frame can happen before
>+// FrameEncoded() for a previous one, but calls themselves must be both
>+// synchronized (e.g. run on a task queue) and in order (per type).
>+
>+// Two different flavors of temporal layers are currently available:
>+// kFixedPattern uses a fixed repeating pattern of 1-4 layers.
>+// kBitrateDynamic can allocate frames dynamically to 1 or 2 layers, based on
>+// the bitrate produced.
>+enum class Vp8TemporalLayersType { kFixedPattern, kBitrateDynamic };
>+
>+struct CodecSpecificInfoVP8;
>+
>+struct Vp8EncoderConfig {
>+  static constexpr size_t kMaxPeriodicity = 16;
>+  static constexpr size_t kMaxLayers = 5;
>+
>+  // Number of active temporal layers. Set to 0 if not used.
>+  uint32_t ts_number_layers;
>+  // Arrays of length |ts_number_layers|, indicating (cumulative) target bitrate
>+  // and rate decimator (e.g. 4 if every 4th frame is in the given layer) for
>+  // each active temporal layer, starting with temporal id 0.
>+  uint32_t ts_target_bitrate[kMaxLayers];
>+  uint32_t ts_rate_decimator[kMaxLayers];
>+
>+  // The periodicity of the temporal pattern. Set to 0 if not used.
>+  uint32_t ts_periodicity;
>+  // Array of length |ts_periodicity| indicating the sequence of temporal id's
>+  // to assign to incoming frames.
>+  uint32_t ts_layer_id[kMaxPeriodicity];
>+
>+  // Target bitrate, in bps.
>+  uint32_t rc_target_bitrate;
>+
>+  // Clamp QP to min/max. Use 0 to disable clamping.
>+  uint32_t rc_min_quantizer;
>+  uint32_t rc_max_quantizer;
>+};
>+
>+// Defined bit-maskable reference to the three buffers available in VP8.
>+enum class Vp8BufferReference : uint8_t {
>+  kNone = 0,
>+  kLast = 1,
>+  kGolden = 2,
>+  kAltref = 4
>+};
>+
>+// This interface defines a way of getting the encoder settings needed to
>+// realize a temporal layer structure.
>+class Vp8TemporalLayers {
>+ public:
>+  enum BufferFlags : int {
>+    kNone = 0,
>+    kReference = 1,
>+    kUpdate = 2,
>+    kReferenceAndUpdate = kReference | kUpdate,
>+  };
>+  enum FreezeEntropy { kFreezeEntropy };
>+
>+  struct FrameConfig {
>+    FrameConfig();
>+
>+    FrameConfig(BufferFlags last, BufferFlags golden, BufferFlags arf);
>+    FrameConfig(BufferFlags last,
>+                BufferFlags golden,
>+                BufferFlags arf,
>+                FreezeEntropy);
>+
>+    bool drop_frame;
>+    BufferFlags last_buffer_flags;
>+    BufferFlags golden_buffer_flags;
>+    BufferFlags arf_buffer_flags;
>+
>+    // The encoder layer ID is used to utilize the correct bitrate allocator
>+    // inside the encoder. It does not control references nor determine which
>+    // "actual" temporal layer this is. The packetizer temporal index determines
>+    // which layer the encoded frame should be packetized into.
>+    // Normally these are the same, but current temporal-layer strategies for
>+    // screenshare use one bitrate allocator for all layers, but attempt to
>+    // packetize / utilize references to split a stream into multiple layers,
>+    // with different quantizer settings, to hit target bitrate.
>+    // TODO(pbos): Screenshare layers are being reconsidered at the time of
>+    // writing, we might be able to remove this distinction, and have a temporal
>+    // layer imply both (the normal case).
>+    int encoder_layer_id;
>+    int packetizer_temporal_idx;
>+
>+    bool layer_sync;
>+
>+    bool freeze_entropy;
>+
>+    // Indicates in which order the encoder should search the reference buffers
>+    // when doing motion prediction. Set to kNone to use unspecified order. Any
>+    // buffer indicated here must not have the corresponding no_ref bit set.
>+    // If all three buffers can be reference, the one not listed here should be
>+    // searched last.
>+    Vp8BufferReference first_reference;
>+    Vp8BufferReference second_reference;
>+
>+   private:
>+    FrameConfig(BufferFlags last,
>+                BufferFlags golden,
>+                BufferFlags arf,
>+                bool freeze_entropy);
>+  };
>+
>+  virtual ~Vp8TemporalLayers() = default;
>+
>+  // If this method returns true, the encoder is free to drop frames for
>+  // instance in an effort to uphold encoding bitrate.
>+  // If this return false, the encoder must not drop any frames unless:
>+  //  1. Requested to do so via FrameConfig.drop_frame
>+  //  2. The frame to be encoded is requested to be a keyframe
>+  //  3. The encoded detected a large overshoot and decided to drop and then
>+  //     re-encode the image at a low bitrate. In this case the encoder should
>+  //     call OnEncodeDone() once with size = 0 to indicate drop, and then call
>+  //     OnEncodeDone() again when the frame has actually been encoded.
>+  virtual bool SupportsEncoderFrameDropping() const = 0;
>+
>+  // New target bitrate, per temporal layer.
>+  virtual void OnRatesUpdated(const std::vector<uint32_t>& bitrates_bps,
>+                              int framerate_fps) = 0;
>+
>+  // Called by the encoder before encoding a frame. |cfg| contains the current
>+  // configuration. If the TemporalLayers instance wishes any part of that
>+  // to be changed before the encode step, |cfg| should be changed and then
>+  // return true. If false is returned, the encoder will proceed without
>+  // updating the configuration.
>+  virtual bool UpdateConfiguration(Vp8EncoderConfig* cfg) = 0;
>+
>+  // Returns the recommended VP8 encode flags needed, and moves the temporal
>+  // pattern to the next frame.
>+  // The timestamp may be used as both a time and a unique identifier, and so
>+  // the caller must make sure no two frames use the same timestamp.
>+  // The timestamp uses a 90kHz RTP clock.
>+  // After calling this method, first call the actual encoder with the provided
>+  // frame configuration, and then OnEncodeDone() below.
>+  virtual FrameConfig UpdateLayerConfig(uint32_t rtp_timestamp) = 0;
>+
>+  // Called after the encode step is done. |rtp_timestamp| must match the
>+  // parameter use in the UpdateLayerConfig() call.
>+  // |is_keyframe| must be true iff the encoder decided to encode this frame as
>+  // a keyframe.
>+  // If the encoder decided to drop this frame, |size_bytes| must be set to 0,
>+  // otherwise it should indicate the size in bytes of the encoded frame.
>+  // If |size_bytes| > 0, and |vp8_info| is not null, the TemporalLayers
>+  // instance my update |vp8_info| with codec specific data such as temporal id.
>+  // Some fields of this struct may have already been populated by the encoder,
>+  // check before overwriting.
>+  // If |size_bytes| > 0, |qp| should indicate the frame-level QP this frame was
>+  // encoded at. If the encoder does not support extracting this, |qp| should be
>+  // set to 0.
>+  virtual void OnEncodeDone(uint32_t rtp_timestamp,
>+                            size_t size_bytes,
>+                            bool is_keyframe,
>+                            int qp,
>+                            CodecSpecificInfoVP8* vp8_info) = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_CODECS_VP8_TEMPORAL_LAYERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/BUILD.gn
>index 91cece93cb9aae50ab51aba645391de535d153d1..c045af637651d37e881ae31a89d7237bbfe96b4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/BUILD.gn
>@@ -26,19 +26,13 @@ rtc_static_library("audio") {
>     "audio_transport_impl.h",
>     "channel_receive.cc",
>     "channel_receive.h",
>-    "channel_receive_proxy.cc",
>-    "channel_receive_proxy.h",
>     "channel_send.cc",
>     "channel_send.h",
>-    "channel_send_proxy.cc",
>-    "channel_send_proxy.h",
>     "conversion.h",
>     "null_audio_poller.cc",
>     "null_audio_poller.h",
>     "remix_resample.cc",
>     "remix_resample.h",
>-    "time_interval.cc",
>-    "time_interval.h",
>     "transport_feedback_packet_loss_tracker.cc",
>     "transport_feedback_packet_loss_tracker.h",
>   ]
>@@ -49,7 +43,6 @@ rtc_static_library("audio") {
>   }
> 
>   deps = [
>-    "..:webrtc_common",
>     "../api:array_view",
>     "../api:call_api",
>     "../api:libjingle_peerconnection_api",
>@@ -65,12 +58,14 @@ rtc_static_library("audio") {
>     "../common_audio:common_audio_c",
>     "../logging:rtc_event_audio",
>     "../logging:rtc_event_log_api",
>+    "../logging:rtc_stream_config",
>     "../modules/audio_coding",
>+    "../modules/audio_coding:audio_encoder_cng",
>     "../modules/audio_coding:audio_format_conversion",
>     "../modules/audio_coding:audio_network_adaptor_config",
>-    "../modules/audio_coding:cng",
>     "../modules/audio_device",
>     "../modules/audio_processing",
>+    "../modules/audio_processing:api",
>     "../modules/bitrate_controller:bitrate_controller",
>     "../modules/pacing:pacing",
>     "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
>@@ -128,22 +123,34 @@ if (rtc_include_tests) {
>       "mock_voe_channel_proxy.h",
>       "remix_resample_unittest.cc",
>       "test/audio_stats_test.cc",
>-      "time_interval_unittest.cc",
>+      "test/media_transport_test.cc",
>       "transport_feedback_packet_loss_tracker_unittest.cc",
>     ]
>     deps = [
>       ":audio",
>       ":audio_end_to_end_test",
>+      "../api:loopback_media_transport",
>       "../api:mock_audio_mixer",
>+      "../api:mock_frame_decryptor",
>+      "../api:mock_frame_encryptor",
>       "../api/audio:audio_frame_api",
>+      "../api/audio_codecs:audio_codecs_api",
>+      "../api/audio_codecs/opus:audio_decoder_opus",
>+      "../api/audio_codecs/opus:audio_encoder_opus",
>       "../api/units:time_delta",
>+      "../call:mock_bitrate_allocator",
>       "../call:mock_call_interfaces",
>       "../call:mock_rtp_interfaces",
>       "../call:rtp_interfaces",
>       "../call:rtp_receiver",
>       "../common_audio",
>       "../logging:mocks",
>+      "../logging:rtc_event_log_api",
>       "../modules/audio_device:mock_audio_device",
>+      "../rtc_base:rtc_base_tests_utils",
>+
>+      # For TestAudioDeviceModule
>+      "../modules/audio_device:audio_device_impl",
>       "../modules/audio_mixer:audio_mixer_impl",
>       "../modules/audio_processing:audio_processing_statistics",
>       "../modules/audio_processing:mocks",
>@@ -151,9 +158,9 @@ if (rtc_include_tests) {
>       "../modules/pacing:pacing",
>       "../modules/rtp_rtcp:mock_rtp_rtcp",
>       "../modules/rtp_rtcp:rtp_rtcp_format",
>+      "../modules/utility",
>       "../rtc_base:checks",
>       "../rtc_base:rtc_base_approved",
>-      "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:rtc_task_queue",
>       "../rtc_base:safe_compare",
>       "../system_wrappers:system_wrappers",
>@@ -265,6 +272,7 @@ if (rtc_include_tests) {
>       "../test:single_threaded_task_queue",
>       "../test:test_common",
>       "../test:test_main",
>+      "../test:test_support",
>       "//testing/gtest",
>       "//third_party/abseil-cpp/absl/memory",
>     ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.cc
>index 563e8a036a303fde7879caeda6c523b8c2d043b9..8d4afe08e3767cbebb64d6dbde4a72cd23632ad3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.cc
>@@ -13,14 +13,17 @@
> #include <string>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
>+#include "api/array_view.h"
>+#include "api/audio_codecs/audio_format.h"
> #include "api/call/audio_sink.h"
>+#include "api/rtpparameters.h"
> #include "audio/audio_send_stream.h"
> #include "audio/audio_state.h"
>-#include "audio/channel_receive_proxy.h"
>+#include "audio/channel_receive.h"
> #include "audio/conversion.h"
>+#include "call/rtp_config.h"
> #include "call/rtp_stream_receiver_controller_interface.h"
>-#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
>-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/strings/string_builder.h"
>@@ -53,6 +56,7 @@ std::string AudioReceiveStream::Config::ToString() const {
>   ss << "{rtp: " << rtp.ToString();
>   ss << ", rtcp_send_transport: "
>      << (rtcp_send_transport ? "(Transport)" : "null");
>+  ss << ", media_transport: " << (media_transport ? "(Transport)" : "null");
>   if (!sync_group.empty()) {
>     ss << ", sync_group: " << sync_group;
>   }
>@@ -62,7 +66,7 @@ std::string AudioReceiveStream::Config::ToString() const {
> 
> namespace internal {
> namespace {
>-std::unique_ptr<voe::ChannelReceiveProxy> CreateChannelAndProxy(
>+std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
>     webrtc::AudioState* audio_state,
>     ProcessThread* module_process_thread,
>     const webrtc::AudioReceiveStream::Config& config,
>@@ -70,13 +74,13 @@ std::unique_ptr<voe::ChannelReceiveProxy> CreateChannelAndProxy(
>   RTC_DCHECK(audio_state);
>   internal::AudioState* internal_audio_state =
>       static_cast<internal::AudioState*>(audio_state);
>-  return absl::make_unique<voe::ChannelReceiveProxy>(
>-      absl::make_unique<voe::ChannelReceive>(
>-          module_process_thread, internal_audio_state->audio_device_module(),
>-          config.rtcp_send_transport, event_log, config.rtp.remote_ssrc,
>-          config.jitter_buffer_max_packets,
>-          config.jitter_buffer_fast_accelerate, config.decoder_factory,
>-          config.codec_pair_id, config.frame_decryptor));
>+  return voe::CreateChannelReceive(
>+      module_process_thread, internal_audio_state->audio_device_module(),
>+      config.media_transport, config.rtcp_send_transport, event_log,
>+      config.rtp.remote_ssrc, config.jitter_buffer_max_packets,
>+      config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
>+      config.decoder_factory, config.codec_pair_id, config.frame_decryptor,
>+      config.crypto_options);
> }
> }  // namespace
> 
>@@ -92,10 +96,10 @@ AudioReceiveStream::AudioReceiveStream(
>                          config,
>                          audio_state,
>                          event_log,
>-                         CreateChannelAndProxy(audio_state.get(),
>-                                               module_process_thread,
>-                                               config,
>-                                               event_log)) {}
>+                         CreateChannelReceive(audio_state.get(),
>+                                              module_process_thread,
>+                                              config,
>+                                              event_log)) {}
> 
> AudioReceiveStream::AudioReceiveStream(
>     RtpStreamReceiverControllerInterface* receiver_controller,
>@@ -103,25 +107,26 @@ AudioReceiveStream::AudioReceiveStream(
>     const webrtc::AudioReceiveStream::Config& config,
>     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
>     webrtc::RtcEventLog* event_log,
>-    std::unique_ptr<voe::ChannelReceiveProxy> channel_proxy)
>-    : audio_state_(audio_state), channel_proxy_(std::move(channel_proxy)) {
>+    std::unique_ptr<voe::ChannelReceiveInterface> channel_receive)
>+    : audio_state_(audio_state), channel_receive_(std::move(channel_receive)) {
>   RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc;
>-  RTC_DCHECK(receiver_controller);
>-  RTC_DCHECK(packet_router);
>   RTC_DCHECK(config.decoder_factory);
>   RTC_DCHECK(config.rtcp_send_transport);
>   RTC_DCHECK(audio_state_);
>-  RTC_DCHECK(channel_proxy_);
>+  RTC_DCHECK(channel_receive_);
> 
>   module_process_thread_checker_.DetachFromThread();
> 
>-  // Configure bandwidth estimation.
>-  channel_proxy_->RegisterReceiverCongestionControlObjects(packet_router);
>-
>-  // Register with transport.
>-  rtp_stream_receiver_ = receiver_controller->CreateReceiver(
>-      config.rtp.remote_ssrc, channel_proxy_.get());
>+  if (!config.media_transport) {
>+    RTC_DCHECK(receiver_controller);
>+    RTC_DCHECK(packet_router);
>+    // Configure bandwidth estimation.
>+    channel_receive_->RegisterReceiverCongestionControlObjects(packet_router);
> 
>+    // Register with transport.
>+    rtp_stream_receiver_ = receiver_controller->CreateReceiver(
>+        config.rtp.remote_ssrc, channel_receive_.get());
>+  }
>   ConfigureStream(this, config, true);
> }
> 
>@@ -129,8 +134,10 @@ AudioReceiveStream::~AudioReceiveStream() {
>   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << config_.rtp.remote_ssrc;
>   Stop();
>-  channel_proxy_->DisassociateSendChannel();
>-  channel_proxy_->ResetReceiverCongestionControlObjects();
>+  channel_receive_->SetAssociatedSendChannel(nullptr);
>+  if (!config_.media_transport) {
>+    channel_receive_->ResetReceiverCongestionControlObjects();
>+  }
> }
> 
> void AudioReceiveStream::Reconfigure(
>@@ -144,7 +151,7 @@ void AudioReceiveStream::Start() {
>   if (playing_) {
>     return;
>   }
>-  channel_proxy_->StartPlayout();
>+  channel_receive_->StartPlayout();
>   playing_ = true;
>   audio_state()->AddReceivingStream(this);
> }
>@@ -154,7 +161,7 @@ void AudioReceiveStream::Stop() {
>   if (!playing_) {
>     return;
>   }
>-  channel_proxy_->StopPlayout();
>+  channel_receive_->StopPlayout();
>   playing_ = false;
>   audio_state()->RemoveReceivingStream(this);
> }
>@@ -165,11 +172,11 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
>   stats.remote_ssrc = config_.rtp.remote_ssrc;
> 
>   webrtc::CallReceiveStatistics call_stats =
>-      channel_proxy_->GetRTCPStatistics();
>+      channel_receive_->GetRTCPStatistics();
>   // TODO(solenberg): Don't return here if we can't get the codec - return the
>   //                  stats we *can* get.
>   webrtc::CodecInst codec_inst = {0};
>-  if (!channel_proxy_->GetRecCodec(&codec_inst)) {
>+  if (!channel_receive_->GetRecCodec(&codec_inst)) {
>     return stats;
>   }
> 
>@@ -186,13 +193,13 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
>   if (codec_inst.plfreq / 1000 > 0) {
>     stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000);
>   }
>-  stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate();
>-  stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange();
>-  stats.total_output_energy = channel_proxy_->GetTotalOutputEnergy();
>-  stats.total_output_duration = channel_proxy_->GetTotalOutputDuration();
>+  stats.delay_estimate_ms = channel_receive_->GetDelayEstimate();
>+  stats.audio_level = channel_receive_->GetSpeechOutputLevelFullRange();
>+  stats.total_output_energy = channel_receive_->GetTotalOutputEnergy();
>+  stats.total_output_duration = channel_receive_->GetTotalOutputDuration();
> 
>   // Get jitter buffer and total delay (alg + jitter + playout) stats.
>-  auto ns = channel_proxy_->GetNetworkStatistics();
>+  auto ns = channel_receive_->GetNetworkStatistics();
>   stats.jitter_buffer_ms = ns.currentBufferSize;
>   stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
>   stats.total_samples_received = ns.totalSamplesReceived;
>@@ -207,8 +214,10 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
>   stats.secondary_discarded_rate = Q14ToFloat(ns.currentSecondaryDiscardedRate);
>   stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
>   stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
>+  stats.jitter_buffer_flushes = ns.packetBufferFlushes;
>+  stats.delayed_packet_outage_samples = ns.delayedPacketOutageSamples;
> 
>-  auto ds = channel_proxy_->GetDecodingCallStatistics();
>+  auto ds = channel_receive_->GetDecodingCallStatistics();
>   stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;
>   stats.decoding_calls_to_neteq = ds.calls_to_neteq;
>   stats.decoding_normal = ds.decoded_normal;
>@@ -222,23 +231,23 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
> 
> void AudioReceiveStream::SetSink(AudioSinkInterface* sink) {
>   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>-  channel_proxy_->SetSink(sink);
>+  channel_receive_->SetSink(sink);
> }
> 
> void AudioReceiveStream::SetGain(float gain) {
>   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>-  channel_proxy_->SetChannelOutputVolumeScaling(gain);
>+  channel_receive_->SetChannelOutputVolumeScaling(gain);
> }
> 
> std::vector<RtpSource> AudioReceiveStream::GetSources() const {
>   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>-  return channel_proxy_->GetSources();
>+  return channel_receive_->GetSources();
> }
> 
> AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
>     int sample_rate_hz,
>     AudioFrame* audio_frame) {
>-  return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
>+  return channel_receive_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
> }
> 
> int AudioReceiveStream::Ssrc() const {
>@@ -246,7 +255,7 @@ int AudioReceiveStream::Ssrc() const {
> }
> 
> int AudioReceiveStream::PreferredSampleRate() const {
>-  return channel_proxy_->PreferredSampleRate();
>+  return channel_receive_->PreferredSampleRate();
> }
> 
> int AudioReceiveStream::id() const {
>@@ -256,32 +265,29 @@ int AudioReceiveStream::id() const {
> 
> absl::optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
>   RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
>-  absl::optional<Syncable::Info> info = channel_proxy_->GetSyncInfo();
>+  absl::optional<Syncable::Info> info = channel_receive_->GetSyncInfo();
> 
>   if (!info)
>     return absl::nullopt;
> 
>-  info->current_delay_ms = channel_proxy_->GetDelayEstimate();
>+  info->current_delay_ms = channel_receive_->GetDelayEstimate();
>   return info;
> }
> 
> uint32_t AudioReceiveStream::GetPlayoutTimestamp() const {
>   // Called on video capture thread.
>-  return channel_proxy_->GetPlayoutTimestamp();
>+  return channel_receive_->GetPlayoutTimestamp();
> }
> 
> void AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
>   RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
>-  return channel_proxy_->SetMinimumPlayoutDelay(delay_ms);
>+  return channel_receive_->SetMinimumPlayoutDelay(delay_ms);
> }
> 
> void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
>   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>-  if (send_stream) {
>-    channel_proxy_->AssociateSendChannel(send_stream->GetChannelProxy());
>-  } else {
>-    channel_proxy_->DisassociateSendChannel();
>-  }
>+  channel_receive_->SetAssociatedSendChannel(
>+      send_stream ? send_stream->GetChannel() : nullptr);
>   associated_send_stream_ = send_stream;
> }
> 
>@@ -294,7 +300,7 @@ bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
>   // calls on the worker thread. We should move towards always using a network
>   // thread. Then this check can be enabled.
>   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
>-  return channel_proxy_->ReceivedRTCPPacket(packet, length);
>+  return channel_receive_->ReceivedRTCPPacket(packet, length);
> }
> 
> void AudioReceiveStream::OnRtpPacket(const RtpPacketReceived& packet) {
>@@ -302,7 +308,7 @@ void AudioReceiveStream::OnRtpPacket(const RtpPacketReceived& packet) {
>   // calls on the worker thread. We should move towards always using a network
>   // thread. Then this check can be enabled.
>   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
>-  channel_proxy_->OnRtpPacket(packet);
>+  channel_receive_->OnRtpPacket(packet);
> }
> 
> const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
>@@ -328,7 +334,7 @@ void AudioReceiveStream::ConfigureStream(AudioReceiveStream* stream,
>   RTC_LOG(LS_INFO) << "AudioReceiveStream::ConfigureStream: "
>                    << new_config.ToString();
>   RTC_DCHECK(stream);
>-  const auto& channel_proxy = stream->channel_proxy_;
>+  const auto& channel_receive = stream->channel_receive_;
>   const auto& old_config = stream->config_;
> 
>   // Configuration parameters which cannot be changed.
>@@ -342,7 +348,7 @@ void AudioReceiveStream::ConfigureStream(AudioReceiveStream* stream,
>              old_config.decoder_factory == new_config.decoder_factory);
> 
>   if (first_time || old_config.rtp.local_ssrc != new_config.rtp.local_ssrc) {
>-    channel_proxy->SetLocalSSRC(new_config.rtp.local_ssrc);
>+    channel_receive->SetLocalSSRC(new_config.rtp.local_ssrc);
>   }
> 
>   if (!first_time) {
>@@ -354,11 +360,11 @@ void AudioReceiveStream::ConfigureStream(AudioReceiveStream* stream,
>   // using the actual packet size for the configured codec.
>   if (first_time || old_config.rtp.nack.rtp_history_ms !=
>                         new_config.rtp.nack.rtp_history_ms) {
>-    channel_proxy->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
>-                                 new_config.rtp.nack.rtp_history_ms / 20);
>+    channel_receive->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
>+                                   new_config.rtp.nack.rtp_history_ms / 20);
>   }
>   if (first_time || old_config.decoder_map != new_config.decoder_map) {
>-    channel_proxy->SetReceiveCodecs(new_config.decoder_map);
>+    channel_receive->SetReceiveCodecs(new_config.decoder_map);
>   }
> 
>   stream->config_ = new_config;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.h
>index e982b04d54b8e7f273f338959d820676a8a94eb6..86bcb1c6a2db425ef46b236e86b426b09e3d7337 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.h
>@@ -18,7 +18,6 @@
> #include "api/rtp_headers.h"
> #include "audio/audio_state.h"
> #include "call/audio_receive_stream.h"
>-#include "call/rtp_packet_sink_interface.h"
> #include "call/syncable.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/thread_checker.h"
>@@ -32,7 +31,7 @@ class RtpStreamReceiverControllerInterface;
> class RtpStreamReceiverInterface;
> 
> namespace voe {
>-class ChannelReceiveProxy;
>+class ChannelReceiveInterface;
> }  // namespace voe
> 
> namespace internal {
>@@ -48,13 +47,14 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
>                      const webrtc::AudioReceiveStream::Config& config,
>                      const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
>                      webrtc::RtcEventLog* event_log);
>-  // For unit tests, which need to supply a mock channel proxy.
>-  AudioReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
>-                     PacketRouter* packet_router,
>-                     const webrtc::AudioReceiveStream::Config& config,
>-                     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
>-                     webrtc::RtcEventLog* event_log,
>-                     std::unique_ptr<voe::ChannelReceiveProxy> channel_proxy);
>+  // For unit tests, which need to supply a mock channel receive.
>+  AudioReceiveStream(
>+      RtpStreamReceiverControllerInterface* receiver_controller,
>+      PacketRouter* packet_router,
>+      const webrtc::AudioReceiveStream::Config& config,
>+      const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
>+      webrtc::RtcEventLog* event_log,
>+      std::unique_ptr<voe::ChannelReceiveInterface> channel_receive);
>   ~AudioReceiveStream() override;
> 
>   // webrtc::AudioReceiveStream implementation.
>@@ -101,7 +101,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
>   rtc::ThreadChecker module_process_thread_checker_;
>   webrtc::AudioReceiveStream::Config config_;
>   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
>-  std::unique_ptr<voe::ChannelReceiveProxy> channel_proxy_;
>+  const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_;
>   AudioSendStream* associated_send_stream_ = nullptr;
> 
>   bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream_unittest.cc
>index 97c42c4383f8436342b477036f6dc57f05979433..742281085665152728efc2d1789a83a713654696 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream_unittest.cc
>@@ -13,6 +13,7 @@
> #include <vector>
> 
> #include "api/test/mock_audio_mixer.h"
>+#include "api/test/mock_frame_decryptor.h"
> #include "audio/audio_receive_stream.h"
> #include "audio/conversion.h"
> #include "audio/mock_voe_channel_proxy.h"
>@@ -82,16 +83,16 @@ struct ConfigHelper {
>         new rtc::RefCountedObject<testing::NiceMock<MockAudioDeviceModule>>();
>     audio_state_ = AudioState::Create(config);
> 
>-    channel_proxy_ = new testing::StrictMock<MockChannelReceiveProxy>();
>-    EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
>-    EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1);
>-    EXPECT_CALL(*channel_proxy_,
>+    channel_receive_ = new testing::StrictMock<MockChannelReceive>();
>+    EXPECT_CALL(*channel_receive_, SetLocalSSRC(kLocalSsrc)).Times(1);
>+    EXPECT_CALL(*channel_receive_, SetNACKStatus(true, 15)).Times(1);
>+    EXPECT_CALL(*channel_receive_,
>                 RegisterReceiverCongestionControlObjects(&packet_router_))
>         .Times(1);
>-    EXPECT_CALL(*channel_proxy_, ResetReceiverCongestionControlObjects())
>+    EXPECT_CALL(*channel_receive_, ResetReceiverCongestionControlObjects())
>         .Times(1);
>-    EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1);
>-    EXPECT_CALL(*channel_proxy_, SetReceiveCodecs(_))
>+    EXPECT_CALL(*channel_receive_, SetAssociatedSendChannel(nullptr)).Times(1);
>+    EXPECT_CALL(*channel_receive_, SetReceiveCodecs(_))
>         .WillRepeatedly(Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
>           EXPECT_THAT(codecs, testing::IsEmpty());
>         }));
>@@ -113,33 +114,33 @@ struct ConfigHelper {
>         new internal::AudioReceiveStream(
>             &rtp_stream_receiver_controller_, &packet_router_, stream_config_,
>             audio_state_, &event_log_,
>-            std::unique_ptr<voe::ChannelReceiveProxy>(channel_proxy_)));
>+            std::unique_ptr<voe::ChannelReceiveInterface>(channel_receive_)));
>   }
> 
>   AudioReceiveStream::Config& config() { return stream_config_; }
>   rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; }
>-  MockChannelReceiveProxy* channel_proxy() { return channel_proxy_; }
>+  MockChannelReceive* channel_receive() { return channel_receive_; }
> 
>   void SetupMockForGetStats() {
>     using testing::DoAll;
>     using testing::SetArgPointee;
> 
>-    ASSERT_TRUE(channel_proxy_);
>-    EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
>+    ASSERT_TRUE(channel_receive_);
>+    EXPECT_CALL(*channel_receive_, GetRTCPStatistics())
>         .WillOnce(Return(kCallStats));
>-    EXPECT_CALL(*channel_proxy_, GetDelayEstimate())
>+    EXPECT_CALL(*channel_receive_, GetDelayEstimate())
>         .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
>-    EXPECT_CALL(*channel_proxy_, GetSpeechOutputLevelFullRange())
>+    EXPECT_CALL(*channel_receive_, GetSpeechOutputLevelFullRange())
>         .WillOnce(Return(kSpeechOutputLevel));
>-    EXPECT_CALL(*channel_proxy_, GetTotalOutputEnergy())
>+    EXPECT_CALL(*channel_receive_, GetTotalOutputEnergy())
>         .WillOnce(Return(kTotalOutputEnergy));
>-    EXPECT_CALL(*channel_proxy_, GetTotalOutputDuration())
>+    EXPECT_CALL(*channel_receive_, GetTotalOutputDuration())
>         .WillOnce(Return(kTotalOutputDuration));
>-    EXPECT_CALL(*channel_proxy_, GetNetworkStatistics())
>+    EXPECT_CALL(*channel_receive_, GetNetworkStatistics())
>         .WillOnce(Return(kNetworkStats));
>-    EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics())
>+    EXPECT_CALL(*channel_receive_, GetDecodingCallStatistics())
>         .WillOnce(Return(kAudioDecodeStats));
>-    EXPECT_CALL(*channel_proxy_, GetRecCodec(_))
>+    EXPECT_CALL(*channel_receive_, GetRecCodec(_))
>         .WillOnce(DoAll(SetArgPointee<0>(kCodecInst), Return(true)));
>   }
> 
>@@ -149,7 +150,7 @@ struct ConfigHelper {
>   rtc::scoped_refptr<AudioState> audio_state_;
>   rtc::scoped_refptr<MockAudioMixer> audio_mixer_;
>   AudioReceiveStream::Config stream_config_;
>-  testing::StrictMock<MockChannelReceiveProxy>* channel_proxy_ = nullptr;
>+  testing::StrictMock<MockChannelReceive>* channel_receive_ = nullptr;
>   RtpStreamReceiverController rtp_stream_receiver_controller_;
>   MockTransport rtcp_send_transport_;
> };
>@@ -216,7 +217,7 @@ TEST(AudioReceiveStreamTest, ConfigToString) {
>       "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, transport_cc: off, nack: "
>       "{rtp_history_ms: 0}, extensions: [{uri: "
>       "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 3}]}, "
>-      "rtcp_send_transport: null}",
>+      "rtcp_send_transport: null, media_transport: null}",
>       config.ToString());
> }
> 
>@@ -238,7 +239,7 @@ TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
>   ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size()));
>   parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
> 
>-  EXPECT_CALL(*helper.channel_proxy(),
>+  EXPECT_CALL(*helper.channel_receive(),
>               OnRtpPacket(testing::Ref(parsed_packet)));
> 
>   recv_stream->OnRtpPacket(parsed_packet);
>@@ -249,7 +250,7 @@ TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
>   helper.config().rtp.transport_cc = true;
>   auto recv_stream = helper.CreateAudioReceiveStream();
>   std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
>-  EXPECT_CALL(*helper.channel_proxy(),
>+  EXPECT_CALL(*helper.channel_receive(),
>               ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size()))
>       .WillOnce(Return(true));
>   EXPECT_TRUE(recv_stream->DeliverRtcp(&rtcp_packet[0], rtcp_packet.size()));
>@@ -311,7 +312,7 @@ TEST(AudioReceiveStreamTest, GetStats) {
> TEST(AudioReceiveStreamTest, SetGain) {
>   ConfigHelper helper;
>   auto recv_stream = helper.CreateAudioReceiveStream();
>-  EXPECT_CALL(*helper.channel_proxy(),
>+  EXPECT_CALL(*helper.channel_receive(),
>               SetChannelOutputVolumeScaling(FloatEq(0.765f)));
>   recv_stream->SetGain(0.765f);
> }
>@@ -322,10 +323,10 @@ TEST(AudioReceiveStreamTest, StreamsShouldBeAddedToMixerOnceOnStart) {
>   auto recv_stream1 = helper1.CreateAudioReceiveStream();
>   auto recv_stream2 = helper2.CreateAudioReceiveStream();
> 
>-  EXPECT_CALL(*helper1.channel_proxy(), StartPlayout()).Times(1);
>-  EXPECT_CALL(*helper2.channel_proxy(), StartPlayout()).Times(1);
>-  EXPECT_CALL(*helper1.channel_proxy(), StopPlayout()).Times(1);
>-  EXPECT_CALL(*helper2.channel_proxy(), StopPlayout()).Times(1);
>+  EXPECT_CALL(*helper1.channel_receive(), StartPlayout()).Times(1);
>+  EXPECT_CALL(*helper2.channel_receive(), StartPlayout()).Times(1);
>+  EXPECT_CALL(*helper1.channel_receive(), StopPlayout()).Times(1);
>+  EXPECT_CALL(*helper2.channel_receive(), StopPlayout()).Times(1);
>   EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream1.get()))
>       .WillOnce(Return(true));
>   EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream2.get()))
>@@ -366,12 +367,32 @@ TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) {
>                    kTransportSequenceNumberId + 1));
>   new_config.decoder_map.emplace(1, SdpAudioFormat("foo", 8000, 1));
> 
>-  MockChannelReceiveProxy& channel_proxy = *helper.channel_proxy();
>-  EXPECT_CALL(channel_proxy, SetLocalSSRC(kLocalSsrc + 1)).Times(1);
>-  EXPECT_CALL(channel_proxy, SetNACKStatus(true, 15 + 1)).Times(1);
>-  EXPECT_CALL(channel_proxy, SetReceiveCodecs(new_config.decoder_map));
>+  MockChannelReceive& channel_receive = *helper.channel_receive();
>+  EXPECT_CALL(channel_receive, SetLocalSSRC(kLocalSsrc + 1)).Times(1);
>+  EXPECT_CALL(channel_receive, SetNACKStatus(true, 15 + 1)).Times(1);
>+  EXPECT_CALL(channel_receive, SetReceiveCodecs(new_config.decoder_map));
> 
>   recv_stream->Reconfigure(new_config);
> }
>+
>+TEST(AudioReceiveStreamTest, ReconfigureWithFrameDecryptor) {
>+  ConfigHelper helper;
>+  auto recv_stream = helper.CreateAudioReceiveStream();
>+
>+  auto new_config_0 = helper.config();
>+  rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_0(
>+      new rtc::RefCountedObject<MockFrameDecryptor>());
>+  new_config_0.frame_decryptor = mock_frame_decryptor_0;
>+
>+  recv_stream->Reconfigure(new_config_0);
>+
>+  auto new_config_1 = helper.config();
>+  rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_1(
>+      new rtc::RefCountedObject<MockFrameDecryptor>());
>+  new_config_1.frame_decryptor = mock_frame_decryptor_1;
>+  new_config_1.crypto_options.sframe.require_frame_encryption = true;
>+  recv_stream->Reconfigure(new_config_1);
>+}
>+
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.cc
>index e82264a83cb1cb285e83e83adc40955c605e6cb5..75e6efb9e800fd0918ccbb5be9f1af7da8891b60 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.cc
>@@ -15,12 +15,22 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>-
>+#include "api/audio_codecs/audio_encoder.h"
>+#include "api/audio_codecs/audio_encoder_factory.h"
>+#include "api/audio_codecs/audio_format.h"
>+#include "api/call/transport.h"
>+#include "api/crypto/frameencryptorinterface.h"
> #include "audio/audio_state.h"
>-#include "audio/channel_send_proxy.h"
>+#include "audio/channel_send.h"
> #include "audio/conversion.h"
>+#include "call/rtp_config.h"
> #include "call/rtp_transport_controller_send_interface.h"
>+#include "common_audio/vad/include/vad.h"
>+#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "logging/rtc_event_log/rtc_stream_config.h"
> #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
>+#include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/event.h"
> #include "rtc_base/function_view.h"
>@@ -38,130 +48,132 @@ constexpr size_t kPacketLossTrackerMaxWindowSizeMs = 15000;
> constexpr size_t kPacketLossRateMinNumAckedPackets = 50;
> constexpr size_t kRecoverablePacketLossRateMinNumAckedPairs = 40;
> 
>-void CallEncoder(const std::unique_ptr<voe::ChannelSendProxy>& channel_proxy,
>+void CallEncoder(const std::unique_ptr<voe::ChannelSendInterface>& channel_send,
>                  rtc::FunctionView<void(AudioEncoder*)> lambda) {
>-  channel_proxy->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
>+  channel_send->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
>     RTC_DCHECK(encoder_ptr);
>     lambda(encoder_ptr->get());
>   });
> }
> 
>-std::unique_ptr<voe::ChannelSendProxy> CreateChannelAndProxy(
>-    rtc::TaskQueue* worker_queue,
>-    ProcessThread* module_process_thread,
>-    RtcpRttStats* rtcp_rtt_stats,
>-    RtcEventLog* event_log,
>-    FrameEncryptorInterface* frame_encryptor) {
>-  return absl::make_unique<voe::ChannelSendProxy>(
>-      absl::make_unique<voe::ChannelSend>(worker_queue, module_process_thread,
>-                                          rtcp_rtt_stats, event_log,
>-                                          frame_encryptor));
>-}
>-}  // namespace
>-
>-// Helper class to track the actively sending lifetime of this stream.
>-class AudioSendStream::TimedTransport : public Transport {
>- public:
>-  TimedTransport(Transport* transport, TimeInterval* time_interval)
>-      : transport_(transport), lifetime_(time_interval) {}
>-  bool SendRtp(const uint8_t* packet,
>-               size_t length,
>-               const PacketOptions& options) {
>-    if (lifetime_) {
>-      lifetime_->Extend();
>+void UpdateEventLogStreamConfig(RtcEventLog* event_log,
>+                                const AudioSendStream::Config& config,
>+                                const AudioSendStream::Config* old_config) {
>+  using SendCodecSpec = AudioSendStream::Config::SendCodecSpec;
>+  // Only update if any of the things we log have changed.
>+  auto payload_types_equal = [](const absl::optional<SendCodecSpec>& a,
>+                                const absl::optional<SendCodecSpec>& b) {
>+    if (a.has_value() && b.has_value()) {
>+      return a->format.name == b->format.name &&
>+             a->payload_type == b->payload_type;
>     }
>-    return transport_->SendRtp(packet, length, options);
>+    return !a.has_value() && !b.has_value();
>+  };
>+
>+  if (old_config && config.rtp.ssrc == old_config->rtp.ssrc &&
>+      config.rtp.extensions == old_config->rtp.extensions &&
>+      payload_types_equal(config.send_codec_spec,
>+                          old_config->send_codec_spec)) {
>+    return;
>   }
>-  bool SendRtcp(const uint8_t* packet, size_t length) {
>-    return transport_->SendRtcp(packet, length);
>+
>+  auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
>+  rtclog_config->local_ssrc = config.rtp.ssrc;
>+  rtclog_config->rtp_extensions = config.rtp.extensions;
>+  if (config.send_codec_spec) {
>+    rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name,
>+                                       config.send_codec_spec->payload_type, 0);
>   }
>-  ~TimedTransport() {}
>+  event_log->Log(absl::make_unique<RtcEventAudioSendStreamConfig>(
>+      std::move(rtclog_config)));
>+}
> 
>- private:
>-  Transport* transport_;
>-  TimeInterval* lifetime_;
>-};
>+}  // namespace
> 
> AudioSendStream::AudioSendStream(
>     const webrtc::AudioSendStream::Config& config,
>     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
>     rtc::TaskQueue* worker_queue,
>     ProcessThread* module_process_thread,
>-    RtpTransportControllerSendInterface* transport,
>-    BitrateAllocator* bitrate_allocator,
>+    RtpTransportControllerSendInterface* rtp_transport,
>+    BitrateAllocatorInterface* bitrate_allocator,
>     RtcEventLog* event_log,
>     RtcpRttStats* rtcp_rtt_stats,
>-    const absl::optional<RtpState>& suspended_rtp_state,
>-    TimeInterval* overall_call_lifetime)
>+    const absl::optional<RtpState>& suspended_rtp_state)
>     : AudioSendStream(config,
>                       audio_state,
>                       worker_queue,
>-                      transport,
>+                      rtp_transport,
>                       bitrate_allocator,
>                       event_log,
>                       rtcp_rtt_stats,
>                       suspended_rtp_state,
>-                      overall_call_lifetime,
>-                      CreateChannelAndProxy(worker_queue,
>-                                            module_process_thread,
>-                                            rtcp_rtt_stats,
>-                                            event_log,
>-                                            config.frame_encryptor)) {}
>+                      voe::CreateChannelSend(worker_queue,
>+                                             module_process_thread,
>+                                             config.media_transport,
>+                                             config.send_transport,
>+                                             rtcp_rtt_stats,
>+                                             event_log,
>+                                             config.frame_encryptor,
>+                                             config.crypto_options,
>+                                             config.rtp.extmap_allow_mixed,
>+                                             config.rtcp_report_interval_ms)) {}
> 
> AudioSendStream::AudioSendStream(
>     const webrtc::AudioSendStream::Config& config,
>     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
>     rtc::TaskQueue* worker_queue,
>-    RtpTransportControllerSendInterface* transport,
>-    BitrateAllocator* bitrate_allocator,
>+    RtpTransportControllerSendInterface* rtp_transport,
>+    BitrateAllocatorInterface* bitrate_allocator,
>     RtcEventLog* event_log,
>     RtcpRttStats* rtcp_rtt_stats,
>     const absl::optional<RtpState>& suspended_rtp_state,
>-    TimeInterval* overall_call_lifetime,
>-    std::unique_ptr<voe::ChannelSendProxy> channel_proxy)
>+    std::unique_ptr<voe::ChannelSendInterface> channel_send)
>     : worker_queue_(worker_queue),
>-      config_(Config(nullptr)),
>+      config_(Config(/*send_transport=*/nullptr,
>+                     /*media_transport=*/nullptr)),
>       audio_state_(audio_state),
>-      channel_proxy_(std::move(channel_proxy)),
>+      channel_send_(std::move(channel_send)),
>       event_log_(event_log),
>       bitrate_allocator_(bitrate_allocator),
>-      transport_(transport),
>+      rtp_transport_(rtp_transport),
>       packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs,
>                            kPacketLossRateMinNumAckedPackets,
>                            kRecoverablePacketLossRateMinNumAckedPairs),
>       rtp_rtcp_module_(nullptr),
>-      suspended_rtp_state_(suspended_rtp_state),
>-      overall_call_lifetime_(overall_call_lifetime) {
>+      suspended_rtp_state_(suspended_rtp_state) {
>   RTC_LOG(LS_INFO) << "AudioSendStream: " << config.rtp.ssrc;
>   RTC_DCHECK(worker_queue_);
>   RTC_DCHECK(audio_state_);
>-  RTC_DCHECK(channel_proxy_);
>+  RTC_DCHECK(channel_send_);
>   RTC_DCHECK(bitrate_allocator_);
>-  RTC_DCHECK(transport);
>-  RTC_DCHECK(overall_call_lifetime_);
>+  // TODO(nisse): Eventually, we should have only media_transport. But for the
>+  // time being, we can have either. When media transport is injected, there
>+  // should be no rtp_transport, and below check should be strengthened to XOR
>+  // (either rtp_transport or media_transport but not both).
>+  RTC_DCHECK(rtp_transport || config.media_transport);
> 
>-  channel_proxy_->SetRTCPStatus(true);
>-  rtp_rtcp_module_ = channel_proxy_->GetRtpRtcp();
>+  rtp_rtcp_module_ = channel_send_->GetRtpRtcp();
>   RTC_DCHECK(rtp_rtcp_module_);
> 
>   ConfigureStream(this, config, true);
> 
>   pacer_thread_checker_.DetachFromThread();
>-  // Signal congestion controller this object is ready for OnPacket* callbacks.
>-  transport_->RegisterPacketFeedbackObserver(this);
>+  if (rtp_transport_) {
>+    // Signal congestion controller this object is ready for OnPacket*
>+    // callbacks.
>+    rtp_transport_->RegisterPacketFeedbackObserver(this);
>+  }
> }
> 
> AudioSendStream::~AudioSendStream() {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.rtp.ssrc;
>   RTC_DCHECK(!sending_);
>-  transport_->DeRegisterPacketFeedbackObserver(this);
>-  channel_proxy_->RegisterTransport(nullptr);
>-  channel_proxy_->ResetSenderCongestionControlObjects();
>-  // Lifetime can only be updated after deregistering
>-  // |timed_send_transport_adapter_| in the underlying channel object to avoid
>-  // data races in |active_lifetime_|.
>-  overall_call_lifetime_->Extend(active_lifetime_);
>+  if (rtp_transport_) {
>+    rtp_transport_->DeRegisterPacketFeedbackObserver(this);
>+    channel_send_->ResetSenderCongestionControlObjects();
>+  }
> }
> 
> const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const {
>@@ -196,51 +208,42 @@ void AudioSendStream::ConfigureStream(
>     bool first_time) {
>   RTC_LOG(LS_INFO) << "AudioSendStream::ConfigureStream: "
>                    << new_config.ToString();
>-  const auto& channel_proxy = stream->channel_proxy_;
>+  UpdateEventLogStreamConfig(stream->event_log_, new_config,
>+                             first_time ? nullptr : &stream->config_);
>+
>+  const auto& channel_send = stream->channel_send_;
>   const auto& old_config = stream->config_;
> 
>+  // Configuration parameters which cannot be changed.
>+  RTC_DCHECK(first_time ||
>+             old_config.send_transport == new_config.send_transport);
>+
>   if (first_time || old_config.rtp.ssrc != new_config.rtp.ssrc) {
>-    channel_proxy->SetLocalSSRC(new_config.rtp.ssrc);
>+    channel_send->SetLocalSSRC(new_config.rtp.ssrc);
>     if (stream->suspended_rtp_state_) {
>       stream->rtp_rtcp_module_->SetRtpState(*stream->suspended_rtp_state_);
>     }
>   }
>   if (first_time || old_config.rtp.c_name != new_config.rtp.c_name) {
>-    channel_proxy->SetRTCP_CNAME(new_config.rtp.c_name);
>-  }
>-  // TODO(solenberg): Config NACK history window (which is a packet count),
>-  // using the actual packet size for the configured codec.
>-  if (first_time || old_config.rtp.nack.rtp_history_ms !=
>-                        new_config.rtp.nack.rtp_history_ms) {
>-    channel_proxy->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
>-                                 new_config.rtp.nack.rtp_history_ms / 20);
>-  }
>-
>-  if (first_time || new_config.send_transport != old_config.send_transport) {
>-    if (old_config.send_transport) {
>-      channel_proxy->RegisterTransport(nullptr);
>-    }
>-    if (new_config.send_transport) {
>-      stream->timed_send_transport_adapter_.reset(new TimedTransport(
>-          new_config.send_transport, &stream->active_lifetime_));
>-    } else {
>-      stream->timed_send_transport_adapter_.reset(nullptr);
>-    }
>-    channel_proxy->RegisterTransport(
>-        stream->timed_send_transport_adapter_.get());
>+    channel_send->SetRTCP_CNAME(new_config.rtp.c_name);
>   }
> 
>   // Enable the frame encryptor if a new frame encryptor has been provided.
>   if (first_time || new_config.frame_encryptor != old_config.frame_encryptor) {
>-    channel_proxy->SetFrameEncryptor(new_config.frame_encryptor);
>+    channel_send->SetFrameEncryptor(new_config.frame_encryptor);
>+  }
>+
>+  if (first_time ||
>+      new_config.rtp.extmap_allow_mixed != old_config.rtp.extmap_allow_mixed) {
>+    channel_send->SetExtmapAllowMixed(new_config.rtp.extmap_allow_mixed);
>   }
> 
>   const ExtensionIds old_ids = FindExtensionIds(old_config.rtp.extensions);
>   const ExtensionIds new_ids = FindExtensionIds(new_config.rtp.extensions);
>   // Audio level indication
>   if (first_time || new_ids.audio_level != old_ids.audio_level) {
>-    channel_proxy->SetSendAudioLevelIndicationStatus(new_ids.audio_level != 0,
>-                                                     new_ids.audio_level);
>+    channel_send->SetSendAudioLevelIndicationStatus(new_ids.audio_level != 0,
>+                                                    new_ids.audio_level);
>   }
>   bool transport_seq_num_id_changed =
>       new_ids.transport_sequence_number != old_ids.transport_sequence_number;
>@@ -248,7 +251,7 @@ void AudioSendStream::ConfigureStream(
>       (transport_seq_num_id_changed &&
>        !webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC"))) {
>     if (!first_time) {
>-      channel_proxy->ResetSenderCongestionControlObjects();
>+      channel_send->ResetSenderCongestionControlObjects();
>     }
> 
>     RtcpBandwidthObserver* bandwidth_observer = nullptr;
>@@ -256,24 +259,26 @@ void AudioSendStream::ConfigureStream(
>         new_ids.transport_sequence_number != 0 &&
>         !webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC");
>     if (has_transport_sequence_number) {
>-      channel_proxy->EnableSendTransportSequenceNumber(
>+      channel_send->EnableSendTransportSequenceNumber(
>           new_ids.transport_sequence_number);
>       // Probing in application limited region is only used in combination with
>       // send side congestion control, wich depends on feedback packets which
>       // requires transport sequence numbers to be enabled.
>-      stream->transport_->EnablePeriodicAlrProbing(true);
>-      bandwidth_observer = stream->transport_->GetBandwidthObserver();
>+      if (stream->rtp_transport_) {
>+        stream->rtp_transport_->EnablePeriodicAlrProbing(true);
>+        bandwidth_observer = stream->rtp_transport_->GetBandwidthObserver();
>+      }
>+    }
>+    if (stream->rtp_transport_) {
>+      channel_send->RegisterSenderCongestionControlObjects(
>+          stream->rtp_transport_, bandwidth_observer);
>     }
>-
>-    channel_proxy->RegisterSenderCongestionControlObjects(stream->transport_,
>-                                                          bandwidth_observer);
>   }
>-
>   // MID RTP header extension.
>   if ((first_time || new_ids.mid != old_ids.mid ||
>        new_config.rtp.mid != old_config.rtp.mid) &&
>       new_ids.mid != 0 && !new_config.rtp.mid.empty()) {
>-    channel_proxy->SetMid(new_config.rtp.mid, new_ids.mid);
>+    channel_send->SetMid(new_config.rtp.mid, new_ids.mid);
>   }
> 
>   if (!ReconfigureSendCodec(stream, new_config)) {
>@@ -296,16 +301,20 @@ void AudioSendStream::Start() {
>       FindExtensionIds(config_.rtp.extensions).transport_sequence_number != 0 &&
>       !webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC");
>   if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1 &&
>+      !config_.has_dscp &&
>       (has_transport_sequence_number ||
>        !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe") ||
>        webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC"))) {
>     // Audio BWE is enabled.
>-    transport_->packet_sender()->SetAccountForAudioPackets(true);
>+    rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
>+    rtp_rtcp_module_->SetAsPartOfAllocation(true);
>     ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps,
>                              config_.bitrate_priority,
>                              has_transport_sequence_number);
>+  } else {
>+    rtp_rtcp_module_->SetAsPartOfAllocation(false);
>   }
>-  channel_proxy_->StartSend();
>+  channel_send_->StartSend();
>   sending_ = true;
>   audio_state()->AddSendingStream(this, encoder_sample_rate_hz_,
>                                   encoder_num_channels_);
>@@ -318,14 +327,14 @@ void AudioSendStream::Stop() {
>   }
> 
>   RemoveBitrateObserver();
>-  channel_proxy_->StopSend();
>+  channel_send_->StopSend();
>   sending_ = false;
>   audio_state()->RemoveSendingStream(this);
> }
> 
> void AudioSendStream::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
>   RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
>-  channel_proxy_->ProcessAndEncodeAudio(std::move(audio_frame));
>+  channel_send_->ProcessAndEncodeAudio(std::move(audio_frame));
> }
> 
> bool AudioSendStream::SendTelephoneEvent(int payload_type,
>@@ -333,14 +342,14 @@ bool AudioSendStream::SendTelephoneEvent(int payload_type,
>                                          int event,
>                                          int duration_ms) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type,
>-                                                          payload_frequency) &&
>-         channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
>+  return channel_send_->SetSendTelephoneEventPayloadType(payload_type,
>+                                                         payload_frequency) &&
>+         channel_send_->SendTelephoneEventOutband(event, duration_ms);
> }
> 
> void AudioSendStream::SetMuted(bool muted) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_proxy_->SetInputMute(muted);
>+  channel_send_->SetInputMute(muted);
> }
> 
> webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
>@@ -352,8 +361,9 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   webrtc::AudioSendStream::Stats stats;
>   stats.local_ssrc = config_.rtp.ssrc;
>+  stats.target_bitrate_bps = channel_send_->GetBitrate();
> 
>-  webrtc::CallSendStatistics call_stats = channel_proxy_->GetRTCPStatistics();
>+  webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
>   stats.bytes_sent = call_stats.bytesSent;
>   stats.packets_sent = call_stats.packetsSent;
>   // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
>@@ -367,7 +377,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
>     stats.codec_payload_type = spec.payload_type;
> 
>     // Get data from the last remote RTCP report.
>-    for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) {
>+    for (const auto& block : channel_send_->GetRemoteRTCPReportBlocks()) {
>       // Lookup report for send ssrc only.
>       if (block.source_SSRC == stats.local_ssrc) {
>         stats.packets_lost = block.cumulative_num_packets_lost;
>@@ -389,7 +399,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
>   stats.total_input_duration = input_stats.total_duration;
> 
>   stats.typing_noise_detected = audio_state()->typing_noise_detected();
>-  stats.ana_statistics = channel_proxy_->GetANAStatistics();
>+  stats.ana_statistics = channel_send_->GetANAStatistics();
>   RTC_DCHECK(audio_state_->audio_processing());
>   stats.apm_statistics =
>       audio_state_->audio_processing()->GetStatistics(has_remote_tracks);
>@@ -406,27 +416,24 @@ bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
>   // calls on the worker thread. We should move towards always using a network
>   // thread. Then this check can be enabled.
>   // RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
>-  return channel_proxy_->ReceivedRTCPPacket(packet, length);
>+  return channel_send_->ReceivedRTCPPacket(packet, length);
> }
> 
>-uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
>-                                           uint8_t fraction_loss,
>-                                           int64_t rtt,
>-                                           int64_t bwe_period_ms) {
>+uint32_t AudioSendStream::OnBitrateUpdated(BitrateAllocationUpdate update) {
>   // A send stream may be allocated a bitrate of zero if the allocator decides
>   // to disable it. For now we ignore this decision and keep sending on min
>   // bitrate.
>-  if (bitrate_bps == 0) {
>-    bitrate_bps = config_.min_bitrate_bps;
>+  if (update.target_bitrate.IsZero()) {
>+    update.target_bitrate = DataRate::bps(config_.min_bitrate_bps);
>   }
>-  RTC_DCHECK_GE(bitrate_bps, static_cast<uint32_t>(config_.min_bitrate_bps));
>+  RTC_DCHECK_GE(update.target_bitrate.bps<int>(), config_.min_bitrate_bps);
>   // The bitrate allocator might allocate an higher than max configured bitrate
>   // if there is room, to allow for, as example, extra FEC. Ignore that for now.
>-  const uint32_t max_bitrate_bps = config_.max_bitrate_bps;
>-  if (bitrate_bps > max_bitrate_bps)
>-    bitrate_bps = max_bitrate_bps;
>+  const DataRate max_bitrate = DataRate::bps(config_.max_bitrate_bps);
>+  if (update.target_bitrate > max_bitrate)
>+    update.target_bitrate = max_bitrate;
> 
>-  channel_proxy_->SetBitrate(bitrate_bps, bwe_period_ms);
>+  channel_send_->OnBitrateAllocation(update);
> 
>   // The amount of audio protection is not exposed by the encoder, hence
>   // always returning 0.
>@@ -460,25 +467,24 @@ void AudioSendStream::OnPacketFeedbackVector(
>   // the previously sent value is no longer relevant. This will be taken care
>   // of with some refactoring which is now being done.
>   if (plr) {
>-    channel_proxy_->OnTwccBasedUplinkPacketLossRate(*plr);
>+    channel_send_->OnTwccBasedUplinkPacketLossRate(*plr);
>   }
>   if (rplr) {
>-    channel_proxy_->OnRecoverableUplinkPacketLossRate(*rplr);
>+    channel_send_->OnRecoverableUplinkPacketLossRate(*rplr);
>   }
> }
> 
> void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_proxy_->SetTransportOverhead(transport_overhead_per_packet);
>+  channel_send_->SetTransportOverhead(transport_overhead_per_packet);
> }
> 
> RtpState AudioSendStream::GetRtpState() const {
>   return rtp_rtcp_module_->GetRtpState();
> }
> 
>-const voe::ChannelSendProxy& AudioSendStream::GetChannelProxy() const {
>-  RTC_DCHECK(channel_proxy_.get());
>-  return *channel_proxy_.get();
>+const voe::ChannelSendInterface* AudioSendStream::GetChannel() const {
>+  return channel_send_.get();
> }
> 
> internal::AudioState* AudioSendStream::audio_state() {
>@@ -549,12 +555,12 @@ bool AudioSendStream::SetupSendCodec(AudioSendStream* stream,
> 
>   // Wrap the encoder in a an AudioEncoderCNG, if VAD is enabled.
>   if (spec.cng_payload_type) {
>-    AudioEncoderCng::Config cng_config;
>+    AudioEncoderCngConfig cng_config;
>     cng_config.num_channels = encoder->NumChannels();
>     cng_config.payload_type = *spec.cng_payload_type;
>     cng_config.speech_encoder = std::move(encoder);
>     cng_config.vad_mode = Vad::kVadNormal;
>-    encoder.reset(new AudioEncoderCng(std::move(cng_config)));
>+    encoder = CreateComfortNoiseEncoder(std::move(cng_config));
> 
>     stream->RegisterCngPayloadType(
>         *spec.cng_payload_type,
>@@ -563,8 +569,8 @@ bool AudioSendStream::SetupSendCodec(AudioSendStream* stream,
> 
>   stream->StoreEncoderProperties(encoder->SampleRateHz(),
>                                  encoder->NumChannels());
>-  stream->channel_proxy_->SetEncoder(new_config.send_codec_spec->payload_type,
>-                                     std::move(encoder));
>+  stream->channel_send_->SetEncoder(new_config.send_codec_spec->payload_type,
>+                                    std::move(encoder));
>   return true;
> }
> 
>@@ -610,7 +616,7 @@ bool AudioSendStream::ReconfigureSendCodec(AudioSendStream* stream,
>   if (!do_not_update_target_bitrate && new_target_bitrate_bps &&
>       new_target_bitrate_bps !=
>           old_config.send_codec_spec->target_bitrate_bps) {
>-    CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
>+    CallEncoder(stream->channel_send_, [&](AudioEncoder* encoder) {
>       encoder->OnReceivedTargetAudioBitrate(*new_target_bitrate_bps);
>     });
>   }
>@@ -628,7 +634,7 @@ void AudioSendStream::ReconfigureANA(AudioSendStream* stream,
>     return;
>   }
>   if (new_config.audio_network_adaptor_config) {
>-    CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
>+    CallEncoder(stream->channel_send_, [&](AudioEncoder* encoder) {
>       if (encoder->EnableAudioNetworkAdaptor(
>               *new_config.audio_network_adaptor_config, stream->event_log_)) {
>         RTC_DLOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
>@@ -638,7 +644,7 @@ void AudioSendStream::ReconfigureANA(AudioSendStream* stream,
>       }
>     });
>   } else {
>-    CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
>+    CallEncoder(stream->channel_send_, [&](AudioEncoder* encoder) {
>       encoder->DisableAudioNetworkAdaptor();
>     });
>     RTC_DLOG(LS_INFO) << "Audio network adaptor disabled on SSRC "
>@@ -662,7 +668,7 @@ void AudioSendStream::ReconfigureCNG(AudioSendStream* stream,
>   }
> 
>   // Wrap or unwrap the encoder in an AudioEncoderCNG.
>-  stream->channel_proxy_->ModifyEncoder(
>+  stream->channel_send_->ModifyEncoder(
>       [&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
>         std::unique_ptr<AudioEncoder> old_encoder(std::move(*encoder_ptr));
>         auto sub_encoders = old_encoder->ReclaimContainedEncoders();
>@@ -676,12 +682,12 @@ void AudioSendStream::ReconfigureCNG(AudioSendStream* stream,
>           old_encoder = std::move(tmp);
>         }
>         if (new_config.send_codec_spec->cng_payload_type) {
>-          AudioEncoderCng::Config config;
>+          AudioEncoderCngConfig config;
>           config.speech_encoder = std::move(old_encoder);
>           config.num_channels = config.speech_encoder->NumChannels();
>           config.payload_type = *new_config.send_codec_spec->cng_payload_type;
>           config.vad_mode = Vad::kVadNormal;
>-          encoder_ptr->reset(new AudioEncoderCng(std::move(config)));
>+          *encoder_ptr = CreateComfortNoiseEncoder(std::move(config));
>         } else {
>           *encoder_ptr = std::move(old_encoder);
>         }
>@@ -708,13 +714,18 @@ void AudioSendStream::ReconfigureBitrateObserver(
> 
>   bool has_transport_sequence_number = new_transport_seq_num_id != 0;
>   if (new_config.min_bitrate_bps != -1 && new_config.max_bitrate_bps != -1 &&
>+      !new_config.has_dscp &&
>       (has_transport_sequence_number ||
>        !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
>+    stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(true);
>     stream->ConfigureBitrateObserver(
>         new_config.min_bitrate_bps, new_config.max_bitrate_bps,
>         new_config.bitrate_priority, has_transport_sequence_number);
>+    stream->rtp_rtcp_module_->SetAsPartOfAllocation(true);
>   } else {
>+    stream->rtp_transport_->packet_sender()->SetAccountForAudioPackets(false);
>     stream->RemoveBitrateObserver();
>+    stream->rtp_rtcp_module_->SetAsPartOfAllocation(false);
>   }
> }
> 
>@@ -724,7 +735,7 @@ void AudioSendStream::ConfigureBitrateObserver(int min_bitrate_bps,
>                                                bool has_packet_feedback) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   RTC_DCHECK_GE(max_bitrate_bps, min_bitrate_bps);
>-  rtc::Event thread_sync_event(false /* manual_reset */, false);
>+  rtc::Event thread_sync_event;
>   worker_queue_->PostTask([&] {
>     // We may get a callback immediately as the observer is registered, so make
>     // sure the bitrate limits in config_ are up-to-date.
>@@ -744,7 +755,7 @@ void AudioSendStream::ConfigureBitrateObserver(int min_bitrate_bps,
> 
> void AudioSendStream::RemoveBitrateObserver() {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  rtc::Event thread_sync_event(false /* manual_reset */, false);
>+  rtc::Event thread_sync_event;
>   worker_queue_->PostTask([this, &thread_sync_event] {
>     bitrate_allocator_->RemoveObserver(this);
>     thread_sync_event.Set();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.h
>index 1ea676b65db745221861aa51d4309498c6aac91a..bf94901ef05750f3068dee903a6d0f5cbef6df5c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.h
>@@ -14,7 +14,7 @@
> #include <memory>
> #include <vector>
> 
>-#include "audio/time_interval.h"
>+#include "audio/channel_send.h"
> #include "audio/transport_feedback_packet_loss_tracker.h"
> #include "call/audio_send_stream.h"
> #include "call/audio_state.h"
>@@ -30,10 +30,6 @@ class RtcpBandwidthObserver;
> class RtcpRttStats;
> class RtpTransportControllerSendInterface;
> 
>-namespace voe {
>-class ChannelSendProxy;
>-}  // namespace voe
>-
> namespace internal {
> class AudioState;
> 
>@@ -45,23 +41,21 @@ class AudioSendStream final : public webrtc::AudioSendStream,
>                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
>                   rtc::TaskQueue* worker_queue,
>                   ProcessThread* module_process_thread,
>-                  RtpTransportControllerSendInterface* transport,
>-                  BitrateAllocator* bitrate_allocator,
>+                  RtpTransportControllerSendInterface* rtp_transport,
>+                  BitrateAllocatorInterface* bitrate_allocator,
>                   RtcEventLog* event_log,
>                   RtcpRttStats* rtcp_rtt_stats,
>-                  const absl::optional<RtpState>& suspended_rtp_state,
>-                  TimeInterval* overall_call_lifetime);
>-  // For unit tests, which need to supply a mock channel proxy.
>+                  const absl::optional<RtpState>& suspended_rtp_state);
>+  // For unit tests, which need to supply a mock ChannelSend.
>   AudioSendStream(const webrtc::AudioSendStream::Config& config,
>                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
>                   rtc::TaskQueue* worker_queue,
>-                  RtpTransportControllerSendInterface* transport,
>-                  BitrateAllocator* bitrate_allocator,
>+                  RtpTransportControllerSendInterface* rtp_transport,
>+                  BitrateAllocatorInterface* bitrate_allocator,
>                   RtcEventLog* event_log,
>                   RtcpRttStats* rtcp_rtt_stats,
>                   const absl::optional<RtpState>& suspended_rtp_state,
>-                  TimeInterval* overall_call_lifetime,
>-                  std::unique_ptr<voe::ChannelSendProxy> channel_proxy);
>+                  std::unique_ptr<voe::ChannelSendInterface> channel_send);
>   ~AudioSendStream() override;
> 
>   // webrtc::AudioSendStream implementation.
>@@ -83,10 +77,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
>   bool DeliverRtcp(const uint8_t* packet, size_t length);
> 
>   // Implements BitrateAllocatorObserver.
>-  uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
>-                            uint8_t fraction_loss,
>-                            int64_t rtt,
>-                            int64_t bwe_period_ms) override;
>+  uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override;
> 
>   // From PacketFeedbackObserver.
>   void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;
>@@ -96,7 +87,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
>   void SetTransportOverhead(int transport_overhead_per_packet);
> 
>   RtpState GetRtpState() const;
>-  const voe::ChannelSendProxy& GetChannelProxy() const;
>+  const voe::ChannelSendInterface* GetChannel() const;
> 
>  private:
>   class TimedTransport;
>@@ -133,15 +124,15 @@ class AudioSendStream final : public webrtc::AudioSendStream,
>   rtc::TaskQueue* worker_queue_;
>   webrtc::AudioSendStream::Config config_;
>   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
>-  std::unique_ptr<voe::ChannelSendProxy> channel_proxy_;
>+  const std::unique_ptr<voe::ChannelSendInterface> channel_send_;
>   RtcEventLog* const event_log_;
> 
>   int encoder_sample_rate_hz_ = 0;
>   size_t encoder_num_channels_ = 0;
>   bool sending_ = false;
> 
>-  BitrateAllocator* const bitrate_allocator_;
>-  RtpTransportControllerSendInterface* const transport_;
>+  BitrateAllocatorInterface* const bitrate_allocator_;
>+  RtpTransportControllerSendInterface* const rtp_transport_;
> 
>   rtc::CriticalSection packet_loss_tracker_cs_;
>   TransportFeedbackPacketLossTracker packet_loss_tracker_
>@@ -150,10 +141,6 @@ class AudioSendStream final : public webrtc::AudioSendStream,
>   RtpRtcp* rtp_rtcp_module_;
>   absl::optional<RtpState> const suspended_rtp_state_;
> 
>-  std::unique_ptr<TimedTransport> timed_send_transport_adapter_;
>-  TimeInterval active_lifetime_;
>-  TimeInterval* overall_call_lifetime_ = nullptr;
>-
>   // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
>   // reserved for padding and MUST NOT be used as a local identifier.
>   // So it should be safe to use 0 here to indicate "not configured".
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_unittest.cc
>index 0a954f86d629c5595d2014d74c1378268de1e6e0..e400ada0e5af409b42ac68b4a668bf1a34b41da5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_unittest.cc
>@@ -13,7 +13,7 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>-#include "api/units/time_delta.h"
>+#include "api/test/mock_frame_encryptor.h"
> #include "audio/audio_send_stream.h"
> #include "audio/audio_state.h"
> #include "audio/conversion.h"
>@@ -27,12 +27,10 @@
> #include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h"
> #include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
> #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
>-#include "rtc_base/fakeclock.h"
> #include "rtc_base/task_queue.h"
> #include "test/gtest.h"
> #include "test/mock_audio_encoder.h"
> #include "test/mock_audio_encoder_factory.h"
>-#include "test/mock_transport.h"
> 
> namespace webrtc {
> namespace test {
>@@ -41,6 +39,7 @@ namespace {
> using testing::_;
> using testing::Eq;
> using testing::Ne;
>+using testing::Field;
> using testing::Invoke;
> using testing::Return;
> using testing::StrEq;
>@@ -128,7 +127,7 @@ rtc::scoped_refptr<MockAudioEncoderFactory> SetupEncoderFactoryMock() {
> 
> struct ConfigHelper {
>   ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
>-      : stream_config_(nullptr),
>+      : stream_config_(/*send_transport=*/nullptr, /*media_transport=*/nullptr),
>         audio_processing_(new rtc::RefCountedObject<MockAudioProcessing>()),
>         bitrate_allocator_(&limit_observer_),
>         worker_queue_("ConfigHelper_worker_queue"),
>@@ -142,7 +141,7 @@ struct ConfigHelper {
>         new rtc::RefCountedObject<MockAudioDeviceModule>();
>     audio_state_ = AudioState::Create(config);
> 
>-    SetupDefaultChannelProxy(audio_bwe_enabled);
>+    SetupDefaultChannelSend(audio_bwe_enabled);
>     SetupMockForSetupSendCodec(expect_set_encoder_call);
> 
>     // Use ISAC as default codec so as to prevent unnecessary |channel_proxy_|
>@@ -150,7 +149,6 @@ struct ConfigHelper {
>     stream_config_.send_codec_spec =
>         AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
>     stream_config_.rtp.ssrc = kSsrc;
>-    stream_config_.rtp.nack.rtp_history_ms = 200;
>     stream_config_.rtp.c_name = kCName;
>     stream_config_.rtp.extensions.push_back(
>         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
>@@ -167,8 +165,7 @@ struct ConfigHelper {
>         new internal::AudioSendStream(
>             stream_config_, audio_state_, &worker_queue_, &rtp_transport_,
>             &bitrate_allocator_, &event_log_, &rtcp_rtt_stats_, absl::nullopt,
>-            &active_lifetime_,
>-            std::unique_ptr<voe::ChannelSendProxy>(channel_proxy_)));
>+            std::unique_ptr<voe::ChannelSendInterface>(channel_send_)));
>   }
> 
>   AudioSendStream::Config& config() { return stream_config_; }
>@@ -176,9 +173,8 @@ struct ConfigHelper {
>     return *static_cast<MockAudioEncoderFactory*>(
>         stream_config_.encoder_factory.get());
>   }
>-  MockChannelSendProxy* channel_proxy() { return channel_proxy_; }
>+  MockChannelSend* channel_send() { return channel_send_; }
>   RtpTransportControllerSendInterface* transport() { return &rtp_transport_; }
>-  TimeInterval* active_lifetime() { return &active_lifetime_; }
> 
>   static void AddBweToConfig(AudioSendStream::Config* config) {
>     config->rtp.extensions.push_back(RtpExtension(
>@@ -186,47 +182,40 @@ struct ConfigHelper {
>     config->send_codec_spec->transport_cc_enabled = true;
>   }
> 
>-  void SetupDefaultChannelProxy(bool audio_bwe_enabled) {
>-    EXPECT_TRUE(channel_proxy_ == nullptr);
>-    channel_proxy_ = new testing::StrictMock<MockChannelSendProxy>();
>-    EXPECT_CALL(*channel_proxy_, GetRtpRtcp()).WillRepeatedly(Invoke([this]() {
>+  void SetupDefaultChannelSend(bool audio_bwe_enabled) {
>+    EXPECT_TRUE(channel_send_ == nullptr);
>+    channel_send_ = new testing::StrictMock<MockChannelSend>();
>+    EXPECT_CALL(*channel_send_, GetRtpRtcp()).WillRepeatedly(Invoke([this]() {
>       return &this->rtp_rtcp_;
>     }));
>-    EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
>-    EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
>-    EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
>-    EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 10)).Times(1);
>-    EXPECT_CALL(*channel_proxy_, SetFrameEncryptor(nullptr)).Times(1);
>-    EXPECT_CALL(*channel_proxy_,
>+    EXPECT_CALL(*channel_send_, SetLocalSSRC(kSsrc)).Times(1);
>+    EXPECT_CALL(*channel_send_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
>+    EXPECT_CALL(*channel_send_, SetFrameEncryptor(_)).Times(1);
>+    EXPECT_CALL(*channel_send_, SetExtmapAllowMixed(false)).Times(1);
>+    EXPECT_CALL(*channel_send_,
>                 SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
>         .Times(1);
>     EXPECT_CALL(rtp_transport_, GetBandwidthObserver())
>         .WillRepeatedly(Return(&bandwidth_observer_));
>     if (audio_bwe_enabled) {
>-      EXPECT_CALL(*channel_proxy_,
>+      EXPECT_CALL(*channel_send_,
>                   EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
>           .Times(1);
>-      EXPECT_CALL(*channel_proxy_,
>+      EXPECT_CALL(*channel_send_,
>                   RegisterSenderCongestionControlObjects(
>                       &rtp_transport_, Eq(&bandwidth_observer_)))
>           .Times(1);
>     } else {
>-      EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects(
>-                                       &rtp_transport_, Eq(nullptr)))
>+      EXPECT_CALL(*channel_send_, RegisterSenderCongestionControlObjects(
>+                                      &rtp_transport_, Eq(nullptr)))
>           .Times(1);
>     }
>-    EXPECT_CALL(*channel_proxy_, ResetSenderCongestionControlObjects())
>-        .Times(1);
>-    {
>-      ::testing::InSequence unregister_on_destruction;
>-      EXPECT_CALL(*channel_proxy_, RegisterTransport(_)).Times(1);
>-      EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(1);
>-    }
>+    EXPECT_CALL(*channel_send_, ResetSenderCongestionControlObjects()).Times(1);
>   }
> 
>   void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
>     if (expect_set_encoder_call) {
>-      EXPECT_CALL(*channel_proxy_, SetEncoderForMock(_, _))
>+      EXPECT_CALL(*channel_send_, SetEncoderForMock(_, _))
>           .WillOnce(Invoke(
>               [this](int payload_type, std::unique_ptr<AudioEncoder>* encoder) {
>                 this->audio_encoder_ = std::move(*encoder);
>@@ -237,7 +226,7 @@ struct ConfigHelper {
> 
>   void SetupMockForModifyEncoder() {
>     // Let ModifyEncoder to invoke mock audio encoder.
>-    EXPECT_CALL(*channel_proxy_, ModifyEncoder(_))
>+    EXPECT_CALL(*channel_send_, ModifyEncoder(_))
>         .WillRepeatedly(Invoke(
>             [this](rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
>                        modifier) {
>@@ -247,13 +236,13 @@ struct ConfigHelper {
>   }
> 
>   void SetupMockForSendTelephoneEvent() {
>-    EXPECT_TRUE(channel_proxy_);
>-    EXPECT_CALL(*channel_proxy_, SetSendTelephoneEventPayloadType(
>-                                     kTelephoneEventPayloadType,
>-                                     kTelephoneEventPayloadFrequency))
>+    EXPECT_TRUE(channel_send_);
>+    EXPECT_CALL(*channel_send_, SetSendTelephoneEventPayloadType(
>+                                    kTelephoneEventPayloadType,
>+                                    kTelephoneEventPayloadFrequency))
>         .WillOnce(Return(true));
>     EXPECT_CALL(
>-        *channel_proxy_,
>+        *channel_send_,
>         SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
>         .WillOnce(Return(true));
>   }
>@@ -271,13 +260,14 @@ struct ConfigHelper {
>     block.fraction_lost = 0;
>     report_blocks.push_back(block);  // Duplicate SSRC, bad fraction_lost.
> 
>-    EXPECT_TRUE(channel_proxy_);
>-    EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
>+    EXPECT_TRUE(channel_send_);
>+    EXPECT_CALL(*channel_send_, GetRTCPStatistics())
>         .WillRepeatedly(Return(kCallStats));
>-    EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
>+    EXPECT_CALL(*channel_send_, GetRemoteRTCPReportBlocks())
>         .WillRepeatedly(Return(report_blocks));
>-    EXPECT_CALL(*channel_proxy_, GetANAStatistics())
>+    EXPECT_CALL(*channel_send_, GetANAStatistics())
>         .WillRepeatedly(Return(ANAStats()));
>+    EXPECT_CALL(*channel_send_, GetBitrate()).WillRepeatedly(Return(0));
> 
>     audio_processing_stats_.echo_return_loss = kEchoReturnLoss;
>     audio_processing_stats_.echo_return_loss_enhancement =
>@@ -297,10 +287,9 @@ struct ConfigHelper {
>  private:
>   rtc::scoped_refptr<AudioState> audio_state_;
>   AudioSendStream::Config stream_config_;
>-  testing::StrictMock<MockChannelSendProxy>* channel_proxy_ = nullptr;
>+  testing::StrictMock<MockChannelSend>* channel_send_ = nullptr;
>   rtc::scoped_refptr<MockAudioProcessing> audio_processing_;
>   AudioProcessingStats audio_processing_stats_;
>-  TimeInterval active_lifetime_;
>   testing::StrictMock<MockRtcpBandwidthObserver> bandwidth_observer_;
>   testing::NiceMock<MockRtcEventLog> event_log_;
>   testing::NiceMock<MockRtpTransportControllerSend> rtp_transport_;
>@@ -316,7 +305,8 @@ struct ConfigHelper {
> }  // namespace
> 
> TEST(AudioSendStreamTest, ConfigToString) {
>-  AudioSendStream::Config config(nullptr);
>+  AudioSendStream::Config config(/*send_transport=*/nullptr,
>+                                 /*media_transport=*/nullptr);
>   config.rtp.ssrc = kSsrc;
>   config.rtp.c_name = kCName;
>   config.min_bitrate_bps = 12000;
>@@ -327,12 +317,15 @@ TEST(AudioSendStreamTest, ConfigToString) {
>   config.send_codec_spec->transport_cc_enabled = false;
>   config.send_codec_spec->cng_payload_type = 42;
>   config.encoder_factory = MockAudioEncoderFactory::CreateUnusedFactory();
>+  config.rtp.extmap_allow_mixed = true;
>   config.rtp.extensions.push_back(
>       RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
>+  config.rtcp_report_interval_ms = 2500;
>   EXPECT_EQ(
>-      "{rtp: {ssrc: 1234, extensions: [{uri: "
>-      "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], nack: "
>-      "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: null, "
>+      "{rtp: {ssrc: 1234, extmap-allow-mixed: true, extensions: [{uri: "
>+      "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], "
>+      "c_name: foo_name}, rtcp_report_interval_ms: 2500, "
>+      "send_transport: null, media_transport: null, "
>       "min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
>       "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
>       "cng_payload_type: 42, payload_type: 103, "
>@@ -358,7 +351,7 @@ TEST(AudioSendStreamTest, SendTelephoneEvent) {
> TEST(AudioSendStreamTest, SetMuted) {
>   ConfigHelper helper(false, true);
>   auto send_stream = helper.CreateAudioSendStream();
>-  EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
>+  EXPECT_CALL(*helper.channel_send(), SetInputMute(true));
>   send_stream->SetMuted(true);
> }
> 
>@@ -448,7 +441,7 @@ TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
>   helper.config().send_codec_spec->cng_payload_type = 105;
>   using ::testing::Invoke;
>   std::unique_ptr<AudioEncoder> stolen_encoder;
>-  EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _))
>+  EXPECT_CALL(*helper.channel_send(), SetEncoderForMock(_, _))
>       .WillOnce(
>           Invoke([&stolen_encoder](int payload_type,
>                                    std::unique_ptr<AudioEncoder>* encoder) {
>@@ -468,17 +461,31 @@ TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
> TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
>   ConfigHelper helper(false, true);
>   auto send_stream = helper.CreateAudioSendStream();
>-  EXPECT_CALL(*helper.channel_proxy(),
>-              SetBitrate(helper.config().max_bitrate_bps, _));
>-  send_stream->OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50,
>-                                6000);
>+  EXPECT_CALL(*helper.channel_send(),
>+              OnBitrateAllocation(
>+                  Field(&BitrateAllocationUpdate::target_bitrate,
>+                        Eq(DataRate::bps(helper.config().max_bitrate_bps)))));
>+  BitrateAllocationUpdate update;
>+  update.target_bitrate = DataRate::bps(helper.config().max_bitrate_bps + 5000);
>+  update.packet_loss_ratio = 0;
>+  update.round_trip_time = TimeDelta::ms(50);
>+  update.bwe_period = TimeDelta::ms(6000);
>+  send_stream->OnBitrateUpdated(update);
> }
> 
> TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
>   ConfigHelper helper(false, true);
>   auto send_stream = helper.CreateAudioSendStream();
>-  EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
>-  send_stream->OnBitrateUpdated(50000, 0.0, 50, 5000);
>+
>+  EXPECT_CALL(*helper.channel_send(),
>+              OnBitrateAllocation(Field(&BitrateAllocationUpdate::bwe_period,
>+                                        Eq(TimeDelta::ms(5000)))));
>+  BitrateAllocationUpdate update;
>+  update.target_bitrate = DataRate::bps(helper.config().max_bitrate_bps + 5000);
>+  update.packet_loss_ratio = 0;
>+  update.round_trip_time = TimeDelta::ms(50);
>+  update.bwe_period = TimeDelta::ms(5000);
>+  send_stream->OnBitrateUpdated(update);
> }
> 
> // Test that AudioSendStream doesn't recreate the encoder unnecessarily.
>@@ -489,7 +496,7 @@ TEST(AudioSendStreamTest, DontRecreateEncoder) {
>   // to be correct, it's instead set-up manually here. Otherwise a simple change
>   // to ConfigHelper (say to WillRepeatedly) would silently make this test
>   // useless.
>-  EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _))
>+  EXPECT_CALL(*helper.channel_send(), SetEncoderForMock(_, _))
>       .WillOnce(Return(true));
> 
>   helper.config().send_codec_spec =
>@@ -504,46 +511,44 @@ TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) {
>   auto send_stream = helper.CreateAudioSendStream();
>   auto new_config = helper.config();
>   ConfigHelper::AddBweToConfig(&new_config);
>-  EXPECT_CALL(*helper.channel_proxy(),
>+  EXPECT_CALL(*helper.channel_send(),
>               EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
>       .Times(1);
>   {
>     ::testing::InSequence seq;
>-    EXPECT_CALL(*helper.channel_proxy(), ResetSenderCongestionControlObjects())
>+    EXPECT_CALL(*helper.channel_send(), ResetSenderCongestionControlObjects())
>         .Times(1);
>-    EXPECT_CALL(*helper.channel_proxy(), RegisterSenderCongestionControlObjects(
>-                                             helper.transport(), Ne(nullptr)))
>+    EXPECT_CALL(*helper.channel_send(), RegisterSenderCongestionControlObjects(
>+                                            helper.transport(), Ne(nullptr)))
>         .Times(1);
>   }
>   send_stream->Reconfigure(new_config);
> }
> 
>-// Checks that AudioSendStream logs the times at which RTP packets are sent
>-// through its interface.
>-TEST(AudioSendStreamTest, UpdateLifetime) {
>+// Validates that reconfiguring the AudioSendStream with a Frame encryptor
>+// correctly reconfigures on the object without crashing.
>+TEST(AudioSendStreamTest, ReconfigureWithFrameEncryptor) {
>   ConfigHelper helper(false, true);
>+  auto send_stream = helper.CreateAudioSendStream();
>+  auto new_config = helper.config();
> 
>-  MockTransport mock_transport;
>-  helper.config().send_transport = &mock_transport;
>+  rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_0(
>+      new rtc::RefCountedObject<MockFrameEncryptor>());
>+  new_config.frame_encryptor = mock_frame_encryptor_0;
>+  EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(Ne(nullptr))).Times(1);
>+  send_stream->Reconfigure(new_config);
> 
>-  Transport* registered_transport;
>-  ON_CALL(*helper.channel_proxy(), RegisterTransport(_))
>-      .WillByDefault(Invoke([&registered_transport](Transport* transport) {
>-        registered_transport = transport;
>-      }));
>+  // Not updating the frame encryptor shouldn't force it to reconfigure.
>+  EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(_)).Times(0);
>+  send_stream->Reconfigure(new_config);
> 
>-  rtc::ScopedFakeClock fake_clock;
>-  constexpr int64_t kTimeBetweenSendRtpCallsMs = 100;
>-  {
>-    auto send_stream = helper.CreateAudioSendStream();
>-    EXPECT_CALL(mock_transport, SendRtp(_, _, _)).Times(2);
>-    const PacketOptions options;
>-    registered_transport->SendRtp(nullptr, 0, options);
>-    fake_clock.AdvanceTime(TimeDelta::ms(kTimeBetweenSendRtpCallsMs));
>-    registered_transport->SendRtp(nullptr, 0, options);
>-  }
>-  EXPECT_TRUE(!helper.active_lifetime()->Empty());
>-  EXPECT_EQ(helper.active_lifetime()->Length(), kTimeBetweenSendRtpCallsMs);
>+  // Updating frame encryptor to a new object should force a call to the proxy.
>+  rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_1(
>+      new rtc::RefCountedObject<MockFrameEncryptor>());
>+  new_config.frame_encryptor = mock_frame_encryptor_1;
>+  new_config.crypto_options.sframe.require_frame_encryption = true;
>+  EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(Ne(nullptr))).Times(1);
>+  send_stream->Reconfigure(new_config);
> }
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.cc
>index 94ef6cb68a775d57d7b38f3db3997ac59902ca20..539456e2f49ae6f5cce0a2eef7a52d379ba728d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.cc
>@@ -17,7 +17,7 @@
> #include "audio/remix_resample.h"
> #include "audio/utility/audio_frame_operations.h"
> #include "call/audio_send_stream.h"
>-#include "rtc_base/logging.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive.cc
>index e9f7503474f5f543fce4bd1d47cd54064aac19e6..483147fa9eed00e4e60fb1b4bdb0531560f911da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive.cc
>@@ -18,14 +18,20 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>+#include "audio/audio_level.h"
> #include "audio/channel_send.h"
> #include "audio/utility/audio_frame_operations.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
>+#include "modules/audio_coding/include/audio_coding_module.h"
> #include "modules/audio_device/include/audio_device.h"
> #include "modules/pacing/packet_router.h"
> #include "modules/rtp_rtcp/include/receive_statistics.h"
>+#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
>+#include "modules/rtp_rtcp/source/contributing_sources.h"
>+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "modules/utility/include/process_thread.h"
> #include "rtc_base/checks.h"
>@@ -33,6 +39,8 @@
> #include "rtc_base/format_macros.h"
> #include "rtc_base/location.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+#include "rtc_base/race_checker.h"
> #include "rtc_base/thread_checker.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/metrics.h"
>@@ -50,13 +58,228 @@ constexpr int64_t kMinRetransmissionWindowMs = 30;
> constexpr int kVoiceEngineMinMinPlayoutDelayMs = 0;
> constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
> 
>-}  // namespace
>+webrtc::FrameType WebrtcFrameTypeForMediaTransportFrameType(
>+    MediaTransportEncodedAudioFrame::FrameType frame_type) {
>+  switch (frame_type) {
>+    case MediaTransportEncodedAudioFrame::FrameType::kSpeech:
>+      return kAudioFrameSpeech;
>+      break;
>+
>+    case MediaTransportEncodedAudioFrame::FrameType::
>+        kDiscountinuousTransmission:
>+      return kAudioFrameCN;
>+      break;
>+  }
>+}
>+
>+WebRtcRTPHeader CreateWebrtcRTPHeaderForMediaTransportFrame(
>+    const MediaTransportEncodedAudioFrame& frame,
>+    uint64_t channel_id) {
>+  webrtc::WebRtcRTPHeader webrtc_header = {};
>+  webrtc_header.header.payloadType = frame.payload_type();
>+  webrtc_header.header.payload_type_frequency = frame.sampling_rate_hz();
>+  webrtc_header.header.timestamp = frame.starting_sample_index();
>+  webrtc_header.header.sequenceNumber = frame.sequence_number();
>+
>+  webrtc_header.frameType =
>+      WebrtcFrameTypeForMediaTransportFrameType(frame.frame_type());
>+
>+  webrtc_header.header.ssrc = static_cast<uint32_t>(channel_id);
>+
>+  // The rest are initialized by the RTPHeader constructor.
>+  return webrtc_header;
>+}
>+
>+class ChannelReceive : public ChannelReceiveInterface,
>+                       public MediaTransportAudioSinkInterface {
>+ public:
>+  // Used for receive streams.
>+  ChannelReceive(ProcessThread* module_process_thread,
>+                 AudioDeviceModule* audio_device_module,
>+                 MediaTransportInterface* media_transport,
>+                 Transport* rtcp_send_transport,
>+                 RtcEventLog* rtc_event_log,
>+                 uint32_t remote_ssrc,
>+                 size_t jitter_buffer_max_packets,
>+                 bool jitter_buffer_fast_playout,
>+                 int jitter_buffer_min_delay_ms,
>+                 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
>+                 absl::optional<AudioCodecPairId> codec_pair_id,
>+                 rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
>+                 const webrtc::CryptoOptions& crypto_options);
>+  ~ChannelReceive() override;
>+
>+  void SetSink(AudioSinkInterface* sink) override;
>+
>+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
>+
>+  // API methods
>+
>+  void StartPlayout() override;
>+  void StopPlayout() override;
>+
>+  // Codecs
>+  bool GetRecCodec(CodecInst* codec) const override;
>+
>+  bool ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
>+
>+  // RtpPacketSinkInterface.
>+  void OnRtpPacket(const RtpPacketReceived& packet) override;
>+
>+  // Muting, Volume and Level.
>+  void SetChannelOutputVolumeScaling(float scaling) override;
>+  int GetSpeechOutputLevelFullRange() const override;
>+  // See description of "totalAudioEnergy" in the WebRTC stats spec:
>+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
>+  double GetTotalOutputEnergy() const override;
>+  double GetTotalOutputDuration() const override;
>+
>+  // Stats.
>+  NetworkStatistics GetNetworkStatistics() const override;
>+  AudioDecodingCallStats GetDecodingCallStatistics() const override;
>+
>+  // Audio+Video Sync.
>+  uint32_t GetDelayEstimate() const override;
>+  void SetMinimumPlayoutDelay(int delayMs) override;
>+  uint32_t GetPlayoutTimestamp() const override;
>+
>+  // Produces the transport-related timestamps; current_delay_ms is left unset.
>+  absl::optional<Syncable::Info> GetSyncInfo() const override;
>+
>+  // RTP+RTCP
>+  void SetLocalSSRC(unsigned int ssrc) override;
>+
>+  void RegisterReceiverCongestionControlObjects(
>+      PacketRouter* packet_router) override;
>+  void ResetReceiverCongestionControlObjects() override;
>+
>+  CallReceiveStatistics GetRTCPStatistics() const override;
>+  void SetNACKStatus(bool enable, int maxNumberOfPackets) override;
>+
>+  AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
>+      int sample_rate_hz,
>+      AudioFrame* audio_frame) override;
>+
>+  int PreferredSampleRate() const override;
>+
>+  // Associate to a send channel.
>+  // Used for obtaining RTT for a receive-only channel.
>+  void SetAssociatedSendChannel(const ChannelSendInterface* channel) override;
>+
>+  std::vector<RtpSource> GetSources() const override;
>+
>+ private:
>+  bool ReceivePacket(const uint8_t* packet,
>+                     size_t packet_length,
>+                     const RTPHeader& header);
>+  int ResendPackets(const uint16_t* sequence_numbers, int length);
>+  void UpdatePlayoutTimestamp(bool rtcp);
>+
>+  int GetRtpTimestampRateHz() const;
>+  int64_t GetRTT() const;
>+
>+  // MediaTransportAudioSinkInterface override;
>+  void OnData(uint64_t channel_id,
>+              MediaTransportEncodedAudioFrame frame) override;
>+
>+  int32_t OnReceivedPayloadData(const uint8_t* payloadData,
>+                                size_t payloadSize,
>+                                const WebRtcRTPHeader* rtpHeader);
>+
>+  bool Playing() const {
>+    rtc::CritScope lock(&playing_lock_);
>+    return playing_;
>+  }
>+
>+  // Thread checkers document and lock usage of some methods to specific threads
>+  // we know about. The goal is to eventually split up voe::ChannelReceive into
>+  // parts with single-threaded semantics, and thereby reduce the need for
>+  // locks.
>+  rtc::ThreadChecker worker_thread_checker_;
>+  rtc::ThreadChecker module_process_thread_checker_;
>+  // Methods accessed from audio and video threads are checked for sequential-
>+  // only access. We don't necessarily own and control these threads, so thread
>+  // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
>+  // audio thread to another, but access is still sequential.
>+  rtc::RaceChecker audio_thread_race_checker_;
>+  rtc::RaceChecker video_capture_thread_race_checker_;
>+  rtc::CriticalSection _callbackCritSect;
>+  rtc::CriticalSection volume_settings_critsect_;
>+
>+  rtc::CriticalSection playing_lock_;
>+  bool playing_ RTC_GUARDED_BY(&playing_lock_) = false;
>+
>+  RtcEventLog* const event_log_;
>+
>+  // Indexed by payload type.
>+  std::map<uint8_t, int> payload_type_frequencies_;
>+
>+  std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
>+  std::unique_ptr<RtpRtcp> _rtpRtcpModule;
>+  const uint32_t remote_ssrc_;
>+
>+  // Info for GetSources and GetSyncInfo is updated on network or worker thread,
>+  // queried on the worker thread.
>+  rtc::CriticalSection rtp_sources_lock_;
>+  ContributingSources contributing_sources_ RTC_GUARDED_BY(&rtp_sources_lock_);
>+  absl::optional<uint32_t> last_received_rtp_timestamp_
>+      RTC_GUARDED_BY(&rtp_sources_lock_);
>+  absl::optional<int64_t> last_received_rtp_system_time_ms_
>+      RTC_GUARDED_BY(&rtp_sources_lock_);
>+  absl::optional<uint8_t> last_received_rtp_audio_level_
>+      RTC_GUARDED_BY(&rtp_sources_lock_);
>+
>+  std::unique_ptr<AudioCodingModule> audio_coding_;
>+  AudioSinkInterface* audio_sink_ = nullptr;
>+  AudioLevel _outputAudioLevel;
>+
>+  RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
>+
>+  // Timestamp of the audio pulled from NetEq.
>+  absl::optional<uint32_t> jitter_buffer_playout_timestamp_;
>+
>+  rtc::CriticalSection video_sync_lock_;
>+  uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_);
>+  uint32_t playout_delay_ms_ RTC_GUARDED_BY(video_sync_lock_);
>+
>+  rtc::CriticalSection ts_stats_lock_;
>+
>+  std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
>+  // The rtp timestamp of the first played out audio frame.
>+  int64_t capture_start_rtp_time_stamp_;
>+  // The capture ntp time (in local timebase) of the first played out audio
>+  // frame.
>+  int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_);
>+
>+  // uses
>+  ProcessThread* _moduleProcessThreadPtr;
>+  AudioDeviceModule* _audioDeviceModulePtr;
>+  float _outputGain RTC_GUARDED_BY(volume_settings_critsect_);
>+
>+  // An associated send channel.
>+  rtc::CriticalSection assoc_send_channel_lock_;
>+  const ChannelSendInterface* associated_send_channel_
>+      RTC_GUARDED_BY(assoc_send_channel_lock_);
>+
>+  PacketRouter* packet_router_ = nullptr;
>+
>+  rtc::ThreadChecker construction_thread_;
>+
>+  MediaTransportInterface* const media_transport_;
>+
>+  // E2EE Audio Frame Decryption
>+  rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
>+  webrtc::CryptoOptions crypto_options_;
>+};
> 
> int32_t ChannelReceive::OnReceivedPayloadData(
>     const uint8_t* payloadData,
>     size_t payloadSize,
>     const WebRtcRTPHeader* rtpHeader) {
>-  if (!channel_state_.Get().playing) {
>+  // We should not be receiving any RTP packets if media_transport is set.
>+  RTC_CHECK(!media_transport_);
>+
>+  if (!Playing()) {
>     // Avoid inserting into NetEQ when we are not playing. Count the
>     // packet as discarded.
>     return 0;
>@@ -82,14 +305,35 @@ int32_t ChannelReceive::OnReceivedPayloadData(
>   return 0;
> }
> 
>+// MediaTransportAudioSinkInterface override.
>+void ChannelReceive::OnData(uint64_t channel_id,
>+                            MediaTransportEncodedAudioFrame frame) {
>+  RTC_CHECK(media_transport_);
>+
>+  if (!Playing()) {
>+    // Avoid inserting into NetEQ when we are not playing. Count the
>+    // packet as discarded.
>+    return;
>+  }
>+
>+  // Send encoded audio frame to Decoder / NetEq.
>+  if (audio_coding_->IncomingPacket(
>+          frame.encoded_data().data(), frame.encoded_data().size(),
>+          CreateWebrtcRTPHeaderForMediaTransportFrame(frame, channel_id)) !=
>+      0) {
>+    RTC_DLOG(LS_ERROR) << "ChannelReceive::OnData: unable to "
>+                          "push data to the ACM";
>+  }
>+}
>+
> AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
>     int sample_rate_hz,
>     AudioFrame* audio_frame) {
>+  RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
>   audio_frame->sample_rate_hz_ = sample_rate_hz;
> 
>-  unsigned int ssrc;
>-  RTC_CHECK_EQ(GetRemoteSSRC(ssrc), 0);
>-  event_log_->Log(absl::make_unique<RtcEventAudioPlayout>(ssrc));
>+  event_log_->Log(absl::make_unique<RtcEventAudioPlayout>(remote_ssrc_));
>+
>   // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
>   bool muted;
>   if (audio_coding_->PlayoutData10Ms(audio_frame->sample_rate_hz_, audio_frame,
>@@ -191,6 +435,7 @@ AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
> }
> 
> int ChannelReceive::PreferredSampleRate() const {
>+  RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
>   // Return the bigger of playout and receive frequency in the ACM.
>   return std::max(audio_coding_->ReceiveFrequency(),
>                   audio_coding_->PlayoutFrequency());
>@@ -199,14 +444,17 @@ int ChannelReceive::PreferredSampleRate() const {
> ChannelReceive::ChannelReceive(
>     ProcessThread* module_process_thread,
>     AudioDeviceModule* audio_device_module,
>+    MediaTransportInterface* media_transport,
>     Transport* rtcp_send_transport,
>     RtcEventLog* rtc_event_log,
>     uint32_t remote_ssrc,
>     size_t jitter_buffer_max_packets,
>     bool jitter_buffer_fast_playout,
>+    int jitter_buffer_min_delay_ms,
>     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
>     absl::optional<AudioCodecPairId> codec_pair_id,
>-    FrameDecryptorInterface* frame_decryptor)
>+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
>+    const webrtc::CryptoOptions& crypto_options)
>     : event_log_(rtc_event_log),
>       rtp_receive_statistics_(
>           ReceiveStatistics::Create(Clock::GetRealTimeClock())),
>@@ -222,7 +470,12 @@ ChannelReceive::ChannelReceive(
>       _audioDeviceModulePtr(audio_device_module),
>       _outputGain(1.0f),
>       associated_send_channel_(nullptr),
>-      frame_decryptor_(frame_decryptor) {
>+      media_transport_(media_transport),
>+      frame_decryptor_(frame_decryptor),
>+      crypto_options_(crypto_options) {
>+  // TODO(nisse): Use _moduleProcessThreadPtr instead?
>+  module_process_thread_checker_.DetachFromThread();
>+
>   RTC_DCHECK(module_process_thread);
>   RTC_DCHECK(audio_device_module);
>   AudioCodingModule::Config acm_config;
>@@ -230,6 +483,7 @@ ChannelReceive::ChannelReceive(
>   acm_config.neteq_config.codec_pair_id = codec_pair_id;
>   acm_config.neteq_config.max_packets_in_buffer = jitter_buffer_max_packets;
>   acm_config.neteq_config.enable_fast_accelerate = jitter_buffer_fast_playout;
>+  acm_config.neteq_config.min_delay_ms = jitter_buffer_min_delay_ms;
>   acm_config.neteq_config.enable_muted_state = true;
>   audio_coding_.reset(AudioCodingModule::Create(acm_config));
> 
>@@ -238,9 +492,7 @@ ChannelReceive::ChannelReceive(
>   rtp_receive_statistics_->EnableRetransmitDetection(remote_ssrc_, true);
>   RtpRtcp::Configuration configuration;
>   configuration.audio = true;
>-  // TODO(nisse): Also set receiver_only = true, but that seems to break RTT
>-  // estimation, resulting in test failures for
>-  // PeerConnectionIntegrationTest.GetCaptureStartNtpTimeWithOldStatsApi
>+  configuration.receiver_only = true;
>   configuration.outgoing_transport = rtcp_send_transport;
>   configuration.receive_statistics = rtp_receive_statistics_.get();
> 
>@@ -249,26 +501,9 @@ ChannelReceive::ChannelReceive(
>   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
>   _rtpRtcpModule->SetSendingMediaStatus(false);
>   _rtpRtcpModule->SetRemoteSSRC(remote_ssrc_);
>-  Init();
>-}
>-
>-ChannelReceive::~ChannelReceive() {
>-  Terminate();
>-  RTC_DCHECK(!channel_state_.Get().playing);
>-}
>-
>-void ChannelReceive::Init() {
>-  channel_state_.Reset();
> 
>-  // --- Add modules to process thread (for periodic schedulation)
>   _moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE);
> 
>-  // --- ACM initialization
>-  int error = audio_coding_->InitializeReceiver();
>-  RTC_DCHECK_EQ(0, error);
>-
>-  // --- RTP/RTCP module initialization
>-
>   // Ensure that RTCP is enabled by default for the created channel.
>   // Note that, the module will keep generating RTCP until it is explicitly
>   // disabled by the user.
>@@ -276,60 +511,54 @@ void ChannelReceive::Init() {
>   // be transmitted since the Transport object will then be invalid.
>   // RTCP is enabled by default.
>   _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
>+
>+  if (media_transport_) {
>+    media_transport_->SetReceiveAudioSink(this);
>+  }
> }
> 
>-void ChannelReceive::Terminate() {
>+ChannelReceive::~ChannelReceive() {
>   RTC_DCHECK(construction_thread_.CalledOnValidThread());
>-  // Must be called on the same thread as Init().
>-  rtp_receive_statistics_->RegisterRtcpStatisticsCallback(NULL);
>+
>+  if (media_transport_) {
>+    media_transport_->SetReceiveAudioSink(nullptr);
>+  }
> 
>   StopPlayout();
> 
>-  // The order to safely shutdown modules in a channel is:
>-  // 1. De-register callbacks in modules
>-  // 2. De-register modules in process thread
>-  // 3. Destroy modules
>   int error = audio_coding_->RegisterTransportCallback(NULL);
>   RTC_DCHECK_EQ(0, error);
> 
>-  // De-register modules in process thread
>   if (_moduleProcessThreadPtr)
>     _moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get());
>-
>-  // End of modules shutdown
> }
> 
> void ChannelReceive::SetSink(AudioSinkInterface* sink) {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   rtc::CritScope cs(&_callbackCritSect);
>   audio_sink_ = sink;
> }
> 
>-int32_t ChannelReceive::StartPlayout() {
>-  if (channel_state_.Get().playing) {
>-    return 0;
>-  }
>-
>-  channel_state_.SetPlaying(true);
>-
>-  return 0;
>+void ChannelReceive::StartPlayout() {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>+  rtc::CritScope lock(&playing_lock_);
>+  playing_ = true;
> }
> 
>-int32_t ChannelReceive::StopPlayout() {
>-  if (!channel_state_.Get().playing) {
>-    return 0;
>-  }
>-
>-  channel_state_.SetPlaying(false);
>+void ChannelReceive::StopPlayout() {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>+  rtc::CritScope lock(&playing_lock_);
>+  playing_ = false;
>   _outputAudioLevel.Clear();
>-
>-  return 0;
> }
> 
>-int32_t ChannelReceive::GetRecCodec(CodecInst& codec) {
>-  return (audio_coding_->ReceiveCodec(&codec));
>+bool ChannelReceive::GetRecCodec(CodecInst* codec) const {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>+  return (audio_coding_->ReceiveCodec(codec) == 0);
> }
> 
> std::vector<webrtc::RtpSource> ChannelReceive::GetSources() const {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   int64_t now_ms = rtc::TimeMillis();
>   std::vector<RtpSource> sources;
>   {
>@@ -347,6 +576,7 @@ std::vector<webrtc::RtpSource> ChannelReceive::GetSources() const {
> 
> void ChannelReceive::SetReceiveCodecs(
>     const std::map<int, SdpAudioFormat>& codecs) {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   for (const auto& kv : codecs) {
>     RTC_DCHECK_GE(kv.second.clockrate_hz, 1000);
>     payload_type_frequencies_[kv.first] = kv.second.clockrate_hz;
>@@ -354,7 +584,7 @@ void ChannelReceive::SetReceiveCodecs(
>   audio_coding_->SetReceiveCodecs(codecs);
> }
> 
>-// TODO(nisse): Move receive logic up to AudioReceiveStream.
>+// May be called on either worker thread or network thread.
> void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
>   int64_t now_ms = rtc::TimeMillis();
>   uint8_t audio_level;
>@@ -369,7 +599,9 @@ void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
>     if (has_audio_level)
>       last_received_rtp_audio_level_ = audio_level;
>     std::vector<uint32_t> csrcs = packet.Csrcs();
>-    contributing_sources_.Update(now_ms, csrcs);
>+    contributing_sources_.Update(
>+        now_ms, csrcs,
>+        has_audio_level ? absl::optional<uint8_t>(audio_level) : absl::nullopt);
>   }
> 
>   // Store playout timestamp for the received RTP packet
>@@ -429,6 +661,10 @@ bool ChannelReceive::ReceivePacket(const uint8_t* packet,
>     // Update the final payload.
>     payload = decrypted_audio_payload.data();
>     payload_data_length = decrypted_audio_payload.size();
>+  } else if (crypto_options_.sframe.require_frame_encryption) {
>+    RTC_DLOG(LS_ERROR)
>+        << "FrameDecryptor required but not set, dropping packet";
>+    payload_data_length = 0;
>   }
> 
>   if (payload_data_length == 0) {
>@@ -439,7 +675,9 @@ bool ChannelReceive::ReceivePacket(const uint8_t* packet,
>                                &webrtc_rtp_header);
> }
> 
>-int32_t ChannelReceive::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
>+// May be called on either worker thread or network thread.
>+// TODO(nisse): Drop always-true return value.
>+bool ChannelReceive::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
>   // Store playout timestamp for the received RTCP packet
>   UpdatePlayoutTimestamp(true);
> 
>@@ -449,7 +687,7 @@ int32_t ChannelReceive::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
>   int64_t rtt = GetRTT();
>   if (rtt == 0) {
>     // Waiting for valid RTT.
>-    return 0;
>+    return true;
>   }
> 
>   int64_t nack_window_ms = rtt;
>@@ -465,46 +703,45 @@ int32_t ChannelReceive::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
>   if (0 != _rtpRtcpModule->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL,
>                                      &rtp_timestamp)) {
>     // Waiting for RTCP.
>-    return 0;
>+    return true;
>   }
> 
>   {
>     rtc::CritScope lock(&ts_stats_lock_);
>     ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
>   }
>-  return 0;
>+  return true;
> }
> 
> int ChannelReceive::GetSpeechOutputLevelFullRange() const {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   return _outputAudioLevel.LevelFullRange();
> }
> 
> double ChannelReceive::GetTotalOutputEnergy() const {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   return _outputAudioLevel.TotalEnergy();
> }
> 
> double ChannelReceive::GetTotalOutputDuration() const {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   return _outputAudioLevel.TotalDuration();
> }
> 
> void ChannelReceive::SetChannelOutputVolumeScaling(float scaling) {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   rtc::CritScope cs(&volume_settings_critsect_);
>   _outputGain = scaling;
> }
> 
>-int ChannelReceive::SetLocalSSRC(unsigned int ssrc) {
>+void ChannelReceive::SetLocalSSRC(uint32_t ssrc) {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   _rtpRtcpModule->SetSSRC(ssrc);
>-  return 0;
>-}
>-
>-// TODO(nisse): Pass ssrc in return value instead.
>-int ChannelReceive::GetRemoteSSRC(unsigned int& ssrc) {
>-  ssrc = remote_ssrc_;
>-  return 0;
> }
> 
> void ChannelReceive::RegisterReceiverCongestionControlObjects(
>     PacketRouter* packet_router) {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   RTC_DCHECK(packet_router);
>   RTC_DCHECK(!packet_router_);
>   constexpr bool remb_candidate = false;
>@@ -513,13 +750,16 @@ void ChannelReceive::RegisterReceiverCongestionControlObjects(
> }
> 
> void ChannelReceive::ResetReceiverCongestionControlObjects() {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   RTC_DCHECK(packet_router_);
>   packet_router_->RemoveReceiveRtpModule(_rtpRtcpModule.get());
>   packet_router_ = nullptr;
> }
> 
>-int ChannelReceive::GetRTPStatistics(CallReceiveStatistics& stats) {
>+CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   // --- RtcpStatistics
>+  CallReceiveStatistics stats;
> 
>   // The jitter statistics is updated for each received RTP packet and is
>   // based on received packets.
>@@ -556,14 +796,15 @@ int ChannelReceive::GetRTPStatistics(CallReceiveStatistics& stats) {
>     rtc::CritScope lock(&ts_stats_lock_);
>     stats.capture_start_ntp_time_ms_ = capture_start_ntp_time_ms_;
>   }
>-  return 0;
>+  return stats;
> }
> 
>-void ChannelReceive::SetNACKStatus(bool enable, int maxNumberOfPackets) {
>+void ChannelReceive::SetNACKStatus(bool enable, int max_packets) {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   // None of these functions can fail.
>-  rtp_receive_statistics_->SetMaxReorderingThreshold(maxNumberOfPackets);
>+  rtp_receive_statistics_->SetMaxReorderingThreshold(max_packets);
>   if (enable)
>-    audio_coding_->EnableNack(maxNumberOfPackets);
>+    audio_coding_->EnableNack(max_packets);
>   else
>     audio_coding_->DisableNack();
> }
>@@ -574,54 +815,61 @@ int ChannelReceive::ResendPackets(const uint16_t* sequence_numbers,
>   return _rtpRtcpModule->SendNACK(sequence_numbers, length);
> }
> 
>-void ChannelReceive::SetAssociatedSendChannel(ChannelSend* channel) {
>+void ChannelReceive::SetAssociatedSendChannel(
>+    const ChannelSendInterface* channel) {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   rtc::CritScope lock(&assoc_send_channel_lock_);
>   associated_send_channel_ = channel;
> }
> 
>-int ChannelReceive::GetNetworkStatistics(NetworkStatistics& stats) {
>-  return audio_coding_->GetNetworkStatistics(&stats);
>+NetworkStatistics ChannelReceive::GetNetworkStatistics() const {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>+  NetworkStatistics stats;
>+  int error = audio_coding_->GetNetworkStatistics(&stats);
>+  RTC_DCHECK_EQ(0, error);
>+  return stats;
> }
> 
>-void ChannelReceive::GetDecodingCallStatistics(
>-    AudioDecodingCallStats* stats) const {
>-  audio_coding_->GetDecodingCallStatistics(stats);
>+AudioDecodingCallStats ChannelReceive::GetDecodingCallStatistics() const {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>+  AudioDecodingCallStats stats;
>+  audio_coding_->GetDecodingCallStatistics(&stats);
>+  return stats;
> }
> 
> uint32_t ChannelReceive::GetDelayEstimate() const {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() ||
>+             module_process_thread_checker_.CalledOnValidThread());
>   rtc::CritScope lock(&video_sync_lock_);
>   return audio_coding_->FilteredCurrentDelayMs() + playout_delay_ms_;
> }
> 
>-int ChannelReceive::SetMinimumPlayoutDelay(int delayMs) {
>-  if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
>-      (delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) {
>+void ChannelReceive::SetMinimumPlayoutDelay(int delay_ms) {
>+  RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
>+  // Limit to range accepted by both VoE and ACM, so we're at least getting as
>+  // close as possible, instead of failing.
>+  delay_ms = rtc::SafeClamp(delay_ms, 0, 10000);
>+  if ((delay_ms < kVoiceEngineMinMinPlayoutDelayMs) ||
>+      (delay_ms > kVoiceEngineMaxMinPlayoutDelayMs)) {
>     RTC_DLOG(LS_ERROR) << "SetMinimumPlayoutDelay() invalid min delay";
>-    return -1;
>+    return;
>   }
>-  if (audio_coding_->SetMinimumPlayoutDelay(delayMs) != 0) {
>+  if (audio_coding_->SetMinimumPlayoutDelay(delay_ms) != 0) {
>     RTC_DLOG(LS_ERROR)
>         << "SetMinimumPlayoutDelay() failed to set min playout delay";
>-    return -1;
>   }
>-  return 0;
> }
> 
>-int ChannelReceive::GetPlayoutTimestamp(unsigned int& timestamp) {
>-  uint32_t playout_timestamp_rtp = 0;
>+uint32_t ChannelReceive::GetPlayoutTimestamp() const {
>+  RTC_DCHECK_RUNS_SERIALIZED(&video_capture_thread_race_checker_);
>   {
>     rtc::CritScope lock(&video_sync_lock_);
>-    playout_timestamp_rtp = playout_timestamp_rtp_;
>-  }
>-  if (playout_timestamp_rtp == 0) {
>-    RTC_DLOG(LS_ERROR) << "GetPlayoutTimestamp() failed to retrieve timestamp";
>-    return -1;
>+    return playout_timestamp_rtp_;
>   }
>-  timestamp = playout_timestamp_rtp;
>-  return 0;
> }
> 
> absl::optional<Syncable::Info> ChannelReceive::GetSyncInfo() const {
>+  RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
>   Syncable::Info info;
>   if (_rtpRtcpModule->RemoteNTP(&info.capture_time_ntp_secs,
>                                 &info.capture_time_ntp_frac, nullptr, nullptr,
>@@ -683,6 +931,14 @@ int ChannelReceive::GetRtpTimestampRateHz() const {
> }
> 
> int64_t ChannelReceive::GetRTT() const {
>+  if (media_transport_) {
>+    auto target_rate = media_transport_->GetLatestTargetTransferRate();
>+    if (target_rate.has_value()) {
>+      return target_rate->network_estimate.round_trip_time.ms();
>+    }
>+
>+    return 0;
>+  }
>   RtcpMode method = _rtpRtcpModule->RTCP();
>   if (method == RtcpMode::kOff) {
>     return 0;
>@@ -705,6 +961,8 @@ int64_t ChannelReceive::GetRTT() const {
>   int64_t avg_rtt = 0;
>   int64_t max_rtt = 0;
>   int64_t min_rtt = 0;
>+  // TODO(nisse): This method computes RTT based on sender reports, even though
>+  // a receive stream is not supposed to do that.
>   if (_rtpRtcpModule->RTT(remote_ssrc_, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
>       0) {
>     return 0;
>@@ -712,5 +970,29 @@ int64_t ChannelReceive::GetRTT() const {
>   return rtt;
> }
> 
>+}  // namespace
>+
>+std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
>+    ProcessThread* module_process_thread,
>+    AudioDeviceModule* audio_device_module,
>+    MediaTransportInterface* media_transport,
>+    Transport* rtcp_send_transport,
>+    RtcEventLog* rtc_event_log,
>+    uint32_t remote_ssrc,
>+    size_t jitter_buffer_max_packets,
>+    bool jitter_buffer_fast_playout,
>+    int jitter_buffer_min_delay_ms,
>+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
>+    absl::optional<AudioCodecPairId> codec_pair_id,
>+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
>+    const webrtc::CryptoOptions& crypto_options) {
>+  return absl::make_unique<ChannelReceive>(
>+      module_process_thread, audio_device_module, media_transport,
>+      rtcp_send_transport, rtc_event_log, remote_ssrc,
>+      jitter_buffer_max_packets, jitter_buffer_fast_playout,
>+      jitter_buffer_min_delay_ms, decoder_factory, codec_pair_id,
>+      frame_decryptor, crypto_options);
>+}
>+
> }  // namespace voe
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive.h
>index 82eb4dff0c4b55749a3fe69665a137dd4f33b397..90276234a137a16c29ab14396d08173a6bbd1701 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive.h
>@@ -17,22 +17,18 @@
> 
> #include "absl/types/optional.h"
> #include "api/audio/audio_mixer.h"
>+#include "api/audio_codecs/audio_decoder_factory.h"
> #include "api/call/audio_sink.h"
> #include "api/call/transport.h"
>+#include "api/crypto/cryptooptions.h"
>+#include "api/media_transport_interface.h"
> #include "api/rtpreceiverinterface.h"
>-#include "audio/audio_level.h"
>+#include "call/rtp_packet_sink_interface.h"
> #include "call/syncable.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/include/audio_coding_module.h"
>-#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
>-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
>-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
>-#include "modules/rtp_rtcp/source/contributing_sources.h"
>-#include "rtc_base/criticalsection.h"
>-#include "rtc_base/thread_checker.h"
> 
> // TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence
>-// warnings about use of unsigned short, and non-const reference arguments.
>+// warnings about use of unsigned short.
> // These need cleanup, in a separate cl.
> 
> namespace rtc {
>@@ -66,201 +62,85 @@ struct CallReceiveStatistics {
> 
> namespace voe {
> 
>-class ChannelSend;
>+class ChannelSendInterface;
> 
>-// Helper class to simplify locking scheme for members that are accessed from
>-// multiple threads.
>-// Example: a member can be set on thread T1 and read by an internal audio
>-// thread T2. Accessing the member via this class ensures that we are
>-// safe and also avoid TSan v2 warnings.
>-class ChannelReceiveState {
>- public:
>-  struct State {
>-    bool playing = false;
>-  };
>-
>-  ChannelReceiveState() {}
>-  virtual ~ChannelReceiveState() {}
>-
>-  void Reset() {
>-    rtc::CritScope lock(&lock_);
>-    state_ = State();
>-  }
>-
>-  State Get() const {
>-    rtc::CritScope lock(&lock_);
>-    return state_;
>-  }
>-
>-  void SetPlaying(bool enable) {
>-    rtc::CritScope lock(&lock_);
>-    state_.playing = enable;
>-  }
>+// Interface class needed for AudioReceiveStream tests that use a
>+// MockChannelReceive.
> 
>- private:
>-  rtc::CriticalSection lock_;
>-  State state_;
>-};
>-
>-class ChannelReceive : public RtpData {
>+class ChannelReceiveInterface : public RtpPacketSinkInterface {
>  public:
>-  // Used for receive streams.
>-  ChannelReceive(ProcessThread* module_process_thread,
>-                 AudioDeviceModule* audio_device_module,
>-                 Transport* rtcp_send_transport,
>-                 RtcEventLog* rtc_event_log,
>-                 uint32_t remote_ssrc,
>-                 size_t jitter_buffer_max_packets,
>-                 bool jitter_buffer_fast_playout,
>-                 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
>-                 absl::optional<AudioCodecPairId> codec_pair_id,
>-                 FrameDecryptorInterface* frame_decryptor);
>-  virtual ~ChannelReceive();
>+  virtual ~ChannelReceiveInterface() = default;
> 
>-  void SetSink(AudioSinkInterface* sink);
>+  virtual void SetSink(AudioSinkInterface* sink) = 0;
> 
>-  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
>+  virtual void SetReceiveCodecs(
>+      const std::map<int, SdpAudioFormat>& codecs) = 0;
> 
>-  // API methods
>+  virtual void StartPlayout() = 0;
>+  virtual void StopPlayout() = 0;
> 
>-  // VoEBase
>-  int32_t StartPlayout();
>-  int32_t StopPlayout();
>+  virtual bool GetRecCodec(CodecInst* codec) const = 0;
> 
>-  // Codecs
>-  int32_t GetRecCodec(CodecInst& codec);  // NOLINT
>+  virtual bool ReceivedRTCPPacket(const uint8_t* data, size_t length) = 0;
> 
>-  // TODO(nisse, solenberg): Delete when VoENetwork is deleted.
>-  int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
>-  void OnRtpPacket(const RtpPacketReceived& packet);
>-
>-  // Muting, Volume and Level.
>-  void SetChannelOutputVolumeScaling(float scaling);
>-  int GetSpeechOutputLevelFullRange() const;
>+  virtual void SetChannelOutputVolumeScaling(float scaling) = 0;
>+  virtual int GetSpeechOutputLevelFullRange() const = 0;
>   // See description of "totalAudioEnergy" in the WebRTC stats spec:
>   // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
>-  double GetTotalOutputEnergy() const;
>-  double GetTotalOutputDuration() const;
>+  virtual double GetTotalOutputEnergy() const = 0;
>+  virtual double GetTotalOutputDuration() const = 0;
> 
>   // Stats.
>-  int GetNetworkStatistics(NetworkStatistics& stats);  // NOLINT
>-  void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
>+  virtual NetworkStatistics GetNetworkStatistics() const = 0;
>+  virtual AudioDecodingCallStats GetDecodingCallStatistics() const = 0;
> 
>   // Audio+Video Sync.
>-  uint32_t GetDelayEstimate() const;
>-  int SetMinimumPlayoutDelay(int delayMs);
>-  int GetPlayoutTimestamp(unsigned int& timestamp);  // NOLINT
>+  virtual uint32_t GetDelayEstimate() const = 0;
>+  virtual void SetMinimumPlayoutDelay(int delay_ms) = 0;
>+  virtual uint32_t GetPlayoutTimestamp() const = 0;
> 
>   // Produces the transport-related timestamps; current_delay_ms is left unset.
>-  absl::optional<Syncable::Info> GetSyncInfo() const;
>+  virtual absl::optional<Syncable::Info> GetSyncInfo() const = 0;
> 
>   // RTP+RTCP
>-  int SetLocalSSRC(unsigned int ssrc);
>-
>-  void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
>-  void ResetReceiverCongestionControlObjects();
>+  virtual void SetLocalSSRC(uint32_t ssrc) = 0;
> 
>-  int GetRTPStatistics(CallReceiveStatistics& stats);  // NOLINT
>-  void SetNACKStatus(bool enable, int maxNumberOfPackets);
>+  virtual void RegisterReceiverCongestionControlObjects(
>+      PacketRouter* packet_router) = 0;
>+  virtual void ResetReceiverCongestionControlObjects() = 0;
> 
>-  // From RtpData in the RTP/RTCP module
>-  int32_t OnReceivedPayloadData(const uint8_t* payloadData,
>-                                size_t payloadSize,
>-                                const WebRtcRTPHeader* rtpHeader) override;
>+  virtual CallReceiveStatistics GetRTCPStatistics() const = 0;
>+  virtual void SetNACKStatus(bool enable, int max_packets) = 0;
> 
>-  // From AudioMixer::Source.
>-  AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
>+  virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
>       int sample_rate_hz,
>-      AudioFrame* audio_frame);
>+      AudioFrame* audio_frame) = 0;
> 
>-  int PreferredSampleRate() const;
>+  virtual int PreferredSampleRate() const = 0;
> 
>   // Associate to a send channel.
>   // Used for obtaining RTT for a receive-only channel.
>-  void SetAssociatedSendChannel(ChannelSend* channel);
>-
>-  std::vector<RtpSource> GetSources() const;
>-
>- private:
>-  void Init();
>-  void Terminate();
>-
>-  int GetRemoteSSRC(unsigned int& ssrc);  // NOLINT
>-
>-  bool ReceivePacket(const uint8_t* packet,
>-                     size_t packet_length,
>-                     const RTPHeader& header);
>-  int ResendPackets(const uint16_t* sequence_numbers, int length);
>-  void UpdatePlayoutTimestamp(bool rtcp);
>-
>-  int GetRtpTimestampRateHz() const;
>-  int64_t GetRTT() const;
>-
>-  rtc::CriticalSection _callbackCritSect;
>-  rtc::CriticalSection volume_settings_critsect_;
>-
>-  ChannelReceiveState channel_state_;
>-
>-  RtcEventLog* const event_log_;
>+  virtual void SetAssociatedSendChannel(
>+      const ChannelSendInterface* channel) = 0;
> 
>-  // Indexed by payload type.
>-  std::map<uint8_t, int> payload_type_frequencies_;
>-
>-  std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
>-  std::unique_ptr<RtpRtcp> _rtpRtcpModule;
>-  const uint32_t remote_ssrc_;
>-
>-  // Info for GetSources and GetSyncInfo is updated on network or worker thread,
>-  // queried on the worker thread.
>-  rtc::CriticalSection rtp_sources_lock_;
>-  ContributingSources contributing_sources_ RTC_GUARDED_BY(&rtp_sources_lock_);
>-  absl::optional<uint32_t> last_received_rtp_timestamp_
>-      RTC_GUARDED_BY(&rtp_sources_lock_);
>-  absl::optional<int64_t> last_received_rtp_system_time_ms_
>-      RTC_GUARDED_BY(&rtp_sources_lock_);
>-  absl::optional<uint8_t> last_received_rtp_audio_level_
>-      RTC_GUARDED_BY(&rtp_sources_lock_);
>-
>-  std::unique_ptr<AudioCodingModule> audio_coding_;
>-  AudioSinkInterface* audio_sink_ = nullptr;
>-  AudioLevel _outputAudioLevel;
>-
>-  RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
>-
>-  // Timestamp of the audio pulled from NetEq.
>-  absl::optional<uint32_t> jitter_buffer_playout_timestamp_;
>-
>-  rtc::CriticalSection video_sync_lock_;
>-  uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_);
>-  uint32_t playout_delay_ms_ RTC_GUARDED_BY(video_sync_lock_);
>-
>-  rtc::CriticalSection ts_stats_lock_;
>-
>-  std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
>-  // The rtp timestamp of the first played out audio frame.
>-  int64_t capture_start_rtp_time_stamp_;
>-  // The capture ntp time (in local timebase) of the first played out audio
>-  // frame.
>-  int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_);
>-
>-  // uses
>-  ProcessThread* _moduleProcessThreadPtr;
>-  AudioDeviceModule* _audioDeviceModulePtr;
>-  float _outputGain RTC_GUARDED_BY(volume_settings_critsect_);
>-
>-  // An associated send channel.
>-  rtc::CriticalSection assoc_send_channel_lock_;
>-  ChannelSend* associated_send_channel_
>-      RTC_GUARDED_BY(assoc_send_channel_lock_);
>-
>-  PacketRouter* packet_router_ = nullptr;
>-
>-  rtc::ThreadChecker construction_thread_;
>-
>-  // E2EE Audio Frame Decryption
>-  FrameDecryptorInterface* frame_decryptor_ = nullptr;
>+  virtual std::vector<RtpSource> GetSources() const = 0;
> };
> 
>+std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
>+    ProcessThread* module_process_thread,
>+    AudioDeviceModule* audio_device_module,
>+    MediaTransportInterface* media_transport,
>+    Transport* rtcp_send_transport,
>+    RtcEventLog* rtc_event_log,
>+    uint32_t remote_ssrc,
>+    size_t jitter_buffer_max_packets,
>+    bool jitter_buffer_fast_playout,
>+    int jitter_buffer_min_delay_ms,
>+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
>+    absl::optional<AudioCodecPairId> codec_pair_id,
>+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
>+    const webrtc::CryptoOptions& crypto_options);
>+
> }  // namespace voe
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive_proxy.cc
>deleted file mode 100644
>index b1c1c458326ac9ad92f2e59383f04137c89c6295..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive_proxy.cc
>+++ /dev/null
>@@ -1,197 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "audio/channel_receive_proxy.h"
>-
>-#include <utility>
>-
>-#include "api/call/audio_sink.h"
>-#include "audio/channel_send_proxy.h"
>-#include "call/rtp_transport_controller_send_interface.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/numerics/safe_minmax.h"
>-
>-namespace webrtc {
>-namespace voe {
>-ChannelReceiveProxy::ChannelReceiveProxy() {}
>-
>-ChannelReceiveProxy::ChannelReceiveProxy(
>-    std::unique_ptr<ChannelReceive> channel)
>-    : channel_(std::move(channel)) {
>-  RTC_DCHECK(channel_);
>-  module_process_thread_checker_.DetachFromThread();
>-}
>-
>-ChannelReceiveProxy::~ChannelReceiveProxy() {}
>-
>-void ChannelReceiveProxy::SetLocalSSRC(uint32_t ssrc) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  int error = channel_->SetLocalSSRC(ssrc);
>-  RTC_DCHECK_EQ(0, error);
>-}
>-
>-void ChannelReceiveProxy::SetNACKStatus(bool enable, int max_packets) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetNACKStatus(enable, max_packets);
>-}
>-
>-CallReceiveStatistics ChannelReceiveProxy::GetRTCPStatistics() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  CallReceiveStatistics stats = {0};
>-  int error = channel_->GetRTPStatistics(stats);
>-  RTC_DCHECK_EQ(0, error);
>-  return stats;
>-}
>-
>-bool ChannelReceiveProxy::ReceivedRTCPPacket(const uint8_t* packet,
>-                                             size_t length) {
>-  // May be called on either worker thread or network thread.
>-  return channel_->ReceivedRTCPPacket(packet, length) == 0;
>-}
>-
>-void ChannelReceiveProxy::RegisterReceiverCongestionControlObjects(
>-    PacketRouter* packet_router) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->RegisterReceiverCongestionControlObjects(packet_router);
>-}
>-
>-void ChannelReceiveProxy::ResetReceiverCongestionControlObjects() {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->ResetReceiverCongestionControlObjects();
>-}
>-
>-NetworkStatistics ChannelReceiveProxy::GetNetworkStatistics() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  NetworkStatistics stats = {0};
>-  int error = channel_->GetNetworkStatistics(stats);
>-  RTC_DCHECK_EQ(0, error);
>-  return stats;
>-}
>-
>-AudioDecodingCallStats ChannelReceiveProxy::GetDecodingCallStatistics() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  AudioDecodingCallStats stats;
>-  channel_->GetDecodingCallStatistics(&stats);
>-  return stats;
>-}
>-
>-int ChannelReceiveProxy::GetSpeechOutputLevelFullRange() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return channel_->GetSpeechOutputLevelFullRange();
>-}
>-
>-double ChannelReceiveProxy::GetTotalOutputEnergy() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return channel_->GetTotalOutputEnergy();
>-}
>-
>-double ChannelReceiveProxy::GetTotalOutputDuration() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return channel_->GetTotalOutputDuration();
>-}
>-
>-uint32_t ChannelReceiveProxy::GetDelayEstimate() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() ||
>-             module_process_thread_checker_.CalledOnValidThread());
>-  return channel_->GetDelayEstimate();
>-}
>-
>-void ChannelReceiveProxy::SetReceiveCodecs(
>-    const std::map<int, SdpAudioFormat>& codecs) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetReceiveCodecs(codecs);
>-}
>-
>-void ChannelReceiveProxy::SetSink(AudioSinkInterface* sink) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetSink(sink);
>-}
>-
>-void ChannelReceiveProxy::OnRtpPacket(const RtpPacketReceived& packet) {
>-  // May be called on either worker thread or network thread.
>-  channel_->OnRtpPacket(packet);
>-}
>-
>-void ChannelReceiveProxy::SetChannelOutputVolumeScaling(float scaling) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetChannelOutputVolumeScaling(scaling);
>-}
>-
>-AudioMixer::Source::AudioFrameInfo ChannelReceiveProxy::GetAudioFrameWithInfo(
>-    int sample_rate_hz,
>-    AudioFrame* audio_frame) {
>-  RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
>-  return channel_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
>-}
>-
>-int ChannelReceiveProxy::PreferredSampleRate() const {
>-  RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
>-  return channel_->PreferredSampleRate();
>-}
>-
>-void ChannelReceiveProxy::AssociateSendChannel(
>-    const ChannelSendProxy& send_channel_proxy) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetAssociatedSendChannel(send_channel_proxy.GetChannel());
>-}
>-
>-void ChannelReceiveProxy::DisassociateSendChannel() {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetAssociatedSendChannel(nullptr);
>-}
>-
>-absl::optional<Syncable::Info> ChannelReceiveProxy::GetSyncInfo() const {
>-  RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
>-  return channel_->GetSyncInfo();
>-}
>-
>-uint32_t ChannelReceiveProxy::GetPlayoutTimestamp() const {
>-  RTC_DCHECK_RUNS_SERIALIZED(&video_capture_thread_race_checker_);
>-  unsigned int timestamp = 0;
>-  int error = channel_->GetPlayoutTimestamp(timestamp);
>-  RTC_DCHECK(!error || timestamp == 0);
>-  return timestamp;
>-}
>-
>-void ChannelReceiveProxy::SetMinimumPlayoutDelay(int delay_ms) {
>-  RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
>-  // Limit to range accepted by both VoE and ACM, so we're at least getting as
>-  // close as possible, instead of failing.
>-  delay_ms = rtc::SafeClamp(delay_ms, 0, 10000);
>-  int error = channel_->SetMinimumPlayoutDelay(delay_ms);
>-  if (0 != error) {
>-    RTC_LOG(LS_WARNING) << "Error setting minimum playout delay.";
>-  }
>-}
>-
>-bool ChannelReceiveProxy::GetRecCodec(CodecInst* codec_inst) const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return channel_->GetRecCodec(*codec_inst) == 0;
>-}
>-
>-std::vector<RtpSource> ChannelReceiveProxy::GetSources() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return channel_->GetSources();
>-}
>-
>-void ChannelReceiveProxy::StartPlayout() {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  int error = channel_->StartPlayout();
>-  RTC_DCHECK_EQ(0, error);
>-}
>-
>-void ChannelReceiveProxy::StopPlayout() {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  int error = channel_->StopPlayout();
>-  RTC_DCHECK_EQ(0, error);
>-}
>-}  // namespace voe
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive_proxy.h
>deleted file mode 100644
>index 8ebacc300dcfe3a725e9c940903fd08b9dfb7557..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_receive_proxy.h
>+++ /dev/null
>@@ -1,109 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef AUDIO_CHANNEL_RECEIVE_PROXY_H_
>-#define AUDIO_CHANNEL_RECEIVE_PROXY_H_
>-
>-#include <map>
>-#include <memory>
>-#include <vector>
>-
>-#include "api/audio/audio_mixer.h"
>-#include "api/rtpreceiverinterface.h"
>-#include "audio/channel_receive.h"
>-#include "call/rtp_packet_sink_interface.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/race_checker.h"
>-#include "rtc_base/thread_checker.h"
>-
>-namespace webrtc {
>-
>-class AudioSinkInterface;
>-class PacketRouter;
>-class RtpPacketReceived;
>-class Transport;
>-
>-namespace voe {
>-
>-class ChannelSendProxy;
>-
>-// This class provides the "view" of a voe::Channel that we need to implement
>-// webrtc::AudioReceiveStream. It serves two purposes:
>-//  1. Allow mocking just the interfaces used, instead of the entire
>-//     voe::Channel class.
>-//  2. Provide a refined interface for the stream classes, including assumptions
>-//     on return values and input adaptation.
>-class ChannelReceiveProxy : public RtpPacketSinkInterface {
>- public:
>-  ChannelReceiveProxy();
>-  explicit ChannelReceiveProxy(std::unique_ptr<ChannelReceive> channel);
>-  virtual ~ChannelReceiveProxy();
>-
>-  // Shared with ChannelSendProxy
>-  virtual void SetLocalSSRC(uint32_t ssrc);
>-  virtual void SetNACKStatus(bool enable, int max_packets);
>-  virtual CallReceiveStatistics GetRTCPStatistics() const;
>-  virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
>-
>-  virtual void RegisterReceiverCongestionControlObjects(
>-      PacketRouter* packet_router);
>-  virtual void ResetReceiverCongestionControlObjects();
>-  virtual NetworkStatistics GetNetworkStatistics() const;
>-  virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
>-  virtual int GetSpeechOutputLevelFullRange() const;
>-  // See description of "totalAudioEnergy" in the WebRTC stats spec:
>-  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
>-  virtual double GetTotalOutputEnergy() const;
>-  virtual double GetTotalOutputDuration() const;
>-  virtual uint32_t GetDelayEstimate() const;
>-  virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
>-  virtual void SetSink(AudioSinkInterface* sink);
>-
>-  // Implements RtpPacketSinkInterface
>-  void OnRtpPacket(const RtpPacketReceived& packet) override;
>-
>-  virtual void SetChannelOutputVolumeScaling(float scaling);
>-  virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
>-      int sample_rate_hz,
>-      AudioFrame* audio_frame);
>-  virtual int PreferredSampleRate() const;
>-  virtual void AssociateSendChannel(const ChannelSendProxy& send_channel_proxy);
>-  virtual void DisassociateSendChannel();
>-
>-  // Produces the transport-related timestamps; current_delay_ms is left unset.
>-  absl::optional<Syncable::Info> GetSyncInfo() const;
>-  virtual uint32_t GetPlayoutTimestamp() const;
>-  virtual void SetMinimumPlayoutDelay(int delay_ms);
>-  virtual bool GetRecCodec(CodecInst* codec_inst) const;
>-  virtual std::vector<webrtc::RtpSource> GetSources() const;
>-  virtual void StartPlayout();
>-  virtual void StopPlayout();
>-
>- private:
>-  // Thread checkers document and lock usage of some methods on voe::Channel to
>-  // specific threads we know about. The goal is to eventually split up
>-  // voe::Channel into parts with single-threaded semantics, and thereby reduce
>-  // the need for locks.
>-  rtc::ThreadChecker worker_thread_checker_;
>-  rtc::ThreadChecker module_process_thread_checker_;
>-  // Methods accessed from audio and video threads are checked for sequential-
>-  // only access. We don't necessarily own and control these threads, so thread
>-  // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
>-  // audio thread to another, but access is still sequential.
>-  rtc::RaceChecker audio_thread_race_checker_;
>-  rtc::RaceChecker video_capture_thread_race_checker_;
>-  std::unique_ptr<ChannelReceive> channel_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelReceiveProxy);
>-};
>-}  // namespace voe
>-}  // namespace webrtc
>-
>-#endif  // AUDIO_CHANNEL_RECEIVE_PROXY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send.cc
>index d3748b39909a493e32ad2196df926644aa852b98..c458fe465b07080d4f12632b81f817b62a3d1444 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send.cc
>@@ -19,19 +19,25 @@
> 
> #include "absl/memory/memory.h"
> #include "api/array_view.h"
>+#include "api/call/transport.h"
> #include "api/crypto/frameencryptorinterface.h"
> #include "audio/utility/audio_frame_operations.h"
> #include "call/rtp_transport_controller_send_interface.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
>+#include "modules/audio_coding/include/audio_coding_module.h"
>+#include "modules/audio_processing/rms_level.h"
> #include "modules/pacing/packet_router.h"
> #include "modules/utility/include/process_thread.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/format_macros.h"
> #include "rtc_base/location.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/race_checker.h"
> #include "rtc_base/rate_limiter.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/thread_checker.h"
>@@ -47,7 +53,259 @@ namespace {
> constexpr int64_t kMaxRetransmissionWindowMs = 1000;
> constexpr int64_t kMinRetransmissionWindowMs = 30;
> 
>-}  // namespace
>+MediaTransportEncodedAudioFrame::FrameType
>+MediaTransportFrameTypeForWebrtcFrameType(webrtc::FrameType frame_type) {
>+  switch (frame_type) {
>+    case kAudioFrameSpeech:
>+      return MediaTransportEncodedAudioFrame::FrameType::kSpeech;
>+      break;
>+
>+    case kAudioFrameCN:
>+      return MediaTransportEncodedAudioFrame::FrameType::
>+          kDiscontinuousTransmission;
>+      break;
>+
>+    default:
>+      RTC_CHECK(false) << "Unexpected frame type=" << frame_type;
>+      break;
>+  }
>+}
>+
>+class RtpPacketSenderProxy;
>+class TransportFeedbackProxy;
>+class TransportSequenceNumberProxy;
>+class VoERtcpObserver;
>+
>+class ChannelSend
>+    : public ChannelSendInterface,
>+      public Transport,
>+      public OverheadObserver,
>+      public AudioPacketizationCallback,  // receive encoded packets from the
>+                                          // ACM
>+      public TargetTransferRateObserver {
>+ public:
>+  // TODO(nisse): Make OnUplinkPacketLossRate public, and delete friend
>+  // declaration.
>+  friend class VoERtcpObserver;
>+
>+  ChannelSend(rtc::TaskQueue* encoder_queue,
>+              ProcessThread* module_process_thread,
>+              MediaTransportInterface* media_transport,
>+              Transport* rtp_transport,
>+              RtcpRttStats* rtcp_rtt_stats,
>+              RtcEventLog* rtc_event_log,
>+              FrameEncryptorInterface* frame_encryptor,
>+              const webrtc::CryptoOptions& crypto_options,
>+              bool extmap_allow_mixed,
>+              int rtcp_report_interval_ms);
>+
>+  ~ChannelSend() override;
>+
>+  // Send using this encoder, with this payload type.
>+  bool SetEncoder(int payload_type,
>+                  std::unique_ptr<AudioEncoder> encoder) override;
>+  void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
>+                         modifier) override;
>+
>+  // API methods
>+  void StartSend() override;
>+  void StopSend() override;
>+
>+  // Codecs
>+  void OnBitrateAllocation(BitrateAllocationUpdate update) override;
>+  int GetBitrate() const override;
>+
>+  // Network
>+  bool ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
>+
>+  // Muting, Volume and Level.
>+  void SetInputMute(bool enable) override;
>+
>+  // Stats.
>+  ANAStats GetANAStatistics() const override;
>+
>+  // Used by AudioSendStream.
>+  RtpRtcp* GetRtpRtcp() const override;
>+
>+  // DTMF.
>+  bool SendTelephoneEventOutband(int event, int duration_ms) override;
>+  bool SetSendTelephoneEventPayloadType(int payload_type,
>+                                        int payload_frequency) override;
>+
>+  // RTP+RTCP
>+  void SetLocalSSRC(uint32_t ssrc) override;
>+  void SetMid(const std::string& mid, int extension_id) override;
>+  void SetExtmapAllowMixed(bool extmap_allow_mixed) override;
>+  void SetSendAudioLevelIndicationStatus(bool enable, int id) override;
>+  void EnableSendTransportSequenceNumber(int id) override;
>+
>+  void RegisterSenderCongestionControlObjects(
>+      RtpTransportControllerSendInterface* transport,
>+      RtcpBandwidthObserver* bandwidth_observer) override;
>+  void ResetSenderCongestionControlObjects() override;
>+  void SetRTCP_CNAME(absl::string_view c_name) override;
>+  std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const override;
>+  CallSendStatistics GetRTCPStatistics() const override;
>+
>+  // ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
>+  // which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
>+  // the actual processing of the audio takes place. The processing mainly
>+  // consists of encoding and preparing the result for sending by adding it to a
>+  // send queue.
>+  // The main reason for using a task queue here is to release the native,
>+  // OS-specific, audio capture thread as soon as possible to ensure that it
>+  // can go back to sleep and be prepared to deliver an new captured audio
>+  // packet.
>+  void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) override;
>+
>+  void SetTransportOverhead(size_t transport_overhead_per_packet) override;
>+
>+  // The existence of this function alongside OnUplinkPacketLossRate is
>+  // a compromise. We want the encoder to be agnostic of the PLR source, but
>+  // we also don't want it to receive conflicting information from TWCC and
>+  // from RTCP-XR.
>+  void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) override;
>+
>+  void OnRecoverableUplinkPacketLossRate(
>+      float recoverable_packet_loss_rate) override;
>+
>+  int64_t GetRTT() const override;
>+
>+  // E2EE Custom Audio Frame Encryption
>+  void SetFrameEncryptor(
>+      rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override;
>+
>+ private:
>+  class ProcessAndEncodeAudioTask;
>+
>+  // From AudioPacketizationCallback in the ACM
>+  int32_t SendData(FrameType frameType,
>+                   uint8_t payloadType,
>+                   uint32_t timeStamp,
>+                   const uint8_t* payloadData,
>+                   size_t payloadSize,
>+                   const RTPFragmentationHeader* fragmentation) override;
>+
>+  // From Transport (called by the RTP/RTCP module)
>+  bool SendRtp(const uint8_t* data,
>+               size_t len,
>+               const PacketOptions& packet_options) override;
>+  bool SendRtcp(const uint8_t* data, size_t len) override;
>+
>+  // From OverheadObserver in the RTP/RTCP module
>+  void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
>+
>+  void OnUplinkPacketLossRate(float packet_loss_rate);
>+  bool InputMute() const;
>+
>+  int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type, int id);
>+
>+  void UpdateOverheadForEncoder()
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_);
>+
>+  int32_t SendRtpAudio(FrameType frameType,
>+                       uint8_t payloadType,
>+                       uint32_t timeStamp,
>+                       rtc::ArrayView<const uint8_t> payload,
>+                       const RTPFragmentationHeader* fragmentation);
>+
>+  int32_t SendMediaTransportAudio(FrameType frameType,
>+                                  uint8_t payloadType,
>+                                  uint32_t timeStamp,
>+                                  rtc::ArrayView<const uint8_t> payload,
>+                                  const RTPFragmentationHeader* fragmentation);
>+
>+  // Return media transport or nullptr if using RTP.
>+  MediaTransportInterface* media_transport() { return media_transport_; }
>+
>+  // Called on the encoder task queue when a new input audio frame is ready
>+  // for encoding.
>+  void ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input);
>+
>+  void OnReceivedRtt(int64_t rtt_ms);
>+
>+  void OnTargetTransferRate(TargetTransferRate) override;
>+
>+  // Thread checkers document and lock usage of some methods on voe::Channel to
>+  // specific threads we know about. The goal is to eventually split up
>+  // voe::Channel into parts with single-threaded semantics, and thereby reduce
>+  // the need for locks.
>+  rtc::ThreadChecker worker_thread_checker_;
>+  rtc::ThreadChecker module_process_thread_checker_;
>+  // Methods accessed from audio and video threads are checked for sequential-
>+  // only access. We don't necessarily own and control these threads, so thread
>+  // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
>+  // audio thread to another, but access is still sequential.
>+  rtc::RaceChecker audio_thread_race_checker_;
>+
>+  rtc::CriticalSection _callbackCritSect;
>+  rtc::CriticalSection volume_settings_critsect_;
>+
>+  bool sending_ RTC_GUARDED_BY(&worker_thread_checker_) = false;
>+
>+  RtcEventLog* const event_log_;
>+
>+  std::unique_ptr<RtpRtcp> _rtpRtcpModule;
>+
>+  std::unique_ptr<AudioCodingModule> audio_coding_;
>+  uint32_t _timeStamp RTC_GUARDED_BY(encoder_queue_);
>+
>+  uint16_t send_sequence_number_;
>+
>+  // uses
>+  ProcessThread* const _moduleProcessThreadPtr;
>+  Transport* const _transportPtr;  // WebRtc socket or external transport
>+  RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
>+  bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);
>+  bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_);
>+  // VoeRTP_RTCP
>+  // TODO(henrika): can today be accessed on the main thread and on the
>+  // task queue; hence potential race.
>+  bool _includeAudioLevelIndication;
>+  size_t transport_overhead_per_packet_
>+      RTC_GUARDED_BY(overhead_per_packet_lock_);
>+  size_t rtp_overhead_per_packet_ RTC_GUARDED_BY(overhead_per_packet_lock_);
>+  rtc::CriticalSection overhead_per_packet_lock_;
>+  // RtcpBandwidthObserver
>+  const std::unique_ptr<VoERtcpObserver> rtcp_observer_;
>+
>+  PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) =
>+      nullptr;
>+  const std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
>+  const std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
>+  const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
>+  const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
>+
>+  rtc::ThreadChecker construction_thread_;
>+
>+  const bool use_twcc_plr_for_ana_;
>+
>+  rtc::CriticalSection encoder_queue_lock_;
>+  bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false;
>+  rtc::TaskQueue* const encoder_queue_ = nullptr;
>+
>+  MediaTransportInterface* const media_transport_;
>+  int media_transport_sequence_number_ RTC_GUARDED_BY(encoder_queue_) = 0;
>+
>+  rtc::CriticalSection media_transport_lock_;
>+  // Currently set by SetLocalSSRC.
>+  uint64_t media_transport_channel_id_ RTC_GUARDED_BY(&media_transport_lock_) =
>+      0;
>+  // Cache payload type and sampling frequency from most recent call to
>+  // SetEncoder. Needed to set MediaTransportEncodedAudioFrame metadata, and
>+  // invalidate on encoder change.
>+  int media_transport_payload_type_ RTC_GUARDED_BY(&media_transport_lock_);
>+  int media_transport_sampling_frequency_
>+      RTC_GUARDED_BY(&media_transport_lock_);
>+
>+  // E2EE Audio Frame Encryption
>+  rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_;
>+  // E2EE Frame Encryption Options
>+  const webrtc::CryptoOptions crypto_options_;
>+
>+  rtc::CriticalSection bitrate_crit_section_;
>+  int configured_bitrate_bps_ RTC_GUARDED_BY(bitrate_crit_section_) = 0;
>+};
> 
> const int kTelephoneEventAttenuationdB = 10;
> 
>@@ -254,6 +512,23 @@ int32_t ChannelSend::SendData(FrameType frameType,
>                               size_t payloadSize,
>                               const RTPFragmentationHeader* fragmentation) {
>   RTC_DCHECK_RUN_ON(encoder_queue_);
>+  rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
>+
>+  if (media_transport() != nullptr) {
>+    return SendMediaTransportAudio(frameType, payloadType, timeStamp, payload,
>+                                   fragmentation);
>+  } else {
>+    return SendRtpAudio(frameType, payloadType, timeStamp, payload,
>+                        fragmentation);
>+  }
>+}
>+
>+int32_t ChannelSend::SendRtpAudio(FrameType frameType,
>+                                  uint8_t payloadType,
>+                                  uint32_t timeStamp,
>+                                  rtc::ArrayView<const uint8_t> payload,
>+                                  const RTPFragmentationHeader* fragmentation) {
>+  RTC_DCHECK_RUN_ON(encoder_queue_);
>   if (_includeAudioLevelIndication) {
>     // Store current audio level in the RTP/RTCP module.
>     // The level will be used in combination with voice-activity state
>@@ -268,16 +543,15 @@ int32_t ChannelSend::SendData(FrameType frameType,
>     // TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
>     // Allocate a buffer to hold the maximum possible encrypted payload.
>     size_t max_ciphertext_size = frame_encryptor_->GetMaxCiphertextByteSize(
>-        cricket::MEDIA_TYPE_AUDIO, payloadSize);
>+        cricket::MEDIA_TYPE_AUDIO, payload.size());
>     encrypted_audio_payload.SetSize(max_ciphertext_size);
> 
>     // Encrypt the audio payload into the buffer.
>     size_t bytes_written = 0;
>     int encrypt_status = frame_encryptor_->Encrypt(
>         cricket::MEDIA_TYPE_AUDIO, _rtpRtcpModule->SSRC(),
>-        /*additional_data=*/nullptr,
>-        rtc::ArrayView<const uint8_t>(payloadData, payloadSize),
>-        encrypted_audio_payload, &bytes_written);
>+        /*additional_data=*/nullptr, payload, encrypted_audio_payload,
>+        &bytes_written);
>     if (encrypt_status != 0) {
>       RTC_DLOG(LS_ERROR) << "Channel::SendData() failed encrypt audio payload: "
>                          << encrypt_status;
>@@ -286,19 +560,23 @@ int32_t ChannelSend::SendData(FrameType frameType,
>     // Resize the buffer to the exact number of bytes actually used.
>     encrypted_audio_payload.SetSize(bytes_written);
>     // Rewrite the payloadData and size to the new encrypted payload.
>-    payloadData = encrypted_audio_payload.data();
>-    payloadSize = encrypted_audio_payload.size();
>+    payload = encrypted_audio_payload;
>+  } else if (crypto_options_.sframe.require_frame_encryption) {
>+    RTC_DLOG(LS_ERROR) << "Channel::SendData() failed sending audio payload: "
>+                       << "A frame encryptor is required but one is not set.";
>+    return -1;
>   }
> 
>   // Push data from ACM to RTP/RTCP-module to deliver audio frame for
>   // packetization.
>   // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
>-  if (!_rtpRtcpModule->SendOutgoingData(
>-          (FrameType&)frameType, payloadType, timeStamp,
>-          // Leaving the time when this frame was
>-          // received from the capture device as
>-          // undefined for voice for now.
>-          -1, payloadData, payloadSize, fragmentation, nullptr, nullptr)) {
>+  if (!_rtpRtcpModule->SendOutgoingData((FrameType&)frameType, payloadType,
>+                                        timeStamp,
>+                                        // Leaving the time when this frame was
>+                                        // received from the capture device as
>+                                        // undefined for voice for now.
>+                                        -1, payload.data(), payload.size(),
>+                                        fragmentation, nullptr, nullptr)) {
>     RTC_DLOG(LS_ERROR)
>         << "ChannelSend::SendData() failed to send data to RTP/RTCP module";
>     return -1;
>@@ -307,9 +585,68 @@ int32_t ChannelSend::SendData(FrameType frameType,
>   return 0;
> }
> 
>+int32_t ChannelSend::SendMediaTransportAudio(
>+    FrameType frameType,
>+    uint8_t payloadType,
>+    uint32_t timeStamp,
>+    rtc::ArrayView<const uint8_t> payload,
>+    const RTPFragmentationHeader* fragmentation) {
>+  RTC_DCHECK_RUN_ON(encoder_queue_);
>+  // TODO(nisse): Use null _transportPtr for MediaTransport.
>+  // RTC_DCHECK(_transportPtr == nullptr);
>+  uint64_t channel_id;
>+  int sampling_rate_hz;
>+  {
>+    rtc::CritScope cs(&media_transport_lock_);
>+    if (media_transport_payload_type_ != payloadType) {
>+      // Payload type is being changed, media_transport_sampling_frequency_,
>+      // no longer current.
>+      return -1;
>+    }
>+    sampling_rate_hz = media_transport_sampling_frequency_;
>+    channel_id = media_transport_channel_id_;
>+  }
>+  const MediaTransportEncodedAudioFrame frame(
>+      /*sampling_rate_hz=*/sampling_rate_hz,
>+
>+      // TODO(nisse): Timestamp and sample index are the same for all supported
>+      // audio codecs except G722. Refactor audio coding module to only use
>+      // sample index, and leave translation to RTP time, when needed, for
>+      // RTP-specific code.
>+      /*starting_sample_index=*/timeStamp,
>+
>+      // Sample count isn't conveniently available from the AudioCodingModule,
>+      // and needs some refactoring to wire up in a good way. For now, left as
>+      // zero.
>+      /*sample_count=*/0,
>+
>+      /*sequence_number=*/media_transport_sequence_number_,
>+      MediaTransportFrameTypeForWebrtcFrameType(frameType), payloadType,
>+      std::vector<uint8_t>(payload.begin(), payload.end()));
>+
>+  // TODO(nisse): Introduce a MediaTransportSender object bound to a specific
>+  // channel id.
>+  RTCError rtc_error =
>+      media_transport()->SendAudioFrame(channel_id, std::move(frame));
>+
>+  if (!rtc_error.ok()) {
>+    RTC_LOG(LS_ERROR) << "Failed to send frame, rtc_error="
>+                      << ToString(rtc_error.type()) << ", "
>+                      << rtc_error.message();
>+    return -1;
>+  }
>+
>+  ++media_transport_sequence_number_;
>+
>+  return 0;
>+}
>+
> bool ChannelSend::SendRtp(const uint8_t* data,
>                           size_t len,
>                           const PacketOptions& options) {
>+  // We should not be sending RTP packets if media transport is available.
>+  RTC_CHECK(!media_transport());
>+
>   rtc::CritScope cs(&_callbackCritSect);
> 
>   if (_transportPtr == NULL) {
>@@ -343,23 +680,22 @@ bool ChannelSend::SendRtcp(const uint8_t* data, size_t len) {
>   return true;
> }
> 
>-int ChannelSend::PreferredSampleRate() const {
>-  // Return the bigger of playout and receive frequency in the ACM.
>-  return std::max(audio_coding_->ReceiveFrequency(),
>-                  audio_coding_->PlayoutFrequency());
>-}
>-
> ChannelSend::ChannelSend(rtc::TaskQueue* encoder_queue,
>                          ProcessThread* module_process_thread,
>+                         MediaTransportInterface* media_transport,
>+                         Transport* rtp_transport,
>                          RtcpRttStats* rtcp_rtt_stats,
>                          RtcEventLog* rtc_event_log,
>-                         FrameEncryptorInterface* frame_encryptor)
>+                         FrameEncryptorInterface* frame_encryptor,
>+                         const webrtc::CryptoOptions& crypto_options,
>+                         bool extmap_allow_mixed,
>+                         int rtcp_report_interval_ms)
>     : event_log_(rtc_event_log),
>       _timeStamp(0),  // This is just an offset, RTP module will add it's own
>                       // random offset
>       send_sequence_number_(0),
>       _moduleProcessThreadPtr(module_process_thread),
>-      _transportPtr(NULL),
>+      _transportPtr(rtp_transport),
>       input_mute_(false),
>       previous_frame_muted_(false),
>       _includeAudioLevelIndication(false),
>@@ -374,49 +710,56 @@ ChannelSend::ChannelSend(rtc::TaskQueue* encoder_queue,
>       use_twcc_plr_for_ana_(
>           webrtc::field_trial::FindFullName("UseTwccPlrForAna") == "Enabled"),
>       encoder_queue_(encoder_queue),
>-      frame_encryptor_(frame_encryptor) {
>+      media_transport_(media_transport),
>+      frame_encryptor_(frame_encryptor),
>+      crypto_options_(crypto_options) {
>   RTC_DCHECK(module_process_thread);
>   RTC_DCHECK(encoder_queue);
>+  module_process_thread_checker_.DetachFromThread();
>+
>   audio_coding_.reset(AudioCodingModule::Create(AudioCodingModule::Config()));
> 
>   RtpRtcp::Configuration configuration;
>+
>+  // We gradually remove codepaths that depend on RTP when using media
>+  // transport. All of this logic should be moved to the future
>+  // RTPMediaTransport. In this case it means that overhead and bandwidth
>+  // observers should not be called when using media transport.
>+  if (!media_transport_) {
>+    configuration.overhead_observer = this;
>+    configuration.bandwidth_callback = rtcp_observer_.get();
>+    configuration.transport_feedback_callback = feedback_observer_proxy_.get();
>+  }
>+
>   configuration.audio = true;
>   configuration.outgoing_transport = this;
>-  configuration.overhead_observer = this;
>-  configuration.bandwidth_callback = rtcp_observer_.get();
> 
>   configuration.paced_sender = rtp_packet_sender_proxy_.get();
>   configuration.transport_sequence_number_allocator =
>       seq_num_allocator_proxy_.get();
>-  configuration.transport_feedback_callback = feedback_observer_proxy_.get();
> 
>   configuration.event_log = event_log_;
>   configuration.rtt_stats = rtcp_rtt_stats;
>   configuration.retransmission_rate_limiter =
>       retransmission_rate_limiter_.get();
>+  configuration.extmap_allow_mixed = extmap_allow_mixed;
>+  configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
> 
>   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
>   _rtpRtcpModule->SetSendingMediaStatus(false);
>-  Init();
>-}
> 
>-ChannelSend::~ChannelSend() {
>-  Terminate();
>-  RTC_DCHECK(!channel_state_.Get().sending);
>-}
>-
>-void ChannelSend::Init() {
>-  channel_state_.Reset();
>+  // We want to invoke the 'TargetRateObserver' and |OnOverheadChanged|
>+  // callbacks after the audio_coding_ is fully initialized.
>+  if (media_transport_) {
>+    RTC_DLOG(LS_INFO) << "Setting media_transport_ rate observers.";
>+    media_transport_->AddTargetTransferRateObserver(this);
>+    OnOverheadChanged(media_transport_->GetAudioPacketOverhead());
>+  } else {
>+    RTC_DLOG(LS_INFO) << "Not setting media_transport_ rate observers.";
>+  }
> 
>-  // --- Add modules to process thread (for periodic schedulation)
>   _moduleProcessThreadPtr->RegisterModule(_rtpRtcpModule.get(), RTC_FROM_HERE);
> 
>-  // --- ACM initialization
>-  int error = audio_coding_->InitializeReceiver();
>-  RTC_DCHECK_EQ(0, error);
>-
>-  // --- RTP/RTCP module initialization
>-
>   // Ensure that RTCP is enabled by default for the created channel.
>   // Note that, the module will keep generating RTCP until it is explicitly
>   // disabled by the user.
>@@ -425,36 +768,30 @@ void ChannelSend::Init() {
>   // RTCP is enabled by default.
>   _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
> 
>-  // --- Register all permanent callbacks
>-  error = audio_coding_->RegisterTransportCallback(this);
>+  int error = audio_coding_->RegisterTransportCallback(this);
>   RTC_DCHECK_EQ(0, error);
> }
> 
>-void ChannelSend::Terminate() {
>+ChannelSend::~ChannelSend() {
>   RTC_DCHECK(construction_thread_.CalledOnValidThread());
>-  // Must be called on the same thread as Init().
>+
>+  if (media_transport_) {
>+    media_transport_->RemoveTargetTransferRateObserver(this);
>+  }
> 
>   StopSend();
> 
>-  // The order to safely shutdown modules in a channel is:
>-  // 1. De-register callbacks in modules
>-  // 2. De-register modules in process thread
>-  // 3. Destroy modules
>   int error = audio_coding_->RegisterTransportCallback(NULL);
>   RTC_DCHECK_EQ(0, error);
> 
>-  // De-register modules in process thread
>   if (_moduleProcessThreadPtr)
>     _moduleProcessThreadPtr->DeRegisterModule(_rtpRtcpModule.get());
>-
>-  // End of modules shutdown
> }
> 
>-int32_t ChannelSend::StartSend() {
>-  if (channel_state_.Get().sending) {
>-    return 0;
>-  }
>-  channel_state_.SetSending(true);
>+void ChannelSend::StartSend() {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>+  RTC_DCHECK(!sending_);
>+  sending_ = true;
> 
>   // Resume the previous sequence number which was reset by StopSend(). This
>   // needs to be done before |sending| is set to true on the RTP/RTCP module.
>@@ -462,26 +799,21 @@ int32_t ChannelSend::StartSend() {
>     _rtpRtcpModule->SetSequenceNumber(send_sequence_number_);
>   }
>   _rtpRtcpModule->SetSendingMediaStatus(true);
>-  if (_rtpRtcpModule->SetSendingStatus(true) != 0) {
>-    RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending";
>-    _rtpRtcpModule->SetSendingMediaStatus(false);
>-    rtc::CritScope cs(&_callbackCritSect);
>-    channel_state_.SetSending(false);
>-    return -1;
>-  }
>+  int ret = _rtpRtcpModule->SetSendingStatus(true);
>+  RTC_DCHECK_EQ(0, ret);
>   {
>     // It is now OK to start posting tasks to the encoder task queue.
>     rtc::CritScope cs(&encoder_queue_lock_);
>     encoder_queue_is_active_ = true;
>   }
>-  return 0;
> }
> 
> void ChannelSend::StopSend() {
>-  if (!channel_state_.Get().sending) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>+  if (!sending_) {
>     return;
>   }
>-  channel_state_.SetSending(false);
>+  sending_ = false;
> 
>   // Post a task to the encoder thread which sets an event when the task is
>   // executed. We know that no more encoding tasks will be added to the task
>@@ -491,7 +823,7 @@ void ChannelSend::StopSend() {
>   // to acccess and invalid channel object.
>   RTC_DCHECK(encoder_queue_);
> 
>-  rtc::Event flush(false, false);
>+  rtc::Event flush;
>   {
>     // Clear |encoder_queue_is_active_| under lock to prevent any other tasks
>     // than this final "flush task" to be posted on the queue.
>@@ -519,6 +851,7 @@ void ChannelSend::StopSend() {
> 
> bool ChannelSend::SetEncoder(int payload_type,
>                              std::unique_ptr<AudioEncoder> encoder) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   RTC_DCHECK_GE(payload_type, 0);
>   RTC_DCHECK_LE(payload_type, 127);
>   // TODO(ossu): Make CodecInsts up, for now: one for the RTP/RTCP module and
>@@ -549,25 +882,48 @@ bool ChannelSend::SetEncoder(int payload_type,
>     }
>   }
> 
>+  if (media_transport_) {
>+    rtc::CritScope cs(&media_transport_lock_);
>+    media_transport_payload_type_ = payload_type;
>+    // TODO(nisse): Currently broken for G722, since timestamps passed through
>+    // encoder use RTP clock rather than sample count, and they differ for G722.
>+    media_transport_sampling_frequency_ = encoder->RtpTimestampRateHz();
>+  }
>   audio_coding_->SetEncoder(std::move(encoder));
>   return true;
> }
> 
> void ChannelSend::ModifyEncoder(
>     rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   audio_coding_->ModifyEncoder(modifier);
> }
> 
>-void ChannelSend::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) {
>+void ChannelSend::OnBitrateAllocation(BitrateAllocationUpdate update) {
>+  // This method can be called on the worker thread, module process thread
>+  // or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged.
>+  // TODO(solenberg): Figure out a good way to check this or enforce calling
>+  // rules.
>+  // RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() ||
>+  //            module_process_thread_checker_.CalledOnValidThread());
>+  rtc::CritScope lock(&bitrate_crit_section_);
>+
>   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
>     if (*encoder) {
>-      (*encoder)->OnReceivedUplinkBandwidth(bitrate_bps, probing_interval_ms);
>+      (*encoder)->OnReceivedUplinkAllocation(update);
>     }
>   });
>-  retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
>+  retransmission_rate_limiter_->SetMaxRate(update.target_bitrate.bps());
>+  configured_bitrate_bps_ = update.target_bitrate.bps();
>+}
>+
>+int ChannelSend::GetBitrate() const {
>+  rtc::CritScope lock(&bitrate_crit_section_);
>+  return configured_bitrate_bps_;
> }
> 
> void ChannelSend::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   if (!use_twcc_plr_for_ana_)
>     return;
>   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
>@@ -579,6 +935,7 @@ void ChannelSend::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {
> 
> void ChannelSend::OnRecoverableUplinkPacketLossRate(
>     float recoverable_packet_loss_rate) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
>     if (*encoder) {
>       (*encoder)->OnReceivedUplinkRecoverablePacketLossFraction(
>@@ -597,47 +954,22 @@ void ChannelSend::OnUplinkPacketLossRate(float packet_loss_rate) {
>   });
> }
> 
>-bool ChannelSend::EnableAudioNetworkAdaptor(const std::string& config_string) {
>-  bool success = false;
>-  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
>-    if (*encoder) {
>-      success =
>-          (*encoder)->EnableAudioNetworkAdaptor(config_string, event_log_);
>-    }
>-  });
>-  return success;
>-}
>-
>-void ChannelSend::DisableAudioNetworkAdaptor() {
>-  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
>-    if (*encoder)
>-      (*encoder)->DisableAudioNetworkAdaptor();
>-  });
>-}
>-
>-void ChannelSend::SetReceiverFrameLengthRange(int min_frame_length_ms,
>-                                              int max_frame_length_ms) {
>-  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
>-    if (*encoder) {
>-      (*encoder)->SetReceiverFrameLengthRange(min_frame_length_ms,
>-                                              max_frame_length_ms);
>-    }
>-  });
>-}
>-
>-void ChannelSend::RegisterTransport(Transport* transport) {
>-  rtc::CritScope cs(&_callbackCritSect);
>-  _transportPtr = transport;
>-}
>+// TODO(nisse): Delete always-true return value.
>+bool ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
>+  // May be called on either worker thread or network thread.
>+  if (media_transport_) {
>+    // Ignore RTCP packets while media transport is used.
>+    // Those packets should not arrive, but we are seeing occasional packets.
>+    return 0;
>+  }
> 
>-int32_t ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
>   // Deliver RTCP packet to RTP/RTCP module for parsing
>   _rtpRtcpModule->IncomingRtcpPacket(data, length);
> 
>   int64_t rtt = GetRTT();
>   if (rtt == 0) {
>     // Waiting for valid RTT.
>-    return 0;
>+    return true;
>   }
> 
>   int64_t nack_window_ms = rtt;
>@@ -648,16 +980,12 @@ int32_t ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
>   }
>   retransmission_rate_limiter_->SetWindowSize(nack_window_ms);
> 
>-  // Invoke audio encoders OnReceivedRtt().
>-  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
>-    if (*encoder)
>-      (*encoder)->OnReceivedRtt(rtt);
>-  });
>-
>-  return 0;
>+  OnReceivedRtt(rtt);
>+  return true;
> }
> 
> void ChannelSend::SetInputMute(bool enable) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   rtc::CritScope cs(&volume_settings_critsect_);
>   input_mute_ = enable;
> }
>@@ -667,24 +995,26 @@ bool ChannelSend::InputMute() const {
>   return input_mute_;
> }
> 
>-int ChannelSend::SendTelephoneEventOutband(int event, int duration_ms) {
>+bool ChannelSend::SendTelephoneEventOutband(int event, int duration_ms) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   RTC_DCHECK_LE(0, event);
>   RTC_DCHECK_GE(255, event);
>   RTC_DCHECK_LE(0, duration_ms);
>   RTC_DCHECK_GE(65535, duration_ms);
>-  if (!Sending()) {
>-    return -1;
>+  if (!sending_) {
>+    return false;
>   }
>   if (_rtpRtcpModule->SendTelephoneEventOutband(
>           event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
>     RTC_DLOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event";
>-    return -1;
>+    return false;
>   }
>-  return 0;
>+  return true;
> }
> 
>-int ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
>-                                                  int payload_frequency) {
>+bool ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
>+                                                   int payload_frequency) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   RTC_DCHECK_LE(0, payload_type);
>   RTC_DCHECK_GE(127, payload_type);
>   CodecInst codec = {0};
>@@ -697,34 +1027,44 @@ int ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
>       RTC_DLOG(LS_ERROR)
>           << "SetSendTelephoneEventPayloadType() failed to register "
>              "send payload type";
>-      return -1;
>+      return false;
>     }
>   }
>-  return 0;
>+  return true;
> }
> 
>-int ChannelSend::SetLocalSSRC(unsigned int ssrc) {
>-  if (channel_state_.Get().sending) {
>-    RTC_DLOG(LS_ERROR) << "SetLocalSSRC() already sending";
>-    return -1;
>+void ChannelSend::SetLocalSSRC(uint32_t ssrc) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>+  RTC_DCHECK(!sending_);
>+
>+  if (media_transport_) {
>+    rtc::CritScope cs(&media_transport_lock_);
>+    media_transport_channel_id_ = ssrc;
>   }
>   _rtpRtcpModule->SetSSRC(ssrc);
>-  return 0;
> }
> 
> void ChannelSend::SetMid(const std::string& mid, int extension_id) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   int ret = SetSendRtpHeaderExtension(true, kRtpExtensionMid, extension_id);
>   RTC_DCHECK_EQ(0, ret);
>   _rtpRtcpModule->SetMid(mid);
> }
> 
>-int ChannelSend::SetSendAudioLevelIndicationStatus(bool enable,
>-                                                   unsigned char id) {
>+void ChannelSend::SetExtmapAllowMixed(bool extmap_allow_mixed) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>+  _rtpRtcpModule->SetExtmapAllowMixed(extmap_allow_mixed);
>+}
>+
>+void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   _includeAudioLevelIndication = enable;
>-  return SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
>+  int ret = SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
>+  RTC_DCHECK_EQ(0, ret);
> }
> 
> void ChannelSend::EnableSendTransportSequenceNumber(int id) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   int ret =
>       SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id);
>   RTC_DCHECK_EQ(0, ret);
>@@ -733,6 +1073,7 @@ void ChannelSend::EnableSendTransportSequenceNumber(int id) {
> void ChannelSend::RegisterSenderCongestionControlObjects(
>     RtpTransportControllerSendInterface* transport,
>     RtcpBandwidthObserver* bandwidth_observer) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   RtpPacketSender* rtp_packet_sender = transport->packet_sender();
>   TransportFeedbackObserver* transport_feedback_observer =
>       transport->transport_feedback_observer();
>@@ -754,6 +1095,7 @@ void ChannelSend::RegisterSenderCongestionControlObjects(
> }
> 
> void ChannelSend::ResetSenderCongestionControlObjects() {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   RTC_DCHECK(packet_router_);
>   _rtpRtcpModule->SetStorePacketsStatus(false, 600);
>   rtcp_observer_->SetBandwidthObserver(nullptr);
>@@ -764,35 +1106,25 @@ void ChannelSend::ResetSenderCongestionControlObjects() {
>   rtp_packet_sender_proxy_->SetPacketSender(nullptr);
> }
> 
>-void ChannelSend::SetRTCPStatus(bool enable) {
>-  _rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff);
>-}
>-
>-int ChannelSend::SetRTCP_CNAME(const char cName[256]) {
>-  if (_rtpRtcpModule->SetCNAME(cName) != 0) {
>-    RTC_DLOG(LS_ERROR) << "SetRTCP_CNAME() failed to set RTCP CNAME";
>-    return -1;
>-  }
>-  return 0;
>+void ChannelSend::SetRTCP_CNAME(absl::string_view c_name) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>+  // Note: SetCNAME() accepts a c string of length at most 255.
>+  const std::string c_name_limited(c_name.substr(0, 255));
>+  int ret = _rtpRtcpModule->SetCNAME(c_name_limited.c_str()) != 0;
>+  RTC_DCHECK_EQ(0, ret) << "SetRTCP_CNAME() failed to set RTCP CNAME";
> }
> 
>-int ChannelSend::GetRemoteRTCPReportBlocks(
>-    std::vector<ReportBlock>* report_blocks) {
>-  if (report_blocks == NULL) {
>-    RTC_DLOG(LS_ERROR) << "GetRemoteRTCPReportBlock()s invalid report_blocks.";
>-    return -1;
>-  }
>-
>+std::vector<ReportBlock> ChannelSend::GetRemoteRTCPReportBlocks() const {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   // Get the report blocks from the latest received RTCP Sender or Receiver
>   // Report. Each element in the vector contains the sender's SSRC and a
>   // report block according to RFC 3550.
>   std::vector<RTCPReportBlock> rtcp_report_blocks;
>-  if (_rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks) != 0) {
>-    return -1;
>-  }
> 
>-  if (rtcp_report_blocks.empty())
>-    return 0;
>+  int ret = _rtpRtcpModule->RemoteRTCPStat(&rtcp_report_blocks);
>+  RTC_DCHECK_EQ(0, ret);
>+
>+  std::vector<ReportBlock> report_blocks;
> 
>   std::vector<RTCPReportBlock>::const_iterator it = rtcp_report_blocks.begin();
>   for (; it != rtcp_report_blocks.end(); ++it) {
>@@ -806,19 +1138,16 @@ int ChannelSend::GetRemoteRTCPReportBlocks(
>     report_block.interarrival_jitter = it->jitter;
>     report_block.last_SR_timestamp = it->last_sender_report_timestamp;
>     report_block.delay_since_last_SR = it->delay_since_last_sender_report;
>-    report_blocks->push_back(report_block);
>+    report_blocks.push_back(report_block);
>   }
>-  return 0;
>+  return report_blocks;
> }
> 
>-int ChannelSend::GetRTPStatistics(CallSendStatistics& stats) {
>-  // --- RtcpStatistics
>-
>-  // --- RTT
>+CallSendStatistics ChannelSend::GetRTCPStatistics() const {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>+  CallSendStatistics stats = {0};
>   stats.rttMs = GetRTT();
> 
>-  // --- Data counters
>-
>   size_t bytesSent(0);
>   uint32_t packetsSent(0);
> 
>@@ -831,24 +1160,12 @@ int ChannelSend::GetRTPStatistics(CallSendStatistics& stats) {
>   stats.bytesSent = bytesSent;
>   stats.packetsSent = packetsSent;
> 
>-  return 0;
>-}
>-
>-void ChannelSend::SetNACKStatus(bool enable, int maxNumberOfPackets) {
>-  // None of these functions can fail.
>-  if (enable)
>-    audio_coding_->EnableNack(maxNumberOfPackets);
>-  else
>-    audio_coding_->DisableNack();
>-}
>-
>-// Called when we are missing one or more packets.
>-int ChannelSend::ResendPackets(const uint16_t* sequence_numbers, int length) {
>-  return _rtpRtcpModule->SendNACK(sequence_numbers, length);
>+  return stats;
> }
> 
> void ChannelSend::ProcessAndEncodeAudio(
>     std::unique_ptr<AudioFrame> audio_frame) {
>+  RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
>   // Avoid posting any new tasks if sending was already stopped in StopSend().
>   rtc::CritScope cs(&encoder_queue_lock_);
>   if (!encoder_queue_is_active_) {
>@@ -914,6 +1231,7 @@ void ChannelSend::UpdateOverheadForEncoder() {
> }
> 
> void ChannelSend::SetTransportOverhead(size_t transport_overhead_per_packet) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   rtc::CritScope cs(&overhead_per_packet_lock_);
>   transport_overhead_per_packet_ = transport_overhead_per_packet;
>   UpdateOverheadForEncoder();
>@@ -927,36 +1245,42 @@ void ChannelSend::OnOverheadChanged(size_t overhead_bytes_per_packet) {
> }
> 
> ANAStats ChannelSend::GetANAStatistics() const {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   return audio_coding_->GetANAStats();
> }
> 
> RtpRtcp* ChannelSend::GetRtpRtcp() const {
>+  RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
>   return _rtpRtcpModule.get();
> }
> 
> int ChannelSend::SetSendRtpHeaderExtension(bool enable,
>                                            RTPExtensionType type,
>-                                           unsigned char id) {
>+                                           int id) {
>   int error = 0;
>   _rtpRtcpModule->DeregisterSendRtpHeaderExtension(type);
>   if (enable) {
>-    error = _rtpRtcpModule->RegisterSendRtpHeaderExtension(type, id);
>+    // TODO(nisse): RtpRtcp::RegisterSendRtpHeaderExtension to take an int
>+    // argument. Currently it wants an uint8_t.
>+    error = _rtpRtcpModule->RegisterSendRtpHeaderExtension(
>+        type, rtc::dchecked_cast<uint8_t>(id));
>   }
>   return error;
> }
> 
>-int ChannelSend::GetRtpTimestampRateHz() const {
>-  const auto format = audio_coding_->ReceiveFormat();
>-  // Default to the playout frequency if we've not gotten any packets yet.
>-  // TODO(ossu): Zero clockrate can only happen if we've added an external
>-  // decoder for a format we don't support internally. Remove once that way of
>-  // adding decoders is gone!
>-  return (format && format->clockrate_hz != 0)
>-             ? format->clockrate_hz
>-             : audio_coding_->PlayoutFrequency();
>-}
>-
> int64_t ChannelSend::GetRTT() const {
>+  if (media_transport_) {
>+    // GetRTT is generally used in the RTCP codepath, where media transport is
>+    // not present and so it shouldn't be needed. But it's also invoked in
>+    // 'GetStats' method, and for now returning media transport RTT here gives
>+    // us "free" rtt stats for media transport.
>+    auto target_rate = media_transport_->GetLatestTargetTransferRate();
>+    if (target_rate.has_value()) {
>+      return target_rate.value().network_estimate.round_trip_time.ms();
>+    }
>+
>+    return 0;
>+  }
>   RtcpMode method = _rtpRtcpModule->RTCP();
>   if (method == RtcpMode::kOff) {
>     return 0;
>@@ -981,16 +1305,52 @@ int64_t ChannelSend::GetRTT() const {
>   return rtt;
> }
> 
>-void ChannelSend::SetFrameEncryptor(FrameEncryptorInterface* frame_encryptor) {
>+void ChannelSend::SetFrameEncryptor(
>+    rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
>+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   rtc::CritScope cs(&encoder_queue_lock_);
>   if (encoder_queue_is_active_) {
>     encoder_queue_->PostTask([this, frame_encryptor]() {
>-      this->frame_encryptor_ = frame_encryptor;
>+      this->frame_encryptor_ = std::move(frame_encryptor);
>     });
>   } else {
>-    frame_encryptor_ = frame_encryptor;
>+    frame_encryptor_ = std::move(frame_encryptor);
>   }
> }
> 
>+void ChannelSend::OnTargetTransferRate(TargetTransferRate rate) {
>+  RTC_DCHECK(media_transport_);
>+  OnReceivedRtt(rate.network_estimate.round_trip_time.ms());
>+}
>+
>+void ChannelSend::OnReceivedRtt(int64_t rtt_ms) {
>+  // Invoke audio encoders OnReceivedRtt().
>+  audio_coding_->ModifyEncoder(
>+      [rtt_ms](std::unique_ptr<AudioEncoder>* encoder) {
>+        if (*encoder) {
>+          (*encoder)->OnReceivedRtt(rtt_ms);
>+        }
>+      });
>+}
>+
>+}  // namespace
>+
>+std::unique_ptr<ChannelSendInterface> CreateChannelSend(
>+    rtc::TaskQueue* encoder_queue,
>+    ProcessThread* module_process_thread,
>+    MediaTransportInterface* media_transport,
>+    Transport* rtp_transport,
>+    RtcpRttStats* rtcp_rtt_stats,
>+    RtcEventLog* rtc_event_log,
>+    FrameEncryptorInterface* frame_encryptor,
>+    const webrtc::CryptoOptions& crypto_options,
>+    bool extmap_allow_mixed,
>+    int rtcp_report_interval_ms) {
>+  return absl::make_unique<ChannelSend>(
>+      encoder_queue, module_process_thread, media_transport, rtp_transport,
>+      rtcp_rtt_stats, rtc_event_log, frame_encryptor, crypto_options,
>+      extmap_allow_mixed, rtcp_report_interval_ms);
>+}
>+
> }  // namespace voe
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send.h
>index ef92f8e84eb60f08fbb27209fa796a5628e3d82a..083e9a60293067bf60fe2ab8675658ebc648b11b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send.h
>@@ -11,42 +11,26 @@
> #ifndef AUDIO_CHANNEL_SEND_H_
> #define AUDIO_CHANNEL_SEND_H_
> 
>-#include <map>
> #include <memory>
> #include <string>
> #include <vector>
> 
> #include "api/audio/audio_frame.h"
> #include "api/audio_codecs/audio_encoder.h"
>-#include "api/call/transport.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/audio_coding/include/audio_coding_module.h"
>-#include "modules/audio_processing/rms_level.h"
>+#include "api/crypto/cryptooptions.h"
>+#include "api/media_transport_interface.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
>-#include "rtc_base/criticalsection.h"
>+#include "rtc_base/function_view.h"
> #include "rtc_base/task_queue.h"
>-#include "rtc_base/thread_checker.h"
>-
>-// TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence
>-// warnings about use of unsigned short, and non-const reference arguments.
>-// These need cleanup, in a separate cl.
>-
>-namespace rtc {
>-class TimestampWrapAroundHandler;
>-}
> 
> namespace webrtc {
> 
> class FrameEncryptorInterface;
>-class PacketRouter;
> class ProcessThread;
>-class RateLimiter;
> class RtcEventLog;
> class RtpRtcp;
> class RtpTransportControllerSendInterface;
> 
>-struct SenderInfo;
>-
> struct CallSendStatistics {
>   int64_t rttMs;
>   size_t bytesSent;
>@@ -67,239 +51,77 @@ struct ReportBlock {
> 
> namespace voe {
> 
>-class RtpPacketSenderProxy;
>-class TransportFeedbackProxy;
>-class TransportSequenceNumberProxy;
>-class VoERtcpObserver;
>-
>-// Helper class to simplify locking scheme for members that are accessed from
>-// multiple threads.
>-// Example: a member can be set on thread T1 and read by an internal audio
>-// thread T2. Accessing the member via this class ensures that we are
>-// safe and also avoid TSan v2 warnings.
>-class ChannelSendState {
>- public:
>-  struct State {
>-    bool sending = false;
>-  };
>-
>-  ChannelSendState() {}
>-  virtual ~ChannelSendState() {}
>-
>-  void Reset() {
>-    rtc::CritScope lock(&lock_);
>-    state_ = State();
>-  }
>-
>-  State Get() const {
>-    rtc::CritScope lock(&lock_);
>-    return state_;
>-  }
>-
>-  void SetSending(bool enable) {
>-    rtc::CritScope lock(&lock_);
>-    state_.sending = enable;
>-  }
>-
>- private:
>-  rtc::CriticalSection lock_;
>-  State state_;
>-};
>-
>-class ChannelSend
>-    : public Transport,
>-      public AudioPacketizationCallback,  // receive encoded packets from the
>-                                          // ACM
>-      public OverheadObserver {
>+class ChannelSendInterface {
>  public:
>-  // TODO(nisse): Make OnUplinkPacketLossRate public, and delete friend
>-  // declaration.
>-  friend class VoERtcpObserver;
>-
>-  ChannelSend(rtc::TaskQueue* encoder_queue,
>-              ProcessThread* module_process_thread,
>-              RtcpRttStats* rtcp_rtt_stats,
>-              RtcEventLog* rtc_event_log,
>-              FrameEncryptorInterface* frame_encryptor);
>-
>-  virtual ~ChannelSend();
>+  virtual ~ChannelSendInterface() = default;
> 
>-  // Send using this encoder, with this payload type.
>-  bool SetEncoder(int payload_type, std::unique_ptr<AudioEncoder> encoder);
>-  void ModifyEncoder(
>-      rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
>+  virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length) = 0;
> 
>-  // API methods
>+  virtual CallSendStatistics GetRTCPStatistics() const = 0;
> 
>-  // VoEBase
>-  int32_t StartSend();
>-  void StopSend();
>+  virtual bool SetEncoder(int payload_type,
>+                          std::unique_ptr<AudioEncoder> encoder) = 0;
>+  virtual void ModifyEncoder(
>+      rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0;
> 
>-  // Codecs
>-  void SetBitRate(int bitrate_bps, int64_t probing_interval_ms);
>-  bool EnableAudioNetworkAdaptor(const std::string& config_string);
>-  void DisableAudioNetworkAdaptor();
>-
>-  // TODO(nisse): Modifies decoder, but not used?
>-  void SetReceiverFrameLengthRange(int min_frame_length_ms,
>-                                   int max_frame_length_ms);
>-
>-  // Network
>-  void RegisterTransport(Transport* transport);
>-  // TODO(nisse, solenberg): Delete when VoENetwork is deleted.
>-  int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
>-
>-  // Muting, Volume and Level.
>-  void SetInputMute(bool enable);
>-
>-  // Stats.
>-  ANAStats GetANAStatistics() const;
>-
>-  // Used by AudioSendStream.
>-  RtpRtcp* GetRtpRtcp() const;
>-
>-  // DTMF.
>-  int SendTelephoneEventOutband(int event, int duration_ms);
>-  int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
>-
>-  // RTP+RTCP
>-  int SetLocalSSRC(unsigned int ssrc);
>-
>-  void SetMid(const std::string& mid, int extension_id);
>-  int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
>-  void EnableSendTransportSequenceNumber(int id);
>-
>-  void RegisterSenderCongestionControlObjects(
>+  virtual void SetLocalSSRC(uint32_t ssrc) = 0;
>+  virtual void SetMid(const std::string& mid, int extension_id) = 0;
>+  virtual void SetRTCP_CNAME(absl::string_view c_name) = 0;
>+  virtual void SetExtmapAllowMixed(bool extmap_allow_mixed) = 0;
>+  virtual void SetSendAudioLevelIndicationStatus(bool enable, int id) = 0;
>+  virtual void EnableSendTransportSequenceNumber(int id) = 0;
>+  virtual void RegisterSenderCongestionControlObjects(
>       RtpTransportControllerSendInterface* transport,
>-      RtcpBandwidthObserver* bandwidth_observer);
>-  void ResetSenderCongestionControlObjects();
>-  void SetRTCPStatus(bool enable);
>-  int SetRTCP_CNAME(const char cName[256]);
>-  int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
>-  int GetRTPStatistics(CallSendStatistics& stats);  // NOLINT
>-  void SetNACKStatus(bool enable, int maxNumberOfPackets);
>-
>-  // From AudioPacketizationCallback in the ACM
>-  int32_t SendData(FrameType frameType,
>-                   uint8_t payloadType,
>-                   uint32_t timeStamp,
>-                   const uint8_t* payloadData,
>-                   size_t payloadSize,
>-                   const RTPFragmentationHeader* fragmentation) override;
>-
>-  // From Transport (called by the RTP/RTCP module)
>-  bool SendRtp(const uint8_t* data,
>-               size_t len,
>-               const PacketOptions& packet_options) override;
>-  bool SendRtcp(const uint8_t* data, size_t len) override;
>-
>-  int PreferredSampleRate() const;
>-
>-  bool Sending() const { return channel_state_.Get().sending; }
>-  RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
>-
>-  // ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
>-  // which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
>-  // the actual processing of the audio takes place. The processing mainly
>-  // consists of encoding and preparing the result for sending by adding it to a
>-  // send queue.
>-  // The main reason for using a task queue here is to release the native,
>-  // OS-specific, audio capture thread as soon as possible to ensure that it
>-  // can go back to sleep and be prepared to deliver an new captured audio
>-  // packet.
>-  void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame);
>-
>-  void SetTransportOverhead(size_t transport_overhead_per_packet);
>-
>-  // From OverheadObserver in the RTP/RTCP module
>-  void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
>-
>-  // The existence of this function alongside OnUplinkPacketLossRate is
>-  // a compromise. We want the encoder to be agnostic of the PLR source, but
>-  // we also don't want it to receive conflicting information from TWCC and
>-  // from RTCP-XR.
>-  void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
>-
>-  void OnRecoverableUplinkPacketLossRate(float recoverable_packet_loss_rate);
>-
>-  int64_t GetRTT() const;
>-
>-  // E2EE Custom Audio Frame Encryption
>-  void SetFrameEncryptor(FrameEncryptorInterface* frame_encryptor);
>-
>- private:
>-  class ProcessAndEncodeAudioTask;
>-
>-  void Init();
>-  void Terminate();
>-
>-  void OnUplinkPacketLossRate(float packet_loss_rate);
>-  bool InputMute() const;
>-
>-  int ResendPackets(const uint16_t* sequence_numbers, int length);
>-
>-  int SetSendRtpHeaderExtension(bool enable,
>-                                RTPExtensionType type,
>-                                unsigned char id);
>-
>-  void UpdateOverheadForEncoder()
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_);
>-
>-  int GetRtpTimestampRateHz() const;
>-
>-  // Called on the encoder task queue when a new input audio frame is ready
>-  // for encoding.
>-  void ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input);
>-
>-  rtc::CriticalSection _callbackCritSect;
>-  rtc::CriticalSection volume_settings_critsect_;
>-
>-  ChannelSendState channel_state_;
>-
>-  RtcEventLog* const event_log_;
>-
>-  std::unique_ptr<RtpRtcp> _rtpRtcpModule;
>-
>-  std::unique_ptr<AudioCodingModule> audio_coding_;
>-  uint32_t _timeStamp RTC_GUARDED_BY(encoder_queue_);
>-
>-  uint16_t send_sequence_number_;
>-
>-  // uses
>-  ProcessThread* _moduleProcessThreadPtr;
>-  Transport* _transportPtr;  // WebRtc socket or external transport
>-  RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
>-  bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);
>-  bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_);
>-  // VoeRTP_RTCP
>-  // TODO(henrika): can today be accessed on the main thread and on the
>-  // task queue; hence potential race.
>-  bool _includeAudioLevelIndication;
>-  size_t transport_overhead_per_packet_
>-      RTC_GUARDED_BY(overhead_per_packet_lock_);
>-  size_t rtp_overhead_per_packet_ RTC_GUARDED_BY(overhead_per_packet_lock_);
>-  rtc::CriticalSection overhead_per_packet_lock_;
>-  // RtcpBandwidthObserver
>-  std::unique_ptr<VoERtcpObserver> rtcp_observer_;
>-
>-  PacketRouter* packet_router_ = nullptr;
>-  std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
>-  std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
>-  std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
>-  std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
>-
>-  rtc::ThreadChecker construction_thread_;
>-
>-  const bool use_twcc_plr_for_ana_;
>-
>-  rtc::CriticalSection encoder_queue_lock_;
>-  bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false;
>-  rtc::TaskQueue* encoder_queue_ = nullptr;
>-
>-  // E2EE Audio Frame Encryption
>-  FrameEncryptorInterface* frame_encryptor_ = nullptr;
>+      RtcpBandwidthObserver* bandwidth_observer) = 0;
>+  virtual void ResetSenderCongestionControlObjects() = 0;
>+  virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const = 0;
>+  virtual ANAStats GetANAStatistics() const = 0;
>+  virtual bool SetSendTelephoneEventPayloadType(int payload_type,
>+                                                int payload_frequency) = 0;
>+  virtual bool SendTelephoneEventOutband(int event, int duration_ms) = 0;
>+  virtual void OnBitrateAllocation(BitrateAllocationUpdate update) = 0;
>+  virtual int GetBitrate() const = 0;
>+  virtual void SetInputMute(bool muted) = 0;
>+
>+  virtual void ProcessAndEncodeAudio(
>+      std::unique_ptr<AudioFrame> audio_frame) = 0;
>+  virtual void SetTransportOverhead(size_t transport_overhead_per_packet) = 0;
>+  virtual RtpRtcp* GetRtpRtcp() const = 0;
>+
>+  virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) = 0;
>+  virtual void OnRecoverableUplinkPacketLossRate(
>+      float recoverable_packet_loss_rate) = 0;
>+  // In RTP we currently rely on RTCP packets (|ReceivedRTCPPacket|) to inform
>+  // about RTT.
>+  // In media transport we rely on the TargetTransferRateObserver instead.
>+  // In other words, if you are using RTP, you should expect
>+  // |ReceivedRTCPPacket| to be called, if you are using media transport,
>+  // |OnTargetTransferRate| will be called.
>+  //
>+  // In future, RTP media will move to the media transport implementation and
>+  // these conditions will be removed.
>+  // Returns the RTT in milliseconds.
>+  virtual int64_t GetRTT() const = 0;
>+  virtual void StartSend() = 0;
>+  virtual void StopSend() = 0;
>+
>+  // E2EE Custom Audio Frame Encryption (Optional)
>+  virtual void SetFrameEncryptor(
>+      rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) = 0;
> };
> 
>+std::unique_ptr<ChannelSendInterface> CreateChannelSend(
>+    rtc::TaskQueue* encoder_queue,
>+    ProcessThread* module_process_thread,
>+    MediaTransportInterface* media_transport,
>+    Transport* rtp_transport,
>+    RtcpRttStats* rtcp_rtt_stats,
>+    RtcEventLog* rtc_event_log,
>+    FrameEncryptorInterface* frame_encryptor,
>+    const webrtc::CryptoOptions& crypto_options,
>+    bool extmap_allow_mixed,
>+    int rtcp_report_interval_ms);
>+
> }  // namespace voe
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send_proxy.cc
>deleted file mode 100644
>index 8091bdc671beeb89350a6a83aac294f126039884..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send_proxy.cc
>+++ /dev/null
>@@ -1,207 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "audio/channel_send_proxy.h"
>-
>-#include <utility>
>-
>-#include "api/call/audio_sink.h"
>-#include "call/rtp_transport_controller_send_interface.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/numerics/safe_minmax.h"
>-
>-namespace webrtc {
>-namespace voe {
>-ChannelSendProxy::ChannelSendProxy() {}
>-
>-ChannelSendProxy::ChannelSendProxy(std::unique_ptr<ChannelSend> channel)
>-    : channel_(std::move(channel)) {
>-  RTC_DCHECK(channel_);
>-  module_process_thread_checker_.DetachFromThread();
>-}
>-
>-ChannelSendProxy::~ChannelSendProxy() {}
>-
>-void ChannelSendProxy::SetLocalSSRC(uint32_t ssrc) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  int error = channel_->SetLocalSSRC(ssrc);
>-  RTC_DCHECK_EQ(0, error);
>-}
>-
>-void ChannelSendProxy::SetNACKStatus(bool enable, int max_packets) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetNACKStatus(enable, max_packets);
>-}
>-
>-CallSendStatistics ChannelSendProxy::GetRTCPStatistics() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  CallSendStatistics stats = {0};
>-  int error = channel_->GetRTPStatistics(stats);
>-  RTC_DCHECK_EQ(0, error);
>-  return stats;
>-}
>-
>-void ChannelSendProxy::RegisterTransport(Transport* transport) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->RegisterTransport(transport);
>-}
>-
>-bool ChannelSendProxy::ReceivedRTCPPacket(const uint8_t* packet,
>-                                          size_t length) {
>-  // May be called on either worker thread or network thread.
>-  return channel_->ReceivedRTCPPacket(packet, length) == 0;
>-}
>-
>-bool ChannelSendProxy::SetEncoder(int payload_type,
>-                                  std::unique_ptr<AudioEncoder> encoder) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return channel_->SetEncoder(payload_type, std::move(encoder));
>-}
>-
>-void ChannelSendProxy::ModifyEncoder(
>-    rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->ModifyEncoder(modifier);
>-}
>-
>-void ChannelSendProxy::SetRTCPStatus(bool enable) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetRTCPStatus(enable);
>-}
>-
>-void ChannelSendProxy::SetMid(const std::string& mid, int extension_id) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetMid(mid, extension_id);
>-}
>-
>-void ChannelSendProxy::SetRTCP_CNAME(const std::string& c_name) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  // Note: VoERTP_RTCP::SetRTCP_CNAME() accepts a char[256] array.
>-  std::string c_name_limited = c_name.substr(0, 255);
>-  int error = channel_->SetRTCP_CNAME(c_name_limited.c_str());
>-  RTC_DCHECK_EQ(0, error);
>-}
>-
>-void ChannelSendProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  int error = channel_->SetSendAudioLevelIndicationStatus(enable, id);
>-  RTC_DCHECK_EQ(0, error);
>-}
>-
>-void ChannelSendProxy::EnableSendTransportSequenceNumber(int id) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->EnableSendTransportSequenceNumber(id);
>-}
>-
>-void ChannelSendProxy::RegisterSenderCongestionControlObjects(
>-    RtpTransportControllerSendInterface* transport,
>-    RtcpBandwidthObserver* bandwidth_observer) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->RegisterSenderCongestionControlObjects(transport,
>-                                                   bandwidth_observer);
>-}
>-
>-void ChannelSendProxy::ResetSenderCongestionControlObjects() {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->ResetSenderCongestionControlObjects();
>-}
>-
>-std::vector<ReportBlock> ChannelSendProxy::GetRemoteRTCPReportBlocks() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  std::vector<webrtc::ReportBlock> blocks;
>-  int error = channel_->GetRemoteRTCPReportBlocks(&blocks);
>-  RTC_DCHECK_EQ(0, error);
>-  return blocks;
>-}
>-
>-ANAStats ChannelSendProxy::GetANAStatistics() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return channel_->GetANAStatistics();
>-}
>-
>-bool ChannelSendProxy::SetSendTelephoneEventPayloadType(int payload_type,
>-                                                        int payload_frequency) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return channel_->SetSendTelephoneEventPayloadType(payload_type,
>-                                                    payload_frequency) == 0;
>-}
>-
>-bool ChannelSendProxy::SendTelephoneEventOutband(int event, int duration_ms) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return channel_->SendTelephoneEventOutband(event, duration_ms) == 0;
>-}
>-
>-void ChannelSendProxy::SetBitrate(int bitrate_bps,
>-                                  int64_t probing_interval_ms) {
>-  // This method can be called on the worker thread, module process thread
>-  // or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged.
>-  // TODO(solenberg): Figure out a good way to check this or enforce calling
>-  // rules.
>-  // RTC_DCHECK(worker_thread_checker_.CalledOnValidThread() ||
>-  //            module_process_thread_checker_.CalledOnValidThread());
>-  channel_->SetBitRate(bitrate_bps, probing_interval_ms);
>-}
>-
>-void ChannelSendProxy::SetInputMute(bool muted) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetInputMute(muted);
>-}
>-
>-void ChannelSendProxy::ProcessAndEncodeAudio(
>-    std::unique_ptr<AudioFrame> audio_frame) {
>-  RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
>-  return channel_->ProcessAndEncodeAudio(std::move(audio_frame));
>-}
>-
>-void ChannelSendProxy::SetTransportOverhead(int transport_overhead_per_packet) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetTransportOverhead(transport_overhead_per_packet);
>-}
>-
>-RtpRtcp* ChannelSendProxy::GetRtpRtcp() const {
>-  RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
>-  return channel_->GetRtpRtcp();
>-}
>-
>-void ChannelSendProxy::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->OnTwccBasedUplinkPacketLossRate(packet_loss_rate);
>-}
>-
>-void ChannelSendProxy::OnRecoverableUplinkPacketLossRate(
>-    float recoverable_packet_loss_rate) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->OnRecoverableUplinkPacketLossRate(recoverable_packet_loss_rate);
>-}
>-
>-void ChannelSendProxy::StartSend() {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  int error = channel_->StartSend();
>-  RTC_DCHECK_EQ(0, error);
>-}
>-
>-void ChannelSendProxy::StopSend() {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->StopSend();
>-}
>-
>-ChannelSend* ChannelSendProxy::GetChannel() const {
>-  return channel_.get();
>-}
>-
>-void ChannelSendProxy::SetFrameEncryptor(
>-    FrameEncryptorInterface* frame_encryptor) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetFrameEncryptor(frame_encryptor);
>-}
>-
>-}  // namespace voe
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send_proxy.h
>deleted file mode 100644
>index 1b8b4a02ce5eecd8d69c886e52081efedeb9ab56..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_send_proxy.h
>+++ /dev/null
>@@ -1,111 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef AUDIO_CHANNEL_SEND_PROXY_H_
>-#define AUDIO_CHANNEL_SEND_PROXY_H_
>-
>-#include <memory>
>-#include <string>
>-#include <vector>
>-
>-#include "api/audio_codecs/audio_encoder.h"
>-#include "audio/channel_send.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/race_checker.h"
>-#include "rtc_base/thread_checker.h"
>-
>-namespace webrtc {
>-
>-class FrameEncryptorInterface;
>-class RtcpBandwidthObserver;
>-class RtpRtcp;
>-class RtpTransportControllerSendInterface;
>-class Transport;
>-
>-namespace voe {
>-
>-// This class provides the "view" of a voe::Channel that we need to implement
>-// webrtc::AudioSendStream. It serves two purposes:
>-//  1. Allow mocking just the interfaces used, instead of the entire
>-//     voe::Channel class.
>-//  2. Provide a refined interface for the stream classes, including assumptions
>-//     on return values and input adaptation.
>-class ChannelSendProxy {
>- public:
>-  ChannelSendProxy();
>-  explicit ChannelSendProxy(std::unique_ptr<ChannelSend> channel);
>-  virtual ~ChannelSendProxy();
>-
>-  // Shared with ChannelReceiveProxy
>-  virtual void SetLocalSSRC(uint32_t ssrc);
>-  virtual void SetNACKStatus(bool enable, int max_packets);
>-  virtual CallSendStatistics GetRTCPStatistics() const;
>-  virtual void RegisterTransport(Transport* transport);
>-  virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
>-
>-  virtual bool SetEncoder(int payload_type,
>-                          std::unique_ptr<AudioEncoder> encoder);
>-  virtual void ModifyEncoder(
>-      rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
>-
>-  virtual void SetRTCPStatus(bool enable);
>-  virtual void SetMid(const std::string& mid, int extension_id);
>-  virtual void SetRTCP_CNAME(const std::string& c_name);
>-  virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
>-  virtual void EnableSendTransportSequenceNumber(int id);
>-  virtual void RegisterSenderCongestionControlObjects(
>-      RtpTransportControllerSendInterface* transport,
>-      RtcpBandwidthObserver* bandwidth_observer);
>-  virtual void ResetSenderCongestionControlObjects();
>-  virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
>-  virtual ANAStats GetANAStatistics() const;
>-  virtual bool SetSendTelephoneEventPayloadType(int payload_type,
>-                                                int payload_frequency);
>-  virtual bool SendTelephoneEventOutband(int event, int duration_ms);
>-  virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
>-  virtual void SetInputMute(bool muted);
>-
>-  virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame);
>-  virtual void SetTransportOverhead(int transport_overhead_per_packet);
>-  virtual RtpRtcp* GetRtpRtcp() const;
>-
>-  virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
>-  virtual void OnRecoverableUplinkPacketLossRate(
>-      float recoverable_packet_loss_rate);
>-  virtual void StartSend();
>-  virtual void StopSend();
>-
>-  // Needed by ChannelReceiveProxy::AssociateSendChannel.
>-  virtual ChannelSend* GetChannel() const;
>-
>-  // E2EE Custom Audio Frame Encryption (Optional)
>-  virtual void SetFrameEncryptor(FrameEncryptorInterface* frame_encryptor);
>-
>- private:
>-  // Thread checkers document and lock usage of some methods on voe::Channel to
>-  // specific threads we know about. The goal is to eventually split up
>-  // voe::Channel into parts with single-threaded semantics, and thereby reduce
>-  // the need for locks.
>-  rtc::ThreadChecker worker_thread_checker_;
>-  rtc::ThreadChecker module_process_thread_checker_;
>-  // Methods accessed from audio and video threads are checked for sequential-
>-  // only access. We don't necessarily own and control these threads, so thread
>-  // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
>-  // audio thread to another, but access is still sequential.
>-  rtc::RaceChecker audio_thread_race_checker_;
>-  rtc::RaceChecker video_capture_thread_race_checker_;
>-  std::unique_ptr<ChannelSend> channel_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelSendProxy);
>-};
>-}  // namespace voe
>-}  // namespace webrtc
>-
>-#endif  // AUDIO_CHANNEL_SEND_PROXY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/mock_voe_channel_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/mock_voe_channel_proxy.h
>index 910858fe130cac8d94f84cb0c59b8b1aff5e2f68..eee25c5a2d532cb245d99cf4dd4873e974392401 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/mock_voe_channel_proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/mock_voe_channel_proxy.h
>@@ -16,15 +16,16 @@
> #include <string>
> #include <vector>
> 
>-#include "audio/channel_receive_proxy.h"
>-#include "audio/channel_send_proxy.h"
>+#include "api/test/mock_frame_encryptor.h"
>+#include "audio/channel_receive.h"
>+#include "audio/channel_send.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "test/gmock.h"
> 
> namespace webrtc {
> namespace test {
> 
>-class MockChannelReceiveProxy : public voe::ChannelReceiveProxy {
>+class MockChannelReceive : public voe::ChannelReceiveInterface {
>  public:
>   MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
>   MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
>@@ -46,10 +47,10 @@ class MockChannelReceiveProxy : public voe::ChannelReceiveProxy {
>                AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
>                                                   AudioFrame* audio_frame));
>   MOCK_CONST_METHOD0(PreferredSampleRate, int());
>-  MOCK_METHOD1(AssociateSendChannel,
>-               void(const voe::ChannelSendProxy& send_channel_proxy));
>-  MOCK_METHOD0(DisassociateSendChannel, void());
>+  MOCK_METHOD1(SetAssociatedSendChannel,
>+               void(const voe::ChannelSendInterface* send_channel));
>   MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
>+  MOCK_CONST_METHOD0(GetSyncInfo, absl::optional<Syncable::Info>());
>   MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
>   MOCK_CONST_METHOD1(GetRecCodec, bool(CodecInst* codec_inst));
>   MOCK_METHOD1(SetReceiveCodecs,
>@@ -59,7 +60,7 @@ class MockChannelReceiveProxy : public voe::ChannelReceiveProxy {
>   MOCK_METHOD0(StopPlayout, void());
> };
> 
>-class MockChannelSendProxy : public voe::ChannelSendProxy {
>+class MockChannelSend : public voe::ChannelSendInterface {
>  public:
>   // GMock doesn't like move-only types, like std::unique_ptr.
>   virtual bool SetEncoder(int payload_type,
>@@ -71,10 +72,10 @@ class MockChannelSendProxy : public voe::ChannelSendProxy {
>   MOCK_METHOD1(
>       ModifyEncoder,
>       void(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier));
>-  MOCK_METHOD1(SetRTCPStatus, void(bool enable));
>+  MOCK_METHOD2(SetMid, void(const std::string& mid, int extension_id));
>   MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
>-  MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
>-  MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
>+  MOCK_METHOD1(SetRTCP_CNAME, void(absl::string_view c_name));
>+  MOCK_METHOD1(SetExtmapAllowMixed, void(bool extmap_allow_mixed));
>   MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
>   MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
>   MOCK_METHOD2(RegisterSenderCongestionControlObjects,
>@@ -87,9 +88,8 @@ class MockChannelSendProxy : public voe::ChannelSendProxy {
>   MOCK_METHOD2(SetSendTelephoneEventPayloadType,
>                bool(int payload_type, int payload_frequency));
>   MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
>-  MOCK_METHOD2(SetBitrate, void(int bitrate_bps, int64_t probing_interval_ms));
>+  MOCK_METHOD1(OnBitrateAllocation, void(BitrateAllocationUpdate update));
>   MOCK_METHOD1(SetInputMute, void(bool muted));
>-  MOCK_METHOD1(RegisterTransport, void(Transport* transport));
>   MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
>   // GMock doesn't like move-only types, like std::unique_ptr.
>   virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) {
>@@ -97,15 +97,19 @@ class MockChannelSendProxy : public voe::ChannelSendProxy {
>   }
>   MOCK_METHOD1(ProcessAndEncodeAudioForMock,
>                void(std::unique_ptr<AudioFrame>* audio_frame));
>-  MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
>+  MOCK_METHOD1(SetTransportOverhead,
>+               void(size_t transport_overhead_per_packet));
>   MOCK_CONST_METHOD0(GetRtpRtcp, RtpRtcp*());
>+  MOCK_CONST_METHOD0(GetBitrate, int());
>   MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate));
>   MOCK_METHOD1(OnRecoverableUplinkPacketLossRate,
>                void(float recoverable_packet_loss_rate));
>+  MOCK_CONST_METHOD0(GetRTT, int64_t());
>   MOCK_METHOD0(StartSend, void());
>   MOCK_METHOD0(StopSend, void());
>-  MOCK_METHOD1(SetFrameEncryptor,
>-               void(FrameEncryptorInterface* frame_encryptor));
>+  MOCK_METHOD1(
>+      SetFrameEncryptor,
>+      void(rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor));
> };
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/null_audio_poller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/null_audio_poller.cc
>index c22b3d879115811ce6d0d1d274db1bfd3718c27d..d2b1199ed6dd27d58d3d7e486728e49f3c318c5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/null_audio_poller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/null_audio_poller.cc
>@@ -9,8 +9,13 @@
>  */
> 
> #include "audio/null_audio_poller.h"
>-#include "rtc_base/logging.h"
>+
>+#include <stddef.h>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/location.h"
> #include "rtc_base/thread.h"
>+#include "rtc_base/timeutils.h"
> 
> namespace webrtc {
> namespace internal {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/null_audio_poller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/null_audio_poller.h
>index afb6edbaf16132f787088acc6f366bd1cb624708..f91eb7d6c0782448cc9a7a6943b5542bf3b7c11e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/null_audio_poller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/null_audio_poller.h
>@@ -11,8 +11,11 @@
> #ifndef AUDIO_NULL_AUDIO_POLLER_H_
> #define AUDIO_NULL_AUDIO_POLLER_H_
> 
>+#include <stdint.h>
>+
> #include "modules/audio_device/include/audio_device_defines.h"
> #include "rtc_base/messagehandler.h"
>+#include "rtc_base/messagequeue.h"
> #include "rtc_base/thread_checker.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>index 97222e947a4e62f345c7304d7b692af9c0da5b64..b1158da213f966b44340d6220fc9e04f662a7ebc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>@@ -13,10 +13,7 @@
> #include "api/audio/audio_frame.h"
> #include "audio/utility/audio_frame_operations.h"
> #include "common_audio/resampler/include/push_resampler.h"
>-#include "common_audio/signal_processing/include/signal_processing_library.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
> 
> namespace webrtc {
> namespace voe {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.cc
>index 8ec2a7cf4a574e153caa143ee647f4133fc9ee92..9fa95b17b8e46d94eebd641c37c02261ccf80d47 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.cc
>@@ -123,8 +123,8 @@ class NoBandwidthDropAfterDtx : public AudioBweTest {
>     return test::ResourcePath("voice_engine/audio_dtx16", "wav");
>   }
> 
>-  DefaultNetworkSimulationConfig GetNetworkPipeConfig() override {
>-    DefaultNetworkSimulationConfig pipe_config;
>+  BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() override {
>+    BuiltInNetworkBehaviorConfig pipe_config;
>     pipe_config.link_capacity_kbps = 50;
>     pipe_config.queue_length_packets = 1500;
>     pipe_config.queue_delay_ms = 300;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.h
>index c70ad1c91b93f771b580f460aff631e2a6cd6b1a..4de2cab31613bec35968111a10dc55e3b995a7d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.h
>@@ -27,7 +27,7 @@ class AudioBweTest : public test::EndToEndTest {
>  protected:
>   virtual std::string AudioInputFile() = 0;
> 
>-  virtual DefaultNetworkSimulationConfig GetNetworkPipeConfig() = 0;
>+  virtual BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() = 0;
> 
>   size_t GetNumVideoStreams() const override;
>   size_t GetNumAudioStreams() const override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.cc
>index 02abe734bbfe1f2ec688b402ca122e2e5d48d596..5e187ad8f8980912935ad6a0bfa90cc09b332e45 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.cc
>@@ -28,8 +28,8 @@ constexpr int kSampleRate = 48000;
> AudioEndToEndTest::AudioEndToEndTest()
>     : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
> 
>-DefaultNetworkSimulationConfig AudioEndToEndTest::GetNetworkPipeConfig() const {
>-  return DefaultNetworkSimulationConfig();
>+BuiltInNetworkBehaviorConfig AudioEndToEndTest::GetNetworkPipeConfig() const {
>+  return BuiltInNetworkBehaviorConfig();
> }
> 
> size_t AudioEndToEndTest::GetNumVideoStreams() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.h
>index 9dda834d22b4ed6e5ee2a0f2f48723cf4c00c002..ba1e0c7d6ef8eec770f3bcdd813bfb22745a4802 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.h
>@@ -29,7 +29,7 @@ class AudioEndToEndTest : public test::EndToEndTest {
>   const AudioSendStream* send_stream() const { return send_stream_; }
>   const AudioReceiveStream* receive_stream() const { return receive_stream_; }
> 
>-  virtual DefaultNetworkSimulationConfig GetNetworkPipeConfig() const;
>+  virtual BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() const;
> 
>   size_t GetNumVideoStreams() const override;
>   size_t GetNumAudioStreams() const override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_stats_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_stats_test.cc
>index 019679e0782f86b0b76a4b9031eebc793a3e2349..556a16dfa6d505633f9972d09652bcc17f019bad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_stats_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_stats_test.cc
>@@ -32,8 +32,8 @@ class NoLossTest : public AudioEndToEndTest {
> 
>   NoLossTest() = default;
> 
>-  DefaultNetworkSimulationConfig GetNetworkPipeConfig() const override {
>-    DefaultNetworkSimulationConfig pipe_config;
>+  BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() const override {
>+    BuiltInNetworkBehaviorConfig pipe_config;
>     pipe_config.queue_delay_ms = kRttMs / 2;
>     return pipe_config;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.cc
>index 16a3b91e877cdd88d6f93f8ed6f22b9bc92ab437..857f983d57775a1a47b10f80980d68a9e8de25df 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.cc
>@@ -14,14 +14,15 @@
> #include "system_wrappers/include/sleep.h"
> #include "test/testsupport/fileutils.h"
> 
>-DEFINE_int(sample_rate_hz,
>-           16000,
>-           "Sample rate (Hz) of the produced audio files.");
>+WEBRTC_DEFINE_int(sample_rate_hz,
>+                  16000,
>+                  "Sample rate (Hz) of the produced audio files.");
> 
>-DEFINE_bool(quick,
>-            false,
>-            "Don't do the full audio recording. "
>-            "Used to quickly check that the test runs without crashing.");
>+WEBRTC_DEFINE_bool(
>+    quick,
>+    false,
>+    "Don't do the full audio recording. "
>+    "Used to quickly check that the test runs without crashing.");
> 
> namespace webrtc {
> namespace test {
>@@ -89,8 +90,8 @@ class Mobile2GNetworkTest : public AudioQualityTest {
>          {{"maxaveragebitrate", "6000"}, {"ptime", "60"}, {"stereo", "1"}}});
>   }
> 
>-  DefaultNetworkSimulationConfig GetNetworkPipeConfig() const override {
>-    DefaultNetworkSimulationConfig pipe_config;
>+  BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() const override {
>+    BuiltInNetworkBehaviorConfig pipe_config;
>     pipe_config.link_capacity_kbps = 12;
>     pipe_config.queue_length_packets = 1500;
>     pipe_config.queue_delay_ms = 400;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/media_transport_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/media_transport_test.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..b4af4a08819138bf696926133f04e1edee3a6020
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/media_transport_test.cc
>@@ -0,0 +1,143 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/audio_codecs/audio_decoder_factory_template.h"
>+#include "api/audio_codecs/audio_encoder_factory_template.h"
>+#include "api/audio_codecs/opus/audio_decoder_opus.h"
>+#include "api/audio_codecs/opus/audio_encoder_opus.h"
>+#include "api/test/loopback_media_transport.h"
>+#include "api/test/mock_audio_mixer.h"
>+#include "audio/audio_receive_stream.h"
>+#include "audio/audio_send_stream.h"
>+#include "call/test/mock_bitrate_allocator.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "modules/audio_device/include/test_audio_device.h"
>+#include "modules/audio_mixer/audio_mixer_impl.h"
>+#include "modules/audio_processing/include/mock_audio_processing.h"
>+#include "modules/utility/include/process_thread.h"
>+#include "rtc_base/task_queue.h"
>+#include "rtc_base/timeutils.h"
>+#include "test/gtest.h"
>+#include "test/mock_transport.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+constexpr int kPayloadTypeOpus = 17;
>+constexpr int kSamplingFrequency = 48000;
>+constexpr int kNumChannels = 2;
>+constexpr int kWantedSamples = 3000;
>+constexpr int kTestTimeoutMs = 2 * rtc::kNumMillisecsPerSec;
>+
>+class TestRenderer : public TestAudioDeviceModule::Renderer {
>+ public:
>+  TestRenderer(int sampling_frequency, int num_channels, size_t wanted_samples)
>+      : sampling_frequency_(sampling_frequency),
>+        num_channels_(num_channels),
>+        wanted_samples_(wanted_samples) {}
>+  ~TestRenderer() override = default;
>+
>+  int SamplingFrequency() const override { return sampling_frequency_; }
>+  int NumChannels() const override { return num_channels_; }
>+
>+  bool Render(rtc::ArrayView<const int16_t> data) override {
>+    if (data.size() >= wanted_samples_) {
>+      return false;
>+    }
>+    wanted_samples_ -= data.size();
>+    return true;
>+  }
>+
>+ private:
>+  const int sampling_frequency_;
>+  const int num_channels_;
>+  size_t wanted_samples_;
>+};
>+
>+}  // namespace
>+
>+TEST(AudioWithMediaTransport, DeliversAudio) {
>+  std::unique_ptr<rtc::Thread> transport_thread = rtc::Thread::Create();
>+  transport_thread->Start();
>+  MediaTransportPair transport_pair(transport_thread.get());
>+  MockTransport rtcp_send_transport;
>+  MockTransport send_transport;
>+  std::unique_ptr<RtcEventLog> null_event_log = RtcEventLog::CreateNull();
>+  MockBitrateAllocator bitrate_allocator;
>+
>+  rtc::scoped_refptr<TestAudioDeviceModule> audio_device =
>+      TestAudioDeviceModule::CreateTestAudioDeviceModule(
>+          TestAudioDeviceModule::CreatePulsedNoiseCapturer(
>+              /* max_amplitude= */ 10000, kSamplingFrequency, kNumChannels),
>+          absl::make_unique<TestRenderer>(kSamplingFrequency, kNumChannels,
>+                                          kWantedSamples));
>+
>+  AudioState::Config audio_config;
>+  audio_config.audio_mixer = AudioMixerImpl::Create();
>+  // TODO(nisse): Is a mock AudioProcessing enough?
>+  audio_config.audio_processing =
>+      new rtc::RefCountedObject<MockAudioProcessing>();
>+  audio_config.audio_device_module = audio_device;
>+  rtc::scoped_refptr<AudioState> audio_state = AudioState::Create(audio_config);
>+
>+  // TODO(nisse): Use some lossless codec?
>+  const SdpAudioFormat audio_format("opus", kSamplingFrequency, kNumChannels);
>+
>+  // Setup receive stream;
>+  webrtc::AudioReceiveStream::Config receive_config;
>+  // TODO(nisse): Update AudioReceiveStream to not require rtcp_send_transport
>+  // when a MediaTransport is provided.
>+  receive_config.rtcp_send_transport = &rtcp_send_transport;
>+  receive_config.media_transport = transport_pair.first();
>+  receive_config.decoder_map.emplace(kPayloadTypeOpus, audio_format);
>+  receive_config.decoder_factory =
>+      CreateAudioDecoderFactory<AudioDecoderOpus>();
>+
>+  std::unique_ptr<ProcessThread> receive_process_thread =
>+      ProcessThread::Create("audio recv thread");
>+
>+  webrtc::internal::AudioReceiveStream receive_stream(
>+      /*rtp_stream_receiver_controller=*/nullptr,
>+      /*packet_router=*/nullptr, receive_process_thread.get(), receive_config,
>+      audio_state, null_event_log.get());
>+
>+  // TODO(nisse): Update AudioSendStream to not require send_transport when a
>+  // MediaTransport is provided.
>+  AudioSendStream::Config send_config(&send_transport, transport_pair.second());
>+  send_config.send_codec_spec =
>+      AudioSendStream::Config::SendCodecSpec(kPayloadTypeOpus, audio_format);
>+  send_config.encoder_factory = CreateAudioEncoderFactory<AudioEncoderOpus>();
>+  rtc::TaskQueue send_tq("audio send queue");
>+  std::unique_ptr<ProcessThread> send_process_thread =
>+      ProcessThread::Create("audio send thread");
>+  webrtc::internal::AudioSendStream send_stream(
>+      send_config, audio_state, &send_tq, send_process_thread.get(),
>+      /*transport=*/nullptr, &bitrate_allocator, null_event_log.get(),
>+      /*rtcp_rtt_stats=*/nullptr, absl::optional<RtpState>());
>+
>+  audio_device->Init();  // Starts thread.
>+  audio_device->RegisterAudioCallback(audio_state->audio_transport());
>+
>+  receive_stream.Start();
>+  send_stream.Start();
>+  audio_device->StartPlayout();
>+  audio_device->StartRecording();
>+
>+  EXPECT_TRUE(audio_device->WaitForPlayoutEnd(kTestTimeoutMs));
>+
>+  audio_device->StopRecording();
>+  audio_device->StopPlayout();
>+  receive_stream.Stop();
>+  send_stream.Stop();
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval.cc
>deleted file mode 100644
>index cc103408a3a2467e5a0a02f88e390d74813c6c1d..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval.cc
>+++ /dev/null
>@@ -1,56 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "audio/time_interval.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/timeutils.h"
>-
>-namespace webrtc {
>-
>-TimeInterval::TimeInterval() = default;
>-TimeInterval::~TimeInterval() = default;
>-
>-void TimeInterval::Extend() {
>-  Extend(rtc::TimeMillis());
>-}
>-
>-void TimeInterval::Extend(int64_t time) {
>-  if (!interval_) {
>-    interval_.emplace(time, time);
>-  } else {
>-    if (time < interval_->first) {
>-      interval_->first = time;
>-    }
>-    if (time > interval_->last) {
>-      interval_->last = time;
>-    }
>-  }
>-}
>-
>-void TimeInterval::Extend(const TimeInterval& other_interval) {
>-  if (!other_interval.Empty()) {
>-    Extend(other_interval.interval_->first);
>-    Extend(other_interval.interval_->last);
>-  }
>-}
>-
>-bool TimeInterval::Empty() const {
>-  return !interval_;
>-}
>-
>-int64_t TimeInterval::Length() const {
>-  RTC_DCHECK(interval_);
>-  return interval_->last - interval_->first;
>-}
>-
>-TimeInterval::Interval::Interval(int64_t first, int64_t last)
>-    : first(first), last(last) {}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval.h
>deleted file mode 100644
>index 79fe29d9d569a1056d59131d5876e2c90b96d499..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval.h
>+++ /dev/null
>@@ -1,65 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef AUDIO_TIME_INTERVAL_H_
>-#define AUDIO_TIME_INTERVAL_H_
>-
>-#include <stdint.h>
>-
>-#include "absl/types/optional.h"
>-
>-namespace webrtc {
>-
>-// This class logs the first and last time its Extend() function is called.
>-//
>-// This class is not thread-safe; Extend() calls should only be made by a
>-// single thread at a time, such as within a lock or destructor.
>-//
>-// Example usage:
>-//   // let x < y < z < u < v
>-//   rtc::TimeInterval interval;
>-//   ...  //   interval.Extend(); // at time x
>-//   ...
>-//   interval.Extend(); // at time y
>-//   ...
>-//   interval.Extend(); // at time u
>-//   ...
>-//   interval.Extend(z); // at time v
>-//   ...
>-//   if (!interval.Empty()) {
>-//     int64_t active_time = interval.Length(); // returns (u - x)
>-//   }
>-class TimeInterval {
>- public:
>-  TimeInterval();
>-  ~TimeInterval();
>-  // Extend the interval with the current time.
>-  void Extend();
>-  // Extend the interval with a given time.
>-  void Extend(int64_t time);
>-  // Take the convex hull with another interval.
>-  void Extend(const TimeInterval& other_interval);
>-  // True iff Extend has never been called.
>-  bool Empty() const;
>-  // Returns the time between the first and the last tick, in milliseconds.
>-  int64_t Length() const;
>-
>- private:
>-  struct Interval {
>-    Interval(int64_t first, int64_t last);
>-
>-    int64_t first, last;
>-  };
>-  absl::optional<Interval> interval_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // AUDIO_TIME_INTERVAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval_unittest.cc
>deleted file mode 100644
>index deff6e363da2161dbdee680719872fb296526e97..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval_unittest.cc
>+++ /dev/null
>@@ -1,48 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "audio/time_interval.h"
>-#include "api/units/time_delta.h"
>-#include "rtc_base/fakeclock.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-
>-TEST(TimeIntervalTest, TimeInMs) {
>-  rtc::ScopedFakeClock fake_clock;
>-  TimeInterval interval;
>-  interval.Extend();
>-  fake_clock.AdvanceTime(TimeDelta::ms(100));
>-  interval.Extend();
>-  EXPECT_EQ(interval.Length(), 100);
>-}
>-
>-TEST(TimeIntervalTest, Empty) {
>-  TimeInterval interval;
>-  EXPECT_TRUE(interval.Empty());
>-  interval.Extend();
>-  EXPECT_FALSE(interval.Empty());
>-  interval.Extend(200);
>-  EXPECT_FALSE(interval.Empty());
>-}
>-
>-TEST(TimeIntervalTest, MonotoneIncreasing) {
>-  const size_t point_count = 7;
>-  const int64_t interval_points[] = {3, 2, 5, 0, 4, 1, 6};
>-  const int64_t interval_differences[] = {0, 1, 3, 5, 5, 5, 6};
>-  TimeInterval interval;
>-  EXPECT_TRUE(interval.Empty());
>-  for (size_t i = 0; i < point_count; ++i) {
>-    interval.Extend(interval_points[i]);
>-    EXPECT_EQ(interval_differences[i], interval.Length());
>-  }
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.cc
>index c7acd766c4485925b4da7a9c028137d35def9fca..f41439b494fcd38b3a3a1675b63386fafaa4de8b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.cc
>@@ -10,11 +10,11 @@
> 
> #include "audio/transport_feedback_packet_loss_tracker.h"
> 
>+#include <iterator>
> #include <limits>
> #include <utility>
> 
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/mod_ops.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/BUILD.gn
>index 76c09a51a380e1c84655afa096e84e1014f7c7eb..11a65bdd4638ad5bfbef56bf9e9341f2ecff3b86 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/BUILD.gn
>@@ -21,7 +21,6 @@ rtc_static_library("audio_frame_operations") {
>   ]
> 
>   deps = [
>-    "../..:webrtc_common",
>     "../../api/audio:audio_frame_api",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.cc
>index fb1f3b07711b79dd4bc059bc1921df28fea26bed..1a8232b02b3ae4bed9b2dcf21e8c1361eb1cd801 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.cc
>@@ -12,6 +12,7 @@
> 
> #include <string.h>
> #include <algorithm>
>+#include <cstdint>
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.h
>index 599352356f62f39346df89e0297c6c21152a6719..c1445b691098e39934affdaf9a839d142eaad523 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.h
>@@ -12,6 +12,7 @@
> #define AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
> 
> #include <stddef.h>
>+#include <stdint.h>
> 
> #include "api/audio/audio_frame.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/build_overrides/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/build_overrides/OWNERS
>index 524e2676ff52bfef38930036abfef85b26072bf4..5465ed8e6a869e1e742474024df997b40ac525bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/build_overrides/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/build_overrides/OWNERS
>@@ -1 +1,2 @@
>+mbonadei@webrtc.org
> phoglund@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/build_overrides/build.gni b/Source/ThirdParty/libwebrtc/Source/webrtc/build_overrides/build.gni
>index 81cb3e73ae2329f92de21978552a8778ab82f214..669044db81cb2185be374b0b10027093c5683c35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/build_overrides/build.gni
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/build_overrides/build.gni
>@@ -12,8 +12,8 @@ enable_java_templates = true
> # Some non-Chromium builds don't use Chromium's third_party/binutils.
> linux_use_bundled_binutils_override = true
> 
>-# Variable that can be used to support multiple build scenarios, like having
>-# Chromium specific targets in a client project's GN file etc.
>+# Don't set this variable to true when building stadalone WebRTC, it is
>+# only needed to support both WebRTC standalone and Chromium builds.
> build_with_chromium = false
> 
> # Use our own suppressions files.
>@@ -42,7 +42,7 @@ if (host_os == "mac") {
>                         [ target_os ],
>                         "value")
>   assert(_result != 2,
>-         "Do not allow building targets with the default" +
>+         "Do not allow building targets with the default " +
>              "hermetic toolchain if the minimum OS version is not met.")
>   use_system_xcode = _result == 0
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/call/BUILD.gn
>index 611d9e8adb13c7b529108fc7cfd314b08616b0dd..f6ba4ec8490829f44af38e25458d0b69889eb014 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/BUILD.gn
>@@ -38,11 +38,14 @@ rtc_source_set("call_interfaces") {
>     "../api/audio_codecs:audio_codecs_api",
>     "../api/transport:network_control",
>     "../modules/audio_device:audio_device",
>+    "../modules/audio_processing:api",
>     "../modules/audio_processing:audio_processing",
>     "../modules/audio_processing:audio_processing_statistics",
>     "../rtc_base:audio_format_to_string",
>+    "../rtc_base:checks",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/network:sent_packet",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -161,6 +164,9 @@ rtc_source_set("bitrate_allocator") {
>     "bitrate_allocator.h",
>   ]
>   deps = [
>+    "../api:bitrate_allocation",
>+    "../api/units:data_rate",
>+    "../api/units:time_delta",
>     "../modules/bitrate_controller",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>@@ -193,7 +199,6 @@ rtc_static_library("call") {
>     ":rtp_sender",
>     ":simulated_network",
>     ":video_stream_api",
>-    "..:webrtc_common",
>     "../api:callfactory_api",
>     "../api:simulated_network_api",
>     "../api:transport_api",
>@@ -217,6 +222,7 @@ rtc_static_library("call") {
>     "../rtc_base:rtc_task_queue",
>     "../rtc_base:safe_minmax",
>     "../rtc_base:sequenced_task_checker",
>+    "../rtc_base/experiments:field_trial_parser",
>     "../rtc_base/synchronization:rw_lock_wrapper",
>     "../system_wrappers",
>     "../system_wrappers:field_trial",
>@@ -236,7 +242,6 @@ rtc_source_set("video_stream_api") {
>   ]
>   deps = [
>     ":rtp_interfaces",
>-    "../:webrtc_common",
>     "../api:libjingle_peerconnection_api",
>     "../api:transport_api",
>     "../api/video:video_frame",
>@@ -257,6 +262,9 @@ rtc_source_set("simulated_network") {
>   ]
>   deps = [
>     "../api:simulated_network_api",
>+    "../api/units:data_rate",
>+    "../api/units:data_size",
>+    "../api/units:time_delta",
>     "../rtc_base:rtc_base_approved",
>     "//third_party/abseil-cpp/absl/memory",
>     "//third_party/abseil-cpp/absl/types:optional",
>@@ -283,10 +291,10 @@ rtc_source_set("fake_network") {
>     ":call_interfaces",
>     ":simulated_network",
>     ":simulated_packet_receiver",
>-    "..:webrtc_common",
>     "../api:simulated_network_api",
>     "../api:transport_api",
>     "../modules:module_api",
>+    "../modules/utility",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:sequenced_task_checker",
>     "../system_wrappers",
>@@ -322,8 +330,9 @@ if (rtc_include_tests) {
>       ":rtp_receiver",
>       ":rtp_sender",
>       ":simulated_network",
>-      "..:webrtc_common",
>       "../api:array_view",
>+      "../api:fake_media_transport",
>+      "../api:fake_media_transport",
>       "../api:libjingle_peerconnection_api",
>       "../api:mock_audio_mixer",
>       "../api/audio_codecs:builtin_audio_decoder_factory",
>@@ -359,6 +368,7 @@ if (rtc_include_tests) {
>       "//testing/gmock",
>       "//testing/gtest",
>       "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -378,12 +388,13 @@ if (rtc_include_tests) {
>       ":call_interfaces",
>       ":simulated_network",
>       ":video_stream_api",
>-      "..:webrtc_common",
>       "../api:simulated_network_api",
>       "../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../api/video:builtin_video_bitrate_allocator_factory",
>       "../api/video:video_bitrate_allocation",
>       "../api/video_codecs:video_codecs_api",
>       "../logging:rtc_event_log_api",
>+      "../logging:rtc_event_log_impl_output",
>       "../modules/audio_coding",
>       "../modules/audio_device",
>       "../modules/audio_device:audio_device_impl",
>@@ -426,6 +437,7 @@ if (rtc_include_tests) {
>       "../modules/pacing",
>       "../rtc_base:rate_limiter",
>       "../rtc_base:rtc_base",
>+      "../rtc_base/network:sent_packet",
>       "../test:test_support",
>     ]
>   }
>@@ -437,7 +449,7 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       ":bitrate_allocator",
>-      "//test:test_support",
>+      "../test:test_support",
>     ]
>   }
>   rtc_source_set("mock_call_interfaces") {
>@@ -448,11 +460,14 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       ":call_interfaces",
>-      "//test:test_support",
>+      "../test:test_support",
>     ]
>   }
> 
>   rtc_test("fake_network_unittests") {
>+    sources = [
>+      "test/fake_network_pipe_unittest.cc",
>+    ]
>     deps = [
>       ":call_interfaces",
>       ":fake_network",
>@@ -462,10 +477,8 @@ if (rtc_include_tests) {
>       "../system_wrappers",
>       "../test:test_common",
>       "../test:test_main",
>+      "../test:test_support",
>       "//testing/gtest",
>     ]
>-    sources = [
>-      "test/fake_network_pipe_unittest.cc",
>-    ]
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.h
>index 9c890b1cda375853141f84bb6dd14a799aabb1f6..36cc05939674bef1b125b51a4d311efbe28443e7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.h
>@@ -19,10 +19,11 @@
> #include "absl/types/optional.h"
> #include "api/audio_codecs/audio_decoder_factory.h"
> #include "api/call/transport.h"
>+#include "api/crypto/cryptooptions.h"
>+#include "api/media_transport_interface.h"
> #include "api/rtpparameters.h"
> #include "api/rtpreceiverinterface.h"
> #include "call/rtp_config.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/scoped_ref_ptr.h"
> 
> namespace webrtc {
>@@ -62,6 +63,7 @@ class AudioReceiveStream {
>     float secondary_discarded_rate = 0.0f;
>     float accelerate_rate = 0.0f;
>     float preemptive_expand_rate = 0.0f;
>+    uint64_t delayed_packet_outage_samples = 0;
>     int32_t decoding_calls_to_silence_generator = 0;
>     int32_t decoding_calls_to_neteq = 0;
>     int32_t decoding_normal = 0;
>@@ -70,6 +72,7 @@ class AudioReceiveStream {
>     int32_t decoding_plc_cng = 0;
>     int32_t decoding_muted_output = 0;
>     int64_t capture_start_ntp_time_ms = 0;
>+    uint64_t jitter_buffer_flushes = 0;
>   };
> 
>   struct Config {
>@@ -106,9 +109,12 @@ class AudioReceiveStream {
> 
>     Transport* rtcp_send_transport = nullptr;
> 
>+    MediaTransportInterface* media_transport = nullptr;
>+
>     // NetEq settings.
>     size_t jitter_buffer_max_packets = 50;
>     bool jitter_buffer_fast_accelerate = false;
>+    int jitter_buffer_min_delay_ms = 0;
> 
>     // Identifier for an A/V synchronization group. Empty string to disable.
>     // TODO(pbos): Synchronize streams in a sync group, not just one video
>@@ -122,6 +128,9 @@ class AudioReceiveStream {
> 
>     absl::optional<AudioCodecPairId> codec_pair_id;
> 
>+    // Per PeerConnection crypto options.
>+    webrtc::CryptoOptions crypto_options;
>+
>     // An optional custom frame decryptor that allows the entire frame to be
>     // decrypted in whatever way the caller choses. This is not required by
>     // default.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.cc
>index e2cf20cf84480e6f47ec7d33470205c30b45652c..303b49cbfb319667218785d824067b77aa8e9579 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.cc
>@@ -9,6 +9,9 @@
>  */
> 
> #include "call/audio_send_stream.h"
>+
>+#include <stddef.h>
>+
> #include "rtc_base/stringencode.h"
> #include "rtc_base/strings/audio_format_to_string.h"
> #include "rtc_base/strings/string_builder.h"
>@@ -18,8 +21,12 @@ namespace webrtc {
> AudioSendStream::Stats::Stats() = default;
> AudioSendStream::Stats::~Stats() = default;
> 
>+AudioSendStream::Config::Config(Transport* send_transport,
>+                                MediaTransportInterface* media_transport)
>+    : send_transport(send_transport), media_transport(media_transport) {}
>+
> AudioSendStream::Config::Config(Transport* send_transport)
>-    : send_transport(send_transport) {}
>+    : Config(send_transport, nullptr) {}
> 
> AudioSendStream::Config::~Config() = default;
> 
>@@ -27,7 +34,9 @@ std::string AudioSendStream::Config::ToString() const {
>   char buf[1024];
>   rtc::SimpleStringBuilder ss(buf);
>   ss << "{rtp: " << rtp.ToString();
>+  ss << ", rtcp_report_interval_ms: " << rtcp_report_interval_ms;
>   ss << ", send_transport: " << (send_transport ? "(Transport)" : "null");
>+  ss << ", media_transport: " << (media_transport ? "(Transport)" : "null");
>   ss << ", min_bitrate_bps: " << min_bitrate_bps;
>   ss << ", max_bitrate_bps: " << max_bitrate_bps;
>   ss << ", send_codec_spec: "
>@@ -44,6 +53,7 @@ std::string AudioSendStream::Config::Rtp::ToString() const {
>   char buf[1024];
>   rtc::SimpleStringBuilder ss(buf);
>   ss << "{ssrc: " << ssrc;
>+  ss << ", extmap-allow-mixed: " << (extmap_allow_mixed ? "true" : "false");
>   ss << ", extensions: [";
>   for (size_t i = 0; i < extensions.size(); ++i) {
>     ss << extensions[i].ToString();
>@@ -52,7 +62,6 @@ std::string AudioSendStream::Config::Rtp::ToString() const {
>     }
>   }
>   ss << ']';
>-  ss << ", nack: " << nack.ToString();
>   ss << ", c_name: " << c_name;
>   ss << '}';
>   return ss.str();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.h
>index 61e25312229e90015ae476faec28ffe3a361ed0a..c996dab83d70005505229e244af493e7e8d2f82f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.h
>@@ -21,7 +21,9 @@
> #include "api/audio_codecs/audio_encoder_factory.h"
> #include "api/audio_codecs/audio_format.h"
> #include "api/call/transport.h"
>+#include "api/crypto/cryptooptions.h"
> #include "api/crypto/frameencryptorinterface.h"
>+#include "api/media_transport_interface.h"
> #include "api/rtpparameters.h"
> #include "call/rtp_config.h"
> #include "modules/audio_processing/include/audio_processing_statistics.h"
>@@ -57,10 +59,13 @@ class AudioSendStream {
> 
>     ANAStats ana_statistics;
>     AudioProcessingStats apm_statistics;
>+
>+    int64_t target_bitrate_bps = 0;
>   };
> 
>   struct Config {
>     Config() = delete;
>+    Config(Transport* send_transport, MediaTransportInterface* media_transport);
>     explicit Config(Transport* send_transport);
>     ~Config();
>     std::string ToString() const;
>@@ -78,20 +83,25 @@ class AudioSendStream {
>       // included in the list of extensions.
>       std::string mid;
> 
>+      // Corresponds to the SDP attribute extmap-allow-mixed.
>+      bool extmap_allow_mixed = false;
>+
>       // RTP header extensions used for the sent stream.
>       std::vector<RtpExtension> extensions;
> 
>-      // See NackConfig for description.
>-      NackConfig nack;
>-
>       // RTCP CNAME, see RFC 3550.
>       std::string c_name;
>     } rtp;
> 
>+    // Time interval between RTCP report for audio
>+    int rtcp_report_interval_ms = 5000;
>+
>     // Transport for outgoing packets. The transport is expected to exist for
>     // the entire life of the AudioSendStream and is owned by the API client.
>     Transport* send_transport = nullptr;
> 
>+    MediaTransportInterface* media_transport = nullptr;
>+
>     // Bitrate limits used for variable audio bitrate streams. Set both to -1 to
>     // disable audio bitrate adaptation.
>     // Note: This is still an experimental feature and not ready for real usage.
>@@ -99,6 +109,7 @@ class AudioSendStream {
>     int max_bitrate_bps = -1;
> 
>     double bitrate_priority = 1.0;
>+    bool has_dscp = false;
> 
>     // Defines whether to turn on audio network adaptor, and defines its config
>     // string.
>@@ -130,6 +141,9 @@ class AudioSendStream {
>     // Track ID as specified during track creation.
>     std::string track_id;
> 
>+    // Per PeerConnection crypto options.
>+    webrtc::CryptoOptions crypto_options;
>+
>     // An optional custom frame encryptor that allows the entire frame to be
>     // encryptor in whatever way the caller choses. This is not required by
>     // default.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.cc
>index acb19ae6fb2bff0bcda025f38e95670fcaf0e28a..660981c30e399edeb069a151640c6d35788edbf9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.cc
>@@ -16,6 +16,8 @@
> #include <memory>
> #include <utility>
> 
>+#include "api/units/data_rate.h"
>+#include "api/units/time_delta.h"
> #include "modules/bitrate_controller/include/bitrate_controller.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>@@ -50,10 +52,12 @@ double MediaRatio(uint32_t allocated_bitrate, uint32_t protection_bitrate) {
> 
> BitrateAllocator::BitrateAllocator(LimitObserver* limit_observer)
>     : limit_observer_(limit_observer),
>-      last_bitrate_bps_(0),
>+      last_target_bps_(0),
>+      last_link_capacity_bps_(0),
>       last_non_zero_bitrate_bps_(kDefaultBitrateBps),
>       last_fraction_loss_(0),
>       last_rtt_(0),
>+      last_bwe_period_ms_(1000),
>       num_pause_events_(0),
>       clock_(Clock::GetRealTimeClock()),
>       last_bwe_log_time_(0),
>@@ -88,11 +92,13 @@ uint8_t BitrateAllocator::GetTransmissionMaxBitrateMultiplier() {
> }
> 
> void BitrateAllocator::OnNetworkChanged(uint32_t target_bitrate_bps,
>+                                        uint32_t link_capacity_bps,
>                                         uint8_t fraction_loss,
>                                         int64_t rtt,
>                                         int64_t bwe_period_ms) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  last_bitrate_bps_ = target_bitrate_bps;
>+  last_target_bps_ = target_bitrate_bps;
>+  last_link_capacity_bps_ = link_capacity_bps;
>   last_non_zero_bitrate_bps_ =
>       target_bitrate_bps > 0 ? target_bitrate_bps : last_non_zero_bitrate_bps_;
>   last_fraction_loss_ = fraction_loss;
>@@ -107,11 +113,18 @@ void BitrateAllocator::OnNetworkChanged(uint32_t target_bitrate_bps,
>   }
> 
>   ObserverAllocation allocation = AllocateBitrates(target_bitrate_bps);
>+  ObserverAllocation bandwidth_allocation = AllocateBitrates(link_capacity_bps);
> 
>   for (auto& config : bitrate_observer_configs_) {
>     uint32_t allocated_bitrate = allocation[config.observer];
>-    uint32_t protection_bitrate = config.observer->OnBitrateUpdated(
>-        allocated_bitrate, last_fraction_loss_, last_rtt_, last_bwe_period_ms_);
>+    uint32_t allocated_bandwidth = bandwidth_allocation[config.observer];
>+    BitrateAllocationUpdate update;
>+    update.target_bitrate = DataRate::bps(allocated_bitrate);
>+    update.link_capacity = DataRate::bps(allocated_bandwidth);
>+    update.packet_loss_ratio = last_fraction_loss_ / 256.0;
>+    update.round_trip_time = TimeDelta::ms(last_rtt_);
>+    update.bwe_period = TimeDelta::ms(last_bwe_period_ms_);
>+    uint32_t protection_bitrate = config.observer->OnBitrateUpdated(update);
> 
>     if (allocated_bitrate == 0 && config.allocated_bitrate_bps > 0) {
>       if (target_bitrate_bps > 0)
>@@ -163,15 +176,22 @@ void BitrateAllocator::AddObserver(BitrateAllocatorObserver* observer,
>         config.bitrate_priority, config.has_packet_feedback));
>   }
> 
>-  ObserverAllocation allocation;
>-  if (last_bitrate_bps_ > 0) {
>+  if (last_target_bps_ > 0) {
>     // Calculate a new allocation and update all observers.
>-    allocation = AllocateBitrates(last_bitrate_bps_);
>+
>+    ObserverAllocation allocation = AllocateBitrates(last_target_bps_);
>+    ObserverAllocation bandwidth_allocation =
>+        AllocateBitrates(last_link_capacity_bps_);
>     for (auto& config : bitrate_observer_configs_) {
>       uint32_t allocated_bitrate = allocation[config.observer];
>-      uint32_t protection_bitrate = config.observer->OnBitrateUpdated(
>-          allocated_bitrate, last_fraction_loss_, last_rtt_,
>-          last_bwe_period_ms_);
>+      uint32_t bandwidth = bandwidth_allocation[config.observer];
>+      BitrateAllocationUpdate update;
>+      update.target_bitrate = DataRate::bps(allocated_bitrate);
>+      update.link_capacity = DataRate::bps(bandwidth);
>+      update.packet_loss_ratio = last_fraction_loss_ / 256.0;
>+      update.round_trip_time = TimeDelta::ms(last_rtt_);
>+      update.bwe_period = TimeDelta::ms(last_bwe_period_ms_);
>+      uint32_t protection_bitrate = config.observer->OnBitrateUpdated(update);
>       config.allocated_bitrate_bps = allocated_bitrate;
>       if (allocated_bitrate > 0)
>         config.media_ratio = MediaRatio(allocated_bitrate, protection_bitrate);
>@@ -180,9 +200,14 @@ void BitrateAllocator::AddObserver(BitrateAllocatorObserver* observer,
>     // Currently, an encoder is not allowed to produce frames.
>     // But we still have to return the initial config bitrate + let the
>     // observer know that it can not produce frames.
>-    allocation = AllocateBitrates(last_non_zero_bitrate_bps_);
>-    observer->OnBitrateUpdated(0, last_fraction_loss_, last_rtt_,
>-                               last_bwe_period_ms_);
>+
>+    BitrateAllocationUpdate update;
>+    update.target_bitrate = DataRate::Zero();
>+    update.link_capacity = DataRate::Zero();
>+    update.packet_loss_ratio = last_fraction_loss_ / 256.0;
>+    update.round_trip_time = TimeDelta::ms(last_rtt_);
>+    update.bwe_period = TimeDelta::ms(last_bwe_period_ms_);
>+    observer->OnBitrateUpdated(update);
>   }
>   UpdateAllocationLimits();
> }
>@@ -248,7 +273,8 @@ void BitrateAllocator::RemoveObserver(BitrateAllocatorObserver* observer) {
>   UpdateAllocationLimits();
> }
> 
>-int BitrateAllocator::GetStartBitrate(BitrateAllocatorObserver* observer) {
>+int BitrateAllocator::GetStartBitrate(
>+    BitrateAllocatorObserver* observer) const {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>   const auto& it = FindObserverConfig(observer);
>   if (it == bitrate_observer_configs_.end()) {
>@@ -272,6 +298,17 @@ void BitrateAllocator::SetBitrateAllocationStrategy(
>   bitrate_allocation_strategy_ = std::move(bitrate_allocation_strategy);
> }
> 
>+BitrateAllocator::ObserverConfigs::const_iterator
>+BitrateAllocator::FindObserverConfig(
>+    const BitrateAllocatorObserver* observer) const {
>+  for (auto it = bitrate_observer_configs_.begin();
>+       it != bitrate_observer_configs_.end(); ++it) {
>+    if (it->observer == observer)
>+      return it;
>+  }
>+  return bitrate_observer_configs_.end();
>+}
>+
> BitrateAllocator::ObserverConfigs::iterator
> BitrateAllocator::FindObserverConfig(const BitrateAllocatorObserver* observer) {
>   for (auto it = bitrate_observer_configs_.begin();
>@@ -283,19 +320,19 @@ BitrateAllocator::FindObserverConfig(const BitrateAllocatorObserver* observer) {
> }
> 
> BitrateAllocator::ObserverAllocation BitrateAllocator::AllocateBitrates(
>-    uint32_t bitrate) {
>+    uint32_t bitrate) const {
>   if (bitrate_observer_configs_.empty())
>     return ObserverAllocation();
> 
>   if (bitrate_allocation_strategy_ != nullptr) {
>-    std::vector<const rtc::BitrateAllocationStrategy::TrackConfig*>
>-        track_configs(bitrate_observer_configs_.size());
>-    int i = 0;
>-    for (const auto& c : bitrate_observer_configs_) {
>-      track_configs[i++] = &c;
>-    }
>+    // Note: This intentionally causes slicing, we only copy the fields in
>+    // ObserverConfig that are inherited from TrackConfig.
>+    std::vector<rtc::BitrateAllocationStrategy::TrackConfig> track_configs(
>+        bitrate_observer_configs_.begin(), bitrate_observer_configs_.end());
>+
>     std::vector<uint32_t> track_allocations =
>-        bitrate_allocation_strategy_->AllocateBitrates(bitrate, track_configs);
>+        bitrate_allocation_strategy_->AllocateBitrates(
>+            bitrate, std::move(track_configs));
>     // The strategy should return allocation for all tracks.
>     RTC_CHECK(track_allocations.size() == bitrate_observer_configs_.size());
>     ObserverAllocation allocation;
>@@ -331,7 +368,8 @@ BitrateAllocator::ObserverAllocation BitrateAllocator::AllocateBitrates(
>   return MaxRateAllocation(bitrate, sum_max_bitrates);
> }
> 
>-BitrateAllocator::ObserverAllocation BitrateAllocator::ZeroRateAllocation() {
>+BitrateAllocator::ObserverAllocation BitrateAllocator::ZeroRateAllocation()
>+    const {
>   ObserverAllocation allocation;
>   for (const auto& observer_config : bitrate_observer_configs_)
>     allocation[observer_config.observer] = 0;
>@@ -339,7 +377,7 @@ BitrateAllocator::ObserverAllocation BitrateAllocator::ZeroRateAllocation() {
> }
> 
> BitrateAllocator::ObserverAllocation BitrateAllocator::LowRateAllocation(
>-    uint32_t bitrate) {
>+    uint32_t bitrate) const {
>   ObserverAllocation allocation;
>   // Start by allocating bitrate to observers enforcing a min bitrate, hence
>   // remaining_bitrate might turn negative.
>@@ -400,7 +438,7 @@ BitrateAllocator::ObserverAllocation BitrateAllocator::LowRateAllocation(
> // min_bitrate_bps values, until one of the observers hits its max_bitrate_bps.
> BitrateAllocator::ObserverAllocation BitrateAllocator::NormalRateAllocation(
>     uint32_t bitrate,
>-    uint32_t sum_min_bitrates) {
>+    uint32_t sum_min_bitrates) const {
>   ObserverAllocation allocation;
>   ObserverAllocation observers_capacities;
>   for (const auto& observer_config : bitrate_observer_configs_) {
>@@ -420,7 +458,7 @@ BitrateAllocator::ObserverAllocation BitrateAllocator::NormalRateAllocation(
> 
> BitrateAllocator::ObserverAllocation BitrateAllocator::MaxRateAllocation(
>     uint32_t bitrate,
>-    uint32_t sum_max_bitrates) {
>+    uint32_t sum_max_bitrates) const {
>   ObserverAllocation allocation;
> 
>   for (const auto& observer_config : bitrate_observer_configs_) {
>@@ -457,10 +495,11 @@ uint32_t BitrateAllocator::ObserverConfig::MinBitrateWithHysteresis() const {
>   return min_bitrate;
> }
> 
>-void BitrateAllocator::DistributeBitrateEvenly(uint32_t bitrate,
>-                                               bool include_zero_allocations,
>-                                               int max_multiplier,
>-                                               ObserverAllocation* allocation) {
>+void BitrateAllocator::DistributeBitrateEvenly(
>+    uint32_t bitrate,
>+    bool include_zero_allocations,
>+    int max_multiplier,
>+    ObserverAllocation* allocation) const {
>   RTC_DCHECK_EQ(allocation->size(), bitrate_observer_configs_.size());
> 
>   ObserverSortingMap list_max_bitrates;
>@@ -491,8 +530,9 @@ void BitrateAllocator::DistributeBitrateEvenly(uint32_t bitrate,
>   }
> }
> 
>-bool BitrateAllocator::EnoughBitrateForAllObservers(uint32_t bitrate,
>-                                                    uint32_t sum_min_bitrates) {
>+bool BitrateAllocator::EnoughBitrateForAllObservers(
>+    uint32_t bitrate,
>+    uint32_t sum_min_bitrates) const {
>   if (bitrate < sum_min_bitrates)
>     return false;
> 
>@@ -511,7 +551,7 @@ bool BitrateAllocator::EnoughBitrateForAllObservers(uint32_t bitrate,
> void BitrateAllocator::DistributeBitrateRelatively(
>     uint32_t remaining_bitrate,
>     const ObserverAllocation& observers_capacities,
>-    ObserverAllocation* allocation) {
>+    ObserverAllocation* allocation) const {
>   RTC_DCHECK_EQ(allocation->size(), bitrate_observer_configs_.size());
>   RTC_DCHECK_EQ(observers_capacities.size(), bitrate_observer_configs_.size());
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.h
>index 47df7e01512eaa327d7016022eb185fe25148582..059f77b967179ab02354e001026cd5e43149d66c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.h
>@@ -19,6 +19,7 @@
> #include <utility>
> #include <vector>
> 
>+#include "api/call/bitrate_allocation.h"
> #include "rtc_base/bitrateallocationstrategy.h"
> #include "rtc_base/sequenced_task_checker.h"
> 
>@@ -34,10 +35,7 @@ class BitrateAllocatorObserver {
>  public:
>   // Returns the amount of protection used by the BitrateAllocatorObserver
>   // implementation, as bitrate in bps.
>-  virtual uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
>-                                    uint8_t fraction_loss,
>-                                    int64_t rtt,
>-                                    int64_t bwe_period_ms) = 0;
>+  virtual uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) = 0;
> 
>  protected:
>   virtual ~BitrateAllocatorObserver() {}
>@@ -73,7 +71,7 @@ class BitrateAllocatorInterface {
>   virtual void AddObserver(BitrateAllocatorObserver* observer,
>                            MediaStreamAllocationConfig config) = 0;
>   virtual void RemoveObserver(BitrateAllocatorObserver* observer) = 0;
>-  virtual int GetStartBitrate(BitrateAllocatorObserver* observer) = 0;
>+  virtual int GetStartBitrate(BitrateAllocatorObserver* observer) const = 0;
> 
>  protected:
>   virtual ~BitrateAllocatorInterface() = default;
>@@ -104,6 +102,7 @@ class BitrateAllocator : public BitrateAllocatorInterface {
> 
>   // Allocate target_bitrate across the registered BitrateAllocatorObservers.
>   void OnNetworkChanged(uint32_t target_bitrate_bps,
>+                        uint32_t link_capacity_bps,
>                         uint8_t fraction_loss,
>                         int64_t rtt,
>                         int64_t bwe_period_ms);
>@@ -120,7 +119,7 @@ class BitrateAllocator : public BitrateAllocatorInterface {
> 
>   // Returns initial bitrate allocated for |observer|. If |observer| is not in
>   // the list of added observers, a best guess is returned.
>-  int GetStartBitrate(BitrateAllocatorObserver* observer) override;
>+  int GetStartBitrate(BitrateAllocatorObserver* observer) const override;
> 
>   // Sets external allocation strategy. If strategy is not set default WebRTC
>   // allocation mechanism will be used. The strategy may be changed during call.
>@@ -171,31 +170,34 @@ class BitrateAllocator : public BitrateAllocatorInterface {
>   void UpdateAllocationLimits() RTC_RUN_ON(&sequenced_checker_);
> 
>   typedef std::vector<ObserverConfig> ObserverConfigs;
>+  ObserverConfigs::const_iterator FindObserverConfig(
>+      const BitrateAllocatorObserver* observer) const
>+      RTC_RUN_ON(&sequenced_checker_);
>   ObserverConfigs::iterator FindObserverConfig(
>       const BitrateAllocatorObserver* observer) RTC_RUN_ON(&sequenced_checker_);
> 
>   typedef std::multimap<uint32_t, const ObserverConfig*> ObserverSortingMap;
>   typedef std::map<BitrateAllocatorObserver*, int> ObserverAllocation;
> 
>-  ObserverAllocation AllocateBitrates(uint32_t bitrate)
>+  ObserverAllocation AllocateBitrates(uint32_t bitrate) const
>       RTC_RUN_ON(&sequenced_checker_);
> 
>   // Allocates zero bitrate to all observers.
>-  ObserverAllocation ZeroRateAllocation() RTC_RUN_ON(&sequenced_checker_);
>+  ObserverAllocation ZeroRateAllocation() const RTC_RUN_ON(&sequenced_checker_);
>   // Allocates bitrate to observers when there isn't enough to allocate the
>   // minimum to all observers.
>-  ObserverAllocation LowRateAllocation(uint32_t bitrate)
>+  ObserverAllocation LowRateAllocation(uint32_t bitrate) const
>       RTC_RUN_ON(&sequenced_checker_);
>   // Allocates bitrate to all observers when the available bandwidth is enough
>   // to allocate the minimum to all observers but not enough to allocate the
>   // max bitrate of each observer.
>   ObserverAllocation NormalRateAllocation(uint32_t bitrate,
>-                                          uint32_t sum_min_bitrates)
>+                                          uint32_t sum_min_bitrates) const
>       RTC_RUN_ON(&sequenced_checker_);
>   // Allocates bitrate to observers when there is enough available bandwidth
>   // for all observers to be allocated their max bitrate.
>   ObserverAllocation MaxRateAllocation(uint32_t bitrate,
>-                                       uint32_t sum_max_bitrates)
>+                                       uint32_t sum_max_bitrates) const
>       RTC_RUN_ON(&sequenced_checker_);
> 
>   // Splits |bitrate| evenly to observers already in |allocation|.
>@@ -205,9 +207,10 @@ class BitrateAllocator : public BitrateAllocatorInterface {
>   void DistributeBitrateEvenly(uint32_t bitrate,
>                                bool include_zero_allocations,
>                                int max_multiplier,
>-                               ObserverAllocation* allocation)
>+                               ObserverAllocation* allocation) const
>       RTC_RUN_ON(&sequenced_checker_);
>-  bool EnoughBitrateForAllObservers(uint32_t bitrate, uint32_t sum_min_bitrates)
>+  bool EnoughBitrateForAllObservers(uint32_t bitrate,
>+                                    uint32_t sum_min_bitrates) const
>       RTC_RUN_ON(&sequenced_checker_);
> 
>   // From the available |bitrate|, each observer will be allocated a
>@@ -219,7 +222,7 @@ class BitrateAllocator : public BitrateAllocatorInterface {
>   void DistributeBitrateRelatively(
>       uint32_t bitrate,
>       const ObserverAllocation& observers_capacities,
>-      ObserverAllocation* allocation) RTC_RUN_ON(&sequenced_checker_);
>+      ObserverAllocation* allocation) const RTC_RUN_ON(&sequenced_checker_);
> 
>   // Allow packets to be transmitted in up to 2 times max video bitrate if the
>   // bandwidth estimate allows it.
>@@ -232,7 +235,8 @@ class BitrateAllocator : public BitrateAllocatorInterface {
>   LimitObserver* const limit_observer_ RTC_GUARDED_BY(&sequenced_checker_);
>   // Stored in a list to keep track of the insertion order.
>   ObserverConfigs bitrate_observer_configs_ RTC_GUARDED_BY(&sequenced_checker_);
>-  uint32_t last_bitrate_bps_ RTC_GUARDED_BY(&sequenced_checker_);
>+  uint32_t last_target_bps_ RTC_GUARDED_BY(&sequenced_checker_);
>+  uint32_t last_link_capacity_bps_ RTC_GUARDED_BY(&sequenced_checker_);
>   uint32_t last_non_zero_bitrate_bps_ RTC_GUARDED_BY(&sequenced_checker_);
>   uint8_t last_fraction_loss_ RTC_GUARDED_BY(&sequenced_checker_);
>   int64_t last_rtt_ RTC_GUARDED_BY(&sequenced_checker_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator_unittest.cc
>index 2961fd409ce98923487a2c3dc64b93da964724ad..0771f798edcd8b19d379759bc8e44b08e23c36f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator_unittest.cc
>@@ -59,15 +59,13 @@ class TestBitrateObserver : public BitrateAllocatorObserver {
>     protection_ratio_ = protection_ratio;
>   }
> 
>-  uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
>-                            uint8_t fraction_loss,
>-                            int64_t rtt,
>-                            int64_t probing_interval_ms) override {
>-    last_bitrate_bps_ = bitrate_bps;
>-    last_fraction_loss_ = fraction_loss;
>-    last_rtt_ms_ = rtt;
>-    last_probing_interval_ms_ = probing_interval_ms;
>-    return bitrate_bps * protection_ratio_;
>+  uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override {
>+    last_bitrate_bps_ = update.target_bitrate.bps();
>+    last_fraction_loss_ =
>+        rtc::dchecked_cast<uint8_t>(update.packet_loss_ratio * 256);
>+    last_rtt_ms_ = update.round_trip_time.ms();
>+    last_probing_interval_ms_ = update.bwe_period.ms();
>+    return update.target_bitrate.bps() * protection_ratio_;
>   }
>   uint32_t last_bitrate_bps_;
>   uint8_t last_fraction_loss_;
>@@ -76,6 +74,18 @@ class TestBitrateObserver : public BitrateAllocatorObserver {
>   double protection_ratio_;
> };
> 
>+class BitrateAllocatorForTest : public BitrateAllocator {
>+ public:
>+  using BitrateAllocator::BitrateAllocator;
>+  void OnNetworkChanged(uint32_t target_bitrate_bps,
>+                        uint8_t fraction_loss,
>+                        int64_t rtt,
>+                        int64_t bwe_period_ms) {
>+    BitrateAllocator::OnNetworkChanged(target_bitrate_bps, target_bitrate_bps,
>+                                       fraction_loss, rtt, bwe_period_ms);
>+  }
>+};
>+
> namespace {
> constexpr int64_t kDefaultProbingIntervalMs = 3000;
> const double kDefaultBitratePriority = 1.0;
>@@ -83,7 +93,8 @@ const double kDefaultBitratePriority = 1.0;
> 
> class BitrateAllocatorTest : public ::testing::Test {
>  protected:
>-  BitrateAllocatorTest() : allocator_(new BitrateAllocator(&limit_observer_)) {
>+  BitrateAllocatorTest()
>+      : allocator_(new BitrateAllocatorForTest(&limit_observer_)) {
>     allocator_->OnNetworkChanged(300000u, 0, 0, kDefaultProbingIntervalMs);
>   }
>   ~BitrateAllocatorTest() {}
>@@ -100,7 +111,7 @@ class BitrateAllocatorTest : public ::testing::Test {
>   }
> 
>   NiceMock<MockLimitObserver> limit_observer_;
>-  std::unique_ptr<BitrateAllocator> allocator_;
>+  std::unique_ptr<BitrateAllocatorForTest> allocator_;
> };
> 
> TEST_F(BitrateAllocatorTest, UpdatingBitrateObserver) {
>@@ -216,7 +227,7 @@ TEST_F(BitrateAllocatorTest, RemoveObserverTriggersLimitObserver) {
> class BitrateAllocatorTestNoEnforceMin : public ::testing::Test {
>  protected:
>   BitrateAllocatorTestNoEnforceMin()
>-      : allocator_(new BitrateAllocator(&limit_observer_)) {
>+      : allocator_(new BitrateAllocatorForTest(&limit_observer_)) {
>     allocator_->OnNetworkChanged(300000u, 0, 0, kDefaultProbingIntervalMs);
>   }
>   ~BitrateAllocatorTestNoEnforceMin() {}
>@@ -232,7 +243,7 @@ class BitrateAllocatorTestNoEnforceMin : public ::testing::Test {
>                    enforce_min_bitrate, track_id, bitrate_priority, false});
>   }
>   NiceMock<MockLimitObserver> limit_observer_;
>-  std::unique_ptr<BitrateAllocator> allocator_;
>+  std::unique_ptr<BitrateAllocatorForTest> allocator_;
> };
> 
> // The following three tests verify enforcing a minimum bitrate works as
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_estimator_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_estimator_tests.cc
>index 766e38c9788cc409ed479898ca1b731c33c47734..0862827d3a24a396d30dc774247c5a6c21bbe227 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_estimator_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_estimator_tests.cc
>@@ -46,8 +46,6 @@ class LogObserver {
>  private:
>   class Callback : public rtc::LogSink {
>    public:
>-    Callback() : done_(false, false) {}
>-
>     void OnLogMessage(const std::string& message) override {
>       rtc::CritScope lock(&crit_sect_);
>       // Ignore log lines that are due to missing AST extensions, these are
>@@ -111,14 +109,14 @@ class BitrateEstimatorTest : public test::CallTest {
>           &task_queue_,
>           absl::make_unique<FakeNetworkPipe>(
>               Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                             DefaultNetworkSimulationConfig())),
>+                                             BuiltInNetworkBehaviorConfig())),
>           sender_call_.get(), payload_type_map_));
>       send_transport_->SetReceiver(receiver_call_->Receiver());
>       receive_transport_.reset(new test::DirectTransport(
>           &task_queue_,
>           absl::make_unique<FakeNetworkPipe>(
>               Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                             DefaultNetworkSimulationConfig())),
>+                                             BuiltInNetworkBehaviorConfig())),
>           receiver_call_.get(), payload_type_map_));
>       receive_transport_->SetReceiver(sender_call_->Receiver());
> 
>@@ -126,6 +124,8 @@ class BitrateEstimatorTest : public test::CallTest {
>       video_send_config.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
>       video_send_config.encoder_settings.encoder_factory =
>           &fake_encoder_factory_;
>+      video_send_config.encoder_settings.bitrate_allocator_factory =
>+          bitrate_allocator_factory_.get();
>       video_send_config.rtp.payload_name = "FAKE";
>       video_send_config.rtp.payload_type = kFakeVideoSendPayloadType;
>       SetVideoSendConfig(video_send_config);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.cc
>index 635874b2d624a652c963af0c9c88701ecf4ccaa3..95f64e92499058d25f2295e599636c9ba9ef7a60 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.cc
>@@ -22,7 +22,6 @@
> #include "audio/audio_receive_stream.h"
> #include "audio/audio_send_stream.h"
> #include "audio/audio_state.h"
>-#include "audio/time_interval.h"
> #include "call/bitrate_allocator.h"
> #include "call/call.h"
> #include "call/flexfec_receive_stream_impl.h"
>@@ -30,7 +29,6 @@
> #include "call/rtp_stream_receiver_controller.h"
> #include "call/rtp_transport_controller_send.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
>-#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
> #include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
> #include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
> #include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
>@@ -56,6 +54,7 @@
> #include "rtc_base/synchronization/rw_lock_wrapper.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/thread_annotations.h"
>+#include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/cpu_info.h"
>@@ -146,18 +145,6 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
>   return rtclog_config;
> }
> 
>-std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
>-    const AudioSendStream::Config& config) {
>-  auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
>-  rtclog_config->local_ssrc = config.rtp.ssrc;
>-  rtclog_config->rtp_extensions = config.rtp.extensions;
>-  if (config.send_codec_spec) {
>-    rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name,
>-                                       config.send_codec_spec->payload_type, 0);
>-  }
>-  return rtclog_config;
>-}
>-
> }  // namespace
> 
> namespace internal {
>@@ -236,6 +223,15 @@ class Call final : public webrtc::Call,
>                                  uint32_t allocated_without_feedback_bps,
>                                  bool has_packet_feedback) override;
> 
>+  // This method is invoked when the media transport is created and when the
>+  // media transport is being destructed.
>+  // We only allow one media transport per connection.
>+  //
>+  // It should be called with non-null argument at most once, and if it was
>+  // called with non-null argument, it has to be called with a null argument
>+  // at least once after that.
>+  void MediaTransportChange(MediaTransportInterface* media_transport) override;
>+
>  private:
>   DeliveryStatus DeliverRtcp(MediaType media_type,
>                              const uint8_t* packet,
>@@ -256,6 +252,10 @@ class Call final : public webrtc::Call,
>   void UpdateHistograms();
>   void UpdateAggregateNetworkState();
> 
>+  //  If |media_transport| is not null, it registers the rate observer for the
>+  //  media transport.
>+  void RegisterRateObserver() RTC_LOCKS_EXCLUDED(target_observer_crit_);
>+
>   Clock* const clock_;
> 
>   const int num_cpu_cores_;
>@@ -346,7 +346,6 @@ class Call final : public webrtc::Call,
>   absl::optional<int64_t> last_received_rtp_audio_ms_;
>   absl::optional<int64_t> first_received_rtp_video_ms_;
>   absl::optional<int64_t> last_received_rtp_video_ms_;
>-  TimeInterval sent_rtp_audio_timer_ms_;
> 
>   rtc::CriticalSection last_bandwidth_bps_crit_;
>   uint32_t last_bandwidth_bps_ RTC_GUARDED_BY(&last_bandwidth_bps_crit_);
>@@ -374,6 +373,15 @@ class Call final : public webrtc::Call,
>   // Declared last since it will issue callbacks from a task queue. Declaring it
>   // last ensures that it is destroyed first and any running tasks are finished.
>   std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;
>+
>+  // This is a precaution, since |MediaTransportChange| is not guaranteed to be
>+  // invoked on a particular thread.
>+  rtc::CriticalSection target_observer_crit_;
>+  bool is_target_rate_observer_registered_
>+      RTC_GUARDED_BY(&target_observer_crit_) = false;
>+  MediaTransportInterface* media_transport_
>+      RTC_GUARDED_BY(&target_observer_crit_) = nullptr;
>+
>   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
> };
> }  // namespace internal
>@@ -444,7 +452,6 @@ Call::Call(const Call::Config& config,
>       video_send_delay_stats_(new SendDelayStats(clock_)),
>       start_ms_(clock_->TimeInMilliseconds()) {
>   RTC_DCHECK(config.event_log != nullptr);
>-  transport_send->RegisterTargetTransferRateObserver(this);
>   transport_send_ = std::move(transport_send);
>   transport_send_ptr_ = transport_send_.get();
> 
>@@ -486,6 +493,43 @@ Call::~Call() {
>   UpdateHistograms();
> }
> 
>+void Call::RegisterRateObserver() {
>+  rtc::CritScope lock(&target_observer_crit_);
>+
>+  if (is_target_rate_observer_registered_) {
>+    return;
>+  }
>+
>+  is_target_rate_observer_registered_ = true;
>+
>+  if (media_transport_) {
>+    media_transport_->AddTargetTransferRateObserver(this);
>+  } else {
>+    transport_send_ptr_->RegisterTargetTransferRateObserver(this);
>+  }
>+}
>+
>+void Call::MediaTransportChange(MediaTransportInterface* media_transport) {
>+  rtc::CritScope lock(&target_observer_crit_);
>+
>+  if (is_target_rate_observer_registered_) {
>+    // Only used to unregister rate observer from media transport. Registration
>+    // happens when the stream is created.
>+    if (!media_transport && media_transport_) {
>+      media_transport_->RemoveTargetTransferRateObserver(this);
>+      media_transport_ = nullptr;
>+      is_target_rate_observer_registered_ = false;
>+    }
>+  } else if (media_transport) {
>+    RTC_DCHECK(media_transport_ == nullptr ||
>+               media_transport_ == media_transport)
>+        << "media_transport_=" << (media_transport_ != nullptr)
>+        << ", (media_transport_==media_transport)="
>+        << (media_transport_ == media_transport);
>+    media_transport_ = media_transport;
>+  }
>+}
>+
> void Call::UpdateHistograms() {
>   RTC_HISTOGRAM_COUNTS_100000(
>       "WebRTC.Call.LifetimeInSeconds",
>@@ -495,11 +539,6 @@ void Call::UpdateHistograms() {
> void Call::UpdateSendHistograms(int64_t first_sent_packet_ms) {
>   if (first_sent_packet_ms == -1)
>     return;
>-  if (!sent_rtp_audio_timer_ms_.Empty()) {
>-    RTC_HISTOGRAM_COUNTS_100000(
>-        "WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds",
>-        sent_rtp_audio_timer_ms_.Length() / 1000);
>-  }
>   int64_t elapsed_sec =
>       (clock_->TimeInMilliseconds() - first_sent_packet_ms) / 1000;
>   if (elapsed_sec < metrics::kMinRunTimeInSeconds)
>@@ -578,9 +617,16 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
>     const webrtc::AudioSendStream::Config& config) {
>   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
>-  event_log_->Log(absl::make_unique<RtcEventAudioSendStreamConfig>(
>-      CreateRtcLogStreamConfig(config)));
> 
>+  {
>+    rtc::CritScope lock(&target_observer_crit_);
>+    RTC_DCHECK(media_transport_ == config.media_transport);
>+  }
>+
>+  RegisterRateObserver();
>+
>+  // Stream config is logged in AudioSendStream::ConfigureStream, as it may
>+  // change during the stream's lifetime.
>   absl::optional<RtpState> suspended_rtp_state;
>   {
>     const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);
>@@ -596,7 +642,7 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
>       config, config_.audio_state, transport_send_ptr_->GetWorkerQueue(),
>       module_process_thread_.get(), transport_send_ptr_,
>       bitrate_allocator_.get(), event_log_, call_stats_.get(),
>-      suspended_rtp_state, &sent_rtp_audio_timer_ms_);
>+      suspended_rtp_state);
>   {
>     WriteLockScoped write_lock(*send_crit_);
>     RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
>@@ -708,6 +754,8 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
>   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
> 
>+  RegisterRateObserver();
>+
>   video_send_delay_stats_->AddSsrcs(config);
>   for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size();
>        ++ssrc_index) {
>@@ -1044,6 +1092,18 @@ void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
> }
> 
> void Call::OnTargetTransferRate(TargetTransferRate msg) {
>+  // TODO(bugs.webrtc.org/9719)
>+  // Call::OnTargetTransferRate requires that on target transfer rate is invoked
>+  // from the worker queue (because bitrate_allocator_ requires it). Media
>+  // transport does not guarantee the callback on the worker queue.
>+  // When the threading model for MediaTransportInterface is update, reconsider
>+  // changing this implementation.
>+  if (!transport_send_ptr_->GetWorkerQueue()->IsCurrent()) {
>+    transport_send_ptr_->GetWorkerQueue()->PostTask(
>+        [this, msg] { this->OnTargetTransferRate(msg); });
>+    return;
>+  }
>+
>   uint32_t target_bitrate_bps = msg.target_rate.bps();
>   int loss_ratio_255 = msg.network_estimate.loss_rate_ratio * 255;
>   uint8_t fraction_loss =
>@@ -1057,8 +1117,9 @@ void Call::OnTargetTransferRate(TargetTransferRate msg) {
>   }
>   // For controlling the rate of feedback messages.
>   receive_side_cc_.OnBitrateChanged(target_bitrate_bps);
>-  bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss,
>-                                       rtt_ms, probing_interval_ms);
>+  bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, bandwidth_bps,
>+                                       fraction_loss, rtt_ms,
>+                                       probing_interval_ms);
> 
>   // Ignore updates if bitrate is zero (the aggregate network state is down).
>   if (target_bitrate_bps == 0) {
>@@ -1215,8 +1276,10 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
> 
>   if (packet_time_us != -1) {
>     if (receive_time_calculator_) {
>+      // Repair packet_time_us for clock resets by comparing a new read of
>+      // the same clock (TimeUTCMicros) to a monotonic clock reading.
>       packet_time_us = receive_time_calculator_->ReconcileReceiveTimes(
>-          packet_time_us, clock_->TimeInMicroseconds());
>+          packet_time_us, rtc::TimeUTCMicros(), clock_->TimeInMicroseconds());
>     }
>     parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
>   } else {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.h
>index 4167296d756c547f434efd6c553ae6971609b3e4..c0e31c0902b3c96dcd0107c7a83e8c5c8f6c814c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.h
>@@ -24,11 +24,10 @@
> #include "call/rtp_transport_controller_send_interface.h"
> #include "call/video_receive_stream.h"
> #include "call/video_send_stream.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/bitrateallocationstrategy.h"
> #include "rtc_base/copyonwritebuffer.h"
>+#include "rtc_base/network/sent_packet.h"
> #include "rtc_base/networkroute.h"
>-#include "rtc_base/socket.h"
> 
> namespace webrtc {
> 
>@@ -58,6 +57,11 @@ class Call {
> 
>   virtual AudioSendStream* CreateAudioSendStream(
>       const AudioSendStream::Config& config) = 0;
>+
>+  // Gets called when media transport is created or removed.
>+  virtual void MediaTransportChange(
>+      MediaTransportInterface* media_transport_interface) = 0;
>+
>   virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
> 
>   virtual AudioReceiveStream* CreateAudioReceiveStream(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.cc
>index d3425aa57997a96ef29702b08e130ec4a2b95a8c..b149c889eae028b9dd3983c6e03491516f25d66a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.cc
>@@ -10,6 +10,8 @@
> 
> #include "call/call_config.h"
> 
>+#include "rtc_base/checks.h"
>+
> namespace webrtc {
> 
> CallConfig::CallConfig(RtcEventLog* event_log) : event_log(event_log) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_perf_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_perf_tests.cc
>index 8fcd566f81a8241de0ad2fa6b1a5f64e20971afb..72aa78c5f7770eac1334163917e2fc70cc9c0bee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_perf_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_perf_tests.cc
>@@ -16,6 +16,7 @@
> #include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/test/simulated_network.h"
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "api/video/video_bitrate_allocation.h"
> #include "api/video_codecs/video_encoder_config.h"
> #include "call/call.h"
>@@ -64,7 +65,7 @@ class CallPerfTest : public test::CallTest {
> 
>   void TestMinTransmitBitrate(bool pad_to_min_bitrate);
> 
>-  void TestCaptureNtpTime(const DefaultNetworkSimulationConfig& net_config,
>+  void TestCaptureNtpTime(const BuiltInNetworkBehaviorConfig& net_config,
>                           int threshold_ms,
>                           int start_time_ms,
>                           int run_time_ms);
>@@ -152,7 +153,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
>   const uint32_t kAudioSendSsrc = 1234;
>   const uint32_t kAudioRecvSsrc = 5678;
> 
>-  DefaultNetworkSimulationConfig audio_net_config;
>+  BuiltInNetworkBehaviorConfig audio_net_config;
>   audio_net_config.queue_delay_ms = 500;
>   audio_net_config.loss_percent = 5;
> 
>@@ -217,7 +218,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
>         test::PacketTransport::kSender, video_pt_map,
>         absl::make_unique<FakeNetworkPipe>(
>             Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())));
>+                                           BuiltInNetworkBehaviorConfig())));
>     video_send_transport->SetReceiver(receiver_call_->Receiver());
> 
>     receive_transport = absl::make_unique<test::PacketTransport>(
>@@ -225,13 +226,14 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
>         test::PacketTransport::kReceiver, payload_type_map_,
>         absl::make_unique<FakeNetworkPipe>(
>             Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())));
>+                                           BuiltInNetworkBehaviorConfig())));
>     receive_transport->SetReceiver(sender_call_->Receiver());
> 
>     CreateSendConfig(1, 0, 0, video_send_transport.get());
>     CreateMatchingReceiveConfigs(receive_transport.get());
> 
>-    AudioSendStream::Config audio_send_config(audio_send_transport.get());
>+    AudioSendStream::Config audio_send_config(audio_send_transport.get(),
>+                                              /*media_transport=*/nullptr);
>     audio_send_config.rtp.ssrc = kAudioSendSsrc;
>     audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
>         kAudioSendPayloadType, {"ISAC", 16000, 1});
>@@ -337,14 +339,14 @@ TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift) {
> }
> 
> void CallPerfTest::TestCaptureNtpTime(
>-    const DefaultNetworkSimulationConfig& net_config,
>+    const BuiltInNetworkBehaviorConfig& net_config,
>     int threshold_ms,
>     int start_time_ms,
>     int run_time_ms) {
>   class CaptureNtpTimeObserver : public test::EndToEndTest,
>                                  public rtc::VideoSinkInterface<VideoFrame> {
>    public:
>-    CaptureNtpTimeObserver(const DefaultNetworkSimulationConfig& net_config,
>+    CaptureNtpTimeObserver(const BuiltInNetworkBehaviorConfig& net_config,
>                            int threshold_ms,
>                            int start_time_ms,
>                            int run_time_ms)
>@@ -461,7 +463,7 @@ void CallPerfTest::TestCaptureNtpTime(
>     }
> 
>     rtc::CriticalSection crit_;
>-    const DefaultNetworkSimulationConfig net_config_;
>+    const BuiltInNetworkBehaviorConfig net_config_;
>     Clock* const clock_;
>     int threshold_ms_;
>     int start_time_ms_;
>@@ -478,10 +480,10 @@ void CallPerfTest::TestCaptureNtpTime(
>   RunBaseTest(&test);
> }
> 
>-// Flaky tests, disabled on Mac due to webrtc:8291.
>-#if !(defined(WEBRTC_MAC))
>+// Flaky tests, disabled on Mac and Windows due to webrtc:8291.
>+#if !(defined(WEBRTC_MAC) || defined(WEBRTC_WIN))
> TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
>-  DefaultNetworkSimulationConfig net_config;
>+  BuiltInNetworkBehaviorConfig net_config;
>   net_config.queue_delay_ms = 100;
>   // TODO(wu): lower the threshold as the calculation/estimatation becomes more
>   // accurate.
>@@ -492,7 +494,7 @@ TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
> }
> 
> TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) {
>-  DefaultNetworkSimulationConfig net_config;
>+  BuiltInNetworkBehaviorConfig net_config;
>   net_config.queue_delay_ms = 100;
>   net_config.delay_standard_deviation_ms = 10;
>   // TODO(wu): lower the threshold as the calculation/estimatation becomes more
>@@ -724,12 +726,13 @@ TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
>     BitrateObserver()
>         : EndToEndTest(kDefaultTimeoutMs),
>           FakeEncoder(Clock::GetRealTimeClock()),
>-          time_to_reconfigure_(false, false),
>           encoder_inits_(0),
>           last_set_bitrate_kbps_(0),
>           send_stream_(nullptr),
>           frame_generator_(nullptr),
>-          encoder_factory_(this) {}
>+          encoder_factory_(this),
>+          bitrate_allocator_factory_(
>+              CreateBuiltinVideoBitrateAllocatorFactory()) {}
> 
>     int32_t InitEncode(const VideoCodec* config,
>                        int32_t number_of_cores,
>@@ -767,8 +770,9 @@ TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
>       return FakeEncoder::SetRateAllocation(rate_allocation, framerate);
>     }
> 
>-    void ModifySenderCallConfig(Call::Config* config) override {
>-      config->bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
>+    void ModifySenderBitrateConfig(
>+        BitrateConstraints* bitrate_config) override {
>+      bitrate_config->start_bitrate_bps = kInitialBitrateKbps * 1000;
>     }
> 
>     void ModifyVideoConfigs(
>@@ -776,6 +780,8 @@ TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>       send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+      send_config->encoder_settings.bitrate_allocator_factory =
>+          bitrate_allocator_factory_.get();
>       encoder_config->max_bitrate_bps = 2 * kReconfigureThresholdKbps * 1000;
>       encoder_config->video_stream_factory =
>           new rtc::RefCountedObject<VideoStreamFactory>();
>@@ -811,6 +817,7 @@ TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
>     VideoSendStream* send_stream_;
>     test::FrameGeneratorCapturer* frame_generator_;
>     test::VideoEncoderProxyFactory encoder_factory_;
>+    std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
>     VideoEncoderConfig encoder_config_;
>   } test;
> 
>@@ -865,8 +872,8 @@ void CallPerfTest::TestMinAudioVideoBitrate(
>           max_bwe_(max_bwe) {}
> 
>    protected:
>-    DefaultNetworkSimulationConfig GetFakeNetworkPipeConfig() {
>-      DefaultNetworkSimulationConfig pipe_config;
>+    BuiltInNetworkBehaviorConfig GetFakeNetworkPipeConfig() {
>+      BuiltInNetworkBehaviorConfig pipe_config;
>       pipe_config.link_capacity_kbps = test_bitrate_from_;
>       return pipe_config;
>     }
>@@ -903,7 +910,7 @@ void CallPerfTest::TestMinAudioVideoBitrate(
>                ? test_bitrate <= test_bitrate_to_
>                : test_bitrate >= test_bitrate_to_;
>            test_bitrate += test_bitrate_step_) {
>-        DefaultNetworkSimulationConfig pipe_config;
>+        BuiltInNetworkBehaviorConfig pipe_config;
>         pipe_config.link_capacity_kbps = test_bitrate;
>         send_simulated_network_->SetConfig(pipe_config);
>         receive_simulated_network_->SetConfig(pipe_config);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_unittest.cc
>index 8ae04008f9ca3ba84c2973ff2cadb1301bb18b64..770f255313cc1d47b8989c43c31008605e20e11d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_unittest.cc
>@@ -15,6 +15,7 @@
> 
> #include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
>+#include "api/test/fake_media_transport.h"
> #include "api/test/mock_audio_mixer.h"
> #include "audio/audio_receive_stream.h"
> #include "audio/audio_send_stream.h"
>@@ -62,7 +63,8 @@ TEST(CallTest, ConstructDestruct) {
> 
> TEST(CallTest, CreateDestroy_AudioSendStream) {
>   CallHelper call;
>-  AudioSendStream::Config config(nullptr);
>+  AudioSendStream::Config config(/*send_transport=*/nullptr,
>+                                 /*media_transport=*/nullptr);
>   config.rtp.ssrc = 42;
>   AudioSendStream* stream = call->CreateAudioSendStream(config);
>   EXPECT_NE(stream, nullptr);
>@@ -84,7 +86,8 @@ TEST(CallTest, CreateDestroy_AudioReceiveStream) {
> 
> TEST(CallTest, CreateDestroy_AudioSendStreams) {
>   CallHelper call;
>-  AudioSendStream::Config config(nullptr);
>+  AudioSendStream::Config config(/*send_transport=*/nullptr,
>+                                 /*media_transport=*/nullptr);
>   std::list<AudioSendStream*> streams;
>   for (int i = 0; i < 2; ++i) {
>     for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
>@@ -142,7 +145,8 @@ TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) {
>   AudioReceiveStream* recv_stream = call->CreateAudioReceiveStream(recv_config);
>   EXPECT_NE(recv_stream, nullptr);
> 
>-  AudioSendStream::Config send_config(nullptr);
>+  AudioSendStream::Config send_config(/*send_transport=*/nullptr,
>+                                      /*media_transport=*/nullptr);
>   send_config.rtp.ssrc = 777;
>   AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
>   EXPECT_NE(send_stream, nullptr);
>@@ -160,7 +164,8 @@ TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) {
> 
> TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
>   CallHelper call;
>-  AudioSendStream::Config send_config(nullptr);
>+  AudioSendStream::Config send_config(/*send_transport=*/nullptr,
>+                                      /*media_transport=*/nullptr);
>   send_config.rtp.ssrc = 777;
>   AudioSendStream* send_stream = call->CreateAudioSendStream(send_config);
>   EXPECT_NE(send_stream, nullptr);
>@@ -263,7 +268,8 @@ TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
>   CallHelper call;
> 
>   auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) {
>-    AudioSendStream::Config config(nullptr);
>+    AudioSendStream::Config config(/*send_transport=*/nullptr,
>+                                   /*media_transport=*/nullptr);
>     config.rtp.ssrc = ssrc;
>     AudioSendStream* stream = call->CreateAudioSendStream(config);
>     const RtpState rtp_state =
>@@ -284,4 +290,27 @@ TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
>   EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent);
> }
> 
>+TEST(CallTest, RegisterMediaTransportBitrateCallbacksInCreateStream) {
>+  CallHelper call;
>+  MediaTransportSettings settings;
>+  webrtc::FakeMediaTransport fake_media_transport(settings);
>+
>+  EXPECT_EQ(0, fake_media_transport.target_rate_observers_size());
>+  AudioSendStream::Config config(/*send_transport=*/nullptr,
>+                                 /*media_transport=*/&fake_media_transport);
>+
>+  call->MediaTransportChange(&fake_media_transport);
>+  AudioSendStream* stream = call->CreateAudioSendStream(config);
>+
>+  // We get 2 subscribers: one subscriber from call.cc, and one from
>+  // ChannelSend.
>+  EXPECT_EQ(2, fake_media_transport.target_rate_observers_size());
>+
>+  call->DestroyAudioSendStream(stream);
>+  EXPECT_EQ(1, fake_media_transport.target_rate_observers_size());
>+
>+  call->MediaTransportChange(nullptr);
>+  EXPECT_EQ(0, fake_media_transport.target_rate_observers_size());
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.cc
>index fa7d0fac6696376cdcc39ce9eafd791aa4f5ef16..ab057be028bb5fc4fea7cb3680fc41a23e358433 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.cc
>@@ -30,7 +30,7 @@ bool ParseConfigParam(std::string exp_name, int* field) {
>   return (sscanf(group.c_str(), "%d", field) == 1);
> }
> 
>-absl::optional<webrtc::DefaultNetworkSimulationConfig> ParseDegradationConfig(
>+absl::optional<webrtc::BuiltInNetworkBehaviorConfig> ParseDegradationConfig(
>     bool send) {
>   std::string exp_prefix = "WebRTCFakeNetwork";
>   if (send) {
>@@ -39,7 +39,7 @@ absl::optional<webrtc::DefaultNetworkSimulationConfig> ParseDegradationConfig(
>     exp_prefix += "Receive";
>   }
> 
>-  webrtc::DefaultNetworkSimulationConfig config;
>+  webrtc::BuiltInNetworkBehaviorConfig config;
>   bool configured = false;
>   configured |=
>       ParseConfigParam(exp_prefix + "DelayMs", &config.queue_delay_ms);
>@@ -63,15 +63,15 @@ absl::optional<webrtc::DefaultNetworkSimulationConfig> ParseDegradationConfig(
>   configured |= ParseConfigParam(exp_prefix + "AvgBurstLossLength",
>                                  &config.avg_burst_loss_length);
>   return configured
>-             ? absl::optional<webrtc::DefaultNetworkSimulationConfig>(config)
>+             ? absl::optional<webrtc::BuiltInNetworkBehaviorConfig>(config)
>              : absl::nullopt;
> }
> }  // namespace
> 
> Call* CallFactory::CreateCall(const Call::Config& config) {
>-  absl::optional<webrtc::DefaultNetworkSimulationConfig>
>-      send_degradation_config = ParseDegradationConfig(true);
>-  absl::optional<webrtc::DefaultNetworkSimulationConfig>
>+  absl::optional<webrtc::BuiltInNetworkBehaviorConfig> send_degradation_config =
>+      ParseDegradationConfig(true);
>+  absl::optional<webrtc::BuiltInNetworkBehaviorConfig>
>       receive_degradation_config = ParseDegradationConfig(false);
> 
>   if (send_degradation_config || receive_degradation_config) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.cc
>index 8181310e78ccbcb7752e29fa32e431f01d540050..a7ef41d490579d27581f6e7556c29a3a6327cb3c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.cc
>@@ -17,8 +17,8 @@
> namespace webrtc {
> DegradedCall::DegradedCall(
>     std::unique_ptr<Call> call,
>-    absl::optional<DefaultNetworkSimulationConfig> send_config,
>-    absl::optional<DefaultNetworkSimulationConfig> receive_config)
>+    absl::optional<BuiltInNetworkBehaviorConfig> send_config,
>+    absl::optional<BuiltInNetworkBehaviorConfig> receive_config)
>     : clock_(Clock::GetRealTimeClock()),
>       call_(std::move(call)),
>       send_config_(send_config),
>@@ -183,6 +183,8 @@ bool DegradedCall::SendRtp(const uint8_t* packet,
>     rtc::SentPacket sent_packet;
>     sent_packet.packet_id = options.packet_id;
>     sent_packet.send_time_ms = clock_->TimeInMilliseconds();
>+    sent_packet.info.included_in_feedback = options.included_in_feedback;
>+    sent_packet.info.included_in_allocation = options.included_in_allocation;
>     sent_packet.info.packet_size_bytes = length;
>     sent_packet.info.packet_type = rtc::PacketType::kData;
>     call_->OnSentPacket(sent_packet);
>@@ -213,4 +215,10 @@ PacketReceiver::DeliveryStatus DegradedCall::DeliverPacket(
>   return status;
> }
> 
>+void DegradedCall::MediaTransportChange(
>+    MediaTransportInterface* media_transport) {
>+  // TODO(bugs.webrtc.org/9719) We should add support for media transport here
>+  // at some point.
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.h
>index 3c0b80df86ad35ae9d22fa239417af3bf9d69976..d78b1d1026cfe8b1e1658db98c47429936e238c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.h
>@@ -28,8 +28,8 @@ class DegradedCall : public Call, private Transport, private PacketReceiver {
>  public:
>   explicit DegradedCall(
>       std::unique_ptr<Call> call,
>-      absl::optional<DefaultNetworkSimulationConfig> send_config,
>-      absl::optional<DefaultNetworkSimulationConfig> receive_config);
>+      absl::optional<BuiltInNetworkBehaviorConfig> send_config,
>+      absl::optional<BuiltInNetworkBehaviorConfig> receive_config);
>   ~DegradedCall() override;
> 
>   // Implements Call.
>@@ -91,13 +91,14 @@ class DegradedCall : public Call, private Transport, private PacketReceiver {
>   Clock* const clock_;
>   const std::unique_ptr<Call> call_;
> 
>-  const absl::optional<DefaultNetworkSimulationConfig> send_config_;
>+  void MediaTransportChange(MediaTransportInterface* media_transport) override;
>+  const absl::optional<BuiltInNetworkBehaviorConfig> send_config_;
>   const std::unique_ptr<ProcessThread> send_process_thread_;
>   SimulatedNetwork* send_simulated_network_;
>   std::unique_ptr<FakeNetworkPipe> send_pipe_;
>   size_t num_send_streams_;
> 
>-  const absl::optional<DefaultNetworkSimulationConfig> receive_config_;
>+  const absl::optional<BuiltInNetworkBehaviorConfig> receive_config_;
>   SimulatedNetwork* receive_simulated_network_;
>   std::unique_ptr<FakeNetworkPipe> receive_pipe_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.cc
>index e702c1b6f5c7ef8b7385cacddd0c9481d24a0717..3bd2e8f48b16a22c6143f5b676ed8ef92b9b8d47 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.cc
>@@ -18,6 +18,7 @@
> #include "call/call.h"
> #include "call/fake_network_pipe.h"
> #include "call/simulated_network.h"
>+#include "modules/utility/include/process_thread.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/clock.h"
> 
>@@ -90,7 +91,6 @@ FakeNetworkPipe::FakeNetworkPipe(
>       dropped_packets_(0),
>       sent_packets_(0),
>       total_packet_delay_us_(0),
>-      next_process_time_us_(clock_->TimeInMicroseconds()),
>       last_log_time_us_(clock_->TimeInMicroseconds()) {}
> 
> FakeNetworkPipe::FakeNetworkPipe(
>@@ -105,7 +105,6 @@ FakeNetworkPipe::FakeNetworkPipe(
>       dropped_packets_(0),
>       sent_packets_(0),
>       total_packet_delay_us_(0),
>-      next_process_time_us_(clock_->TimeInMicroseconds()),
>       last_log_time_us_(clock_->TimeInMicroseconds()) {}
> 
> FakeNetworkPipe::~FakeNetworkPipe() = default;
>@@ -154,8 +153,8 @@ bool FakeNetworkPipe::EnqueuePacket(rtc::CopyOnWriteBuffer packet,
>                                     bool is_rtcp,
>                                     MediaType media_type,
>                                     absl::optional<int64_t> packet_time_us) {
>-  int64_t time_now_us = clock_->TimeInMicroseconds();
>   rtc::CritScope crit(&process_lock_);
>+  int64_t time_now_us = clock_->TimeInMicroseconds();
>   size_t packet_size = packet.size();
>   NetworkPacket net_packet(std::move(packet), time_now_us, time_now_us, options,
>                            is_rtcp, media_type, packet_time_us);
>@@ -169,6 +168,12 @@ bool FakeNetworkPipe::EnqueuePacket(rtc::CopyOnWriteBuffer packet,
>     packets_in_flight_.pop_back();
>     ++dropped_packets_;
>   }
>+  if (network_behavior_->NextDeliveryTimeUs()) {
>+    rtc::CritScope crit(&process_thread_lock_);
>+    if (process_thread_)
>+      process_thread_->WakeUp(nullptr);
>+  }
>+
>   return sent;
> }
> 
>@@ -201,10 +206,11 @@ size_t FakeNetworkPipe::SentPackets() {
> }
> 
> void FakeNetworkPipe::Process() {
>-  int64_t time_now_us = clock_->TimeInMicroseconds();
>+  int64_t time_now_us;
>   std::queue<NetworkPacket> packets_to_deliver;
>   {
>     rtc::CritScope crit(&process_lock_);
>+    time_now_us = clock_->TimeInMicroseconds();
>     if (time_now_us - last_log_time_us_ > kLogIntervalMs * 1000) {
>       int64_t queueing_delay_us = 0;
>       if (!packets_in_flight_.empty())
>@@ -250,9 +256,11 @@ void FakeNetworkPipe::Process() {
>         // arrived, due to NetworkProcess being called too late. For stats, use
>         // the time it should have been on the link.
>         total_packet_delay_us_ += added_delay_us;
>+        ++sent_packets_;
>+      } else {
>+        ++dropped_packets_;
>       }
>     }
>-    sent_packets_ += packets_to_deliver.size();
>   }
> 
>   rtc::CritScope crit(&config_lock_);
>@@ -261,10 +269,6 @@ void FakeNetworkPipe::Process() {
>     packets_to_deliver.pop();
>     DeliverNetworkPacket(&packet);
>   }
>-  absl::optional<int64_t> delivery_us = network_behavior_->NextDeliveryTimeUs();
>-  next_process_time_us_ = delivery_us
>-                              ? *delivery_us
>-                              : time_now_us + kDefaultProcessIntervalMs * 1000;
> }
> 
> void FakeNetworkPipe::DeliverNetworkPacket(NetworkPacket* packet) {
>@@ -291,8 +295,17 @@ void FakeNetworkPipe::DeliverNetworkPacket(NetworkPacket* packet) {
> 
> int64_t FakeNetworkPipe::TimeUntilNextProcess() {
>   rtc::CritScope crit(&process_lock_);
>-  int64_t delay_us = next_process_time_us_ - clock_->TimeInMicroseconds();
>-  return std::max<int64_t>((delay_us + 500) / 1000, 0);
>+  absl::optional<int64_t> delivery_us = network_behavior_->NextDeliveryTimeUs();
>+  if (delivery_us) {
>+    int64_t delay_us = *delivery_us - clock_->TimeInMicroseconds();
>+    return std::max<int64_t>((delay_us + 500) / 1000, 0);
>+  }
>+  return kDefaultProcessIntervalMs;
>+}
>+
>+void FakeNetworkPipe::ProcessThreadAttached(ProcessThread* process_thread) {
>+  rtc::CritScope cs(&process_thread_lock_);
>+  process_thread_ = process_thread;
> }
> 
> bool FakeNetworkPipe::HasTransport() const {
>@@ -316,31 +329,8 @@ void FakeNetworkPipe::ResetStats() {
>   total_packet_delay_us_ = 0;
> }
> 
>-void FakeNetworkPipe::AddToPacketDropCount() {
>-  rtc::CritScope crit(&process_lock_);
>-  ++dropped_packets_;
>-}
>-
>-void FakeNetworkPipe::AddToPacketSentCount(int count) {
>-  rtc::CritScope crit(&process_lock_);
>-  sent_packets_ += count;
>-}
>-
>-void FakeNetworkPipe::AddToTotalDelay(int delay_us) {
>-  rtc::CritScope crit(&process_lock_);
>-  total_packet_delay_us_ += delay_us;
>-}
>-
> int64_t FakeNetworkPipe::GetTimeInMicroseconds() const {
>   return clock_->TimeInMicroseconds();
> }
> 
>-bool FakeNetworkPipe::ShouldProcess(int64_t time_now_us) const {
>-  return time_now_us >= next_process_time_us_;
>-}
>-
>-void FakeNetworkPipe::SetTimeToNextProcess(int64_t skip_us) {
>-  next_process_time_us_ += skip_us;
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.h
>index f85e82efc5ee46edc40a9d10e5dc9a2642f0f966..b1b4cee14d0e61d101927c94e9b7b4e5818fb718 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.h
>@@ -23,7 +23,6 @@
> #include "api/test/simulated_network.h"
> #include "call/call.h"
> #include "call/simulated_packet_receiver.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
>@@ -140,6 +139,7 @@ class FakeNetworkPipe : public webrtc::SimulatedPacketReceiverInterface,
>   // packets ready to be delivered.
>   void Process() override;
>   int64_t TimeUntilNextProcess() override;
>+  void ProcessThreadAttached(ProcessThread* process_thread) override;
> 
>   // Get statistics.
>   float PercentageLoss();
>@@ -150,9 +150,6 @@ class FakeNetworkPipe : public webrtc::SimulatedPacketReceiverInterface,
> 
>  protected:
>   void DeliverPacketWithLock(NetworkPacket* packet);
>-  void AddToPacketDropCount();
>-  void AddToPacketSentCount(int count);
>-  void AddToTotalDelay(int delay_us);
>   int64_t GetTimeInMicroseconds() const;
>   bool ShouldProcess(int64_t time_now_us) const;
>   void SetTimeToNextProcess(int64_t skip_us);
>@@ -197,6 +194,9 @@ class FakeNetworkPipe : public webrtc::SimulatedPacketReceiverInterface,
>   // processes, such as the packet queues.
>   rtc::CriticalSection process_lock_;
> 
>+  rtc::CriticalSection process_thread_lock_;
>+  ProcessThread* process_thread_ RTC_GUARDED_BY(process_thread_lock_) = nullptr;
>+
>   // Packets  are added at the back of the deque, this makes the deque ordered
>   // by increasing send time. The common case when removing packets from the
>   // deque is removing early packets, which will be close to the front of the
>@@ -210,9 +210,6 @@ class FakeNetworkPipe : public webrtc::SimulatedPacketReceiverInterface,
>   size_t dropped_packets_ RTC_GUARDED_BY(process_lock_);
>   size_t sent_packets_ RTC_GUARDED_BY(process_lock_);
>   int64_t total_packet_delay_us_ RTC_GUARDED_BY(process_lock_);
>-
>-  int64_t next_process_time_us_;
>-
>   int64_t last_log_time_us_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.cc
>index ab138368ba0ea998166c6a9f92226e6d84d298da..ab6dde37b46d769cdd9cb6e31d1cca355ff85da1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.cc
>@@ -10,6 +10,8 @@
> 
> #include "call/flexfec_receive_stream.h"
> 
>+#include "rtc_base/checks.h"
>+
> namespace webrtc {
> 
> FlexfecReceiveStream::Config::Config(Transport* rtcp_send_transport)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.h
>index ccc301430e844a388bdca8179f22b480081cd7d3..949ad742aa03f66d721ab50b875e1bcbaf5b3a29 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.h
>@@ -12,7 +12,6 @@
> #define CALL_FLEXFEC_RECEIVE_STREAM_H_
> 
> #include <stdint.h>
>-
> #include <string>
> #include <vector>
> 
>@@ -20,7 +19,6 @@
> #include "api/rtp_headers.h"
> #include "api/rtpparameters.h"
> #include "call/rtp_packet_sink_interface.h"
>-#include "common_types.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.h
>index f05e409d79d136c98c082d509cf906d37d756a06..02a01550def65234a89417668d784c555cc6b03c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.h
>@@ -16,7 +16,6 @@
> #include <vector>
> 
> #include "api/mediatypes.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/copyonwritebuffer.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.cc
>index 8eb451ac04f73a7b85aebc82effe40b7b156999b..9735421f30ec51cb0ebc5f61e56fced0525de60b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.cc
>@@ -11,7 +11,9 @@
> #include "call/rampup_tests.h"
> 
> #include "call/fake_network_pipe.h"
>+#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/flags.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/platform_thread.h"
> #include "rtc_base/stringencode.h"
>@@ -38,6 +40,10 @@ std::vector<uint32_t> GenerateSsrcs(size_t num_streams, uint32_t ssrc_offset) {
> }
> }  // namespace
> 
>+WEBRTC_DEFINE_string(ramp_dump_name,
>+                     "",
>+                     "Filename for dumped received RTP stream.");
>+
> RampUpTester::RampUpTester(size_t num_video_streams,
>                            size_t num_audio_streams,
>                            size_t num_flexfec_streams,
>@@ -48,7 +54,6 @@ RampUpTester::RampUpTester(size_t num_video_streams,
>                            bool red,
>                            bool report_perf_stats)
>     : EndToEndTest(test::CallTest::kLongTimeoutMs),
>-      stop_event_(false, false),
>       clock_(Clock::GetRealTimeClock()),
>       num_video_streams_(num_video_streams),
>       num_audio_streams_(num_audio_streams),
>@@ -79,11 +84,12 @@ RampUpTester::RampUpTester(size_t num_video_streams,
> 
> RampUpTester::~RampUpTester() {}
> 
>-void RampUpTester::ModifySenderCallConfig(Call::Config* config) {
>+void RampUpTester::ModifySenderBitrateConfig(
>+    BitrateConstraints* bitrate_config) {
>   if (start_bitrate_bps_ != 0) {
>-    config->bitrate_config.start_bitrate_bps = start_bitrate_bps_;
>+    bitrate_config->start_bitrate_bps = start_bitrate_bps_;
>   }
>-  config->bitrate_config.min_bitrate_bps = 10000;
>+  bitrate_config->min_bitrate_bps = 10000;
> }
> 
> void RampUpTester::OnVideoStreamsCreated(
>@@ -448,8 +454,9 @@ void RampUpDownUpTester::PollStats() {
>   } while (!stop_event_.Wait(kPollIntervalMs));
> }
> 
>-void RampUpDownUpTester::ModifyReceiverCallConfig(Call::Config* config) {
>-  config->bitrate_config.min_bitrate_bps = 10000;
>+void RampUpDownUpTester::ModifyReceiverBitrateConfig(
>+    BitrateConstraints* bitrate_config) {
>+  bitrate_config->min_bitrate_bps = 10000;
> }
> 
> std::string RampUpDownUpTester::GetModifierString() const {
>@@ -566,7 +573,23 @@ void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) {
> 
> class RampUpTest : public test::CallTest {
>  public:
>-  RampUpTest() {}
>+  RampUpTest() {
>+    std::string dump_name(FLAG_ramp_dump_name);
>+    if (!dump_name.empty()) {
>+      send_event_log_ = RtcEventLog::Create(RtcEventLog::EncodingType::Legacy);
>+      recv_event_log_ = RtcEventLog::Create(RtcEventLog::EncodingType::Legacy);
>+      bool event_log_started =
>+          send_event_log_->StartLogging(
>+              absl::make_unique<RtcEventLogOutputFile>(
>+                  dump_name + ".send.rtc.dat", RtcEventLog::kUnlimitedOutput),
>+              RtcEventLog::kImmediateOutput) &&
>+          recv_event_log_->StartLogging(
>+              absl::make_unique<RtcEventLogOutputFile>(
>+                  dump_name + ".recv.rtc.dat", RtcEventLog::kUnlimitedOutput),
>+              RtcEventLog::kImmediateOutput);
>+      RTC_DCHECK(event_log_started);
>+    }
>+  }
> };
> 
> static const uint32_t kStartBitrateBps = 60000;
>@@ -667,13 +690,7 @@ TEST_F(RampUpTest, TransportSequenceNumberSimulcastRedRtx) {
>   RunBaseTest(&test);
> }
> 
>-// TODO(bugs.webrtc.org/8878)
>-#if defined(WEBRTC_MAC)
>-#define MAYBE_AudioTransportSequenceNumber DISABLED_AudioTransportSequenceNumber
>-#else
>-#define MAYBE_AudioTransportSequenceNumber AudioTransportSequenceNumber
>-#endif
>-TEST_F(RampUpTest, MAYBE_AudioTransportSequenceNumber) {
>+TEST_F(RampUpTest, AudioTransportSequenceNumber) {
>   RampUpTester test(0, 1, 0, 300000, 10000,
>                     RtpExtension::kTransportSequenceNumberUri, false, false,
>                     false);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.h
>index 6cc65ce2a2243488ec91a5de3c337ae315f90997..b7d4af5f5a8898fafa0b6b3a1345706cb82c9759 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.h
>@@ -67,7 +67,7 @@ class RampUpTester : public test::EndToEndTest {
> 
>   rtc::Event stop_event_;
>   Clock* const clock_;
>-  DefaultNetworkSimulationConfig forward_transport_config_;
>+  BuiltInNetworkBehaviorConfig forward_transport_config_;
>   const size_t num_video_streams_;
>   const size_t num_audio_streams_;
>   const size_t num_flexfec_streams_;
>@@ -83,7 +83,7 @@ class RampUpTester : public test::EndToEndTest {
>   typedef std::map<uint32_t, uint32_t> SsrcMap;
>   class VideoStreamFactory;
> 
>-  void ModifySenderCallConfig(Call::Config* config) override;
>+  void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) override;
>   void OnVideoStreamsCreated(
>       VideoSendStream* send_stream,
>       const std::vector<VideoReceiveStream*>& receive_streams) override;
>@@ -142,7 +142,7 @@ class RampUpDownUpTester : public RampUpTester {
>     kTransitionToNextState,
>   };
> 
>-  void ModifyReceiverCallConfig(Call::Config* config);
>+  void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config) override;
> 
>   std::string GetModifierString() const;
>   int GetExpectedHighBitrate() const;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.cc
>index 16c6a43cc3ffc05492916a194604af5dc7fc1ff2..b7056732e8b65a8822c91ac5fd56d51c9416d38d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.cc
>@@ -9,59 +9,112 @@
>  */
> 
> #include "call/receive_time_calculator.h"
>+
>+#include <string>
>+
> #include "absl/memory/memory.h"
>+#include "rtc_base/experiments/field_trial_parser.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_minmax.h"
> #include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> namespace {
>-using ::webrtc::field_trial::FindFullName;
> using ::webrtc::field_trial::IsEnabled;
> 
>-const char kBweReceiveTimeCorrection[] = "WebRTC-BweReceiveTimeCorrection";
>+const char kBweReceiveTimeCorrection[] = "WebRTC-Bwe-ReceiveTimeFix";
> }  // namespace
> 
>-ReceiveTimeCalculator::ReceiveTimeCalculator(int64_t min_delta_ms,
>-                                             int64_t max_delta_diff_ms)
>-    : min_delta_us_(min_delta_ms * 1000),
>-      max_delta_diff_us_(max_delta_diff_ms * 1000) {}
>+ReceiveTimeCalculatorConfig::ReceiveTimeCalculatorConfig()
>+    : max_packet_time_repair("maxrep", TimeDelta::ms(2000)),
>+      stall_threshold("stall", TimeDelta::ms(5)),
>+      tolerance("tol", TimeDelta::ms(1)),
>+      max_stall("maxstall", TimeDelta::seconds(5)) {
>+  std::string trial_string =
>+      field_trial::FindFullName(kBweReceiveTimeCorrection);
>+  ParseFieldTrial(
>+      {&max_packet_time_repair, &stall_threshold, &tolerance, &max_stall},
>+      trial_string);
>+}
>+ReceiveTimeCalculatorConfig::ReceiveTimeCalculatorConfig(
>+    const ReceiveTimeCalculatorConfig&) = default;
>+ReceiveTimeCalculatorConfig::~ReceiveTimeCalculatorConfig() = default;
>+
>+ReceiveTimeCalculator::ReceiveTimeCalculator()
>+    : config_(ReceiveTimeCalculatorConfig()) {}
> 
> std::unique_ptr<ReceiveTimeCalculator>
> ReceiveTimeCalculator::CreateFromFieldTrial() {
>   if (!IsEnabled(kBweReceiveTimeCorrection))
>     return nullptr;
>-  int min, max;
>-  if (sscanf(FindFullName(kBweReceiveTimeCorrection).c_str(), "Enabled,%d,%d",
>-             &min, &max) != 2) {
>-    RTC_LOG(LS_WARNING) << "Invalid number of parameters provided.";
>-    return nullptr;
>-  }
>-  return absl::make_unique<ReceiveTimeCalculator>(min, max);
>+  return absl::make_unique<ReceiveTimeCalculator>();
> }
> 
>-int64_t ReceiveTimeCalculator::ReconcileReceiveTimes(int64_t packet_time_us_,
>-                                                     int64_t safe_time_us_) {
>-  if (!receive_time_offset_us_) {
>-    receive_time_offset_us_ = safe_time_us_ - packet_time_us_;
>-  } else {
>-    int64_t safe_delta_us = safe_time_us_ - last_safe_time_us_;
>-    int64_t packet_delta_us_ = packet_time_us_ - last_packet_time_us_;
>-    int64_t delta_diff = packet_delta_us_ - safe_delta_us;
>-    // Packet time should not decrease significantly, a large decrease indicates
>-    // a reset of the packet time clock and we should reset the offest
>-    // parameter. The safe reference time can increase in large jumps if the
>-    // thread measuring it is backgrounded for longer periods. But if the packet
>-    // time increases significantly more than the safe time, it indicates a
>-    // clock reset and we should reset the offset.
>-
>-    if (packet_delta_us_ < min_delta_us_ || delta_diff > max_delta_diff_us_) {
>-      RTC_LOG(LS_WARNING) << "Received a clock jump of " << delta_diff
>-                          << " resetting offset";
>-      receive_time_offset_us_ = safe_time_us_ - packet_time_us_;
>+int64_t ReceiveTimeCalculator::ReconcileReceiveTimes(int64_t packet_time_us,
>+                                                     int64_t system_time_us,
>+                                                     int64_t safe_time_us) {
>+  int64_t stall_time_us = system_time_us - packet_time_us;
>+  if (total_system_time_passed_us_ < config_.stall_threshold->us()) {
>+    stall_time_us = rtc::SafeMin(stall_time_us, config_.max_stall->us());
>+  }
>+  int64_t corrected_time_us = safe_time_us - stall_time_us;
>+
>+  if (last_packet_time_us_ == -1 && stall_time_us < 0) {
>+    static_clock_offset_us_ = stall_time_us;
>+    corrected_time_us += static_clock_offset_us_;
>+  } else if (last_packet_time_us_ > 0) {
>+    // All repairs depend on variables being intialized
>+    int64_t packet_time_delta_us = packet_time_us - last_packet_time_us_;
>+    int64_t system_time_delta_us = system_time_us - last_system_time_us_;
>+    int64_t safe_time_delta_us = safe_time_us - last_safe_time_us_;
>+
>+    // Repair backwards clock resets during initial stall. In this case, the
>+    // reset is observed only in packet time but never in system time.
>+    if (system_time_delta_us < 0)
>+      total_system_time_passed_us_ += config_.stall_threshold->us();
>+    else
>+      total_system_time_passed_us_ += system_time_delta_us;
>+    if (packet_time_delta_us < 0 &&
>+        total_system_time_passed_us_ < config_.stall_threshold->us()) {
>+      static_clock_offset_us_ -= packet_time_delta_us;
>+    }
>+    corrected_time_us += static_clock_offset_us_;
>+
>+    // Detect resets inbetween clock readings in socket and app.
>+    bool forward_clock_reset =
>+        corrected_time_us + config_.tolerance->us() < last_corrected_time_us_;
>+    bool obvious_backward_clock_reset = system_time_us < packet_time_us;
>+
>+    // Harder case with backward clock reset during stall, the reset being
>+    // smaller than the stall. Compensate throughout the stall.
>+    bool small_backward_clock_reset =
>+        !obvious_backward_clock_reset &&
>+        safe_time_delta_us > system_time_delta_us + config_.tolerance->us();
>+    bool stall_start =
>+        packet_time_delta_us >= 0 &&
>+        system_time_delta_us > packet_time_delta_us + config_.tolerance->us();
>+    bool stall_is_over = safe_time_delta_us > config_.stall_threshold->us();
>+    bool packet_time_caught_up =
>+        packet_time_delta_us < 0 && system_time_delta_us >= 0;
>+    if (stall_start && small_backward_clock_reset)
>+      small_reset_during_stall_ = true;
>+    else if (stall_is_over || packet_time_caught_up)
>+      small_reset_during_stall_ = false;
>+
>+    // If resets are detected, advance time by (capped) packet time increase.
>+    if (forward_clock_reset || obvious_backward_clock_reset ||
>+        small_reset_during_stall_) {
>+      corrected_time_us = last_corrected_time_us_ +
>+                          rtc::SafeClamp(packet_time_delta_us, 0,
>+                                         config_.max_packet_time_repair->us());
>     }
>   }
>-  last_packet_time_us_ = packet_time_us_;
>-  last_safe_time_us_ = safe_time_us_;
>-  return packet_time_us_ + *receive_time_offset_us_;
>+
>+  last_corrected_time_us_ = corrected_time_us;
>+  last_packet_time_us_ = packet_time_us;
>+  last_system_time_us_ = system_time_us;
>+  last_safe_time_us_ = safe_time_us;
>+  return corrected_time_us;
> }
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.h
>index a8217cd101259f8b381adb082279d8375dc21f14..269d4ef91009084d20dc04f00102426080c4267c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.h
>@@ -10,13 +10,24 @@
> #ifndef CALL_RECEIVE_TIME_CALCULATOR_H_
> #define CALL_RECEIVE_TIME_CALCULATOR_H_
> 
>-#include <stdint.h>
> #include <memory>
> 
>-#include "absl/types/optional.h"
>+#include "rtc_base/experiments/field_trial_units.h"
> 
> namespace webrtc {
> 
>+struct ReceiveTimeCalculatorConfig {
>+  ReceiveTimeCalculatorConfig();
>+  ReceiveTimeCalculatorConfig(const ReceiveTimeCalculatorConfig&);
>+  ReceiveTimeCalculatorConfig& operator=(const ReceiveTimeCalculatorConfig&) =
>+      default;
>+  ~ReceiveTimeCalculatorConfig();
>+  FieldTrialParameter<TimeDelta> max_packet_time_repair;
>+  FieldTrialParameter<TimeDelta> stall_threshold;
>+  FieldTrialParameter<TimeDelta> tolerance;
>+  FieldTrialParameter<TimeDelta> max_stall;
>+};
>+
> // The receive time calculator serves the purpose of combining packet time
> // stamps with a safely incremental clock. This assumes that the packet time
> // stamps are based on lower layer timestamps that have more accurate time
>@@ -28,20 +39,20 @@ namespace webrtc {
> class ReceiveTimeCalculator {
>  public:
>   static std::unique_ptr<ReceiveTimeCalculator> CreateFromFieldTrial();
>-  // The min delta is used to correct for jumps backwards in time, to allow some
>-  // packet reordering a small negative value is appropriate to use. The max
>-  // delta difference is used as margin when detecting when packet time
>-  // increases more than the safe clock. This should be larger than the largest
>-  // expected sysmtem induced delay in the safe clock timestamp.
>-  ReceiveTimeCalculator(int64_t min_delta_ms, int64_t max_delta_diff_ms);
>-  int64_t ReconcileReceiveTimes(int64_t packet_time_us_, int64_t safe_time_us_);
>+  ReceiveTimeCalculator();
>+  int64_t ReconcileReceiveTimes(int64_t packet_time_us_,
>+                                int64_t system_time_us_,
>+                                int64_t safe_time_us_);
> 
>  private:
>-  const int64_t min_delta_us_;
>-  const int64_t max_delta_diff_us_;
>-  absl::optional<int64_t> receive_time_offset_us_;
>-  int64_t last_packet_time_us_ = 0;
>-  int64_t last_safe_time_us_ = 0;
>+  int64_t last_corrected_time_us_ = -1;
>+  int64_t last_packet_time_us_ = -1;
>+  int64_t last_system_time_us_ = -1;
>+  int64_t last_safe_time_us_ = -1;
>+  int64_t total_system_time_passed_us_ = 0;
>+  int64_t static_clock_offset_us_ = 0;
>+  int64_t small_reset_during_stall_ = false;
>+  ReceiveTimeCalculatorConfig config_;
> };
> }  // namespace webrtc
> #endif  // CALL_RECEIVE_TIME_CALCULATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator_unittest.cc
>index 92d5a279eae11242b786b77c15fe425edfa29ba2..38ef54ece794112cc9dad815c24b49513c70d3aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator_unittest.cc
>@@ -10,65 +10,237 @@
> 
> #include "call/receive_time_calculator.h"
> 
>+#include <algorithm>
>+#include <iostream>
>+#include <tuple>
>+#include <vector>
>+
>+#include "absl/memory/memory.h"
>+#include "absl/types/optional.h"
>+#include "rtc_base/random.h"
>+#include "rtc_base/timeutils.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> namespace test {
> namespace {
> 
>-int64_t ReconcileMs(ReceiveTimeCalculator* calc,
>-                    int64_t packet_time_ms,
>-                    int64_t safe_time_ms) {
>-  return calc->ReconcileReceiveTimes(packet_time_ms * 1000,
>-                                     safe_time_ms * 1000) /
>-         1000;
>-}
>-}  // namespace
>+class EmulatedClock {
>+ public:
>+  explicit EmulatedClock(int seed, float drift = 0.0f)
>+      : random_(seed), clock_us_(random_.Rand<uint32_t>()), drift_(drift) {}
>+  virtual ~EmulatedClock() = default;
>+  int64_t GetClockUs() const { return clock_us_; }
> 
>-TEST(ReceiveTimeCalculatorTest, UsesSmallerIncrements) {
>-  int64_t kMinDeltaMs = -20;
>-  int64_t kMaxDeltaDiffMs = 100;
>-  ReceiveTimeCalculator calc(kMinDeltaMs, kMaxDeltaDiffMs);
>-  // Initialize offset.
>-  ReconcileMs(&calc, 10000, 20000);
>-
>-  EXPECT_EQ(ReconcileMs(&calc, 10010, 20100), 20010);
>-  EXPECT_EQ(ReconcileMs(&calc, 10020, 20100), 20020);
>-  EXPECT_EQ(ReconcileMs(&calc, 10030, 20100), 20030);
>-
>-  EXPECT_EQ(ReconcileMs(&calc, 10110, 20200), 20110);
>-  EXPECT_EQ(ReconcileMs(&calc, 10120, 20200), 20120);
>-  EXPECT_EQ(ReconcileMs(&calc, 10130, 20200), 20130);
>-
>-  // Small jumps backwards are let trough, they might be due to reordering.
>-  EXPECT_EQ(ReconcileMs(&calc, 10120, 20200), 20120);
>-  // The safe clock might be smaller than the packet clock.
>-  EXPECT_EQ(ReconcileMs(&calc, 10210, 20200), 20210);
>-  EXPECT_EQ(ReconcileMs(&calc, 10240, 20200), 20240);
>-}
>+ protected:
>+  int64_t UpdateClock(int64_t time_us) {
>+    if (!last_query_us_)
>+      last_query_us_ = time_us;
>+    int64_t skip_us = time_us - *last_query_us_;
>+    accumulated_drift_us_ += skip_us * drift_;
>+    int64_t drift_correction_us = static_cast<int64_t>(accumulated_drift_us_);
>+    accumulated_drift_us_ -= drift_correction_us;
>+    clock_us_ += skip_us + drift_correction_us;
>+    last_query_us_ = time_us;
>+    return skip_us;
>+  }
>+  Random random_;
> 
>-TEST(ReceiveTimeCalculatorTest, CorrectsJumps) {
>-  int64_t kMinDeltaMs = -20;
>-  int64_t kMaxDeltaDiffMs = 100;
>-  ReceiveTimeCalculator calc(kMinDeltaMs, kMaxDeltaDiffMs);
>-  // Initialize offset.
>-  ReconcileMs(&calc, 10000, 20000);
>-
>-  EXPECT_EQ(ReconcileMs(&calc, 10010, 20100), 20010);
>-  EXPECT_EQ(ReconcileMs(&calc, 10020, 20100), 20020);
>-  EXPECT_EQ(ReconcileMs(&calc, 10030, 20100), 20030);
>-
>-  // Jump forward in time.
>-  EXPECT_EQ(ReconcileMs(&calc, 10240, 20200), 20200);
>-  EXPECT_EQ(ReconcileMs(&calc, 10250, 20200), 20210);
>-  EXPECT_EQ(ReconcileMs(&calc, 10260, 20200), 20220);
>-
>-  // Jump backward in time.
>-  EXPECT_EQ(ReconcileMs(&calc, 10230, 20300), 20300);
>-  EXPECT_EQ(ReconcileMs(&calc, 10240, 20300), 20310);
>-  EXPECT_EQ(ReconcileMs(&calc, 10250, 20300), 20320);
>-}
>+ private:
>+  int64_t clock_us_;
>+  absl::optional<int64_t> last_query_us_;
>+  float drift_;
>+  float accumulated_drift_us_ = 0;
>+};
> 
>-}  // namespace test
>+class EmulatedMonotoneousClock : public EmulatedClock {
>+ public:
>+  explicit EmulatedMonotoneousClock(int seed) : EmulatedClock(seed) {}
>+  ~EmulatedMonotoneousClock() = default;
>+
>+  int64_t Query(int64_t time_us) {
>+    int64_t skip_us = UpdateClock(time_us);
>+
>+    // In a stall
>+    if (stall_recovery_time_us_ > 0) {
>+      if (GetClockUs() > stall_recovery_time_us_) {
>+        stall_recovery_time_us_ = 0;
>+        return GetClockUs();
>+      } else {
>+        return stall_recovery_time_us_;
>+      }
>+    }
>+
>+    // Check if we enter a stall
>+    for (int k = 0; k < skip_us; ++k) {
>+      if (random_.Rand<double>() < kChanceOfStallPerUs) {
>+        int64_t stall_duration_us =
>+            static_cast<int64_t>(random_.Rand<float>() * kMaxStallDurationUs);
>+        stall_recovery_time_us_ = GetClockUs() + stall_duration_us;
>+        return stall_recovery_time_us_;
>+      }
>+    }
>+    return GetClockUs();
>+  }
>+
>+  void ForceStallUs() {
>+    int64_t stall_duration_us =
>+        static_cast<int64_t>(random_.Rand<float>() * kMaxStallDurationUs);
>+    stall_recovery_time_us_ = GetClockUs() + stall_duration_us;
>+  }
>+
>+  bool Stalled() const { return stall_recovery_time_us_ > 0; }
>+
>+  int64_t GetRemainingStall(int64_t time_us) const {
>+    return stall_recovery_time_us_ > 0 ? stall_recovery_time_us_ - GetClockUs()
>+                                       : 0;
>+  }
>+
>+  const int64_t kMaxStallDurationUs = rtc::kNumMicrosecsPerSec;
>+
>+ private:
>+  const float kChanceOfStallPerUs = 5e-6f;
>+  int64_t stall_recovery_time_us_ = 0;
>+};
> 
>+class EmulatedNonMonotoneousClock : public EmulatedClock {
>+ public:
>+  EmulatedNonMonotoneousClock(int seed, int64_t duration_us, float drift = 0)
>+      : EmulatedClock(seed, drift) {
>+    Pregenerate(duration_us);
>+  }
>+  ~EmulatedNonMonotoneousClock() = default;
>+
>+  void Pregenerate(int64_t duration_us) {
>+    int64_t time_since_reset_us = kMinTimeBetweenResetsUs;
>+    int64_t clock_offset_us = 0;
>+    for (int64_t time_us = 0; time_us < duration_us; time_us += kResolutionUs) {
>+      int64_t skip_us = UpdateClock(time_us);
>+      time_since_reset_us += skip_us;
>+      int64_t reset_us = 0;
>+      if (time_since_reset_us >= kMinTimeBetweenResetsUs) {
>+        for (int k = 0; k < skip_us; ++k) {
>+          if (random_.Rand<double>() < kChanceOfResetPerUs) {
>+            reset_us = static_cast<int64_t>(2 * random_.Rand<float>() *
>+                                            kMaxAbsResetUs) -
>+                       kMaxAbsResetUs;
>+            clock_offset_us += reset_us;
>+            time_since_reset_us = 0;
>+            break;
>+          }
>+        }
>+      }
>+      pregenerated_clock_.emplace_back(GetClockUs() + clock_offset_us);
>+      resets_us_.emplace_back(reset_us);
>+    }
>+  }
>+
>+  int64_t Query(int64_t time_us) {
>+    size_t ixStart =
>+        (last_reset_query_time_us_ + (kResolutionUs >> 1)) / kResolutionUs + 1;
>+    size_t ixEnd = (time_us + (kResolutionUs >> 1)) / kResolutionUs;
>+    if (ixEnd >= pregenerated_clock_.size())
>+      return -1;
>+    last_reset_size_us_ = 0;
>+    for (size_t ix = ixStart; ix <= ixEnd; ++ix) {
>+      if (resets_us_[ix] != 0) {
>+        last_reset_size_us_ = resets_us_[ix];
>+      }
>+    }
>+    last_reset_query_time_us_ = time_us;
>+    return pregenerated_clock_[ixEnd];
>+  }
>+
>+  bool WasReset() const { return last_reset_size_us_ != 0; }
>+  bool WasNegativeReset() const { return last_reset_size_us_ < 0; }
>+  int64_t GetLastResetUs() const { return last_reset_size_us_; }
>+
>+ private:
>+  const float kChanceOfResetPerUs = 1e-6f;
>+  const int64_t kMaxAbsResetUs = rtc::kNumMicrosecsPerSec;
>+  const int64_t kMinTimeBetweenResetsUs = 3 * rtc::kNumMicrosecsPerSec;
>+  const int64_t kResolutionUs = rtc::kNumMicrosecsPerMillisec;
>+  int64_t last_reset_query_time_us_ = 0;
>+  int64_t last_reset_size_us_ = 0;
>+  std::vector<int64_t> pregenerated_clock_;
>+  std::vector<int64_t> resets_us_;
>+};
>+
>+TEST(ClockRepair, NoClockDrift) {
>+  const int kSeeds = 10;
>+  const int kFirstSeed = 1;
>+  const int64_t kRuntimeUs = 10 * rtc::kNumMicrosecsPerSec;
>+  const float kDrift = 0.0f;
>+  const int64_t kMaxPacketInterarrivalUs = 50 * rtc::kNumMicrosecsPerMillisec;
>+  for (int seed = kFirstSeed; seed < kSeeds + kFirstSeed; ++seed) {
>+    EmulatedMonotoneousClock monotone_clock(seed);
>+    EmulatedNonMonotoneousClock non_monotone_clock(
>+        seed + 1, kRuntimeUs + rtc::kNumMicrosecsPerSec, kDrift);
>+    ReceiveTimeCalculator reception_time_tracker;
>+    int64_t corrected_clock_0 = 0;
>+    int64_t reset_during_stall_tol_us = 0;
>+    bool initial_clock_stall = true;
>+    int64_t accumulated_upper_bound_tolerance_us = 0;
>+    int64_t accumulated_lower_bound_tolerance_us = 0;
>+    Random random(1);
>+    monotone_clock.ForceStallUs();
>+    int64_t last_time_us = 0;
>+    bool add_tolerance_on_next_packet = false;
>+    int64_t monotone_noise_us = 1000;
>+
>+    for (int64_t time_us = 0; time_us < kRuntimeUs;
>+         time_us += static_cast<int64_t>(random.Rand<float>() *
>+                                         kMaxPacketInterarrivalUs)) {
>+      int64_t socket_time_us = non_monotone_clock.Query(time_us);
>+      int64_t monotone_us = monotone_clock.Query(time_us) +
>+                            2 * random.Rand<float>() * monotone_noise_us -
>+                            monotone_noise_us;
>+      int64_t system_time_us = non_monotone_clock.Query(
>+          time_us + monotone_clock.GetRemainingStall(time_us));
>+
>+      int64_t corrected_clock_us = reception_time_tracker.ReconcileReceiveTimes(
>+          socket_time_us, system_time_us, monotone_us);
>+      if (time_us == 0)
>+        corrected_clock_0 = corrected_clock_us;
>+
>+      if (add_tolerance_on_next_packet)
>+        accumulated_lower_bound_tolerance_us -= (time_us - last_time_us);
>+
>+      // Perfect repair cannot be achiveved if non-monotone clock resets during
>+      // a monotone clock stall.
>+      add_tolerance_on_next_packet = false;
>+      if (monotone_clock.Stalled() && non_monotone_clock.WasReset()) {
>+        reset_during_stall_tol_us =
>+            std::max(reset_during_stall_tol_us, time_us - last_time_us);
>+        if (non_monotone_clock.WasNegativeReset()) {
>+          add_tolerance_on_next_packet = true;
>+        }
>+        if (initial_clock_stall && !non_monotone_clock.WasNegativeReset()) {
>+          // Positive resets during an initial clock stall cannot be repaired
>+          // and error will propagate through rest of trace.
>+          accumulated_upper_bound_tolerance_us +=
>+              std::abs(non_monotone_clock.GetLastResetUs());
>+        }
>+      } else {
>+        reset_during_stall_tol_us = 0;
>+        initial_clock_stall = false;
>+      }
>+      int64_t err = corrected_clock_us - corrected_clock_0 - time_us;
>+
>+      // Resets during stalls may lead to small errors temporarily.
>+      int64_t lower_tol_us = accumulated_lower_bound_tolerance_us -
>+                             reset_during_stall_tol_us - monotone_noise_us -
>+                             2 * rtc::kNumMicrosecsPerMillisec;
>+      EXPECT_GE(err, lower_tol_us);
>+      int64_t upper_tol_us = accumulated_upper_bound_tolerance_us +
>+                             monotone_noise_us +
>+                             2 * rtc::kNumMicrosecsPerMillisec;
>+      EXPECT_LE(err, upper_tol_us);
>+
>+      last_time_us = time_us;
>+    }
>+  }
>+}
>+}  // namespace
>+}  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.cc
>index d441c05b3db85bc3a6e99b84d498717c4e7e8686..a06d2a4604c403807c0a28a6df68258321438bd2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.cc
>@@ -13,7 +13,6 @@
> #include "api/rtp_headers.h"
> #include "call/rtcp_packet_sink_interface.h"
> #include "call/rtp_rtcp_demuxer_helper.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer_unittest.cc
>index 808364acc5e8434674bc1b29f405ce7f72ace76a..6ceb7ab5d9d69e937f1468a08d2050b17b801059 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer_unittest.cc
>@@ -16,7 +16,6 @@
> #include "absl/memory/memory.h"
> #include "api/rtp_headers.h"
> #include "call/rtcp_packet_sink_interface.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.cc
>index 1445c2552e492948e626998d7e0bc8fabf7914b0..cce78ad64c7d91130ff22e9b40543c32d9463793 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.cc
>@@ -10,6 +10,9 @@
> 
> #include "call/rtp_config.h"
> 
>+#include <cstdint>
>+
>+#include "api/array_view.h"
> #include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
>@@ -60,6 +63,7 @@ std::string RtpConfig::ToString() const {
>      << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
>                                           : "RtcpMode::kReducedSize");
>   ss << ", max_packet_size: " << max_packet_size;
>+  ss << ", extmap-allow-mixed: " << (extmap_allow_mixed ? "true" : "false");
>   ss << ", extensions: [";
>   for (size_t i = 0; i < extensions.size(); ++i) {
>     ss << extensions[i].ToString();
>@@ -108,18 +112,4 @@ std::string RtpConfig::Rtx::ToString() const {
>   ss << '}';
>   return ss.str();
> }
>-
>-RtcpConfig::RtcpConfig() = default;
>-RtcpConfig::RtcpConfig(const RtcpConfig&) = default;
>-RtcpConfig::~RtcpConfig() = default;
>-
>-std::string RtcpConfig::ToString() const {
>-  char buf[1024];
>-  rtc::SimpleStringBuilder ss(buf);
>-  ss << "{video_report_interval_ms: " << video_report_interval_ms;
>-  ss << ", audio_report_interval_ms: " << audio_report_interval_ms;
>-  ss << '}';
>-  return ss.str();
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.h
>index 8f8d03a3a8242a6b19f5c5f65bd1c2b6386b6a05..97ae951765c2bd23e452a1fd140e115a35869bdd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.h
>@@ -11,6 +11,8 @@
> #ifndef CALL_RTP_CONFIG_H_
> #define CALL_RTP_CONFIG_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <string>
> #include <vector>
> 
>@@ -74,6 +76,9 @@ struct RtpConfig {
>   // Max RTP packet size delivered to send transport from VideoEngine.
>   size_t max_packet_size = kDefaultMaxPacketSize;
> 
>+  // Corresponds to the SDP attribute extmap-allow-mixed.
>+  bool extmap_allow_mixed = false;
>+
>   // RTP header extensions to use for this send stream.
>   std::vector<RtpExtension> extensions;
> 
>@@ -129,17 +134,5 @@ struct RtpConfig {
>   // RTCP CNAME, see RFC 3550.
>   std::string c_name;
> };
>-
>-struct RtcpConfig {
>-  RtcpConfig();
>-  RtcpConfig(const RtcpConfig&);
>-  ~RtcpConfig();
>-  std::string ToString() const;
>-
>-  // Time interval between RTCP report for video
>-  int64_t video_report_interval_ms = 1000;
>-  // Time interval between RTCP report for audio
>-  int64_t audio_report_interval_ms = 5000;
>-};
> }  // namespace webrtc
> #endif  // CALL_RTP_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer_unittest.cc
>index b5ede3cd03a0ce5f7dfda815ac89100271b6d536..3b8200e7819889a5163c9ccdd08a7cfa76127fd2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer_unittest.cc
>@@ -17,7 +17,6 @@
> #include "absl/memory/memory.h"
> #include "call/ssrc_binding_observer.h"
> #include "call/test/mock_rtp_packet_sink_interface.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.cc
>index ffbda1a2dbe3a328669184e62af62c97bdaeb836..2a32eef78332e8033edbe8fddeb11b8808b34dbb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.cc
>@@ -144,6 +144,8 @@ RTPVideoHeader RtpPayloadParams::GetRtpVideoHeader(
>   rtp_video_header.rotation = image.rotation_;
>   rtp_video_header.content_type = image.content_type_;
>   rtp_video_header.playout_delay = image.playout_delay_;
>+  rtp_video_header.width = image._encodedWidth;
>+  rtp_video_header.height = image._encodedHeight;
> 
>   SetVideoTiming(image, &rtp_video_header.video_timing);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.h
>index f6829cace5c629e55a2ec9cd5a471cd49a55de89..b91bfe4b2c5db00e02f39c19b92a705646aa0d4e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.h
>@@ -16,7 +16,6 @@
> 
> #include "api/video_codecs/video_encoder.h"
> #include "call/rtp_config.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
> #include "modules/rtp_rtcp/source/rtp_video_header.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params_unittest.cc
>index a3cb379d18a655548682229137e3e8fdbf551b50..9a519ba0a7f730fc58b80e69bf4a13efbb11d73e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params_unittest.cc
>@@ -309,9 +309,13 @@ class RtpPayloadParamsVp8ToGenericTest : public ::testing::Test {
>                        int64_t shared_frame_id,
>                        FrameType frame_type,
>                        LayerSync layer_sync,
>-                       const std::set<int64_t>& expected_deps) {
>+                       const std::set<int64_t>& expected_deps,
>+                       uint16_t width = 0,
>+                       uint16_t height = 0) {
>     EncodedImage encoded_image;
>     encoded_image._frameType = frame_type;
>+    encoded_image._encodedWidth = width;
>+    encoded_image._encodedHeight = height;
> 
>     CodecSpecificInfo codec_info{};
>     codec_info.codecType = kVideoCodecVP8;
>@@ -330,6 +334,9 @@ class RtpPayloadParamsVp8ToGenericTest : public ::testing::Test {
>     std::set<int64_t> actual_deps(header.generic->dependencies.begin(),
>                                   header.generic->dependencies.end());
>     EXPECT_EQ(expected_deps, actual_deps);
>+
>+    EXPECT_EQ(header.width, width);
>+    EXPECT_EQ(header.height, height);
>   }
> 
>  protected:
>@@ -339,13 +346,13 @@ class RtpPayloadParamsVp8ToGenericTest : public ::testing::Test {
> };
> 
> TEST_F(RtpPayloadParamsVp8ToGenericTest, Keyframe) {
>-  ConvertAndCheck(0, 0, kVideoFrameKey, kNoSync, {});
>+  ConvertAndCheck(0, 0, kVideoFrameKey, kNoSync, {}, 480, 360);
>   ConvertAndCheck(0, 1, kVideoFrameDelta, kNoSync, {0});
>-  ConvertAndCheck(0, 2, kVideoFrameKey, kNoSync, {});
>+  ConvertAndCheck(0, 2, kVideoFrameKey, kNoSync, {}, 480, 360);
> }
> 
> TEST_F(RtpPayloadParamsVp8ToGenericTest, TooHighTemporalIndex) {
>-  ConvertAndCheck(0, 0, kVideoFrameKey, kNoSync, {});
>+  ConvertAndCheck(0, 0, kVideoFrameKey, kNoSync, {}, 480, 360);
> 
>   EncodedImage encoded_image;
>   encoded_image._frameType = kVideoFrameDelta;
>@@ -362,7 +369,7 @@ TEST_F(RtpPayloadParamsVp8ToGenericTest, TooHighTemporalIndex) {
> 
> TEST_F(RtpPayloadParamsVp8ToGenericTest, LayerSync) {
>   // 02120212 pattern
>-  ConvertAndCheck(0, 0, kVideoFrameKey, kNoSync, {});
>+  ConvertAndCheck(0, 0, kVideoFrameKey, kNoSync, {}, 480, 360);
>   ConvertAndCheck(2, 1, kVideoFrameDelta, kNoSync, {0});
>   ConvertAndCheck(1, 2, kVideoFrameDelta, kNoSync, {0});
>   ConvertAndCheck(2, 3, kVideoFrameDelta, kNoSync, {0, 1, 2});
>@@ -375,7 +382,7 @@ TEST_F(RtpPayloadParamsVp8ToGenericTest, LayerSync) {
> 
> TEST_F(RtpPayloadParamsVp8ToGenericTest, FrameIdGaps) {
>   // 0101 pattern
>-  ConvertAndCheck(0, 0, kVideoFrameKey, kNoSync, {});
>+  ConvertAndCheck(0, 0, kVideoFrameKey, kNoSync, {}, 480, 360);
>   ConvertAndCheck(1, 1, kVideoFrameDelta, kNoSync, {0});
> 
>   ConvertAndCheck(0, 5, kVideoFrameDelta, kNoSync, {0});
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.cc
>index 54aef21fd9eafc9133c363e556ebf232c6ed4f37..5265c95afb8104128d340f7269d9519d6d77727a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.cc
>@@ -26,11 +26,6 @@ static const size_t kMaxOverheadBytes = 500;
> const char kTaskQueueExperiment[] = "WebRTC-TaskQueueCongestionControl";
> using TaskQueueController = webrtc::webrtc_cc::SendSideCongestionController;
> 
>-bool TaskQueueExperimentEnabled() {
>-  std::string trial = webrtc::field_trial::FindFullName(kTaskQueueExperiment);
>-  return trial.find("Enable") == 0;
>-}
>-
> std::unique_ptr<SendSideCongestionControllerInterface> CreateController(
>     Clock* clock,
>     rtc::TaskQueue* task_queue,
>@@ -70,9 +65,9 @@ RtpTransportControllerSend::RtpTransportControllerSend(
>       retransmission_rate_limiter_(clock, kRetransmitWindowSizeMs),
>       task_queue_("rtp_send_controller") {
>   // Created after task_queue to be able to post to the task queue internally.
>-  send_side_cc_ =
>-      CreateController(clock, &task_queue_, event_log, &pacer_, bitrate_config,
>-                       TaskQueueExperimentEnabled(), controller_factory);
>+  send_side_cc_ = CreateController(
>+      clock, &task_queue_, event_log, &pacer_, bitrate_config,
>+      !field_trial::IsDisabled(kTaskQueueExperiment), controller_factory);
> 
>   process_thread_->RegisterModule(&pacer_, RTC_FROM_HERE);
>   process_thread_->RegisterModule(send_side_cc_.get(), RTC_FROM_HERE);
>@@ -90,18 +85,20 @@ RtpVideoSenderInterface* RtpTransportControllerSend::CreateRtpVideoSender(
>     std::map<uint32_t, RtpState> suspended_ssrcs,
>     const std::map<uint32_t, RtpPayloadState>& states,
>     const RtpConfig& rtp_config,
>-    const RtcpConfig& rtcp_config,
>+    int rtcp_report_interval_ms,
>     Transport* send_transport,
>     const RtpSenderObservers& observers,
>     RtcEventLog* event_log,
>-    std::unique_ptr<FecController> fec_controller) {
>+    std::unique_ptr<FecController> fec_controller,
>+    const RtpSenderFrameEncryptionConfig& frame_encryption_config) {
>   video_rtp_senders_.push_back(absl::make_unique<RtpVideoSender>(
>-      ssrcs, suspended_ssrcs, states, rtp_config, rtcp_config, send_transport,
>-      observers,
>+      ssrcs, suspended_ssrcs, states, rtp_config, rtcp_report_interval_ms,
>+      send_transport, observers,
>       // TODO(holmer): Remove this circular dependency by injecting
>       // the parts of RtpTransportControllerSendInterface that are really used.
>-      this, event_log, &retransmission_rate_limiter_,
>-      std::move(fec_controller)));
>+      this, event_log, &retransmission_rate_limiter_, std::move(fec_controller),
>+      frame_encryption_config.frame_encryptor,
>+      frame_encryption_config.crypto_options));
>   return video_rtp_senders_.back().get();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.h
>index 6aaf29d6bc37b83323c78aa3e45301261f023416..f1c46cc07855ddef7e412c6680d19c5b68c29de4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.h
>@@ -20,7 +20,6 @@
> #include "call/rtp_bitrate_configurator.h"
> #include "call/rtp_transport_controller_send_interface.h"
> #include "call/rtp_video_sender.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/congestion_controller/include/send_side_congestion_controller_interface.h"
> #include "modules/pacing/packet_router.h"
> #include "modules/utility/include/process_thread.h"
>@@ -29,7 +28,9 @@
> #include "rtc_base/task_queue.h"
> 
> namespace webrtc {
>+
> class Clock;
>+class FrameEncryptorInterface;
> class RtcEventLog;
> 
> // TODO(nisse): When we get the underlying transports here, we should
>@@ -52,11 +53,12 @@ class RtpTransportControllerSend final
>       const std::map<uint32_t, RtpPayloadState>&
>           states,  // move states into RtpTransportControllerSend
>       const RtpConfig& rtp_config,
>-      const RtcpConfig& rtcp_config,
>+      int rtcp_report_interval_ms,
>       Transport* send_transport,
>       const RtpSenderObservers& observers,
>       RtcEventLog* event_log,
>-      std::unique_ptr<FecController> fec_controller) override;
>+      std::unique_ptr<FecController> fec_controller,
>+      const RtpSenderFrameEncryptionConfig& frame_encryption_config) override;
>   void DestroyRtpVideoSender(
>       RtpVideoSenderInterface* rtp_video_sender) override;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send_interface.h
>index 5c51c54fe13cf102fcc5afdd02c1a5fdfdb697b7..9868585cdf8113928a1ad63dd65eecc971cc2f7c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send_interface.h
>@@ -20,10 +20,12 @@
> 
> #include "absl/types/optional.h"
> #include "api/bitrate_constraints.h"
>+#include "api/crypto/cryptooptions.h"
> #include "api/fec_controller.h"
> #include "api/transport/bitrate_settings.h"
> #include "call/rtp_config.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
>+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> 
> namespace rtc {
>@@ -35,6 +37,7 @@ namespace webrtc {
> 
> class CallStats;
> class CallStatsObserver;
>+class FrameEncryptorInterface;
> class TargetTransferRateObserver;
> class Transport;
> class Module;
>@@ -62,6 +65,11 @@ struct RtpSenderObservers {
>   SendPacketObserver* send_packet_observer;
> };
> 
>+struct RtpSenderFrameEncryptionConfig {
>+  FrameEncryptorInterface* frame_encryptor = nullptr;
>+  CryptoOptions crypto_options;
>+};
>+
> // An RtpTransportController should own everything related to the RTP
> // transport to/from a remote endpoint. We should have separate
> // interfaces for send and receive side, even if they are implemented
>@@ -97,11 +105,12 @@ class RtpTransportControllerSendInterface {
>       // TODO(holmer): Move states into RtpTransportControllerSend.
>       const std::map<uint32_t, RtpPayloadState>& states,
>       const RtpConfig& rtp_config,
>-      const RtcpConfig& rtcp_config,
>+      int rtcp_report_interval_ms,
>       Transport* send_transport,
>       const RtpSenderObservers& observers,
>       RtcEventLog* event_log,
>-      std::unique_ptr<FecController> fec_controller) = 0;
>+      std::unique_ptr<FecController> fec_controller,
>+      const RtpSenderFrameEncryptionConfig& frame_encryption_config) = 0;
>   virtual void DestroyRtpVideoSender(
>       RtpVideoSenderInterface* rtp_video_sender) = 0;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.cc
>index 10d7e7daeda6d321756caf992845eda8d7cca522..ba1b992e95025aca58bc6cc2cfa1190ceb8381dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.cc
>@@ -40,8 +40,8 @@ static const size_t kPathMTU = 1500;
> 
> std::vector<std::unique_ptr<RtpRtcp>> CreateRtpRtcpModules(
>     const std::vector<uint32_t>& ssrcs,
>-    const std::vector<uint32_t>& protected_media_ssrcs,
>-    const RtcpConfig& rtcp_config,
>+    const RtpConfig& rtp_config,
>+    int rtcp_report_interval_ms,
>     Transport* send_transport,
>     RtcpIntraFrameObserver* intra_frame_callback,
>     RtcpBandwidthObserver* bandwidth_callback,
>@@ -56,8 +56,11 @@ std::vector<std::unique_ptr<RtpRtcp>> CreateRtpRtcpModules(
>     RtcEventLog* event_log,
>     RateLimiter* retransmission_rate_limiter,
>     OverheadObserver* overhead_observer,
>-    RtpKeepAliveConfig keepalive_config) {
>+    RtpKeepAliveConfig keepalive_config,
>+    FrameEncryptorInterface* frame_encryptor,
>+    const CryptoOptions& crypto_options) {
>   RTC_DCHECK_GT(ssrcs.size(), 0);
>+
>   RtpRtcp::Configuration configuration;
>   configuration.audio = false;
>   configuration.receiver_only = false;
>@@ -79,12 +82,15 @@ std::vector<std::unique_ptr<RtpRtcp>> CreateRtpRtcpModules(
>   configuration.retransmission_rate_limiter = retransmission_rate_limiter;
>   configuration.overhead_observer = overhead_observer;
>   configuration.keepalive_config = keepalive_config;
>-  configuration.rtcp_interval_config.video_interval_ms =
>-      rtcp_config.video_report_interval_ms;
>-  configuration.rtcp_interval_config.audio_interval_ms =
>-      rtcp_config.audio_report_interval_ms;
>+  configuration.frame_encryptor = frame_encryptor;
>+  configuration.require_frame_encryption =
>+      crypto_options.sframe.require_frame_encryption;
>+  configuration.extmap_allow_mixed = rtp_config.extmap_allow_mixed;
>+  configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
>+
>   std::vector<std::unique_ptr<RtpRtcp>> modules;
>-  const std::vector<uint32_t>& flexfec_protected_ssrcs = protected_media_ssrcs;
>+  const std::vector<uint32_t>& flexfec_protected_ssrcs =
>+      rtp_config.flexfec.protected_media_ssrcs;
>   for (uint32_t ssrc : ssrcs) {
>     bool enable_flexfec = flexfec_sender != nullptr &&
>                           std::find(flexfec_protected_ssrcs.begin(),
>@@ -177,13 +183,15 @@ RtpVideoSender::RtpVideoSender(
>     std::map<uint32_t, RtpState> suspended_ssrcs,
>     const std::map<uint32_t, RtpPayloadState>& states,
>     const RtpConfig& rtp_config,
>-    const RtcpConfig& rtcp_config,
>+    int rtcp_report_interval_ms,
>     Transport* send_transport,
>     const RtpSenderObservers& observers,
>     RtpTransportControllerSendInterface* transport,
>     RtcEventLog* event_log,
>     RateLimiter* retransmission_limiter,
>-    std::unique_ptr<FecController> fec_controller)
>+    std::unique_ptr<FecController> fec_controller,
>+    FrameEncryptorInterface* frame_encryptor,
>+    const CryptoOptions& crypto_options)
>     : send_side_bwe_with_overhead_(
>           webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
>       active_(false),
>@@ -191,25 +199,26 @@ RtpVideoSender::RtpVideoSender(
>       suspended_ssrcs_(std::move(suspended_ssrcs)),
>       flexfec_sender_(MaybeCreateFlexfecSender(rtp_config, suspended_ssrcs_)),
>       fec_controller_(std::move(fec_controller)),
>-      rtp_modules_(
>-          CreateRtpRtcpModules(ssrcs,
>-                               rtp_config.flexfec.protected_media_ssrcs,
>-                               rtcp_config,
>-                               send_transport,
>-                               observers.intra_frame_callback,
>-                               transport->GetBandwidthObserver(),
>-                               transport,
>-                               observers.rtcp_rtt_stats,
>-                               flexfec_sender_.get(),
>-                               observers.bitrate_observer,
>-                               observers.frame_count_observer,
>-                               observers.rtcp_type_observer,
>-                               observers.send_delay_observer,
>-                               observers.send_packet_observer,
>-                               event_log,
>-                               retransmission_limiter,
>-                               this,
>-                               transport->keepalive_config())),
>+      rtp_modules_(CreateRtpRtcpModules(ssrcs,
>+                                        rtp_config,
>+                                        rtcp_report_interval_ms,
>+                                        send_transport,
>+                                        observers.intra_frame_callback,
>+                                        transport->GetBandwidthObserver(),
>+                                        transport,
>+                                        observers.rtcp_rtt_stats,
>+                                        flexfec_sender_.get(),
>+                                        observers.bitrate_observer,
>+                                        observers.frame_count_observer,
>+                                        observers.rtcp_type_observer,
>+                                        observers.send_delay_observer,
>+                                        observers.send_packet_observer,
>+                                        event_log,
>+                                        retransmission_limiter,
>+                                        this,
>+                                        transport->keepalive_config(),
>+                                        frame_encryptor,
>+                                        crypto_options)),
>       rtp_config_(rtp_config),
>       transport_(transport),
>       transport_overhead_bytes_per_packet_(0),
>@@ -246,9 +255,6 @@ RtpVideoSender::RtpVideoSender(
>   for (size_t i = 0; i < rtp_config_.extensions.size(); ++i) {
>     const std::string& extension = rtp_config_.extensions[i].uri;
>     int id = rtp_config_.extensions[i].id;
>-    // One-byte-extension local identifiers are in the range 1-14 inclusive.
>-    RTC_DCHECK_GE(id, 1);
>-    RTC_DCHECK_LE(id, 14);
>     RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
>     for (auto& rtp_rtcp : rtp_modules_) {
>       RTC_CHECK(rtp_rtcp->RegisterRtpHeaderExtension(extension, id));
>@@ -592,12 +598,14 @@ void RtpVideoSender::OnBitrateUpdated(uint32_t bitrate_bps,
>   rtc::CritScope lock(&crit_);
>   uint32_t payload_bitrate_bps = bitrate_bps;
>   if (send_side_bwe_with_overhead_) {
>-    payload_bitrate_bps -= CalculateOverheadRateBps(
>+    uint32_t overhead_bps = CalculateOverheadRateBps(
>         CalculatePacketRate(
>             bitrate_bps,
>             rtp_config_.max_packet_size + transport_overhead_bytes_per_packet_),
>         overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_,
>         bitrate_bps);
>+    RTC_DCHECK_LE(overhead_bps, bitrate_bps);
>+    payload_bitrate_bps = bitrate_bps - overhead_bps;
>   }
> 
>   // Get the encoder target rate. It is the estimated network rate -
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.h
>index db329be00b04470fb3917f420df485c3d5b8b290..adac46bcd813084654b36cf742117cf52f1a7cfa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.h
>@@ -23,7 +23,6 @@
> #include "call/rtp_payload_params.h"
> #include "call/rtp_transport_controller_send_interface.h"
> #include "call/rtp_video_sender_interface.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/rtp_rtcp/include/flexfec_sender.h"
> #include "modules/rtp_rtcp/source/rtp_video_header.h"
>@@ -36,6 +35,7 @@
> 
> namespace webrtc {
> 
>+class FrameEncryptorInterface;
> class RTPFragmentationHeader;
> class RtpRtcp;
> class RtpTransportControllerSendInterface;
>@@ -53,13 +53,15 @@ class RtpVideoSender : public RtpVideoSenderInterface,
>       std::map<uint32_t, RtpState> suspended_ssrcs,
>       const std::map<uint32_t, RtpPayloadState>& states,
>       const RtpConfig& rtp_config,
>-      const RtcpConfig& rtcp_config,
>+      int rtcp_report_interval_ms,
>       Transport* send_transport,
>       const RtpSenderObservers& observers,
>       RtpTransportControllerSendInterface* transport,
>       RtcEventLog* event_log,
>       RateLimiter* retransmission_limiter,  // move inside RtpTransport
>-      std::unique_ptr<FecController> fec_controller);
>+      std::unique_ptr<FecController> fec_controller,
>+      FrameEncryptorInterface* frame_encryptor,
>+      const CryptoOptions& crypto_options);  // move inside RtpTransport
>   ~RtpVideoSender() override;
> 
>   // RegisterProcessThread register |module_process_thread| with those objects
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_unittest.cc
>index e91bfc4aeb14f268e309eafc0405269d293cc293..8ab89d6104c465abffc9867c4b8d34cccfaabc94 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_unittest.cc
>@@ -102,12 +102,13 @@ class RtpVideoSenderTestFixture {
>     std::map<uint32_t, RtpState> suspended_ssrcs;
>     router_ = absl::make_unique<RtpVideoSender>(
>         config_.rtp.ssrcs, suspended_ssrcs, suspended_payload_states,
>-        config_.rtp, config_.rtcp, &transport_,
>+        config_.rtp, config_.rtcp_report_interval_ms, &transport_,
>         CreateObservers(&call_stats_, &encoder_feedback_, &stats_proxy_,
>                         &stats_proxy_, &stats_proxy_, &stats_proxy_,
>                         &stats_proxy_, &stats_proxy_, &send_delay_stats_),
>         &transport_controller_, &event_log_, &retransmission_rate_limiter_,
>-        absl::make_unique<FecControllerDefault>(&clock_));
>+        absl::make_unique<FecControllerDefault>(&clock_), nullptr,
>+        CryptoOptions{});
>   }
> 
>   RtpVideoSender* router() { return router_.get(); }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.cc
>index ee58dce718248aa106bc3a817a34a921c8929fa6..77be14821c2aef1eb77f5ffa66f17fbe4640b2cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.cc
>@@ -13,6 +13,7 @@
> #include <algorithm>
> #include <cmath>
> #include <utility>
>+#include "api/units/data_rate.h"
> 
> namespace webrtc {
> 
>@@ -26,6 +27,9 @@ SimulatedNetwork::~SimulatedNetwork() = default;
> 
> void SimulatedNetwork::SetConfig(const SimulatedNetwork::Config& config) {
>   rtc::CritScope crit(&config_lock_);
>+  if (config_.link_capacity_kbps != config.link_capacity_kbps) {
>+    reset_capacity_delay_error_ = true;
>+  }
>   config_ = config;  // Shallow copy of the struct.
>   double prob_loss = config.loss_percent / 100.0;
>   if (config_.avg_burst_loss_length == -1) {
>@@ -64,42 +68,45 @@ bool SimulatedNetwork::EnqueuePacket(PacketInFlightInfo packet) {
>     // Too many packet on the link, drop this one.
>     return false;
>   }
>-
>-  // Delay introduced by the link capacity.
>-  int64_t capacity_delay_ms = 0;
>-  if (config.link_capacity_kbps > 0) {
>-    // Using bytes per millisecond to avoid losing precision.
>-    const int64_t bytes_per_millisecond = config.link_capacity_kbps / 8;
>-    // To round to the closest millisecond we add half a milliseconds worth of
>-    // bytes to the delay calculation.
>-    capacity_delay_ms = (packet.size + capacity_delay_error_bytes_ +
>-                         bytes_per_millisecond / 2) /
>-                        bytes_per_millisecond;
>-    capacity_delay_error_bytes_ +=
>-        packet.size - capacity_delay_ms * bytes_per_millisecond;
>-  }
>   int64_t network_start_time_us = packet.send_time_us;
>-
>   {
>     rtc::CritScope crit(&config_lock_);
>+    if (reset_capacity_delay_error_) {
>+      capacity_delay_error_bytes_ = 0;
>+      reset_capacity_delay_error_ = false;
>+    }
>     if (pause_transmission_until_us_) {
>       network_start_time_us =
>           std::max(network_start_time_us, *pause_transmission_until_us_);
>       pause_transmission_until_us_.reset();
>     }
>   }
>+
>+  // Delay introduced by the link capacity.
>+  TimeDelta capacity_delay = TimeDelta::Zero();
>+  if (config.link_capacity_kbps > 0) {
>+    const DataRate link_capacity = DataRate::kbps(config.link_capacity_kbps);
>+    int64_t compensated_size =
>+        static_cast<int64_t>(packet.size) + capacity_delay_error_bytes_;
>+    capacity_delay = DataSize::bytes(compensated_size) / link_capacity;
>+
>+    capacity_delay_error_bytes_ +=
>+        packet.size - (capacity_delay * link_capacity).bytes();
>+  }
>+
>   // Check if there already are packets on the link and change network start
>   // time forward if there is.
>   if (!capacity_link_.empty() &&
>       network_start_time_us < capacity_link_.back().arrival_time_us)
>     network_start_time_us = capacity_link_.back().arrival_time_us;
> 
>-  int64_t arrival_time_us = network_start_time_us + capacity_delay_ms * 1000;
>+  int64_t arrival_time_us = network_start_time_us + capacity_delay.us();
>   capacity_link_.push({packet, arrival_time_us});
>   return true;
> }
> 
> absl::optional<int64_t> SimulatedNetwork::NextDeliveryTimeUs() const {
>+  rtc::CritScope crit(&process_lock_);
>   if (!delay_link_.empty())
>     return delay_link_.begin()->arrival_time_us;
>   return absl::nullopt;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.h
>index 0f8453283a88dd99deedf7c8ddeacc26d292d8ea..e978027092c7793dd3ce430d9bb73ef3775d2ede 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.h
>@@ -28,7 +28,7 @@ namespace webrtc {
> // capacity introduced delay.
> class SimulatedNetwork : public NetworkBehaviorInterface {
>  public:
>-  using Config = DefaultNetworkSimulationConfig;
>+  using Config = BuiltInNetworkBehaviorConfig;
>   explicit SimulatedNetwork(Config config, uint64_t random_seed = 1);
>   ~SimulatedNetwork() override;
> 
>@@ -49,6 +49,7 @@ class SimulatedNetwork : public NetworkBehaviorInterface {
>     int64_t arrival_time_us;
>   };
>   rtc::CriticalSection config_lock_;
>+  bool reset_capacity_delay_error_ RTC_GUARDED_BY(config_lock_) = false;
> 
>   // |process_lock| guards the data structures involved in delay and loss
>   // processes, such as the packet queues.
>@@ -56,7 +57,7 @@ class SimulatedNetwork : public NetworkBehaviorInterface {
>   std::queue<PacketInfo> capacity_link_ RTC_GUARDED_BY(process_lock_);
>   Random random_;
> 
>-  std::deque<PacketInfo> delay_link_;
>+  std::deque<PacketInfo> delay_link_ RTC_GUARDED_BY(process_lock_);
> 
>   // Link configuration.
>   Config config_ RTC_GUARDED_BY(config_lock_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/fake_network_pipe_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/fake_network_pipe_unittest.cc
>index 09e585241a407677af37e9a7c266fdaeb1624074..de5aec71352ccc7ae04881ca61bae3b78c2a6d2c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/fake_network_pipe_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/fake_network_pipe_unittest.cc
>@@ -73,7 +73,7 @@ class FakeNetworkPipeTest : public ::testing::Test {
> 
> // Test the capacity link and verify we get as many packets as we expect.
> TEST_F(FakeNetworkPipeTest, CapacityTest) {
>-  DefaultNetworkSimulationConfig config;
>+  BuiltInNetworkBehaviorConfig config;
>   config.queue_length_packets = 20;
>   config.link_capacity_kbps = 80;
>   MockReceiver receiver;
>@@ -113,7 +113,7 @@ TEST_F(FakeNetworkPipeTest, CapacityTest) {
> 
> // Test the extra network delay.
> TEST_F(FakeNetworkPipeTest, ExtraDelayTest) {
>-  DefaultNetworkSimulationConfig config;
>+  BuiltInNetworkBehaviorConfig config;
>   config.queue_length_packets = 20;
>   config.queue_delay_ms = 100;
>   config.link_capacity_kbps = 80;
>@@ -149,7 +149,7 @@ TEST_F(FakeNetworkPipeTest, ExtraDelayTest) {
> // Test the number of buffers and packets are dropped when sending too many
> // packets too quickly.
> TEST_F(FakeNetworkPipeTest, QueueLengthTest) {
>-  DefaultNetworkSimulationConfig config;
>+  BuiltInNetworkBehaviorConfig config;
>   config.queue_length_packets = 2;
>   config.link_capacity_kbps = 80;
>   MockReceiver receiver;
>@@ -173,7 +173,7 @@ TEST_F(FakeNetworkPipeTest, QueueLengthTest) {
> 
> // Test we get statistics as expected.
> TEST_F(FakeNetworkPipeTest, StatisticsTest) {
>-  DefaultNetworkSimulationConfig config;
>+  BuiltInNetworkBehaviorConfig config;
>   config.queue_length_packets = 2;
>   config.queue_delay_ms = 20;
>   config.link_capacity_kbps = 80;
>@@ -205,7 +205,7 @@ TEST_F(FakeNetworkPipeTest, StatisticsTest) {
> // Change the link capacity half-way through the test and verify that the
> // delivery times change accordingly.
> TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) {
>-  DefaultNetworkSimulationConfig config;
>+  BuiltInNetworkBehaviorConfig config;
>   config.queue_length_packets = 20;
>   config.link_capacity_kbps = 80;
>   MockReceiver receiver;
>@@ -266,7 +266,7 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) {
> // Change the link capacity half-way through the test and verify that the
> // delivery times change accordingly.
> TEST_F(FakeNetworkPipeTest, ChangingCapacityWithPacketsInPipeTest) {
>-  DefaultNetworkSimulationConfig config;
>+  BuiltInNetworkBehaviorConfig config;
>   config.queue_length_packets = 20;
>   config.link_capacity_kbps = 80;
>   MockReceiver receiver;
>@@ -321,7 +321,7 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithPacketsInPipeTest) {
> 
> // At first disallow reordering and then allow reordering.
> TEST_F(FakeNetworkPipeTest, DisallowReorderingThenAllowReordering) {
>-  DefaultNetworkSimulationConfig config;
>+  BuiltInNetworkBehaviorConfig config;
>   config.queue_length_packets = 1000;
>   config.link_capacity_kbps = 800;
>   config.queue_delay_ms = 100;
>@@ -374,7 +374,7 @@ TEST_F(FakeNetworkPipeTest, BurstLoss) {
>   const int kNumPackets = 10000;
>   const int kPacketSize = 10;
> 
>-  DefaultNetworkSimulationConfig config;
>+  BuiltInNetworkBehaviorConfig config;
>   config.queue_length_packets = kNumPackets;
>   config.loss_percent = kLossPercent;
>   config.avg_burst_loss_length = kAvgBurstLength;
>@@ -409,7 +409,7 @@ TEST_F(FakeNetworkPipeTest, BurstLoss) {
> }
> 
> TEST_F(FakeNetworkPipeTest, SetReceiver) {
>-  DefaultNetworkSimulationConfig config;
>+  BuiltInNetworkBehaviorConfig config;
>   config.link_capacity_kbps = 800;
>   MockReceiver receiver;
>   auto simulated_network = absl::make_unique<SimulatedNetwork>(config);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_bitrate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_bitrate_allocator.h
>index 714c02541f172ad5d5cd694a8092ca143b9544c8..f00ed79c59fb78289f8d904d89afe865b59b9e58 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_bitrate_allocator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_bitrate_allocator.h
>@@ -21,7 +21,7 @@ class MockBitrateAllocator : public BitrateAllocatorInterface {
>   MOCK_METHOD2(AddObserver,
>                void(BitrateAllocatorObserver*, MediaStreamAllocationConfig));
>   MOCK_METHOD1(RemoveObserver, void(BitrateAllocatorObserver*));
>-  MOCK_METHOD1(GetStartBitrate, int(BitrateAllocatorObserver*));
>+  MOCK_CONST_METHOD1(GetStartBitrate, int(BitrateAllocatorObserver*));
> };
> }  // namespace webrtc
> #endif  // CALL_TEST_MOCK_BITRATE_ALLOCATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_rtp_transport_controller_send.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_rtp_transport_controller_send.h
>index 37587b2b76b9be19dfa68bc4ae73a458ecbb4e19..4d34d06e0567b99ef40a09357fdec4654134af24 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_rtp_transport_controller_send.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_rtp_transport_controller_send.h
>@@ -17,12 +17,14 @@
> #include <vector>
> 
> #include "api/bitrate_constraints.h"
>+#include "api/crypto/cryptooptions.h"
>+#include "api/crypto/frameencryptorinterface.h"
> #include "call/rtp_transport_controller_send_interface.h"
> #include "modules/congestion_controller/include/network_changed_observer.h"
> #include "modules/pacing/packet_router.h"
>+#include "rtc_base/network/sent_packet.h"
> #include "rtc_base/networkroute.h"
> #include "rtc_base/rate_limiter.h"
>-#include "rtc_base/socket.h"
> #include "test/gmock.h"
> 
> namespace webrtc {
>@@ -30,17 +32,18 @@ namespace webrtc {
> class MockRtpTransportControllerSend
>     : public RtpTransportControllerSendInterface {
>  public:
>-  MOCK_METHOD9(
>+  MOCK_METHOD10(
>       CreateRtpVideoSender,
>       RtpVideoSenderInterface*(const std::vector<uint32_t>&,
>                                std::map<uint32_t, RtpState>,
>                                const std::map<uint32_t, RtpPayloadState>&,
>                                const RtpConfig&,
>-                               const RtcpConfig&,
>+                               int rtcp_report_interval_ms,
>                                Transport*,
>                                const RtpSenderObservers&,
>                                RtcEventLog*,
>-                               std::unique_ptr<FecController>));
>+                               std::unique_ptr<FecController>,
>+                               const RtpSenderFrameEncryptionConfig&));
>   MOCK_METHOD1(DestroyRtpVideoSender, void(RtpVideoSenderInterface*));
>   MOCK_METHOD0(GetWorkerQueue, rtc::TaskQueue*());
>   MOCK_METHOD0(packet_router, PacketRouter*());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.h
>index 49efe4fceaa5ee4647192b08a7f3e05ceaca22a9..2cd5f1631a09b15a0195315b7eb88c3f16fc7179 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.h
>@@ -17,6 +17,7 @@
> #include <vector>
> 
> #include "api/call/transport.h"
>+#include "api/crypto/cryptooptions.h"
> #include "api/rtp_headers.h"
> #include "api/rtpparameters.h"
> #include "api/rtpreceiverinterface.h"
>@@ -25,12 +26,12 @@
> #include "api/video/video_timing.h"
> #include "api/video_codecs/sdp_video_format.h"
> #include "call/rtp_config.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/frame_callback.h"
>+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> 
> namespace webrtc {
> 
>+class FrameDecryptorInterface;
> class RtpPacketSinkInterface;
> class VideoDecoderFactory;
> 
>@@ -215,6 +216,14 @@ class VideoReceiveStream {
>     // TODO(nisse): Used with VideoDecoderFactory::LegacyCreateVideoDecoder.
>     // Delete when that method is retired.
>     std::string stream_id;
>+
>+    // An optional custom frame decryptor that allows the entire frame to be
>+    // decrypted in whatever way the caller choses. This is not required by
>+    // default.
>+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor;
>+
>+    // Per PeerConnection cryptography options.
>+    CryptoOptions crypto_options;
>   };
> 
>   // Starts stream activity.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.cc
>index 6b93fe28238a4b8e35da230fdae455c68dc513e1..c8ba308fa54236a2bf05733a4146b119c4d7371e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.cc
>@@ -9,6 +9,10 @@
>  */
> 
> #include "call/video_send_stream.h"
>+
>+#include <utility>
>+
>+#include "api/crypto/frameencryptorinterface.h"
> #include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
>@@ -75,11 +79,9 @@ std::string VideoSendStream::Config::ToString() const {
>   ss << "{encoder_settings: { experiment_cpu_load_estimator: "
>      << (encoder_settings.experiment_cpu_load_estimator ? "on" : "off") << "}}";
>   ss << ", rtp: " << rtp.ToString();
>-  ss << ", rtcp: " << rtcp.ToString();
>+  ss << ", rtcp_report_interval_ms: " << rtcp_report_interval_ms;
>   ss << ", pre_encode_callback: "
>      << (pre_encode_callback ? "(VideoSinkInterface)" : "nullptr");
>-  ss << ", post_encode_callback: "
>-     << (post_encode_callback ? "(EncodedFrameObserver)" : "nullptr");
>   ss << ", render_delay_ms: " << render_delay_ms;
>   ss << ", target_delay_ms: " << target_delay_ms;
>   ss << ", suspend_below_min_bitrate: "
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h
>index 3ff624d230bd99811cc40ef1d8350b7720ab192a..4d3d9c0c574cb7f76c6e9f9b95140284ef6ebe42 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h
>@@ -11,24 +11,29 @@
> #ifndef CALL_VIDEO_SEND_STREAM_H_
> #define CALL_VIDEO_SEND_STREAM_H_
> 
>+#include <stdint.h>
> #include <map>
> #include <string>
>-#include <utility>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/call/transport.h"
>+#include "api/crypto/cryptooptions.h"
>+#include "api/rtpparameters.h"
>+#include "api/video/video_content_type.h"
>+#include "api/video/video_frame.h"
> #include "api/video/video_sink_interface.h"
> #include "api/video/video_source_interface.h"
> #include "api/video/video_stream_encoder_settings.h"
> #include "api/video_codecs/video_encoder_config.h"
>-#include "api/video_codecs/video_encoder_factory.h"
> #include "call/rtp_config.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/frame_callback.h"
>+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> 
> namespace webrtc {
> 
>+class FrameEncryptorInterface;
>+
> class VideoSendStream {
>  public:
>   struct StreamStats {
>@@ -108,7 +113,8 @@ class VideoSendStream {
> 
>     RtpConfig rtp;
> 
>-    RtcpConfig rtcp;
>+    // Time interval between RTCP report for video
>+    int rtcp_report_interval_ms = 1000;
> 
>     // Transport for outgoing packets.
>     Transport* send_transport = nullptr;
>@@ -117,12 +123,6 @@ class VideoSendStream {
>     // effects, snapshots etc. 'nullptr' disables the callback.
>     rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback = nullptr;
> 
>-    // Called for each encoded frame, e.g. used for file storage. 'nullptr'
>-    // disables the callback. Also measures timing and passes the time
>-    // spent on encoding. This timing will not fire if encoding takes longer
>-    // than the measuring window, since the sample data will have been dropped.
>-    EncodedFrameObserver* post_encode_callback = nullptr;
>-
>     // Expected delay needed by the renderer, i.e. the frame will be delivered
>     // this many milliseconds, if possible, earlier than expected render time.
>     // Only valid if |local_renderer| is set.
>@@ -143,6 +143,14 @@ class VideoSendStream {
>     // Track ID as specified during track creation.
>     std::string track_id;
> 
>+    // An optional custom frame encryptor that allows the entire frame to be
>+    // encrypted in whatever way the caller chooses. This is not required by
>+    // default.
>+    rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor;
>+
>+    // Per PeerConnection cryptography options.
>+    CryptoOptions crypto_options;
>+
>    private:
>     // Access to the copy constructor is private to force use of the Copy()
>     // method for those exceptional cases where we do use it.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/codereview.settings b/Source/ThirdParty/libwebrtc/Source/webrtc/codereview.settings
>index d8cf6125f4e3a522be58dd85cbc788ba82c6ac5c..1724eddc072defe1f41ff0ffde116c30446387d6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/codereview.settings
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/codereview.settings
>@@ -1,5 +1,4 @@
> # This file is used by git-cl to get repository specific information.
>-CC_LIST: webrtc-reviews@webrtc.org
> CODE_REVIEW_SERVER: codereview.webrtc.org
> GERRIT_HOST: True
> PROJECT: webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/BUILD.gn
>index abcfe9ad1683fbbb58707a7db652b39d3e9c2e0b..74b0c6029fde6877d9db30edeb915fa431f7a867 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/BUILD.gn
>@@ -47,10 +47,10 @@ rtc_static_library("common_audio") {
>   deps = [
>     ":common_audio_c",
>     ":sinc_resampler",
>-    "..:webrtc_common",
>     "../rtc_base:checks",
>     "../rtc_base:gtest_prod",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base:sanitizer",
>     "../rtc_base/memory:aligned_array",
>     "../rtc_base/memory:aligned_malloc",
>     "../rtc_base/system:arch",
>@@ -181,7 +181,6 @@ rtc_source_set("common_audio_c") {
>   deps = [
>     ":common_audio_c_arm_asm",
>     ":common_audio_cc",
>-    "..:webrtc_common",
>     "../rtc_base:checks",
>     "../rtc_base:compile_assert_c",
>     "../rtc_base:rtc_base_approved",
>@@ -201,7 +200,6 @@ rtc_source_set("common_audio_cc") {
>   ]
> 
>   deps = [
>-    "..:webrtc_common",
>     "../rtc_base:rtc_base_approved",
>     "../system_wrappers",
>   ]
>@@ -212,7 +210,6 @@ rtc_source_set("sinc_resampler") {
>     "resampler/sinc_resampler.h",
>   ]
>   deps = [
>-    "..:webrtc_common",
>     "../rtc_base:gtest_prod",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base/memory:aligned_malloc",
>@@ -381,7 +378,6 @@ if (rtc_include_tests) {
>       ":fir_filter",
>       ":fir_filter_factory",
>       ":sinc_resampler",
>-      "..:webrtc_common",
>       "../rtc_base:checks",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>@@ -389,6 +385,7 @@ if (rtc_include_tests) {
>       "../system_wrappers:cpu_features_api",
>       "../test:fileutils",
>       "../test:test_main",
>+      "../test:test_support",
>       "//testing/gtest",
>     ]
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.h
>index 769d724e2fb69e0b432933e770caf29cb50c0ff2..24a5e72cee9040890a97e26bb0e023bc1f43869c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.h
>@@ -11,6 +11,7 @@
> #ifndef COMMON_AUDIO_AUDIO_CONVERTER_H_
> #define COMMON_AUDIO_AUDIO_CONVERTER_H_
> 
>+#include <stddef.h>
> #include <memory>
> 
> #include "rtc_base/constructormagic.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.cc
>index 38d231e09d06471f99bd28ff2668f014dfa8e242..b9b8c25e37506ed3899f84f0a9f9dce7c9f01b1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.cc
>@@ -10,6 +10,9 @@
> 
> #include "common_audio/channel_buffer.h"
> 
>+#include <cstdint>
>+
>+#include "common_audio/include/audio_util.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.cc
>index 3418434672ace3e21ab2af0e4c48f773d8faddf1..b6ec27ad26bda6fcb9b0c9248e8b37af3c24ec4d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.cc
>@@ -11,11 +11,8 @@
> #include "common_audio/fir_filter_c.h"
> 
> #include <string.h>
>-
> #include <memory>
> 
>-#include "common_audio/fir_filter_neon.h"
>-#include "common_audio/fir_filter_sse.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.cc
>index 3243b27ef9ad2df624c5ac7ed7ddfc9b77b6512a..19528e312ec54cc6175c358ec3ac79cf24de87bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.cc
>@@ -13,12 +13,12 @@
> #include "common_audio/fir_filter_c.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/system/arch.h"
>-#include "system_wrappers/include/cpu_features_wrapper.h"
> 
> #if defined(WEBRTC_HAS_NEON)
> #include "common_audio/fir_filter_neon.h"
> #elif defined(WEBRTC_ARCH_X86_FAMILY)
> #include "common_audio/fir_filter_sse.h"
>+#include "system_wrappers/include/cpu_features_wrapper.h"  // kSSE2, WebRtc_G...
> #endif
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.h
>index 7707f933d9ee6b7b1305c5694c7bdda6530e01ca..b768a37aa1544b08a6cea3ec13778fc23346d4b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.h
>@@ -11,6 +11,7 @@
> #ifndef COMMON_AUDIO_FIR_FILTER_SSE_H_
> #define COMMON_AUDIO_FIR_FILTER_SSE_H_
> 
>+#include <stddef.h>
> #include <memory>
> 
> #include "common_audio/fir_filter.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/include/audio_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/include/audio_util.h
>index de242a49be8b663484a8b9eb4e5faecb03827af4..bca5718044c90e86191a2218eebcca522efc1612 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/include/audio_util.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/include/audio_util.h
>@@ -11,6 +11,7 @@
> #ifndef COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
> #define COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
> 
>+#include <stdint.h>
> #include <algorithm>
> #include <cmath>
> #include <cstring>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.h
>index a3e4dd16a82c21349efa60828d97aa9c5782f815..4d0d8bf38ee90b62a681bd3c9405802d8f74fdf0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.h
>@@ -11,6 +11,7 @@
> #ifndef COMMON_AUDIO_REAL_FOURIER_H_
> #define COMMON_AUDIO_REAL_FOURIER_H_
> 
>+#include <stddef.h>
> #include <complex>
> #include <memory>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.h
>index bb8eef96dff6ef103d39491fbd09d6db1ea1d072..b36c84f10b6b238212e637ab505b3d584e230ad3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.h
>@@ -11,6 +11,7 @@
> #ifndef COMMON_AUDIO_REAL_FOURIER_OOURA_H_
> #define COMMON_AUDIO_REAL_FOURIER_OOURA_H_
> 
>+#include <stddef.h>
> #include <complex>
> #include <memory>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_resampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_resampler.cc
>index 318d97b72c28ad50f3b0337aaf3b3fd855a542c8..9b89867e5a0dc930335276ee0983c2a7396135ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_resampler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_resampler.cc
>@@ -10,12 +10,12 @@
> 
> #include "common_audio/resampler/include/push_resampler.h"
> 
>+#include <stdint.h>
> #include <string.h>
> 
> #include "absl/container/inlined_vector.h"
> #include "absl/memory/memory.h"
> #include "common_audio/include/audio_util.h"
>-#include "common_audio/resampler/include/resampler.h"
> #include "common_audio/resampler/push_sinc_resampler.h"
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.h
>index 1ffe73f79ba2806daa93268a1f8597dc1162a4b9..db9cdc1fd4b33e8a562bcb3226210be7ad428f5e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.h
>@@ -11,6 +11,8 @@
> #ifndef COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
> #define COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> 
> #include "common_audio/resampler/sinc_resampler.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler.cc
>index aa3a4bac2aa2bc23447ce28f57662cd96b3ed725..e4d2aa2b61559ed4509970f6e5549a27cff00578 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler.cc
>@@ -12,6 +12,7 @@
>  * A wrapper for resampling a numerous amount of sampling combinations.
>  */
> 
>+#include <stdint.h>
> #include <stdlib.h>
> #include <string.h>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.cc
>index 5aa2061ab75ec4db122e09967566d107daf6bc41..46015140303112d3e2c7bd5d917d967d916a2402 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.cc
>@@ -88,13 +88,13 @@
> #include "common_audio/resampler/sinc_resampler.h"
> 
> #include <math.h>
>+#include <stdint.h>
> #include <string.h>
>-
> #include <limits>
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/system/arch.h"
>-#include "system_wrappers/include/cpu_features_wrapper.h"
>+#include "system_wrappers/include/cpu_features_wrapper.h"  // kSSE2, WebRtc_G...
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.h
>index 8a833ce88c99a32f80bdffd82561862b5423a569..0be4318e33922070457de0595b879578ddddb72c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.h
>@@ -14,6 +14,7 @@
> #ifndef COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
> #define COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
> 
>+#include <stddef.h>
> #include <memory>
> 
> #include "rtc_base/constructormagic.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_sse.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_sse.cc
>index 3906a797528a425d03621ea170614fdd0e1bb927..f6a24d0a0e0d34174732e68a91f83204605fab22 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_sse.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_sse.cc
>@@ -11,10 +11,12 @@
> // Modified from the Chromium original:
> // src/media/base/simd/sinc_resampler_sse.cc
> 
>-#include "common_audio/resampler/sinc_resampler.h"
>-
>+#include <stddef.h>
>+#include <stdint.h>
> #include <xmmintrin.h>
> 
>+#include "common_audio/resampler/sinc_resampler.h"
>+
> namespace webrtc {
> 
> float SincResampler::Convolve_SSE(const float* input_ptr,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.cc
>index d426bda2501343cc721a5b2c0f52fb66c035290e..0d5aaa485415904eb3ccdb023fa2b37b81fd875a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.cc
>@@ -12,6 +12,7 @@
> 
> #include <cmath>
> 
>+#include "rtc_base/checks.h"
> #include "rtc_base/timeutils.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.h
>index cff746953aa80ad90cdd5d29e3d915171b35bf23..c467d85426956f888cf025cbcda59b8e40e0d011 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.h
>@@ -11,9 +11,10 @@
> #ifndef COMMON_AUDIO_SMOOTHING_FILTER_H_
> #define COMMON_AUDIO_SMOOTHING_FILTER_H_
> 
>+#include <stdint.h>
>+
> #include "absl/types/optional.h"
> #include "rtc_base/constructormagic.h"
>-#include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad.cc
>index 1cb332a61d887fd24db15aea5b8c1ea7e8915a12..987ed526c00a9793ca97fca30b3248be66b06157 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad.cc
>@@ -12,6 +12,7 @@
> 
> #include <memory>
> 
>+#include "common_audio/vad/include/webrtc_vad.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>index 0209f52d2c73592eaa63da25f2bd91d3e97bee85..90a89281e5cd48dc2e32c20553c00f1d1f8785be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>@@ -10,35 +10,45 @@
> 
> #include "common_audio/wav_file.h"
> 
>+#include <errno.h>
> #include <algorithm>
> #include <cstdio>
>-#include <limits>
>+#include <type_traits>
> 
> #include "common_audio/include/audio_util.h"
> #include "common_audio/wav_header.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/system/arch.h"
> 
> namespace webrtc {
>+namespace {
> 
> // We write 16-bit PCM WAV files.
>-static const WavFormat kWavFormat = kWavFormatPcm;
>-static const size_t kBytesPerSample = 2;
>+constexpr WavFormat kWavFormat = kWavFormatPcm;
>+static_assert(std::is_trivially_destructible<WavFormat>::value, "");
>+constexpr size_t kBytesPerSample = 2;
> 
> // Doesn't take ownership of the file handle and won't close it.
> class ReadableWavFile : public ReadableWav {
>  public:
>   explicit ReadableWavFile(FILE* file) : file_(file) {}
>+  ReadableWavFile(const ReadableWavFile&) = delete;
>+  ReadableWavFile& operator=(const ReadableWavFile&) = delete;
>   size_t Read(void* buf, size_t num_bytes) override {
>     return fread(buf, 1, num_bytes, file_);
>   }
>+  bool Eof() const override { return feof(file_) != 0; }
>+  bool SeekForward(uint32_t num_bytes) override {
>+    return fseek(file_, num_bytes, SEEK_CUR) == 0;
>+  }
> 
>  private:
>   FILE* file_;
> };
> 
>+}  // namespace
>+
> WavReader::WavReader(const std::string& filename)
>     : WavReader(rtc::OpenPlatformFileReadOnly(filename)) {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.cc
>index 8fc5fef5085230668379dec0f29476430552c673..e49b7489b2efa3eaed6abae6d6a7243100dcef36 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.cc
>@@ -14,13 +14,13 @@
> 
> #include "common_audio/wav_header.h"
> 
>-#include <algorithm>
> #include <cstring>
> #include <limits>
> #include <string>
> 
>-#include "common_audio/include/audio_util.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/sanitizer.h"
> #include "rtc_base/system/arch.h"
> 
> namespace webrtc {
>@@ -32,6 +32,11 @@ struct ChunkHeader {
> };
> static_assert(sizeof(ChunkHeader) == 8, "ChunkHeader size");
> 
>+struct RiffHeader {
>+  ChunkHeader header;
>+  uint32_t Format;
>+};
>+
> // We can't nest this definition in WavHeader, because VS2013 gives an error
> // on sizeof(WavHeader::fmt): "error C2070: 'unknown': illegal sizeof operand".
> struct FmtSubchunk {
>@@ -46,11 +51,12 @@ struct FmtSubchunk {
> static_assert(sizeof(FmtSubchunk) == 24, "FmtSubchunk size");
> const uint32_t kFmtSubchunkSize = sizeof(FmtSubchunk) - sizeof(ChunkHeader);
> 
>+// Simple wav header. It does not include chunks that are not essential to read
>+// audio samples.
> struct WavHeader {
>-  struct {
>-    ChunkHeader header;
>-    uint32_t Format;
>-  } riff;
>+  WavHeader(const WavHeader&) = default;
>+  WavHeader& operator=(const WavHeader&) = default;
>+  RiffHeader riff;
>   FmtSubchunk fmt;
>   struct {
>     ChunkHeader header;
>@@ -58,6 +64,87 @@ struct WavHeader {
> };
> static_assert(sizeof(WavHeader) == kWavHeaderSize, "no padding in header");
> 
>+#ifdef WEBRTC_ARCH_LITTLE_ENDIAN
>+static inline void WriteLE16(uint16_t* f, uint16_t x) {
>+  *f = x;
>+}
>+static inline void WriteLE32(uint32_t* f, uint32_t x) {
>+  *f = x;
>+}
>+static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
>+  *f = static_cast<uint32_t>(a) | static_cast<uint32_t>(b) << 8 |
>+       static_cast<uint32_t>(c) << 16 | static_cast<uint32_t>(d) << 24;
>+}
>+
>+static inline uint16_t ReadLE16(uint16_t x) {
>+  return x;
>+}
>+static inline uint32_t ReadLE32(uint32_t x) {
>+  return x;
>+}
>+static inline std::string ReadFourCC(uint32_t x) {
>+  return std::string(reinterpret_cast<char*>(&x), 4);
>+}
>+#else
>+#error "Write be-to-le conversion functions"
>+#endif
>+
>+static inline uint32_t RiffChunkSize(size_t bytes_in_payload) {
>+  return static_cast<uint32_t>(bytes_in_payload + kWavHeaderSize -
>+                               sizeof(ChunkHeader));
>+}
>+
>+static inline uint32_t ByteRate(size_t num_channels,
>+                                int sample_rate,
>+                                size_t bytes_per_sample) {
>+  return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample);
>+}
>+
>+static inline uint16_t BlockAlign(size_t num_channels,
>+                                  size_t bytes_per_sample) {
>+  return static_cast<uint16_t>(num_channels * bytes_per_sample);
>+}
>+
>+// Finds a chunk having the sought ID. If found, then |readable| points to the
>+// first byte of the sought chunk data. If not found, the end of the file is
>+// reached.
>+void FindWaveChunk(ChunkHeader* chunk_header,
>+                   ReadableWav* readable,
>+                   const std::string sought_chunk_id) {
>+  RTC_DCHECK_EQ(sought_chunk_id.size(), 4);
>+  while (!readable->Eof()) {
>+    if (readable->Read(chunk_header, sizeof(*chunk_header)) !=
>+        sizeof(*chunk_header))
>+      return;  // EOF.
>+    if (ReadFourCC(chunk_header->ID) == sought_chunk_id)
>+      return;  // Sought chunk found.
>+    // Ignore current chunk by skipping its payload.
>+    if (!readable->SeekForward(chunk_header->Size))
>+      return;  // EOF or error.
>+  }
>+  return;  // EOF.
>+}
>+
>+bool ReadFmtChunkData(FmtSubchunk* fmt_subchunk, ReadableWav* readable) {
>+  // Reads "fmt " chunk payload.
>+  if (readable->Read(&(fmt_subchunk->AudioFormat), kFmtSubchunkSize) !=
>+      kFmtSubchunkSize)
>+    return false;
>+  const uint32_t fmt_size = ReadLE32(fmt_subchunk->header.Size);
>+  if (fmt_size != kFmtSubchunkSize) {
>+    // There is an optional two-byte extension field permitted to be present
>+    // with PCM, but which must be zero.
>+    int16_t ext_size;
>+    if (kFmtSubchunkSize + sizeof(ext_size) != fmt_size)
>+      return false;
>+    if (readable->Read(&ext_size, sizeof(ext_size)) != sizeof(ext_size))
>+      return false;
>+    if (ext_size != 0)
>+      return false;
>+  }
>+  return true;
>+}
>+
> }  // namespace
> 
> bool CheckWavParameters(size_t num_channels,
>@@ -112,47 +199,6 @@ bool CheckWavParameters(size_t num_channels,
>   return true;
> }
> 
>-#ifdef WEBRTC_ARCH_LITTLE_ENDIAN
>-static inline void WriteLE16(uint16_t* f, uint16_t x) {
>-  *f = x;
>-}
>-static inline void WriteLE32(uint32_t* f, uint32_t x) {
>-  *f = x;
>-}
>-static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
>-  *f = static_cast<uint32_t>(a) | static_cast<uint32_t>(b) << 8 |
>-       static_cast<uint32_t>(c) << 16 | static_cast<uint32_t>(d) << 24;
>-}
>-
>-static inline uint16_t ReadLE16(uint16_t x) {
>-  return x;
>-}
>-static inline uint32_t ReadLE32(uint32_t x) {
>-  return x;
>-}
>-static inline std::string ReadFourCC(uint32_t x) {
>-  return std::string(reinterpret_cast<char*>(&x), 4);
>-}
>-#else
>-#error "Write be-to-le conversion functions"
>-#endif
>-
>-static inline uint32_t RiffChunkSize(size_t bytes_in_payload) {
>-  return static_cast<uint32_t>(bytes_in_payload + kWavHeaderSize -
>-                               sizeof(ChunkHeader));
>-}
>-
>-static inline uint32_t ByteRate(size_t num_channels,
>-                                int sample_rate,
>-                                size_t bytes_per_sample) {
>-  return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample);
>-}
>-
>-static inline uint16_t BlockAlign(size_t num_channels,
>-                                  size_t bytes_per_sample) {
>-  return static_cast<uint16_t>(num_channels * bytes_per_sample);
>-}
>-
> void WriteWavHeader(uint8_t* buf,
>                     size_t num_channels,
>                     int sample_rate,
>@@ -162,7 +208,7 @@ void WriteWavHeader(uint8_t* buf,
>   RTC_CHECK(CheckWavParameters(num_channels, sample_rate, format,
>                                bytes_per_sample, num_samples));
> 
>-  WavHeader header;
>+  auto header = rtc::MsanUninitialized<WavHeader>({});
>   const size_t bytes_in_payload = bytes_per_sample * num_samples;
> 
>   WriteFourCC(&header.riff.header.ID, 'R', 'I', 'F', 'F');
>@@ -194,25 +240,38 @@ bool ReadWavHeader(ReadableWav* readable,
>                    WavFormat* format,
>                    size_t* bytes_per_sample,
>                    size_t* num_samples) {
>-  WavHeader header;
>-  if (readable->Read(&header, kWavHeaderSize - sizeof(header.data)) !=
>-      kWavHeaderSize - sizeof(header.data))
>+  auto header = rtc::MsanUninitialized<WavHeader>({});
>+
>+  // Read RIFF chunk.
>+  if (readable->Read(&header.riff, sizeof(header.riff)) != sizeof(header.riff))
>+    return false;
>+  if (ReadFourCC(header.riff.header.ID) != "RIFF")
>+    return false;
>+  if (ReadFourCC(header.riff.Format) != "WAVE")
>     return false;
> 
>-  const uint32_t fmt_size = ReadLE32(header.fmt.header.Size);
>-  if (fmt_size != kFmtSubchunkSize) {
>-    // There is an optional two-byte extension field permitted to be present
>-    // with PCM, but which must be zero.
>-    int16_t ext_size;
>-    if (kFmtSubchunkSize + sizeof(ext_size) != fmt_size)
>-      return false;
>-    if (readable->Read(&ext_size, sizeof(ext_size)) != sizeof(ext_size))
>-      return false;
>-    if (ext_size != 0)
>-      return false;
>+  // Find "fmt " and "data" chunks. While the official Wave file specification
>+  // does not put requirements on the chunks order, it is uncommon to find the
>+  // "data" chunk before the "fmt " one. The code below fails if this is not the
>+  // case.
>+  FindWaveChunk(&header.fmt.header, readable, "fmt ");
>+  if (ReadFourCC(header.fmt.header.ID) != "fmt ") {
>+    RTC_LOG(LS_ERROR) << "Cannot find 'fmt ' chunk.";
>+    return false;
>+  }
>+  if (!ReadFmtChunkData(&header.fmt, readable)) {
>+    RTC_LOG(LS_ERROR) << "Cannot read 'fmt ' chunk.";
>+    return false;
>+  }
>+  if (readable->Eof()) {
>+    RTC_LOG(LS_ERROR) << "'fmt ' chunk placed after 'data' chunk.";
>+    return false;
>   }
>-  if (readable->Read(&header.data, sizeof(header.data)) != sizeof(header.data))
>+  FindWaveChunk(&header.data.header, readable, "data");
>+  if (ReadFourCC(header.data.header.ID) != "data") {
>+    RTC_LOG(LS_ERROR) << "Cannot find 'data' chunk.";
>     return false;
>+  }
> 
>   // Parse needed fields.
>   *format = static_cast<WavFormat>(ReadLE16(header.fmt.AudioFormat));
>@@ -224,16 +283,6 @@ bool ReadWavHeader(ReadableWav* readable,
>     return false;
>   *num_samples = bytes_in_payload / *bytes_per_sample;
> 
>-  // Sanity check remaining fields.
>-  if (ReadFourCC(header.riff.header.ID) != "RIFF")
>-    return false;
>-  if (ReadFourCC(header.riff.Format) != "WAVE")
>-    return false;
>-  if (ReadFourCC(header.fmt.header.ID) != "fmt ")
>-    return false;
>-  if (ReadFourCC(header.data.header.ID) != "data")
>-    return false;
>-
>   if (ReadLE32(header.riff.header.Size) < RiffChunkSize(bytes_in_payload))
>     return false;
>   if (ReadLE32(header.fmt.ByteRate) !=
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.h
>index 872d3abce5ca4cdc940ca91440441e7237bcbe0b..a519ba556f54f0365494beb637d8d6b7ba249670 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.h
>@@ -21,8 +21,11 @@ static const size_t kWavHeaderSize = 44;
> class ReadableWav {
>  public:
>   // Returns the number of bytes read.
>-  size_t virtual Read(void* buf, size_t num_bytes) = 0;
>-  virtual ~ReadableWav() {}
>+  virtual size_t Read(void* buf, size_t num_bytes) = 0;
>+  // Returns true if the end-of-file has been reached.
>+  virtual bool Eof() const = 0;
>+  virtual bool SeekForward(uint32_t num_bytes) = 0;
>+  virtual ~ReadableWav() = default;
> };
> 
> enum WavFormat {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header_unittest.cc
>index b7169b53c8f1b1a1c1c4fd24bcde860b410817ee..2e2eda5a6f2d08f23cfba5dcd49f9c3fc5a7a0dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header_unittest.cc
>@@ -19,12 +19,6 @@ namespace webrtc {
> // Doesn't take ownership of the buffer.
> class ReadableWavBuffer : public ReadableWav {
>  public:
>-  ReadableWavBuffer(const uint8_t* buf, size_t size)
>-      : buf_(buf),
>-        size_(size),
>-        pos_(0),
>-        buf_exhausted_(false),
>-        check_read_size_(true) {}
>   ReadableWavBuffer(const uint8_t* buf, size_t size, bool check_read_size)
>       : buf_(buf),
>         size_(size),
>@@ -57,6 +51,27 @@ class ReadableWavBuffer : public ReadableWav {
>     return num_bytes;
>   }
> 
>+  bool Eof() const override { return pos_ == size_; }
>+
>+  bool SeekForward(uint32_t num_bytes) override {
>+    // Verify we don't try to read outside of a properly sized header.
>+    if (size_ >= kWavHeaderSize)
>+      EXPECT_GE(size_, pos_ + num_bytes);
>+    EXPECT_FALSE(buf_exhausted_);
>+
>+    const size_t bytes_remaining = size_ - pos_;
>+    if (num_bytes > bytes_remaining) {
>+      // Error: cannot seek beyond EOF.
>+      return false;
>+    }
>+    if (num_bytes == bytes_remaining) {
>+      // There should not be another read attempt after this point.
>+      buf_exhausted_ = true;
>+    }
>+    pos_ += num_bytes;
>+    return true;
>+  }
>+
>  private:
>   const uint8_t* buf_;
>   const size_t size_;
>@@ -103,7 +118,7 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>   // invalid field is indicated in the array name, and in the comments with
>   // *BAD*.
>   {
>-    static const uint8_t kBadRiffID[] = {
>+    constexpr uint8_t kBadRiffID[] = {
>         // clang-format off
>         // clang formatting doesn't respect inline comments.
>       'R', 'i', 'f', 'f',  // *BAD*
>@@ -121,12 +136,13 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>       0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>         // clang-format on
>     };
>-    ReadableWavBuffer r(kBadRiffID, sizeof(kBadRiffID));
>+    ReadableWavBuffer r(kBadRiffID, sizeof(kBadRiffID),
>+                        /*check_read_size=*/false);
>     EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>                                &bytes_per_sample, &num_samples));
>   }
>   {
>-    static const uint8_t kBadBitsPerSample[] = {
>+    constexpr uint8_t kBadBitsPerSample[] = {
>         // clang-format off
>         // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>@@ -144,12 +160,13 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>       0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>         // clang-format on
>     };
>-    ReadableWavBuffer r(kBadBitsPerSample, sizeof(kBadBitsPerSample));
>+    ReadableWavBuffer r(kBadBitsPerSample, sizeof(kBadBitsPerSample),
>+                        /*check_read_size=*/true);
>     EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>                                &bytes_per_sample, &num_samples));
>   }
>   {
>-    static const uint8_t kBadByteRate[] = {
>+    constexpr uint8_t kBadByteRate[] = {
>         // clang-format off
>         // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>@@ -167,12 +184,13 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>       0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>         // clang-format on
>     };
>-    ReadableWavBuffer r(kBadByteRate, sizeof(kBadByteRate));
>+    ReadableWavBuffer r(kBadByteRate, sizeof(kBadByteRate),
>+                        /*check_read_size=*/true);
>     EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>                                &bytes_per_sample, &num_samples));
>   }
>   {
>-    static const uint8_t kBadFmtHeaderSize[] = {
>+    constexpr uint8_t kBadFmtHeaderSize[] = {
>         // clang-format off
>         // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>@@ -191,12 +209,13 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>       0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>         // clang-format on
>     };
>-    ReadableWavBuffer r(kBadFmtHeaderSize, sizeof(kBadFmtHeaderSize), false);
>+    ReadableWavBuffer r(kBadFmtHeaderSize, sizeof(kBadFmtHeaderSize),
>+                        /*check_read_size=*/false);
>     EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>                                &bytes_per_sample, &num_samples));
>   }
>   {
>-    static const uint8_t kNonZeroExtensionField[] = {
>+    constexpr uint8_t kNonZeroExtensionField[] = {
>         // clang-format off
>         // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>@@ -216,12 +235,12 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>         // clang-format on
>     };
>     ReadableWavBuffer r(kNonZeroExtensionField, sizeof(kNonZeroExtensionField),
>-                        false);
>+                        /*check_read_size=*/false);
>     EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>                                &bytes_per_sample, &num_samples));
>   }
>   {
>-    static const uint8_t kMissingDataChunk[] = {
>+    constexpr uint8_t kMissingDataChunk[] = {
>         // clang-format off
>         // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>@@ -237,12 +256,13 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>       8, 0,  // bits per sample: 1 * 8
>         // clang-format on
>     };
>-    ReadableWavBuffer r(kMissingDataChunk, sizeof(kMissingDataChunk));
>+    ReadableWavBuffer r(kMissingDataChunk, sizeof(kMissingDataChunk),
>+                        /*check_read_size=*/true);
>     EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>                                &bytes_per_sample, &num_samples));
>   }
>   {
>-    static const uint8_t kMissingFmtAndDataChunks[] = {
>+    constexpr uint8_t kMissingFmtAndDataChunks[] = {
>         // clang-format off
>         // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>@@ -251,7 +271,8 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>         // clang-format on
>     };
>     ReadableWavBuffer r(kMissingFmtAndDataChunks,
>-                        sizeof(kMissingFmtAndDataChunks));
>+                        sizeof(kMissingFmtAndDataChunks),
>+                        /*check_read_size=*/true);
>     EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>                                &bytes_per_sample, &num_samples));
>   }
>@@ -259,11 +280,11 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
> 
> // Try writing and reading a valid WAV header and make sure it looks OK.
> TEST(WavHeaderTest, WriteAndReadWavHeader) {
>-  static const int kSize = 4 + kWavHeaderSize + 4;
>+  constexpr int kSize = 4 + kWavHeaderSize + 4;
>   uint8_t buf[kSize];
>   memset(buf, 0xa4, sizeof(buf));
>   WriteWavHeader(buf + 4, 17, 12345, kWavFormatALaw, 1, 123457689);
>-  static const uint8_t kExpectedBuf[] = {
>+  constexpr uint8_t kExpectedBuf[] = {
>       // clang-format off
>       // clang formatting doesn't respect inline comments.
>     0xa4, 0xa4, 0xa4, 0xa4,  // untouched bytes before header
>@@ -291,7 +312,8 @@ TEST(WavHeaderTest, WriteAndReadWavHeader) {
>   WavFormat format = kWavFormatPcm;
>   size_t bytes_per_sample = 0;
>   size_t num_samples = 0;
>-  ReadableWavBuffer r(buf + 4, sizeof(buf) - 8);
>+  ReadableWavBuffer r(buf + 4, sizeof(buf) - 8,
>+                      /*check_read_size=*/true);
>   EXPECT_TRUE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>                             &bytes_per_sample, &num_samples));
>   EXPECT_EQ(17u, num_channels);
>@@ -303,24 +325,67 @@ TEST(WavHeaderTest, WriteAndReadWavHeader) {
> 
> // Try reading an atypical but valid WAV header and make sure it's parsed OK.
> TEST(WavHeaderTest, ReadAtypicalWavHeader) {
>-  static const uint8_t kBuf[] = {
>+  constexpr uint8_t kBuf[] = {
>       // clang-format off
>       // clang formatting doesn't respect inline comments.
>     'R', 'I', 'F', 'F',
>-    0x3d, 0xd1, 0x5b, 0x07,  // size of whole file - 8 + an extra 128 bytes of
>-                             // "metadata": 123457689 + 44 - 8 + 128. (atypical)
>+    0xbf, 0xd0, 0x5b, 0x07,  // Size of whole file - 8 + extra 2 bytes of zero
>+                             // extension: 123457689 + 44 - 8 + 2 (atypical).
>     'W', 'A', 'V', 'E',
>     'f', 'm', 't', ' ',
>-    18, 0, 0, 0,  // size of fmt block (with an atypical extension size field)
>-    6, 0,  // format: A-law (6)
>-    17, 0,  // channels: 17
>-    0x39, 0x30, 0, 0,  // sample rate: 12345
>-    0xc9, 0x33, 0x03, 0,  // byte rate: 1 * 17 * 12345
>-    17, 0,  // block align: NumChannels * BytesPerSample
>-    8, 0,  // bits per sample: 1 * 8
>-    0, 0,  // zero extension size field (atypical)
>+    18, 0, 0, 0,             // Size of fmt block (with an atypical extension
>+                             // size field).
>+    6, 0,                    // Format: A-law (6).
>+    17, 0,                   // Channels: 17.
>+    0x39, 0x30, 0, 0,        // Sample rate: 12345.
>+    0xc9, 0x33, 0x03, 0,     // Byte rate: 1 * 17 * 12345.
>+    17, 0,                   // Block align: NumChannels * BytesPerSample.
>+    8, 0,                    // Bits per sample: 1 * 8.
>+    0, 0,                    // Zero extension size field (atypical).
>     'd', 'a', 't', 'a',
>-    0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>+    0x99, 0xd0, 0x5b, 0x07,  // Size of payload: 123457689.
>+      // clang-format on
>+  };
>+
>+  size_t num_channels = 0;
>+  int sample_rate = 0;
>+  WavFormat format = kWavFormatPcm;
>+  size_t bytes_per_sample = 0;
>+  size_t num_samples = 0;
>+  ReadableWavBuffer r(kBuf, sizeof(kBuf), /*check_read_size=*/true);
>+  EXPECT_TRUE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>+                            &bytes_per_sample, &num_samples));
>+  EXPECT_EQ(17u, num_channels);
>+  EXPECT_EQ(12345, sample_rate);
>+  EXPECT_EQ(kWavFormatALaw, format);
>+  EXPECT_EQ(1u, bytes_per_sample);
>+  EXPECT_EQ(123457689u, num_samples);
>+}
>+
>+// Try reading a valid WAV header which contains an optional chunk and make sure
>+// it's parsed OK.
>+TEST(WavHeaderTest, ReadWavHeaderWithOptionalChunk) {
>+  constexpr uint8_t kBuf[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>+    'R', 'I', 'F', 'F',
>+    0xcd, 0xd0, 0x5b, 0x07,  // Size of whole file - 8 + an extra 16 bytes of
>+                             // "metadata" (8 bytes header, 16 bytes payload):
>+                             // 123457689 + 44 - 8 + 16.
>+    'W', 'A', 'V', 'E',
>+    'f', 'm', 't', ' ',
>+    16, 0, 0, 0,             // Size of fmt block.
>+    6, 0,                    // Format: A-law (6).
>+    17, 0,                   // Channels: 17.
>+    0x39, 0x30, 0, 0,        // Sample rate: 12345.
>+    0xc9, 0x33, 0x03, 0,     // Byte rate: 1 * 17 * 12345.
>+    17, 0,                   // Block align: NumChannels * BytesPerSample.
>+    8, 0,                    // Bits per sample: 1 * 8.
>+    'L', 'I', 'S', 'T',      // Metadata chunk ID.
>+    16, 0, 0, 0,             // Metadata chunk payload size.
>+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,  // Metadata (16 bytes).
>+    'd', 'a', 't', 'a',
>+    0x99, 0xd0, 0x5b, 0x07,  // Size of payload: 123457689.
>       // clang-format on
>   };
> 
>@@ -329,7 +394,7 @@ TEST(WavHeaderTest, ReadAtypicalWavHeader) {
>   WavFormat format = kWavFormatPcm;
>   size_t bytes_per_sample = 0;
>   size_t num_samples = 0;
>-  ReadableWavBuffer r(kBuf, sizeof(kBuf));
>+  ReadableWavBuffer r(kBuf, sizeof(kBuf), /*check_read_size=*/true);
>   EXPECT_TRUE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>                             &bytes_per_sample, &num_samples));
>   EXPECT_EQ(17u, num_channels);
>@@ -339,4 +404,37 @@ TEST(WavHeaderTest, ReadAtypicalWavHeader) {
>   EXPECT_EQ(123457689u, num_samples);
> }
> 
>+// Try reading an invalid WAV header which has the the data chunk before the
>+// format one and make sure it's not parsed.
>+TEST(WavHeaderTest, ReadWavHeaderWithDataBeforeFormat) {
>+  constexpr uint8_t kBuf[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>+    'R', 'I', 'F', 'F',
>+    52,  0,   0,   0,    // Size of whole file - 8: 16 + 44 - 8.
>+    'W', 'A', 'V', 'E',
>+    'd', 'a', 't', 'a',
>+    16, 0, 0, 0,         // Data chunk payload size.
>+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,  // Data 16 bytes.
>+    'f', 'm', 't', ' ',
>+    16,  0,   0,   0,    // Size of fmt block.
>+    6,   0,              // Format: A-law (6).
>+    1,   0,              // Channels: 1.
>+    60,  0,   0,   0,    // Sample rate: 60.
>+    60,  0,   0,   0,    // Byte rate: 1 * 1 * 60.
>+    1,   0,              // Block align: NumChannels * BytesPerSample.
>+    8,   0,              // Bits per sample: 1 * 8.
>+      // clang-format on
>+  };
>+
>+  size_t num_channels = 0;
>+  int sample_rate = 0;
>+  WavFormat format = kWavFormatPcm;
>+  size_t bytes_per_sample = 0;
>+  size_t num_samples = 0;
>+  ReadableWavBuffer r(kBuf, sizeof(kBuf), /*check_read_size=*/false);
>+  EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>+                             &bytes_per_sample, &num_samples));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.h
>index 99c4064a3cf660d1326ad590e4f68fa34443ed0a..848b899a0dd1f489c526d0b25451a5504c2b2b38 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.h
>@@ -11,16 +11,15 @@
> #ifndef COMMON_TYPES_H_
> #define COMMON_TYPES_H_
> 
>-#include <stddef.h>
>-#include <string.h>
>-#include <string>
>-#include <vector>
>+#include <stddef.h>  // For size_t
>+#include <cstdint>
> 
>-#include "api/array_view.h"
>+#include "absl/strings/match.h"
> // TODO(sprang): Remove this include when all usage includes it directly.
> #include "api/video/video_bitrate_allocation.h"
>+// TODO(bugs.webrtc.org/7660): Delete include once downstream code is updated.
>+#include "api/video/video_codec_type.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/deprecation.h"
> 
> #if defined(_MSC_VER)
> // Disable "new behavior: elements of array will be default initialized"
>@@ -30,16 +29,6 @@
> 
> #define RTP_PAYLOAD_NAME_SIZE 32u
> 
>-#if defined(WEBRTC_WIN) || defined(WIN32)
>-// Compares two strings without regard to case.
>-#define STR_CASE_CMP(s1, s2) ::_stricmp(s1, s2)
>-// Compares characters of two strings without regard to case.
>-#define STR_NCASE_CMP(s1, s2, n) ::_strnicmp(s1, s2, n)
>-#else
>-#define STR_CASE_CMP(s1, s2) ::strcasecmp(s1, s2)
>-#define STR_NCASE_CMP(s1, s2, n) ::strncasecmp(s1, s2, n)
>-#endif
>-
> namespace webrtc {
> 
> enum FrameType {
>@@ -50,29 +39,6 @@ enum FrameType {
>   kVideoFrameDelta = 4,
> };
> 
>-// Statistics for an RTCP channel
>-struct RtcpStatistics {
>-  RtcpStatistics()
>-      : fraction_lost(0),
>-        packets_lost(0),
>-        extended_highest_sequence_number(0),
>-        jitter(0) {}
>-
>-  uint8_t fraction_lost;
>-  int32_t packets_lost;  // Defined as a 24 bit signed integer in RTCP
>-  uint32_t extended_highest_sequence_number;
>-  uint32_t jitter;
>-};
>-
>-class RtcpStatisticsCallback {
>- public:
>-  virtual ~RtcpStatisticsCallback() {}
>-
>-  virtual void StatisticsUpdated(const RtcpStatistics& statistics,
>-                                 uint32_t ssrc) = 0;
>-  virtual void CNameChanged(const char* cname, uint32_t ssrc) = 0;
>-};
>-
> // Statistics for RTCP packet types.
> struct RtcpPacketTypeCounter {
>   RtcpPacketTypeCounter()
>@@ -207,7 +173,7 @@ struct CodecInst {
> 
>   bool operator==(const CodecInst& other) const {
>     return pltype == other.pltype &&
>-           (STR_CASE_CMP(plname, other.plname) == 0) &&
>+           absl::EqualsIgnoreCase(plname, other.plname) &&
>            plfreq == other.plfreq && pacsize == other.pacsize &&
>            channels == other.channels && rate == other.rate;
>   }
>@@ -218,80 +184,6 @@ struct CodecInst {
> // RTP
> enum { kRtpCsrcSize = 15 };  // RFC 3550 page 13
> 
>-// NETEQ statistics.
>-struct NetworkStatistics {
>-  // current jitter buffer size in ms
>-  uint16_t currentBufferSize;
>-  // preferred (optimal) buffer size in ms
>-  uint16_t preferredBufferSize;
>-  // adding extra delay due to "peaky jitter"
>-  bool jitterPeaksFound;
>-  // Stats below correspond to similarly-named fields in the WebRTC stats spec.
>-  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
>-  uint64_t totalSamplesReceived;
>-  uint64_t concealedSamples;
>-  uint64_t concealmentEvents;
>-  uint64_t jitterBufferDelayMs;
>-  // Stats below DO NOT correspond directly to anything in the WebRTC stats
>-  // Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
>-  uint16_t currentPacketLossRate;
>-  // Late loss rate; fraction between 0 and 1, scaled to Q14.
>-  union {
>-    RTC_DEPRECATED uint16_t currentDiscardRate;
>-  };
>-  // fraction (of original stream) of synthesized audio inserted through
>-  // expansion (in Q14)
>-  uint16_t currentExpandRate;
>-  // fraction (of original stream) of synthesized speech inserted through
>-  // expansion (in Q14)
>-  uint16_t currentSpeechExpandRate;
>-  // fraction of synthesized speech inserted through pre-emptive expansion
>-  // (in Q14)
>-  uint16_t currentPreemptiveRate;
>-  // fraction of data removed through acceleration (in Q14)
>-  uint16_t currentAccelerateRate;
>-  // fraction of data coming from secondary decoding (in Q14)
>-  uint16_t currentSecondaryDecodedRate;
>-  // Fraction of secondary data, including FEC and RED, that is discarded (in
>-  // Q14). Discarding of secondary data can be caused by the reception of the
>-  // primary data, obsoleting the secondary data. It can also be caused by early
>-  // or late arrival of secondary data.
>-  uint16_t currentSecondaryDiscardedRate;
>-  // clock-drift in parts-per-million (negative or positive)
>-  int32_t clockDriftPPM;
>-  // average packet waiting time in the jitter buffer (ms)
>-  int meanWaitingTimeMs;
>-  // median packet waiting time in the jitter buffer (ms)
>-  int medianWaitingTimeMs;
>-  // min packet waiting time in the jitter buffer (ms)
>-  int minWaitingTimeMs;
>-  // max packet waiting time in the jitter buffer (ms)
>-  int maxWaitingTimeMs;
>-  // added samples in off mode due to packet loss
>-  size_t addedSamples;
>-};
>-
>-// Statistics for calls to AudioCodingModule::PlayoutData10Ms().
>-struct AudioDecodingCallStats {
>-  AudioDecodingCallStats()
>-      : calls_to_silence_generator(0),
>-        calls_to_neteq(0),
>-        decoded_normal(0),
>-        decoded_plc(0),
>-        decoded_cng(0),
>-        decoded_plc_cng(0),
>-        decoded_muted_output(0) {}
>-
>-  int calls_to_silence_generator;  // Number of calls where silence generated,
>-                                   // and NetEq was disengaged from decoding.
>-  int calls_to_neteq;              // Number of calls to NetEq.
>-  int decoded_normal;  // Number of calls where audio RTP packet decoded.
>-  int decoded_plc;     // Number of calls resulted in PLC.
>-  int decoded_cng;  // Number of calls where comfort noise generated due to DTX.
>-  int decoded_plc_cng;       // Number of calls resulted where PLC faded to CNG.
>-  int decoded_muted_output;  // Number of calls returning a muted state output.
>-};
>-
> // ==================================================================
> // Video specific types
> // ==================================================================
>@@ -330,18 +222,6 @@ enum Profile {
> 
> }  // namespace H264
> 
>-// Video codec types
>-enum VideoCodecType {
>-  // There are various memset(..., 0, ...) calls in the code that rely on
>-  // kVideoCodecGeneric being zero.
>-  kVideoCodecGeneric = 0,
>-  kVideoCodecVP8,
>-  kVideoCodecVP9,
>-  kVideoCodecH264,
>-  kVideoCodecI420,
>-  kVideoCodecMultiplex,
>-};
>-
> struct SpatialLayer {
>   bool operator==(const SpatialLayer& other) const;
>   bool operator!=(const SpatialLayer& other) const { return !(*this == other); }
>@@ -361,9 +241,6 @@ struct SpatialLayer {
> // settings such as resolution.
> typedef SpatialLayer SimulcastStream;
> 
>-// TODO(sprang): Remove this when downstream projects have been updated.
>-using BitrateAllocation = VideoBitrateAllocation;
>-
> // Bandwidth over-use detector options.  These are used to drive
> // experimentation with bandwidth estimation parameters.
> // See modules/remote_bitrate_estimator/overuse_detector.h
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/BUILD.gn
>index 43780c113d507fb4b6682f9fef83be530baf7597..125bcfc558337e0e23c71e8929f63d088f9d4697 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/BUILD.gn
>@@ -26,7 +26,6 @@ rtc_static_library("common_video") {
>     "h264/sps_vui_rewriter.h",
>     "i420_buffer_pool.cc",
>     "include/bitrate_adjuster.h",
>-    "include/frame_callback.h",
>     "include/i420_buffer_pool.h",
>     "include/incoming_video_stream.h",
>     "include/video_frame.h",
>@@ -34,7 +33,6 @@ rtc_static_library("common_video") {
>     "incoming_video_stream.cc",
>     "libyuv/include/webrtc_libyuv.h",
>     "libyuv/webrtc_libyuv.cc",
>-    "video_frame.cc",
>     "video_frame_buffer.cc",
>     "video_render_frames.cc",
>     "video_render_frames.h",
>@@ -42,6 +40,7 @@ rtc_static_library("common_video") {
> 
>   deps = [
>     "..:webrtc_common",
>+    "../api/video:encoded_image",
>     "../api/video:video_bitrate_allocation",
>     "../api/video:video_bitrate_allocator",
>     "../api/video:video_frame",
>@@ -96,6 +95,7 @@ if (rtc_include_tests) {
>       "../rtc_base:rtc_base_tests_utils",
>       "../test:fileutils",
>       "../test:test_main",
>+      "../test:test_support",
>       "../test:video_test_common",
>       "//testing/gtest",
>       "//third_party/libyuv",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster.cc
>index 163c4b1981474b392bb8707206f7ad5785f969ea..ac21f2bd20387a6731727b48010ebc205ae12de2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster.cc
>@@ -13,7 +13,6 @@
> #include <algorithm>
> #include <cmath>
> 
>-#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/timeutils.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.cc
>index 2e63b9d4a16ef5e61a3cfcb1f5eb04111c9c9459..f3d2f8a6bf804de31433cd2d7cb54aa00a8b53aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.cc
>@@ -9,13 +9,12 @@
>  */
> #include "common_video/h264/h264_bitstream_parser.h"
> 
>-#include <memory>
>+#include <stdlib.h>
>+#include <cstdint>
> #include <vector>
> 
>-#include "rtc_base/bitbuffer.h"
>-#include "rtc_base/checks.h"
>-
> #include "common_video/h264/h264_common.h"
>+#include "rtc_base/bitbuffer.h"
> #include "rtc_base/logging.h"
> 
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.h
>index b3fac7bb215726eff82ff33aa1e2374aceee3232..962c9c16ad2f46fbad7e47d3c264f5866a21ba28 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.h
>@@ -17,10 +17,6 @@
> #include "common_video/h264/pps_parser.h"
> #include "common_video/h264/sps_parser.h"
> 
>-namespace rtc {
>-class BitBufferWriter;
>-}
>-
> namespace webrtc {
> 
> // Stateful H264 bitstream parser (due to SPS/PPS). Used to parse out QP values
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_common.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_common.cc
>index 60ed94d40311675865ef844cf88465686eeb0f51..5e58ba62e91308534fdfbbeaf9427b7df0696caf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_common.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_common.cc
>@@ -10,6 +10,8 @@
> 
> #include "common_video/h264/h264_common.h"
> 
>+#include <cstdint>
>+
> namespace webrtc {
> namespace H264 {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_common.h
>index a58a9f55a59cdda46079222f7a2ba5682ec52341..027833b7cd7c26fe8d0f88c910f1db8894dd6444 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_common.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_common.h
>@@ -11,7 +11,8 @@
> #ifndef COMMON_VIDEO_H264_H264_COMMON_H_
> #define COMMON_VIDEO_H264_H264_COMMON_H_
> 
>-#include <memory>
>+#include <stddef.h>
>+#include <stdint.h>
> #include <vector>
> 
> #include "rtc_base/buffer.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.cc
>index 5bc29f359257d1cdf14deac3b59ceada544698cd..464f6081e31e0c5b653b467ef97b31b8e08f6f02 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.cc
>@@ -10,12 +10,12 @@
> 
> #include "common_video/h264/pps_parser.h"
> 
>-#include <memory>
>+#include <cstdint>
> #include <vector>
> 
> #include "common_video/h264/h264_common.h"
> #include "rtc_base/bitbuffer.h"
>-#include "rtc_base/logging.h"
>+#include "rtc_base/checks.h"
> 
> #define RETURN_EMPTY_ON_FAIL(x) \
>   if (!(x)) {                   \
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.cc
>index b6799a3be40840461afda20a140bc2ef9b64e764..c6f6d47466b15cbf13cdf9f8cff9bc580d06fde8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.cc
>@@ -10,12 +10,11 @@
> 
> #include "common_video/h264/sps_parser.h"
> 
>-#include <memory>
>+#include <cstdint>
> #include <vector>
> 
> #include "common_video/h264/h264_common.h"
> #include "rtc_base/bitbuffer.h"
>-#include "rtc_base/logging.h"
> 
> namespace {
> typedef absl::optional<webrtc::SpsParser::SpsState> OptionalSps;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.cc
>index 749b62e6d21d313af0407436c0dcb09d7052d9e0..3eab11f188cebb0ff8c83b1593d8bc35b522d82d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.cc
>@@ -11,18 +11,17 @@
> 
> #include "common_video/h264/sps_vui_rewriter.h"
> 
>-#include <algorithm>
>-#include <memory>
>+#include <string.h>
>+#include <cstdint>
> #include <vector>
> 
>+#include "common_video/h264/h264_common.h"
>+#include "common_video/h264/sps_parser.h"
> #include "rtc_base/bitbuffer.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_minmax.h"
> 
>-#include "common_video/h264/h264_common.h"
>-#include "common_video/h264/sps_parser.h"
>-
> namespace webrtc {
> 
> // The maximum expected growth from adding a VUI to the SPS. It's actually
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.h
>index 233051d2d61dfb6f85799c30badcd69b86d9d068..20d1dd0cdbfef688007b9c2362846fcae8714d07 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.h
>@@ -12,14 +12,13 @@
> #ifndef COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_
> #define COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "absl/types/optional.h"
> #include "common_video/h264/sps_parser.h"
> #include "rtc_base/buffer.h"
> 
>-namespace rtc {
>-class BitBuffer;
>-}
>-
> namespace webrtc {
> 
> // A class that can parse an SPS block of a NAL unit and if necessary
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/i420_buffer_pool.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/i420_buffer_pool.cc
>index c403eeb4fcba14a499f2b091040cd3158817ddb5..2e6cdc83b51b6bfb4f889f84851889ae1fcb08ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/i420_buffer_pool.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/i420_buffer_pool.cc
>@@ -10,6 +10,8 @@
> 
> #include "common_video/include/i420_buffer_pool.h"
> 
>+#include <limits>
>+
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/bitrate_adjuster.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/bitrate_adjuster.h
>index ee312e4f9a452e759be39dd5a569df1c5c9dfd57..bc2c6bb84936d39b92957f0a482cc6996ed11ed0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/bitrate_adjuster.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/bitrate_adjuster.h
>@@ -11,10 +11,13 @@
> #ifndef COMMON_VIDEO_INCLUDE_BITRATE_ADJUSTER_H_
> #define COMMON_VIDEO_INCLUDE_BITRATE_ADJUSTER_H_
> 
>-#include <functional>
>+#include <stddef.h>
>+#include <stdint.h>
> 
>+#include "absl/types/optional.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/rate_statistics.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/frame_callback.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/frame_callback.h
>deleted file mode 100644
>index a3883c1d8d2f688a3673b0ae362e7ef8bacb3945..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/frame_callback.h
>+++ /dev/null
>@@ -1,59 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef COMMON_VIDEO_INCLUDE_FRAME_CALLBACK_H_
>-#define COMMON_VIDEO_INCLUDE_FRAME_CALLBACK_H_
>-
>-#include <stddef.h>
>-#include <stdint.h>
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-class VideoFrame;
>-
>-struct EncodedFrame {
>- public:
>-  EncodedFrame()
>-      : data_(nullptr),
>-        length_(0),
>-        frame_type_(kEmptyFrame),
>-        stream_id_(0),
>-        timestamp_(0) {}
>-  EncodedFrame(const uint8_t* data,
>-               size_t length,
>-               FrameType frame_type,
>-               size_t stream_id,
>-               uint32_t timestamp)
>-      : data_(data),
>-        length_(length),
>-        frame_type_(frame_type),
>-        stream_id_(stream_id),
>-        timestamp_(timestamp) {}
>-
>-  const uint8_t* data_;
>-  const size_t length_;
>-  const FrameType frame_type_;
>-  const size_t stream_id_;
>-  const uint32_t timestamp_;
>-};
>-
>-class EncodedFrameObserver {
>- public:
>-  virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) = 0;
>-
>- protected:
>-  virtual ~EncodedFrameObserver() {}
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // COMMON_VIDEO_INCLUDE_FRAME_CALLBACK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/i420_buffer_pool.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/i420_buffer_pool.h
>index 79f7eec23af08fd0ea656874066659fd00286d87..2dcee195fad1e73b28b2cb0ec41e784f6374258a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/i420_buffer_pool.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/i420_buffer_pool.h
>@@ -11,12 +11,13 @@
> #ifndef COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
> #define COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
> 
>-#include <limits>
>+#include <stddef.h>
> #include <list>
> 
> #include "api/video/i420_buffer.h"
> #include "rtc_base/race_checker.h"
> #include "rtc_base/refcountedobject.h"
>+#include "rtc_base/scoped_ref_ptr.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/incoming_video_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/incoming_video_stream.h
>index 8063061d31327228022d9e9b544d5b4477d3a00f..fe8d10e216a0dfbd08a68a8be9d0a6ef19759ff8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/incoming_video_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/incoming_video_stream.h
>@@ -11,10 +11,14 @@
> #ifndef COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
> #define COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
> 
>+#include <stdint.h>
>+
>+#include "api/video/video_frame.h"
> #include "api/video/video_sink_interface.h"
> #include "common_video/video_render_frames.h"
> #include "rtc_base/race_checker.h"
> #include "rtc_base/task_queue.h"
>+#include "rtc_base/thread_checker.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame.h
>index 565870e4dc36471824a86bf2ce44e0c1666767b6..ba280f2a8cdcbf86c8dbb0528ae82a973f25f07e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame.h
>@@ -11,91 +11,7 @@
> #ifndef COMMON_VIDEO_INCLUDE_VIDEO_FRAME_H_
> #define COMMON_VIDEO_INCLUDE_VIDEO_FRAME_H_
> 
>-// TODO(nisse): This header file should eventually be deleted. The
>-// EncodedImage class stays in this file until we have figured out how
>-// to refactor and clean up related interfaces, at which point it
>-// should be moved to somewhere under api/.
>-
>-#include "absl/types/optional.h"
>-#include "api/video/video_content_type.h"
>-#include "api/video/video_rotation.h"
>-#include "api/video/video_timing.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-// TODO(pbos): Rename EncodedFrame and reformat this class' members.
>-class EncodedImage {
>- public:
>-  static const size_t kBufferPaddingBytesH264;
>-
>-  // Some decoders require encoded image buffers to be padded with a small
>-  // number of additional bytes (due to over-reading byte readers).
>-  static size_t GetBufferPaddingBytes(VideoCodecType codec_type);
>-
>-  EncodedImage();
>-  EncodedImage(const EncodedImage&);
>-  EncodedImage(uint8_t* buffer, size_t length, size_t size);
>-
>-  // TODO(nisse): Change style to timestamp(), set_timestamp(), for consistency
>-  // with the VideoFrame class.
>-  // Set frame timestamp (90kHz).
>-  void SetTimestamp(uint32_t timestamp) { timestamp_rtp_ = timestamp; }
>-
>-  // Get frame timestamp (90kHz).
>-  uint32_t Timestamp() const { return timestamp_rtp_; }
>-
>-  void SetEncodeTime(int64_t encode_start_ms, int64_t encode_finish_ms);
>-
>-  absl::optional<int> SpatialIndex() const {
>-    if (spatial_index_ < 0)
>-      return absl::nullopt;
>-    return spatial_index_;
>-  }
>-  void SetSpatialIndex(absl::optional<int> spatial_index) {
>-    RTC_DCHECK_GE(spatial_index.value_or(0), 0);
>-    RTC_DCHECK_LT(spatial_index.value_or(0), kMaxSpatialLayers);
>-    spatial_index_ = spatial_index.value_or(-1);
>-  }
>-
>-  uint32_t _encodedWidth = 0;
>-  uint32_t _encodedHeight = 0;
>-  // NTP time of the capture time in local timebase in milliseconds.
>-  int64_t ntp_time_ms_ = 0;
>-  int64_t capture_time_ms_ = 0;
>-  FrameType _frameType = kVideoFrameDelta;
>-  uint8_t* _buffer;
>-  size_t _length;
>-  size_t _size;
>-  VideoRotation rotation_ = kVideoRotation_0;
>-  VideoContentType content_type_ = VideoContentType::UNSPECIFIED;
>-  bool _completeFrame = false;
>-  int qp_ = -1;  // Quantizer value.
>-
>-  // When an application indicates non-zero values here, it is taken as an
>-  // indication that all future frames will be constrained with those limits
>-  // until the application indicates a change again.
>-  PlayoutDelay playout_delay_ = {-1, -1};
>-
>-  struct Timing {
>-    uint8_t flags = VideoSendTiming::kInvalid;
>-    int64_t encode_start_ms = 0;
>-    int64_t encode_finish_ms = 0;
>-    int64_t packetization_finish_ms = 0;
>-    int64_t pacer_exit_ms = 0;
>-    int64_t network_timestamp_ms = 0;
>-    int64_t network2_timestamp_ms = 0;
>-    int64_t receive_start_ms = 0;
>-    int64_t receive_finish_ms = 0;
>-  } timing_;
>-
>- private:
>-  uint32_t timestamp_rtp_ = 0;
>-  // -1 means not set. Use a plain int rather than optional, to keep this class
>-  // copyable with memcpy.
>-  int spatial_index_ = -1;
>-};
>-
>-}  // namespace webrtc
>+// TODO(nisse): Delete this file, after downstream code is updated.
>+#include "api/video/encoded_image.h"
> 
> #endif  // COMMON_VIDEO_INCLUDE_VIDEO_FRAME_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame_buffer.h
>index 11bb8125316aac912d1ff859aaadd730a821cf8a..f813851b9a9ea71ad577e490ffd66c1e717868a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame_buffer.h
>@@ -11,51 +11,15 @@
> #ifndef COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
> #define COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
> 
>-#include <memory>
>+#include <stdint.h>
> 
> #include "api/video/video_frame_buffer.h"
> #include "rtc_base/callback.h"
>+#include "rtc_base/refcountedobject.h"
> #include "rtc_base/scoped_ref_ptr.h"
> 
> namespace webrtc {
> 
>-// Deprecated. Please use WrapI420Buffer(...) instead.
>-class WrappedI420Buffer : public I420BufferInterface {
>- public:
>-  WrappedI420Buffer(int width,
>-                    int height,
>-                    const uint8_t* y_plane,
>-                    int y_stride,
>-                    const uint8_t* u_plane,
>-                    int u_stride,
>-                    const uint8_t* v_plane,
>-                    int v_stride,
>-                    const rtc::Callback0<void>& no_longer_used);
>-  int width() const override;
>-  int height() const override;
>-
>-  const uint8_t* DataY() const override;
>-  const uint8_t* DataU() const override;
>-  const uint8_t* DataV() const override;
>-  int StrideY() const override;
>-  int StrideU() const override;
>-  int StrideV() const override;
>-
>- private:
>-  friend class rtc::RefCountedObject<WrappedI420Buffer>;
>-  ~WrappedI420Buffer() override;
>-
>-  const int width_;
>-  const int height_;
>-  const uint8_t* const y_plane_;
>-  const uint8_t* const u_plane_;
>-  const uint8_t* const v_plane_;
>-  const int y_stride_;
>-  const int u_stride_;
>-  const int v_stride_;
>-  rtc::Callback0<void> no_longer_used_cb_;
>-};
>-
> rtc::scoped_refptr<I420BufferInterface> WrapI420Buffer(
>     int width,
>     int height,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/incoming_video_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/incoming_video_stream.cc
>index efca514556698f6f83bd1137e78063442c094939..c8b12e747ff6c8c979a2d0599b3608e12f679e33 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/incoming_video_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/incoming_video_stream.cc
>@@ -11,9 +11,11 @@
> #include "common_video/include/incoming_video_stream.h"
> 
> #include <memory>
>+#include <utility>
> 
>+#include "absl/types/optional.h"
> #include "common_video/video_render_frames.h"
>-#include "rtc_base/timeutils.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/trace_event.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h
>index 4cbfec20187e2204027270d802891fd1c7f39ee0..d3248340d875fea8175183572bbb23e9b6deb7ec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h
>@@ -15,11 +15,14 @@
> #ifndef COMMON_VIDEO_LIBYUV_INCLUDE_WEBRTC_LIBYUV_H_
> #define COMMON_VIDEO_LIBYUV_INCLUDE_WEBRTC_LIBYUV_H_
> 
>+#include <stdint.h>
> #include <stdio.h>
> #include <vector>
> 
> #include "api/video/video_frame.h"
>-#include "common_types.h"  // NOLINT(build/include)  // VideoTypes.
>+#include "api/video/video_frame_buffer.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "rtc_base/scoped_ref_ptr.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/webrtc_libyuv.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/webrtc_libyuv.cc
>index 6bdfabd5e9c76f9ec72898ab20e6b0f902175f6b..6e3fb3ed35a4a403aa5235e9781e95b102341080 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/webrtc_libyuv.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/webrtc_libyuv.cc
>@@ -124,29 +124,6 @@ int ExtractBuffer(const VideoFrame& input_frame, size_t size, uint8_t* buffer) {
>                        buffer);
> }
> 
>-int ConvertNV12ToRGB565(const uint8_t* src_frame,
>-                        uint8_t* dst_frame,
>-                        int width,
>-                        int height) {
>-  int abs_height = (height < 0) ? -height : height;
>-  const uint8_t* yplane = src_frame;
>-  const uint8_t* uvInterlaced = src_frame + (width * abs_height);
>-
>-  return libyuv::NV12ToRGB565(yplane, width, uvInterlaced, (width + 1) >> 1,
>-                              dst_frame, width, width, height);
>-}
>-
>-int ConvertRGB24ToARGB(const uint8_t* src_frame,
>-                       uint8_t* dst_frame,
>-                       int width,
>-                       int height,
>-                       int dst_stride) {
>-  if (dst_stride == 0)
>-    dst_stride = width;
>-  return libyuv::RGB24ToARGB(src_frame, width, dst_frame, dst_stride, width,
>-                             height);
>-}
>-
> int ConvertVideoType(VideoType video_type) {
>   switch (video_type) {
>     case VideoType::kUnknown:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame.cc
>deleted file mode 100644
>index f4618146711183cfaa37ee257dd2320acbf7fa55..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame.cc
>+++ /dev/null
>@@ -1,47 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "common_video/include/video_frame.h"
>-
>-#include <string.h>
>-
>-#include <algorithm>  // swap
>-
>-#include "rtc_base/bind.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-
>-// FFmpeg's decoder, used by H264DecoderImpl, requires up to 8 bytes padding due
>-// to optimized bitstream readers. See avcodec_decode_video2.
>-const size_t EncodedImage::kBufferPaddingBytesH264 = 8;
>-
>-size_t EncodedImage::GetBufferPaddingBytes(VideoCodecType codec_type) {
>-  switch (codec_type) {
>-    case kVideoCodecH264:
>-      return kBufferPaddingBytesH264;
>-    default:
>-      return 0;
>-  }
>-}
>-
>-EncodedImage::EncodedImage() : EncodedImage(nullptr, 0, 0) {}
>-
>-EncodedImage::EncodedImage(const EncodedImage&) = default;
>-
>-EncodedImage::EncodedImage(uint8_t* buffer, size_t length, size_t size)
>-    : _buffer(buffer), _length(length), _size(size) {}
>-
>-void EncodedImage::SetEncodeTime(int64_t encode_start_ms,
>-                                 int64_t encode_finish_ms) {
>-  timing_.encode_start_ms = encode_start_ms;
>-  timing_.encode_finish_ms = encode_finish_ms;
>-}
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>index be0b4927bbefb4cc0d73f1342382b85c0d99b787..5f89b4acb96d947da246c4ce42f913fc558970f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>@@ -203,57 +203,6 @@ rtc::scoped_refptr<I420BufferInterface> I010BufferBase::ToI420() {
> 
> }  // namespace
> 
>-WrappedI420Buffer::WrappedI420Buffer(int width,
>-                                     int height,
>-                                     const uint8_t* y_plane,
>-                                     int y_stride,
>-                                     const uint8_t* u_plane,
>-                                     int u_stride,
>-                                     const uint8_t* v_plane,
>-                                     int v_stride,
>-                                     const rtc::Callback0<void>& no_longer_used)
>-    : width_(width),
>-      height_(height),
>-      y_plane_(y_plane),
>-      u_plane_(u_plane),
>-      v_plane_(v_plane),
>-      y_stride_(y_stride),
>-      u_stride_(u_stride),
>-      v_stride_(v_stride),
>-      no_longer_used_cb_(no_longer_used) {}
>-
>-WrappedI420Buffer::~WrappedI420Buffer() {
>-  no_longer_used_cb_();
>-}
>-
>-int WrappedI420Buffer::width() const {
>-  return width_;
>-}
>-
>-int WrappedI420Buffer::height() const {
>-  return height_;
>-}
>-
>-const uint8_t* WrappedI420Buffer::DataY() const {
>-  return y_plane_;
>-}
>-const uint8_t* WrappedI420Buffer::DataU() const {
>-  return u_plane_;
>-}
>-const uint8_t* WrappedI420Buffer::DataV() const {
>-  return v_plane_;
>-}
>-
>-int WrappedI420Buffer::StrideY() const {
>-  return y_stride_;
>-}
>-int WrappedI420Buffer::StrideU() const {
>-  return u_stride_;
>-}
>-int WrappedI420Buffer::StrideV() const {
>-  return v_stride_;
>-}
>-
> rtc::scoped_refptr<I420BufferInterface> WrapI420Buffer(
>     int width,
>     int height,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.cc
>index 5a0b0b00712d5b44008b363f23eb498fa487949f..8152625f5b2ef4d5cf644e811f739380e3c7e283 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.cc
>@@ -10,8 +10,10 @@
> 
> #include "common_video/video_render_frames.h"
> 
>+#include <type_traits>
> #include <utility>
> 
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/metrics.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.h
>index 2226577dc3ef1ea5d5d211949514cc957f7ff0fc..2c4bdd9177dcd025f6508e2dd911c44ae5057c42 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.h
>@@ -11,8 +11,8 @@
> #ifndef COMMON_VIDEO_VIDEO_RENDER_FRAMES_H_
> #define COMMON_VIDEO_VIDEO_RENDER_FRAMES_H_
> 
>+#include <stddef.h>
> #include <stdint.h>
>-
> #include <list>
> 
> #include "absl/types/optional.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/rtpplay.exe b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/rtpplay.exe
>new file mode 100644
>index 0000000000000000000000000000000000000000..6f938c8fb220f73406e23295817e71c5fde58d4f
>GIT binary patch
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>zcr%CvgVcnvS}dGJu!gJ&iv-^av2qK#lz#|V7jnP-;VMEkR83SQXb1v-`lQ;Zw}St^
>zTkqG0^-*1<PwKP!yuPR}gU@(F7wZyzAAIZ}E7%GFPrH^CZbeuPttM8a6=fmI1fOyv
>z*-Cbh-DE#GOpcNwa*~_{O}$7ilN+R%l#u(xpg}YkW<CvR6B<dQC<0GeQk%8|#)+c|
>zG>ImIKhcx+p{d|8rqfKYJtOHDkgCaaD)<I@bT*w!=hKCBDg6*;PK9(oJxq&$pU%?r
>U)W|HGkE0-vln?~y{o&_70meto0{{R3
>
>literal 0
>HcmV?d00001
>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/BUILD.gn
>index 43d1dfe9f71298d395ea2d976318f251cfb2e077..4ed07af9830a5293819af7f7d0c4fb28caefa102 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/BUILD.gn
>@@ -60,6 +60,32 @@ group("examples") {
>   }
> }
> 
>+rtc_source_set("read_auth_file") {
>+  testonly = true
>+  sources = [
>+    "turnserver/read_auth_file.cc",
>+    "turnserver/read_auth_file.h",
>+  ]
>+  deps = [
>+    "../rtc_base:rtc_base",
>+  ]
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_test("examples_unittests") {
>+    testonly = true
>+    sources = [
>+      "turnserver/read_auth_file_unittest.cc",
>+    ]
>+    deps = [
>+      ":read_auth_file",
>+      "../test:test_main",
>+      "//test:test_support",
>+      "//testing/gtest",
>+    ]
>+  }
>+}
>+
> if (is_android) {
>   rtc_android_apk("AppRTCMobile") {
>     testonly = true
>@@ -166,15 +192,12 @@ if (is_android) {
>       "../build/android/adb_reverse_forwarder.py",
>       "../examples/androidtests/video_quality_loopback_test.py",
>       "../resources/reference_video_640x360_30fps.y4m",
>-      "../rtc_tools/barcode_tools/barcode_decoder.py",
>-      "../rtc_tools/barcode_tools/helper_functions.py",
>       "../rtc_tools/compare_videos.py",
>       "../rtc_tools/testing/prebuilt_apprtc.zip",
>       "../rtc_tools/testing/golang/linux/go.tar.gz",
>       "../rtc_tools/testing/build_apprtc.py",
>       "../rtc_tools/testing/utils.py",
>       "../tools_webrtc/video_quality_toolchain/linux/ffmpeg",
>-      "../tools_webrtc/video_quality_toolchain/linux/zxing",
>       "${root_out_dir}/frame_analyzer_host",
>     ]
>   }
>@@ -480,8 +503,10 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>         "../api:libjingle_peerconnection_api",
>         "../api/audio_codecs:builtin_audio_decoder_factory",
>         "../api/audio_codecs:builtin_audio_encoder_factory",
>+        "../api/video:builtin_video_bitrate_allocator_factory",
>         "../logging:rtc_event_log_impl_base",
>         "../media:rtc_audio_video",
>+        "../modules/audio_processing:api",
>         "../modules/audio_processing:audio_processing",
>         "../pc:libjingle_peerconnection",
>         "../rtc_base:rtc_base",
>@@ -625,12 +650,13 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>       rtc_ios_xctest_test("apprtcmobile_tests") {
>         info_plist = "objc/AppRTCMobile/ios/Info.plist"
>         sources = [
>-          "objc/AppRTCMobile/ios/main.m",
>+          "objc/AppRTCMobile/tests/main.mm",
>         ]
>         deps = [
>           ":AppRTCMobile_lib",
>           ":apprtcmobile_test_sources",
>           "../sdk:framework_objc",
>+          "//test:test_support",
>         ]
>         ldflags = [ "-all_load" ]
>       }
>@@ -655,11 +681,14 @@ if (is_linux || is_win) {
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>     }
>     deps = [
>+      "../api:create_peerconnection_factory",
>       "../api:libjingle_peerconnection_api",
>       "../api/video:video_frame_i420",
>       "../rtc_base:checks",
>       "../rtc_base:stringutils",
>       "../rtc_base/third_party/sigslot",
>+      "../system_wrappers:field_trial",
>+      "../test:field_trial",
>     ]
>     if (is_win) {
>       sources += [
>@@ -696,6 +725,7 @@ if (is_linux || is_win) {
>       "../api/video_codecs:builtin_video_encoder_factory",
>       "../media:rtc_audio_video",
>       "../modules/audio_device:audio_device",
>+      "../modules/audio_processing:api",
>       "../modules/audio_processing:audio_processing",
>       "../modules/video_capture:video_capture_module",
>       "../pc:libjingle_peerconnection",
>@@ -718,10 +748,11 @@ if (is_linux || is_win) {
>       "peerconnection/server/utils.h",
>     ]
>     deps = [
>-      "..:webrtc_common",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:stringutils",
>       "../rtc_tools:command_line_parser",
>+      "../system_wrappers:field_trial",
>+      "../test:field_trial",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -750,6 +781,7 @@ if (is_linux || is_win) {
>       "turnserver/turnserver_main.cc",
>     ]
>     deps = [
>+      ":read_auth_file",
>       "../p2p:rtc_p2p",
>       "../pc:rtc_pc",
>       "../rtc_base:rtc_base",
>@@ -807,6 +839,7 @@ if (is_win || is_android) {
>       configs += [ "//build/config/win:windowed" ]
>     }
>     deps = [
>+      "../api:create_peerconnection_factory",
>       "../api:libjingle_peerconnection_api",
>       "../api/audio_codecs:builtin_audio_decoder_factory",
>       "../api/audio_codecs:builtin_audio_encoder_factory",
>@@ -816,6 +849,7 @@ if (is_win || is_android) {
>       "../media:rtc_media",
>       "../media:rtc_media_base",
>       "../modules/audio_device:audio_device",
>+      "../modules/audio_processing:api",
>       "../modules/audio_processing:audio_processing",
>       "../modules/video_capture:video_capture_module",
>       "../pc:libjingle_peerconnection",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/DEPS
>index 6a9a9f09cffef37ffc32156a3bc5bdc9b63aa2a8..5b378455900b537142e9a81ea769a6a1f27e7bba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/DEPS
>@@ -8,5 +8,6 @@ include_rules = [
>   "+p2p",
>   "+pc",
>   "+sdk/objc",
>+  "+system_wrappers/include",
>   "+third_party/libyuv",
> ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/OWNERS
>new file mode 100644
>index 0000000000000000000000000000000000000000..3c4e54174e60f6590dcca98a9806777cc1f9c118
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/OWNERS
>@@ -0,0 +1 @@
>+sakal@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/AndroidManifest.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/AndroidManifest.xml
>index d6e0ff0417444db9d19c20f27c60cd8f46f9fe10..576449bb14e9edefe405dcb47ae5a123c419661d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/AndroidManifest.xml
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/AndroidManifest.xml
>@@ -18,7 +18,8 @@
>     <uses-permission android:name="android.permission.INTERNET" />
>     <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
>     <uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" />
>-    <uses-permission android:name="android.permission.CAPTURE_VIDEO_OUTPUT" />
>+    <uses-permission tools:ignore="ProtectedPermissions"
>+                     android:name="android.permission.CAPTURE_VIDEO_OUTPUT" />
> 
>     
>     <application android:label="@string/app_name"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/layout/activity_connect.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/layout/activity_connect.xml
>index 83707f72ed07f7cbd71445ea4ce4c378426bd963..3c591896444a825d9c0dd88a275a03e47a778151 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/layout/activity_connect.xml
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/layout/activity_connect.xml
>@@ -1,6 +1,7 @@
> <?xml version="1.0" encoding="utf-8"?>
> <LinearLayout
>     xmlns:android="http://schemas.android.com/apk/res/android"
>+    xmlns:tools="http://schemas.android.com/tools"
>     android:layout_margin="16dp"
>     android:layout_width="match_parent"
>     android:layout_height="match_parent"
>@@ -21,7 +22,9 @@
>         android:gravity="center"
>         android:layout_marginBottom="8dp">
> 
>+        
>         <EditText
>+            tools:ignore="LabelFor"
>             android:id="@+id/room_edittext"
>             android:layout_width="0dp"
>             android:layout_height="wrap_content"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java
>index ccc2c3bd93d8070e4363671e2734785074bb9eac..43961506ab93b32d5eb37c72c7356aa6facbb259 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java
>@@ -171,7 +171,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>   private RoomConnectionParameters roomConnectionParameters;
>   @Nullable
>   private PeerConnectionParameters peerConnectionParameters;
>-  private boolean iceConnected;
>+  private boolean connected;
>   private boolean isError;
>   private boolean callControlFragmentVisible = true;
>   private long callStartedTimeMs;
>@@ -203,7 +203,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>     getWindow().getDecorView().setSystemUiVisibility(getSystemUiVisibility());
>     setContentView(R.layout.activity_call);
> 
>-    iceConnected = false;
>+    connected = false;
>     signalingParameters = null;
> 
>     // Create UI controls.
>@@ -553,7 +553,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
> 
>   // Helper functions.
>   private void toggleCallControlFragmentVisibility() {
>-    if (!iceConnected || !callFragment.isAdded()) {
>+    if (!connected || !callFragment.isAdded()) {
>       return;
>     }
>     // Show/hide call control fragment
>@@ -649,7 +649,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>       audioManager.stop();
>       audioManager = null;
>     }
>-    if (iceConnected && !isError) {
>+    if (connected && !isError) {
>       setResult(RESULT_OK);
>     } else {
>       setResult(RESULT_CANCELED);
>@@ -911,8 +911,6 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>       @Override
>       public void run() {
>         logAndToast("ICE connected, delay=" + delta + "ms");
>-        iceConnected = true;
>-        callConnected();
>       }
>     });
>   }
>@@ -923,7 +921,30 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>       @Override
>       public void run() {
>         logAndToast("ICE disconnected");
>-        iceConnected = false;
>+      }
>+    });
>+  }
>+
>+  @Override
>+  public void onConnected() {
>+    final long delta = System.currentTimeMillis() - callStartedTimeMs;
>+    runOnUiThread(new Runnable() {
>+      @Override
>+      public void run() {
>+        logAndToast("DTLS connected, delay=" + delta + "ms");
>+        connected = true;
>+        callConnected();
>+      }
>+    });
>+  }
>+
>+  @Override
>+  public void onDisconnected() {
>+    runOnUiThread(new Runnable() {
>+      @Override
>+      public void run() {
>+        logAndToast("DTLS disconnected");
>+        connected = false;
>         disconnect();
>       }
>     });
>@@ -937,7 +958,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>     runOnUiThread(new Runnable() {
>       @Override
>       public void run() {
>-        if (!isError && iceConnected) {
>+        if (!isError && connected) {
>           hudFragment.updateEncoderStatistics(reports);
>         }
>       }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
>index 94135ded4d648713ce54a52ef39ed92aff709122..f85a788f3d16c6709549d4c8c3efe98f9c8ff2e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
>@@ -52,6 +52,7 @@ import org.webrtc.MediaStreamTrack;
> import org.webrtc.MediaStreamTrack.MediaType;
> import org.webrtc.PeerConnection;
> import org.webrtc.PeerConnection.IceConnectionState;
>+import org.webrtc.PeerConnection.PeerConnectionState;
> import org.webrtc.PeerConnectionFactory;
> import org.webrtc.RtpParameters;
> import org.webrtc.RtpReceiver;
>@@ -293,11 +294,23 @@ public class PeerConnectionClient {
>     void onIceConnected();
> 
>     /**
>-     * Callback fired once connection is closed (IceConnectionState is
>+     * Callback fired once connection is disconnected (IceConnectionState is
>      * DISCONNECTED).
>      */
>     void onIceDisconnected();
> 
>+    /**
>+     * Callback fired once DTLS connection is established (PeerConnectionState
>+     * is CONNECTED).
>+     */
>+    void onConnected();
>+
>+    /**
>+     * Callback fired once DTLS connection is disconnected (PeerConnectionState
>+     * is DISCONNECTED).
>+     */
>+    void onDisconnected();
>+
>     /**
>      * Callback fired once peer connection is closed.
>      */
>@@ -1263,6 +1276,20 @@ public class PeerConnectionClient {
>       });
>     }
> 
>+    @Override
>+    public void onConnectionChange(final PeerConnection.PeerConnectionState newState) {
>+      executor.execute(() -> {
>+        Log.d(TAG, "PeerConnectionState: " + newState);
>+        if (newState == PeerConnectionState.CONNECTED) {
>+          events.onConnected();
>+        } else if (newState == PeerConnectionState.DISCONNECTED) {
>+          events.onDisconnected();
>+        } else if (newState == PeerConnectionState.FAILED) {
>+          reportError("DTLS connection failed.");
>+        }
>+      });
>+    }
>+
>     @Override
>     public void onIceGatheringChange(PeerConnection.IceGatheringState newState) {
>       Log.d(TAG, "IceGatheringState: " + newState);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/OWNERS
>new file mode 100644
>index 0000000000000000000000000000000000000000..3c4e54174e60f6590dcca98a9806777cc1f9c118
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/OWNERS
>@@ -0,0 +1 @@
>+sakal@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java
>index 8c5f38ccb31b16afe5ba3ee8b28447f7f6f7953f..a8806ec5f7d67c16b09fc37bd61cd4d3d3a9ccd3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java
>@@ -38,10 +38,10 @@ public class TCPChannelClientTest {
>    * How long we wait before trying to connect to the server. Note: was
>    * previously only 10, which was too short (tests were flaky).
>    */
>-  private static final int SERVER_WAIT = 100;
>-  private static final int CONNECT_TIMEOUT = 100;
>-  private static final int SEND_TIMEOUT = 100;
>-  private static final int DISCONNECT_TIMEOUT = 100;
>+  private static final int SERVER_WAIT = 300;
>+  private static final int CONNECT_TIMEOUT = 1000;
>+  private static final int SEND_TIMEOUT = 1000;
>+  private static final int DISCONNECT_TIMEOUT = 1000;
>   private static final int TERMINATION_TIMEOUT = 1000;
>   private static final String TEST_MESSAGE_SERVER = "Hello, Server!";
>   private static final String TEST_MESSAGE_CLIENT = "Hello, Client!";
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/BUILD.gn
>index 5568eb11dde3ebf3b951b02a62aa6855132865e1..7928640cc6f5caaeaea9c6a7156cd0d6613f9c74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/BUILD.gn
>@@ -49,10 +49,12 @@ if (is_android) {
>       "//api:libjingle_peerconnection_api",
>       "//api/audio_codecs:builtin_audio_decoder_factory",
>       "//api/audio_codecs:builtin_audio_encoder_factory",
>+      "//api/video:builtin_video_bitrate_allocator_factory",
>       "//logging:rtc_event_log_impl_base",
>       "//media:rtc_audio_video",
>       "//media:rtc_internal_video_codecs",
>       "//modules/audio_processing",
>+      "//modules/audio_processing:api",
>       "//modules/utility:utility",
>       "//pc:libjingle_peerconnection",
>       "//rtc_base:rtc_base",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/androidcallclient.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/androidcallclient.cc
>index 005f369a6d104622dddb0bdbef8071a812f21233..a1394cd01c523dc6904830f3ac9af9876c49d5ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/androidcallclient.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/androidcallclient.cc
>@@ -16,6 +16,7 @@
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/peerconnectioninterface.h"
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "examples/androidnativeapi/generated_jni/jni/CallClient_jni.h"
> #include "media/engine/internaldecoderfactory.h"
> #include "media/engine/internalencoderfactory.h"
>@@ -99,7 +100,8 @@ void AndroidCallClient::Call(JNIEnv* env,
>   remote_sink_ = webrtc::JavaToNativeVideoSink(env, remote_sink.obj());
> 
>   video_source_ = webrtc::CreateJavaVideoSource(env, signaling_thread_.get(),
>-                                                false /* is_screencast */);
>+                                                /* is_screencast= */ false,
>+                                                /* align_timestamps= */ true);
> 
>   CreatePeerConnection();
>   Connect();
>@@ -159,6 +161,7 @@ void AndroidCallClient::CreatePeerConnectionFactory() {
>           webrtc::CreateBuiltinAudioDecoderFactory(),
>           absl::make_unique<webrtc::InternalEncoderFactory>(),
>           absl::make_unique<webrtc::InternalDecoderFactory>(),
>+          webrtc::CreateBuiltinVideoBitrateAllocatorFactory(),
>           nullptr /* audio_mixer */, webrtc::AudioProcessingBuilder().Create());
>   RTC_LOG(LS_INFO) << "Media engine created: " << media_engine.get();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/AndroidManifest.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/AndroidManifest.xml
>index 7cd52bc531d74b9e861f0b5c3062c95a3a4564ae..dae2e980a6a60a7f34feab8f93f771f4fdc1e904 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/AndroidManifest.xml
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/AndroidManifest.xml
>@@ -14,7 +14,7 @@
>       package="org.appspot.apprtc.test">
> 
>     <uses-permission android:name="android.permission.RUN_INSTRUMENTATION" />
>-    <uses-sdk android:minSdkVersion="13" android:targetSdkVersion="21" />
>+    <uses-sdk android:minSdkVersion="16" android:targetSdkVersion="21" />
> 
>     <application>
>         <uses-library android:name="android.test.runner" />
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
>index e2674017e456e059761baf502fa17905269a010b..c165f59b033d53fdd092af733a180cc7da465efc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
>@@ -183,6 +183,16 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>     }
>   }
> 
>+  @Override
>+  public void onConnected() {
>+    Log.d(TAG, "DTLS Connected");
>+  }
>+
>+  @Override
>+  public void onDisconnected() {
>+    Log.d(TAG, "DTLS Disconnected");
>+  }
>+
>   @Override
>   public void onPeerConnectionClosed() {
>     Log.d(TAG, "PeerConnection closed");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/video_quality_loopback_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/video_quality_loopback_test.py
>index 19dd37c3d36bfc84ecc9824aaa6607f73c21edd3..fc198a826f7962444b6485868b4b20a3d5d6e297 100755
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/video_quality_loopback_test.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/video_quality_loopback_test.py
>@@ -190,20 +190,13 @@ def RunTest(android_device, adb_path, build_dir, temp_dir, num_retries,
>   # Run comparison script.
>   compare_script = os.path.join(SRC_DIR, 'rtc_tools', 'compare_videos.py')
>   frame_analyzer = os.path.join(build_dir, 'frame_analyzer_host')
>-  zxing_path = os.path.join(TOOLCHAIN_DIR, 'zxing')
>-  stats_file_ref = os.path.join(temp_dir, 'stats_ref.txt')
>-  stats_file_test = os.path.join(temp_dir, 'stats_test.txt')
> 
>   args = [
>       '--ref_video', reference_video_yuv,
>       '--test_video', test_video_yuv,
>       '--yuv_frame_width', '640',
>       '--yuv_frame_height', '360',
>-      '--stats_file_ref', stats_file_ref,
>-      '--stats_file_test', stats_file_test,
>       '--frame_analyzer', frame_analyzer,
>-      '--ffmpeg_path', ffmpeg_path,
>-      '--zxing_path', zxing_path,
>   ]
>   if chartjson_result_file:
>     args.extend(['--chartjson_result_file', chartjson_result_file])
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.m
>index 9a0d3695f91615469c990d1122d1aeecdbff60e5..8b3d105797e2fff6861aba5dea3e10ee99ca111a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.m
>@@ -400,10 +400,15 @@ - (void)peerConnection:(RTCPeerConnection *)peerConnection
>     didChangeIceConnectionState:(RTCIceConnectionState)newState {
>   RTCLog(@"ICE state changed: %ld", (long)newState);
>   dispatch_async(dispatch_get_main_queue(), ^{
>-    [_delegate appClient:self didChangeConnectionState:newState];
>+    [self.delegate appClient:self didChangeConnectionState:newState];
>   });
> }
> 
>+- (void)peerConnection:(RTCPeerConnection *)peerConnection
>+    didChangeConnectionState:(RTCPeerConnectionState)newState {
>+  RTCLog(@"ICE+DTLS state changed: %ld", (long)newState);
>+}
>+
> - (void)peerConnection:(RTCPeerConnection *)peerConnection
>     didChangeIceGatheringState:(RTCIceGatheringState)newState {
>   RTCLog(@"ICE gathering state changed: %ld", (long)newState);
>@@ -450,16 +455,16 @@ - (void)peerConnection:(RTCPeerConnection *)peerConnection
>           [[NSError alloc] initWithDomain:kARDAppClientErrorDomain
>                                      code:kARDAppClientErrorCreateSDP
>                                  userInfo:userInfo];
>-      [_delegate appClient:self didError:sdpError];
>+      [self.delegate appClient:self didError:sdpError];
>       return;
>     }
>     __weak ARDAppClient *weakSelf = self;
>-    [_peerConnection setLocalDescription:sdp
>-                       completionHandler:^(NSError *error) {
>-                         ARDAppClient *strongSelf = weakSelf;
>-                         [strongSelf peerConnection:strongSelf.peerConnection
>-                             didSetSessionDescriptionWithError:error];
>-                       }];
>+    [self.peerConnection setLocalDescription:sdp
>+                           completionHandler:^(NSError *error) {
>+                             ARDAppClient *strongSelf = weakSelf;
>+                             [strongSelf peerConnection:strongSelf.peerConnection
>+                                 didSetSessionDescriptionWithError:error];
>+                           }];
>     ARDSessionDescriptionMessage *message =
>         [[ARDSessionDescriptionMessage alloc] initWithDescription:sdp];
>     [self sendSignalingMessage:message];
>@@ -480,22 +485,21 @@ - (void)peerConnection:(RTCPeerConnection *)peerConnection
>           [[NSError alloc] initWithDomain:kARDAppClientErrorDomain
>                                      code:kARDAppClientErrorSetSDP
>                                  userInfo:userInfo];
>-      [_delegate appClient:self didError:sdpError];
>+      [self.delegate appClient:self didError:sdpError];
>       return;
>     }
>     // If we're answering and we've just set the remote offer we need to create
>     // an answer and set the local description.
>-    if (!_isInitiator && !_peerConnection.localDescription) {
>+    if (!self.isInitiator && !self.peerConnection.localDescription) {
>       RTCMediaConstraints *constraints = [self defaultAnswerConstraints];
>       __weak ARDAppClient *weakSelf = self;
>-      [_peerConnection answerForConstraints:constraints
>-                          completionHandler:^(RTCSessionDescription *sdp,
>-                                              NSError *error) {
>-        ARDAppClient *strongSelf = weakSelf;
>-        [strongSelf peerConnection:strongSelf.peerConnection
>-            didCreateSessionDescription:sdp
>-                                  error:error];
>-      }];
>+      [self.peerConnection answerForConstraints:constraints
>+                              completionHandler:^(RTCSessionDescription *sdp, NSError *error) {
>+                                ARDAppClient *strongSelf = weakSelf;
>+                                [strongSelf peerConnection:strongSelf.peerConnection
>+                                    didCreateSessionDescription:sdp
>+                                                          error:error];
>+                              }];
>     }
>   });
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.m
>index cfe2ec5cd828569007f239b1debd7738ce1dbd9e..f9a1920399211bf7a63c0adfa77f321c7eda4393 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDStatsBuilder.m
>@@ -167,23 +167,43 @@ - (int)calculateAvgQP {
>   return deltaFramesEncoded != 0 ? deltaQPSum / deltaFramesEncoded : 0;
> }
> 
>+- (void)updateBweStatOfKey:(NSString *)key value:(NSString *)value {
>+  if ([key isEqualToString:@"googAvailableSendBandwidth"]) {
>+    _availableSendBw = [ARDBitrateTracker bitrateStringForBitrate:value.doubleValue];
>+  } else if ([key isEqualToString:@"googAvailableReceiveBandwidth"]) {
>+    _availableRecvBw = [ARDBitrateTracker bitrateStringForBitrate:value.doubleValue];
>+  } else if ([key isEqualToString:@"googActualEncBitrate"]) {
>+    _actualEncBitrate = [ARDBitrateTracker bitrateStringForBitrate:value.doubleValue];
>+  } else if ([key isEqualToString:@"googTargetEncBitrate"]) {
>+    _targetEncBitrate = [ARDBitrateTracker bitrateStringForBitrate:value.doubleValue];
>+  }
>+}
>+
> - (void)parseBweStatsReport:(RTCLegacyStatsReport *)statsReport {
>-  [statsReport.values enumerateKeysAndObjectsUsingBlock:^(
>-      NSString *key, NSString *value, BOOL *stop) {
>-    if ([key isEqualToString:@"googAvailableSendBandwidth"]) {
>-      _availableSendBw =
>-          [ARDBitrateTracker bitrateStringForBitrate:value.doubleValue];
>-    } else if ([key isEqualToString:@"googAvailableReceiveBandwidth"]) {
>-      _availableRecvBw =
>-          [ARDBitrateTracker bitrateStringForBitrate:value.doubleValue];
>-    } else if ([key isEqualToString:@"googActualEncBitrate"]) {
>-      _actualEncBitrate =
>-          [ARDBitrateTracker bitrateStringForBitrate:value.doubleValue];
>-    } else if ([key isEqualToString:@"googTargetEncBitrate"]) {
>-      _targetEncBitrate =
>-          [ARDBitrateTracker bitrateStringForBitrate:value.doubleValue];
>-    }
>-  }];
>+  [statsReport.values
>+      enumerateKeysAndObjectsUsingBlock:^(NSString *key, NSString *value, BOOL *stop) {
>+        [self updateBweStatOfKey:key value:value];
>+      }];
>+}
>+
>+- (void)updateConnectionStatOfKey:(NSString *)key value:(NSString *)value {
>+  if ([key isEqualToString:@"googRtt"]) {
>+    _connRtt = value;
>+  } else if ([key isEqualToString:@"googLocalCandidateType"]) {
>+    _localCandType = value;
>+  } else if ([key isEqualToString:@"googRemoteCandidateType"]) {
>+    _remoteCandType = value;
>+  } else if ([key isEqualToString:@"googTransportType"]) {
>+    _transportType = value;
>+  } else if ([key isEqualToString:@"bytesReceived"]) {
>+    NSInteger byteCount = value.integerValue;
>+    [_connRecvBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>+    _connRecvBitrate = _connRecvBitrateTracker.bitrateString;
>+  } else if ([key isEqualToString:@"bytesSent"]) {
>+    NSInteger byteCount = value.integerValue;
>+    [_connSendBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>+    _connSendBitrate = _connSendBitrateTracker.bitrateString;
>+  }
> }
> 
> - (void)parseConnectionStatsReport:(RTCLegacyStatsReport *)statsReport {
>@@ -191,26 +211,10 @@ - (void)parseConnectionStatsReport:(RTCLegacyStatsReport *)statsReport {
>   if (![activeConnection isEqualToString:@"true"]) {
>     return;
>   }
>-  [statsReport.values enumerateKeysAndObjectsUsingBlock:^(
>-      NSString *key, NSString *value, BOOL *stop) {
>-    if ([key isEqualToString:@"googRtt"]) {
>-      _connRtt = value;
>-    } else if ([key isEqualToString:@"googLocalCandidateType"]) {
>-      _localCandType = value;
>-    } else if ([key isEqualToString:@"googRemoteCandidateType"]) {
>-      _remoteCandType = value;
>-    } else if ([key isEqualToString:@"googTransportType"]) {
>-      _transportType = value;
>-    } else if ([key isEqualToString:@"bytesReceived"]) {
>-      NSInteger byteCount = value.integerValue;
>-      [_connRecvBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>-      _connRecvBitrate = _connRecvBitrateTracker.bitrateString;
>-    } else if ([key isEqualToString:@"bytesSent"]) {
>-      NSInteger byteCount = value.integerValue;
>-      [_connSendBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>-      _connSendBitrate = _connSendBitrateTracker.bitrateString;
>-    }
>-  }];
>+  [statsReport.values
>+      enumerateKeysAndObjectsUsingBlock:^(NSString *key, NSString *value, BOOL *stop) {
>+        [self updateConnectionStatOfKey:key value:value];
>+      }];
> }
> 
> - (void)parseSendSsrcStatsReport:(RTCLegacyStatsReport *)statsReport {
>@@ -224,50 +228,58 @@ - (void)parseSendSsrcStatsReport:(RTCLegacyStatsReport *)statsReport {
>   }
> }
> 
>+- (void)updateAudioSendStatOfKey:(NSString *)key value:(NSString *)value {
>+  if ([key isEqualToString:@"googCodecName"]) {
>+    _audioSendCodec = value;
>+  } else if ([key isEqualToString:@"bytesSent"]) {
>+    NSInteger byteCount = value.integerValue;
>+    [_audioSendBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>+    _audioSendBitrate = _audioSendBitrateTracker.bitrateString;
>+  }
>+}
>+
> - (void)parseAudioSendStatsReport:(RTCLegacyStatsReport *)statsReport {
>-  [statsReport.values enumerateKeysAndObjectsUsingBlock:^(
>-      NSString *key, NSString *value, BOOL *stop) {
>-    if ([key isEqualToString:@"googCodecName"]) {
>-      _audioSendCodec = value;
>-    } else if ([key isEqualToString:@"bytesSent"]) {
>-      NSInteger byteCount = value.integerValue;
>-      [_audioSendBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>-      _audioSendBitrate = _audioSendBitrateTracker.bitrateString;
>-    }
>-  }];
>+  [statsReport.values
>+      enumerateKeysAndObjectsUsingBlock:^(NSString *key, NSString *value, BOOL *stop) {
>+        [self updateAudioSendStatOfKey:key value:value];
>+      }];
>+}
>+
>+- (void)updateVideoSendStatOfKey:(NSString *)key value:(NSString *)value {
>+  if ([key isEqualToString:@"googCodecName"]) {
>+    _videoSendCodec = value;
>+  } else if ([key isEqualToString:@"googFrameHeightInput"]) {
>+    _videoInputHeight = value;
>+  } else if ([key isEqualToString:@"googFrameWidthInput"]) {
>+    _videoInputWidth = value;
>+  } else if ([key isEqualToString:@"googFrameRateInput"]) {
>+    _videoInputFps = value;
>+  } else if ([key isEqualToString:@"googFrameHeightSent"]) {
>+    _videoSendHeight = value;
>+  } else if ([key isEqualToString:@"googFrameWidthSent"]) {
>+    _videoSendWidth = value;
>+  } else if ([key isEqualToString:@"googFrameRateSent"]) {
>+    _videoSendFps = value;
>+  } else if ([key isEqualToString:@"googAvgEncodeMs"]) {
>+    _videoEncodeMs = value;
>+  } else if ([key isEqualToString:@"bytesSent"]) {
>+    NSInteger byteCount = value.integerValue;
>+    [_videoSendBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>+    _videoSendBitrate = _videoSendBitrateTracker.bitrateString;
>+  } else if ([key isEqualToString:@"qpSum"]) {
>+    _oldVideoQPSum = _videoQPSum;
>+    _videoQPSum = value.integerValue;
>+  } else if ([key isEqualToString:@"framesEncoded"]) {
>+    _oldFramesEncoded = _framesEncoded;
>+    _framesEncoded = value.integerValue;
>+  }
> }
> 
> - (void)parseVideoSendStatsReport:(RTCLegacyStatsReport *)statsReport {
>-  [statsReport.values enumerateKeysAndObjectsUsingBlock:^(
>-      NSString *key, NSString *value, BOOL *stop) {
>-    if ([key isEqualToString:@"googCodecName"]) {
>-      _videoSendCodec = value;
>-    } else if ([key isEqualToString:@"googFrameHeightInput"]) {
>-      _videoInputHeight = value;
>-    } else if ([key isEqualToString:@"googFrameWidthInput"]) {
>-      _videoInputWidth = value;
>-    } else if ([key isEqualToString:@"googFrameRateInput"]) {
>-      _videoInputFps = value;
>-    } else if ([key isEqualToString:@"googFrameHeightSent"]) {
>-      _videoSendHeight = value;
>-    } else if ([key isEqualToString:@"googFrameWidthSent"]) {
>-      _videoSendWidth = value;
>-    } else if ([key isEqualToString:@"googFrameRateSent"]) {
>-      _videoSendFps = value;
>-    } else if ([key isEqualToString:@"googAvgEncodeMs"]) {
>-      _videoEncodeMs = value;
>-    } else if ([key isEqualToString:@"bytesSent"]) {
>-      NSInteger byteCount = value.integerValue;
>-      [_videoSendBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>-      _videoSendBitrate = _videoSendBitrateTracker.bitrateString;
>-    } else if ([key isEqualToString:@"qpSum"]) {
>-      _oldVideoQPSum = _videoQPSum;
>-      _videoQPSum = value.integerValue;
>-    } else if ([key isEqualToString:@"framesEncoded"]) {
>-      _oldFramesEncoded = _framesEncoded;
>-      _framesEncoded = value.integerValue;
>-    }
>-  }];
>+  [statsReport.values
>+      enumerateKeysAndObjectsUsingBlock:^(NSString *key, NSString *value, BOOL *stop) {
>+        [self updateVideoSendStatOfKey:key value:value];
>+      }];
> }
> 
> - (void)parseRecvSsrcStatsReport:(RTCLegacyStatsReport *)statsReport {
>@@ -281,44 +293,52 @@ - (void)parseRecvSsrcStatsReport:(RTCLegacyStatsReport *)statsReport {
>   }
> }
> 
>+- (void)updateAudioRecvStatOfKey:(NSString *)key value:(NSString *)value {
>+  if ([key isEqualToString:@"googCodecName"]) {
>+    _audioRecvCodec = value;
>+  } else if ([key isEqualToString:@"bytesReceived"]) {
>+    NSInteger byteCount = value.integerValue;
>+    [_audioRecvBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>+    _audioRecvBitrate = _audioRecvBitrateTracker.bitrateString;
>+  } else if ([key isEqualToString:@"googSpeechExpandRate"]) {
>+    _audioExpandRate = value;
>+  } else if ([key isEqualToString:@"googCurrentDelayMs"]) {
>+    _audioCurrentDelay = value;
>+  }
>+}
>+
> - (void)parseAudioRecvStatsReport:(RTCLegacyStatsReport *)statsReport {
>-  [statsReport.values enumerateKeysAndObjectsUsingBlock:^(
>-      NSString *key, NSString *value, BOOL *stop) {
>-    if ([key isEqualToString:@"googCodecName"]) {
>-      _audioRecvCodec = value;
>-    } else if ([key isEqualToString:@"bytesReceived"]) {
>-      NSInteger byteCount = value.integerValue;
>-      [_audioRecvBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>-      _audioRecvBitrate = _audioRecvBitrateTracker.bitrateString;
>-    } else if ([key isEqualToString:@"googSpeechExpandRate"]) {
>-      _audioExpandRate = value;
>-    } else if ([key isEqualToString:@"googCurrentDelayMs"]) {
>-      _audioCurrentDelay = value;
>-    }
>-  }];
>+  [statsReport.values
>+      enumerateKeysAndObjectsUsingBlock:^(NSString *key, NSString *value, BOOL *stop) {
>+        [self updateAudioRecvStatOfKey:key value:value];
>+      }];
>+}
>+
>+- (void)updateVideoRecvStatOfKey:(NSString *)key value:(NSString *)value {
>+  if ([key isEqualToString:@"googFrameHeightReceived"]) {
>+    _videoRecvHeight = value;
>+  } else if ([key isEqualToString:@"googFrameWidthReceived"]) {
>+    _videoRecvWidth = value;
>+  } else if ([key isEqualToString:@"googFrameRateReceived"]) {
>+    _videoRecvFps = value;
>+  } else if ([key isEqualToString:@"googFrameRateDecoded"]) {
>+    _videoDecodedFps = value;
>+  } else if ([key isEqualToString:@"googFrameRateOutput"]) {
>+    _videoOutputFps = value;
>+  } else if ([key isEqualToString:@"googDecodeMs"]) {
>+    _videoDecodeMs = value;
>+  } else if ([key isEqualToString:@"bytesReceived"]) {
>+    NSInteger byteCount = value.integerValue;
>+    [_videoRecvBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>+    _videoRecvBitrate = _videoRecvBitrateTracker.bitrateString;
>+  }
> }
> 
> - (void)parseVideoRecvStatsReport:(RTCLegacyStatsReport *)statsReport {
>-  [statsReport.values enumerateKeysAndObjectsUsingBlock:^(
>-      NSString *key, NSString *value, BOOL *stop) {
>-    if ([key isEqualToString:@"googFrameHeightReceived"]) {
>-      _videoRecvHeight = value;
>-    } else if ([key isEqualToString:@"googFrameWidthReceived"]) {
>-      _videoRecvWidth = value;
>-    } else if ([key isEqualToString:@"googFrameRateReceived"]) {
>-      _videoRecvFps = value;
>-    } else if ([key isEqualToString:@"googFrameRateDecoded"]) {
>-      _videoDecodedFps = value;
>-    } else if ([key isEqualToString:@"googFrameRateOutput"]) {
>-      _videoOutputFps = value;
>-    } else if ([key isEqualToString:@"googDecodeMs"]) {
>-      _videoDecodeMs = value;
>-    } else if ([key isEqualToString:@"bytesReceived"]) {
>-      NSInteger byteCount = value.integerValue;
>-      [_videoRecvBitrateTracker updateBitrateWithCurrentByteCount:byteCount];
>-      _videoRecvBitrate = _videoRecvBitrateTracker.bitrateString;
>-    }
>-  }];
>+  [statsReport.values
>+      enumerateKeysAndObjectsUsingBlock:^(NSString *key, NSString *value, BOOL *stop) {
>+        [self updateVideoRecvStatOfKey:key value:value];
>+      }];
> }
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m
>index fae3f7b8aa7267c6db45bef262ed9b3560c6b20d..63b3dd76cad025d9bc985110adc523314428ae07 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDMainViewController.m
>@@ -32,14 +32,17 @@ @interface ARDMainViewController () <
>     ARDMainViewDelegate,
>     ARDVideoCallViewControllerDelegate,
>     RTCAudioSessionDelegate>
>+@property(nonatomic, strong) ARDMainView *mainView;
>+@property(nonatomic, strong) AVAudioPlayer *audioPlayer;
> @end
> 
> @implementation ARDMainViewController {
>-  ARDMainView *_mainView;
>-  AVAudioPlayer *_audioPlayer;
>   BOOL _useManualAudio;
> }
> 
>+@synthesize mainView = _mainView;
>+@synthesize audioPlayer = _audioPlayer;
>+
> - (void)viewDidLoad {
>   [super viewDidLoad];
>   if ([[[NSProcessInfo processInfo] arguments] containsObject:loopbackLaunchProcessArgument]) {
>@@ -165,13 +168,13 @@ - (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session {
>   // Stop playback on main queue and then configure WebRTC.
>   [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeMain
>                                block:^{
>-    if (_mainView.isAudioLoopPlaying) {
>-      RTCLog(@"Stopping audio loop due to WebRTC start.");
>-      [_audioPlayer stop];
>-    }
>-    RTCLog(@"Setting isAudioEnabled to YES.");
>-    session.isAudioEnabled = YES;
>-  }];
>+                                 if (self.mainView.isAudioLoopPlaying) {
>+                                   RTCLog(@"Stopping audio loop due to WebRTC start.");
>+                                   [self.audioPlayer stop];
>+                                 }
>+                                 RTCLog(@"Setting isAudioEnabled to YES.");
>+                                 session.isAudioEnabled = YES;
>+                               }];
> }
> 
> - (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
>index 2827bf94423cb023e45ea984f5e77687ae764a0a..f7f4a877ef0aa1ed131a71229b4aff207dbacc2a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
>@@ -27,6 +27,7 @@ @interface ARDVideoCallViewController () <ARDAppClientDelegate,
>                                           RTCAudioSessionDelegate>
> @property(nonatomic, strong) RTCVideoTrack *remoteVideoTrack;
> @property(nonatomic, readonly) ARDVideoCallView *videoCallView;
>+@property(nonatomic, assign) AVAudioSessionPortOverride portOverride;
> @end
> 
> @implementation ARDVideoCallViewController {
>@@ -34,12 +35,12 @@ @implementation ARDVideoCallViewController {
>   RTCVideoTrack *_remoteVideoTrack;
>   ARDCaptureController *_captureController;
>   ARDFileCaptureController *_fileCaptureController NS_AVAILABLE_IOS(10);
>-  AVAudioSessionPortOverride _portOverride;
> }
> 
> @synthesize videoCallView = _videoCallView;
> @synthesize remoteVideoTrack = _remoteVideoTrack;
> @synthesize delegate = _delegate;
>+@synthesize portOverride = _portOverride;
> 
> - (instancetype)initForRoom:(NSString *)room
>                  isLoopback:(BOOL)isLoopback
>@@ -168,7 +169,7 @@ - (void)videoCallViewDidChangeRoute:(ARDVideoCallView *)view {
>     [session lockForConfiguration];
>     NSError *error = nil;
>     if ([session overrideOutputAudioPort:override error:&error]) {
>-      _portOverride = override;
>+      self.portOverride = override;
>     } else {
>       RTCLogError(@"Error overriding output port: %@",
>                   error.localizedDescription);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.m
>index 9f5ed40b1b41c28ea1cf1ed73af365d5461b7bc2..a972a20e4ea898964b1ebe0222a8d2c440bbbfad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/mac/APPRTCViewController.m
>@@ -39,6 +39,7 @@ @interface APPRTCMainView : NSView
> @property(nonatomic, weak) id<APPRTCMainViewDelegate> delegate;
> @property(nonatomic, readonly) NSView<RTCVideoRenderer>* localVideoView;
> @property(nonatomic, readonly) NSView<RTCVideoRenderer>* remoteVideoView;
>+@property(nonatomic, readonly) NSTextView* logView;
> 
> - (void)displayLogMessage:(NSString*)message;
> 
>@@ -52,7 +53,6 @@ @implementation APPRTCMainView  {
>   NSButton* _connectButton;
>   NSButton* _loopbackButton;
>   NSTextField* _roomField;
>-  NSTextView* _logView;
>   CGSize _localVideoSize;
>   CGSize _remoteVideoSize;
> }
>@@ -60,14 +60,13 @@ @implementation APPRTCMainView  {
> @synthesize delegate = _delegate;
> @synthesize localVideoView = _localVideoView;
> @synthesize remoteVideoView = _remoteVideoView;
>-
>+@synthesize logView = _logView;
> 
> - (void)displayLogMessage:(NSString *)message {
>   dispatch_async(dispatch_get_main_queue(), ^{
>-    _logView.string =
>-        [NSString stringWithFormat:@"%@%@\n", _logView.string, message];
>-    NSRange range = NSMakeRange(_logView.string.length, 0);
>-    [_logView scrollRangeToVisible:range];
>+    self.logView.string = [NSString stringWithFormat:@"%@%@\n", self.logView.string, message];
>+    NSRange range = NSMakeRange(self.logView.string.length, 0);
>+    [self.logView scrollRangeToVisible:range];
>   });
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/tests/main.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/tests/main.mm
>new file mode 100644
>index 0000000000000000000000000000000000000000..3625ffd7bf81ca83ee266bfa199fa9ef3048035d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/tests/main.mm
>@@ -0,0 +1,21 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <UIKit/UIKit.h>
>+
>+#include "test/ios/coverage_util_ios.h"
>+
>+int main(int argc, char* argv[]) {
>+  rtc::test::ConfigureCoverageReportPath();
>+
>+  @autoreleasepool {
>+    return UIApplicationMain(argc, argv, nil, nil);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/objccallclient.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/objccallclient.mm
>index c384da3ef8cbbb776dd05fc54055c2099db111cf..c7b2af42123e495d7e72ae66cecfd157e32381d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/objccallclient.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/objccallclient.mm
>@@ -21,6 +21,7 @@
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/peerconnectioninterface.h"
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "media/engine/webrtcmediaengine.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "sdk/objc/native/api/video_capturer.h"
>@@ -116,12 +117,16 @@ void ObjCCallClient::CreatePeerConnectionFactory() {
>   std::unique_ptr<webrtc::VideoEncoderFactory> videoEncoderFactory =
>       webrtc::ObjCToNativeVideoEncoderFactory([[RTCDefaultVideoEncoderFactory alloc] init]);
> 
>+  std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> videoBitrateAllocatorFactory =
>+      webrtc::CreateBuiltinVideoBitrateAllocatorFactory();
>+
>   std::unique_ptr<cricket::MediaEngineInterface> media_engine =
>       cricket::WebRtcMediaEngineFactory::Create(nullptr /* adm */,
>                                                 webrtc::CreateBuiltinAudioEncoderFactory(),
>                                                 webrtc::CreateBuiltinAudioDecoderFactory(),
>                                                 std::move(videoEncoderFactory),
>                                                 std::move(videoDecoderFactory),
>+                                                std::move(videoBitrateAllocatorFactory),
>                                                 nullptr /* audio_mixer */,
>                                                 webrtc::AudioProcessingBuilder().Create());
>   RTC_LOG(LS_INFO) << "Media engine created: " << media_engine.get();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.cc
>index c25259ce34df35ca4c384568f8ef9248fca7d02a..a781c4a1874aa32b0d9cf8307bd550aa0966888d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.cc
>@@ -16,6 +16,7 @@
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/create_peerconnection_factory.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "examples/peerconnection/client/defaults.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/flagdefs.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/flagdefs.h
>index 564e0e95f90c7c55fc9325f93fd02c7e9865524f..51f9c9a4255a4aae3faf480f655b0a775acf41e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/flagdefs.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/flagdefs.h
>@@ -19,20 +19,29 @@ extern const uint16_t kDefaultServerPort;  // From defaults.[h|cc]
> // header file so that they can be shared across the different main.cc's
> // for each platform.
> 
>-DEFINE_bool(help, false, "Prints this message");
>-DEFINE_bool(autoconnect,
>-            false,
>-            "Connect to the server without user "
>-            "intervention.");
>-DEFINE_string(server, "localhost", "The server to connect to.");
>-DEFINE_int(port,
>-           kDefaultServerPort,
>-           "The port on which the server is listening.");
>-DEFINE_bool(
>+WEBRTC_DEFINE_bool(help, false, "Prints this message");
>+WEBRTC_DEFINE_bool(autoconnect,
>+                   false,
>+                   "Connect to the server without user "
>+                   "intervention.");
>+WEBRTC_DEFINE_string(server, "localhost", "The server to connect to.");
>+WEBRTC_DEFINE_int(port,
>+                  kDefaultServerPort,
>+                  "The port on which the server is listening.");
>+WEBRTC_DEFINE_bool(
>     autocall,
>     false,
>     "Call the first available other client on "
>     "the server without user intervention.  Note: this flag should only be set "
>     "to true on one of the two clients.");
> 
>+WEBRTC_DEFINE_string(
>+    force_fieldtrials,
>+    "",
>+    "Field trials control experimental features. This flag specifies the field "
>+    "trials in effect. E.g. running with "
>+    "--force_fieldtrials=WebRTC-FooFeature/Enabled/ "
>+    "will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
>+    "trials are separated by \"/\"");
>+
> #endif  // EXAMPLES_PEERCONNECTION_CLIENT_FLAGDEFS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main.cc
>index 50179c4ba5510cf1ed7a3c2eda6fa00af7dc7b18..6c34683d2ea91d61051e78de09e7fdd2e9e38401 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main.cc
>@@ -17,6 +17,8 @@
> 
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/thread.h"
>+#include "system_wrappers/include/field_trial.h"
>+#include "test/field_trial.h"
> 
> class CustomSocketServer : public rtc::PhysicalSocketServer {
>  public:
>@@ -75,6 +77,11 @@ int main(int argc, char* argv[]) {
>     return 0;
>   }
> 
>+  webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
>+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
>+  // the application.
>+  webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
>+
>   // Abort if the user specifies a port that is outside the allowed
>   // range [1, 65535].
>   if ((FLAG_port < 1) || (FLAG_port > 65535)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.cc
>index 52b0d88523c3ca823dc98707412538db0df9aae5..2f7777d1eea1e6e38d1bb90fe9f5f6023862291c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.cc
>@@ -18,11 +18,8 @@
> #include "examples/peerconnection/client/defaults.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/stringutils.h"
> #include "third_party/libyuv/include/libyuv/convert_from.h"
> 
>-using rtc::sprintfn;
>-
> namespace {
> 
> //
>@@ -150,7 +147,7 @@ GtkMainWnd::GtkMainWnd(const char* server,
>       autoconnect_(autoconnect),
>       autocall_(autocall) {
>   char buffer[10];
>-  sprintfn(buffer, sizeof(buffer), "%i", port);
>+  snprintf(buffer, sizeof(buffer), "%i", port);
>   port_ = buffer;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main.cc
>index 25dde1ef6585d82578a79f45452bea4b6d616fef..f92e606c338119e22e43b11357ffb8bac230aa6c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main.cc
>@@ -16,6 +16,8 @@
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/win32socketinit.h"
> #include "rtc_base/win32socketserver.h"
>+#include "system_wrappers/include/field_trial.h"
>+#include "test/field_trial.h"
> 
> int PASCAL wWinMain(HINSTANCE instance,
>                     HINSTANCE prev_instance,
>@@ -36,6 +38,11 @@ int PASCAL wWinMain(HINSTANCE instance,
>     return 0;
>   }
> 
>+  webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
>+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
>+  // the application.
>+  webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
>+
>   // Abort if the user specifies a port that is outside the allowed
>   // range [1, 65535].
>   if ((FLAG_port < 1) || (FLAG_port > 65535)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.cc
>index 3ad248843a815b0a9005682e4ff01a3d04c9899c..8edd1e76f350314de0304035a193cd2101a79388 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.cc
>@@ -17,14 +17,11 @@
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/stringutils.h"
> #include "third_party/libyuv/include/libyuv/convert_argb.h"
> 
> ATOM MainWnd::wnd_class_ = 0;
> const wchar_t MainWnd::kClassName[] = L"WebRTC_MainWnd";
> 
>-using rtc::sprintfn;
>-
> namespace {
> 
> const char kConnecting[] = "Connecting... ";
>@@ -86,8 +83,8 @@ MainWnd::MainWnd(const char* server,
>       server_(server),
>       auto_connect_(auto_connect),
>       auto_call_(auto_call) {
>-  char buffer[10] = {0};
>-  sprintfn(buffer, sizeof(buffer), "%i", port);
>+  char buffer[10];
>+  snprintf(buffer, sizeof(buffer), "%i", port);
>   port_ = buffer;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.cc
>index bb79d167911198dc556139d9460bd731b37cc778..f173e42793a25aed8c772fec4c5721b1c86d828e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.cc
>@@ -14,14 +14,11 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/nethelpers.h"
>-#include "rtc_base/stringutils.h"
> 
> #ifdef WIN32
> #include "rtc_base/win32socketserver.h"
> #endif
> 
>-using rtc::sprintfn;
>-
> namespace {
> 
> // This is our magical hangup signal.
>@@ -136,7 +133,7 @@ void PeerConnectionClient::DoConnect() {
>   hanging_get_.reset(CreateClientSocket(server_address_.ipaddr().family()));
>   InitSocketSignals();
>   char buffer[1024];
>-  sprintfn(buffer, sizeof(buffer), "GET /sign_in?%s HTTP/1.0\r\n\r\n",
>+  snprintf(buffer, sizeof(buffer), "GET /sign_in?%s HTTP/1.0\r\n\r\n",
>            client_name_.c_str());
>   onconnect_data_ = buffer;
> 
>@@ -158,9 +155,9 @@ bool PeerConnectionClient::SendToPeer(int peer_id, const std::string& message) {
>     return false;
> 
>   char headers[1024];
>-  sprintfn(headers, sizeof(headers),
>+  snprintf(headers, sizeof(headers),
>            "POST /message?peer_id=%i&to=%i HTTP/1.0\r\n"
>-           "Content-Length: %i\r\n"
>+           "Content-Length: %zu\r\n"
>            "Content-Type: text/plain\r\n"
>            "\r\n",
>            my_id_, peer_id, message.length());
>@@ -190,7 +187,7 @@ bool PeerConnectionClient::SignOut() {
> 
>     if (my_id_ != -1) {
>       char buffer[1024];
>-      sprintfn(buffer, sizeof(buffer),
>+      snprintf(buffer, sizeof(buffer),
>                "GET /sign_out?peer_id=%i HTTP/1.0\r\n\r\n", my_id_);
>       onconnect_data_ = buffer;
>       return ConnectControlSocket();
>@@ -237,7 +234,7 @@ void PeerConnectionClient::OnConnect(rtc::AsyncSocket* socket) {
> 
> void PeerConnectionClient::OnHangingGetConnect(rtc::AsyncSocket* socket) {
>   char buffer[1024];
>-  sprintfn(buffer, sizeof(buffer), "GET /wait?peer_id=%i HTTP/1.0\r\n\r\n",
>+  snprintf(buffer, sizeof(buffer), "GET /wait?peer_id=%i HTTP/1.0\r\n\r\n",
>            my_id_);
>   int len = static_cast<int>(strlen(buffer));
>   int sent = socket->Send(buffer, len);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/main.cc
>index 075b6ea2d3f7a06f7186876b37155e2e5e4dd542..5214ca5ca551154cf92a1955fece100d1a417189 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/main.cc
>@@ -18,7 +18,19 @@
> #include "examples/peerconnection/server/data_socket.h"
> #include "examples/peerconnection/server/peer_channel.h"
> #include "examples/peerconnection/server/utils.h"
>+#include "rtc_base/flags.h"
> #include "rtc_tools/simple_command_line_parser.h"
>+#include "system_wrappers/include/field_trial.h"
>+#include "test/field_trial.h"
>+
>+WEBRTC_DEFINE_string(
>+    force_fieldtrials,
>+    "",
>+    "Field trials control experimental features. This flag specifies the field "
>+    "trials in effect. E.g. running with "
>+    "--force_fieldtrials=WebRTC-FooFeature/Enabled/ "
>+    "will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
>+    "trials are separated by \"/\"");
> 
> static const size_t kMaxConnections = (FD_SETSIZE - 2);
> 
>@@ -62,6 +74,11 @@ int main(int argc, char* argv[]) {
>     return 0;
>   }
> 
>+  webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
>+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
>+  // the application.
>+  webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
>+
>   int port = strtol((parser.GetFlag("port")).c_str(), NULL, 10);
> 
>   // Abort if the user specifies a port that is outside the allowed
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.cc
>index 6ef08d7b21d73b6f2299adb69ce72b2faf2ef850..b23b7e04e21d6bbc0b9ce5d22075e0735fdf17f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.cc
>@@ -19,9 +19,6 @@
> #include "examples/peerconnection/server/data_socket.h"
> #include "examples/peerconnection/server/utils.h"
> #include "rtc_base/stringencode.h"
>-#include "rtc_base/stringutils.h"
>-
>-using rtc::sprintfn;
> 
> // Set to the peer id of the originator when messages are being
> // exchanged between peers, but set to the id of the receiving peer
>@@ -98,7 +95,7 @@ std::string ChannelMember::GetEntry() const {
> 
>   // name, 11-digit int, 1-digit bool, newline, null
>   char entry[kMaxNameLength + 15];
>-  sprintfn(entry, sizeof(entry), "%s,%d,%d\n",
>+  snprintf(entry, sizeof(entry), "%s,%d,%d\n",
>            name_.substr(0, kMaxNameLength).c_str(), id_, connected_);
>   return entry;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunprober/main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunprober/main.cc
>index 45a76f523743e6fafa1be85f449badfa3c7f6855..6da96e469982f05e73baab2b348a2124f54a4707 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunprober/main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunprober/main.cc
>@@ -14,7 +14,7 @@
> 
> #include <map>
> #include <memory>
>-#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
>+#include <sstream>
> 
> #include "p2p/base/basicpacketsocketfactory.h"
> #include "p2p/stunprober/stunprober.h"
>@@ -33,16 +33,21 @@
> using stunprober::StunProber;
> using stunprober::AsyncCallback;
> 
>-DEFINE_bool(help, false, "Prints this message");
>-DEFINE_int(interval, 10, "Interval of consecutive stun pings in milliseconds");
>-DEFINE_bool(shared_socket, false, "Share socket mode for different remote IPs");
>-DEFINE_int(pings_per_ip,
>-           10,
>-           "Number of consecutive stun pings to send for each IP");
>-DEFINE_int(timeout,
>-           1000,
>-           "Milliseconds of wait after the last ping sent before exiting");
>-DEFINE_string(
>+WEBRTC_DEFINE_bool(help, false, "Prints this message");
>+WEBRTC_DEFINE_int(interval,
>+                  10,
>+                  "Interval of consecutive stun pings in milliseconds");
>+WEBRTC_DEFINE_bool(shared_socket,
>+                   false,
>+                   "Share socket mode for different remote IPs");
>+WEBRTC_DEFINE_int(pings_per_ip,
>+                  10,
>+                  "Number of consecutive stun pings to send for each IP");
>+WEBRTC_DEFINE_int(
>+    timeout,
>+    1000,
>+    "Milliseconds of wait after the last ping sent before exiting");
>+WEBRTC_DEFINE_string(
>     servers,
>     "stun.l.google.com:19302,stun1.l.google.com:19302,stun2.l.google.com:19302",
>     "Comma separated STUN server addresses with ports");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/read_auth_file.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/read_auth_file.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..de5465e82fe8339b01d527cf4642e03a06bf662a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/read_auth_file.cc
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "examples/turnserver/read_auth_file.h"
>+#include "rtc_base/stringencode.h"
>+
>+namespace webrtc_examples {
>+
>+std::map<std::string, std::string> ReadAuthFile(std::istream* s) {
>+  std::map<std::string, std::string> name_to_key;
>+  for (std::string line; std::getline(*s, line);) {
>+    const size_t sep = line.find('=');
>+    if (sep == std::string::npos)
>+      continue;
>+    char buf[32];
>+    size_t len = rtc::hex_decode(buf, sizeof(buf), line.data() + sep + 1,
>+                                 line.size() - sep - 1);
>+    if (len > 0) {
>+      name_to_key.emplace(line.substr(0, sep), std::string(buf, len));
>+    }
>+  }
>+  return name_to_key;
>+}
>+
>+}  // namespace webrtc_examples
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/read_auth_file.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/read_auth_file.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..1c139c99240cae9d1a1f7ba483e595c331f1c0a1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/read_auth_file.h
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef EXAMPLES_TURNSERVER_READ_AUTH_FILE_H_
>+#define EXAMPLES_TURNSERVER_READ_AUTH_FILE_H_
>+
>+#include <istream>
>+#include <map>
>+#include <string>
>+
>+namespace webrtc_examples {
>+
>+std::map<std::string, std::string> ReadAuthFile(std::istream* s);
>+
>+}  // namespace webrtc_examples
>+
>+#endif  // EXAMPLES_TURNSERVER_READ_AUTH_FILE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/read_auth_file_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/read_auth_file_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..4a6f332c27a57669f4f2b45592ece3f32ac6bc22
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/read_auth_file_unittest.cc
>@@ -0,0 +1,44 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <sstream>
>+
>+#include "examples/turnserver/read_auth_file.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc_examples {
>+
>+TEST(ReadAuthFile, HandlesEmptyFile) {
>+  std::istringstream empty;
>+  auto map = ReadAuthFile(&empty);
>+  EXPECT_TRUE(map.empty());
>+}
>+
>+TEST(ReadAuthFile, RecognizesValidUser) {
>+  std::istringstream file("foo=deadbeaf\n");
>+  auto map = ReadAuthFile(&file);
>+  ASSERT_NE(map.find("foo"), map.end());
>+  EXPECT_EQ(map["foo"], "\xde\xad\xbe\xaf");
>+}
>+
>+TEST(ReadAuthFile, EmptyValueForInvalidHex) {
>+  std::istringstream file(
>+      "foo=deadbeaf\n"
>+      "bar=xxxxinvalidhex\n"
>+      "baz=cafe\n");
>+  auto map = ReadAuthFile(&file);
>+  ASSERT_NE(map.find("foo"), map.end());
>+  EXPECT_EQ(map["foo"], "\xde\xad\xbe\xaf");
>+  EXPECT_EQ(map.find("bar"), map.end());
>+  ASSERT_NE(map.find("baz"), map.end());
>+  EXPECT_EQ(map["baz"], "\xca\xfe");
>+}
>+
>+}  // namespace webrtc_examples
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/turnserver_main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/turnserver_main.cc
>index edc0b699b4c7628d35451e98f0a8e9f943f3736c..850fc2d9c89a215b9e12e9880585ac0419b94f47 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/turnserver_main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/turnserver_main.cc
>@@ -8,39 +8,44 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <iostream>  // NOLINT
>+#include <fstream>
>+#include <iostream>
>+#include <map>
>+#include <string>
> 
>+#include "examples/turnserver/read_auth_file.h"
> #include "p2p/base/basicpacketsocketfactory.h"
> #include "p2p/base/turnserver.h"
> #include "rtc_base/asyncudpsocket.h"
>-#include "rtc_base/optionsfile.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/thread.h"
> 
>-static const char kSoftware[] = "libjingle TurnServer";
>+namespace {
>+const char kSoftware[] = "libjingle TurnServer";
> 
> class TurnFileAuth : public cricket::TurnAuthInterface {
>  public:
>-  explicit TurnFileAuth(const std::string& path) : file_(path) { file_.Load(); }
>+  explicit TurnFileAuth(std::map<std::string, std::string> name_to_key)
>+      : name_to_key_(std::move(name_to_key)) {}
>+
>   virtual bool GetKey(const std::string& username,
>                       const std::string& realm,
>                       std::string* key) {
>     // File is stored as lines of <username>=<HA1>.
>     // Generate HA1 via "echo -n "<username>:<realm>:<password>" | md5sum"
>-    std::string hex;
>-    bool ret = file_.GetStringValue(username, &hex);
>-    if (ret) {
>-      char buf[32];
>-      size_t len = rtc::hex_decode(buf, sizeof(buf), hex);
>-      *key = std::string(buf, len);
>-    }
>-    return ret;
>+    auto it = name_to_key_.find(username);
>+    if (it == name_to_key_.end())
>+      return false;
>+    *key = it->second;
>+    return true;
>   }
> 
>  private:
>-  rtc::OptionsFile file_;
>+  const std::map<std::string, std::string> name_to_key_;
> };
> 
>+}  // namespace
>+
> int main(int argc, char* argv[]) {
>   if (argc != 5) {
>     std::cerr << "usage: turnserver int-addr ext-ip realm auth-file"
>@@ -70,7 +75,11 @@ int main(int argc, char* argv[]) {
>   }
> 
>   cricket::TurnServer server(main);
>-  TurnFileAuth auth(argv[4]);
>+  std::fstream auth_file(argv[4], std::fstream::in);
>+
>+  TurnFileAuth auth(auth_file.is_open()
>+                        ? webrtc_examples::ReadAuthFile(&auth_file)
>+                        : std::map<std::string, std::string>());
>   server.set_realm(argv[3]);
>   server.set_software(kSoftware);
>   server.set_auth_hook(&auth);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.cc
>index 36e1937ead7d92c4e11256d1768657e3b7fc265e..bf6a6827b03071cf6648b7a1f2948bf48156514b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.cc
>@@ -15,6 +15,7 @@
> #include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/create_peerconnection_factory.h"
> #include "api/videosourceproxy.h"
> #include "media/engine/internaldecoderfactory.h"
> #include "media/engine/internalencoderfactory.h"
>@@ -448,7 +449,8 @@ void SimplePeerConnection::AddStreams(bool audio_only) {
> 
>     rtc::scoped_refptr<webrtc::jni::AndroidVideoTrackSource> source(
>         new rtc::RefCountedObject<webrtc::jni::AndroidVideoTrackSource>(
>-            g_signaling_thread.get(), env, false));
>+            g_signaling_thread.get(), env, /* is_screencast= */ false,
>+            /* align_timestamps= */ true));
>     rtc::scoped_refptr<webrtc::VideoTrackSourceProxy> proxy_source =
>         webrtc::VideoTrackSourceProxy::Create(g_signaling_thread.get(),
>                                               g_worker_thread.get(), source);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/cq.cfg b/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/cq.cfg
>index c7b5ef923b8ade4266773b1d3908cecf51bc4743..1c0f0228f811d4dc174b383d906f4c3b05cf34ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/cq.cfg
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/cq.cfg
>@@ -36,7 +36,6 @@ verifiers {
>       builders { name: "ios_arm64_rel" }
>       builders { name: "ios_dbg" }
>       builders { name: "ios_rel" }
>-      builders { name: "ios32_sim_ios9_dbg" }
>       builders { name: "ios64_sim_ios9_dbg" }
>       builders { name: "ios64_sim_ios10_dbg" }
>       builders { name: "ios_api_framework" }
>@@ -63,7 +62,12 @@ verifiers {
>       builders { name: "mac_chromium_compile" }
>       builders { name: "mac_compile_dbg" }
>       builders { name: "mac_rel" }
>-      builders { name: "presubmit" }
>+      builders {
>+        name: "presubmit"
>+        # Presubmit builder should be re-run every time CQ is triggered
>+        # for last minute lint, OWNERS, etc checks.
>+        disable_reuse: true
>+      }
>       builders { name: "win_asan" }
>       builders { name: "win_chromium_compile" }
>       builders { name: "win_clang_dbg" }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/BUILD.gn
>index 3871193badcd7f681077ed70298801b0df51f03d..575952c315f9c3182054c9987237caa39cf5d2b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/BUILD.gn
>@@ -42,6 +42,7 @@ rtc_source_set("rtc_event_log_api") {
>     "../rtc_base:ptr_util",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:rtc_task_queue",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> }
> 
>@@ -53,7 +54,6 @@ rtc_source_set("rtc_stream_config") {
> 
>   deps = [
>     ":rtc_event_log_api",
>-    "..:webrtc_common",
>     "../api:libjingle_peerconnection_api",
>   ]
> }
>@@ -87,6 +87,7 @@ rtc_source_set("rtc_event_audio") {
>     ":rtc_event_log_api",
>     ":rtc_stream_config",
>     "../modules/audio_coding:audio_network_adaptor_config",
>+    "../rtc_base:checks",
>     "../rtc_base:ptr_util",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
>@@ -130,6 +131,7 @@ rtc_source_set("rtc_event_rtp_rtcp") {
>     ":rtc_event_log_api",
>     "../api:array_view",
>     "../modules/rtp_rtcp:rtp_rtcp_format",
>+    "../rtc_base:checks",
>     "../rtc_base:ptr_util",
>     "../rtc_base:rtc_base_approved",
>     "//third_party/abseil-cpp/absl/memory",
>@@ -147,40 +149,62 @@ rtc_source_set("rtc_event_video") {
>   deps = [
>     ":rtc_event_log_api",
>     ":rtc_stream_config",
>+    "../rtc_base:checks",
>     "../rtc_base:ptr_util",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
> }
> 
>+# TODO(eladalon): Break down into (1) encoder and (2) decoder; we don't need
>+# the decoder code in the WebRTC library, only in unit tests and tools.
> rtc_static_library("rtc_event_log_impl_encoder") {
>   visibility = [ "*" ]
>   sources = [
>-    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
>-    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.h",
>+    "rtc_event_log/encoder/blob_encoding.cc",
>+    "rtc_event_log/encoder/blob_encoding.h",
>+    "rtc_event_log/encoder/delta_encoding.cc",
>+    "rtc_event_log/encoder/delta_encoding.h",
>+    "rtc_event_log/encoder/rtc_event_log_encoder_common.cc",
>+    "rtc_event_log/encoder/rtc_event_log_encoder_common.h",
>+    "rtc_event_log/encoder/varint.cc",
>+    "rtc_event_log/encoder/varint.h",
>   ]
> 
>   defines = []
> 
>   deps = [
>-    ":ice_log",
>-    ":rtc_event_audio",
>-    ":rtc_event_bwe",
>-    ":rtc_event_log_api",
>-    ":rtc_event_log_impl_output",
>-    ":rtc_event_pacing",
>-    ":rtc_event_rtp_rtcp",
>-    ":rtc_event_video",
>-    ":rtc_stream_config",
>-    "../modules/audio_coding:audio_network_adaptor",
>-    "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
>-    "../modules/rtp_rtcp:rtp_rtcp_format",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings:strings",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>   if (rtc_enable_protobuf) {
>     defines += [ "ENABLE_RTC_EVENT_LOG" ]
>-    deps += [ ":rtc_event_log_proto" ]
>+    deps += [
>+      ":ice_log",
>+      ":rtc_event_audio",
>+      ":rtc_event_bwe",
>+      ":rtc_event_log2_proto",
>+      ":rtc_event_log_api",
>+      ":rtc_event_log_impl_output",
>+      ":rtc_event_log_proto",
>+      ":rtc_event_pacing",
>+      ":rtc_event_rtp_rtcp",
>+      ":rtc_event_video",
>+      ":rtc_stream_config",
>+      "../api:array_view",
>+      "../modules/audio_coding:audio_network_adaptor",
>+      "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
>+      "../modules/rtp_rtcp:rtp_rtcp_format",
>+    ]
>+    sources += [
>+      "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
>+      "rtc_event_log/encoder/rtc_event_log_encoder_legacy.h",
>+      "rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc",
>+      "rtc_event_log/encoder/rtc_event_log_encoder_new_format.h",
>+    ]
>   }
> }
> 
>@@ -211,8 +235,7 @@ rtc_static_library("rtc_event_log_impl_base") {
>   deps = [
>     ":ice_log",
>     ":rtc_event_log_api",
>-    ":rtc_event_log_impl_encoder",
>-    ":rtc_event_log_impl_output",
>+    "../api:libjingle_logging_api",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:rtc_task_queue",
>@@ -223,7 +246,7 @@ rtc_static_library("rtc_event_log_impl_base") {
> 
>   if (rtc_enable_protobuf) {
>     defines += [ "ENABLE_RTC_EVENT_LOG" ]
>-    deps += [ ":rtc_event_log_proto" ]
>+    deps += [ ":rtc_event_log_impl_encoder" ]
>   }
> }
> 
>@@ -246,6 +269,7 @@ rtc_source_set("fake_rtc_event_log") {
> 
> if (rtc_enable_protobuf) {
>   proto_library("rtc_event_log_proto") {
>+    visibility = [ "*" ]
>     sources = [
>       "rtc_event_log/rtc_event_log.proto",
>     ]
>@@ -253,6 +277,7 @@ if (rtc_enable_protobuf) {
>   }
> 
>   proto_library("rtc_event_log2_proto") {
>+    visibility = [ "*" ]
>     sources = [
>       "rtc_event_log/rtc_event_log2.proto",
>     ]
>@@ -262,8 +287,6 @@ if (rtc_enable_protobuf) {
>   rtc_static_library("rtc_event_log_parser") {
>     visibility = [ "*" ]
>     sources = [
>-      "rtc_event_log/rtc_event_log_parser.cc",
>-      "rtc_event_log/rtc_event_log_parser.h",
>       "rtc_event_log/rtc_event_log_parser_new.cc",
>       "rtc_event_log/rtc_event_log_parser_new.h",
>       "rtc_event_log/rtc_event_processor.h",
>@@ -274,13 +297,13 @@ if (rtc_enable_protobuf) {
>       ":rtc_event_bwe",
>       ":rtc_event_log2_proto",
>       ":rtc_event_log_api",
>+      ":rtc_event_log_impl_encoder",
>       ":rtc_event_log_proto",
>       ":rtc_stream_config",
>-      "..:webrtc_common",
>       "../api:libjingle_peerconnection_api",
>       "../call:video_stream_api",
>       "../modules/audio_coding:audio_network_adaptor",
>-      "../modules/congestion_controller:transport_feedback",
>+      "../modules/congestion_controller/rtp:transport_feedback",
>       "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
>       "../modules/rtp_rtcp",
>       "../modules/rtp_rtcp:rtp_rtcp_format",
>@@ -298,6 +321,9 @@ if (rtc_enable_protobuf) {
>       assert(rtc_enable_protobuf)
>       defines = [ "ENABLE_RTC_EVENT_LOG" ]
>       sources = [
>+        "rtc_event_log/encoder/blob_encoding_unittest.cc",
>+        "rtc_event_log/encoder/delta_encoding_unittest.cc",
>+        "rtc_event_log/encoder/rtc_event_log_encoder_common_unittest.cc",
>         "rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc",
>         "rtc_event_log/output/rtc_event_log_output_file_unittest.cc",
>         "rtc_event_log/rtc_event_log_unittest.cc",
>@@ -309,6 +335,7 @@ if (rtc_enable_protobuf) {
>         ":ice_log",
>         ":rtc_event_audio",
>         ":rtc_event_bwe",
>+        ":rtc_event_log2_proto",
>         ":rtc_event_log_api",
>         ":rtc_event_log_impl_base",
>         ":rtc_event_log_impl_encoder",
>@@ -332,6 +359,7 @@ if (rtc_enable_protobuf) {
>         "../test:test_support",
>         "//testing/gtest",
>         "//third_party/abseil-cpp/absl/memory",
>+        "//third_party/abseil-cpp/absl/types:optional",
>       ]
>       if (!build_with_chromium && is_clang) {
>         # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -362,49 +390,6 @@ if (rtc_enable_protobuf) {
>       }
>     }
>   }
>-
>-  if (rtc_include_tests) {
>-    rtc_executable("rtc_event_log2text") {
>-      testonly = true
>-      sources = [
>-        "rtc_event_log/rtc_event_log2text.cc",
>-      ]
>-      deps = [
>-        ":rtc_event_log_api",
>-        ":rtc_event_log_parser",
>-        "../:webrtc_common",
>-        "../call:video_stream_api",
>-        "../modules/rtp_rtcp:rtp_rtcp_format",
>-        "../rtc_base:checks",
>-        "../rtc_base:protobuf_utils",
>-        "../rtc_base:rtc_base_approved",
>-        "../rtc_base:stringutils",
>-
>-        # TODO(kwiberg): Remove this dependency.
>-        "../api/audio_codecs:audio_codecs_api",
>-        "../modules/audio_coding:audio_network_adaptor_config",
>-        "../modules/rtp_rtcp",
>-      ]
>-      if (!build_with_chromium && is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>-    }
>-  }
>-  if (rtc_include_tests) {
>-    rtc_executable("rtc_event_log2stats") {
>-      testonly = true
>-      sources = [
>-        "rtc_event_log/rtc_event_log2stats.cc",
>-      ]
>-      deps = [
>-        ":rtc_event_log_api",
>-        ":rtc_event_log_proto",
>-        "../rtc_base:checks",
>-        "../rtc_base:rtc_base_approved",
>-      ]
>-    }
>-  }
> }
> 
> rtc_source_set("ice_log") {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/blob_encoding.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/blob_encoding.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..ee137f13fc1a402f6d808387a76a409494585fe8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/blob_encoding.cc
>@@ -0,0 +1,110 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/encoder/blob_encoding.h"
>+
>+#include <algorithm>
>+
>+#include "logging/rtc_event_log/encoder/varint.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+std::string EncodeBlobs(const std::vector<std::string>& blobs) {
>+  RTC_DCHECK(!blobs.empty());
>+
>+  size_t result_length_bound = kMaxVarIntLengthBytes * blobs.size();
>+  for (const auto& blob : blobs) {
>+    // Providing an input so long that it would cause a wrap-around is an error.
>+    RTC_DCHECK_GE(result_length_bound + blob.length(), result_length_bound);
>+    result_length_bound += blob.length();
>+  }
>+
>+  std::string result;
>+  result.reserve(result_length_bound);
>+
>+  // First, encode all of the lengths.
>+  for (absl::string_view blob : blobs) {
>+    result += EncodeVarInt(blob.length());
>+  }
>+
>+  // Second, encode the actual blobs.
>+  for (absl::string_view blob : blobs) {
>+    result.append(blob.data(), blob.length());
>+  }
>+
>+  RTC_DCHECK_LE(result.size(), result_length_bound);
>+  return result;
>+}
>+
>+std::vector<absl::string_view> DecodeBlobs(absl::string_view encoded_blobs,
>+                                           size_t num_of_blobs) {
>+  if (encoded_blobs.empty()) {
>+    RTC_LOG(LS_WARNING) << "Corrupt input; empty input.";
>+    return std::vector<absl::string_view>();
>+  }
>+
>+  if (num_of_blobs == 0u) {
>+    RTC_LOG(LS_WARNING)
>+        << "Corrupt input; number of blobs must be greater than 0.";
>+    return std::vector<absl::string_view>();
>+  }
>+
>+  size_t read_idx = 0;
>+
>+  // Read the lengths of all blobs.
>+  std::vector<uint64_t> lengths(num_of_blobs);
>+  for (size_t i = 0; i < num_of_blobs; ++i) {
>+    if (read_idx >= encoded_blobs.length()) {
>+      RTC_DCHECK_EQ(read_idx, encoded_blobs.length());
>+      RTC_LOG(LS_WARNING) << "Corrupt input; excessive number of blobs.";
>+      return std::vector<absl::string_view>();
>+    }
>+
>+    const size_t read_bytes =
>+        DecodeVarInt(encoded_blobs.substr(read_idx), &lengths[i]);
>+    if (read_bytes == 0) {
>+      RTC_LOG(LS_WARNING) << "Corrupt input; varint decoding failed.";
>+      return std::vector<absl::string_view>();
>+    }
>+
>+    read_idx += read_bytes;
>+
>+    // Note: It might be that read_idx == encoded_blobs.length(), if this
>+    // is the last iteration, and all of the blobs are the empty string.
>+    RTC_DCHECK_LE(read_idx, encoded_blobs.length());
>+  }
>+
>+  // Read the blobs themselves.
>+  std::vector<absl::string_view> blobs(num_of_blobs);
>+  for (size_t i = 0; i < num_of_blobs; ++i) {
>+    if (read_idx + lengths[i] < read_idx) {  // Wrap-around detection.
>+      RTC_LOG(LS_WARNING) << "Corrupt input; unreasonably large blob sequence.";
>+      return std::vector<absl::string_view>();
>+    }
>+
>+    if (read_idx + lengths[i] > encoded_blobs.length()) {
>+      RTC_LOG(LS_WARNING) << "Corrupt input; blob sizes exceed input size.";
>+      return std::vector<absl::string_view>();
>+    }
>+
>+    blobs[i] = encoded_blobs.substr(read_idx, lengths[i]);
>+    read_idx += lengths[i];
>+  }
>+
>+  if (read_idx != encoded_blobs.length()) {
>+    RTC_LOG(LS_WARNING) << "Corrupt input; unrecognized trailer.";
>+    return std::vector<absl::string_view>();
>+  }
>+
>+  return blobs;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/blob_encoding.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/blob_encoding.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..32a10d277f7fd6ae17030a774a03e8b71d960d8b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/blob_encoding.h
>@@ -0,0 +1,51 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef LOGGING_RTC_EVENT_LOG_ENCODER_BLOB_ENCODING_H_
>+#define LOGGING_RTC_EVENT_LOG_ENCODER_BLOB_ENCODING_H_
>+
>+#include <string>
>+#include <vector>
>+
>+#include "absl/strings/string_view.h"
>+
>+namespace webrtc {
>+
>+// Encode/decode a sequence of strings, whose length is not known to be
>+// discernable from the blob itself (i.e. without being transmitted OOB),
>+// in a way that would allow us to separate them again on the decoding side.
>+// The number of blobs is assumed to be transmitted OOB. For example, if
>+// multiple sequences of different blobs are sent, but all sequences contain
>+// the same number of blobs, it is beneficial to not encode the number of blobs.
>+//
>+// EncodeBlobs() must be given a non-empty vector. The blobs themselves may
>+// be equal to "", though.
>+// EncodeBlobs() may not fail.
>+// EncodeBlobs() never returns the empty string.
>+//
>+// Calling DecodeBlobs() on an empty string, or with |num_of_blobs| set to 0,
>+// is an error.
>+// DecodeBlobs() returns an empty vector if it fails, e.g. due to a mismatch
>+// between |num_of_blobs| and |encoded_blobs|, which can happen if
>+// |encoded_blobs| is corrupted.
>+// When successful, DecodeBlobs() returns a vector of string_view objects,
>+// which refer to the original input (|encoded_blobs|), and therefore may
>+// not outlive it.
>+//
>+// Note that the returned std::string might have been reserved for significantly
>+// more memory than it ends up using. If the caller to EncodeBlobs() intends
>+// to store the result long-term, he should consider shrink_to_fit()-ing it.
>+std::string EncodeBlobs(const std::vector<std::string>& blobs);
>+std::vector<absl::string_view> DecodeBlobs(absl::string_view encoded_blobs,
>+                                           size_t num_of_blobs);
>+
>+}  // namespace webrtc
>+
>+#endif  // LOGGING_RTC_EVENT_LOG_ENCODER_BLOB_ENCODING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/blob_encoding_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/blob_encoding_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..1d08e0feb8707f7610738a9b9e2dcdbf5841ae73
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/blob_encoding_unittest.cc
>@@ -0,0 +1,152 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/encoder/blob_encoding.h"
>+
>+#include <string>
>+#include <vector>
>+
>+#include "logging/rtc_event_log/encoder/varint.h"
>+#include "rtc_base/checks.h"
>+#include "test/gtest.h"
>+
>+using CharT = std::string::value_type;
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+void TestEncodingAndDecoding(const std::vector<std::string>& blobs) {
>+  RTC_DCHECK(!blobs.empty());
>+
>+  const std::string encoded = EncodeBlobs(blobs);
>+  ASSERT_FALSE(encoded.empty());
>+
>+  const std::vector<absl::string_view> decoded =
>+      DecodeBlobs(encoded, blobs.size());
>+
>+  ASSERT_EQ(decoded.size(), blobs.size());
>+  for (size_t i = 0; i < decoded.size(); ++i) {
>+    ASSERT_EQ(decoded[i], blobs[i]);
>+  }
>+}
>+
>+void TestGracefulErrorHandling(absl::string_view encoded_blobs,
>+                               size_t num_of_blobs) {
>+  const std::vector<absl::string_view> decoded =
>+      DecodeBlobs(encoded_blobs, num_of_blobs);
>+  EXPECT_TRUE(decoded.empty());
>+}
>+
>+}  // namespace
>+
>+TEST(BlobEncoding, EmptyBlob) {
>+  TestEncodingAndDecoding({""});
>+}
>+
>+TEST(BlobEncoding, SingleCharacterBlob) {
>+  TestEncodingAndDecoding({"a"});
>+}
>+
>+TEST(BlobEncoding, LongBlob) {
>+  std::string blob = "";
>+  for (size_t i = 0; i < 100000; ++i) {
>+    blob += std::to_string(i + 1) + " Mississippi\n";
>+  }
>+  TestEncodingAndDecoding({blob});
>+}
>+
>+TEST(BlobEncoding, BlobsOfVariousLengths) {
>+  constexpr size_t kJump = 0xf032d;  // Arbitrary.
>+  constexpr size_t kMax = 0xffffff;  // Arbitrary.
>+
>+  std::string blob;
>+  blob.reserve(kMax);
>+
>+  for (size_t i = 0; i < kMax; i += kJump) {
>+    blob.append(kJump, 'x');
>+    TestEncodingAndDecoding({blob});
>+  }
>+}
>+
>+TEST(BlobEncoding, MultipleBlobs) {
>+  std::vector<std::string> blobs;
>+  for (size_t i = 0; i < 100000; ++i) {
>+    blobs.push_back(std::to_string(i + 1) + " Mississippi\n");
>+  }
>+  TestEncodingAndDecoding(blobs);
>+}
>+
>+TEST(BlobEncoding, DecodeBlobsHandlesErrorsGracefullyEmptyInput) {
>+  TestGracefulErrorHandling("", 1);
>+}
>+
>+TEST(BlobEncoding, DecodeBlobsHandlesErrorsGracefullyZeroBlobs) {
>+  const std::string encoded = EncodeBlobs({"a"});
>+  ASSERT_FALSE(encoded.empty());
>+  TestGracefulErrorHandling(encoded, 0);
>+}
>+
>+TEST(BlobEncoding, DecodeBlobsHandlesErrorsGracefullyBlobLengthTooSmall) {
>+  std::string encoded = EncodeBlobs({"ab"});
>+  ASSERT_FALSE(encoded.empty());
>+  ASSERT_EQ(encoded[0], 0x02);
>+  encoded[0] = 0x01;
>+  TestGracefulErrorHandling(encoded, 1);
>+}
>+
>+TEST(BlobEncoding, DecodeBlobsHandlesErrorsGracefullyBlobLengthTooLarge) {
>+  std::string encoded = EncodeBlobs({"a"});
>+  ASSERT_FALSE(encoded.empty());
>+  ASSERT_EQ(encoded[0], 0x01);
>+  encoded[0] = 0x02;
>+  TestGracefulErrorHandling(encoded, 1);
>+}
>+
>+TEST(BlobEncoding,
>+     DecodeBlobsHandlesErrorsGracefullyNumberOfBlobsIncorrectlyHigh) {
>+  const std::vector<std::string> blobs = {"a", "b"};
>+  const std::string encoded = EncodeBlobs(blobs);
>+  // Test focus - two empty strings encoded, but DecodeBlobs() told way more
>+  // blobs are in the strings than could be expected.
>+  TestGracefulErrorHandling(encoded, 1000);
>+
>+  // Test sanity - show that DecodeBlobs() would have worked if it got the
>+  // correct input.
>+  TestEncodingAndDecoding(blobs);
>+}
>+
>+TEST(BlobEncoding, DecodeBlobsHandlesErrorsGracefullyDefectiveVarInt) {
>+  std::string defective_varint;
>+  for (size_t i = 0; i < kMaxVarIntLengthBytes; ++i) {
>+    ASSERT_LE(kMaxVarIntLengthBytes, 0xffu);
>+    defective_varint += static_cast<CharT>(static_cast<size_t>(0x80u) | i);
>+  }
>+  defective_varint += 0x01u;
>+
>+  const std::string defective_encoded = defective_varint + "whatever";
>+
>+  TestGracefulErrorHandling(defective_encoded, 1);
>+}
>+
>+TEST(BlobEncoding, DecodeBlobsHandlesErrorsGracefullyLengthSumWrapAround) {
>+  std::string max_size_varint;
>+  for (size_t i = 0; i < kMaxVarIntLengthBytes - 1; ++i) {
>+    max_size_varint += 0xffu;
>+  }
>+  max_size_varint += 0x7fu;
>+
>+  const std::string defective_encoded =
>+      max_size_varint + max_size_varint + "whatever";
>+
>+  TestGracefulErrorHandling(defective_encoded, 2);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/delta_encoding.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/delta_encoding.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..5e334b98d0009badf43daa4505a3d0e3fa99f338
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/delta_encoding.cc
>@@ -0,0 +1,974 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/encoder/delta_encoding.h"
>+
>+#include <algorithm>
>+#include <limits>
>+#include <memory>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "logging/rtc_event_log/encoder/varint.h"
>+#include "rtc_base/bitbuffer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+// TODO(eladalon): Only build the decoder in tools and unit tests.
>+
>+bool g_force_unsigned_for_testing = false;
>+bool g_force_signed_for_testing = false;
>+
>+size_t BitsToBytes(size_t bits) {
>+  return (bits / 8) + (bits % 8 > 0 ? 1 : 0);
>+}
>+
>+// TODO(eladalon): Replace by something more efficient.
>+uint64_t UnsignedBitWidth(uint64_t input, bool zero_val_as_zero_width = false) {
>+  if (zero_val_as_zero_width && input == 0) {
>+    return 0;
>+  }
>+
>+  uint64_t width = 0;
>+  do {  // input == 0 -> width == 1
>+    width += 1;
>+    input >>= 1;
>+  } while (input != 0);
>+  return width;
>+}
>+
>+uint64_t SignedBitWidth(uint64_t max_pos_magnitude,
>+                        uint64_t max_neg_magnitude) {
>+  const uint64_t bitwidth_pos = UnsignedBitWidth(max_pos_magnitude, true);
>+  const uint64_t bitwidth_neg =
>+      (max_neg_magnitude > 0) ? UnsignedBitWidth(max_neg_magnitude - 1, true)
>+                              : 0;
>+  return 1 + std::max(bitwidth_pos, bitwidth_neg);
>+}
>+
>+// Return the maximum integer of a given bit width.
>+// Examples:
>+// MaxUnsignedValueOfBitWidth(1) = 0x01
>+// MaxUnsignedValueOfBitWidth(6) = 0x3f
>+// MaxUnsignedValueOfBitWidth(8) = 0xff
>+// MaxUnsignedValueOfBitWidth(32) = 0xffffffff
>+uint64_t MaxUnsignedValueOfBitWidth(uint64_t bit_width) {
>+  RTC_DCHECK_GE(bit_width, 1);
>+  RTC_DCHECK_LE(bit_width, 64);
>+  return (bit_width == 64) ? std::numeric_limits<uint64_t>::max()
>+                           : ((static_cast<uint64_t>(1) << bit_width) - 1);
>+}
>+
>+// Computes the delta between |previous| and |current|, under the assumption
>+// that wrap-around occurs after |width| is exceeded.
>+uint64_t UnsignedDelta(uint64_t previous, uint64_t current, uint64_t bit_mask) {
>+  return (current - previous) & bit_mask;
>+}
>+
>+// Determines the encoding type (e.g. fixed-size encoding).
>+// Given an encoding type, may also distinguish between some variants of it
>+// (e.g. which fields of the fixed-size encoding are explicitly mentioned by
>+// the header, and which are implicitly assumed to hold certain default values).
>+enum class EncodingType {
>+  kFixedSizeUnsignedDeltasNoEarlyWrapNoOpt = 0,
>+  kFixedSizeSignedDeltasEarlyWrapAndOptSupported = 1,
>+  kReserved1 = 2,
>+  kReserved2 = 3,
>+  kNumberOfEncodingTypes  // Keep last
>+};
>+
>+// The width of each field in the encoding header. Note that this is the
>+// width in case the field exists; not all fields occur in all encoding types.
>+constexpr size_t kBitsInHeaderForEncodingType = 2;
>+constexpr size_t kBitsInHeaderForDeltaWidthBits = 6;
>+constexpr size_t kBitsInHeaderForSignedDeltas = 1;
>+constexpr size_t kBitsInHeaderForValuesOptional = 1;
>+constexpr size_t kBitsInHeaderForValueWidthBits = 6;
>+
>+static_assert(static_cast<size_t>(EncodingType::kNumberOfEncodingTypes) <=
>+                  1 << kBitsInHeaderForEncodingType,
>+              "Not all encoding types fit.");
>+
>+// Default values for when the encoding header does not specify explicitly.
>+constexpr bool kDefaultSignedDeltas = false;
>+constexpr bool kDefaultValuesOptional = false;
>+constexpr uint64_t kDefaultValueWidthBits = 64;
>+
>+// Wrap BitBufferWriter and extend its functionality by (1) keeping track of
>+// the number of bits written and (2) owning its buffer.
>+class BitWriter final {
>+ public:
>+  explicit BitWriter(size_t byte_count)
>+      : buffer_(byte_count, '\0'),
>+        bit_writer_(reinterpret_cast<uint8_t*>(&buffer_[0]), buffer_.size()),
>+        written_bits_(0),
>+        valid_(true) {
>+    RTC_DCHECK_GT(byte_count, 0);
>+  }
>+
>+  void WriteBits(uint64_t val, size_t bit_count) {
>+    RTC_DCHECK(valid_);
>+    const bool success = bit_writer_.WriteBits(val, bit_count);
>+    RTC_DCHECK(success);
>+    written_bits_ += bit_count;
>+  }
>+
>+  void WriteBits(const std::string& input) {
>+    RTC_DCHECK(valid_);
>+    for (std::string::value_type c : input) {
>+      WriteBits(c, 8 * sizeof(std::string::value_type));
>+    }
>+  }
>+
>+  // Returns everything that was written so far.
>+  // Nothing more may be written after this is called.
>+  std::string GetString() {
>+    RTC_DCHECK(valid_);
>+    valid_ = false;
>+
>+    buffer_.resize(BitsToBytes(written_bits_));
>+    written_bits_ = 0;
>+
>+    std::string result;
>+    std::swap(buffer_, result);
>+    return result;
>+  }
>+
>+ private:
>+  std::string buffer_;
>+  rtc::BitBufferWriter bit_writer_;
>+  // Note: Counting bits instead of bytes wraps around earlier than it has to,
>+  // which means the maximum length is lower than it could be. We don't expect
>+  // to go anywhere near the limit, though, so this is good enough.
>+  size_t written_bits_;
>+  bool valid_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(BitWriter);
>+};
>+
>+// Parameters for fixed-size delta-encoding/decoding.
>+// These are tailored for the sequence which will be encoded (e.g. widths).
>+class FixedLengthEncodingParameters final {
>+ public:
>+  static bool ValidParameters(uint64_t delta_width_bits,
>+                              bool signed_deltas,
>+                              bool values_optional,
>+                              uint64_t value_width_bits) {
>+    return (1 <= delta_width_bits && delta_width_bits <= 64 &&
>+            1 <= value_width_bits && value_width_bits <= 64 &&
>+            delta_width_bits <= value_width_bits);
>+  }
>+
>+  FixedLengthEncodingParameters(uint64_t delta_width_bits,
>+                                bool signed_deltas,
>+                                bool values_optional,
>+                                uint64_t value_width_bits)
>+      : delta_width_bits_(delta_width_bits),
>+        signed_deltas_(signed_deltas),
>+        values_optional_(values_optional),
>+        value_width_bits_(value_width_bits),
>+        delta_mask_(MaxUnsignedValueOfBitWidth(delta_width_bits_)),
>+        value_mask_(MaxUnsignedValueOfBitWidth(value_width_bits_)) {
>+    RTC_DCHECK(ValidParameters(delta_width_bits, signed_deltas, values_optional,
>+                               value_width_bits));
>+  }
>+
>+  // Number of bits necessary to hold the widest(*) of the deltas between the
>+  // values in the sequence.
>+  // (*) - Widest might not be the largest, if signed deltas are used.
>+  uint64_t delta_width_bits() const { return delta_width_bits_; }
>+
>+  // Whether deltas are signed.
>+  bool signed_deltas() const { return signed_deltas_; }
>+
>+  // Whether the values of the sequence are optional. That is, it may be
>+  // that some of them do not have a value (not even a sentinel value indicating
>+  // invalidity).
>+  bool values_optional() const { return values_optional_; }
>+
>+  // Number of bits necessary to hold the largest value in the sequence.
>+  uint64_t value_width_bits() const { return value_width_bits_; }
>+
>+  // Masks where only the bits relevant to the deltas/values are turned on.
>+  uint64_t delta_mask() const { return delta_mask_; }
>+  uint64_t value_mask() const { return value_mask_; }
>+
>+  void SetSignedDeltas(bool signed_deltas) { signed_deltas_ = signed_deltas; }
>+  void SetDeltaWidthBits(uint64_t delta_width_bits) {
>+    delta_width_bits_ = delta_width_bits;
>+    delta_mask_ = MaxUnsignedValueOfBitWidth(delta_width_bits);
>+  }
>+
>+ private:
>+  uint64_t delta_width_bits_;  // Normally const, but mutable in tests.
>+  bool signed_deltas_;         // Normally const, but mutable in tests.
>+  const bool values_optional_;
>+  const uint64_t value_width_bits_;
>+
>+  uint64_t delta_mask_;  // Normally const, but mutable in tests.
>+  const uint64_t value_mask_;
>+};
>+
>+// Performs delta-encoding of a single (non-empty) sequence of values, using
>+// an encoding where all deltas are encoded using the same number of bits.
>+// (With the exception of optional elements; those are encoded as a bit vector
>+// with one bit per element, plus a fixed number of bits for every element that
>+// has a value.)
>+class FixedLengthDeltaEncoder final {
>+ public:
>+  // See webrtc::EncodeDeltas() for general details.
>+  // This function return a bit pattern that would allow the decoder to
>+  // determine whether it was produced by FixedLengthDeltaEncoder, and can
>+  // therefore be decoded by FixedLengthDeltaDecoder, or whether it was produced
>+  // by a different encoder.
>+  static std::string EncodeDeltas(
>+      absl::optional<uint64_t> base,
>+      const std::vector<absl::optional<uint64_t>>& values);
>+
>+ private:
>+  // Calculate min/max values of unsigned/signed deltas, given the bit width
>+  // of all the values in the series.
>+  static void CalculateMinAndMaxDeltas(
>+      absl::optional<uint64_t> base,
>+      const std::vector<absl::optional<uint64_t>>& values,
>+      uint64_t bit_width,
>+      uint64_t* max_unsigned_delta,
>+      uint64_t* max_pos_signed_delta,
>+      uint64_t* min_neg_signed_delta);
>+
>+  // No effect outside of unit tests.
>+  // In unit tests, may lead to forcing signed/unsigned deltas, etc.
>+  static void ConsiderTestOverrides(FixedLengthEncodingParameters* params,
>+                                    uint64_t delta_width_bits_signed,
>+                                    uint64_t delta_width_bits_unsigned);
>+
>+  // FixedLengthDeltaEncoder objects are to be created by EncodeDeltas() and
>+  // released by it before it returns. They're mostly a convenient way to
>+  // avoid having to pass a lot of state between different functions.
>+  // Therefore, it was deemed acceptable to let them have a reference to
>+  // |values|, whose lifetime must exceed the lifetime of |this|.
>+  FixedLengthDeltaEncoder(const FixedLengthEncodingParameters& params,
>+                          absl::optional<uint64_t> base,
>+                          const std::vector<absl::optional<uint64_t>>& values,
>+                          size_t existent_values_count);
>+
>+  // Perform delta-encoding using the parameters given to the ctor on the
>+  // sequence of values given to the ctor.
>+  std::string Encode();
>+
>+  // Exact lengths.
>+  size_t OutputLengthBytes(size_t existent_values_count) const;
>+  size_t HeaderLengthBits() const;
>+  size_t EncodedDeltasLengthBits(size_t existent_values_count) const;
>+
>+  // Encode the compression parameters into the stream.
>+  void EncodeHeader();
>+
>+  // Encode a given delta into the stream.
>+  void EncodeDelta(uint64_t previous, uint64_t current);
>+  void EncodeUnsignedDelta(uint64_t previous, uint64_t current);
>+  void EncodeSignedDelta(uint64_t previous, uint64_t current);
>+
>+  // The parameters according to which encoding will be done (width of
>+  // fields, whether signed deltas should be used, etc.)
>+  const FixedLengthEncodingParameters params_;
>+
>+  // The encoding scheme assumes that at least one value is transmitted OOB,
>+  // so that the first value can be encoded as a delta from that OOB value,
>+  // which is |base_|.
>+  const absl::optional<uint64_t> base_;
>+
>+  // The values to be encoded.
>+  // Note: This is a non-owning reference. See comment above ctor for details.
>+  const std::vector<absl::optional<uint64_t>>& values_;
>+
>+  // Buffer into which encoded values will be written.
>+  // This is created dynmically as a way to enforce that the rest of the
>+  // ctor has finished running when this is constructed, so that the lower
>+  // bound on the buffer size would be guaranteed correct.
>+  std::unique_ptr<BitWriter> writer_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(FixedLengthDeltaEncoder);
>+};
>+
>+// TODO(eladalon): Reduce the number of passes.
>+std::string FixedLengthDeltaEncoder::EncodeDeltas(
>+    absl::optional<uint64_t> base,
>+    const std::vector<absl::optional<uint64_t>>& values) {
>+  RTC_DCHECK(!values.empty());
>+
>+  // As a special case, if all of the elements are identical to the base,
>+  // (including, for optional fields, about their existence/non-existence),
>+  // the empty string is used to signal that.
>+  if (std::all_of(
>+          values.cbegin(), values.cend(),
>+          [base](absl::optional<uint64_t> val) { return val == base; })) {
>+    return std::string();
>+  }
>+
>+  bool non_decreasing = true;
>+  uint64_t max_value_including_base = base.value_or(0u);
>+  size_t existent_values_count = 0;
>+  {
>+    uint64_t previous = base.value_or(0u);
>+    for (size_t i = 0; i < values.size(); ++i) {
>+      if (!values[i].has_value()) {
>+        continue;
>+      }
>+      ++existent_values_count;
>+      non_decreasing &= (previous <= values[i].value());
>+      max_value_including_base =
>+          std::max(max_value_including_base, values[i].value());
>+      previous = values[i].value();
>+    }
>+  }
>+
>+  // If the sequence is non-decreasing, it may be assumed to have width = 64;
>+  // there's no reason to encode the actual max width in the encoding header.
>+  const uint64_t value_width_bits =
>+      non_decreasing ? 64 : UnsignedBitWidth(max_value_including_base);
>+
>+  uint64_t max_unsigned_delta;
>+  uint64_t max_pos_signed_delta;
>+  uint64_t min_neg_signed_delta;
>+  CalculateMinAndMaxDeltas(base, values, value_width_bits, &max_unsigned_delta,
>+                           &max_pos_signed_delta, &min_neg_signed_delta);
>+
>+  const uint64_t delta_width_bits_unsigned =
>+      UnsignedBitWidth(max_unsigned_delta);
>+  const uint64_t delta_width_bits_signed =
>+      SignedBitWidth(max_pos_signed_delta, min_neg_signed_delta);
>+
>+  // Note: Preference for unsigned if the two have the same width (efficiency).
>+  const bool signed_deltas =
>+      delta_width_bits_signed < delta_width_bits_unsigned;
>+  const uint64_t delta_width_bits =
>+      signed_deltas ? delta_width_bits_signed : delta_width_bits_unsigned;
>+
>+  const bool values_optional =
>+      !base.has_value() || (existent_values_count < values.size());
>+
>+  FixedLengthEncodingParameters params(delta_width_bits, signed_deltas,
>+                                       values_optional, value_width_bits);
>+
>+  // No effect in production.
>+  ConsiderTestOverrides(&params, delta_width_bits_signed,
>+                        delta_width_bits_unsigned);
>+
>+  FixedLengthDeltaEncoder encoder(params, base, values, existent_values_count);
>+  return encoder.Encode();
>+}
>+
>+void FixedLengthDeltaEncoder::CalculateMinAndMaxDeltas(
>+    absl::optional<uint64_t> base,
>+    const std::vector<absl::optional<uint64_t>>& values,
>+    uint64_t bit_width,
>+    uint64_t* max_unsigned_delta_out,
>+    uint64_t* max_pos_signed_delta_out,
>+    uint64_t* min_neg_signed_delta_out) {
>+  RTC_DCHECK(!values.empty());
>+  RTC_DCHECK(max_unsigned_delta_out);
>+  RTC_DCHECK(max_pos_signed_delta_out);
>+  RTC_DCHECK(min_neg_signed_delta_out);
>+
>+  const uint64_t bit_mask = MaxUnsignedValueOfBitWidth(bit_width);
>+
>+  uint64_t max_unsigned_delta = 0;
>+  uint64_t max_pos_signed_delta = 0;
>+  uint64_t min_neg_signed_delta = 0;
>+
>+  absl::optional<uint64_t> prev = base;
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    if (!values[i].has_value()) {
>+      continue;
>+    }
>+
>+    if (!prev.has_value()) {
>+      // If the base is non-existent, the first existent value is encoded as
>+      // a varint, rather than as a delta.
>+      RTC_DCHECK(!base.has_value());
>+      prev = values[i];
>+      continue;
>+    }
>+
>+    const uint64_t current = values[i].value();
>+
>+    const uint64_t forward_delta = UnsignedDelta(*prev, current, bit_mask);
>+    const uint64_t backward_delta = UnsignedDelta(current, *prev, bit_mask);
>+
>+    max_unsigned_delta = std::max(max_unsigned_delta, forward_delta);
>+
>+    if (forward_delta < backward_delta) {
>+      max_pos_signed_delta = std::max(max_pos_signed_delta, forward_delta);
>+    } else {
>+      min_neg_signed_delta = std::max(min_neg_signed_delta, backward_delta);
>+    }
>+
>+    prev = current;
>+  }
>+
>+  *max_unsigned_delta_out = max_unsigned_delta;
>+  *max_pos_signed_delta_out = max_pos_signed_delta;
>+  *min_neg_signed_delta_out = min_neg_signed_delta;
>+}
>+
>+void FixedLengthDeltaEncoder::ConsiderTestOverrides(
>+    FixedLengthEncodingParameters* params,
>+    uint64_t delta_width_bits_signed,
>+    uint64_t delta_width_bits_unsigned) {
>+  if (g_force_unsigned_for_testing) {
>+    params->SetDeltaWidthBits(delta_width_bits_unsigned);
>+    params->SetSignedDeltas(false);
>+  } else if (g_force_signed_for_testing) {
>+    params->SetDeltaWidthBits(delta_width_bits_signed);
>+    params->SetSignedDeltas(true);
>+  } else {
>+    // Unchanged.
>+  }
>+}
>+
>+FixedLengthDeltaEncoder::FixedLengthDeltaEncoder(
>+    const FixedLengthEncodingParameters& params,
>+    absl::optional<uint64_t> base,
>+    const std::vector<absl::optional<uint64_t>>& values,
>+    size_t existent_values_count)
>+    : params_(params), base_(base), values_(values) {
>+  RTC_DCHECK(!values_.empty());
>+  writer_ =
>+      absl::make_unique<BitWriter>(OutputLengthBytes(existent_values_count));
>+}
>+
>+std::string FixedLengthDeltaEncoder::Encode() {
>+  EncodeHeader();
>+
>+  if (params_.values_optional()) {
>+    // Encode which values exist and which don't.
>+    for (absl::optional<uint64_t> value : values_) {
>+      writer_->WriteBits(value.has_value() ? 1u : 0u, 1);
>+    }
>+  }
>+
>+  absl::optional<uint64_t> previous = base_;
>+  for (absl::optional<uint64_t> value : values_) {
>+    if (!value.has_value()) {
>+      RTC_DCHECK(params_.values_optional());
>+      continue;
>+    }
>+
>+    if (!previous.has_value()) {
>+      // If the base is non-existent, the first existent value is encoded as
>+      // a varint, rather than as a delta.
>+      RTC_DCHECK(!base_.has_value());
>+      writer_->WriteBits(EncodeVarInt(value.value()));
>+    } else {
>+      EncodeDelta(previous.value(), value.value());
>+    }
>+
>+    previous = value;
>+  }
>+
>+  return writer_->GetString();
>+}
>+
>+size_t FixedLengthDeltaEncoder::OutputLengthBytes(
>+    size_t existent_values_count) const {
>+  return BitsToBytes(HeaderLengthBits() +
>+                     EncodedDeltasLengthBits(existent_values_count));
>+}
>+
>+size_t FixedLengthDeltaEncoder::HeaderLengthBits() const {
>+  if (params_.signed_deltas() == kDefaultSignedDeltas &&
>+      params_.values_optional() == kDefaultValuesOptional &&
>+      params_.value_width_bits() == kDefaultValueWidthBits) {
>+    return kBitsInHeaderForEncodingType + kBitsInHeaderForDeltaWidthBits;
>+  } else {
>+    return kBitsInHeaderForEncodingType + kBitsInHeaderForDeltaWidthBits +
>+           kBitsInHeaderForSignedDeltas + kBitsInHeaderForValuesOptional +
>+           kBitsInHeaderForValueWidthBits;
>+  }
>+}
>+
>+size_t FixedLengthDeltaEncoder::EncodedDeltasLengthBits(
>+    size_t existent_values_count) const {
>+  if (!params_.values_optional()) {
>+    return values_.size() * params_.delta_width_bits();
>+  } else {
>+    RTC_DCHECK_EQ(std::count_if(values_.begin(), values_.end(),
>+                                [](absl::optional<uint64_t> val) {
>+                                  return val.has_value();
>+                                }),
>+                  existent_values_count);
>+    // One bit for each delta, to indicate if the value exists, and delta_width
>+    // for each existent value, to indicate the delta itself.
>+    // If base_ is non-existent, the first value (if any) is encoded as a varint
>+    // rather than as a delta.
>+    const size_t existence_bitmap_size_bits = 1 * values_.size();
>+    const bool first_value_is_varint =
>+        !base_.has_value() && existent_values_count >= 1;
>+    const size_t first_value_varint_size_bits = 8 * kMaxVarIntLengthBytes;
>+    const size_t deltas_count = existent_values_count - first_value_is_varint;
>+    const size_t deltas_size_bits = deltas_count * params_.delta_width_bits();
>+    return existence_bitmap_size_bits + first_value_varint_size_bits +
>+           deltas_size_bits;
>+  }
>+}
>+
>+void FixedLengthDeltaEncoder::EncodeHeader() {
>+  RTC_DCHECK(writer_);
>+
>+  const EncodingType encoding_type =
>+      (params_.value_width_bits() == kDefaultValueWidthBits &&
>+       params_.signed_deltas() == kDefaultSignedDeltas &&
>+       params_.values_optional() == kDefaultValuesOptional)
>+          ? EncodingType::kFixedSizeUnsignedDeltasNoEarlyWrapNoOpt
>+          : EncodingType::kFixedSizeSignedDeltasEarlyWrapAndOptSupported;
>+
>+  writer_->WriteBits(static_cast<uint64_t>(encoding_type),
>+                     kBitsInHeaderForEncodingType);
>+
>+  // Note: Since it's meaningless for a field to be of width 0, when it comes
>+  // to fields that relate widths, we encode  width 1 as 0, width 2 as 1,
>+
>+  writer_->WriteBits(params_.delta_width_bits() - 1,
>+                     kBitsInHeaderForDeltaWidthBits);
>+
>+  if (encoding_type == EncodingType::kFixedSizeUnsignedDeltasNoEarlyWrapNoOpt) {
>+    return;
>+  }
>+
>+  writer_->WriteBits(static_cast<uint64_t>(params_.signed_deltas()),
>+                     kBitsInHeaderForSignedDeltas);
>+  writer_->WriteBits(static_cast<uint64_t>(params_.values_optional()),
>+                     kBitsInHeaderForValuesOptional);
>+  writer_->WriteBits(params_.value_width_bits() - 1,
>+                     kBitsInHeaderForValueWidthBits);
>+}
>+
>+void FixedLengthDeltaEncoder::EncodeDelta(uint64_t previous, uint64_t current) {
>+  if (params_.signed_deltas()) {
>+    EncodeSignedDelta(previous, current);
>+  } else {
>+    EncodeUnsignedDelta(previous, current);
>+  }
>+}
>+
>+void FixedLengthDeltaEncoder::EncodeUnsignedDelta(uint64_t previous,
>+                                                  uint64_t current) {
>+  RTC_DCHECK(writer_);
>+  const uint64_t delta = UnsignedDelta(previous, current, params_.value_mask());
>+  writer_->WriteBits(delta, params_.delta_width_bits());
>+}
>+
>+void FixedLengthDeltaEncoder::EncodeSignedDelta(uint64_t previous,
>+                                                uint64_t current) {
>+  RTC_DCHECK(writer_);
>+
>+  const uint64_t forward_delta =
>+      UnsignedDelta(previous, current, params_.value_mask());
>+  const uint64_t backward_delta =
>+      UnsignedDelta(current, previous, params_.value_mask());
>+
>+  uint64_t delta;
>+  if (forward_delta <= backward_delta) {
>+    delta = forward_delta;
>+  } else {
>+    // Compute the unsigned representation of a negative delta.
>+    // This is the two's complement representation of this negative value,
>+    // when deltas are of width params_.delta_mask().
>+    RTC_DCHECK_GE(params_.delta_mask(), backward_delta);
>+    RTC_DCHECK_LT(params_.delta_mask() - backward_delta, params_.delta_mask());
>+    delta = params_.delta_mask() - backward_delta + 1;
>+    RTC_DCHECK_LE(delta, params_.delta_mask());
>+  }
>+
>+  writer_->WriteBits(delta, params_.delta_width_bits());
>+}
>+
>+// Perform decoding of a a delta-encoded stream, extracting the original
>+// sequence of values.
>+class FixedLengthDeltaDecoder final {
>+ public:
>+  // Checks whether FixedLengthDeltaDecoder is a suitable decoder for this
>+  // bitstream. Note that this does NOT imply that stream is valid, and will
>+  // be decoded successfully. It DOES imply that all other decoder classes
>+  // will fail to decode this input, though.
>+  static bool IsSuitableDecoderFor(const std::string& input);
>+
>+  // Assuming that |input| is the result of fixed-size delta-encoding
>+  // that took place with the same value to |base| and over |num_of_deltas|
>+  // original values, this will return the sequence of original values.
>+  // If an error occurs (can happen if |input| is corrupt), an empty
>+  // vector will be returned.
>+  static std::vector<absl::optional<uint64_t>> DecodeDeltas(
>+      const std::string& input,
>+      absl::optional<uint64_t> base,
>+      size_t num_of_deltas);
>+
>+ private:
>+  // Reads the encoding header in |input| and returns a FixedLengthDeltaDecoder
>+  // with the corresponding configuration, that can be used to decode the
>+  // values in |input|.
>+  // If the encoding header is corrupt (contains an illegal configuration),
>+  // nullptr will be returned.
>+  // When a valid FixedLengthDeltaDecoder is returned, this does not mean that
>+  // the entire stream is free of error. Rather, only the encoding header is
>+  // examined and guaranteed.
>+  static std::unique_ptr<FixedLengthDeltaDecoder> Create(
>+      const std::string& input,
>+      absl::optional<uint64_t> base,
>+      size_t num_of_deltas);
>+
>+  // FixedLengthDeltaDecoder objects are to be created by DecodeDeltas() and
>+  // released by it before it returns. They're mostly a convenient way to
>+  // avoid having to pass a lot of state between different functions.
>+  // Therefore, it was deemed acceptable that |reader| does not own the buffer
>+  // it reads, meaning the lifetime of |this| must not exceed the lifetime
>+  // of |reader|'s underlying buffer.
>+  FixedLengthDeltaDecoder(std::unique_ptr<rtc::BitBuffer> reader,
>+                          const FixedLengthEncodingParameters& params,
>+                          absl::optional<uint64_t> base,
>+                          size_t num_of_deltas);
>+
>+  // Perform the decoding using the parameters given to the ctor.
>+  std::vector<absl::optional<uint64_t>> Decode();
>+
>+  // Decode a varint and write it to |output|. Return value indicates success
>+  // or failure. In case of failure, no guarantees are made about the contents
>+  // of |output| or the results of additional reads.
>+  bool ParseVarInt(uint64_t* output);
>+
>+  // Attempt to parse a delta from the input reader.
>+  // Returns true/false for success/failure.
>+  // Writes the delta into |delta| if successful.
>+  bool ParseDelta(uint64_t* delta);
>+
>+  // Add |delta| to |base| to produce the next value in a sequence.
>+  // The delta is applied as signed/unsigned depending on the parameters
>+  // given to the ctor. Wrap-around is taken into account according to the
>+  // values' width, as specified by the aforementioned encoding parameters.
>+  uint64_t ApplyDelta(uint64_t base, uint64_t delta) const;
>+
>+  // Helpers for ApplyDelta().
>+  uint64_t ApplyUnsignedDelta(uint64_t base, uint64_t delta) const;
>+  uint64_t ApplySignedDelta(uint64_t base, uint64_t delta) const;
>+
>+  // Reader of the input stream to be decoded. Does not own that buffer.
>+  // See comment above ctor for details.
>+  const std::unique_ptr<rtc::BitBuffer> reader_;
>+
>+  // The parameters according to which encoding will be done (width of
>+  // fields, whether signed deltas should be used, etc.)
>+  const FixedLengthEncodingParameters params_;
>+
>+  // The encoding scheme assumes that at least one value is transmitted OOB,
>+  // so that the first value can be encoded as a delta from that OOB value,
>+  // which is |base_|.
>+  const absl::optional<uint64_t> base_;
>+
>+  // The number of values to be known to be decoded.
>+  const size_t num_of_deltas_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(FixedLengthDeltaDecoder);
>+};
>+
>+bool FixedLengthDeltaDecoder::IsSuitableDecoderFor(const std::string& input) {
>+  if (input.length() < kBitsInHeaderForEncodingType) {
>+    return false;
>+  }
>+
>+  rtc::BitBuffer reader(reinterpret_cast<const uint8_t*>(&input[0]),
>+                        kBitsInHeaderForEncodingType);
>+
>+  uint32_t encoding_type_bits;
>+  const bool result =
>+      reader.ReadBits(&encoding_type_bits, kBitsInHeaderForEncodingType);
>+  RTC_DCHECK(result);
>+
>+  const auto encoding_type = static_cast<EncodingType>(encoding_type_bits);
>+  return encoding_type ==
>+             EncodingType::kFixedSizeUnsignedDeltasNoEarlyWrapNoOpt ||
>+         encoding_type ==
>+             EncodingType::kFixedSizeSignedDeltasEarlyWrapAndOptSupported;
>+}
>+
>+std::vector<absl::optional<uint64_t>> FixedLengthDeltaDecoder::DecodeDeltas(
>+    const std::string& input,
>+    absl::optional<uint64_t> base,
>+    size_t num_of_deltas) {
>+  auto decoder = FixedLengthDeltaDecoder::Create(input, base, num_of_deltas);
>+  if (!decoder) {
>+    return std::vector<absl::optional<uint64_t>>();
>+  }
>+
>+  return decoder->Decode();
>+}
>+
>+std::unique_ptr<FixedLengthDeltaDecoder> FixedLengthDeltaDecoder::Create(
>+    const std::string& input,
>+    absl::optional<uint64_t> base,
>+    size_t num_of_deltas) {
>+  if (input.length() < kBitsInHeaderForEncodingType) {
>+    return nullptr;
>+  }
>+
>+  auto reader = absl::make_unique<rtc::BitBuffer>(
>+      reinterpret_cast<const uint8_t*>(&input[0]), input.length());
>+
>+  // Encoding type
>+  uint32_t encoding_type_bits;
>+  const bool result =
>+      reader->ReadBits(&encoding_type_bits, kBitsInHeaderForEncodingType);
>+  RTC_DCHECK(result);
>+  const EncodingType encoding = static_cast<EncodingType>(encoding_type_bits);
>+  if (encoding != EncodingType::kFixedSizeUnsignedDeltasNoEarlyWrapNoOpt &&
>+      encoding !=
>+          EncodingType::kFixedSizeSignedDeltasEarlyWrapAndOptSupported) {
>+    RTC_LOG(LS_WARNING) << "Unrecognized encoding type.";
>+    return nullptr;
>+  }
>+
>+  uint32_t read_buffer;
>+
>+  // delta_width_bits
>+  if (!reader->ReadBits(&read_buffer, kBitsInHeaderForDeltaWidthBits)) {
>+    return nullptr;
>+  }
>+  RTC_DCHECK_LE(read_buffer, 64 - 1);  // See encoding for -1's rationale.
>+  const uint64_t delta_width_bits =
>+      read_buffer + 1;  // See encoding for +1's rationale.
>+
>+  // signed_deltas, values_optional, value_width_bits
>+  bool signed_deltas;
>+  bool values_optional;
>+  uint64_t value_width_bits;
>+  if (encoding == EncodingType::kFixedSizeUnsignedDeltasNoEarlyWrapNoOpt) {
>+    signed_deltas = kDefaultSignedDeltas;
>+    values_optional = kDefaultValuesOptional;
>+    value_width_bits = kDefaultValueWidthBits;
>+  } else {
>+    // signed_deltas
>+    if (!reader->ReadBits(&read_buffer, kBitsInHeaderForSignedDeltas)) {
>+      return nullptr;
>+    }
>+    signed_deltas = rtc::dchecked_cast<bool>(read_buffer);
>+
>+    // values_optional
>+    if (!reader->ReadBits(&read_buffer, kBitsInHeaderForValuesOptional)) {
>+      return nullptr;
>+    }
>+    RTC_DCHECK_LE(read_buffer, 1);
>+    values_optional = rtc::dchecked_cast<bool>(read_buffer);
>+
>+    // value_width_bits
>+    if (!reader->ReadBits(&read_buffer, kBitsInHeaderForValueWidthBits)) {
>+      return nullptr;
>+    }
>+    RTC_DCHECK_LE(read_buffer, 64 - 1);  // See encoding for -1's rationale.
>+    value_width_bits = read_buffer + 1;  // See encoding for +1's rationale.
>+  }
>+
>+  // Note: Because of the way the parameters are read, it is not possible
>+  // for illegal values to be read. We check nevertheless, in case the code
>+  // changes in the future in a way that breaks this promise.
>+  if (!FixedLengthEncodingParameters::ValidParameters(
>+          delta_width_bits, signed_deltas, values_optional, value_width_bits)) {
>+    RTC_LOG(LS_WARNING) << "Corrupt log; illegal encoding parameters.";
>+    return nullptr;
>+  }
>+
>+  FixedLengthEncodingParameters params(delta_width_bits, signed_deltas,
>+                                       values_optional, value_width_bits);
>+  return absl::WrapUnique(new FixedLengthDeltaDecoder(std::move(reader), params,
>+                                                      base, num_of_deltas));
>+}
>+
>+FixedLengthDeltaDecoder::FixedLengthDeltaDecoder(
>+    std::unique_ptr<rtc::BitBuffer> reader,
>+    const FixedLengthEncodingParameters& params,
>+    absl::optional<uint64_t> base,
>+    size_t num_of_deltas)
>+    : reader_(std::move(reader)),
>+      params_(params),
>+      base_(base),
>+      num_of_deltas_(num_of_deltas) {
>+  RTC_DCHECK(reader_);
>+}
>+
>+std::vector<absl::optional<uint64_t>> FixedLengthDeltaDecoder::Decode() {
>+  RTC_DCHECK(reader_);
>+
>+  std::vector<bool> existing_values(num_of_deltas_);
>+  if (params_.values_optional()) {
>+    for (size_t i = 0; i < num_of_deltas_; ++i) {
>+      uint32_t exists;
>+      if (!reader_->ReadBits(&exists, 1u)) {
>+        RTC_LOG(LS_WARNING) << "Failed to read existence-indicating bit.";
>+        return std::vector<absl::optional<uint64_t>>();
>+      }
>+      RTC_DCHECK_LE(exists, 1u);
>+      existing_values[i] = (exists == 1);
>+    }
>+  } else {
>+    std::fill(existing_values.begin(), existing_values.end(), true);
>+  }
>+
>+  absl::optional<uint64_t> previous = base_;
>+  std::vector<absl::optional<uint64_t>> values(num_of_deltas_);
>+
>+  for (size_t i = 0; i < num_of_deltas_; ++i) {
>+    if (!existing_values[i]) {
>+      RTC_DCHECK(params_.values_optional());
>+      continue;
>+    }
>+
>+    if (!previous) {
>+      // If the base is non-existent, the first existent value is encoded as
>+      // a varint, rather than as a delta.
>+      RTC_DCHECK(!base_.has_value());
>+      uint64_t first_value;
>+      if (!ParseVarInt(&first_value)) {
>+        RTC_LOG(LS_WARNING) << "Failed to read first value.";
>+        return std::vector<absl::optional<uint64_t>>();
>+      }
>+      values[i] = first_value;
>+    } else {
>+      uint64_t delta;
>+      if (!ParseDelta(&delta)) {
>+        return std::vector<absl::optional<uint64_t>>();
>+      }
>+      values[i] = ApplyDelta(previous.value(), delta);
>+    }
>+
>+    previous = values[i];
>+  }
>+
>+  return values;
>+}
>+
>+bool FixedLengthDeltaDecoder::ParseVarInt(uint64_t* output) {
>+  RTC_DCHECK(reader_);
>+  return DecodeVarInt(reader_.get(), output) != 0;
>+}
>+
>+bool FixedLengthDeltaDecoder::ParseDelta(uint64_t* delta) {
>+  RTC_DCHECK(reader_);
>+
>+  // BitBuffer and BitBufferWriter read/write higher bits before lower bits.
>+
>+  const size_t lower_bit_count =
>+      std::min<uint64_t>(params_.delta_width_bits(), 32u);
>+  const size_t higher_bit_count = (params_.delta_width_bits() <= 32u)
>+                                      ? 0
>+                                      : params_.delta_width_bits() - 32u;
>+
>+  uint32_t lower_bits;
>+  uint32_t higher_bits;
>+
>+  if (higher_bit_count > 0) {
>+    if (!reader_->ReadBits(&higher_bits, higher_bit_count)) {
>+      RTC_LOG(LS_WARNING) << "Failed to read higher half of delta.";
>+      return false;
>+    }
>+  } else {
>+    higher_bits = 0;
>+  }
>+
>+  if (!reader_->ReadBits(&lower_bits, lower_bit_count)) {
>+    RTC_LOG(LS_WARNING) << "Failed to read lower half of delta.";
>+    return false;
>+  }
>+
>+  const uint64_t lower_bits_64 = static_cast<uint64_t>(lower_bits);
>+  const uint64_t higher_bits_64 = static_cast<uint64_t>(higher_bits);
>+
>+  *delta = (higher_bits_64 << 32) | lower_bits_64;
>+  return true;
>+}
>+
>+uint64_t FixedLengthDeltaDecoder::ApplyDelta(uint64_t base,
>+                                             uint64_t delta) const {
>+  RTC_DCHECK_LE(base, MaxUnsignedValueOfBitWidth(params_.value_width_bits()));
>+  RTC_DCHECK_LE(delta, MaxUnsignedValueOfBitWidth(params_.delta_width_bits()));
>+  return params_.signed_deltas() ? ApplySignedDelta(base, delta)
>+                                 : ApplyUnsignedDelta(base, delta);
>+}
>+
>+uint64_t FixedLengthDeltaDecoder::ApplyUnsignedDelta(uint64_t base,
>+                                                     uint64_t delta) const {
>+  // Note: May still be used if signed deltas used.
>+  RTC_DCHECK_LE(base, MaxUnsignedValueOfBitWidth(params_.value_width_bits()));
>+  RTC_DCHECK_LE(delta, MaxUnsignedValueOfBitWidth(params_.delta_width_bits()));
>+  return (base + delta) & params_.value_mask();
>+}
>+
>+uint64_t FixedLengthDeltaDecoder::ApplySignedDelta(uint64_t base,
>+                                                   uint64_t delta) const {
>+  RTC_DCHECK(params_.signed_deltas());
>+  RTC_DCHECK_LE(base, MaxUnsignedValueOfBitWidth(params_.value_width_bits()));
>+  RTC_DCHECK_LE(delta, MaxUnsignedValueOfBitWidth(params_.delta_width_bits()));
>+
>+  const uint64_t top_bit = static_cast<uint64_t>(1)
>+                           << (params_.delta_width_bits() - 1);
>+
>+  const bool positive_delta = ((delta & top_bit) == 0);
>+  if (positive_delta) {
>+    return ApplyUnsignedDelta(base, delta);
>+  }
>+
>+  const uint64_t delta_abs = (~delta & params_.delta_mask()) + 1;
>+  return (base - delta_abs) & params_.value_mask();
>+}
>+
>+}  // namespace
>+
>+std::string EncodeDeltas(absl::optional<uint64_t> base,
>+                         const std::vector<absl::optional<uint64_t>>& values) {
>+  // TODO(eladalon): Support additional encodings.
>+  return FixedLengthDeltaEncoder::EncodeDeltas(base, values);
>+}
>+
>+std::vector<absl::optional<uint64_t>> DecodeDeltas(
>+    const std::string& input,
>+    absl::optional<uint64_t> base,
>+    size_t num_of_deltas) {
>+  RTC_DCHECK_GT(num_of_deltas, 0);  // Allows empty vector to indicate error.
>+
>+  // The empty string is a special case indicating that all values were equal
>+  // to the base.
>+  if (input.empty()) {
>+    std::vector<absl::optional<uint64_t>> result(num_of_deltas);
>+    std::fill(result.begin(), result.end(), base);
>+    return result;
>+  }
>+
>+  if (FixedLengthDeltaDecoder::IsSuitableDecoderFor(input)) {
>+    return FixedLengthDeltaDecoder::DecodeDeltas(input, base, num_of_deltas);
>+  }
>+
>+  RTC_LOG(LS_WARNING) << "Could not decode delta-encoded stream.";
>+  return std::vector<absl::optional<uint64_t>>();
>+}
>+
>+void SetFixedLengthEncoderDeltaSignednessForTesting(bool signedness) {
>+  g_force_unsigned_for_testing = !signedness;
>+  g_force_signed_for_testing = signedness;
>+}
>+
>+void UnsetFixedLengthEncoderDeltaSignednessForTesting() {
>+  g_force_unsigned_for_testing = false;
>+  g_force_signed_for_testing = false;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/delta_encoding.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/delta_encoding.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..9e5e8b7f43cdd4b9d4290fbd4fc860df566d51e1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/delta_encoding.h
>@@ -0,0 +1,45 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef LOGGING_RTC_EVENT_LOG_ENCODER_DELTA_ENCODING_H_
>+#define LOGGING_RTC_EVENT_LOG_ENCODER_DELTA_ENCODING_H_
>+
>+#include <string>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+
>+namespace webrtc {
>+
>+// Encode |values| as a sequence of deltas following on |base| and return it.
>+// If all of the values were equal to the base, an empty string will be
>+// returned; this is a valid encoding of that edge case.
>+// |base| is not guaranteed to be written into |output|, and must therefore
>+// be provided separately to the decoder.
>+// This function never fails.
>+// TODO(eladalon): Split into optional and non-optional variants (efficiency).
>+std::string EncodeDeltas(absl::optional<uint64_t> base,
>+                         const std::vector<absl::optional<uint64_t>>& values);
>+
>+// EncodeDeltas() and DecodeDeltas() are inverse operations;
>+// invoking DecodeDeltas() over the output of EncodeDeltas(), will return
>+// the input originally given to EncodeDeltas().
>+// |num_of_deltas| must be greater than zero. If input is not a valid encoding
>+// of |num_of_deltas| elements based on |base|, the function returns an empty
>+// vector, which signals an error.
>+// TODO(eladalon): Split into optional and non-optional variants (efficiency).
>+std::vector<absl::optional<uint64_t>> DecodeDeltas(
>+    const std::string& input,
>+    absl::optional<uint64_t> base,
>+    size_t num_of_deltas);
>+
>+}  // namespace webrtc
>+
>+#endif  // LOGGING_RTC_EVENT_LOG_ENCODER_DELTA_ENCODING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/delta_encoding_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/delta_encoding_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..ca44010edac3005af51f90cddf593824f455fa14
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/delta_encoding_unittest.cc
>@@ -0,0 +1,693 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/encoder/delta_encoding.h"
>+
>+#include <algorithm>
>+#include <limits>
>+#include <numeric>
>+#include <string>
>+#include <tuple>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/random.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+void SetFixedLengthEncoderDeltaSignednessForTesting(bool signedness);
>+void UnsetFixedLengthEncoderDeltaSignednessForTesting();
>+
>+namespace {
>+
>+enum class DeltaSignedness { kNoOverride, kForceUnsigned, kForceSigned };
>+
>+void MaybeSetSignedness(DeltaSignedness signedness) {
>+  switch (signedness) {
>+    case DeltaSignedness::kNoOverride:
>+      UnsetFixedLengthEncoderDeltaSignednessForTesting();
>+      return;
>+    case DeltaSignedness::kForceUnsigned:
>+      SetFixedLengthEncoderDeltaSignednessForTesting(false);
>+      return;
>+    case DeltaSignedness::kForceSigned:
>+      SetFixedLengthEncoderDeltaSignednessForTesting(true);
>+      return;
>+  }
>+  RTC_NOTREACHED();
>+}
>+
>+uint64_t RandomWithMaxBitWidth(Random* prng, uint64_t max_width) {
>+  RTC_DCHECK_GE(max_width, 1u);
>+  RTC_DCHECK_LE(max_width, 64u);
>+
>+  const uint64_t low = prng->Rand(std::numeric_limits<uint32_t>::max());
>+  const uint64_t high =
>+      max_width > 32u ? prng->Rand(std::numeric_limits<uint32_t>::max()) : 0u;
>+
>+  const uint64_t random_before_mask = (high << 32) | low;
>+
>+  if (max_width < 64) {
>+    return random_before_mask & ((static_cast<uint64_t>(1) << max_width) - 1);
>+  } else {
>+    return random_before_mask;
>+  }
>+}
>+
>+// Encodes |values| based on |base|, then decodes the result and makes sure
>+// that it is equal to the original input.
>+// If |encoded_string| is non-null, the encoded result will also be written
>+// into it.
>+void TestEncodingAndDecoding(
>+    absl::optional<uint64_t> base,
>+    const std::vector<absl::optional<uint64_t>>& values,
>+    std::string* encoded_string = nullptr) {
>+  const std::string encoded = EncodeDeltas(base, values);
>+  if (encoded_string) {
>+    *encoded_string = encoded;
>+  }
>+
>+  const std::vector<absl::optional<uint64_t>> decoded =
>+      DecodeDeltas(encoded, base, values.size());
>+
>+  EXPECT_EQ(decoded, values);
>+}
>+
>+std::vector<absl::optional<uint64_t>> CreateSequenceByFirstValue(
>+    uint64_t first,
>+    size_t sequence_length) {
>+  std::vector<absl::optional<uint64_t>> sequence(sequence_length);
>+  std::iota(sequence.begin(), sequence.end(), first);
>+  return sequence;
>+}
>+
>+std::vector<absl::optional<uint64_t>> CreateSequenceByLastValue(
>+    uint64_t last,
>+    size_t num_values) {
>+  const uint64_t first = last - num_values + 1;
>+  std::vector<absl::optional<uint64_t>> result(num_values);
>+  std::iota(result.begin(), result.end(), first);
>+  return result;
>+}
>+
>+// If |sequence_length| is greater than the number of deltas, the sequence of
>+// deltas will wrap around.
>+std::vector<absl::optional<uint64_t>> CreateSequenceByOptionalDeltas(
>+    uint64_t first,
>+    const std::vector<absl::optional<uint64_t>>& deltas,
>+    size_t sequence_length) {
>+  RTC_DCHECK_GE(sequence_length, 1);
>+
>+  std::vector<absl::optional<uint64_t>> sequence(sequence_length);
>+
>+  uint64_t previous = first;
>+  for (size_t i = 0, next_delta_index = 0; i < sequence.size(); ++i) {
>+    if (deltas[next_delta_index].has_value()) {
>+      sequence[i] =
>+          absl::optional<uint64_t>(previous + deltas[next_delta_index].value());
>+      previous = sequence[i].value();
>+    }
>+    next_delta_index = (next_delta_index + 1) % deltas.size();
>+  }
>+
>+  return sequence;
>+}
>+
>+size_t EncodingLengthUpperBound(size_t delta_max_bit_width,
>+                                size_t num_of_deltas,
>+                                DeltaSignedness signedness_override) {
>+  absl::optional<size_t> smallest_header_size_bytes;
>+  switch (signedness_override) {
>+    case DeltaSignedness::kNoOverride:
>+    case DeltaSignedness::kForceUnsigned:
>+      smallest_header_size_bytes = 1;
>+      break;
>+    case DeltaSignedness::kForceSigned:
>+      smallest_header_size_bytes = 2;
>+      break;
>+  }
>+  RTC_DCHECK(smallest_header_size_bytes);
>+
>+  return delta_max_bit_width * num_of_deltas + *smallest_header_size_bytes;
>+}
>+
>+// If |sequence_length| is greater than the number of deltas, the sequence of
>+// deltas will wrap around.
>+std::vector<absl::optional<uint64_t>> CreateSequenceByDeltas(
>+    uint64_t first,
>+    const std::vector<uint64_t>& deltas,
>+    size_t sequence_length) {
>+  RTC_DCHECK(!deltas.empty());
>+  std::vector<absl::optional<uint64_t>> optional_deltas(deltas.size());
>+  for (size_t i = 0; i < deltas.size(); ++i) {
>+    optional_deltas[i] = absl::optional<uint64_t>(deltas[i]);
>+  }
>+  return CreateSequenceByOptionalDeltas(first, optional_deltas,
>+                                        sequence_length);
>+}
>+
>+// Tests of the delta encoding, parameterized by the number of values
>+// in the sequence created by the test.
>+class DeltaEncodingTest
>+    : public ::testing::TestWithParam<
>+          std::tuple<DeltaSignedness, size_t, bool, uint64_t>> {
>+ public:
>+  DeltaEncodingTest()
>+      : signedness_(std::get<0>(GetParam())),
>+        num_of_values_(std::get<1>(GetParam())),
>+        optional_values_(std::get<2>(GetParam())),
>+        partial_random_seed_(std::get<3>(GetParam())) {
>+    MaybeSetSignedness(signedness_);
>+  }
>+
>+  ~DeltaEncodingTest() override = default;
>+
>+  // Running with the same seed for all variants would make all tests start
>+  // with the same sequence; avoid this by making the seed different.
>+  uint64_t Seed() const {
>+    // Multiply everything but by different primes to produce unique results.
>+    return 2 * static_cast<uint64_t>(signedness_) + 3 * num_of_values_ +
>+           5 * optional_values_ + 7 * partial_random_seed_;
>+  }
>+
>+  const DeltaSignedness signedness_;
>+  const uint64_t num_of_values_;
>+  const bool optional_values_;
>+  const uint64_t partial_random_seed_;  // Explained where it's used.
>+};
>+
>+TEST_P(DeltaEncodingTest, AllValuesEqualToExistentBaseValue) {
>+  const absl::optional<uint64_t> base(3432);
>+  std::vector<absl::optional<uint64_t>> values(num_of_values_);
>+  std::fill(values.begin(), values.end(), base);
>+  std::string encoded;
>+  TestEncodingAndDecoding(base, values, &encoded);
>+
>+  // Additional requirement - the encoding should be efficient in this
>+  // case - the empty string will be used.
>+  EXPECT_TRUE(encoded.empty());
>+}
>+
>+TEST_P(DeltaEncodingTest, AllValuesEqualToNonExistentBaseValue) {
>+  if (!optional_values_) {
>+    return;  // Test irrelevant for this case.
>+  }
>+
>+  const absl::optional<uint64_t> base;
>+  std::vector<absl::optional<uint64_t>> values(num_of_values_);
>+  std::fill(values.begin(), values.end(), base);
>+  std::string encoded;
>+  TestEncodingAndDecoding(base, values, &encoded);
>+
>+  // Additional requirement - the encoding should be efficient in this
>+  // case - the empty string will be used.
>+  EXPECT_TRUE(encoded.empty());
>+}
>+
>+TEST_P(DeltaEncodingTest, BaseNonExistentButSomeOtherValuesExist) {
>+  if (!optional_values_) {
>+    return;  // Test irrelevant for this case.
>+  }
>+
>+  const absl::optional<uint64_t> base;
>+  std::vector<absl::optional<uint64_t>> values(num_of_values_);
>+
>+  Random prng(Seed());
>+
>+  const uint64_t max_bit_width = 1 + prng.Rand(63);  // [1, 64]
>+
>+  for (size_t i = 0; i < values.size();) {
>+    // Leave a random number of values as non-existent.
>+    const size_t non_existent_count = prng.Rand(values.size() - i - 1);
>+    i += non_existent_count;
>+
>+    // Assign random values to a random number of values. (At least one, to
>+    // prevent this iteration of the outer loop from being a no-op.)
>+    const size_t existent_count =
>+        std::max<size_t>(prng.Rand(values.size() - i - 1), 1);
>+    for (size_t j = 0; j < existent_count; ++j) {
>+      values[i + j] = RandomWithMaxBitWidth(&prng, max_bit_width);
>+    }
>+    i += existent_count;
>+  }
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+TEST_P(DeltaEncodingTest, MinDeltaNoWrapAround) {
>+  const absl::optional<uint64_t> base(3432);
>+
>+  auto values = CreateSequenceByFirstValue(base.value() + 1, num_of_values_);
>+  ASSERT_GT(values[values.size() - 1], base) << "Sanity; must not wrap around";
>+
>+  if (optional_values_) {
>+    // Arbitrarily make one of the values non-existent, to force
>+    // optional-supporting encoding.
>+    values[0] = absl::optional<uint64_t>();
>+  }
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+TEST_P(DeltaEncodingTest, BigDeltaNoWrapAround) {
>+  const uint64_t kBigDelta = 132828;
>+  const absl::optional<uint64_t> base(3432);
>+
>+  auto values =
>+      CreateSequenceByFirstValue(base.value() + kBigDelta, num_of_values_);
>+  ASSERT_GT(values[values.size() - 1], base) << "Sanity; must not wrap around";
>+
>+  if (optional_values_) {
>+    // Arbitrarily make one of the values non-existent, to force
>+    // optional-supporting encoding.
>+    values[0] = absl::optional<uint64_t>();
>+  }
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+TEST_P(DeltaEncodingTest, MaxDeltaNoWrapAround) {
>+  const absl::optional<uint64_t> base(3432);
>+
>+  auto values = CreateSequenceByLastValue(std::numeric_limits<uint64_t>::max(),
>+                                          num_of_values_);
>+  ASSERT_GT(values[values.size() - 1], base) << "Sanity; must not wrap around";
>+
>+  if (optional_values_) {
>+    // Arbitrarily make one of the values non-existent, to force
>+    // optional-supporting encoding.
>+    values[0] = absl::optional<uint64_t>();
>+  }
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+TEST_P(DeltaEncodingTest, SmallDeltaWithWrapAroundComparedToBase) {
>+  if (optional_values_ && num_of_values_ == 1) {
>+    return;  // Inapplicable
>+  }
>+
>+  const absl::optional<uint64_t> base(std::numeric_limits<uint64_t>::max());
>+
>+  auto values = CreateSequenceByDeltas(*base, {1, 10, 3}, num_of_values_);
>+  ASSERT_LT(values[0], base) << "Sanity; must wrap around";
>+
>+  if (optional_values_) {
>+    // Arbitrarily make one of the values non-existent, to force
>+    // optional-supporting encoding.
>+    values[1] = absl::optional<uint64_t>();
>+  }
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+TEST_P(DeltaEncodingTest, SmallDeltaWithWrapAroundInValueSequence) {
>+  if (num_of_values_ == 1 || (optional_values_ && num_of_values_ < 3)) {
>+    return;  // Inapplicable.
>+  }
>+
>+  const absl::optional<uint64_t> base(std::numeric_limits<uint64_t>::max() - 2);
>+
>+  auto values = CreateSequenceByDeltas(*base, {1, 10, 3}, num_of_values_);
>+  ASSERT_LT(values[values.size() - 1], values[0]) << "Sanity; must wrap around";
>+
>+  if (optional_values_) {
>+    // Arbitrarily make one of the values non-existent, to force
>+    // optional-supporting encoding.
>+    RTC_DCHECK_GT(values.size() - 1, 1u);  // Wrap around not cancelled.
>+    values[1] = absl::optional<uint64_t>();
>+  }
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+// Suppress "integral constant overflow" warning; this is the test's focus.
>+#ifdef _MSC_VER
>+#pragma warning(push)
>+#pragma warning(disable : 4307)
>+#endif
>+TEST_P(DeltaEncodingTest, BigDeltaWithWrapAroundComparedToBase) {
>+  if (optional_values_ && num_of_values_ == 1) {
>+    return;  // Inapplicable
>+  }
>+
>+  const uint64_t kBigDelta = 132828;
>+  const absl::optional<uint64_t> base(std::numeric_limits<uint64_t>::max() -
>+                                      kBigDelta + 3);
>+
>+  auto values =
>+      CreateSequenceByFirstValue(base.value() + kBigDelta, num_of_values_);
>+  ASSERT_LT(values[0], base.value()) << "Sanity; must wrap around";
>+
>+  if (optional_values_) {
>+    // Arbitrarily make one of the values non-existent, to force
>+    // optional-supporting encoding.
>+    values[1] = absl::optional<uint64_t>();
>+  }
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+TEST_P(DeltaEncodingTest, BigDeltaWithWrapAroundInValueSequence) {
>+  if (num_of_values_ == 1 || (optional_values_ && num_of_values_ < 3)) {
>+    return;  // Inapplicable.
>+  }
>+
>+  const uint64_t kBigDelta = 132828;
>+  const absl::optional<uint64_t> base(std::numeric_limits<uint64_t>::max() -
>+                                      kBigDelta + 3);
>+
>+  auto values = CreateSequenceByFirstValue(std::numeric_limits<uint64_t>::max(),
>+                                           num_of_values_);
>+  ASSERT_LT(values[values.size() - 1], values[0]) << "Sanity; must wrap around";
>+
>+  if (optional_values_) {
>+    // Arbitrarily make one of the values non-existent, to force
>+    // optional-supporting encoding.
>+    RTC_DCHECK_GT(values.size() - 1, 1u);  // Wrap around not cancelled.
>+    values[1] = absl::optional<uint64_t>();
>+  }
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+#ifdef _MSC_VER
>+#pragma warning(pop)
>+#endif
>+
>+TEST_P(DeltaEncodingTest, MaxDeltaWithWrapAroundComparedToBase) {
>+  if (optional_values_ && num_of_values_ == 1) {
>+    return;  // Inapplicable
>+  }
>+
>+  const absl::optional<uint64_t> base(3432);
>+  auto values = CreateSequenceByFirstValue(*base - 1, num_of_values_);
>+
>+  if (optional_values_) {
>+    // Arbitrarily make one of the values non-existent, to force
>+    // optional-supporting encoding.
>+    values[1] = absl::optional<uint64_t>();
>+  }
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+TEST_P(DeltaEncodingTest, MaxDeltaWithWrapAroundInValueSequence) {
>+  if (num_of_values_ == 1 || (optional_values_ && num_of_values_ < 3)) {
>+    return;  // Inapplicable.
>+  }
>+
>+  const absl::optional<uint64_t> base(3432);
>+
>+  auto values = CreateSequenceByDeltas(
>+      *base, {0, std::numeric_limits<uint64_t>::max(), 3}, num_of_values_);
>+  // Wraps around continuously by virtue of being max(); will not ASSERT.
>+
>+  if (optional_values_) {
>+    // Arbitrarily make one of the values non-existent, to force
>+    // optional-supporting encoding.
>+    RTC_DCHECK_GT(values.size() - 1, 1u);  // Wrap around not cancelled.
>+    values[1] = absl::optional<uint64_t>();
>+  }
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+// If num_of_values_ == 1, a zero delta will yield an empty string; that's
>+// already covered by AllValuesEqualToExistentBaseValue, but it doesn't hurt to
>+// test again. For all other cases, we have a new test.
>+TEST_P(DeltaEncodingTest, ZeroDelta) {
>+  const absl::optional<uint64_t> base(3432);
>+
>+  // Arbitrary sequence of deltas with intentional zero deltas, as well as
>+  // consecutive zeros.
>+  const std::vector<uint64_t> deltas = {0,      312, 11, 1,  1, 0, 0, 12,
>+                                        400321, 3,   3,  12, 5, 0, 6};
>+  auto values = CreateSequenceByDeltas(base.value(), deltas, num_of_values_);
>+
>+  if (optional_values_) {
>+    // Arbitrarily make one of the values non-existent, to force
>+    // optional-supporting encoding.
>+    values[0] = absl::optional<uint64_t>();
>+  }
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+INSTANTIATE_TEST_CASE_P(
>+    SignednessOverrideAndNumberOfValuesInSequence,
>+    DeltaEncodingTest,
>+    ::testing::Combine(::testing::Values(DeltaSignedness::kNoOverride,
>+                                         DeltaSignedness::kForceUnsigned,
>+                                         DeltaSignedness::kForceSigned),
>+                       ::testing::Values(1, 2, 100, 10000),
>+                       ::testing::Bool(),
>+                       ::testing::Values(10, 20, 30)));
>+
>+// Tests over the quality of the compression (as opposed to its correctness).
>+// Not to be confused with tests of runtime efficiency.
>+class DeltaEncodingCompressionQualityTest
>+    : public ::testing::TestWithParam<
>+          std::tuple<DeltaSignedness, uint64_t, uint64_t, uint64_t>> {
>+ public:
>+  DeltaEncodingCompressionQualityTest()
>+      : signedness_(std::get<0>(GetParam())),
>+        delta_max_bit_width_(std::get<1>(GetParam())),
>+        num_of_values_(std::get<2>(GetParam())),
>+        partial_random_seed_(std::get<3>(GetParam())) {
>+    MaybeSetSignedness(signedness_);
>+  }
>+
>+  ~DeltaEncodingCompressionQualityTest() override = default;
>+
>+  // Running with the same seed for all variants would make all tests start
>+  // with the same sequence; avoid this by making the seed different.
>+  uint64_t Seed() const {
>+    // Multiply everything but by different primes to produce unique results.
>+    return 2 * static_cast<uint64_t>(signedness_) + 3 * delta_max_bit_width_ +
>+           5 * delta_max_bit_width_ + 7 * num_of_values_ +
>+           11 * partial_random_seed_;
>+  }
>+
>+  const DeltaSignedness signedness_;
>+  const uint64_t delta_max_bit_width_;
>+  const uint64_t num_of_values_;
>+  const uint64_t partial_random_seed_;  // Explained where it's used.
>+};
>+
>+// If no wrap-around occurs in the stream, the width of the values does not
>+// matter to compression performance; only the deltas matter.
>+TEST_P(DeltaEncodingCompressionQualityTest,
>+       BaseDoesNotAffectEfficiencyIfNoWrapAround) {
>+  // 1. Bases which will not produce a wrap-around.
>+  // 2. The last base - 0xffffffffffffffff - does cause a wrap-around, but
>+  //    that still works, because the width is 64 anyway, and does not
>+  //    need to be conveyed explicitly in the encoding header.
>+  const uint64_t bases[] = {0, 0x55, 0xffffffff,
>+                            std::numeric_limits<uint64_t>::max()};
>+  const size_t kIntendedWrapAroundBaseIndex = arraysize(bases);
>+
>+  std::vector<uint64_t> deltas(num_of_values_);
>+
>+  // Allows us to make sure that the deltas do not produce a wrap-around.
>+  uint64_t last_element[arraysize(bases)];
>+  memcpy(last_element, bases, sizeof(bases));
>+
>+  // Avoid empty |deltas| due to first element causing wrap-around.
>+  deltas[0] = 1;
>+  for (size_t i = 0; i < arraysize(last_element); ++i) {
>+    last_element[i] += 1;
>+  }
>+
>+  Random prng(Seed());
>+
>+  for (size_t i = 1; i < deltas.size(); ++i) {
>+    const uint64_t delta = RandomWithMaxBitWidth(&prng, delta_max_bit_width_);
>+
>+    bool wrap_around = false;
>+    for (size_t j = 0; j < arraysize(last_element); ++j) {
>+      if (j == kIntendedWrapAroundBaseIndex) {
>+        continue;
>+      }
>+
>+      last_element[j] += delta;
>+      if (last_element[j] < bases[j]) {
>+        wrap_around = true;
>+        break;
>+      }
>+    }
>+
>+    if (wrap_around) {
>+      deltas.resize(i);
>+      break;
>+    }
>+
>+    deltas[i] = delta;
>+  }
>+
>+  std::string encodings[arraysize(bases)];
>+
>+  for (size_t i = 0; i < arraysize(bases); ++i) {
>+    const auto values =
>+        CreateSequenceByDeltas(bases[i], deltas, num_of_values_);
>+    // Produce the encoding and write it to encodings[i].
>+    // By using TestEncodingAndDecoding() to do this, we also sanity-test
>+    // the encoding/decoding, though that is not the test's focus.
>+    TestEncodingAndDecoding(bases[i], values, &encodings[i]);
>+    EXPECT_LE(encodings[i].length(),
>+              EncodingLengthUpperBound(delta_max_bit_width_, num_of_values_,
>+                                       signedness_));
>+  }
>+
>+  // Test focus - all of the encodings should be the same, as they are based
>+  // on the same delta sequence, and do not contain a wrap-around.
>+  for (size_t i = 1; i < arraysize(encodings); ++i) {
>+    EXPECT_EQ(encodings[i], encodings[0]);
>+  }
>+}
>+
>+INSTANTIATE_TEST_CASE_P(
>+    SignednessOverrideAndDeltaMaxBitWidthAndNumberOfValuesInSequence,
>+    DeltaEncodingCompressionQualityTest,
>+    ::testing::Combine(
>+        ::testing::Values(DeltaSignedness::kNoOverride,
>+                          DeltaSignedness::kForceUnsigned,
>+                          DeltaSignedness::kForceSigned),
>+        ::testing::Values(1, 4, 8, 15, 16, 17, 31, 32, 33, 63, 64),
>+        ::testing::Values(1, 2, 100, 10000),
>+        ::testing::Values(11, 12, 13)));
>+
>+// Similar to DeltaEncodingTest, but instead of semi-surgically producing
>+// specific cases, produce large amount of semi-realistic inputs.
>+class DeltaEncodingFuzzerLikeTest
>+    : public ::testing::TestWithParam<
>+          std::tuple<DeltaSignedness, uint64_t, uint64_t, bool, uint64_t>> {
>+ public:
>+  DeltaEncodingFuzzerLikeTest()
>+      : signedness_(std::get<0>(GetParam())),
>+        delta_max_bit_width_(std::get<1>(GetParam())),
>+        num_of_values_(std::get<2>(GetParam())),
>+        optional_values_(std::get<3>(GetParam())),
>+        partial_random_seed_(std::get<4>(GetParam())) {
>+    MaybeSetSignedness(signedness_);
>+  }
>+
>+  ~DeltaEncodingFuzzerLikeTest() override = default;
>+
>+  // Running with the same seed for all variants would make all tests start
>+  // with the same sequence; avoid this by making the seed different.
>+  uint64_t Seed() const {
>+    // Multiply everything but by different primes to produce unique results.
>+    return 2 * static_cast<uint64_t>(signedness_) + 3 * delta_max_bit_width_ +
>+           5 * delta_max_bit_width_ + 7 * num_of_values_ +
>+           11 * static_cast<uint64_t>(optional_values_) +
>+           13 * partial_random_seed_;
>+  }
>+
>+  const DeltaSignedness signedness_;
>+  const uint64_t delta_max_bit_width_;
>+  const uint64_t num_of_values_;
>+  const bool optional_values_;
>+  const uint64_t partial_random_seed_;  // Explained where it's used.
>+};
>+
>+TEST_P(DeltaEncodingFuzzerLikeTest, Test) {
>+  const absl::optional<uint64_t> base(3432);
>+
>+  Random prng(Seed());
>+  std::vector<absl::optional<uint64_t>> deltas(num_of_values_);
>+  for (size_t i = 0; i < deltas.size(); ++i) {
>+    if (!optional_values_ || prng.Rand<bool>()) {
>+      deltas[i] = RandomWithMaxBitWidth(&prng, delta_max_bit_width_);
>+    }
>+  }
>+  const auto values =
>+      CreateSequenceByOptionalDeltas(base.value(), deltas, num_of_values_);
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+INSTANTIATE_TEST_CASE_P(
>+    SignednessOverrideAndDeltaMaxBitWidthAndNumberOfValuesInSequence,
>+    DeltaEncodingFuzzerLikeTest,
>+    ::testing::Combine(
>+        ::testing::Values(DeltaSignedness::kNoOverride,
>+                          DeltaSignedness::kForceUnsigned,
>+                          DeltaSignedness::kForceSigned),
>+        ::testing::Values(1, 4, 8, 15, 16, 17, 31, 32, 33, 63, 64),
>+        ::testing::Values(1, 2, 100, 10000),
>+        ::testing::Bool(),
>+        ::testing::Values(21, 22, 23)));
>+
>+class DeltaEncodingSpecificEdgeCasesTest
>+    : public ::testing::TestWithParam<
>+          std::tuple<DeltaSignedness, uint64_t, bool>> {
>+ public:
>+  DeltaEncodingSpecificEdgeCasesTest() {
>+    UnsetFixedLengthEncoderDeltaSignednessForTesting();
>+  }
>+
>+  ~DeltaEncodingSpecificEdgeCasesTest() override = default;
>+};
>+
>+// This case is special because it produces identical forward/backward deltas.
>+TEST_F(DeltaEncodingSpecificEdgeCasesTest, SignedDeltaWithOnlyTopBitOn) {
>+  MaybeSetSignedness(DeltaSignedness::kForceSigned);
>+
>+  const absl::optional<uint64_t> base(3432);
>+
>+  const uint64_t delta = static_cast<uint64_t>(1) << 63;
>+  const std::vector<absl::optional<uint64_t>> values = {base.value() + delta};
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+TEST_F(DeltaEncodingSpecificEdgeCasesTest, MaximumUnsignedDelta) {
>+  MaybeSetSignedness(DeltaSignedness::kForceUnsigned);
>+
>+  const absl::optional<uint64_t> base((static_cast<uint64_t>(1) << 63) + 0x123);
>+
>+  const std::vector<absl::optional<uint64_t>> values = {base.value() - 1};
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+// Check that, if all deltas are set to -1, things still work.
>+TEST_P(DeltaEncodingSpecificEdgeCasesTest, ReverseSequence) {
>+  MaybeSetSignedness(std::get<0>(GetParam()));
>+  const uint64_t width = std::get<1>(GetParam());
>+  const bool wrap_around = std::get<2>(GetParam());
>+
>+  const uint64_t value_mask = (width == 64)
>+                                  ? std::numeric_limits<uint64_t>::max()
>+                                  : ((static_cast<uint64_t>(1) << width) - 1);
>+
>+  const uint64_t base = wrap_around ? 1u : (0xf82d3 & value_mask);
>+  const std::vector<absl::optional<uint64_t>> values = {
>+      (base - 1u) & value_mask, (base - 2u) & value_mask,
>+      (base - 3u) & value_mask};
>+
>+  TestEncodingAndDecoding(base, values);
>+}
>+
>+INSTANTIATE_TEST_CASE_P(
>+    _,
>+    DeltaEncodingSpecificEdgeCasesTest,
>+    ::testing::Combine(
>+        ::testing::Values(DeltaSignedness::kNoOverride,
>+                          DeltaSignedness::kForceUnsigned,
>+                          DeltaSignedness::kForceSigned),
>+        ::testing::Values(1, 4, 8, 15, 16, 17, 31, 32, 33, 63, 64),
>+        ::testing::Bool()));
>+
>+}  // namespace
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h
>index beb711c9dd82f72b95f5dc85f102e1648116f8bb..6ce750f67e760a1dc27ead8f6d8fbeaa2e36ee86 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h
>@@ -22,7 +22,8 @@ class RtcEventLogEncoder {
>  public:
>   virtual ~RtcEventLogEncoder() = default;
> 
>-  virtual std::string EncodeLogStart(int64_t timestamp_us) = 0;
>+  virtual std::string EncodeLogStart(int64_t timestamp_us,
>+                                     int64_t utc_time_us) = 0;
>   virtual std::string EncodeLogEnd(int64_t timestamp_us) = 0;
> 
>   virtual std::string EncodeBatch(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_common.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_common.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..7aea47611d6c3da6c55735b296f7b19ec9bb43ca
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_common.cc
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h"
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace {
>+// We use 0x3fff because that gives decent precision (compared to the underlying
>+// measurement producing the packet loss fraction) on the one hand, while
>+// allowing us to use no more than 2 bytes in varint form on the other hand.
>+// (We might also fixed-size encode using at most 14 bits.)
>+constexpr uint32_t kPacketLossFractionRange = (1 << 14) - 1;  // 0x3fff
>+constexpr float kPacketLossFractionRangeFloat =
>+    static_cast<float>(kPacketLossFractionRange);
>+}  // namespace
>+
>+uint32_t ConvertPacketLossFractionToProtoFormat(float packet_loss_fraction) {
>+  RTC_DCHECK_GE(packet_loss_fraction, 0);
>+  RTC_DCHECK_LE(packet_loss_fraction, 1);
>+  return static_cast<uint32_t>(packet_loss_fraction * kPacketLossFractionRange);
>+}
>+
>+bool ParsePacketLossFractionFromProtoFormat(uint32_t proto_packet_loss_fraction,
>+                                            float* output) {
>+  if (proto_packet_loss_fraction >= kPacketLossFractionRange) {
>+    return false;
>+  }
>+  *output = proto_packet_loss_fraction / kPacketLossFractionRangeFloat;
>+  return true;
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..429f8ed2ad73fc5a9eaaa76fcda120f5cb52c699
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h
>@@ -0,0 +1,93 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_COMMON_H_
>+#define LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_COMMON_H_
>+
>+#include <stdint.h>
>+
>+#include <limits>
>+#include <type_traits>
>+
>+namespace webrtc {
>+
>+// Convert between the packet fraction loss (a floating point number in
>+// the range [0.0, 1.0]), and a uint32_t with up to a fixed number of bits.
>+// The latter can be more efficiently stored in a protobuf and/or delta-encoded.
>+uint32_t ConvertPacketLossFractionToProtoFormat(float packet_loss_fraction);
>+bool ParsePacketLossFractionFromProtoFormat(uint32_t proto_packet_loss_fraction,
>+                                            float* output);
>+
>+}  // namespace webrtc
>+
>+namespace webrtc_event_logging {
>+
>+// Produce an unsigned representation of a signed integer. On two's complement
>+// machines, this is equivalent to:
>+// static_cast<uint64_t>(static_cast<std::make_unsigned<T>>(y))
>+template <typename T>
>+uint64_t ToUnsigned(T y) {
>+  static_assert(std::is_integral<T>::value, "");
>+  static_assert(std::is_signed<T>::value, "");
>+
>+  // Note that a signed integer whose width is N bits, has N-1 digits.
>+  static_assert(std::numeric_limits<T>::digits < 64, "");
>+
>+  constexpr T MIN_T = std::numeric_limits<T>::min();
>+  constexpr T MAX_T = std::numeric_limits<T>::max();
>+
>+  static_assert(MAX_T + MIN_T + 1 >= 0, "MAX_T >= abs(MIN_T) - 1");
>+
>+  if (y >= 0) {
>+    return static_cast<uint64_t>(y);
>+  } else {
>+    // y is in the range [MIN_T, -1], so (y - MIN_T) is in the
>+    // range [0, abs(MIN_T) - 1]. This is representable in a T
>+    // because MAX_T >= abs(MIN_T) - 1, as per the static_assert above.
>+    return static_cast<uint64_t>(MAX_T) + 1 + static_cast<uint64_t>(y - MIN_T);
>+  }
>+}
>+
>+// Assuming x = ToUnsigned(y), return |y|.
>+// Note: static_cast<T>(x) would work on most platforms and compilers, but
>+// involves undefined behavior. This function is well-defined, and can be
>+// optimized to a noop for 64 bit types, or a few arithmetic
>+// instructions and a single conditional jump for narrower types.
>+template <typename T>
>+bool ToSigned(uint64_t x, T* y) {
>+  static_assert(std::is_integral<T>::value, "");
>+  static_assert(std::is_signed<T>::value, "");
>+
>+  // Note that a signed integer whose width is N bits, has N-1 digits.
>+  static_assert(std::numeric_limits<T>::digits < 64, "");
>+
>+  constexpr T MIN_T = std::numeric_limits<T>::min();
>+  constexpr T MAX_T = std::numeric_limits<T>::max();
>+
>+  using UNSIGNED_T = typename std::make_unsigned<T>::type;
>+  constexpr auto MAX_UNSIGNED_T = std::numeric_limits<UNSIGNED_T>::max();
>+  if (x > static_cast<uint64_t>(MAX_UNSIGNED_T)) {
>+    return false;  // |x| cannot be represented using a T.
>+  }
>+
>+  if (x <= static_cast<uint64_t>(MAX_T)) {
>+    // The original value was positive, so it is safe to just static_cast.
>+    *y = static_cast<T>(x);
>+  } else {  // x > static_cast<uint64_t>(MAX_T)
>+    const uint64_t neg_x = x - static_cast<uint64_t>(MAX_T) - 1;
>+    *y = static_cast<T>(neg_x) + MIN_T;
>+  }
>+
>+  return true;
>+}
>+
>+}  // namespace webrtc_event_logging
>+
>+#endif  // LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_COMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_common_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_common_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..a7d94406ae0d2f6e1cb1e656a71f8b471e83fcdb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_common_unittest.cc
>@@ -0,0 +1,83 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h"
>+
>+#include <limits>
>+#include <type_traits>
>+#include <vector>
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc_event_logging {
>+namespace {
>+
>+template <typename T>
>+class SignednessConversionTest : public testing::Test {
>+ public:
>+  static_assert(std::is_integral<T>::value, "");
>+  static_assert(std::is_signed<T>::value, "");
>+};
>+
>+TYPED_TEST_CASE_P(SignednessConversionTest);
>+
>+TYPED_TEST_P(SignednessConversionTest, CorrectlyConvertsLegalValues) {
>+  using T = TypeParam;
>+  std::vector<T> legal_values = {std::numeric_limits<T>::min(),
>+                                 std::numeric_limits<T>::min() + 1,
>+                                 -1,
>+                                 0,
>+                                 1,
>+                                 std::numeric_limits<T>::max() - 1,
>+                                 std::numeric_limits<T>::max()};
>+  for (T val : legal_values) {
>+    const auto unsigned_val = ToUnsigned(val);
>+    T signed_val;
>+    ASSERT_TRUE(ToSigned<T>(unsigned_val, &signed_val))
>+        << "Failed on " << static_cast<uint64_t>(unsigned_val) << ".";
>+    EXPECT_EQ(val, signed_val)
>+        << "Failed on " << static_cast<uint64_t>(unsigned_val) << ".";
>+  }
>+}
>+
>+TYPED_TEST_P(SignednessConversionTest, FailsOnConvertingIllegalValues) {
>+  using T = TypeParam;
>+
>+  // Note that a signed integer whose width is N bits, has N-1 digits.
>+  constexpr bool width_is_64 = std::numeric_limits<T>::digits == 63;
>+
>+  if (width_is_64) {
>+    return;  // Test irrelevant; illegal values do not exist.
>+  }
>+
>+  const uint64_t max_legal_value = ToUnsigned(static_cast<T>(-1));
>+
>+  const std::vector<uint64_t> illegal_values = {
>+      max_legal_value + 1u, max_legal_value + 2u,
>+      std::numeric_limits<uint64_t>::max() - 1u,
>+      std::numeric_limits<uint64_t>::max()};
>+
>+  for (uint64_t unsigned_val : illegal_values) {
>+    T signed_val;
>+    EXPECT_FALSE(ToSigned<T>(unsigned_val, &signed_val))
>+        << "Failed on " << static_cast<uint64_t>(unsigned_val) << ".";
>+  }
>+}
>+
>+REGISTER_TYPED_TEST_CASE_P(SignednessConversionTest,
>+                           CorrectlyConvertsLegalValues,
>+                           FailsOnConvertingIllegalValues);
>+
>+using Types = ::testing::Types<int8_t, int16_t, int32_t, int64_t>;
>+
>+INSTANTIATE_TYPED_TEST_CASE_P(_, SignednessConversionTest, Types);
>+
>+}  // namespace
>+}  // namespace webrtc_event_logging
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
>index 7ef8d677c609072e106b04e7251d6b2a48cee8d0..29636ede922f2d44928e3730d1e25e6c2ab7f6b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
>@@ -47,7 +47,6 @@
> #include "rtc_base/ignore_wundef.h"
> #include "rtc_base/logging.h"
> 
>-#ifdef ENABLE_RTC_EVENT_LOG
> 
> // *.pb.h files are generated at build-time by the protobuf compiler.
> RTC_PUSH_IGNORING_WUNDEF()
>@@ -223,7 +222,8 @@ ConvertIceCandidatePairEventType(IceCandidatePairEventType type) {
> 
> }  // namespace
> 
>-std::string RtcEventLogEncoderLegacy::EncodeLogStart(int64_t timestamp_us) {
>+std::string RtcEventLogEncoderLegacy::EncodeLogStart(int64_t timestamp_us,
>+                                                     int64_t utc_time_us) {
>   rtclog::Event rtclog_event;
>   rtclog_event.set_timestamp_us(timestamp_us);
>   rtclog_event.set_type(rtclog::Event::LOG_START);
>@@ -357,37 +357,37 @@ std::string RtcEventLogEncoderLegacy::Encode(const RtcEvent& event) {
> std::string RtcEventLogEncoderLegacy::EncodeAlrState(
>     const RtcEventAlrState& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::ALR_STATE_EVENT);
> 
>   auto* alr_state = rtclog_event.mutable_alr_state();
>-  alr_state->set_in_alr(event.in_alr_);
>+  alr_state->set_in_alr(event.in_alr());
>   return Serialize(&rtclog_event);
> }
> 
> std::string RtcEventLogEncoderLegacy::EncodeAudioNetworkAdaptation(
>     const RtcEventAudioNetworkAdaptation& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
> 
>   auto* audio_network_adaptation =
>       rtclog_event.mutable_audio_network_adaptation();
>-  if (event.config_->bitrate_bps)
>-    audio_network_adaptation->set_bitrate_bps(*event.config_->bitrate_bps);
>-  if (event.config_->frame_length_ms)
>+  if (event.config().bitrate_bps)
>+    audio_network_adaptation->set_bitrate_bps(*event.config().bitrate_bps);
>+  if (event.config().frame_length_ms)
>     audio_network_adaptation->set_frame_length_ms(
>-        *event.config_->frame_length_ms);
>-  if (event.config_->uplink_packet_loss_fraction) {
>+        *event.config().frame_length_ms);
>+  if (event.config().uplink_packet_loss_fraction) {
>     audio_network_adaptation->set_uplink_packet_loss_fraction(
>-        *event.config_->uplink_packet_loss_fraction);
>+        *event.config().uplink_packet_loss_fraction);
>   }
>-  if (event.config_->enable_fec)
>-    audio_network_adaptation->set_enable_fec(*event.config_->enable_fec);
>-  if (event.config_->enable_dtx)
>-    audio_network_adaptation->set_enable_dtx(*event.config_->enable_dtx);
>-  if (event.config_->num_channels)
>-    audio_network_adaptation->set_num_channels(*event.config_->num_channels);
>+  if (event.config().enable_fec)
>+    audio_network_adaptation->set_enable_fec(*event.config().enable_fec);
>+  if (event.config().enable_dtx)
>+    audio_network_adaptation->set_enable_dtx(*event.config().enable_dtx);
>+  if (event.config().num_channels)
>+    audio_network_adaptation->set_num_channels(*event.config().num_channels);
> 
>   return Serialize(&rtclog_event);
> }
>@@ -395,11 +395,11 @@ std::string RtcEventLogEncoderLegacy::EncodeAudioNetworkAdaptation(
> std::string RtcEventLogEncoderLegacy::EncodeAudioPlayout(
>     const RtcEventAudioPlayout& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT);
> 
>   auto* playout_event = rtclog_event.mutable_audio_playout_event();
>-  playout_event->set_local_ssrc(event.ssrc_);
>+  playout_event->set_local_ssrc(event.ssrc());
> 
>   return Serialize(&rtclog_event);
> }
>@@ -407,15 +407,15 @@ std::string RtcEventLogEncoderLegacy::EncodeAudioPlayout(
> std::string RtcEventLogEncoderLegacy::EncodeAudioReceiveStreamConfig(
>     const RtcEventAudioReceiveStreamConfig& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
> 
>   rtclog::AudioReceiveConfig* receiver_config =
>       rtclog_event.mutable_audio_receiver_config();
>-  receiver_config->set_remote_ssrc(event.config_->remote_ssrc);
>-  receiver_config->set_local_ssrc(event.config_->local_ssrc);
>+  receiver_config->set_remote_ssrc(event.config().remote_ssrc);
>+  receiver_config->set_local_ssrc(event.config().local_ssrc);
> 
>-  for (const auto& e : event.config_->rtp_extensions) {
>+  for (const auto& e : event.config().rtp_extensions) {
>     rtclog::RtpHeaderExtension* extension =
>         receiver_config->add_header_extensions();
>     extension->set_name(e.uri);
>@@ -428,15 +428,15 @@ std::string RtcEventLogEncoderLegacy::EncodeAudioReceiveStreamConfig(
> std::string RtcEventLogEncoderLegacy::EncodeAudioSendStreamConfig(
>     const RtcEventAudioSendStreamConfig& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
> 
>   rtclog::AudioSendConfig* sender_config =
>       rtclog_event.mutable_audio_sender_config();
> 
>-  sender_config->set_ssrc(event.config_->local_ssrc);
>+  sender_config->set_ssrc(event.config().local_ssrc);
> 
>-  for (const auto& e : event.config_->rtp_extensions) {
>+  for (const auto& e : event.config().rtp_extensions) {
>     rtclog::RtpHeaderExtension* extension =
>         sender_config->add_header_extensions();
>     extension->set_name(e.uri);
>@@ -449,12 +449,12 @@ std::string RtcEventLogEncoderLegacy::EncodeAudioSendStreamConfig(
> std::string RtcEventLogEncoderLegacy::EncodeBweUpdateDelayBased(
>     const RtcEventBweUpdateDelayBased& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::DELAY_BASED_BWE_UPDATE);
> 
>   auto* bwe_event = rtclog_event.mutable_delay_based_bwe_update();
>-  bwe_event->set_bitrate_bps(event.bitrate_bps_);
>-  bwe_event->set_detector_state(ConvertDetectorState(event.detector_state_));
>+  bwe_event->set_bitrate_bps(event.bitrate_bps());
>+  bwe_event->set_detector_state(ConvertDetectorState(event.detector_state()));
> 
>   return Serialize(&rtclog_event);
> }
>@@ -462,13 +462,13 @@ std::string RtcEventLogEncoderLegacy::EncodeBweUpdateDelayBased(
> std::string RtcEventLogEncoderLegacy::EncodeBweUpdateLossBased(
>     const RtcEventBweUpdateLossBased& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::LOSS_BASED_BWE_UPDATE);
> 
>   auto* bwe_event = rtclog_event.mutable_loss_based_bwe_update();
>-  bwe_event->set_bitrate_bps(event.bitrate_bps_);
>-  bwe_event->set_fraction_loss(event.fraction_loss_);
>-  bwe_event->set_total_packets(event.total_packets_);
>+  bwe_event->set_bitrate_bps(event.bitrate_bps());
>+  bwe_event->set_fraction_loss(event.fraction_loss());
>+  bwe_event->set_total_packets(event.total_packets());
> 
>   return Serialize(&rtclog_event);
> }
>@@ -476,15 +476,15 @@ std::string RtcEventLogEncoderLegacy::EncodeBweUpdateLossBased(
> std::string RtcEventLogEncoderLegacy::EncodeIceCandidatePairConfig(
>     const RtcEventIceCandidatePairConfig& event) {
>   rtclog::Event encoded_rtc_event;
>-  encoded_rtc_event.set_timestamp_us(event.timestamp_us_);
>+  encoded_rtc_event.set_timestamp_us(event.timestamp_us());
>   encoded_rtc_event.set_type(rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG);
> 
>   auto* encoded_ice_event =
>       encoded_rtc_event.mutable_ice_candidate_pair_config();
>   encoded_ice_event->set_config_type(
>-      ConvertIceCandidatePairConfigType(event.type_));
>-  encoded_ice_event->set_candidate_pair_id(event.candidate_pair_id_);
>-  const auto& desc = event.candidate_pair_desc_;
>+      ConvertIceCandidatePairConfigType(event.type()));
>+  encoded_ice_event->set_candidate_pair_id(event.candidate_pair_id());
>+  const auto& desc = event.candidate_pair_desc();
>   encoded_ice_event->set_local_candidate_type(
>       ConvertIceCandidateType(desc.local_candidate_type));
>   encoded_ice_event->set_local_relay_protocol(
>@@ -505,28 +505,28 @@ std::string RtcEventLogEncoderLegacy::EncodeIceCandidatePairConfig(
> std::string RtcEventLogEncoderLegacy::EncodeIceCandidatePairEvent(
>     const RtcEventIceCandidatePair& event) {
>   rtclog::Event encoded_rtc_event;
>-  encoded_rtc_event.set_timestamp_us(event.timestamp_us_);
>+  encoded_rtc_event.set_timestamp_us(event.timestamp_us());
>   encoded_rtc_event.set_type(rtclog::Event::ICE_CANDIDATE_PAIR_EVENT);
> 
>   auto* encoded_ice_event =
>       encoded_rtc_event.mutable_ice_candidate_pair_event();
>   encoded_ice_event->set_event_type(
>-      ConvertIceCandidatePairEventType(event.type_));
>-  encoded_ice_event->set_candidate_pair_id(event.candidate_pair_id_);
>+      ConvertIceCandidatePairEventType(event.type()));
>+  encoded_ice_event->set_candidate_pair_id(event.candidate_pair_id());
>   return Serialize(&encoded_rtc_event);
> }
> 
> std::string RtcEventLogEncoderLegacy::EncodeProbeClusterCreated(
>     const RtcEventProbeClusterCreated& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT);
> 
>   auto* probe_cluster = rtclog_event.mutable_probe_cluster();
>-  probe_cluster->set_id(event.id_);
>-  probe_cluster->set_bitrate_bps(event.bitrate_bps_);
>-  probe_cluster->set_min_packets(event.min_probes_);
>-  probe_cluster->set_min_bytes(event.min_bytes_);
>+  probe_cluster->set_id(event.id());
>+  probe_cluster->set_bitrate_bps(event.bitrate_bps());
>+  probe_cluster->set_min_packets(event.min_probes());
>+  probe_cluster->set_min_bytes(event.min_bytes());
> 
>   return Serialize(&rtclog_event);
> }
>@@ -534,12 +534,12 @@ std::string RtcEventLogEncoderLegacy::EncodeProbeClusterCreated(
> std::string RtcEventLogEncoderLegacy::EncodeProbeResultFailure(
>     const RtcEventProbeResultFailure& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT);
> 
>   auto* probe_result = rtclog_event.mutable_probe_result();
>-  probe_result->set_id(event.id_);
>-  probe_result->set_result(ConvertProbeResultType(event.failure_reason_));
>+  probe_result->set_id(event.id());
>+  probe_result->set_result(ConvertProbeResultType(event.failure_reason()));
> 
>   return Serialize(&rtclog_event);
> }
>@@ -547,70 +547,71 @@ std::string RtcEventLogEncoderLegacy::EncodeProbeResultFailure(
> std::string RtcEventLogEncoderLegacy::EncodeProbeResultSuccess(
>     const RtcEventProbeResultSuccess& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT);
> 
>   auto* probe_result = rtclog_event.mutable_probe_result();
>-  probe_result->set_id(event.id_);
>+  probe_result->set_id(event.id());
>   probe_result->set_result(rtclog::BweProbeResult::SUCCESS);
>-  probe_result->set_bitrate_bps(event.bitrate_bps_);
>+  probe_result->set_bitrate_bps(event.bitrate_bps());
> 
>   return Serialize(&rtclog_event);
> }
> 
> std::string RtcEventLogEncoderLegacy::EncodeRtcpPacketIncoming(
>     const RtcEventRtcpPacketIncoming& event) {
>-  return EncodeRtcpPacket(event.timestamp_us_, event.packet_, true);
>+  return EncodeRtcpPacket(event.timestamp_us(), event.packet(), true);
> }
> 
> std::string RtcEventLogEncoderLegacy::EncodeRtcpPacketOutgoing(
>     const RtcEventRtcpPacketOutgoing& event) {
>-  return EncodeRtcpPacket(event.timestamp_us_, event.packet_, false);
>+  return EncodeRtcpPacket(event.timestamp_us(), event.packet(), false);
> }
> 
> std::string RtcEventLogEncoderLegacy::EncodeRtpPacketIncoming(
>     const RtcEventRtpPacketIncoming& event) {
>-  return EncodeRtpPacket(event.timestamp_us_, event.header_,
>-                         event.packet_length_, PacedPacketInfo::kNotAProbe,
>+  return EncodeRtpPacket(event.timestamp_us(), event.header(),
>+                         event.packet_length(), PacedPacketInfo::kNotAProbe,
>                          true);
> }
> 
> std::string RtcEventLogEncoderLegacy::EncodeRtpPacketOutgoing(
>     const RtcEventRtpPacketOutgoing& event) {
>-  return EncodeRtpPacket(event.timestamp_us_, event.header_,
>-                         event.packet_length_, event.probe_cluster_id_, false);
>+  return EncodeRtpPacket(event.timestamp_us(), event.header(),
>+                         event.packet_length(), event.probe_cluster_id(),
>+                         false);
> }
> 
> std::string RtcEventLogEncoderLegacy::EncodeVideoReceiveStreamConfig(
>     const RtcEventVideoReceiveStreamConfig& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
> 
>   rtclog::VideoReceiveConfig* receiver_config =
>       rtclog_event.mutable_video_receiver_config();
>-  receiver_config->set_remote_ssrc(event.config_->remote_ssrc);
>-  receiver_config->set_local_ssrc(event.config_->local_ssrc);
>+  receiver_config->set_remote_ssrc(event.config().remote_ssrc);
>+  receiver_config->set_local_ssrc(event.config().local_ssrc);
> 
>   // TODO(perkj): Add field for rsid.
>-  receiver_config->set_rtcp_mode(ConvertRtcpMode(event.config_->rtcp_mode));
>-  receiver_config->set_remb(event.config_->remb);
>+  receiver_config->set_rtcp_mode(ConvertRtcpMode(event.config().rtcp_mode));
>+  receiver_config->set_remb(event.config().remb);
> 
>-  for (const auto& e : event.config_->rtp_extensions) {
>+  for (const auto& e : event.config().rtp_extensions) {
>     rtclog::RtpHeaderExtension* extension =
>         receiver_config->add_header_extensions();
>     extension->set_name(e.uri);
>     extension->set_id(e.id);
>   }
> 
>-  for (const auto& d : event.config_->codecs) {
>+  for (const auto& d : event.config().codecs) {
>     rtclog::DecoderConfig* decoder = receiver_config->add_decoders();
>     decoder->set_name(d.payload_name);
>     decoder->set_payload_type(d.payload_type);
>     if (d.rtx_payload_type != 0) {
>       rtclog::RtxMap* rtx = receiver_config->add_rtx_map();
>       rtx->set_payload_type(d.payload_type);
>-      rtx->mutable_config()->set_rtx_ssrc(event.config_->rtx_ssrc);
>+      rtx->mutable_config()->set_rtx_ssrc(event.config().rtx_ssrc);
>       rtx->mutable_config()->set_rtx_payload_type(d.rtx_payload_type);
>     }
>   }
>@@ -621,19 +622,19 @@ std::string RtcEventLogEncoderLegacy::EncodeVideoReceiveStreamConfig(
> std::string RtcEventLogEncoderLegacy::EncodeVideoSendStreamConfig(
>     const RtcEventVideoSendStreamConfig& event) {
>   rtclog::Event rtclog_event;
>-  rtclog_event.set_timestamp_us(event.timestamp_us_);
>+  rtclog_event.set_timestamp_us(event.timestamp_us());
>   rtclog_event.set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
> 
>   rtclog::VideoSendConfig* sender_config =
>       rtclog_event.mutable_video_sender_config();
> 
>   // TODO(perkj): rtclog::VideoSendConfig should only contain one SSRC.
>-  sender_config->add_ssrcs(event.config_->local_ssrc);
>-  if (event.config_->rtx_ssrc != 0) {
>-    sender_config->add_rtx_ssrcs(event.config_->rtx_ssrc);
>+  sender_config->add_ssrcs(event.config().local_ssrc);
>+  if (event.config().rtx_ssrc != 0) {
>+    sender_config->add_rtx_ssrcs(event.config().rtx_ssrc);
>   }
> 
>-  for (const auto& e : event.config_->rtp_extensions) {
>+  for (const auto& e : event.config().rtp_extensions) {
>     rtclog::RtpHeaderExtension* extension =
>         sender_config->add_header_extensions();
>     extension->set_name(e.uri);
>@@ -642,13 +643,13 @@ std::string RtcEventLogEncoderLegacy::EncodeVideoSendStreamConfig(
> 
>   // TODO(perkj): rtclog::VideoSendConfig should contain many possible codec
>   // configurations.
>-  for (const auto& codec : event.config_->codecs) {
>+  for (const auto& codec : event.config().codecs) {
>     sender_config->set_rtx_payload_type(codec.rtx_payload_type);
>     rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
>     encoder->set_name(codec.payload_name);
>     encoder->set_payload_type(codec.payload_type);
> 
>-    if (event.config_->codecs.size() > 1) {
>+    if (event.config().codecs.size() > 1) {
>       RTC_LOG(WARNING)
>           << "LogVideoSendStreamConfig currently only supports one "
>           << "codec. Logging codec :" << codec.payload_name;
>@@ -756,5 +757,3 @@ std::string RtcEventLogEncoderLegacy::Serialize(rtclog::Event* event) {
> }
> 
> }  // namespace webrtc
>-
>-#endif  // ENABLE_RTC_EVENT_LOG
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
>index 87db039e3aff3f16eb9caeac5722e11567da55d0..3105dc1e68c6b938ea890c7d261b9f2766702bf7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
>@@ -18,8 +18,6 @@
> #include "logging/rtc_event_log/encoder/rtc_event_log_encoder.h"
> #include "rtc_base/buffer.h"
> 
>-#if defined(ENABLE_RTC_EVENT_LOG)
>-
> namespace webrtc {
> 
> namespace rtclog {
>@@ -52,7 +50,8 @@ class RtcEventLogEncoderLegacy final : public RtcEventLogEncoder {
>  public:
>   ~RtcEventLogEncoderLegacy() override = default;
> 
>-  std::string EncodeLogStart(int64_t timestamp_us) override;
>+  std::string EncodeLogStart(int64_t timestamp_us,
>+                             int64_t utc_time_us) override;
>   std::string EncodeLogEnd(int64_t timestamp_us) override;
> 
>   std::string EncodeBatch(
>@@ -105,6 +104,4 @@ class RtcEventLogEncoderLegacy final : public RtcEventLogEncoder {
> 
> }  // namespace webrtc
> 
>-#endif  // ENABLE_RTC_EVENT_LOG
>-
> #endif  // LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_LEGACY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..79dc379f48b41a460d5f462172c214f1227598c2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
>@@ -0,0 +1,1336 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h"
>+
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "logging/rtc_event_log/encoder/blob_encoding.h"
>+#include "logging/rtc_event_log/encoder/delta_encoding.h"
>+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h"
>+#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
>+#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
>+#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
>+#include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
>+#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
>+#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
>+#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
>+#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
>+#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
>+#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
>+#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
>+#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
>+#include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
>+#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
>+#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
>+#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
>+#include "logging/rtc_event_log/rtc_stream_config.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
>+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>+#include "modules/rtp_rtcp/include/rtp_cvo.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
>+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
>+#include "modules/rtp_rtcp/source/rtp_packet.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/ignore_wundef.h"
>+#include "rtc_base/logging.h"
>+
>+// *.pb.h files are generated at build-time by the protobuf compiler.
>+RTC_PUSH_IGNORING_WUNDEF()
>+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
>+#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log2.pb.h"
>+#else
>+#include "logging/rtc_event_log/rtc_event_log2.pb.h"
>+#endif
>+RTC_POP_IGNORING_WUNDEF()
>+
>+using webrtc_event_logging::ToUnsigned;
>+
>+namespace webrtc {
>+
>+namespace {
>+rtclog2::DelayBasedBweUpdates::DetectorState ConvertToProtoFormat(
>+    BandwidthUsage state) {
>+  switch (state) {
>+    case BandwidthUsage::kBwNormal:
>+      return rtclog2::DelayBasedBweUpdates::BWE_NORMAL;
>+    case BandwidthUsage::kBwUnderusing:
>+      return rtclog2::DelayBasedBweUpdates::BWE_UNDERUSING;
>+    case BandwidthUsage::kBwOverusing:
>+      return rtclog2::DelayBasedBweUpdates::BWE_OVERUSING;
>+    case BandwidthUsage::kLast:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog2::DelayBasedBweUpdates::BWE_UNKNOWN_STATE;
>+}
>+
>+rtclog2::BweProbeResultFailure::FailureReason ConvertToProtoFormat(
>+    ProbeFailureReason failure_reason) {
>+  switch (failure_reason) {
>+    case ProbeFailureReason::kInvalidSendReceiveInterval:
>+      return rtclog2::BweProbeResultFailure::INVALID_SEND_RECEIVE_INTERVAL;
>+    case ProbeFailureReason::kInvalidSendReceiveRatio:
>+      return rtclog2::BweProbeResultFailure::INVALID_SEND_RECEIVE_RATIO;
>+    case ProbeFailureReason::kTimeout:
>+      return rtclog2::BweProbeResultFailure::TIMEOUT;
>+    case ProbeFailureReason::kLast:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog2::BweProbeResultFailure::UNKNOWN;
>+}
>+
>+// Returns true if there are recognized extensions that we should log
>+// and false if there are no extensions or all extensions are types we don't
>+// log. The protobuf representation of the header configs is written to
>+// |proto_config|.
>+bool ConvertToProtoFormat(const std::vector<RtpExtension>& extensions,
>+                          rtclog2::RtpHeaderExtensionConfig* proto_config) {
>+  size_t unknown_extensions = 0;
>+  for (auto& extension : extensions) {
>+    if (extension.uri == RtpExtension::kAudioLevelUri) {
>+      proto_config->set_audio_level_id(extension.id);
>+    } else if (extension.uri == RtpExtension::kTimestampOffsetUri) {
>+      proto_config->set_transmission_time_offset_id(extension.id);
>+    } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
>+      proto_config->set_absolute_send_time_id(extension.id);
>+    } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
>+      proto_config->set_transport_sequence_number_id(extension.id);
>+    } else if (extension.uri == RtpExtension::kVideoRotationUri) {
>+      proto_config->set_video_rotation_id(extension.id);
>+    } else {
>+      ++unknown_extensions;
>+    }
>+  }
>+  return unknown_extensions < extensions.size();
>+}
>+
>+rtclog2::IceCandidatePairConfig::IceCandidatePairConfigType
>+ConvertToProtoFormat(IceCandidatePairConfigType type) {
>+  switch (type) {
>+    case IceCandidatePairConfigType::kAdded:
>+      return rtclog2::IceCandidatePairConfig::ADDED;
>+    case IceCandidatePairConfigType::kUpdated:
>+      return rtclog2::IceCandidatePairConfig::UPDATED;
>+    case IceCandidatePairConfigType::kDestroyed:
>+      return rtclog2::IceCandidatePairConfig::DESTROYED;
>+    case IceCandidatePairConfigType::kSelected:
>+      return rtclog2::IceCandidatePairConfig::SELECTED;
>+    case IceCandidatePairConfigType::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog2::IceCandidatePairConfig::UNKNOWN_CONFIG_TYPE;
>+}
>+
>+rtclog2::IceCandidatePairConfig::IceCandidateType ConvertToProtoFormat(
>+    IceCandidateType type) {
>+  switch (type) {
>+    case IceCandidateType::kUnknown:
>+      return rtclog2::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE;
>+    case IceCandidateType::kLocal:
>+      return rtclog2::IceCandidatePairConfig::LOCAL;
>+    case IceCandidateType::kStun:
>+      return rtclog2::IceCandidatePairConfig::STUN;
>+    case IceCandidateType::kPrflx:
>+      return rtclog2::IceCandidatePairConfig::PRFLX;
>+    case IceCandidateType::kRelay:
>+      return rtclog2::IceCandidatePairConfig::RELAY;
>+    case IceCandidateType::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog2::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE;
>+}
>+
>+rtclog2::IceCandidatePairConfig::Protocol ConvertToProtoFormat(
>+    IceCandidatePairProtocol protocol) {
>+  switch (protocol) {
>+    case IceCandidatePairProtocol::kUnknown:
>+      return rtclog2::IceCandidatePairConfig::UNKNOWN_PROTOCOL;
>+    case IceCandidatePairProtocol::kUdp:
>+      return rtclog2::IceCandidatePairConfig::UDP;
>+    case IceCandidatePairProtocol::kTcp:
>+      return rtclog2::IceCandidatePairConfig::TCP;
>+    case IceCandidatePairProtocol::kSsltcp:
>+      return rtclog2::IceCandidatePairConfig::SSLTCP;
>+    case IceCandidatePairProtocol::kTls:
>+      return rtclog2::IceCandidatePairConfig::TLS;
>+    case IceCandidatePairProtocol::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog2::IceCandidatePairConfig::UNKNOWN_PROTOCOL;
>+}
>+
>+rtclog2::IceCandidatePairConfig::AddressFamily ConvertToProtoFormat(
>+    IceCandidatePairAddressFamily address_family) {
>+  switch (address_family) {
>+    case IceCandidatePairAddressFamily::kUnknown:
>+      return rtclog2::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY;
>+    case IceCandidatePairAddressFamily::kIpv4:
>+      return rtclog2::IceCandidatePairConfig::IPV4;
>+    case IceCandidatePairAddressFamily::kIpv6:
>+      return rtclog2::IceCandidatePairConfig::IPV6;
>+    case IceCandidatePairAddressFamily::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog2::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY;
>+}
>+
>+rtclog2::IceCandidatePairConfig::NetworkType ConvertToProtoFormat(
>+    IceCandidateNetworkType network_type) {
>+  switch (network_type) {
>+    case IceCandidateNetworkType::kUnknown:
>+      return rtclog2::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE;
>+    case IceCandidateNetworkType::kEthernet:
>+      return rtclog2::IceCandidatePairConfig::ETHERNET;
>+    case IceCandidateNetworkType::kLoopback:
>+      return rtclog2::IceCandidatePairConfig::LOOPBACK;
>+    case IceCandidateNetworkType::kWifi:
>+      return rtclog2::IceCandidatePairConfig::WIFI;
>+    case IceCandidateNetworkType::kVpn:
>+      return rtclog2::IceCandidatePairConfig::VPN;
>+    case IceCandidateNetworkType::kCellular:
>+      return rtclog2::IceCandidatePairConfig::CELLULAR;
>+    case IceCandidateNetworkType::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog2::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE;
>+}
>+
>+rtclog2::IceCandidatePairEvent::IceCandidatePairEventType ConvertToProtoFormat(
>+    IceCandidatePairEventType type) {
>+  switch (type) {
>+    case IceCandidatePairEventType::kCheckSent:
>+      return rtclog2::IceCandidatePairEvent::CHECK_SENT;
>+    case IceCandidatePairEventType::kCheckReceived:
>+      return rtclog2::IceCandidatePairEvent::CHECK_RECEIVED;
>+    case IceCandidatePairEventType::kCheckResponseSent:
>+      return rtclog2::IceCandidatePairEvent::CHECK_RESPONSE_SENT;
>+    case IceCandidatePairEventType::kCheckResponseReceived:
>+      return rtclog2::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED;
>+    case IceCandidatePairEventType::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog2::IceCandidatePairEvent::UNKNOWN_CHECK_TYPE;
>+}
>+
>+// Copies all RTCP blocks except APP, SDES and unknown from |packet| to
>+// |buffer|. |buffer| must have space for |IP_PACKET_SIZE| bytes. |packet| must
>+// be at most |IP_PACKET_SIZE| bytes long.
>+size_t RemoveNonWhitelistedRtcpBlocks(const rtc::Buffer& packet,
>+                                      uint8_t* buffer) {
>+  RTC_DCHECK(packet.size() <= IP_PACKET_SIZE);
>+  RTC_DCHECK(buffer != nullptr);
>+  rtcp::CommonHeader header;
>+  const uint8_t* block_begin = packet.data();
>+  const uint8_t* packet_end = packet.data() + packet.size();
>+  size_t buffer_length = 0;
>+  while (block_begin < packet_end) {
>+    if (!header.Parse(block_begin, packet_end - block_begin)) {
>+      break;  // Incorrect message header.
>+    }
>+    const uint8_t* next_block = header.NextPacket();
>+    RTC_DCHECK_GT(next_block, block_begin);
>+    RTC_DCHECK_LE(next_block, packet_end);
>+    size_t block_size = next_block - block_begin;
>+    switch (header.type()) {
>+      case rtcp::Bye::kPacketType:
>+      case rtcp::ExtendedJitterReport::kPacketType:
>+      case rtcp::ExtendedReports::kPacketType:
>+      case rtcp::Psfb::kPacketType:
>+      case rtcp::ReceiverReport::kPacketType:
>+      case rtcp::Rtpfb::kPacketType:
>+      case rtcp::SenderReport::kPacketType:
>+        // We log sender reports, receiver reports, bye messages
>+        // inter-arrival jitter, third-party loss reports, payload-specific
>+        // feedback and extended reports.
>+        // TODO(terelius): As an optimization, don't copy anything if all blocks
>+        // in the packet are whitelisted types.
>+        memcpy(buffer + buffer_length, block_begin, block_size);
>+        buffer_length += block_size;
>+        break;
>+      case rtcp::App::kPacketType:
>+      case rtcp::Sdes::kPacketType:
>+      default:
>+        // We don't log sender descriptions, application defined messages
>+        // or message blocks of unknown type.
>+        break;
>+    }
>+
>+    block_begin += block_size;
>+  }
>+  return buffer_length;
>+}
>+
>+template <typename EventType, typename ProtoType>
>+void EncodeRtcpPacket(rtc::ArrayView<const EventType*> batch,
>+                      ProtoType* proto_batch) {
>+  if (batch.size() == 0) {
>+    return;
>+  }
>+
>+  // Base event
>+  const EventType* const base_event = batch[0];
>+  proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+  {
>+    uint8_t buffer[IP_PACKET_SIZE];
>+    size_t buffer_length =
>+        RemoveNonWhitelistedRtcpBlocks(base_event->packet(), buffer);
>+    proto_batch->set_raw_packet(buffer, buffer_length);
>+  }
>+
>+  if (batch.size() == 1) {
>+    return;
>+  }
>+
>+  // Delta encoding
>+  proto_batch->set_number_of_deltas(batch.size() - 1);
>+  std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
>+  std::string encoded_deltas;
>+
>+  // timestamp_ms
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    values[i] = ToUnsigned(event->timestamp_ms());
>+  }
>+  encoded_deltas = EncodeDeltas(ToUnsigned(base_event->timestamp_ms()), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_timestamp_ms_deltas(encoded_deltas);
>+  }
>+
>+  // raw_packet
>+  std::vector<std::string> scrubed_packets(batch.size() - 1);
>+  for (size_t i = 0; i < scrubed_packets.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    scrubed_packets[i].resize(event->packet().size());
>+    static_assert(sizeof(std::string::value_type) == sizeof(uint8_t), "");
>+    const size_t buffer_length = RemoveNonWhitelistedRtcpBlocks(
>+        event->packet(), reinterpret_cast<uint8_t*>(&scrubed_packets[i][0]));
>+    if (buffer_length < event->packet().size()) {
>+      scrubed_packets[i].resize(buffer_length);
>+    }
>+  }
>+  proto_batch->set_raw_packet_blobs(EncodeBlobs(scrubed_packets));
>+}
>+
>+template <typename EventType, typename ProtoType>
>+void EncodeRtpPacket(const std::vector<const EventType*>& batch,
>+                     ProtoType* proto_batch) {
>+  if (batch.size() == 0) {
>+    return;
>+  }
>+
>+  // Base event
>+  const EventType* const base_event = batch[0];
>+  proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+  proto_batch->set_marker(base_event->header().Marker());
>+  // TODO(terelius): Is payload type needed?
>+  proto_batch->set_payload_type(base_event->header().PayloadType());
>+  proto_batch->set_sequence_number(base_event->header().SequenceNumber());
>+  proto_batch->set_rtp_timestamp(base_event->header().Timestamp());
>+  proto_batch->set_ssrc(base_event->header().Ssrc());
>+  proto_batch->set_payload_size(base_event->payload_length());
>+  proto_batch->set_header_size(base_event->header_length());
>+  proto_batch->set_padding_size(base_event->padding_length());
>+
>+  // Add header extensions (base event).
>+  absl::optional<uint64_t> base_transport_sequence_number;
>+  {
>+    uint16_t seqnum;
>+    if (base_event->header().template GetExtension<TransportSequenceNumber>(
>+            &seqnum)) {
>+      proto_batch->set_transport_sequence_number(seqnum);
>+      base_transport_sequence_number = seqnum;
>+    }
>+  }
>+
>+  absl::optional<uint64_t> unsigned_base_transmission_time_offset;
>+  {
>+    int32_t offset;
>+    if (base_event->header().template GetExtension<TransmissionOffset>(
>+            &offset)) {
>+      proto_batch->set_transmission_time_offset(offset);
>+      unsigned_base_transmission_time_offset = ToUnsigned(offset);
>+    }
>+  }
>+
>+  absl::optional<uint64_t> base_absolute_send_time;
>+  {
>+    uint32_t sendtime;
>+    if (base_event->header().template GetExtension<AbsoluteSendTime>(
>+            &sendtime)) {
>+      proto_batch->set_absolute_send_time(sendtime);
>+      base_absolute_send_time = sendtime;
>+    }
>+  }
>+
>+  absl::optional<uint64_t> base_video_rotation;
>+  {
>+    VideoRotation video_rotation;
>+    if (base_event->header().template GetExtension<VideoOrientation>(
>+            &video_rotation)) {
>+      proto_batch->set_video_rotation(
>+          ConvertVideoRotationToCVOByte(video_rotation));
>+      base_video_rotation = ConvertVideoRotationToCVOByte(video_rotation);
>+    }
>+  }
>+
>+  absl::optional<uint64_t> base_audio_level;
>+  absl::optional<uint64_t> base_voice_activity;
>+  {
>+    bool voice_activity;
>+    uint8_t audio_level;
>+    if (base_event->header().template GetExtension<AudioLevel>(&voice_activity,
>+                                                               &audio_level)) {
>+      RTC_DCHECK_LE(audio_level, 0x7Fu);
>+      base_audio_level = audio_level;
>+      proto_batch->set_audio_level(audio_level);
>+
>+      base_voice_activity = voice_activity;
>+      proto_batch->set_voice_activity(voice_activity);
>+    }
>+  }
>+
>+  if (batch.size() == 1) {
>+    return;
>+  }
>+
>+  // Delta encoding
>+  proto_batch->set_number_of_deltas(batch.size() - 1);
>+  std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
>+  std::string encoded_deltas;
>+
>+  // timestamp_ms (event)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    values[i] = ToUnsigned(event->timestamp_ms());
>+  }
>+  encoded_deltas = EncodeDeltas(ToUnsigned(base_event->timestamp_ms()), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_timestamp_ms_deltas(encoded_deltas);
>+  }
>+
>+  // marker (RTP base)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    values[i] = event->header().Marker();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->header().Marker(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_marker_deltas(encoded_deltas);
>+  }
>+
>+  // payload_type (RTP base)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    values[i] = event->header().PayloadType();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->header().PayloadType(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_payload_type_deltas(encoded_deltas);
>+  }
>+
>+  // sequence_number (RTP base)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    values[i] = event->header().SequenceNumber();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->header().SequenceNumber(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_sequence_number_deltas(encoded_deltas);
>+  }
>+
>+  // rtp_timestamp (RTP base)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    values[i] = event->header().Timestamp();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->header().Timestamp(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_rtp_timestamp_deltas(encoded_deltas);
>+  }
>+
>+  // ssrc (RTP base)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    values[i] = event->header().Ssrc();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->header().Ssrc(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_ssrc_deltas(encoded_deltas);
>+  }
>+
>+  // payload_size (RTP base)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    values[i] = event->payload_length();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->payload_length(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_payload_size_deltas(encoded_deltas);
>+  }
>+
>+  // header_size (RTP base)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    values[i] = event->header_length();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->header_length(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_header_size_deltas(encoded_deltas);
>+  }
>+
>+  // padding_size (RTP base)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    values[i] = event->padding_length();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->padding_length(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_padding_size_deltas(encoded_deltas);
>+  }
>+
>+  // transport_sequence_number (RTP extension)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    uint16_t seqnum;
>+    if (event->header().template GetExtension<TransportSequenceNumber>(
>+            &seqnum)) {
>+      values[i] = seqnum;
>+    } else {
>+      values[i].reset();
>+    }
>+  }
>+  encoded_deltas = EncodeDeltas(base_transport_sequence_number, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_transport_sequence_number_deltas(encoded_deltas);
>+  }
>+
>+  // transmission_time_offset (RTP extension)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    int32_t offset;
>+    if (event->header().template GetExtension<TransmissionOffset>(&offset)) {
>+      values[i] = ToUnsigned(offset);
>+    } else {
>+      values[i].reset();
>+    }
>+  }
>+  encoded_deltas = EncodeDeltas(unsigned_base_transmission_time_offset, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_transmission_time_offset_deltas(encoded_deltas);
>+  }
>+
>+  // absolute_send_time (RTP extension)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    uint32_t sendtime;
>+    if (event->header().template GetExtension<AbsoluteSendTime>(&sendtime)) {
>+      values[i] = sendtime;
>+    } else {
>+      values[i].reset();
>+    }
>+  }
>+  encoded_deltas = EncodeDeltas(base_absolute_send_time, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_absolute_send_time_deltas(encoded_deltas);
>+  }
>+
>+  // video_rotation (RTP extension)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    VideoRotation video_rotation;
>+    if (event->header().template GetExtension<VideoOrientation>(
>+            &video_rotation)) {
>+      values[i] = ConvertVideoRotationToCVOByte(video_rotation);
>+    } else {
>+      values[i].reset();
>+    }
>+  }
>+  encoded_deltas = EncodeDeltas(base_video_rotation, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_video_rotation_deltas(encoded_deltas);
>+  }
>+
>+  // audio_level (RTP extension)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    bool voice_activity;
>+    uint8_t audio_level;
>+    if (event->header().template GetExtension<AudioLevel>(&voice_activity,
>+                                                          &audio_level)) {
>+      RTC_DCHECK_LE(audio_level, 0x7Fu);
>+      values[i] = audio_level;
>+    } else {
>+      values[i].reset();
>+    }
>+  }
>+  encoded_deltas = EncodeDeltas(base_audio_level, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_audio_level_deltas(encoded_deltas);
>+  }
>+
>+  // voice_activity (RTP extension)
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const EventType* event = batch[i + 1];
>+    bool voice_activity;
>+    uint8_t audio_level;
>+    if (event->header().template GetExtension<AudioLevel>(&voice_activity,
>+                                                          &audio_level)) {
>+      RTC_DCHECK_LE(audio_level, 0x7Fu);
>+      values[i] = voice_activity;
>+    } else {
>+      values[i].reset();
>+    }
>+  }
>+  encoded_deltas = EncodeDeltas(base_voice_activity, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_voice_activity_deltas(encoded_deltas);
>+  }
>+}
>+}  // namespace
>+
>+std::string RtcEventLogEncoderNewFormat::EncodeLogStart(int64_t timestamp_us,
>+                                                        int64_t utc_time_us) {
>+  rtclog2::EventStream event_stream;
>+  rtclog2::BeginLogEvent* proto_batch = event_stream.add_begin_log_events();
>+  proto_batch->set_timestamp_ms(timestamp_us / 1000);
>+  proto_batch->set_version(2);
>+  proto_batch->set_utc_time_ms(utc_time_us / 1000);
>+  return event_stream.SerializeAsString();
>+}
>+
>+std::string RtcEventLogEncoderNewFormat::EncodeLogEnd(int64_t timestamp_us) {
>+  rtclog2::EventStream event_stream;
>+  rtclog2::EndLogEvent* proto_batch = event_stream.add_end_log_events();
>+  proto_batch->set_timestamp_ms(timestamp_us / 1000);
>+  return event_stream.SerializeAsString();
>+}
>+
>+std::string RtcEventLogEncoderNewFormat::EncodeBatch(
>+    std::deque<std::unique_ptr<RtcEvent>>::const_iterator begin,
>+    std::deque<std::unique_ptr<RtcEvent>>::const_iterator end) {
>+  rtclog2::EventStream event_stream;
>+  std::string encoded_output;
>+
>+  {
>+    std::vector<const RtcEventAlrState*> alr_state_events;
>+    std::vector<const RtcEventAudioNetworkAdaptation*>
>+        audio_network_adaptation_events;
>+    std::vector<const RtcEventAudioPlayout*> audio_playout_events;
>+    std::vector<const RtcEventAudioReceiveStreamConfig*>
>+        audio_recv_stream_configs;
>+    std::vector<const RtcEventAudioSendStreamConfig*> audio_send_stream_configs;
>+    std::vector<const RtcEventBweUpdateDelayBased*> bwe_delay_based_updates;
>+    std::vector<const RtcEventBweUpdateLossBased*> bwe_loss_based_updates;
>+    std::vector<const RtcEventProbeClusterCreated*>
>+        probe_cluster_created_events;
>+    std::vector<const RtcEventProbeResultFailure*> probe_result_failure_events;
>+    std::vector<const RtcEventProbeResultSuccess*> probe_result_success_events;
>+    std::vector<const RtcEventRtcpPacketIncoming*> incoming_rtcp_packets;
>+    std::vector<const RtcEventRtcpPacketOutgoing*> outgoing_rtcp_packets;
>+    std::map<uint32_t /* SSRC */, std::vector<const RtcEventRtpPacketIncoming*>>
>+        incoming_rtp_packets;
>+    std::map<uint32_t /* SSRC */, std::vector<const RtcEventRtpPacketOutgoing*>>
>+        outgoing_rtp_packets;
>+    std::vector<const RtcEventVideoReceiveStreamConfig*>
>+        video_recv_stream_configs;
>+    std::vector<const RtcEventVideoSendStreamConfig*> video_send_stream_configs;
>+    std::vector<const RtcEventIceCandidatePairConfig*> ice_candidate_configs;
>+    std::vector<const RtcEventIceCandidatePair*> ice_candidate_events;
>+
>+    for (auto it = begin; it != end; ++it) {
>+      switch ((*it)->GetType()) {
>+        case RtcEvent::Type::AlrStateEvent: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventAlrState* const>(it->get());
>+          alr_state_events.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::AudioNetworkAdaptation: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventAudioNetworkAdaptation* const>(
>+                  it->get());
>+          audio_network_adaptation_events.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::AudioPlayout: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventAudioPlayout* const>(it->get());
>+          audio_playout_events.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::AudioReceiveStreamConfig: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventAudioReceiveStreamConfig* const>(
>+                  it->get());
>+          audio_recv_stream_configs.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::AudioSendStreamConfig: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventAudioSendStreamConfig* const>(
>+                  it->get());
>+          audio_send_stream_configs.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::BweUpdateDelayBased: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventBweUpdateDelayBased* const>(it->get());
>+          bwe_delay_based_updates.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::BweUpdateLossBased: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventBweUpdateLossBased* const>(it->get());
>+          bwe_loss_based_updates.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::ProbeClusterCreated: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventProbeClusterCreated* const>(it->get());
>+          probe_cluster_created_events.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::ProbeResultFailure: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventProbeResultFailure* const>(it->get());
>+          probe_result_failure_events.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::ProbeResultSuccess: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventProbeResultSuccess* const>(it->get());
>+          probe_result_success_events.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::RtcpPacketIncoming: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventRtcpPacketIncoming* const>(it->get());
>+          incoming_rtcp_packets.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::RtcpPacketOutgoing: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventRtcpPacketOutgoing* const>(it->get());
>+          outgoing_rtcp_packets.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::RtpPacketIncoming: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventRtpPacketIncoming* const>(it->get());
>+          auto& v = incoming_rtp_packets[rtc_event->header().Ssrc()];
>+          v.emplace_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::RtpPacketOutgoing: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventRtpPacketOutgoing* const>(it->get());
>+          auto& v = outgoing_rtp_packets[rtc_event->header().Ssrc()];
>+          v.emplace_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::VideoReceiveStreamConfig: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventVideoReceiveStreamConfig* const>(
>+                  it->get());
>+          video_recv_stream_configs.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::VideoSendStreamConfig: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventVideoSendStreamConfig* const>(
>+                  it->get());
>+          video_send_stream_configs.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::IceCandidatePairConfig: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventIceCandidatePairConfig* const>(
>+                  it->get());
>+          ice_candidate_configs.push_back(rtc_event);
>+          break;
>+        }
>+        case RtcEvent::Type::IceCandidatePairEvent: {
>+          auto* rtc_event =
>+              static_cast<const RtcEventIceCandidatePair* const>(it->get());
>+          ice_candidate_events.push_back(rtc_event);
>+          break;
>+        }
>+      }
>+    }
>+
>+    EncodeAlrState(alr_state_events, &event_stream);
>+    EncodeAudioNetworkAdaptation(audio_network_adaptation_events,
>+                                 &event_stream);
>+    EncodeAudioPlayout(audio_playout_events, &event_stream);
>+    EncodeAudioRecvStreamConfig(audio_recv_stream_configs, &event_stream);
>+    EncodeAudioSendStreamConfig(audio_send_stream_configs, &event_stream);
>+    EncodeBweUpdateDelayBased(bwe_delay_based_updates, &event_stream);
>+    EncodeBweUpdateLossBased(bwe_loss_based_updates, &event_stream);
>+    EncodeProbeClusterCreated(probe_cluster_created_events, &event_stream);
>+    EncodeProbeResultFailure(probe_result_failure_events, &event_stream);
>+    EncodeProbeResultSuccess(probe_result_success_events, &event_stream);
>+    EncodeRtcpPacketIncoming(incoming_rtcp_packets, &event_stream);
>+    EncodeRtcpPacketOutgoing(outgoing_rtcp_packets, &event_stream);
>+    EncodeRtpPacketIncoming(incoming_rtp_packets, &event_stream);
>+    EncodeRtpPacketOutgoing(outgoing_rtp_packets, &event_stream);
>+    EncodeVideoRecvStreamConfig(video_recv_stream_configs, &event_stream);
>+    EncodeVideoSendStreamConfig(video_send_stream_configs, &event_stream);
>+    EncodeIceCandidatePairConfig(ice_candidate_configs, &event_stream);
>+    EncodeIceCandidatePairEvent(ice_candidate_events, &event_stream);
>+  }  // Deallocate the temporary vectors.
>+
>+  return event_stream.SerializeAsString();
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeAlrState(
>+    rtc::ArrayView<const RtcEventAlrState*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (const RtcEventAlrState* base_event : batch) {
>+    rtclog2::AlrState* proto_batch = event_stream->add_alr_states();
>+    proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+    proto_batch->set_in_alr(base_event->in_alr());
>+  }
>+  // TODO(terelius): Should we delta-compress this event type?
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeAudioNetworkAdaptation(
>+    rtc::ArrayView<const RtcEventAudioNetworkAdaptation*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  if (batch.size() == 0)
>+    return;
>+
>+  // Base event
>+  const RtcEventAudioNetworkAdaptation* const base_event = batch[0];
>+  rtclog2::AudioNetworkAdaptations* proto_batch =
>+      event_stream->add_audio_network_adaptations();
>+  proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+  if (base_event->config().bitrate_bps.has_value())
>+    proto_batch->set_bitrate_bps(base_event->config().bitrate_bps.value());
>+  if (base_event->config().frame_length_ms.has_value()) {
>+    proto_batch->set_frame_length_ms(
>+        base_event->config().frame_length_ms.value());
>+  }
>+  absl::optional<uint64_t> base_uplink_packet_loss_fraction;
>+  if (base_event->config().uplink_packet_loss_fraction.has_value()) {
>+    base_uplink_packet_loss_fraction = ConvertPacketLossFractionToProtoFormat(
>+        base_event->config().uplink_packet_loss_fraction.value());
>+    proto_batch->set_uplink_packet_loss_fraction(
>+        base_uplink_packet_loss_fraction.value());
>+  }
>+  if (base_event->config().enable_fec.has_value())
>+    proto_batch->set_enable_fec(base_event->config().enable_fec.value());
>+  if (base_event->config().enable_dtx.has_value())
>+    proto_batch->set_enable_dtx(base_event->config().enable_dtx.value());
>+  // Note that |num_channels_deltas| encodes N as N-1, to keep deltas smaller,
>+  // but there's no reason to do the same for the base event's value, since
>+  // no bits will be spared.
>+  if (base_event->config().num_channels.has_value())
>+    proto_batch->set_num_channels(base_event->config().num_channels.value());
>+
>+  if (batch.size() == 1)
>+    return;
>+
>+  // Delta encoding
>+  proto_batch->set_number_of_deltas(batch.size() - 1);
>+  std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
>+  std::string encoded_deltas;
>+
>+  // timestamp_ms
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
>+    values[i] = ToUnsigned(event->timestamp_ms());
>+  }
>+  encoded_deltas = EncodeDeltas(ToUnsigned(base_event->timestamp_ms()), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_timestamp_ms_deltas(encoded_deltas);
>+  }
>+
>+  // bitrate_bps
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
>+    if (event->config().bitrate_bps.has_value()) {
>+      values[i] = ToUnsigned(event->config().bitrate_bps.value());
>+    } else {
>+      values[i].reset();
>+    }
>+  }
>+  const absl::optional<uint64_t> unsigned_base_bitrate_bps =
>+      base_event->config().bitrate_bps.has_value()
>+          ? ToUnsigned(base_event->config().bitrate_bps.value())
>+          : absl::optional<uint64_t>();
>+  encoded_deltas = EncodeDeltas(unsigned_base_bitrate_bps, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_bitrate_bps_deltas(encoded_deltas);
>+  }
>+
>+  // frame_length_ms
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
>+    if (event->config().frame_length_ms.has_value()) {
>+      values[i] = ToUnsigned(event->config().frame_length_ms.value());
>+    } else {
>+      values[i].reset();
>+    }
>+  }
>+  const absl::optional<uint64_t> unsigned_base_frame_length_ms =
>+      base_event->config().frame_length_ms.has_value()
>+          ? ToUnsigned(base_event->config().frame_length_ms.value())
>+          : absl::optional<uint64_t>();
>+  encoded_deltas = EncodeDeltas(unsigned_base_frame_length_ms, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_frame_length_ms_deltas(encoded_deltas);
>+  }
>+
>+  // uplink_packet_loss_fraction
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
>+    if (event->config().uplink_packet_loss_fraction.has_value()) {
>+      values[i] = ConvertPacketLossFractionToProtoFormat(
>+          event->config().uplink_packet_loss_fraction.value());
>+    } else {
>+      values[i].reset();
>+    }
>+  }
>+  encoded_deltas = EncodeDeltas(base_uplink_packet_loss_fraction, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_uplink_packet_loss_fraction_deltas(encoded_deltas);
>+  }
>+
>+  // enable_fec
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
>+    values[i] = event->config().enable_fec;
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->config().enable_fec, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_enable_fec_deltas(encoded_deltas);
>+  }
>+
>+  // enable_dtx
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
>+    values[i] = event->config().enable_dtx;
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->config().enable_dtx, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_enable_dtx_deltas(encoded_deltas);
>+  }
>+
>+  // num_channels
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventAudioNetworkAdaptation* event = batch[i + 1];
>+    const absl::optional<size_t> num_channels = event->config().num_channels;
>+    if (num_channels.has_value()) {
>+      // Since the number of channels is always greater than 0, we can encode
>+      // N channels as N-1, thereby making sure that we get smaller deltas.
>+      // That is, a toggle of 1->2->1 can be encoded as deltas vector (1, 1),
>+      // rather than as (1, 3) or (1, -1), either of which would require two
>+      // bits per delta.
>+      RTC_DCHECK_GT(num_channels.value(), 0u);
>+      values[i] = num_channels.value() - 1;
>+    } else {
>+      values[i].reset();
>+    }
>+  }
>+  // In the base event, N channels encoded as N channels, but for delta
>+  // compression purposes, also shifted down by 1.
>+  absl::optional<size_t> shifted_base_num_channels;
>+  if (base_event->config().num_channels.has_value()) {
>+    RTC_DCHECK_GT(base_event->config().num_channels.value(), 0u);
>+    shifted_base_num_channels = base_event->config().num_channels.value() - 1;
>+  }
>+  encoded_deltas = EncodeDeltas(shifted_base_num_channels, values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_num_channels_deltas(encoded_deltas);
>+  }
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeAudioPlayout(
>+    rtc::ArrayView<const RtcEventAudioPlayout*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  if (batch.size() == 0)
>+    return;
>+
>+  // Base event
>+  const RtcEventAudioPlayout* const base_event = batch[0];
>+  rtclog2::AudioPlayoutEvents* proto_batch =
>+      event_stream->add_audio_playout_events();
>+  proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+  proto_batch->set_local_ssrc(base_event->ssrc());
>+
>+  if (batch.size() == 1)
>+    return;
>+
>+  // Delta encoding
>+  proto_batch->set_number_of_deltas(batch.size() - 1);
>+  std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
>+  std::string encoded_deltas;
>+
>+  // timestamp_ms
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventAudioPlayout* event = batch[i + 1];
>+    values[i] = ToUnsigned(event->timestamp_ms());
>+  }
>+  encoded_deltas = EncodeDeltas(ToUnsigned(base_event->timestamp_ms()), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_timestamp_ms_deltas(encoded_deltas);
>+  }
>+
>+  // local_ssrc
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventAudioPlayout* event = batch[i + 1];
>+    values[i] = event->ssrc();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->ssrc(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_local_ssrc_deltas(encoded_deltas);
>+  }
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeAudioRecvStreamConfig(
>+    rtc::ArrayView<const RtcEventAudioReceiveStreamConfig*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (const RtcEventAudioReceiveStreamConfig* base_event : batch) {
>+    rtclog2::AudioRecvStreamConfig* proto_batch =
>+        event_stream->add_audio_recv_stream_configs();
>+    proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+    proto_batch->set_remote_ssrc(base_event->config().remote_ssrc);
>+    proto_batch->set_local_ssrc(base_event->config().local_ssrc);
>+
>+    rtclog2::RtpHeaderExtensionConfig* proto_config =
>+        proto_batch->mutable_header_extensions();
>+    bool has_recognized_extensions =
>+        ConvertToProtoFormat(base_event->config().rtp_extensions, proto_config);
>+    if (!has_recognized_extensions)
>+      proto_batch->clear_header_extensions();
>+  }
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeAudioSendStreamConfig(
>+    rtc::ArrayView<const RtcEventAudioSendStreamConfig*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (const RtcEventAudioSendStreamConfig* base_event : batch) {
>+    rtclog2::AudioSendStreamConfig* proto_batch =
>+        event_stream->add_audio_send_stream_configs();
>+    proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+    proto_batch->set_ssrc(base_event->config().local_ssrc);
>+
>+    rtclog2::RtpHeaderExtensionConfig* proto_config =
>+        proto_batch->mutable_header_extensions();
>+    bool has_recognized_extensions =
>+        ConvertToProtoFormat(base_event->config().rtp_extensions, proto_config);
>+    if (!has_recognized_extensions)
>+      proto_batch->clear_header_extensions();
>+  }
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeBweUpdateDelayBased(
>+    rtc::ArrayView<const RtcEventBweUpdateDelayBased*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  if (batch.size() == 0)
>+    return;
>+
>+  // Base event
>+  const RtcEventBweUpdateDelayBased* const base_event = batch[0];
>+  rtclog2::DelayBasedBweUpdates* proto_batch =
>+      event_stream->add_delay_based_bwe_updates();
>+  proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+  proto_batch->set_bitrate_bps(base_event->bitrate_bps());
>+  proto_batch->set_detector_state(
>+      ConvertToProtoFormat(base_event->detector_state()));
>+
>+  if (batch.size() == 1)
>+    return;
>+
>+  // Delta encoding
>+  proto_batch->set_number_of_deltas(batch.size() - 1);
>+  std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
>+  std::string encoded_deltas;
>+
>+  // timestamp_ms
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventBweUpdateDelayBased* event = batch[i + 1];
>+    values[i] = ToUnsigned(event->timestamp_ms());
>+  }
>+  encoded_deltas = EncodeDeltas(ToUnsigned(base_event->timestamp_ms()), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_timestamp_ms_deltas(encoded_deltas);
>+  }
>+
>+  // bitrate_bps
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventBweUpdateDelayBased* event = batch[i + 1];
>+    values[i] = event->bitrate_bps();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->bitrate_bps(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_bitrate_bps_deltas(encoded_deltas);
>+  }
>+
>+  // detector_state
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventBweUpdateDelayBased* event = batch[i + 1];
>+    values[i] =
>+        static_cast<uint64_t>(ConvertToProtoFormat(event->detector_state()));
>+  }
>+  encoded_deltas = EncodeDeltas(
>+      static_cast<uint64_t>(ConvertToProtoFormat(base_event->detector_state())),
>+      values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_detector_state_deltas(encoded_deltas);
>+  }
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeBweUpdateLossBased(
>+    rtc::ArrayView<const RtcEventBweUpdateLossBased*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  if (batch.size() == 0)
>+    return;
>+
>+  // Base event
>+  const RtcEventBweUpdateLossBased* const base_event = batch[0];
>+  rtclog2::LossBasedBweUpdates* proto_batch =
>+      event_stream->add_loss_based_bwe_updates();
>+  proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+  proto_batch->set_bitrate_bps(base_event->bitrate_bps());
>+  proto_batch->set_fraction_loss(base_event->fraction_loss());
>+  proto_batch->set_total_packets(base_event->total_packets());
>+
>+  if (batch.size() == 1)
>+    return;
>+
>+  // Delta encoding
>+  proto_batch->set_number_of_deltas(batch.size() - 1);
>+  std::vector<absl::optional<uint64_t>> values(batch.size() - 1);
>+  std::string encoded_deltas;
>+
>+  // timestamp_ms
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventBweUpdateLossBased* event = batch[i + 1];
>+    values[i] = ToUnsigned(event->timestamp_ms());
>+  }
>+  encoded_deltas = EncodeDeltas(ToUnsigned(base_event->timestamp_ms()), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_timestamp_ms_deltas(encoded_deltas);
>+  }
>+
>+  // bitrate_bps
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventBweUpdateLossBased* event = batch[i + 1];
>+    values[i] = event->bitrate_bps();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->bitrate_bps(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_bitrate_bps_deltas(encoded_deltas);
>+  }
>+
>+  // fraction_loss
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventBweUpdateLossBased* event = batch[i + 1];
>+    values[i] = event->fraction_loss();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->fraction_loss(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_fraction_loss_deltas(encoded_deltas);
>+  }
>+
>+  // total_packets
>+  for (size_t i = 0; i < values.size(); ++i) {
>+    const RtcEventBweUpdateLossBased* event = batch[i + 1];
>+    values[i] = event->total_packets();
>+  }
>+  encoded_deltas = EncodeDeltas(base_event->total_packets(), values);
>+  if (!encoded_deltas.empty()) {
>+    proto_batch->set_total_packets_deltas(encoded_deltas);
>+  }
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeProbeClusterCreated(
>+    rtc::ArrayView<const RtcEventProbeClusterCreated*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (const RtcEventProbeClusterCreated* base_event : batch) {
>+    rtclog2::BweProbeCluster* proto_batch = event_stream->add_probe_clusters();
>+    proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+    proto_batch->set_id(base_event->id());
>+    proto_batch->set_bitrate_bps(base_event->bitrate_bps());
>+    proto_batch->set_min_packets(base_event->min_probes());
>+    proto_batch->set_min_bytes(base_event->min_bytes());
>+  }
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeProbeResultFailure(
>+    rtc::ArrayView<const RtcEventProbeResultFailure*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (const RtcEventProbeResultFailure* base_event : batch) {
>+    rtclog2::BweProbeResultFailure* proto_batch =
>+        event_stream->add_probe_failure();
>+    proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+    proto_batch->set_id(base_event->id());
>+    proto_batch->set_failure(
>+        ConvertToProtoFormat(base_event->failure_reason()));
>+  }
>+  // TODO(terelius): Should we delta-compress this event type?
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeProbeResultSuccess(
>+    rtc::ArrayView<const RtcEventProbeResultSuccess*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (const RtcEventProbeResultSuccess* base_event : batch) {
>+    rtclog2::BweProbeResultSuccess* proto_batch =
>+        event_stream->add_probe_success();
>+    proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+    proto_batch->set_id(base_event->id());
>+    proto_batch->set_bitrate_bps(base_event->bitrate_bps());
>+  }
>+  // TODO(terelius): Should we delta-compress this event type?
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeRtcpPacketIncoming(
>+    rtc::ArrayView<const RtcEventRtcpPacketIncoming*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  if (batch.empty()) {
>+    return;
>+  }
>+  EncodeRtcpPacket(batch, event_stream->add_incoming_rtcp_packets());
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeRtcpPacketOutgoing(
>+    rtc::ArrayView<const RtcEventRtcpPacketOutgoing*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  if (batch.empty()) {
>+    return;
>+  }
>+  EncodeRtcpPacket(batch, event_stream->add_outgoing_rtcp_packets());
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeRtpPacketIncoming(
>+    const std::map<uint32_t, std::vector<const RtcEventRtpPacketIncoming*>>&
>+        batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (auto it : batch) {
>+    RTC_DCHECK(!it.second.empty());
>+    EncodeRtpPacket(it.second, event_stream->add_incoming_rtp_packets());
>+  }
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeRtpPacketOutgoing(
>+    const std::map<uint32_t, std::vector<const RtcEventRtpPacketOutgoing*>>&
>+        batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (auto it : batch) {
>+    RTC_DCHECK(!it.second.empty());
>+    EncodeRtpPacket(it.second, event_stream->add_outgoing_rtp_packets());
>+  }
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeVideoRecvStreamConfig(
>+    rtc::ArrayView<const RtcEventVideoReceiveStreamConfig*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (const RtcEventVideoReceiveStreamConfig* base_event : batch) {
>+    rtclog2::VideoRecvStreamConfig* proto_batch =
>+        event_stream->add_video_recv_stream_configs();
>+    proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+    proto_batch->set_remote_ssrc(base_event->config().remote_ssrc);
>+    proto_batch->set_local_ssrc(base_event->config().local_ssrc);
>+    proto_batch->set_rtx_ssrc(base_event->config().rtx_ssrc);
>+
>+    rtclog2::RtpHeaderExtensionConfig* proto_config =
>+        proto_batch->mutable_header_extensions();
>+    bool has_recognized_extensions =
>+        ConvertToProtoFormat(base_event->config().rtp_extensions, proto_config);
>+    if (!has_recognized_extensions)
>+      proto_batch->clear_header_extensions();
>+  }
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeVideoSendStreamConfig(
>+    rtc::ArrayView<const RtcEventVideoSendStreamConfig*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (const RtcEventVideoSendStreamConfig* base_event : batch) {
>+    rtclog2::VideoSendStreamConfig* proto_batch =
>+        event_stream->add_video_send_stream_configs();
>+    proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+    proto_batch->set_ssrc(base_event->config().local_ssrc);
>+    proto_batch->set_rtx_ssrc(base_event->config().rtx_ssrc);
>+
>+    rtclog2::RtpHeaderExtensionConfig* proto_config =
>+        proto_batch->mutable_header_extensions();
>+    bool has_recognized_extensions =
>+        ConvertToProtoFormat(base_event->config().rtp_extensions, proto_config);
>+    if (!has_recognized_extensions)
>+      proto_batch->clear_header_extensions();
>+  }
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeIceCandidatePairConfig(
>+    rtc::ArrayView<const RtcEventIceCandidatePairConfig*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (const RtcEventIceCandidatePairConfig* base_event : batch) {
>+    rtclog2::IceCandidatePairConfig* proto_batch =
>+        event_stream->add_ice_candidate_configs();
>+
>+    proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+    proto_batch->set_config_type(ConvertToProtoFormat(base_event->type()));
>+    proto_batch->set_candidate_pair_id(base_event->candidate_pair_id());
>+    const auto& desc = base_event->candidate_pair_desc();
>+    proto_batch->set_local_candidate_type(
>+        ConvertToProtoFormat(desc.local_candidate_type));
>+    proto_batch->set_local_relay_protocol(
>+        ConvertToProtoFormat(desc.local_relay_protocol));
>+    proto_batch->set_local_network_type(
>+        ConvertToProtoFormat(desc.local_network_type));
>+    proto_batch->set_local_address_family(
>+        ConvertToProtoFormat(desc.local_address_family));
>+    proto_batch->set_remote_candidate_type(
>+        ConvertToProtoFormat(desc.remote_candidate_type));
>+    proto_batch->set_remote_address_family(
>+        ConvertToProtoFormat(desc.remote_address_family));
>+    proto_batch->set_candidate_pair_protocol(
>+        ConvertToProtoFormat(desc.candidate_pair_protocol));
>+  }
>+  // TODO(terelius): Should we delta-compress this event type?
>+}
>+
>+void RtcEventLogEncoderNewFormat::EncodeIceCandidatePairEvent(
>+    rtc::ArrayView<const RtcEventIceCandidatePair*> batch,
>+    rtclog2::EventStream* event_stream) {
>+  for (const RtcEventIceCandidatePair* base_event : batch) {
>+    rtclog2::IceCandidatePairEvent* proto_batch =
>+        event_stream->add_ice_candidate_events();
>+
>+    proto_batch->set_timestamp_ms(base_event->timestamp_ms());
>+
>+    proto_batch->set_event_type(ConvertToProtoFormat(base_event->type()));
>+    proto_batch->set_candidate_pair_id(base_event->candidate_pair_id());
>+  }
>+  // TODO(terelius): Should we delta-compress this event type?
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..7dfa490114b4eb883bc7896151498e98110a8806
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h
>@@ -0,0 +1,127 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_NEW_FORMAT_H_
>+#define LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_NEW_FORMAT_H_
>+
>+#include <deque>
>+#include <map>
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "api/array_view.h"
>+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder.h"
>+
>+namespace webrtc {
>+
>+namespace rtclog2 {
>+class EventStream;  // Auto-generated from protobuf.
>+}  // namespace rtclog2
>+
>+class RtcEventAlrState;
>+class RtcEventAudioNetworkAdaptation;
>+class RtcEventAudioPlayout;
>+class RtcEventAudioReceiveStreamConfig;
>+class RtcEventAudioSendStreamConfig;
>+class RtcEventBweUpdateDelayBased;
>+class RtcEventBweUpdateLossBased;
>+class RtcEventLoggingStarted;
>+class RtcEventLoggingStopped;
>+class RtcEventProbeClusterCreated;
>+class RtcEventProbeResultFailure;
>+class RtcEventProbeResultSuccess;
>+class RtcEventRtcpPacketIncoming;
>+class RtcEventRtcpPacketOutgoing;
>+class RtcEventRtpPacketIncoming;
>+class RtcEventRtpPacketOutgoing;
>+class RtcEventVideoReceiveStreamConfig;
>+class RtcEventVideoSendStreamConfig;
>+class RtcEventIceCandidatePairConfig;
>+class RtcEventIceCandidatePair;
>+class RtpPacket;
>+
>+class RtcEventLogEncoderNewFormat final : public RtcEventLogEncoder {
>+ public:
>+  ~RtcEventLogEncoderNewFormat() override = default;
>+
>+  std::string EncodeBatch(
>+      std::deque<std::unique_ptr<RtcEvent>>::const_iterator begin,
>+      std::deque<std::unique_ptr<RtcEvent>>::const_iterator end) override;
>+
>+  std::string EncodeLogStart(int64_t timestamp_us,
>+                             int64_t utc_time_us) override;
>+  std::string EncodeLogEnd(int64_t timestamp_us) override;
>+
>+ private:
>+  // Encoding entry-point for the various RtcEvent subclasses.
>+  void EncodeAlrState(rtc::ArrayView<const RtcEventAlrState*> batch,
>+                      rtclog2::EventStream* event_stream);
>+  void EncodeAudioNetworkAdaptation(
>+      rtc::ArrayView<const RtcEventAudioNetworkAdaptation*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeAudioPlayout(rtc::ArrayView<const RtcEventAudioPlayout*> batch,
>+                          rtclog2::EventStream* event_stream);
>+  void EncodeAudioRecvStreamConfig(
>+      rtc::ArrayView<const RtcEventAudioReceiveStreamConfig*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeAudioSendStreamConfig(
>+      rtc::ArrayView<const RtcEventAudioSendStreamConfig*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeBweUpdateDelayBased(
>+      rtc::ArrayView<const RtcEventBweUpdateDelayBased*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeBweUpdateLossBased(
>+      rtc::ArrayView<const RtcEventBweUpdateLossBased*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeLoggingStarted(rtc::ArrayView<const RtcEventLoggingStarted*> batch,
>+                            rtclog2::EventStream* event_stream);
>+  void EncodeLoggingStopped(rtc::ArrayView<const RtcEventLoggingStopped*> batch,
>+                            rtclog2::EventStream* event_stream);
>+  void EncodeProbeClusterCreated(
>+      rtc::ArrayView<const RtcEventProbeClusterCreated*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeProbeResultFailure(
>+      rtc::ArrayView<const RtcEventProbeResultFailure*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeProbeResultSuccess(
>+      rtc::ArrayView<const RtcEventProbeResultSuccess*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeRtcpPacketIncoming(
>+      rtc::ArrayView<const RtcEventRtcpPacketIncoming*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeRtcpPacketOutgoing(
>+      rtc::ArrayView<const RtcEventRtcpPacketOutgoing*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeRtpPacketIncoming(
>+      const std::map<uint32_t, std::vector<const RtcEventRtpPacketIncoming*>>&
>+          batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeRtpPacketOutgoing(
>+      const std::map<uint32_t, std::vector<const RtcEventRtpPacketOutgoing*>>&
>+          batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeVideoRecvStreamConfig(
>+      rtc::ArrayView<const RtcEventVideoReceiveStreamConfig*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeVideoSendStreamConfig(
>+      rtc::ArrayView<const RtcEventVideoSendStreamConfig*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeIceCandidatePairConfig(
>+      rtc::ArrayView<const RtcEventIceCandidatePairConfig*> batch,
>+      rtclog2::EventStream* event_stream);
>+  void EncodeIceCandidatePairEvent(
>+      rtc::ArrayView<const RtcEventIceCandidatePair*> batch,
>+      rtclog2::EventStream* event_stream);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_NEW_FORMAT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
>index 990fa6080bb94613d543474460b87f8a3c46c08e..856bec66e7ccc081c36407a8c2641d15da47cab4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
>@@ -11,9 +11,11 @@
> #include <deque>
> #include <limits>
> #include <string>
>+#include <tuple>
> 
> #include "absl/memory/memory.h"
> #include "logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h"
>+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h"
> #include "logging/rtc_event_log/events/rtc_event_alr_state.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
>@@ -39,156 +41,357 @@
> #include "test/gtest.h"
> 
> namespace webrtc {
>-
>-
>-class RtcEventLogEncoderTest : public testing::TestWithParam<int> {
>+class RtcEventLogEncoderTest
>+    : public testing::TestWithParam<std::tuple<int, bool, size_t, bool>> {
>  protected:
>   RtcEventLogEncoderTest()
>-      : encoder_(new RtcEventLogEncoderLegacy),
>-        seed_(GetParam()),
>+      : seed_(std::get<0>(GetParam())),
>         prng_(seed_),
>-        gen_(seed_ * 880001UL) {}
>+        new_encoding_(std::get<1>(GetParam())),
>+        event_count_(std::get<2>(GetParam())),
>+        force_repeated_fields_(std::get<3>(GetParam())),
>+        gen_(seed_ * 880001UL),
>+        verifier_(new_encoding_ ? RtcEventLog::EncodingType::NewFormat
>+                                : RtcEventLog::EncodingType::Legacy) {
>+    if (new_encoding_)
>+      encoder_ = absl::make_unique<RtcEventLogEncoderNewFormat>();
>+    else
>+      encoder_ = absl::make_unique<RtcEventLogEncoderLegacy>();
>+  }
>   ~RtcEventLogEncoderTest() override = default;
> 
>   // ANA events have some optional fields, so we want to make sure that we get
>   // correct behavior both when all of the values are there, as well as when
>   // only some.
>   void TestRtcEventAudioNetworkAdaptation(
>-      std::unique_ptr<AudioEncoderRuntimeConfig> runtime_config);
>+      const std::vector<std::unique_ptr<RtcEventAudioNetworkAdaptation>>&);
>+
>+  template <typename EventType>
>+  std::unique_ptr<EventType> NewRtpPacket(
>+      uint32_t ssrc,
>+      const RtpHeaderExtensionMap& extension_map);
>+
>+  template <typename ParsedType>
>+  const std::vector<ParsedType>* GetRtpPacketsBySsrc(
>+      const ParsedRtcEventLogNew* parsed_log,
>+      uint32_t ssrc);
>+
>+  template <typename EventType, typename ParsedType>
>+  void TestRtpPackets();
> 
>   std::deque<std::unique_ptr<RtcEvent>> history_;
>-  // TODO(eladalon): Once we have more than one possible encoder, parameterize
>-  // encoder selection.
>   std::unique_ptr<RtcEventLogEncoder> encoder_;
>   ParsedRtcEventLogNew parsed_log_;
>   const uint64_t seed_;
>   Random prng_;
>+  const bool new_encoding_;
>+  const size_t event_count_;
>+  const bool force_repeated_fields_;
>   test::EventGenerator gen_;
>+  test::EventVerifier verifier_;
> };
> 
>-TEST_P(RtcEventLogEncoderTest, RtcEventAlrState) {
>-  std::unique_ptr<RtcEventAlrState> event = gen_.NewAlrState();
>-  history_.push_back(event->Copy());
>+void RtcEventLogEncoderTest::TestRtcEventAudioNetworkAdaptation(
>+    const std::vector<std::unique_ptr<RtcEventAudioNetworkAdaptation>>&
>+        events) {
>+  ASSERT_TRUE(history_.empty()) << "Function should be called once per test.";
>+
>+  for (auto& event : events) {
>+    history_.push_back(event->Copy());
>+  }
> 
>   std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>   ASSERT_TRUE(parsed_log_.ParseString(encoded));
>-  const auto& alr_state_events = parsed_log_.alr_state_events();
>+  const auto& ana_configs = parsed_log_.audio_network_adaptation_events();
> 
>-  ASSERT_EQ(alr_state_events.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedAlrStateEvent(*event, alr_state_events[0]));
>+  ASSERT_EQ(ana_configs.size(), events.size());
>+  for (size_t i = 0; i < events.size(); ++i) {
>+    verifier_.VerifyLoggedAudioNetworkAdaptationEvent(*events[i],
>+                                                      ana_configs[i]);
>+  }
> }
> 
>-void RtcEventLogEncoderTest::TestRtcEventAudioNetworkAdaptation(
>-    std::unique_ptr<AudioEncoderRuntimeConfig> runtime_config) {
>-  // This function is called repeatedly. Clear state between calls.
>-  history_.clear();
>-  auto original_runtime_config = *runtime_config;
>-  auto event = absl::make_unique<RtcEventAudioNetworkAdaptation>(
>-      std::move(runtime_config));
>-  const int64_t timestamp_us = event->timestamp_us_;
>-  history_.push_back(std::move(event));
>+template <>
>+std::unique_ptr<RtcEventRtpPacketIncoming> RtcEventLogEncoderTest::NewRtpPacket(
>+    uint32_t ssrc,
>+    const RtpHeaderExtensionMap& extension_map) {
>+  return gen_.NewRtpPacketIncoming(ssrc, extension_map, false);
>+}
>+
>+template <>
>+std::unique_ptr<RtcEventRtpPacketOutgoing> RtcEventLogEncoderTest::NewRtpPacket(
>+    uint32_t ssrc,
>+    const RtpHeaderExtensionMap& extension_map) {
>+  return gen_.NewRtpPacketOutgoing(ssrc, extension_map, false);
>+}
>+
>+template <>
>+const std::vector<LoggedRtpPacketIncoming>*
>+RtcEventLogEncoderTest::GetRtpPacketsBySsrc(
>+    const ParsedRtcEventLogNew* parsed_log,
>+    uint32_t ssrc) {
>+  const auto& incoming_streams = parsed_log->incoming_rtp_packets_by_ssrc();
>+  for (const auto& stream : incoming_streams) {
>+    if (stream.ssrc == ssrc) {
>+      return &stream.incoming_packets;
>+    }
>+  }
>+  return nullptr;
>+}
>+
>+template <>
>+const std::vector<LoggedRtpPacketOutgoing>*
>+RtcEventLogEncoderTest::GetRtpPacketsBySsrc(
>+    const ParsedRtcEventLogNew* parsed_log,
>+    uint32_t ssrc) {
>+  const auto& outgoing_streams = parsed_log->outgoing_rtp_packets_by_ssrc();
>+  for (const auto& stream : outgoing_streams) {
>+    if (stream.ssrc == ssrc) {
>+      return &stream.outgoing_packets;
>+    }
>+  }
>+  return nullptr;
>+}
>+
>+template <typename EventType, typename ParsedType>
>+void RtcEventLogEncoderTest::TestRtpPackets() {
>+  // SSRCs will be randomly assigned out of this small pool, significant only
>+  // in that it also covers such edge cases as SSRC = 0 and SSRC = 0xffffffff.
>+  // The pool is intentionally small, so as to produce collisions.
>+  const std::vector<uint32_t> kSsrcPool = {0x00000000, 0x12345678, 0xabcdef01,
>+                                           0xffffffff, 0x20171024, 0x19840730,
>+                                           0x19831230};
>+
>+  // TODO(terelius): Test extensions for legacy encoding, too.
>+  RtpHeaderExtensionMap extension_map;
>+  if (new_encoding_) {
>+    extension_map = gen_.NewRtpHeaderExtensionMap(true);
>+  }
>+
>+  // Simulate |event_count_| RTP packets, with SSRCs assigned randomly
>+  // out of the small pool above.
>+  std::map<uint32_t, std::vector<std::unique_ptr<EventType>>> events_by_ssrc;
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    const uint32_t ssrc = kSsrcPool[prng_.Rand(kSsrcPool.size() - 1)];
>+    std::unique_ptr<EventType> event =
>+        (events_by_ssrc[ssrc].empty() || !force_repeated_fields_)
>+            ? NewRtpPacket<EventType>(ssrc, extension_map)
>+            : events_by_ssrc[ssrc][0]->Copy();
>+    history_.push_back(event->Copy());
>+    events_by_ssrc[ssrc].emplace_back(std::move(event));
>+  }
> 
>+  // Encode and parse.
>   std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>   ASSERT_TRUE(parsed_log_.ParseString(encoded));
>-  const auto& ana_configs = parsed_log_.audio_network_adaptation_events();
>-  ASSERT_EQ(ana_configs.size(), 1u);
> 
>-  EXPECT_EQ(ana_configs[0].timestamp_us, timestamp_us);
>-  EXPECT_EQ(ana_configs[0].config, original_runtime_config);
>+  // For each SSRC, make sure the RTP packets associated with it to have been
>+  // correctly encoded and parsed.
>+  for (auto it = events_by_ssrc.begin(); it != events_by_ssrc.end(); ++it) {
>+    const uint32_t ssrc = it->first;
>+    const auto& original_packets = it->second;
>+    const std::vector<ParsedType>* parsed_rtp_packets =
>+        GetRtpPacketsBySsrc<ParsedType>(&parsed_log_, ssrc);
>+    ASSERT_NE(parsed_rtp_packets, nullptr);
>+    ASSERT_EQ(original_packets.size(), parsed_rtp_packets->size());
>+    for (size_t i = 0; i < original_packets.size(); ++i) {
>+      verifier_.VerifyLoggedRtpPacket<EventType, ParsedType>(
>+          *original_packets[i], (*parsed_rtp_packets)[i]);
>+    }
>+  }
>+}
>+
>+TEST_P(RtcEventLogEncoderTest, RtcEventAlrState) {
>+  std::vector<std::unique_ptr<RtcEventAlrState>> events(event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    events[i] = (i == 0 || !force_repeated_fields_) ? gen_.NewAlrState()
>+                                                    : events[0]->Copy();
>+    history_.push_back(events[i]->Copy());
>+  }
>+
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& alr_state_events = parsed_log_.alr_state_events();
>+
>+  ASSERT_EQ(alr_state_events.size(), event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    verifier_.VerifyLoggedAlrStateEvent(*events[i], alr_state_events[i]);
>+  }
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationBitrate) {
>-  auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>-  const int bitrate_bps = rtc::checked_cast<int>(
>-      prng_.Rand(0, std::numeric_limits<int32_t>::max()));
>-  runtime_config->bitrate_bps = bitrate_bps;
>-  TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
>+  std::vector<std::unique_ptr<RtcEventAudioNetworkAdaptation>> events(
>+      event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    if (i == 0 || !force_repeated_fields_) {
>+      auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>+      const int bitrate_bps = rtc::checked_cast<int>(
>+          prng_.Rand(0, std::numeric_limits<int32_t>::max()));
>+      runtime_config->bitrate_bps = bitrate_bps;
>+      events[i] = absl::make_unique<RtcEventAudioNetworkAdaptation>(
>+          std::move(runtime_config));
>+    } else {
>+      events[i] = events[0]->Copy();
>+    }
>+  }
>+  TestRtcEventAudioNetworkAdaptation(events);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationFrameLength) {
>-  auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>-  const int frame_length_ms = prng_.Rand(1, 1000);
>-  runtime_config->frame_length_ms = frame_length_ms;
>-  TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
>+  std::vector<std::unique_ptr<RtcEventAudioNetworkAdaptation>> events(
>+      event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    if (i == 0 || !force_repeated_fields_) {
>+      auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>+      const int frame_length_ms = prng_.Rand(1, 1000);
>+      runtime_config->frame_length_ms = frame_length_ms;
>+      events[i] = absl::make_unique<RtcEventAudioNetworkAdaptation>(
>+          std::move(runtime_config));
>+    } else {
>+      events[i] = events[0]->Copy();
>+    }
>+  }
>+  TestRtcEventAudioNetworkAdaptation(events);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationPacketLoss) {
>-  // To simplify the test, we just check powers of two.
>-  const float plr = std::pow(0.5f, prng_.Rand(1, 8));
>-  auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>-  runtime_config->uplink_packet_loss_fraction = plr;
>-  TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
>+  std::vector<std::unique_ptr<RtcEventAudioNetworkAdaptation>> events(
>+      event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    if (i == 0 || !force_repeated_fields_) {
>+      // To simplify the test, we just check powers of two.
>+      const float plr = std::pow(0.5f, prng_.Rand(1, 8));
>+      auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>+      runtime_config->uplink_packet_loss_fraction = plr;
>+      events[i] = absl::make_unique<RtcEventAudioNetworkAdaptation>(
>+          std::move(runtime_config));
>+    } else {
>+      events[i] = events[0]->Copy();
>+    }
>+  }
>+  TestRtcEventAudioNetworkAdaptation(events);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationFec) {
>-  // The test might be trivially passing for one of the two boolean values, so
>-  // for safety's sake, we test both.
>-  for (bool fec_enabled : {false, true}) {
>-    auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>-    runtime_config->enable_fec = fec_enabled;
>-    TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
>+  std::vector<std::unique_ptr<RtcEventAudioNetworkAdaptation>> events(
>+      event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    if (i == 0 || !force_repeated_fields_) {
>+      auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>+      runtime_config->enable_fec = prng_.Rand<bool>();
>+      events[i] = absl::make_unique<RtcEventAudioNetworkAdaptation>(
>+          std::move(runtime_config));
>+    } else {
>+      events[i] = events[0]->Copy();
>+    }
>   }
>+  TestRtcEventAudioNetworkAdaptation(events);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationDtx) {
>-  // The test might be trivially passing for one of the two boolean values, so
>-  // for safety's sake, we test both.
>-  for (bool dtx_enabled : {false, true}) {
>-    auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>-    runtime_config->enable_dtx = dtx_enabled;
>-    TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
>+  std::vector<std::unique_ptr<RtcEventAudioNetworkAdaptation>> events(
>+      event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    if (i == 0 || !force_repeated_fields_) {
>+      auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>+      runtime_config->enable_dtx = prng_.Rand<bool>();
>+      events[i] = absl::make_unique<RtcEventAudioNetworkAdaptation>(
>+          std::move(runtime_config));
>+    } else {
>+      events[i] = events[0]->Copy();
>+    }
>   }
>+  TestRtcEventAudioNetworkAdaptation(events);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationChannels) {
>-  // The test might be trivially passing for one of the two possible values, so
>-  // for safety's sake, we test both.
>-  for (size_t channels : {1, 2}) {
>-    auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>-    runtime_config->num_channels = channels;
>-    TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
>+  std::vector<std::unique_ptr<RtcEventAudioNetworkAdaptation>> events(
>+      event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    if (i == 0 || !force_repeated_fields_) {
>+      auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>+      runtime_config->num_channels = prng_.Rand(1, 2);
>+      events[i] = absl::make_unique<RtcEventAudioNetworkAdaptation>(
>+          std::move(runtime_config));
>+    } else {
>+      events[i] = events[0]->Copy();
>+    }
>   }
>+  TestRtcEventAudioNetworkAdaptation(events);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationAll) {
>-  const int bitrate_bps = rtc::checked_cast<int>(
>-      prng_.Rand(0, std::numeric_limits<int32_t>::max()));
>-  const int frame_length_ms = prng_.Rand(1, 1000);
>-  const float plr = std::pow(0.5f, prng_.Rand(1, 8));
>-  for (bool fec_enabled : {false, true}) {
>-    for (bool dtx_enabled : {false, true}) {
>-      for (size_t channels : {1, 2}) {
>-        auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>-        runtime_config->bitrate_bps = bitrate_bps;
>-        runtime_config->frame_length_ms = frame_length_ms;
>-        runtime_config->uplink_packet_loss_fraction = plr;
>-        runtime_config->enable_fec = fec_enabled;
>-        runtime_config->enable_dtx = dtx_enabled;
>-        runtime_config->num_channels = channels;
>-
>-        TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
>-      }
>+  std::vector<std::unique_ptr<RtcEventAudioNetworkAdaptation>> events(
>+      event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    if (i == 0 || !force_repeated_fields_) {
>+      auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>+      runtime_config->bitrate_bps = rtc::checked_cast<int>(
>+          prng_.Rand(0, std::numeric_limits<int32_t>::max()));
>+      runtime_config->frame_length_ms = prng_.Rand(1, 1000);
>+      runtime_config->uplink_packet_loss_fraction =
>+          std::pow(0.5f, prng_.Rand(1, 8));
>+      runtime_config->enable_fec = prng_.Rand<bool>();
>+      runtime_config->enable_dtx = prng_.Rand<bool>();
>+      runtime_config->num_channels = prng_.Rand(1, 2);
>+      events[i] = absl::make_unique<RtcEventAudioNetworkAdaptation>(
>+          std::move(runtime_config));
>+    } else {
>+      events[i] = events[0]->Copy();
>     }
>   }
>+  TestRtcEventAudioNetworkAdaptation(events);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioPlayout) {
>-  uint32_t ssrc = prng_.Rand<uint32_t>();
>-  std::unique_ptr<RtcEventAudioPlayout> event = gen_.NewAudioPlayout(ssrc);
>-  history_.push_back(event->Copy());
>+  // SSRCs will be randomly assigned out of this small pool, significant only
>+  // in that it also covers such edge cases as SSRC = 0 and SSRC = 0xffffffff.
>+  // The pool is intentionally small, so as to produce collisions.
>+  const std::vector<uint32_t> kSsrcPool = {0x00000000, 0x12345678, 0xabcdef01,
>+                                           0xffffffff, 0x20171024, 0x19840730,
>+                                           0x19831230};
>+
>+  std::map<uint32_t, std::vector<std::unique_ptr<RtcEventAudioPlayout>>>
>+      original_events_by_ssrc;
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    const uint32_t ssrc = kSsrcPool[prng_.Rand(kSsrcPool.size() - 1)];
>+    std::unique_ptr<RtcEventAudioPlayout> event =
>+        (original_events_by_ssrc[ssrc].empty() || !force_repeated_fields_)
>+            ? gen_.NewAudioPlayout(ssrc)
>+            : original_events_by_ssrc[ssrc][0]->Copy();
>+    history_.push_back(event->Copy());
>+    original_events_by_ssrc[ssrc].push_back(std::move(event));
>+  }
> 
>   std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>   ASSERT_TRUE(parsed_log_.ParseString(encoded));
> 
>-  const auto& playout_events = parsed_log_.audio_playout_events();
>-  ASSERT_EQ(playout_events.size(), 1u);
>-  const auto playout_stream = playout_events.find(ssrc);
>-  ASSERT_TRUE(playout_stream != playout_events.end());
>-  ASSERT_EQ(playout_stream->second.size(), 1u);
>-  LoggedAudioPlayoutEvent playout_event = playout_stream->second[0];
>-  EXPECT_TRUE(test::VerifyLoggedAudioPlayoutEvent(*event, playout_event));
>+  const auto& parsed_playout_events_by_ssrc =
>+      parsed_log_.audio_playout_events();
>+
>+  // Same number of distinct SSRCs.
>+  ASSERT_EQ(parsed_playout_events_by_ssrc.size(),
>+            original_events_by_ssrc.size());
>+
>+  for (auto& original_event_it : original_events_by_ssrc) {
>+    const uint32_t ssrc = original_event_it.first;
>+    const auto& original_playout_events = original_event_it.second;
>+
>+    const auto& parsed_event_it = parsed_playout_events_by_ssrc.find(ssrc);
>+    ASSERT_TRUE(parsed_event_it != parsed_playout_events_by_ssrc.end());
>+    const auto& parsed_playout_events = parsed_event_it->second;
>+
>+    // Same number playout events for the SSRC under examination.
>+    ASSERT_EQ(original_playout_events.size(), parsed_playout_events.size());
>+
>+    for (size_t i = 0; i < original_playout_events.size(); ++i) {
>+      verifier_.VerifyLoggedAudioPlayoutEvent(*original_playout_events[i],
>+                                              parsed_playout_events[i]);
>+    }
>+  }
> }
> 
>+// TODO(eladalon/terelius): Test with multiple events in the batch.
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioReceiveStreamConfig) {
>   uint32_t ssrc = prng_.Rand<uint32_t>();
>   RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>@@ -201,9 +404,10 @@ TEST_P(RtcEventLogEncoderTest, RtcEventAudioReceiveStreamConfig) {
>   const auto& audio_recv_configs = parsed_log_.audio_recv_configs();
> 
>   ASSERT_EQ(audio_recv_configs.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedAudioRecvConfig(*event, audio_recv_configs[0]));
>+  verifier_.VerifyLoggedAudioRecvConfig(*event, audio_recv_configs[0]);
> }
> 
>+// TODO(eladalon/terelius): Test with multiple events in the batch.
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioSendStreamConfig) {
>   uint32_t ssrc = prng_.Rand<uint32_t>();
>   RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>@@ -216,37 +420,51 @@ TEST_P(RtcEventLogEncoderTest, RtcEventAudioSendStreamConfig) {
>   const auto& audio_send_configs = parsed_log_.audio_send_configs();
> 
>   ASSERT_EQ(audio_send_configs.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedAudioSendConfig(*event, audio_send_configs[0]));
>+  verifier_.VerifyLoggedAudioSendConfig(*event, audio_send_configs[0]);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventBweUpdateDelayBased) {
>-  std::unique_ptr<RtcEventBweUpdateDelayBased> event =
>-      gen_.NewBweUpdateDelayBased();
>-  history_.push_back(event->Copy());
>+  std::vector<std::unique_ptr<RtcEventBweUpdateDelayBased>> events(
>+      event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    events[i] = (i == 0 || !force_repeated_fields_)
>+                    ? gen_.NewBweUpdateDelayBased()
>+                    : events[0]->Copy();
>+    history_.push_back(events[i]->Copy());
>+  }
> 
>   std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>   ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+
>   const auto& bwe_delay_updates = parsed_log_.bwe_delay_updates();
>+  ASSERT_EQ(bwe_delay_updates.size(), event_count_);
> 
>-  ASSERT_EQ(bwe_delay_updates.size(), 1u);
>-  EXPECT_TRUE(
>-      test::VerifyLoggedBweDelayBasedUpdate(*event, bwe_delay_updates[0]));
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    verifier_.VerifyLoggedBweDelayBasedUpdate(*events[i], bwe_delay_updates[i]);
>+  }
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventBweUpdateLossBased) {
>-  std::unique_ptr<RtcEventBweUpdateLossBased> event =
>-      gen_.NewBweUpdateLossBased();
>-  history_.push_back(event->Copy());
>+  std::vector<std::unique_ptr<RtcEventBweUpdateLossBased>> events(event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    events[i] = (i == 0 || !force_repeated_fields_)
>+                    ? gen_.NewBweUpdateLossBased()
>+                    : events[0]->Copy();
>+    history_.push_back(events[i]->Copy());
>+  }
> 
>   std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>   ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+
>   const auto& bwe_loss_updates = parsed_log_.bwe_loss_updates();
>+  ASSERT_EQ(bwe_loss_updates.size(), event_count_);
> 
>-  ASSERT_EQ(bwe_loss_updates.size(), 1u);
>-  EXPECT_TRUE(
>-      test::VerifyLoggedBweLossBasedUpdate(*event, bwe_loss_updates[0]));
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    verifier_.VerifyLoggedBweLossBasedUpdate(*events[i], bwe_loss_updates[i]);
>+  }
> }
> 
>+// TODO(eladalon/terelius): Test with multiple events in the batch.
> TEST_P(RtcEventLogEncoderTest, RtcEventIceCandidatePairConfig) {
>   std::unique_ptr<RtcEventIceCandidatePairConfig> event =
>       gen_.NewIceCandidatePairConfig();
>@@ -258,10 +476,11 @@ TEST_P(RtcEventLogEncoderTest, RtcEventIceCandidatePairConfig) {
>       parsed_log_.ice_candidate_pair_configs();
> 
>   ASSERT_EQ(ice_candidate_pair_configs.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedIceCandidatePairConfig(
>-      *event, ice_candidate_pair_configs[0]));
>+  verifier_.VerifyLoggedIceCandidatePairConfig(*event,
>+                                               ice_candidate_pair_configs[0]);
> }
> 
>+// TODO(eladalon/terelius): Test with multiple events in the batch.
> TEST_P(RtcEventLogEncoderTest, RtcEventIceCandidatePair) {
>   std::unique_ptr<RtcEventIceCandidatePair> event = gen_.NewIceCandidatePair();
>   history_.push_back(event->Copy());
>@@ -272,18 +491,21 @@ TEST_P(RtcEventLogEncoderTest, RtcEventIceCandidatePair) {
>       parsed_log_.ice_candidate_pair_events();
> 
>   ASSERT_EQ(ice_candidate_pair_events.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedIceCandidatePairEvent(
>-      *event, ice_candidate_pair_events[0]));
>+  verifier_.VerifyLoggedIceCandidatePairEvent(*event,
>+                                              ice_candidate_pair_events[0]);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventLoggingStarted) {
>   const int64_t timestamp_us = rtc::TimeMicros();
>+  const int64_t utc_time_us = rtc::TimeUTCMicros();
> 
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->EncodeLogStart(timestamp_us)));
>+  ASSERT_TRUE(parsed_log_.ParseString(
>+      encoder_->EncodeLogStart(timestamp_us, utc_time_us)));
>   const auto& start_log_events = parsed_log_.start_log_events();
> 
>   ASSERT_EQ(start_log_events.size(), 1u);
>-  EXPECT_EQ(start_log_events[0].timestamp_us, timestamp_us);
>+  verifier_.VerifyLoggedStartEvent(timestamp_us, utc_time_us,
>+                                   start_log_events[0]);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventLoggingStopped) {
>@@ -293,9 +515,10 @@ TEST_P(RtcEventLogEncoderTest, RtcEventLoggingStopped) {
>   const auto& stop_log_events = parsed_log_.stop_log_events();
> 
>   ASSERT_EQ(stop_log_events.size(), 1u);
>-  EXPECT_EQ(stop_log_events[0].timestamp_us, timestamp_us);
>+  verifier_.VerifyLoggedStopEvent(timestamp_us, stop_log_events[0]);
> }
> 
>+// TODO(eladalon/terelius): Test with multiple events in the batch.
> TEST_P(RtcEventLogEncoderTest, RtcEventProbeClusterCreated) {
>   std::unique_ptr<RtcEventProbeClusterCreated> event =
>       gen_.NewProbeClusterCreated();
>@@ -307,10 +530,11 @@ TEST_P(RtcEventLogEncoderTest, RtcEventProbeClusterCreated) {
>       parsed_log_.bwe_probe_cluster_created_events();
> 
>   ASSERT_EQ(bwe_probe_cluster_created_events.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedBweProbeClusterCreatedEvent(
>-      *event, bwe_probe_cluster_created_events[0]));
>+  verifier_.VerifyLoggedBweProbeClusterCreatedEvent(
>+      *event, bwe_probe_cluster_created_events[0]);
> }
> 
>+// TODO(eladalon/terelius): Test with multiple events in the batch.
> TEST_P(RtcEventLogEncoderTest, RtcEventProbeResultFailure) {
>   std::unique_ptr<RtcEventProbeResultFailure> event =
>       gen_.NewProbeResultFailure();
>@@ -321,10 +545,11 @@ TEST_P(RtcEventLogEncoderTest, RtcEventProbeResultFailure) {
>   const auto& bwe_probe_failure_events = parsed_log_.bwe_probe_failure_events();
> 
>   ASSERT_EQ(bwe_probe_failure_events.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedBweProbeFailureEvent(
>-      *event, bwe_probe_failure_events[0]));
>+  verifier_.VerifyLoggedBweProbeFailureEvent(*event,
>+                                             bwe_probe_failure_events[0]);
> }
> 
>+// TODO(eladalon/terelius): Test with multiple events in the batch.
> TEST_P(RtcEventLogEncoderTest, RtcEventProbeResultSuccess) {
>   std::unique_ptr<RtcEventProbeResultSuccess> event =
>       gen_.NewProbeResultSuccess();
>@@ -335,78 +560,67 @@ TEST_P(RtcEventLogEncoderTest, RtcEventProbeResultSuccess) {
>   const auto& bwe_probe_success_events = parsed_log_.bwe_probe_success_events();
> 
>   ASSERT_EQ(bwe_probe_success_events.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedBweProbeSuccessEvent(
>-      *event, bwe_probe_success_events[0]));
>+  verifier_.VerifyLoggedBweProbeSuccessEvent(*event,
>+                                             bwe_probe_success_events[0]);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventRtcpPacketIncoming) {
>-  std::unique_ptr<RtcEventRtcpPacketIncoming> event =
>-      gen_.NewRtcpPacketIncoming();
>-  history_.push_back(event->Copy());
>+  if (!new_encoding_ && force_repeated_fields_) {
>+    // The old encoding does not work with duplicated packets. Since the legacy
>+    // encoding is being phased out, we will not fix this.
>+    return;
>+  }
>+
>+  std::vector<std::unique_ptr<RtcEventRtcpPacketIncoming>> events(event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    events[i] = (i == 0 || !force_repeated_fields_)
>+                    ? gen_.NewRtcpPacketIncoming()
>+                    : events[0]->Copy();
>+    history_.push_back(events[i]->Copy());
>+  }
> 
>   std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>   ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+
>   const auto& incoming_rtcp_packets = parsed_log_.incoming_rtcp_packets();
>+  ASSERT_EQ(incoming_rtcp_packets.size(), event_count_);
> 
>-  ASSERT_EQ(incoming_rtcp_packets.size(), 1u);
>-  EXPECT_TRUE(
>-      test::VerifyLoggedRtcpPacketIncoming(*event, incoming_rtcp_packets[0]));
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    verifier_.VerifyLoggedRtcpPacketIncoming(*events[i],
>+                                             incoming_rtcp_packets[i]);
>+  }
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventRtcpPacketOutgoing) {
>-  std::unique_ptr<RtcEventRtcpPacketOutgoing> event =
>-      gen_.NewRtcpPacketOutgoing();
>-  history_.push_back(event->Copy());
>+  std::vector<std::unique_ptr<RtcEventRtcpPacketOutgoing>> events(event_count_);
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    events[i] = (i == 0 || !force_repeated_fields_)
>+                    ? gen_.NewRtcpPacketOutgoing()
>+                    : events[0]->Copy();
>+    history_.push_back(events[i]->Copy());
>+  }
> 
>   std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>   ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+
>   const auto& outgoing_rtcp_packets = parsed_log_.outgoing_rtcp_packets();
>+  ASSERT_EQ(outgoing_rtcp_packets.size(), event_count_);
> 
>-  ASSERT_EQ(outgoing_rtcp_packets.size(), 1u);
>-  EXPECT_TRUE(
>-      test::VerifyLoggedRtcpPacketOutgoing(*event, outgoing_rtcp_packets[0]));
>+  for (size_t i = 0; i < event_count_; ++i) {
>+    verifier_.VerifyLoggedRtcpPacketOutgoing(*events[i],
>+                                             outgoing_rtcp_packets[i]);
>+  }
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventRtpPacketIncoming) {
>-  uint32_t ssrc = prng_.Rand<uint32_t>();
>-  RtpHeaderExtensionMap extension_map;  // TODO(terelius): Test extensions too.
>-  std::unique_ptr<RtcEventRtpPacketIncoming> event =
>-      gen_.NewRtpPacketIncoming(ssrc, extension_map);
>-  history_.push_back(event->Copy());
>-
>-  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>-  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>-  const auto& incoming_rtp_packets_by_ssrc =
>-      parsed_log_.incoming_rtp_packets_by_ssrc();
>-
>-  ASSERT_EQ(incoming_rtp_packets_by_ssrc.size(), 1u);
>-  const auto& stream = incoming_rtp_packets_by_ssrc[0];
>-  EXPECT_EQ(stream.ssrc, ssrc);
>-  ASSERT_EQ(stream.incoming_packets.size(), 1u);
>-  EXPECT_TRUE(
>-      test::VerifyLoggedRtpPacketIncoming(*event, stream.incoming_packets[0]));
>+  TestRtpPackets<RtcEventRtpPacketIncoming, LoggedRtpPacketIncoming>();
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventRtpPacketOutgoing) {
>-  uint32_t ssrc = prng_.Rand<uint32_t>();
>-  RtpHeaderExtensionMap extension_map;  // TODO(terelius): Test extensions too.
>-  std::unique_ptr<RtcEventRtpPacketOutgoing> event =
>-      gen_.NewRtpPacketOutgoing(ssrc, extension_map);
>-  history_.push_back(event->Copy());
>-
>-  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>-  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>-  const auto& outgoing_rtp_packets_by_ssrc =
>-      parsed_log_.outgoing_rtp_packets_by_ssrc();
>-
>-  ASSERT_EQ(outgoing_rtp_packets_by_ssrc.size(), 1u);
>-  const auto& stream = outgoing_rtp_packets_by_ssrc[0];
>-  EXPECT_EQ(stream.ssrc, ssrc);
>-  ASSERT_EQ(stream.outgoing_packets.size(), 1u);
>-  EXPECT_TRUE(
>-      test::VerifyLoggedRtpPacketOutgoing(*event, stream.outgoing_packets[0]));
>+  TestRtpPackets<RtcEventRtpPacketOutgoing, LoggedRtpPacketOutgoing>();
> }
> 
>+// TODO(eladalon/terelius): Test with multiple events in the batch.
> TEST_P(RtcEventLogEncoderTest, RtcEventVideoReceiveStreamConfig) {
>   uint32_t ssrc = prng_.Rand<uint32_t>();
>   RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>@@ -419,9 +633,10 @@ TEST_P(RtcEventLogEncoderTest, RtcEventVideoReceiveStreamConfig) {
>   const auto& video_recv_configs = parsed_log_.video_recv_configs();
> 
>   ASSERT_EQ(video_recv_configs.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedVideoRecvConfig(*event, video_recv_configs[0]));
>+  verifier_.VerifyLoggedVideoRecvConfig(*event, video_recv_configs[0]);
> }
> 
>+// TODO(eladalon/terelius): Test with multiple events in the batch.
> TEST_P(RtcEventLogEncoderTest, RtcEventVideoSendStreamConfig) {
>   uint32_t ssrc = prng_.Rand<uint32_t>();
>   RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>@@ -434,11 +649,15 @@ TEST_P(RtcEventLogEncoderTest, RtcEventVideoSendStreamConfig) {
>   const auto& video_send_configs = parsed_log_.video_send_configs();
> 
>   ASSERT_EQ(video_send_configs.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedVideoSendConfig(*event, video_send_configs[0]));
>+  verifier_.VerifyLoggedVideoSendConfig(*event, video_send_configs[0]);
> }
> 
>-INSTANTIATE_TEST_CASE_P(RandomSeeds,
>-                        RtcEventLogEncoderTest,
>-                        ::testing::Values(1, 2, 3, 4, 5));
>+INSTANTIATE_TEST_CASE_P(
>+    RandomSeeds,
>+    RtcEventLogEncoderTest,
>+    ::testing::Combine(/* Random seed*: */ ::testing::Values(1, 2, 3, 4, 5),
>+                       /* Encoding: */ ::testing::Bool(),
>+                       /* Event count: */ ::testing::Values(1, 2, 10, 100),
>+                       /* Repeated fields: */ ::testing::Bool()));
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/varint.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/varint.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..41e29e8480bd5aa11ca2e1b64057bda86dcf43ba
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/varint.cc
>@@ -0,0 +1,80 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/encoder/varint.h"
>+
>+#include "rtc_base/checks.h"
>+
>+// TODO(eladalon): Add unit tests.
>+
>+namespace webrtc {
>+
>+const size_t kMaxVarIntLengthBytes = 10;  // ceil(64 / 7.0) is 10.
>+
>+std::string EncodeVarInt(uint64_t input) {
>+  std::string output;
>+  output.reserve(kMaxVarIntLengthBytes);
>+
>+  do {
>+    uint8_t byte = static_cast<uint8_t>(input & 0x7f);
>+    input >>= 7;
>+    if (input > 0) {
>+      byte |= 0x80;
>+    }
>+    output += byte;
>+  } while (input > 0);
>+
>+  RTC_DCHECK_GE(output.size(), 1u);
>+  RTC_DCHECK_LE(output.size(), kMaxVarIntLengthBytes);
>+
>+  return output;
>+}
>+
>+// There is some code duplication between the flavors of this function.
>+// For performance's sake, it's best to just keep it.
>+size_t DecodeVarInt(absl::string_view input, uint64_t* output) {
>+  RTC_DCHECK(output);
>+
>+  uint64_t decoded = 0;
>+  for (size_t i = 0; i < input.length() && i < kMaxVarIntLengthBytes; ++i) {
>+    decoded += (static_cast<uint64_t>(input[i] & 0x7f)
>+                << static_cast<uint64_t>(7 * i));
>+    if (!(input[i] & 0x80)) {
>+      *output = decoded;
>+      return i + 1;
>+    }
>+  }
>+
>+  return 0;
>+}
>+
>+// There is some code duplication between the flavors of this function.
>+// For performance's sake, it's best to just keep it.
>+size_t DecodeVarInt(rtc::BitBuffer* input, uint64_t* output) {
>+  RTC_DCHECK(output);
>+
>+  uint64_t decoded = 0;
>+  for (size_t i = 0; i < kMaxVarIntLengthBytes; ++i) {
>+    uint8_t byte;
>+    if (!input->ReadUInt8(&byte)) {
>+      return 0;
>+    }
>+    decoded +=
>+        (static_cast<uint64_t>(byte & 0x7f) << static_cast<uint64_t>(7 * i));
>+    if (!(byte & 0x80)) {
>+      *output = decoded;
>+      return i + 1;
>+    }
>+  }
>+
>+  return 0;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/varint.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/varint.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..95494ae753fc2587e7cbb0e3facc94ecf13682bf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/varint.h
>@@ -0,0 +1,45 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef LOGGING_RTC_EVENT_LOG_ENCODER_VARINT_H_
>+#define LOGGING_RTC_EVENT_LOG_ENCODER_VARINT_H_
>+
>+#include <string>
>+
>+#include "absl/strings/string_view.h"
>+#include "rtc_base/bitbuffer.h"
>+
>+namespace webrtc {
>+
>+extern const size_t kMaxVarIntLengthBytes;
>+
>+// Encode a given uint64_t as a varint. From least to most significant,
>+// each batch of seven bits are put into the lower bits of a byte, and the last
>+// remaining bit in that byte (the highest one) marks whether additional bytes
>+// follow (which happens if and only if there are other bits in |input| which
>+// are non-zero).
>+// Notes: If input == 0, one byte is used. If input is uint64_t::max, exactly
>+// kMaxVarIntLengthBytes are used.
>+std::string EncodeVarInt(uint64_t input);
>+
>+// Inverse of EncodeVarInt().
>+// If decoding is successful, a non-zero number is returned, indicating the
>+// number of bytes read from |input|, and the decoded varint is written
>+// into |output|.
>+// If not successful, 0 is returned, and |output| is not modified.
>+size_t DecodeVarInt(absl::string_view input, uint64_t* output);
>+
>+// Same as other version, but uses a rtc::BitBuffer for input.
>+// Some bits may be consumed even if a varint fails to be read.
>+size_t DecodeVarInt(rtc::BitBuffer* input, uint64_t* output);
>+
>+}  // namespace webrtc
>+
>+#endif  // LOGGING_RTC_EVENT_LOG_ENCODER_VARINT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event.h
>index 21d6e48e0f52904500492f3aeaa9ec6d3b967f47..7ae63758b158ab306e11f0550c898d8dc58d7ae3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event.h
>@@ -57,12 +57,13 @@ class RtcEvent {
> 
>   virtual bool IsConfigEvent() const = 0;
> 
>-  virtual std::unique_ptr<RtcEvent> Copy() const = 0;
>-
>-  const int64_t timestamp_us_;
>+  int64_t timestamp_ms() const { return timestamp_us_ / 1000; }
>+  int64_t timestamp_us() const { return timestamp_us_; }
> 
>  protected:
>   explicit RtcEvent(int64_t timestamp_us) : timestamp_us_(timestamp_us) {}
>+
>+  const int64_t timestamp_us_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.cc
>index c1f1b39532033010b5c6c81b4f48656c1685d66c..8e5c2ca82f71cc9fdeff8bb5d756ba64da72e616 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.cc
>@@ -28,8 +28,8 @@ bool RtcEventAlrState::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventAlrState::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventAlrState(*this));
>+std::unique_ptr<RtcEventAlrState> RtcEventAlrState::Copy() const {
>+  return absl::WrapUnique<RtcEventAlrState>(new RtcEventAlrState(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.h
>index e737c915e1d561118fc0eb46470e8dae6ff68c73..9769585d3d6826477d2e8a7567618047a3158808 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.h
>@@ -26,12 +26,14 @@ class RtcEventAlrState final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventAlrState> Copy() const;
> 
>-  const bool in_alr_;
>+  bool in_alr() const { return in_alr_; }
> 
>  private:
>   RtcEventAlrState(const RtcEventAlrState& other);
>+
>+  const bool in_alr_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc
>index dcf2d7de59224af2d64c111cebd6fb445ad5279a..571b9a93d3eb7dee9e62c037ed9684988d3e48f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc
>@@ -14,12 +14,15 @@
> 
> #include "absl/memory/memory.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
> RtcEventAudioNetworkAdaptation::RtcEventAudioNetworkAdaptation(
>     std::unique_ptr<AudioEncoderRuntimeConfig> config)
>-    : config_(std::move(config)) {}
>+    : config_(std::move(config)) {
>+  RTC_DCHECK(config_);
>+}
> 
> RtcEventAudioNetworkAdaptation::RtcEventAudioNetworkAdaptation(
>     const RtcEventAudioNetworkAdaptation& other)
>@@ -36,7 +39,8 @@ bool RtcEventAudioNetworkAdaptation::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventAudioNetworkAdaptation::Copy() const {
>+std::unique_ptr<RtcEventAudioNetworkAdaptation>
>+RtcEventAudioNetworkAdaptation::Copy() const {
>   return absl::WrapUnique(new RtcEventAudioNetworkAdaptation(*this));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
>index 65fbad8621d03ea7d6f2ba26cc5f7592255a1624..ec6ca1125c40a3e6ae2f95e7571927b451adb5ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
>@@ -29,12 +29,14 @@ class RtcEventAudioNetworkAdaptation final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventAudioNetworkAdaptation> Copy() const;
> 
>-  const std::unique_ptr<const AudioEncoderRuntimeConfig> config_;
>+  const AudioEncoderRuntimeConfig& config() const { return *config_; }
> 
>  private:
>   RtcEventAudioNetworkAdaptation(const RtcEventAudioNetworkAdaptation& other);
>+
>+  const std::unique_ptr<const AudioEncoderRuntimeConfig> config_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.cc
>index a9643ba868c038d81d0a3557239f9d8f2a0e3fc8..6c4aa98d3e238cac3253df2533da0c1063f22288 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.cc
>@@ -27,8 +27,9 @@ bool RtcEventAudioPlayout::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventAudioPlayout::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventAudioPlayout(*this));
>+std::unique_ptr<RtcEventAudioPlayout> RtcEventAudioPlayout::Copy() const {
>+  return absl::WrapUnique<RtcEventAudioPlayout>(
>+      new RtcEventAudioPlayout(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h
>index 3080f5b9f0e72eb1add3a9d216e65ced7ecf5363..4b40f5c02728aa6b4f203ddc01f6c97335d5df8f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h
>@@ -11,6 +11,7 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_PLAYOUT_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_PLAYOUT_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "logging/rtc_event_log/events/rtc_event.h"
>@@ -26,12 +27,14 @@ class RtcEventAudioPlayout final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventAudioPlayout> Copy() const;
> 
>-  const uint32_t ssrc_;
>+  uint32_t ssrc() const { return ssrc_; }
> 
>  private:
>   RtcEventAudioPlayout(const RtcEventAudioPlayout& other);
>+
>+  const uint32_t ssrc_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.cc
>index fee7c3c766c3d1da48beb88c07e50e03a0ed0555..fdef393d546c1a891662a2554d2d0779a91775e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.cc
>@@ -14,12 +14,15 @@
> 
> #include "absl/memory/memory.h"
> #include "logging/rtc_event_log/rtc_stream_config.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
> RtcEventAudioReceiveStreamConfig::RtcEventAudioReceiveStreamConfig(
>     std::unique_ptr<rtclog::StreamConfig> config)
>-    : config_(std::move(config)) {}
>+    : config_(std::move(config)) {
>+  RTC_DCHECK(config_);
>+}
> 
> RtcEventAudioReceiveStreamConfig::RtcEventAudioReceiveStreamConfig(
>     const RtcEventAudioReceiveStreamConfig& other)
>@@ -36,9 +39,9 @@ bool RtcEventAudioReceiveStreamConfig::IsConfigEvent() const {
>   return true;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventAudioReceiveStreamConfig::Copy() const {
>-  auto config_copy = absl::make_unique<rtclog::StreamConfig>(*config_);
>-  return absl::WrapUnique<RtcEvent>(
>+std::unique_ptr<RtcEventAudioReceiveStreamConfig>
>+RtcEventAudioReceiveStreamConfig::Copy() const {
>+  return absl::WrapUnique<RtcEventAudioReceiveStreamConfig>(
>       new RtcEventAudioReceiveStreamConfig(*this));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h
>index 275765031436209d299fe7add7561dd3c3418b55..2b73f63f000dff06f94f39bba0cc3b3a76348fa2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h
>@@ -31,13 +31,15 @@ class RtcEventAudioReceiveStreamConfig final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventAudioReceiveStreamConfig> Copy() const;
> 
>-  const std::unique_ptr<const rtclog::StreamConfig> config_;
>+  const rtclog::StreamConfig& config() const { return *config_; }
> 
>  private:
>   RtcEventAudioReceiveStreamConfig(
>       const RtcEventAudioReceiveStreamConfig& other);
>+
>+  const std::unique_ptr<const rtclog::StreamConfig> config_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.cc
>index 47939c7bdc32ef51af0b9092d0499e5ba88e4944..f1a85bff6946500446fe8e318b9aa0fae1c3c431 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.cc
>@@ -14,12 +14,15 @@
> 
> #include "absl/memory/memory.h"
> #include "logging/rtc_event_log/rtc_stream_config.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
> RtcEventAudioSendStreamConfig::RtcEventAudioSendStreamConfig(
>     std::unique_ptr<rtclog::StreamConfig> config)
>-    : config_(std::move(config)) {}
>+    : config_(std::move(config)) {
>+  RTC_DCHECK(config_);
>+}
> 
> RtcEventAudioSendStreamConfig::RtcEventAudioSendStreamConfig(
>     const RtcEventAudioSendStreamConfig& other)
>@@ -36,9 +39,10 @@ bool RtcEventAudioSendStreamConfig::IsConfigEvent() const {
>   return true;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventAudioSendStreamConfig::Copy() const {
>-  auto config_copy = absl::make_unique<rtclog::StreamConfig>(*config_);
>-  return absl::WrapUnique<RtcEvent>(new RtcEventAudioSendStreamConfig(*this));
>+std::unique_ptr<RtcEventAudioSendStreamConfig>
>+RtcEventAudioSendStreamConfig::Copy() const {
>+  return absl::WrapUnique<RtcEventAudioSendStreamConfig>(
>+      new RtcEventAudioSendStreamConfig(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h
>index b248a61915019e97a6e758977ba602a95c0dfc31..c0efa950901bbedc205de431d5de1a68d626dafb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h
>@@ -31,12 +31,14 @@ class RtcEventAudioSendStreamConfig final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventAudioSendStreamConfig> Copy() const;
> 
>-  const std::unique_ptr<const rtclog::StreamConfig> config_;
>+  const rtclog::StreamConfig& config() const { return *config_; }
> 
>  private:
>   RtcEventAudioSendStreamConfig(const RtcEventAudioSendStreamConfig& other);
>+
>+  const std::unique_ptr<const rtclog::StreamConfig> config_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.cc
>index 8c17971260520bbaf32a23610de7d11aa56fd359..dcc87421f8737d7e33fc6de2cf6e306dcf1f2be3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.cc
>@@ -36,8 +36,10 @@ bool RtcEventBweUpdateDelayBased::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventBweUpdateDelayBased::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventBweUpdateDelayBased(*this));
>+std::unique_ptr<RtcEventBweUpdateDelayBased> RtcEventBweUpdateDelayBased::Copy()
>+    const {
>+  return absl::WrapUnique<RtcEventBweUpdateDelayBased>(
>+      new RtcEventBweUpdateDelayBased(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h
>index 3f47dc7f97045d279eda22bd9bb8a4fb621d983b..a97d49f5a54e47bc0982bf9455b1d3dd4141c93a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h
>@@ -11,6 +11,7 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_DELAY_BASED_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_DELAY_BASED_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "logging/rtc_event_log/events/rtc_event.h"
>@@ -29,13 +30,16 @@ class RtcEventBweUpdateDelayBased final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventBweUpdateDelayBased> Copy() const;
> 
>-  const int32_t bitrate_bps_;
>-  const BandwidthUsage detector_state_;
>+  int32_t bitrate_bps() const { return bitrate_bps_; }
>+  BandwidthUsage detector_state() const { return detector_state_; }
> 
>  private:
>   RtcEventBweUpdateDelayBased(const RtcEventBweUpdateDelayBased& other);
>+
>+  const int32_t bitrate_bps_;
>+  const BandwidthUsage detector_state_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.cc
>index b0e72ddb0aa794fd67333b06524713a18cdf698c..8453238cf903b38e914529131108eb8a5852e995 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.cc
>@@ -38,8 +38,10 @@ bool RtcEventBweUpdateLossBased::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventBweUpdateLossBased::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventBweUpdateLossBased(*this));
>+std::unique_ptr<RtcEventBweUpdateLossBased> RtcEventBweUpdateLossBased::Copy()
>+    const {
>+  return absl::WrapUnique<RtcEventBweUpdateLossBased>(
>+      new RtcEventBweUpdateLossBased(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h
>index 2d4ffda1b1ea6ed66d01800de749d6a1d42887d8..b4f88775fb72eaa9618aaa919ec136cc5d83fea8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h
>@@ -11,6 +11,7 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_LOSS_BASED_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_LOSS_BASED_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "logging/rtc_event_log/events/rtc_event.h"
>@@ -28,14 +29,18 @@ class RtcEventBweUpdateLossBased final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventBweUpdateLossBased> Copy() const;
> 
>-  const int32_t bitrate_bps_;
>-  const uint8_t fraction_loss_;
>-  const int32_t total_packets_;
>+  int32_t bitrate_bps() const { return bitrate_bps_; }
>+  uint8_t fraction_loss() const { return fraction_loss_; }
>+  int32_t total_packets() const { return total_packets_; }
> 
>  private:
>   RtcEventBweUpdateLossBased(const RtcEventBweUpdateLossBased& other);
>+
>+  const int32_t bitrate_bps_;
>+  const uint8_t fraction_loss_;
>+  const int32_t total_packets_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.cc
>index fa32de68d5eab7c1a2ff7e5736753b0150b26a01..9c31737ea5ee611351807e1b04057d0c5627ec62 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.cc
>@@ -35,8 +35,10 @@ bool RtcEventIceCandidatePair::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventIceCandidatePair::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventIceCandidatePair(*this));
>+std::unique_ptr<RtcEventIceCandidatePair> RtcEventIceCandidatePair::Copy()
>+    const {
>+  return absl::WrapUnique<RtcEventIceCandidatePair>(
>+      new RtcEventIceCandidatePair(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h
>index f49a0d1792cb431befc37badc6b1693dfab3ab11..7414fec79b1e8b062d204d5786746df1473ca5c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h
>@@ -36,13 +36,16 @@ class RtcEventIceCandidatePair final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventIceCandidatePair> Copy() const;
> 
>-  const IceCandidatePairEventType type_;
>-  const uint32_t candidate_pair_id_;
>+  IceCandidatePairEventType type() const { return type_; }
>+  uint32_t candidate_pair_id() const { return candidate_pair_id_; }
> 
>  private:
>   RtcEventIceCandidatePair(const RtcEventIceCandidatePair& other);
>+
>+  const IceCandidatePairEventType type_;
>+  const uint32_t candidate_pair_id_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc
>index f89de7740c7be98ebd62f8f0a6eef3c25d087dfb..fbb8a73dfbea59f66ff3bf56e6454c74454f55d6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc
>@@ -64,8 +64,10 @@ bool RtcEventIceCandidatePairConfig::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventIceCandidatePairConfig::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventIceCandidatePairConfig(*this));
>+std::unique_ptr<RtcEventIceCandidatePairConfig>
>+RtcEventIceCandidatePairConfig::Copy() const {
>+  return absl::WrapUnique<RtcEventIceCandidatePairConfig>(
>+      new RtcEventIceCandidatePairConfig(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h
>index 1f6eb8ceb0fffe8e87d1fe489e26e23ef2bb96ec..40bdbaa771e81c0cfb6a9d0132b7cdef42b9cd70 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h
>@@ -92,14 +92,20 @@ class RtcEventIceCandidatePairConfig final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventIceCandidatePairConfig> Copy() const;
> 
>-  const IceCandidatePairConfigType type_;
>-  const uint32_t candidate_pair_id_;
>-  const IceCandidatePairDescription candidate_pair_desc_;
>+  IceCandidatePairConfigType type() const { return type_; }
>+  uint32_t candidate_pair_id() const { return candidate_pair_id_; }
>+  const IceCandidatePairDescription& candidate_pair_desc() const {
>+    return candidate_pair_desc_;
>+  }
> 
>  private:
>   RtcEventIceCandidatePairConfig(const RtcEventIceCandidatePairConfig& other);
>+
>+  const IceCandidatePairConfigType type_;
>+  const uint32_t candidate_pair_id_;
>+  const IceCandidatePairDescription candidate_pair_desc_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.cc
>index a60765dbd0f1beff9bdbbb44f134d24cff8f4e5e..c11a6ce78eb56e0bf9363b4cf8a1a6b2aacdfe4e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.cc
>@@ -39,8 +39,10 @@ bool RtcEventProbeClusterCreated::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventProbeClusterCreated::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventProbeClusterCreated(*this));
>+std::unique_ptr<RtcEventProbeClusterCreated> RtcEventProbeClusterCreated::Copy()
>+    const {
>+  return absl::WrapUnique<RtcEventProbeClusterCreated>(
>+      new RtcEventProbeClusterCreated(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h
>index 3e8c21f5c9219839673334eb0816dca1251450f0..ad757edd530e0cf85a15e18e44c25cf15d401865 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h
>@@ -11,6 +11,7 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_CLUSTER_CREATED_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_CLUSTER_CREATED_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "logging/rtc_event_log/events/rtc_event.h"
>@@ -29,15 +30,20 @@ class RtcEventProbeClusterCreated final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventProbeClusterCreated> Copy() const;
>+
>+  int32_t id() const { return id_; }
>+  int32_t bitrate_bps() const { return bitrate_bps_; }
>+  uint32_t min_probes() const { return min_probes_; }
>+  uint32_t min_bytes() const { return min_bytes_; }
>+
>+ private:
>+  RtcEventProbeClusterCreated(const RtcEventProbeClusterCreated& other);
> 
>   const int32_t id_;
>   const int32_t bitrate_bps_;
>   const uint32_t min_probes_;
>   const uint32_t min_bytes_;
>-
>- private:
>-  RtcEventProbeClusterCreated(const RtcEventProbeClusterCreated& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc
>index aa6091ca775b54b92db908a71e011546565f28b6..295003ae8d049114114e2494f48e7058b1b7b428 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc
>@@ -33,8 +33,10 @@ bool RtcEventProbeResultFailure::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventProbeResultFailure::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventProbeResultFailure(*this));
>+std::unique_ptr<RtcEventProbeResultFailure> RtcEventProbeResultFailure::Copy()
>+    const {
>+  return absl::WrapUnique<RtcEventProbeResultFailure>(
>+      new RtcEventProbeResultFailure(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.h
>index 181c6940370ddd0636c99e21e552b965c34f2898..0c40f0cd1b11eb9309fc6a6acf7f9c5bdb67ecce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.h
>@@ -11,6 +11,7 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_RESULT_FAILURE_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_RESULT_FAILURE_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "logging/rtc_event_log/events/rtc_event.h"
>@@ -33,13 +34,16 @@ class RtcEventProbeResultFailure final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventProbeResultFailure> Copy() const;
> 
>-  const int32_t id_;
>-  const ProbeFailureReason failure_reason_;
>+  int32_t id() const { return id_; }
>+  ProbeFailureReason failure_reason() const { return failure_reason_; }
> 
>  private:
>   RtcEventProbeResultFailure(const RtcEventProbeResultFailure& other);
>+
>+  const int32_t id_;
>+  const ProbeFailureReason failure_reason_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.cc
>index 55907295debd613d93508dbe75b6d10e6cc3a2cc..d5f9e2f780ca977566f41c4c3334a5e1442d68dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.cc
>@@ -32,8 +32,10 @@ bool RtcEventProbeResultSuccess::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventProbeResultSuccess::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventProbeResultSuccess(*this));
>+std::unique_ptr<RtcEventProbeResultSuccess> RtcEventProbeResultSuccess::Copy()
>+    const {
>+  return absl::WrapUnique<RtcEventProbeResultSuccess>(
>+      new RtcEventProbeResultSuccess(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.h
>index 15ff1833bf04d1c62ada136ed3803dbfdcbd59a9..a08dadd212ba823f058fd2925328db60036ac3e0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.h
>@@ -11,6 +11,7 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_RESULT_SUCCESS_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_RESULT_SUCCESS_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "logging/rtc_event_log/events/rtc_event.h"
>@@ -26,13 +27,16 @@ class RtcEventProbeResultSuccess final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventProbeResultSuccess> Copy() const;
> 
>-  const int32_t id_;
>-  const int32_t bitrate_bps_;
>+  int32_t id() const { return id_; }
>+  int32_t bitrate_bps() const { return bitrate_bps_; }
> 
>  private:
>   RtcEventProbeResultSuccess(const RtcEventProbeResultSuccess& other);
>+
>+  const int32_t id_;
>+  const int32_t bitrate_bps_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc
>index 119938c7f71884809d5482b0008411c160cd76e3..45a418f1f69c2bc786086cfde9909fa71bf4282f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc
>@@ -33,8 +33,10 @@ bool RtcEventRtcpPacketIncoming::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventRtcpPacketIncoming::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventRtcpPacketIncoming(*this));
>+std::unique_ptr<RtcEventRtcpPacketIncoming> RtcEventRtcpPacketIncoming::Copy()
>+    const {
>+  return absl::WrapUnique<RtcEventRtcpPacketIncoming>(
>+      new RtcEventRtcpPacketIncoming(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h
>index 6d904953a48a8876c2370ebd7e08927233c2677e..8394fe005cd56382acf63a12e390938e06da4f18 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h
>@@ -11,6 +11,7 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTCP_PACKET_INCOMING_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTCP_PACKET_INCOMING_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "api/array_view.h"
>@@ -28,12 +29,14 @@ class RtcEventRtcpPacketIncoming final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventRtcpPacketIncoming> Copy() const;
> 
>-  rtc::Buffer packet_;
>+  const rtc::Buffer& packet() const { return packet_; }
> 
>  private:
>   RtcEventRtcpPacketIncoming(const RtcEventRtcpPacketIncoming& other);
>+
>+  rtc::Buffer packet_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc
>index 4085ac0f31362aef9d5e57f2129095de93e6a744..b583e5614a2040ce8ea724cf0ed96a8eb3fe2449 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc
>@@ -33,8 +33,10 @@ bool RtcEventRtcpPacketOutgoing::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventRtcpPacketOutgoing::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventRtcpPacketOutgoing(*this));
>+std::unique_ptr<RtcEventRtcpPacketOutgoing> RtcEventRtcpPacketOutgoing::Copy()
>+    const {
>+  return absl::WrapUnique<RtcEventRtcpPacketOutgoing>(
>+      new RtcEventRtcpPacketOutgoing(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h
>index 3aa9e7199ad633c1e7661b492613abdf6008e8de..b47b85d0b3635d18b071e465d6b864ffec09099f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h
>@@ -11,6 +11,7 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTCP_PACKET_OUTGOING_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTCP_PACKET_OUTGOING_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "api/array_view.h"
>@@ -28,12 +29,14 @@ class RtcEventRtcpPacketOutgoing final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventRtcpPacketOutgoing> Copy() const;
> 
>-  rtc::Buffer packet_;
>+  const rtc::Buffer& packet() const { return packet_; }
> 
>  private:
>   RtcEventRtcpPacketOutgoing(const RtcEventRtcpPacketOutgoing& other);
>+
>+  rtc::Buffer packet_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.cc
>index 8e15d021751b685cf9a5ccead4b2178e818792a9..898c0aaf8ee64f15b0f4ac6e8799346dddbcb428 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.cc
>@@ -12,18 +12,26 @@
> 
> #include "absl/memory/memory.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
> RtcEventRtpPacketIncoming::RtcEventRtpPacketIncoming(
>     const RtpPacketReceived& packet)
>-    : packet_length_(packet.size()) {
>+    : payload_length_(packet.payload_size()),
>+      header_length_(packet.headers_size()),
>+      padding_length_(packet.padding_size()) {
>   header_.CopyHeaderFrom(packet);
>+  RTC_DCHECK_EQ(packet.size(),
>+                payload_length_ + header_length_ + padding_length_);
> }
> 
> RtcEventRtpPacketIncoming::RtcEventRtpPacketIncoming(
>     const RtcEventRtpPacketIncoming& other)
>-    : RtcEvent(other.timestamp_us_), packet_length_(other.packet_length_) {
>+    : RtcEvent(other.timestamp_us_),
>+      payload_length_(other.payload_length_),
>+      header_length_(other.header_length_),
>+      padding_length_(other.padding_length_) {
>   header_.CopyHeaderFrom(other.header_);
> }
> 
>@@ -37,8 +45,10 @@ bool RtcEventRtpPacketIncoming::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventRtpPacketIncoming::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventRtpPacketIncoming(*this));
>+std::unique_ptr<RtcEventRtpPacketIncoming> RtcEventRtpPacketIncoming::Copy()
>+    const {
>+  return absl::WrapUnique<RtcEventRtpPacketIncoming>(
>+      new RtcEventRtpPacketIncoming(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h
>index 6b141b6d57622e6d29b19a61749fb6cb336139c5..1e357351b3cf26da031c8f1b98f95f2389faff75 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h
>@@ -29,13 +29,24 @@ class RtcEventRtpPacketIncoming final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventRtpPacketIncoming> Copy() const;
> 
>-  RtpPacket header_;            // Only the packet's header will be stored here.
>-  const size_t packet_length_;  // Length before stripping away all but header.
>+  size_t packet_length() const {
>+    return payload_length_ + header_length_ + padding_length_;
>+  }
>+
>+  const RtpPacket& header() const { return header_; }
>+  size_t payload_length() const { return payload_length_; }
>+  size_t header_length() const { return header_length_; }
>+  size_t padding_length() const { return padding_length_; }
> 
>  private:
>   RtcEventRtpPacketIncoming(const RtcEventRtpPacketIncoming& other);
>+
>+  RtpPacket header_;            // Only the packet's header will be stored here.
>+  const size_t payload_length_;  // Media payload, excluding header and padding.
>+  const size_t header_length_;   // RTP header.
>+  const size_t padding_length_;  // RTP padding.
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc
>index 103440ad6e9661daded0fe1feac25a022df0da3f..050474edda7c02c0cff4f34730b0ad6a7f4d9f7d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc
>@@ -12,20 +12,28 @@
> 
> #include "absl/memory/memory.h"
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
> RtcEventRtpPacketOutgoing::RtcEventRtpPacketOutgoing(
>     const RtpPacketToSend& packet,
>     int probe_cluster_id)
>-    : packet_length_(packet.size()), probe_cluster_id_(probe_cluster_id) {
>+    : payload_length_(packet.payload_size()),
>+      header_length_(packet.headers_size()),
>+      padding_length_(packet.padding_size()),
>+      probe_cluster_id_(probe_cluster_id) {
>   header_.CopyHeaderFrom(packet);
>+  RTC_DCHECK_EQ(packet.size(),
>+                payload_length_ + header_length_ + padding_length_);
> }
> 
> RtcEventRtpPacketOutgoing::RtcEventRtpPacketOutgoing(
>     const RtcEventRtpPacketOutgoing& other)
>     : RtcEvent(other.timestamp_us_),
>-      packet_length_(other.packet_length_),
>+      payload_length_(other.payload_length_),
>+      header_length_(other.header_length_),
>+      padding_length_(other.padding_length_),
>       probe_cluster_id_(other.probe_cluster_id_) {
>   header_.CopyHeaderFrom(other.header_);
> }
>@@ -40,8 +48,10 @@ bool RtcEventRtpPacketOutgoing::IsConfigEvent() const {
>   return false;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventRtpPacketOutgoing::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventRtpPacketOutgoing(*this));
>+std::unique_ptr<RtcEventRtpPacketOutgoing> RtcEventRtpPacketOutgoing::Copy()
>+    const {
>+  return absl::WrapUnique<RtcEventRtpPacketOutgoing>(
>+      new RtcEventRtpPacketOutgoing(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h
>index 171a21c420e41676f3f33050cacc1c6e2d61c455..ebddc14231f3af193f2d0223d337966f017c5048 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h
>@@ -30,14 +30,27 @@ class RtcEventRtpPacketOutgoing final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventRtpPacketOutgoing> Copy() const;
> 
>-  RtpPacket header_;            // Only the packet's header will be stored here.
>-  const size_t packet_length_;  // Length before stripping away all but header.
>-  const int probe_cluster_id_;
>+  size_t packet_length() const {
>+    return payload_length_ + header_length_ + padding_length_;
>+  }
>+
>+  const RtpPacket& header() const { return header_; }
>+  size_t payload_length() const { return payload_length_; }
>+  size_t header_length() const { return header_length_; }
>+  size_t padding_length() const { return padding_length_; }
>+  int probe_cluster_id() const { return probe_cluster_id_; }
> 
>  private:
>   RtcEventRtpPacketOutgoing(const RtcEventRtpPacketOutgoing& other);
>+
>+  RtpPacket header_;            // Only the packet's header will be stored here.
>+  const size_t payload_length_;  // Media payload, excluding header and padding.
>+  const size_t header_length_;   // RTP header.
>+  const size_t padding_length_;  // RTP padding.
>+  // TODO(eladalon): Delete |probe_cluster_id_| along with legacy encoding.
>+  const int probe_cluster_id_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.cc
>index 7f4dd9b7b807c4b12fd389bd426d4fdeb6c58640..5dec97bf83772e912b1986b02221c55d9e88e2a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.cc
>@@ -13,12 +13,15 @@
> #include <utility>
> 
> #include "absl/memory/memory.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
> RtcEventVideoReceiveStreamConfig::RtcEventVideoReceiveStreamConfig(
>     std::unique_ptr<rtclog::StreamConfig> config)
>-    : config_(std::move(config)) {}
>+    : config_(std::move(config)) {
>+  RTC_DCHECK(config_);
>+}
> 
> RtcEventVideoReceiveStreamConfig::RtcEventVideoReceiveStreamConfig(
>     const RtcEventVideoReceiveStreamConfig& other)
>@@ -35,8 +38,9 @@ bool RtcEventVideoReceiveStreamConfig::IsConfigEvent() const {
>   return true;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventVideoReceiveStreamConfig::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(
>+std::unique_ptr<RtcEventVideoReceiveStreamConfig>
>+RtcEventVideoReceiveStreamConfig::Copy() const {
>+  return absl::WrapUnique<RtcEventVideoReceiveStreamConfig>(
>       new RtcEventVideoReceiveStreamConfig(*this));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h
>index c6263c3fdbe22c7ccb691d7be882550235ff4482..801ba7ddd44d81468e4154f61c271a625e12babc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h
>@@ -28,13 +28,15 @@ class RtcEventVideoReceiveStreamConfig final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventVideoReceiveStreamConfig> Copy() const;
> 
>-  const std::unique_ptr<const rtclog::StreamConfig> config_;
>+  const rtclog::StreamConfig& config() const { return *config_; }
> 
>  private:
>   RtcEventVideoReceiveStreamConfig(
>       const RtcEventVideoReceiveStreamConfig& other);
>+
>+  const std::unique_ptr<const rtclog::StreamConfig> config_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.cc
>index b4b78a8a52a75c13ccec6f714a13e30535465bbd..dc4b167438cba793e1aec888adc16472cbf89c5f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.cc
>@@ -35,8 +35,10 @@ bool RtcEventVideoSendStreamConfig::IsConfigEvent() const {
>   return true;
> }
> 
>-std::unique_ptr<RtcEvent> RtcEventVideoSendStreamConfig::Copy() const {
>-  return absl::WrapUnique<RtcEvent>(new RtcEventVideoSendStreamConfig(*this));
>+std::unique_ptr<RtcEventVideoSendStreamConfig>
>+RtcEventVideoSendStreamConfig::Copy() const {
>+  return absl::WrapUnique<RtcEventVideoSendStreamConfig>(
>+      new RtcEventVideoSendStreamConfig(*this));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h
>index 3472f8855be669dd13ca68eb651e18853a53eb0f..fe274c8c3a55fd350f83ad500cea482062b296cb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h
>@@ -28,12 +28,14 @@ class RtcEventVideoSendStreamConfig final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>-  std::unique_ptr<RtcEvent> Copy() const override;
>+  std::unique_ptr<RtcEventVideoSendStreamConfig> Copy() const;
> 
>-  const std::unique_ptr<const rtclog::StreamConfig> config_;
>+  const rtclog::StreamConfig& config() const { return *config_; }
> 
>  private:
>   RtcEventVideoSendStreamConfig(const RtcEventVideoSendStreamConfig& other);
>+
>+  const std::unique_ptr<const rtclog::StreamConfig> config_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.cc
>index 73a598d6b1e32a8c418a23328165dd8e2b12c8ee..b48fe9234fd74fc72a8911ff4baac55a71604bda 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.cc
>@@ -10,8 +10,14 @@
> 
> #include "logging/rtc_event_log/rtc_event_log.h"
> 
>+#include "absl/memory/memory.h"
>+
> namespace webrtc {
> 
>+std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
>+  return absl::make_unique<RtcEventLogNullImpl>();
>+}
>+
> bool RtcEventLogNullImpl::StartLogging(
>     std::unique_ptr<RtcEventLogOutput> output,
>     int64_t output_period_ms) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h
>index 0c7140690241d808740a1940e749189e9e64c5c2..7d900c00686f860728c9202004af81bfa1df88ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h
>@@ -11,9 +11,8 @@
> #ifndef LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
> #define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
> 
>+#include <stdint.h>
> #include <memory>
>-#include <string>
>-#include <utility>
> 
> #include "api/rtceventlogoutput.h"
> #include "logging/rtc_event_log/events/rtc_event.h"
>@@ -29,10 +28,9 @@ class RtcEventLog {
>   enum : size_t { kUnlimitedOutput = 0 };
>   enum : int64_t { kImmediateOutput = 0 };
> 
>-  // TODO(eladalon): Two stages are upcoming.
>-  // 1. Extend this to actually support the new encoding.
>-  // 2. Get rid of the legacy encoding, allowing us to get rid of this enum.
>-  enum class EncodingType { Legacy };
>+  // TODO(eladalon):  Get rid of the legacy encoding and this enum once all
>+  // clients have migrated to the new format.
>+  enum class EncodingType { Legacy, NewFormat };
> 
>   virtual ~RtcEventLog() {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2.proto b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2.proto
>index 4cac4553ced4347ba3a27d3bb08af4152d371725..995853b565cc1e18d30ed8bacd180af5d75ad5cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2.proto
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2.proto
>@@ -10,20 +10,17 @@ package webrtc.rtclog2;
> // single EventStream object containing the same events. Hence, it is not
> // necessary to wait for the entire log to be complete before beginning to
> // write it to a file.
>+// Note: For all X_deltas fields, we rely on the default value being an
>+// empty string.
> message EventStream {
>   // Deprecated - Maintained for compatibility with the old event log.
>-  // TODO(terelius): Maybe we can remove this and instead check the stream for
>-  // presence of a version field. That requires a custom protobuf parser, but we
>-  // have that already anyway.
>   repeated Event stream = 1 [deprecated = true];
>-  // required - The version number must be 2 for this version of the event log.
>-  optional uint32 version = 2;
>-  repeated IncomingRtpPackets incoming_rtp_packets = 3;
>-  repeated OutgoingRtpPackets outgoing_rtp_packets = 4;
>-  repeated IncomingRtcpPackets incoming_rtcp_packets = 5;
>-  repeated OutgoingRtcpPackets outgoing_rtcp_packets = 6;
>-  repeated AudioPlayoutEvents audio_playout_events = 7;
>-  // The field tags 8-15 are reserved for the most common events
>+  repeated IncomingRtpPackets incoming_rtp_packets = 2;
>+  repeated OutgoingRtpPackets outgoing_rtp_packets = 3;
>+  repeated IncomingRtcpPackets incoming_rtcp_packets = 4;
>+  repeated OutgoingRtcpPackets outgoing_rtcp_packets = 5;
>+  repeated AudioPlayoutEvents audio_playout_events = 6;
>+  // The field tags 7-15 are reserved for the most common events.
>   repeated BeginLogEvent begin_log_events = 16;
>   repeated EndLogEvent end_log_events = 17;
>   repeated LossBasedBweUpdates loss_based_bwe_updates = 18;
>@@ -32,6 +29,9 @@ message EventStream {
>   repeated BweProbeCluster probe_clusters = 21;
>   repeated BweProbeResultSuccess probe_success = 22;
>   repeated BweProbeResultFailure probe_failure = 23;
>+  repeated AlrState alr_states = 24;
>+  repeated IceCandidatePairConfig ice_candidate_configs = 25;
>+  repeated IceCandidatePairEvent ice_candidate_events = 26;
> 
>   repeated AudioRecvStreamConfig audio_recv_stream_configs = 101;
>   repeated AudioSendStreamConfig audio_send_stream_configs = 102;
>@@ -45,136 +45,206 @@ message Event {
> }
> 
> message IncomingRtpPackets {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>-  // RTP marker bit, used to label boundaries within e.g. video frames.
>+  // required - RTP marker bit, used to label boundaries between video frames.
>   optional bool marker = 2;
> 
>-  // RTP payload type.
>+  // required - RTP payload type.
>   optional uint32 payload_type = 3;
> 
>-  // RTP sequence number.
>+  // required - RTP sequence number.
>   optional uint32 sequence_number = 4;
> 
>-  // RTP monotonic clock timestamp (not actual time).
>+  // required - RTP monotonic clock timestamp (not actual time).
>   optional fixed32 rtp_timestamp = 5;
> 
>-  // Synchronization source of this packet's RTP stream.
>+  // required - Synchronization source of this packet's RTP stream.
>   optional fixed32 ssrc = 6;
> 
>   // TODO(terelius/dinor): Add CSRCs. Field number 7 reserved for this purpose.
> 
>-  // required - The size of the packet including both payload and header.
>-  optional uint32 packet_size = 8;
>+  // required - The size (in bytes) of the media payload, not including
>+  // RTP header or padding. The packet size is the sum of payload, header and
>+  // padding.
>+  optional uint32 payload_size = 8;
>+
>+  // required - The size (in bytes) of the RTP header.
>+  optional uint32 header_size = 9;
>+
>+  // required - The size (in bytes) of the padding.
>+  optional uint32 padding_size = 10;
>+
>+  // optional - required if the batch contains delta encoded events.
>+  optional uint32 number_of_deltas = 11;
>+
>+  // Field numbers 12-14 reserved for future use.
> 
>   // Optional header extensions.
>-  optional int32 transmission_time_offset = 9;
>-  optional uint32 absolute_send_time = 10;
>-  optional uint32 transport_sequence_number = 11;
>-  optional uint32 audio_level = 12;
>-  // TODO(terelius): Add header extensions like video rotation, playout delay?
>-
>-  // Delta encodings
>-  optional bytes timestamp_deltas_ms = 101;
>+  optional uint32 transport_sequence_number = 15;
>+  optional int32 transmission_time_offset = 16;
>+  optional uint32 absolute_send_time = 17;
>+  optional uint32 video_rotation = 18;
>+  // |audio_level| and |voice_activity| are always used in conjunction.
>+  optional uint32 audio_level = 19;
>+  optional bool voice_activity = 20;
>+  // TODO(terelius): Add other header extensions like playout delay?
>+
>+  // Delta encodings.
>+  optional bytes timestamp_ms_deltas = 101;
>   optional bytes marker_deltas = 102;
>   optional bytes payload_type_deltas = 103;
>   optional bytes sequence_number_deltas = 104;
>   optional bytes rtp_timestamp_deltas = 105;
>+  // Field number 107 reserved for CSRC.
>   optional bytes ssrc_deltas = 106;
>-  optional bytes packet_size_deltas = 107;
>-  optional bytes transmission_time_offset_deltas = 108;
>-  optional bytes absolute_send_time_deltas = 109;
>-  optional bytes transport_sequence_number_deltas = 110;
>-  optional bytes audio_level_deltas = 111;
>+  optional bytes payload_size_deltas = 108;
>+  optional bytes header_size_deltas = 109;
>+  optional bytes padding_size_deltas = 110;
>+  // Field number 111-114 reserved for future use.
>+  optional bytes transport_sequence_number_deltas = 115;
>+  optional bytes transmission_time_offset_deltas = 116;
>+  optional bytes absolute_send_time_deltas = 117;
>+  optional bytes video_rotation_deltas = 118;
>+  // |audio_level| and |voice_activity| are always used in conjunction.
>+  optional bytes audio_level_deltas = 119;
>+  optional bytes voice_activity_deltas = 120;
> }
> 
> message OutgoingRtpPackets {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>-  // RTP marker bit, used to label boundaries within e.g. video frames.
>+  // required - RTP marker bit, used to label boundaries between video frames.
>   optional bool marker = 2;
> 
>+  // required - RTP payload type.
>   optional uint32 payload_type = 3;
> 
>-  // RTP sequence number.
>+  // required - RTP sequence number.
>   optional uint32 sequence_number = 4;
> 
>-  // RTP monotonic clock timestamp (not actual time).
>+  // required - RTP monotonic clock timestamp (not actual time).
>   optional fixed32 rtp_timestamp = 5;
> 
>-  // Synchronization source of this packet's RTP stream.
>+  // required - Synchronization source of this packet's RTP stream.
>   optional fixed32 ssrc = 6;
> 
>   // TODO(terelius/dinor): Add CSRCs. Field number 7 reserved for this purpose.
> 
>-  // required - The size of the packet including both payload and header.
>-  optional uint32 packet_size = 8;
>+  // required - The size (in bytes) of the media payload, not including
>+  // RTP header or padding. The packet size is the sum of payload, header and
>+  // padding.
>+  optional uint32 payload_size = 8;
>+
>+  // required - The size (in bytes) of the RTP header.
>+  optional uint32 header_size = 9;
>+
>+  // required - The size (in bytes) of the padding.
>+  optional uint32 padding_size = 10;
>+
>+  // optional - required if the batch contains delta encoded events.
>+  optional uint32 number_of_deltas = 11;
>+
>+  // Field numbers 12-14 reserved for future use.
> 
>   // Optional header extensions.
>-  optional int32 transmission_time_offset = 9;
>-  optional uint32 absolute_send_time = 10;
>-  optional uint32 transport_sequence_number = 11;
>-  optional uint32 audio_level = 12;
>-  // TODO(terelius): Add header extensions like video rotation, playout delay?
>-
>-  // Delta encodings
>-  optional bytes timestamp_deltas_ms = 101;
>+  optional uint32 transport_sequence_number = 15;
>+  optional int32 transmission_time_offset = 16;
>+  optional uint32 absolute_send_time = 17;
>+  optional uint32 video_rotation = 18;
>+  // |audio_level| and |voice_activity| are always used in conjunction.
>+  optional uint32 audio_level = 19;
>+  optional bool voice_activity = 20;
>+  // TODO(terelius): Add other header extensions like playout delay?
>+
>+  // Delta encodings.
>+  optional bytes timestamp_ms_deltas = 101;
>   optional bytes marker_deltas = 102;
>   optional bytes payload_type_deltas = 103;
>   optional bytes sequence_number_deltas = 104;
>   optional bytes rtp_timestamp_deltas = 105;
>   optional bytes ssrc_deltas = 106;
>-  optional bytes packet_size_deltas = 107;
>-  optional bytes probe_cluster_id_deltas = 108;
>-  optional bytes transmission_time_offset_deltas = 109;
>-  optional bytes absolute_send_time_deltas = 110;
>-  optional bytes transport_sequence_number_deltas = 111;
>+  // Field number 107 reserved for CSRC.
>+  optional bytes payload_size_deltas = 108;
>+  optional bytes header_size_deltas = 109;
>+  optional bytes padding_size_deltas = 110;
>+  // Field number 111-114 reserved for future use.
>+  optional bytes transport_sequence_number_deltas = 115;
>+  optional bytes transmission_time_offset_deltas = 116;
>+  optional bytes absolute_send_time_deltas = 117;
>+  optional bytes video_rotation_deltas = 118;
>+  // |audio_level| and |voice_activity| are always used in conjunction.
>+  optional bytes audio_level_deltas = 119;
>+  optional bytes voice_activity_deltas = 120;
> }
> 
> message IncomingRtcpPackets {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>   // required - The whole packet including both payload and header.
>   optional bytes raw_packet = 2;
>   // TODO(terelius): Feasible to log parsed RTCP instead?
> 
>-  // Delta encodings
>-  optional bytes timestamp_deltas_ms = 101;
>-  optional bytes raw_packet_deltas = 102;
>+  // optional - required if the batch contains delta encoded events.
>+  optional uint32 number_of_deltas = 3;
>+
>+  // Delta/blob encodings.
>+  optional bytes timestamp_ms_deltas = 101;
>+  optional bytes raw_packet_blobs = 102;
> }
> 
> message OutgoingRtcpPackets {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>   // required - The whole packet including both payload and header.
>   optional bytes raw_packet = 2;
>   // TODO(terelius): Feasible to log parsed RTCP instead?
> 
>-  // Delta encodings
>-  optional bytes timestamp_deltas_ms = 101;
>-  optional bytes raw_packet_deltas = 102;
>+  // optional - required if the batch contains delta encoded events.
>+  optional uint32 number_of_deltas = 3;
>+
>+  // Delta/blob encodings.
>+  optional bytes timestamp_ms_deltas = 101;
>+  optional bytes raw_packet_blobs = 102;
> }
> 
> message AudioPlayoutEvents {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>   // required - The SSRC of the audio stream associated with the playout event.
>   optional uint32 local_ssrc = 2;
> 
>-  // Delta encodings
>-  optional bytes timestamp_deltas_ms = 101;
>+  // optional - required if the batch contains delta encoded events.
>+  optional uint32 number_of_deltas = 3;
>+
>+  // Delta encodings.
>+  optional bytes timestamp_ms_deltas = 101;
>   optional bytes local_ssrc_deltas = 102;
> }
> 
> message BeginLogEvent {
>+  // required
>   optional int64 timestamp_ms = 1;
>+
>+  // required
>+  optional uint32 version = 2;
>+
>+  // required
>+  optional int64 utc_time_ms = 3;
> }
> 
> message EndLogEvent {
>+  // required
>   optional int64 timestamp_ms = 1;
> }
> 
> message LossBasedBweUpdates {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>   // TODO(terelius): Update log interface to unsigned.
>@@ -191,29 +261,37 @@ message LossBasedBweUpdates {
>   // required - Total number of packets that the BWE update is based on.
>   optional uint32 total_packets = 4;
> 
>-  // Delta encodings
>-  optional bytes timestamp_deltas_ms = 101;
>-  optional bytes bitrate_deltas_bps = 102;
>+  // optional - required if the batch contains delta encoded events.
>+  optional uint32 number_of_deltas = 5;
>+
>+  // Delta encodings.
>+  optional bytes timestamp_ms_deltas = 101;
>+  optional bytes bitrate_bps_deltas = 102;
>   optional bytes fraction_loss_deltas = 103;
>   optional bytes total_packets_deltas = 104;
> }
> 
> message DelayBasedBweUpdates {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>   // required - Bandwidth estimate (in bps) after the update.
>   optional uint32 bitrate_bps = 2;
> 
>   enum DetectorState {
>-    BWE_NORMAL = 0;
>-    BWE_UNDERUSING = 1;
>-    BWE_OVERUSING = 2;
>+    BWE_UNKNOWN_STATE = 0;
>+    BWE_NORMAL = 1;
>+    BWE_UNDERUSING = 2;
>+    BWE_OVERUSING = 3;
>   }
>   optional DetectorState detector_state = 3;
> 
>-  // Delta encodings
>-  optional bytes timestamp_deltas_ms = 101;
>-  optional bytes bitrate_deltas_bps = 102;
>+  // optional - required if the batch contains delta encoded events.
>+  optional uint32 number_of_deltas = 4;
>+
>+  // Delta encodings.
>+  optional bytes timestamp_ms_deltas = 101;
>+  optional bytes bitrate_bps_deltas = 102;
>   optional bytes detector_state_deltas = 103;
> }
> 
>@@ -221,15 +299,17 @@ message DelayBasedBweUpdates {
> message RtpHeaderExtensionConfig {
>   // Optional IDs for the header extensions. Each ID is a 4-bit number that is
>   // only set if that extension is configured.
>-  // TODO(terelius): Can we skip transmission_time_offset? When is it used?
>+  // TODO: Can we skip audio level?
>   optional int32 transmission_time_offset_id = 1;
>   optional int32 absolute_send_time_id = 2;
>   optional int32 transport_sequence_number_id = 3;
>-  optional int32 audio_level_id = 4;
>-  // TODO(terelius): Add video_rotation and playout delay?
>+  optional int32 video_rotation_id = 4;
>+  optional int32 audio_level_id = 5;
>+  // TODO(terelius): Add other header extensions like playout delay?
> }
> 
> message VideoRecvStreamConfig {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>   // required - Synchronization source (stream identifier) to be received.
>@@ -238,42 +318,38 @@ message VideoRecvStreamConfig {
>   // required - Sender SSRC used for sending RTCP (such as receiver reports).
>   optional uint32 local_ssrc = 3;
> 
>-  // required if RTX is configured
>+  // optional - required if RTX is configured. SSRC for the RTX stream.
>   optional uint32 rtx_ssrc = 4;
> 
>-  // optional - RTP source stream ID
>-  optional bytes rsid = 5;
>-
>   // IDs for the header extension we care about. Only required if there are
>   // header extensions configured.
>-  optional RtpHeaderExtensionConfig header_extensions = 6;
>+  optional RtpHeaderExtensionConfig header_extensions = 5;
> 
>   // TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
>   // used, we also need a map between RTP payload type and RTX payload type.
> }
> 
> message VideoSendStreamConfig {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>-  // Synchronization source (stream identifier) for outgoing stream.
>-  // One stream can have several ssrcs for e.g. simulcast.
>+  // required - Synchronization source (stream identifier) for outgoing stream.
>+  // When using simulcast, a separate config should be logged for each stream.
>   optional uint32 ssrc = 2;
> 
>-  // SSRC for the RTX stream
>+  // optional - required if RTX is configured. SSRC for the RTX stream.
>   optional uint32 rtx_ssrc = 3;
> 
>-  // RTP source stream ID
>-  optional bytes rsid = 4;
>-
>   // IDs for the header extension we care about. Only required if there are
>   // header extensions configured.
>-  optional RtpHeaderExtensionConfig header_extensions = 5;
>+  optional RtpHeaderExtensionConfig header_extensions = 4;
> 
>   // TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
>   // used, we also need a map between RTP payload type and RTX payload type.
> }
> 
> message AudioRecvStreamConfig {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>   // required - Synchronization source (stream identifier) to be received.
>@@ -284,38 +360,33 @@ message AudioRecvStreamConfig {
> 
>   // Field number 4 reserved for RTX SSRC.
> 
>-  // optional - RTP source stream ID
>-  optional bytes rsid = 5;
>-
>   // IDs for the header extension we care about. Only required if there are
>   // header extensions configured.
>-  optional RtpHeaderExtensionConfig header_extensions = 6;
>+  optional RtpHeaderExtensionConfig header_extensions = 5;
> 
>   // TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
>   // used, we also need a map between RTP payload type and RTX payload type.
> }
> 
> message AudioSendStreamConfig {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>-  // Synchronization source (stream identifier) for outgoing stream.
>-  // One stream can have several ssrcs for e.g. simulcast.
>+  // required - Synchronization source (stream identifier) for outgoing stream.
>   optional uint32 ssrc = 2;
> 
>-  // Field number 3 reserved for RTX SSRC
>-
>-  // RTP source stream ID
>-  optional bytes rsid = 4;
>+  // Field number 3 reserved for RTX SSRC.
> 
>   // IDs for the header extension we care about. Only required if there are
>   // header extensions configured.
>-  optional RtpHeaderExtensionConfig header_extensions = 5;
>+  optional RtpHeaderExtensionConfig header_extensions = 4;
> 
>   // TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
>   // used, we also need a map between RTP payload type and RTX payload type.
> }
> 
> message AudioNetworkAdaptations {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>   // Bit rate that the audio encoder is operating at.
>@@ -328,7 +399,12 @@ message AudioNetworkAdaptations {
> 
>   // Packet loss fraction that the encoder's forward error correction (FEC) is
>   // optimized for.
>-  optional float uplink_packet_loss_fraction = 4;
>+  // Instead of encoding a float, we encode a value between 0 and 16383, which
>+  // if divided by 16383, will give a value close to the original float.
>+  // The value 16383 (2^14 - 1) was chosen so that it would give good precision
>+  // on the one hand, and would be encodable with two bytes in varint form
>+  // on the other hand.
>+  optional uint32 uplink_packet_loss_fraction = 4;
> 
>   // Whether forward error correction (FEC) is turned on or off.
>   optional bool enable_fec = 5;
>@@ -339,10 +415,13 @@ message AudioNetworkAdaptations {
>   // Number of audio channels that each encoded packet consists of.
>   optional uint32 num_channels = 7;
> 
>-  // Delta encodings
>-  optional bytes timestamp_deltas_ms = 101;
>-  optional bytes bitrate_deltas_bps = 102;
>-  optional bytes frame_length_deltas_ms = 103;
>+  // optional - required if the batch contains delta encoded events.
>+  optional uint32 number_of_deltas = 8;
>+
>+  // Delta encodings.
>+  optional bytes timestamp_ms_deltas = 101;
>+  optional bytes bitrate_bps_deltas = 102;
>+  optional bytes frame_length_ms_deltas = 103;
>   optional bytes uplink_packet_loss_fraction_deltas = 104;
>   optional bytes enable_fec_deltas = 105;
>   optional bytes enable_dtx_deltas = 106;
>@@ -350,6 +429,7 @@ message AudioNetworkAdaptations {
> }
> 
> message BweProbeCluster {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>   // required - The id of this probe cluster.
>@@ -366,6 +446,7 @@ message BweProbeCluster {
> }
> 
> message BweProbeResultSuccess {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>   // required - The id of this probe cluster.
>@@ -376,6 +457,7 @@ message BweProbeResultSuccess {
> }
> 
> message BweProbeResultFailure {
>+  // required
>   optional int64 timestamp_ms = 1;
> 
>   // required - The id of this probe cluster.
>@@ -391,3 +473,101 @@ message BweProbeResultFailure {
>   // required
>   optional FailureReason failure = 3;
> }
>+
>+message AlrState {
>+  // required
>+  optional int64 timestamp_ms = 1;
>+
>+  // required - True if the send rate is application limited.
>+  optional bool in_alr = 2;
>+}
>+
>+message IceCandidatePairConfig {
>+  enum IceCandidatePairConfigType {
>+    UNKNOWN_CONFIG_TYPE = 0;
>+    ADDED = 1;
>+    UPDATED = 2;
>+    DESTROYED = 3;
>+    SELECTED = 4;
>+  }
>+
>+  enum IceCandidateType {
>+    UNKNOWN_CANDIDATE_TYPE = 0;
>+    LOCAL = 1;
>+    STUN = 2;
>+    PRFLX = 3;
>+    RELAY = 4;
>+  }
>+
>+  enum Protocol {
>+    UNKNOWN_PROTOCOL = 0;
>+    UDP = 1;
>+    TCP = 2;
>+    SSLTCP = 3;
>+    TLS = 4;
>+  }
>+
>+  enum AddressFamily {
>+    UNKNOWN_ADDRESS_FAMILY = 0;
>+    IPV4 = 1;
>+    IPV6 = 2;
>+  }
>+
>+  enum NetworkType {
>+    UNKNOWN_NETWORK_TYPE = 0;
>+    ETHERNET = 1;
>+    WIFI = 2;
>+    CELLULAR = 3;
>+    VPN = 4;
>+    LOOPBACK = 5;
>+  }
>+
>+  // required
>+  optional int64 timestamp_ms = 1;
>+
>+  // required
>+  optional IceCandidatePairConfigType config_type = 2;
>+
>+  // required
>+  optional uint32 candidate_pair_id = 3;
>+
>+  // required
>+  optional IceCandidateType local_candidate_type = 4;
>+
>+  // required
>+  optional Protocol local_relay_protocol = 5;
>+
>+  // required
>+  optional NetworkType local_network_type = 6;
>+
>+  // required
>+  optional AddressFamily local_address_family = 7;
>+
>+  // required
>+  optional IceCandidateType remote_candidate_type = 8;
>+
>+  // required
>+  optional AddressFamily remote_address_family = 9;
>+
>+  // required
>+  optional Protocol candidate_pair_protocol = 10;
>+}
>+
>+message IceCandidatePairEvent {
>+  enum IceCandidatePairEventType {
>+    UNKNOWN_CHECK_TYPE = 0;
>+    CHECK_SENT = 1;
>+    CHECK_RECEIVED = 2;
>+    CHECK_RESPONSE_SENT = 3;
>+    CHECK_RESPONSE_RECEIVED = 4;
>+  }
>+
>+  // required
>+  optional int64 timestamp_ms = 1;
>+
>+  // required
>+  optional IceCandidatePairEventType event_type = 2;
>+
>+  // required
>+  optional uint32 candidate_pair_id = 3;
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
>index a3d3456a7bdeab7e2bcd9d32a1bfac2c21f9d378..da7df64faddf61a13e41d12fbd58e72f456a0feb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
>@@ -32,31 +32,32 @@ namespace {
> 
> using MediaType = webrtc::ParsedRtcEventLogNew::MediaType;
> 
>-DEFINE_bool(
>+WEBRTC_DEFINE_bool(
>     audio,
>     true,
>     "Use --noaudio to exclude audio packets from the converted RTPdump file.");
>-DEFINE_bool(
>+WEBRTC_DEFINE_bool(
>     video,
>     true,
>     "Use --novideo to exclude video packets from the converted RTPdump file.");
>-DEFINE_bool(
>+WEBRTC_DEFINE_bool(
>     data,
>     true,
>     "Use --nodata to exclude data packets from the converted RTPdump file.");
>-DEFINE_bool(
>+WEBRTC_DEFINE_bool(
>     rtp,
>     true,
>     "Use --nortp to exclude RTP packets from the converted RTPdump file.");
>-DEFINE_bool(
>+WEBRTC_DEFINE_bool(
>     rtcp,
>     true,
>     "Use --nortcp to exclude RTCP packets from the converted RTPdump file.");
>-DEFINE_string(ssrc,
>-              "",
>-              "Store only packets with this SSRC (decimal or hex, the latter "
>-              "starting with 0x).");
>-DEFINE_bool(help, false, "Prints this message.");
>+WEBRTC_DEFINE_string(
>+    ssrc,
>+    "",
>+    "Store only packets with this SSRC (decimal or hex, the latter "
>+    "starting with 0x).");
>+WEBRTC_DEFINE_bool(help, false, "Prints this message.");
> 
> // Parses the input string for a valid SSRC. If a valid SSRC is found, it is
> // written to the output variable |ssrc|, and true is returned. Otherwise,
>@@ -97,9 +98,10 @@ bool ShouldSkipStream(MediaType media_type,
> // Convert a LoggedRtpPacketIncoming to a test::RtpPacket. Header extension IDs
> // are allocated according to the provided extension map. This might not match
> // the extension map used in the actual call.
>-void ConvertRtpPacket(const webrtc::LoggedRtpPacketIncoming& incoming,
>-                      const webrtc::RtpHeaderExtensionMap default_extension_map,
>-                      webrtc::test::RtpPacket* packet) {
>+void ConvertRtpPacket(
>+    const webrtc::LoggedRtpPacketIncoming& incoming,
>+    const webrtc::RtpHeaderExtensionMap& default_extension_map,
>+    webrtc::test::RtpPacket* packet) {
>   webrtc::RtpPacket reconstructed_packet(&default_extension_map);
> 
>   reconstructed_packet.SetMarker(incoming.rtp.header.markerBit);
>@@ -196,8 +198,6 @@ int main(int argc, char* argv[]) {
>     return -1;
>   }
> 
>-  std::cout << "Found " << parsed_stream.GetNumberOfEvents()
>-            << " events in the input file." << std::endl;
>   int rtp_counter = 0, rtcp_counter = 0;
>   bool header_only = false;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2stats.cc
>deleted file mode 100644
>index 928c829f49f13c0c51ac90855c56f4833d397ca3..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2stats.cc
>+++ /dev/null
>@@ -1,265 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <inttypes.h>
>-#include <stdio.h>
>-
>-#include <fstream>
>-#include <iostream>
>-#include <map>
>-#include <string>
>-#include <tuple>
>-#include <utility>
>-#include <vector>
>-
>-#include "logging/rtc_event_log/rtc_event_log.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/flags.h"
>-#include "rtc_base/ignore_wundef.h"
>-#include "rtc_base/logging.h"
>-
>-// Files generated at build-time by the protobuf compiler.
>-RTC_PUSH_IGNORING_WUNDEF()
>-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
>-#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
>-#else
>-#include "logging/rtc_event_log/rtc_event_log.pb.h"
>-#endif
>-RTC_POP_IGNORING_WUNDEF()
>-
>-namespace {
>-
>-DEFINE_bool(help, false, "Prints this message.");
>-
>-struct Stats {
>-  int count = 0;
>-  size_t total_size = 0;
>-};
>-
>-// We are duplicating some parts of the parser here because we want to get
>-// access to raw protobuf events.
>-std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
>-  uint64_t varint = 0;
>-  for (size_t bytes_read = 0; bytes_read < 10; ++bytes_read) {
>-    // The most significant bit of each byte is 0 if it is the last byte in
>-    // the varint and 1 otherwise. Thus, we take the 7 least significant bits
>-    // of each byte and shift them 7 bits for each byte read previously to get
>-    // the (unsigned) integer.
>-    int byte = stream.get();
>-    if (stream.eof()) {
>-      return std::make_pair(varint, false);
>-    }
>-    RTC_DCHECK_GE(byte, 0);
>-    RTC_DCHECK_LE(byte, 255);
>-    varint |= static_cast<uint64_t>(byte & 0x7F) << (7 * bytes_read);
>-    if ((byte & 0x80) == 0) {
>-      return std::make_pair(varint, true);
>-    }
>-  }
>-  return std::make_pair(varint, false);
>-}
>-
>-bool ParseEvents(const std::string& filename,
>-                 std::vector<webrtc::rtclog::Event>* events) {
>-  std::ifstream stream(filename, std::ios_base::in | std::ios_base::binary);
>-  if (!stream.good() || !stream.is_open()) {
>-    RTC_LOG(LS_WARNING) << "Could not open file for reading.";
>-    return false;
>-  }
>-
>-  const size_t kMaxEventSize = (1u << 16) - 1;
>-  std::vector<char> tmp_buffer(kMaxEventSize);
>-  uint64_t tag;
>-  uint64_t message_length;
>-  bool success;
>-
>-  RTC_DCHECK(stream.good());
>-
>-  while (1) {
>-    // Check whether we have reached end of file.
>-    stream.peek();
>-    if (stream.eof()) {
>-      return true;
>-    }
>-
>-    // Read the next message tag. The tag number is defined as
>-    // (fieldnumber << 3) | wire_type. In our case, the field number is
>-    // supposed to be 1 and the wire type for an length-delimited field is 2.
>-    const uint64_t kExpectedTag = (1 << 3) | 2;
>-    std::tie(tag, success) = ParseVarInt(stream);
>-    if (!success) {
>-      RTC_LOG(LS_WARNING)
>-          << "Missing field tag from beginning of protobuf event.";
>-      return false;
>-    } else if (tag != kExpectedTag) {
>-      RTC_LOG(LS_WARNING)
>-          << "Unexpected field tag at beginning of protobuf event.";
>-      return false;
>-    }
>-
>-    // Read the length field.
>-    std::tie(message_length, success) = ParseVarInt(stream);
>-    if (!success) {
>-      RTC_LOG(LS_WARNING) << "Missing message length after protobuf field tag.";
>-      return false;
>-    } else if (message_length > kMaxEventSize) {
>-      RTC_LOG(LS_WARNING) << "Protobuf message length is too large.";
>-      return false;
>-    }
>-
>-    // Read the next protobuf event to a temporary char buffer.
>-    stream.read(tmp_buffer.data(), message_length);
>-    if (stream.gcount() != static_cast<int>(message_length)) {
>-      RTC_LOG(LS_WARNING) << "Failed to read protobuf message from file.";
>-      return false;
>-    }
>-
>-    // Parse the protobuf event from the buffer.
>-    webrtc::rtclog::Event event;
>-    if (!event.ParseFromArray(tmp_buffer.data(), message_length)) {
>-      RTC_LOG(LS_WARNING) << "Failed to parse protobuf message.";
>-      return false;
>-    }
>-    events->push_back(event);
>-  }
>-}
>-
>-// TODO(terelius): Should this be placed in some utility file instead?
>-std::string EventTypeToString(webrtc::rtclog::Event::EventType event_type) {
>-  switch (event_type) {
>-    case webrtc::rtclog::Event::UNKNOWN_EVENT:
>-      return "UNKNOWN_EVENT";
>-    case webrtc::rtclog::Event::LOG_START:
>-      return "LOG_START";
>-    case webrtc::rtclog::Event::LOG_END:
>-      return "LOG_END";
>-    case webrtc::rtclog::Event::RTP_EVENT:
>-      return "RTP_EVENT";
>-    case webrtc::rtclog::Event::RTCP_EVENT:
>-      return "RTCP_EVENT";
>-    case webrtc::rtclog::Event::AUDIO_PLAYOUT_EVENT:
>-      return "AUDIO_PLAYOUT_EVENT";
>-    case webrtc::rtclog::Event::LOSS_BASED_BWE_UPDATE:
>-      return "LOSS_BASED_BWE_UPDATE";
>-    case webrtc::rtclog::Event::DELAY_BASED_BWE_UPDATE:
>-      return "DELAY_BASED_BWE_UPDATE";
>-    case webrtc::rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
>-      return "VIDEO_RECV_CONFIG";
>-    case webrtc::rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
>-      return "VIDEO_SEND_CONFIG";
>-    case webrtc::rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
>-      return "AUDIO_RECV_CONFIG";
>-    case webrtc::rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
>-      return "AUDIO_SEND_CONFIG";
>-    case webrtc::rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
>-      return "AUDIO_NETWORK_ADAPTATION";
>-    case webrtc::rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT:
>-      return "BWE_PROBE_CREATED";
>-    case webrtc::rtclog::Event::BWE_PROBE_RESULT_EVENT:
>-      return "BWE_PROBE_RESULT";
>-    case webrtc::rtclog::Event::ALR_STATE_EVENT:
>-      return "ALR_STATE_EVENT";
>-    case webrtc::rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG:
>-      return "ICE_CANDIDATE_PAIR_CONFIG";
>-    case webrtc::rtclog::Event::ICE_CANDIDATE_PAIR_EVENT:
>-      return "ICE_CANDIDATE_PAIR_EVENT";
>-  }
>-  RTC_NOTREACHED();
>-  return "UNKNOWN_EVENT";
>-}
>-
>-}  // namespace
>-
>-// This utility will print basic information about each packet to stdout.
>-// Note that parser will assert if the protobuf event is missing some required
>-// fields and we attempt to access them. We don't handle this at the moment.
>-int main(int argc, char* argv[]) {
>-  std::string program_name = argv[0];
>-  std::string usage =
>-      "Tool for file usage statistics from an RtcEventLog.\n"
>-      "Run " +
>-      program_name +
>-      " --help for usage.\n"
>-      "Example usage:\n" +
>-      program_name + " input.rel\n";
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
>-      argc != 2) {
>-    std::cout << usage;
>-    if (FLAG_help) {
>-      rtc::FlagList::Print(nullptr, false);
>-      return 0;
>-    }
>-    return 1;
>-  }
>-  std::string file_name = argv[1];
>-
>-  std::vector<webrtc::rtclog::Event> events;
>-  if (!ParseEvents(file_name, &events)) {
>-    RTC_LOG(LS_ERROR) << "Failed to parse event log.";
>-    return -1;
>-  }
>-
>-  // Get file size
>-  FILE* file = fopen(file_name.c_str(), "rb");
>-  fseek(file, 0L, SEEK_END);
>-  int64_t file_size = ftell(file);
>-  fclose(file);
>-
>-  // We are deliberately using low level protobuf functions to get the stats
>-  // since the convenience functions in the parser would CHECK that the events
>-  // are well formed.
>-  std::map<webrtc::rtclog::Event::EventType, Stats> stats;
>-  int malformed_events = 0;
>-  size_t malformed_event_size = 0;
>-  size_t accumulated_event_size = 0;
>-  for (const webrtc::rtclog::Event& event : events) {
>-    size_t serialized_size = event.ByteSizeLong();
>-    // When the event is written on the disk, it is part of an EventStream
>-    // object. The event stream will prepend a 1 byte field number/wire type,
>-    // and a varint encoding (base 128) of the event length.
>-    serialized_size =
>-        1 + (1 + (serialized_size > 127) + (serialized_size > 16383)) +
>-        serialized_size;
>-
>-    if (event.has_type() && event.has_timestamp_us()) {
>-      stats[event.type()].count++;
>-      stats[event.type()].total_size += serialized_size;
>-    } else {
>-      // The event is missing the type or the timestamp field.
>-      malformed_events++;
>-      malformed_event_size += serialized_size;
>-    }
>-    accumulated_event_size += serialized_size;
>-  }
>-
>-  printf("Type                  \tCount\tTotal size\tAverage size\tPercent\n");
>-  printf(
>-      "-----------------------------------------------------------------------"
>-      "\n");
>-  for (const auto it : stats) {
>-    printf("%-22s\t%5d\t%10zu\t%12.2lf\t%7.2lf\n",
>-           EventTypeToString(it.first).c_str(), it.second.count,
>-           it.second.total_size,
>-           static_cast<double>(it.second.total_size) / it.second.count,
>-           static_cast<double>(it.second.total_size) / file_size * 100);
>-  }
>-  if (malformed_events != 0) {
>-    printf("%-22s\t%5d\t%10zu\t%12.2lf\t%7.2lf\n", "MALFORMED",
>-           malformed_events, malformed_event_size,
>-           static_cast<double>(malformed_event_size) / malformed_events,
>-           static_cast<double>(malformed_event_size) / file_size * 100);
>-  }
>-  if (file_size - accumulated_event_size != 0) {
>-    printf("WARNING: %" PRId64 " bytes not accounted for\n",
>-           file_size - accumulated_event_size);
>-  }
>-
>-  return 0;
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2text.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
>deleted file mode 100644
>index 67cb6ecdf4b5a5cbbfbd38693c520d2dde381b6f..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
>+++ /dev/null
>@@ -1,881 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <string.h>
>-
>-#include <iomanip>  // setfill, setw
>-#include <iostream>
>-#include <map>
>-#include <string>
>-#include <utility>  // pair
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
>-#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
>-#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
>-#include "modules/rtp_rtcp/source/rtp_utility.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/flags.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/strings/string_builder.h"
>-
>-namespace {
>-
>-DEFINE_bool(unknown, true, "Use --nounknown to exclude unknown events.");
>-DEFINE_bool(startstop, true, "Use --nostartstop to exclude start/stop events.");
>-DEFINE_bool(config, true, "Use --noconfig to exclude stream configurations.");
>-DEFINE_bool(bwe, true, "Use --nobwe to exclude BWE events.");
>-DEFINE_bool(incoming, true, "Use --noincoming to exclude incoming packets.");
>-DEFINE_bool(outgoing, true, "Use --nooutgoing to exclude packets.");
>-// TODO(terelius): Note that the media type doesn't work with outgoing packets.
>-DEFINE_bool(audio, true, "Use --noaudio to exclude audio packets.");
>-// TODO(terelius): Note that the media type doesn't work with outgoing packets.
>-DEFINE_bool(video, true, "Use --novideo to exclude video packets.");
>-// TODO(terelius): Note that the media type doesn't work with outgoing packets.
>-DEFINE_bool(data, true, "Use --nodata to exclude data packets.");
>-DEFINE_bool(rtp, true, "Use --nortp to exclude RTP packets.");
>-DEFINE_bool(rtcp, true, "Use --nortcp to exclude RTCP packets.");
>-DEFINE_bool(playout, true, "Use --noplayout to exclude audio playout events.");
>-DEFINE_bool(ana, true, "Use --noana to exclude ANA events.");
>-DEFINE_bool(probe, true, "Use --noprobe to exclude probe events.");
>-DEFINE_bool(ice, true, "Use --noice to exclude ICE events.");
>-
>-DEFINE_bool(print_full_packets,
>-            false,
>-            "Print the full RTP headers and RTCP packets in hex.");
>-
>-// TODO(terelius): Allow a list of SSRCs.
>-DEFINE_string(ssrc,
>-              "",
>-              "Print only packets with this SSRC (decimal or hex, the latter "
>-              "starting with 0x).");
>-DEFINE_bool(help, false, "Prints this message.");
>-
>-using MediaType = webrtc::ParsedRtcEventLogNew::MediaType;
>-
>-static uint32_t filtered_ssrc = 0;
>-
>-// Parses the input string for a valid SSRC. If a valid SSRC is found, it is
>-// written to the static global variable |filtered_ssrc|, and true is returned.
>-// Otherwise, false is returned.
>-// The empty string must be validated as true, because it is the default value
>-// of the command-line flag. In this case, no value is written to the output
>-// variable.
>-bool ParseSsrc(std::string str) {
>-  // If the input string starts with 0x or 0X it indicates a hexadecimal number.
>-  auto read_mode = std::dec;
>-  if (str.size() > 2 &&
>-      (str.substr(0, 2) == "0x" || str.substr(0, 2) == "0X")) {
>-    read_mode = std::hex;
>-    str = str.substr(2);
>-  }
>-  std::stringstream ss(str);
>-  ss >> read_mode >> filtered_ssrc;
>-  return str.empty() || (!ss.fail() && ss.eof());
>-}
>-
>-bool ExcludePacket(webrtc::PacketDirection direction,
>-                   MediaType media_type,
>-                   uint32_t packet_ssrc) {
>-  if (!FLAG_outgoing && direction == webrtc::kOutgoingPacket)
>-    return true;
>-  if (!FLAG_incoming && direction == webrtc::kIncomingPacket)
>-    return true;
>-  if (!FLAG_audio && media_type == MediaType::AUDIO)
>-    return true;
>-  if (!FLAG_video && media_type == MediaType::VIDEO)
>-    return true;
>-  if (!FLAG_data && media_type == MediaType::DATA)
>-    return true;
>-  if (strlen(FLAG_ssrc) > 0 && packet_ssrc != filtered_ssrc)
>-    return true;
>-  return false;
>-}
>-
>-const char* StreamInfo(webrtc::PacketDirection direction,
>-                       MediaType media_type) {
>-  if (direction == webrtc::kOutgoingPacket) {
>-    if (media_type == MediaType::AUDIO)
>-      return "(out,audio)";
>-    else if (media_type == MediaType::VIDEO)
>-      return "(out,video)";
>-    else if (media_type == MediaType::DATA)
>-      return "(out,data)";
>-    else
>-      return "(out)";
>-  }
>-  if (direction == webrtc::kIncomingPacket) {
>-    if (media_type == MediaType::AUDIO)
>-      return "(in,audio)";
>-    else if (media_type == MediaType::VIDEO)
>-      return "(in,video)";
>-    else if (media_type == MediaType::DATA)
>-      return "(in,data)";
>-    else
>-      return "(in)";
>-  }
>-  return "(unknown)";
>-}
>-
>-// Return default values for header extensions, to use on streams without stored
>-// mapping data. Currently this only applies to audio streams, since the mapping
>-// is not stored in the event log.
>-// TODO(ivoc): Remove this once this mapping is stored in the event log for
>-//             audio streams. Tracking bug: webrtc:6399
>-webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
>-  webrtc::RtpHeaderExtensionMap default_map;
>-  default_map.Register<webrtc::AudioLevel>(
>-      webrtc::RtpExtension::kAudioLevelDefaultId);
>-  default_map.Register<webrtc::TransmissionOffset>(
>-      webrtc::RtpExtension::kTimestampOffsetDefaultId);
>-  default_map.Register<webrtc::AbsoluteSendTime>(
>-      webrtc::RtpExtension::kAbsSendTimeDefaultId);
>-  default_map.Register<webrtc::VideoOrientation>(
>-      webrtc::RtpExtension::kVideoRotationDefaultId);
>-  default_map.Register<webrtc::VideoContentTypeExtension>(
>-      webrtc::RtpExtension::kVideoContentTypeDefaultId);
>-  default_map.Register<webrtc::VideoTimingExtension>(
>-      webrtc::RtpExtension::kVideoTimingDefaultId);
>-  default_map.Register<webrtc::FrameMarkingExtension>(
>-      webrtc::RtpExtension::kFrameMarkingDefaultId);
>-  default_map.Register<webrtc::TransportSequenceNumber>(
>-      webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
>-  default_map.Register<webrtc::PlayoutDelayLimits>(
>-      webrtc::RtpExtension::kPlayoutDelayDefaultId);
>-  return default_map;
>-}
>-
>-void PrintSenderReport(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>-                       const webrtc::rtcp::CommonHeader& rtcp_block,
>-                       uint64_t log_timestamp,
>-                       webrtc::PacketDirection direction) {
>-  webrtc::rtcp::SenderReport sr;
>-  if (!sr.Parse(rtcp_block))
>-    return;
>-  MediaType media_type =
>-      parsed_stream.GetMediaType(sr.sender_ssrc(), direction);
>-  if (ExcludePacket(direction, media_type, sr.sender_ssrc()))
>-    return;
>-  std::cout << log_timestamp << "\t"
>-            << "RTCP_SR" << StreamInfo(direction, media_type)
>-            << "\tssrc=" << sr.sender_ssrc()
>-            << "\ttimestamp=" << sr.rtp_timestamp() << std::endl;
>-}
>-
>-void PrintReceiverReport(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>-                         const webrtc::rtcp::CommonHeader& rtcp_block,
>-                         uint64_t log_timestamp,
>-                         webrtc::PacketDirection direction) {
>-  webrtc::rtcp::ReceiverReport rr;
>-  if (!rr.Parse(rtcp_block))
>-    return;
>-  MediaType media_type =
>-      parsed_stream.GetMediaType(rr.sender_ssrc(), direction);
>-  if (ExcludePacket(direction, media_type, rr.sender_ssrc()))
>-    return;
>-  std::cout << log_timestamp << "\t"
>-            << "RTCP_RR" << StreamInfo(direction, media_type)
>-            << "\tssrc=" << rr.sender_ssrc() << std::endl;
>-}
>-
>-void PrintXr(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>-             const webrtc::rtcp::CommonHeader& rtcp_block,
>-             uint64_t log_timestamp,
>-             webrtc::PacketDirection direction) {
>-  webrtc::rtcp::ExtendedReports xr;
>-  if (!xr.Parse(rtcp_block))
>-    return;
>-  MediaType media_type =
>-      parsed_stream.GetMediaType(xr.sender_ssrc(), direction);
>-  if (ExcludePacket(direction, media_type, xr.sender_ssrc()))
>-    return;
>-  std::cout << log_timestamp << "\t"
>-            << "RTCP_XR" << StreamInfo(direction, media_type)
>-            << "\tssrc=" << xr.sender_ssrc() << std::endl;
>-}
>-
>-void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block,
>-               uint64_t log_timestamp,
>-               webrtc::PacketDirection direction) {
>-  std::cout << log_timestamp << "\t"
>-            << "RTCP_SDES" << StreamInfo(direction, MediaType::ANY)
>-            << std::endl;
>-  RTC_NOTREACHED() << "SDES should have been redacted when writing the log";
>-}
>-
>-void PrintBye(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>-              const webrtc::rtcp::CommonHeader& rtcp_block,
>-              uint64_t log_timestamp,
>-              webrtc::PacketDirection direction) {
>-  webrtc::rtcp::Bye bye;
>-  if (!bye.Parse(rtcp_block))
>-    return;
>-  MediaType media_type =
>-      parsed_stream.GetMediaType(bye.sender_ssrc(), direction);
>-  if (ExcludePacket(direction, media_type, bye.sender_ssrc()))
>-    return;
>-  std::cout << log_timestamp << "\t"
>-            << "RTCP_BYE" << StreamInfo(direction, media_type)
>-            << "\tssrc=" << bye.sender_ssrc() << std::endl;
>-}
>-
>-void PrintRtpFeedback(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>-                      const webrtc::rtcp::CommonHeader& rtcp_block,
>-                      uint64_t log_timestamp,
>-                      webrtc::PacketDirection direction) {
>-  switch (rtcp_block.fmt()) {
>-    case webrtc::rtcp::Nack::kFeedbackMessageType: {
>-      webrtc::rtcp::Nack nack;
>-      if (!nack.Parse(rtcp_block))
>-        return;
>-      MediaType media_type =
>-          parsed_stream.GetMediaType(nack.sender_ssrc(), direction);
>-      if (ExcludePacket(direction, media_type, nack.sender_ssrc()))
>-        return;
>-      std::cout << log_timestamp << "\t"
>-                << "RTCP_NACK" << StreamInfo(direction, media_type)
>-                << "\tssrc=" << nack.sender_ssrc() << std::endl;
>-      break;
>-    }
>-    case webrtc::rtcp::Tmmbr::kFeedbackMessageType: {
>-      webrtc::rtcp::Tmmbr tmmbr;
>-      if (!tmmbr.Parse(rtcp_block))
>-        return;
>-      MediaType media_type =
>-          parsed_stream.GetMediaType(tmmbr.sender_ssrc(), direction);
>-      if (ExcludePacket(direction, media_type, tmmbr.sender_ssrc()))
>-        return;
>-      std::cout << log_timestamp << "\t"
>-                << "RTCP_TMMBR" << StreamInfo(direction, media_type)
>-                << "\tssrc=" << tmmbr.sender_ssrc() << std::endl;
>-      break;
>-    }
>-    case webrtc::rtcp::Tmmbn::kFeedbackMessageType: {
>-      webrtc::rtcp::Tmmbn tmmbn;
>-      if (!tmmbn.Parse(rtcp_block))
>-        return;
>-      MediaType media_type =
>-          parsed_stream.GetMediaType(tmmbn.sender_ssrc(), direction);
>-      if (ExcludePacket(direction, media_type, tmmbn.sender_ssrc()))
>-        return;
>-      std::cout << log_timestamp << "\t"
>-                << "RTCP_TMMBN" << StreamInfo(direction, media_type)
>-                << "\tssrc=" << tmmbn.sender_ssrc() << std::endl;
>-      break;
>-    }
>-    case webrtc::rtcp::RapidResyncRequest::kFeedbackMessageType: {
>-      webrtc::rtcp::RapidResyncRequest sr_req;
>-      if (!sr_req.Parse(rtcp_block))
>-        return;
>-      MediaType media_type =
>-          parsed_stream.GetMediaType(sr_req.sender_ssrc(), direction);
>-      if (ExcludePacket(direction, media_type, sr_req.sender_ssrc()))
>-        return;
>-      std::cout << log_timestamp << "\t"
>-                << "RTCP_SRREQ" << StreamInfo(direction, media_type)
>-                << "\tssrc=" << sr_req.sender_ssrc() << std::endl;
>-      break;
>-    }
>-    case webrtc::rtcp::TransportFeedback::kFeedbackMessageType: {
>-      webrtc::rtcp::TransportFeedback transport_feedback;
>-      if (!transport_feedback.Parse(rtcp_block))
>-        return;
>-      MediaType media_type = parsed_stream.GetMediaType(
>-          transport_feedback.sender_ssrc(), direction);
>-      if (ExcludePacket(direction, media_type,
>-                        transport_feedback.sender_ssrc()))
>-        return;
>-      std::cout << log_timestamp << "\t"
>-                << "RTCP_NEWFB" << StreamInfo(direction, media_type)
>-                << "\tsender_ssrc=" << transport_feedback.sender_ssrc()
>-                << "\tmedia_ssrc=" << transport_feedback.media_ssrc()
>-                << std::endl;
>-      break;
>-    }
>-    default:
>-      std::cout << log_timestamp << "\t"
>-                << "RTCP_RTPFB(UNKNOWN)" << std::endl;
>-      break;
>-  }
>-}
>-
>-void PrintPsFeedback(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>-                     const webrtc::rtcp::CommonHeader& rtcp_block,
>-                     uint64_t log_timestamp,
>-                     webrtc::PacketDirection direction) {
>-  switch (rtcp_block.fmt()) {
>-    case webrtc::rtcp::Pli::kFeedbackMessageType: {
>-      webrtc::rtcp::Pli pli;
>-      if (!pli.Parse(rtcp_block))
>-        return;
>-      MediaType media_type =
>-          parsed_stream.GetMediaType(pli.sender_ssrc(), direction);
>-      if (ExcludePacket(direction, media_type, pli.sender_ssrc()))
>-        return;
>-      std::cout << log_timestamp << "\t"
>-                << "RTCP_PLI" << StreamInfo(direction, media_type)
>-                << "\tssrc=" << pli.sender_ssrc() << std::endl;
>-      break;
>-    }
>-    case webrtc::rtcp::Fir::kFeedbackMessageType: {
>-      webrtc::rtcp::Fir fir;
>-      if (!fir.Parse(rtcp_block))
>-        return;
>-      MediaType media_type =
>-          parsed_stream.GetMediaType(fir.sender_ssrc(), direction);
>-      if (ExcludePacket(direction, media_type, fir.sender_ssrc()))
>-        return;
>-      std::cout << log_timestamp << "\t"
>-                << "RTCP_FIR" << StreamInfo(direction, media_type)
>-                << "\tssrc=" << fir.sender_ssrc() << std::endl;
>-      break;
>-    }
>-    case webrtc::rtcp::Remb::kFeedbackMessageType: {
>-      webrtc::rtcp::Remb remb;
>-      if (!remb.Parse(rtcp_block))
>-        return;
>-      MediaType media_type =
>-          parsed_stream.GetMediaType(remb.sender_ssrc(), direction);
>-      if (ExcludePacket(direction, media_type, remb.sender_ssrc()))
>-        return;
>-      std::cout << log_timestamp << "\t"
>-                << "RTCP_REMB" << StreamInfo(direction, media_type)
>-                << "\tssrc=" << remb.sender_ssrc() << std::endl;
>-      break;
>-    }
>-    default:
>-      std::cout << log_timestamp << "\t"
>-                << "RTCP_PSFB(UNKNOWN)" << std::endl;
>-      break;
>-  }
>-}
>-
>-enum class InputSource {
>-  STDIN,
>-  FILE,
>-};
>-
>-void PrintUsageGuide(const std::string& program_name) {
>-  std::cout
>-      << "Tool for printing packet information from an RtcEventLog as text.\n"
>-      << "* Run " + program_name + " --help for usage.\n"
>-      << "* Example usage for parsing a file:\n"
>-      << "  " << program_name + " input.rel\n"
>-      << "* Example usage for parsing the stdin:\n"
>-      << "  " << program_name + "\n";
>-}
>-
>-// TODO(eladalon): Return a stream or file descriptor instead.
>-InputSource ParseCommandLineFlags(int argc, char* argv[]) {
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) != 0) {
>-    PrintUsageGuide(argv[0]);
>-    exit(-1);
>-  }
>-
>-  if (FLAG_help) {
>-    PrintUsageGuide(argv[0]);
>-    std::cout << std::endl;
>-    rtc::FlagList::Print(nullptr, false);
>-    exit(0);
>-  }
>-
>-  switch (argc) {
>-    case 1:
>-      return InputSource::STDIN;
>-    case 2:
>-      return InputSource::FILE;
>-    default:
>-      PrintUsageGuide(argv[0]);
>-      exit(-1);
>-  }
>-}
>-
>-}  // namespace
>-
>-// This utility will print basic information about each packet to stdout.
>-// Note that parser will assert if the protobuf event is missing some required
>-// fields and we attempt to access them. We don't handle this at the moment.
>-int main(int argc, char* argv[]) {
>-  InputSource input_source = ParseCommandLineFlags(argc, argv);
>-
>-  if (strlen(FLAG_ssrc) > 0)
>-    RTC_CHECK(ParseSsrc(FLAG_ssrc)) << "Flag verification has failed.";
>-
>-  webrtc::RtpHeaderExtensionMap default_map = GetDefaultHeaderExtensionMap();
>-  bool default_map_used = false;
>-
>-  webrtc::ParsedRtcEventLogNew parsed_stream;
>-
>-  switch (input_source) {
>-    case InputSource::STDIN: {
>-      if (!parsed_stream.ParseStream(std::cin)) {
>-        std::cerr << "Error while parsing input stream." << std::endl;
>-        return -1;
>-      }
>-      break;
>-    }
>-    case InputSource::FILE: {
>-      if (!parsed_stream.ParseFile(argv[1])) {
>-        std::cerr << "Error while parsing input file: " << argv[1] << std::endl;
>-        return -1;
>-      }
>-      break;
>-    }
>-    default: { RTC_NOTREACHED() << "Unsupported input source."; }
>-  }
>-
>-  for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
>-    bool event_recognized = false;
>-    switch (parsed_stream.GetEventType(i)) {
>-      case webrtc::ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT: {
>-        if (FLAG_unknown) {
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tUNKNOWN_EVENT"
>-                    << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::LOG_START: {
>-        if (FLAG_startstop) {
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tLOG_START"
>-                    << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::LOG_END: {
>-        if (FLAG_startstop) {
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tLOG_END"
>-                    << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::RTP_EVENT: {
>-        if (FLAG_rtp) {
>-          size_t header_length;
>-          size_t total_length;
>-          uint8_t header[IP_PACKET_SIZE];
>-          webrtc::PacketDirection direction;
>-          const webrtc::RtpHeaderExtensionMap* extension_map =
>-              parsed_stream.GetRtpHeader(i, &direction, header, &header_length,
>-                                         &total_length, nullptr);
>-
>-          if (extension_map == nullptr) {
>-            extension_map = &default_map;
>-            if (!default_map_used)
>-              RTC_LOG(LS_WARNING) << "Using default header extension map";
>-            default_map_used = true;
>-          }
>-
>-          // Parse header to get SSRC and RTP time.
>-          webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
>-          webrtc::RTPHeader parsed_header;
>-          rtp_parser.Parse(&parsed_header, extension_map);
>-          MediaType media_type =
>-              parsed_stream.GetMediaType(parsed_header.ssrc, direction);
>-
>-          if (ExcludePacket(direction, media_type, parsed_header.ssrc)) {
>-            event_recognized = true;
>-            break;
>-          }
>-
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
>-                    << StreamInfo(direction, media_type)
>-                    << "\tssrc=" << parsed_header.ssrc
>-                    << "\ttimestamp=" << parsed_header.timestamp;
>-          if (parsed_header.extension.hasAbsoluteSendTime) {
>-            std::cout << "\tAbsSendTime="
>-                      << parsed_header.extension.absoluteSendTime;
>-          }
>-          if (parsed_header.extension.hasVideoContentType) {
>-            std::cout << "\tContentType="
>-                      << static_cast<int>(
>-                             parsed_header.extension.videoContentType);
>-          }
>-          if (parsed_header.extension.hasVideoRotation) {
>-            std::cout << "\tRotation="
>-                      << static_cast<int>(
>-                             parsed_header.extension.videoRotation);
>-          }
>-          if (parsed_header.extension.hasTransportSequenceNumber) {
>-            std::cout << "\tTransportSeq="
>-                      << parsed_header.extension.transportSequenceNumber;
>-          }
>-          if (parsed_header.extension.hasTransmissionTimeOffset) {
>-            std::cout << "\tTransmTimeOffset="
>-                      << parsed_header.extension.transmissionTimeOffset;
>-          }
>-          if (parsed_header.extension.hasAudioLevel) {
>-            std::cout << "\tAudioLevel="
>-                      << static_cast<int>(parsed_header.extension.audioLevel);
>-          }
>-          std::cout << std::endl;
>-          if (FLAG_print_full_packets) {
>-            // TODO(terelius): Rewrite this file to use printf instead of cout.
>-            std::cout << "\t\t" << std::hex;
>-            char prev_fill = std::cout.fill('0');
>-            for (size_t i = 0; i < header_length; i++) {
>-              std::cout << std::setw(2) << static_cast<unsigned>(header[i]);
>-              if (i % 4 == 3)
>-                std::cout << " ";  // Separator between 32-bit words.
>-            }
>-            std::cout.fill(prev_fill);
>-            std::cout << std::dec << std::endl;
>-          }
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::RTCP_EVENT: {
>-        if (FLAG_rtcp) {
>-          size_t length;
>-          uint8_t packet[IP_PACKET_SIZE];
>-          webrtc::PacketDirection direction;
>-          parsed_stream.GetRtcpPacket(i, &direction, packet, &length);
>-
>-          webrtc::rtcp::CommonHeader rtcp_block;
>-          const uint8_t* packet_end = packet + length;
>-          for (const uint8_t* next_block = packet; next_block != packet_end;
>-               next_block = rtcp_block.NextPacket()) {
>-            ptrdiff_t remaining_blocks_size = packet_end - next_block;
>-            RTC_DCHECK_GT(remaining_blocks_size, 0);
>-            if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
>-              RTC_LOG(LS_WARNING) << "Failed to parse RTCP";
>-              break;
>-            }
>-
>-            uint64_t log_timestamp = parsed_stream.GetTimestamp(i);
>-            switch (rtcp_block.type()) {
>-              case webrtc::rtcp::SenderReport::kPacketType:
>-                PrintSenderReport(parsed_stream, rtcp_block, log_timestamp,
>-                                  direction);
>-                break;
>-              case webrtc::rtcp::ReceiverReport::kPacketType:
>-                PrintReceiverReport(parsed_stream, rtcp_block, log_timestamp,
>-                                    direction);
>-                break;
>-              case webrtc::rtcp::Sdes::kPacketType:
>-                PrintSdes(rtcp_block, log_timestamp, direction);
>-                break;
>-              case webrtc::rtcp::ExtendedReports::kPacketType:
>-                PrintXr(parsed_stream, rtcp_block, log_timestamp, direction);
>-                break;
>-              case webrtc::rtcp::Bye::kPacketType:
>-                PrintBye(parsed_stream, rtcp_block, log_timestamp, direction);
>-                break;
>-              case webrtc::rtcp::Rtpfb::kPacketType:
>-                PrintRtpFeedback(parsed_stream, rtcp_block, log_timestamp,
>-                                 direction);
>-                break;
>-              case webrtc::rtcp::Psfb::kPacketType:
>-                PrintPsFeedback(parsed_stream, rtcp_block, log_timestamp,
>-                                direction);
>-                break;
>-              default:
>-                break;
>-            }
>-            if (FLAG_print_full_packets) {
>-              std::cout << "\t\t" << std::hex;
>-              char prev_fill = std::cout.fill('0');
>-              for (const uint8_t* p = next_block; p < rtcp_block.NextPacket();
>-                   p++) {
>-                std::cout << std::setw(2) << static_cast<unsigned>(*p);
>-                ptrdiff_t chars_printed = p - next_block;
>-                if (chars_printed % 4 == 3)
>-                  std::cout << " ";  // Separator between 32-bit words.
>-              }
>-              std::cout.fill(prev_fill);
>-              std::cout << std::dec << std::endl;
>-            }
>-          }
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT: {
>-        if (FLAG_playout) {
>-          auto audio_playout = parsed_stream.GetAudioPlayout(i);
>-          std::cout << audio_playout.log_time_us() << "\tAUDIO_PLAYOUT"
>-                    << "\tssrc=" << audio_playout.ssrc << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::LOSS_BASED_BWE_UPDATE: {
>-        if (FLAG_bwe) {
>-          auto bwe_update = parsed_stream.GetLossBasedBweUpdate(i);
>-          std::cout << bwe_update.log_time_us() << "\tBWE(LOSS_BASED)"
>-                    << "\tbitrate_bps=" << bwe_update.bitrate_bps
>-                    << "\tfraction_lost="
>-                    << static_cast<unsigned>(bwe_update.fraction_lost)
>-                    << "\texpected_packets=" << bwe_update.expected_packets
>-                    << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::DELAY_BASED_BWE_UPDATE: {
>-        if (FLAG_bwe) {
>-          auto bwe_update = parsed_stream.GetDelayBasedBweUpdate(i);
>-          std::cout << bwe_update.log_time_us() << "\tBWE(DELAY_BASED)"
>-                    << "\tbitrate_bps=" << bwe_update.bitrate_bps
>-                    << "\tdetector_state="
>-                    << static_cast<int>(bwe_update.detector_state) << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::
>-          VIDEO_RECEIVER_CONFIG_EVENT: {
>-        if (FLAG_config && FLAG_video && FLAG_incoming) {
>-          webrtc::rtclog::StreamConfig config =
>-              parsed_stream.GetVideoReceiveConfig(i);
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
>-                    << "\tssrc=" << config.remote_ssrc
>-                    << "\tfeedback_ssrc=" << config.local_ssrc;
>-          std::cout << "\textensions={";
>-          for (const auto& extension : config.rtp_extensions) {
>-            std::cout << extension.ToString() << ",";
>-          }
>-          std::cout << "}";
>-          std::cout << "\tcodecs={";
>-          for (const auto& codec : config.codecs) {
>-            std::cout << "{name: " << codec.payload_name
>-                      << ", payload_type: " << codec.payload_type
>-                      << ", rtx_payload_type: " << codec.rtx_payload_type
>-                      << "}";
>-          }
>-          std::cout << "}" << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::VIDEO_SENDER_CONFIG_EVENT: {
>-        if (FLAG_config && FLAG_video && FLAG_outgoing) {
>-          std::vector<webrtc::rtclog::StreamConfig> configs =
>-              parsed_stream.GetVideoSendConfig(i);
>-          for (const auto& config : configs) {
>-            std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
>-            std::cout << "\tssrcs=" << config.local_ssrc;
>-            std::cout << "\trtx_ssrcs=" << config.rtx_ssrc;
>-            std::cout << "\textensions={";
>-            for (const auto& extension : config.rtp_extensions) {
>-              std::cout << extension.ToString() << ",";
>-            }
>-            std::cout << "}";
>-            std::cout << "\tcodecs={";
>-            for (const auto& codec : config.codecs) {
>-              std::cout << "{name: " << codec.payload_name
>-                        << ", payload_type: " << codec.payload_type
>-                        << ", rtx_payload_type: " << codec.rtx_payload_type
>-                        << "}";
>-            }
>-            std::cout << "}" << std::endl;
>-          }
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::
>-          AUDIO_RECEIVER_CONFIG_EVENT: {
>-        if (FLAG_config && FLAG_audio && FLAG_incoming) {
>-          webrtc::rtclog::StreamConfig config =
>-              parsed_stream.GetAudioReceiveConfig(i);
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
>-                    << "\tssrc=" << config.remote_ssrc
>-                    << "\tfeedback_ssrc=" << config.local_ssrc;
>-          std::cout << "\textensions={";
>-          for (const auto& extension : config.rtp_extensions) {
>-            std::cout << extension.ToString() << ",";
>-          }
>-          std::cout << "}";
>-          std::cout << "\tcodecs={";
>-          for (const auto& codec : config.codecs) {
>-            std::cout << "{name: " << codec.payload_name
>-                      << ", payload_type: " << codec.payload_type
>-                      << ", rtx_payload_type: " << codec.rtx_payload_type
>-                      << "}";
>-          }
>-          std::cout << "}" << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::AUDIO_SENDER_CONFIG_EVENT: {
>-        if (FLAG_config && FLAG_audio && FLAG_outgoing) {
>-          webrtc::rtclog::StreamConfig config =
>-              parsed_stream.GetAudioSendConfig(i);
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
>-                    << "\tssrc=" << config.local_ssrc;
>-          std::cout << "\textensions={";
>-          for (const auto& extension : config.rtp_extensions) {
>-            std::cout << extension.ToString() << ",";
>-          }
>-          std::cout << "}";
>-          std::cout << "\tcodecs={";
>-          for (const auto& codec : config.codecs) {
>-            std::cout << "{name: " << codec.payload_name
>-                      << ", payload_type: " << codec.payload_type
>-                      << ", rtx_payload_type: " << codec.rtx_payload_type
>-                      << "}";
>-          }
>-          std::cout << "}" << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::
>-          AUDIO_NETWORK_ADAPTATION_EVENT: {
>-        if (FLAG_ana) {
>-          auto ana_event = parsed_stream.GetAudioNetworkAdaptation(i);
>-          char buffer[300];
>-          rtc::SimpleStringBuilder builder(buffer);
>-          builder << parsed_stream.GetTimestamp(i) << "\tANA_UPDATE";
>-          if (ana_event.config.bitrate_bps) {
>-            builder << "\tbitrate_bps=" << *ana_event.config.bitrate_bps;
>-          }
>-          if (ana_event.config.frame_length_ms) {
>-            builder << "\tframe_length_ms="
>-                    << *ana_event.config.frame_length_ms;
>-          }
>-          if (ana_event.config.uplink_packet_loss_fraction) {
>-            builder << "\tuplink_packet_loss_fraction="
>-                    << *ana_event.config.uplink_packet_loss_fraction;
>-          }
>-          if (ana_event.config.enable_fec) {
>-            builder << "\tenable_fec=" << *ana_event.config.enable_fec;
>-          }
>-          if (ana_event.config.enable_dtx) {
>-            builder << "\tenable_dtx=" << *ana_event.config.enable_dtx;
>-          }
>-          if (ana_event.config.num_channels) {
>-            builder << "\tnum_channels=" << *ana_event.config.num_channels;
>-          }
>-          std::cout << builder.str() << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::
>-          BWE_PROBE_CLUSTER_CREATED_EVENT: {
>-        if (FLAG_probe) {
>-          auto probe_event = parsed_stream.GetBweProbeClusterCreated(i);
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_CREATED("
>-                    << probe_event.id << ")"
>-                    << "\tbitrate_bps=" << probe_event.bitrate_bps
>-                    << "\tmin_packets=" << probe_event.min_packets
>-                    << "\tmin_bytes=" << probe_event.min_bytes << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::BWE_PROBE_FAILURE_EVENT: {
>-        if (FLAG_probe) {
>-          webrtc::LoggedBweProbeFailureEvent probe_result =
>-              parsed_stream.GetBweProbeFailure(i);
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_FAILURE("
>-                    << probe_result.id << ")"
>-                    << "\tfailure_reason="
>-                    << static_cast<int>(probe_result.failure_reason)
>-                    << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::BWE_PROBE_SUCCESS_EVENT: {
>-        if (FLAG_probe) {
>-          webrtc::LoggedBweProbeSuccessEvent probe_result =
>-              parsed_stream.GetBweProbeSuccess(i);
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_SUCCESS("
>-                    << probe_result.id << ")"
>-                    << "\tbitrate_bps=" << probe_result.bitrate_bps
>-                    << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::ALR_STATE_EVENT: {
>-        if (FLAG_bwe) {
>-          webrtc::LoggedAlrStateEvent alr_state = parsed_stream.GetAlrState(i);
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tALR_STATE"
>-                    << "\tin_alr=" << alr_state.in_alr << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_CONFIG: {
>-        if (FLAG_ice) {
>-          webrtc::LoggedIceCandidatePairConfig ice_cp_config =
>-              parsed_stream.GetIceCandidatePairConfig(i);
>-          // TODO(qingsi): convert the numeric representation of states to text
>-          std::cout << parsed_stream.GetTimestamp(i)
>-                    << "\tICE_CANDIDATE_PAIR_CONFIG"
>-                    << "\ttype=" << static_cast<int>(ice_cp_config.type)
>-                    << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-
>-      case webrtc::ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_EVENT: {
>-        if (FLAG_ice) {
>-          webrtc::LoggedIceCandidatePairEvent ice_cp_event =
>-              parsed_stream.GetIceCandidatePairEvent(i);
>-          // TODO(qingsi): convert the numeric representation of states to text
>-          std::cout << parsed_stream.GetTimestamp(i)
>-                    << "\tICE_CANDIDATE_PAIR_EVENT"
>-                    << "\ttype=" << static_cast<int>(ice_cp_event.type)
>-                    << std::endl;
>-        }
>-        event_recognized = true;
>-        break;
>-      }
>-    }
>-
>-    if (!event_recognized) {
>-      std::cout << "Unrecognized event ("
>-                << static_cast<int>(parsed_stream.GetEventType(i)) << ")"
>-                << std::endl;
>-    }
>-  }
>-  return 0;
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_impl.cc
>index 554a0283fd6f9737aa04e2e2a84f1d8888442146..c022a3d4185cf143b94e0e281bc6fe725cee3672 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_impl.cc
>@@ -18,8 +18,9 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>+#include "api/rtceventlogoutput.h"
> #include "logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h"
>-#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
>+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/event.h"
>@@ -64,7 +65,11 @@ std::unique_ptr<RtcEventLogEncoder> CreateEncoder(
>     RtcEventLog::EncodingType type) {
>   switch (type) {
>     case RtcEventLog::EncodingType::Legacy:
>+      RTC_LOG(LS_INFO) << "Creating legacy encoder for RTC event log.";
>       return absl::make_unique<RtcEventLogEncoderLegacy>();
>+    case RtcEventLog::EncodingType::NewFormat:
>+      RTC_LOG(LS_INFO) << "Creating new format encoder for RTC event log.";
>+      return absl::make_unique<RtcEventLogEncoderNewFormat>();
>     default:
>       RTC_LOG(LS_ERROR) << "Unknown RtcEventLog encoder type (" << int(type)
>                         << ")";
>@@ -168,20 +173,19 @@ bool RtcEventLogImpl::StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>     return false;
>   }
> 
>-  // TODO(terelius): The mapping between log timestamps and UTC should be stored
>-  // in the event_log START event.
>   const int64_t timestamp_us = rtc::TimeMicros();
>   const int64_t utc_time_us = rtc::TimeUTCMicros();
>   RTC_LOG(LS_INFO) << "Starting WebRTC event log. (Timestamp, UTC) = "
>                    << "(" << timestamp_us << ", " << utc_time_us << ").";
> 
>   // Binding to |this| is safe because |this| outlives the |task_queue_|.
>-  auto start = [this, timestamp_us](std::unique_ptr<RtcEventLogOutput> output) {
>+  auto start = [this, timestamp_us,
>+                utc_time_us](std::unique_ptr<RtcEventLogOutput> output) {
>     RTC_DCHECK_RUN_ON(task_queue_.get());
>     RTC_DCHECK(output->IsActive());
>     event_output_ = std::move(output);
>     num_config_events_written_ = 0;
>-    WriteToOutput(event_encoder_->EncodeLogStart(timestamp_us));
>+    WriteToOutput(event_encoder_->EncodeLogStart(timestamp_us, utc_time_us));
>     LogEventsFromMemoryToOutput();
>   };
> 
>@@ -197,7 +201,7 @@ void RtcEventLogImpl::StopLogging() {
> 
>   RTC_LOG(LS_INFO) << "Stopping WebRTC event log.";
> 
>-  rtc::Event output_stopped(true, false);
>+  rtc::Event output_stopped;
> 
>   // Binding to |this| is safe because |this| outlives the |task_queue_|.
>   task_queue_->PostTask([this, &output_stopped]() {
>@@ -372,8 +376,4 @@ std::unique_ptr<RtcEventLog> RtcEventLog::Create(
> #endif  // ENABLE_RTC_EVENT_LOG
> }
> 
>-std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
>-  return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
>deleted file mode 100644
>index 802af189aad59270668f57c335d22c31d46b35ed..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
>+++ /dev/null
>@@ -1,851 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "logging/rtc_event_log/rtc_event_log_parser.h"
>-
>-#include <stdint.h>
>-#include <string.h>
>-
>-#include <algorithm>
>-#include <fstream>
>-#include <istream>
>-#include <map>
>-#include <utility>
>-
>-#include "logging/rtc_event_log/rtc_event_log.h"
>-#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
>-#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/protobuf_utils.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
>-  switch (rtcp_mode) {
>-    case rtclog::VideoReceiveConfig::RTCP_COMPOUND:
>-      return RtcpMode::kCompound;
>-    case rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE:
>-      return RtcpMode::kReducedSize;
>-  }
>-  RTC_NOTREACHED();
>-  return RtcpMode::kOff;
>-}
>-
>-ParsedRtcEventLog::EventType GetRuntimeEventType(
>-    rtclog::Event::EventType event_type) {
>-  switch (event_type) {
>-    case rtclog::Event::UNKNOWN_EVENT:
>-      return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
>-    case rtclog::Event::LOG_START:
>-      return ParsedRtcEventLog::EventType::LOG_START;
>-    case rtclog::Event::LOG_END:
>-      return ParsedRtcEventLog::EventType::LOG_END;
>-    case rtclog::Event::RTP_EVENT:
>-      return ParsedRtcEventLog::EventType::RTP_EVENT;
>-    case rtclog::Event::RTCP_EVENT:
>-      return ParsedRtcEventLog::EventType::RTCP_EVENT;
>-    case rtclog::Event::AUDIO_PLAYOUT_EVENT:
>-      return ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT;
>-    case rtclog::Event::LOSS_BASED_BWE_UPDATE:
>-      return ParsedRtcEventLog::EventType::LOSS_BASED_BWE_UPDATE;
>-    case rtclog::Event::DELAY_BASED_BWE_UPDATE:
>-      return ParsedRtcEventLog::EventType::DELAY_BASED_BWE_UPDATE;
>-    case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
>-      return ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
>-    case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
>-      return ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT;
>-    case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
>-      return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
>-    case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
>-      return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT;
>-    case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
>-      return ParsedRtcEventLog::EventType::AUDIO_NETWORK_ADAPTATION_EVENT;
>-    case rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT:
>-      return ParsedRtcEventLog::EventType::BWE_PROBE_CLUSTER_CREATED_EVENT;
>-    case rtclog::Event::BWE_PROBE_RESULT_EVENT:
>-      return ParsedRtcEventLog::EventType::BWE_PROBE_RESULT_EVENT;
>-    case rtclog::Event::ALR_STATE_EVENT:
>-      return ParsedRtcEventLog::EventType::ALR_STATE_EVENT;
>-    case rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG:
>-      return ParsedRtcEventLog::EventType::ICE_CANDIDATE_PAIR_CONFIG;
>-    case rtclog::Event::ICE_CANDIDATE_PAIR_EVENT:
>-      return ParsedRtcEventLog::EventType::ICE_CANDIDATE_PAIR_EVENT;
>-  }
>-  return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
>-}
>-
>-BandwidthUsage GetRuntimeDetectorState(
>-    rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
>-  switch (detector_state) {
>-    case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
>-      return BandwidthUsage::kBwNormal;
>-    case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
>-      return BandwidthUsage::kBwUnderusing;
>-    case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
>-      return BandwidthUsage::kBwOverusing;
>-  }
>-  RTC_NOTREACHED();
>-  return BandwidthUsage::kBwNormal;
>-}
>-
>-IceCandidatePairConfigType GetRuntimeIceCandidatePairConfigType(
>-    rtclog::IceCandidatePairConfig::IceCandidatePairConfigType type) {
>-  switch (type) {
>-    case rtclog::IceCandidatePairConfig::ADDED:
>-      return IceCandidatePairConfigType::kAdded;
>-    case rtclog::IceCandidatePairConfig::UPDATED:
>-      return IceCandidatePairConfigType::kUpdated;
>-    case rtclog::IceCandidatePairConfig::DESTROYED:
>-      return IceCandidatePairConfigType::kDestroyed;
>-    case rtclog::IceCandidatePairConfig::SELECTED:
>-      return IceCandidatePairConfigType::kSelected;
>-  }
>-  RTC_NOTREACHED();
>-  return IceCandidatePairConfigType::kAdded;
>-}
>-
>-IceCandidateType GetRuntimeIceCandidateType(
>-    rtclog::IceCandidatePairConfig::IceCandidateType type) {
>-  switch (type) {
>-    case rtclog::IceCandidatePairConfig::LOCAL:
>-      return IceCandidateType::kLocal;
>-    case rtclog::IceCandidatePairConfig::STUN:
>-      return IceCandidateType::kStun;
>-    case rtclog::IceCandidatePairConfig::PRFLX:
>-      return IceCandidateType::kPrflx;
>-    case rtclog::IceCandidatePairConfig::RELAY:
>-      return IceCandidateType::kRelay;
>-    case rtclog::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE:
>-      return IceCandidateType::kUnknown;
>-  }
>-  RTC_NOTREACHED();
>-  return IceCandidateType::kUnknown;
>-}
>-
>-IceCandidatePairProtocol GetRuntimeIceCandidatePairProtocol(
>-    rtclog::IceCandidatePairConfig::Protocol protocol) {
>-  switch (protocol) {
>-    case rtclog::IceCandidatePairConfig::UDP:
>-      return IceCandidatePairProtocol::kUdp;
>-    case rtclog::IceCandidatePairConfig::TCP:
>-      return IceCandidatePairProtocol::kTcp;
>-    case rtclog::IceCandidatePairConfig::SSLTCP:
>-      return IceCandidatePairProtocol::kSsltcp;
>-    case rtclog::IceCandidatePairConfig::TLS:
>-      return IceCandidatePairProtocol::kTls;
>-    case rtclog::IceCandidatePairConfig::UNKNOWN_PROTOCOL:
>-      return IceCandidatePairProtocol::kUnknown;
>-  }
>-  RTC_NOTREACHED();
>-  return IceCandidatePairProtocol::kUnknown;
>-}
>-
>-IceCandidatePairAddressFamily GetRuntimeIceCandidatePairAddressFamily(
>-    rtclog::IceCandidatePairConfig::AddressFamily address_family) {
>-  switch (address_family) {
>-    case rtclog::IceCandidatePairConfig::IPV4:
>-      return IceCandidatePairAddressFamily::kIpv4;
>-    case rtclog::IceCandidatePairConfig::IPV6:
>-      return IceCandidatePairAddressFamily::kIpv6;
>-    case rtclog::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY:
>-      return IceCandidatePairAddressFamily::kUnknown;
>-  }
>-  RTC_NOTREACHED();
>-  return IceCandidatePairAddressFamily::kUnknown;
>-}
>-
>-IceCandidateNetworkType GetRuntimeIceCandidateNetworkType(
>-    rtclog::IceCandidatePairConfig::NetworkType network_type) {
>-  switch (network_type) {
>-    case rtclog::IceCandidatePairConfig::ETHERNET:
>-      return IceCandidateNetworkType::kEthernet;
>-    case rtclog::IceCandidatePairConfig::LOOPBACK:
>-      return IceCandidateNetworkType::kLoopback;
>-    case rtclog::IceCandidatePairConfig::WIFI:
>-      return IceCandidateNetworkType::kWifi;
>-    case rtclog::IceCandidatePairConfig::VPN:
>-      return IceCandidateNetworkType::kVpn;
>-    case rtclog::IceCandidatePairConfig::CELLULAR:
>-      return IceCandidateNetworkType::kCellular;
>-    case rtclog::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE:
>-      return IceCandidateNetworkType::kUnknown;
>-  }
>-  RTC_NOTREACHED();
>-  return IceCandidateNetworkType::kUnknown;
>-}
>-
>-IceCandidatePairEventType GetRuntimeIceCandidatePairEventType(
>-    rtclog::IceCandidatePairEvent::IceCandidatePairEventType type) {
>-  switch (type) {
>-    case rtclog::IceCandidatePairEvent::CHECK_SENT:
>-      return IceCandidatePairEventType::kCheckSent;
>-    case rtclog::IceCandidatePairEvent::CHECK_RECEIVED:
>-      return IceCandidatePairEventType::kCheckReceived;
>-    case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_SENT:
>-      return IceCandidatePairEventType::kCheckResponseSent;
>-    case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED:
>-      return IceCandidatePairEventType::kCheckResponseReceived;
>-  }
>-  RTC_NOTREACHED();
>-  return IceCandidatePairEventType::kCheckSent;
>-}
>-
>-std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
>-  uint64_t varint = 0;
>-  for (size_t bytes_read = 0; bytes_read < 10; ++bytes_read) {
>-    // The most significant bit of each byte is 0 if it is the last byte in
>-    // the varint and 1 otherwise. Thus, we take the 7 least significant bits
>-    // of each byte and shift them 7 bits for each byte read previously to get
>-    // the (unsigned) integer.
>-    int byte = stream.get();
>-    if (stream.eof()) {
>-      return std::make_pair(varint, false);
>-    }
>-    RTC_DCHECK_GE(byte, 0);
>-    RTC_DCHECK_LE(byte, 255);
>-    varint |= static_cast<uint64_t>(byte & 0x7F) << (7 * bytes_read);
>-    if ((byte & 0x80) == 0) {
>-      return std::make_pair(varint, true);
>-    }
>-  }
>-  return std::make_pair(varint, false);
>-}
>-
>-void GetHeaderExtensions(std::vector<RtpExtension>* header_extensions,
>-                         const RepeatedPtrField<rtclog::RtpHeaderExtension>&
>-                             proto_header_extensions) {
>-  header_extensions->clear();
>-  for (auto& p : proto_header_extensions) {
>-    RTC_CHECK(p.has_name());
>-    RTC_CHECK(p.has_id());
>-    const std::string& name = p.name();
>-    int id = p.id();
>-    header_extensions->push_back(RtpExtension(name, id));
>-  }
>-}
>-
>-}  // namespace
>-
>-ParsedRtcEventLog::ParsedRtcEventLog() = default;
>-ParsedRtcEventLog::~ParsedRtcEventLog() = default;
>-
>-ParsedRtcEventLog::BweProbeResultEvent::BweProbeResultEvent() = default;
>-ParsedRtcEventLog::BweProbeResultEvent::BweProbeResultEvent(
>-    const BweProbeResultEvent&) = default;
>-
>-bool ParsedRtcEventLog::ParseFile(const std::string& filename) {
>-  std::ifstream file(filename, std::ios_base::in | std::ios_base::binary);
>-  if (!file.good() || !file.is_open()) {
>-    RTC_LOG(LS_WARNING) << "Could not open file for reading.";
>-    return false;
>-  }
>-
>-  return ParseStream(file);
>-}
>-
>-bool ParsedRtcEventLog::ParseString(const std::string& s) {
>-  std::istringstream stream(s, std::ios_base::in | std::ios_base::binary);
>-  return ParseStream(stream);
>-}
>-
>-bool ParsedRtcEventLog::ParseStream(std::istream& stream) {
>-  events_.clear();
>-  const size_t kMaxEventSize = (1u << 16) - 1;
>-  std::vector<char> tmp_buffer(kMaxEventSize);
>-  uint64_t tag;
>-  uint64_t message_length;
>-  bool success;
>-
>-  RTC_DCHECK(stream.good());
>-
>-  while (1) {
>-    // Check whether we have reached end of file.
>-    stream.peek();
>-    if (stream.eof()) {
>-      // Process all extensions maps for faster look-up later.
>-      for (auto& event_stream : streams_) {
>-        rtp_extensions_maps_[StreamId(event_stream.ssrc,
>-                                      event_stream.direction)] =
>-            &event_stream.rtp_extensions_map;
>-      }
>-      return true;
>-    }
>-
>-    // Read the next message tag. The tag number is defined as
>-    // (fieldnumber << 3) | wire_type. In our case, the field number is
>-    // supposed to be 1 and the wire type for an
>-    // length-delimited field is 2.
>-    const uint64_t kExpectedTag = (1 << 3) | 2;
>-    std::tie(tag, success) = ParseVarInt(stream);
>-    if (!success) {
>-      RTC_LOG(LS_WARNING)
>-          << "Missing field tag from beginning of protobuf event.";
>-      return false;
>-    } else if (tag != kExpectedTag) {
>-      RTC_LOG(LS_WARNING)
>-          << "Unexpected field tag at beginning of protobuf event.";
>-      return false;
>-    }
>-
>-    // Read the length field.
>-    std::tie(message_length, success) = ParseVarInt(stream);
>-    if (!success) {
>-      RTC_LOG(LS_WARNING) << "Missing message length after protobuf field tag.";
>-      return false;
>-    } else if (message_length > kMaxEventSize) {
>-      RTC_LOG(LS_WARNING) << "Protobuf message length is too large.";
>-      return false;
>-    }
>-
>-    // Read the next protobuf event to a temporary char buffer.
>-    stream.read(tmp_buffer.data(), message_length);
>-    if (stream.gcount() != static_cast<int>(message_length)) {
>-      RTC_LOG(LS_WARNING) << "Failed to read protobuf message from file.";
>-      return false;
>-    }
>-
>-    // Parse the protobuf event from the buffer.
>-    rtclog::Event event;
>-    if (!event.ParseFromArray(tmp_buffer.data(), message_length)) {
>-      RTC_LOG(LS_WARNING) << "Failed to parse protobuf message.";
>-      return false;
>-    }
>-
>-    EventType type = GetRuntimeEventType(event.type());
>-    switch (type) {
>-      case VIDEO_RECEIVER_CONFIG_EVENT: {
>-        rtclog::StreamConfig config = GetVideoReceiveConfig(event);
>-        streams_.emplace_back(config.remote_ssrc, MediaType::VIDEO,
>-                              kIncomingPacket,
>-                              RtpHeaderExtensionMap(config.rtp_extensions));
>-        streams_.emplace_back(config.local_ssrc, MediaType::VIDEO,
>-                              kOutgoingPacket,
>-                              RtpHeaderExtensionMap(config.rtp_extensions));
>-        break;
>-      }
>-      case VIDEO_SENDER_CONFIG_EVENT: {
>-        std::vector<rtclog::StreamConfig> configs = GetVideoSendConfig(event);
>-        for (size_t i = 0; i < configs.size(); i++) {
>-          streams_.emplace_back(
>-              configs[i].local_ssrc, MediaType::VIDEO, kOutgoingPacket,
>-              RtpHeaderExtensionMap(configs[i].rtp_extensions));
>-
>-          streams_.emplace_back(
>-              configs[i].rtx_ssrc, MediaType::VIDEO, kOutgoingPacket,
>-              RtpHeaderExtensionMap(configs[i].rtp_extensions));
>-        }
>-        break;
>-      }
>-      case AUDIO_RECEIVER_CONFIG_EVENT: {
>-        rtclog::StreamConfig config = GetAudioReceiveConfig(event);
>-        streams_.emplace_back(config.remote_ssrc, MediaType::AUDIO,
>-                              kIncomingPacket,
>-                              RtpHeaderExtensionMap(config.rtp_extensions));
>-        streams_.emplace_back(config.local_ssrc, MediaType::AUDIO,
>-                              kOutgoingPacket,
>-                              RtpHeaderExtensionMap(config.rtp_extensions));
>-        break;
>-      }
>-      case AUDIO_SENDER_CONFIG_EVENT: {
>-        rtclog::StreamConfig config = GetAudioSendConfig(event);
>-        streams_.emplace_back(config.local_ssrc, MediaType::AUDIO,
>-                              kOutgoingPacket,
>-                              RtpHeaderExtensionMap(config.rtp_extensions));
>-        break;
>-      }
>-      default:
>-        break;
>-    }
>-
>-    events_.push_back(event);
>-  }
>-}
>-
>-size_t ParsedRtcEventLog::GetNumberOfEvents() const {
>-  return events_.size();
>-}
>-
>-int64_t ParsedRtcEventLog::GetTimestamp(size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  RTC_CHECK(event.has_timestamp_us());
>-  return event.timestamp_us();
>-}
>-
>-ParsedRtcEventLog::EventType ParsedRtcEventLog::GetEventType(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  RTC_CHECK(event.has_type());
>-  return GetRuntimeEventType(event.type());
>-}
>-
>-// The header must have space for at least IP_PACKET_SIZE bytes.
>-webrtc::RtpHeaderExtensionMap* ParsedRtcEventLog::GetRtpHeader(
>-    size_t index,
>-    PacketDirection* incoming,
>-    uint8_t* header,
>-    size_t* header_length,
>-    size_t* total_length,
>-    int* probe_cluster_id) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::RTP_EVENT);
>-  RTC_CHECK(event.has_rtp_packet());
>-  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
>-  // Get direction of packet.
>-  RTC_CHECK(rtp_packet.has_incoming());
>-  if (incoming != nullptr) {
>-    *incoming = rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
>-  }
>-  // Get packet length.
>-  RTC_CHECK(rtp_packet.has_packet_length());
>-  if (total_length != nullptr) {
>-    *total_length = rtp_packet.packet_length();
>-  }
>-  // Get header length.
>-  RTC_CHECK(rtp_packet.has_header());
>-  if (header_length != nullptr) {
>-    *header_length = rtp_packet.header().size();
>-  }
>-  // Get header contents.
>-  if (header != nullptr) {
>-    const size_t kMinRtpHeaderSize = 12;
>-    RTC_CHECK_GE(rtp_packet.header().size(), kMinRtpHeaderSize);
>-    RTC_CHECK_LE(rtp_packet.header().size(),
>-                 static_cast<size_t>(IP_PACKET_SIZE));
>-    memcpy(header, rtp_packet.header().data(), rtp_packet.header().size());
>-    uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(header + 8);
>-    StreamId stream_id(
>-        ssrc, rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket);
>-    auto it = rtp_extensions_maps_.find(stream_id);
>-    if (it != rtp_extensions_maps_.end()) {
>-      return it->second;
>-    }
>-  }
>-  if (probe_cluster_id != nullptr) {
>-    if (rtp_packet.has_probe_cluster_id()) {
>-      *probe_cluster_id = rtp_packet.probe_cluster_id();
>-      RTC_CHECK_NE(*probe_cluster_id, PacedPacketInfo::kNotAProbe);
>-    } else {
>-      *probe_cluster_id = PacedPacketInfo::kNotAProbe;
>-    }
>-  }
>-  return nullptr;
>-}
>-
>-// The packet must have space for at least IP_PACKET_SIZE bytes.
>-void ParsedRtcEventLog::GetRtcpPacket(size_t index,
>-                                      PacketDirection* incoming,
>-                                      uint8_t* packet,
>-                                      size_t* length) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::RTCP_EVENT);
>-  RTC_CHECK(event.has_rtcp_packet());
>-  const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
>-  // Get direction of packet.
>-  RTC_CHECK(rtcp_packet.has_incoming());
>-  if (incoming != nullptr) {
>-    *incoming = rtcp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
>-  }
>-  // Get packet length.
>-  RTC_CHECK(rtcp_packet.has_packet_data());
>-  if (length != nullptr) {
>-    *length = rtcp_packet.packet_data().size();
>-  }
>-  // Get packet contents.
>-  if (packet != nullptr) {
>-    RTC_CHECK_LE(rtcp_packet.packet_data().size(),
>-                 static_cast<unsigned>(IP_PACKET_SIZE));
>-    memcpy(packet, rtcp_packet.packet_data().data(),
>-           rtcp_packet.packet_data().size());
>-  }
>-}
>-
>-rtclog::StreamConfig ParsedRtcEventLog::GetVideoReceiveConfig(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  return GetVideoReceiveConfig(events_[index]);
>-}
>-
>-rtclog::StreamConfig ParsedRtcEventLog::GetVideoReceiveConfig(
>-    const rtclog::Event& event) const {
>-  rtclog::StreamConfig config;
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
>-  RTC_CHECK(event.has_video_receiver_config());
>-  const rtclog::VideoReceiveConfig& receiver_config =
>-      event.video_receiver_config();
>-  // Get SSRCs.
>-  RTC_CHECK(receiver_config.has_remote_ssrc());
>-  config.remote_ssrc = receiver_config.remote_ssrc();
>-  RTC_CHECK(receiver_config.has_local_ssrc());
>-  config.local_ssrc = receiver_config.local_ssrc();
>-  config.rtx_ssrc = 0;
>-  // Get RTCP settings.
>-  RTC_CHECK(receiver_config.has_rtcp_mode());
>-  config.rtcp_mode = GetRuntimeRtcpMode(receiver_config.rtcp_mode());
>-  RTC_CHECK(receiver_config.has_remb());
>-  config.remb = receiver_config.remb();
>-
>-  // Get RTX map.
>-  std::map<uint32_t, const rtclog::RtxConfig> rtx_map;
>-  for (int i = 0; i < receiver_config.rtx_map_size(); i++) {
>-    const rtclog::RtxMap& map = receiver_config.rtx_map(i);
>-    RTC_CHECK(map.has_payload_type());
>-    RTC_CHECK(map.has_config());
>-    RTC_CHECK(map.config().has_rtx_ssrc());
>-    RTC_CHECK(map.config().has_rtx_payload_type());
>-    rtx_map.insert(std::make_pair(map.payload_type(), map.config()));
>-  }
>-
>-  // Get header extensions.
>-  GetHeaderExtensions(&config.rtp_extensions,
>-                      receiver_config.header_extensions());
>-  // Get decoders.
>-  config.codecs.clear();
>-  for (int i = 0; i < receiver_config.decoders_size(); i++) {
>-    RTC_CHECK(receiver_config.decoders(i).has_name());
>-    RTC_CHECK(receiver_config.decoders(i).has_payload_type());
>-    int rtx_payload_type = 0;
>-    auto rtx_it = rtx_map.find(receiver_config.decoders(i).payload_type());
>-    if (rtx_it != rtx_map.end()) {
>-      rtx_payload_type = rtx_it->second.rtx_payload_type();
>-      if (config.rtx_ssrc != 0 &&
>-          config.rtx_ssrc != rtx_it->second.rtx_ssrc()) {
>-        RTC_LOG(LS_WARNING)
>-            << "RtcEventLog protobuf contained different SSRCs for "
>-               "different received RTX payload types. Will only use "
>-               "rtx_ssrc = "
>-            << config.rtx_ssrc << ".";
>-      } else {
>-        config.rtx_ssrc = rtx_it->second.rtx_ssrc();
>-      }
>-    }
>-    config.codecs.emplace_back(receiver_config.decoders(i).name(),
>-                               receiver_config.decoders(i).payload_type(),
>-                               rtx_payload_type);
>-  }
>-  return config;
>-}
>-
>-std::vector<rtclog::StreamConfig> ParsedRtcEventLog::GetVideoSendConfig(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  return GetVideoSendConfig(events_[index]);
>-}
>-
>-std::vector<rtclog::StreamConfig> ParsedRtcEventLog::GetVideoSendConfig(
>-    const rtclog::Event& event) const {
>-  std::vector<rtclog::StreamConfig> configs;
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
>-  RTC_CHECK(event.has_video_sender_config());
>-  const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
>-  if (sender_config.rtx_ssrcs_size() > 0 &&
>-      sender_config.ssrcs_size() != sender_config.rtx_ssrcs_size()) {
>-    RTC_LOG(WARNING)
>-        << "VideoSendConfig is configured for RTX but the number of "
>-           "SSRCs doesn't match the number of RTX SSRCs.";
>-  }
>-  configs.resize(sender_config.ssrcs_size());
>-  for (int i = 0; i < sender_config.ssrcs_size(); i++) {
>-    // Get SSRCs.
>-    configs[i].local_ssrc = sender_config.ssrcs(i);
>-    if (sender_config.rtx_ssrcs_size() > 0 &&
>-        i < sender_config.rtx_ssrcs_size()) {
>-      RTC_CHECK(sender_config.has_rtx_payload_type());
>-      configs[i].rtx_ssrc = sender_config.rtx_ssrcs(i);
>-    }
>-    // Get header extensions.
>-    GetHeaderExtensions(&configs[i].rtp_extensions,
>-                        sender_config.header_extensions());
>-
>-    // Get the codec.
>-    RTC_CHECK(sender_config.has_encoder());
>-    RTC_CHECK(sender_config.encoder().has_name());
>-    RTC_CHECK(sender_config.encoder().has_payload_type());
>-    configs[i].codecs.emplace_back(
>-        sender_config.encoder().name(), sender_config.encoder().payload_type(),
>-        sender_config.has_rtx_payload_type() ? sender_config.rtx_payload_type()
>-                                             : 0);
>-  }
>-  return configs;
>-}
>-
>-rtclog::StreamConfig ParsedRtcEventLog::GetAudioReceiveConfig(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  return GetAudioReceiveConfig(events_[index]);
>-}
>-
>-rtclog::StreamConfig ParsedRtcEventLog::GetAudioReceiveConfig(
>-    const rtclog::Event& event) const {
>-  rtclog::StreamConfig config;
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
>-  RTC_CHECK(event.has_audio_receiver_config());
>-  const rtclog::AudioReceiveConfig& receiver_config =
>-      event.audio_receiver_config();
>-  // Get SSRCs.
>-  RTC_CHECK(receiver_config.has_remote_ssrc());
>-  config.remote_ssrc = receiver_config.remote_ssrc();
>-  RTC_CHECK(receiver_config.has_local_ssrc());
>-  config.local_ssrc = receiver_config.local_ssrc();
>-  // Get header extensions.
>-  GetHeaderExtensions(&config.rtp_extensions,
>-                      receiver_config.header_extensions());
>-  return config;
>-}
>-
>-rtclog::StreamConfig ParsedRtcEventLog::GetAudioSendConfig(size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  return GetAudioSendConfig(events_[index]);
>-}
>-
>-rtclog::StreamConfig ParsedRtcEventLog::GetAudioSendConfig(
>-    const rtclog::Event& event) const {
>-  rtclog::StreamConfig config;
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
>-  RTC_CHECK(event.has_audio_sender_config());
>-  const rtclog::AudioSendConfig& sender_config = event.audio_sender_config();
>-  // Get SSRCs.
>-  RTC_CHECK(sender_config.has_ssrc());
>-  config.local_ssrc = sender_config.ssrc();
>-  // Get header extensions.
>-  GetHeaderExtensions(&config.rtp_extensions,
>-                      sender_config.header_extensions());
>-  return config;
>-}
>-
>-void ParsedRtcEventLog::GetAudioPlayout(size_t index, uint32_t* ssrc) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_PLAYOUT_EVENT);
>-  RTC_CHECK(event.has_audio_playout_event());
>-  const rtclog::AudioPlayoutEvent& loss_event = event.audio_playout_event();
>-  RTC_CHECK(loss_event.has_local_ssrc());
>-  if (ssrc != nullptr) {
>-    *ssrc = loss_event.local_ssrc();
>-  }
>-}
>-
>-void ParsedRtcEventLog::GetLossBasedBweUpdate(size_t index,
>-                                              int32_t* bitrate_bps,
>-                                              uint8_t* fraction_loss,
>-                                              int32_t* total_packets) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::LOSS_BASED_BWE_UPDATE);
>-  RTC_CHECK(event.has_loss_based_bwe_update());
>-  const rtclog::LossBasedBweUpdate& loss_event = event.loss_based_bwe_update();
>-  RTC_CHECK(loss_event.has_bitrate_bps());
>-  if (bitrate_bps != nullptr) {
>-    *bitrate_bps = loss_event.bitrate_bps();
>-  }
>-  RTC_CHECK(loss_event.has_fraction_loss());
>-  if (fraction_loss != nullptr) {
>-    *fraction_loss = loss_event.fraction_loss();
>-  }
>-  RTC_CHECK(loss_event.has_total_packets());
>-  if (total_packets != nullptr) {
>-    *total_packets = loss_event.total_packets();
>-  }
>-}
>-
>-ParsedRtcEventLog::BweDelayBasedUpdate
>-ParsedRtcEventLog::GetDelayBasedBweUpdate(size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::DELAY_BASED_BWE_UPDATE);
>-  RTC_CHECK(event.has_delay_based_bwe_update());
>-  const rtclog::DelayBasedBweUpdate& delay_event =
>-      event.delay_based_bwe_update();
>-
>-  BweDelayBasedUpdate res;
>-  res.timestamp = GetTimestamp(index);
>-  RTC_CHECK(delay_event.has_bitrate_bps());
>-  res.bitrate_bps = delay_event.bitrate_bps();
>-  RTC_CHECK(delay_event.has_detector_state());
>-  res.detector_state = GetRuntimeDetectorState(delay_event.detector_state());
>-  return res;
>-}
>-
>-void ParsedRtcEventLog::GetAudioNetworkAdaptation(
>-    size_t index,
>-    AudioEncoderRuntimeConfig* config) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
>-  RTC_CHECK(event.has_audio_network_adaptation());
>-  const rtclog::AudioNetworkAdaptation& ana_event =
>-      event.audio_network_adaptation();
>-  if (ana_event.has_bitrate_bps())
>-    config->bitrate_bps = ana_event.bitrate_bps();
>-  if (ana_event.has_enable_fec())
>-    config->enable_fec = ana_event.enable_fec();
>-  if (ana_event.has_enable_dtx())
>-    config->enable_dtx = ana_event.enable_dtx();
>-  if (ana_event.has_frame_length_ms())
>-    config->frame_length_ms = ana_event.frame_length_ms();
>-  if (ana_event.has_num_channels())
>-    config->num_channels = ana_event.num_channels();
>-  if (ana_event.has_uplink_packet_loss_fraction())
>-    config->uplink_packet_loss_fraction =
>-        ana_event.uplink_packet_loss_fraction();
>-}
>-
>-ParsedRtcEventLog::BweProbeClusterCreatedEvent
>-ParsedRtcEventLog::GetBweProbeClusterCreated(size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT);
>-  RTC_CHECK(event.has_probe_cluster());
>-  const rtclog::BweProbeCluster& pcc_event = event.probe_cluster();
>-  BweProbeClusterCreatedEvent res;
>-  res.timestamp = GetTimestamp(index);
>-  RTC_CHECK(pcc_event.has_id());
>-  res.id = pcc_event.id();
>-  RTC_CHECK(pcc_event.has_bitrate_bps());
>-  res.bitrate_bps = pcc_event.bitrate_bps();
>-  RTC_CHECK(pcc_event.has_min_packets());
>-  res.min_packets = pcc_event.min_packets();
>-  RTC_CHECK(pcc_event.has_min_bytes());
>-  res.min_bytes = pcc_event.min_bytes();
>-  return res;
>-}
>-
>-ParsedRtcEventLog::BweProbeResultEvent ParsedRtcEventLog::GetBweProbeResult(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PROBE_RESULT_EVENT);
>-  RTC_CHECK(event.has_probe_result());
>-  const rtclog::BweProbeResult& pr_event = event.probe_result();
>-  BweProbeResultEvent res;
>-  res.timestamp = GetTimestamp(index);
>-  RTC_CHECK(pr_event.has_id());
>-  res.id = pr_event.id();
>-
>-  RTC_CHECK(pr_event.has_result());
>-  if (pr_event.result() == rtclog::BweProbeResult::SUCCESS) {
>-    RTC_CHECK(pr_event.has_bitrate_bps());
>-    res.bitrate_bps = pr_event.bitrate_bps();
>-  } else if (pr_event.result() ==
>-             rtclog::BweProbeResult::INVALID_SEND_RECEIVE_INTERVAL) {
>-    res.failure_reason = ProbeFailureReason::kInvalidSendReceiveInterval;
>-  } else if (pr_event.result() ==
>-             rtclog::BweProbeResult::INVALID_SEND_RECEIVE_RATIO) {
>-    res.failure_reason = ProbeFailureReason::kInvalidSendReceiveRatio;
>-  } else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
>-    res.failure_reason = ProbeFailureReason::kTimeout;
>-  } else {
>-    RTC_NOTREACHED();
>-  }
>-
>-  return res;
>-}
>-
>-ParsedRtcEventLog::AlrStateEvent ParsedRtcEventLog::GetAlrState(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  RTC_CHECK(event.has_type());
>-  RTC_CHECK_EQ(event.type(), rtclog::Event::ALR_STATE_EVENT);
>-  RTC_CHECK(event.has_alr_state());
>-  const rtclog::AlrState& alr_event = event.alr_state();
>-  AlrStateEvent res;
>-  res.timestamp = GetTimestamp(index);
>-  RTC_CHECK(alr_event.has_in_alr());
>-  res.in_alr = alr_event.in_alr();
>-
>-  return res;
>-}
>-
>-ParsedRtcEventLog::IceCandidatePairConfig
>-ParsedRtcEventLog::GetIceCandidatePairConfig(size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& rtc_event = events_[index];
>-  RTC_CHECK(rtc_event.has_type());
>-  RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG);
>-  IceCandidatePairConfig res;
>-  const rtclog::IceCandidatePairConfig& config =
>-      rtc_event.ice_candidate_pair_config();
>-  res.timestamp = GetTimestamp(index);
>-  RTC_CHECK(config.has_config_type());
>-  res.type = GetRuntimeIceCandidatePairConfigType(config.config_type());
>-  RTC_CHECK(config.has_candidate_pair_id());
>-  res.candidate_pair_id = config.candidate_pair_id();
>-  RTC_CHECK(config.has_local_candidate_type());
>-  res.local_candidate_type =
>-      GetRuntimeIceCandidateType(config.local_candidate_type());
>-  RTC_CHECK(config.has_local_relay_protocol());
>-  res.local_relay_protocol =
>-      GetRuntimeIceCandidatePairProtocol(config.local_relay_protocol());
>-  RTC_CHECK(config.has_local_network_type());
>-  res.local_network_type =
>-      GetRuntimeIceCandidateNetworkType(config.local_network_type());
>-  RTC_CHECK(config.has_local_address_family());
>-  res.local_address_family =
>-      GetRuntimeIceCandidatePairAddressFamily(config.local_address_family());
>-  RTC_CHECK(config.has_remote_candidate_type());
>-  res.remote_candidate_type =
>-      GetRuntimeIceCandidateType(config.remote_candidate_type());
>-  RTC_CHECK(config.has_remote_address_family());
>-  res.remote_address_family =
>-      GetRuntimeIceCandidatePairAddressFamily(config.remote_address_family());
>-  RTC_CHECK(config.has_candidate_pair_protocol());
>-  res.candidate_pair_protocol =
>-      GetRuntimeIceCandidatePairProtocol(config.candidate_pair_protocol());
>-  return res;
>-}
>-
>-ParsedRtcEventLog::IceCandidatePairEvent
>-ParsedRtcEventLog::GetIceCandidatePairEvent(size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& rtc_event = events_[index];
>-  RTC_CHECK(rtc_event.has_type());
>-  RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_EVENT);
>-  IceCandidatePairEvent res;
>-  const rtclog::IceCandidatePairEvent& event =
>-      rtc_event.ice_candidate_pair_event();
>-  res.timestamp = GetTimestamp(index);
>-  RTC_CHECK(event.has_event_type());
>-  res.type = GetRuntimeIceCandidatePairEventType(event.event_type());
>-  RTC_CHECK(event.has_candidate_pair_id());
>-  res.candidate_pair_id = event.candidate_pair_id();
>-  return res;
>-}
>-
>-// Returns the MediaType for registered SSRCs. Search from the end to use last
>-// registered types first.
>-ParsedRtcEventLog::MediaType ParsedRtcEventLog::GetMediaType(
>-    uint32_t ssrc,
>-    PacketDirection direction) const {
>-  for (auto rit = streams_.rbegin(); rit != streams_.rend(); ++rit) {
>-    if (rit->ssrc == ssrc && rit->direction == direction)
>-      return rit->media_type;
>-  }
>-  return MediaType::ANY;
>-}
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
>deleted file mode 100644
>index c93ec6da7c361ababadc5f2634410f028acb8b68..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
>+++ /dev/null
>@@ -1,259 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-#ifndef LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
>-#define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
>-
>-#include <map>
>-#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
>-#include <string>
>-#include <utility>  // pair
>-#include <vector>
>-
>-#include "call/video_receive_stream.h"
>-#include "call/video_send_stream.h"
>-#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
>-#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
>-#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
>-#include "logging/rtc_event_log/rtc_event_log.h"
>-#include "logging/rtc_event_log/rtc_stream_config.h"
>-#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
>-#include "modules/rtp_rtcp/source/byte_io.h"
>-#include "rtc_base/ignore_wundef.h"
>-
>-// Files generated at build-time by the protobuf compiler.
>-RTC_PUSH_IGNORING_WUNDEF()
>-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
>-#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
>-#else
>-#include "logging/rtc_event_log/rtc_event_log.pb.h"
>-#endif
>-RTC_POP_IGNORING_WUNDEF()
>-
>-namespace webrtc {
>-
>-enum class BandwidthUsage;
>-enum class MediaType;
>-
>-struct AudioEncoderRuntimeConfig;
>-
>-class ParsedRtcEventLog {
>-  friend class RtcEventLogTestHelper;
>-
>- public:
>-  ParsedRtcEventLog();
>-  ~ParsedRtcEventLog();
>-
>-  struct BweProbeClusterCreatedEvent {
>-    uint64_t timestamp;
>-    uint32_t id;
>-    uint64_t bitrate_bps;
>-    uint32_t min_packets;
>-    uint32_t min_bytes;
>-  };
>-
>-  struct BweProbeResultEvent {
>-    BweProbeResultEvent();
>-    BweProbeResultEvent(const BweProbeResultEvent&);
>-
>-    uint64_t timestamp;
>-    uint32_t id;
>-    absl::optional<uint64_t> bitrate_bps;
>-    absl::optional<ProbeFailureReason> failure_reason;
>-  };
>-
>-  struct BweDelayBasedUpdate {
>-    uint64_t timestamp;
>-    int32_t bitrate_bps;
>-    BandwidthUsage detector_state;
>-  };
>-
>-  struct AlrStateEvent {
>-    uint64_t timestamp;
>-    bool in_alr;
>-  };
>-
>-  struct IceCandidatePairConfig {
>-    uint64_t timestamp;
>-    IceCandidatePairConfigType type;
>-    uint32_t candidate_pair_id;
>-    IceCandidateType local_candidate_type;
>-    IceCandidatePairProtocol local_relay_protocol;
>-    IceCandidateNetworkType local_network_type;
>-    IceCandidatePairAddressFamily local_address_family;
>-    IceCandidateType remote_candidate_type;
>-    IceCandidatePairAddressFamily remote_address_family;
>-    IceCandidatePairProtocol candidate_pair_protocol;
>-  };
>-
>-  struct IceCandidatePairEvent {
>-    uint64_t timestamp;
>-    IceCandidatePairEventType type;
>-    uint32_t candidate_pair_id;
>-  };
>-
>-  enum EventType {
>-    UNKNOWN_EVENT = 0,
>-    LOG_START = 1,
>-    LOG_END = 2,
>-    RTP_EVENT = 3,
>-    RTCP_EVENT = 4,
>-    AUDIO_PLAYOUT_EVENT = 5,
>-    LOSS_BASED_BWE_UPDATE = 6,
>-    DELAY_BASED_BWE_UPDATE = 7,
>-    VIDEO_RECEIVER_CONFIG_EVENT = 8,
>-    VIDEO_SENDER_CONFIG_EVENT = 9,
>-    AUDIO_RECEIVER_CONFIG_EVENT = 10,
>-    AUDIO_SENDER_CONFIG_EVENT = 11,
>-    AUDIO_NETWORK_ADAPTATION_EVENT = 16,
>-    BWE_PROBE_CLUSTER_CREATED_EVENT = 17,
>-    BWE_PROBE_RESULT_EVENT = 18,
>-    ALR_STATE_EVENT = 19,
>-    ICE_CANDIDATE_PAIR_CONFIG = 20,
>-    ICE_CANDIDATE_PAIR_EVENT = 21,
>-  };
>-
>-  enum class MediaType { ANY, AUDIO, VIDEO, DATA };
>-
>-  // Reads an RtcEventLog file and returns true if parsing was successful.
>-  bool ParseFile(const std::string& file_name);
>-
>-  // Reads an RtcEventLog from a string and returns true if successful.
>-  bool ParseString(const std::string& s);
>-
>-  // Reads an RtcEventLog from an istream and returns true if successful.
>-  bool ParseStream(std::istream& stream);
>-
>-  // Returns the number of events in an EventStream.
>-  size_t GetNumberOfEvents() const;
>-
>-  // Reads the arrival timestamp (in microseconds) from a rtclog::Event.
>-  int64_t GetTimestamp(size_t index) const;
>-
>-  // Reads the event type of the rtclog::Event at |index|.
>-  EventType GetEventType(size_t index) const;
>-
>-  // Reads the header, direction, header length and packet length from the RTP
>-  // event at |index|, and stores the values in the corresponding output
>-  // parameters. Each output parameter can be set to nullptr if that value
>-  // isn't needed.
>-  // NB: The header must have space for at least IP_PACKET_SIZE bytes.
>-  // Returns: a pointer to a header extensions map acquired from parsing
>-  // corresponding Audio/Video Sender/Receiver config events.
>-  // Warning: if the same SSRC is reused by both video and audio streams during
>-  // call, extensions maps may be incorrect (the last one would be returned).
>-  webrtc::RtpHeaderExtensionMap* GetRtpHeader(size_t index,
>-                                              PacketDirection* incoming,
>-                                              uint8_t* header,
>-                                              size_t* header_length,
>-                                              size_t* total_length,
>-                                              int* probe_cluster_id) const;
>-
>-  // Reads packet, direction and packet length from the RTCP event at |index|,
>-  // and stores the values in the corresponding output parameters.
>-  // Each output parameter can be set to nullptr if that value isn't needed.
>-  // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
>-  void GetRtcpPacket(size_t index,
>-                     PacketDirection* incoming,
>-                     uint8_t* packet,
>-                     size_t* length) const;
>-
>-  // Reads a video receive config event to a StreamConfig struct.
>-  // Only the fields that are stored in the protobuf will be written.
>-  rtclog::StreamConfig GetVideoReceiveConfig(size_t index) const;
>-
>-  // Reads a video send config event to a StreamConfig struct. If the proto
>-  // contains multiple SSRCs and RTX SSRCs (this used to be the case for
>-  // simulcast streams) then we return one StreamConfig per SSRC,RTX_SSRC pair.
>-  // Only the fields that are stored in the protobuf will be written.
>-  std::vector<rtclog::StreamConfig> GetVideoSendConfig(size_t index) const;
>-
>-  // Reads a audio receive config event to a StreamConfig struct.
>-  // Only the fields that are stored in the protobuf will be written.
>-  rtclog::StreamConfig GetAudioReceiveConfig(size_t index) const;
>-
>-  // Reads a config event to a StreamConfig struct.
>-  // Only the fields that are stored in the protobuf will be written.
>-  rtclog::StreamConfig GetAudioSendConfig(size_t index) const;
>-
>-  // Reads the SSRC from the audio playout event at |index|. The SSRC is stored
>-  // in the output parameter ssrc. The output parameter can be set to nullptr
>-  // and in that case the function only asserts that the event is well formed.
>-  void GetAudioPlayout(size_t index, uint32_t* ssrc) const;
>-
>-  // Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
>-  // expected packets from the loss based BWE event at |index| and stores the
>-  // values in
>-  // the corresponding output parameters. Each output parameter can be set to
>-  // nullptr if that
>-  // value isn't needed.
>-  void GetLossBasedBweUpdate(size_t index,
>-                             int32_t* bitrate_bps,
>-                             uint8_t* fraction_loss,
>-                             int32_t* total_packets) const;
>-
>-  // Reads bitrate and detector_state from the delay based BWE event at |index|
>-  // and stores the values in the corresponding output parameters. Each output
>-  // parameter can be set to nullptr if that
>-  // value isn't needed.
>-  BweDelayBasedUpdate GetDelayBasedBweUpdate(size_t index) const;
>-
>-  // Reads a audio network adaptation event to a (non-NULL)
>-  // AudioEncoderRuntimeConfig struct. Only the fields that are
>-  // stored in the protobuf will be written.
>-  void GetAudioNetworkAdaptation(size_t index,
>-                                 AudioEncoderRuntimeConfig* config) const;
>-
>-  BweProbeClusterCreatedEvent GetBweProbeClusterCreated(size_t index) const;
>-
>-  BweProbeResultEvent GetBweProbeResult(size_t index) const;
>-
>-  MediaType GetMediaType(uint32_t ssrc, PacketDirection direction) const;
>-
>-  AlrStateEvent GetAlrState(size_t index) const;
>-
>-  IceCandidatePairConfig GetIceCandidatePairConfig(size_t index) const;
>-  IceCandidatePairEvent GetIceCandidatePairEvent(size_t index) const;
>-
>- private:
>-  rtclog::StreamConfig GetVideoReceiveConfig(const rtclog::Event& event) const;
>-  std::vector<rtclog::StreamConfig> GetVideoSendConfig(
>-      const rtclog::Event& event) const;
>-  rtclog::StreamConfig GetAudioReceiveConfig(const rtclog::Event& event) const;
>-  rtclog::StreamConfig GetAudioSendConfig(const rtclog::Event& event) const;
>-
>-  std::vector<rtclog::Event> events_;
>-
>-  struct Stream {
>-    Stream(uint32_t ssrc,
>-           MediaType media_type,
>-           webrtc::PacketDirection direction,
>-           webrtc::RtpHeaderExtensionMap map)
>-        : ssrc(ssrc),
>-          media_type(media_type),
>-          direction(direction),
>-          rtp_extensions_map(map) {}
>-    uint32_t ssrc;
>-    MediaType media_type;
>-    webrtc::PacketDirection direction;
>-    webrtc::RtpHeaderExtensionMap rtp_extensions_map;
>-  };
>-
>-  // All configured streams found in the event log.
>-  std::vector<Stream> streams_;
>-
>-  // To find configured extensions map for given stream, what are needed to
>-  // parse a header.
>-  typedef std::pair<uint32_t, webrtc::PacketDirection> StreamId;
>-  std::map<StreamId, webrtc::RtpHeaderExtensionMap*> rtp_extensions_maps_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.cc
>index 5fa69ed637f65eac09129b46f82b8595f4138457..7278ea18b1d33c45c3d8e77e00e12bfcee98e399 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.cc
>@@ -24,21 +24,30 @@
> #include "absl/types/optional.h"
> #include "api/rtp_headers.h"
> #include "api/rtpparameters.h"
>+#include "logging/rtc_event_log/encoder/blob_encoding.h"
>+#include "logging/rtc_event_log/encoder/delta_encoding.h"
>+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_common.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
>-#include "modules/congestion_controller/transport_feedback_adapter.h"
>+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>+#include "modules/rtp_rtcp/include/rtp_cvo.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/protobuf_utils.h"
> 
>+using webrtc_event_logging::ToSigned;
>+using webrtc_event_logging::ToUnsigned;
>+
> namespace webrtc {
> 
> namespace {
>+// Conversion functions for legacy wire format.
> RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
>   switch (rtcp_mode) {
>     case rtclog::VideoReceiveConfig::RTCP_COMPOUND:
>@@ -50,52 +59,6 @@ RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
>   return RtcpMode::kOff;
> }
> 
>-ParsedRtcEventLogNew::EventType GetRuntimeEventType(
>-    rtclog::Event::EventType event_type) {
>-  switch (event_type) {
>-    case rtclog::Event::UNKNOWN_EVENT:
>-      return ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT;
>-    case rtclog::Event::LOG_START:
>-      return ParsedRtcEventLogNew::EventType::LOG_START;
>-    case rtclog::Event::LOG_END:
>-      return ParsedRtcEventLogNew::EventType::LOG_END;
>-    case rtclog::Event::RTP_EVENT:
>-      return ParsedRtcEventLogNew::EventType::RTP_EVENT;
>-    case rtclog::Event::RTCP_EVENT:
>-      return ParsedRtcEventLogNew::EventType::RTCP_EVENT;
>-    case rtclog::Event::AUDIO_PLAYOUT_EVENT:
>-      return ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT;
>-    case rtclog::Event::LOSS_BASED_BWE_UPDATE:
>-      return ParsedRtcEventLogNew::EventType::LOSS_BASED_BWE_UPDATE;
>-    case rtclog::Event::DELAY_BASED_BWE_UPDATE:
>-      return ParsedRtcEventLogNew::EventType::DELAY_BASED_BWE_UPDATE;
>-    case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
>-      return ParsedRtcEventLogNew::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
>-    case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
>-      return ParsedRtcEventLogNew::EventType::VIDEO_SENDER_CONFIG_EVENT;
>-    case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
>-      return ParsedRtcEventLogNew::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
>-    case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
>-      return ParsedRtcEventLogNew::EventType::AUDIO_SENDER_CONFIG_EVENT;
>-    case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
>-      return ParsedRtcEventLogNew::EventType::AUDIO_NETWORK_ADAPTATION_EVENT;
>-    case rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT:
>-      return ParsedRtcEventLogNew::EventType::BWE_PROBE_CLUSTER_CREATED_EVENT;
>-    case rtclog::Event::BWE_PROBE_RESULT_EVENT:
>-      // Probe successes and failures are currently stored in the same proto
>-      // message, we are moving towards separate messages. Probe results
>-      // therefore need special treatment in the parser.
>-      return ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT;
>-    case rtclog::Event::ALR_STATE_EVENT:
>-      return ParsedRtcEventLogNew::EventType::ALR_STATE_EVENT;
>-    case rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG:
>-      return ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_CONFIG;
>-    case rtclog::Event::ICE_CANDIDATE_PAIR_EVENT:
>-      return ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_EVENT;
>-  }
>-  return ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT;
>-}
>-
> BandwidthUsage GetRuntimeDetectorState(
>     rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
>   switch (detector_state) {
>@@ -212,6 +175,175 @@ IceCandidatePairEventType GetRuntimeIceCandidatePairEventType(
>   return IceCandidatePairEventType::kCheckSent;
> }
> 
>+// Conversion functions for version 2 of the wire format.
>+BandwidthUsage GetRuntimeDetectorState(
>+    rtclog2::DelayBasedBweUpdates::DetectorState detector_state) {
>+  switch (detector_state) {
>+    case rtclog2::DelayBasedBweUpdates::BWE_NORMAL:
>+      return BandwidthUsage::kBwNormal;
>+    case rtclog2::DelayBasedBweUpdates::BWE_UNDERUSING:
>+      return BandwidthUsage::kBwUnderusing;
>+    case rtclog2::DelayBasedBweUpdates::BWE_OVERUSING:
>+      return BandwidthUsage::kBwOverusing;
>+    case rtclog2::DelayBasedBweUpdates::BWE_UNKNOWN_STATE:
>+      break;
>+  }
>+  RTC_NOTREACHED();
>+  return BandwidthUsage::kBwNormal;
>+}
>+
>+ProbeFailureReason GetRuntimeProbeFailureReason(
>+    rtclog2::BweProbeResultFailure::FailureReason failure) {
>+  switch (failure) {
>+    case rtclog2::BweProbeResultFailure::INVALID_SEND_RECEIVE_INTERVAL:
>+      return ProbeFailureReason::kInvalidSendReceiveInterval;
>+    case rtclog2::BweProbeResultFailure::INVALID_SEND_RECEIVE_RATIO:
>+      return ProbeFailureReason::kInvalidSendReceiveRatio;
>+    case rtclog2::BweProbeResultFailure::TIMEOUT:
>+      return ProbeFailureReason::kTimeout;
>+    case rtclog2::BweProbeResultFailure::UNKNOWN:
>+      break;
>+  }
>+  RTC_NOTREACHED();
>+  return ProbeFailureReason::kTimeout;
>+}
>+
>+IceCandidatePairConfigType GetRuntimeIceCandidatePairConfigType(
>+    rtclog2::IceCandidatePairConfig::IceCandidatePairConfigType type) {
>+  switch (type) {
>+    case rtclog2::IceCandidatePairConfig::ADDED:
>+      return IceCandidatePairConfigType::kAdded;
>+    case rtclog2::IceCandidatePairConfig::UPDATED:
>+      return IceCandidatePairConfigType::kUpdated;
>+    case rtclog2::IceCandidatePairConfig::DESTROYED:
>+      return IceCandidatePairConfigType::kDestroyed;
>+    case rtclog2::IceCandidatePairConfig::SELECTED:
>+      return IceCandidatePairConfigType::kSelected;
>+    case rtclog2::IceCandidatePairConfig::UNKNOWN_CONFIG_TYPE:
>+      break;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairConfigType::kAdded;
>+}
>+
>+IceCandidateType GetRuntimeIceCandidateType(
>+    rtclog2::IceCandidatePairConfig::IceCandidateType type) {
>+  switch (type) {
>+    case rtclog2::IceCandidatePairConfig::LOCAL:
>+      return IceCandidateType::kLocal;
>+    case rtclog2::IceCandidatePairConfig::STUN:
>+      return IceCandidateType::kStun;
>+    case rtclog2::IceCandidatePairConfig::PRFLX:
>+      return IceCandidateType::kPrflx;
>+    case rtclog2::IceCandidatePairConfig::RELAY:
>+      return IceCandidateType::kRelay;
>+    case rtclog2::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE:
>+      return IceCandidateType::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidateType::kUnknown;
>+}
>+
>+IceCandidatePairProtocol GetRuntimeIceCandidatePairProtocol(
>+    rtclog2::IceCandidatePairConfig::Protocol protocol) {
>+  switch (protocol) {
>+    case rtclog2::IceCandidatePairConfig::UDP:
>+      return IceCandidatePairProtocol::kUdp;
>+    case rtclog2::IceCandidatePairConfig::TCP:
>+      return IceCandidatePairProtocol::kTcp;
>+    case rtclog2::IceCandidatePairConfig::SSLTCP:
>+      return IceCandidatePairProtocol::kSsltcp;
>+    case rtclog2::IceCandidatePairConfig::TLS:
>+      return IceCandidatePairProtocol::kTls;
>+    case rtclog2::IceCandidatePairConfig::UNKNOWN_PROTOCOL:
>+      return IceCandidatePairProtocol::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairProtocol::kUnknown;
>+}
>+
>+IceCandidatePairAddressFamily GetRuntimeIceCandidatePairAddressFamily(
>+    rtclog2::IceCandidatePairConfig::AddressFamily address_family) {
>+  switch (address_family) {
>+    case rtclog2::IceCandidatePairConfig::IPV4:
>+      return IceCandidatePairAddressFamily::kIpv4;
>+    case rtclog2::IceCandidatePairConfig::IPV6:
>+      return IceCandidatePairAddressFamily::kIpv6;
>+    case rtclog2::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY:
>+      return IceCandidatePairAddressFamily::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairAddressFamily::kUnknown;
>+}
>+
>+IceCandidateNetworkType GetRuntimeIceCandidateNetworkType(
>+    rtclog2::IceCandidatePairConfig::NetworkType network_type) {
>+  switch (network_type) {
>+    case rtclog2::IceCandidatePairConfig::ETHERNET:
>+      return IceCandidateNetworkType::kEthernet;
>+    case rtclog2::IceCandidatePairConfig::LOOPBACK:
>+      return IceCandidateNetworkType::kLoopback;
>+    case rtclog2::IceCandidatePairConfig::WIFI:
>+      return IceCandidateNetworkType::kWifi;
>+    case rtclog2::IceCandidatePairConfig::VPN:
>+      return IceCandidateNetworkType::kVpn;
>+    case rtclog2::IceCandidatePairConfig::CELLULAR:
>+      return IceCandidateNetworkType::kCellular;
>+    case rtclog2::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE:
>+      return IceCandidateNetworkType::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidateNetworkType::kUnknown;
>+}
>+
>+IceCandidatePairEventType GetRuntimeIceCandidatePairEventType(
>+    rtclog2::IceCandidatePairEvent::IceCandidatePairEventType type) {
>+  switch (type) {
>+    case rtclog2::IceCandidatePairEvent::CHECK_SENT:
>+      return IceCandidatePairEventType::kCheckSent;
>+    case rtclog2::IceCandidatePairEvent::CHECK_RECEIVED:
>+      return IceCandidatePairEventType::kCheckReceived;
>+    case rtclog2::IceCandidatePairEvent::CHECK_RESPONSE_SENT:
>+      return IceCandidatePairEventType::kCheckResponseSent;
>+    case rtclog2::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED:
>+      return IceCandidatePairEventType::kCheckResponseReceived;
>+    case rtclog2::IceCandidatePairEvent::UNKNOWN_CHECK_TYPE:
>+      break;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairEventType::kCheckSent;
>+}
>+
>+std::vector<RtpExtension> GetRuntimeRtpHeaderExtensionConfig(
>+    const rtclog2::RtpHeaderExtensionConfig& proto_header_extensions) {
>+  std::vector<RtpExtension> rtp_extensions;
>+  if (proto_header_extensions.has_transmission_time_offset_id()) {
>+    rtp_extensions.emplace_back(
>+        RtpExtension::kTimestampOffsetUri,
>+        proto_header_extensions.transmission_time_offset_id());
>+  }
>+  if (proto_header_extensions.has_absolute_send_time_id()) {
>+    rtp_extensions.emplace_back(
>+        RtpExtension::kAbsSendTimeUri,
>+        proto_header_extensions.absolute_send_time_id());
>+  }
>+  if (proto_header_extensions.has_transport_sequence_number_id()) {
>+    rtp_extensions.emplace_back(
>+        RtpExtension::kTransportSequenceNumberUri,
>+        proto_header_extensions.transport_sequence_number_id());
>+  }
>+  if (proto_header_extensions.has_audio_level_id()) {
>+    rtp_extensions.emplace_back(RtpExtension::kAudioLevelUri,
>+                                proto_header_extensions.audio_level_id());
>+  }
>+  if (proto_header_extensions.has_video_rotation_id()) {
>+    rtp_extensions.emplace_back(RtpExtension::kVideoRotationUri,
>+                                proto_header_extensions.video_rotation_id());
>+  }
>+  return rtp_extensions;
>+}
>+// End of conversion functions.
>+
> // Reads a VarInt from |stream| and returns it. Also writes the read bytes to
> // |buffer| starting |bytes_written| bytes into the buffer. |bytes_written| is
> // incremented for each written byte.
>@@ -271,15 +403,322 @@ void SortPacketFeedbackVectorWithLoss(std::vector<PacketFeedback>* vec) {
>   std::sort(vec->begin(), vec->end(), LossHandlingPacketFeedbackComparator());
> }
> 
>+template <typename ProtoType, typename LoggedType>
>+void StoreRtpPackets(
>+    const ProtoType& proto,
>+    std::map<uint32_t, std::vector<LoggedType>>* rtp_packets_map) {
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  RTC_CHECK(proto.has_marker());
>+  RTC_CHECK(proto.has_payload_type());
>+  RTC_CHECK(proto.has_sequence_number());
>+  RTC_CHECK(proto.has_rtp_timestamp());
>+  RTC_CHECK(proto.has_ssrc());
>+  RTC_CHECK(proto.has_payload_size());
>+  RTC_CHECK(proto.has_header_size());
>+  RTC_CHECK(proto.has_padding_size());
>+
>+  // Base event
>+  {
>+    RTPHeader header;
>+    header.markerBit = rtc::checked_cast<bool>(proto.marker());
>+    header.payloadType = rtc::checked_cast<uint8_t>(proto.payload_type());
>+    header.sequenceNumber =
>+        rtc::checked_cast<uint16_t>(proto.sequence_number());
>+    header.timestamp = rtc::checked_cast<uint32_t>(proto.rtp_timestamp());
>+    header.ssrc = rtc::checked_cast<uint32_t>(proto.ssrc());
>+    header.numCSRCs = 0;  // TODO(terelius): Implement CSRC.
>+    header.paddingLength = rtc::checked_cast<size_t>(proto.padding_size());
>+    header.headerLength = rtc::checked_cast<size_t>(proto.header_size());
>+    // TODO(terelius): Should we implement payload_type_frequency?
>+    if (proto.has_transport_sequence_number()) {
>+      header.extension.hasTransportSequenceNumber = true;
>+      header.extension.transportSequenceNumber =
>+          rtc::checked_cast<uint16_t>(proto.transport_sequence_number());
>+    }
>+    if (proto.has_transmission_time_offset()) {
>+      header.extension.hasTransmissionTimeOffset = true;
>+      header.extension.transmissionTimeOffset =
>+          rtc::checked_cast<int32_t>(proto.transmission_time_offset());
>+    }
>+    if (proto.has_absolute_send_time()) {
>+      header.extension.hasAbsoluteSendTime = true;
>+      header.extension.absoluteSendTime =
>+          rtc::checked_cast<uint32_t>(proto.absolute_send_time());
>+    }
>+    if (proto.has_video_rotation()) {
>+      header.extension.hasVideoRotation = true;
>+      header.extension.videoRotation = ConvertCVOByteToVideoRotation(
>+          rtc::checked_cast<uint8_t>(proto.video_rotation()));
>+    }
>+    if (proto.has_audio_level()) {
>+      RTC_CHECK(proto.has_voice_activity());
>+      header.extension.hasAudioLevel = true;
>+      header.extension.voiceActivity =
>+          rtc::checked_cast<bool>(proto.voice_activity());
>+      const uint8_t audio_level =
>+          rtc::checked_cast<uint8_t>(proto.audio_level());
>+      RTC_CHECK_LE(audio_level, 0x7Fu);
>+      header.extension.audioLevel = audio_level;
>+    } else {
>+      RTC_CHECK(!proto.has_voice_activity());
>+    }
>+    (*rtp_packets_map)[header.ssrc].emplace_back(
>+        proto.timestamp_ms() * 1000, header, proto.header_size(),
>+        proto.payload_size() + header.headerLength + header.paddingLength);
>+  }
>+
>+  const size_t number_of_deltas =
>+      proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
>+  if (number_of_deltas == 0) {
>+    return;
>+  }
>+
>+  // timestamp_ms (event)
>+  std::vector<absl::optional<uint64_t>> timestamp_ms_values =
>+      DecodeDeltas(proto.timestamp_ms_deltas(),
>+                   ToUnsigned(proto.timestamp_ms()), number_of_deltas);
>+  RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
>+
>+  // marker (RTP base)
>+  std::vector<absl::optional<uint64_t>> marker_values =
>+      DecodeDeltas(proto.marker_deltas(), proto.marker(), number_of_deltas);
>+  RTC_CHECK_EQ(marker_values.size(), number_of_deltas);
>+
>+  // payload_type (RTP base)
>+  std::vector<absl::optional<uint64_t>> payload_type_values = DecodeDeltas(
>+      proto.payload_type_deltas(), proto.payload_type(), number_of_deltas);
>+  RTC_CHECK_EQ(payload_type_values.size(), number_of_deltas);
>+
>+  // sequence_number (RTP base)
>+  std::vector<absl::optional<uint64_t>> sequence_number_values =
>+      DecodeDeltas(proto.sequence_number_deltas(), proto.sequence_number(),
>+                   number_of_deltas);
>+  RTC_CHECK_EQ(sequence_number_values.size(), number_of_deltas);
>+
>+  // rtp_timestamp (RTP base)
>+  std::vector<absl::optional<uint64_t>> rtp_timestamp_values = DecodeDeltas(
>+      proto.rtp_timestamp_deltas(), proto.rtp_timestamp(), number_of_deltas);
>+  RTC_CHECK_EQ(rtp_timestamp_values.size(), number_of_deltas);
>+
>+  // ssrc (RTP base)
>+  std::vector<absl::optional<uint64_t>> ssrc_values =
>+      DecodeDeltas(proto.ssrc_deltas(), proto.ssrc(), number_of_deltas);
>+  RTC_CHECK_EQ(ssrc_values.size(), number_of_deltas);
>+
>+  // payload_size (RTP base)
>+  std::vector<absl::optional<uint64_t>> payload_size_values = DecodeDeltas(
>+      proto.payload_size_deltas(), proto.payload_size(), number_of_deltas);
>+  RTC_CHECK_EQ(payload_size_values.size(), number_of_deltas);
>+
>+  // header_size (RTP base)
>+  std::vector<absl::optional<uint64_t>> header_size_values = DecodeDeltas(
>+      proto.header_size_deltas(), proto.header_size(), number_of_deltas);
>+  RTC_CHECK_EQ(header_size_values.size(), number_of_deltas);
>+
>+  // padding_size (RTP base)
>+  std::vector<absl::optional<uint64_t>> padding_size_values = DecodeDeltas(
>+      proto.padding_size_deltas(), proto.padding_size(), number_of_deltas);
>+  RTC_CHECK_EQ(padding_size_values.size(), number_of_deltas);
>+
>+  // transport_sequence_number (RTP extension)
>+  std::vector<absl::optional<uint64_t>> transport_sequence_number_values;
>+  {
>+    const absl::optional<uint64_t> base_transport_sequence_number =
>+        proto.has_transport_sequence_number()
>+            ? proto.transport_sequence_number()
>+            : absl::optional<uint64_t>();
>+    transport_sequence_number_values =
>+        DecodeDeltas(proto.transport_sequence_number_deltas(),
>+                     base_transport_sequence_number, number_of_deltas);
>+    RTC_CHECK_EQ(transport_sequence_number_values.size(), number_of_deltas);
>+  }
>+
>+  // transmission_time_offset (RTP extension)
>+  std::vector<absl::optional<uint64_t>> transmission_time_offset_values;
>+  {
>+    const absl::optional<uint64_t> unsigned_base_transmission_time_offset =
>+        proto.has_transmission_time_offset()
>+            ? ToUnsigned(proto.transmission_time_offset())
>+            : absl::optional<uint64_t>();
>+    transmission_time_offset_values =
>+        DecodeDeltas(proto.transmission_time_offset_deltas(),
>+                     unsigned_base_transmission_time_offset, number_of_deltas);
>+    RTC_CHECK_EQ(transmission_time_offset_values.size(), number_of_deltas);
>+  }
>+
>+  // absolute_send_time (RTP extension)
>+  std::vector<absl::optional<uint64_t>> absolute_send_time_values;
>+  {
>+    const absl::optional<uint64_t> base_absolute_send_time =
>+        proto.has_absolute_send_time() ? proto.absolute_send_time()
>+                                       : absl::optional<uint64_t>();
>+    absolute_send_time_values =
>+        DecodeDeltas(proto.absolute_send_time_deltas(), base_absolute_send_time,
>+                     number_of_deltas);
>+    RTC_CHECK_EQ(absolute_send_time_values.size(), number_of_deltas);
>+  }
>+
>+  // video_rotation (RTP extension)
>+  std::vector<absl::optional<uint64_t>> video_rotation_values;
>+  {
>+    const absl::optional<uint64_t> base_video_rotation =
>+        proto.has_video_rotation() ? proto.video_rotation()
>+                                   : absl::optional<uint64_t>();
>+    video_rotation_values = DecodeDeltas(proto.video_rotation_deltas(),
>+                                         base_video_rotation, number_of_deltas);
>+    RTC_CHECK_EQ(video_rotation_values.size(), number_of_deltas);
>+  }
>+
>+  // audio_level (RTP extension)
>+  std::vector<absl::optional<uint64_t>> audio_level_values;
>+  {
>+    const absl::optional<uint64_t> base_audio_level =
>+        proto.has_audio_level() ? proto.audio_level()
>+                                : absl::optional<uint64_t>();
>+    audio_level_values = DecodeDeltas(proto.audio_level_deltas(),
>+                                      base_audio_level, number_of_deltas);
>+    RTC_CHECK_EQ(audio_level_values.size(), number_of_deltas);
>+  }
>+
>+  // voice_activity (RTP extension)
>+  std::vector<absl::optional<uint64_t>> voice_activity_values;
>+  {
>+    const absl::optional<uint64_t> base_voice_activity =
>+        proto.has_voice_activity() ? proto.voice_activity()
>+                                   : absl::optional<uint64_t>();
>+    voice_activity_values = DecodeDeltas(proto.voice_activity_deltas(),
>+                                         base_voice_activity, number_of_deltas);
>+    RTC_CHECK_EQ(voice_activity_values.size(), number_of_deltas);
>+  }
>+
>+  // Delta decoding
>+  for (size_t i = 0; i < number_of_deltas; ++i) {
>+    RTC_CHECK(timestamp_ms_values[i].has_value());
>+    RTC_CHECK(marker_values[i].has_value());
>+    RTC_CHECK(payload_type_values[i].has_value());
>+    RTC_CHECK(sequence_number_values[i].has_value());
>+    RTC_CHECK(rtp_timestamp_values[i].has_value());
>+    RTC_CHECK(ssrc_values[i].has_value());
>+    RTC_CHECK(payload_size_values[i].has_value());
>+    RTC_CHECK(header_size_values[i].has_value());
>+    RTC_CHECK(padding_size_values[i].has_value());
>+
>+    int64_t timestamp_ms;
>+    RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
>+
>+    RTPHeader header;
>+    header.markerBit = rtc::checked_cast<bool>(*marker_values[i]);
>+    header.payloadType = rtc::checked_cast<uint8_t>(*payload_type_values[i]);
>+    header.sequenceNumber =
>+        rtc::checked_cast<uint16_t>(*sequence_number_values[i]);
>+    header.timestamp = rtc::checked_cast<uint32_t>(*rtp_timestamp_values[i]);
>+    header.ssrc = rtc::checked_cast<uint32_t>(*ssrc_values[i]);
>+    header.numCSRCs = 0;  // TODO(terelius): Implement CSRC.
>+    header.paddingLength = rtc::checked_cast<size_t>(*padding_size_values[i]);
>+    header.headerLength = rtc::checked_cast<size_t>(*header_size_values[i]);
>+    // TODO(terelius): Should we implement payload_type_frequency?
>+    if (transport_sequence_number_values.size() > i &&
>+        transport_sequence_number_values[i].has_value()) {
>+      header.extension.hasTransportSequenceNumber = true;
>+      header.extension.transportSequenceNumber = rtc::checked_cast<uint16_t>(
>+          transport_sequence_number_values[i].value());
>+    }
>+    if (transmission_time_offset_values.size() > i &&
>+        transmission_time_offset_values[i].has_value()) {
>+      header.extension.hasTransmissionTimeOffset = true;
>+      int32_t transmission_time_offset;
>+      RTC_CHECK(ToSigned(transmission_time_offset_values[i].value(),
>+                         &transmission_time_offset));
>+      header.extension.transmissionTimeOffset = transmission_time_offset;
>+    }
>+    if (absolute_send_time_values.size() > i &&
>+        absolute_send_time_values[i].has_value()) {
>+      header.extension.hasAbsoluteSendTime = true;
>+      header.extension.absoluteSendTime =
>+          rtc::checked_cast<uint32_t>(absolute_send_time_values[i].value());
>+    }
>+    if (video_rotation_values.size() > i &&
>+        video_rotation_values[i].has_value()) {
>+      header.extension.hasVideoRotation = true;
>+      header.extension.videoRotation = ConvertCVOByteToVideoRotation(
>+          rtc::checked_cast<uint8_t>(video_rotation_values[i].value()));
>+    }
>+    if (audio_level_values.size() > i && audio_level_values[i].has_value()) {
>+      RTC_CHECK(voice_activity_values.size() > i &&
>+                voice_activity_values[i].has_value());
>+      header.extension.hasAudioLevel = true;
>+      header.extension.voiceActivity =
>+          rtc::checked_cast<bool>(voice_activity_values[i].value());
>+      const uint8_t audio_level =
>+          rtc::checked_cast<uint8_t>(audio_level_values[i].value());
>+      RTC_CHECK_LE(audio_level, 0x7Fu);
>+      header.extension.audioLevel = audio_level;
>+    } else {
>+      RTC_CHECK(voice_activity_values.size() <= i ||
>+                !voice_activity_values[i].has_value());
>+    }
>+    (*rtp_packets_map)[header.ssrc].emplace_back(
>+        1000 * timestamp_ms, header, header.headerLength,
>+        payload_size_values[i].value() + header.headerLength +
>+            header.paddingLength);
>+  }
>+}
>+
>+template <typename ProtoType, typename LoggedType>
>+void StoreRtcpPackets(const ProtoType& proto,
>+                      std::vector<LoggedType>* rtcp_packets) {
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  RTC_CHECK(proto.has_raw_packet());
>+
>+  // Base event
>+  rtcp_packets->emplace_back(proto.timestamp_ms() * 1000, proto.raw_packet());
>+
>+  const size_t number_of_deltas =
>+      proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
>+  if (number_of_deltas == 0) {
>+    return;
>+  }
>+
>+  // timestamp_ms
>+  std::vector<absl::optional<uint64_t>> timestamp_ms_values =
>+      DecodeDeltas(proto.timestamp_ms_deltas(),
>+                   ToUnsigned(proto.timestamp_ms()), number_of_deltas);
>+  RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
>+
>+  // raw_packet
>+  RTC_CHECK(proto.has_raw_packet_blobs());
>+  std::vector<absl::string_view> raw_packet_values =
>+      DecodeBlobs(proto.raw_packet_blobs(), number_of_deltas);
>+  RTC_CHECK_EQ(raw_packet_values.size(), number_of_deltas);
>+
>+  // Delta decoding
>+  for (size_t i = 0; i < number_of_deltas; ++i) {
>+    RTC_CHECK(timestamp_ms_values[i].has_value());
>+    int64_t timestamp_ms;
>+    RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
>+
>+    rtcp_packets->emplace_back(
>+        1000 * timestamp_ms,
>+        reinterpret_cast<const uint8_t*>(raw_packet_values[i].data()),
>+        raw_packet_values[i].size());
>+  }
>+}
>+
> }  // namespace
> 
> LoggedRtcpPacket::LoggedRtcpPacket(uint64_t timestamp_us,
>                                    const uint8_t* packet,
>                                    size_t total_length)
>     : timestamp_us(timestamp_us), raw_data(packet, packet + total_length) {}
>+LoggedRtcpPacket::LoggedRtcpPacket(uint64_t timestamp_us,
>+                                   const std::string& packet)
>+    : timestamp_us(timestamp_us), raw_data(packet.size()) {
>+  memcpy(raw_data.data(), packet.data(), packet.size());
>+}
> LoggedRtcpPacket::LoggedRtcpPacket(const LoggedRtcpPacket& rhs) = default;
> LoggedRtcpPacket::~LoggedRtcpPacket() = default;
> 
>+LoggedVideoSendConfig::LoggedVideoSendConfig() = default;
> LoggedVideoSendConfig::LoggedVideoSendConfig(
>     int64_t timestamp_us,
>     const std::vector<rtclog::StreamConfig>& configs)
>@@ -361,7 +800,6 @@ ParsedRtcEventLogNew::ParsedRtcEventLogNew(
> }
> 
> void ParsedRtcEventLogNew::Clear() {
>-  events_.clear();
>   default_extension_map_ = GetDefaultHeaderExtensionMap();
> 
>   incoming_rtx_ssrcs_.clear();
>@@ -478,9 +916,8 @@ bool ParsedRtcEventLogNew::ParseStream(
> 
> bool ParsedRtcEventLogNew::ParseStreamInternal(
>     std::istream& stream) {  // no-presubmit-check TODO(webrtc:8982)
>-  const size_t kMaxEventSize = (1u << 16) - 1;
>-  const size_t kMaxVarintSize = 10;
>-  std::vector<char> buffer(kMaxEventSize + 2 * kMaxVarintSize);
>+  constexpr uint64_t kMaxEventSize = 10000000;  // Sanity check.
>+  std::vector<char> buffer(0xFFFF);
> 
>   RTC_DCHECK(stream.good());
> 
>@@ -491,10 +928,12 @@ bool ParsedRtcEventLogNew::ParseStreamInternal(
>       break;
>     }
> 
>-    // Read the next message tag. The tag number is defined as
>-    // (fieldnumber << 3) | wire_type. In our case, the field number is
>-    // supposed to be 1 and the wire type for a length-delimited field is 2.
>-    const uint64_t kExpectedV1Tag = (1 << 3) | 2;
>+    // Read the next message tag. Protobuf defines the message tag as
>+    // (field_number << 3) | wire_type. In the legacy encoding, the field number
>+    // is supposed to be 1 and the wire type for a length-delimited field is 2.
>+    // In the new encoding we still expect the wire type to be 2, but the field
>+    // number will be greater than 1.
>+    constexpr uint64_t kExpectedV1Tag = (1 << 3) | 2;
>     size_t bytes_written = 0;
>     absl::optional<uint64_t> tag =
>         ParseVarInt(stream, buffer.data(), &bytes_written);
>@@ -502,9 +941,13 @@ bool ParsedRtcEventLogNew::ParseStreamInternal(
>       RTC_LOG(LS_WARNING)
>           << "Missing field tag from beginning of protobuf event.";
>       return false;
>-    } else if (*tag != kExpectedV1Tag) {
>-      RTC_LOG(LS_WARNING)
>-          << "Unexpected field tag at beginning of protobuf event.";
>+    }
>+    constexpr uint64_t kWireTypeMask = 0x07;
>+    const uint64_t wire_type = *tag & kWireTypeMask;
>+    if (wire_type != 2) {
>+      RTC_LOG(LS_WARNING) << "Expected field tag with wire type 2 (length "
>+                             "delimited message). Found wire type "
>+                          << wire_type;
>       return false;
>     }
> 
>@@ -520,6 +963,8 @@ bool ParsedRtcEventLogNew::ParseStreamInternal(
>     }
> 
>     // Read the next protobuf event to a temporary char buffer.
>+    if (buffer.size() < bytes_written + *message_length)
>+      buffer.resize(bytes_written + *message_length);
>     stream.read(buffer.data() + bytes_written, *message_length);
>     if (stream.gcount() != static_cast<int>(*message_length)) {
>       RTC_LOG(LS_WARNING) << "Failed to read protobuf message from file.";
>@@ -527,21 +972,32 @@ bool ParsedRtcEventLogNew::ParseStreamInternal(
>     }
>     size_t buffer_size = bytes_written + *message_length;
> 
>-    // Parse the protobuf event from the buffer.
>-    rtclog::EventStream event_stream;
>-    if (!event_stream.ParseFromArray(buffer.data(), buffer_size)) {
>-      RTC_LOG(LS_WARNING) << "Failed to parse protobuf message.";
>-      return false;
>-    }
>+    if (*tag == kExpectedV1Tag) {
>+      // Parse the protobuf event from the buffer.
>+      rtclog::EventStream event_stream;
>+      if (!event_stream.ParseFromArray(buffer.data(), buffer_size)) {
>+        RTC_LOG(LS_WARNING)
>+            << "Failed to parse legacy-format protobuf message.";
>+        return false;
>+      }
> 
>-    RTC_CHECK_EQ(event_stream.stream_size(), 1);
>-    StoreParsedEvent(event_stream.stream(0));
>-    events_.push_back(event_stream.stream(0));
>+      RTC_CHECK_EQ(event_stream.stream_size(), 1);
>+      StoreParsedLegacyEvent(event_stream.stream(0));
>+    } else {
>+      // Parse the protobuf event from the buffer.
>+      rtclog2::EventStream event_stream;
>+      if (!event_stream.ParseFromArray(buffer.data(), buffer_size)) {
>+        RTC_LOG(LS_WARNING) << "Failed to parse new-format protobuf message.";
>+        return false;
>+      }
>+      StoreParsedNewFormatEvent(event_stream);
>+    }
>   }
>   return true;
> }
> 
>-void ParsedRtcEventLogNew::StoreParsedEvent(const rtclog::Event& event) {
>+void ParsedRtcEventLogNew::StoreParsedLegacyEvent(const rtclog::Event& event) {
>+  RTC_CHECK(event.has_type());
>   if (event.type() != rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT &&
>       event.type() != rtclog::Event::VIDEO_SENDER_CONFIG_EVENT &&
>       event.type() != rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT &&
>@@ -554,55 +1010,63 @@ void ParsedRtcEventLogNew::StoreParsedEvent(const rtclog::Event& event) {
>     last_timestamp_ = std::max(last_timestamp_, timestamp);
>   }
> 
>-  switch (GetEventType(event)) {
>-    case ParsedRtcEventLogNew::EventType::VIDEO_RECEIVER_CONFIG_EVENT: {
>+  switch (event.type()) {
>+    case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT: {
>       rtclog::StreamConfig config = GetVideoReceiveConfig(event);
>       video_recv_configs_.emplace_back(GetTimestamp(event), config);
>-      incoming_rtp_extensions_maps_[config.remote_ssrc] =
>-          RtpHeaderExtensionMap(config.rtp_extensions);
>-      // TODO(terelius): I don't understand the reason for configuring header
>-      // extensions for the local SSRC. I think it should be removed, but for
>-      // now I want to preserve the previous functionality.
>-      incoming_rtp_extensions_maps_[config.local_ssrc] =
>-          RtpHeaderExtensionMap(config.rtp_extensions);
>+      if (!config.rtp_extensions.empty()) {
>+        incoming_rtp_extensions_maps_[config.remote_ssrc] =
>+            RtpHeaderExtensionMap(config.rtp_extensions);
>+        // TODO(terelius): I don't understand the reason for configuring header
>+        // extensions for the local SSRC. I think it should be removed, but for
>+        // now I want to preserve the previous functionality.
>+        incoming_rtp_extensions_maps_[config.local_ssrc] =
>+            RtpHeaderExtensionMap(config.rtp_extensions);
>+      }
>       incoming_video_ssrcs_.insert(config.remote_ssrc);
>       incoming_video_ssrcs_.insert(config.rtx_ssrc);
>       incoming_rtx_ssrcs_.insert(config.rtx_ssrc);
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::VIDEO_SENDER_CONFIG_EVENT: {
>+    case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT: {
>       std::vector<rtclog::StreamConfig> configs = GetVideoSendConfig(event);
>       video_send_configs_.emplace_back(GetTimestamp(event), configs);
>       for (const auto& config : configs) {
>-        outgoing_rtp_extensions_maps_[config.local_ssrc] =
>-            RtpHeaderExtensionMap(config.rtp_extensions);
>-        outgoing_rtp_extensions_maps_[config.rtx_ssrc] =
>-            RtpHeaderExtensionMap(config.rtp_extensions);
>+        if (!config.rtp_extensions.empty()) {
>+          outgoing_rtp_extensions_maps_[config.local_ssrc] =
>+              RtpHeaderExtensionMap(config.rtp_extensions);
>+          outgoing_rtp_extensions_maps_[config.rtx_ssrc] =
>+              RtpHeaderExtensionMap(config.rtp_extensions);
>+        }
>         outgoing_video_ssrcs_.insert(config.local_ssrc);
>         outgoing_video_ssrcs_.insert(config.rtx_ssrc);
>         outgoing_rtx_ssrcs_.insert(config.rtx_ssrc);
>       }
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::AUDIO_RECEIVER_CONFIG_EVENT: {
>+    case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT: {
>       rtclog::StreamConfig config = GetAudioReceiveConfig(event);
>       audio_recv_configs_.emplace_back(GetTimestamp(event), config);
>-      incoming_rtp_extensions_maps_[config.remote_ssrc] =
>-          RtpHeaderExtensionMap(config.rtp_extensions);
>-      incoming_rtp_extensions_maps_[config.local_ssrc] =
>-          RtpHeaderExtensionMap(config.rtp_extensions);
>+      if (!config.rtp_extensions.empty()) {
>+        incoming_rtp_extensions_maps_[config.remote_ssrc] =
>+            RtpHeaderExtensionMap(config.rtp_extensions);
>+        incoming_rtp_extensions_maps_[config.local_ssrc] =
>+            RtpHeaderExtensionMap(config.rtp_extensions);
>+      }
>       incoming_audio_ssrcs_.insert(config.remote_ssrc);
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::AUDIO_SENDER_CONFIG_EVENT: {
>+    case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT: {
>       rtclog::StreamConfig config = GetAudioSendConfig(event);
>       audio_send_configs_.emplace_back(GetTimestamp(event), config);
>-      outgoing_rtp_extensions_maps_[config.local_ssrc] =
>-          RtpHeaderExtensionMap(config.rtp_extensions);
>+      if (!config.rtp_extensions.empty()) {
>+        outgoing_rtp_extensions_maps_[config.local_ssrc] =
>+            RtpHeaderExtensionMap(config.rtp_extensions);
>+      }
>       outgoing_audio_ssrcs_.insert(config.local_ssrc);
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::RTP_EVENT: {
>+    case rtclog::Event::RTP_EVENT: {
>       PacketDirection direction;
>       uint8_t header[IP_PACKET_SIZE];
>       size_t header_length;
>@@ -645,7 +1109,7 @@ void ParsedRtcEventLogNew::StoreParsedEvent(const rtclog::Event& event) {
>       }
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::RTCP_EVENT: {
>+    case rtclog::Event::RTCP_EVENT: {
>       PacketDirection direction;
>       uint8_t packet[IP_PACKET_SIZE];
>       size_t total_length;
>@@ -731,113 +1195,76 @@ void ParsedRtcEventLogNew::StoreParsedEvent(const rtclog::Event& event) {
>       }
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::LOG_START: {
>+    case rtclog::Event::LOG_START: {
>       start_log_events_.push_back(LoggedStartEvent(GetTimestamp(event)));
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::LOG_END: {
>+    case rtclog::Event::LOG_END: {
>       stop_log_events_.push_back(LoggedStopEvent(GetTimestamp(event)));
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT: {
>+    case rtclog::Event::AUDIO_PLAYOUT_EVENT: {
>       LoggedAudioPlayoutEvent playout_event = GetAudioPlayout(event);
>       audio_playout_events_[playout_event.ssrc].push_back(playout_event);
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::LOSS_BASED_BWE_UPDATE: {
>+    case rtclog::Event::LOSS_BASED_BWE_UPDATE: {
>       bwe_loss_updates_.push_back(GetLossBasedBweUpdate(event));
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::DELAY_BASED_BWE_UPDATE: {
>+    case rtclog::Event::DELAY_BASED_BWE_UPDATE: {
>       bwe_delay_updates_.push_back(GetDelayBasedBweUpdate(event));
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::AUDIO_NETWORK_ADAPTATION_EVENT: {
>+    case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT: {
>       LoggedAudioNetworkAdaptationEvent ana_event =
>           GetAudioNetworkAdaptation(event);
>       audio_network_adaptation_events_.push_back(ana_event);
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::BWE_PROBE_CLUSTER_CREATED_EVENT: {
>+    case rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT: {
>       bwe_probe_cluster_created_events_.push_back(
>           GetBweProbeClusterCreated(event));
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::BWE_PROBE_FAILURE_EVENT: {
>-      bwe_probe_failure_events_.push_back(GetBweProbeFailure(event));
>-      break;
>-    }
>-    case ParsedRtcEventLogNew::EventType::BWE_PROBE_SUCCESS_EVENT: {
>-      bwe_probe_success_events_.push_back(GetBweProbeSuccess(event));
>+    case rtclog::Event::BWE_PROBE_RESULT_EVENT: {
>+      // Probe successes and failures are currently stored in the same proto
>+      // message, we are moving towards separate messages. Probe results
>+      // therefore need special treatment in the parser.
>+      RTC_CHECK(event.has_probe_result());
>+      RTC_CHECK(event.probe_result().has_result());
>+      if (event.probe_result().result() == rtclog::BweProbeResult::SUCCESS) {
>+        bwe_probe_success_events_.push_back(GetBweProbeSuccess(event));
>+      } else {
>+        bwe_probe_failure_events_.push_back(GetBweProbeFailure(event));
>+      }
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::ALR_STATE_EVENT: {
>+    case rtclog::Event::ALR_STATE_EVENT: {
>       alr_state_events_.push_back(GetAlrState(event));
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_CONFIG: {
>+    case rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG: {
>       ice_candidate_pair_configs_.push_back(GetIceCandidatePairConfig(event));
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_EVENT: {
>+    case rtclog::Event::ICE_CANDIDATE_PAIR_EVENT: {
>       ice_candidate_pair_events_.push_back(GetIceCandidatePairEvent(event));
>       break;
>     }
>-    case ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT: {
>+    case rtclog::Event::UNKNOWN_EVENT: {
>       break;
>     }
>   }
> }
> 
>-size_t ParsedRtcEventLogNew::GetNumberOfEvents() const {
>-  return events_.size();
>-}
>-
>-int64_t ParsedRtcEventLogNew::GetTimestamp(size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  return GetTimestamp(event);
>-}
> 
> int64_t ParsedRtcEventLogNew::GetTimestamp(const rtclog::Event& event) const {
>   RTC_CHECK(event.has_timestamp_us());
>   return event.timestamp_us();
> }
> 
>-ParsedRtcEventLogNew::EventType ParsedRtcEventLogNew::GetEventType(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  return GetEventType(event);
>-}
>-
>-ParsedRtcEventLogNew::EventType ParsedRtcEventLogNew::GetEventType(
>-    const rtclog::Event& event) const {
>-  RTC_CHECK(event.has_type());
>-  if (event.type() == rtclog::Event::BWE_PROBE_RESULT_EVENT) {
>-    RTC_CHECK(event.has_probe_result());
>-    RTC_CHECK(event.probe_result().has_result());
>-    if (event.probe_result().result() == rtclog::BweProbeResult::SUCCESS)
>-      return ParsedRtcEventLogNew::EventType::BWE_PROBE_SUCCESS_EVENT;
>-    return ParsedRtcEventLogNew::EventType::BWE_PROBE_FAILURE_EVENT;
>-  }
>-  return GetRuntimeEventType(event.type());
>-}
>-
> // The header must have space for at least IP_PACKET_SIZE bytes.
>-const webrtc::RtpHeaderExtensionMap* ParsedRtcEventLogNew::GetRtpHeader(
>-    size_t index,
>-    PacketDirection* incoming,
>-    uint8_t* header,
>-    size_t* header_length,
>-    size_t* total_length,
>-    int* probe_cluster_id) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  return GetRtpHeader(event, incoming, header, header_length, total_length,
>-                      probe_cluster_id);
>-}
>-
> const webrtc::RtpHeaderExtensionMap* ParsedRtcEventLogNew::GetRtpHeader(
>     const rtclog::Event& event,
>     PacketDirection* incoming,
>@@ -899,15 +1326,6 @@ const webrtc::RtpHeaderExtensionMap* ParsedRtcEventLogNew::GetRtpHeader(
> }
> 
> // The packet must have space for at least IP_PACKET_SIZE bytes.
>-void ParsedRtcEventLogNew::GetRtcpPacket(size_t index,
>-                                         PacketDirection* incoming,
>-                                         uint8_t* packet,
>-                                         size_t* length) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  GetRtcpPacket(event, incoming, packet, length);
>-}
>-
> void ParsedRtcEventLogNew::GetRtcpPacket(const rtclog::Event& event,
>                                          PacketDirection* incoming,
>                                          uint8_t* packet,
>@@ -935,12 +1353,6 @@ void ParsedRtcEventLogNew::GetRtcpPacket(const rtclog::Event& event,
>   }
> }
> 
>-rtclog::StreamConfig ParsedRtcEventLogNew::GetVideoReceiveConfig(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  return GetVideoReceiveConfig(events_[index]);
>-}
>-
> rtclog::StreamConfig ParsedRtcEventLogNew::GetVideoReceiveConfig(
>     const rtclog::Event& event) const {
>   rtclog::StreamConfig config;
>@@ -1002,12 +1414,6 @@ rtclog::StreamConfig ParsedRtcEventLogNew::GetVideoReceiveConfig(
>   return config;
> }
> 
>-std::vector<rtclog::StreamConfig> ParsedRtcEventLogNew::GetVideoSendConfig(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  return GetVideoSendConfig(events_[index]);
>-}
>-
> std::vector<rtclog::StreamConfig> ParsedRtcEventLogNew::GetVideoSendConfig(
>     const rtclog::Event& event) const {
>   std::vector<rtclog::StreamConfig> configs;
>@@ -1046,12 +1452,6 @@ std::vector<rtclog::StreamConfig> ParsedRtcEventLogNew::GetVideoSendConfig(
>   return configs;
> }
> 
>-rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioReceiveConfig(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  return GetAudioReceiveConfig(events_[index]);
>-}
>-
> rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioReceiveConfig(
>     const rtclog::Event& event) const {
>   rtclog::StreamConfig config;
>@@ -1071,12 +1471,6 @@ rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioReceiveConfig(
>   return config;
> }
> 
>-rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioSendConfig(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  return GetAudioSendConfig(events_[index]);
>-}
>-
> rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioSendConfig(
>     const rtclog::Event& event) const {
>   rtclog::StreamConfig config;
>@@ -1093,13 +1487,6 @@ rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioSendConfig(
>   return config;
> }
> 
>-LoggedAudioPlayoutEvent ParsedRtcEventLogNew::GetAudioPlayout(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  return GetAudioPlayout(event);
>-}
>-
> LoggedAudioPlayoutEvent ParsedRtcEventLogNew::GetAudioPlayout(
>     const rtclog::Event& event) const {
>   RTC_CHECK(event.has_type());
>@@ -1113,13 +1500,6 @@ LoggedAudioPlayoutEvent ParsedRtcEventLogNew::GetAudioPlayout(
>   return res;
> }
> 
>-LoggedBweLossBasedUpdate ParsedRtcEventLogNew::GetLossBasedBweUpdate(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  return GetLossBasedBweUpdate(event);
>-}
>-
> LoggedBweLossBasedUpdate ParsedRtcEventLogNew::GetLossBasedBweUpdate(
>     const rtclog::Event& event) const {
>   RTC_CHECK(event.has_type());
>@@ -1138,13 +1518,6 @@ LoggedBweLossBasedUpdate ParsedRtcEventLogNew::GetLossBasedBweUpdate(
>   return bwe_update;
> }
> 
>-LoggedBweDelayBasedUpdate ParsedRtcEventLogNew::GetDelayBasedBweUpdate(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  return GetDelayBasedBweUpdate(event);
>-}
>-
> LoggedBweDelayBasedUpdate ParsedRtcEventLogNew::GetDelayBasedBweUpdate(
>     const rtclog::Event& event) const {
>   RTC_CHECK(event.has_type());
>@@ -1162,13 +1535,6 @@ LoggedBweDelayBasedUpdate ParsedRtcEventLogNew::GetDelayBasedBweUpdate(
>   return res;
> }
> 
>-LoggedAudioNetworkAdaptationEvent
>-ParsedRtcEventLogNew::GetAudioNetworkAdaptation(size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  return GetAudioNetworkAdaptation(event);
>-}
>-
> LoggedAudioNetworkAdaptationEvent
> ParsedRtcEventLogNew::GetAudioNetworkAdaptation(
>     const rtclog::Event& event) const {
>@@ -1196,13 +1562,6 @@ ParsedRtcEventLogNew::GetAudioNetworkAdaptation(
>   return res;
> }
> 
>-LoggedBweProbeClusterCreatedEvent
>-ParsedRtcEventLogNew::GetBweProbeClusterCreated(size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  return GetBweProbeClusterCreated(event);
>-}
>-
> LoggedBweProbeClusterCreatedEvent
> ParsedRtcEventLogNew::GetBweProbeClusterCreated(
>     const rtclog::Event& event) const {
>@@ -1223,13 +1582,6 @@ ParsedRtcEventLogNew::GetBweProbeClusterCreated(
>   return res;
> }
> 
>-LoggedBweProbeFailureEvent ParsedRtcEventLogNew::GetBweProbeFailure(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  return GetBweProbeFailure(event);
>-}
>-
> LoggedBweProbeFailureEvent ParsedRtcEventLogNew::GetBweProbeFailure(
>     const rtclog::Event& event) const {
>   RTC_CHECK(event.has_type());
>@@ -1260,13 +1612,6 @@ LoggedBweProbeFailureEvent ParsedRtcEventLogNew::GetBweProbeFailure(
>   return res;
> }
> 
>-LoggedBweProbeSuccessEvent ParsedRtcEventLogNew::GetBweProbeSuccess(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  return GetBweProbeSuccess(event);
>-}
>-
> LoggedBweProbeSuccessEvent ParsedRtcEventLogNew::GetBweProbeSuccess(
>     const rtclog::Event& event) const {
>   RTC_CHECK(event.has_type());
>@@ -1286,12 +1631,6 @@ LoggedBweProbeSuccessEvent ParsedRtcEventLogNew::GetBweProbeSuccess(
>   return res;
> }
> 
>-LoggedAlrStateEvent ParsedRtcEventLogNew::GetAlrState(size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& event = events_[index];
>-  return GetAlrState(event);
>-}
>-
> LoggedAlrStateEvent ParsedRtcEventLogNew::GetAlrState(
>     const rtclog::Event& event) const {
>   RTC_CHECK(event.has_type());
>@@ -1306,13 +1645,6 @@ LoggedAlrStateEvent ParsedRtcEventLogNew::GetAlrState(
>   return res;
> }
> 
>-LoggedIceCandidatePairConfig ParsedRtcEventLogNew::GetIceCandidatePairConfig(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& rtc_event = events_[index];
>-  return GetIceCandidatePairConfig(rtc_event);
>-}
>-
> LoggedIceCandidatePairConfig ParsedRtcEventLogNew::GetIceCandidatePairConfig(
>     const rtclog::Event& rtc_event) const {
>   RTC_CHECK(rtc_event.has_type());
>@@ -1349,13 +1681,6 @@ LoggedIceCandidatePairConfig ParsedRtcEventLogNew::GetIceCandidatePairConfig(
>   return res;
> }
> 
>-LoggedIceCandidatePairEvent ParsedRtcEventLogNew::GetIceCandidatePairEvent(
>-    size_t index) const {
>-  RTC_CHECK_LT(index, GetNumberOfEvents());
>-  const rtclog::Event& rtc_event = events_[index];
>-  return GetIceCandidatePairEvent(rtc_event);
>-}
>-
> LoggedIceCandidatePairEvent ParsedRtcEventLogNew::GetIceCandidatePairEvent(
>     const rtclog::Event& rtc_event) const {
>   RTC_CHECK(rtc_event.has_type());
>@@ -1438,7 +1763,8 @@ const std::vector<MatchedSendArrivalTimes> GetNetworkTrace(
>     clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
>     if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
>       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
>-      feedback_adapter.OnTransportFeedback(rtcp_iterator->transport_feedback);
>+      feedback_adapter.ProcessTransportFeedback(
>+          rtcp_iterator->transport_feedback);
>       std::vector<PacketFeedback> feedback =
>           feedback_adapter.GetTransportFeedbackVector();
>       SortPacketFeedbackVectorWithLoss(&feedback);
>@@ -1452,14 +1778,25 @@ const std::vector<MatchedSendArrivalTimes> GetNetworkTrace(
>     if (clock.TimeInMicroseconds() >= NextRtpTime()) {
>       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
>       const RtpPacketType& rtp_packet = *rtp_iterator->second;
>+      rtc::SentPacket sent_packet;
>+      sent_packet.send_time_ms = rtp_packet.rtp.log_time_ms();
>+      sent_packet.info.packet_size_bytes = rtp_packet.rtp.total_length;
>       if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
>         feedback_adapter.AddPacket(
>             rtp_packet.rtp.header.ssrc,
>             rtp_packet.rtp.header.extension.transportSequenceNumber,
>             rtp_packet.rtp.total_length, PacedPacketInfo());
>-        feedback_adapter.OnSentPacket(
>-            rtp_packet.rtp.header.extension.transportSequenceNumber,
>-            rtp_packet.rtp.log_time_ms());
>+        sent_packet.packet_id =
>+            rtp_packet.rtp.header.extension.transportSequenceNumber;
>+        sent_packet.info.included_in_feedback = true;
>+        sent_packet.info.included_in_allocation = true;
>+        feedback_adapter.ProcessSentPacket(sent_packet);
>+      } else {
>+        sent_packet.info.included_in_feedback = false;
>+        // TODO(srte): Make it possible to indicate that all packets are part of
>+        // allocation.
>+        sent_packet.info.included_in_allocation = false;
>+        feedback_adapter.ProcessSentPacket(sent_packet);
>       }
>       ++rtp_iterator;
>     }
>@@ -1468,4 +1805,613 @@ const std::vector<MatchedSendArrivalTimes> GetNetworkTrace(
>   return rtp_rtcp_matched;
> }
> 
>+// Helper functions for new format start here
>+void ParsedRtcEventLogNew::StoreParsedNewFormatEvent(
>+    const rtclog2::EventStream& stream) {
>+  RTC_DCHECK_EQ(stream.stream_size(), 0);
>+
>+  RTC_DCHECK_EQ(
>+      stream.incoming_rtp_packets_size() + stream.outgoing_rtp_packets_size() +
>+          stream.incoming_rtcp_packets_size() +
>+          stream.outgoing_rtcp_packets_size() +
>+          stream.audio_playout_events_size() + stream.begin_log_events_size() +
>+          stream.end_log_events_size() + stream.loss_based_bwe_updates_size() +
>+          stream.delay_based_bwe_updates_size() +
>+          stream.audio_network_adaptations_size() +
>+          stream.probe_clusters_size() + stream.probe_success_size() +
>+          stream.probe_failure_size() + stream.alr_states_size() +
>+          stream.ice_candidate_configs_size() +
>+          stream.ice_candidate_events_size() +
>+          stream.audio_recv_stream_configs_size() +
>+          stream.audio_send_stream_configs_size() +
>+          stream.video_recv_stream_configs_size() +
>+          stream.video_send_stream_configs_size(),
>+      1u);
>+
>+  if (stream.incoming_rtp_packets_size() == 1) {
>+    StoreIncomingRtpPackets(stream.incoming_rtp_packets(0));
>+  } else if (stream.outgoing_rtp_packets_size() == 1) {
>+    StoreOutgoingRtpPackets(stream.outgoing_rtp_packets(0));
>+  } else if (stream.incoming_rtcp_packets_size() == 1) {
>+    StoreIncomingRtcpPackets(stream.incoming_rtcp_packets(0));
>+  } else if (stream.outgoing_rtcp_packets_size() == 1) {
>+    StoreOutgoingRtcpPackets(stream.outgoing_rtcp_packets(0));
>+  } else if (stream.audio_playout_events_size() == 1) {
>+    StoreAudioPlayoutEvent(stream.audio_playout_events(0));
>+  } else if (stream.begin_log_events_size() == 1) {
>+    StoreStartEvent(stream.begin_log_events(0));
>+  } else if (stream.end_log_events_size() == 1) {
>+    StoreStopEvent(stream.end_log_events(0));
>+  } else if (stream.loss_based_bwe_updates_size() == 1) {
>+    StoreBweLossBasedUpdate(stream.loss_based_bwe_updates(0));
>+  } else if (stream.delay_based_bwe_updates_size() == 1) {
>+    StoreBweDelayBasedUpdate(stream.delay_based_bwe_updates(0));
>+  } else if (stream.audio_network_adaptations_size() == 1) {
>+    StoreAudioNetworkAdaptationEvent(stream.audio_network_adaptations(0));
>+  } else if (stream.probe_clusters_size() == 1) {
>+    StoreBweProbeClusterCreated(stream.probe_clusters(0));
>+  } else if (stream.probe_success_size() == 1) {
>+    StoreBweProbeSuccessEvent(stream.probe_success(0));
>+  } else if (stream.probe_failure_size() == 1) {
>+    StoreBweProbeFailureEvent(stream.probe_failure(0));
>+  } else if (stream.alr_states_size() == 1) {
>+    StoreAlrStateEvent(stream.alr_states(0));
>+  } else if (stream.ice_candidate_configs_size() == 1) {
>+    StoreIceCandidatePairConfig(stream.ice_candidate_configs(0));
>+  } else if (stream.ice_candidate_events_size() == 1) {
>+    StoreIceCandidateEvent(stream.ice_candidate_events(0));
>+  } else if (stream.audio_recv_stream_configs_size() == 1) {
>+    StoreAudioRecvConfig(stream.audio_recv_stream_configs(0));
>+  } else if (stream.audio_send_stream_configs_size() == 1) {
>+    StoreAudioSendConfig(stream.audio_send_stream_configs(0));
>+  } else if (stream.video_recv_stream_configs_size() == 1) {
>+    StoreVideoRecvConfig(stream.video_recv_stream_configs(0));
>+  } else if (stream.video_send_stream_configs_size() == 1) {
>+    StoreVideoSendConfig(stream.video_send_stream_configs(0));
>+  } else {
>+    RTC_NOTREACHED();
>+  }
>+}
>+
>+void ParsedRtcEventLogNew::StoreAlrStateEvent(const rtclog2::AlrState& proto) {
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  RTC_CHECK(proto.has_in_alr());
>+  LoggedAlrStateEvent alr_event;
>+  alr_event.timestamp_us = proto.timestamp_ms() * 1000;
>+  alr_event.in_alr = proto.in_alr();
>+
>+  alr_state_events_.push_back(alr_event);
>+  // TODO(terelius): Should we delta encode this event type?
>+}
>+
>+void ParsedRtcEventLogNew::StoreAudioPlayoutEvent(
>+    const rtclog2::AudioPlayoutEvents& proto) {
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  RTC_CHECK(proto.has_local_ssrc());
>+
>+  // Base event
>+  auto map_it = audio_playout_events_[proto.local_ssrc()];
>+  audio_playout_events_[proto.local_ssrc()].emplace_back(
>+      1000 * proto.timestamp_ms(), proto.local_ssrc());
>+
>+  const size_t number_of_deltas =
>+      proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
>+  if (number_of_deltas == 0) {
>+    return;
>+  }
>+
>+  // timestamp_ms
>+  std::vector<absl::optional<uint64_t>> timestamp_ms_values =
>+      DecodeDeltas(proto.timestamp_ms_deltas(),
>+                   ToUnsigned(proto.timestamp_ms()), number_of_deltas);
>+  RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
>+
>+  // local_ssrc
>+  std::vector<absl::optional<uint64_t>> local_ssrc_values = DecodeDeltas(
>+      proto.local_ssrc_deltas(), proto.local_ssrc(), number_of_deltas);
>+  RTC_CHECK_EQ(local_ssrc_values.size(), number_of_deltas);
>+
>+  // Delta decoding
>+  for (size_t i = 0; i < number_of_deltas; ++i) {
>+    RTC_CHECK(timestamp_ms_values[i].has_value());
>+    RTC_CHECK(local_ssrc_values[i].has_value());
>+    RTC_CHECK_LE(local_ssrc_values[i].value(),
>+                 std::numeric_limits<uint32_t>::max());
>+
>+    int64_t timestamp_ms;
>+    RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
>+
>+    const uint32_t local_ssrc =
>+        static_cast<uint32_t>(local_ssrc_values[i].value());
>+    audio_playout_events_[local_ssrc].emplace_back(1000 * timestamp_ms,
>+                                                   local_ssrc);
>+  }
>+}
>+
>+void ParsedRtcEventLogNew::StoreIncomingRtpPackets(
>+    const rtclog2::IncomingRtpPackets& proto) {
>+  StoreRtpPackets(proto, &incoming_rtp_packets_map_);
>+}
>+
>+void ParsedRtcEventLogNew::StoreOutgoingRtpPackets(
>+    const rtclog2::OutgoingRtpPackets& proto) {
>+  StoreRtpPackets(proto, &outgoing_rtp_packets_map_);
>+}
>+
>+void ParsedRtcEventLogNew::StoreIncomingRtcpPackets(
>+    const rtclog2::IncomingRtcpPackets& proto) {
>+  StoreRtcpPackets(proto, &incoming_rtcp_packets_);
>+}
>+
>+void ParsedRtcEventLogNew::StoreOutgoingRtcpPackets(
>+    const rtclog2::OutgoingRtcpPackets& proto) {
>+  StoreRtcpPackets(proto, &outgoing_rtcp_packets_);
>+}
>+
>+void ParsedRtcEventLogNew::StoreStartEvent(
>+    const rtclog2::BeginLogEvent& proto) {
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  RTC_CHECK(proto.has_version());
>+  RTC_CHECK(proto.has_utc_time_ms());
>+  RTC_CHECK_EQ(proto.version(), 2);
>+  LoggedStartEvent start_event(proto.timestamp_ms() * 1000,
>+                               proto.utc_time_ms());
>+
>+  start_log_events_.push_back(start_event);
>+}
>+
>+void ParsedRtcEventLogNew::StoreStopEvent(const rtclog2::EndLogEvent& proto) {
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  LoggedStopEvent stop_event(proto.timestamp_ms() * 1000);
>+
>+  stop_log_events_.push_back(stop_event);
>+}
>+
>+void ParsedRtcEventLogNew::StoreBweLossBasedUpdate(
>+    const rtclog2::LossBasedBweUpdates& proto) {
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  RTC_CHECK(proto.has_bitrate_bps());
>+  RTC_CHECK(proto.has_fraction_loss());
>+  RTC_CHECK(proto.has_total_packets());
>+
>+  // Base event
>+  bwe_loss_updates_.emplace_back(1000 * proto.timestamp_ms(),
>+                                 proto.bitrate_bps(), proto.fraction_loss(),
>+                                 proto.total_packets());
>+
>+  const size_t number_of_deltas =
>+      proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
>+  if (number_of_deltas == 0) {
>+    return;
>+  }
>+
>+  // timestamp_ms
>+  std::vector<absl::optional<uint64_t>> timestamp_ms_values =
>+      DecodeDeltas(proto.timestamp_ms_deltas(),
>+                   ToUnsigned(proto.timestamp_ms()), number_of_deltas);
>+  RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
>+
>+  // bitrate_bps
>+  std::vector<absl::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
>+      proto.bitrate_bps_deltas(), proto.bitrate_bps(), number_of_deltas);
>+  RTC_CHECK_EQ(bitrate_bps_values.size(), number_of_deltas);
>+
>+  // fraction_loss
>+  std::vector<absl::optional<uint64_t>> fraction_loss_values = DecodeDeltas(
>+      proto.fraction_loss_deltas(), proto.fraction_loss(), number_of_deltas);
>+  RTC_CHECK_EQ(fraction_loss_values.size(), number_of_deltas);
>+
>+  // total_packets
>+  std::vector<absl::optional<uint64_t>> total_packets_values = DecodeDeltas(
>+      proto.total_packets_deltas(), proto.total_packets(), number_of_deltas);
>+  RTC_CHECK_EQ(total_packets_values.size(), number_of_deltas);
>+
>+  // Delta decoding
>+  for (size_t i = 0; i < number_of_deltas; ++i) {
>+    RTC_CHECK(timestamp_ms_values[i].has_value());
>+    int64_t timestamp_ms;
>+    RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
>+
>+    RTC_CHECK(bitrate_bps_values[i].has_value());
>+    RTC_CHECK_LE(bitrate_bps_values[i].value(),
>+                 std::numeric_limits<uint32_t>::max());
>+    const uint32_t bitrate_bps =
>+        static_cast<uint32_t>(bitrate_bps_values[i].value());
>+
>+    RTC_CHECK(fraction_loss_values[i].has_value());
>+    RTC_CHECK_LE(fraction_loss_values[i].value(),
>+                 std::numeric_limits<uint32_t>::max());
>+    const uint32_t fraction_loss =
>+        static_cast<uint32_t>(fraction_loss_values[i].value());
>+
>+    RTC_CHECK(total_packets_values[i].has_value());
>+    RTC_CHECK_LE(total_packets_values[i].value(),
>+                 std::numeric_limits<uint32_t>::max());
>+    const uint32_t total_packets =
>+        static_cast<uint32_t>(total_packets_values[i].value());
>+
>+    bwe_loss_updates_.emplace_back(1000 * timestamp_ms, bitrate_bps,
>+                                   fraction_loss, total_packets);
>+  }
>+}
>+
>+void ParsedRtcEventLogNew::StoreBweDelayBasedUpdate(
>+    const rtclog2::DelayBasedBweUpdates& proto) {
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  RTC_CHECK(proto.has_bitrate_bps());
>+  RTC_CHECK(proto.has_detector_state());
>+
>+  // Base event
>+  const BandwidthUsage base_detector_state =
>+      GetRuntimeDetectorState(proto.detector_state());
>+  bwe_delay_updates_.emplace_back(1000 * proto.timestamp_ms(),
>+                                  proto.bitrate_bps(), base_detector_state);
>+
>+  const size_t number_of_deltas =
>+      proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
>+  if (number_of_deltas == 0) {
>+    return;
>+  }
>+
>+  // timestamp_ms
>+  std::vector<absl::optional<uint64_t>> timestamp_ms_values =
>+      DecodeDeltas(proto.timestamp_ms_deltas(),
>+                   ToUnsigned(proto.timestamp_ms()), number_of_deltas);
>+  RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
>+
>+  // bitrate_bps
>+  std::vector<absl::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
>+      proto.bitrate_bps_deltas(), proto.bitrate_bps(), number_of_deltas);
>+  RTC_CHECK_EQ(bitrate_bps_values.size(), number_of_deltas);
>+
>+  // detector_state
>+  std::vector<absl::optional<uint64_t>> detector_state_values = DecodeDeltas(
>+      proto.detector_state_deltas(),
>+      static_cast<uint64_t>(proto.detector_state()), number_of_deltas);
>+  RTC_CHECK_EQ(detector_state_values.size(), number_of_deltas);
>+
>+  // Delta decoding
>+  for (size_t i = 0; i < number_of_deltas; ++i) {
>+    RTC_CHECK(timestamp_ms_values[i].has_value());
>+    int64_t timestamp_ms;
>+    RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
>+
>+    RTC_CHECK(bitrate_bps_values[i].has_value());
>+    RTC_CHECK_LE(bitrate_bps_values[i].value(),
>+                 std::numeric_limits<uint32_t>::max());
>+    const uint32_t bitrate_bps =
>+        static_cast<uint32_t>(bitrate_bps_values[i].value());
>+
>+    RTC_CHECK(detector_state_values[i].has_value());
>+    const auto detector_state =
>+        static_cast<rtclog2::DelayBasedBweUpdates::DetectorState>(
>+            detector_state_values[i].value());
>+
>+    bwe_delay_updates_.emplace_back(1000 * timestamp_ms, bitrate_bps,
>+                                    GetRuntimeDetectorState(detector_state));
>+  }
>+}
>+
>+void ParsedRtcEventLogNew::StoreBweProbeClusterCreated(
>+    const rtclog2::BweProbeCluster& proto) {
>+  LoggedBweProbeClusterCreatedEvent probe_cluster;
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  probe_cluster.timestamp_us = proto.timestamp_ms() * 1000;
>+  RTC_CHECK(proto.has_id());
>+  probe_cluster.id = proto.id();
>+  RTC_CHECK(proto.has_bitrate_bps());
>+  probe_cluster.bitrate_bps = proto.bitrate_bps();
>+  RTC_CHECK(proto.has_min_packets());
>+  probe_cluster.min_packets = proto.min_packets();
>+  RTC_CHECK(proto.has_min_bytes());
>+  probe_cluster.min_bytes = proto.min_bytes();
>+
>+  bwe_probe_cluster_created_events_.push_back(probe_cluster);
>+
>+  // TODO(terelius): Should we delta encode this event type?
>+}
>+
>+void ParsedRtcEventLogNew::StoreBweProbeSuccessEvent(
>+    const rtclog2::BweProbeResultSuccess& proto) {
>+  LoggedBweProbeSuccessEvent probe_result;
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  probe_result.timestamp_us = proto.timestamp_ms() * 1000;
>+  RTC_CHECK(proto.has_id());
>+  probe_result.id = proto.id();
>+  RTC_CHECK(proto.has_bitrate_bps());
>+  probe_result.bitrate_bps = proto.bitrate_bps();
>+
>+  bwe_probe_success_events_.push_back(probe_result);
>+
>+  // TODO(terelius): Should we delta encode this event type?
>+}
>+
>+void ParsedRtcEventLogNew::StoreBweProbeFailureEvent(
>+    const rtclog2::BweProbeResultFailure& proto) {
>+  LoggedBweProbeFailureEvent probe_result;
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  probe_result.timestamp_us = proto.timestamp_ms() * 1000;
>+  RTC_CHECK(proto.has_id());
>+  probe_result.id = proto.id();
>+  RTC_CHECK(proto.has_failure());
>+  probe_result.failure_reason = GetRuntimeProbeFailureReason(proto.failure());
>+
>+  bwe_probe_failure_events_.push_back(probe_result);
>+
>+  // TODO(terelius): Should we delta encode this event type?
>+}
>+
>+void ParsedRtcEventLogNew::StoreAudioNetworkAdaptationEvent(
>+    const rtclog2::AudioNetworkAdaptations& proto) {
>+  RTC_CHECK(proto.has_timestamp_ms());
>+
>+  // Base event
>+  {
>+    AudioEncoderRuntimeConfig runtime_config;
>+    if (proto.has_bitrate_bps()) {
>+      runtime_config.bitrate_bps = proto.bitrate_bps();
>+    }
>+    if (proto.has_frame_length_ms()) {
>+      runtime_config.frame_length_ms = proto.frame_length_ms();
>+    }
>+    if (proto.has_uplink_packet_loss_fraction()) {
>+      float uplink_packet_loss_fraction;
>+      RTC_CHECK(ParsePacketLossFractionFromProtoFormat(
>+          proto.uplink_packet_loss_fraction(), &uplink_packet_loss_fraction));
>+      runtime_config.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
>+    }
>+    if (proto.has_enable_fec()) {
>+      runtime_config.enable_fec = proto.enable_fec();
>+    }
>+    if (proto.has_enable_dtx()) {
>+      runtime_config.enable_dtx = proto.enable_dtx();
>+    }
>+    if (proto.has_num_channels()) {
>+      // Note: Encoding N as N-1 only done for |num_channels_deltas|.
>+      runtime_config.num_channels = proto.num_channels();
>+    }
>+    audio_network_adaptation_events_.emplace_back(1000 * proto.timestamp_ms(),
>+                                                  runtime_config);
>+  }
>+
>+  const size_t number_of_deltas =
>+      proto.has_number_of_deltas() ? proto.number_of_deltas() : 0u;
>+  if (number_of_deltas == 0) {
>+    return;
>+  }
>+
>+  // timestamp_ms
>+  std::vector<absl::optional<uint64_t>> timestamp_ms_values =
>+      DecodeDeltas(proto.timestamp_ms_deltas(),
>+                   ToUnsigned(proto.timestamp_ms()), number_of_deltas);
>+  RTC_CHECK_EQ(timestamp_ms_values.size(), number_of_deltas);
>+
>+  // bitrate_bps
>+  const absl::optional<uint64_t> unsigned_base_bitrate_bps =
>+      proto.has_bitrate_bps()
>+          ? absl::optional<uint64_t>(ToUnsigned(proto.bitrate_bps()))
>+          : absl::optional<uint64_t>();
>+  std::vector<absl::optional<uint64_t>> bitrate_bps_values = DecodeDeltas(
>+      proto.bitrate_bps_deltas(), unsigned_base_bitrate_bps, number_of_deltas);
>+  RTC_CHECK_EQ(bitrate_bps_values.size(), number_of_deltas);
>+
>+  // frame_length_ms
>+  const absl::optional<uint64_t> unsigned_base_frame_length_ms =
>+      proto.has_frame_length_ms()
>+          ? absl::optional<uint64_t>(ToUnsigned(proto.frame_length_ms()))
>+          : absl::optional<uint64_t>();
>+  std::vector<absl::optional<uint64_t>> frame_length_ms_values =
>+      DecodeDeltas(proto.frame_length_ms_deltas(),
>+                   unsigned_base_frame_length_ms, number_of_deltas);
>+  RTC_CHECK_EQ(frame_length_ms_values.size(), number_of_deltas);
>+
>+  // uplink_packet_loss_fraction
>+  const absl::optional<uint64_t> uplink_packet_loss_fraction =
>+      proto.has_uplink_packet_loss_fraction()
>+          ? absl::optional<uint64_t>(proto.uplink_packet_loss_fraction())
>+          : absl::optional<uint64_t>();
>+  std::vector<absl::optional<uint64_t>> uplink_packet_loss_fraction_values =
>+      DecodeDeltas(proto.uplink_packet_loss_fraction_deltas(),
>+                   uplink_packet_loss_fraction, number_of_deltas);
>+  RTC_CHECK_EQ(uplink_packet_loss_fraction_values.size(), number_of_deltas);
>+
>+  // enable_fec
>+  const absl::optional<uint64_t> enable_fec =
>+      proto.has_enable_fec() ? absl::optional<uint64_t>(proto.enable_fec())
>+                             : absl::optional<uint64_t>();
>+  std::vector<absl::optional<uint64_t>> enable_fec_values =
>+      DecodeDeltas(proto.enable_fec_deltas(), enable_fec, number_of_deltas);
>+  RTC_CHECK_EQ(enable_fec_values.size(), number_of_deltas);
>+
>+  // enable_dtx
>+  const absl::optional<uint64_t> enable_dtx =
>+      proto.has_enable_dtx() ? absl::optional<uint64_t>(proto.enable_dtx())
>+                             : absl::optional<uint64_t>();
>+  std::vector<absl::optional<uint64_t>> enable_dtx_values =
>+      DecodeDeltas(proto.enable_dtx_deltas(), enable_dtx, number_of_deltas);
>+  RTC_CHECK_EQ(enable_dtx_values.size(), number_of_deltas);
>+
>+  // num_channels
>+  // Note: For delta encoding, all num_channel values, including the base,
>+  // were shifted down by one, but in the base event, they were not.
>+  // We likewise shift the base event down by one, to get the same base as
>+  // encoding had, but then shift all of the values (except the base) back up
>+  //  to their original value.
>+  absl::optional<uint64_t> shifted_base_num_channels;
>+  if (proto.has_num_channels()) {
>+    shifted_base_num_channels =
>+        absl::optional<uint64_t>(proto.num_channels() - 1);
>+  }
>+  std::vector<absl::optional<uint64_t>> num_channels_values = DecodeDeltas(
>+      proto.num_channels_deltas(), shifted_base_num_channels, number_of_deltas);
>+  for (size_t i = 0; i < num_channels_values.size(); ++i) {
>+    if (num_channels_values[i].has_value()) {
>+      num_channels_values[i] = num_channels_values[i].value() + 1;
>+    }
>+  }
>+  RTC_CHECK_EQ(num_channels_values.size(), number_of_deltas);
>+
>+  // Delta decoding
>+  for (size_t i = 0; i < number_of_deltas; ++i) {
>+    RTC_CHECK(timestamp_ms_values[i].has_value());
>+    int64_t timestamp_ms;
>+    RTC_CHECK(ToSigned(timestamp_ms_values[i].value(), &timestamp_ms));
>+
>+    AudioEncoderRuntimeConfig runtime_config;
>+    if (bitrate_bps_values[i].has_value()) {
>+      int signed_bitrate_bps;
>+      RTC_CHECK(ToSigned(bitrate_bps_values[i].value(), &signed_bitrate_bps));
>+      runtime_config.bitrate_bps = signed_bitrate_bps;
>+    }
>+    if (frame_length_ms_values[i].has_value()) {
>+      int signed_frame_length_ms;
>+      RTC_CHECK(
>+          ToSigned(frame_length_ms_values[i].value(), &signed_frame_length_ms));
>+      runtime_config.frame_length_ms = signed_frame_length_ms;
>+    }
>+    if (uplink_packet_loss_fraction_values[i].has_value()) {
>+      float uplink_packet_loss_fraction;
>+      RTC_CHECK(ParsePacketLossFractionFromProtoFormat(
>+          rtc::checked_cast<uint32_t>(
>+              uplink_packet_loss_fraction_values[i].value()),
>+          &uplink_packet_loss_fraction));
>+      runtime_config.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
>+    }
>+    if (enable_fec_values[i].has_value()) {
>+      runtime_config.enable_fec =
>+          rtc::checked_cast<bool>(enable_fec_values[i].value());
>+    }
>+    if (enable_dtx_values[i].has_value()) {
>+      runtime_config.enable_dtx =
>+          rtc::checked_cast<bool>(enable_dtx_values[i].value());
>+    }
>+    if (num_channels_values[i].has_value()) {
>+      runtime_config.num_channels =
>+          rtc::checked_cast<size_t>(num_channels_values[i].value());
>+    }
>+    audio_network_adaptation_events_.emplace_back(1000 * timestamp_ms,
>+                                                  runtime_config);
>+  }
>+}
>+
>+void ParsedRtcEventLogNew::StoreIceCandidatePairConfig(
>+    const rtclog2::IceCandidatePairConfig& proto) {
>+  LoggedIceCandidatePairConfig ice_config;
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  ice_config.timestamp_us = proto.timestamp_ms() * 1000;
>+
>+  RTC_CHECK(proto.has_config_type());
>+  ice_config.type = GetRuntimeIceCandidatePairConfigType(proto.config_type());
>+  RTC_CHECK(proto.has_candidate_pair_id());
>+  ice_config.candidate_pair_id = proto.candidate_pair_id();
>+  RTC_CHECK(proto.has_local_candidate_type());
>+  ice_config.local_candidate_type =
>+      GetRuntimeIceCandidateType(proto.local_candidate_type());
>+  RTC_CHECK(proto.has_local_relay_protocol());
>+  ice_config.local_relay_protocol =
>+      GetRuntimeIceCandidatePairProtocol(proto.local_relay_protocol());
>+  RTC_CHECK(proto.has_local_network_type());
>+  ice_config.local_network_type =
>+      GetRuntimeIceCandidateNetworkType(proto.local_network_type());
>+  RTC_CHECK(proto.has_local_address_family());
>+  ice_config.local_address_family =
>+      GetRuntimeIceCandidatePairAddressFamily(proto.local_address_family());
>+  RTC_CHECK(proto.has_remote_candidate_type());
>+  ice_config.remote_candidate_type =
>+      GetRuntimeIceCandidateType(proto.remote_candidate_type());
>+  RTC_CHECK(proto.has_remote_address_family());
>+  ice_config.remote_address_family =
>+      GetRuntimeIceCandidatePairAddressFamily(proto.remote_address_family());
>+  RTC_CHECK(proto.has_candidate_pair_protocol());
>+  ice_config.candidate_pair_protocol =
>+      GetRuntimeIceCandidatePairProtocol(proto.candidate_pair_protocol());
>+
>+  ice_candidate_pair_configs_.push_back(ice_config);
>+
>+  // TODO(terelius): Should we delta encode this event type?
>+}
>+
>+void ParsedRtcEventLogNew::StoreIceCandidateEvent(
>+    const rtclog2::IceCandidatePairEvent& proto) {
>+  LoggedIceCandidatePairEvent ice_event;
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  ice_event.timestamp_us = proto.timestamp_ms() * 1000;
>+  RTC_CHECK(proto.has_event_type());
>+  ice_event.type = GetRuntimeIceCandidatePairEventType(proto.event_type());
>+  RTC_CHECK(proto.has_candidate_pair_id());
>+  ice_event.candidate_pair_id = proto.candidate_pair_id();
>+
>+  ice_candidate_pair_events_.push_back(ice_event);
>+
>+  // TODO(terelius): Should we delta encode this event type?
>+}
>+
>+void ParsedRtcEventLogNew::StoreVideoRecvConfig(
>+    const rtclog2::VideoRecvStreamConfig& proto) {
>+  LoggedVideoRecvConfig stream;
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  stream.timestamp_us = proto.timestamp_ms() * 1000;
>+  RTC_CHECK(proto.has_remote_ssrc());
>+  stream.config.remote_ssrc = proto.remote_ssrc();
>+  RTC_CHECK(proto.has_local_ssrc());
>+  stream.config.local_ssrc = proto.local_ssrc();
>+  if (proto.has_rtx_ssrc()) {
>+    stream.config.rtx_ssrc = proto.rtx_ssrc();
>+  }
>+  if (proto.has_header_extensions()) {
>+    stream.config.rtp_extensions =
>+        GetRuntimeRtpHeaderExtensionConfig(proto.header_extensions());
>+  }
>+  video_recv_configs_.push_back(stream);
>+}
>+
>+void ParsedRtcEventLogNew::StoreVideoSendConfig(
>+    const rtclog2::VideoSendStreamConfig& proto) {
>+  LoggedVideoSendConfig stream;
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  stream.timestamp_us = proto.timestamp_ms() * 1000;
>+  rtclog::StreamConfig config;
>+  RTC_CHECK(proto.has_ssrc());
>+  config.local_ssrc = proto.ssrc();
>+  if (proto.has_rtx_ssrc()) {
>+    config.rtx_ssrc = proto.rtx_ssrc();
>+  }
>+  if (proto.has_header_extensions()) {
>+    config.rtp_extensions =
>+        GetRuntimeRtpHeaderExtensionConfig(proto.header_extensions());
>+  }
>+  stream.configs.push_back(config);
>+  video_send_configs_.push_back(stream);
>+}
>+
>+void ParsedRtcEventLogNew::StoreAudioRecvConfig(
>+    const rtclog2::AudioRecvStreamConfig& proto) {
>+  LoggedAudioRecvConfig stream;
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  stream.timestamp_us = proto.timestamp_ms() * 1000;
>+  RTC_CHECK(proto.has_remote_ssrc());
>+  stream.config.remote_ssrc = proto.remote_ssrc();
>+  RTC_CHECK(proto.has_local_ssrc());
>+  stream.config.local_ssrc = proto.local_ssrc();
>+  if (proto.has_header_extensions()) {
>+    stream.config.rtp_extensions =
>+        GetRuntimeRtpHeaderExtensionConfig(proto.header_extensions());
>+  }
>+  audio_recv_configs_.push_back(stream);
>+}
>+
>+void ParsedRtcEventLogNew::StoreAudioSendConfig(
>+    const rtclog2::AudioSendStreamConfig& proto) {
>+  LoggedAudioSendConfig stream;
>+  RTC_CHECK(proto.has_timestamp_ms());
>+  stream.timestamp_us = proto.timestamp_ms() * 1000;
>+  RTC_CHECK(proto.has_ssrc());
>+  stream.config.local_ssrc = proto.ssrc();
>+  if (proto.has_header_extensions()) {
>+    stream.config.rtp_extensions =
>+        GetRuntimeRtpHeaderExtensionConfig(proto.header_extensions());
>+  }
>+  audio_send_configs_.push_back(stream);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.h
>index 5ab0c15b92a79e56c4c09cdf0d4b04eef12f0676..50e3b9922de26adc8de8000d0ea6f4ec4cbb5b31 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.h
>@@ -25,7 +25,7 @@
> #include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "logging/rtc_event_log/rtc_stream_config.h"
>-#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
>@@ -39,8 +39,10 @@
> RTC_PUSH_IGNORING_WUNDEF()
> #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
> #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
>+#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log2.pb.h"
> #else
> #include "logging/rtc_event_log/rtc_event_log.pb.h"
>+#include "logging/rtc_event_log/rtc_event_log2.pb.h"
> #endif
> RTC_POP_IGNORING_WUNDEF()
> 
>@@ -55,70 +57,136 @@ struct AudioEncoderRuntimeConfig;
> // considered to outweigh the added memory and runtime overhead incurred by
> // adding a vptr.
> struct LoggedAlrStateEvent {
>-  int64_t timestamp_us;
>-  bool in_alr;
>+  LoggedAlrStateEvent() = default;
>+  LoggedAlrStateEvent(int64_t timestamp_us, bool in_alr)
>+      : timestamp_us(timestamp_us), in_alr(in_alr) {}
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  bool in_alr;
> };
> 
> struct LoggedAudioPlayoutEvent {
>-  int64_t timestamp_us;
>-  uint32_t ssrc;
>+  LoggedAudioPlayoutEvent() = default;
>+  LoggedAudioPlayoutEvent(int64_t timestamp_us, uint32_t ssrc)
>+      : timestamp_us(timestamp_us), ssrc(ssrc) {}
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  uint32_t ssrc;
> };
> 
> struct LoggedAudioNetworkAdaptationEvent {
>-  int64_t timestamp_us;
>-  AudioEncoderRuntimeConfig config;
>+  LoggedAudioNetworkAdaptationEvent() = default;
>+  LoggedAudioNetworkAdaptationEvent(int64_t timestamp_us,
>+                                    const AudioEncoderRuntimeConfig& config)
>+      : timestamp_us(timestamp_us), config(config) {}
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  AudioEncoderRuntimeConfig config;
> };
> 
> struct LoggedBweDelayBasedUpdate {
>+  LoggedBweDelayBasedUpdate() = default;
>+  LoggedBweDelayBasedUpdate(int64_t timestamp_us,
>+                            int32_t bitrate_bps,
>+                            BandwidthUsage detector_state)
>+      : timestamp_us(timestamp_us),
>+        bitrate_bps(bitrate_bps),
>+        detector_state(detector_state) {}
>+
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>   int64_t timestamp_us;
>   int32_t bitrate_bps;
>   BandwidthUsage detector_state;
>-  int64_t log_time_us() const { return timestamp_us; }
>-  int64_t log_time_ms() const { return timestamp_us / 1000; }
> };
> 
> struct LoggedBweLossBasedUpdate {
>+  LoggedBweLossBasedUpdate() = default;
>+  LoggedBweLossBasedUpdate(int64_t timestamp_us,
>+                           int32_t bitrate_bps,
>+                           uint8_t fraction_lost,
>+                           int32_t expected_packets)
>+      : timestamp_us(timestamp_us),
>+        bitrate_bps(bitrate_bps),
>+        fraction_lost(fraction_lost),
>+        expected_packets(expected_packets) {}
>+
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>   int64_t timestamp_us;
>   int32_t bitrate_bps;
>   uint8_t fraction_lost;
>   int32_t expected_packets;
>-  int64_t log_time_us() const { return timestamp_us; }
>-  int64_t log_time_ms() const { return timestamp_us / 1000; }
> };
> 
> struct LoggedBweProbeClusterCreatedEvent {
>+  LoggedBweProbeClusterCreatedEvent() = default;
>+  LoggedBweProbeClusterCreatedEvent(int64_t timestamp_us,
>+                                    int32_t id,
>+                                    int32_t bitrate_bps,
>+                                    uint32_t min_packets,
>+                                    uint32_t min_bytes)
>+      : timestamp_us(timestamp_us),
>+        id(id),
>+        bitrate_bps(bitrate_bps),
>+        min_packets(min_packets),
>+        min_bytes(min_bytes) {}
>+
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>   int64_t timestamp_us;
>   int32_t id;
>   int32_t bitrate_bps;
>   uint32_t min_packets;
>   uint32_t min_bytes;
>-  int64_t log_time_us() const { return timestamp_us; }
>-  int64_t log_time_ms() const { return timestamp_us / 1000; }
> };
> 
> struct LoggedBweProbeSuccessEvent {
>+  LoggedBweProbeSuccessEvent() = default;
>+  LoggedBweProbeSuccessEvent(int64_t timestamp_us,
>+                             int32_t id,
>+                             int32_t bitrate_bps)
>+      : timestamp_us(timestamp_us), id(id), bitrate_bps(bitrate_bps) {}
>+
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>   int64_t timestamp_us;
>   int32_t id;
>   int32_t bitrate_bps;
>-  int64_t log_time_us() const { return timestamp_us; }
>-  int64_t log_time_ms() const { return timestamp_us / 1000; }
> };
> 
> struct LoggedBweProbeFailureEvent {
>+  LoggedBweProbeFailureEvent() = default;
>+  LoggedBweProbeFailureEvent(int64_t timestamp_us,
>+                             int32_t id,
>+                             ProbeFailureReason failure_reason)
>+      : timestamp_us(timestamp_us), id(id), failure_reason(failure_reason) {}
>+
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>   int64_t timestamp_us;
>   int32_t id;
>   ProbeFailureReason failure_reason;
>-  int64_t log_time_us() const { return timestamp_us; }
>-  int64_t log_time_ms() const { return timestamp_us / 1000; }
> };
> 
> struct LoggedIceCandidatePairConfig {
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>   int64_t timestamp_us;
>   IceCandidatePairConfigType type;
>   uint32_t candidate_pair_id;
>@@ -129,16 +197,23 @@ struct LoggedIceCandidatePairConfig {
>   IceCandidateType remote_candidate_type;
>   IceCandidatePairAddressFamily remote_address_family;
>   IceCandidatePairProtocol candidate_pair_protocol;
>-  int64_t log_time_us() const { return timestamp_us; }
>-  int64_t log_time_ms() const { return timestamp_us / 1000; }
> };
> 
> struct LoggedIceCandidatePairEvent {
>+  LoggedIceCandidatePairEvent() = default;
>+  LoggedIceCandidatePairEvent(int64_t timestamp_us,
>+                              IceCandidatePairEventType type,
>+                              uint32_t candidate_pair_id)
>+      : timestamp_us(timestamp_us),
>+        type(type),
>+        candidate_pair_id(candidate_pair_id) {}
>+
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>   int64_t timestamp_us;
>   IceCandidatePairEventType type;
>   uint32_t candidate_pair_id;
>-  int64_t log_time_us() const { return timestamp_us; }
>-  int64_t log_time_ms() const { return timestamp_us / 1000; }
> };
> 
> struct LoggedRtpPacket {
>@@ -150,13 +225,15 @@ struct LoggedRtpPacket {
>         header(header),
>         header_length(header_length),
>         total_length(total_length) {}
>+
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>   int64_t timestamp_us;
>   // TODO(terelius): This allocates space for 15 CSRCs even if none are used.
>   RTPHeader header;
>   size_t header_length;
>   size_t total_length;
>-  int64_t log_time_us() const { return timestamp_us; }
>-  int64_t log_time_ms() const { return timestamp_us / 1000; }
> };
> 
> struct LoggedRtpPacketIncoming {
>@@ -165,9 +242,10 @@ struct LoggedRtpPacketIncoming {
>                           size_t header_length,
>                           size_t total_length)
>       : rtp(timestamp_us, header, header_length, total_length) {}
>-  LoggedRtpPacket rtp;
>   int64_t log_time_us() const { return rtp.timestamp_us; }
>   int64_t log_time_ms() const { return rtp.timestamp_us / 1000; }
>+
>+  LoggedRtpPacket rtp;
> };
> 
> struct LoggedRtpPacketOutgoing {
>@@ -176,21 +254,25 @@ struct LoggedRtpPacketOutgoing {
>                           size_t header_length,
>                           size_t total_length)
>       : rtp(timestamp_us, header, header_length, total_length) {}
>-  LoggedRtpPacket rtp;
>   int64_t log_time_us() const { return rtp.timestamp_us; }
>   int64_t log_time_ms() const { return rtp.timestamp_us / 1000; }
>+
>+  LoggedRtpPacket rtp;
> };
> 
> struct LoggedRtcpPacket {
>   LoggedRtcpPacket(uint64_t timestamp_us,
>                    const uint8_t* packet,
>                    size_t total_length);
>+  LoggedRtcpPacket(uint64_t timestamp_us, const std::string& packet);
>   LoggedRtcpPacket(const LoggedRtcpPacket&);
>   ~LoggedRtcpPacket();
>-  int64_t timestamp_us;
>-  std::vector<uint8_t> raw_data;
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  std::vector<uint8_t> raw_data;
> };
> 
> struct LoggedRtcpPacketIncoming {
>@@ -198,9 +280,13 @@ struct LoggedRtcpPacketIncoming {
>                            const uint8_t* packet,
>                            size_t total_length)
>       : rtcp(timestamp_us, packet, total_length) {}
>-  LoggedRtcpPacket rtcp;
>+  LoggedRtcpPacketIncoming(uint64_t timestamp_us, const std::string& packet)
>+      : rtcp(timestamp_us, packet) {}
>+
>   int64_t log_time_us() const { return rtcp.timestamp_us; }
>   int64_t log_time_ms() const { return rtcp.timestamp_us / 1000; }
>+
>+  LoggedRtcpPacket rtcp;
> };
> 
> struct LoggedRtcpPacketOutgoing {
>@@ -208,97 +294,150 @@ struct LoggedRtcpPacketOutgoing {
>                            const uint8_t* packet,
>                            size_t total_length)
>       : rtcp(timestamp_us, packet, total_length) {}
>-  LoggedRtcpPacket rtcp;
>+  LoggedRtcpPacketOutgoing(uint64_t timestamp_us, const std::string& packet)
>+      : rtcp(timestamp_us, packet) {}
>+
>   int64_t log_time_us() const { return rtcp.timestamp_us; }
>   int64_t log_time_ms() const { return rtcp.timestamp_us / 1000; }
>+
>+  LoggedRtcpPacket rtcp;
> };
> 
> struct LoggedRtcpPacketReceiverReport {
>-  int64_t timestamp_us;
>-  rtcp::ReceiverReport rr;
>+  LoggedRtcpPacketReceiverReport() = default;
>+  LoggedRtcpPacketReceiverReport(int64_t timestamp_us,
>+                                 const rtcp::ReceiverReport& rr)
>+      : timestamp_us(timestamp_us), rr(rr) {}
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  rtcp::ReceiverReport rr;
> };
> 
> struct LoggedRtcpPacketSenderReport {
>-  int64_t timestamp_us;
>-  rtcp::SenderReport sr;
>+  LoggedRtcpPacketSenderReport() = default;
>+  LoggedRtcpPacketSenderReport(int64_t timestamp_us,
>+                               const rtcp::SenderReport& sr)
>+      : timestamp_us(timestamp_us), sr(sr) {}
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  rtcp::SenderReport sr;
> };
> 
> struct LoggedRtcpPacketRemb {
>-  int64_t timestamp_us;
>-  rtcp::Remb remb;
>+  LoggedRtcpPacketRemb() = default;
>+  LoggedRtcpPacketRemb(int64_t timestamp_us, const rtcp::Remb& remb)
>+      : timestamp_us(timestamp_us), remb(remb) {}
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  rtcp::Remb remb;
> };
> 
> struct LoggedRtcpPacketNack {
>-  int64_t timestamp_us;
>-  rtcp::Nack nack;
>+  LoggedRtcpPacketNack() = default;
>+  LoggedRtcpPacketNack(int64_t timestamp_us, const rtcp::Nack& nack)
>+      : timestamp_us(timestamp_us), nack(nack) {}
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  rtcp::Nack nack;
> };
> 
> struct LoggedRtcpPacketTransportFeedback {
>-  int64_t timestamp_us;
>-  rtcp::TransportFeedback transport_feedback;
>+  LoggedRtcpPacketTransportFeedback() = default;
>+  LoggedRtcpPacketTransportFeedback(
>+      int64_t timestamp_us,
>+      const rtcp::TransportFeedback& transport_feedback)
>+      : timestamp_us(timestamp_us), transport_feedback(transport_feedback) {}
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  rtcp::TransportFeedback transport_feedback;
> };
> 
> struct LoggedStartEvent {
>   explicit LoggedStartEvent(int64_t timestamp_us)
>-      : timestamp_us(timestamp_us) {}
>-  int64_t timestamp_us;
>+      : LoggedStartEvent(timestamp_us, timestamp_us / 1000) {}
>+
>+  LoggedStartEvent(int64_t timestamp_us, int64_t utc_start_time_ms)
>+      : timestamp_us(timestamp_us), utc_start_time_ms(utc_start_time_ms) {}
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  int64_t utc_start_time_ms;
> };
> 
> struct LoggedStopEvent {
>   explicit LoggedStopEvent(int64_t timestamp_us) : timestamp_us(timestamp_us) {}
>-  int64_t timestamp_us;
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
> };
> 
> struct LoggedAudioRecvConfig {
>+  LoggedAudioRecvConfig() = default;
>   LoggedAudioRecvConfig(int64_t timestamp_us, const rtclog::StreamConfig config)
>       : timestamp_us(timestamp_us), config(config) {}
>-  int64_t timestamp_us;
>-  rtclog::StreamConfig config;
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  rtclog::StreamConfig config;
> };
> 
> struct LoggedAudioSendConfig {
>+  LoggedAudioSendConfig() = default;
>   LoggedAudioSendConfig(int64_t timestamp_us, const rtclog::StreamConfig config)
>       : timestamp_us(timestamp_us), config(config) {}
>-  int64_t timestamp_us;
>-  rtclog::StreamConfig config;
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  rtclog::StreamConfig config;
> };
> 
> struct LoggedVideoRecvConfig {
>+  LoggedVideoRecvConfig() = default;
>   LoggedVideoRecvConfig(int64_t timestamp_us, const rtclog::StreamConfig config)
>       : timestamp_us(timestamp_us), config(config) {}
>-  int64_t timestamp_us;
>-  rtclog::StreamConfig config;
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  rtclog::StreamConfig config;
> };
> 
> struct LoggedVideoSendConfig {
>+  LoggedVideoSendConfig();
>   LoggedVideoSendConfig(int64_t timestamp_us,
>                         const std::vector<rtclog::StreamConfig>& configs);
>   LoggedVideoSendConfig(const LoggedVideoSendConfig&);
>   ~LoggedVideoSendConfig();
>-  int64_t timestamp_us;
>-  std::vector<rtclog::StreamConfig> configs;
>+
>   int64_t log_time_us() const { return timestamp_us; }
>   int64_t log_time_ms() const { return timestamp_us / 1000; }
>+
>+  int64_t timestamp_us;
>+  std::vector<rtclog::StreamConfig> configs;
> };
> 
> template <typename T>
>@@ -477,30 +616,6 @@ class ParsedRtcEventLogNew {
>   friend class RtcEventLogTestHelper;
> 
>  public:
>-  ~ParsedRtcEventLogNew();
>-
>-  enum class EventType {
>-    UNKNOWN_EVENT = 0,
>-    LOG_START = 1,
>-    LOG_END = 2,
>-    RTP_EVENT = 3,
>-    RTCP_EVENT = 4,
>-    AUDIO_PLAYOUT_EVENT = 5,
>-    LOSS_BASED_BWE_UPDATE = 6,
>-    DELAY_BASED_BWE_UPDATE = 7,
>-    VIDEO_RECEIVER_CONFIG_EVENT = 8,
>-    VIDEO_SENDER_CONFIG_EVENT = 9,
>-    AUDIO_RECEIVER_CONFIG_EVENT = 10,
>-    AUDIO_SENDER_CONFIG_EVENT = 11,
>-    AUDIO_NETWORK_ADAPTATION_EVENT = 16,
>-    BWE_PROBE_CLUSTER_CREATED_EVENT = 17,
>-    BWE_PROBE_FAILURE_EVENT = 18,
>-    BWE_PROBE_SUCCESS_EVENT = 19,
>-    ALR_STATE_EVENT = 20,
>-    ICE_CANDIDATE_PAIR_CONFIG = 21,
>-    ICE_CANDIDATE_PAIR_EVENT = 22,
>-  };
>-
>   enum class MediaType { ANY, AUDIO, VIDEO, DATA };
>   enum class UnconfiguredHeaderExtensions {
>     kDontParse,
>@@ -541,6 +656,8 @@ class ParsedRtcEventLogNew {
>       UnconfiguredHeaderExtensions parse_unconfigured_header_extensions =
>           UnconfiguredHeaderExtensions::kDontParse);
> 
>+  ~ParsedRtcEventLogNew();
>+
>   // Clears previously parsed events and resets the ParsedRtcEventLogNew to an
>   // empty state.
>   void Clear();
>@@ -555,111 +672,8 @@ class ParsedRtcEventLogNew {
>   bool ParseStream(
>       std::istream& stream);  // no-presubmit-check TODO(webrtc:8982)
> 
>-  // Returns the number of events in an EventStream.
>-  size_t GetNumberOfEvents() const;
>-
>-  // Reads the arrival timestamp (in microseconds) from a rtclog::Event.
>-  int64_t GetTimestamp(size_t index) const;
>-  int64_t GetTimestamp(const rtclog::Event& event) const;
>-
>-  // Reads the event type of the rtclog::Event at |index|.
>-  EventType GetEventType(size_t index) const;
>-  EventType GetEventType(const rtclog::Event& event) const;
>-
>-  // Reads the header, direction, header length and packet length from the RTP
>-  // event at |index|, and stores the values in the corresponding output
>-  // parameters. Each output parameter can be set to nullptr if that value
>-  // isn't needed.
>-  // NB: The header must have space for at least IP_PACKET_SIZE bytes.
>-  // Returns: a pointer to a header extensions map acquired from parsing
>-  // corresponding Audio/Video Sender/Receiver config events.
>-  // Warning: if the same SSRC is reused by both video and audio streams during
>-  // call, extensions maps may be incorrect (the last one would be returned).
>-  const webrtc::RtpHeaderExtensionMap* GetRtpHeader(
>-      size_t index,
>-      PacketDirection* incoming,
>-      uint8_t* header,
>-      size_t* header_length,
>-      size_t* total_length,
>-      int* probe_cluster_id) const;
>-  const webrtc::RtpHeaderExtensionMap* GetRtpHeader(
>-      const rtclog::Event& event,
>-      PacketDirection* incoming,
>-      uint8_t* header,
>-      size_t* header_length,
>-      size_t* total_length,
>-      int* probe_cluster_id) const;
>-
>-  // Reads packet, direction and packet length from the RTCP event at |index|,
>-  // and stores the values in the corresponding output parameters.
>-  // Each output parameter can be set to nullptr if that value isn't needed.
>-  // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
>-  void GetRtcpPacket(size_t index,
>-                     PacketDirection* incoming,
>-                     uint8_t* packet,
>-                     size_t* length) const;
>-  void GetRtcpPacket(const rtclog::Event& event,
>-                     PacketDirection* incoming,
>-                     uint8_t* packet,
>-                     size_t* length) const;
>-
>-  // Reads a video receive config event to a StreamConfig struct.
>-  // Only the fields that are stored in the protobuf will be written.
>-  rtclog::StreamConfig GetVideoReceiveConfig(size_t index) const;
>-
>-  // Reads a video send config event to a StreamConfig struct. If the proto
>-  // contains multiple SSRCs and RTX SSRCs (this used to be the case for
>-  // simulcast streams) then we return one StreamConfig per SSRC,RTX_SSRC pair.
>-  // Only the fields that are stored in the protobuf will be written.
>-  std::vector<rtclog::StreamConfig> GetVideoSendConfig(size_t index) const;
>-
>-  // Reads a audio receive config event to a StreamConfig struct.
>-  // Only the fields that are stored in the protobuf will be written.
>-  rtclog::StreamConfig GetAudioReceiveConfig(size_t index) const;
>-
>-  // Reads a config event to a StreamConfig struct.
>-  // Only the fields that are stored in the protobuf will be written.
>-  rtclog::StreamConfig GetAudioSendConfig(size_t index) const;
>-
>-  // Reads the SSRC from the audio playout event at |index|. The SSRC is stored
>-  // in the output parameter ssrc. The output parameter can be set to nullptr
>-  // and in that case the function only asserts that the event is well formed.
>-  LoggedAudioPlayoutEvent GetAudioPlayout(size_t index) const;
>-
>-  // Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
>-  // expected packets from the loss based BWE event at |index| and stores the
>-  // values in
>-  // the corresponding output parameters. Each output parameter can be set to
>-  // nullptr if that
>-  // value isn't needed.
>-  LoggedBweLossBasedUpdate GetLossBasedBweUpdate(size_t index) const;
>-
>-  // Reads bitrate and detector_state from the delay based BWE event at |index|
>-  // and stores the values in the corresponding output parameters. Each output
>-  // parameter can be set to nullptr if that
>-  // value isn't needed.
>-  LoggedBweDelayBasedUpdate GetDelayBasedBweUpdate(size_t index) const;
>-
>-  // Reads a audio network adaptation event to a (non-NULL)
>-  // AudioEncoderRuntimeConfig struct. Only the fields that are
>-  // stored in the protobuf will be written.
>-  LoggedAudioNetworkAdaptationEvent GetAudioNetworkAdaptation(
>-      size_t index) const;
>-
>-  LoggedBweProbeClusterCreatedEvent GetBweProbeClusterCreated(
>-      size_t index) const;
>-
>-  LoggedBweProbeFailureEvent GetBweProbeFailure(size_t index) const;
>-  LoggedBweProbeSuccessEvent GetBweProbeSuccess(size_t index) const;
>-
>   MediaType GetMediaType(uint32_t ssrc, PacketDirection direction) const;
> 
>-  LoggedAlrStateEvent GetAlrState(size_t index) const;
>-
>-  LoggedIceCandidatePairConfig GetIceCandidatePairConfig(size_t index) const;
>-
>-  LoggedIceCandidatePairEvent GetIceCandidatePairEvent(size_t index) const;
>-
>   // Configured SSRCs.
>   const std::set<uint32_t>& incoming_rtx_ssrcs() const {
>     return incoming_rtx_ssrcs_;
>@@ -841,7 +855,36 @@ class ParsedRtcEventLogNew {
>   bool ParseStreamInternal(
>       std::istream& stream);  // no-presubmit-check TODO(webrtc:8982)
> 
>-  void StoreParsedEvent(const rtclog::Event& event);
>+  void StoreParsedLegacyEvent(const rtclog::Event& event);
>+
>+  // Reads the arrival timestamp (in microseconds) from a rtclog::Event.
>+  int64_t GetTimestamp(const rtclog::Event& event) const;
>+
>+  // Reads the header, direction, header length and packet length from the RTP
>+  // event at |index|, and stores the values in the corresponding output
>+  // parameters. Each output parameter can be set to nullptr if that value
>+  // isn't needed.
>+  // NB: The header must have space for at least IP_PACKET_SIZE bytes.
>+  // Returns: a pointer to a header extensions map acquired from parsing
>+  // corresponding Audio/Video Sender/Receiver config events.
>+  // Warning: if the same SSRC is reused by both video and audio streams during
>+  // call, extensions maps may be incorrect (the last one would be returned).
>+  const webrtc::RtpHeaderExtensionMap* GetRtpHeader(
>+      const rtclog::Event& event,
>+      PacketDirection* incoming,
>+      uint8_t* header,
>+      size_t* header_length,
>+      size_t* total_length,
>+      int* probe_cluster_id) const;
>+
>+  // Reads packet, direction and packet length from the RTCP event at |index|,
>+  // and stores the values in the corresponding output parameters.
>+  // Each output parameter can be set to nullptr if that value isn't needed.
>+  // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
>+  void GetRtcpPacket(const rtclog::Event& event,
>+                     PacketDirection* incoming,
>+                     uint8_t* packet,
>+                     size_t* length) const;
> 
>   rtclog::StreamConfig GetVideoReceiveConfig(const rtclog::Event& event) const;
>   std::vector<rtclog::StreamConfig> GetVideoSendConfig(
>@@ -873,7 +916,31 @@ class ParsedRtcEventLogNew {
>   LoggedIceCandidatePairEvent GetIceCandidatePairEvent(
>       const rtclog::Event& event) const;
> 
>-  std::vector<rtclog::Event> events_;
>+  // Parsing functions for new format.
>+  void StoreParsedNewFormatEvent(const rtclog2::EventStream& event);
>+  void StoreIncomingRtpPackets(const rtclog2::IncomingRtpPackets& proto);
>+  void StoreOutgoingRtpPackets(const rtclog2::OutgoingRtpPackets& proto);
>+  void StoreIncomingRtcpPackets(const rtclog2::IncomingRtcpPackets& proto);
>+  void StoreOutgoingRtcpPackets(const rtclog2::OutgoingRtcpPackets& proto);
>+  void StoreAudioPlayoutEvent(const rtclog2::AudioPlayoutEvents& proto);
>+  void StoreStartEvent(const rtclog2::BeginLogEvent& proto);
>+  void StoreStopEvent(const rtclog2::EndLogEvent& proto);
>+  void StoreBweLossBasedUpdate(const rtclog2::LossBasedBweUpdates& proto);
>+  void StoreBweDelayBasedUpdate(const rtclog2::DelayBasedBweUpdates& proto);
>+  void StoreAudioNetworkAdaptationEvent(
>+      const rtclog2::AudioNetworkAdaptations& proto);
>+  void StoreBweProbeClusterCreated(const rtclog2::BweProbeCluster& proto);
>+  void StoreBweProbeSuccessEvent(const rtclog2::BweProbeResultSuccess& proto);
>+  void StoreBweProbeFailureEvent(const rtclog2::BweProbeResultFailure& proto);
>+  void StoreAlrStateEvent(const rtclog2::AlrState& proto);
>+  void StoreIceCandidatePairConfig(
>+      const rtclog2::IceCandidatePairConfig& proto);
>+  void StoreIceCandidateEvent(const rtclog2::IceCandidatePairEvent& proto);
>+  void StoreAudioRecvConfig(const rtclog2::AudioRecvStreamConfig& proto);
>+  void StoreAudioSendConfig(const rtclog2::AudioSendStreamConfig& proto);
>+  void StoreVideoRecvConfig(const rtclog2::VideoRecvStreamConfig& proto);
>+  void StoreVideoSendConfig(const rtclog2::VideoSendStreamConfig& proto);
>+  // End of new parsing functions.
> 
>   struct Stream {
>     Stream(uint32_t ssrc,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
>index e238f84884618ac19073e7304e01ee1746e4bc91..f7b7898993f7c92b9157d9a7918e924d8110cb87 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
>@@ -101,13 +101,16 @@ struct EventCounts {
> };
> 
> class RtcEventLogSession
>-    : public ::testing::TestWithParam<std::tuple<uint64_t, int64_t>> {
>+    : public ::testing::TestWithParam<
>+          std::tuple<uint64_t, int64_t, RtcEventLog::EncodingType>> {
>  public:
>   RtcEventLogSession()
>       : seed_(std::get<0>(GetParam())),
>         prng_(seed_),
>+        output_period_ms_(std::get<1>(GetParam())),
>+        encoding_type_(std::get<2>(GetParam())),
>         gen_(seed_ * 880001UL),
>-        output_period_ms_(std::get<1>(GetParam())) {
>+        verifier_(encoding_type_) {
>     clock_.SetTimeMicros(prng_.Rand<uint32_t>());
>     // Find the name of the current test, in order to use it as a temporary
>     // filename.
>@@ -166,12 +169,15 @@ class RtcEventLogSession
>   std::vector<std::unique_ptr<RtcEventRtcpPacketOutgoing>> outgoing_rtcp_list_;
> 
>   int64_t start_time_us_;
>+  int64_t utc_start_time_us_;
>   int64_t stop_time_us_;
> 
>   const uint64_t seed_;
>   Random prng_;
>+  const int64_t output_period_ms_;
>+  const RtcEventLog::EncodingType encoding_type_;
>   test::EventGenerator gen_;
>-  int64_t output_period_ms_;
>+  test::EventVerifier verifier_;
>   rtc::ScopedFakeClock clock_;
>   std::string temp_filename_;
> };
>@@ -284,8 +290,7 @@ void RtcEventLogSession::WriteLog(EventCounts count,
>   // Maybe always use the ScopedFakeClock, but conditionally SleepMs()?
> 
>   // The log file will be flushed to disk when the event_log goes out of scope.
>-  std::unique_ptr<RtcEventLog> event_log(
>-      RtcEventLog::Create(RtcEventLog::EncodingType::Legacy));
>+  std::unique_ptr<RtcEventLog> event_log(RtcEventLog::Create(encoding_type_));
> 
>   // We can't send or receive packets without configured streams.
>   RTC_CHECK_GE(count.video_recv_streams, 1);
>@@ -306,6 +311,7 @@ void RtcEventLogSession::WriteLog(EventCounts count,
>           absl::make_unique<RtcEventLogOutputFile>(temp_filename_, 10000000),
>           output_period_ms_);
>       start_time_us_ = rtc::TimeMicros();
>+      utc_start_time_us_ = rtc::TimeUTCMicros();
>     }
> 
>     clock_.AdvanceTimeMicros(prng_.Rand(20) * 1000);
>@@ -465,19 +471,18 @@ void RtcEventLogSession::ReadAndVerifyLog() {
>   // Start and stop events.
>   auto& parsed_start_log_events = parsed_log.start_log_events();
>   ASSERT_EQ(parsed_start_log_events.size(), static_cast<size_t>(1));
>-  EXPECT_TRUE(
>-      test::VerifyLoggedStartEvent(start_time_us_, parsed_start_log_events[0]));
>+  verifier_.VerifyLoggedStartEvent(start_time_us_, utc_start_time_us_,
>+                                   parsed_start_log_events[0]);
> 
>   auto& parsed_stop_log_events = parsed_log.stop_log_events();
>   ASSERT_EQ(parsed_stop_log_events.size(), static_cast<size_t>(1));
>-  EXPECT_TRUE(
>-      test::VerifyLoggedStopEvent(stop_time_us_, parsed_stop_log_events[0]));
>+  verifier_.VerifyLoggedStopEvent(stop_time_us_, parsed_stop_log_events[0]);
> 
>   auto& parsed_alr_state_events = parsed_log.alr_state_events();
>   ASSERT_EQ(parsed_alr_state_events.size(), alr_state_list_.size());
>   for (size_t i = 0; i < parsed_alr_state_events.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedAlrStateEvent(*alr_state_list_[i],
>-                                                parsed_alr_state_events[i]));
>+    verifier_.VerifyLoggedAlrStateEvent(*alr_state_list_[i],
>+                                        parsed_alr_state_events[i]);
>   }
> 
>   const auto& parsed_audio_playout_map = parsed_log.audio_playout_events();
>@@ -488,8 +493,8 @@ void RtcEventLogSession::ReadAndVerifyLog() {
>     const auto& audio_playout_stream = audio_playout_map_[ssrc];
>     ASSERT_EQ(parsed_audio_playout_stream.size(), audio_playout_stream.size());
>     for (size_t i = 0; i < parsed_audio_playout_map.size(); i++) {
>-      EXPECT_TRUE(test::VerifyLoggedAudioPlayoutEvent(
>-          *audio_playout_stream[i], parsed_audio_playout_stream[i]));
>+      verifier_.VerifyLoggedAudioPlayoutEvent(*audio_playout_stream[i],
>+                                              parsed_audio_playout_stream[i]);
>     }
>   }
> 
>@@ -498,22 +503,22 @@ void RtcEventLogSession::ReadAndVerifyLog() {
>   ASSERT_EQ(parsed_audio_network_adaptation_events.size(),
>             ana_configs_list_.size());
>   for (size_t i = 0; i < parsed_audio_network_adaptation_events.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedAudioNetworkAdaptationEvent(
>-        *ana_configs_list_[i], parsed_audio_network_adaptation_events[i]));
>+    verifier_.VerifyLoggedAudioNetworkAdaptationEvent(
>+        *ana_configs_list_[i], parsed_audio_network_adaptation_events[i]);
>   }
> 
>   auto& parsed_bwe_delay_updates = parsed_log.bwe_delay_updates();
>   ASSERT_EQ(parsed_bwe_delay_updates.size(), bwe_delay_list_.size());
>   for (size_t i = 0; i < parsed_bwe_delay_updates.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedBweDelayBasedUpdate(
>-        *bwe_delay_list_[i], parsed_bwe_delay_updates[i]));
>+    verifier_.VerifyLoggedBweDelayBasedUpdate(*bwe_delay_list_[i],
>+                                              parsed_bwe_delay_updates[i]);
>   }
> 
>   auto& parsed_bwe_loss_updates = parsed_log.bwe_loss_updates();
>   ASSERT_EQ(parsed_bwe_loss_updates.size(), bwe_loss_list_.size());
>   for (size_t i = 0; i < parsed_bwe_loss_updates.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedBweLossBasedUpdate(
>-        *bwe_loss_list_[i], parsed_bwe_loss_updates[i]));
>+    verifier_.VerifyLoggedBweLossBasedUpdate(*bwe_loss_list_[i],
>+                                             parsed_bwe_loss_updates[i]);
>   }
> 
>   auto& parsed_bwe_probe_cluster_created_events =
>@@ -521,30 +526,30 @@ void RtcEventLogSession::ReadAndVerifyLog() {
>   ASSERT_EQ(parsed_bwe_probe_cluster_created_events.size(),
>             probe_creation_list_.size());
>   for (size_t i = 0; i < parsed_bwe_probe_cluster_created_events.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedBweProbeClusterCreatedEvent(
>-        *probe_creation_list_[i], parsed_bwe_probe_cluster_created_events[i]));
>+    verifier_.VerifyLoggedBweProbeClusterCreatedEvent(
>+        *probe_creation_list_[i], parsed_bwe_probe_cluster_created_events[i]);
>   }
> 
>   auto& parsed_bwe_probe_failure_events = parsed_log.bwe_probe_failure_events();
>   ASSERT_EQ(parsed_bwe_probe_failure_events.size(), probe_failure_list_.size());
>   for (size_t i = 0; i < parsed_bwe_probe_failure_events.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedBweProbeFailureEvent(
>-        *probe_failure_list_[i], parsed_bwe_probe_failure_events[i]));
>+    verifier_.VerifyLoggedBweProbeFailureEvent(
>+        *probe_failure_list_[i], parsed_bwe_probe_failure_events[i]);
>   }
> 
>   auto& parsed_bwe_probe_success_events = parsed_log.bwe_probe_success_events();
>   ASSERT_EQ(parsed_bwe_probe_success_events.size(), probe_success_list_.size());
>   for (size_t i = 0; i < parsed_bwe_probe_success_events.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedBweProbeSuccessEvent(
>-        *probe_success_list_[i], parsed_bwe_probe_success_events[i]));
>+    verifier_.VerifyLoggedBweProbeSuccessEvent(
>+        *probe_success_list_[i], parsed_bwe_probe_success_events[i]);
>   }
> 
>   auto& parsed_ice_candidate_pair_configs =
>       parsed_log.ice_candidate_pair_configs();
>   ASSERT_EQ(parsed_ice_candidate_pair_configs.size(), ice_config_list_.size());
>   for (size_t i = 0; i < parsed_ice_candidate_pair_configs.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedIceCandidatePairConfig(
>-        *ice_config_list_[i], parsed_ice_candidate_pair_configs[i]));
>+    verifier_.VerifyLoggedIceCandidatePairConfig(
>+        *ice_config_list_[i], parsed_ice_candidate_pair_configs[i]);
>   }
> 
>   auto& parsed_ice_candidate_pair_events =
>@@ -552,8 +557,8 @@ void RtcEventLogSession::ReadAndVerifyLog() {
>   ASSERT_EQ(parsed_ice_candidate_pair_events.size(),
>             parsed_ice_candidate_pair_events.size());
>   for (size_t i = 0; i < parsed_ice_candidate_pair_events.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedIceCandidatePairEvent(
>-        *ice_event_list_[i], parsed_ice_candidate_pair_events[i]));
>+    verifier_.VerifyLoggedIceCandidatePairEvent(
>+        *ice_event_list_[i], parsed_ice_candidate_pair_events[i]);
>   }
> 
>   auto& parsed_incoming_rtp_packets_by_ssrc =
>@@ -566,8 +571,8 @@ void RtcEventLogSession::ReadAndVerifyLog() {
>     const auto& rtp_stream = incoming_rtp_map_[ssrc];
>     ASSERT_EQ(parsed_rtp_stream.size(), rtp_stream.size());
>     for (size_t i = 0; i < parsed_rtp_stream.size(); i++) {
>-      EXPECT_TRUE(test::VerifyLoggedRtpPacketIncoming(*rtp_stream[i],
>-                                                      parsed_rtp_stream[i]));
>+      verifier_.VerifyLoggedRtpPacketIncoming(*rtp_stream[i],
>+                                              parsed_rtp_stream[i]);
>     }
>   }
> 
>@@ -581,47 +586,47 @@ void RtcEventLogSession::ReadAndVerifyLog() {
>     const auto& rtp_stream = outgoing_rtp_map_[ssrc];
>     ASSERT_EQ(parsed_rtp_stream.size(), rtp_stream.size());
>     for (size_t i = 0; i < parsed_rtp_stream.size(); i++) {
>-      EXPECT_TRUE(test::VerifyLoggedRtpPacketOutgoing(*rtp_stream[i],
>-                                                      parsed_rtp_stream[i]));
>+      verifier_.VerifyLoggedRtpPacketOutgoing(*rtp_stream[i],
>+                                              parsed_rtp_stream[i]);
>     }
>   }
> 
>   auto& parsed_incoming_rtcp_packets = parsed_log.incoming_rtcp_packets();
>   ASSERT_EQ(parsed_incoming_rtcp_packets.size(), incoming_rtcp_list_.size());
>   for (size_t i = 0; i < parsed_incoming_rtcp_packets.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedRtcpPacketIncoming(
>-        *incoming_rtcp_list_[i], parsed_incoming_rtcp_packets[i]));
>+    verifier_.VerifyLoggedRtcpPacketIncoming(*incoming_rtcp_list_[i],
>+                                             parsed_incoming_rtcp_packets[i]);
>   }
> 
>   auto& parsed_outgoing_rtcp_packets = parsed_log.outgoing_rtcp_packets();
>   ASSERT_EQ(parsed_outgoing_rtcp_packets.size(), outgoing_rtcp_list_.size());
>   for (size_t i = 0; i < parsed_outgoing_rtcp_packets.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedRtcpPacketOutgoing(
>-        *outgoing_rtcp_list_[i], parsed_outgoing_rtcp_packets[i]));
>+    verifier_.VerifyLoggedRtcpPacketOutgoing(*outgoing_rtcp_list_[i],
>+                                             parsed_outgoing_rtcp_packets[i]);
>   }
>   auto& parsed_audio_recv_configs = parsed_log.audio_recv_configs();
>   ASSERT_EQ(parsed_audio_recv_configs.size(), audio_recv_config_list_.size());
>   for (size_t i = 0; i < parsed_audio_recv_configs.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedAudioRecvConfig(
>-        *audio_recv_config_list_[i], parsed_audio_recv_configs[i]));
>+    verifier_.VerifyLoggedAudioRecvConfig(*audio_recv_config_list_[i],
>+                                          parsed_audio_recv_configs[i]);
>   }
>   auto& parsed_audio_send_configs = parsed_log.audio_send_configs();
>   ASSERT_EQ(parsed_audio_send_configs.size(), audio_send_config_list_.size());
>   for (size_t i = 0; i < parsed_audio_send_configs.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedAudioSendConfig(
>-        *audio_send_config_list_[i], parsed_audio_send_configs[i]));
>+    verifier_.VerifyLoggedAudioSendConfig(*audio_send_config_list_[i],
>+                                          parsed_audio_send_configs[i]);
>   }
>   auto& parsed_video_recv_configs = parsed_log.video_recv_configs();
>   ASSERT_EQ(parsed_video_recv_configs.size(), video_recv_config_list_.size());
>   for (size_t i = 0; i < parsed_video_recv_configs.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedVideoRecvConfig(
>-        *video_recv_config_list_[i], parsed_video_recv_configs[i]));
>+    verifier_.VerifyLoggedVideoRecvConfig(*video_recv_config_list_[i],
>+                                          parsed_video_recv_configs[i]);
>   }
>   auto& parsed_video_send_configs = parsed_log.video_send_configs();
>   ASSERT_EQ(parsed_video_send_configs.size(), video_send_config_list_.size());
>   for (size_t i = 0; i < parsed_video_send_configs.size(); i++) {
>-    EXPECT_TRUE(test::VerifyLoggedVideoSendConfig(
>-        *video_send_config_list_[i], parsed_video_send_configs[i]));
>+    verifier_.VerifyLoggedVideoSendConfig(*video_send_config_list_[i],
>+                                          parsed_video_send_configs[i]);
>   }
> 
>   // Clean up temporary file - can be pretty slow.
>@@ -680,7 +685,25 @@ TEST_P(RtcEventLogSession, StartLoggingInTheMiddle) {
>   ReadAndVerifyLog();
> }
> 
>-TEST(RtcEventLogTest, CircularBufferKeepsMostRecentEvents) {
>+INSTANTIATE_TEST_CASE_P(
>+    RtcEventLogTest,
>+    RtcEventLogSession,
>+    ::testing::Combine(
>+        ::testing::Values(1234567, 7654321),
>+        ::testing::Values(RtcEventLog::kImmediateOutput, 1, 5),
>+        ::testing::Values(RtcEventLog::EncodingType::Legacy,
>+                          RtcEventLog::EncodingType::NewFormat)));
>+
>+class RtcEventLogCircularBufferTest
>+    : public ::testing::TestWithParam<RtcEventLog::EncodingType> {
>+ public:
>+  RtcEventLogCircularBufferTest()
>+      : encoding_type_(GetParam()), verifier_(encoding_type_) {}
>+  const RtcEventLog::EncodingType encoding_type_;
>+  const test::EventVerifier verifier_;
>+};
>+
>+TEST_P(RtcEventLogCircularBufferTest, KeepsMostRecentEvents) {
>   // TODO(terelius): Maybe make a separate RtcEventLogImplTest that can access
>   // the size of the cyclic buffer?
>   constexpr size_t kNumEvents = 20000;
>@@ -699,8 +722,7 @@ TEST(RtcEventLogTest, CircularBufferKeepsMostRecentEvents) {
> 
>   // When log_dumper goes out of scope, it causes the log file to be flushed
>   // to disk.
>-  std::unique_ptr<RtcEventLog> log_dumper(
>-      RtcEventLog::Create(RtcEventLog::EncodingType::Legacy));
>+  std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create(encoding_type_));
> 
>   for (size_t i = 0; i < kNumEvents; i++) {
>     // The purpose of the test is to verify that the log can handle
>@@ -714,6 +736,7 @@ TEST(RtcEventLogTest, CircularBufferKeepsMostRecentEvents) {
>     fake_clock->AdvanceTimeMicros(10000);
>   }
>   int64_t start_time_us = rtc::TimeMicros();
>+  int64_t utc_start_time_us = rtc::TimeUTCMicros();
>   log_dumper->StartLogging(
>       absl::make_unique<RtcEventLogOutputFile>(temp_filename, 10000000),
>       RtcEventLog::kImmediateOutput);
>@@ -727,11 +750,12 @@ TEST(RtcEventLogTest, CircularBufferKeepsMostRecentEvents) {
> 
>   const auto& start_log_events = parsed_log.start_log_events();
>   ASSERT_EQ(start_log_events.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedStartEvent(start_time_us, start_log_events[0]));
>+  verifier_.VerifyLoggedStartEvent(start_time_us, utc_start_time_us,
>+                                   start_log_events[0]);
> 
>   const auto& stop_log_events = parsed_log.stop_log_events();
>   ASSERT_EQ(stop_log_events.size(), 1u);
>-  EXPECT_TRUE(test::VerifyLoggedStopEvent(stop_time_us, stop_log_events[0]));
>+  verifier_.VerifyLoggedStopEvent(stop_time_us, stop_log_events[0]);
> 
>   const auto& probe_success_events = parsed_log.bwe_probe_success_events();
>   // If the following fails, it probably means that kNumEvents isn't larger
>@@ -752,19 +776,20 @@ TEST(RtcEventLogTest, CircularBufferKeepsMostRecentEvents) {
>   fake_clock->SetTimeMicros(first_timestamp_us);
>   for (size_t i = 1; i < probe_success_events.size(); i++) {
>     fake_clock->AdvanceTimeMicros(10000);
>-    ASSERT_TRUE(test::VerifyLoggedBweProbeSuccessEvent(
>+    verifier_.VerifyLoggedBweProbeSuccessEvent(
>         RtcEventProbeResultSuccess(first_id + i, first_bitrate_bps + i * 1000),
>-        probe_success_events[i]));
>+        probe_success_events[i]);
>   }
> }
> 
>+INSTANTIATE_TEST_CASE_P(
>+    RtcEventLogTest,
>+    RtcEventLogCircularBufferTest,
>+    ::testing::Values(RtcEventLog::EncodingType::Legacy,
>+                      RtcEventLog::EncodingType::NewFormat));
>+
> // TODO(terelius): Verify parser behavior if the timestamps are not
> // monotonically increasing in the log.
> 
>-INSTANTIATE_TEST_CASE_P(
>-    RtcEventLogTest,
>-    RtcEventLogSession,
>-    ::testing::Combine(::testing::Values(1234567, 7654321),
>-                       ::testing::Values(RtcEventLog::kImmediateOutput, 1, 5)));
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
>index 9f6240cbe1e6bf77771df1b137dbe9d37f10b638..e87808d125c3e493c11f15c77a9daf01643b527c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
>@@ -12,12 +12,15 @@
> 
> #include <string.h>  // memcmp
> 
>+#include <algorithm>
> #include <limits>
> #include <memory>
> #include <numeric>
>+#include <string>
> #include <utility>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/rtp_rtcp/include/rtp_cvo.h"
>@@ -25,6 +28,7 @@
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "rtc_base/checks.h"
>+#include "test/gtest.h"
> 
> namespace webrtc {
> 
>@@ -62,6 +66,16 @@ void ShuffleInPlace(Random* prng, rtc::ArrayView<T> array) {
>     std::swap(array[i], array[other]);
>   }
> }
>+
>+absl::optional<int> GetExtensionId(const std::vector<RtpExtension>& extensions,
>+                                   const std::string& uri) {
>+  for (const auto& extension : extensions) {
>+    if (extension.uri == uri)
>+      return extension.id;
>+  }
>+  return absl::nullopt;
>+}
>+
> }  // namespace
> 
> std::unique_ptr<RtcEventAlrState> EventGenerator::NewAlrState() {
>@@ -266,7 +280,8 @@ void EventGenerator::RandomizeRtpPacket(
>     size_t padding_size,
>     uint32_t ssrc,
>     const RtpHeaderExtensionMap& extension_map,
>-    RtpPacket* rtp_packet) {
>+    RtpPacket* rtp_packet,
>+    bool all_configured_exts) {
>   constexpr int kMaxPayloadType = 127;
>   rtp_packet->SetPayloadType(prng_.Rand(kMaxPayloadType));
>   rtp_packet->SetMarker(prng_.Rand<bool>());
>@@ -281,16 +296,30 @@ void EventGenerator::RandomizeRtpPacket(
>   }
>   rtp_packet->SetCsrcs(csrcs);
> 
>-  if (extension_map.IsRegistered(TransmissionOffset::kId))
>+  if (extension_map.IsRegistered(TransmissionOffset::kId) &&
>+      (all_configured_exts || prng_.Rand<bool>())) {
>     rtp_packet->SetExtension<TransmissionOffset>(prng_.Rand(0x00ffffff));
>-  if (extension_map.IsRegistered(AudioLevel::kId))
>+  }
>+
>+  if (extension_map.IsRegistered(AudioLevel::kId) &&
>+      (all_configured_exts || prng_.Rand<bool>())) {
>     rtp_packet->SetExtension<AudioLevel>(prng_.Rand<bool>(), prng_.Rand(127));
>-  if (extension_map.IsRegistered(AbsoluteSendTime::kId))
>+  }
>+
>+  if (extension_map.IsRegistered(AbsoluteSendTime::kId) &&
>+      (all_configured_exts || prng_.Rand<bool>())) {
>     rtp_packet->SetExtension<AbsoluteSendTime>(prng_.Rand(0x00ffffff));
>-  if (extension_map.IsRegistered(VideoOrientation::kId))
>+  }
>+
>+  if (extension_map.IsRegistered(VideoOrientation::kId) &&
>+      (all_configured_exts || prng_.Rand<bool>())) {
>     rtp_packet->SetExtension<VideoOrientation>(prng_.Rand(3));
>-  if (extension_map.IsRegistered(TransportSequenceNumber::kId))
>+  }
>+
>+  if (extension_map.IsRegistered(TransportSequenceNumber::kId) &&
>+      (all_configured_exts || prng_.Rand<bool>())) {
>     rtp_packet->SetExtension<TransportSequenceNumber>(prng_.Rand<uint16_t>());
>+  }
> 
>   RTC_CHECK_LE(rtp_packet->headers_size() + payload_size, IP_PACKET_SIZE);
> 
>@@ -299,12 +328,13 @@ void EventGenerator::RandomizeRtpPacket(
>   for (size_t i = 0; i < payload_size; i++) {
>     payload[i] = prng_.Rand<uint8_t>();
>   }
>-  RTC_CHECK(rtp_packet->SetPadding(padding_size, &prng_));
>+  RTC_CHECK(rtp_packet->SetPadding(padding_size));
> }
> 
> std::unique_ptr<RtcEventRtpPacketIncoming> EventGenerator::NewRtpPacketIncoming(
>     uint32_t ssrc,
>-    const RtpHeaderExtensionMap& extension_map) {
>+    const RtpHeaderExtensionMap& extension_map,
>+    bool all_configured_exts) {
>   constexpr size_t kMaxPaddingLength = 224;
>   const bool padding = prng_.Rand(0, 9) == 0;  // Let padding be 10% probable.
>   const size_t padding_size = !padding ? 0u : prng_.Rand(0u, kMaxPaddingLength);
>@@ -325,14 +355,15 @@ std::unique_ptr<RtcEventRtpPacketIncoming> EventGenerator::NewRtpPacketIncoming(
> 
>   RtpPacketReceived rtp_packet(&extension_map);
>   RandomizeRtpPacket(payload_size, padding_size, ssrc, extension_map,
>-                     &rtp_packet);
>+                     &rtp_packet, all_configured_exts);
> 
>   return absl::make_unique<RtcEventRtpPacketIncoming>(rtp_packet);
> }
> 
> std::unique_ptr<RtcEventRtpPacketOutgoing> EventGenerator::NewRtpPacketOutgoing(
>     uint32_t ssrc,
>-    const RtpHeaderExtensionMap& extension_map) {
>+    const RtpHeaderExtensionMap& extension_map,
>+    bool all_configured_exts) {
>   constexpr size_t kMaxPaddingLength = 224;
>   const bool padding = prng_.Rand(0, 9) == 0;  // Let padding be 10% probable.
>   const size_t padding_size = !padding ? 0u : prng_.Rand(0u, kMaxPaddingLength);
>@@ -354,33 +385,34 @@ std::unique_ptr<RtcEventRtpPacketOutgoing> EventGenerator::NewRtpPacketOutgoing(
>   RtpPacketToSend rtp_packet(&extension_map,
>                              kMaxHeaderSize + payload_size + padding_size);
>   RandomizeRtpPacket(payload_size, padding_size, ssrc, extension_map,
>-                     &rtp_packet);
>+                     &rtp_packet, all_configured_exts);
> 
>   int probe_cluster_id = prng_.Rand(0, 100000);
>   return absl::make_unique<RtcEventRtpPacketOutgoing>(rtp_packet,
>                                                       probe_cluster_id);
> }
> 
>-RtpHeaderExtensionMap EventGenerator::NewRtpHeaderExtensionMap() {
>+RtpHeaderExtensionMap EventGenerator::NewRtpHeaderExtensionMap(
>+    bool configure_all) {
>   RtpHeaderExtensionMap extension_map;
>   std::vector<int> id(RtpExtension::kOneByteHeaderExtensionMaxId -
>                       RtpExtension::kMinId + 1);
>   std::iota(id.begin(), id.end(), RtpExtension::kMinId);
>   ShuffleInPlace(&prng_, rtc::ArrayView<int>(id));
> 
>-  if (prng_.Rand<bool>()) {
>+  if (configure_all || prng_.Rand<bool>()) {
>     extension_map.Register<AudioLevel>(id[0]);
>   }
>-  if (prng_.Rand<bool>()) {
>+  if (configure_all || prng_.Rand<bool>()) {
>     extension_map.Register<TransmissionOffset>(id[1]);
>   }
>-  if (prng_.Rand<bool>()) {
>+  if (configure_all || prng_.Rand<bool>()) {
>     extension_map.Register<AbsoluteSendTime>(id[2]);
>   }
>-  if (prng_.Rand<bool>()) {
>+  if (configure_all || prng_.Rand<bool>()) {
>     extension_map.Register<VideoOrientation>(id[3]);
>   }
>-  if (prng_.Rand<bool>()) {
>+  if (configure_all || prng_.Rand<bool>()) {
>     extension_map.Register<TransportSequenceNumber>(id[4]);
>   }
> 
>@@ -469,396 +501,317 @@ EventGenerator::NewVideoSendStreamConfig(
>   return absl::make_unique<RtcEventVideoSendStreamConfig>(std::move(config));
> }
> 
>-bool VerifyLoggedAlrStateEvent(const RtcEventAlrState& original_event,
>-                               const LoggedAlrStateEvent& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-  if (original_event.in_alr_ != logged_event.in_alr)
>-    return false;
>-  return true;
>+void EventVerifier::VerifyLoggedAlrStateEvent(
>+    const RtcEventAlrState& original_event,
>+    const LoggedAlrStateEvent& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+  EXPECT_EQ(original_event.in_alr(), logged_event.in_alr);
> }
> 
>-bool VerifyLoggedAudioPlayoutEvent(
>+void EventVerifier::VerifyLoggedAudioPlayoutEvent(
>     const RtcEventAudioPlayout& original_event,
>-    const LoggedAudioPlayoutEvent& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-  if (original_event.ssrc_ != logged_event.ssrc)
>-    return false;
>-  return true;
>+    const LoggedAudioPlayoutEvent& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+  EXPECT_EQ(original_event.ssrc(), logged_event.ssrc);
> }
> 
>-bool VerifyLoggedAudioNetworkAdaptationEvent(
>+void EventVerifier::VerifyLoggedAudioNetworkAdaptationEvent(
>     const RtcEventAudioNetworkAdaptation& original_event,
>-    const LoggedAudioNetworkAdaptationEvent& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-
>-  if (original_event.config_->bitrate_bps != logged_event.config.bitrate_bps)
>-    return false;
>-  if (original_event.config_->enable_dtx != logged_event.config.enable_dtx)
>-    return false;
>-  if (original_event.config_->enable_fec != logged_event.config.enable_fec)
>-    return false;
>-  if (original_event.config_->frame_length_ms !=
>-      logged_event.config.frame_length_ms)
>-    return false;
>-  if (original_event.config_->num_channels != logged_event.config.num_channels)
>-    return false;
>-  if (original_event.config_->uplink_packet_loss_fraction !=
>-      logged_event.config.uplink_packet_loss_fraction)
>-    return false;
>-
>-  return true;
>-}
>-
>-bool VerifyLoggedBweDelayBasedUpdate(
>+    const LoggedAudioNetworkAdaptationEvent& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+
>+  EXPECT_EQ(original_event.config().bitrate_bps,
>+            logged_event.config.bitrate_bps);
>+  EXPECT_EQ(original_event.config().enable_dtx, logged_event.config.enable_dtx);
>+  EXPECT_EQ(original_event.config().enable_fec, logged_event.config.enable_fec);
>+  EXPECT_EQ(original_event.config().frame_length_ms,
>+            logged_event.config.frame_length_ms);
>+  EXPECT_EQ(original_event.config().num_channels,
>+            logged_event.config.num_channels);
>+
>+  // uplink_packet_loss_fraction
>+  ASSERT_EQ(original_event.config().uplink_packet_loss_fraction.has_value(),
>+            logged_event.config.uplink_packet_loss_fraction.has_value());
>+  if (original_event.config().uplink_packet_loss_fraction.has_value()) {
>+    const float original =
>+        original_event.config().uplink_packet_loss_fraction.value();
>+    const float logged =
>+        logged_event.config.uplink_packet_loss_fraction.value();
>+    const float uplink_packet_loss_fraction_delta = std::abs(original - logged);
>+    EXPECT_LE(uplink_packet_loss_fraction_delta, 0.0001f);
>+  }
>+}
>+
>+void EventVerifier::VerifyLoggedBweDelayBasedUpdate(
>     const RtcEventBweUpdateDelayBased& original_event,
>-    const LoggedBweDelayBasedUpdate& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-  if (original_event.bitrate_bps_ != logged_event.bitrate_bps)
>-    return false;
>-  if (original_event.detector_state_ != logged_event.detector_state)
>-    return false;
>-  return true;
>+    const LoggedBweDelayBasedUpdate& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+  EXPECT_EQ(original_event.bitrate_bps(), logged_event.bitrate_bps);
>+  EXPECT_EQ(original_event.detector_state(), logged_event.detector_state);
> }
> 
>-bool VerifyLoggedBweLossBasedUpdate(
>+void EventVerifier::VerifyLoggedBweLossBasedUpdate(
>     const RtcEventBweUpdateLossBased& original_event,
>-    const LoggedBweLossBasedUpdate& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-  if (original_event.bitrate_bps_ != logged_event.bitrate_bps)
>-    return false;
>-  if (original_event.fraction_loss_ != logged_event.fraction_lost)
>-    return false;
>-  if (original_event.total_packets_ != logged_event.expected_packets)
>-    return false;
>-  return true;
>-}
>-
>-bool VerifyLoggedBweProbeClusterCreatedEvent(
>+    const LoggedBweLossBasedUpdate& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+  EXPECT_EQ(original_event.bitrate_bps(), logged_event.bitrate_bps);
>+  EXPECT_EQ(original_event.fraction_loss(), logged_event.fraction_lost);
>+  EXPECT_EQ(original_event.total_packets(), logged_event.expected_packets);
>+}
>+
>+void EventVerifier::VerifyLoggedBweProbeClusterCreatedEvent(
>     const RtcEventProbeClusterCreated& original_event,
>-    const LoggedBweProbeClusterCreatedEvent& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-  if (original_event.id_ != logged_event.id)
>-    return false;
>-  if (original_event.bitrate_bps_ != logged_event.bitrate_bps)
>-    return false;
>-  if (original_event.min_probes_ != logged_event.min_packets)
>-    return false;
>-  if (original_event.min_bytes_ != logged_event.min_bytes)
>-    return false;
>-
>-  return true;
>-}
>-
>-bool VerifyLoggedBweProbeFailureEvent(
>+    const LoggedBweProbeClusterCreatedEvent& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+  EXPECT_EQ(original_event.id(), logged_event.id);
>+  EXPECT_EQ(original_event.bitrate_bps(), logged_event.bitrate_bps);
>+  EXPECT_EQ(original_event.min_probes(), logged_event.min_packets);
>+  EXPECT_EQ(original_event.min_bytes(), logged_event.min_bytes);
>+}
>+
>+void EventVerifier::VerifyLoggedBweProbeFailureEvent(
>     const RtcEventProbeResultFailure& original_event,
>-    const LoggedBweProbeFailureEvent& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-  if (original_event.id_ != logged_event.id)
>-    return false;
>-  if (original_event.failure_reason_ != logged_event.failure_reason)
>-    return false;
>-  return true;
>+    const LoggedBweProbeFailureEvent& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+  EXPECT_EQ(original_event.id(), logged_event.id);
>+  EXPECT_EQ(original_event.failure_reason(), logged_event.failure_reason);
> }
> 
>-bool VerifyLoggedBweProbeSuccessEvent(
>+void EventVerifier::VerifyLoggedBweProbeSuccessEvent(
>     const RtcEventProbeResultSuccess& original_event,
>-    const LoggedBweProbeSuccessEvent& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-  if (original_event.id_ != logged_event.id)
>-    return false;
>-  if (original_event.bitrate_bps_ != logged_event.bitrate_bps)
>-    return false;
>-  return true;
>+    const LoggedBweProbeSuccessEvent& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+  EXPECT_EQ(original_event.id(), logged_event.id);
>+  EXPECT_EQ(original_event.bitrate_bps(), logged_event.bitrate_bps);
> }
> 
>-bool VerifyLoggedIceCandidatePairConfig(
>+void EventVerifier::VerifyLoggedIceCandidatePairConfig(
>     const RtcEventIceCandidatePairConfig& original_event,
>-    const LoggedIceCandidatePairConfig& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-
>-  if (original_event.type_ != logged_event.type)
>-    return false;
>-  if (original_event.candidate_pair_id_ != logged_event.candidate_pair_id)
>-    return false;
>-  if (original_event.candidate_pair_desc_.local_candidate_type !=
>-      logged_event.local_candidate_type)
>-    return false;
>-  if (original_event.candidate_pair_desc_.local_relay_protocol !=
>-      logged_event.local_relay_protocol)
>-    return false;
>-  if (original_event.candidate_pair_desc_.local_network_type !=
>-      logged_event.local_network_type)
>-    return false;
>-  if (original_event.candidate_pair_desc_.local_address_family !=
>-      logged_event.local_address_family)
>-    return false;
>-  if (original_event.candidate_pair_desc_.remote_candidate_type !=
>-      logged_event.remote_candidate_type)
>-    return false;
>-  if (original_event.candidate_pair_desc_.remote_address_family !=
>-      logged_event.remote_address_family)
>-    return false;
>-  if (original_event.candidate_pair_desc_.candidate_pair_protocol !=
>-      logged_event.candidate_pair_protocol)
>-    return false;
>-
>-  return true;
>-}
>-
>-bool VerifyLoggedIceCandidatePairEvent(
>+    const LoggedIceCandidatePairConfig& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+
>+  EXPECT_EQ(original_event.type(), logged_event.type);
>+  EXPECT_EQ(original_event.candidate_pair_id(), logged_event.candidate_pair_id);
>+  EXPECT_EQ(original_event.candidate_pair_desc().local_candidate_type,
>+            logged_event.local_candidate_type);
>+  EXPECT_EQ(original_event.candidate_pair_desc().local_relay_protocol,
>+            logged_event.local_relay_protocol);
>+  EXPECT_EQ(original_event.candidate_pair_desc().local_network_type,
>+            logged_event.local_network_type);
>+  EXPECT_EQ(original_event.candidate_pair_desc().local_address_family,
>+            logged_event.local_address_family);
>+  EXPECT_EQ(original_event.candidate_pair_desc().remote_candidate_type,
>+            logged_event.remote_candidate_type);
>+  EXPECT_EQ(original_event.candidate_pair_desc().remote_address_family,
>+            logged_event.remote_address_family);
>+  EXPECT_EQ(original_event.candidate_pair_desc().candidate_pair_protocol,
>+            logged_event.candidate_pair_protocol);
>+}
>+
>+void EventVerifier::VerifyLoggedIceCandidatePairEvent(
>     const RtcEventIceCandidatePair& original_event,
>-    const LoggedIceCandidatePairEvent& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-
>-  if (original_event.type_ != logged_event.type)
>-    return false;
>-  if (original_event.candidate_pair_id_ != logged_event.candidate_pair_id)
>-    return false;
>+    const LoggedIceCandidatePairEvent& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
> 
>-  return true;
>+  EXPECT_EQ(original_event.type(), logged_event.type);
>+  EXPECT_EQ(original_event.candidate_pair_id(), logged_event.candidate_pair_id);
> }
> 
>-bool VerifyLoggedRtpHeader(const RtpPacket& original_header,
>+void VerifyLoggedRtpHeader(const RtpPacket& original_header,
>                            const RTPHeader& logged_header) {
>   // Standard RTP header.
>-  if (original_header.Marker() != logged_header.markerBit)
>-    return false;
>-  if (original_header.PayloadType() != logged_header.payloadType)
>-    return false;
>-  if (original_header.SequenceNumber() != logged_header.sequenceNumber)
>-    return false;
>-  if (original_header.Timestamp() != logged_header.timestamp)
>-    return false;
>-  if (original_header.Ssrc() != logged_header.ssrc)
>-    return false;
>-  if (original_header.Csrcs().size() != logged_header.numCSRCs)
>-    return false;
>-  for (size_t i = 0; i < logged_header.numCSRCs; i++) {
>-    if (original_header.Csrcs()[i] != logged_header.arrOfCSRCs[i])
>-      return false;
>-  }
>+  EXPECT_EQ(original_header.Marker(), logged_header.markerBit);
>+  EXPECT_EQ(original_header.PayloadType(), logged_header.payloadType);
>+  EXPECT_EQ(original_header.SequenceNumber(), logged_header.sequenceNumber);
>+  EXPECT_EQ(original_header.Timestamp(), logged_header.timestamp);
>+  EXPECT_EQ(original_header.Ssrc(), logged_header.ssrc);
> 
>-  if (original_header.headers_size() != logged_header.headerLength)
>-    return false;
>+  EXPECT_EQ(original_header.headers_size(), logged_header.headerLength);
> 
>   // TransmissionOffset header extension.
>-  if (original_header.HasExtension<TransmissionOffset>() !=
>-      logged_header.extension.hasTransmissionTimeOffset)
>-    return false;
>+  ASSERT_EQ(original_header.HasExtension<TransmissionOffset>(),
>+            logged_header.extension.hasTransmissionTimeOffset);
>   if (logged_header.extension.hasTransmissionTimeOffset) {
>     int32_t offset;
>-    original_header.GetExtension<TransmissionOffset>(&offset);
>-    if (offset != logged_header.extension.transmissionTimeOffset)
>-      return false;
>+    ASSERT_TRUE(original_header.GetExtension<TransmissionOffset>(&offset));
>+    EXPECT_EQ(offset, logged_header.extension.transmissionTimeOffset);
>   }
> 
>   // AbsoluteSendTime header extension.
>-  if (original_header.HasExtension<AbsoluteSendTime>() !=
>-      logged_header.extension.hasAbsoluteSendTime)
>-    return false;
>+  ASSERT_EQ(original_header.HasExtension<AbsoluteSendTime>(),
>+            logged_header.extension.hasAbsoluteSendTime);
>   if (logged_header.extension.hasAbsoluteSendTime) {
>     uint32_t sendtime;
>-    original_header.GetExtension<AbsoluteSendTime>(&sendtime);
>-    if (sendtime != logged_header.extension.absoluteSendTime)
>-      return false;
>+    ASSERT_TRUE(original_header.GetExtension<AbsoluteSendTime>(&sendtime));
>+    EXPECT_EQ(sendtime, logged_header.extension.absoluteSendTime);
>   }
> 
>   // TransportSequenceNumber header extension.
>-  if (original_header.HasExtension<TransportSequenceNumber>() !=
>-      logged_header.extension.hasTransportSequenceNumber)
>-    return false;
>+  ASSERT_EQ(original_header.HasExtension<TransportSequenceNumber>(),
>+            logged_header.extension.hasTransportSequenceNumber);
>   if (logged_header.extension.hasTransportSequenceNumber) {
>     uint16_t seqnum;
>-    original_header.GetExtension<TransportSequenceNumber>(&seqnum);
>-    if (seqnum != logged_header.extension.transportSequenceNumber)
>-      return false;
>+    ASSERT_TRUE(original_header.GetExtension<TransportSequenceNumber>(&seqnum));
>+    EXPECT_EQ(seqnum, logged_header.extension.transportSequenceNumber);
>   }
> 
>   // AudioLevel header extension.
>-  if (original_header.HasExtension<AudioLevel>() !=
>-      logged_header.extension.hasAudioLevel)
>-    return false;
>+  ASSERT_EQ(original_header.HasExtension<AudioLevel>(),
>+            logged_header.extension.hasAudioLevel);
>   if (logged_header.extension.hasAudioLevel) {
>     bool voice_activity;
>     uint8_t audio_level;
>-    original_header.GetExtension<AudioLevel>(&voice_activity, &audio_level);
>-    if (voice_activity != logged_header.extension.voiceActivity)
>-      return false;
>-    if (audio_level != logged_header.extension.audioLevel)
>-      return false;
>+    ASSERT_TRUE(original_header.GetExtension<AudioLevel>(&voice_activity,
>+                                                         &audio_level));
>+    EXPECT_EQ(voice_activity, logged_header.extension.voiceActivity);
>+    EXPECT_EQ(audio_level, logged_header.extension.audioLevel);
>   }
> 
>   // VideoOrientation header extension.
>-  if (original_header.HasExtension<VideoOrientation>() !=
>-      logged_header.extension.hasVideoRotation)
>-    return false;
>+  ASSERT_EQ(original_header.HasExtension<VideoOrientation>(),
>+            logged_header.extension.hasVideoRotation);
>   if (logged_header.extension.hasVideoRotation) {
>     uint8_t rotation;
>-    original_header.GetExtension<VideoOrientation>(&rotation);
>-    if (ConvertCVOByteToVideoRotation(rotation) !=
>-        logged_header.extension.videoRotation)
>-      return false;
>+    ASSERT_TRUE(original_header.GetExtension<VideoOrientation>(&rotation));
>+    EXPECT_EQ(ConvertCVOByteToVideoRotation(rotation),
>+              logged_header.extension.videoRotation);
>   }
>-
>-  return true;
> }
> 
>-bool VerifyLoggedRtpPacketIncoming(
>+void EventVerifier::VerifyLoggedRtpPacketIncoming(
>     const RtcEventRtpPacketIncoming& original_event,
>-    const LoggedRtpPacketIncoming& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-
>-  if (original_event.header_.headers_size() != logged_event.rtp.header_length)
>-    return false;
>-
>-  if (original_event.packet_length_ != logged_event.rtp.total_length)
>-    return false;
>-
>-  if ((original_event.header_.data()[0] & 0x20) != 0 &&  // has padding
>-      original_event.packet_length_ - original_event.header_.headers_size() !=
>-          logged_event.rtp.header.paddingLength) {
>-    // Currently, RTC eventlog encoder-parser can only maintain padding length
>-    // if packet is full padding.
>-    // TODO(webrtc:9730): Change the condition to something like
>-    // original_event.padding_length_ != logged_event.rtp.header.paddingLength.
>-    return false;
>-  }
>+    const LoggedRtpPacketIncoming& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+
>+  EXPECT_EQ(original_event.header().headers_size(),
>+            logged_event.rtp.header_length);
>+
>+  EXPECT_EQ(original_event.packet_length(), logged_event.rtp.total_length);
> 
>-  if (!VerifyLoggedRtpHeader(original_event.header_, logged_event.rtp.header))
>-    return false;
>+  // Currently, RTC eventlog encoder-parser can only maintain padding length
>+  // if packet is full padding.
>+  EXPECT_EQ(original_event.padding_length(),
>+            logged_event.rtp.header.paddingLength);
> 
>-  return true;
>+  VerifyLoggedRtpHeader(original_event.header(), logged_event.rtp.header);
> }
> 
>-bool VerifyLoggedRtpPacketOutgoing(
>+void EventVerifier::VerifyLoggedRtpPacketOutgoing(
>     const RtcEventRtpPacketOutgoing& original_event,
>-    const LoggedRtpPacketOutgoing& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-
>-  if (original_event.header_.headers_size() != logged_event.rtp.header_length)
>-    return false;
>-
>-  if (original_event.packet_length_ != logged_event.rtp.total_length)
>-    return false;
>-
>-  if ((original_event.header_.data()[0] & 0x20) != 0 &&  // has padding
>-      original_event.packet_length_ - original_event.header_.headers_size() !=
>-          logged_event.rtp.header.paddingLength) {
>-    // Currently, RTC eventlog encoder-parser can only maintain padding length
>-    // if packet is full padding.
>-    // TODO(webrtc:9730): Change the condition to something like
>-    // original_event.padding_length_ != logged_event.rtp.header.paddingLength.
>-    return false;
>-  }
>+    const LoggedRtpPacketOutgoing& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+
>+  EXPECT_EQ(original_event.header().headers_size(),
>+            logged_event.rtp.header_length);
>+
>+  EXPECT_EQ(original_event.packet_length(), logged_event.rtp.total_length);
>+
>+  // Currently, RTC eventlog encoder-parser can only maintain padding length
>+  // if packet is full padding.
>+  EXPECT_EQ(original_event.padding_length(),
>+            logged_event.rtp.header.paddingLength);
> 
>   // TODO(terelius): Probe cluster ID isn't parsed, used or tested. Unless
>   // someone has a strong reason to keep it, it'll be removed.
> 
>-  if (!VerifyLoggedRtpHeader(original_event.header_, logged_event.rtp.header))
>-    return false;
>-
>-  return true;
>+  VerifyLoggedRtpHeader(original_event.header(), logged_event.rtp.header);
> }
> 
>-bool VerifyLoggedRtcpPacketIncoming(
>+void EventVerifier::VerifyLoggedRtcpPacketIncoming(
>     const RtcEventRtcpPacketIncoming& original_event,
>-    const LoggedRtcpPacketIncoming& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-
>-  if (original_event.packet_.size() != logged_event.rtcp.raw_data.size())
>-    return false;
>-  if (memcmp(original_event.packet_.data(), logged_event.rtcp.raw_data.data(),
>-             original_event.packet_.size()) != 0) {
>-    return false;
>-  }
>-  return true;
>+    const LoggedRtcpPacketIncoming& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+
>+  ASSERT_EQ(original_event.packet().size(), logged_event.rtcp.raw_data.size());
>+  EXPECT_EQ(
>+      memcmp(original_event.packet().data(), logged_event.rtcp.raw_data.data(),
>+             original_event.packet().size()),
>+      0);
> }
> 
>-bool VerifyLoggedRtcpPacketOutgoing(
>+void EventVerifier::VerifyLoggedRtcpPacketOutgoing(
>     const RtcEventRtcpPacketOutgoing& original_event,
>-    const LoggedRtcpPacketOutgoing& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-
>-  if (original_event.packet_.size() != logged_event.rtcp.raw_data.size())
>-    return false;
>-  if (memcmp(original_event.packet_.data(), logged_event.rtcp.raw_data.data(),
>-             original_event.packet_.size()) != 0) {
>-    return false;
>+    const LoggedRtcpPacketOutgoing& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+
>+  ASSERT_EQ(original_event.packet().size(), logged_event.rtcp.raw_data.size());
>+  EXPECT_EQ(
>+      memcmp(original_event.packet().data(), logged_event.rtcp.raw_data.data(),
>+             original_event.packet().size()),
>+      0);
>+}
>+
>+void EventVerifier::VerifyLoggedStartEvent(
>+    int64_t start_time_us,
>+    int64_t utc_start_time_us,
>+    const LoggedStartEvent& logged_event) const {
>+  EXPECT_EQ(start_time_us / 1000, logged_event.log_time_ms());
>+  if (encoding_type_ == RtcEventLog::EncodingType::NewFormat) {
>+    EXPECT_EQ(utc_start_time_us / 1000, logged_event.utc_start_time_ms);
>   }
>-  return true;
> }
> 
>-bool VerifyLoggedStartEvent(int64_t start_time_us,
>-                            const LoggedStartEvent& logged_event) {
>-  if (start_time_us != logged_event.log_time_us())
>-    return false;
>-  return true;
>+void EventVerifier::VerifyLoggedStopEvent(
>+    int64_t stop_time_us,
>+    const LoggedStopEvent& logged_event) const {
>+  EXPECT_EQ(stop_time_us / 1000, logged_event.log_time_ms());
> }
> 
>-bool VerifyLoggedStopEvent(int64_t stop_time_us,
>-                           const LoggedStopEvent& logged_event) {
>-  if (stop_time_us != logged_event.log_time_us())
>-    return false;
>-  return true;
>+void VerifyLoggedStreamConfig(const rtclog::StreamConfig& original_config,
>+                              const rtclog::StreamConfig& logged_config) {
>+  EXPECT_EQ(original_config.local_ssrc, logged_config.local_ssrc);
>+  EXPECT_EQ(original_config.remote_ssrc, logged_config.remote_ssrc);
>+  EXPECT_EQ(original_config.rtx_ssrc, logged_config.rtx_ssrc);
>+
>+  EXPECT_EQ(original_config.rtp_extensions.size(),
>+            logged_config.rtp_extensions.size());
>+  size_t recognized_extensions = 0;
>+  for (size_t i = 0; i < kMaxNumExtensions; i++) {
>+    auto original_id =
>+        GetExtensionId(original_config.rtp_extensions, kExtensions[i].name);
>+    auto logged_id =
>+        GetExtensionId(logged_config.rtp_extensions, kExtensions[i].name);
>+    EXPECT_EQ(original_id, logged_id)
>+        << "IDs for " << kExtensions[i].name << " don't match. Original ID "
>+        << original_id.value_or(-1) << ". Parsed ID " << logged_id.value_or(-1)
>+        << ".";
>+    if (original_id) {
>+      recognized_extensions++;
>+    }
>+  }
>+  EXPECT_EQ(recognized_extensions, original_config.rtp_extensions.size());
> }
> 
>-bool VerifyLoggedAudioRecvConfig(
>+void EventVerifier::VerifyLoggedAudioRecvConfig(
>     const RtcEventAudioReceiveStreamConfig& original_event,
>-    const LoggedAudioRecvConfig& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-  if (*original_event.config_ != logged_event.config)
>-    return false;
>-  return true;
>+    const LoggedAudioRecvConfig& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+  VerifyLoggedStreamConfig(original_event.config(), logged_event.config);
> }
> 
>-bool VerifyLoggedAudioSendConfig(
>+void EventVerifier::VerifyLoggedAudioSendConfig(
>     const RtcEventAudioSendStreamConfig& original_event,
>-    const LoggedAudioSendConfig& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-  if (*original_event.config_ != logged_event.config)
>-    return false;
>-  return true;
>+    const LoggedAudioSendConfig& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+  VerifyLoggedStreamConfig(original_event.config(), logged_event.config);
> }
> 
>-bool VerifyLoggedVideoRecvConfig(
>+void EventVerifier::VerifyLoggedVideoRecvConfig(
>     const RtcEventVideoReceiveStreamConfig& original_event,
>-    const LoggedVideoRecvConfig& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>-  if (*original_event.config_ != logged_event.config)
>-    return false;
>-  return true;
>+    const LoggedVideoRecvConfig& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>+  VerifyLoggedStreamConfig(original_event.config(), logged_event.config);
> }
> 
>-bool VerifyLoggedVideoSendConfig(
>+void EventVerifier::VerifyLoggedVideoSendConfig(
>     const RtcEventVideoSendStreamConfig& original_event,
>-    const LoggedVideoSendConfig& logged_event) {
>-  if (original_event.timestamp_us_ != logged_event.log_time_us())
>-    return false;
>+    const LoggedVideoSendConfig& logged_event) const {
>+  EXPECT_EQ(original_event.timestamp_ms(), logged_event.log_time_ms());
>   // TODO(terelius): In the past, we allowed storing multiple RtcStreamConfigs
>   // in the same RtcEventVideoSendStreamConfig. Look into whether we should drop
>   // backwards compatibility in the parser.
>-  if (logged_event.configs.size() != 1)
>-    return false;
>-  if (*original_event.config_ != logged_event.configs[0])
>-    return false;
>-  return true;
>+  ASSERT_EQ(logged_event.configs.size(), 1u);
>+  VerifyLoggedStreamConfig(original_event.config(), logged_event.configs[0]);
> }
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
>index f4d18765287be4dc156633ec3bb48d3c8927d522..81760b75553608affaaae360d82ae0b908f0e8bc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
>@@ -33,6 +33,7 @@
> #include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
> #include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
> #include "logging/rtc_event_log/rtc_event_log_parser_new.h"
>+#include "logging/rtc_event_log/rtc_stream_config.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
>@@ -70,21 +71,32 @@ class EventGenerator {
> 
>   std::unique_ptr<RtcEventRtcpPacketOutgoing> NewRtcpPacketOutgoing();
> 
>+  // |all_configured_exts| determines whether the RTP packet exhibits all
>+  // configured extensions, or a random subset thereof.
>   void RandomizeRtpPacket(size_t payload_size,
>                           size_t padding_size,
>                           uint32_t ssrc,
>                           const RtpHeaderExtensionMap& extension_map,
>-                          RtpPacket* rtp_packet);
>+                          RtpPacket* rtp_packet,
>+                          bool all_configured_exts);
> 
>+  // |all_configured_exts| determines whether the RTP packet exhibits all
>+  // configured extensions, or a random subset thereof.
>   std::unique_ptr<RtcEventRtpPacketIncoming> NewRtpPacketIncoming(
>       uint32_t ssrc,
>-      const RtpHeaderExtensionMap& extension_map);
>+      const RtpHeaderExtensionMap& extension_map,
>+      bool all_configured_exts = true);
> 
>+  // |all_configured_exts| determines whether the RTP packet exhibits all
>+  // configured extensions, or a random subset thereof.
>   std::unique_ptr<RtcEventRtpPacketOutgoing> NewRtpPacketOutgoing(
>       uint32_t ssrc,
>-      const RtpHeaderExtensionMap& extension_map);
>+      const RtpHeaderExtensionMap& extension_map,
>+      bool all_configured_exts = true);
> 
>-  RtpHeaderExtensionMap NewRtpHeaderExtensionMap();
>+  // |configure_all| determines whether all supported extensions are configured,
>+  // or a random subset.
>+  RtpHeaderExtensionMap NewRtpHeaderExtensionMap(bool configure_all = false);
> 
>   std::unique_ptr<RtcEventAudioReceiveStreamConfig> NewAudioReceiveStreamConfig(
>       uint32_t ssrc,
>@@ -110,80 +122,111 @@ class EventGenerator {
>   Random prng_;
> };
> 
>-bool VerifyLoggedAlrStateEvent(const RtcEventAlrState& original_event,
>-                               const LoggedAlrStateEvent& logged_event);
>-
>-bool VerifyLoggedAudioPlayoutEvent(const RtcEventAudioPlayout& original_event,
>-                                   const LoggedAudioPlayoutEvent& logged_event);
>-
>-bool VerifyLoggedAudioNetworkAdaptationEvent(
>-    const RtcEventAudioNetworkAdaptation& original_event,
>-    const LoggedAudioNetworkAdaptationEvent& logged_event);
>-
>-bool VerifyLoggedBweDelayBasedUpdate(
>-    const RtcEventBweUpdateDelayBased& original_event,
>-    const LoggedBweDelayBasedUpdate& logged_event);
>-
>-bool VerifyLoggedBweLossBasedUpdate(
>-    const RtcEventBweUpdateLossBased& original_event,
>-    const LoggedBweLossBasedUpdate& logged_event);
>-
>-bool VerifyLoggedBweProbeClusterCreatedEvent(
>-    const RtcEventProbeClusterCreated& original_event,
>-    const LoggedBweProbeClusterCreatedEvent& logged_event);
>-
>-bool VerifyLoggedBweProbeFailureEvent(
>-    const RtcEventProbeResultFailure& original_event,
>-    const LoggedBweProbeFailureEvent& logged_event);
>-
>-bool VerifyLoggedBweProbeSuccessEvent(
>-    const RtcEventProbeResultSuccess& original_event,
>-    const LoggedBweProbeSuccessEvent& logged_event);
>-
>-bool VerifyLoggedIceCandidatePairConfig(
>-    const RtcEventIceCandidatePairConfig& original_event,
>-    const LoggedIceCandidatePairConfig& logged_event);
>-
>-bool VerifyLoggedIceCandidatePairEvent(
>-    const RtcEventIceCandidatePair& original_event,
>-    const LoggedIceCandidatePairEvent& logged_event);
>-
>-bool VerifyLoggedRtpPacketIncoming(
>-    const RtcEventRtpPacketIncoming& original_event,
>-    const LoggedRtpPacketIncoming& logged_event);
>-
>-bool VerifyLoggedRtpPacketOutgoing(
>-    const RtcEventRtpPacketOutgoing& original_event,
>-    const LoggedRtpPacketOutgoing& logged_event);
>-
>-bool VerifyLoggedRtcpPacketIncoming(
>-    const RtcEventRtcpPacketIncoming& original_event,
>-    const LoggedRtcpPacketIncoming& logged_event);
>-
>-bool VerifyLoggedRtcpPacketOutgoing(
>-    const RtcEventRtcpPacketOutgoing& original_event,
>-    const LoggedRtcpPacketOutgoing& logged_event);
>-
>-bool VerifyLoggedStartEvent(int64_t start_time_us,
>-                            const LoggedStartEvent& logged_event);
>-bool VerifyLoggedStopEvent(int64_t stop_time_us,
>-                           const LoggedStopEvent& logged_event);
>-
>-bool VerifyLoggedAudioRecvConfig(
>-    const RtcEventAudioReceiveStreamConfig& original_event,
>-    const LoggedAudioRecvConfig& logged_event);
>-
>-bool VerifyLoggedAudioSendConfig(
>-    const RtcEventAudioSendStreamConfig& original_event,
>-    const LoggedAudioSendConfig& logged_event);
>-
>-bool VerifyLoggedVideoRecvConfig(
>-    const RtcEventVideoReceiveStreamConfig& original_event,
>-    const LoggedVideoRecvConfig& logged_event);
>+class EventVerifier {
>+ public:
>+  explicit EventVerifier(RtcEventLog::EncodingType encoding_type)
>+      : encoding_type_(encoding_type) {}
>+
>+  void VerifyLoggedAlrStateEvent(const RtcEventAlrState& original_event,
>+                                 const LoggedAlrStateEvent& logged_event) const;
>+
>+  void VerifyLoggedAudioPlayoutEvent(
>+      const RtcEventAudioPlayout& original_event,
>+      const LoggedAudioPlayoutEvent& logged_event) const;
>+
>+  void VerifyLoggedAudioNetworkAdaptationEvent(
>+      const RtcEventAudioNetworkAdaptation& original_event,
>+      const LoggedAudioNetworkAdaptationEvent& logged_event) const;
>+
>+  void VerifyLoggedBweDelayBasedUpdate(
>+      const RtcEventBweUpdateDelayBased& original_event,
>+      const LoggedBweDelayBasedUpdate& logged_event) const;
>+
>+  void VerifyLoggedBweLossBasedUpdate(
>+      const RtcEventBweUpdateLossBased& original_event,
>+      const LoggedBweLossBasedUpdate& logged_event) const;
>+
>+  void VerifyLoggedBweProbeClusterCreatedEvent(
>+      const RtcEventProbeClusterCreated& original_event,
>+      const LoggedBweProbeClusterCreatedEvent& logged_event) const;
>+
>+  void VerifyLoggedBweProbeFailureEvent(
>+      const RtcEventProbeResultFailure& original_event,
>+      const LoggedBweProbeFailureEvent& logged_event) const;
>+
>+  void VerifyLoggedBweProbeSuccessEvent(
>+      const RtcEventProbeResultSuccess& original_event,
>+      const LoggedBweProbeSuccessEvent& logged_event) const;
>+
>+  void VerifyLoggedIceCandidatePairConfig(
>+      const RtcEventIceCandidatePairConfig& original_event,
>+      const LoggedIceCandidatePairConfig& logged_event) const;
>+
>+  void VerifyLoggedIceCandidatePairEvent(
>+      const RtcEventIceCandidatePair& original_event,
>+      const LoggedIceCandidatePairEvent& logged_event) const;
>+
>+  void VerifyLoggedRtpPacketIncoming(
>+      const RtcEventRtpPacketIncoming& original_event,
>+      const LoggedRtpPacketIncoming& logged_event) const;
>+
>+  void VerifyLoggedRtpPacketOutgoing(
>+      const RtcEventRtpPacketOutgoing& original_event,
>+      const LoggedRtpPacketOutgoing& logged_event) const;
>+
>+  template <typename EventType, typename ParsedType>
>+  void VerifyLoggedRtpPacket(const EventType& original_event,
>+                             const ParsedType& logged_event) {
>+    static_assert(sizeof(ParsedType) == 0,
>+                  "You have to use one of the two defined template "
>+                  "specializations of VerifyLoggedRtpPacket");
>+  }
>+
>+  template <>
>+  void VerifyLoggedRtpPacket(const RtcEventRtpPacketIncoming& original_event,
>+                             const LoggedRtpPacketIncoming& logged_event) {
>+    VerifyLoggedRtpPacketIncoming(original_event, logged_event);
>+  }
>+
>+  template <>
>+  void VerifyLoggedRtpPacket(const RtcEventRtpPacketOutgoing& original_event,
>+                             const LoggedRtpPacketOutgoing& logged_event) {
>+    VerifyLoggedRtpPacketOutgoing(original_event, logged_event);
>+  }
>+
>+  void VerifyLoggedRtcpPacketIncoming(
>+      const RtcEventRtcpPacketIncoming& original_event,
>+      const LoggedRtcpPacketIncoming& logged_event) const;
>+
>+  void VerifyLoggedRtcpPacketOutgoing(
>+      const RtcEventRtcpPacketOutgoing& original_event,
>+      const LoggedRtcpPacketOutgoing& logged_event) const;
>+
>+  void VerifyLoggedStartEvent(int64_t start_time_us,
>+                              int64_t utc_start_time_us,
>+                              const LoggedStartEvent& logged_event) const;
>+  void VerifyLoggedStopEvent(int64_t stop_time_us,
>+                             const LoggedStopEvent& logged_event) const;
>+
>+  void VerifyLoggedAudioRecvConfig(
>+      const RtcEventAudioReceiveStreamConfig& original_event,
>+      const LoggedAudioRecvConfig& logged_event) const;
>+
>+  void VerifyLoggedAudioSendConfig(
>+      const RtcEventAudioSendStreamConfig& original_event,
>+      const LoggedAudioSendConfig& logged_event) const;
>+
>+  void VerifyLoggedVideoRecvConfig(
>+      const RtcEventVideoReceiveStreamConfig& original_event,
>+      const LoggedVideoRecvConfig& logged_event) const;
>+
>+  void VerifyLoggedVideoSendConfig(
>+      const RtcEventVideoSendStreamConfig& original_event,
>+      const LoggedVideoSendConfig& logged_event) const;
> 
>-bool VerifyLoggedVideoSendConfig(
>-    const RtcEventVideoSendStreamConfig& original_event,
>-    const LoggedVideoSendConfig& logged_event);
>+ private:
>+  RtcEventLog::EncodingType encoding_type_;
>+};
> 
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.h
>index d95c8359ece829f70527ba418cb2348c9e576e88..a9da4814e6b388481022bafc4b5aab7f8d494df0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.h
>@@ -11,12 +11,12 @@
> #ifndef LOGGING_RTC_EVENT_LOG_RTC_STREAM_CONFIG_H_
> #define LOGGING_RTC_EVENT_LOG_RTC_STREAM_CONFIG_H_
> 
>+#include <stdint.h>
> #include <string>
> #include <vector>
> 
> #include "api/rtp_headers.h"
> #include "api/rtpparameters.h"
>-#include "common_types.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace rtclog {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/media/BUILD.gn
>index 858b272f5580dc2a4b16efff37d4087b4ce6ec18..b6f98c67b7278227d17c6676464d9f821369e3b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/BUILD.gn
>@@ -42,6 +42,7 @@ rtc_source_set("rtc_h264_profile_id") {
>     "..:webrtc_common",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -136,7 +137,9 @@ rtc_static_library("rtc_media_base") {
>     "../modules/audio_processing:audio_processing_statistics",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/system:rtc_export",
>     "../rtc_base/third_party/sigslot",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>@@ -159,6 +162,52 @@ rtc_static_library("rtc_constants") {
>   ]
> }
> 
>+rtc_static_library("rtc_simulcast_encoder_adapter") {
>+  visibility = [ "*" ]
>+  defines = []
>+  libs = []
>+  sources = [
>+    "engine/simulcast_encoder_adapter.cc",
>+    "engine/simulcast_encoder_adapter.h",
>+  ]
>+  deps = [
>+    "../api/video:video_bitrate_allocation",
>+    "../api/video:video_frame_i420",
>+    "../api/video_codecs:video_codecs_api",
>+    "../modules/video_coding:video_codec_interface",
>+    "../modules/video_coding:video_coding_utility",
>+    "../rtc_base:checks",
>+    "../rtc_base:rtc_base_approved",
>+    "../rtc_base:sequenced_task_checker",
>+    "../system_wrappers",
>+    "../system_wrappers:field_trial",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+    "//third_party/libyuv",
>+  ]
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+}
>+
>+rtc_static_library("rtc_vp8_encoder_simulcast_proxy") {
>+  visibility = [ "*" ]
>+  defines = []
>+  libs = []
>+  sources = [
>+    "engine/vp8_encoder_simulcast_proxy.cc",
>+    "engine/vp8_encoder_simulcast_proxy.h",
>+  ]
>+  deps = [
>+    ":rtc_simulcast_encoder_adapter",
>+    "../api/video_codecs:video_codecs_api",
>+  ]
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+}
>+
> rtc_static_library("rtc_internal_video_codecs") {
>   visibility = [ "*" ]
>   allow_poison = [ "software_video_codecs" ]
>@@ -166,8 +215,9 @@ rtc_static_library("rtc_internal_video_codecs") {
>   libs = []
>   deps = [
>     ":rtc_h264_profile_id",
>+    ":rtc_simulcast_encoder_adapter",
>+    ":rtc_vp8_encoder_simulcast_proxy",
>     "../modules/video_coding:video_codec_interface",
>-    "../system_wrappers:field_trial",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
>   sources = [
>@@ -183,9 +233,9 @@ rtc_static_library("rtc_internal_video_codecs") {
>     "engine/scopedvideodecoder.h",
>     "engine/scopedvideoencoder.cc",
>     "engine/scopedvideoencoder.h",
>-    "engine/simulcast_encoder_adapter.cc",
>-    "engine/simulcast_encoder_adapter.h",
>-    "engine/vp8_encoder_simulcast_proxy.cc",
>+
>+    # TODO(bugs.webrtc.org/7925): stop exporting this header once downstream
>+    # targets depend on :rtc_vp8_encoder_simulcast_proxy directly.
>     "engine/vp8_encoder_simulcast_proxy.h",
>     "engine/webrtcvideodecoderfactory.h",
>     "engine/webrtcvideoencoderfactory.h",
>@@ -202,24 +252,20 @@ rtc_static_library("rtc_internal_video_codecs") {
>   deps += [
>     ":rtc_constants",
>     ":rtc_media_base",
>-    "..:webrtc_common",
>     "../api/video:video_bitrate_allocation",
>-    "../api/video:video_frame_i420",
>+    "../api/video:video_frame",
>     "../api/video_codecs:rtc_software_fallback_wrappers",
>     "../api/video_codecs:video_codecs_api",
>     "../call:call_interfaces",
>     "../call:video_stream_api",
>-    "../modules/video_coding:video_coding_utility",
>     "../modules/video_coding:webrtc_h264",
>     "../modules/video_coding:webrtc_multiplex",
>     "../modules/video_coding:webrtc_vp8",
>     "../modules/video_coding:webrtc_vp9",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>-    "../rtc_base:sequenced_task_checker",
>-    "../system_wrappers",
>-    "//third_party/abseil-cpp/absl/types:optional",
>-    "//third_party/libyuv",
>+    "../rtc_base/system:rtc_export",
>+    "//third_party/abseil-cpp/absl/strings",
>   ]
> }
> 
>@@ -232,12 +278,15 @@ rtc_static_library("rtc_audio_video") {
>   defines = []
>   libs = []
>   deps = [
>+    "../api/video:video_bitrate_allocator_factory",
>+    "../modules/audio_processing:api",
>     "../modules/audio_processing/aec_dump:aec_dump",
>     "../modules/video_coding:video_codec_interface",
>     "../modules/video_coding:video_coding",
>     "../modules/video_coding:video_coding_utility",
>     "../rtc_base:audio_format_to_string",
>     "../rtc_base:checks",
>+    "../rtc_base/system:rtc_export",
>     "../rtc_base/third_party/base64",
>     "../system_wrappers:field_trial",
>     "../system_wrappers:metrics",
>@@ -305,6 +354,7 @@ rtc_static_library("rtc_audio_video") {
>     "../api:libjingle_peerconnection_api",
>     "../api:transport_api",
>     "../api/audio_codecs:audio_codecs_api",
>+    "../api/video:builtin_video_bitrate_allocator_factory",
>     "../api/video:video_frame",
>     "../api/video:video_frame_i420",
>     "../api/video_codecs:rtc_software_fallback_wrappers",
>@@ -322,7 +372,10 @@ rtc_static_library("rtc_audio_video") {
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_task_queue",
>     "../rtc_base:stringutils",
>+    "../rtc_base/experiments:normalize_simulcast_size_experiment",
>     "../system_wrappers",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -391,24 +444,29 @@ if (rtc_include_tests) {
>     include_dirs = []
>     deps = [
>       ":rtc_audio_video",
>+      ":rtc_simulcast_encoder_adapter",
>       "../api:libjingle_peerconnection_api",
>       "../api/video:video_frame_i420",
>       "../call:video_stream_api",
>       "../common_video:common_video",
>       "../modules/audio_coding:rent_a_codec",
>+      "../modules/audio_processing:api",
>       "../modules/audio_processing:audio_processing",
>       "../modules/rtp_rtcp:rtp_rtcp_format",
>       "../modules/video_coding:video_codec_interface",
>       "../modules/video_coding:video_coding_utility",
>       "../p2p:rtc_p2p",
>       "../rtc_base:checks",
>+      "../rtc_base:gunit_helpers",
>       "../rtc_base:rtc_task_queue",
>       "../rtc_base:stringutils",
>       "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/strings",
>     ]
>     sources = [
>       "base/fakeframesource.cc",
>       "base/fakeframesource.h",
>+      "base/fakemediaengine.cc",
>       "base/fakemediaengine.h",
>       "base/fakenetworkinterface.h",
>       "base/fakertp.cc",
>@@ -450,7 +508,6 @@ if (rtc_include_tests) {
>       "../call:mock_rtp_interfaces",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>-      "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:rtc_task_queue_for_test",
>       "../rtc_base/third_party/sigslot",
>       "../test:test_support",
>@@ -486,8 +543,10 @@ if (rtc_include_tests) {
>       ":rtc_audio_video",
>       ":rtc_constants",
>       ":rtc_data",
>+      "../api/test/video:function_video_factory",
>       "../api/units:time_delta",
>       "../api/video:video_frame_i420",
>+      "../modules/audio_processing:api",
>       "../modules/audio_processing:mocks",
>       "../modules/rtp_rtcp",
>       "../modules/video_coding:video_codec_interface",
>@@ -495,11 +554,13 @@ if (rtc_include_tests) {
>       "../pc:rtc_pc",
>       "../pc:rtc_pc_base",
>       "../rtc_base:checks",
>+      "../rtc_base:gunit_helpers",
>+      "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:rtc_task_queue",
>       "../rtc_base:stringutils",
>       "../test:field_trial",
>-      "../test:test_common",
>       "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/strings",
>     ]
>     sources = [
>       "base/codec_unittest.cc",
>@@ -568,13 +629,18 @@ if (rtc_include_tests) {
>       ":rtc_media",
>       ":rtc_media_base",
>       ":rtc_media_tests_utils",
>+      ":rtc_simulcast_encoder_adapter",
>+      ":rtc_vp8_encoder_simulcast_proxy",
>       ":rtc_vp9_profile",
>       "../api:create_simulcast_test_fixture_api",
>       "../api:libjingle_peerconnection_api",
>+      "../api:mock_video_bitrate_allocator",
>+      "../api:mock_video_bitrate_allocator_factory",
>       "../api:mock_video_codec_factory",
>       "../api:simulcast_test_fixture_api",
>       "../api/audio_codecs:builtin_audio_decoder_factory",
>       "../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../api/video:builtin_video_bitrate_allocator_factory",
>       "../api/video:video_bitrate_allocation",
>       "../api/video:video_frame",
>       "../api/video_codecs:builtin_video_decoder_factory",
>@@ -592,7 +658,6 @@ if (rtc_include_tests) {
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_main",
>-      "../rtc_base:rtc_base_tests_utils",
>       "../test:audio_codec_mocks",
>       "../test:test_support",
>       "../test:video_test_common",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/media/OWNERS
>index 634eaf2bfa79e1871a7b0608cf5c0205562b7c52..4544e9f4806138670823db84ba3ebbaa4d47f331 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/OWNERS
>@@ -4,6 +4,7 @@ sprang@webrtc.org
> magjed@webrtc.org
> mflodman@webrtc.org
> perkj@webrtc.org
>+shampson@webrtc.org
> solenberg@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.cc
>index caf32121125b1dbec0ca8cf20f96189489315fe8..0a1c7156af80e0599d19e52752b8e2d731a51b3c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.cc
>@@ -12,13 +12,13 @@
> 
> #include <algorithm>
> 
>+#include "absl/strings/match.h"
> #include "media/base/h264_profile_level_id.h"
> #include "media/base/vp9_profile.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/strings/string_builder.h"
>-#include "rtc_base/stringutils.h"
> 
> namespace cricket {
> 
>@@ -26,8 +26,8 @@ FeedbackParams::FeedbackParams() = default;
> FeedbackParams::~FeedbackParams() = default;
> 
> bool FeedbackParam::operator==(const FeedbackParam& other) const {
>-  return _stricmp(other.id().c_str(), id().c_str()) == 0 &&
>-         _stricmp(other.param().c_str(), param().c_str()) == 0;
>+  return absl::EqualsIgnoreCase(other.id(), id()) &&
>+         absl::EqualsIgnoreCase(other.param(), param());
> }
> 
> bool FeedbackParams::operator==(const FeedbackParams& other) const {
>@@ -97,7 +97,7 @@ bool Codec::Matches(const Codec& codec) const {
>   const int kMaxStaticPayloadId = 95;
>   return (id <= kMaxStaticPayloadId || codec.id <= kMaxStaticPayloadId)
>              ? (id == codec.id)
>-             : (_stricmp(name.c_str(), codec.name.c_str()) == 0);
>+             : (absl::EqualsIgnoreCase(name, codec.name));
> }
> 
> bool Codec::GetParam(const std::string& name, std::string* out) const {
>@@ -235,7 +235,7 @@ VideoCodec& VideoCodec::operator=(const VideoCodec& c) = default;
> VideoCodec& VideoCodec::operator=(VideoCodec&& c) = default;
> 
> void VideoCodec::SetDefaultParameters() {
>-  if (_stricmp(kH264CodecName, name.c_str()) == 0) {
>+  if (absl::EqualsIgnoreCase(kH264CodecName, name)) {
>     // This default is set for all H.264 codecs created because
>     // that was the default before packetization mode support was added.
>     // TODO(hta): Move this to the places that create VideoCodecs from
>@@ -268,10 +268,10 @@ static bool IsSameH264PacketizationMode(const CodecParameterMap& ours,
> bool VideoCodec::Matches(const VideoCodec& other) const {
>   if (!Codec::Matches(other))
>     return false;
>-  if (CodecNamesEq(name.c_str(), kH264CodecName))
>+  if (absl::EqualsIgnoreCase(name, kH264CodecName))
>     return webrtc::H264::IsSameH264Profile(params, other.params) &&
>            IsSameH264PacketizationMode(params, other.params);
>-  if (CodecNamesEq(name.c_str(), kVp9CodecName))
>+  if (absl::EqualsIgnoreCase(name, kVp9CodecName))
>     return webrtc::IsSameVP9Profile(params, other.params);
>   return true;
> }
>@@ -285,16 +285,16 @@ VideoCodec VideoCodec::CreateRtxCodec(int rtx_payload_type,
> 
> VideoCodec::CodecType VideoCodec::GetCodecType() const {
>   const char* payload_name = name.c_str();
>-  if (_stricmp(payload_name, kRedCodecName) == 0) {
>+  if (absl::EqualsIgnoreCase(payload_name, kRedCodecName)) {
>     return CODEC_RED;
>   }
>-  if (_stricmp(payload_name, kUlpfecCodecName) == 0) {
>+  if (absl::EqualsIgnoreCase(payload_name, kUlpfecCodecName)) {
>     return CODEC_ULPFEC;
>   }
>-  if (_stricmp(payload_name, kFlexfecCodecName) == 0) {
>+  if (absl::EqualsIgnoreCase(payload_name, kFlexfecCodecName)) {
>     return CODEC_FLEXFEC;
>   }
>-  if (_stricmp(payload_name, kRtxCodecName) == 0) {
>+  if (absl::EqualsIgnoreCase(payload_name, kRtxCodecName)) {
>     return CODEC_RTX;
>   }
> 
>@@ -362,14 +362,6 @@ bool HasTransportCc(const Codec& codec) {
>       FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
> }
> 
>-bool CodecNamesEq(const std::string& name1, const std::string& name2) {
>-  return CodecNamesEq(name1.c_str(), name2.c_str());
>-}
>-
>-bool CodecNamesEq(const char* name1, const char* name2) {
>-  return _stricmp(name1, name2) == 0;
>-}
>-
> const VideoCodec* FindMatchingCodec(
>     const std::vector<VideoCodec>& supported_codecs,
>     const VideoCodec& codec) {
>@@ -387,12 +379,12 @@ bool IsSameCodec(const std::string& name1,
>                  const std::string& name2,
>                  const CodecParameterMap& params2) {
>   // If different names (case insensitive), then not same formats.
>-  if (!CodecNamesEq(name1, name2))
>+  if (!absl::EqualsIgnoreCase(name1, name2))
>     return false;
>   // For every format besides H264 and VP9, comparing names is enough.
>-  if (CodecNamesEq(name1.c_str(), kH264CodecName))
>+  if (absl::EqualsIgnoreCase(name1, kH264CodecName))
>     return webrtc::H264::IsSameH264Profile(params1, params2);
>-  if (CodecNamesEq(name1.c_str(), kVp9CodecName))
>+  if (absl::EqualsIgnoreCase(name1, kVp9CodecName))
>     return webrtc::IsSameVP9Profile(params1, params2);
>   return true;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.h
>index db133f6e27f644f51e0d79da829cc253f0013922..0ca6535287beeb29ff5598df1f25cbaf14090d53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.h
>@@ -18,8 +18,8 @@
> 
> #include "api/rtpparameters.h"
> #include "api/video_codecs/sdp_video_format.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "media/base/mediaconstants.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace cricket {
> 
>@@ -62,7 +62,7 @@ class FeedbackParams {
>   std::vector<FeedbackParam> params_;
> };
> 
>-struct Codec {
>+struct RTC_EXPORT Codec {
>   int id;
>   std::string name;
>   int clockrate;
>@@ -142,7 +142,7 @@ struct AudioCodec : public Codec {
>   bool operator!=(const AudioCodec& c) const { return !(*this == c); }
> };
> 
>-struct VideoCodec : public Codec {
>+struct RTC_EXPORT VideoCodec : public Codec {
>   // Creates a codec with the given parameters.
>   VideoCodec(int id, const std::string& name);
>   // Creates a codec with the given name and empty id.
>@@ -215,8 +215,6 @@ const Codec* FindCodecById(const std::vector<Codec>& codecs, int payload_type) {
>   return nullptr;
> }
> 
>-bool CodecNamesEq(const std::string& name1, const std::string& name2);
>-bool CodecNamesEq(const char* name1, const char* name2);
> bool HasNack(const Codec& codec);
> bool HasRemb(const Codec& codec);
> bool HasRrtr(const Codec& codec);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakemediaengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakemediaengine.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..b48206fd77772ac9388b0860e29f558850fe339a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakemediaengine.cc
>@@ -0,0 +1,598 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "media/base/fakemediaengine.h"
>+
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "absl/strings/match.h"
>+#include "rtc_base/checks.h"
>+
>+namespace cricket {
>+
>+FakeVoiceMediaChannel::DtmfInfo::DtmfInfo(uint32_t ssrc,
>+                                          int event_code,
>+                                          int duration)
>+    : ssrc(ssrc), event_code(event_code), duration(duration) {}
>+
>+FakeVoiceMediaChannel::VoiceChannelAudioSink::VoiceChannelAudioSink(
>+    AudioSource* source)
>+    : source_(source) {
>+  source_->SetSink(this);
>+}
>+FakeVoiceMediaChannel::VoiceChannelAudioSink::~VoiceChannelAudioSink() {
>+  if (source_) {
>+    source_->SetSink(nullptr);
>+  }
>+}
>+void FakeVoiceMediaChannel::VoiceChannelAudioSink::OnData(
>+    const void* audio_data,
>+    int bits_per_sample,
>+    int sample_rate,
>+    size_t number_of_channels,
>+    size_t number_of_frames) {}
>+void FakeVoiceMediaChannel::VoiceChannelAudioSink::OnClose() {
>+  source_ = nullptr;
>+}
>+AudioSource* FakeVoiceMediaChannel::VoiceChannelAudioSink::source() const {
>+  return source_;
>+}
>+
>+FakeVoiceMediaChannel::FakeVoiceMediaChannel(FakeVoiceEngine* engine,
>+                                             const AudioOptions& options)
>+    : engine_(engine), max_bps_(-1) {
>+  output_scalings_[0] = 1.0;  // For default channel.
>+  SetOptions(options);
>+}
>+FakeVoiceMediaChannel::~FakeVoiceMediaChannel() {
>+  if (engine_) {
>+    engine_->UnregisterChannel(this);
>+  }
>+}
>+const std::vector<AudioCodec>& FakeVoiceMediaChannel::recv_codecs() const {
>+  return recv_codecs_;
>+}
>+const std::vector<AudioCodec>& FakeVoiceMediaChannel::send_codecs() const {
>+  return send_codecs_;
>+}
>+const std::vector<AudioCodec>& FakeVoiceMediaChannel::codecs() const {
>+  return send_codecs();
>+}
>+const std::vector<FakeVoiceMediaChannel::DtmfInfo>&
>+FakeVoiceMediaChannel::dtmf_info_queue() const {
>+  return dtmf_info_queue_;
>+}
>+const AudioOptions& FakeVoiceMediaChannel::options() const {
>+  return options_;
>+}
>+int FakeVoiceMediaChannel::max_bps() const {
>+  return max_bps_;
>+}
>+bool FakeVoiceMediaChannel::SetSendParameters(
>+    const AudioSendParameters& params) {
>+  set_send_rtcp_parameters(params.rtcp);
>+  return (SetSendCodecs(params.codecs) &&
>+          SetSendExtmapAllowMixed(params.extmap_allow_mixed) &&
>+          SetSendRtpHeaderExtensions(params.extensions) &&
>+          SetMaxSendBandwidth(params.max_bandwidth_bps) &&
>+          SetOptions(params.options));
>+}
>+bool FakeVoiceMediaChannel::SetRecvParameters(
>+    const AudioRecvParameters& params) {
>+  set_recv_rtcp_parameters(params.rtcp);
>+  return (SetRecvCodecs(params.codecs) &&
>+          SetRecvRtpHeaderExtensions(params.extensions));
>+}
>+void FakeVoiceMediaChannel::SetPlayout(bool playout) {
>+  set_playout(playout);
>+}
>+void FakeVoiceMediaChannel::SetSend(bool send) {
>+  set_sending(send);
>+}
>+bool FakeVoiceMediaChannel::SetAudioSend(uint32_t ssrc,
>+                                         bool enable,
>+                                         const AudioOptions* options,
>+                                         AudioSource* source) {
>+  if (!SetLocalSource(ssrc, source)) {
>+    return false;
>+  }
>+  if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, !enable)) {
>+    return false;
>+  }
>+  if (enable && options) {
>+    return SetOptions(*options);
>+  }
>+  return true;
>+}
>+bool FakeVoiceMediaChannel::HasSource(uint32_t ssrc) const {
>+  return local_sinks_.find(ssrc) != local_sinks_.end();
>+}
>+bool FakeVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
>+  if (!RtpHelper<VoiceMediaChannel>::AddRecvStream(sp))
>+    return false;
>+  output_scalings_[sp.first_ssrc()] = 1.0;
>+  return true;
>+}
>+bool FakeVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
>+  if (!RtpHelper<VoiceMediaChannel>::RemoveRecvStream(ssrc))
>+    return false;
>+  output_scalings_.erase(ssrc);
>+  return true;
>+}
>+bool FakeVoiceMediaChannel::CanInsertDtmf() {
>+  for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
>+       it != send_codecs_.end(); ++it) {
>+    // Find the DTMF telephone event "codec".
>+    if (absl::EqualsIgnoreCase(it->name, "telephone-event")) {
>+      return true;
>+    }
>+  }
>+  return false;
>+}
>+bool FakeVoiceMediaChannel::InsertDtmf(uint32_t ssrc,
>+                                       int event_code,
>+                                       int duration) {
>+  dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration));
>+  return true;
>+}
>+bool FakeVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
>+  if (0 == ssrc) {
>+    std::map<uint32_t, double>::iterator it;
>+    for (it = output_scalings_.begin(); it != output_scalings_.end(); ++it) {
>+      it->second = volume;
>+    }
>+    return true;
>+  } else if (output_scalings_.find(ssrc) != output_scalings_.end()) {
>+    output_scalings_[ssrc] = volume;
>+    return true;
>+  }
>+  return false;
>+}
>+bool FakeVoiceMediaChannel::GetOutputVolume(uint32_t ssrc, double* volume) {
>+  if (output_scalings_.find(ssrc) == output_scalings_.end())
>+    return false;
>+  *volume = output_scalings_[ssrc];
>+  return true;
>+}
>+bool FakeVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
>+  return false;
>+}
>+void FakeVoiceMediaChannel::SetRawAudioSink(
>+    uint32_t ssrc,
>+    std::unique_ptr<webrtc::AudioSinkInterface> sink) {
>+  sink_ = std::move(sink);
>+}
>+std::vector<webrtc::RtpSource> FakeVoiceMediaChannel::GetSources(
>+    uint32_t ssrc) const {
>+  return std::vector<webrtc::RtpSource>();
>+}
>+bool FakeVoiceMediaChannel::SetRecvCodecs(
>+    const std::vector<AudioCodec>& codecs) {
>+  if (fail_set_recv_codecs()) {
>+    // Fake the failure in SetRecvCodecs.
>+    return false;
>+  }
>+  recv_codecs_ = codecs;
>+  return true;
>+}
>+bool FakeVoiceMediaChannel::SetSendCodecs(
>+    const std::vector<AudioCodec>& codecs) {
>+  if (fail_set_send_codecs()) {
>+    // Fake the failure in SetSendCodecs.
>+    return false;
>+  }
>+  send_codecs_ = codecs;
>+  return true;
>+}
>+bool FakeVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
>+  max_bps_ = bps;
>+  return true;
>+}
>+bool FakeVoiceMediaChannel::SetOptions(const AudioOptions& options) {
>+  // Does a "merge" of current options and set options.
>+  options_.SetAll(options);
>+  return true;
>+}
>+bool FakeVoiceMediaChannel::SetLocalSource(uint32_t ssrc, AudioSource* source) {
>+  auto it = local_sinks_.find(ssrc);
>+  if (source) {
>+    if (it != local_sinks_.end()) {
>+      RTC_CHECK(it->second->source() == source);
>+    } else {
>+      local_sinks_.insert(std::make_pair(
>+          ssrc, absl::make_unique<VoiceChannelAudioSink>(source)));
>+    }
>+  } else {
>+    if (it != local_sinks_.end()) {
>+      local_sinks_.erase(it);
>+    }
>+  }
>+  return true;
>+}
>+
>+bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
>+                     uint32_t ssrc,
>+                     int event_code,
>+                     int duration) {
>+  return (info.duration == duration && info.event_code == event_code &&
>+          info.ssrc == ssrc);
>+}
>+
>+FakeVideoMediaChannel::FakeVideoMediaChannel(FakeVideoEngine* engine,
>+                                             const VideoOptions& options)
>+    : engine_(engine), max_bps_(-1) {
>+  SetOptions(options);
>+}
>+FakeVideoMediaChannel::~FakeVideoMediaChannel() {
>+  if (engine_) {
>+    engine_->UnregisterChannel(this);
>+  }
>+}
>+const std::vector<VideoCodec>& FakeVideoMediaChannel::recv_codecs() const {
>+  return recv_codecs_;
>+}
>+const std::vector<VideoCodec>& FakeVideoMediaChannel::send_codecs() const {
>+  return send_codecs_;
>+}
>+const std::vector<VideoCodec>& FakeVideoMediaChannel::codecs() const {
>+  return send_codecs();
>+}
>+bool FakeVideoMediaChannel::rendering() const {
>+  return playout();
>+}
>+const VideoOptions& FakeVideoMediaChannel::options() const {
>+  return options_;
>+}
>+const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
>+FakeVideoMediaChannel::sinks() const {
>+  return sinks_;
>+}
>+int FakeVideoMediaChannel::max_bps() const {
>+  return max_bps_;
>+}
>+bool FakeVideoMediaChannel::SetSendParameters(
>+    const VideoSendParameters& params) {
>+  set_send_rtcp_parameters(params.rtcp);
>+  return (SetSendCodecs(params.codecs) &&
>+          SetSendExtmapAllowMixed(params.extmap_allow_mixed) &&
>+          SetSendRtpHeaderExtensions(params.extensions) &&
>+          SetMaxSendBandwidth(params.max_bandwidth_bps));
>+}
>+bool FakeVideoMediaChannel::SetRecvParameters(
>+    const VideoRecvParameters& params) {
>+  set_recv_rtcp_parameters(params.rtcp);
>+  return (SetRecvCodecs(params.codecs) &&
>+          SetRecvRtpHeaderExtensions(params.extensions));
>+}
>+bool FakeVideoMediaChannel::AddSendStream(const StreamParams& sp) {
>+  return RtpHelper<VideoMediaChannel>::AddSendStream(sp);
>+}
>+bool FakeVideoMediaChannel::RemoveSendStream(uint32_t ssrc) {
>+  return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
>+}
>+bool FakeVideoMediaChannel::GetSendCodec(VideoCodec* send_codec) {
>+  if (send_codecs_.empty()) {
>+    return false;
>+  }
>+  *send_codec = send_codecs_[0];
>+  return true;
>+}
>+bool FakeVideoMediaChannel::SetSink(
>+    uint32_t ssrc,
>+    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
>+  if (ssrc != 0 && sinks_.find(ssrc) == sinks_.end()) {
>+    return false;
>+  }
>+  if (ssrc != 0) {
>+    sinks_[ssrc] = sink;
>+  }
>+  return true;
>+}
>+bool FakeVideoMediaChannel::HasSink(uint32_t ssrc) const {
>+  return sinks_.find(ssrc) != sinks_.end() && sinks_.at(ssrc) != nullptr;
>+}
>+bool FakeVideoMediaChannel::SetSend(bool send) {
>+  return set_sending(send);
>+}
>+bool FakeVideoMediaChannel::SetVideoSend(
>+    uint32_t ssrc,
>+    const VideoOptions* options,
>+    rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
>+  if (options) {
>+    if (!SetOptions(*options)) {
>+      return false;
>+    }
>+  }
>+  sources_[ssrc] = source;
>+  return true;
>+}
>+bool FakeVideoMediaChannel::HasSource(uint32_t ssrc) const {
>+  return sources_.find(ssrc) != sources_.end() && sources_.at(ssrc) != nullptr;
>+}
>+bool FakeVideoMediaChannel::AddRecvStream(const StreamParams& sp) {
>+  if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp))
>+    return false;
>+  sinks_[sp.first_ssrc()] = NULL;
>+  return true;
>+}
>+bool FakeVideoMediaChannel::RemoveRecvStream(uint32_t ssrc) {
>+  if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc))
>+    return false;
>+  sinks_.erase(ssrc);
>+  return true;
>+}
>+void FakeVideoMediaChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
>+}
>+bool FakeVideoMediaChannel::GetStats(VideoMediaInfo* info) {
>+  return false;
>+}
>+std::vector<webrtc::RtpSource> FakeVideoMediaChannel::GetSources(
>+    uint32_t ssrc) const {
>+  return {};
>+}
>+bool FakeVideoMediaChannel::SetRecvCodecs(
>+    const std::vector<VideoCodec>& codecs) {
>+  if (fail_set_recv_codecs()) {
>+    // Fake the failure in SetRecvCodecs.
>+    return false;
>+  }
>+  recv_codecs_ = codecs;
>+  return true;
>+}
>+bool FakeVideoMediaChannel::SetSendCodecs(
>+    const std::vector<VideoCodec>& codecs) {
>+  if (fail_set_send_codecs()) {
>+    // Fake the failure in SetSendCodecs.
>+    return false;
>+  }
>+  send_codecs_ = codecs;
>+
>+  return true;
>+}
>+bool FakeVideoMediaChannel::SetOptions(const VideoOptions& options) {
>+  options_ = options;
>+  return true;
>+}
>+bool FakeVideoMediaChannel::SetMaxSendBandwidth(int bps) {
>+  max_bps_ = bps;
>+  return true;
>+}
>+
>+FakeDataMediaChannel::FakeDataMediaChannel(void* unused,
>+                                           const DataOptions& options)
>+    : send_blocked_(false), max_bps_(-1) {}
>+FakeDataMediaChannel::~FakeDataMediaChannel() {}
>+const std::vector<DataCodec>& FakeDataMediaChannel::recv_codecs() const {
>+  return recv_codecs_;
>+}
>+const std::vector<DataCodec>& FakeDataMediaChannel::send_codecs() const {
>+  return send_codecs_;
>+}
>+const std::vector<DataCodec>& FakeDataMediaChannel::codecs() const {
>+  return send_codecs();
>+}
>+int FakeDataMediaChannel::max_bps() const {
>+  return max_bps_;
>+}
>+bool FakeDataMediaChannel::SetSendParameters(const DataSendParameters& params) {
>+  set_send_rtcp_parameters(params.rtcp);
>+  return (SetSendCodecs(params.codecs) &&
>+          SetMaxSendBandwidth(params.max_bandwidth_bps));
>+}
>+bool FakeDataMediaChannel::SetRecvParameters(const DataRecvParameters& params) {
>+  set_recv_rtcp_parameters(params.rtcp);
>+  return SetRecvCodecs(params.codecs);
>+}
>+bool FakeDataMediaChannel::SetSend(bool send) {
>+  return set_sending(send);
>+}
>+bool FakeDataMediaChannel::SetReceive(bool receive) {
>+  set_playout(receive);
>+  return true;
>+}
>+bool FakeDataMediaChannel::AddRecvStream(const StreamParams& sp) {
>+  if (!RtpHelper<DataMediaChannel>::AddRecvStream(sp))
>+    return false;
>+  return true;
>+}
>+bool FakeDataMediaChannel::RemoveRecvStream(uint32_t ssrc) {
>+  if (!RtpHelper<DataMediaChannel>::RemoveRecvStream(ssrc))
>+    return false;
>+  return true;
>+}
>+bool FakeDataMediaChannel::SendData(const SendDataParams& params,
>+                                    const rtc::CopyOnWriteBuffer& payload,
>+                                    SendDataResult* result) {
>+  if (send_blocked_) {
>+    *result = SDR_BLOCK;
>+    return false;
>+  } else {
>+    last_sent_data_params_ = params;
>+    last_sent_data_ = std::string(payload.data<char>(), payload.size());
>+    return true;
>+  }
>+}
>+SendDataParams FakeDataMediaChannel::last_sent_data_params() {
>+  return last_sent_data_params_;
>+}
>+std::string FakeDataMediaChannel::last_sent_data() {
>+  return last_sent_data_;
>+}
>+bool FakeDataMediaChannel::is_send_blocked() {
>+  return send_blocked_;
>+}
>+void FakeDataMediaChannel::set_send_blocked(bool blocked) {
>+  send_blocked_ = blocked;
>+}
>+bool FakeDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) {
>+  if (fail_set_recv_codecs()) {
>+    // Fake the failure in SetRecvCodecs.
>+    return false;
>+  }
>+  recv_codecs_ = codecs;
>+  return true;
>+}
>+bool FakeDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) {
>+  if (fail_set_send_codecs()) {
>+    // Fake the failure in SetSendCodecs.
>+    return false;
>+  }
>+  send_codecs_ = codecs;
>+  return true;
>+}
>+bool FakeDataMediaChannel::SetMaxSendBandwidth(int bps) {
>+  max_bps_ = bps;
>+  return true;
>+}
>+
>+FakeVoiceEngine::FakeVoiceEngine() : fail_create_channel_(false) {
>+  // Add a fake audio codec. Note that the name must not be "" as there are
>+  // sanity checks against that.
>+  codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1));
>+}
>+RtpCapabilities FakeVoiceEngine::GetCapabilities() const {
>+  return RtpCapabilities();
>+}
>+void FakeVoiceEngine::Init() {}
>+rtc::scoped_refptr<webrtc::AudioState> FakeVoiceEngine::GetAudioState() const {
>+  return rtc::scoped_refptr<webrtc::AudioState>();
>+}
>+VoiceMediaChannel* FakeVoiceEngine::CreateMediaChannel(
>+    webrtc::Call* call,
>+    const MediaConfig& config,
>+    const AudioOptions& options,
>+    const webrtc::CryptoOptions& crypto_options) {
>+  if (fail_create_channel_) {
>+    return nullptr;
>+  }
>+
>+  FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options);
>+  channels_.push_back(ch);
>+  return ch;
>+}
>+FakeVoiceMediaChannel* FakeVoiceEngine::GetChannel(size_t index) {
>+  return (channels_.size() > index) ? channels_[index] : NULL;
>+}
>+void FakeVoiceEngine::UnregisterChannel(VoiceMediaChannel* channel) {
>+  channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
>+}
>+const std::vector<AudioCodec>& FakeVoiceEngine::send_codecs() const {
>+  return codecs_;
>+}
>+const std::vector<AudioCodec>& FakeVoiceEngine::recv_codecs() const {
>+  return codecs_;
>+}
>+void FakeVoiceEngine::SetCodecs(const std::vector<AudioCodec>& codecs) {
>+  codecs_ = codecs;
>+}
>+int FakeVoiceEngine::GetInputLevel() {
>+  return 0;
>+}
>+bool FakeVoiceEngine::StartAecDump(rtc::PlatformFile file,
>+                                   int64_t max_size_bytes) {
>+  return false;
>+}
>+void FakeVoiceEngine::StopAecDump() {}
>+bool FakeVoiceEngine::StartRtcEventLog(rtc::PlatformFile file,
>+                                       int64_t max_size_bytes) {
>+  return false;
>+}
>+void FakeVoiceEngine::StopRtcEventLog() {}
>+
>+FakeVideoEngine::FakeVideoEngine()
>+    : capture_(false), fail_create_channel_(false) {
>+  // Add a fake video codec. Note that the name must not be "" as there are
>+  // sanity checks against that.
>+  codecs_.push_back(VideoCodec(0, "fake_video_codec"));
>+}
>+RtpCapabilities FakeVideoEngine::GetCapabilities() const {
>+  return RtpCapabilities();
>+}
>+bool FakeVideoEngine::SetOptions(const VideoOptions& options) {
>+  options_ = options;
>+  return true;
>+}
>+VideoMediaChannel* FakeVideoEngine::CreateMediaChannel(
>+    webrtc::Call* call,
>+    const MediaConfig& config,
>+    const VideoOptions& options,
>+    const webrtc::CryptoOptions& crypto_options) {
>+  if (fail_create_channel_) {
>+    return nullptr;
>+  }
>+
>+  FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this, options);
>+  channels_.emplace_back(ch);
>+  return ch;
>+}
>+FakeVideoMediaChannel* FakeVideoEngine::GetChannel(size_t index) {
>+  return (channels_.size() > index) ? channels_[index] : nullptr;
>+}
>+void FakeVideoEngine::UnregisterChannel(VideoMediaChannel* channel) {
>+  auto it = std::find(channels_.begin(), channels_.end(), channel);
>+  RTC_DCHECK(it != channels_.end());
>+  channels_.erase(it);
>+}
>+std::vector<VideoCodec> FakeVideoEngine::codecs() const {
>+  return codecs_;
>+}
>+void FakeVideoEngine::SetCodecs(const std::vector<VideoCodec> codecs) {
>+  codecs_ = codecs;
>+}
>+bool FakeVideoEngine::SetCapture(bool capture) {
>+  capture_ = capture;
>+  return true;
>+}
>+
>+FakeMediaEngine::FakeMediaEngine()
>+    : CompositeMediaEngine(absl::make_unique<FakeVoiceEngine>(),
>+                           absl::make_unique<FakeVideoEngine>()),
>+      voice_(static_cast<FakeVoiceEngine*>(&voice())),
>+      video_(static_cast<FakeVideoEngine*>(&video())) {}
>+FakeMediaEngine::~FakeMediaEngine() {}
>+void FakeMediaEngine::SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
>+  voice_->SetCodecs(codecs);
>+}
>+void FakeMediaEngine::SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
>+  video_->SetCodecs(codecs);
>+}
>+
>+FakeVoiceMediaChannel* FakeMediaEngine::GetVoiceChannel(size_t index) {
>+  return voice_->GetChannel(index);
>+}
>+FakeVideoMediaChannel* FakeMediaEngine::GetVideoChannel(size_t index) {
>+  return video_->GetChannel(index);
>+}
>+
>+void FakeMediaEngine::set_fail_create_channel(bool fail) {
>+  voice_->fail_create_channel_ = fail;
>+  video_->fail_create_channel_ = fail;
>+}
>+
>+DataMediaChannel* FakeDataEngine::CreateChannel(const MediaConfig& config) {
>+  FakeDataMediaChannel* ch = new FakeDataMediaChannel(this, DataOptions());
>+  channels_.push_back(ch);
>+  return ch;
>+}
>+FakeDataMediaChannel* FakeDataEngine::GetChannel(size_t index) {
>+  return (channels_.size() > index) ? channels_[index] : NULL;
>+}
>+void FakeDataEngine::UnregisterChannel(DataMediaChannel* channel) {
>+  channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
>+}
>+void FakeDataEngine::SetDataCodecs(const std::vector<DataCodec>& data_codecs) {
>+  data_codecs_ = data_codecs;
>+}
>+const std::vector<DataCodec>& FakeDataEngine::data_codecs() {
>+  return data_codecs_;
>+}
>+
>+}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakemediaengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakemediaengine.h
>index 62bd6ba5ba50f214a69f9f71bb436f3e9f161dea..e3a0638715282cba7a21bb1f75dc1ea66fcd6406 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakemediaengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakemediaengine.h
>@@ -17,10 +17,8 @@
> #include <set>
> #include <string>
> #include <tuple>
>-#include <utility>
> #include <vector>
> 
>-#include "absl/memory/memory.h"
> #include "api/call/audio_sink.h"
> #include "media/base/audiosource.h"
> #include "media/base/mediaengine.h"
>@@ -28,10 +26,8 @@
> #include "media/base/streamparams.h"
> #include "media/engine/webrtcvideoengine.h"
> #include "modules/audio_processing/include/audio_processing.h"
>-#include "rtc_base/checks.h"
> #include "rtc_base/copyonwritebuffer.h"
> #include "rtc_base/networkroute.h"
>-#include "rtc_base/stringutils.h"
> 
> using webrtc::RtpExtension;
> 
>@@ -128,7 +124,7 @@ class RtpHelper : public Base {
>     }
>     receive_streams_.push_back(sp);
>     rtp_receive_parameters_[sp.first_ssrc()] =
>-        CreateRtpParametersWithOneEncoding();
>+        CreateRtpParametersWithEncodings(sp);
>     return true;
>   }
>   virtual bool RemoveRecvStream(uint32_t ssrc) {
>@@ -256,6 +252,12 @@ class RtpHelper : public Base {
>     recv_extensions_ = extensions;
>     return true;
>   }
>+  bool SetSendExtmapAllowMixed(bool extmap_allow_mixed) {
>+    if (Base::ExtmapAllowMixed() != extmap_allow_mixed) {
>+      Base::SetExtmapAllowMixed(extmap_allow_mixed);
>+    }
>+    return true;
>+  }
>   bool SetSendRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) {
>     send_extensions_ = extensions;
>     return true;
>@@ -267,11 +269,11 @@ class RtpHelper : public Base {
>     recv_rtcp_parameters_ = params;
>   }
>   virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                const rtc::PacketTime& packet_time) {
>+                                int64_t packet_time_us) {
>     rtp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
>   }
>   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
>-                              const rtc::PacketTime& packet_time) {
>+                              int64_t packet_time_us) {
>     rtcp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
>   }
>   virtual void OnReadyToSend(bool ready) { ready_to_send_ = ready; }
>@@ -312,187 +314,72 @@ class RtpHelper : public Base {
> class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
>  public:
>   struct DtmfInfo {
>-    DtmfInfo(uint32_t ssrc, int event_code, int duration)
>-        : ssrc(ssrc), event_code(event_code), duration(duration) {}
>+    DtmfInfo(uint32_t ssrc, int event_code, int duration);
>     uint32_t ssrc;
>     int event_code;
>     int duration;
>   };
>   explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
>-                                 const AudioOptions& options)
>-      : engine_(engine), max_bps_(-1) {
>-    output_scalings_[0] = 1.0;  // For default channel.
>-    SetOptions(options);
>-  }
>+                                 const AudioOptions& options);
>   ~FakeVoiceMediaChannel();
>-  const std::vector<AudioCodec>& recv_codecs() const { return recv_codecs_; }
>-  const std::vector<AudioCodec>& send_codecs() const { return send_codecs_; }
>-  const std::vector<AudioCodec>& codecs() const { return send_codecs(); }
>-  const std::vector<DtmfInfo>& dtmf_info_queue() const {
>-    return dtmf_info_queue_;
>-  }
>-  const AudioOptions& options() const { return options_; }
>-  int max_bps() const { return max_bps_; }
>-  virtual bool SetSendParameters(const AudioSendParameters& params) {
>-    set_send_rtcp_parameters(params.rtcp);
>-    return (SetSendCodecs(params.codecs) &&
>-            SetSendRtpHeaderExtensions(params.extensions) &&
>-            SetMaxSendBandwidth(params.max_bandwidth_bps) &&
>-            SetOptions(params.options));
>-  }
>-
>-  virtual bool SetRecvParameters(const AudioRecvParameters& params) {
>-    set_recv_rtcp_parameters(params.rtcp);
>-    return (SetRecvCodecs(params.codecs) &&
>-            SetRecvRtpHeaderExtensions(params.extensions));
>-  }
>-
>-  virtual void SetPlayout(bool playout) { set_playout(playout); }
>-  virtual void SetSend(bool send) { set_sending(send); }
>-  virtual bool SetAudioSend(uint32_t ssrc,
>-                            bool enable,
>-                            const AudioOptions* options,
>-                            AudioSource* source) {
>-    if (!SetLocalSource(ssrc, source)) {
>-      return false;
>-    }
>-    if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, !enable)) {
>-      return false;
>-    }
>-    if (enable && options) {
>-      return SetOptions(*options);
>-    }
>-    return true;
>-  }
>+  const std::vector<AudioCodec>& recv_codecs() const;
>+  const std::vector<AudioCodec>& send_codecs() const;
>+  const std::vector<AudioCodec>& codecs() const;
>+  const std::vector<DtmfInfo>& dtmf_info_queue() const;
>+  const AudioOptions& options() const;
>+  int max_bps() const;
>+  bool SetSendParameters(const AudioSendParameters& params) override;
> 
>-  bool HasSource(uint32_t ssrc) const {
>-    return local_sinks_.find(ssrc) != local_sinks_.end();
>-  }
>+  bool SetRecvParameters(const AudioRecvParameters& params) override;
> 
>-  virtual bool AddRecvStream(const StreamParams& sp) {
>-    if (!RtpHelper<VoiceMediaChannel>::AddRecvStream(sp))
>-      return false;
>-    output_scalings_[sp.first_ssrc()] = 1.0;
>-    return true;
>-  }
>-  virtual bool RemoveRecvStream(uint32_t ssrc) {
>-    if (!RtpHelper<VoiceMediaChannel>::RemoveRecvStream(ssrc))
>-      return false;
>-    output_scalings_.erase(ssrc);
>-    return true;
>-  }
>+  void SetPlayout(bool playout) override;
>+  void SetSend(bool send) override;
>+  bool SetAudioSend(uint32_t ssrc,
>+                    bool enable,
>+                    const AudioOptions* options,
>+                    AudioSource* source) override;
> 
>-  virtual bool CanInsertDtmf() {
>-    for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
>-         it != send_codecs_.end(); ++it) {
>-      // Find the DTMF telephone event "codec".
>-      if (_stricmp(it->name.c_str(), "telephone-event") == 0) {
>-        return true;
>-      }
>-    }
>-    return false;
>-  }
>-  virtual bool InsertDtmf(uint32_t ssrc, int event_code, int duration) {
>-    dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration));
>-    return true;
>-  }
>+  bool HasSource(uint32_t ssrc) const;
> 
>-  virtual bool SetOutputVolume(uint32_t ssrc, double volume) {
>-    if (0 == ssrc) {
>-      std::map<uint32_t, double>::iterator it;
>-      for (it = output_scalings_.begin(); it != output_scalings_.end(); ++it) {
>-        it->second = volume;
>-      }
>-      return true;
>-    } else if (output_scalings_.find(ssrc) != output_scalings_.end()) {
>-      output_scalings_[ssrc] = volume;
>-      return true;
>-    }
>-    return false;
>-  }
>-  bool GetOutputVolume(uint32_t ssrc, double* volume) {
>-    if (output_scalings_.find(ssrc) == output_scalings_.end())
>-      return false;
>-    *volume = output_scalings_[ssrc];
>-    return true;
>-  }
>+  bool AddRecvStream(const StreamParams& sp) override;
>+  bool RemoveRecvStream(uint32_t ssrc) override;
>+
>+  bool CanInsertDtmf() override;
>+  bool InsertDtmf(uint32_t ssrc, int event_code, int duration) override;
>+
>+  bool SetOutputVolume(uint32_t ssrc, double volume) override;
>+  bool GetOutputVolume(uint32_t ssrc, double* volume);
> 
>-  virtual bool GetStats(VoiceMediaInfo* info) { return false; }
>+  bool GetStats(VoiceMediaInfo* info) override;
> 
>-  virtual void SetRawAudioSink(
>+  void SetRawAudioSink(
>       uint32_t ssrc,
>-      std::unique_ptr<webrtc::AudioSinkInterface> sink) {
>-    sink_ = std::move(sink);
>-  }
>+      std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
> 
>-  virtual std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const {
>-    return std::vector<webrtc::RtpSource>();
>-  }
>+  std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
> 
>  private:
>   class VoiceChannelAudioSink : public AudioSource::Sink {
>    public:
>-    explicit VoiceChannelAudioSink(AudioSource* source) : source_(source) {
>-      source_->SetSink(this);
>-    }
>-    virtual ~VoiceChannelAudioSink() {
>-      if (source_) {
>-        source_->SetSink(nullptr);
>-      }
>-    }
>+    explicit VoiceChannelAudioSink(AudioSource* source);
>+    ~VoiceChannelAudioSink() override;
>     void OnData(const void* audio_data,
>                 int bits_per_sample,
>                 int sample_rate,
>                 size_t number_of_channels,
>-                size_t number_of_frames) override {}
>-    void OnClose() override { source_ = nullptr; }
>-    AudioSource* source() const { return source_; }
>+                size_t number_of_frames) override;
>+    void OnClose() override;
>+    AudioSource* source() const;
> 
>    private:
>     AudioSource* source_;
>   };
> 
>-  bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) {
>-    if (fail_set_recv_codecs()) {
>-      // Fake the failure in SetRecvCodecs.
>-      return false;
>-    }
>-    recv_codecs_ = codecs;
>-    return true;
>-  }
>-  bool SetSendCodecs(const std::vector<AudioCodec>& codecs) {
>-    if (fail_set_send_codecs()) {
>-      // Fake the failure in SetSendCodecs.
>-      return false;
>-    }
>-    send_codecs_ = codecs;
>-    return true;
>-  }
>-  bool SetMaxSendBandwidth(int bps) {
>-    max_bps_ = bps;
>-    return true;
>-  }
>-  bool SetOptions(const AudioOptions& options) {
>-    // Does a "merge" of current options and set options.
>-    options_.SetAll(options);
>-    return true;
>-  }
>-  bool SetLocalSource(uint32_t ssrc, AudioSource* source) {
>-    auto it = local_sinks_.find(ssrc);
>-    if (source) {
>-      if (it != local_sinks_.end()) {
>-        RTC_CHECK(it->second->source() == source);
>-      } else {
>-        local_sinks_.insert(std::make_pair(
>-            ssrc, absl::make_unique<VoiceChannelAudioSink>(source)));
>-      }
>-    } else {
>-      if (it != local_sinks_.end()) {
>-        local_sinks_.erase(it);
>-      }
>-    }
>-    return true;
>-  }
>+  bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
>+  bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
>+  bool SetMaxSendBandwidth(int bps);
>+  bool SetOptions(const AudioOptions& options);
>+  bool SetLocalSource(uint32_t ssrc, AudioSource* source);
> 
>   FakeVoiceEngine* engine_;
>   std::vector<AudioCodec> recv_codecs_;
>@@ -506,136 +393,55 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
> };
> 
> // A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
>-inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
>-                            uint32_t ssrc,
>-                            int event_code,
>-                            int duration) {
>-  return (info.duration == duration && info.event_code == event_code &&
>-          info.ssrc == ssrc);
>-}
>+bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
>+                     uint32_t ssrc,
>+                     int event_code,
>+                     int duration);
> 
> class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
>  public:
>-  FakeVideoMediaChannel(FakeVideoEngine* engine, const VideoOptions& options)
>-      : engine_(engine), max_bps_(-1) {
>-    SetOptions(options);
>-  }
>+  FakeVideoMediaChannel(FakeVideoEngine* engine, const VideoOptions& options);
> 
>   ~FakeVideoMediaChannel();
> 
>-  const std::vector<VideoCodec>& recv_codecs() const { return recv_codecs_; }
>-  const std::vector<VideoCodec>& send_codecs() const { return send_codecs_; }
>-  const std::vector<VideoCodec>& codecs() const { return send_codecs(); }
>-  bool rendering() const { return playout(); }
>-  const VideoOptions& options() const { return options_; }
>+  const std::vector<VideoCodec>& recv_codecs() const;
>+  const std::vector<VideoCodec>& send_codecs() const;
>+  const std::vector<VideoCodec>& codecs() const;
>+  bool rendering() const;
>+  const VideoOptions& options() const;
>   const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
>-  sinks() const {
>-    return sinks_;
>-  }
>-  int max_bps() const { return max_bps_; }
>-  bool SetSendParameters(const VideoSendParameters& params) override {
>-    set_send_rtcp_parameters(params.rtcp);
>-    return (SetSendCodecs(params.codecs) &&
>-            SetSendRtpHeaderExtensions(params.extensions) &&
>-            SetMaxSendBandwidth(params.max_bandwidth_bps));
>-  }
>-  bool SetRecvParameters(const VideoRecvParameters& params) override {
>-    set_recv_rtcp_parameters(params.rtcp);
>-    return (SetRecvCodecs(params.codecs) &&
>-            SetRecvRtpHeaderExtensions(params.extensions));
>-  }
>-  bool AddSendStream(const StreamParams& sp) override {
>-    return RtpHelper<VideoMediaChannel>::AddSendStream(sp);
>-  }
>-  bool RemoveSendStream(uint32_t ssrc) override {
>-    return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
>-  }
>-
>-  bool GetSendCodec(VideoCodec* send_codec) override {
>-    if (send_codecs_.empty()) {
>-      return false;
>-    }
>-    *send_codec = send_codecs_[0];
>-    return true;
>-  }
>+  sinks() const;
>+  int max_bps() const;
>+  bool SetSendParameters(const VideoSendParameters& params) override;
>+  bool SetRecvParameters(const VideoRecvParameters& params) override;
>+  bool AddSendStream(const StreamParams& sp) override;
>+  bool RemoveSendStream(uint32_t ssrc) override;
>+
>+  bool GetSendCodec(VideoCodec* send_codec) override;
>   bool SetSink(uint32_t ssrc,
>-               rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override {
>-    if (ssrc != 0 && sinks_.find(ssrc) == sinks_.end()) {
>-      return false;
>-    }
>-    if (ssrc != 0) {
>-      sinks_[ssrc] = sink;
>-    }
>-    return true;
>-  }
>-  bool HasSink(uint32_t ssrc) const {
>-    return sinks_.find(ssrc) != sinks_.end() && sinks_.at(ssrc) != nullptr;
>-  }
>+               rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
>+  bool HasSink(uint32_t ssrc) const;
> 
>-  bool SetSend(bool send) override { return set_sending(send); }
>+  bool SetSend(bool send) override;
>   bool SetVideoSend(
>       uint32_t ssrc,
>       const VideoOptions* options,
>-      rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override {
>-    if (options) {
>-      if (!SetOptions(*options)) {
>-        return false;
>-      }
>-    }
>-    sources_[ssrc] = source;
>-    return true;
>-  }
>+      rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override;
> 
>-  bool HasSource(uint32_t ssrc) const {
>-    return sources_.find(ssrc) != sources_.end() &&
>-           sources_.at(ssrc) != nullptr;
>-  }
>-  bool AddRecvStream(const StreamParams& sp) override {
>-    if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp))
>-      return false;
>-    sinks_[sp.first_ssrc()] = NULL;
>-    return true;
>-  }
>-  bool RemoveRecvStream(uint32_t ssrc) override {
>-    if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc))
>-      return false;
>-    sinks_.erase(ssrc);
>-    return true;
>-  }
>+  bool HasSource(uint32_t ssrc) const;
>+  bool AddRecvStream(const StreamParams& sp) override;
>+  bool RemoveRecvStream(uint32_t ssrc) override;
> 
>-  void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override {}
>-  bool GetStats(VideoMediaInfo* info) override { return false; }
>+  void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
>+  bool GetStats(VideoMediaInfo* info) override;
> 
>-  std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override {
>-    return {};
>-  }
>+  std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
> 
>  private:
>-  bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
>-    if (fail_set_recv_codecs()) {
>-      // Fake the failure in SetRecvCodecs.
>-      return false;
>-    }
>-    recv_codecs_ = codecs;
>-    return true;
>-  }
>-  bool SetSendCodecs(const std::vector<VideoCodec>& codecs) {
>-    if (fail_set_send_codecs()) {
>-      // Fake the failure in SetSendCodecs.
>-      return false;
>-    }
>-    send_codecs_ = codecs;
>-
>-    return true;
>-  }
>-  bool SetOptions(const VideoOptions& options) {
>-    options_ = options;
>-    return true;
>-  }
>-  bool SetMaxSendBandwidth(int bps) {
>-    max_bps_ = bps;
>-    return true;
>-  }
>+  bool SetRecvCodecs(const std::vector<VideoCodec>& codecs);
>+  bool SetSendCodecs(const std::vector<VideoCodec>& codecs);
>+  bool SetOptions(const VideoOptions& options);
>+  bool SetMaxSendBandwidth(int bps);
> 
>   FakeVideoEngine* engine_;
>   std::vector<VideoCodec> recv_codecs_;
>@@ -652,78 +458,33 @@ class DataOptions {};
> 
> class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
>  public:
>-  explicit FakeDataMediaChannel(void* unused, const DataOptions& options)
>-      : send_blocked_(false), max_bps_(-1) {}
>-  ~FakeDataMediaChannel() {}
>-  const std::vector<DataCodec>& recv_codecs() const { return recv_codecs_; }
>-  const std::vector<DataCodec>& send_codecs() const { return send_codecs_; }
>-  const std::vector<DataCodec>& codecs() const { return send_codecs(); }
>-  int max_bps() const { return max_bps_; }
>-
>-  virtual bool SetSendParameters(const DataSendParameters& params) {
>-    set_send_rtcp_parameters(params.rtcp);
>-    return (SetSendCodecs(params.codecs) &&
>-            SetMaxSendBandwidth(params.max_bandwidth_bps));
>-  }
>-  virtual bool SetRecvParameters(const DataRecvParameters& params) {
>-    set_recv_rtcp_parameters(params.rtcp);
>-    return SetRecvCodecs(params.codecs);
>-  }
>-  virtual bool SetSend(bool send) { return set_sending(send); }
>-  virtual bool SetReceive(bool receive) {
>-    set_playout(receive);
>-    return true;
>-  }
>-  virtual bool AddRecvStream(const StreamParams& sp) {
>-    if (!RtpHelper<DataMediaChannel>::AddRecvStream(sp))
>-      return false;
>-    return true;
>-  }
>-  virtual bool RemoveRecvStream(uint32_t ssrc) {
>-    if (!RtpHelper<DataMediaChannel>::RemoveRecvStream(ssrc))
>-      return false;
>-    return true;
>-  }
>-
>-  virtual bool SendData(const SendDataParams& params,
>-                        const rtc::CopyOnWriteBuffer& payload,
>-                        SendDataResult* result) {
>-    if (send_blocked_) {
>-      *result = SDR_BLOCK;
>-      return false;
>-    } else {
>-      last_sent_data_params_ = params;
>-      last_sent_data_ = std::string(payload.data<char>(), payload.size());
>-      return true;
>-    }
>-  }
>-
>-  SendDataParams last_sent_data_params() { return last_sent_data_params_; }
>-  std::string last_sent_data() { return last_sent_data_; }
>-  bool is_send_blocked() { return send_blocked_; }
>-  void set_send_blocked(bool blocked) { send_blocked_ = blocked; }
>+  explicit FakeDataMediaChannel(void* unused, const DataOptions& options);
>+  ~FakeDataMediaChannel();
>+  const std::vector<DataCodec>& recv_codecs() const;
>+  const std::vector<DataCodec>& send_codecs() const;
>+  const std::vector<DataCodec>& codecs() const;
>+  int max_bps() const;
>+
>+  bool SetSendParameters(const DataSendParameters& params) override;
>+  bool SetRecvParameters(const DataRecvParameters& params) override;
>+  bool SetSend(bool send) override;
>+  bool SetReceive(bool receive) override;
>+  bool AddRecvStream(const StreamParams& sp) override;
>+  bool RemoveRecvStream(uint32_t ssrc) override;
>+
>+  bool SendData(const SendDataParams& params,
>+                const rtc::CopyOnWriteBuffer& payload,
>+                SendDataResult* result) override;
>+
>+  SendDataParams last_sent_data_params();
>+  std::string last_sent_data();
>+  bool is_send_blocked();
>+  void set_send_blocked(bool blocked);
> 
>  private:
>-  bool SetRecvCodecs(const std::vector<DataCodec>& codecs) {
>-    if (fail_set_recv_codecs()) {
>-      // Fake the failure in SetRecvCodecs.
>-      return false;
>-    }
>-    recv_codecs_ = codecs;
>-    return true;
>-  }
>-  bool SetSendCodecs(const std::vector<DataCodec>& codecs) {
>-    if (fail_set_send_codecs()) {
>-      // Fake the failure in SetSendCodecs.
>-      return false;
>-    }
>-    send_codecs_ = codecs;
>-    return true;
>-  }
>-  bool SetMaxSendBandwidth(int bps) {
>-    max_bps_ = bps;
>-    return true;
>-  }
>+  bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
>+  bool SetSendCodecs(const std::vector<DataCodec>& codecs);
>+  bool SetMaxSendBandwidth(int bps);
> 
>   std::vector<DataCodec> recv_codecs_;
>   std::vector<DataCodec> send_codecs_;
>@@ -733,233 +494,98 @@ class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
>   int max_bps_;
> };
> 
>-// A base class for all of the shared parts between FakeVoiceEngine
>-// and FakeVideoEngine.
>-class FakeBaseEngine {
>- public:
>-  FakeBaseEngine() : options_changed_(false), fail_create_channel_(false) {}
>-  void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
>-
>-  RtpCapabilities GetCapabilities() const { return capabilities_; }
>-  void set_rtp_header_extensions(const std::vector<RtpExtension>& extensions) {
>-    capabilities_.header_extensions = extensions;
>-  }
>-
>-  void set_rtp_header_extensions(
>-      const std::vector<cricket::RtpHeaderExtension>& extensions) {
>-    for (const cricket::RtpHeaderExtension& ext : extensions) {
>-      RtpExtension webrtc_ext;
>-      webrtc_ext.uri = ext.uri;
>-      webrtc_ext.id = ext.id;
>-      capabilities_.header_extensions.push_back(webrtc_ext);
>-    }
>-  }
>-
>- protected:
>-  // Flag used by optionsmessagehandler_unittest for checking whether any
>-  // relevant setting has been updated.
>-  // TODO(thaloun): Replace with explicit checks of before & after values.
>-  bool options_changed_;
>-  bool fail_create_channel_;
>-  RtpCapabilities capabilities_;
>-};
>-
>-class FakeVoiceEngine : public FakeBaseEngine {
>+class FakeVoiceEngine : public VoiceEngineInterface {
>  public:
>-  FakeVoiceEngine() {
>-    // Add a fake audio codec. Note that the name must not be "" as there are
>-    // sanity checks against that.
>-    codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1));
>-  }
>-  void Init() {}
>-  rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
>-    return rtc::scoped_refptr<webrtc::AudioState>();
>-  }
>-
>-  VoiceMediaChannel* CreateChannel(webrtc::Call* call,
>-                                   const MediaConfig& config,
>-                                   const AudioOptions& options) {
>-    if (fail_create_channel_) {
>-      return nullptr;
>-    }
>-
>-    FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options);
>-    channels_.push_back(ch);
>-    return ch;
>-  }
>-  FakeVoiceMediaChannel* GetChannel(size_t index) {
>-    return (channels_.size() > index) ? channels_[index] : NULL;
>-  }
>-  void UnregisterChannel(VoiceMediaChannel* channel) {
>-    channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
>-  }
>+  FakeVoiceEngine();
>+  RtpCapabilities GetCapabilities() const override;
>+  void Init() override;
>+  rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
>+
>+  VoiceMediaChannel* CreateMediaChannel(
>+      webrtc::Call* call,
>+      const MediaConfig& config,
>+      const AudioOptions& options,
>+      const webrtc::CryptoOptions& crypto_options) override;
>+  FakeVoiceMediaChannel* GetChannel(size_t index);
>+  void UnregisterChannel(VoiceMediaChannel* channel);
> 
>   // TODO(ossu): For proper testing, These should either individually settable
>   //             or the voice engine should reference mockable factories.
>-  const std::vector<AudioCodec>& send_codecs() { return codecs_; }
>-  const std::vector<AudioCodec>& recv_codecs() { return codecs_; }
>-  void SetCodecs(const std::vector<AudioCodec>& codecs) { codecs_ = codecs; }
>-
>-  int GetInputLevel() { return 0; }
>-
>-  bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) {
>-    return false;
>-  }
>-
>-  void StopAecDump() {}
>-
>-  bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes) {
>-    return false;
>-  }
>-
>-  void StopRtcEventLog() {}
>+  const std::vector<AudioCodec>& send_codecs() const override;
>+  const std::vector<AudioCodec>& recv_codecs() const override;
>+  void SetCodecs(const std::vector<AudioCodec>& codecs);
>+  int GetInputLevel();
>+  bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
>+  void StopAecDump() override;
>+  bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes);
>+  void StopRtcEventLog();
> 
>  private:
>   std::vector<FakeVoiceMediaChannel*> channels_;
>   std::vector<AudioCodec> codecs_;
>+  bool fail_create_channel_;
> 
>   friend class FakeMediaEngine;
> };
> 
>-class FakeVideoEngine : public FakeBaseEngine {
>+class FakeVideoEngine : public VideoEngineInterface {
>  public:
>-  FakeVideoEngine() : capture_(false) {
>-    // Add a fake video codec. Note that the name must not be "" as there are
>-    // sanity checks against that.
>-    codecs_.push_back(VideoCodec(0, "fake_video_codec"));
>-  }
>-
>-  bool SetOptions(const VideoOptions& options) {
>-    options_ = options;
>-    options_changed_ = true;
>-    return true;
>-  }
>-
>-  VideoMediaChannel* CreateChannel(webrtc::Call* call,
>-                                   const MediaConfig& config,
>-                                   const VideoOptions& options) {
>-    if (fail_create_channel_) {
>-      return nullptr;
>-    }
>-
>-    FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this, options);
>-    channels_.emplace_back(ch);
>-    return ch;
>-  }
>-
>-  FakeVideoMediaChannel* GetChannel(size_t index) {
>-    return (channels_.size() > index) ? channels_[index] : nullptr;
>-  }
>-
>-  void UnregisterChannel(VideoMediaChannel* channel) {
>-    auto it = std::find(channels_.begin(), channels_.end(), channel);
>-    RTC_DCHECK(it != channels_.end());
>-    channels_.erase(it);
>-  }
>-
>-  const std::vector<VideoCodec>& codecs() const { return codecs_; }
>-
>-  void SetCodecs(const std::vector<VideoCodec> codecs) { codecs_ = codecs; }
>-
>-  bool SetCapture(bool capture) {
>-    capture_ = capture;
>-    return true;
>-  }
>+  FakeVideoEngine();
>+  RtpCapabilities GetCapabilities() const override;
>+  bool SetOptions(const VideoOptions& options);
>+  VideoMediaChannel* CreateMediaChannel(
>+      webrtc::Call* call,
>+      const MediaConfig& config,
>+      const VideoOptions& options,
>+      const webrtc::CryptoOptions& crypto_options) override;
>+  FakeVideoMediaChannel* GetChannel(size_t index);
>+  void UnregisterChannel(VideoMediaChannel* channel);
>+  std::vector<VideoCodec> codecs() const override;
>+  void SetCodecs(const std::vector<VideoCodec> codecs);
>+  bool SetCapture(bool capture);
> 
>  private:
>   std::vector<FakeVideoMediaChannel*> channels_;
>   std::vector<VideoCodec> codecs_;
>   bool capture_;
>   VideoOptions options_;
>+  bool fail_create_channel_;
> 
>   friend class FakeMediaEngine;
> };
> 
>-class FakeMediaEngine
>-    : public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
>+class FakeMediaEngine : public CompositeMediaEngine {
>  public:
>-  FakeMediaEngine()
>-      : CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine>(std::tuple<>(),
>-                                                               std::tuple<>()) {
>-  }
>+  FakeMediaEngine();
> 
>-  virtual ~FakeMediaEngine() {}
>+  ~FakeMediaEngine() override;
> 
>-  void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
>-    voice().SetCodecs(codecs);
>-  }
>-  void SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
>-    video().SetCodecs(codecs);
>-  }
>+  void SetAudioCodecs(const std::vector<AudioCodec>& codecs);
>+  void SetVideoCodecs(const std::vector<VideoCodec>& codecs);
> 
>-  void SetAudioRtpHeaderExtensions(
>-      const std::vector<RtpExtension>& extensions) {
>-    voice().set_rtp_header_extensions(extensions);
>-  }
>-  void SetVideoRtpHeaderExtensions(
>-      const std::vector<RtpExtension>& extensions) {
>-    video().set_rtp_header_extensions(extensions);
>-  }
>+  FakeVoiceMediaChannel* GetVoiceChannel(size_t index);
>+  FakeVideoMediaChannel* GetVideoChannel(size_t index);
> 
>-  void SetAudioRtpHeaderExtensions(
>-      const std::vector<cricket::RtpHeaderExtension>& extensions) {
>-    voice().set_rtp_header_extensions(extensions);
>-  }
>-  void SetVideoRtpHeaderExtensions(
>-      const std::vector<cricket::RtpHeaderExtension>& extensions) {
>-    video().set_rtp_header_extensions(extensions);
>-  }
>+  void set_fail_create_channel(bool fail);
> 
>-  FakeVoiceMediaChannel* GetVoiceChannel(size_t index) {
>-    return voice().GetChannel(index);
>-  }
>-  FakeVideoMediaChannel* GetVideoChannel(size_t index) {
>-    return video().GetChannel(index);
>-  }
>-
>-  bool capture() const { return video().capture_; }
>-  bool options_changed() const { return video().options_changed_; }
>-  void clear_options_changed() { video().options_changed_ = false; }
>-  void set_fail_create_channel(bool fail) {
>-    voice().set_fail_create_channel(fail);
>-    video().set_fail_create_channel(fail);
>-  }
>+ private:
>+  FakeVoiceEngine* const voice_;
>+  FakeVideoEngine* const video_;
> };
> 
> // Have to come afterwards due to declaration order
>-inline FakeVoiceMediaChannel::~FakeVoiceMediaChannel() {
>-  if (engine_) {
>-    engine_->UnregisterChannel(this);
>-  }
>-}
>-
>-inline FakeVideoMediaChannel::~FakeVideoMediaChannel() {
>-  if (engine_) {
>-    engine_->UnregisterChannel(this);
>-  }
>-}
> 
> class FakeDataEngine : public DataEngineInterface {
>  public:
>-  virtual DataMediaChannel* CreateChannel(const MediaConfig& config) {
>-    FakeDataMediaChannel* ch = new FakeDataMediaChannel(this, DataOptions());
>-    channels_.push_back(ch);
>-    return ch;
>-  }
>+  DataMediaChannel* CreateChannel(const MediaConfig& config) override;
> 
>-  FakeDataMediaChannel* GetChannel(size_t index) {
>-    return (channels_.size() > index) ? channels_[index] : NULL;
>-  }
>+  FakeDataMediaChannel* GetChannel(size_t index);
> 
>-  void UnregisterChannel(DataMediaChannel* channel) {
>-    channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
>-  }
>+  void UnregisterChannel(DataMediaChannel* channel);
> 
>-  virtual void SetDataCodecs(const std::vector<DataCodec>& data_codecs) {
>-    data_codecs_ = data_codecs;
>-  }
>+  void SetDataCodecs(const std::vector<DataCodec>& data_codecs);
> 
>-  virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
>+  const std::vector<DataCodec>& data_codecs() override;
> 
>  private:
>   std::vector<FakeDataMediaChannel*> channels_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakenetworkinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakenetworkinterface.h
>index 0f14659c4606c9df6a760e6ee3383b2f97ac9a59..dfd340772b693392b15d3544b7c4c6d1d22fce55 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakenetworkinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakenetworkinterface.h
>@@ -169,9 +169,9 @@ class FakeNetworkInterface : public MediaChannel::NetworkInterface,
>         static_cast<rtc::TypedMessageData<rtc::CopyOnWriteBuffer>*>(msg->pdata);
>     if (dest_) {
>       if (msg->message_id == ST_RTP) {
>-        dest_->OnPacketReceived(&msg_data->data(), rtc::CreatePacketTime(0));
>+        dest_->OnPacketReceived(&msg_data->data(), rtc::TimeMicros());
>       } else {
>-        dest_->OnRtcpReceived(&msg_data->data(), rtc::CreatePacketTime(0));
>+        dest_->OnRtcpReceived(&msg_data->data(), rtc::TimeMicros());
>       }
>     }
>     delete msg_data;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideocapturer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideocapturer.cc
>index 94900ba37b6a54b00fa6267b69dbeb5238318457..b5d0189608a92352b8ed450eb3e67528032ffcaa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideocapturer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideocapturer.cc
>@@ -11,6 +11,7 @@
> #include "media/base/fakevideocapturer.h"
> 
> #include "rtc_base/arraysize.h"
>+#include "rtc_base/timeutils.h"
> 
> namespace cricket {
> 
>@@ -95,7 +96,7 @@ cricket::CaptureState FakeVideoCapturer::Start(
>   SetCaptureState(cricket::CS_RUNNING);
>   frame_source_ = absl::make_unique<FakeFrameSource>(
>       format.width, format.height,
>-      format.interval / rtc::kNumNanosecsPerMicrosec);
>+      format.interval / rtc::kNumNanosecsPerMicrosec, rtc::TimeMicros());
>   frame_source_->SetRotation(rotation_);
>   return cricket::CS_RUNNING;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.h
>index b4ff88369c61489f4f851b95c546c271a9c0fdf8..6a3e7cdb321966f693bf83fb90a61a94c4739b85 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.h
>@@ -16,6 +16,7 @@
> 
> #include "absl/types/optional.h"
> #include "common_types.h"  // NOLINT(build/include)
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> namespace H264 {
>@@ -61,18 +62,19 @@ absl::optional<ProfileLevelId> ParseProfileLevelId(const char* str);
> // contained in an SDP key-value map. A default profile level id will be
> // returned if the profile-level-id key is missing. Nothing will be returned if
> // the key is present but the string is invalid.
>-absl::optional<ProfileLevelId> ParseSdpProfileLevelId(
>+RTC_EXPORT absl::optional<ProfileLevelId> ParseSdpProfileLevelId(
>     const CodecParameterMap& params);
> 
> // Given that a decoder supports up to a given frame size (in pixels) at up to a
> // given number of frames per second, return the highest H.264 level where it
> // can guarantee that it will be able to support all valid encoded streams that
> // are within that level.
>-absl::optional<Level> SupportedLevel(int max_frame_pixel_count, float max_fps);
>+RTC_EXPORT absl::optional<Level> SupportedLevel(int max_frame_pixel_count,
>+                                                float max_fps);
> 
> // Returns canonical string representation as three hex bytes of the profile
> // level id, or returns nothing for invalid profile level ids.
>-absl::optional<std::string> ProfileLevelIdToString(
>+RTC_EXPORT absl::optional<std::string> ProfileLevelIdToString(
>     const ProfileLevelId& profile_level_id);
> 
> // Generate codec parameters that will be used as answer in an SDP negotiation
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.cc
>index cba3be30ed1616ce3375acaa0a9d57637698cf9e..0634b2e575f8b2ebaffc2299e12b7e7d0944fb74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.cc
>@@ -16,16 +16,19 @@ VideoOptions::VideoOptions() = default;
> VideoOptions::~VideoOptions() = default;
> 
> MediaChannel::MediaChannel(const MediaConfig& config)
>-    : enable_dscp_(config.enable_dscp), network_interface_(NULL) {}
>+    : enable_dscp_(config.enable_dscp) {}
> 
>-MediaChannel::MediaChannel() : enable_dscp_(false), network_interface_(NULL) {}
>+MediaChannel::MediaChannel() : enable_dscp_(false) {}
> 
> MediaChannel::~MediaChannel() {}
> 
>-void MediaChannel::SetInterface(NetworkInterface* iface) {
>+void MediaChannel::SetInterface(
>+    NetworkInterface* iface,
>+    webrtc::MediaTransportInterface* media_transport) {
>   rtc::CritScope cs(&network_interface_crit_);
>   network_interface_ = iface;
>-  SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT);
>+  media_transport_ = media_transport;
>+  UpdateDscp();
> }
> 
> int MediaChannel::GetRtpSendTimeExtnId() const {
>@@ -84,6 +87,10 @@ std::map<std::string, std::string> AudioSendParameters::ToStringMap() const {
>   return params;
> }
> 
>+cricket::MediaType VoiceMediaChannel::media_type() const {
>+  return cricket::MediaType::MEDIA_TYPE_AUDIO;
>+}
>+
> VideoSendParameters::VideoSendParameters() = default;
> VideoSendParameters::~VideoSendParameters() = default;
> 
>@@ -93,11 +100,19 @@ std::map<std::string, std::string> VideoSendParameters::ToStringMap() const {
>   return params;
> }
> 
>+cricket::MediaType VideoMediaChannel::media_type() const {
>+  return cricket::MediaType::MEDIA_TYPE_VIDEO;
>+}
>+
> DataMediaChannel::DataMediaChannel() = default;
> DataMediaChannel::DataMediaChannel(const MediaConfig& config)
>     : MediaChannel(config) {}
> DataMediaChannel::~DataMediaChannel() = default;
> 
>+cricket::MediaType DataMediaChannel::media_type() const {
>+  return cricket::MediaType::MEDIA_TYPE_DATA;
>+}
>+
> bool DataMediaChannel::GetStats(DataMediaInfo* info) {
>   return true;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>index ff3368cf401d5f78af0779c547cf0b2ed7cb282e..fca6ee4dd9be5a55fb25857028e300a4f8e7a085 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>@@ -22,6 +22,7 @@
> #include "api/audio_options.h"
> #include "api/crypto/framedecryptorinterface.h"
> #include "api/crypto/frameencryptorinterface.h"
>+#include "api/media_transport_interface.h"
> #include "api/rtcerror.h"
> #include "api/rtpparameters.h"
> #include "api/rtpreceiverinterface.h"
>@@ -183,14 +184,22 @@ class MediaChannel : public sigslot::has_slots<> {
>   MediaChannel();
>   ~MediaChannel() override;
> 
>-  // Sets the abstract interface class for sending RTP/RTCP data.
>-  virtual void SetInterface(NetworkInterface* iface);
>+  virtual cricket::MediaType media_type() const = 0;
>+
>+  // Sets the abstract interface class for sending RTP/RTCP data and
>+  // interface for media transport (experimental). If media transport is
>+  // provided, it should be used instead of RTP/RTCP.
>+  // TODO(sukhanov): Currently media transport can co-exist with RTP/RTCP, but
>+  // in the future we will refactor code to send all frames with media
>+  // transport.
>+  virtual void SetInterface(NetworkInterface* iface,
>+                            webrtc::MediaTransportInterface* media_transport);
>   // Called when a RTP packet is received.
>   virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                const rtc::PacketTime& packet_time) = 0;
>+                                int64_t packet_time_us) = 0;
>   // Called when a RTCP packet is received.
>   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
>-                              const rtc::PacketTime& packet_time) = 0;
>+                              int64_t packet_time_us) = 0;
>   // Called when the socket's ability to send has changed.
>   virtual void OnReadyToSend(bool ready) = 0;
>   // Called when the network route used for sending packets changed.
>@@ -251,14 +260,28 @@ class MediaChannel : public sigslot::has_slots<> {
>     return network_interface_->SetOption(type, opt, option);
>   }
> 
>+  webrtc::MediaTransportInterface* media_transport() {
>+    return media_transport_;
>+  }
>+
>+  // Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
>+  // Set to true if it's allowed to mix one- and two-byte RTP header extensions
>+  // in the same stream. The setter and getter must only be called from
>+  // worker_thread.
>+  void SetExtmapAllowMixed(bool extmap_allow_mixed) {
>+    extmap_allow_mixed_ = extmap_allow_mixed;
>+  }
>+  bool ExtmapAllowMixed() const { return extmap_allow_mixed_; }
>+
>  protected:
>   virtual rtc::DiffServCodePoint PreferredDscp() const;
> 
>   bool DscpEnabled() const { return enable_dscp_; }
> 
>- private:
>   // This method sets DSCP |value| on both RTP and RTCP channels.
>-  int SetDscp(rtc::DiffServCodePoint value) {
>+  int UpdateDscp() {
>+    rtc::DiffServCodePoint value =
>+        enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT;
>     int ret;
>     ret = SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_DSCP, value);
>     if (ret == 0) {
>@@ -267,6 +290,7 @@ class MediaChannel : public sigslot::has_slots<> {
>     return ret;
>   }
> 
>+ private:
>   bool DoSendPacket(rtc::CopyOnWriteBuffer* packet,
>                     bool rtcp,
>                     const rtc::PacketOptions& options) {
>@@ -283,7 +307,9 @@ class MediaChannel : public sigslot::has_slots<> {
>   // from any MediaEngine threads. This critical section is to protect accessing
>   // of network_interface_ object.
>   rtc::CriticalSection network_interface_crit_;
>-  NetworkInterface* network_interface_;
>+  NetworkInterface* network_interface_ = nullptr;
>+  webrtc::MediaTransportInterface* media_transport_ = nullptr;
>+  bool extmap_allow_mixed_ = false;
> };
> 
> // The stats information is structured as follows:
>@@ -402,14 +428,6 @@ struct VoiceSenderInfo : public MediaSenderInfo {
>   // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
>   double total_input_energy = 0.0;
>   double total_input_duration = 0.0;
>-  // TODO(bugs.webrtc.org/8572): Remove APM stats from this struct, since they
>-  // are no longer needed now that we have apm_statistics.
>-  int echo_delay_median_ms = 0;
>-  int echo_delay_std_ms = 0;
>-  int echo_return_loss = 0;
>-  int echo_return_loss_enhancement = 0;
>-  float residual_echo_likelihood = 0.0f;
>-  float residual_echo_likelihood_recent_max = 0.0f;
>   bool typing_noise_detected = false;
>   webrtc::ANAStats ana_statistics;
>   webrtc::AudioProcessingStats apm_statistics;
>@@ -458,6 +476,10 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
>   int decoding_muted_output = 0;
>   // Estimated capture start time in NTP time in ms.
>   int64_t capture_start_ntp_time_ms = -1;
>+  // Count of the number of buffer flushes.
>+  uint64_t jitter_buffer_flushes = 0;
>+  // Number of samples expanded due to delayed packets.
>+  uint64_t delayed_packet_outage_samples = 0;
> };
> 
> struct VideoSenderInfo : public MediaSenderInfo {
>@@ -647,12 +669,14 @@ struct RtpSendParameters : RtpParameters<Codec> {
>   // This is the value to be sent in the MID RTP header extension (if the header
>   // extension in included in the list of extensions).
>   std::string mid;
>+  bool extmap_allow_mixed = false;
> 
>  protected:
>   std::map<std::string, std::string> ToStringMap() const override {
>     auto params = RtpParameters<Codec>::ToStringMap();
>     params["max_bandwidth_bps"] = rtc::ToString(max_bandwidth_bps);
>     params["mid"] = (mid.empty() ? "<not set>" : mid);
>+    params["extmap-allow-mixed"] = extmap_allow_mixed ? "true" : "false";
>     return params;
>   }
> };
>@@ -674,6 +698,8 @@ class VoiceMediaChannel : public MediaChannel {
>   explicit VoiceMediaChannel(const MediaConfig& config)
>       : MediaChannel(config) {}
>   ~VoiceMediaChannel() override {}
>+
>+  cricket::MediaType media_type() const override;
>   virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
>   virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
>   virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
>@@ -746,6 +772,7 @@ class VideoMediaChannel : public MediaChannel {
>       : MediaChannel(config) {}
>   ~VideoMediaChannel() override {}
> 
>+  cricket::MediaType media_type() const override;
>   virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
>   virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
>   virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
>@@ -857,6 +884,7 @@ class DataMediaChannel : public MediaChannel {
>   explicit DataMediaChannel(const MediaConfig& config);
>   ~DataMediaChannel() override;
> 
>+  cricket::MediaType media_type() const override;
>   virtual bool SetSendParameters(const DataSendParameters& params) = 0;
>   virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconfig.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconfig.h
>index eda387e319112b321f2fd8e68b68f23d5b0bf089..c8fc5212a0f8cbc5b27386b587ef46053ca5fb79 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconfig.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconfig.h
>@@ -59,8 +59,17 @@ struct MediaConfig {
>     // TODO(bugs.webrtc.org/8504): If all goes well, the flag will be removed
>     // together with the old method of estimation.
>     bool experiment_cpu_load_estimator = false;
>+
>+    // Time interval between RTCP report for video
>+    int rtcp_report_interval_ms = 1000;
>   } video;
> 
>+  // Audio-specific config.
>+  struct Audio {
>+    // Time interval between RTCP report for audio
>+    int rtcp_report_interval_ms = 5000;
>+  } audio;
>+
>   bool operator==(const MediaConfig& o) const {
>     return enable_dscp == o.enable_dscp &&
>            video.enable_cpu_adaptation == o.video.enable_cpu_adaptation &&
>@@ -71,7 +80,9 @@ struct MediaConfig {
>            video.periodic_alr_bandwidth_probing ==
>                o.video.periodic_alr_bandwidth_probing &&
>            video.experiment_cpu_load_estimator ==
>-               o.video.experiment_cpu_load_estimator;
>+               o.video.experiment_cpu_load_estimator &&
>+           video.rtcp_report_interval_ms == o.video.rtcp_report_interval_ms &&
>+           audio.rtcp_report_interval_ms == o.audio.rtcp_report_interval_ms;
>   }
> 
>   bool operator!=(const MediaConfig& o) const { return !(*this == o); }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.h
>index 14c49baf72cbad1c5a92028b479a0b173c0dabcd..c4a3406fd85a23a8aaba3403c319c6e811f5a6d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.h
>@@ -13,6 +13,8 @@
> 
> #include <string>
> 
>+#include "rtc_base/system/rtc_export.h"
>+
> // This file contains constants related to media.
> 
> namespace cricket {
>@@ -127,12 +129,12 @@ extern const char kComfortNoiseCodecName[];
> 
> extern const char kVp8CodecName[];
> extern const char kVp9CodecName[];
>-extern const char kH264CodecName[];
>+RTC_EXPORT extern const char kH264CodecName[];
> 
> // RFC 6184 RTP Payload Format for H.264 video
>-extern const char kH264FmtpProfileLevelId[];
>-extern const char kH264FmtpLevelAsymmetryAllowed[];
>-extern const char kH264FmtpPacketizationMode[];
>+RTC_EXPORT extern const char kH264FmtpProfileLevelId[];
>+RTC_EXPORT extern const char kH264FmtpLevelAsymmetryAllowed[];
>+RTC_EXPORT extern const char kH264FmtpPacketizationMode[];
> extern const char kH264FmtpSpropParameterSets[];
> extern const char kH264ProfileLevelConstrainedBaseline[];
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.cc
>index 76cac4b438342e0abe226a72765d04629d5823b7..d5198fe64f1901e003df4822939834dcf1bcef6a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.cc
>@@ -10,6 +10,8 @@
> 
> #include "media/base/mediaengine.h"
> 
>+#include <utility>
>+
> #include "api/video/video_bitrate_allocation.h"
> #include "rtc_base/stringencode.h"
> 
>@@ -104,4 +106,33 @@ webrtc::RTCError ValidateRtpParameters(
>   return webrtc::RTCError::OK();
> }
> 
>+CompositeMediaEngine::CompositeMediaEngine(
>+    std::unique_ptr<VoiceEngineInterface> voice_engine,
>+    std::unique_ptr<VideoEngineInterface> video_engine)
>+    : voice_engine_(std::move(voice_engine)),
>+      video_engine_(std::move(video_engine)) {}
>+
>+CompositeMediaEngine::~CompositeMediaEngine() = default;
>+
>+bool CompositeMediaEngine::Init() {
>+  voice().Init();
>+  return true;
>+}
>+
>+VoiceEngineInterface& CompositeMediaEngine::voice() {
>+  return *voice_engine_.get();
>+}
>+
>+VideoEngineInterface& CompositeMediaEngine::video() {
>+  return *video_engine_.get();
>+}
>+
>+const VoiceEngineInterface& CompositeMediaEngine::voice() const {
>+  return *voice_engine_.get();
>+}
>+
>+const VideoEngineInterface& CompositeMediaEngine::video() const {
>+  return *video_engine_.get();
>+}
>+
> };  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.h
>index 01c6fb65269ea2e726124de9abb7289d56654bdb..01300d48ac370b7a95a6daa788a6f492c46c7a3c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.h
>@@ -15,13 +15,13 @@
> #include <CoreAudio/CoreAudio.h>
> #endif
> 
>+#include <memory>
> #include <string>
>-#include <tuple>
>-#include <utility>
> #include <vector>
> 
> #include "api/audio_codecs/audio_decoder_factory.h"
> #include "api/audio_codecs/audio_encoder_factory.h"
>+#include "api/crypto/cryptooptions.h"
> #include "api/rtpparameters.h"
> #include "call/audio_state.h"
> #include "media/base/codec.h"
>@@ -48,37 +48,30 @@ struct RtpCapabilities {
>   std::vector<webrtc::RtpExtension> header_extensions;
> };
> 
>-// MediaEngineInterface is an abstraction of a media engine which can be
>-// subclassed to support different media componentry backends.
>-// It supports voice and video operations in the same class to facilitate
>-// proper synchronization between both media types.
>-class MediaEngineInterface {
>+class VoiceEngineInterface {
>  public:
>-  virtual ~MediaEngineInterface() {}
>+  VoiceEngineInterface() = default;
>+  virtual ~VoiceEngineInterface() = default;
>+  RTC_DISALLOW_COPY_AND_ASSIGN(VoiceEngineInterface);
> 
>   // Initialization
>   // Starts the engine.
>-  virtual bool Init() = 0;
>+  virtual void Init() = 0;
>+
>   // TODO(solenberg): Remove once VoE API refactoring is done.
>   virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
> 
>   // MediaChannel creation
>   // Creates a voice media channel. Returns NULL on failure.
>-  virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
>-                                           const MediaConfig& config,
>-                                           const AudioOptions& options) = 0;
>-  // Creates a video media channel, paired with the specified voice channel.
>-  // Returns NULL on failure.
>-  virtual VideoMediaChannel* CreateVideoChannel(
>+  virtual VoiceMediaChannel* CreateMediaChannel(
>       webrtc::Call* call,
>       const MediaConfig& config,
>-      const VideoOptions& options) = 0;
>+      const AudioOptions& options,
>+      const webrtc::CryptoOptions& crypto_options) = 0;
> 
>-  virtual const std::vector<AudioCodec>& audio_send_codecs() = 0;
>-  virtual const std::vector<AudioCodec>& audio_recv_codecs() = 0;
>-  virtual RtpCapabilities GetAudioCapabilities() = 0;
>-  virtual std::vector<VideoCodec> video_codecs() = 0;
>-  virtual RtpCapabilities GetVideoCapabilities() = 0;
>+  virtual const std::vector<AudioCodec>& send_codecs() const = 0;
>+  virtual const std::vector<AudioCodec>& recv_codecs() const = 0;
>+  virtual RtpCapabilities GetCapabilities() const = 0;
> 
>   // Starts AEC dump using existing file, a maximum file size in bytes can be
>   // specified. Logging is stopped just before the size limit is exceeded.
>@@ -89,69 +82,66 @@ class MediaEngineInterface {
>   virtual void StopAecDump() = 0;
> };
> 
>+class VideoEngineInterface {
>+ public:
>+  VideoEngineInterface() = default;
>+  virtual ~VideoEngineInterface() = default;
>+  RTC_DISALLOW_COPY_AND_ASSIGN(VideoEngineInterface);
>+
>+  // Creates a video media channel, paired with the specified voice channel.
>+  // Returns NULL on failure.
>+  virtual VideoMediaChannel* CreateMediaChannel(
>+      webrtc::Call* call,
>+      const MediaConfig& config,
>+      const VideoOptions& options,
>+      const webrtc::CryptoOptions& crypto_options) = 0;
>+
>+  virtual std::vector<VideoCodec> codecs() const = 0;
>+  virtual RtpCapabilities GetCapabilities() const = 0;
>+};
>+
>+// MediaEngineInterface is an abstraction of a media engine which can be
>+// subclassed to support different media componentry backends.
>+// It supports voice and video operations in the same class to facilitate
>+// proper synchronization between both media types.
>+class MediaEngineInterface {
>+ public:
>+  virtual ~MediaEngineInterface() {}
>+
>+  // Initialization
>+  // Starts the engine.
>+  virtual bool Init() = 0;
>+  virtual VoiceEngineInterface& voice() = 0;
>+  virtual VideoEngineInterface& video() = 0;
>+  virtual const VoiceEngineInterface& voice() const = 0;
>+  virtual const VideoEngineInterface& video() const = 0;
>+};
>+
> // CompositeMediaEngine constructs a MediaEngine from separate
> // voice and video engine classes.
>-template <class VOICE, class VIDEO>
> class CompositeMediaEngine : public MediaEngineInterface {
>  public:
>-  template <class... Args1, class... Args2>
>-  CompositeMediaEngine(std::tuple<Args1...> first_args,
>-                       std::tuple<Args2...> second_args)
>-      : engines_(std::piecewise_construct,
>-                 std::move(first_args),
>-                 std::move(second_args)) {}
>-
>-  virtual ~CompositeMediaEngine() {}
>-  virtual bool Init() {
>-    voice().Init();
>-    return true;
>-  }
>-
>-  virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
>-    return voice().GetAudioState();
>-  }
>-  virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
>-                                           const MediaConfig& config,
>-                                           const AudioOptions& options) {
>-    return voice().CreateChannel(call, config, options);
>-  }
>-  virtual VideoMediaChannel* CreateVideoChannel(webrtc::Call* call,
>-                                                const MediaConfig& config,
>-                                                const VideoOptions& options) {
>-    return video().CreateChannel(call, config, options);
>-  }
>-
>-  virtual const std::vector<AudioCodec>& audio_send_codecs() {
>-    return voice().send_codecs();
>-  }
>-  virtual const std::vector<AudioCodec>& audio_recv_codecs() {
>-    return voice().recv_codecs();
>-  }
>-  virtual RtpCapabilities GetAudioCapabilities() {
>-    return voice().GetCapabilities();
>-  }
>-  virtual std::vector<VideoCodec> video_codecs() { return video().codecs(); }
>-  virtual RtpCapabilities GetVideoCapabilities() {
>-    return video().GetCapabilities();
>-  }
>-
>-  virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) {
>-    return voice().StartAecDump(file, max_size_bytes);
>-  }
>-
>-  virtual void StopAecDump() { voice().StopAecDump(); }
>-
>- protected:
>-  VOICE& voice() { return engines_.first; }
>-  VIDEO& video() { return engines_.second; }
>-  const VOICE& voice() const { return engines_.first; }
>-  const VIDEO& video() const { return engines_.second; }
>+  CompositeMediaEngine(std::unique_ptr<VoiceEngineInterface> audio_engine,
>+                       std::unique_ptr<VideoEngineInterface> video_engine);
>+  ~CompositeMediaEngine() override;
>+  bool Init() override;
>+
>+  VoiceEngineInterface& voice() override;
>+  VideoEngineInterface& video() override;
>+  const VoiceEngineInterface& voice() const override;
>+  const VideoEngineInterface& video() const override;
> 
>  private:
>-  std::pair<VOICE, VIDEO> engines_;
>+  std::unique_ptr<VoiceEngineInterface> voice_engine_;
>+  std::unique_ptr<VideoEngineInterface> video_engine_;
> };
> 
>-enum DataChannelType { DCT_NONE = 0, DCT_RTP = 1, DCT_SCTP = 2 };
>+enum DataChannelType {
>+  DCT_NONE = 0,
>+  DCT_RTP = 1,
>+  DCT_SCTP = 2,
>+  DCT_MEDIA_TRANSPORT = 3
>+};
> 
> class DataEngineInterface {
>  public:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.cc
>index fc2b73141edd75e08e6d66b35b64389cfda8f3e6..c3367c0a1f60b494f7a3313d1f552cb7ad6eded2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.cc
>@@ -12,6 +12,7 @@
> 
> #include <map>
> 
>+#include "absl/strings/match.h"
> #include "media/base/codec.h"
> #include "media/base/mediaconstants.h"
> #include "media/base/rtputils.h"
>@@ -21,7 +22,6 @@
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/sanitizer.h"
>-#include "rtc_base/stringutils.h"
> 
> namespace cricket {
> 
>@@ -47,7 +47,7 @@ DataMediaChannel* RtpDataEngine::CreateChannel(const MediaConfig& config) {
> static const DataCodec* FindCodecByName(const std::vector<DataCodec>& codecs,
>                                         const std::string& name) {
>   for (const DataCodec& codec : codecs) {
>-    if (_stricmp(name.c_str(), codec.name.c_str()) == 0)
>+    if (absl::EqualsIgnoreCase(name, codec.name))
>       return &codec;
>   }
>   return nullptr;
>@@ -194,7 +194,7 @@ bool RtpDataMediaChannel::RemoveRecvStream(uint32_t ssrc) {
> }
> 
> void RtpDataMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                           const rtc::PacketTime& packet_time) {
>+                                           int64_t /* packet_time_us */) {
>   RtpHeader header;
>   if (!GetRtpHeader(packet->cdata(), packet->size(), &header)) {
>     return;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.h
>index 966d0c81514157d2e07a3c77d54f63f32e30b6f3..42e919039d013f9cfa86379e3aaaa2dad277c782 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.h
>@@ -82,9 +82,9 @@ class RtpDataMediaChannel : public DataMediaChannel {
>     return true;
>   }
>   virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                const rtc::PacketTime& packet_time);
>+                                int64_t packet_time_us);
>   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
>-                              const rtc::PacketTime& packet_time) {}
>+                              int64_t packet_time_us) {}
>   virtual void OnReadyToSend(bool ready) {}
>   virtual bool SendData(const SendDataParams& params,
>                         const rtc::CopyOnWriteBuffer& payload,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine_unittest.cc
>index c15c55f5e6274e88d357f9d960de6c16a7e88517..9b8e10b45007cced6698a5e244203e08b6dd45d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine_unittest.cc
>@@ -73,7 +73,7 @@ class RtpDataMediaChannelTest : public testing::Test {
>     cricket::MediaConfig config;
>     cricket::RtpDataMediaChannel* channel =
>         static_cast<cricket::RtpDataMediaChannel*>(dme->CreateChannel(config));
>-    channel->SetInterface(iface_.get());
>+    channel->SetInterface(iface_.get(), /*media_transport=*/nullptr);
>     channel->SignalDataReceived.connect(receiver_.get(),
>                                         &FakeDataReceiver::OnDataReceived);
>     return channel;
>@@ -317,13 +317,13 @@ TEST_F(RtpDataMediaChannelTest, ReceiveData) {
>   std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
> 
>   // SetReceived not called.
>-  dmc->OnPacketReceived(&packet, rtc::PacketTime());
>+  dmc->OnPacketReceived(&packet, /* packet_time_us */ -1);
>   EXPECT_FALSE(HasReceivedData());
> 
>   dmc->SetReceive(true);
> 
>   // Unknown payload id
>-  dmc->OnPacketReceived(&packet, rtc::PacketTime());
>+  dmc->OnPacketReceived(&packet, /* packet_time_us */ -1);
>   EXPECT_FALSE(HasReceivedData());
> 
>   cricket::DataCodec codec;
>@@ -334,7 +334,7 @@ TEST_F(RtpDataMediaChannelTest, ReceiveData) {
>   ASSERT_TRUE(dmc->SetRecvParameters(parameters));
> 
>   // Unknown stream
>-  dmc->OnPacketReceived(&packet, rtc::PacketTime());
>+  dmc->OnPacketReceived(&packet, /* packet_time_us */ -1);
>   EXPECT_FALSE(HasReceivedData());
> 
>   cricket::StreamParams stream;
>@@ -342,7 +342,7 @@ TEST_F(RtpDataMediaChannelTest, ReceiveData) {
>   ASSERT_TRUE(dmc->AddRecvStream(stream));
> 
>   // Finally works!
>-  dmc->OnPacketReceived(&packet, rtc::PacketTime());
>+  dmc->OnPacketReceived(&packet, /* packet_time_us */ -1);
>   EXPECT_TRUE(HasReceivedData());
>   EXPECT_EQ("abcde", GetReceivedData());
>   EXPECT_EQ(5U, GetReceivedDataLen());
>@@ -355,6 +355,6 @@ TEST_F(RtpDataMediaChannelTest, InvalidRtpPackets) {
>   std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
> 
>   // Too short
>-  dmc->OnPacketReceived(&packet, rtc::PacketTime());
>+  dmc->OnPacketReceived(&packet, /* packet_time_us */ -1);
>   EXPECT_FALSE(HasReceivedData());
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.h
>index 0fd9a59816b6e7e4c1a8ab3c24e954db010d6b28..c5fe14fb3d381788937d15ab0027d1b256c856b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.h
>@@ -12,6 +12,7 @@
> #define MEDIA_BASE_RTPUTILS_H_
> 
> #include "rtc_base/byteorder.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace rtc {
> struct PacketTimeUpdateParams;
>@@ -68,9 +69,9 @@ bool IsValidRtpRtcpPacketSize(bool rtcp, size_t size);
> const char* RtpRtcpStringLiteral(bool rtcp);
> 
> // Verifies that a packet has a valid RTP header.
>-bool ValidateRtpHeader(const uint8_t* rtp,
>-                       size_t length,
>-                       size_t* header_length);
>+bool RTC_EXPORT ValidateRtpHeader(const uint8_t* rtp,
>+                                  size_t length,
>+                                  size_t* header_length);
> 
> // Helper method which updates the absolute send time extension if present.
> bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
>@@ -80,10 +81,11 @@ bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
> 
> // Applies specified |options| to the packet. It updates the absolute send time
> // extension header if it is present present then updates HMAC.
>-bool ApplyPacketOptions(uint8_t* data,
>-                        size_t length,
>-                        const rtc::PacketTimeUpdateParams& packet_time_params,
>-                        uint64_t time_us);
>+bool RTC_EXPORT
>+ApplyPacketOptions(uint8_t* data,
>+                   size_t length,
>+                   const rtc::PacketTimeUpdateParams& packet_time_params,
>+                   uint64_t time_us);
> 
> }  // namespace cricket
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.cc
>index 93c2bc98aef32b870772b35150b2849ef69067f5..512dc91f93c31b33fea3c454e827fc21e99c24d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.cc
>@@ -10,90 +10,14 @@
> 
> #include "media/base/testutils.h"
> 
>-#include <math.h>
> #include <algorithm>
> #include <memory>
> 
> #include "api/video/video_frame.h"
> #include "media/base/videocapturer.h"
>-#include "rtc_base/bytebuffer.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/stream.h"
>-#include "rtc_base/stringutils.h"
>-#include "rtc_base/testutils.h"
> 
> namespace cricket {
> 
>-/////////////////////////////////////////////////////////////////////////
>-// Implementation of RawRtpPacket
>-/////////////////////////////////////////////////////////////////////////
>-void RawRtpPacket::WriteToByteBuffer(uint32_t in_ssrc,
>-                                     rtc::ByteBufferWriter* buf) const {
>-  if (!buf)
>-    return;
>-
>-  buf->WriteUInt8(ver_to_cc);
>-  buf->WriteUInt8(m_to_pt);
>-  buf->WriteUInt16(sequence_number);
>-  buf->WriteUInt32(timestamp);
>-  buf->WriteUInt32(in_ssrc);
>-  buf->WriteBytes(payload, sizeof(payload));
>-}
>-
>-bool RawRtpPacket::ReadFromByteBuffer(rtc::ByteBufferReader* buf) {
>-  if (!buf)
>-    return false;
>-
>-  bool ret = true;
>-  ret &= buf->ReadUInt8(&ver_to_cc);
>-  ret &= buf->ReadUInt8(&m_to_pt);
>-  ret &= buf->ReadUInt16(&sequence_number);
>-  ret &= buf->ReadUInt32(&timestamp);
>-  ret &= buf->ReadUInt32(&ssrc);
>-  ret &= buf->ReadBytes(payload, sizeof(payload));
>-  return ret;
>-}
>-
>-bool RawRtpPacket::SameExceptSeqNumTimestampSsrc(const RawRtpPacket& packet,
>-                                                 uint16_t seq,
>-                                                 uint32_t ts,
>-                                                 uint32_t ssc) const {
>-  return sequence_number == seq && timestamp == ts &&
>-         ver_to_cc == packet.ver_to_cc && m_to_pt == packet.m_to_pt &&
>-         ssrc == ssc && 0 == memcmp(payload, packet.payload, sizeof(payload));
>-}
>-
>-/////////////////////////////////////////////////////////////////////////
>-// Implementation of RawRtcpPacket
>-/////////////////////////////////////////////////////////////////////////
>-void RawRtcpPacket::WriteToByteBuffer(rtc::ByteBufferWriter* buf) const {
>-  if (!buf)
>-    return;
>-
>-  buf->WriteUInt8(ver_to_count);
>-  buf->WriteUInt8(type);
>-  buf->WriteUInt16(length);
>-  buf->WriteBytes(payload, sizeof(payload));
>-}
>-
>-bool RawRtcpPacket::ReadFromByteBuffer(rtc::ByteBufferReader* buf) {
>-  if (!buf)
>-    return false;
>-
>-  bool ret = true;
>-  ret &= buf->ReadUInt8(&ver_to_count);
>-  ret &= buf->ReadUInt8(&type);
>-  ret &= buf->ReadUInt16(&length);
>-  ret &= buf->ReadBytes(payload, sizeof(payload));
>-  return ret;
>-}
>-
>-bool RawRtcpPacket::EqualsTo(const RawRtcpPacket& packet) const {
>-  return ver_to_count == packet.ver_to_count && type == packet.type &&
>-         length == packet.length &&
>-         0 == memcmp(payload, packet.payload, sizeof(payload));
>-}
>-
> // Implementation of VideoCaptureListener.
> VideoCapturerListener::VideoCapturerListener(VideoCapturer* capturer)
>     : capturer_(capturer),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.h
>index c84696236b4a402fc4c46e341da29836136d3536..6b63ffd180003646caa54af755754b7856b5e835 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.h
>@@ -20,12 +20,6 @@
> #include "rtc_base/arraysize.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
> 
>-namespace rtc {
>-class ByteBufferReader;
>-class ByteBufferWriter;
>-class StreamInterface;
>-}  // namespace rtc
>-
> namespace webrtc {
> class VideoFrame;
> }
>@@ -43,40 +37,6 @@ inline std::vector<T> MakeVector(const T a[], size_t s) {
> }
> #define MAKE_VECTOR(a) cricket::MakeVector(a, arraysize(a))
> 
>-struct RtpDumpPacket;
>-class RtpDumpWriter;
>-
>-struct RawRtpPacket {
>-  void WriteToByteBuffer(uint32_t in_ssrc, rtc::ByteBufferWriter* buf) const;
>-  bool ReadFromByteBuffer(rtc::ByteBufferReader* buf);
>-  // Check if this packet is the same as the specified packet except the
>-  // sequence number and timestamp, which should be the same as the specified
>-  // parameters.
>-  bool SameExceptSeqNumTimestampSsrc(const RawRtpPacket& packet,
>-                                     uint16_t seq,
>-                                     uint32_t ts,
>-                                     uint32_t ssc) const;
>-  int size() const { return 28; }
>-
>-  uint8_t ver_to_cc;
>-  uint8_t m_to_pt;
>-  uint16_t sequence_number;
>-  uint32_t timestamp;
>-  uint32_t ssrc;
>-  char payload[16];
>-};
>-
>-struct RawRtcpPacket {
>-  void WriteToByteBuffer(rtc::ByteBufferWriter* buf) const;
>-  bool ReadFromByteBuffer(rtc::ByteBufferReader* buf);
>-  bool EqualsTo(const RawRtcpPacket& packet) const;
>-
>-  uint8_t ver_to_count;
>-  uint8_t type;
>-  uint16_t length;
>-  char payload[16];
>-};
>-
> // Test helper for testing VideoCapturer implementations.
> class VideoCapturerListener
>     : public sigslot::has_slots<>,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/turnutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/turnutils.h
>index 13ed26b7a5ff7cc27569c266a8dd64522cffb623..7aa0651cea3ea559f4beb13b53b422d157f4ea2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/turnutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/turnutils.h
>@@ -14,16 +14,18 @@
> #include <cstddef>
> #include <cstdint>
> 
>+#include "rtc_base/system/rtc_export.h"
>+
> namespace cricket {
> 
> struct PacketOptions;
> 
> // Finds data location within a TURN Channel Message or TURN Send Indication
> // message.
>-bool UnwrapTurnPacket(const uint8_t* packet,
>-                      size_t packet_size,
>-                      size_t* content_position,
>-                      size_t* content_size);
>+bool RTC_EXPORT UnwrapTurnPacket(const uint8_t* packet,
>+                                 size_t packet_size,
>+                                 size_t* content_position,
>+                                 size_t* content_size);
> 
> }  // namespace cricket
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.cc
>index 840101bcee3403d64bb8f91f75c4d51ebd04b72d..b2ed333614930887663b574f3fe58c05cb621095 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.cc
>@@ -169,8 +169,18 @@ bool VideoAdapter::AdaptFrameResolution(int in_width,
>   // OnOutputFormatRequest and OnResolutionFramerateRequest.
>   int max_pixel_count = resolution_request_max_pixel_count_;
> 
>-  if (max_pixel_count_)
>-    max_pixel_count = std::min(max_pixel_count, *max_pixel_count_);
>+  // Select target aspect ratio and max pixel count depending on input frame
>+  // orientation.
>+  absl::optional<std::pair<int, int>> target_aspect_ratio;
>+  if (in_width > in_height) {
>+    target_aspect_ratio = target_landscape_aspect_ratio_;
>+    if (max_landscape_pixel_count_)
>+      max_pixel_count = std::min(max_pixel_count, *max_landscape_pixel_count_);
>+  } else {
>+    target_aspect_ratio = target_portrait_aspect_ratio_;
>+    if (max_portrait_pixel_count_)
>+      max_pixel_count = std::min(max_pixel_count, *max_portrait_pixel_count_);
>+  }
> 
>   int target_pixel_count =
>       std::min(resolution_request_target_pixel_count_, max_pixel_count);
>@@ -195,19 +205,14 @@ bool VideoAdapter::AdaptFrameResolution(int in_width,
>   }
> 
>   // Calculate how the input should be cropped.
>-  if (!target_aspect_ratio_ || target_aspect_ratio_->first <= 0 ||
>-      target_aspect_ratio_->second <= 0) {
>+  if (!target_aspect_ratio || target_aspect_ratio->first <= 0 ||
>+      target_aspect_ratio->second <= 0) {
>     *cropped_width = in_width;
>     *cropped_height = in_height;
>   } else {
>-    // Adjust |target_aspect_ratio_| orientation to match input.
>-    if ((in_width > in_height) !=
>-        (target_aspect_ratio_->first > target_aspect_ratio_->second)) {
>-      std::swap(target_aspect_ratio_->first, target_aspect_ratio_->second);
>-    }
>     const float requested_aspect =
>-        target_aspect_ratio_->first /
>-        static_cast<float>(target_aspect_ratio_->second);
>+        target_aspect_ratio->first /
>+        static_cast<float>(target_aspect_ratio->second);
>     *cropped_width =
>         std::min(in_width, static_cast<int>(in_height * requested_aspect));
>     *cropped_height =
>@@ -274,9 +279,33 @@ void VideoAdapter::OnOutputFormatRequest(
>     const absl::optional<std::pair<int, int>>& target_aspect_ratio,
>     const absl::optional<int>& max_pixel_count,
>     const absl::optional<int>& max_fps) {
>+  absl::optional<std::pair<int, int>> target_landscape_aspect_ratio;
>+  absl::optional<std::pair<int, int>> target_portrait_aspect_ratio;
>+  if (target_aspect_ratio && target_aspect_ratio->first > 0 &&
>+      target_aspect_ratio->second > 0) {
>+    // Maintain input orientation.
>+    const int max_side =
>+        std::max(target_aspect_ratio->first, target_aspect_ratio->second);
>+    const int min_side =
>+        std::min(target_aspect_ratio->first, target_aspect_ratio->second);
>+    target_landscape_aspect_ratio = std::make_pair(max_side, min_side);
>+    target_portrait_aspect_ratio = std::make_pair(min_side, max_side);
>+  }
>+  OnOutputFormatRequest(target_landscape_aspect_ratio, max_pixel_count,
>+                        target_portrait_aspect_ratio, max_pixel_count, max_fps);
>+}
>+
>+void VideoAdapter::OnOutputFormatRequest(
>+    const absl::optional<std::pair<int, int>>& target_landscape_aspect_ratio,
>+    const absl::optional<int>& max_landscape_pixel_count,
>+    const absl::optional<std::pair<int, int>>& target_portrait_aspect_ratio,
>+    const absl::optional<int>& max_portrait_pixel_count,
>+    const absl::optional<int>& max_fps) {
>   rtc::CritScope cs(&critical_section_);
>-  target_aspect_ratio_ = target_aspect_ratio;
>-  max_pixel_count_ = max_pixel_count;
>+  target_landscape_aspect_ratio_ = target_landscape_aspect_ratio;
>+  max_landscape_pixel_count_ = max_landscape_pixel_count;
>+  target_portrait_aspect_ratio_ = target_portrait_aspect_ratio;
>+  max_portrait_pixel_count_ = max_portrait_pixel_count;
>   max_fps_ = max_fps;
>   next_frame_timestamp_ns_ = absl::nullopt;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.h
>index bb9ab8026f088acd94329491ca25d4a62ea946fa..1864d5cc0c7bf358d95bbda3c5fbe205407809ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.h
>@@ -68,6 +68,17 @@ class VideoAdapter {
>       const absl::optional<int>& max_pixel_count,
>       const absl::optional<int>& max_fps);
> 
>+  // Same as above, but allows setting two different target aspect ratios
>+  // depending on incoming frame orientation. This gives more fine-grained
>+  // control and can e.g. be used to force landscape video to be cropped to
>+  // portrait video.
>+  void OnOutputFormatRequest(
>+      const absl::optional<std::pair<int, int>>& target_landscape_aspect_ratio,
>+      const absl::optional<int>& max_landscape_pixel_count,
>+      const absl::optional<std::pair<int, int>>& target_portrait_aspect_ratio,
>+      const absl::optional<int>& max_portrait_pixel_count,
>+      const absl::optional<int>& max_fps);
>+
>   // Requests the output frame size from |AdaptFrameResolution| to have as close
>   // as possible to |target_pixel_count| pixels (if set) but no more than
>   // |max_pixel_count|.
>@@ -100,9 +111,14 @@ class VideoAdapter {
>   // Max number of pixels/fps requested via calls to OnOutputFormatRequest,
>   // OnResolutionFramerateRequest respectively.
>   // The adapted output format is the minimum of these.
>-  absl::optional<std::pair<int, int>> target_aspect_ratio_
>+  absl::optional<std::pair<int, int>> target_landscape_aspect_ratio_
>+      RTC_GUARDED_BY(critical_section_);
>+  absl::optional<int> max_landscape_pixel_count_
>+      RTC_GUARDED_BY(critical_section_);
>+  absl::optional<std::pair<int, int>> target_portrait_aspect_ratio_
>+      RTC_GUARDED_BY(critical_section_);
>+  absl::optional<int> max_portrait_pixel_count_
>       RTC_GUARDED_BY(critical_section_);
>-  absl::optional<int> max_pixel_count_ RTC_GUARDED_BY(critical_section_);
>   absl::optional<int> max_fps_ RTC_GUARDED_BY(critical_section_);
>   int resolution_request_target_pixel_count_ RTC_GUARDED_BY(critical_section_);
>   int resolution_request_max_pixel_count_ RTC_GUARDED_BY(critical_section_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter_unittest.cc
>index 67a662639e5514e44438d2a5e01686a091f0be58..c600fc2c0730c9916c4a4a9c17b3826a7ae8921b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter_unittest.cc
>@@ -18,7 +18,6 @@
> #include "absl/memory/memory.h"
> #include "media/base/fakeframesource.h"
> #include "media/base/mediachannel.h"
>-#include "media/base/testutils.h"
> #include "media/base/videoadapter.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>@@ -1112,4 +1111,61 @@ TEST_P(VideoAdapterTest, TestAdaptToMax) {
>   EXPECT_EQ(640, out_width_);
>   EXPECT_EQ(360, out_height_);
> }
>+
>+// Test adjusting to 16:9 in landscape, and 9:16 in portrait.
>+TEST(VideoAdapterTestMultipleOrientation, TestNormal) {
>+  VideoAdapter video_adapter;
>+  video_adapter.OnOutputFormatRequest(std::make_pair(640, 360), 640 * 360,
>+                                      std::make_pair(360, 640), 360 * 640, 30);
>+
>+  int cropped_width;
>+  int cropped_height;
>+  int out_width;
>+  int out_height;
>+  EXPECT_TRUE(video_adapter.AdaptFrameResolution(
>+      /* in_width= */ 640, /* in_height= */ 480, /* in_timestamp_ns= */ 0,
>+      &cropped_width, &cropped_height, &out_width, &out_height));
>+  EXPECT_EQ(640, cropped_width);
>+  EXPECT_EQ(360, cropped_height);
>+  EXPECT_EQ(640, out_width);
>+  EXPECT_EQ(360, out_height);
>+
>+  EXPECT_TRUE(video_adapter.AdaptFrameResolution(
>+      /* in_width= */ 480, /* in_height= */ 640,
>+      /* in_timestamp_ns= */ rtc::kNumNanosecsPerSec / 30, &cropped_width,
>+      &cropped_height, &out_width, &out_height));
>+  EXPECT_EQ(360, cropped_width);
>+  EXPECT_EQ(640, cropped_height);
>+  EXPECT_EQ(360, out_width);
>+  EXPECT_EQ(640, out_height);
>+}
>+
>+// Force output to be 9:16, even for landscape input.
>+TEST(VideoAdapterTestMultipleOrientation, TestForcePortrait) {
>+  VideoAdapter video_adapter;
>+  video_adapter.OnOutputFormatRequest(std::make_pair(360, 640), 640 * 360,
>+                                      std::make_pair(360, 640), 360 * 640, 30);
>+
>+  int cropped_width;
>+  int cropped_height;
>+  int out_width;
>+  int out_height;
>+  EXPECT_TRUE(video_adapter.AdaptFrameResolution(
>+      /* in_width= */ 640, /* in_height= */ 480, /* in_timestamp_ns= */ 0,
>+      &cropped_width, &cropped_height, &out_width, &out_height));
>+  EXPECT_EQ(270, cropped_width);
>+  EXPECT_EQ(480, cropped_height);
>+  EXPECT_EQ(270, out_width);
>+  EXPECT_EQ(480, out_height);
>+
>+  EXPECT_TRUE(video_adapter.AdaptFrameResolution(
>+      /* in_width= */ 480, /* in_height= */ 640,
>+      /* in_timestamp_ns= */ rtc::kNumNanosecsPerSec / 30, &cropped_width,
>+      &cropped_height, &out_width, &out_height));
>+  EXPECT_EQ(360, cropped_width);
>+  EXPECT_EQ(640, cropped_height);
>+  EXPECT_EQ(360, out_width);
>+  EXPECT_EQ(640, out_height);
>+}
>+
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.h
>index 826e38fa5a1acbdb1628e7745a10d5204986fb52..32742062aa928c0217885c3d0aa386ba918e1746 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.h
>@@ -26,6 +26,7 @@
> #include "media/base/videocommon.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/system/rtc_export.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread_checker.h"
> #include "rtc_base/timestampaligner.h"
>@@ -72,8 +73,9 @@ enum CaptureState {
> //   media engine thread). Hence, the VideoCapture subclasses dont need to be
> //   thread safe.
> //
>-class VideoCapturer : public sigslot::has_slots<>,
>-                      public rtc::VideoSourceInterface<webrtc::VideoFrame> {
>+class RTC_EXPORT VideoCapturer
>+    : public sigslot::has_slots<>,
>+      public rtc::VideoSourceInterface<webrtc::VideoFrame> {
>  public:
>   VideoCapturer();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer_unittest.cc
>index 080f4a08caeab048d6fa1e3beb9dbd08467d9f3f..07344ca7ac85ae2451b497fa49e6740e758bcfd1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer_unittest.cc
>@@ -15,7 +15,6 @@
> 
> #include "media/base/fakevideocapturer.h"
> #include "media/base/fakevideorenderer.h"
>-#include "media/base/testutils.h"
> #include "media/base/videocapturer.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.h
>index 0ffd989cb6515f0e5e9d22001b87cbf645cc7998..6447cba3b2b69ac53f589a77cbac55024b3ea64c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.h
>@@ -17,6 +17,7 @@
> 
> #include <string>
> 
>+#include "rtc_base/system/rtc_export.h"
> #include "rtc_base/timeutils.h"
> 
> namespace cricket {
>@@ -149,7 +150,7 @@ struct VideoFormatPod {
>   uint32_t fourcc;  // Color space. FOURCC_ANY means that any color space is OK.
> };
> 
>-struct VideoFormat : VideoFormatPod {
>+struct RTC_EXPORT VideoFormat : VideoFormatPod {
>   static const int64_t kMinimumInterval =
>       rtc::kNumNanosecsPerSec / 10000;  // 10k fps.
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.cc
>index a28a912762a729e62796740aeb84815c3b991c2a..c9ccaa9f9d7dec1f0591935ddf960f9c0fd6c3ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.cc
>@@ -15,7 +15,7 @@
> namespace webrtc {
> 
> // Profile information for VP9 video.
>-const char kVP9FmtpProfileId[] = "x-google-profile-id";
>+const char kVP9FmtpProfileId[] = "profile-id";
> 
> std::string VP9ProfileToString(VP9Profile profile) {
>   switch (profile) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.h
>index 5cc1d166a8789b1829d7aa2d91951ca5efe08258..755c46da2cbdb963b5aedccae4e9596f8564f204 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.h
>@@ -16,7 +16,6 @@
> 
> #include "absl/types/optional.h"
> #include "api/video_codecs/sdp_video_format.h"
>-#include "common_types.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/adm_helpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/adm_helpers.h
>index c6ea3a209f2c9af3632bc48c39376ad4c95af280..2a35d26b4716140533e96a6af8a7de35df76442c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/adm_helpers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/adm_helpers.h
>@@ -11,8 +11,6 @@
> #ifndef MEDIA_ENGINE_ADM_HELPERS_H_
> #define MEDIA_ENGINE_ADM_HELPERS_H_
> 
>-#include "common_types.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> class AudioDeviceModule;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers_unittest.cc
>index 464e053dc0a8d16bb6e1d2bff69a993b2d4ef92c..70ac0c1c21e8a535bcdb6a03099d83cd35639c1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers_unittest.cc
>@@ -164,11 +164,4 @@ TEST(ApmHelpersTest, TypingDetectionStatus_EnableDisable) {
>   EXPECT_EQ(VoiceDetection::kVeryLowLikelihood, vd->likelihood());
>   EXPECT_FALSE(vd->is_enabled());
> }
>-
>-// TODO(solenberg): Move this test to a better place - added here for the sake
>-// of duplicating all relevant tests from audio_processing_test.cc.
>-TEST(ApmHelpersTest, HighPassFilter_DefaultMode) {
>-  TestHelper helper;
>-  EXPECT_FALSE(helper.apm()->high_pass_filter()->is_enabled());
>-}
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/constants.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/constants.cc
>index 736685fc79f3684f4df4195eeb4f48503ae45815..fb3ac698e6170278e5b8e81377ed78edb2dad6db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/constants.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/constants.cc
>@@ -14,5 +14,6 @@ namespace cricket {
> 
> const int kMinVideoBitrateBps = 30000;
> const int kVideoMtu = 1200;
>-const int kVideoRtpBufferSize = 65536;
>+const int kVideoRtpSendBufferSize = 65536;
>+const int kVideoRtpRecvBufferSize = 262144;
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/constants.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/constants.h
>index ea6c0754a65d972ebf8617bdb14aa869a18a6a24..b136060546100194ed6fe51478feb5b1583839e6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/constants.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/constants.h
>@@ -14,7 +14,8 @@
> namespace cricket {
> 
> extern const int kVideoMtu;
>-extern const int kVideoRtpBufferSize;
>+extern const int kVideoRtpSendBufferSize;
>+extern const int kVideoRtpRecvBufferSize;
> 
> extern const char kH264CodecName[];
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.cc
>index 7da1b0fe2fd7c2b76382e21ffdd12338d15e8868..297a143425b4c1ea41cc051afa43ad9135c3cd33 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.cc
>@@ -15,6 +15,7 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>+#include "absl/strings/match.h"
> #include "api/video_codecs/video_decoder_factory.h"
> #include "api/video_codecs/video_decoder_software_fallback_wrapper.h"
> #include "api/video_codecs/video_encoder_factory.h"
>@@ -130,7 +131,7 @@ class EncoderAdapter : public webrtc::VideoEncoderFactory {
>     if (IsFormatSupported(internal_encoder_factory_->GetSupportedFormats(),
>                           format)) {
>       internal_encoder =
>-          CodecNamesEq(format.name.c_str(), kVp8CodecName)
>+          absl::EqualsIgnoreCase(format.name, kVp8CodecName)
>               ? absl::make_unique<webrtc::VP8EncoderSimulcastProxy>(
>                     internal_encoder_factory_.get(), format)
>               : internal_encoder_factory_->CreateVideoEncoder(format);
>@@ -141,7 +142,7 @@ class EncoderAdapter : public webrtc::VideoEncoderFactory {
>     if (IsFormatSupported(external_encoder_factory_->GetSupportedFormats(),
>                           format)) {
>       external_encoder =
>-          CodecNamesEq(format.name.c_str(), kVp8CodecName)
>+          absl::EqualsIgnoreCase(format.name, kVp8CodecName)
>               ? absl::make_unique<webrtc::SimulcastEncoderAdapter>(
>                     external_encoder_factory_.get(), format)
>               : external_encoder_factory_->CreateVideoEncoder(format);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.h
>index 5bd3580a6f10425d30fd95d4381368ba687ecbbf..27383ab386e9f3983de4325733dd1e486db65c99 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.h
>@@ -13,6 +13,8 @@
> 
> #include <memory>
> 
>+#include "rtc_base/system/rtc_export.h"
>+
> namespace webrtc {
> class VideoEncoderFactory;
> class VideoDecoderFactory;
>@@ -27,10 +29,12 @@ class WebRtcVideoDecoderFactory;
> // new type of codec factories. The purpose of these functions is to provide an
> // easy way for clients to migrate to the API with new factory types.
> // TODO(magjed): Remove once old factories are gone, webrtc:7925.
>-std::unique_ptr<webrtc::VideoEncoderFactory> ConvertVideoEncoderFactory(
>+RTC_EXPORT std::unique_ptr<webrtc::VideoEncoderFactory>
>+ConvertVideoEncoderFactory(
>     std::unique_ptr<WebRtcVideoEncoderFactory> external_encoder_factory);
> 
>-std::unique_ptr<webrtc::VideoDecoderFactory> ConvertVideoDecoderFactory(
>+RTC_EXPORT std::unique_ptr<webrtc::VideoDecoderFactory>
>+ConvertVideoDecoderFactory(
>     std::unique_ptr<WebRtcVideoDecoderFactory> external_decoder_factory);
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.cc
>index ce99b9c1034c09215408948836e4c9443112af42..6c5b8c74b8f08d1829f5852efb3fd45919ea14c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.cc
>@@ -123,6 +123,7 @@ FakeVideoSendStream::FakeVideoSendStream(
>       source_(nullptr),
>       num_swapped_frames_(0) {
>   RTC_DCHECK(config.encoder_settings.encoder_factory != nullptr);
>+  RTC_DCHECK(config.encoder_settings.bitrate_allocator_factory != nullptr);
>   ReconfigureVideoEncoder(std::move(encoder_config));
> }
> 
>@@ -643,4 +644,7 @@ void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
>   }
> }
> 
>+void FakeCall::MediaTransportChange(
>+    webrtc::MediaTransportInterface* media_transport_interface) {}
>+
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.h
>index dbcedb8f4108fe55f1a2181b88a0316c9b20889b..1b6deb0702da7f35e7425ae02f663052915be177 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.h
>@@ -273,6 +273,9 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
>   int GetNumCreatedReceiveStreams() const;
>   void SetStats(const webrtc::Call::Stats& stats);
> 
>+  void MediaTransportChange(
>+      webrtc::MediaTransportInterface* media_transport_interface) override;
>+
>  private:
>   webrtc::AudioSendStream* CreateAudioSendStream(
>       const webrtc::AudioSendStream::Config& config) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideocapturemodule.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideocapturemodule.h
>index 47f1393c3f3819e851710bb84bb281a4ebd40d38..7556811f2de01962bfe014e5056324c2d68400fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideocapturemodule.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideocapturemodule.h
>@@ -14,7 +14,6 @@
> #include <vector>
> 
> #include "api/video/i420_buffer.h"
>-#include "media/base/testutils.h"
> #include "media/engine/webrtcvideocapturer.h"
> #include "rtc_base/task_queue_for_test.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.cc
>index 7081b64933a956f2c06aa5a86284574b10f3db72..690f3e7f831eca760a75c5114434ccde6b914bd1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.cc
>@@ -10,6 +10,7 @@
> 
> #include "media/engine/fakewebrtcvideoengine.h"
> 
>+#include "absl/strings/match.h"
> #include "media/base/codec.h"
> #include "media/engine/simulcast_encoder_adapter.h"
> #include "media/engine/webrtcvideodecoderfactory.h"
>@@ -128,9 +129,7 @@ FakeWebRtcVideoDecoderFactory::decoders() {
> // Encoder.
> FakeWebRtcVideoEncoder::FakeWebRtcVideoEncoder(
>     FakeWebRtcVideoEncoderFactory* factory)
>-    : init_encode_event_(false, false),
>-      num_frames_encoded_(0),
>-      factory_(factory) {}
>+    : num_frames_encoded_(0), factory_(factory) {}
> 
> FakeWebRtcVideoEncoder::~FakeWebRtcVideoEncoder() {
>   if (factory_) {
>@@ -167,11 +166,6 @@ int32_t FakeWebRtcVideoEncoder::Release() {
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-int32_t FakeWebRtcVideoEncoder::SetChannelParameters(uint32_t packetLoss,
>-                                                     int64_t rtt) {
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
> int32_t FakeWebRtcVideoEncoder::SetRateAllocation(
>     const webrtc::VideoBitrateAllocation& allocation,
>     uint32_t framerate) {
>@@ -194,8 +188,7 @@ int FakeWebRtcVideoEncoder::GetNumEncodedFrames() {
> 
> // Video encoder factory.
> FakeWebRtcVideoEncoderFactory::FakeWebRtcVideoEncoderFactory()
>-    : created_video_encoder_event_(false, false),
>-      num_created_encoders_(0),
>+    : num_created_encoders_(0),
>       encoders_have_internal_sources_(false),
>       vp8_factory_mode_(false) {}
> 
>@@ -218,7 +211,7 @@ FakeWebRtcVideoEncoderFactory::CreateVideoEncoder(
>   rtc::CritScope lock(&crit_);
>   std::unique_ptr<webrtc::VideoEncoder> encoder;
>   if (IsFormatSupported(formats_, format)) {
>-    if (CodecNamesEq(format.name.c_str(), kVp8CodecName) &&
>+    if (absl::EqualsIgnoreCase(format.name, kVp8CodecName) &&
>         !vp8_factory_mode_) {
>       // The simulcast adapter will ask this factory for multiple VP8
>       // encoders. Enter vp8_factory_mode so that we now create these encoders
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.h
>index f4287a31d26026fdee8931be9347db489846ae23..6ed92a789b57e2dd790980d7a9c8962ba247f098 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.h
>@@ -86,7 +86,6 @@ class FakeWebRtcVideoEncoder : public webrtc::VideoEncoder {
>   int32_t RegisterEncodeCompleteCallback(
>       webrtc::EncodedImageCallback* callback) override;
>   int32_t Release() override;
>-  int32_t SetChannelParameters(uint32_t packetLoss, int64_t rtt) override;
>   int32_t SetRateAllocation(const webrtc::VideoBitrateAllocation& allocation,
>                             uint32_t framerate) override;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>index df74773c86db26ee374841247aa10bb400b197ac..722413338e51198d2fc616af59e99f9f760f7bc7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>@@ -10,6 +10,7 @@
> 
> #include "media/engine/internaldecoderfactory.h"
> 
>+#include "absl/strings/match.h"
> #include "api/video_codecs/sdp_video_format.h"
> #include "media/base/mediaconstants.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
>@@ -55,11 +56,11 @@ std::unique_ptr<VideoDecoder> InternalDecoderFactory::CreateVideoDecoder(
>     return nullptr;
>   }
> 
>-  if (cricket::CodecNamesEq(format.name, cricket::kVp8CodecName))
>+  if (absl::EqualsIgnoreCase(format.name, cricket::kVp8CodecName))
>     return VP8Decoder::Create();
>-  if (cricket::CodecNamesEq(format.name, cricket::kVp9CodecName))
>+  if (absl::EqualsIgnoreCase(format.name, cricket::kVp9CodecName))
>     return VP9Decoder::Create();
>-  if (cricket::CodecNamesEq(format.name, cricket::kH264CodecName))
>+  if (absl::EqualsIgnoreCase(format.name, cricket::kH264CodecName))
>     return H264Decoder::Create();
> 
>   RTC_NOTREACHED();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>index e6c3c2e72fd632eac71eb38dfab61327a18b82df..e81b73d5e3d6cf15ac38cf7d35535d15d4220c87 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>@@ -12,6 +12,7 @@
> 
> #include <utility>
> 
>+#include "absl/strings/match.h"
> #include "api/video_codecs/sdp_video_format.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>@@ -41,11 +42,11 @@ VideoEncoderFactory::CodecInfo InternalEncoderFactory::QueryVideoEncoder(
> 
> std::unique_ptr<VideoEncoder> InternalEncoderFactory::CreateVideoEncoder(
>     const SdpVideoFormat& format) {
>-  if (cricket::CodecNamesEq(format.name, cricket::kVp8CodecName))
>+  if (absl::EqualsIgnoreCase(format.name, cricket::kVp8CodecName))
>     return VP8Encoder::Create();
>-  if (cricket::CodecNamesEq(format.name, cricket::kVp9CodecName))
>+  if (absl::EqualsIgnoreCase(format.name, cricket::kVp9CodecName))
>     return VP9Encoder::Create(cricket::VideoCodec(format));
>-  if (cricket::CodecNamesEq(format.name, cricket::kH264CodecName))
>+  if (absl::EqualsIgnoreCase(format.name, cricket::kH264CodecName))
>     return H264Encoder::Create(cricket::VideoCodec(format));
>   RTC_LOG(LS_ERROR) << "Trying to created encoder of unsupported format "
>                     << format.name;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.cc
>index 236a2e81acc722d848554a87adf4840c24d75f94..b3913243edeb05d163dd3a2ab873fc00f200cf49 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.cc
>@@ -12,6 +12,7 @@
> 
> #include <utility>
> 
>+#include "absl/strings/match.h"
> #include "api/video_codecs/sdp_video_format.h"
> #include "media/base/codec.h"
> #include "media/base/mediaconstants.h"
>@@ -22,8 +23,8 @@
> namespace {
> 
> bool IsMultiplexCodec(const cricket::VideoCodec& codec) {
>-  return cricket::CodecNamesEq(codec.name.c_str(),
>-                               cricket::kMultiplexCodecName);
>+  return absl::EqualsIgnoreCase(codec.name.c_str(),
>+                                cricket::kMultiplexCodecName);
> }
> 
> }  // anonymous namespace
>@@ -42,7 +43,7 @@ std::vector<SdpVideoFormat> MultiplexEncoderFactory::GetSupportedFormats()
>     const {
>   std::vector<SdpVideoFormat> formats = factory_->GetSupportedFormats();
>   for (const auto& format : formats) {
>-    if (cricket::CodecNamesEq(format.name, kMultiplexAssociatedCodecName)) {
>+    if (absl::EqualsIgnoreCase(format.name, kMultiplexAssociatedCodecName)) {
>       SdpVideoFormat multiplex_format = format;
>       multiplex_format.parameters[cricket::kCodecParamAssociatedCodecName] =
>           format.name;
>@@ -88,7 +89,7 @@ std::vector<SdpVideoFormat> MultiplexDecoderFactory::GetSupportedFormats()
>     const {
>   std::vector<SdpVideoFormat> formats = factory_->GetSupportedFormats();
>   for (const auto& format : formats) {
>-    if (cricket::CodecNamesEq(format.name, kMultiplexAssociatedCodecName)) {
>+    if (absl::EqualsIgnoreCase(format.name, kMultiplexAssociatedCodecName)) {
>       SdpVideoFormat multiplex_format = format;
>       multiplex_format.parameters[cricket::kCodecParamAssociatedCodecName] =
>           format.name;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.h
>index 7a851c8c1ecfc4203ca83fae91ac86eac77fe67a..ae099a4ef3e78b1efadfb192d76903015dcb5349 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.h
>@@ -16,6 +16,7 @@
> 
> #include "api/video_codecs/video_decoder_factory.h"
> #include "api/video_codecs/video_encoder_factory.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> // Multiplex codec is a completely modular/optional codec that allows users to
>@@ -36,7 +37,7 @@ namespace webrtc {
> // and decoder instance(s) using these factories.
> // - Use Multiplex*coderFactory classes in CreatePeerConnectionFactory() calls.
> // - Select "multiplex" codec in SDP negotiation.
>-class MultiplexEncoderFactory : public VideoEncoderFactory {
>+class RTC_EXPORT MultiplexEncoderFactory : public VideoEncoderFactory {
>  public:
>   // |supports_augmenting_data| defines if the encoder would support augmenting
>   // data. If set, the encoder expects to receive video frame buffers of type
>@@ -54,7 +55,7 @@ class MultiplexEncoderFactory : public VideoEncoderFactory {
>   const bool supports_augmenting_data_;
> };
> 
>-class MultiplexDecoderFactory : public VideoDecoderFactory {
>+class RTC_EXPORT MultiplexDecoderFactory : public VideoDecoderFactory {
>  public:
>   // |supports_augmenting_data| defines if the decoder would support augmenting
>   // data. If set, the decoder is expected to output video frame buffers of type
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine.h
>index 9af0f9b130c6c2e4564534e9022de07e757fc8c1..62326dd4ddd43b39698edd122de0cc1addef5a9c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine.h
>@@ -30,15 +30,19 @@ class WebRtcVideoEncoderFactory;
> 
> // Video engine implementation that does nothing and can be used in
> // CompositeMediaEngine.
>-class NullWebRtcVideoEngine {
>+class NullWebRtcVideoEngine : public VideoEngineInterface {
>  public:
>-  std::vector<VideoCodec> codecs() const { return std::vector<VideoCodec>(); }
>+  std::vector<VideoCodec> codecs() const override {
>+    return std::vector<VideoCodec>();
>+  }
> 
>-  RtpCapabilities GetCapabilities() const { return RtpCapabilities(); }
>+  RtpCapabilities GetCapabilities() const override { return RtpCapabilities(); }
> 
>-  VideoMediaChannel* CreateChannel(webrtc::Call* call,
>-                                   const MediaConfig& config,
>-                                   const VideoOptions& options) {
>+  VideoMediaChannel* CreateMediaChannel(
>+      webrtc::Call* call,
>+      const MediaConfig& config,
>+      const VideoOptions& options,
>+      const webrtc::CryptoOptions& crypto_options) override {
>     return nullptr;
>   }
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine_unittest.cc
>index 6c740b8c8c372ab8a290c31bdcae575c855b2adf..e718d1bb697e21822728fa8c1f45ecc5e5d86334 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine_unittest.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "media/engine/nullwebrtcvideoengine.h"
>+#include "absl/memory/memory.h"
> #include "media/engine/webrtcvoiceengine.h"
> #include "modules/audio_device/include/mock_audio_device.h"
> #include "modules/audio_processing/include/audio_processing.h"
>@@ -18,8 +19,7 @@
> 
> namespace cricket {
> 
>-class WebRtcMediaEngineNullVideo
>-    : public CompositeMediaEngine<WebRtcVoiceEngine, NullWebRtcVideoEngine> {
>+class WebRtcMediaEngineNullVideo : public CompositeMediaEngine {
>  public:
>   WebRtcMediaEngineNullVideo(
>       webrtc::AudioDeviceModule* adm,
>@@ -27,13 +27,13 @@ class WebRtcMediaEngineNullVideo
>           audio_encoder_factory,
>       const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
>           audio_decoder_factory)
>-      : CompositeMediaEngine<WebRtcVoiceEngine, NullWebRtcVideoEngine>(
>-            std::forward_as_tuple(adm,
>-                                  audio_encoder_factory,
>-                                  audio_decoder_factory,
>-                                  nullptr,
>-                                  webrtc::AudioProcessingBuilder().Create()),
>-            std::forward_as_tuple()) {}
>+      : CompositeMediaEngine(absl::make_unique<WebRtcVoiceEngine>(
>+                                 adm,
>+                                 audio_encoder_factory,
>+                                 audio_decoder_factory,
>+                                 nullptr,
>+                                 webrtc::AudioProcessingBuilder().Create()),
>+                             absl::make_unique<NullWebRtcVideoEngine>()) {}
> };
> 
> // Simple test to check if NullWebRtcVideoEngine implements the methods
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.cc
>index 5d2fb6757c445ad212517900e894dc8409df5655..ae4e309ef4900d3b917f89ad0618fdced27219a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.cc
>@@ -12,8 +12,8 @@
> 
> #include <utility>
> 
>+#include "absl/strings/ascii.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "media/base/mediaconstants.h"
> 
> namespace cricket {
>@@ -140,7 +140,8 @@ bool PayloadTypeMapper::SdpAudioFormatOrdering::operator()(
>     const webrtc::SdpAudioFormat& b) const {
>   if (a.clockrate_hz == b.clockrate_hz) {
>     if (a.num_channels == b.num_channels) {
>-      int name_cmp = STR_CASE_CMP(a.name.c_str(), b.name.c_str());
>+      int name_cmp =
>+          absl::AsciiStrToLower(a.name).compare(absl::AsciiStrToLower(b.name));
>       if (name_cmp == 0)
>         return a.parameters < b.parameters;
>       return name_cmp < 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideoencoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideoencoder.cc
>index 026654e07204d044b69f5cb89666f78b92ab31c6..db1148bf30fe2009304b57a7972bb8ff1ccf6350 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideoencoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideoencoder.cc
>@@ -33,14 +33,10 @@ class ScopedVideoEncoder : public webrtc::VideoEncoder {
>   int32_t Encode(const webrtc::VideoFrame& frame,
>                  const webrtc::CodecSpecificInfo* codec_specific_info,
>                  const std::vector<webrtc::FrameType>* frame_types) override;
>-  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>   int32_t SetRates(uint32_t bitrate, uint32_t framerate) override;
>   int32_t SetRateAllocation(const webrtc::VideoBitrateAllocation& allocation,
>                             uint32_t framerate) override;
>-  ScalingSettings GetScalingSettings() const override;
>-  bool SupportsNativeHandle() const override;
>-  const char* ImplementationName() const override;
>-
>+  EncoderInfo GetEncoderInfo() const override;
>   ~ScopedVideoEncoder() override;
> 
>  private:
>@@ -75,11 +71,6 @@ int32_t ScopedVideoEncoder::Encode(
>   return encoder_->Encode(frame, codec_specific_info, frame_types);
> }
> 
>-int32_t ScopedVideoEncoder::SetChannelParameters(uint32_t packet_loss,
>-                                                 int64_t rtt) {
>-  return encoder_->SetChannelParameters(packet_loss, rtt);
>-}
>-
> int32_t ScopedVideoEncoder::SetRates(uint32_t bitrate, uint32_t framerate) {
>   return encoder_->SetRates(bitrate, framerate);
> }
>@@ -90,17 +81,8 @@ int32_t ScopedVideoEncoder::SetRateAllocation(
>   return encoder_->SetRateAllocation(allocation, framerate);
> }
> 
>-webrtc::VideoEncoder::ScalingSettings ScopedVideoEncoder::GetScalingSettings()
>-    const {
>-  return encoder_->GetScalingSettings();
>-}
>-
>-bool ScopedVideoEncoder::SupportsNativeHandle() const {
>-  return encoder_->SupportsNativeHandle();
>-}
>-
>-const char* ScopedVideoEncoder::ImplementationName() const {
>-  return encoder_->ImplementationName();
>+webrtc::VideoEncoder::EncoderInfo ScopedVideoEncoder::GetEncoderInfo() const {
>+  return encoder_->GetEncoderInfo();
> }
> 
> ScopedVideoEncoder::~ScopedVideoEncoder() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.cc
>index c42e1dcae290840a9b398cc830d12c81cf6161c6..de3d8df097f1505d01f64ce30fe7a6e022dc0fe4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.cc
>@@ -18,6 +18,7 @@
> #include "media/engine/simulcast.h"
> #include "modules/video_coding/utility/simulcast_rate_allocator.h"
> #include "rtc_base/arraysize.h"
>+#include "rtc_base/experiments/normalize_simulcast_size_experiment.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/field_trial.h"
> 
>@@ -140,7 +141,13 @@ int FindSimulcastFormatIndex(int width, int height, size_t max_layers) {
> // Simulcast stream width and height must both be dividable by
> // |2 ^ (simulcast_layers - 1)|.
> int NormalizeSimulcastSize(int size, size_t simulcast_layers) {
>-  const int base2_exponent = static_cast<int>(simulcast_layers) - 1;
>+  int base2_exponent = static_cast<int>(simulcast_layers) - 1;
>+  const absl::optional<int> experimental_base2_exponent =
>+      webrtc::NormalizeSimulcastSizeExperiment::GetBase2Exponent();
>+  if (experimental_base2_exponent &&
>+      (size > (1 << *experimental_base2_exponent))) {
>+    base2_exponent = *experimental_base2_exponent;
>+  }
>   return ((size >> base2_exponent) << base2_exponent);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.cc
>index 30da68ec85ca8b7fa0de71a71c65b8cd4d1725c8..6241ff9d36d691d039cdfec6c6c6666122d5967d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.cc
>@@ -17,9 +17,9 @@
> #include "api/video/i420_buffer.h"
> #include "api/video/video_bitrate_allocation.h"
> #include "api/video_codecs/video_encoder_factory.h"
>-#include "media/engine/scopedvideoencoder.h"
> #include "modules/video_coding/utility/simulcast_rate_allocator.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/field_trial.h"
> #include "third_party/libyuv/include/libyuv/scale.h"
>@@ -127,9 +127,9 @@ SimulcastEncoderAdapter::SimulcastEncoderAdapter(VideoEncoderFactory* factory,
>       factory_(factory),
>       video_format_(format),
>       encoded_complete_callback_(nullptr),
>-      implementation_name_("SimulcastEncoderAdapter"),
>       experimental_boosted_screenshare_qp_(GetScreenshareBoostedQpValue()) {
>   RTC_DCHECK(factory_);
>+  encoder_info_.implementation_name = "SimulcastEncoderAdapter";
> 
>   // The adapter is typically created on the worker thread, but operated on
>   // the encoder task queue.
>@@ -203,7 +203,8 @@ int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
>     start_bitrates.push_back(stream_bitrate);
>   }
> 
>-  std::string implementation_name;
>+  encoder_info_.supports_native_handle = true;
>+  encoder_info_.scaling_settings.thresholds = absl::nullopt;
>   // Create |number_of_streams| of encoder instances and init them.
>   for (int i = 0; i < number_of_streams; ++i) {
>     VideoCodec stream_codec;
>@@ -212,6 +213,7 @@ int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
>     if (!doing_simulcast) {
>       stream_codec = codec_;
>       stream_codec.numberOfSimulcastStreams = 1;
>+
>     } else {
>       // Cap start bitrate to the min bitrate in order to avoid strange codec
>       // behavior. Since sending will be false, this should not matter.
>@@ -247,6 +249,7 @@ int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
>       Release();
>       return ret;
>     }
>+
>     std::unique_ptr<EncodedImageCallback> callback(
>         new AdapterEncodedImageCallback(this, i));
>     encoder->RegisterEncodeCompleteCallback(callback.get());
>@@ -254,17 +257,45 @@ int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
>                               stream_codec.width, stream_codec.height,
>                               send_stream);
> 
>-    if (i != 0) {
>-      implementation_name += ", ";
>+    if (!doing_simulcast) {
>+      // Without simulcast, just pass through the encoder info from the one
>+      // active encoder.
>+      encoder_info_ = streaminfos_[0].encoder->GetEncoderInfo();
>+    } else {
>+      const EncoderInfo encoder_impl_info =
>+          streaminfos_[i].encoder->GetEncoderInfo();
>+
>+      if (i == 0) {
>+        // Quality scaling not enabled for simulcast.
>+        encoder_info_.scaling_settings = VideoEncoder::ScalingSettings::kOff;
>+
>+        // Encoder name indicates names of all sub-encoders.
>+        encoder_info_.implementation_name = "SimulcastEncoderAdapter (";
>+        encoder_info_.implementation_name +=
>+            encoder_impl_info.implementation_name;
>+
>+        encoder_info_.supports_native_handle =
>+            encoder_impl_info.supports_native_handle;
>+        encoder_info_.has_trusted_rate_controller =
>+            encoder_impl_info.has_trusted_rate_controller;
>+      } else {
>+        encoder_info_.implementation_name += ", ";
>+        encoder_info_.implementation_name +=
>+            encoder_impl_info.implementation_name;
>+
>+        // Native handle supported only if all encoders supports it.
>+        encoder_info_.supports_native_handle &=
>+            encoder_impl_info.supports_native_handle;
>+
>+        // Trusted rate controller only if all encoders have it.
>+        encoder_info_.has_trusted_rate_controller &=
>+            encoder_impl_info.has_trusted_rate_controller;
>+      }
>     }
>-    implementation_name += streaminfos_[i].encoder->ImplementationName();
>   }
> 
>   if (doing_simulcast) {
>-    implementation_name_ =
>-        "SimulcastEncoderAdapter (" + implementation_name + ")";
>-  } else {
>-    implementation_name_ = implementation_name;
>+    encoder_info_.implementation_name += ")";
>   }
> 
>   // To save memory, don't store encoders that we don't use.
>@@ -377,15 +408,6 @@ int SimulcastEncoderAdapter::RegisterEncodeCompleteCallback(
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-int SimulcastEncoderAdapter::SetChannelParameters(uint32_t packet_loss,
>-                                                  int64_t rtt) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_queue_);
>-  for (size_t stream_idx = 0; stream_idx < streaminfos_.size(); ++stream_idx) {
>-    streaminfos_[stream_idx].encoder->SetChannelParameters(packet_loss, rtt);
>-  }
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
> int SimulcastEncoderAdapter::SetRateAllocation(
>     const VideoBitrateAllocation& bitrate,
>     uint32_t new_framerate) {
>@@ -452,7 +474,6 @@ EncodedImageCallback::Result SimulcastEncoderAdapter::OnEncodedImage(
>     const RTPFragmentationHeader* fragmentation) {
>   EncodedImage stream_image(encodedImage);
>   CodecSpecificInfo stream_codec_specific = *codecSpecificInfo;
>-  stream_codec_specific.codec_name = implementation_name_.c_str();
> 
>   stream_image.SetSpatialIndex(stream_idx);
> 
>@@ -517,32 +538,8 @@ void SimulcastEncoderAdapter::DestroyStoredEncoders() {
>   }
> }
> 
>-bool SimulcastEncoderAdapter::SupportsNativeHandle() const {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_queue_);
>-  // We should not be calling this method before streaminfos_ are configured.
>-  RTC_DCHECK(!streaminfos_.empty());
>-  for (const auto& streaminfo : streaminfos_) {
>-    if (!streaminfo.encoder->SupportsNativeHandle()) {
>-      return false;
>-    }
>-  }
>-  return true;
>-}
>-
>-VideoEncoder::ScalingSettings SimulcastEncoderAdapter::GetScalingSettings()
>-    const {
>-  // TODO(brandtr): Investigate why the sequence checker below fails on mac.
>-  // RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_queue_);
>-  // Turn off quality scaling for simulcast.
>-  if (!Initialized() || NumberOfStreams(codec_) != 1) {
>-    return VideoEncoder::ScalingSettings::kOff;
>-  }
>-  return streaminfos_[0].encoder->GetScalingSettings();
>-}
>-
>-const char* SimulcastEncoderAdapter::ImplementationName() const {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_queue_);
>-  return implementation_name_.c_str();
>+VideoEncoder::EncoderInfo SimulcastEncoderAdapter::GetEncoderInfo() const {
>+  return encoder_info_;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.h
>index 52643aa79b16270019f596e789daae99591a94b5..783c342d6b9fc4a289be7758af93adc64ddd7c3d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.h
>@@ -19,7 +19,7 @@
> #include <vector>
> 
> #include "absl/types/optional.h"
>-#include "media/engine/webrtcvideoencoderfactory.h"
>+#include "api/video_codecs/sdp_video_format.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/sequenced_task_checker.h"
>@@ -48,7 +48,6 @@ class SimulcastEncoderAdapter : public VideoEncoder {
>              const CodecSpecificInfo* codec_specific_info,
>              const std::vector<FrameType>* frame_types) override;
>   int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
>-  int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>   int SetRateAllocation(const VideoBitrateAllocation& bitrate,
>                         uint32_t new_framerate) override;
> 
>@@ -61,10 +60,7 @@ class SimulcastEncoderAdapter : public VideoEncoder {
>       const CodecSpecificInfo* codec_specific_info,
>       const RTPFragmentationHeader* fragmentation);
> 
>-  VideoEncoder::ScalingSettings GetScalingSettings() const override;
>-
>-  bool SupportsNativeHandle() const override;
>-  const char* ImplementationName() const override;
>+  EncoderInfo GetEncoderInfo() const override;
> 
>  private:
>   struct StreamInfo {
>@@ -104,7 +100,7 @@ class SimulcastEncoderAdapter : public VideoEncoder {
>   VideoCodec codec_;
>   std::vector<StreamInfo> streaminfos_;
>   EncodedImageCallback* encoded_complete_callback_;
>-  std::string implementation_name_;
>+  EncoderInfo encoder_info_;
> 
>   // Used for checking the single-threaded access of the encoder interface.
>   rtc::SequencedTaskChecker encoder_queue_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter_unittest.cc
>index 5b7d185b8cc6ebaa2c244880ec34e05e60fb8fc3..59541755fa51dc377b6e3e63710b433ebc334bfc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter_unittest.cc
>@@ -15,16 +15,17 @@
> #include "absl/memory/memory.h"
> #include "api/test/create_simulcast_test_fixture.h"
> #include "api/test/simulcast_test_fixture.h"
>+#include "api/test/video/function_video_decoder_factory.h"
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "api/video_codecs/sdp_video_format.h"
> #include "api/video_codecs/video_encoder_factory.h"
> #include "common_video/include/video_frame_buffer.h"
>+#include "media/base/mediaconstants.h"
> #include "media/engine/internalencoderfactory.h"
> #include "media/engine/simulcast_encoder_adapter.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/utility/simulcast_test_fixture_impl.h"
>-#include "test/function_video_decoder_factory.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>@@ -55,7 +56,6 @@ std::unique_ptr<SimulcastTestFixture> CreateSpecificSimulcastTestFixture(
>                                     std::move(decoder_factory),
>                                     SdpVideoFormat(cricket::kVp8CodecName));
> }
>-
> }  // namespace
> 
> TEST(SimulcastEncoderAdapterSimulcastTest, TestKeyFrameRequestsOnAllStreams) {
>@@ -144,6 +144,12 @@ TEST(SimulcastEncoderAdapterSimulcastTest,
>   fixture->TestSpatioTemporalLayers321PatternEncoder();
> }
> 
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestDecodeWidthHeightSet) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestDecodeWidthHeightSet();
>+}
>+
> class MockVideoEncoder;
> 
> class MockVideoEncoderFactory : public VideoEncoderFactory {
>@@ -170,7 +176,9 @@ class MockVideoEncoderFactory : public VideoEncoderFactory {
> class MockVideoEncoder : public VideoEncoder {
>  public:
>   explicit MockVideoEncoder(MockVideoEncoderFactory* factory)
>-      : factory_(factory), callback_(nullptr) {}
>+      : factory_(factory),
>+        scaling_settings_(VideoEncoder::ScalingSettings::kOff),
>+        callback_(nullptr) {}
> 
>   // TODO(nisse): Valid overrides commented out, because the gmock
>   // methods don't use any override declarations, and we want to avoid
>@@ -203,10 +211,13 @@ class MockVideoEncoder : public VideoEncoder {
>     return 0;
>   }
> 
>-  MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
>-
>-  bool SupportsNativeHandle() const /* override */ {
>-    return supports_native_handle_;
>+  EncoderInfo GetEncoderInfo() const override {
>+    EncoderInfo info;
>+    info.supports_native_handle = supports_native_handle_;
>+    info.implementation_name = implementation_name_;
>+    info.scaling_settings = scaling_settings_;
>+    info.has_trusted_rate_controller = has_trusted_rate_controller_;
>+    return info;
>   }
> 
>   virtual ~MockVideoEncoder() { factory_->DestroyVideoEncoder(this); }
>@@ -228,17 +239,30 @@ class MockVideoEncoder : public VideoEncoder {
>     supports_native_handle_ = enabled;
>   }
> 
>+  void set_implementation_name(const std::string& name) {
>+    implementation_name_ = name;
>+  }
>+
>   void set_init_encode_return_value(int32_t value) {
>     init_encode_return_value_ = value;
>   }
> 
>-  VideoBitrateAllocation last_set_bitrate() const { return last_set_bitrate_; }
>+  void set_scaling_settings(const VideoEncoder::ScalingSettings& settings) {
>+    scaling_settings_ = settings;
>+  }
> 
>-  MOCK_CONST_METHOD0(ImplementationName, const char*());
>+  void set_has_trusted_rate_controller(bool trusted) {
>+    has_trusted_rate_controller_ = trusted;
>+  }
>+
>+  VideoBitrateAllocation last_set_bitrate() const { return last_set_bitrate_; }
> 
>  private:
>   MockVideoEncoderFactory* const factory_;
>   bool supports_native_handle_ = false;
>+  std::string implementation_name_ = "unknown";
>+  VideoEncoder::ScalingSettings scaling_settings_;
>+  bool has_trusted_rate_controller_ = false;
>   int32_t init_encode_return_value_ = 0;
>   VideoBitrateAllocation last_set_bitrate_;
> 
>@@ -260,7 +284,7 @@ std::unique_ptr<VideoEncoder> MockVideoEncoderFactory::CreateVideoEncoder(
>   const char* encoder_name = encoder_names_.empty()
>                                  ? "codec_implementation_name"
>                                  : encoder_names_[encoders_.size()];
>-  ON_CALL(*encoder, ImplementationName()).WillByDefault(Return(encoder_name));
>+  encoder->set_implementation_name(encoder_name);
>   encoders_.push_back(encoder.get());
>   return encoder;
> }
>@@ -302,15 +326,6 @@ class TestSimulcastEncoderAdapterFakeHelper {
>     return new SimulcastEncoderAdapter(factory_.get(), SdpVideoFormat("VP8"));
>   }
> 
>-  void ExpectCallSetChannelParameters(uint32_t packetLoss, int64_t rtt) {
>-    EXPECT_TRUE(!factory_->encoders().empty());
>-    for (size_t i = 0; i < factory_->encoders().size(); ++i) {
>-      EXPECT_CALL(*factory_->encoders()[i],
>-                  SetChannelParameters(packetLoss, rtt))
>-          .Times(1);
>-    }
>-  }
>-
>   MockVideoEncoderFactory* factory() { return factory_.get(); }
> 
>  private:
>@@ -459,14 +474,6 @@ TEST_F(TestSimulcastEncoderAdapterFake, Reinit) {
>   EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
> }
> 
>-TEST_F(TestSimulcastEncoderAdapterFake, SetChannelParameters) {
>-  SetupCodec();
>-  const uint32_t packetLoss = 5;
>-  const int64_t rtt = 30;
>-  helper_->ExpectCallSetChannelParameters(packetLoss, rtt);
>-  adapter_->SetChannelParameters(packetLoss, rtt);
>-}
>-
> TEST_F(TestSimulcastEncoderAdapterFake, EncodedCallbackForDifferentEncoders) {
>   SetupCodec();
> 
>@@ -728,9 +735,11 @@ TEST_F(TestSimulcastEncoderAdapterFake, SupportsNativeHandleForSingleStreams) {
>   adapter_->RegisterEncodeCompleteCallback(this);
>   ASSERT_EQ(1u, helper_->factory()->encoders().size());
>   helper_->factory()->encoders()[0]->set_supports_native_handle(true);
>-  EXPECT_TRUE(adapter_->SupportsNativeHandle());
>+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>+  EXPECT_TRUE(adapter_->GetEncoderInfo().supports_native_handle);
>   helper_->factory()->encoders()[0]->set_supports_native_handle(false);
>-  EXPECT_FALSE(adapter_->SupportsNativeHandle());
>+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>+  EXPECT_FALSE(adapter_->GetEncoderInfo().supports_native_handle);
> }
> 
> TEST_F(TestSimulcastEncoderAdapterFake, SetRatesUnderMinBitrate) {
>@@ -763,7 +772,8 @@ TEST_F(TestSimulcastEncoderAdapterFake, SetRatesUnderMinBitrate) {
> }
> 
> TEST_F(TestSimulcastEncoderAdapterFake, SupportsImplementationName) {
>-  EXPECT_STREQ("SimulcastEncoderAdapter", adapter_->ImplementationName());
>+  EXPECT_EQ("SimulcastEncoderAdapter",
>+            adapter_->GetEncoderInfo().implementation_name);
>   SimulcastTestFixtureImpl::DefaultSettings(
>       &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>       kVideoCodecVP8);
>@@ -773,8 +783,8 @@ TEST_F(TestSimulcastEncoderAdapterFake, SupportsImplementationName) {
>   encoder_names.push_back("codec3");
>   helper_->factory()->SetEncoderNames(encoder_names);
>   EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>-  EXPECT_STREQ("SimulcastEncoderAdapter (codec1, codec2, codec3)",
>-               adapter_->ImplementationName());
>+  EXPECT_EQ("SimulcastEncoderAdapter (codec1, codec2, codec3)",
>+            adapter_->GetEncoderInfo().implementation_name);
> 
>   // Single streams should not expose "SimulcastEncoderAdapter" in name.
>   EXPECT_EQ(0, adapter_->Release());
>@@ -782,7 +792,7 @@ TEST_F(TestSimulcastEncoderAdapterFake, SupportsImplementationName) {
>   EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>   adapter_->RegisterEncodeCompleteCallback(this);
>   ASSERT_EQ(1u, helper_->factory()->encoders().size());
>-  EXPECT_STREQ("codec1", adapter_->ImplementationName());
>+  EXPECT_EQ("codec1", adapter_->GetEncoderInfo().implementation_name);
> }
> 
> TEST_F(TestSimulcastEncoderAdapterFake,
>@@ -798,10 +808,11 @@ TEST_F(TestSimulcastEncoderAdapterFake,
>     encoder->set_supports_native_handle(true);
>   // If one encoder doesn't support it, then overall support is disabled.
>   helper_->factory()->encoders()[0]->set_supports_native_handle(false);
>-  EXPECT_FALSE(adapter_->SupportsNativeHandle());
>+  EXPECT_FALSE(adapter_->GetEncoderInfo().supports_native_handle);
>   // Once all do, then the adapter claims support.
>   helper_->factory()->encoders()[0]->set_supports_native_handle(true);
>-  EXPECT_TRUE(adapter_->SupportsNativeHandle());
>+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>+  EXPECT_TRUE(adapter_->GetEncoderInfo().supports_native_handle);
> }
> 
> // TODO(nisse): Reuse definition in webrtc/test/fake_texture_handle.h.
>@@ -837,7 +848,8 @@ TEST_F(TestSimulcastEncoderAdapterFake,
>   ASSERT_EQ(3u, helper_->factory()->encoders().size());
>   for (MockVideoEncoder* encoder : helper_->factory()->encoders())
>     encoder->set_supports_native_handle(true);
>-  EXPECT_TRUE(adapter_->SupportsNativeHandle());
>+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>+  EXPECT_TRUE(adapter_->GetEncoderInfo().supports_native_handle);
> 
>   rtc::scoped_refptr<VideoFrameBuffer> buffer(
>       new rtc::RefCountedObject<FakeNativeBufferNoI420>(1280, 720));
>@@ -948,5 +960,50 @@ TEST_F(TestSimulcastEncoderAdapterFake, ActivatesCorrectStreamsInInitEncode) {
>   frame_types.resize(3, kVideoFrameKey);
>   EXPECT_EQ(0, adapter_->Encode(input_frame, nullptr, &frame_types));
> }
>+
>+TEST_F(TestSimulcastEncoderAdapterFake, TrustedRateControl) {
>+  // Set up common settings for three streams.
>+  SimulcastTestFixtureImpl::DefaultSettings(
>+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>+      kVideoCodecVP8);
>+  rate_allocator_.reset(new SimulcastRateAllocator(codec_));
>+  adapter_->RegisterEncodeCompleteCallback(this);
>+
>+  // Only enough start bitrate for the lowest stream.
>+  ASSERT_EQ(3u, codec_.numberOfSimulcastStreams);
>+  codec_.startBitrate = codec_.simulcastStream[0].targetBitrate +
>+                        codec_.simulcastStream[1].minBitrate - 1;
>+
>+  // Input data.
>+  rtc::scoped_refptr<VideoFrameBuffer> buffer(I420Buffer::Create(1280, 720));
>+  VideoFrame input_frame(buffer, 100, 1000, kVideoRotation_180);
>+
>+  // No encoder trusted, so simulcast adapter should not be either.
>+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>+  EXPECT_FALSE(adapter_->GetEncoderInfo().has_trusted_rate_controller);
>+
>+  // Encode with three streams.
>+  std::vector<MockVideoEncoder*> original_encoders =
>+      helper_->factory()->encoders();
>+
>+  // All encoders are trusted, so simulcast adapter should be too.
>+  original_encoders[0]->set_has_trusted_rate_controller(true);
>+  original_encoders[1]->set_has_trusted_rate_controller(true);
>+  original_encoders[2]->set_has_trusted_rate_controller(true);
>+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>+  EXPECT_TRUE(adapter_->GetEncoderInfo().has_trusted_rate_controller);
>+
>+  // One encoder not trusted, so simulcast adapter should not be either.
>+  original_encoders[2]->set_has_trusted_rate_controller(false);
>+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>+  EXPECT_FALSE(adapter_->GetEncoderInfo().has_trusted_rate_controller);
>+
>+  // No encoder trusted, so simulcast adapter should not be either.
>+  original_encoders[0]->set_has_trusted_rate_controller(false);
>+  original_encoders[1]->set_has_trusted_rate_controller(false);
>+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>+  EXPECT_FALSE(adapter_->GetEncoderInfo().has_trusted_rate_controller);
>+}
>+
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_unittest.cc
>index f41636869789931739d6f93ab08e16bb598660fc..0c278160689dfe3639cf8d0914991024d5c2761b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_unittest.cc
>@@ -179,7 +179,7 @@ TEST(SimulcastTest, GetConfigWithNormalizedResolution) {
>       kMaxLayers, 640 + 1, 360 + 1, kMaxBitrateBps, kBitratePriority, kQpMax,
>       kMaxFps, !kScreenshare);
> 
>-  // Must be dividable by |2 ^ (num_layers - 1)|.
>+  // Must be divisible by |2 ^ (num_layers - 1)|.
>   EXPECT_EQ(kMaxLayers, streams.size());
>   EXPECT_EQ(320u, streams[0].width);
>   EXPECT_EQ(180u, streams[0].height);
>@@ -187,6 +187,40 @@ TEST(SimulcastTest, GetConfigWithNormalizedResolution) {
>   EXPECT_EQ(360u, streams[1].height);
> }
> 
>+TEST(SimulcastTest, GetConfigWithNormalizedResolutionDivisibleBy4) {
>+  test::ScopedFieldTrials field_trials(
>+      "WebRTC-NormalizeSimulcastResolution/Enabled-2/");
>+
>+  const size_t kMaxLayers = 2;
>+  std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
>+      kMaxLayers, 709, 501, kMaxBitrateBps, kBitratePriority, kQpMax, kMaxFps,
>+      !kScreenshare);
>+
>+  // Must be divisible by |2 ^ 2|.
>+  EXPECT_EQ(kMaxLayers, streams.size());
>+  EXPECT_EQ(354u, streams[0].width);
>+  EXPECT_EQ(250u, streams[0].height);
>+  EXPECT_EQ(708u, streams[1].width);
>+  EXPECT_EQ(500u, streams[1].height);
>+}
>+
>+TEST(SimulcastTest, GetConfigWithNormalizedResolutionDivisibleBy8) {
>+  test::ScopedFieldTrials field_trials(
>+      "WebRTC-NormalizeSimulcastResolution/Enabled-3/");
>+
>+  const size_t kMaxLayers = 2;
>+  std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
>+      kMaxLayers, 709, 501, kMaxBitrateBps, kBitratePriority, kQpMax, kMaxFps,
>+      !kScreenshare);
>+
>+  // Must be divisible by |2 ^ 3|.
>+  EXPECT_EQ(kMaxLayers, streams.size());
>+  EXPECT_EQ(352u, streams[0].width);
>+  EXPECT_EQ(248u, streams[0].height);
>+  EXPECT_EQ(704u, streams[1].width);
>+  EXPECT_EQ(496u, streams[1].height);
>+}
>+
> TEST(SimulcastTest, GetConfigForScreenshare) {
>   const size_t kMaxLayers = 3;
>   std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.cc
>index 4d858630b5824d704ee89c1b80416470db9a2e86..1a2f57725cec25aee430b468dc1592785b932050 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.cc
>@@ -10,9 +10,7 @@
> 
> #include "media/engine/vp8_encoder_simulcast_proxy.h"
> 
>-#include "media/engine/scopedvideoencoder.h"
> #include "media/engine/simulcast_encoder_adapter.h"
>-#include "rtc_base/checks.h"
> 
> namespace webrtc {
> VP8EncoderSimulcastProxy::VP8EncoderSimulcastProxy(VideoEncoderFactory* factory,
>@@ -57,28 +55,14 @@ int VP8EncoderSimulcastProxy::RegisterEncodeCompleteCallback(
>   return encoder_->RegisterEncodeCompleteCallback(callback);
> }
> 
>-int VP8EncoderSimulcastProxy::SetChannelParameters(uint32_t packet_loss,
>-                                                   int64_t rtt) {
>-  return encoder_->SetChannelParameters(packet_loss, rtt);
>-}
>-
> int VP8EncoderSimulcastProxy::SetRateAllocation(
>     const VideoBitrateAllocation& bitrate,
>     uint32_t new_framerate) {
>   return encoder_->SetRateAllocation(bitrate, new_framerate);
> }
> 
>-VideoEncoder::ScalingSettings VP8EncoderSimulcastProxy::GetScalingSettings()
>-    const {
>-  return encoder_->GetScalingSettings();
>-}
>-
>-bool VP8EncoderSimulcastProxy::SupportsNativeHandle() const {
>-  return encoder_->SupportsNativeHandle();
>-}
>-
>-const char* VP8EncoderSimulcastProxy::ImplementationName() const {
>-  return encoder_->ImplementationName();
>+VideoEncoder::EncoderInfo VP8EncoderSimulcastProxy::GetEncoderInfo() const {
>+  return encoder_->GetEncoderInfo();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.h
>index 685bcbed6ef846d79afbd3f8339890808f652a00..d04624f26c5d861507e68a682a22cefce8066c41 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.h
>@@ -16,8 +16,8 @@
> #include <vector>
> 
> #include "api/video_codecs/sdp_video_format.h"
>+#include "api/video_codecs/video_encoder.h"
> #include "api/video_codecs/video_encoder_factory.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8.h"
> 
> namespace webrtc {
> 
>@@ -42,14 +42,9 @@ class VP8EncoderSimulcastProxy : public VideoEncoder {
>              const CodecSpecificInfo* codec_specific_info,
>              const std::vector<FrameType>* frame_types) override;
>   int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
>-  int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>   int SetRateAllocation(const VideoBitrateAllocation& bitrate,
>                         uint32_t new_framerate) override;
>-
>-  VideoEncoder::ScalingSettings GetScalingSettings() const override;
>-
>-  bool SupportsNativeHandle() const override;
>-  const char* ImplementationName() const override;
>+  EncoderInfo GetEncoderInfo() const override;
> 
>  private:
>   VideoEncoderFactory* const factory_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy_unittest.cc
>index d1dc9a31df66bc5933faf0b8b750456d56a9bc73..d1b26bbe93e6913a7a9b27971e2c083971eb5552 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy_unittest.cc
>@@ -10,12 +10,11 @@
>  */
> 
> #include "media/engine/vp8_encoder_simulcast_proxy.h"
>-
> #include <string>
> 
> #include "api/test/mock_video_encoder_factory.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "media/engine/webrtcvideoencoderfactory.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>@@ -52,9 +51,7 @@ class MockEncoder : public VideoEncoder {
>               const CodecSpecificInfo* codecSpecificInfo,
>               const std::vector<FrameType>* frame_types) /* override */);
> 
>-  MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
>-
>-  MOCK_CONST_METHOD0(ImplementationName, const char*());
>+  MOCK_CONST_METHOD0(GetEncoderInfo, VideoEncoder::EncoderInfo(void));
> };
> 
> TEST(VP8EncoderSimulcastProxy, ChoosesCorrectImplementation) {
>@@ -94,8 +91,10 @@ TEST(VP8EncoderSimulcastProxy, ChoosesCorrectImplementation) {
> 
>   EXPECT_CALL(*mock_encoder, InitEncode(_, _, _))
>       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
>-  EXPECT_CALL(*mock_encoder, ImplementationName())
>-      .WillRepeatedly(Return(kImplementationName.c_str()));
>+  VideoEncoder::EncoderInfo encoder_info;
>+  encoder_info.implementation_name = kImplementationName;
>+  EXPECT_CALL(*mock_encoder, GetEncoderInfo())
>+      .WillRepeatedly(Return(encoder_info));
> 
>   EXPECT_CALL(simulcast_factory, CreateVideoEncoderProxy(_))
>       .Times(1)
>@@ -105,7 +104,8 @@ TEST(VP8EncoderSimulcastProxy, ChoosesCorrectImplementation) {
>                                                    SdpVideoFormat("VP8"));
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>             simulcast_enabled_proxy.InitEncode(&codec_settings, 4, 1200));
>-  EXPECT_EQ(kImplementationName, simulcast_enabled_proxy.ImplementationName());
>+  EXPECT_EQ(kImplementationName,
>+            simulcast_enabled_proxy.GetEncoderInfo().implementation_name);
> 
>   NiceMock<MockEncoder>* mock_encoder1 = new NiceMock<MockEncoder>();
>   NiceMock<MockEncoder>* mock_encoder2 = new NiceMock<MockEncoder>();
>@@ -116,23 +116,23 @@ TEST(VP8EncoderSimulcastProxy, ChoosesCorrectImplementation) {
>   EXPECT_CALL(*mock_encoder1, InitEncode(_, _, _))
>       .WillOnce(
>           Return(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED));
>-  EXPECT_CALL(*mock_encoder1, ImplementationName())
>-      .WillRepeatedly(Return(kImplementationName.c_str()));
>+  EXPECT_CALL(*mock_encoder1, GetEncoderInfo())
>+      .WillRepeatedly(Return(encoder_info));
> 
>   EXPECT_CALL(*mock_encoder2, InitEncode(_, _, _))
>       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
>-  EXPECT_CALL(*mock_encoder2, ImplementationName())
>-      .WillRepeatedly(Return(kImplementationName.c_str()));
>+  EXPECT_CALL(*mock_encoder2, GetEncoderInfo())
>+      .WillRepeatedly(Return(encoder_info));
> 
>   EXPECT_CALL(*mock_encoder3, InitEncode(_, _, _))
>       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
>-  EXPECT_CALL(*mock_encoder3, ImplementationName())
>-      .WillRepeatedly(Return(kImplementationName.c_str()));
>+  EXPECT_CALL(*mock_encoder3, GetEncoderInfo())
>+      .WillRepeatedly(Return(encoder_info));
> 
>   EXPECT_CALL(*mock_encoder4, InitEncode(_, _, _))
>       .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
>-  EXPECT_CALL(*mock_encoder4, ImplementationName())
>-      .WillRepeatedly(Return(kImplementationName.c_str()));
>+  EXPECT_CALL(*mock_encoder4, GetEncoderInfo())
>+      .WillRepeatedly(Return(encoder_info));
> 
>   EXPECT_CALL(nonsimulcast_factory, CreateVideoEncoderProxy(_))
>       .Times(4)
>@@ -146,12 +146,38 @@ TEST(VP8EncoderSimulcastProxy, ChoosesCorrectImplementation) {
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>             simulcast_disabled_proxy.InitEncode(&codec_settings, 4, 1200));
>   EXPECT_EQ(kSimulcastAdaptedImplementationName,
>-            simulcast_disabled_proxy.ImplementationName());
>+            simulcast_disabled_proxy.GetEncoderInfo().implementation_name);
> 
>   // Cleanup.
>   simulcast_enabled_proxy.Release();
>   simulcast_disabled_proxy.Release();
> }
> 
>+TEST(VP8EncoderSimulcastProxy, ForwardsTrustedSetting) {
>+  NiceMock<MockEncoder>* mock_encoder = new NiceMock<MockEncoder>();
>+  NiceMock<MockVideoEncoderFactory> simulcast_factory;
>+
>+  EXPECT_CALL(*mock_encoder, InitEncode(_, _, _))
>+      .WillOnce(Return(WEBRTC_VIDEO_CODEC_OK));
>+
>+  EXPECT_CALL(simulcast_factory, CreateVideoEncoderProxy(_))
>+      .Times(1)
>+      .WillOnce(Return(mock_encoder));
>+
>+  VP8EncoderSimulcastProxy simulcast_enabled_proxy(&simulcast_factory,
>+                                                   SdpVideoFormat("VP8"));
>+  VideoCodec codec_settings;
>+  webrtc::test::CodecSettings(kVideoCodecVP8, &codec_settings);
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            simulcast_enabled_proxy.InitEncode(&codec_settings, 4, 1200));
>+
>+  VideoEncoder::EncoderInfo info;
>+  info.has_trusted_rate_controller = true;
>+  EXPECT_CALL(*mock_encoder, GetEncoderInfo()).WillRepeatedly(Return(info));
>+
>+  EXPECT_TRUE(
>+      simulcast_enabled_proxy.GetEncoderInfo().has_trusted_rate_controller);
>+}
>+
> }  // namespace testing
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.cc
>index e60c592a3fabeed12eb58fed9a6b6ecd67fdf035..bc7d3c3e124ffefea36b021e5bc635a3602eb629 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.cc
>@@ -11,10 +11,10 @@
> #include "media/engine/webrtcmediaengine.h"
> 
> #include <algorithm>
>-#include <memory>
>-#include <tuple>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "api/video_codecs/video_decoder_factory.h"
> #include "api/video_codecs/video_encoder_factory.h"
> #include "media/engine/webrtcvoiceengine.h"
>@@ -38,23 +38,24 @@ MediaEngineInterface* CreateWebRtcMediaEngine(
>         audio_decoder_factory,
>     WebRtcVideoEncoderFactory* video_encoder_factory,
>     WebRtcVideoDecoderFactory* video_decoder_factory,
>+    std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+        video_bitrate_allocator_factory,
>     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
>     rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
>+  std::unique_ptr<VideoEngineInterface> video_engine;
> #ifdef HAVE_WEBRTC_VIDEO
>-  typedef WebRtcVideoEngine VideoEngine;
>-  std::tuple<std::unique_ptr<WebRtcVideoEncoderFactory>,
>-             std::unique_ptr<WebRtcVideoDecoderFactory>>
>-      video_args(
>-          (std::unique_ptr<WebRtcVideoEncoderFactory>(video_encoder_factory)),
>-          (std::unique_ptr<WebRtcVideoDecoderFactory>(video_decoder_factory)));
>+  video_engine = absl::make_unique<WebRtcVideoEngine>(
>+      std::unique_ptr<WebRtcVideoEncoderFactory>(video_encoder_factory),
>+      std::unique_ptr<WebRtcVideoDecoderFactory>(video_decoder_factory),
>+      std::move(video_bitrate_allocator_factory));
> #else
>-  typedef NullWebRtcVideoEngine VideoEngine;
>-  std::tuple<> video_args;
>+  video_engine = absl::make_unique<NullWebRtcVideoEngine>();
> #endif
>-  return new CompositeMediaEngine<WebRtcVoiceEngine, VideoEngine>(
>-      std::forward_as_tuple(adm, audio_encoder_factory, audio_decoder_factory,
>-                            audio_mixer, audio_processing),
>-      std::move(video_args));
>+  return new CompositeMediaEngine(
>+      absl::make_unique<WebRtcVoiceEngine>(adm, audio_encoder_factory,
>+                                           audio_decoder_factory, audio_mixer,
>+                                           audio_processing),
>+      std::move(video_engine));
> }
> 
> }  // namespace
>@@ -67,10 +68,43 @@ MediaEngineInterface* WebRtcMediaEngineFactory::Create(
>         audio_decoder_factory,
>     WebRtcVideoEncoderFactory* video_encoder_factory,
>     WebRtcVideoDecoderFactory* video_decoder_factory) {
>-  return CreateWebRtcMediaEngine(adm, audio_encoder_factory,
>-                                 audio_decoder_factory, video_encoder_factory,
>-                                 video_decoder_factory, nullptr,
>-                                 webrtc::AudioProcessingBuilder().Create());
>+  return WebRtcMediaEngineFactory::Create(
>+      adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
>+      video_decoder_factory,
>+      webrtc::CreateBuiltinVideoBitrateAllocatorFactory());
>+}
>+
>+MediaEngineInterface* WebRtcMediaEngineFactory::Create(
>+    webrtc::AudioDeviceModule* adm,
>+    const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
>+        audio_encoder_factory,
>+    const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
>+        audio_decoder_factory,
>+    WebRtcVideoEncoderFactory* video_encoder_factory,
>+    WebRtcVideoDecoderFactory* video_decoder_factory,
>+    std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+        video_bitrate_allocator_factory) {
>+  return CreateWebRtcMediaEngine(
>+      adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
>+      video_decoder_factory, std::move(video_bitrate_allocator_factory),
>+      nullptr, webrtc::AudioProcessingBuilder().Create());
>+}
>+
>+MediaEngineInterface* WebRtcMediaEngineFactory::Create(
>+    webrtc::AudioDeviceModule* adm,
>+    const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
>+        audio_encoder_factory,
>+    const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
>+        audio_decoder_factory,
>+    WebRtcVideoEncoderFactory* video_encoder_factory,
>+    WebRtcVideoDecoderFactory* video_decoder_factory,
>+    rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
>+    rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
>+  return WebRtcMediaEngineFactory::Create(
>+      adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
>+      video_decoder_factory,
>+      webrtc::CreateBuiltinVideoBitrateAllocatorFactory(), audio_mixer,
>+      audio_processing);
> }
> 
> MediaEngineInterface* WebRtcMediaEngineFactory::Create(
>@@ -81,11 +115,14 @@ MediaEngineInterface* WebRtcMediaEngineFactory::Create(
>         audio_decoder_factory,
>     WebRtcVideoEncoderFactory* video_encoder_factory,
>     WebRtcVideoDecoderFactory* video_decoder_factory,
>+    std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+        video_bitrate_allocator_factory,
>     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
>     rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
>   return CreateWebRtcMediaEngine(
>       adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
>-      video_decoder_factory, audio_mixer, audio_processing);
>+      video_decoder_factory, std::move(video_bitrate_allocator_factory),
>+      audio_mixer, audio_processing);
> }
> #endif
> 
>@@ -97,22 +134,35 @@ std::unique_ptr<MediaEngineInterface> WebRtcMediaEngineFactory::Create(
>     std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
>     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
>     rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
>+  return WebRtcMediaEngineFactory::Create(
>+      adm, audio_encoder_factory, audio_decoder_factory,
>+      std::move(video_encoder_factory), std::move(video_decoder_factory),
>+      webrtc::CreateBuiltinVideoBitrateAllocatorFactory(), audio_mixer,
>+      audio_processing);
>+}
>+
>+std::unique_ptr<MediaEngineInterface> WebRtcMediaEngineFactory::Create(
>+    rtc::scoped_refptr<webrtc::AudioDeviceModule> adm,
>+    rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory,
>+    std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
>+    std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
>+    std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+        video_bitrate_allocator_factory,
>+    rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
>+    rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
> #ifdef HAVE_WEBRTC_VIDEO
>-  typedef WebRtcVideoEngine VideoEngine;
>-  std::tuple<std::unique_ptr<webrtc::VideoEncoderFactory>,
>-             std::unique_ptr<webrtc::VideoDecoderFactory>>
>-      video_args(std::move(video_encoder_factory),
>-                 std::move(video_decoder_factory));
>+  auto video_engine = absl::make_unique<WebRtcVideoEngine>(
>+      std::move(video_encoder_factory), std::move(video_decoder_factory),
>+      std::move(video_bitrate_allocator_factory));
> #else
>-  typedef NullWebRtcVideoEngine VideoEngine;
>-  std::tuple<> video_args;
>+  auto video_engine = absl::make_unique<NullWebRtcVideoEngine>();
> #endif
>-  return std::unique_ptr<MediaEngineInterface>(
>-      new CompositeMediaEngine<WebRtcVoiceEngine, VideoEngine>(
>-          std::forward_as_tuple(adm, audio_encoder_factory,
>-                                audio_decoder_factory, audio_mixer,
>-                                audio_processing),
>-          std::move(video_args)));
>+  return std::unique_ptr<MediaEngineInterface>(new CompositeMediaEngine(
>+      absl::make_unique<WebRtcVoiceEngine>(adm, audio_encoder_factory,
>+                                           audio_decoder_factory, audio_mixer,
>+                                           audio_processing),
>+      std::move(video_engine)));
> }
> 
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.h
>index d82eccf9dc347faf96f8631fdf264b9022173af9..fbf4466732715f33278d85fd3a56cbf42dc3b2fb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.h
>@@ -25,6 +25,7 @@ class AudioMixer;
> class AudioProcessing;
> class VideoDecoderFactory;
> class VideoEncoderFactory;
>+class VideoBitrateAllocatorFactory;
> }  // namespace webrtc
> namespace cricket {
> class WebRtcVideoDecoderFactory;
>@@ -50,6 +51,29 @@ class WebRtcMediaEngineFactory {
>           audio_decoder_factory,
>       WebRtcVideoEncoderFactory* video_encoder_factory,
>       WebRtcVideoDecoderFactory* video_decoder_factory);
>+
>+  static MediaEngineInterface* Create(
>+      webrtc::AudioDeviceModule* adm,
>+      const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
>+          audio_encoder_factory,
>+      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
>+          audio_decoder_factory,
>+      WebRtcVideoEncoderFactory* video_encoder_factory,
>+      WebRtcVideoDecoderFactory* video_decoder_factory,
>+      rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
>+      rtc::scoped_refptr<webrtc::AudioProcessing> apm);
>+
>+  static MediaEngineInterface* Create(
>+      webrtc::AudioDeviceModule* adm,
>+      const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
>+          audio_encoder_factory,
>+      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
>+          audio_decoder_factory,
>+      WebRtcVideoEncoderFactory* video_encoder_factory,
>+      WebRtcVideoDecoderFactory* video_decoder_factory,
>+      std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+          video_bitrate_allocator_factory);
>+
>   static MediaEngineInterface* Create(
>       webrtc::AudioDeviceModule* adm,
>       const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
>@@ -58,6 +82,8 @@ class WebRtcMediaEngineFactory {
>           audio_decoder_factory,
>       WebRtcVideoEncoderFactory* video_encoder_factory,
>       WebRtcVideoDecoderFactory* video_decoder_factory,
>+      std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+          video_bitrate_allocator_factory,
>       rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
>       rtc::scoped_refptr<webrtc::AudioProcessing> apm);
> #endif
>@@ -73,6 +99,17 @@ class WebRtcMediaEngineFactory {
>       std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
>       rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
>       rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing);
>+
>+  static std::unique_ptr<MediaEngineInterface> Create(
>+      rtc::scoped_refptr<webrtc::AudioDeviceModule> adm,
>+      rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
>+      rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory,
>+      std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
>+      std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
>+      std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+          video_bitrate_allocator_factory,
>+      rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
>+      rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing);
> };
> 
> // Verify that extension IDs are within 1-byte extension range and are not
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine_unittest.cc
>index 4fa9a6b5ee35d89cbfc6854919d1bf44ad539640..eec7dc39c4881c369d4be642c9f6c3cdb8310044 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine_unittest.cc
>@@ -12,6 +12,7 @@
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "media/engine/webrtcmediaengine.h"
>@@ -249,4 +250,15 @@ TEST(WebRtcMediaEngineFactoryTest, CreateWithBuiltinDecoders) {
>   EXPECT_TRUE(engine);
> }
> 
>+TEST(WebRtcMediaEngineFactoryTest, CreateWithVideoBitrateFactory) {
>+  std::unique_ptr<MediaEngineInterface> engine(WebRtcMediaEngineFactory::Create(
>+      nullptr /* adm */, webrtc::CreateBuiltinAudioEncoderFactory(),
>+      webrtc::CreateBuiltinAudioDecoderFactory(),
>+      webrtc::CreateBuiltinVideoEncoderFactory(),
>+      webrtc::CreateBuiltinVideoDecoderFactory(),
>+      webrtc::CreateBuiltinVideoBitrateAllocatorFactory(),
>+      nullptr /* audio_mixer */, webrtc::AudioProcessingBuilder().Create()));
>+  EXPECT_TRUE(engine);
>+}
>+
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h
>index 016cb09da000f474fadd65622934dd6308a950e0..f00d9a69e0e266d589d2ff9472bd54a4f3ee279d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideodecoderfactory.h
>@@ -13,9 +13,10 @@
> 
> #include <string>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "api/video/video_codec_type.h"
> #include "media/base/codec.h"
> #include "rtc_base/refcount.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> class VideoDecoder;
>@@ -29,7 +30,7 @@ struct VideoDecoderParams {
> 
> // Deprecated. Use webrtc::VideoDecoderFactory instead.
> // https://bugs.chromium.org/p/webrtc/issues/detail?id=7925
>-class WebRtcVideoDecoderFactory {
>+class RTC_EXPORT WebRtcVideoDecoderFactory {
>  public:
>   // Caller takes the ownership of the returned object and it should be released
>   // by calling DestroyVideoDecoder().
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h
>index 97ac03b473f8f0c78c84ee865036560cad619213..d39e0fd5f95ae4540801503f2d558fd80f20b775 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoencoderfactory.h
>@@ -14,8 +14,9 @@
> #include <string>
> #include <vector>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "api/video/video_codec_type.h"
> #include "media/base/codec.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> class VideoEncoder;
>@@ -25,7 +26,7 @@ namespace cricket {
> 
> // Deprecated. Use webrtc::VideoEncoderFactory instead.
> // https://bugs.chromium.org/p/webrtc/issues/detail?id=7925
>-class WebRtcVideoEncoderFactory {
>+class RTC_EXPORT WebRtcVideoEncoderFactory {
>  public:
>   virtual ~WebRtcVideoEncoderFactory() {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.cc
>index 40b0c45a3d832c1166ac3744efad65e7fe036d01..a75db73697069b937fa88b617c3e3d9a4f4b48f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.cc
>@@ -16,6 +16,7 @@
> #include <string>
> #include <utility>
> 
>+#include "absl/strings/match.h"
> #include "api/video_codecs/sdp_video_format.h"
> #include "api/video_codecs/video_decoder_factory.h"
> #include "api/video_codecs/video_encoder.h"
>@@ -111,8 +112,8 @@ std::vector<VideoCodec> AssignPayloadTypesAndDefaultCodecs(
>     }
> 
>     // Add associated RTX codec for non-FEC codecs.
>-    if (!CodecNamesEq(codec.name, kUlpfecCodecName) &&
>-        !CodecNamesEq(codec.name, kFlexfecCodecName)) {
>+    if (!absl::EqualsIgnoreCase(codec.name, kUlpfecCodecName) &&
>+        !absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName)) {
>       output_codecs.push_back(
>           VideoCodec::CreateRtxCodec(payload_type, codec.id));
> 
>@@ -147,8 +148,8 @@ int GetMaxFramerate(const webrtc::VideoEncoderConfig& encoder_config,
> }
> 
> bool IsTemporalLayersSupported(const std::string& codec_name) {
>-  return CodecNamesEq(codec_name, kVp8CodecName) ||
>-         CodecNamesEq(codec_name, kVp9CodecName);
>+  return absl::EqualsIgnoreCase(codec_name, kVp8CodecName) ||
>+         absl::EqualsIgnoreCase(codec_name, kVp9CodecName);
> }
> 
> static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) {
>@@ -220,9 +221,9 @@ static bool ValidateStreamParams(const StreamParams& sp) {
> // Returns true if the given codec is disallowed from doing simulcast.
> bool IsCodecBlacklistedForSimulcast(const std::string& codec_name) {
>   return webrtc::field_trial::IsEnabled("WebRTC-H264Simulcast")
>-    ? CodecNamesEq(codec_name, kVp9CodecName)
>-    : CodecNamesEq(codec_name, kH264CodecName) ||
>-      CodecNamesEq(codec_name, kVp9CodecName);
>+             ? absl::EqualsIgnoreCase(codec_name, kVp9CodecName)
>+             : absl::EqualsIgnoreCase(codec_name, kH264CodecName) ||
>+                   absl::EqualsIgnoreCase(codec_name, kVp9CodecName);
> }
> 
> // The selected thresholds for QVGA and VGA corresponded to a QP around 10.
>@@ -337,14 +338,14 @@ WebRtcVideoChannel::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
>     denoising = parameters_.options.video_noise_reduction.value_or(false);
>   }
> 
>-  if (CodecNamesEq(codec.name, kH264CodecName)) {
>+  if (absl::EqualsIgnoreCase(codec.name, kH264CodecName)) {
>     webrtc::VideoCodecH264 h264_settings =
>         webrtc::VideoEncoder::GetDefaultH264Settings();
>     h264_settings.frameDroppingOn = frame_dropping;
>     return new rtc::RefCountedObject<
>         webrtc::VideoEncoderConfig::H264EncoderSpecificSettings>(h264_settings);
>   }
>-  if (CodecNamesEq(codec.name, kVp8CodecName)) {
>+  if (absl::EqualsIgnoreCase(codec.name, kVp8CodecName)) {
>     webrtc::VideoCodecVP8 vp8_settings =
>         webrtc::VideoEncoder::GetDefaultVp8Settings();
>     vp8_settings.automaticResizeOn = automatic_resize;
>@@ -354,7 +355,7 @@ WebRtcVideoChannel::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
>     return new rtc::RefCountedObject<
>         webrtc::VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8_settings);
>   }
>-  if (CodecNamesEq(codec.name, kVp9CodecName)) {
>+  if (absl::EqualsIgnoreCase(codec.name, kVp9CodecName)) {
>     webrtc::VideoCodecVP9 vp9_settings =
>         webrtc::VideoEncoder::GetDefaultVp9Settings();
>     const size_t default_num_spatial_layers =
>@@ -436,20 +437,26 @@ void DefaultUnsignalledSsrcHandler::SetDefaultSink(
> #if defined(USE_BUILTIN_SW_CODECS)
> WebRtcVideoEngine::WebRtcVideoEngine(
>     std::unique_ptr<WebRtcVideoEncoderFactory> external_video_encoder_factory,
>-    std::unique_ptr<WebRtcVideoDecoderFactory> external_video_decoder_factory)
>+    std::unique_ptr<WebRtcVideoDecoderFactory> external_video_decoder_factory,
>+    std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+        video_bitrate_allocator_factory)
>     : decoder_factory_(ConvertVideoDecoderFactory(
>           std::move(external_video_decoder_factory))),
>       encoder_factory_(ConvertVideoEncoderFactory(
>-          std::move(external_video_encoder_factory))) {
>+          std::move(external_video_encoder_factory))),
>+      bitrate_allocator_factory_(std::move(video_bitrate_allocator_factory)) {
>   RTC_LOG(LS_INFO) << "WebRtcVideoEngine::WebRtcVideoEngine()";
> }
> #endif
> 
> WebRtcVideoEngine::WebRtcVideoEngine(
>     std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
>-    std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory)
>+    std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
>+    std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+        video_bitrate_allocator_factory)
>     : decoder_factory_(std::move(video_decoder_factory)),
>-      encoder_factory_(std::move(video_encoder_factory)) {
>+      encoder_factory_(std::move(video_encoder_factory)),
>+      bitrate_allocator_factory_(std::move(video_bitrate_allocator_factory)) {
>   RTC_LOG(LS_INFO) << "WebRtcVideoEngine::WebRtcVideoEngine()";
> }
> 
>@@ -457,15 +464,16 @@ WebRtcVideoEngine::~WebRtcVideoEngine() {
>   RTC_LOG(LS_INFO) << "WebRtcVideoEngine::~WebRtcVideoEngine";
> }
> 
>-WebRtcVideoChannel* WebRtcVideoEngine::CreateChannel(
>+VideoMediaChannel* WebRtcVideoEngine::CreateMediaChannel(
>     webrtc::Call* call,
>     const MediaConfig& config,
>-    const VideoOptions& options) {
>-  RTC_LOG(LS_INFO) << "CreateChannel. Options: " << options.ToString();
>-  return new WebRtcVideoChannel(call, config, options, encoder_factory_.get(),
>-                                decoder_factory_.get());
>+    const VideoOptions& options,
>+    const webrtc::CryptoOptions& crypto_options) {
>+  RTC_LOG(LS_INFO) << "CreateMediaChannel. Options: " << options.ToString();
>+  return new WebRtcVideoChannel(call, config, options, crypto_options,
>+                                encoder_factory_.get(), decoder_factory_.get(),
>+                                bitrate_allocator_factory_.get());
> }
>-
> std::vector<VideoCodec> WebRtcVideoEngine::codecs() const {
>   return AssignPayloadTypesAndDefaultCodecs(encoder_factory_.get());
> }
>@@ -496,6 +504,11 @@ RtpCapabilities WebRtcVideoEngine::GetCapabilities() const {
>   capabilities.header_extensions.push_back(
>       webrtc::RtpExtension(webrtc::RtpExtension::kFrameMarkingUri,
>                            webrtc::RtpExtension::kFrameMarkingDefaultId));
>+  if (webrtc::field_trial::IsEnabled("WebRTC-GenericDescriptorAdvertised")) {
>+    capabilities.header_extensions.push_back(webrtc::RtpExtension(
>+        webrtc::RtpExtension::kGenericFrameDescriptorUri,
>+        webrtc::RtpExtension::kGenericFrameDescriptorDefaultId));
>+  }
>   // TODO(bugs.webrtc.org/4050): Add MID header extension as capability once MID
>   // demuxing is completed.
>   // capabilities.header_extensions.push_back(webrtc::RtpExtension(
>@@ -507,16 +520,23 @@ WebRtcVideoChannel::WebRtcVideoChannel(
>     webrtc::Call* call,
>     const MediaConfig& config,
>     const VideoOptions& options,
>+    const webrtc::CryptoOptions& crypto_options,
>     webrtc::VideoEncoderFactory* encoder_factory,
>-    webrtc::VideoDecoderFactory* decoder_factory)
>+    webrtc::VideoDecoderFactory* decoder_factory,
>+    webrtc::VideoBitrateAllocatorFactory* bitrate_allocator_factory)
>     : VideoMediaChannel(config),
>       call_(call),
>       unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
>       video_config_(config.video),
>       encoder_factory_(encoder_factory),
>       decoder_factory_(decoder_factory),
>+      bitrate_allocator_factory_(bitrate_allocator_factory),
>+      preferred_dscp_(rtc::DSCP_DEFAULT),
>       default_send_options_(options),
>-      last_stats_log_ms_(-1) {
>+      last_stats_log_ms_(-1),
>+      discard_unknown_ssrc_packets_(webrtc::field_trial::IsEnabled(
>+          "WebRTC-Video-DiscardPacketsWithUnknownSsrc")),
>+      crypto_options_(crypto_options) {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
> 
>   rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
>@@ -612,6 +632,9 @@ bool WebRtcVideoChannel::GetChangedSendParameters(
>     changed_params->codec = selected_send_codec;
> 
>   // Handle RTP header extensions.
>+  if (params.extmap_allow_mixed != ExtmapAllowMixed()) {
>+    changed_params->extmap_allow_mixed = params.extmap_allow_mixed;
>+  }
>   std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
>       params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
>   if (!send_rtp_extensions_ || (*send_rtp_extensions_ != filtered_extensions)) {
>@@ -649,7 +672,7 @@ bool WebRtcVideoChannel::GetChangedSendParameters(
> }
> 
> rtc::DiffServCodePoint WebRtcVideoChannel::PreferredDscp() const {
>-  return rtc::DSCP_AF41;
>+  return preferred_dscp_;
> }
> 
> bool WebRtcVideoChannel::SetSendParameters(const VideoSendParameters& params) {
>@@ -666,6 +689,9 @@ bool WebRtcVideoChannel::SetSendParameters(const VideoSendParameters& params) {
>     RTC_LOG(LS_INFO) << "Using codec: " << codec_settings.codec.ToString();
>   }
> 
>+  if (changed_params.extmap_allow_mixed) {
>+    SetExtmapAllowMixed(*changed_params.extmap_allow_mixed);
>+  }
>   if (changed_params.rtp_header_extensions) {
>     send_rtp_extensions_ = changed_params.rtp_header_extensions;
>   }
>@@ -764,11 +790,34 @@ webrtc::RTCError WebRtcVideoChannel::SetRtpSendParameters(
>   // different order (which should change the send codec).
>   webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
>   if (current_parameters.codecs != parameters.codecs) {
>-    RTC_LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
>-                      << "is not currently supported.";
>+    RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
>+                       << "is not currently supported.";
>     return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
>   }
> 
>+  if (!parameters.encodings.empty()) {
>+    const auto& priority = parameters.encodings[0].network_priority;
>+    rtc::DiffServCodePoint new_dscp = rtc::DSCP_DEFAULT;
>+    if (priority == 0.5 * webrtc::kDefaultBitratePriority) {
>+      new_dscp = rtc::DSCP_CS1;
>+    } else if (priority == webrtc::kDefaultBitratePriority) {
>+      new_dscp = rtc::DSCP_DEFAULT;
>+    } else if (priority == 2.0 * webrtc::kDefaultBitratePriority) {
>+      new_dscp = rtc::DSCP_AF42;
>+    } else if (priority == 4.0 * webrtc::kDefaultBitratePriority) {
>+      new_dscp = rtc::DSCP_AF41;
>+    } else {
>+      RTC_LOG(LS_WARNING) << "Received invalid send network priority: "
>+                          << priority;
>+      return webrtc::RTCError(webrtc::RTCErrorType::INVALID_RANGE);
>+    }
>+
>+    if (new_dscp != preferred_dscp_) {
>+      preferred_dscp_ = new_dscp;
>+      MediaChannel::UpdateDscp();
>+    }
>+  }
>+
>   return it->second->SetRtpParameters(parameters);
> }
> 
>@@ -832,8 +881,8 @@ bool WebRtcVideoChannel::SetRtpReceiveParameters(
> 
>   webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
>   if (current_parameters != parameters) {
>-    RTC_LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
>-                      << "unsupported.";
>+    RTC_DLOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
>+                       << "unsupported.";
>     return false;
>   }
>   return true;
>@@ -949,7 +998,7 @@ bool WebRtcVideoChannel::SetSend(bool send) {
>   TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetSend");
>   RTC_LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
>   if (send && !send_codec_) {
>-    RTC_LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
>+    RTC_DLOG(LS_ERROR) << "SetSend(true) called before setting codec.";
>     return false;
>   }
>   {
>@@ -1028,6 +1077,11 @@ bool WebRtcVideoChannel::AddSendStream(const StreamParams& sp) {
>   config.encoder_settings.experiment_cpu_load_estimator =
>       video_config_.experiment_cpu_load_estimator;
>   config.encoder_settings.encoder_factory = encoder_factory_;
>+  config.encoder_settings.bitrate_allocator_factory =
>+      bitrate_allocator_factory_;
>+  config.crypto_options = crypto_options_;
>+  config.rtp.extmap_allow_mixed = ExtmapAllowMixed();
>+  config.rtcp_report_interval_ms = video_config_.rtcp_report_interval_ms;
> 
>   WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
>       call_, sp, std::move(config), default_send_options_,
>@@ -1144,6 +1198,7 @@ bool WebRtcVideoChannel::AddRecvStream(const StreamParams& sp,
>   webrtc::FlexfecReceiveStream::Config flexfec_config(this);
>   ConfigureReceiverRtp(&config, &flexfec_config, sp);
> 
>+  config.crypto_options = crypto_options_;
>   // TODO(nisse): Rename config variable to avoid negation.
>   config.disable_prerenderer_smoothing =
>       !video_config_.enable_prerenderer_smoothing;
>@@ -1332,10 +1387,10 @@ void WebRtcVideoChannel::FillSendAndReceiveCodecStats(
> }
> 
> void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                          const rtc::PacketTime& packet_time) {
>+                                          int64_t packet_time_us) {
>   const webrtc::PacketReceiver::DeliveryStatus delivery_result =
>       call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
>-                                       packet_time.timestamp);
>+                                       packet_time_us);
>   switch (delivery_result) {
>     case webrtc::PacketReceiver::DELIVERY_OK:
>       return;
>@@ -1345,6 +1400,10 @@ void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>       break;
>   }
> 
>+  if (discard_unknown_ssrc_packets_) {
>+    return;
>+  }
>+
>   uint32_t ssrc = 0;
>   if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
>     return;
>@@ -1379,7 +1438,7 @@ void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>   }
> 
>   if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
>-                                       packet_time.timestamp) !=
>+                                       packet_time_us) !=
>       webrtc::PacketReceiver::DELIVERY_OK) {
>     RTC_LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
>     return;
>@@ -1387,13 +1446,13 @@ void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
> }
> 
> void WebRtcVideoChannel::OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
>-                                        const rtc::PacketTime& packet_time) {
>+                                        int64_t packet_time_us) {
>   // TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver
>   // for both audio and video on the same path. Since BundleFilter doesn't
>   // filter RTCP anymore incoming RTCP packets could've been going to audio (so
>   // logging failures spam the log).
>   call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
>-                                   packet_time.timestamp);
>+                                   packet_time_us);
> }
> 
> void WebRtcVideoChannel::OnReadyToSend(bool ready) {
>@@ -1412,31 +1471,46 @@ void WebRtcVideoChannel::OnNetworkRouteChanged(
>       network_route.packet_overhead);
> }
> 
>-void WebRtcVideoChannel::SetInterface(NetworkInterface* iface) {
>-  MediaChannel::SetInterface(iface);
>+void WebRtcVideoChannel::SetInterface(
>+    NetworkInterface* iface,
>+    webrtc::MediaTransportInterface* media_transport) {
>+  // TODO(sukhanov): Video is not currently supported with media transport.
>+  RTC_CHECK(media_transport == nullptr);
>+
>+  MediaChannel::SetInterface(iface, media_transport);
>   // Set the RTP recv/send buffer to a bigger size.
> 
>-  // The group here can be either a positive integer with an explicit size, in
>-  // which case that is used as size. All other values shall result in the
>-  // default value being used.
>-  const std::string group_name =
>-      webrtc::field_trial::FindFullName("WebRTC-IncreasedReceivebuffers");
>-  int recv_buffer_size = kVideoRtpBufferSize;
>-  if (!group_name.empty() &&
>-      (sscanf(group_name.c_str(), "%d", &recv_buffer_size) != 1 ||
>-       recv_buffer_size <= 0)) {
>-    RTC_LOG(LS_WARNING) << "Invalid receive buffer size: " << group_name;
>-    recv_buffer_size = kVideoRtpBufferSize;
>-  }
>   MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_RCVBUF,
>-                          recv_buffer_size);
>+                          kVideoRtpRecvBufferSize);
> 
>   // Speculative change to increase the outbound socket buffer size.
>   // In b/15152257, we are seeing a significant number of packets discarded
>   // due to lack of socket buffer space, although it's not yet clear what the
>   // ideal value should be.
>   MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_SNDBUF,
>-                          kVideoRtpBufferSize);
>+                          kVideoRtpSendBufferSize);
>+}
>+
>+void WebRtcVideoChannel::SetFrameDecryptor(
>+    uint32_t ssrc,
>+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
>+  rtc::CritScope stream_lock(&stream_crit_);
>+  auto matching_stream = receive_streams_.find(ssrc);
>+  if (matching_stream != receive_streams_.end()) {
>+    matching_stream->second->SetFrameDecryptor(frame_decryptor);
>+  }
>+}
>+
>+void WebRtcVideoChannel::SetFrameEncryptor(
>+    uint32_t ssrc,
>+    rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor) {
>+  rtc::CritScope stream_lock(&stream_crit_);
>+  auto matching_stream = send_streams_.find(ssrc);
>+  if (matching_stream != send_streams_.end()) {
>+    matching_stream->second->SetFrameEncryptor(frame_encryptor);
>+  } else {
>+    RTC_LOG(LS_ERROR) << "No stream found to attach frame encryptor";
>+  }
> }
> 
> absl::optional<uint32_t> WebRtcVideoChannel::GetDefaultReceiveStreamSsrc() {
>@@ -1474,6 +1548,10 @@ bool WebRtcVideoChannel::SendRtp(const uint8_t* data,
>   if (DscpEnabled()) {
>     rtc_options.dscp = PreferredDscp();
>   }
>+  rtc_options.info_signaled_after_sent.included_in_feedback =
>+      options.included_in_feedback;
>+  rtc_options.info_signaled_after_sent.included_in_allocation =
>+      options.included_in_allocation;
>   return MediaChannel::SendPacket(&packet, rtc_options);
> }
> 
>@@ -1483,6 +1561,7 @@ bool WebRtcVideoChannel::SendRtcp(const uint8_t* data, size_t len) {
>   if (DscpEnabled()) {
>     rtc_options.dscp = PreferredDscp();
>   }
>+
>   return MediaChannel::SendRtcp(&packet, rtc_options);
> }
> 
>@@ -1569,7 +1648,6 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
>                                          ? webrtc::RtcpMode::kReducedSize
>                                          : webrtc::RtcpMode::kCompound;
>   parameters_.config.rtp.mid = send_params.mid;
>-
>   rtp_parameters_.rtcp.reduced_size = send_params.rtcp.reduced_size;
> 
>   if (codec_settings) {
>@@ -1699,6 +1777,10 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::SetSendParameters(
>         parameters_.config.rtp.rtcp_mode == webrtc::RtcpMode::kReducedSize;
>     recreate_stream = true;
>   }
>+  if (params.extmap_allow_mixed) {
>+    parameters_.config.rtp.extmap_allow_mixed = *params.extmap_allow_mixed;
>+    recreate_stream = true;
>+  }
>   if (params.rtp_header_extensions) {
>     parameters_.config.rtp.extensions = *params.rtp_header_extensions;
>     rtp_parameters_.header_extensions = *params.rtp_header_extensions;
>@@ -1799,6 +1881,15 @@ WebRtcVideoChannel::WebRtcVideoSendStream::GetRtpParameters() const {
>   return rtp_parameters_;
> }
> 
>+void WebRtcVideoChannel::WebRtcVideoSendStream::SetFrameEncryptor(
>+    rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  parameters_.config.frame_encryptor = frame_encryptor;
>+  if (stream_) {
>+    RecreateWebRtcStream();
>+  }
>+}
>+
> void WebRtcVideoChannel::WebRtcVideoSendStream::UpdateSendState() {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>   if (sending_) {
>@@ -2161,26 +2252,20 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetSources() {
>   return stream_->GetSources();
> }
> 
>-absl::optional<uint32_t>
>-WebRtcVideoChannel::WebRtcVideoReceiveStream::GetFirstPrimarySsrc() const {
>+webrtc::RtpParameters
>+WebRtcVideoChannel::WebRtcVideoReceiveStream::GetRtpParameters() const {
>+  webrtc::RtpParameters rtp_parameters;
>+
>   std::vector<uint32_t> primary_ssrcs;
>   stream_params_.GetPrimarySsrcs(&primary_ssrcs);
>-
>-  if (primary_ssrcs.empty()) {
>-    RTC_LOG(LS_WARNING)
>-        << "Empty primary ssrcs vector, returning empty optional";
>-    return absl::nullopt;
>-  } else {
>-    return primary_ssrcs[0];
>+  for (uint32_t ssrc : primary_ssrcs) {
>+    rtp_parameters.encodings.emplace_back();
>+    rtp_parameters.encodings.back().ssrc = ssrc;
>   }
>-}
> 
>-webrtc::RtpParameters
>-WebRtcVideoChannel::WebRtcVideoReceiveStream::GetRtpParameters() const {
>-  webrtc::RtpParameters rtp_parameters;
>-  rtp_parameters.encodings.emplace_back();
>-  rtp_parameters.encodings[0].ssrc = GetFirstPrimarySsrc();
>   rtp_parameters.header_extensions = config_.rtp.extensions;
>+  rtp_parameters.rtcp.reduced_size =
>+      config_.rtp.rtcp_mode == webrtc::RtcpMode::kReducedSize;
> 
>   return rtp_parameters;
> }
>@@ -2230,9 +2315,9 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetLocalSsrc(
>   // right now this can't be done due to unittests depending on receiving what
>   // they are sending from the same MediaChannel.
>   if (local_ssrc == config_.rtp.local_ssrc) {
>-    RTC_LOG(LS_INFO) << "Ignoring call to SetLocalSsrc because parameters are "
>-                        "unchanged; local_ssrc="
>-                     << local_ssrc;
>+    RTC_DLOG(LS_INFO) << "Ignoring call to SetLocalSsrc because parameters are "
>+                         "unchanged; local_ssrc="
>+                      << local_ssrc;
>     return;
>   }
> 
>@@ -2372,6 +2457,14 @@ bool WebRtcVideoChannel::WebRtcVideoReceiveStream::IsDefaultStream() const {
>   return default_stream_;
> }
> 
>+void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetFrameDecryptor(
>+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
>+  config_.frame_decryptor = frame_decryptor;
>+  if (stream_) {
>+    RecreateWebRtcVideoStream();
>+  }
>+}
>+
> void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetSink(
>     rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
>   rtc::CritScope crit(&sink_lock_);
>@@ -2607,10 +2700,11 @@ std::vector<webrtc::VideoStream> EncoderStreamFactory::CreateEncoderStreams(
>   std::vector<webrtc::VideoStream> layers;
> 
>   if (encoder_config.number_of_streams > 1 ||
>-     ((CodecNamesEq(codec_name_, kVp8CodecName) ||
>-        CodecNamesEq(codec_name_, kH264CodecName)) &&
>+      ((absl::EqualsIgnoreCase(codec_name_, kVp8CodecName) ||
>+        absl::EqualsIgnoreCase(codec_name_, kH264CodecName)) &&
>        is_screenshare_ && screenshare_config_explicitly_enabled_)) {
>-    bool temporal_layers_supported = CodecNamesEq(codec_name_, kVp8CodecName);
>+    bool temporal_layers_supported =
>+        absl::EqualsIgnoreCase(codec_name_, kVp8CodecName);
>     layers = GetSimulcastConfig(encoder_config.number_of_streams, width, height,
>                                 0 /*not used*/, encoder_config.bitrate_priority,
>                                 max_qp_, 0 /*not_used*/, is_screenshare_,
>@@ -2703,7 +2797,7 @@ std::vector<webrtc::VideoStream> EncoderStreamFactory::CreateEncoderStreams(
>   layer.max_qp = max_qp_;
>   layer.bitrate_priority = encoder_config.bitrate_priority;
> 
>-  if (CodecNamesEq(codec_name_, kVp9CodecName)) {
>+  if (absl::EqualsIgnoreCase(codec_name_, kVp9CodecName)) {
>     RTC_DCHECK(encoder_config.encoder_specific_settings);
>     // Use VP9 SVC layering from codec settings which might be initialized
>     // though field trial in ConfigureVideoEncoderSettings.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.h
>index 8a1e6ad4d1703648edcf4570df47a33917d3c61a..3b79ad40b8a2d673fb87f1264ca8b55d724a9f53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.h
>@@ -19,6 +19,7 @@
> 
> #include "absl/types/optional.h"
> #include "api/call/transport.h"
>+#include "api/video/video_bitrate_allocator_factory.h"
> #include "api/video/video_frame.h"
> #include "api/video/video_sink_interface.h"
> #include "api/video/video_source_interface.h"
>@@ -78,43 +79,53 @@ class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
> };
> 
> // WebRtcVideoEngine is used for the new native WebRTC Video API (webrtc:1667).
>-class WebRtcVideoEngine {
>+class WebRtcVideoEngine : public VideoEngineInterface {
>  public:
> #if defined(USE_BUILTIN_SW_CODECS)
>   // Internal SW video codecs will be added on top of the external codecs.
>   WebRtcVideoEngine(
>       std::unique_ptr<WebRtcVideoEncoderFactory> external_video_encoder_factory,
>-      std::unique_ptr<WebRtcVideoDecoderFactory>
>-          external_video_decoder_factory);
>+      std::unique_ptr<WebRtcVideoDecoderFactory> external_video_decoder_factory,
>+      std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+          video_bitrate_allocator_factory);
> #endif
> 
>   // These video codec factories represents all video codecs, i.e. both software
>   // and external hardware codecs.
>   WebRtcVideoEngine(
>       std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
>-      std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory);
>+      std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
>+      std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+          video_bitrate_allocator_factory);
> 
>-  virtual ~WebRtcVideoEngine();
>+  ~WebRtcVideoEngine() override;
> 
>-  WebRtcVideoChannel* CreateChannel(webrtc::Call* call,
>-                                    const MediaConfig& config,
>-                                    const VideoOptions& options);
>+  VideoMediaChannel* CreateMediaChannel(
>+      webrtc::Call* call,
>+      const MediaConfig& config,
>+      const VideoOptions& options,
>+      const webrtc::CryptoOptions& crypto_options) override;
> 
>-  std::vector<VideoCodec> codecs() const;
>-  RtpCapabilities GetCapabilities() const;
>+  std::vector<VideoCodec> codecs() const override;
>+  RtpCapabilities GetCapabilities() const override;
> 
>  private:
>   const std::unique_ptr<webrtc::VideoDecoderFactory> decoder_factory_;
>   const std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory_;
>+  const std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
>+      bitrate_allocator_factory_;
> };
> 
> class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>  public:
>-  WebRtcVideoChannel(webrtc::Call* call,
>-                     const MediaConfig& config,
>-                     const VideoOptions& options,
>-                     webrtc::VideoEncoderFactory* encoder_factory,
>-                     webrtc::VideoDecoderFactory* decoder_factory);
>+  WebRtcVideoChannel(
>+      webrtc::Call* call,
>+      const MediaConfig& config,
>+      const VideoOptions& options,
>+      const webrtc::CryptoOptions& crypto_options,
>+      webrtc::VideoEncoderFactory* encoder_factory,
>+      webrtc::VideoDecoderFactory* decoder_factory,
>+      webrtc::VideoBitrateAllocatorFactory* bitrate_allocator_factory);
>   ~WebRtcVideoChannel() override;
> 
>   // VideoMediaChannel implementation
>@@ -147,13 +158,28 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>   bool GetStats(VideoMediaInfo* info) override;
> 
>   void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                        const rtc::PacketTime& packet_time) override;
>+                        int64_t packet_time_us) override;
>   void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
>-                      const rtc::PacketTime& packet_time) override;
>+                      int64_t packet_time_us) override;
>   void OnReadyToSend(bool ready) override;
>   void OnNetworkRouteChanged(const std::string& transport_name,
>                              const rtc::NetworkRoute& network_route) override;
>-  void SetInterface(NetworkInterface* iface) override;
>+  void SetInterface(NetworkInterface* iface,
>+                    webrtc::MediaTransportInterface* media_transport) override;
>+
>+  // E2E Encrypted Video Frame API
>+  // Set a frame decryptor to a particular ssrc that will intercept all
>+  // incoming video frames and attempt to decrypt them before forwarding the
>+  // result.
>+  void SetFrameDecryptor(uint32_t ssrc,
>+                         rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
>+                             frame_decryptor) override;
>+  // Set a frame encryptor to a particular ssrc that will intercept all
>+  // outgoing video frames and attempt to encrypt them and forward the result
>+  // to the packetizer.
>+  void SetFrameEncryptor(uint32_t ssrc,
>+                         rtc::scoped_refptr<webrtc::FrameEncryptorInterface>
>+                             frame_encryptor) override;
> 
>   // Implemented for VideoMediaChannelTest.
>   bool sending() const { return sending_; }
>@@ -201,6 +227,7 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>     absl::optional<VideoCodecSettings> codec;
>     absl::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
>     absl::optional<std::string> mid;
>+    absl::optional<bool> extmap_allow_mixed;
>     absl::optional<int> max_bandwidth_bps;
>     absl::optional<bool> conference_mode;
>     absl::optional<webrtc::RtcpMode> rtcp_mode;
>@@ -257,6 +284,9 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>     webrtc::RTCError SetRtpParameters(const webrtc::RtpParameters& parameters);
>     webrtc::RtpParameters GetRtpParameters() const;
> 
>+    void SetFrameEncryptor(
>+        rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor);
>+
>     // Implements rtc::VideoSourceInterface<webrtc::VideoFrame>.
>     // WebRtcVideoSendStream acts as a source to the webrtc::VideoSendStream
>     // in |stream_|. This is done to proxy VideoSinkWants from the encoder to
>@@ -371,6 +401,9 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>     void OnFrame(const webrtc::VideoFrame& frame) override;
>     bool IsDefaultStream() const;
> 
>+    void SetFrameDecryptor(
>+        rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor);
>+
>     void SetSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink);
> 
>     VideoReceiverInfo GetVideoReceiverInfo(bool log_stats);
>@@ -387,8 +420,6 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
> 
>     std::string GetCodecNameFromPayloadType(int payload_type);
> 
>-    absl::optional<uint32_t> GetFirstPrimarySsrc() const;
>-
>     webrtc::Call* const call_;
>     const StreamParams stream_params_;
> 
>@@ -466,6 +497,7 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
> 
>   webrtc::VideoEncoderFactory* const encoder_factory_;
>   webrtc::VideoDecoderFactory* const decoder_factory_;
>+  webrtc::VideoBitrateAllocatorFactory* const bitrate_allocator_factory_;
>   std::vector<VideoCodecSettings> recv_codecs_;
>   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
>   // See reason for keeping track of the FlexFEC payload type separately in
>@@ -475,14 +507,19 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>   // TODO(deadbeef): Don't duplicate information between
>   // send_params/recv_params, rtp_extensions, options, etc.
>   VideoSendParameters send_params_;
>+  rtc::DiffServCodePoint preferred_dscp_;
>   VideoOptions default_send_options_;
>   VideoRecvParameters recv_params_;
>   int64_t last_stats_log_ms_;
>+  const bool discard_unknown_ssrc_packets_;
>   // This is a stream param that comes from the remote description, but wasn't
>   // signaled with any a=ssrc lines. It holds information that was signaled
>   // before the unsignaled receive stream is created when the first packet is
>   // received.
>   StreamParams unsignaled_stream_params_;
>+  // Per peer connection crypto options that last for the lifetime of the peer
>+  // connection.
>+  const webrtc::CryptoOptions crypto_options_;
> };
> 
> class EncoderStreamFactory
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine_unittest.cc
>index e87f63ff4a65f2a14546ab1af79a540a4d5ba4c0..ab04fa6af3e5a6651154905b40887029a9165e9d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine_unittest.cc
>@@ -14,10 +14,14 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/strings/match.h"
> #include "api/rtpparameters.h"
>+#include "api/test/mock_video_bitrate_allocator.h"
>+#include "api/test/mock_video_bitrate_allocator_factory.h"
> #include "api/test/mock_video_decoder_factory.h"
> #include "api/test/mock_video_encoder_factory.h"
> #include "api/units/time_delta.h"
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "api/video/video_bitrate_allocation.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
>@@ -101,7 +105,7 @@ bool HasRtxCodec(const std::vector<cricket::VideoCodec>& codecs,
>                  int payload_type) {
>   for (const cricket::VideoCodec& codec : codecs) {
>     int associated_payload_type;
>-    if (cricket::CodecNamesEq(codec.name.c_str(), "rtx") &&
>+    if (absl::EqualsIgnoreCase(codec.name.c_str(), "rtx") &&
>         codec.GetParam(cricket::kCodecParamAssociatedPayloadType,
>                        &associated_payload_type) &&
>         associated_payload_type == payload_type) {
>@@ -213,7 +217,8 @@ class WebRtcVideoEngineTest : public ::testing::Test {
>         engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>(
>                     encoder_factory_),
>                 std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>(
>-                    decoder_factory_)) {
>+                    decoder_factory_),
>+                webrtc::CreateBuiltinVideoBitrateAllocatorFactory()) {
>     // Ensure fake clock doesn't return 0, which will cause some initializations
>     // fail inside RTP senders.
>     fake_clock_.AdvanceTimeMicros(1);
>@@ -318,6 +323,25 @@ TEST_F(WebRtcVideoEngineTest, SupportsVideoRotationHeaderExtension) {
>   FAIL() << "Video Rotation extension not in header-extension list.";
> }
> 
>+class WebRtcVideoEngineTestWithGenericDescriptor
>+    : public WebRtcVideoEngineTest {
>+ public:
>+  WebRtcVideoEngineTestWithGenericDescriptor()
>+      : WebRtcVideoEngineTest("WebRTC-GenericDescriptorAdvertised/Enabled/") {}
>+};
>+
>+TEST_F(WebRtcVideoEngineTestWithGenericDescriptor, AdvertiseGenericDescriptor) {
>+  RtpCapabilities capabilities = engine_.GetCapabilities();
>+  ASSERT_FALSE(capabilities.header_extensions.empty());
>+  for (const RtpExtension& extension : capabilities.header_extensions) {
>+    if (extension.uri == RtpExtension::kGenericFrameDescriptorUri) {
>+      EXPECT_EQ(RtpExtension::kGenericFrameDescriptorDefaultId, extension.id);
>+      return;
>+    }
>+  }
>+  FAIL() << "Generic descriptor extension not in header-extension list.";
>+}
>+
> TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeCapturer) {
>   // Allocate the source first to prevent early destruction before channel's
>   // dtor is called.
>@@ -433,8 +457,8 @@ TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionAfterCapturer) {
> TEST_F(WebRtcVideoEngineTest, SetSendFailsBeforeSettingCodecs) {
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(
>-      engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions()));
>+  std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
>+      call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions()));
> 
>   EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(123)));
> 
>@@ -447,8 +471,8 @@ TEST_F(WebRtcVideoEngineTest, SetSendFailsBeforeSettingCodecs) {
> TEST_F(WebRtcVideoEngineTest, GetStatsWithoutSendCodecsSetDoesNotCrash) {
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(
>-      engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions()));
>+  std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
>+      call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions()));
>   EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(123)));
>   VideoMediaInfo info;
>   channel->GetStats(&info);
>@@ -530,7 +554,7 @@ TEST_F(WebRtcVideoEngineTest, RtxCodecAddedForH264Codec) {
>       codecs, FindMatchingCodec(codecs, cricket::VideoCodec(h264_high))->id));
> }
> 
>-#if !defined(RTC_DISABLE_VP9)
>+#if defined(RTC_ENABLE_VP9)
> TEST_F(WebRtcVideoEngineTest, CanConstructDecoderForVp9EncoderFactory) {
>   encoder_factory_->AddSupportedVideoCodecType("VP9");
> 
>@@ -540,7 +564,7 @@ TEST_F(WebRtcVideoEngineTest, CanConstructDecoderForVp9EncoderFactory) {
>   EXPECT_TRUE(
>       channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
> }
>-#endif  // !defined(RTC_DISABLE_VP9)
>+#endif  // defined(RTC_ENABLE_VP9)
> 
> TEST_F(WebRtcVideoEngineTest, PropagatesInputFrameTimestamp) {
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
>@@ -631,11 +655,11 @@ size_t WebRtcVideoEngineTest::GetEngineCodecIndex(
>   const std::vector<cricket::VideoCodec> codecs = engine_.codecs();
>   for (size_t i = 0; i < codecs.size(); ++i) {
>     const cricket::VideoCodec engine_codec = codecs[i];
>-    if (!CodecNamesEq(name, engine_codec.name))
>+    if (!absl::EqualsIgnoreCase(name, engine_codec.name))
>       continue;
>     // The tests only use H264 Constrained Baseline. Make sure we don't return
>     // an internal H264 codec from the engine with a different H264 profile.
>-    if (CodecNamesEq(name.c_str(), kH264CodecName)) {
>+    if (absl::EqualsIgnoreCase(name.c_str(), kH264CodecName)) {
>       const absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
>           webrtc::H264::ParseSdpProfileLevelId(engine_codec.params);
>       if (profile_level_id->profile !=
>@@ -657,8 +681,8 @@ cricket::VideoCodec WebRtcVideoEngineTest::GetEngineCodec(
> 
> VideoMediaChannel*
> WebRtcVideoEngineTest::SetSendParamsWithAllSupportedCodecs() {
>-  VideoMediaChannel* channel =
>-      engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions());
>+  VideoMediaChannel* channel = engine_.CreateMediaChannel(
>+      call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions());
>   cricket::VideoSendParameters parameters;
>   // We need to look up the codec in the engine to get the correct payload type.
>   for (const webrtc::SdpVideoFormat& format :
>@@ -677,8 +701,8 @@ WebRtcVideoEngineTest::SetSendParamsWithAllSupportedCodecs() {
> 
> VideoMediaChannel* WebRtcVideoEngineTest::SetRecvParamsWithSupportedCodecs(
>     const std::vector<VideoCodec>& codecs) {
>-  VideoMediaChannel* channel =
>-      engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions());
>+  VideoMediaChannel* channel = engine_.CreateMediaChannel(
>+      call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions());
>   cricket::VideoRecvParameters parameters;
>   parameters.codecs = codecs;
>   EXPECT_TRUE(channel->SetRecvParameters(parameters));
>@@ -732,8 +756,8 @@ TEST_F(WebRtcVideoEngineTest, ChannelWithH264CanChangeToVp8) {
>   EXPECT_EQ(cricket::CS_RUNNING,
>             capturer.Start(capturer.GetSupportedFormats()->front()));
> 
>-  std::unique_ptr<VideoMediaChannel> channel(
>-      engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions()));
>+  std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
>+      call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions()));
>   cricket::VideoSendParameters parameters;
>   parameters.codecs.push_back(GetEngineCodec("H264"));
>   EXPECT_TRUE(channel->SetSendParameters(parameters));
>@@ -761,8 +785,8 @@ TEST_F(WebRtcVideoEngineTest,
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
>   encoder_factory_->AddSupportedVideoCodecType("H264");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(
>-      engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions()));
>+  std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
>+      call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions()));
>   cricket::VideoSendParameters parameters;
>   parameters.codecs.push_back(GetEngineCodec("VP8"));
>   EXPECT_TRUE(channel->SetSendParameters(parameters));
>@@ -796,8 +820,8 @@ TEST_F(WebRtcVideoEngineTest,
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
>   encoder_factory_->AddSupportedVideoCodecType("H264");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(
>-      engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions()));
>+  std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
>+      call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions()));
>   cricket::VideoSendParameters parameters;
>   parameters.codecs.push_back(GetEngineCodec("H264"));
>   EXPECT_TRUE(channel->SetSendParameters(parameters));
>@@ -828,8 +852,8 @@ TEST_F(WebRtcVideoEngineTest, SimulcastEnabledForH264BehindFieldTrial) {
>       "WebRTC-H264Simulcast/Enabled/");
>   encoder_factory_->AddSupportedVideoCodecType("H264");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(
>-      engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions()));
>+  std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel(
>+      call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions()));
>   cricket::VideoSendParameters parameters;
>   parameters.codecs.push_back(GetEngineCodec("H264"));
>   EXPECT_TRUE(channel->SetSendParameters(parameters));
>@@ -989,8 +1013,10 @@ TEST_F(WebRtcVideoEngineTest, GetSourcesWithNonExistingSsrc) {
> TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullFactories) {
>   std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory;
>   std::unique_ptr<webrtc::VideoDecoderFactory> decoder_factory;
>+  std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> rate_allocator_factory;
>   WebRtcVideoEngine engine(std::move(encoder_factory),
>-                           std::move(decoder_factory));
>+                           std::move(decoder_factory),
>+                           std::move(rate_allocator_factory));
>   EXPECT_EQ(0u, engine.codecs().size());
> }
> 
>@@ -1000,13 +1026,18 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, EmptyFactories) {
>       new webrtc::MockVideoEncoderFactory();
>   webrtc::MockVideoDecoderFactory* decoder_factory =
>       new webrtc::MockVideoDecoderFactory();
>+  webrtc::MockVideoBitrateAllocatorFactory* rate_allocator_factory =
>+      new webrtc::MockVideoBitrateAllocatorFactory();
>   WebRtcVideoEngine engine(
>       (std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
>-      (std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)));
>+      (std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)),
>+      (std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>(
>+          rate_allocator_factory)));
>   EXPECT_CALL(*encoder_factory, GetSupportedFormats());
>   EXPECT_EQ(0u, engine.codecs().size());
>   EXPECT_CALL(*encoder_factory, Die());
>   EXPECT_CALL(*decoder_factory, Die());
>+  EXPECT_CALL(*rate_allocator_factory, Die());
> }
> 
> // Test full behavior in the video engine when video codec factories of the new
>@@ -1019,9 +1050,17 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
>       new webrtc::MockVideoEncoderFactory();
>   webrtc::MockVideoDecoderFactory* decoder_factory =
>       new webrtc::MockVideoDecoderFactory();
>+  webrtc::MockVideoBitrateAllocatorFactory* rate_allocator_factory =
>+      new webrtc::MockVideoBitrateAllocatorFactory();
>+  EXPECT_CALL(*rate_allocator_factory,
>+              CreateVideoBitrateAllocatorProxy(Field(
>+                  &webrtc::VideoCodec::codecType, webrtc::kVideoCodecVP8)))
>+      .WillOnce(testing::Return(new webrtc::MockVideoBitrateAllocator()));
>   WebRtcVideoEngine engine(
>       (std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
>-      (std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)));
>+      (std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)),
>+      (std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>(
>+          rate_allocator_factory)));
>   const webrtc::SdpVideoFormat vp8_format("VP8");
>   const std::vector<webrtc::SdpVideoFormat> supported_formats = {vp8_format};
>   EXPECT_CALL(*encoder_factory, GetSupportedFormats())
>@@ -1066,7 +1105,7 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
>   EXPECT_CALL(*encoder_factory, QueryVideoEncoder(format))
>       .WillRepeatedly(testing::Return(codec_info));
>   FakeWebRtcVideoEncoder* const encoder = new FakeWebRtcVideoEncoder(nullptr);
>-  rtc::Event encoder_created(false, false);
>+  rtc::Event encoder_created;
>   EXPECT_CALL(*encoder_factory, CreateVideoEncoderProxy(format))
>       .WillOnce(
>           ::testing::DoAll(::testing::InvokeWithoutArgs(
>@@ -1085,8 +1124,8 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
> 
>   // Create send channel.
>   const int send_ssrc = 123;
>-  std::unique_ptr<VideoMediaChannel> send_channel(
>-      engine.CreateChannel(call.get(), GetMediaConfig(), VideoOptions()));
>+  std::unique_ptr<VideoMediaChannel> send_channel(engine.CreateMediaChannel(
>+      call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions()));
>   cricket::VideoSendParameters send_parameters;
>   send_parameters.codecs.push_back(engine_codecs.at(0));
>   EXPECT_TRUE(send_channel->SetSendParameters(send_parameters));
>@@ -1106,8 +1145,8 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
> 
>   // Create recv channel.
>   const int recv_ssrc = 321;
>-  std::unique_ptr<VideoMediaChannel> recv_channel(
>-      engine.CreateChannel(call.get(), GetMediaConfig(), VideoOptions()));
>+  std::unique_ptr<VideoMediaChannel> recv_channel(engine.CreateMediaChannel(
>+      call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions()));
>   cricket::VideoRecvParameters recv_parameters;
>   recv_parameters.codecs.push_back(engine_codecs.at(0));
>   EXPECT_TRUE(recv_channel->SetRecvParameters(recv_parameters));
>@@ -1117,6 +1156,7 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
>   // Remove streams previously added to free the encoder and decoder instance.
>   EXPECT_CALL(*encoder_factory, Die());
>   EXPECT_CALL(*decoder_factory, Die());
>+  EXPECT_CALL(*rate_allocator_factory, Die());
>   EXPECT_TRUE(send_channel->RemoveSendStream(send_ssrc));
>   EXPECT_TRUE(recv_channel->RemoveRecvStream(recv_ssrc));
> }
>@@ -1125,12 +1165,16 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
> TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullDecoder) {
>   // |engine| take ownership of the factories.
>   webrtc::MockVideoEncoderFactory* encoder_factory =
>-      new testing::StrictMock<webrtc::MockVideoEncoderFactory>();
>+      new webrtc::MockVideoEncoderFactory();
>   webrtc::MockVideoDecoderFactory* decoder_factory =
>-      new testing::StrictMock<webrtc::MockVideoDecoderFactory>();
>+      new webrtc::MockVideoDecoderFactory();
>+  webrtc::MockVideoBitrateAllocatorFactory* rate_allocator_factory =
>+      new webrtc::MockVideoBitrateAllocatorFactory();
>   WebRtcVideoEngine engine(
>       (std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
>-      (std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)));
>+      (std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)),
>+      (std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>(
>+          rate_allocator_factory)));
>   const webrtc::SdpVideoFormat vp8_format("VP8");
>   const std::vector<webrtc::SdpVideoFormat> supported_formats = {vp8_format};
>   EXPECT_CALL(*encoder_factory, GetSupportedFormats())
>@@ -1147,8 +1191,8 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullDecoder) {
> 
>   // Create recv channel.
>   const int recv_ssrc = 321;
>-  std::unique_ptr<VideoMediaChannel> recv_channel(
>-      engine.CreateChannel(call.get(), GetMediaConfig(), VideoOptions()));
>+  std::unique_ptr<VideoMediaChannel> recv_channel(engine.CreateMediaChannel(
>+      call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions()));
>   cricket::VideoRecvParameters recv_parameters;
>   recv_parameters.codecs.push_back(engine.codecs().front());
>   EXPECT_TRUE(recv_channel->SetRecvParameters(recv_parameters));
>@@ -1223,7 +1267,8 @@ class WebRtcVideoChannelBaseTest : public testing::Test {
>  protected:
>   WebRtcVideoChannelBaseTest()
>       : engine_(webrtc::CreateBuiltinVideoEncoderFactory(),
>-                webrtc::CreateBuiltinVideoDecoderFactory()) {}
>+                webrtc::CreateBuiltinVideoDecoderFactory(),
>+                webrtc::CreateBuiltinVideoBitrateAllocatorFactory()) {}
> 
>   virtual void SetUp() {
>     // One testcase calls SetUp in a loop, only create call_ once.
>@@ -1236,12 +1281,14 @@ class WebRtcVideoChannelBaseTest : public testing::Test {
>     // needs to be disabled, otherwise, tests which check the size of received
>     // frames become flaky.
>     media_config.video.enable_cpu_adaptation = false;
>-    channel_.reset(engine_.CreateChannel(call_.get(), media_config,
>-                                         cricket::VideoOptions()));
>+    channel_.reset(
>+        static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
>+            call_.get(), media_config, cricket::VideoOptions(),
>+            webrtc::CryptoOptions())));
>     channel_->OnReadyToSend(true);
>     EXPECT_TRUE(channel_.get() != NULL);
>     network_interface_.SetDestination(channel_.get());
>-    channel_->SetInterface(&network_interface_);
>+    channel_->SetInterface(&network_interface_, /*media_transport=*/nullptr);
>     cricket::VideoRecvParameters parameters;
>     parameters.codecs = engine_.codecs();
>     channel_->SetRecvParameters(parameters);
>@@ -1411,7 +1458,7 @@ class WebRtcVideoChannelBaseTest : public testing::Test {
> 
>   cricket::VideoCodec GetEngineCodec(const std::string& name) {
>     for (const cricket::VideoCodec& engine_codec : engine_.codecs()) {
>-      if (CodecNamesEq(name, engine_codec.name))
>+      if (absl::EqualsIgnoreCase(name, engine_codec.name))
>         return engine_codec;
>     }
>     // This point should never be reached.
>@@ -1465,60 +1512,7 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSendSetsTransportBufferSizes) {
>   EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>   EXPECT_TRUE(SetSend(true));
>   EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
>-  EXPECT_EQ(64 * 1024, network_interface_.recvbuf_size());
>-}
>-
>-// Test that we properly set the send and recv buffer sizes when overriding
>-// via field trials.
>-TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSize) {
>-  // Set field trial to override the default recv buffer size, and then re-run
>-  // setup where the interface is created and configured.
>-  const int kCustomRecvBufferSize = 123456;
>-  webrtc::test::ScopedFieldTrials field_trial(
>-      "WebRTC-IncreasedReceivebuffers/123456/");
>-  SetUp();
>-
>-  EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>-  EXPECT_TRUE(SetSend(true));
>-  EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
>-  EXPECT_EQ(kCustomRecvBufferSize, network_interface_.recvbuf_size());
>-}
>-
>-// Test that we properly set the send and recv buffer sizes when overriding
>-// via field trials with suffix.
>-TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSizeWithSuffix) {
>-  // Set field trial to override the default recv buffer size, and then re-run
>-  // setup where the interface is created and configured.
>-  const int kCustomRecvBufferSize = 123456;
>-  webrtc::test::ScopedFieldTrials field_trial(
>-      "WebRTC-IncreasedReceivebuffers/123456_Dogfood/");
>-  SetUp();
>-
>-  EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>-  EXPECT_TRUE(SetSend(true));
>-  EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
>-  EXPECT_EQ(kCustomRecvBufferSize, network_interface_.recvbuf_size());
>-}
>-
>-// Test that we properly set the send and recv buffer sizes when overriding
>-// via field trials that don't make any sense.
>-TEST_F(WebRtcVideoChannelBaseTest, InvalidRecvBufferSize) {
>-  // Set bogus field trial values to override the default recv buffer size, and
>-  // then re-run setup where the interface is created and configured. The
>-  // default value should still be used.
>-
>-  for (std::string group : {" ", "NotANumber", "-1", "0"}) {
>-    std::string field_trial_string = "WebRTC-IncreasedReceivebuffers/";
>-    field_trial_string += group;
>-    field_trial_string += "/";
>-    webrtc::test::ScopedFieldTrials field_trial(field_trial_string);
>-    SetUp();
>-
>-    EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
>-    EXPECT_EQ(64 * 1024, network_interface_.recvbuf_size());
>-  }
>+  EXPECT_EQ(256 * 1024, network_interface_.recvbuf_size());
> }
> 
> // Test that stats work properly for a 1-1 call.
>@@ -1757,7 +1751,7 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSink) {
>   EXPECT_TRUE(SetDefaultCodec());
>   EXPECT_TRUE(SetSend(true));
>   EXPECT_EQ(0, renderer_.num_rendered_frames());
>-  channel_->OnPacketReceived(&packet1, rtc::PacketTime());
>+  channel_->OnPacketReceived(&packet1, /* packet_time_us */ -1);
>   EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>   EXPECT_TRUE(SendFrame());
>   EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>@@ -2046,8 +2040,9 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
> #endif
> 
>     fake_call_.reset(new FakeCall());
>-    channel_.reset(engine_.CreateChannel(fake_call_.get(), GetMediaConfig(),
>-                                         VideoOptions()));
>+    channel_.reset(engine_.CreateMediaChannel(fake_call_.get(),
>+                                              GetMediaConfig(), VideoOptions(),
>+                                              webrtc::CryptoOptions()));
>     channel_->OnReadyToSend(true);
>     last_ssrc_ = 123;
>     send_parameters_.codecs = engine_.codecs();
>@@ -2121,6 +2116,29 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
>                     &BitrateConstraints::max_bitrate_bps, max_bitrate_bps)));
>   }
> 
>+  void TestExtmapAllowMixedCaller(bool extmap_allow_mixed) {
>+    // For a caller, the answer will be applied in set remote description
>+    // where SetSendParameters() is called.
>+    EXPECT_TRUE(
>+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>+    send_parameters_.extmap_allow_mixed = extmap_allow_mixed;
>+    EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
>+    const webrtc::VideoSendStream::Config& config =
>+        fake_call_->GetVideoSendStreams()[0]->GetConfig();
>+    EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
>+  }
>+
>+  void TestExtmapAllowMixedCallee(bool extmap_allow_mixed) {
>+    // For a callee, the answer will be applied in set local description
>+    // where SetExtmapAllowMixed() and AddSendStream() are called.
>+    channel_->SetExtmapAllowMixed(extmap_allow_mixed);
>+    EXPECT_TRUE(
>+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>+    const webrtc::VideoSendStream::Config& config =
>+        fake_call_->GetVideoSendStreams()[0]->GetConfig();
>+    EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
>+  }
>+
>   void TestSetSendRtpHeaderExtensions(const std::string& ext_uri) {
>     // Enable extension.
>     const int id = 1;
>@@ -2377,6 +2395,20 @@ TEST_F(WebRtcVideoChannelTest, RecvStreamNoRtx) {
>   ASSERT_EQ(0U, recv_stream->GetConfig().rtp.rtx_ssrc);
> }
> 
>+// Test propagation of extmap allow mixed setting.
>+TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedAsCaller) {
>+  TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/true);
>+}
>+TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedDisabledAsCaller) {
>+  TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/false);
>+}
>+TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedAsCallee) {
>+  TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/true);
>+}
>+TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedDisabledAsCallee) {
>+  TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/false);
>+}
>+
> TEST_F(WebRtcVideoChannelTest, NoHeaderExtesionsByDefault) {
>   FakeVideoSendStream* send_stream =
>       AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcs1[0]));
>@@ -2853,8 +2885,8 @@ TEST_F(WebRtcVideoChannelTest, SetMediaConfigSuspendBelowMinBitrate) {
>   MediaConfig media_config = GetMediaConfig();
>   media_config.video.suspend_below_min_bitrate = true;
> 
>-  channel_.reset(
>-      engine_.CreateChannel(fake_call_.get(), media_config, VideoOptions()));
>+  channel_.reset(engine_.CreateMediaChannel(
>+      fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions()));
>   channel_->OnReadyToSend(true);
> 
>   channel_->SetSendParameters(send_parameters_);
>@@ -2863,8 +2895,8 @@ TEST_F(WebRtcVideoChannelTest, SetMediaConfigSuspendBelowMinBitrate) {
>   EXPECT_TRUE(stream->GetConfig().suspend_below_min_bitrate);
> 
>   media_config.video.suspend_below_min_bitrate = false;
>-  channel_.reset(
>-      engine_.CreateChannel(fake_call_.get(), media_config, VideoOptions()));
>+  channel_.reset(engine_.CreateMediaChannel(
>+      fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions()));
>   channel_->OnReadyToSend(true);
> 
>   channel_->SetSendParameters(send_parameters_);
>@@ -3204,8 +3236,8 @@ TEST_F(WebRtcVideoChannelTest, AdaptsOnOveruseAndChangeResolution) {
>   parameters.codecs.push_back(codec);
> 
>   MediaConfig media_config = GetMediaConfig();
>-  channel_.reset(
>-      engine_.CreateChannel(fake_call_.get(), media_config, VideoOptions()));
>+  channel_.reset(engine_.CreateMediaChannel(
>+      fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions()));
>   channel_->OnReadyToSend(true);
>   ASSERT_TRUE(channel_->SetSendParameters(parameters));
> 
>@@ -3285,8 +3317,8 @@ TEST_F(WebRtcVideoChannelTest, PreviousAdaptationDoesNotApplyToScreenshare) {
> 
>   MediaConfig media_config = GetMediaConfig();
>   media_config.video.enable_cpu_adaptation = true;
>-  channel_.reset(
>-      engine_.CreateChannel(fake_call_.get(), media_config, VideoOptions()));
>+  channel_.reset(engine_.CreateMediaChannel(
>+      fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions()));
>   channel_->OnReadyToSend(true);
>   ASSERT_TRUE(channel_->SetSendParameters(parameters));
> 
>@@ -3359,8 +3391,8 @@ void WebRtcVideoChannelTest::TestDegradationPreference(
> 
>   MediaConfig media_config = GetMediaConfig();
>   media_config.video.enable_cpu_adaptation = true;
>-  channel_.reset(
>-      engine_.CreateChannel(fake_call_.get(), media_config, VideoOptions()));
>+  channel_.reset(engine_.CreateMediaChannel(
>+      fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions()));
>   channel_->OnReadyToSend(true);
> 
>   EXPECT_TRUE(channel_->SetSendParameters(parameters));
>@@ -3394,8 +3426,8 @@ void WebRtcVideoChannelTest::TestCpuAdaptation(bool enable_overuse,
>   if (enable_overuse) {
>     media_config.video.enable_cpu_adaptation = true;
>   }
>-  channel_.reset(
>-      engine_.CreateChannel(fake_call_.get(), media_config, VideoOptions()));
>+  channel_.reset(engine_.CreateMediaChannel(
>+      fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions()));
>   channel_->OnReadyToSend(true);
> 
>   EXPECT_TRUE(channel_->SetSendParameters(parameters));
>@@ -4575,31 +4607,55 @@ TEST_F(WebRtcVideoChannelTest, TestSetDscpOptions) {
>       new cricket::FakeNetworkInterface);
>   MediaConfig config;
>   std::unique_ptr<cricket::WebRtcVideoChannel> channel;
>+  webrtc::RtpParameters parameters;
> 
>-  channel.reset(static_cast<cricket::WebRtcVideoChannel*>(
>-      engine_.CreateChannel(call_.get(), config, VideoOptions())));
>-  channel->SetInterface(network_interface.get());
>+  channel.reset(
>+      static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
>+          call_.get(), config, VideoOptions(), webrtc::CryptoOptions())));
>+  channel->SetInterface(network_interface.get(), /*media_transport=*/nullptr);
>   // Default value when DSCP is disabled should be DSCP_DEFAULT.
>   EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
> 
>+  // Default value when DSCP is enabled is also DSCP_DEFAULT, until it is set
>+  // through rtp parameters.
>   config.enable_dscp = true;
>-  channel.reset(static_cast<cricket::WebRtcVideoChannel*>(
>-      engine_.CreateChannel(call_.get(), config, VideoOptions())));
>-  channel->SetInterface(network_interface.get());
>+  channel.reset(
>+      static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
>+          call_.get(), config, VideoOptions(), webrtc::CryptoOptions())));
>+  channel->SetInterface(network_interface.get(), /*media_transport=*/nullptr);
>+  EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
>+
>+  // Create a send stream to configure
>+  EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
>+  parameters = channel->GetRtpSendParameters(kSsrc);
>+  ASSERT_FALSE(parameters.encodings.empty());
>+
>+  // Various priorities map to various dscp values.
>+  parameters.encodings[0].network_priority = 4.0;
>+  ASSERT_TRUE(channel->SetRtpSendParameters(kSsrc, parameters).ok());
>   EXPECT_EQ(rtc::DSCP_AF41, network_interface->dscp());
>+  parameters.encodings[0].network_priority = 0.5;
>+  ASSERT_TRUE(channel->SetRtpSendParameters(kSsrc, parameters).ok());
>+  EXPECT_EQ(rtc::DSCP_CS1, network_interface->dscp());
>+
>+  // A bad priority does not change the dscp value.
>+  parameters.encodings[0].network_priority = 0.0;
>+  ASSERT_FALSE(channel->SetRtpSendParameters(kSsrc, parameters).ok());
>+  EXPECT_EQ(rtc::DSCP_CS1, network_interface->dscp());
> 
>   // Packets should also self-identify their dscp in PacketOptions.
>   const uint8_t kData[10] = {0};
>   EXPECT_TRUE(static_cast<webrtc::Transport*>(channel.get())
>                   ->SendRtcp(kData, sizeof(kData)));
>-  EXPECT_EQ(rtc::DSCP_AF41, network_interface->options().dscp);
>+  EXPECT_EQ(rtc::DSCP_CS1, network_interface->options().dscp);
> 
>   // Verify that setting the option to false resets the
>   // DiffServCodePoint.
>   config.enable_dscp = false;
>-  channel.reset(static_cast<cricket::WebRtcVideoChannel*>(
>-      engine_.CreateChannel(call_.get(), config, VideoOptions())));
>-  channel->SetInterface(network_interface.get());
>+  channel.reset(
>+      static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel(
>+          call_.get(), config, VideoOptions(), webrtc::CryptoOptions())));
>+  channel->SetInterface(network_interface.get(), /*media_transport=*/nullptr);
>   EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
> }
> 
>@@ -4957,8 +5013,7 @@ TEST_F(WebRtcVideoChannelTest, DefaultReceiveStreamReconfiguresToUseRtx) {
>   memset(data, 0, sizeof(data));
>   rtc::SetBE32(&data[8], ssrcs[0]);
>   rtc::CopyOnWriteBuffer packet(data, kDataLength);
>-  rtc::PacketTime packet_time;
>-  channel_->OnPacketReceived(&packet, packet_time);
>+  channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
> 
>   ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
>       << "No default receive stream created.";
>@@ -5116,8 +5171,7 @@ TEST_F(WebRtcVideoChannelTest, RecvUnsignaledSsrcWithSignaledStreamId) {
>   memset(data, 0, sizeof(data));
>   rtc::SetBE32(&data[8], kIncomingUnsignalledSsrc);
>   rtc::CopyOnWriteBuffer packet(data, kDataLength);
>-  rtc::PacketTime packet_time;
>-  channel_->OnPacketReceived(&packet, packet_time);
>+  channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
> 
>   // The stream should now be created with the appropriate sync label.
>   EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
>@@ -5130,7 +5184,7 @@ TEST_F(WebRtcVideoChannelTest, RecvUnsignaledSsrcWithSignaledStreamId) {
>   ASSERT_TRUE(channel_->RemoveRecvStream(kIncomingUnsignalledSsrc));
>   EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
> 
>-  channel_->OnPacketReceived(&packet, packet_time);
>+  channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
>   EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
>   EXPECT_TRUE(
>       fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group.empty());
>@@ -5156,8 +5210,7 @@ void WebRtcVideoChannelTest::TestReceiveUnsignaledSsrcPacket(
>   rtc::Set8(data, 1, payload_type);
>   rtc::SetBE32(&data[8], kIncomingUnsignalledSsrc);
>   rtc::CopyOnWriteBuffer packet(data, kDataLength);
>-  rtc::PacketTime packet_time;
>-  channel_->OnPacketReceived(&packet, packet_time);
>+  channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
> 
>   if (expect_created_receive_stream) {
>     EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size())
>@@ -5170,6 +5223,18 @@ void WebRtcVideoChannelTest::TestReceiveUnsignaledSsrcPacket(
>   }
> }
> 
>+class WebRtcVideoChannelDiscardUnknownSsrcTest : public WebRtcVideoChannelTest {
>+ public:
>+  WebRtcVideoChannelDiscardUnknownSsrcTest()
>+      : WebRtcVideoChannelTest(
>+            "WebRTC-Video-DiscardPacketsWithUnknownSsrc/Enabled/") {}
>+};
>+
>+TEST_F(WebRtcVideoChannelDiscardUnknownSsrcTest, NoUnsignalledStreamCreated) {
>+  TestReceiveUnsignaledSsrcPacket(GetEngineCodec("VP8").id,
>+                                  false /* expect_created_receive_stream */);
>+}
>+
> TEST_F(WebRtcVideoChannelTest, Vp8PacketCreatesUnsignalledStream) {
>   TestReceiveUnsignaledSsrcPacket(GetEngineCodec("VP8").id,
>                                   true /* expect_created_receive_stream */);
>@@ -5232,8 +5297,7 @@ TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) {
>   rtpHeader.ssrc = kIncomingUnsignalledSsrc + 1;
>   cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
>   rtc::CopyOnWriteBuffer packet(data, sizeof(data));
>-  rtc::PacketTime packet_time;
>-  channel_->OnPacketReceived(&packet, packet_time);
>+  channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
>   // VP8 packet should create default receive stream.
>   ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
>   FakeVideoReceiveStream* recv_stream = fake_call_->GetVideoReceiveStreams()[0];
>@@ -5249,7 +5313,7 @@ TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) {
>   rtpHeader.ssrc = kIncomingUnsignalledSsrc + 2;
>   cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
>   rtc::CopyOnWriteBuffer packet2(data, sizeof(data));
>-  channel_->OnPacketReceived(&packet2, packet_time);
>+  channel_->OnPacketReceived(&packet2, /* packet_time_us */ -1);
>   // VP9 packet should replace the default receive SSRC.
>   ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
>   recv_stream = fake_call_->GetVideoReceiveStreams()[0];
>@@ -5266,7 +5330,7 @@ TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) {
>   rtpHeader.ssrc = kIncomingUnsignalledSsrc + 3;
>   cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
>   rtc::CopyOnWriteBuffer packet3(data, sizeof(data));
>-  channel_->OnPacketReceived(&packet3, packet_time);
>+  channel_->OnPacketReceived(&packet3, /* packet_time_us */ -1);
>   // H264 packet should replace the default receive SSRC.
>   ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
>   recv_stream = fake_call_->GetVideoReceiveStreams()[0];
>@@ -5300,8 +5364,7 @@ TEST_F(WebRtcVideoChannelTest,
>   rtp_header.ssrc = kSsrcs3[0];
>   cricket::SetRtpHeader(data, sizeof(data), rtp_header);
>   rtc::CopyOnWriteBuffer packet(data, sizeof(data));
>-  rtc::PacketTime packet_time;
>-  channel_->OnPacketReceived(&packet, packet_time);
>+  channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
>   // Default receive stream should be created.
>   ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
>   FakeVideoReceiveStream* recv_stream0 =
>@@ -5319,7 +5382,7 @@ TEST_F(WebRtcVideoChannelTest,
>   rtp_header.ssrc = kSsrcs3[1];
>   cricket::SetRtpHeader(data, sizeof(data), rtp_header);
>   packet.SetData(data, sizeof(data));
>-  channel_->OnPacketReceived(&packet, packet_time);
>+  channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
>   // New default receive stream should be created, but old stream should remain.
>   ASSERT_EQ(2u, fake_call_->GetVideoReceiveStreams().size());
>   EXPECT_EQ(recv_stream0, fake_call_->GetVideoReceiveStreams()[0]);
>@@ -5646,6 +5709,28 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) {
>   EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, nullptr, nullptr));
> }
> 
>+// RTCRtpEncodingParameters.network_priority must be one of a few values
>+// derived from the default priority, corresponding to very-low, low, medium,
>+// or high.
>+TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersInvalidNetworkPriority) {
>+  AddSendStream();
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(1UL, parameters.encodings.size());
>+  EXPECT_EQ(webrtc::kDefaultBitratePriority,
>+            parameters.encodings[0].network_priority);
>+
>+  double good_values[] = {0.5, 1.0, 2.0, 4.0};
>+  double bad_values[] = {-1.0, 0.0, 0.49, 0.51, 1.1, 3.99, 4.1, 5.0};
>+  for (auto it : good_values) {
>+    parameters.encodings[0].network_priority = it;
>+    EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+  }
>+  for (auto it : bad_values) {
>+    parameters.encodings[0].network_priority = it;
>+    EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+  }
>+}
>+
> TEST_F(WebRtcVideoChannelTest, GetAndSetRtpSendParametersMaxFramerate) {
>   const size_t kNumSimulcastStreams = 3;
>   SetUpSimulcast(true, false);
>@@ -6634,8 +6719,7 @@ TEST_F(WebRtcVideoChannelTest, GetRtpReceiveParametersWithUnsignaledSsrc) {
>   rtpHeader.ssrc = kIncomingUnsignalledSsrc;
>   cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
>   rtc::CopyOnWriteBuffer packet(data, sizeof(data));
>-  rtc::PacketTime packet_time;
>-  channel_->OnPacketReceived(&packet, packet_time);
>+  channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
> 
>   // The |ssrc| member should still be unset.
>   rtp_parameters = channel_->GetRtpReceiveParameters(0);
>@@ -6699,16 +6783,21 @@ class WebRtcVideoChannelSimulcastTest : public testing::Test {
>       : fake_call_(),
>         encoder_factory_(new cricket::FakeWebRtcVideoEncoderFactory),
>         decoder_factory_(new cricket::FakeWebRtcVideoDecoderFactory),
>+        mock_rate_allocator_factory_(
>+            new webrtc::MockVideoBitrateAllocatorFactory),
>         engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>(
>                     encoder_factory_),
>                 std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>(
>-                    decoder_factory_)),
>+                    decoder_factory_),
>+                std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>(
>+                    mock_rate_allocator_factory_)),
>         last_ssrc_(0) {}
> 
>   void SetUp() override {
>     encoder_factory_->AddSupportedVideoCodecType("VP8");
>-    channel_.reset(
>-        engine_.CreateChannel(&fake_call_, GetMediaConfig(), VideoOptions()));
>+    channel_.reset(engine_.CreateMediaChannel(&fake_call_, GetMediaConfig(),
>+                                              VideoOptions(),
>+                                              webrtc::CryptoOptions()));
>     channel_->OnReadyToSend(true);
>     last_ssrc_ = 123;
>   }
>@@ -6858,6 +6947,7 @@ class WebRtcVideoChannelSimulcastTest : public testing::Test {
>   FakeCall fake_call_;
>   cricket::FakeWebRtcVideoEncoderFactory* encoder_factory_;
>   cricket::FakeWebRtcVideoDecoderFactory* decoder_factory_;
>+  webrtc::MockVideoBitrateAllocatorFactory* mock_rate_allocator_factory_;
>   WebRtcVideoEngine engine_;
>   std::unique_ptr<VideoMediaChannel> channel_;
>   uint32_t last_ssrc_;
>@@ -6905,16 +6995,20 @@ TEST_F(WebRtcVideoChannelSimulcastTest,
>                           false);
> }
> 
>-// The fake clock needs to be initialize before the call.
>-// So defer creating call in base class.
>-class WebRtcVideoChannelTestWithClock : public WebRtcVideoChannelBaseTest {
>+class WebRtcVideoFakeClock {
>  public:
>-  WebRtcVideoChannelTestWithClock() {
>+  WebRtcVideoFakeClock() {
>     fake_clock_.AdvanceTime(webrtc::TimeDelta::ms(1));  // avoid time=0
>   }
>   rtc::ScopedFakeClock fake_clock_;
> };
> 
>+// The fake clock needs to be initialized before the call, and not
>+// destroyed until after all threads spawned by the test have been stopped.
>+// This mixin ensures that.
>+class WebRtcVideoChannelTestWithClock : public WebRtcVideoFakeClock,
>+                                        public WebRtcVideoChannelBaseTest {};
>+
> TEST_F(WebRtcVideoChannelTestWithClock, GetSources) {
>   uint8_t data1[] = {0x80, 0x00, 0x00, 0x00, 0x00, 0x00,
>                      0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.cc
>index 8ae7b991e805392dc9254aea84b1386186b73ff9..1660bd86e114aafbcec1647d6388b915aabf4da0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.cc
>@@ -19,8 +19,10 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/strings/match.h"
> #include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/call/audio_sink.h"
>+#include "api/media_transport_interface.h"
> #include "media/base/audiosource.h"
> #include "media/base/mediaconstants.h"
> #include "media/base/streamparams.h"
>@@ -38,10 +40,8 @@
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/race_checker.h"
>-#include "rtc_base/stringencode.h"
> #include "rtc_base/strings/audio_format_to_string.h"
> #include "rtc_base/strings/string_builder.h"
>-#include "rtc_base/stringutils.h"
> #include "rtc_base/third_party/base64/base64.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/field_trial.h"
>@@ -58,11 +58,6 @@ constexpr int kNackRtpHistoryMs = 5000;
> const int kOpusMinBitrateBps = 6000;
> const int kOpusBitrateFbBps = 32000;
> 
>-// Default audio dscp value.
>-// See http://tools.ietf.org/html/rfc2474 for details.
>-// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
>-const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
>-
> const int kMinTelephoneEventCode = 0;  // RFC4733 (Section 2.3.1)
> const int kMaxTelephoneEventCode = 255;
> 
>@@ -83,12 +78,12 @@ class ProxySink : public webrtc::AudioSinkInterface {
> 
> bool ValidateStreamParams(const StreamParams& sp) {
>   if (sp.ssrcs.empty()) {
>-    RTC_LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
>+    RTC_DLOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
>     return false;
>   }
>   if (sp.ssrcs.size() > 1) {
>-    RTC_LOG(LS_ERROR) << "Multiple SSRCs in stream parameters: "
>-                      << sp.ToString();
>+    RTC_DLOG(LS_ERROR) << "Multiple SSRCs in stream parameters: "
>+                       << sp.ToString();
>     return false;
>   }
>   return true;
>@@ -110,7 +105,7 @@ std::string ToString(const AudioCodec& codec) {
> }
> 
> bool IsCodec(const AudioCodec& codec, const char* ref_name) {
>-  return (_stricmp(codec.name.c_str(), ref_name) == 0);
>+  return absl::EqualsIgnoreCase(codec.name, ref_name);
> }
> 
> bool FindCodec(const std::vector<AudioCodec>& codecs,
>@@ -284,6 +279,7 @@ void WebRtcVoiceEngine::Init() {
>     options.stereo_swapping = false;
>     options.audio_jitter_buffer_max_packets = 50;
>     options.audio_jitter_buffer_fast_accelerate = false;
>+    options.audio_jitter_buffer_min_delay_ms = 0;
>     options.typing_detection = true;
>     options.experimental_agc = false;
>     options.extended_filter_aec = false;
>@@ -303,12 +299,14 @@ rtc::scoped_refptr<webrtc::AudioState> WebRtcVoiceEngine::GetAudioState()
>   return audio_state_;
> }
> 
>-VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
>+VoiceMediaChannel* WebRtcVoiceEngine::CreateMediaChannel(
>     webrtc::Call* call,
>     const MediaConfig& config,
>-    const AudioOptions& options) {
>+    const AudioOptions& options,
>+    const webrtc::CryptoOptions& crypto_options) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return new WebRtcVoiceMediaChannel(this, config, options, call);
>+  return new WebRtcVoiceMediaChannel(this, config, options, crypto_options,
>+                                     call);
> }
> 
> bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
>@@ -320,11 +318,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
>   // Set and adjust echo canceller options.
>   // kEcConference is AEC with high suppression.
>   webrtc::EcModes ec_mode = webrtc::kEcConference;
>-  if (options.aecm_generate_comfort_noise &&
>-      *options.aecm_generate_comfort_noise) {
>-    RTC_LOG(LS_WARNING)
>-        << "Ignoring deprecated mobile AEC setting: comfort noise";
>-  }
> 
> #if defined(WEBRTC_IOS)
>   if (options.ios_force_software_aec_HACK &&
>@@ -490,6 +483,12 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
>     audio_jitter_buffer_fast_accelerate_ =
>         *options.audio_jitter_buffer_fast_accelerate;
>   }
>+  if (options.audio_jitter_buffer_min_delay_ms) {
>+    RTC_LOG(LS_INFO) << "NetEq minimum delay is "
>+                     << *options.audio_jitter_buffer_min_delay_ms;
>+    audio_jitter_buffer_min_delay_ms_ =
>+        *options.audio_jitter_buffer_min_delay_ms;
>+  }
> 
>   if (options.typing_detection) {
>     RTC_LOG(LS_INFO) << "Typing detection is enabled? "
>@@ -713,16 +712,20 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>       const std::string track_id,
>       const absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>&
>           send_codec_spec,
>+      bool extmap_allow_mixed,
>       const std::vector<webrtc::RtpExtension>& extensions,
>       int max_send_bitrate_bps,
>+      int rtcp_report_interval_ms,
>       const absl::optional<std::string>& audio_network_adaptor_config,
>       webrtc::Call* call,
>       webrtc::Transport* send_transport,
>+      webrtc::MediaTransportInterface* media_transport,
>       const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
>       const absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
>-      rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor)
>+      rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor,
>+      const webrtc::CryptoOptions& crypto_options)
>       : call_(call),
>-        config_(send_transport),
>+        config_(send_transport, media_transport),
>         send_side_bwe_with_overhead_(
>             webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
>         max_send_bitrate_bps_(max_send_bitrate_bps),
>@@ -732,12 +735,17 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>     config_.rtp.ssrc = ssrc;
>     config_.rtp.mid = mid;
>     config_.rtp.c_name = c_name;
>+    config_.rtp.extmap_allow_mixed = extmap_allow_mixed;
>     config_.rtp.extensions = extensions;
>+    config_.has_dscp = rtp_parameters_.encodings[0].network_priority !=
>+                       webrtc::kDefaultBitratePriority;
>     config_.audio_network_adaptor_config = audio_network_adaptor_config;
>     config_.encoder_factory = encoder_factory;
>     config_.codec_pair_id = codec_pair_id;
>     config_.track_id = track_id;
>     config_.frame_encryptor = frame_encryptor;
>+    config_.crypto_options = crypto_options;
>+    config_.rtcp_report_interval_ms = rtcp_report_interval_ms;
>     rtp_parameters_.encodings[0].ssrc = ssrc;
>     rtp_parameters_.rtcp.cname = c_name;
>     rtp_parameters_.header_extensions = extensions;
>@@ -768,6 +776,11 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>     ReconfigureAudioSendStream();
>   }
> 
>+  void SetExtmapAllowMixed(bool extmap_allow_mixed) {
>+    config_.rtp.extmap_allow_mixed = extmap_allow_mixed;
>+    ReconfigureAudioSendStream();
>+  }
>+
>   void SetMid(const std::string& mid) {
>     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>     if (config_.rtp.mid == mid) {
>@@ -928,12 +941,16 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>     const absl::optional<int> old_rtp_max_bitrate =
>         rtp_parameters_.encodings[0].max_bitrate_bps;
>     double old_priority = rtp_parameters_.encodings[0].bitrate_priority;
>+    double old_dscp = rtp_parameters_.encodings[0].network_priority;
>     rtp_parameters_ = parameters;
>     config_.bitrate_priority = rtp_parameters_.encodings[0].bitrate_priority;
>+    config_.has_dscp = (rtp_parameters_.encodings[0].network_priority !=
>+                        webrtc::kDefaultBitratePriority);
> 
>     bool reconfigure_send_stream =
>         (rtp_parameters_.encodings[0].max_bitrate_bps != old_rtp_max_bitrate) ||
>-        (rtp_parameters_.encodings[0].bitrate_priority != old_priority);
>+        (rtp_parameters_.encodings[0].bitrate_priority != old_priority) ||
>+        (rtp_parameters_.encodings[0].network_priority != old_dscp);
>     if (rtp_parameters_.encodings[0].max_bitrate_bps != old_rtp_max_bitrate) {
>       // Update the bitrate range.
>       if (send_rate) {
>@@ -969,8 +986,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>     const bool is_opus =
>         config_.send_codec_spec &&
>-        !STR_CASE_CMP(config_.send_codec_spec->format.name.c_str(),
>-                      kOpusCodecName);
>+        absl::EqualsIgnoreCase(config_.send_codec_spec->format.name,
>+                               kOpusCodecName);
>     if (is_opus && webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
>       config_.min_bitrate_bps = kOpusMinBitrateBps;
> 
>@@ -1014,8 +1031,6 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>   void UpdateSendCodecSpec(
>       const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) {
>     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-    config_.rtp.nack.rtp_history_ms =
>-        send_codec_spec.nack_enabled ? kNackRtpHistoryMs : 0;
>     config_.send_codec_spec = send_codec_spec;
>     auto info =
>         config_.encoder_factory->QueryAudioEncoder(send_codec_spec.format);
>@@ -1077,12 +1092,15 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
>       const std::vector<webrtc::RtpExtension>& extensions,
>       webrtc::Call* call,
>       webrtc::Transport* rtcp_send_transport,
>+      webrtc::MediaTransportInterface* media_transport,
>       const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
>       const std::map<int, webrtc::SdpAudioFormat>& decoder_map,
>       absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
>       size_t jitter_buffer_max_packets,
>       bool jitter_buffer_fast_accelerate,
>-      rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor)
>+      int jitter_buffer_min_delay_ms,
>+      rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor,
>+      const webrtc::CryptoOptions& crypto_options)
>       : call_(call), config_() {
>     RTC_DCHECK(call);
>     config_.rtp.remote_ssrc = remote_ssrc;
>@@ -1091,8 +1109,10 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
>     config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
>     config_.rtp.extensions = extensions;
>     config_.rtcp_send_transport = rtcp_send_transport;
>+    config_.media_transport = media_transport;
>     config_.jitter_buffer_max_packets = jitter_buffer_max_packets;
>     config_.jitter_buffer_fast_accelerate = jitter_buffer_fast_accelerate;
>+    config_.jitter_buffer_min_delay_ms = jitter_buffer_min_delay_ms;
>     if (!stream_ids.empty()) {
>       config_.sync_group = stream_ids[0];
>     }
>@@ -1100,6 +1120,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
>     config_.decoder_map = decoder_map;
>     config_.codec_pair_id = codec_pair_id;
>     config_.frame_decryptor = frame_decryptor;
>+    config_.crypto_options = crypto_options;
>     RecreateAudioReceiveStream();
>   }
> 
>@@ -1237,11 +1258,17 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
> };
> 
>-WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
>-                                                 const MediaConfig& config,
>-                                                 const AudioOptions& options,
>-                                                 webrtc::Call* call)
>-    : VoiceMediaChannel(config), engine_(engine), call_(call) {
>+WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(
>+    WebRtcVoiceEngine* engine,
>+    const MediaConfig& config,
>+    const AudioOptions& options,
>+    const webrtc::CryptoOptions& crypto_options,
>+    webrtc::Call* call)
>+    : VoiceMediaChannel(config),
>+      engine_(engine),
>+      call_(call),
>+      audio_config_(config.audio),
>+      crypto_options_(crypto_options) {
>   RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
>   RTC_DCHECK(call);
>   engine->RegisterChannel(this);
>@@ -1264,7 +1291,7 @@ WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
> }
> 
> rtc::DiffServCodePoint WebRtcVoiceMediaChannel::PreferredDscp() const {
>-  return kAudioDscpValue;
>+  return preferred_dscp_;
> }
> 
> bool WebRtcVoiceMediaChannel::SetSendParameters(
>@@ -1283,6 +1310,14 @@ bool WebRtcVoiceMediaChannel::SetSendParameters(
>   if (!ValidateRtpExtensions(params.extensions)) {
>     return false;
>   }
>+
>+  if (ExtmapAllowMixed() != params.extmap_allow_mixed) {
>+    SetExtmapAllowMixed(params.extmap_allow_mixed);
>+    for (auto& it : send_streams_) {
>+      it.second->SetExtmapAllowMixed(params.extmap_allow_mixed);
>+    }
>+  }
>+
>   std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
>       params.extensions, webrtc::RtpExtension::IsSupportedForAudio, true);
>   if (send_rtp_extensions_ != filtered_extensions) {
>@@ -1365,11 +1400,34 @@ webrtc::RTCError WebRtcVoiceMediaChannel::SetRtpSendParameters(
>   // different order (which should change the send codec).
>   webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
>   if (current_parameters.codecs != parameters.codecs) {
>-    RTC_LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
>-                      << "is not currently supported.";
>+    RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
>+                       << "is not currently supported.";
>     return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_PARAMETER);
>   }
> 
>+  if (!parameters.encodings.empty()) {
>+    auto& priority = parameters.encodings[0].network_priority;
>+    rtc::DiffServCodePoint new_dscp = rtc::DSCP_DEFAULT;
>+    if (priority == 0.5 * webrtc::kDefaultBitratePriority) {
>+      new_dscp = rtc::DSCP_CS1;
>+    } else if (priority == 1.0 * webrtc::kDefaultBitratePriority) {
>+      new_dscp = rtc::DSCP_DEFAULT;
>+    } else if (priority == 2.0 * webrtc::kDefaultBitratePriority) {
>+      new_dscp = rtc::DSCP_EF;
>+    } else if (priority == 4.0 * webrtc::kDefaultBitratePriority) {
>+      new_dscp = rtc::DSCP_EF;
>+    } else {
>+      RTC_LOG(LS_WARNING) << "Received invalid send network priority: "
>+                          << priority;
>+      return webrtc::RTCError(webrtc::RTCErrorType::INVALID_RANGE);
>+    }
>+
>+    if (new_dscp != preferred_dscp_) {
>+      preferred_dscp_ = new_dscp;
>+      MediaChannel::UpdateDscp();
>+    }
>+  }
>+
>   // TODO(minyue): The following legacy actions go into
>   // |WebRtcAudioSendStream::SetRtpParameters()| which is called at the end,
>   // though there are two difference:
>@@ -1440,8 +1498,8 @@ bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
> 
>   webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
>   if (current_parameters != parameters) {
>-    RTC_LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
>-                      << "unsupported.";
>+    RTC_DLOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
>+                       << "unsupported.";
>     return false;
>   }
>   return true;
>@@ -1623,17 +1681,17 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
>     // TODO(solenberg): Break out into a separate function?
>     for (const AudioCodec& cn_codec : codecs) {
>       if (IsCodec(cn_codec, kCnCodecName) &&
>-          cn_codec.clockrate == send_codec_spec->format.clockrate_hz) {
>-        switch (cn_codec.clockrate) {
>-          case 8000:
>-          case 16000:
>-          case 32000:
>-            send_codec_spec->cng_payload_type = cn_codec.id;
>-            break;
>-          default:
>-            RTC_LOG(LS_WARNING)
>-                << "CN frequency " << cn_codec.clockrate << " not supported.";
>-            break;
>+          cn_codec.clockrate == send_codec_spec->format.clockrate_hz &&
>+          cn_codec.channels == voice_codec_info->num_channels) {
>+        if (cn_codec.channels != 1) {
>+          RTC_LOG(LS_WARNING)
>+              << "CN #channels " << cn_codec.channels << " not supported.";
>+        } else if (cn_codec.clockrate != 8000 && cn_codec.clockrate != 16000 &&
>+                   cn_codec.clockrate != 32000) {
>+          RTC_LOG(LS_WARNING)
>+              << "CN frequency " << cn_codec.clockrate << " not supported.";
>+        } else {
>+          send_codec_spec->cng_payload_type = cn_codec.id;
>         }
>         break;
>       }
>@@ -1761,9 +1819,11 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
>   absl::optional<std::string> audio_network_adaptor_config =
>       GetAudioNetworkAdaptorConfig(options_);
>   WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
>-      ssrc, mid_, sp.cname, sp.id, send_codec_spec_, send_rtp_extensions_,
>-      max_send_bitrate_bps_, audio_network_adaptor_config, call_, this,
>-      engine()->encoder_factory_, codec_pair_id_, nullptr);
>+      ssrc, mid_, sp.cname, sp.id, send_codec_spec_, ExtmapAllowMixed(),
>+      send_rtp_extensions_, max_send_bitrate_bps_,
>+      audio_config_.rtcp_report_interval_ms, audio_network_adaptor_config,
>+      call_, this, media_transport(), engine()->encoder_factory_,
>+      codec_pair_id_, nullptr, crypto_options_);
>   send_streams_.insert(std::make_pair(ssrc, stream));
> 
>   // At this point the stream's local SSRC has been updated. If it is the first
>@@ -1826,7 +1886,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
> 
>   const uint32_t ssrc = sp.first_ssrc();
>   if (ssrc == 0) {
>-    RTC_LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
>+    RTC_DLOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
>     return false;
>   }
> 
>@@ -1844,13 +1904,15 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
> 
>   // Create a new channel for receiving audio data.
>   recv_streams_.insert(std::make_pair(
>-      ssrc, new WebRtcAudioReceiveStream(
>-                ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
>-                recv_nack_enabled_, sp.stream_ids(), recv_rtp_extensions_,
>-                call_, this, engine()->decoder_factory_, decoder_map_,
>-                codec_pair_id_, engine()->audio_jitter_buffer_max_packets_,
>-                engine()->audio_jitter_buffer_fast_accelerate_,
>-                unsignaled_frame_decryptor_)));
>+      ssrc,
>+      new WebRtcAudioReceiveStream(
>+          ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
>+          recv_nack_enabled_, sp.stream_ids(), recv_rtp_extensions_, call_,
>+          this, media_transport(), engine()->decoder_factory_, decoder_map_,
>+          codec_pair_id_, engine()->audio_jitter_buffer_max_packets_,
>+          engine()->audio_jitter_buffer_fast_accelerate_,
>+          engine()->audio_jitter_buffer_min_delay_ms_,
>+          unsignaled_frame_decryptor_, crypto_options_)));
>   recv_streams_[ssrc]->SetPlayout(playout_);
> 
>   return true;
>@@ -1979,14 +2041,14 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc,
>                                         event, duration);
> }
> 
>-void WebRtcVoiceMediaChannel::OnPacketReceived(
>-    rtc::CopyOnWriteBuffer* packet,
>-    const rtc::PacketTime& packet_time) {
>+void WebRtcVoiceMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                                               int64_t packet_time_us) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
> 
>   webrtc::PacketReceiver::DeliveryStatus delivery_result =
>       call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet,
>-                                       packet_time.timestamp);
>+                                       packet_time_us);
>+
>   if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
>     return;
>   }
>@@ -2017,8 +2079,8 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
>   // Remove oldest unsignaled stream, if we have too many.
>   if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) {
>     uint32_t remove_ssrc = unsignaled_recv_ssrcs_.front();
>-    RTC_LOG(LS_INFO) << "Removing unsignaled receive stream with SSRC="
>-                     << remove_ssrc;
>+    RTC_DLOG(LS_INFO) << "Removing unsignaled receive stream with SSRC="
>+                      << remove_ssrc;
>     RemoveRecvStream(remove_ssrc);
>   }
>   RTC_DCHECK_GE(kMaxUnsignaledRecvStreams, unsignaled_recv_ssrcs_.size());
>@@ -2038,19 +2100,18 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
>     SetRawAudioSink(ssrc, std::move(proxy_sink));
>   }
> 
>-  delivery_result = call_->Receiver()->DeliverPacket(
>-      webrtc::MediaType::AUDIO, *packet, packet_time.timestamp);
>+  delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
>+                                                     *packet, packet_time_us);
>   RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result);
> }
> 
>-void WebRtcVoiceMediaChannel::OnRtcpReceived(
>-    rtc::CopyOnWriteBuffer* packet,
>-    const rtc::PacketTime& packet_time) {
>+void WebRtcVoiceMediaChannel::OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
>+                                             int64_t packet_time_us) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
> 
>   // Forward packet to Call as well.
>   call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet,
>-                                   packet_time.timestamp);
>+                                   packet_time_us);
> }
> 
> void WebRtcVoiceMediaChannel::OnNetworkRouteChanged(
>@@ -2182,6 +2243,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
>     rinfo.secondary_discarded_rate = stats.secondary_discarded_rate;
>     rinfo.accelerate_rate = stats.accelerate_rate;
>     rinfo.preemptive_expand_rate = stats.preemptive_expand_rate;
>+    rinfo.delayed_packet_outage_samples = stats.delayed_packet_outage_samples;
>     rinfo.decoding_calls_to_silence_generator =
>         stats.decoding_calls_to_silence_generator;
>     rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq;
>@@ -2191,6 +2253,8 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
>     rinfo.decoding_plc_cng = stats.decoding_plc_cng;
>     rinfo.decoding_muted_output = stats.decoding_muted_output;
>     rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
>+    rinfo.jitter_buffer_flushes = stats.jitter_buffer_flushes;
>+
>     info->receivers.push_back(rinfo);
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.h
>index bedede12480dc77fe74c30bb54787c19487d8ab4..213f1b393f9e2ca28e059f05cf11a5e3f6c28e44 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.h
>@@ -40,7 +40,7 @@ class WebRtcVoiceMediaChannel;
> 
> // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
> // It uses the WebRtc VoiceEngine library for audio handling.
>-class WebRtcVoiceEngine final {
>+class WebRtcVoiceEngine final : public VoiceEngineInterface {
>   friend class WebRtcVoiceMediaChannel;
> 
>  public:
>@@ -50,19 +50,21 @@ class WebRtcVoiceEngine final {
>       const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
>       rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
>       rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing);
>-  ~WebRtcVoiceEngine();
>+  ~WebRtcVoiceEngine() override;
> 
>   // Does initialization that needs to occur on the worker thread.
>-  void Init();
>+  void Init() override;
> 
>-  rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
>-  VoiceMediaChannel* CreateChannel(webrtc::Call* call,
>-                                   const MediaConfig& config,
>-                                   const AudioOptions& options);
>+  rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
>+  VoiceMediaChannel* CreateMediaChannel(
>+      webrtc::Call* call,
>+      const MediaConfig& config,
>+      const AudioOptions& options,
>+      const webrtc::CryptoOptions& crypto_options) override;
> 
>-  const std::vector<AudioCodec>& send_codecs() const;
>-  const std::vector<AudioCodec>& recv_codecs() const;
>-  RtpCapabilities GetCapabilities() const;
>+  const std::vector<AudioCodec>& send_codecs() const override;
>+  const std::vector<AudioCodec>& recv_codecs() const override;
>+  RtpCapabilities GetCapabilities() const override;
> 
>   // For tracking WebRtc channels. Needed because we have to pause them
>   // all when switching devices.
>@@ -74,10 +76,10 @@ class WebRtcVoiceEngine final {
>   // specified. When the maximum file size is reached, logging is stopped and
>   // the file is closed. If max_size_bytes is set to <= 0, no limit will be
>   // used.
>-  bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
>+  bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
> 
>   // Stops AEC dump.
>-  void StopAecDump();
>+  void StopAecDump() override;
> 
>   const webrtc::AudioProcessing::Config GetApmConfigForTest() const {
>     return apm()->GetConfig();
>@@ -130,6 +132,7 @@ class WebRtcVoiceEngine final {
>   // Jitter buffer settings for new streams.
>   size_t audio_jitter_buffer_max_packets_ = 50;
>   bool audio_jitter_buffer_fast_accelerate_ = false;
>+  int audio_jitter_buffer_min_delay_ms_ = 0;
> 
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine);
> };
>@@ -142,6 +145,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>   WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
>                           const MediaConfig& config,
>                           const AudioOptions& options,
>+                          const webrtc::CryptoOptions& crypto_options,
>                           webrtc::Call* call);
>   ~WebRtcVoiceMediaChannel() override;
> 
>@@ -192,9 +196,9 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>   bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
> 
>   void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                        const rtc::PacketTime& packet_time) override;
>+                        int64_t packet_time_us) override;
>   void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
>-                      const rtc::PacketTime& packet_time) override;
>+                      int64_t packet_time_us) override;
>   void OnNetworkRouteChanged(const std::string& transport_name,
>                              const rtc::NetworkRoute& network_route) override;
>   void OnReadyToSend(bool ready) override;
>@@ -218,6 +222,10 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>     if (DscpEnabled()) {
>       rtc_options.dscp = PreferredDscp();
>     }
>+    rtc_options.info_signaled_after_sent.included_in_feedback =
>+        options.included_in_feedback;
>+    rtc_options.info_signaled_after_sent.included_in_allocation =
>+        options.included_in_allocation;
>     return VoiceMediaChannel::SendPacket(&packet, rtc_options);
>   }
> 
>@@ -227,6 +235,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>     if (DscpEnabled()) {
>       rtc_options.dscp = PreferredDscp();
>     }
>+
>     return VoiceMediaChannel::SendRtcp(&packet, rtc_options);
>   }
> 
>@@ -258,6 +267,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>   std::vector<AudioCodec> recv_codecs_;
> 
>   int max_send_bitrate_bps_ = 0;
>+  rtc::DiffServCodePoint preferred_dscp_ = rtc::DSCP_DEFAULT;
>   AudioOptions options_;
>   absl::optional<int> dtmf_payload_type_;
>   int dtmf_payload_freq_ = -1;
>@@ -268,6 +278,8 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>   bool send_ = false;
>   webrtc::Call* const call_ = nullptr;
> 
>+  const MediaConfig::Audio audio_config_;
>+
>   // Queue of unsignaled SSRCs; oldest at the beginning.
>   std::vector<uint32_t> unsignaled_recv_ssrcs_;
> 
>@@ -302,6 +314,9 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>   const webrtc::AudioCodecPairId codec_pair_id_ =
>       webrtc::AudioCodecPairId::Create();
> 
>+  // Per peer connection crypto options that last for the lifetime of the peer
>+  // connection.
>+  const webrtc::CryptoOptions crypto_options_;
>   // Unsignaled streams have an option to have a frame decryptor set on them.
>   rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
>       unsignaled_frame_decryptor_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine_unittest.cc
>index bb816eed11827f2c0cdc62991643ebdb09835f83..9fd5dfb0ad85c4f7839da8a334ca3e1329302077 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine_unittest.cc
>@@ -11,6 +11,7 @@
> #include <memory>
> #include <utility>
> 
>+#include "absl/strings/match.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/rtpparameters.h"
>@@ -210,8 +211,9 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
> 
>   bool SetupChannel() {
>     EXPECT_CALL(*apm_, SetExtraOptions(testing::_));
>-    channel_ = engine_->CreateChannel(&call_, cricket::MediaConfig(),
>-                                      cricket::AudioOptions());
>+    channel_ = engine_->CreateMediaChannel(&call_, cricket::MediaConfig(),
>+                                           cricket::AudioOptions(),
>+                                           webrtc::CryptoOptions());
>     return (channel_ != nullptr);
>   }
> 
>@@ -253,7 +255,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
> 
>   void DeliverPacket(const void* data, int len) {
>     rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len);
>-    channel_->OnPacketReceived(&packet, rtc::PacketTime());
>+    channel_->OnPacketReceived(&packet, /* packet_time_us */ -1);
>   }
> 
>   void TearDown() override { delete channel_; }
>@@ -349,6 +351,30 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>     EXPECT_EQ(123, telephone_event.duration_ms);
>   }
> 
>+  void TestExtmapAllowMixedCaller(bool extmap_allow_mixed) {
>+    // For a caller, the answer will be applied in set remote description
>+    // where SetSendParameters() is called.
>+    EXPECT_TRUE(SetupChannel());
>+    EXPECT_TRUE(
>+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
>+    send_parameters_.extmap_allow_mixed = extmap_allow_mixed;
>+    SetSendParameters(send_parameters_);
>+    const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX);
>+    EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
>+  }
>+
>+  void TestExtmapAllowMixedCallee(bool extmap_allow_mixed) {
>+    // For a callee, the answer will be applied in set local description
>+    // where SetExtmapAllowMixed() and AddSendStream() are called.
>+    EXPECT_TRUE(SetupChannel());
>+    channel_->SetExtmapAllowMixed(extmap_allow_mixed);
>+    EXPECT_TRUE(
>+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
>+
>+    const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX);
>+    EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
>+  }
>+
>   // Test that send bandwidth is set correctly.
>   // |codec| is the codec under test.
>   // |max_bitrate| is a parameter to set to SetMaxSendBandwidth().
>@@ -735,7 +761,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
> };
> 
> // Tests that we can create and destroy a channel.
>-TEST_F(WebRtcVoiceEngineTestFake, CreateChannel) {
>+TEST_F(WebRtcVoiceEngineTestFake, CreateMediaChannel) {
>   EXPECT_TRUE(SetupChannel());
> }
> 
>@@ -1137,6 +1163,28 @@ TEST_F(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) {
>   EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2]));
> }
> 
>+// RTCRtpEncodingParameters.network_priority must be one of a few values
>+// derived from the default priority, corresponding to very-low, low, medium,
>+// or high.
>+TEST_F(WebRtcVoiceEngineTestFake, SetRtpSendParametersInvalidNetworkPriority) {
>+  EXPECT_TRUE(SetupSendStream());
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
>+  EXPECT_EQ(1UL, parameters.encodings.size());
>+  EXPECT_EQ(webrtc::kDefaultBitratePriority,
>+            parameters.encodings[0].network_priority);
>+
>+  double good_values[] = {0.5, 1.0, 2.0, 4.0};
>+  double bad_values[] = {-1.0, 0.0, 0.49, 0.51, 1.1, 3.99, 4.1, 5.0};
>+  for (auto it : good_values) {
>+    parameters.encodings[0].network_priority = it;
>+    EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
>+  }
>+  for (auto it : bad_values) {
>+    parameters.encodings[0].network_priority = it;
>+    EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
>+  }
>+}
>+
> // Test that GetRtpSendParameters returns the currently configured codecs.
> TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) {
>   EXPECT_TRUE(SetupSendStream());
>@@ -1587,18 +1635,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthForAudioDoesntAffectBwe) {
>   SetSendParameters(send_parameters_);
> }
> 
>-// Test that we can enable NACK with opus as caller.
>-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCaller) {
>-  EXPECT_TRUE(SetupSendStream());
>-  cricket::AudioSendParameters parameters;
>-  parameters.codecs.push_back(kOpusCodec);
>-  parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
>-      cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
>-  EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>-  SetSendParameters(parameters);
>-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>-}
>-
> // Test that we can enable NACK with opus as callee.
> TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
>   EXPECT_TRUE(SetupRecvStream());
>@@ -1613,7 +1649,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
> 
>   EXPECT_TRUE(
>       channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
>-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
> }
> 
> // Test that we can enable NACK on receive streams.
>@@ -1624,29 +1659,11 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) {
>   parameters.codecs.push_back(kOpusCodec);
>   parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
>       cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
>-  EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>   EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
>   SetSendParameters(parameters);
>-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>   EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
> }
> 
>-// Test that we can disable NACK.
>-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNack) {
>-  EXPECT_TRUE(SetupSendStream());
>-  cricket::AudioSendParameters parameters;
>-  parameters.codecs.push_back(kOpusCodec);
>-  parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
>-      cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
>-  SetSendParameters(parameters);
>-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>-
>-  parameters.codecs.clear();
>-  parameters.codecs.push_back(kOpusCodec);
>-  SetSendParameters(parameters);
>-  EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>-}
>-
> // Test that we can disable NACK on receive streams.
> TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) {
>   EXPECT_TRUE(SetupSendStream());
>@@ -1656,13 +1673,11 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) {
>   parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
>       cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
>   SetSendParameters(parameters);
>-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>   EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
> 
>   parameters.codecs.clear();
>   parameters.codecs.push_back(kOpusCodec);
>   SetSendParameters(parameters);
>-  EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>   EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
> }
> 
>@@ -1675,7 +1690,6 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) {
>   parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
>       cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
>   SetSendParameters(parameters);
>-  EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
> 
>   EXPECT_TRUE(AddRecvStream(kSsrcY));
>   EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
>@@ -2799,6 +2813,20 @@ TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) {
>   TestInsertDtmf(kSsrcX, false, kTelephoneEventCodec1);
> }
> 
>+// Test propagation of extmap allow mixed setting.
>+TEST_F(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedAsCaller) {
>+  TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/true);
>+}
>+TEST_F(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedDisabledAsCaller) {
>+  TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/false);
>+}
>+TEST_F(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedAsCallee) {
>+  TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/true);
>+}
>+TEST_F(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedDisabledAsCallee) {
>+  TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/false);
>+}
>+
> TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
>   EXPECT_TRUE(SetupSendStream());
>   EXPECT_TRUE(AddRecvStream(kSsrcY));
>@@ -2914,11 +2942,15 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
>   EXPECT_CALL(*apm_, SetExtraOptions(testing::_)).Times(10);
> 
>   std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel1(
>-      static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel(
>-          &call_, cricket::MediaConfig(), cricket::AudioOptions())));
>+      static_cast<cricket::WebRtcVoiceMediaChannel*>(
>+          engine_->CreateMediaChannel(&call_, cricket::MediaConfig(),
>+                                      cricket::AudioOptions(),
>+                                      webrtc::CryptoOptions())));
>   std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel2(
>-      static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel(
>-          &call_, cricket::MediaConfig(), cricket::AudioOptions())));
>+      static_cast<cricket::WebRtcVoiceMediaChannel*>(
>+          engine_->CreateMediaChannel(&call_, cricket::MediaConfig(),
>+                                      cricket::AudioOptions(),
>+                                      webrtc::CryptoOptions())));
> 
>   // Have to add a stream to make SetSend work.
>   cricket::StreamParams stream1;
>@@ -3017,6 +3049,7 @@ TEST_F(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
>   cricket::FakeNetworkInterface network_interface;
>   cricket::MediaConfig config;
>   std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel;
>+  webrtc::RtpParameters parameters;
> 
>   webrtc::AudioProcessing::Config apm_config;
>   EXPECT_CALL(*apm_, GetConfig())
>@@ -3026,32 +3059,54 @@ TEST_F(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
>   EXPECT_CALL(*apm_, SetExtraOptions(testing::_)).Times(3);
> 
>   channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
>-      engine_->CreateChannel(&call_, config, cricket::AudioOptions())));
>-  channel->SetInterface(&network_interface);
>+      engine_->CreateMediaChannel(&call_, config, cricket::AudioOptions(),
>+                                  webrtc::CryptoOptions())));
>+  channel->SetInterface(&network_interface, /*media_transport=*/nullptr);
>   // Default value when DSCP is disabled should be DSCP_DEFAULT.
>   EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
> 
>   config.enable_dscp = true;
>   channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
>-      engine_->CreateChannel(&call_, config, cricket::AudioOptions())));
>-  channel->SetInterface(&network_interface);
>+      engine_->CreateMediaChannel(&call_, config, cricket::AudioOptions(),
>+                                  webrtc::CryptoOptions())));
>+  channel->SetInterface(&network_interface, /*media_transport=*/nullptr);
>+  EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
>+
>+  // Create a send stream to configure
>+  EXPECT_TRUE(
>+      channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcZ)));
>+  parameters = channel->GetRtpSendParameters(kSsrcZ);
>+  ASSERT_FALSE(parameters.encodings.empty());
>+
>+  // Various priorities map to various dscp values.
>+  parameters.encodings[0].network_priority = 4.0;
>+  ASSERT_TRUE(channel->SetRtpSendParameters(kSsrcZ, parameters).ok());
>   EXPECT_EQ(rtc::DSCP_EF, network_interface.dscp());
>+  parameters.encodings[0].network_priority = 0.5;
>+  ASSERT_TRUE(channel->SetRtpSendParameters(kSsrcZ, parameters).ok());
>+  EXPECT_EQ(rtc::DSCP_CS1, network_interface.dscp());
>+
>+  // A bad priority does not change the dscp value.
>+  parameters.encodings[0].network_priority = 0.0;
>+  ASSERT_FALSE(channel->SetRtpSendParameters(kSsrcZ, parameters).ok());
>+  EXPECT_EQ(rtc::DSCP_CS1, network_interface.dscp());
> 
>   // Packets should also self-identify their dscp in PacketOptions.
>   const uint8_t kData[10] = {0};
>   EXPECT_TRUE(channel->SendRtcp(kData, sizeof(kData)));
>-  EXPECT_EQ(rtc::DSCP_EF, network_interface.options().dscp);
>+  EXPECT_EQ(rtc::DSCP_CS1, network_interface.options().dscp);
> 
>   // Verify that setting the option to false resets the
>   // DiffServCodePoint.
>   config.enable_dscp = false;
>   channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
>-      engine_->CreateChannel(&call_, config, cricket::AudioOptions())));
>-  channel->SetInterface(&network_interface);
>+      engine_->CreateMediaChannel(&call_, config, cricket::AudioOptions(),
>+                                  webrtc::CryptoOptions())));
>+  channel->SetInterface(&network_interface, /*media_transport=*/nullptr);
>   // Default value when DSCP is disabled should be DSCP_DEFAULT.
>   EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
> 
>-  channel->SetInterface(nullptr);
>+  channel->SetInterface(nullptr, nullptr);
> }
> 
> TEST_F(WebRtcVoiceEngineTestFake, SetOutputVolume) {
>@@ -3194,9 +3249,9 @@ TEST_F(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
>   const cricket::FakeAudioReceiveStream* s =
>       call_.GetAudioReceiveStream(kAudioSsrc);
>   EXPECT_EQ(0, s->received_packets());
>-  channel_->OnPacketReceived(&kPcmuPacket, rtc::PacketTime());
>+  channel_->OnPacketReceived(&kPcmuPacket, /* packet_time_us */ -1);
>   EXPECT_EQ(1, s->received_packets());
>-  channel_->OnRtcpReceived(&kRtcpPacket, rtc::PacketTime());
>+  channel_->OnRtcpReceived(&kRtcpPacket, /* packet_time_us */ -1);
>   EXPECT_EQ(2, s->received_packets());
> }
> 
>@@ -3374,8 +3429,9 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
>   webrtc::RtcEventLogNullImpl event_log;
>   std::unique_ptr<webrtc::Call> call(
>       webrtc::Call::Create(webrtc::Call::Config(&event_log)));
>-  cricket::VoiceMediaChannel* channel = engine.CreateChannel(
>-      call.get(), cricket::MediaConfig(), cricket::AudioOptions());
>+  cricket::VoiceMediaChannel* channel = engine.CreateMediaChannel(
>+      call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
>+      webrtc::CryptoOptions());
>   EXPECT_TRUE(channel != nullptr);
>   delete channel;
> }
>@@ -3397,8 +3453,9 @@ TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
>     webrtc::RtcEventLogNullImpl event_log;
>     std::unique_ptr<webrtc::Call> call(
>         webrtc::Call::Create(webrtc::Call::Config(&event_log)));
>-    cricket::VoiceMediaChannel* channel = engine.CreateChannel(
>-        call.get(), cricket::MediaConfig(), cricket::AudioOptions());
>+    cricket::VoiceMediaChannel* channel = engine.CreateMediaChannel(
>+        call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
>+        webrtc::CryptoOptions());
>     EXPECT_TRUE(channel != nullptr);
>     delete channel;
>   }
>@@ -3417,7 +3474,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
>   engine.Init();
>   for (const cricket::AudioCodec& codec : engine.send_codecs()) {
>     auto is_codec = [&codec](const char* name, int clockrate = 0) {
>-      return STR_CASE_CMP(codec.name.c_str(), name) == 0 &&
>+      return absl::EqualsIgnoreCase(codec.name, name) &&
>              (clockrate == 0 || codec.clockrate == clockrate);
>     };
>     if (is_codec("CN", 16000)) {
>@@ -3467,8 +3524,9 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
>   cricket::VoiceMediaChannel* channels[32];
>   size_t num_channels = 0;
>   while (num_channels < arraysize(channels)) {
>-    cricket::VoiceMediaChannel* channel = engine.CreateChannel(
>-        call.get(), cricket::MediaConfig(), cricket::AudioOptions());
>+    cricket::VoiceMediaChannel* channel = engine.CreateMediaChannel(
>+        call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
>+        webrtc::CryptoOptions());
>     if (!channel)
>       break;
>     channels[num_channels++] = channel;
>@@ -3502,7 +3560,8 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
>   std::unique_ptr<webrtc::Call> call(
>       webrtc::Call::Create(webrtc::Call::Config(&event_log)));
>   cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
>-                                           cricket::AudioOptions(), call.get());
>+                                           cricket::AudioOptions(),
>+                                           webrtc::CryptoOptions(), call.get());
>   cricket::AudioRecvParameters parameters;
>   parameters.codecs = engine.recv_codecs();
>   EXPECT_TRUE(channel.SetRecvParameters(parameters));
>@@ -3564,7 +3623,7 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
>   auto find_codec = [&codecs](const webrtc::SdpAudioFormat& format) -> int {
>     for (size_t i = 0; i != codecs.size(); ++i) {
>       const cricket::AudioCodec& codec = codecs[i];
>-      if (STR_CASE_CMP(codec.name.c_str(), format.name.c_str()) == 0 &&
>+      if (absl::EqualsIgnoreCase(codec.name, format.name) &&
>           codec.clockrate == format.clockrate_hz &&
>           codec.channels == format.num_channels) {
>         return rtc::checked_cast<int>(i);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/OWNERS
>index 8262ff750896fb69b1c070d5905861ba46d3f0b4..bc2182b1b60f2c9c43060262adea10a8c90ba415 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/OWNERS
>@@ -1,2 +1 @@
> jeroendb@webrtc.org
>-shampson@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.cc
>index a8b945a6f173d9ba06f29a4bb28c4a5fabc09822..4ffaaf7f5a0448d10fc31ef405cc3ef6b848bd94 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.cc
>@@ -858,7 +858,7 @@ void SctpTransport::OnWritableState(rtc::PacketTransportInternal* transport) {
> void SctpTransport::OnPacketRead(rtc::PacketTransportInternal* transport,
>                                  const char* data,
>                                  size_t len,
>-                                 const rtc::PacketTime& packet_time,
>+                                 const int64_t& /* packet_time_us */,
>                                  int flags) {
>   RTC_DCHECK_RUN_ON(network_thread_);
>   RTC_DCHECK_EQ(transport_, transport);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.h
>index a401d8b795a8aa914ae364ab13552628c5307721..5c2a750e4f3c3220c4e97b4b80716983bdd4305e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.h
>@@ -14,7 +14,7 @@
> #include <errno.h>
> 
> #include <map>
>-#include <memory>  // for unique_ptr.
>+#include <memory>
> #include <set>
> #include <string>
> #include <vector>
>@@ -113,7 +113,7 @@ class SctpTransport : public SctpTransportInternal,
>   virtual void OnPacketRead(rtc::PacketTransportInternal* transport,
>                             const char* data,
>                             size_t len,
>-                            const rtc::PacketTime& packet_time,
>+                            const int64_t& packet_time_us,
>                             int flags);
> 
>   // Methods related to usrsctp callbacks.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransportinternal.h
>index 0380a870b4c0b08d2ce1b46846b588684cba77d0..4c3b542a521d8bb63483720b8064da46495f49a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransportinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransportinternal.h
>@@ -14,7 +14,7 @@
> // TODO(deadbeef): Move SCTP code out of media/, and make it not depend on
> // anything in media/.
> 
>-#include <memory>  // for unique_ptr
>+#include <memory>
> #include <string>
> #include <vector>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/BUILD.gn
>index 8127244462b3357266296af2c3586f15c88bf6d0..ef029914b3093c2be438fa230ebabac1c6f04934 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/BUILD.gn
>@@ -54,6 +54,7 @@ rtc_source_set("module_api") {
>     "../api:libjingle_peerconnection_api",
>     "../modules/rtp_rtcp:rtp_video_header",
>     "../rtc_base:safe_conversions",
>+    "../rtc_base/system:rtc_export",
>   ]
> }
> 
>@@ -236,6 +237,7 @@ if (rtc_include_tests) {
>     deps = [
>       ":module_api",
>       "../test:test_main",
>+      "../test:test_support",
>       "audio_coding:audio_coding_unittests",
>       "audio_device:audio_device_unittests",
>       "audio_mixer:audio_mixer_unittests",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/BUILD.gn
>index 4982d19a7b3036f75089012d1e0e3d7417cbfbc4..df4ba23fff5d37740138c77b802239b4c628315c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/BUILD.gn
>@@ -14,32 +14,6 @@ if (!build_with_mozilla) {
> 
> visibility = [ ":*" ]
> 
>-audio_codec_deps = [
>-  ":cng",
>-  ":g711",
>-  ":pcm16b",
>-]
>-if (rtc_include_ilbc) {
>-  audio_codec_deps += [ ":ilbc" ]
>-}
>-if (rtc_include_opus) {
>-  audio_codec_deps += [ ":webrtc_opus" ]
>-}
>-if (current_cpu == "arm") {
>-  audio_codec_deps += [ ":isac_fix" ]
>-} else {
>-  audio_codec_deps += [ ":isac" ]
>-}
>-audio_codec_deps += [ ":g722" ]
>-if (!build_with_mozilla && !build_with_chromium) {
>-  audio_codec_deps += [ ":red" ]
>-}
>-audio_coding_deps = audio_codec_deps + [
>-                      "../..:webrtc_common",
>-                      "../../common_audio",
>-                      "../../system_wrappers",
>-                    ]
>-
> rtc_static_library("audio_format_conversion") {
>   visibility += webrtc_default_visibility
>   sources = [
>@@ -53,6 +27,7 @@ rtc_static_library("audio_format_conversion") {
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:sanitizer",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -63,7 +38,6 @@ rtc_static_library("rent_a_codec") {
>   # TODO(bugs.webrtc.org/9808): Move to private visibility as soon as that
>   # client code gets updated.
>   visibility += [ "*" ]
>-  allow_poison = [ "audio_codecs" ]
> 
>   sources = [
>     "acm2/acm_codec_database.cc",
>@@ -72,20 +46,18 @@ rtc_static_library("rent_a_codec") {
>     "acm2/rent_a_codec.h",
>   ]
>   deps = [
>-           "../../rtc_base:checks",
>-           "../../api:array_view",
>-           "//third_party/abseil-cpp/absl/types:optional",
>-           "../../api/audio_codecs:audio_codecs_api",
>-           "../..:webrtc_common",
>-           "../../rtc_base:protobuf_utils",
>-           "../../rtc_base:rtc_base_approved",
>-           "../../system_wrappers",
>-           ":audio_coding_module_typedefs",
>-           ":isac_common",
>-           ":isac_fix_c",
>-           ":neteq_decoder_enum",
>-         ] + audio_codec_deps
>-
>+    ":audio_coding_module_typedefs",
>+    ":neteq_decoder_enum",
>+    "../..:webrtc_common",
>+    "../../api:array_view",
>+    "../../api/audio_codecs:audio_codecs_api",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:protobuf_utils",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../system_wrappers",
>+    "//third_party/abseil-cpp/absl/strings",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>   defines = audio_codec_defines
> }
> 
>@@ -95,12 +67,12 @@ rtc_source_set("audio_coding_module_typedefs") {
>   ]
>   deps = [
>     "../..:webrtc_common",
>+    "../../rtc_base:deprecation",
>   ]
> }
> 
> rtc_static_library("audio_coding") {
>   visibility += [ "*" ]
>-  allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
>   sources = [
>     "acm2/acm_receiver.cc",
>     "acm2/acm_receiver.h",
>@@ -109,37 +81,34 @@ rtc_static_library("audio_coding") {
>     "acm2/audio_coding_module.cc",
>     "acm2/call_statistics.cc",
>     "acm2/call_statistics.h",
>-    "acm2/codec_manager.cc",
>-    "acm2/codec_manager.h",
>     "include/audio_coding_module.h",
>   ]
> 
>   defines = []
> 
>-  if (rtc_include_opus) {
>-    public_deps = [
>-      ":webrtc_opus",
>-    ]
>-  }
>-
>-  deps = audio_coding_deps + [
>-           "../../system_wrappers:metrics",
>-           "../../api/audio:audio_frame_api",
>-           "..:module_api",
>-           "../../common_audio:common_audio_c",
>-           "../../rtc_base:deprecation",
>-           "../../rtc_base:checks",
>-           "../../api:array_view",
>-           "../../api/audio_codecs:audio_codecs_api",
>-           ":audio_coding_module_typedefs",
>-           ":neteq",
>-           ":rent_a_codec",
>-           "../../rtc_base:audio_format_to_string",
>-           "../../rtc_base:rtc_base_approved",
>-           "//third_party/abseil-cpp/absl/types:optional",
>-           "../../logging:rtc_event_log_api",
>-         ]
>-  defines = audio_coding_defines
>+  deps = [
>+    ":audio_coding_module_typedefs",
>+    ":neteq",
>+    ":neteq_decoder_enum",
>+    ":rent_a_codec",
>+    "..:module_api",
>+    "..:module_api_public",
>+    "../..:webrtc_common",
>+    "../../api:array_view",
>+    "../../api/audio:audio_frame_api",
>+    "../../api/audio_codecs:audio_codecs_api",
>+    "../../common_audio:common_audio",
>+    "../../common_audio:common_audio_c",
>+    "../../logging:rtc_event_log_api",
>+    "../../rtc_base:audio_format_to_string",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:deprecation",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../system_wrappers",
>+    "../../system_wrappers:metrics",
>+    "//third_party/abseil-cpp/absl/strings",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
> }
> 
> rtc_static_library("legacy_encoded_audio_frame") {
>@@ -150,26 +119,41 @@ rtc_static_library("legacy_encoded_audio_frame") {
>   deps = [
>     "../../api:array_view",
>     "../../api/audio_codecs:audio_codecs_api",
>+    "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>-rtc_static_library("cng") {
>-  visibility += [ "*" ]
>+rtc_static_library("webrtc_cng") {
>+  visibility += webrtc_default_visibility
>   sources = [
>-    "codecs/cng/audio_encoder_cng.cc",
>-    "codecs/cng/audio_encoder_cng.h",
>     "codecs/cng/webrtc_cng.cc",
>     "codecs/cng/webrtc_cng.h",
>   ]
> 
>   deps = [
>-    "../..:webrtc_common",
>     "../../api:array_view",
>-    "../../api/audio_codecs:audio_codecs_api",
>-    "../../common_audio",
>     "../../common_audio:common_audio_c",
>+    "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:safe_conversions",
>+  ]
>+}
>+
>+rtc_static_library("audio_encoder_cng") {
>+  visibility += [ "*" ]
>+  sources = [
>+    "codecs/cng/audio_encoder_cng.cc",
>+    "codecs/cng/audio_encoder_cng.h",
>+  ]
>+
>+  deps = [
>+    ":webrtc_cng",
>+    "../../api/audio_codecs:audio_codecs_api",
>+    "../../common_audio",
>+    "../../rtc_base:checks",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> }
> 
>@@ -181,10 +165,12 @@ rtc_static_library("red") {
>   ]
> 
>   deps = [
>+    "../../api:array_view",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../common_audio",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -201,6 +187,7 @@ rtc_static_library("g711") {
>   deps = [
>     ":legacy_encoded_audio_frame",
>     "../..:webrtc_common",
>+    "../../api:array_view",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>@@ -235,6 +222,7 @@ rtc_static_library("g722") {
>   deps = [
>     ":legacy_encoded_audio_frame",
>     "../..:webrtc_common",
>+    "../../api:array_view",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../api/audio_codecs/g722:audio_encoder_g722_config",
>     "../../rtc_base:checks",
>@@ -270,6 +258,7 @@ rtc_static_library("ilbc") {
>   deps = [
>     ":legacy_encoded_audio_frame",
>     "../..:webrtc_common",
>+    "../../api:array_view",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../api/audio_codecs/ilbc:audio_encoder_ilbc_config",
>     "../../common_audio",
>@@ -786,6 +775,7 @@ rtc_static_library("pcm16b") {
>     ":g711",
>     ":legacy_encoded_audio_frame",
>     "../..:webrtc_common",
>+    "../../api:array_view",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>@@ -820,6 +810,7 @@ rtc_static_library("webrtc_opus") {
>   deps = [
>     ":audio_network_adaptor",
>     "../..:webrtc_common",
>+    "../../api:array_view",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../api/audio_codecs/opus:audio_encoder_opus_config",
>     "../../common_audio",
>@@ -829,6 +820,7 @@ rtc_static_library("webrtc_opus") {
>     "../../rtc_base:safe_minmax",
>     "../../system_wrappers:field_trial",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
>   public_deps = [
>@@ -882,6 +874,7 @@ if (rtc_enable_protobuf) {
>     proto_out_dir = "modules/audio_coding/audio_network_adaptor"
>   }
>   proto_library("ana_config_proto") {
>+    visibility += [ "*" ]
>     sources = [
>       "audio_network_adaptor/config.proto",
>     ]
>@@ -1032,8 +1025,6 @@ rtc_static_library("neteq") {
>     "neteq/random_vector.h",
>     "neteq/red_payload_splitter.cc",
>     "neteq/red_payload_splitter.h",
>-    "neteq/rtcp.cc",
>-    "neteq/rtcp.h",
>     "neteq/statistics_calculator.cc",
>     "neteq/statistics_calculator.h",
>     "neteq/sync_buffer.cc",
>@@ -1048,9 +1039,10 @@ rtc_static_library("neteq") {
> 
>   deps = [
>     ":audio_coding_module_typedefs",
>-    ":cng",
>     ":neteq_decoder_enum",
>+    ":webrtc_cng",
>     "..:module_api",
>+    "..:module_api_public",
>     "../..:webrtc_common",
>     "../../api:array_view",
>     "../../api:libjingle_peerconnection_api",
>@@ -1067,6 +1059,7 @@ rtc_static_library("neteq") {
>     "../../rtc_base/system:fallthrough",
>     "../../system_wrappers:field_trial",
>     "../../system_wrappers:metrics",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -1105,6 +1098,7 @@ rtc_source_set("neteq_tools_minimal") {
>     "../rtp_rtcp",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
>+  defines = audio_codec_defines
> }
> 
> rtc_source_set("neteq_test_tools") {
>@@ -1140,11 +1134,11 @@ rtc_source_set("neteq_test_tools") {
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base",
>     "../../rtc_base:rtc_base_approved",
>-    "../../rtc_base:rtc_base_tests_utils",
>     "../../rtc_base/system:arch",
>     "../../test:rtp_test_utils",
>     "../rtp_rtcp",
>     "../rtp_rtcp:rtp_rtcp_format",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>   public_deps = [
>@@ -1238,6 +1232,7 @@ if (rtc_enable_protobuf) {
>       "../../rtc_base:rtc_base_approved",
>       "../rtp_rtcp",
>       "../rtp_rtcp:rtp_rtcp_format",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>     public_deps = [
>       "../../logging:rtc_event_log_proto",
>@@ -1246,6 +1241,30 @@ if (rtc_enable_protobuf) {
> }
> 
> if (rtc_include_tests) {
>+  audio_coding_deps = [
>+    "../../common_audio",
>+    "../../system_wrappers",
>+    "../..:webrtc_common",
>+    ":audio_encoder_cng",
>+    ":g711",
>+    ":g722",
>+    ":pcm16b",
>+  ]
>+  if (rtc_include_ilbc) {
>+    audio_coding_deps += [ ":ilbc" ]
>+  }
>+  if (rtc_include_opus) {
>+    audio_coding_deps += [ ":webrtc_opus" ]
>+  }
>+  if (current_cpu == "arm") {
>+    audio_coding_deps += [ ":isac_fix" ]
>+  } else {
>+    audio_coding_deps += [ ":isac" ]
>+  }
>+  if (!build_with_mozilla && !build_with_chromium) {
>+    audio_coding_deps += [ ":red" ]
>+  }
>+
>   rtc_source_set("mocks") {
>     testonly = true
>     sources = [
>@@ -1330,10 +1349,11 @@ if (rtc_include_tests) {
>     deps = [
>       ":audio_coding",
>       ":audio_coding_module_typedefs",
>+      ":audio_encoder_cng",
>       ":audio_format_conversion",
>-      ":cng",
>       ":pcm16b_c",
>       ":red",
>+      ":webrtc_opus_c",
>       "..:module_api",
>       "../..:webrtc_common",
>       "../../api/audio:audio_frame_api",
>@@ -1359,6 +1379,7 @@ if (rtc_include_tests) {
>       "../../system_wrappers",
>       "../../test:fileutils",
>       "../../test:test_support",
>+      "//third_party/abseil-cpp/absl/strings",
>       "//third_party/abseil-cpp/absl/types:optional",
>     ]
>     defines = audio_coding_defines
>@@ -1414,6 +1435,7 @@ if (rtc_include_tests) {
>              ":neteq_tools",
>              "../../rtc_base:rtc_base_approved",
>              "../../test:test_support",
>+             "//third_party/abseil-cpp/absl/strings",
>              "//testing/gtest",
>            ]
>   }
>@@ -1477,6 +1499,7 @@ if (rtc_include_tests) {
>              "../../rtc_base/system:arch",
>              "../../test:test_main",
>              "//testing/gtest",
>+             "../../test:test_support",
>            ] + audio_coding_deps
> 
>     data = audio_decoder_unittests_resources
>@@ -1501,7 +1524,7 @@ if (rtc_include_tests) {
>     rtc_source_set("neteq_test_factory") {
>       testonly = true
>       visibility += webrtc_default_visibility
>-      defines = []
>+      defines = audio_codec_defines
>       deps = [
>         "../../rtc_base:checks",
>         "../../test:fileutils",
>@@ -1594,6 +1617,7 @@ if (rtc_include_tests) {
>       "../../api:libjingle_peerconnection_api",
>       "../../rtc_base:rtc_base_approved",
>       "../../test:test_main",
>+      "../../test:test_support",
>       "../audio_processing",
>       "//testing/gtest",
>     ]
>@@ -1661,6 +1685,7 @@ if (rtc_include_tests) {
>     deps = audio_coding_deps + [
>              "//third_party/abseil-cpp/absl/memory",
>              ":audio_coding",
>+             ":audio_encoder_cng",
>              ":neteq_input_audio_tools",
>              "../../api/audio:audio_frame_api",
>              "../../api/audio_codecs/g711:audio_encoder_g711",
>@@ -1756,7 +1781,6 @@ if (rtc_include_tests) {
>       ":neteq_test_support",
>       "../..:webrtc_common",
>       "../../rtc_base:rtc_base_approved",
>-      "../../test:fileutils",
>       "../../test:test_support",
>     ]
>   }
>@@ -1956,6 +1980,7 @@ if (rtc_include_tests) {
>       "../../rtc_base:rtc_base_approved",
>       "../../test:fileutils",
>       "../../test:test_main",
>+      "../../test:test_support",
>       "//testing/gtest",
>     ]
>   }
>@@ -1968,8 +1993,6 @@ if (rtc_include_tests) {
>       "acm2/acm_receiver_unittest.cc",
>       "acm2/audio_coding_module_unittest.cc",
>       "acm2/call_statistics_unittest.cc",
>-      "acm2/codec_manager_unittest.cc",
>-      "acm2/rent_a_codec_unittest.cc",
>       "audio_network_adaptor/audio_network_adaptor_impl_unittest.cc",
>       "audio_network_adaptor/bitrate_controller_unittest.cc",
>       "audio_network_adaptor/channel_controller_unittest.cc",
>@@ -2048,9 +2071,9 @@ if (rtc_include_tests) {
>       ":acm_send_test",
>       ":audio_coding",
>       ":audio_coding_module_typedefs",
>+      ":audio_encoder_cng",
>       ":audio_format_conversion",
>       ":audio_network_adaptor",
>-      ":cng",
>       ":g711",
>       ":ilbc",
>       ":isac",
>@@ -2064,6 +2087,7 @@ if (rtc_include_tests) {
>       ":pcm16b",
>       ":red",
>       ":rent_a_codec",
>+      ":webrtc_cng",
>       ":webrtc_opus",
>       "..:module_api",
>       "../..:webrtc_common",
>@@ -2079,6 +2103,7 @@ if (rtc_include_tests) {
>       "../../logging:mocks",
>       "../../logging:rtc_event_audio",
>       "../../logging:rtc_event_log_api",
>+      "../../modules/rtp_rtcp:rtp_rtcp_format",
>       "../../rtc_base:checks",
>       "../../rtc_base:protobuf_utils",
>       "../../rtc_base:rtc_base",
>@@ -2098,6 +2123,7 @@ if (rtc_include_tests) {
>       "codecs/opus/test:test_unittest",
>       "//testing/gtest",
>       "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
> 
>     defines = audio_coding_defines
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/OWNERS
>index c1adc561b895dd82247d32f3481f24af47c73c89..46f99586842b39151d20ab7c6f63876ee58d0c68 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/OWNERS
>@@ -4,6 +4,7 @@ kwiberg@webrtc.org
> minyue@webrtc.org
> jan.skoglund@webrtc.org
> ossu@webrtc.org
>+ivoc@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
> # structural changes, please get a review from a reviewer in this file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
>index 311af2df3ff6e94fd5933f29ee19a1851065eb14..cada80c061884389a1daa04249099a2e8c4b3ae0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
>@@ -17,7 +17,9 @@
> // references, where appropriate.
> #include "modules/audio_coding/acm2/acm_codec_database.h"
> 
>-#include "rtc_base/checks.h"
>+#include "absl/strings/match.h"
>+#include "api/array_view.h"
>+#include "modules/audio_coding/acm2/rent_a_codec.h"
> 
> #if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
> #error iSAC and iSACFX codecs cannot be enabled at the same time
>@@ -239,12 +241,12 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
>   }
> 
>   // Comfort Noise is special case, packet-size & rate is not checked.
>-  if (STR_CASE_CMP(database_[codec_id].plname, "CN") == 0) {
>+  if (absl::EqualsIgnoreCase(database_[codec_id].plname, "CN")) {
>     return codec_id;
>   }
> 
>   // RED is special case, packet-size & rate is not checked.
>-  if (STR_CASE_CMP(database_[codec_id].plname, "red") == 0) {
>+  if (absl::EqualsIgnoreCase(database_[codec_id].plname, "red")) {
>     return codec_id;
>   }
> 
>@@ -272,12 +274,12 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
> 
>   // Check the validity of rate. Codecs with multiple rates have their own
>   // function for this.
>-  if (STR_CASE_CMP("isac", codec_inst.plname) == 0) {
>+  if (absl::EqualsIgnoreCase("isac", codec_inst.plname)) {
>     return IsISACRateValid(codec_inst.rate) ? codec_id : kInvalidRate;
>-  } else if (STR_CASE_CMP("ilbc", codec_inst.plname) == 0) {
>+  } else if (absl::EqualsIgnoreCase("ilbc", codec_inst.plname)) {
>     return IsILBCRateValid(codec_inst.rate, codec_inst.pacsize) ? codec_id
>                                                                 : kInvalidRate;
>-  } else if (STR_CASE_CMP("opus", codec_inst.plname) == 0) {
>+  } else if (absl::EqualsIgnoreCase("opus", codec_inst.plname)) {
>     return IsOpusRateValid(codec_inst.rate) ? codec_id : kInvalidRate;
>   }
> 
>@@ -296,7 +298,7 @@ int ACMCodecDB::CodecId(const CodecInst& codec_inst) {
> int ACMCodecDB::CodecId(const char* payload_name,
>                         int frequency,
>                         size_t channels) {
>-  for (const CodecInst& ci : RentACodec::Database()) {
>+  for (const CodecInst& ci : database_) {
>     bool name_match = false;
>     bool frequency_match = false;
>     bool channels_match = false;
>@@ -304,10 +306,10 @@ int ACMCodecDB::CodecId(const char* payload_name,
>     // Payload name, sampling frequency and number of channels need to match.
>     // NOTE! If |frequency| is -1, the frequency is not applicable, and is
>     // always treated as true, like for RED.
>-    name_match = (STR_CASE_CMP(ci.plname, payload_name) == 0);
>+    name_match = absl::EqualsIgnoreCase(ci.plname, payload_name);
>     frequency_match = (frequency == ci.plfreq) || (frequency == -1);
>     // The number of channels must match for all codecs but Opus.
>-    if (STR_CASE_CMP(payload_name, "opus") != 0) {
>+    if (!absl::EqualsIgnoreCase(payload_name, "opus")) {
>       channels_match = (channels == ci.channels);
>     } else {
>       // For opus we just check that number of channels is valid.
>@@ -316,7 +318,7 @@ int ACMCodecDB::CodecId(const char* payload_name,
> 
>     if (name_match && frequency_match && channels_match) {
>       // We have found a matching codec in the list.
>-      return &ci - RentACodec::Database().data();
>+      return &ci - database_;
>     }
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.cc
>index 2d7f2960de589bda4d935ac33294a6e59f619fa5..c149ec1b0bc21defe9ad30cc8fb7138ee6d438e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.cc
>@@ -14,6 +14,7 @@
> 
> #include <memory>
> 
>+#include "absl/strings/match.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
>@@ -30,11 +31,11 @@ namespace {
> // Returns true if the codec should be registered, otherwise false. Changes
> // the number of channels for the Opus codec to always be 1.
> bool ModifyAndUseThisCodec(CodecInst* codec_param) {
>-  if (STR_CASE_CMP(codec_param->plname, "CN") == 0 &&
>+  if (absl::EqualsIgnoreCase(codec_param->plname, "CN") &&
>       codec_param->plfreq == 48000)
>     return false;  // Skip 48 kHz comfort noise.
> 
>-  if (STR_CASE_CMP(codec_param->plname, "telephone-event") == 0)
>+  if (absl::EqualsIgnoreCase(codec_param->plname, "telephone-event"))
>     return false;  // Skip DTFM.
> 
>   return true;
>@@ -65,39 +66,43 @@ bool RemapPltypeAndUseThisCodec(const char* plname,
>     return false;  // Don't use non-mono codecs.
> 
>   // Re-map pltypes to those used in the NetEq test files.
>-  if (STR_CASE_CMP(plname, "PCMU") == 0 && plfreq == 8000) {
>+  if (absl::EqualsIgnoreCase(plname, "PCMU") && plfreq == 8000) {
>     *pltype = 0;
>-  } else if (STR_CASE_CMP(plname, "PCMA") == 0 && plfreq == 8000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "PCMA") && plfreq == 8000) {
>     *pltype = 8;
>-  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 8000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "CN") && plfreq == 8000) {
>     *pltype = 13;
>-  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 16000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "CN") && plfreq == 16000) {
>     *pltype = 98;
>-  } else if (STR_CASE_CMP(plname, "CN") == 0 && plfreq == 32000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "CN") && plfreq == 32000) {
>     *pltype = 99;
>-  } else if (STR_CASE_CMP(plname, "ILBC") == 0) {
>+  } else if (absl::EqualsIgnoreCase(plname, "ILBC")) {
>     *pltype = 102;
>-  } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 16000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "ISAC") && plfreq == 16000) {
>     *pltype = 103;
>-  } else if (STR_CASE_CMP(plname, "ISAC") == 0 && plfreq == 32000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "ISAC") && plfreq == 32000) {
>     *pltype = 104;
>-  } else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 8000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
>+             plfreq == 8000) {
>     *pltype = 106;
>-  } else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 16000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
>+             plfreq == 16000) {
>     *pltype = 114;
>-  } else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 32000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
>+             plfreq == 32000) {
>     *pltype = 115;
>-  } else if (STR_CASE_CMP(plname, "telephone-event") == 0 && plfreq == 48000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "telephone-event") &&
>+             plfreq == 48000) {
>     *pltype = 116;
>-  } else if (STR_CASE_CMP(plname, "red") == 0) {
>+  } else if (absl::EqualsIgnoreCase(plname, "red")) {
>     *pltype = 117;
>-  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 8000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "L16") && plfreq == 8000) {
>     *pltype = 93;
>-  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 16000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "L16") && plfreq == 16000) {
>     *pltype = 94;
>-  } else if (STR_CASE_CMP(plname, "L16") == 0 && plfreq == 32000) {
>+  } else if (absl::EqualsIgnoreCase(plname, "L16") && plfreq == 32000) {
>     *pltype = 95;
>-  } else if (STR_CASE_CMP(plname, "G722") == 0) {
>+  } else if (absl::EqualsIgnoreCase(plname, "G722")) {
>     *pltype = 9;
>   } else {
>     // Don't use any other codecs.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.cc
>index f631746b88f4bfb8e50baadbd36e2950e2e43fdb..d3af7c03418851a16c49bfc3254cacb2ab3d0bd7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.cc
>@@ -10,21 +10,21 @@
> 
> #include "modules/audio_coding/acm2/acm_receiver.h"
> 
>-#include <stdlib.h>  // malloc
>-
>-#include <algorithm>  // sort
>+#include <stdlib.h>
>+#include <string.h>
>+#include <cstdint>
> #include <vector>
> 
>+#include "absl/strings/match.h"
>+#include "api/audio/audio_frame.h"
> #include "api/audio_codecs/audio_decoder.h"
>-#include "common_audio/signal_processing/include/signal_processing_library.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/acm2/acm_resampler.h"
> #include "modules/audio_coding/acm2/call_statistics.h"
> #include "modules/audio_coding/acm2/rent_a_codec.h"
> #include "modules/audio_coding/neteq/include/neteq.h"
>+#include "modules/audio_coding/neteq/neteq_decoder_enum.h"
> #include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/format_macros.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/strings/audio_format_to_string.h"
>@@ -92,7 +92,7 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
>     }
>     receive_timestamp = NowInTimestamp(ci->plfreq);
> 
>-    if (STR_CASE_CMP(ci->plname, "cn") == 0) {
>+    if (absl::EqualsIgnoreCase(ci->plname, "cn")) {
>       if (last_audio_decoder_ && last_audio_decoder_->channels > 1) {
>         // This is a CNG and the audio codec is not mono, so skip pushing in
>         // packets into NetEq.
>@@ -345,6 +345,13 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
>   acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
>   acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
>   acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
>+  acm_stat->delayedPacketOutageSamples =
>+      neteq_lifetime_stat.delayed_packet_outage_samples;
>+
>+  NetEqOperationsAndState neteq_operations_and_state =
>+      neteq_->GetOperationsAndState();
>+  acm_stat->packetBufferFlushes =
>+      neteq_operations_and_state.packet_buffer_flushes;
> }
> 
> int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
>@@ -391,7 +398,7 @@ const absl::optional<CodecInst> AcmReceiver::RtpHeaderToDecoder(
>     uint8_t first_payload_byte) const {
>   const absl::optional<CodecInst> ci =
>       neteq_->GetDecoder(rtp_header.payloadType);
>-  if (ci && STR_CASE_CMP(ci->plname, "red") == 0) {
>+  if (ci && absl::EqualsIgnoreCase(ci->plname, "red")) {
>     // This is a RED packet. Get the payload of the audio codec.
>     return neteq_->GetDecoder(first_payload_byte & 0x7f);
>   } else {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.h
>index a2ae723deee6acdbc7f43ff2f322c8ffe7c4248d..8e7d83978c0d7a13a39b62b9219cd4b4e30a30e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
> #define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
> 
>+#include <stdint.h>
> #include <map>
> #include <memory>
> #include <string>
>@@ -18,19 +19,21 @@
> 
> #include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/audio/audio_frame.h"
>-#include "common_audio/vad/include/webrtc_vad.h"
>+#include "api/audio_codecs/audio_decoder.h"
>+#include "api/audio_codecs/audio_format.h"
> #include "modules/audio_coding/acm2/acm_resampler.h"
> #include "modules/audio_coding/acm2/call_statistics.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
>-#include "modules/audio_coding/neteq/include/neteq.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
> 
>+class Clock;
> struct CodecInst;
> class NetEq;
>+struct RTPHeader;
>+struct WebRtcRTPHeader;
> 
> namespace acm2 {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
>index 29f2a45669b3d99f21edc0402d16c1fed0be1f11..46384febb5749f45d0ed2dc71553e944409394ac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
>@@ -73,12 +73,12 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
>     // If we have a compatible CN specification, stack a CNG on top.
>     auto it = cng_payload_types.find(info.sample_rate_hz);
>     if (it != cng_payload_types.end()) {
>-      AudioEncoderCng::Config config;
>+      AudioEncoderCngConfig config;
>       config.speech_encoder = std::move(enc);
>       config.num_channels = 1;
>       config.payload_type = it->second;
>       config.vad_mode = Vad::kVadNormal;
>-      enc = absl::make_unique<AudioEncoderCng>(std::move(config));
>+      enc = CreateComfortNoiseEncoder(std::move(config));
>     }
> 
>     // Actually start using the new encoder.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.cc
>index c0b2064453759cd4fdbf167b1adc5b32407cb4b0..ca3583e32cdea464b9946c6ab9c8c5fe5c374c57 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.cc
>@@ -13,7 +13,6 @@
> #include <assert.h>
> #include <string.h>
> 
>-#include "common_audio/resampler/include/resampler.h"
> #include "rtc_base/logging.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.h
>index 904ea524f8839a604da8a386f7aa5247457adf83..96ba93a7622995ce9aca315bf5aaad93494bd6e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.h
>@@ -11,6 +11,9 @@
> #ifndef MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
> #define MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "common_audio/resampler/include/push_resampler.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
>index 60afeb69bded39cfc37f24fc3f7510539c9bcc12..c0aab3a30fa539a3bb90ea8b130fdb027f57cc50 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
>@@ -10,28 +10,29 @@
> 
> #include "modules/audio_coding/include/audio_coding_module.h"
> 
>+#include <assert.h>
> #include <algorithm>
>+#include <cstdint>
> 
>+#include "absl/strings/match.h"
>+#include "api/array_view.h"
> #include "modules/audio_coding/acm2/acm_receiver.h"
> #include "modules/audio_coding/acm2/acm_resampler.h"
>-#include "modules/audio_coding/acm2/codec_manager.h"
> #include "modules/audio_coding/acm2/rent_a_codec.h"
> #include "modules/include/module_common_types.h"
>+#include "modules/include/module_common_types_public.h"
>+#include "rtc_base/buffer.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/criticalsection.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/metrics.h"
> 
> namespace webrtc {
> 
> namespace {
> 
>-struct EncoderFactory {
>-  AudioEncoder* external_speech_encoder = nullptr;
>-  acm2::CodecManager codec_manager;
>-  acm2::RentACodec rent_a_codec;
>-};
>-
> class AudioCodingModuleImpl final : public AudioCodingModule {
>  public:
>   explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
>@@ -41,12 +42,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
>   //   Sender
>   //
> 
>-  // Can be called multiple times for Codec, CNG, RED.
>-  int RegisterSendCodec(const CodecInst& send_codec) override;
>-
>-  void RegisterExternalSendCodec(
>-      AudioEncoder* external_speech_encoder) override;
>-
>   void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
>                          modifier) override;
> 
>@@ -65,26 +60,10 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
>   // Add 10 ms of raw (PCM) audio data to the encoder.
>   int Add10MsData(const AudioFrame& audio_frame) override;
> 
>-  /////////////////////////////////////////
>-  // (RED) Redundant Coding
>-  //
>-
>-  // Configure RED status i.e. on/off.
>-  int SetREDStatus(bool enable_red) override;
>-
>-  // Get RED status.
>-  bool REDStatus() const override;
>-
>   /////////////////////////////////////////
>   // (FEC) Forward Error Correction (codec internal)
>   //
> 
>-  // Configure FEC status i.e. on/off.
>-  int SetCodecFEC(bool enabled_codec_fec) override;
>-
>-  // Get FEC status.
>-  bool CodecFEC() const override;
>-
>   // Set target packet loss rate
>   int SetPacketLossRate(int loss_rate) override;
> 
>@@ -94,14 +73,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
>   //   (CNG) Comfort Noise Generation
>   //
> 
>-  int SetVAD(bool enable_dtx = true,
>-             bool enable_vad = false,
>-             ACMVADMode mode = VADNormal) override;
>-
>-  int VAD(bool* dtx_enabled,
>-          bool* vad_enabled,
>-          ACMVADMode* mode) const override;
>-
>   int RegisterVADCallback(ACMVADCallback* vad_callback) override;
> 
>   /////////////////////////////////////////
>@@ -122,11 +93,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
>   bool RegisterReceiveCodec(int rtp_payload_type,
>                             const SdpAudioFormat& audio_format) override;
> 
>-  int RegisterReceiveCodec(const CodecInst& receive_codec) override;
>-  int RegisterReceiveCodec(
>-      const CodecInst& receive_codec,
>-      rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) override;
>-
>   int RegisterExternalReceiveCodec(int rtp_payload_type,
>                                    AudioDecoder* external_decoder,
>                                    int sample_rate_hz,
>@@ -214,11 +180,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
>     const std::string histogram_name_;
>   };
> 
>-  int RegisterReceiveCodecUnlocked(
>-      const CodecInst& codec,
>-      rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory)
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
>-
>   int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
>   int Encode(const InputData& input_data)
>@@ -256,12 +217,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
>   acm2::AcmReceiver receiver_;  // AcmReceiver has it's own internal lock.
>   ChangeLogger bitrate_logger_ RTC_GUARDED_BY(acm_crit_sect_);
> 
>-  std::unique_ptr<EncoderFactory> encoder_factory_
>-      RTC_GUARDED_BY(acm_crit_sect_);
>-
>-  // Current encoder stack, either obtained from
>-  // encoder_factory_->rent_a_codec.RentEncoderStack or provided by a call to
>-  // RegisterEncoder.
>+  // Current encoder stack, provided by a call to RegisterEncoder.
>   std::unique_ptr<AudioEncoder> encoder_stack_ RTC_GUARDED_BY(acm_crit_sect_);
> 
>   std::unique_ptr<AudioDecoder> isac_decoder_16k_
>@@ -397,28 +353,6 @@ class RawAudioEncoderWrapper final : public AudioEncoder {
>   AudioEncoder* enc_;
> };
> 
>-// Return false on error.
>-bool CreateSpeechEncoderIfNecessary(EncoderFactory* ef) {
>-  auto* sp = ef->codec_manager.GetStackParams();
>-  if (sp->speech_encoder) {
>-    // Do nothing; we already have a speech encoder.
>-  } else if (ef->codec_manager.GetCodecInst()) {
>-    RTC_DCHECK(!ef->external_speech_encoder);
>-    // We have no speech encoder, but we have a specification for making one.
>-    std::unique_ptr<AudioEncoder> enc =
>-        ef->rent_a_codec.RentEncoder(*ef->codec_manager.GetCodecInst());
>-    if (!enc)
>-      return false;  // Encoder spec was bad.
>-    sp->speech_encoder = std::move(enc);
>-  } else if (ef->external_speech_encoder) {
>-    RTC_DCHECK(!ef->codec_manager.GetCodecInst());
>-    // We have an external speech encoder.
>-    sp->speech_encoder = std::unique_ptr<AudioEncoder>(
>-        new RawAudioEncoderWrapper(ef->external_speech_encoder));
>-  }
>-  return true;
>-}
>-
> void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
>   if (value != last_value_ || first_time_) {
>     first_time_ = false;
>@@ -433,7 +367,6 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
>       expected_in_ts_(0xD87F3F9F),
>       receiver_(config),
>       bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
>-      encoder_factory_(new EncoderFactory),
>       encoder_stack_(nullptr),
>       previous_pltype_(255),
>       receiver_initialized_(false),
>@@ -541,69 +474,29 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
> //   Sender
> //
> 
>-// Can be called multiple times for Codec, CNG, RED.
>-int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) {
>-  rtc::CritScope lock(&acm_crit_sect_);
>-  if (!encoder_factory_->codec_manager.RegisterEncoder(send_codec)) {
>-    return -1;
>-  }
>-  if (encoder_factory_->codec_manager.GetCodecInst()) {
>-    encoder_factory_->external_speech_encoder = nullptr;
>-  }
>-  if (!CreateSpeechEncoderIfNecessary(encoder_factory_.get())) {
>-    return -1;
>-  }
>-  auto* sp = encoder_factory_->codec_manager.GetStackParams();
>-  if (sp->speech_encoder)
>-    encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
>-  return 0;
>-}
>-
>-void AudioCodingModuleImpl::RegisterExternalSendCodec(
>-    AudioEncoder* external_speech_encoder) {
>-  rtc::CritScope lock(&acm_crit_sect_);
>-  encoder_factory_->codec_manager.UnsetCodecInst();
>-  encoder_factory_->external_speech_encoder = external_speech_encoder;
>-  RTC_CHECK(CreateSpeechEncoderIfNecessary(encoder_factory_.get()));
>-  auto* sp = encoder_factory_->codec_manager.GetStackParams();
>-  RTC_CHECK(sp->speech_encoder);
>-  encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
>-}
>-
> void AudioCodingModuleImpl::ModifyEncoder(
>     rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
>   rtc::CritScope lock(&acm_crit_sect_);
>-
>-  // Wipe the encoder factory, so that everything that relies on it will fail.
>-  // We don't want the complexity of supporting swapping back and forth.
>-  if (encoder_factory_) {
>-    encoder_factory_.reset();
>-    RTC_CHECK(!encoder_stack_);  // Ensure we hadn't started using the factory.
>-  }
>-
>   modifier(&encoder_stack_);
> }
> 
> // Get current send codec.
> absl::optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
>   rtc::CritScope lock(&acm_crit_sect_);
>-  if (encoder_factory_) {
>-    auto* ci = encoder_factory_->codec_manager.GetCodecInst();
>-    if (ci) {
>-      return *ci;
>-    }
>-    CreateSpeechEncoderIfNecessary(encoder_factory_.get());
>-    const std::unique_ptr<AudioEncoder>& enc =
>-        encoder_factory_->codec_manager.GetStackParams()->speech_encoder;
>-    if (enc) {
>-      return acm2::CodecManager::ForgeCodecInst(enc.get());
>-    }
>-    return absl::nullopt;
>+  if (encoder_stack_) {
>+    CodecInst ci;
>+    ci.channels = encoder_stack_->NumChannels();
>+    ci.plfreq = encoder_stack_->SampleRateHz();
>+    ci.pacsize = rtc::CheckedDivExact(
>+        static_cast<int>(encoder_stack_->Max10MsFramesInAPacket() * ci.plfreq),
>+        100);
>+    ci.pltype = -1;  // Not valid.
>+    ci.rate = -1;    // Not valid.
>+    static const char kName[] = "external";
>+    memcpy(ci.plname, kName, sizeof(kName));
>+    return ci;
>   } else {
>-    return encoder_stack_
>-               ? absl::optional<CodecInst>(
>-                     acm2::CodecManager::ForgeCodecInst(encoder_stack_.get()))
>-               : absl::nullopt;
>+    return absl::nullopt;
>   }
> }
> 
>@@ -800,59 +693,10 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
>   return 0;
> }
> 
>-/////////////////////////////////////////
>-//   (RED) Redundant Coding
>-//
>-
>-bool AudioCodingModuleImpl::REDStatus() const {
>-  rtc::CritScope lock(&acm_crit_sect_);
>-  return encoder_factory_->codec_manager.GetStackParams()->use_red;
>-}
>-
>-// Configure RED status i.e on/off.
>-int AudioCodingModuleImpl::SetREDStatus(bool enable_red) {
>-#ifdef WEBRTC_CODEC_RED
>-  rtc::CritScope lock(&acm_crit_sect_);
>-  CreateSpeechEncoderIfNecessary(encoder_factory_.get());
>-  if (!encoder_factory_->codec_manager.SetCopyRed(enable_red)) {
>-    return -1;
>-  }
>-  auto* sp = encoder_factory_->codec_manager.GetStackParams();
>-  if (sp->speech_encoder)
>-    encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
>-  return 0;
>-#else
>-  RTC_LOG(LS_WARNING) << "  WEBRTC_CODEC_RED is undefined";
>-  return -1;
>-#endif
>-}
>-
> /////////////////////////////////////////
> //   (FEC) Forward Error Correction (codec internal)
> //
> 
>-bool AudioCodingModuleImpl::CodecFEC() const {
>-  rtc::CritScope lock(&acm_crit_sect_);
>-  return encoder_factory_->codec_manager.GetStackParams()->use_codec_fec;
>-}
>-
>-int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
>-  rtc::CritScope lock(&acm_crit_sect_);
>-  CreateSpeechEncoderIfNecessary(encoder_factory_.get());
>-  if (!encoder_factory_->codec_manager.SetCodecFEC(enable_codec_fec)) {
>-    return -1;
>-  }
>-  auto* sp = encoder_factory_->codec_manager.GetStackParams();
>-  if (sp->speech_encoder)
>-    encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
>-  if (enable_codec_fec) {
>-    return sp->use_codec_fec ? 0 : -1;
>-  } else {
>-    RTC_DCHECK(!sp->use_codec_fec);
>-    return 0;
>-  }
>-}
>-
> int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
>   rtc::CritScope lock(&acm_crit_sect_);
>   if (HaveValidEncoder("SetPacketLossRate")) {
>@@ -861,36 +705,6 @@ int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
>   return 0;
> }
> 
>-/////////////////////////////////////////
>-//   (VAD) Voice Activity Detection
>-//
>-int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
>-                                  bool enable_vad,
>-                                  ACMVADMode mode) {
>-  // Note: |enable_vad| is not used; VAD is enabled based on the DTX setting.
>-  RTC_DCHECK_EQ(enable_dtx, enable_vad);
>-  rtc::CritScope lock(&acm_crit_sect_);
>-  CreateSpeechEncoderIfNecessary(encoder_factory_.get());
>-  if (!encoder_factory_->codec_manager.SetVAD(enable_dtx, mode)) {
>-    return -1;
>-  }
>-  auto* sp = encoder_factory_->codec_manager.GetStackParams();
>-  if (sp->speech_encoder)
>-    encoder_stack_ = encoder_factory_->rent_a_codec.RentEncoderStack(sp);
>-  return 0;
>-}
>-
>-// Get VAD/DTX settings.
>-int AudioCodingModuleImpl::VAD(bool* dtx_enabled,
>-                               bool* vad_enabled,
>-                               ACMVADMode* mode) const {
>-  rtc::CritScope lock(&acm_crit_sect_);
>-  const auto* sp = encoder_factory_->codec_manager.GetStackParams();
>-  *dtx_enabled = *vad_enabled = sp->use_cng;
>-  *mode = sp->vad_mode;
>-  return 0;
>-}
>-
> /////////////////////////////////////////
> //   Receiver
> //
>@@ -949,59 +763,6 @@ bool AudioCodingModuleImpl::RegisterReceiveCodec(
>   return receiver_.AddCodec(rtp_payload_type, audio_format);
> }
> 
>-int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
>-  rtc::CritScope lock(&acm_crit_sect_);
>-  auto* ef = encoder_factory_.get();
>-  return RegisterReceiveCodecUnlocked(
>-      codec, [&] { return ef->rent_a_codec.RentIsacDecoder(codec.plfreq); });
>-}
>-
>-int AudioCodingModuleImpl::RegisterReceiveCodec(
>-    const CodecInst& codec,
>-    rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) {
>-  rtc::CritScope lock(&acm_crit_sect_);
>-  return RegisterReceiveCodecUnlocked(codec, isac_factory);
>-}
>-
>-int AudioCodingModuleImpl::RegisterReceiveCodecUnlocked(
>-    const CodecInst& codec,
>-    rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) {
>-  RTC_DCHECK(receiver_initialized_);
>-  if (codec.channels > 2) {
>-    RTC_LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
>-    return -1;
>-  }
>-
>-  auto codec_id = acm2::RentACodec::CodecIdByParams(codec.plname, codec.plfreq,
>-                                                    codec.channels);
>-  if (!codec_id) {
>-    RTC_LOG_F(LS_ERROR)
>-        << "Wrong codec params to be registered as receive codec";
>-    return -1;
>-  }
>-  auto codec_index = acm2::RentACodec::CodecIndexFromId(*codec_id);
>-  RTC_CHECK(codec_index) << "Invalid codec ID: " << static_cast<int>(*codec_id);
>-
>-  // Check if the payload-type is valid.
>-  if (!acm2::RentACodec::IsPayloadTypeValid(codec.pltype)) {
>-    RTC_LOG_F(LS_ERROR) << "Invalid payload type " << codec.pltype << " for "
>-                        << codec.plname;
>-    return -1;
>-  }
>-
>-  AudioDecoder* isac_decoder = nullptr;
>-  if (STR_CASE_CMP(codec.plname, "isac") == 0) {
>-    std::unique_ptr<AudioDecoder>& saved_isac_decoder =
>-        codec.plfreq == 16000 ? isac_decoder_16k_ : isac_decoder_32k_;
>-    if (!saved_isac_decoder) {
>-      saved_isac_decoder = isac_factory();
>-    }
>-    isac_decoder = saved_isac_decoder.get();
>-  }
>-  return receiver_.AddCodec(*codec_index, codec.pltype, codec.channels,
>-                            codec.plfreq, isac_decoder, codec.plname);
>-}
>-
> int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
>     int rtp_payload_type,
>     AudioDecoder* external_decoder,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
>index 5ac6102c68209988adbb02b7b1ca5cc048ff0527..4e262f7b038a3bdf9f136f3a564a4952a7a26c0f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
>@@ -35,6 +35,7 @@
> #include "modules/audio_coding/neteq/tools/packet.h"
> #include "modules/audio_coding/neteq/tools/rtp_file_source.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/messagedigest.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/platform_thread.h"
>@@ -42,7 +43,6 @@
> #include "rtc_base/system/arch.h"
> #include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/clock.h"
>-#include "system_wrappers/include/event_wrapper.h"
> #include "system_wrappers/include/sleep.h"
> #include "test/gtest.h"
> #include "test/mock_audio_decoder.h"
>@@ -408,12 +408,12 @@ class AudioCodingModuleTestWithComfortNoiseOldApi
>               acm_->RegisterReceiveCodec(
>                   rtp_payload_type, SdpAudioFormat("cn", kSampleRateHz, 1)));
>     acm_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
>-      AudioEncoderCng::Config config;
>+      AudioEncoderCngConfig config;
>       config.speech_encoder = std::move(*enc);
>       config.num_channels = 1;
>       config.payload_type = rtp_payload_type;
>       config.vad_mode = Vad::kVadNormal;
>-      *enc = absl::make_unique<AudioEncoderCng>(std::move(config));
>+      *enc = CreateComfortNoiseEncoder(std::move(config));
>     });
>   }
> 
>@@ -481,7 +481,6 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
>         send_thread_(CbSendThread, this, "send"),
>         insert_packet_thread_(CbInsertPacketThread, this, "insert_packet"),
>         pull_audio_thread_(CbPullAudioThread, this, "pull_audio"),
>-        test_complete_(EventWrapper::Create()),
>         send_count_(0),
>         insert_packet_count_(0),
>         pull_audio_count_(0),
>@@ -512,8 +511,8 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
>     insert_packet_thread_.Stop();
>   }
> 
>-  EventTypeWrapper RunTest() {
>-    return test_complete_->Wait(10 * 60 * 1000);  // 10 minutes' timeout.
>+  bool RunTest() {
>+    return test_complete_.Wait(10 * 60 * 1000);  // 10 minutes' timeout.
>   }
> 
>   virtual bool TestDone() {
>@@ -538,12 +537,12 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
>     SleepMs(1);
>     if (HasFatalFailure()) {
>       // End the test early if a fatal failure (ASSERT_*) has occurred.
>-      test_complete_->Set();
>+      test_complete_.Set();
>     }
>     ++send_count_;
>     InsertAudioAndVerifyEncoding();
>     if (TestDone()) {
>-      test_complete_->Set();
>+      test_complete_.Set();
>     }
>     return true;
>   }
>@@ -592,7 +591,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
>   rtc::PlatformThread send_thread_;
>   rtc::PlatformThread insert_packet_thread_;
>   rtc::PlatformThread pull_audio_thread_;
>-  const std::unique_ptr<EventWrapper> test_complete_;
>+  rtc::Event test_complete_;
>   int send_count_;
>   int insert_packet_count_;
>   int pull_audio_count_ RTC_GUARDED_BY(crit_sect_);
>@@ -607,7 +606,7 @@ class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi {
> #define MAYBE_DoTest DoTest
> #endif
> TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) {
>-  EXPECT_EQ(kEventSignaled, RunTest());
>+  EXPECT_TRUE(RunTest());
> }
> 
> // This is a multi-threaded ACM test using iSAC. The test encodes audio
>@@ -717,7 +716,7 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
> #endif
> #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
> TEST_F(AcmIsacMtTestOldApi, MAYBE_DoTest) {
>-  EXPECT_EQ(kEventSignaled, RunTest());
>+  EXPECT_TRUE(RunTest());
> }
> #endif
> 
>@@ -734,7 +733,6 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
>         codec_registration_thread_(CbCodecRegistrationThread,
>                                    this,
>                                    "codec_registration"),
>-        test_complete_(EventWrapper::Create()),
>         codec_registered_(false),
>         receive_packet_count_(0),
>         next_insert_packet_time_ms_(0),
>@@ -781,8 +779,8 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
>     codec_registration_thread_.Stop();
>   }
> 
>-  EventTypeWrapper RunTest() {
>-    return test_complete_->Wait(10 * 60 * 1000);  // 10 minutes' timeout.
>+  bool RunTest() {
>+    return test_complete_.Wait(10 * 60 * 1000);  // 10 minutes' timeout.
>   }
> 
>   static bool CbReceiveThread(void* context) {
>@@ -845,7 +843,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
>     SleepMs(1);
>     if (HasFatalFailure()) {
>       // End the test early if a fatal failure (ASSERT_*) has occurred.
>-      test_complete_->Set();
>+      test_complete_.Set();
>     }
>     rtc::CritScope lock(&crit_sect_);
>     if (!codec_registered_ &&
>@@ -856,14 +854,14 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
>       codec_registered_ = true;
>     }
>     if (codec_registered_ && receive_packet_count_ > kNumPackets) {
>-      test_complete_->Set();
>+      test_complete_.Set();
>     }
>     return true;
>   }
> 
>   rtc::PlatformThread receive_thread_;
>   rtc::PlatformThread codec_registration_thread_;
>-  const std::unique_ptr<EventWrapper> test_complete_;
>+  rtc::Event test_complete_;
>   rtc::CriticalSection crit_sect_;
>   bool codec_registered_ RTC_GUARDED_BY(crit_sect_);
>   int receive_packet_count_ RTC_GUARDED_BY(crit_sect_);
>@@ -880,7 +878,7 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
> #endif
> #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
> TEST_F(AcmReRegisterIsacMtTestOldApi, MAYBE_DoTest) {
>-  EXPECT_EQ(kEventSignaled, RunTest());
>+  EXPECT_TRUE(RunTest());
> }
> #endif
> 
>@@ -1185,14 +1183,14 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
> 
>     // Extract and verify the audio checksum.
>     std::string checksum_string = audio_checksum.Finish();
>-    EXPECT_EQ(audio_checksum_ref, checksum_string);
>+    ExpectChecksumEq(audio_checksum_ref, checksum_string);
> 
>     // Extract and verify the payload checksum.
>     rtc::Buffer checksum_result(payload_checksum_->Size());
>     payload_checksum_->Finish(checksum_result.data(), checksum_result.size());
>     checksum_string =
>         rtc::hex_encode(checksum_result.data<char>(), checksum_result.size());
>-    EXPECT_EQ(payload_checksum_ref, checksum_string);
>+    ExpectChecksumEq(payload_checksum_ref, checksum_string);
> 
>     // Verify number of packets produced.
>     EXPECT_EQ(expected_packets, packet_count_);
>@@ -1201,6 +1199,18 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
>     remove(output_file_name.c_str());
>   }
> 
>+  // Helper: result must be one the "|"-separated checksums.
>+  void ExpectChecksumEq(std::string ref, std::string result) {
>+    if (ref.size() == result.size()) {
>+      // Only one checksum: clearer message.
>+      EXPECT_EQ(ref, result);
>+    } else {
>+      EXPECT_NE(ref.find(result), std::string::npos)
>+          << result << " must be one of these:\n"
>+          << ref;
>+    }
>+  }
>+
>   // Inherited from test::PacketSource.
>   std::unique_ptr<test::Packet> NextPacket() override {
>     auto packet = send_test_->NextPacket();
>@@ -1438,21 +1448,35 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_G722_stereo_20ms) {
>       50, test::AcmReceiveTestOldApi::kStereoOutput);
> }
> 
>+namespace {
>+// Checksum depends on libopus being compiled with or without SSE.
>+const std::string audio_maybe_sse =
>+    "3e285b74510e62062fbd8142dacd16e9|"
>+    "fd5d57d6d766908e6a7211e2a5c7f78a";
>+const std::string payload_maybe_sse =
>+    "78cf8f03157358acdc69f6835caa0d9b|"
>+    "b693bd95c2ee2354f92340dd09e9da68";
>+// Common checksums.
>+const std::string audio_checksum =
>+    AcmReceiverBitExactnessOldApi::PlatformChecksum(
>+        audio_maybe_sse,
>+        audio_maybe_sse,
>+        "439e97ad1932c49923b5da029c17dd5e",
>+        "038ec90f5f3fc2320f3090f8ecef6bb7",
>+        "038ec90f5f3fc2320f3090f8ecef6bb7");
>+const std::string payload_checksum =
>+    AcmReceiverBitExactnessOldApi::PlatformChecksum(
>+        payload_maybe_sse,
>+        payload_maybe_sse,
>+        "ab88b1a049c36bdfeb7e8b057ef6982a",
>+        "27fef7b799393347ec3b5694369a1c36",
>+        "27fef7b799393347ec3b5694369a1c36");
>+};  // namespace
>+
> TEST_F(AcmSenderBitExactnessOldApi, Opus_stereo_20ms) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 2, 120, 960, 960));
>-  Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
>-          "3e285b74510e62062fbd8142dacd16e9",
>-          "3e285b74510e62062fbd8142dacd16e9",
>-          "439e97ad1932c49923b5da029c17dd5e",
>-          "038ec90f5f3fc2320f3090f8ecef6bb7",
>-          "038ec90f5f3fc2320f3090f8ecef6bb7"),
>-      AcmReceiverBitExactnessOldApi::PlatformChecksum(
>-          "78cf8f03157358acdc69f6835caa0d9b",
>-          "78cf8f03157358acdc69f6835caa0d9b",
>-          "ab88b1a049c36bdfeb7e8b057ef6982a",
>-          "27fef7b799393347ec3b5694369a1c36",
>-          "27fef7b799393347ec3b5694369a1c36"),
>-      50, test::AcmReceiveTestOldApi::kStereoOutput);
>+  Run(audio_checksum, payload_checksum, 50,
>+      test::AcmReceiveTestOldApi::kStereoOutput);
> }
> 
> TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms) {
>@@ -1460,19 +1484,8 @@ TEST_F(AcmSenderBitExactnessNewApi, MAYBE_OpusFromFormat_stereo_20ms) {
>       SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}}));
>   ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(
>       AudioEncoderOpus::MakeAudioEncoder(*config, 120), 120));
>-  Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
>-          "3e285b74510e62062fbd8142dacd16e9",
>-          "3e285b74510e62062fbd8142dacd16e9",
>-          "439e97ad1932c49923b5da029c17dd5e",
>-          "038ec90f5f3fc2320f3090f8ecef6bb7",
>-          "038ec90f5f3fc2320f3090f8ecef6bb7"),
>-      AcmReceiverBitExactnessOldApi::PlatformChecksum(
>-          "78cf8f03157358acdc69f6835caa0d9b",
>-          "78cf8f03157358acdc69f6835caa0d9b",
>-          "ab88b1a049c36bdfeb7e8b057ef6982a",
>-          "27fef7b799393347ec3b5694369a1c36",
>-          "27fef7b799393347ec3b5694369a1c36"),
>-      50, test::AcmReceiveTestOldApi::kStereoOutput);
>+  Run(audio_checksum, payload_checksum, 50,
>+      test::AcmReceiveTestOldApi::kStereoOutput);
> }
> 
> TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms_voip) {
>@@ -1482,15 +1495,19 @@ TEST_F(AcmSenderBitExactnessNewApi, OpusFromFormat_stereo_20ms_voip) {
>   config->application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
>   ASSERT_NO_FATAL_FAILURE(SetUpTestExternalEncoder(
>       AudioEncoderOpus::MakeAudioEncoder(*config, 120), 120));
>+  // Checksum depends on libopus being compiled with or without SSE.
>+  const std::string audio_maybe_sse =
>+      "b0325df4e8104f04e03af23c0b75800e|"
>+      "3cd4e1bc2acd9440bb9e97af34080ffc";
>+  const std::string payload_maybe_sse =
>+      "4eab2259b6fe24c22dd242a113e0b3d9|"
>+      "4fc0af0aa06c26454af09832d3ec1b4e";
>   Run(AcmReceiverBitExactnessOldApi::PlatformChecksum(
>-          "b0325df4e8104f04e03af23c0b75800e",
>-          "b0325df4e8104f04e03af23c0b75800e",
>-          "1c81121f5d9286a5a865d01dbab22ce8",
>+          audio_maybe_sse, audio_maybe_sse, "1c81121f5d9286a5a865d01dbab22ce8",
>           "11d547f89142e9ef03f37d7ca7f32379",
>           "11d547f89142e9ef03f37d7ca7f32379"),
>       AcmReceiverBitExactnessOldApi::PlatformChecksum(
>-          "4eab2259b6fe24c22dd242a113e0b3d9",
>-          "4eab2259b6fe24c22dd242a113e0b3d9",
>+          payload_maybe_sse, payload_maybe_sse,
>           "839ea60399447268ee0f0262a50b75fd",
>           "1815fd5589cad0c6f6cf946c76b81aeb",
>           "1815fd5589cad0c6f6cf946c76b81aeb"),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics.h
>index 9dced6475a80c773f4c14290bf7f770674ae037a..5d94ac453891db8d351b75569145996d4fefea93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics.h
>@@ -12,7 +12,7 @@
> #define MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
> 
> #include "api/audio/audio_frame.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
> 
> //
> // This class is for book keeping of calls to ACM. It is not useful to log API
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.cc
>deleted file mode 100644
>index f29e0f1a1e27b90cc9054d8377eb2f85b4dcd504..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.cc
>+++ /dev/null
>@@ -1,246 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_coding/acm2/codec_manager.h"
>-
>-#include "rtc_base/checks.h"
>-//#include "rtc_base/format_macros.h"
>-#include "modules/audio_coding/acm2/rent_a_codec.h"
>-#include "rtc_base/logging.h"
>-
>-namespace webrtc {
>-namespace acm2 {
>-
>-namespace {
>-
>-// Check if the given codec is a valid to be registered as send codec.
>-int IsValidSendCodec(const CodecInst& send_codec) {
>-  if ((send_codec.channels != 1) && (send_codec.channels != 2)) {
>-    RTC_LOG(LS_ERROR) << "Wrong number of channels (" << send_codec.channels
>-                      << "), only mono and stereo are supported)";
>-    return -1;
>-  }
>-
>-  auto maybe_codec_id = RentACodec::CodecIdByInst(send_codec);
>-  if (!maybe_codec_id) {
>-    RTC_LOG(LS_ERROR) << "Invalid codec setting for the send codec.";
>-    return -1;
>-  }
>-
>-  // Telephone-event cannot be a send codec.
>-  if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) {
>-    RTC_LOG(LS_ERROR) << "telephone-event cannot be a send codec";
>-    return -1;
>-  }
>-
>-  if (!RentACodec::IsSupportedNumChannels(*maybe_codec_id, send_codec.channels)
>-           .value_or(false)) {
>-    RTC_LOG(LS_ERROR) << send_codec.channels
>-                      << " number of channels not supported for "
>-                      << send_codec.plname << ".";
>-    return -1;
>-  }
>-  return RentACodec::CodecIndexFromId(*maybe_codec_id).value_or(-1);
>-}
>-
>-bool IsOpus(const CodecInst& codec) {
>-  return
>-#ifdef WEBRTC_CODEC_OPUS
>-      !STR_CASE_CMP(codec.plname, "opus") ||
>-#endif
>-      false;
>-}
>-
>-}  // namespace
>-
>-CodecManager::CodecManager() {
>-  thread_checker_.DetachFromThread();
>-}
>-
>-CodecManager::~CodecManager() = default;
>-
>-bool CodecManager::RegisterEncoder(const CodecInst& send_codec) {
>-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  int codec_id = IsValidSendCodec(send_codec);
>-
>-  // Check for reported errors from function IsValidSendCodec().
>-  if (codec_id < 0) {
>-    return false;
>-  }
>-
>-  switch (RentACodec::RegisterRedPayloadType(
>-      &codec_stack_params_.red_payload_types, send_codec)) {
>-    case RentACodec::RegistrationResult::kOk:
>-      return true;
>-    case RentACodec::RegistrationResult::kBadFreq:
>-      RTC_LOG(LS_ERROR)
>-          << "RegisterSendCodec() failed, invalid frequency for RED"
>-             " registration";
>-      return false;
>-    case RentACodec::RegistrationResult::kSkip:
>-      break;
>-  }
>-  switch (RentACodec::RegisterCngPayloadType(
>-      &codec_stack_params_.cng_payload_types, send_codec)) {
>-    case RentACodec::RegistrationResult::kOk:
>-      return true;
>-    case RentACodec::RegistrationResult::kBadFreq:
>-      RTC_LOG(LS_ERROR)
>-          << "RegisterSendCodec() failed, invalid frequency for CNG"
>-             " registration";
>-      return false;
>-    case RentACodec::RegistrationResult::kSkip:
>-      break;
>-  }
>-
>-  if (IsOpus(send_codec)) {
>-    // VAD/DTX not supported.
>-    codec_stack_params_.use_cng = false;
>-  }
>-
>-  send_codec_inst_ = send_codec;
>-  recreate_encoder_ = true;  // Caller must recreate it.
>-  return true;
>-}
>-
>-CodecInst CodecManager::ForgeCodecInst(
>-    const AudioEncoder* external_speech_encoder) {
>-  CodecInst ci;
>-  ci.channels = external_speech_encoder->NumChannels();
>-  ci.plfreq = external_speech_encoder->SampleRateHz();
>-  ci.pacsize = rtc::CheckedDivExact(
>-      static_cast<int>(external_speech_encoder->Max10MsFramesInAPacket() *
>-                       ci.plfreq),
>-      100);
>-  ci.pltype = -1;  // Not valid.
>-  ci.rate = -1;    // Not valid.
>-  static const char kName[] = "external";
>-  memcpy(ci.plname, kName, sizeof(kName));
>-  return ci;
>-}
>-
>-bool CodecManager::SetCopyRed(bool enable) {
>-  if (enable && codec_stack_params_.use_codec_fec) {
>-    RTC_LOG(LS_WARNING) << "Codec internal FEC and RED cannot be co-enabled.";
>-    return false;
>-  }
>-  if (enable && send_codec_inst_ &&
>-      codec_stack_params_.red_payload_types.count(send_codec_inst_->plfreq) <
>-          1) {
>-    RTC_LOG(LS_WARNING) << "Cannot enable RED at " << send_codec_inst_->plfreq
>-                        << " Hz.";
>-    return false;
>-  }
>-  codec_stack_params_.use_red = enable;
>-  return true;
>-}
>-
>-bool CodecManager::SetVAD(bool enable, ACMVADMode mode) {
>-  // Sanity check of the mode.
>-  RTC_DCHECK(mode == VADNormal || mode == VADLowBitrate || mode == VADAggr ||
>-             mode == VADVeryAggr);
>-
>-  // Check that the send codec is mono. We don't support VAD/DTX for stereo
>-  // sending.
>-  const bool stereo_send =
>-      codec_stack_params_.speech_encoder
>-          ? (codec_stack_params_.speech_encoder->NumChannels() != 1)
>-          : false;
>-  if (enable && stereo_send) {
>-    RTC_LOG(LS_ERROR) << "VAD/DTX not supported for stereo sending";
>-    return false;
>-  }
>-
>-  // TODO(kwiberg): This doesn't protect Opus when injected as an external
>-  // encoder.
>-  if (send_codec_inst_ && IsOpus(*send_codec_inst_)) {
>-    // VAD/DTX not supported, but don't fail.
>-    enable = false;
>-  }
>-
>-  codec_stack_params_.use_cng = enable;
>-  codec_stack_params_.vad_mode = mode;
>-  return true;
>-}
>-
>-bool CodecManager::SetCodecFEC(bool enable_codec_fec) {
>-  if (enable_codec_fec && codec_stack_params_.use_red) {
>-    RTC_LOG(LS_WARNING) << "Codec internal FEC and RED cannot be co-enabled.";
>-    return false;
>-  }
>-
>-  codec_stack_params_.use_codec_fec = enable_codec_fec;
>-  return true;
>-}
>-
>-bool CodecManager::MakeEncoder(RentACodec* rac, AudioCodingModule* acm) {
>-  RTC_DCHECK(rac);
>-  RTC_DCHECK(acm);
>-
>-  if (!recreate_encoder_) {
>-    bool error = false;
>-    // Try to re-use the speech encoder we've given to the ACM.
>-    acm->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
>-      if (!*encoder) {
>-        // There is no existing encoder.
>-        recreate_encoder_ = true;
>-        return;
>-      }
>-
>-      // Extract the speech encoder from the ACM.
>-      std::unique_ptr<AudioEncoder> enc = std::move(*encoder);
>-      while (true) {
>-        auto sub_enc = enc->ReclaimContainedEncoders();
>-        if (sub_enc.empty()) {
>-          break;
>-        }
>-        RTC_CHECK_EQ(1, sub_enc.size());
>-
>-        // Replace enc with its sub encoder. We need to put the sub encoder in
>-        // a temporary first, since otherwise the old value of enc would be
>-        // destroyed before the new value got assigned, which would be bad
>-        // since the new value is a part of the old value.
>-        auto tmp_enc = std::move(sub_enc[0]);
>-        enc = std::move(tmp_enc);
>-      }
>-
>-      // Wrap it in a new encoder stack and put it back.
>-      codec_stack_params_.speech_encoder = std::move(enc);
>-      *encoder = rac->RentEncoderStack(&codec_stack_params_);
>-      if (!*encoder) {
>-        error = true;
>-      }
>-    });
>-    if (error) {
>-      return false;
>-    }
>-    if (!recreate_encoder_) {
>-      return true;
>-    }
>-  }
>-
>-  if (!send_codec_inst_) {
>-    // We don't have the information we need to create a new speech encoder.
>-    // (This is not an error.)
>-    return true;
>-  }
>-
>-  codec_stack_params_.speech_encoder = rac->RentEncoder(*send_codec_inst_);
>-  auto stack = rac->RentEncoderStack(&codec_stack_params_);
>-  if (!stack) {
>-    return false;
>-  }
>-  acm->SetEncoder(std::move(stack));
>-  recreate_encoder_ = false;
>-  return true;
>-}
>-
>-}  // namespace acm2
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.h
>deleted file mode 100644
>index ffbad9670880508c76f16815769853070571ed8d..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.h
>+++ /dev/null
>@@ -1,75 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
>-#define MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
>-
>-#include <map>
>-
>-#include "absl/types/optional.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/audio_coding/acm2/rent_a_codec.h"
>-#include "modules/audio_coding/include/audio_coding_module.h"
>-#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/thread_checker.h"
>-
>-namespace webrtc {
>-
>-class AudioDecoder;
>-class AudioEncoder;
>-
>-namespace acm2 {
>-
>-class CodecManager final {
>- public:
>-  CodecManager();
>-  ~CodecManager();
>-
>-  // Parses the given specification. On success, returns true and updates the
>-  // stored CodecInst and stack parameters; on error, returns false.
>-  bool RegisterEncoder(const CodecInst& send_codec);
>-
>-  static CodecInst ForgeCodecInst(const AudioEncoder* external_speech_encoder);
>-
>-  const CodecInst* GetCodecInst() const {
>-    return send_codec_inst_ ? &*send_codec_inst_ : nullptr;
>-  }
>-
>-  void UnsetCodecInst() { send_codec_inst_ = absl::nullopt; }
>-
>-  const RentACodec::StackParameters* GetStackParams() const {
>-    return &codec_stack_params_;
>-  }
>-  RentACodec::StackParameters* GetStackParams() { return &codec_stack_params_; }
>-
>-  bool SetCopyRed(bool enable);
>-
>-  bool SetVAD(bool enable, ACMVADMode mode);
>-
>-  bool SetCodecFEC(bool enable_codec_fec);
>-
>-  // Uses the provided Rent-A-Codec to create a new encoder stack, if we have a
>-  // complete specification; if so, it is then passed to set_encoder. On error,
>-  // returns false.
>-  bool MakeEncoder(RentACodec* rac, AudioCodingModule* acm);
>-
>- private:
>-  rtc::ThreadChecker thread_checker_;
>-  absl::optional<CodecInst> send_codec_inst_;
>-  RentACodec::StackParameters codec_stack_params_;
>-  bool recreate_encoder_ = true;  // Need to recreate encoder?
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(CodecManager);
>-};
>-
>-}  // namespace acm2
>-}  // namespace webrtc
>-#endif  // MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
>deleted file mode 100644
>index 6a5ea5f52e2556248d6c07d42ec3fa2b4a3b3875..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
>+++ /dev/null
>@@ -1,70 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-
>-#include "modules/audio_coding/acm2/codec_manager.h"
>-#include "modules/audio_coding/acm2/rent_a_codec.h"
>-#include "test/gtest.h"
>-#include "test/mock_audio_encoder.h"
>-
>-namespace webrtc {
>-namespace acm2 {
>-
>-using ::testing::Return;
>-
>-namespace {
>-
>-// Create a MockAudioEncoder with some reasonable default behavior.
>-std::unique_ptr<MockAudioEncoder> CreateMockEncoder() {
>-  auto enc = std::unique_ptr<MockAudioEncoder>(new MockAudioEncoder);
>-  EXPECT_CALL(*enc, SampleRateHz()).WillRepeatedly(Return(8000));
>-  EXPECT_CALL(*enc, NumChannels()).WillRepeatedly(Return(1));
>-  EXPECT_CALL(*enc, Max10MsFramesInAPacket()).WillRepeatedly(Return(1));
>-  return enc;
>-}
>-
>-}  // namespace
>-
>-TEST(CodecManagerTest, ExternalEncoderFec) {
>-  auto enc0 = CreateMockEncoder();
>-  auto enc1 = CreateMockEncoder();
>-  auto enc2 = CreateMockEncoder();
>-  {
>-    ::testing::InSequence s;
>-    EXPECT_CALL(*enc0, SetFec(false)).WillOnce(Return(true));
>-    EXPECT_CALL(*enc1, SetFec(true)).WillOnce(Return(true));
>-    EXPECT_CALL(*enc2, SetFec(true)).WillOnce(Return(false));
>-  }
>-
>-  CodecManager cm;
>-  RentACodec rac;
>-
>-  // use_codec_fec starts out false.
>-  EXPECT_FALSE(cm.GetStackParams()->use_codec_fec);
>-  cm.GetStackParams()->speech_encoder = std::move(enc0);
>-  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
>-  EXPECT_FALSE(cm.GetStackParams()->use_codec_fec);
>-
>-  // Set it to true.
>-  EXPECT_EQ(true, cm.SetCodecFEC(true));
>-  EXPECT_TRUE(cm.GetStackParams()->use_codec_fec);
>-  cm.GetStackParams()->speech_encoder = std::move(enc1);
>-  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
>-  EXPECT_TRUE(cm.GetStackParams()->use_codec_fec);
>-
>-  // Switch to a codec that doesn't support it.
>-  cm.GetStackParams()->speech_encoder = std::move(enc2);
>-  EXPECT_TRUE(rac.RentEncoderStack(cm.GetStackParams()));
>-  EXPECT_FALSE(cm.GetStackParams()->use_codec_fec);
>-}
>-
>-}  // namespace acm2
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>index 45d78d74a5dd4e8a96e84351107a3b4162dcd9a2..bfddc42bc911876cf59b6eda8507801b1d06a0d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>@@ -13,34 +13,9 @@
> #include <memory>
> #include <utility>
> 
>-#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
>-#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
> #include "rtc_base/logging.h"
>-#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
>-#ifdef WEBRTC_CODEC_ILBC
>-#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"  // nogncheck
>-#endif
>-#ifdef WEBRTC_CODEC_ISACFX
>-#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"  // nogncheck
>-#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"  // nogncheck
>-#endif
>-#ifdef WEBRTC_CODEC_ISAC
>-#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"  // nogncheck
>-#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"  // nogncheck
>-#endif
>-#ifdef WEBRTC_CODEC_OPUS
>-#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
>-#endif
>-#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
>-#ifdef WEBRTC_CODEC_RED
>-#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"  // nogncheck
>-#endif
> #include "modules/audio_coding/acm2/acm_codec_database.h"
> 
>-#if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC)
>-#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
>-#endif
>-
> namespace webrtc {
> namespace acm2 {
> 
>@@ -54,7 +29,8 @@ absl::optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
> 
> absl::optional<CodecInst> RentACodec::CodecInstById(CodecId codec_id) {
>   absl::optional<int> mi = CodecIndexFromId(codec_id);
>-  return mi ? absl::optional<CodecInst>(Database()[*mi]) : absl::nullopt;
>+  return mi ? absl::optional<CodecInst>(ACMCodecDB::database_[*mi])
>+            : absl::nullopt;
> }
> 
> absl::optional<RentACodec::CodecId> RentACodec::CodecIdByInst(
>@@ -80,20 +56,6 @@ absl::optional<CodecInst> RentACodec::CodecInstByParams(
>   return ci;
> }
> 
>-absl::optional<bool> RentACodec::IsSupportedNumChannels(CodecId codec_id,
>-                                                        size_t num_channels) {
>-  auto i = CodecIndexFromId(codec_id);
>-  return i ? absl::optional<bool>(
>-                 ACMCodecDB::codec_settings_[*i].channel_support >=
>-                 num_channels)
>-           : absl::nullopt;
>-}
>-
>-rtc::ArrayView<const CodecInst> RentACodec::Database() {
>-  return rtc::ArrayView<const CodecInst>(ACMCodecDB::database_,
>-                                         NumberOfCodecs());
>-}
>-
> absl::optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
>     CodecId codec_id,
>     size_t num_channels) {
>@@ -106,200 +68,5 @@ absl::optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
>              : ned;
> }
> 
>-RentACodec::RegistrationResult RentACodec::RegisterCngPayloadType(
>-    std::map<int, int>* pt_map,
>-    const CodecInst& codec_inst) {
>-  if (STR_CASE_CMP(codec_inst.plname, "CN") != 0)
>-    return RegistrationResult::kSkip;
>-  switch (codec_inst.plfreq) {
>-    case 8000:
>-    case 16000:
>-    case 32000:
>-    case 48000:
>-      (*pt_map)[codec_inst.plfreq] = codec_inst.pltype;
>-      return RegistrationResult::kOk;
>-    default:
>-      return RegistrationResult::kBadFreq;
>-  }
>-}
>-
>-RentACodec::RegistrationResult RentACodec::RegisterRedPayloadType(
>-    std::map<int, int>* pt_map,
>-    const CodecInst& codec_inst) {
>-  if (STR_CASE_CMP(codec_inst.plname, "RED") != 0)
>-    return RegistrationResult::kSkip;
>-  switch (codec_inst.plfreq) {
>-    case 8000:
>-      (*pt_map)[codec_inst.plfreq] = codec_inst.pltype;
>-      return RegistrationResult::kOk;
>-    default:
>-      return RegistrationResult::kBadFreq;
>-  }
>-}
>-
>-namespace {
>-
>-// Returns a new speech encoder, or null on error.
>-// TODO(kwiberg): Don't handle errors here (bug 5033)
>-std::unique_ptr<AudioEncoder> CreateEncoder(
>-    const CodecInst& speech_inst,
>-    const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo) {
>-#if defined(WEBRTC_CODEC_ISACFX)
>-  if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
>-    return std::unique_ptr<AudioEncoder>(
>-        new AudioEncoderIsacFixImpl(speech_inst, bwinfo));
>-#endif
>-#if defined(WEBRTC_CODEC_ISAC)
>-  if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
>-    return std::unique_ptr<AudioEncoder>(
>-        new AudioEncoderIsacFloatImpl(speech_inst, bwinfo));
>-#endif
>-#ifdef WEBRTC_CODEC_OPUS
>-  if (STR_CASE_CMP(speech_inst.plname, "opus") == 0)
>-    return std::unique_ptr<AudioEncoder>(new AudioEncoderOpusImpl(speech_inst));
>-#endif
>-  if (STR_CASE_CMP(speech_inst.plname, "pcmu") == 0)
>-    return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmU(speech_inst));
>-  if (STR_CASE_CMP(speech_inst.plname, "pcma") == 0)
>-    return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmA(speech_inst));
>-  if (STR_CASE_CMP(speech_inst.plname, "l16") == 0)
>-    return std::unique_ptr<AudioEncoder>(new AudioEncoderPcm16B(speech_inst));
>-#ifdef WEBRTC_CODEC_ILBC
>-  if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0)
>-    return std::unique_ptr<AudioEncoder>(new AudioEncoderIlbcImpl(speech_inst));
>-#endif
>-  if (STR_CASE_CMP(speech_inst.plname, "g722") == 0)
>-    return std::unique_ptr<AudioEncoder>(new AudioEncoderG722Impl(speech_inst));
>-  RTC_LOG_F(LS_ERROR) << "Could not create encoder of type "
>-                      << speech_inst.plname;
>-  return std::unique_ptr<AudioEncoder>();
>-}
>-
>-std::unique_ptr<AudioEncoder> CreateRedEncoder(
>-    std::unique_ptr<AudioEncoder> encoder,
>-    int red_payload_type) {
>-#ifdef WEBRTC_CODEC_RED
>-  AudioEncoderCopyRed::Config config;
>-  config.payload_type = red_payload_type;
>-  config.speech_encoder = std::move(encoder);
>-  return std::unique_ptr<AudioEncoder>(
>-      new AudioEncoderCopyRed(std::move(config)));
>-#else
>-  return std::unique_ptr<AudioEncoder>();
>-#endif
>-}
>-
>-std::unique_ptr<AudioEncoder> CreateCngEncoder(
>-    std::unique_ptr<AudioEncoder> encoder,
>-    int payload_type,
>-    ACMVADMode vad_mode) {
>-  AudioEncoderCng::Config config;
>-  config.num_channels = encoder->NumChannels();
>-  config.payload_type = payload_type;
>-  config.speech_encoder = std::move(encoder);
>-  switch (vad_mode) {
>-    case VADNormal:
>-      config.vad_mode = Vad::kVadNormal;
>-      break;
>-    case VADLowBitrate:
>-      config.vad_mode = Vad::kVadLowBitrate;
>-      break;
>-    case VADAggr:
>-      config.vad_mode = Vad::kVadAggressive;
>-      break;
>-    case VADVeryAggr:
>-      config.vad_mode = Vad::kVadVeryAggressive;
>-      break;
>-    default:
>-      RTC_FATAL();
>-  }
>-  return std::unique_ptr<AudioEncoder>(new AudioEncoderCng(std::move(config)));
>-}
>-
>-std::unique_ptr<AudioDecoder> CreateIsacDecoder(
>-    int sample_rate_hz,
>-    const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo) {
>-#if defined(WEBRTC_CODEC_ISACFX)
>-  return std::unique_ptr<AudioDecoder>(
>-      new AudioDecoderIsacFixImpl(sample_rate_hz, bwinfo));
>-#elif defined(WEBRTC_CODEC_ISAC)
>-  return std::unique_ptr<AudioDecoder>(
>-      new AudioDecoderIsacFloatImpl(sample_rate_hz, bwinfo));
>-#else
>-  RTC_FATAL() << "iSAC is not supported.";
>-  return std::unique_ptr<AudioDecoder>();
>-#endif
>-}
>-
>-}  // namespace
>-
>-RentACodec::RentACodec() {
>-#if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC)
>-  isac_bandwidth_info_ = new LockedIsacBandwidthInfo;
>-#endif
>-}
>-RentACodec::~RentACodec() = default;
>-
>-std::unique_ptr<AudioEncoder> RentACodec::RentEncoder(
>-    const CodecInst& codec_inst) {
>-  return CreateEncoder(codec_inst, isac_bandwidth_info_);
>-}
>-
>-RentACodec::StackParameters::StackParameters() {
>-  // Register the default payload types for RED and CNG.
>-  for (const CodecInst& ci : RentACodec::Database()) {
>-    RentACodec::RegisterCngPayloadType(&cng_payload_types, ci);
>-    RentACodec::RegisterRedPayloadType(&red_payload_types, ci);
>-  }
>-}
>-
>-RentACodec::StackParameters::~StackParameters() = default;
>-
>-std::unique_ptr<AudioEncoder> RentACodec::RentEncoderStack(
>-    StackParameters* param) {
>-  if (!param->speech_encoder)
>-    return nullptr;
>-
>-  if (param->use_codec_fec) {
>-    // Switch FEC on. On failure, remember that FEC is off.
>-    if (!param->speech_encoder->SetFec(true))
>-      param->use_codec_fec = false;
>-  } else {
>-    // Switch FEC off. This shouldn't fail.
>-    const bool success = param->speech_encoder->SetFec(false);
>-    RTC_DCHECK(success);
>-  }
>-
>-  auto pt = [&param](const std::map<int, int>& m) {
>-    auto it = m.find(param->speech_encoder->SampleRateHz());
>-    return it == m.end() ? absl::nullopt : absl::optional<int>(it->second);
>-  };
>-  auto cng_pt = pt(param->cng_payload_types);
>-  param->use_cng =
>-      param->use_cng && cng_pt && param->speech_encoder->NumChannels() == 1;
>-  auto red_pt = pt(param->red_payload_types);
>-  param->use_red = param->use_red && red_pt;
>-
>-  if (param->use_cng || param->use_red) {
>-    // The RED and CNG encoders need to be in sync with the speech encoder, so
>-    // reset the latter to ensure its buffer is empty.
>-    param->speech_encoder->Reset();
>-  }
>-  std::unique_ptr<AudioEncoder> encoder_stack =
>-      std::move(param->speech_encoder);
>-  if (param->use_red) {
>-    encoder_stack = CreateRedEncoder(std::move(encoder_stack), *red_pt);
>-  }
>-  if (param->use_cng) {
>-    encoder_stack =
>-        CreateCngEncoder(std::move(encoder_stack), *cng_pt, param->vad_mode);
>-  }
>-  return encoder_stack;
>-}
>-
>-std::unique_ptr<AudioDecoder> RentACodec::RentIsacDecoder(int sample_rate_hz) {
>-  return CreateIsacDecoder(sample_rate_hz, isac_bandwidth_info_);
>-}
>-
> }  // namespace acm2
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.h
>index b0ad382d8a9b243d2cceb7aa013c50f025010b51..2cf1c6e18c9e6f8b060c05408c625e43da2d862a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.h
>@@ -31,8 +31,7 @@ class LockedIsacBandwidthInfo;
> 
> namespace acm2 {
> 
>-class RentACodec {
>- public:
>+struct RentACodec {
>   enum class CodecId {
> #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
>     kISAC,
>@@ -133,64 +132,9 @@ class RentACodec {
>     return payload_type >= 0 && payload_type <= 127;
>   }
> 
>-  static rtc::ArrayView<const CodecInst> Database();
>-
>-  static absl::optional<bool> IsSupportedNumChannels(CodecId codec_id,
>-                                                     size_t num_channels);
>-
>   static absl::optional<NetEqDecoder> NetEqDecoderFromCodecId(
>       CodecId codec_id,
>       size_t num_channels);
>-
>-  // Parse codec_inst and extract payload types. If the given CodecInst was for
>-  // the wrong sort of codec, return kSkip; otherwise, if the rate was illegal,
>-  // return kBadFreq; otherwise, update the given RTP timestamp rate (Hz) ->
>-  // payload type map and return kOk.
>-  enum class RegistrationResult { kOk, kSkip, kBadFreq };
>-  static RegistrationResult RegisterCngPayloadType(std::map<int, int>* pt_map,
>-                                                   const CodecInst& codec_inst);
>-  static RegistrationResult RegisterRedPayloadType(std::map<int, int>* pt_map,
>-                                                   const CodecInst& codec_inst);
>-
>-  RentACodec();
>-  ~RentACodec();
>-
>-  // Creates and returns an audio encoder built to the given specification.
>-  // Returns null in case of error.
>-  std::unique_ptr<AudioEncoder> RentEncoder(const CodecInst& codec_inst);
>-
>-  struct StackParameters {
>-    StackParameters();
>-    ~StackParameters();
>-
>-    std::unique_ptr<AudioEncoder> speech_encoder;
>-
>-    bool use_codec_fec = false;
>-    bool use_red = false;
>-    bool use_cng = false;
>-    ACMVADMode vad_mode = VADNormal;
>-
>-    // Maps from RTP timestamp rate (in Hz) to payload type.
>-    std::map<int, int> cng_payload_types;
>-    std::map<int, int> red_payload_types;
>-  };
>-
>-  // Creates and returns an audio encoder stack constructed to the given
>-  // specification. If the specification isn't compatible with the encoder, it
>-  // will be changed to match (things will be switched off). The speech encoder
>-  // will be stolen. If the specification isn't complete, returns nullptr.
>-  std::unique_ptr<AudioEncoder> RentEncoderStack(StackParameters* param);
>-
>-  // Creates and returns an iSAC decoder.
>-  std::unique_ptr<AudioDecoder> RentIsacDecoder(int sample_rate_hz);
>-
>- private:
>-  std::unique_ptr<AudioEncoder> speech_encoder_;
>-  std::unique_ptr<AudioEncoder> cng_encoder_;
>-  std::unique_ptr<AudioEncoder> red_encoder_;
>-  rtc::scoped_refptr<LockedIsacBandwidthInfo> isac_bandwidth_info_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(RentACodec);
> };
> 
> }  // namespace acm2
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc
>deleted file mode 100644
>index fd3329cfb11359cf5d27c3417a19983ca816db5e..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc
>+++ /dev/null
>@@ -1,228 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-
>-#include "common_types.h"
>-#include "modules/audio_coding/acm2/rent_a_codec.h"
>-#include "rtc_base/arraysize.h"
>-#include "test/gtest.h"
>-#include "test/mock_audio_encoder.h"
>-
>-namespace webrtc {
>-namespace acm2 {
>-
>-using ::testing::Return;
>-
>-namespace {
>-
>-const int kDataLengthSamples = 80;
>-const int kPacketSizeSamples = 2 * kDataLengthSamples;
>-const int16_t kZeroData[kDataLengthSamples] = {0};
>-const CodecInst kDefaultCodecInst = {0, "pcmu", 8000, kPacketSizeSamples,
>-                                     1, 64000};
>-const int kCngPt = 13;
>-
>-class Marker final {
>- public:
>-  MOCK_METHOD1(Mark, void(std::string desc));
>-};
>-
>-}  // namespace
>-
>-class RentACodecTestF : public ::testing::Test {
>- protected:
>-  void CreateCodec() {
>-    auto speech_encoder = rent_a_codec_.RentEncoder(kDefaultCodecInst);
>-    ASSERT_TRUE(speech_encoder);
>-    RentACodec::StackParameters param;
>-    param.use_cng = true;
>-    param.speech_encoder = std::move(speech_encoder);
>-    encoder_ = rent_a_codec_.RentEncoderStack(&param);
>-  }
>-
>-  void EncodeAndVerify(size_t expected_out_length,
>-                       uint32_t expected_timestamp,
>-                       int expected_payload_type,
>-                       int expected_send_even_if_empty) {
>-    rtc::Buffer out;
>-    AudioEncoder::EncodedInfo encoded_info;
>-    encoded_info = encoder_->Encode(timestamp_, kZeroData, &out);
>-    timestamp_ += kDataLengthSamples;
>-    EXPECT_TRUE(encoded_info.redundant.empty());
>-    EXPECT_EQ(expected_out_length, encoded_info.encoded_bytes);
>-    EXPECT_EQ(expected_timestamp, encoded_info.encoded_timestamp);
>-    if (expected_payload_type >= 0)
>-      EXPECT_EQ(expected_payload_type, encoded_info.payload_type);
>-    if (expected_send_even_if_empty >= 0)
>-      EXPECT_EQ(static_cast<bool>(expected_send_even_if_empty),
>-                encoded_info.send_even_if_empty);
>-  }
>-
>-  RentACodec rent_a_codec_;
>-  std::unique_ptr<AudioEncoder> encoder_;
>-  uint32_t timestamp_ = 0;
>-};
>-
>-// This test verifies that CNG frames are delivered as expected. Since the frame
>-// size is set to 20 ms, we expect the first encode call to produce no output
>-// (which is signaled as 0 bytes output of type kNoEncoding). The next encode
>-// call should produce one SID frame of 9 bytes. The third call should not
>-// result in any output (just like the first one). The fourth and final encode
>-// call should produce an "empty frame", which is like no output, but with
>-// AudioEncoder::EncodedInfo::send_even_if_empty set to true. (The reason to
>-// produce an empty frame is to drive sending of DTMF packets in the RTP/RTCP
>-// module.)
>-TEST_F(RentACodecTestF, VerifyCngFrames) {
>-  CreateCodec();
>-  uint32_t expected_timestamp = timestamp_;
>-  // Verify no frame.
>-  {
>-    SCOPED_TRACE("First encoding");
>-    EncodeAndVerify(0, expected_timestamp, -1, -1);
>-  }
>-
>-  // Verify SID frame delivered.
>-  {
>-    SCOPED_TRACE("Second encoding");
>-    EncodeAndVerify(9, expected_timestamp, kCngPt, 1);
>-  }
>-
>-  // Verify no frame.
>-  {
>-    SCOPED_TRACE("Third encoding");
>-    EncodeAndVerify(0, expected_timestamp, -1, -1);
>-  }
>-
>-  // Verify NoEncoding.
>-  expected_timestamp += 2 * kDataLengthSamples;
>-  {
>-    SCOPED_TRACE("Fourth encoding");
>-    EncodeAndVerify(0, expected_timestamp, kCngPt, 1);
>-  }
>-}
>-
>-TEST(RentACodecTest, ExternalEncoder) {
>-  const int kSampleRateHz = 8000;
>-  auto* external_encoder = new MockAudioEncoder;
>-  EXPECT_CALL(*external_encoder, SampleRateHz())
>-      .WillRepeatedly(Return(kSampleRateHz));
>-  EXPECT_CALL(*external_encoder, NumChannels()).WillRepeatedly(Return(1));
>-  EXPECT_CALL(*external_encoder, SetFec(false)).WillRepeatedly(Return(true));
>-
>-  RentACodec rac;
>-  RentACodec::StackParameters param;
>-  param.speech_encoder = std::unique_ptr<AudioEncoder>(external_encoder);
>-  std::unique_ptr<AudioEncoder> encoder_stack = rac.RentEncoderStack(&param);
>-  EXPECT_EQ(external_encoder, encoder_stack.get());
>-  const int kPacketSizeSamples = kSampleRateHz / 100;
>-  int16_t audio[kPacketSizeSamples] = {0};
>-  rtc::Buffer encoded;
>-  AudioEncoder::EncodedInfo info;
>-
>-  Marker marker;
>-  {
>-    ::testing::InSequence s;
>-    info.encoded_timestamp = 0;
>-    EXPECT_CALL(*external_encoder,
>-                EncodeImpl(0, rtc::ArrayView<const int16_t>(audio), &encoded))
>-        .WillOnce(Return(info));
>-    EXPECT_CALL(marker, Mark("A"));
>-    EXPECT_CALL(marker, Mark("B"));
>-    EXPECT_CALL(marker, Mark("C"));
>-  }
>-
>-  info = encoder_stack->Encode(0, audio, &encoded);
>-  EXPECT_EQ(0u, info.encoded_timestamp);
>-  marker.Mark("A");
>-
>-  // Change to internal encoder.
>-  CodecInst codec_inst = kDefaultCodecInst;
>-  codec_inst.pacsize = kPacketSizeSamples;
>-  param.speech_encoder = rac.RentEncoder(codec_inst);
>-  ASSERT_TRUE(param.speech_encoder);
>-  AudioEncoder* enc = param.speech_encoder.get();
>-  std::unique_ptr<AudioEncoder> stack = rac.RentEncoderStack(&param);
>-  EXPECT_EQ(enc, stack.get());
>-
>-  // Don't expect any more calls to the external encoder.
>-  info = stack->Encode(1, audio, &encoded);
>-  marker.Mark("B");
>-  encoder_stack.reset();
>-  marker.Mark("C");
>-}
>-
>-// Verify that the speech encoder's Reset method is called when CNG or RED
>-// (or both) are switched on, but not when they're switched off.
>-void TestCngAndRedResetSpeechEncoder(bool use_cng, bool use_red) {
>-  auto make_enc = [] {
>-    auto speech_encoder =
>-        std::unique_ptr<MockAudioEncoder>(new MockAudioEncoder);
>-    EXPECT_CALL(*speech_encoder, NumChannels()).WillRepeatedly(Return(1));
>-    EXPECT_CALL(*speech_encoder, Max10MsFramesInAPacket())
>-        .WillRepeatedly(Return(2));
>-    EXPECT_CALL(*speech_encoder, SampleRateHz()).WillRepeatedly(Return(8000));
>-    EXPECT_CALL(*speech_encoder, SetFec(false)).WillRepeatedly(Return(true));
>-    return speech_encoder;
>-  };
>-  auto speech_encoder1 = make_enc();
>-  auto speech_encoder2 = make_enc();
>-  Marker marker;
>-  {
>-    ::testing::InSequence s;
>-    EXPECT_CALL(marker, Mark("disabled"));
>-    EXPECT_CALL(marker, Mark("enabled"));
>-    if (use_cng || use_red)
>-      EXPECT_CALL(*speech_encoder2, Reset());
>-  }
>-
>-  RentACodec::StackParameters param1, param2;
>-  param1.speech_encoder = std::move(speech_encoder1);
>-  param2.speech_encoder = std::move(speech_encoder2);
>-  param2.use_cng = use_cng;
>-  param2.use_red = use_red;
>-  marker.Mark("disabled");
>-  RentACodec rac;
>-  rac.RentEncoderStack(&param1);
>-  marker.Mark("enabled");
>-  rac.RentEncoderStack(&param2);
>-}
>-
>-TEST(RentACodecTest, CngResetsSpeechEncoder) {
>-  TestCngAndRedResetSpeechEncoder(true, false);
>-}
>-
>-TEST(RentACodecTest, RedResetsSpeechEncoder) {
>-  TestCngAndRedResetSpeechEncoder(false, true);
>-}
>-
>-TEST(RentACodecTest, CngAndRedResetsSpeechEncoder) {
>-  TestCngAndRedResetSpeechEncoder(true, true);
>-}
>-
>-TEST(RentACodecTest, NoCngAndRedNoSpeechEncoderReset) {
>-  TestCngAndRedResetSpeechEncoder(false, false);
>-}
>-
>-TEST(RentACodecTest, RentEncoderError) {
>-  const CodecInst codec_inst = {
>-      0, "Robert'); DROP TABLE Students;", 8000, 160, 1, 64000};
>-  RentACodec rent_a_codec;
>-  EXPECT_FALSE(rent_a_codec.RentEncoder(codec_inst));
>-}
>-
>-TEST(RentACodecTest, RentEncoderStackWithoutSpeechEncoder) {
>-  RentACodec::StackParameters sp;
>-  EXPECT_EQ(nullptr, sp.speech_encoder);
>-  EXPECT_EQ(nullptr, RentACodec().RentEncoderStack(&sp));
>-}
>-
>-}  // namespace acm2
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
>index c4832a36de4a557e577e6e5d73476060598288ec..9e47a06f7cac9c3bdfafe549f114c3697fe3232a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
>@@ -10,9 +10,14 @@
> 
> #include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
> 
>+#include <stdint.h>
> #include <utility>
>+#include <vector>
> 
>-#include "rtc_base/logging.h"
>+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
>+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
>+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/field_trial.h"
> 
>@@ -41,17 +46,7 @@ AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl(
>                                    kEventLogMinBitrateChangeBps,
>                                    kEventLogMinBitrateChangeFraction,
>                                    kEventLogMinPacketLossChangeFraction)
>-              : nullptr),
>-      enable_bitrate_adaptation_(
>-          webrtc::field_trial::IsEnabled("WebRTC-Audio-BitrateAdaptation")),
>-      enable_dtx_adaptation_(
>-          webrtc::field_trial::IsEnabled("WebRTC-Audio-DtxAdaptation")),
>-      enable_fec_adaptation_(
>-          webrtc::field_trial::IsEnabled("WebRTC-Audio-FecAdaptation")),
>-      enable_channel_adaptation_(
>-          webrtc::field_trial::IsEnabled("WebRTC-Audio-ChannelAdaptation")),
>-      enable_frame_length_adaptation_(webrtc::field_trial::IsEnabled(
>-          "WebRTC-Audio-FrameLengthAdaptation")) {
>+              : nullptr) {
>   RTC_DCHECK(controller_manager_);
> }
> 
>@@ -152,24 +147,6 @@ AudioEncoderRuntimeConfig AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() {
>   }
>   prev_config_ = config;
> 
>-  // Prevent certain controllers from taking action (determined by field trials)
>-  if (!enable_bitrate_adaptation_ && config.bitrate_bps) {
>-    config.bitrate_bps.reset();
>-  }
>-  if (!enable_dtx_adaptation_ && config.enable_dtx) {
>-    config.enable_dtx.reset();
>-  }
>-  if (!enable_fec_adaptation_ && config.enable_fec) {
>-    config.enable_fec.reset();
>-    config.uplink_packet_loss_fraction.reset();
>-  }
>-  if (!enable_frame_length_adaptation_ && config.frame_length_ms) {
>-    config.frame_length_ms.reset();
>-  }
>-  if (!enable_channel_adaptation_ && config.num_channels) {
>-    config.num_channels.reset();
>-  }
>-
>   if (debug_dump_writer_)
>     debug_dump_writer_->DumpEncoderRuntimeConfig(config, rtc::TimeMillis());
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
>index e208ed2607c344632ba76718fbf0342637eae4ac..4c1c19b7ea28cb1c5e7ad760be0a41ee1add02ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
>@@ -11,17 +11,21 @@
> #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
> #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
> 
>+#include <stdio.h>
> #include <memory>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_encoder.h"
> #include "modules/audio_coding/audio_network_adaptor/controller.h"
>-#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
> #include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
>-#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
>+class ControllerManager;
>+class EventLogWriter;
> class RtcEventLog;
> 
> class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor {
>@@ -79,12 +83,6 @@ class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor {
> 
>   ANAStats stats_;
> 
>-  const bool enable_bitrate_adaptation_;
>-  const bool enable_dtx_adaptation_;
>-  const bool enable_fec_adaptation_;
>-  const bool enable_channel_adaptation_;
>-  const bool enable_frame_length_adaptation_;
>-
>   RTC_DISALLOW_COPY_AND_ASSIGN(AudioNetworkAdaptorImpl);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
>index 5948ac36440da0f02c4a2fc5faddc047efa03954..be9550a7a763e5de345cc2073e0bc65acc5b801a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
>@@ -51,7 +51,7 @@ MATCHER_P(IsRtcEventAnaConfigEqualTo, config, "") {
>     return false;
>   }
>   auto ana_event = static_cast<RtcEventAudioNetworkAdaptation*>(arg);
>-  return *ana_event->config_ == config;
>+  return ana_event->config() == config;
> }
> 
> MATCHER_P(EncoderRuntimeConfigIs, config, "") {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
>index 282f599b6da3025547c5020aed7fd259cbd6dbe0..6b6330b030152b21e2ed8205582d77740fc26d1b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
>@@ -11,7 +11,11 @@
> #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
> #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
> 
>+#include <stddef.h>
>+
>+#include "absl/types/optional.h"
> #include "modules/audio_coding/audio_network_adaptor/controller.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h
>index 23cbef663b4fbfc5551c237278e50f271e5deac1..0d775b1c77f1c97311467d7897baf89ea447b7e7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h
>@@ -11,7 +11,11 @@
> #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
> #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
> 
>+#include <stddef.h>
>+
>+#include "absl/types/optional.h"
> #include "modules/audio_coding/audio_network_adaptor/controller.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
>index 818362ed0da7ea3a3a7cf7a339ab267a6037d035..8e04e8eafc9371f1ff35471906b87fd92b305a78 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
>@@ -10,10 +10,12 @@
> 
> #include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
> 
>+#include "absl/types/optional.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/ignore_wundef.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/protobuf_utils.h"
>+#include "rtc_base/system/file_wrapper.h"
> 
> #if WEBRTC_ENABLE_PROTOBUF
> RTC_PUSH_IGNORING_WUNDEF()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h
>index fb40db28b20cca73e959322f42d4d4dba87afcd0..d3334eca9ef001b52840c4bf507fff0e2cfd10db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h
>@@ -11,7 +11,9 @@
> #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
> #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
> 
>+#include "absl/types/optional.h"
> #include "modules/audio_coding/audio_network_adaptor/controller.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
>index 4a92343d9688bd09aaf98e3520a61579d3096033..7925b891277501de0dc62b704e99d5998c48d433 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
>@@ -9,13 +9,17 @@
>  */
> 
> #include <math.h>
>-
> #include <algorithm>
>+#include <cstdlib>
>+#include <utility>
> 
> #include "absl/memory/memory.h"
>+#include "absl/types/optional.h"
>+#include "logging/rtc_event_log/events/rtc_event.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h
>index fca8e53b664bf1bf3de89e50e98a82e8b0909267..72b5245e115a6ffd50205d10a3c8cba9e1dc899c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
> #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
> 
>-#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
>index df975947ca13a1f43ac8ceb8e75a5bf4ccd52446..42189c3e11c321fa03639a673dca22306779e0cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
>@@ -36,7 +36,7 @@ MATCHER_P(IsRtcEventAnaConfigEqualTo, config, "") {
>     return false;
>   }
>   auto ana_event = static_cast<RtcEventAudioNetworkAdaptation*>(arg);
>-  return *ana_event->config_ == config;
>+  return ana_event->config() == config;
> }
> 
> struct EventLogWriterStates {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
>index 7ab72c9437e37219d24fd1e0a12799d4b5d87d88..936e22429ac88854fcd72eca75217e7d326aa10c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
>@@ -10,7 +10,7 @@
> 
> #include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
> 
>-#include <limits>
>+#include <string>
> #include <utility>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
>index b66883ef552e514b038e82b25e4d6dc9727c64ad..b7d3d56173e6ba8de2d6cb35203b9ce31bd37e2e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
>@@ -13,8 +13,10 @@
> 
> #include <memory>
> 
>+#include "absl/types/optional.h"
> #include "common_audio/smoothing_filter.h"
> #include "modules/audio_coding/audio_network_adaptor/controller.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
> #include "rtc_base/constructormagic.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc
>index c8cfd31607c46e30a4cefc5a41fc349457b26414..6c30b8f2c0743e74008e6a22d41e6018a8c033d6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc
>@@ -10,9 +10,6 @@
> 
> #include "modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
> 
>-#include <limits>
>-#include <utility>
>-
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
>index 9a3c37c3d1d8ce06c208af72720fa7d1686a1e13..421cb70abd18baaabcfc6a0fadf43764f87b05ec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
>@@ -11,9 +11,9 @@
> #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_RPLR_BASED_H_
> #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_RPLR_BASED_H_
> 
>-#include <memory>
>-
>+#include "absl/types/optional.h"
> #include "modules/audio_coding/audio_network_adaptor/controller.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
> #include "rtc_base/constructormagic.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
>index 40e97cb371e43c270f771ecf060e1f0e63615f02..b123c7c70ed5e87f825863c09a2c3fa686664fe8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
>@@ -11,10 +11,10 @@
> #include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
> 
> #include <algorithm>
>+#include <iterator>
> #include <utility>
> 
> #include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
>index f084fd0382f1607959d0c018c5df42b695bab926..f0a5aaba9a2a6a2a7d4653756400414a6e1b8e8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
>@@ -11,10 +11,13 @@
> #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
> #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
> 
>+#include <stddef.h>
> #include <map>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "modules/audio_coding/audio_network_adaptor/controller.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
>index 257a79af18d9194f1d0065eb0aadd8c39689390c..94e8ed961e7e683bf0d89be57b10e2be347cefac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
>@@ -11,6 +11,8 @@
> #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
> #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
> 
>+#include <stddef.h>
>+
> #include "absl/types/optional.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
>index e38aa3333505c76420fb54f7892ef6497f87fcf1..f068301adf9168576cb11306fdbcffa71e564551 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
>@@ -11,11 +11,12 @@
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> 
> #include <string.h>
>+#include <string>
>+#include <utility>
> 
>-#include "absl/types/optional.h"
>+#include "absl/strings/match.h"
> #include "api/array_view.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/sanitizer.h"
> 
> namespace webrtc {
>@@ -41,11 +42,11 @@ CodecInst MakeCodecInst(int payload_type,
> }  // namespace
> 
> SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
>-  if (STR_CASE_CMP(ci.plname, "g722") == 0) {
>+  if (absl::EqualsIgnoreCase(ci.plname, "g722")) {
>     RTC_CHECK_EQ(16000, ci.plfreq);
>     RTC_CHECK(ci.channels == 1 || ci.channels == 2);
>     return {"g722", 8000, ci.channels};
>-  } else if (STR_CASE_CMP(ci.plname, "opus") == 0) {
>+  } else if (absl::EqualsIgnoreCase(ci.plname, "opus")) {
>     RTC_CHECK_EQ(48000, ci.plfreq);
>     RTC_CHECK(ci.channels == 1 || ci.channels == 2);
>     return ci.channels == 1
>@@ -57,12 +58,12 @@ SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
> }
> 
> CodecInst SdpToCodecInst(int payload_type, const SdpAudioFormat& audio_format) {
>-  if (STR_CASE_CMP(audio_format.name.c_str(), "g722") == 0) {
>+  if (absl::EqualsIgnoreCase(audio_format.name, "g722")) {
>     RTC_CHECK_EQ(8000, audio_format.clockrate_hz);
>     RTC_CHECK(audio_format.num_channels == 1 || audio_format.num_channels == 2);
>     return MakeCodecInst(payload_type, "g722", 16000,
>                          audio_format.num_channels);
>-  } else if (STR_CASE_CMP(audio_format.name.c_str(), "opus") == 0) {
>+  } else if (absl::EqualsIgnoreCase(audio_format.name, "opus")) {
>     RTC_CHECK_EQ(48000, audio_format.clockrate_hz);
>     RTC_CHECK_EQ(2, audio_format.num_channels);
>     const int num_channels = [&] {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>index 055190497a54c28436ed401295d2a01516ee4360..476954fe60d06d0478cbb5bd47d27d3c89d9ce9d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>@@ -10,40 +10,72 @@
> 
> #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
> 
>-#include <algorithm>
>-#include <limits>
>+#include <cstdint>
> #include <memory>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
>+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
>+#include "rtc_base/checks.h"
>+
> namespace webrtc {
> 
> namespace {
> 
> const int kMaxFrameSizeMs = 60;
> 
>-}  // namespace
>-
>-AudioEncoderCng::Config::Config() = default;
>-AudioEncoderCng::Config::Config(Config&&) = default;
>-AudioEncoderCng::Config::~Config() = default;
>-
>-bool AudioEncoderCng::Config::IsOk() const {
>-  if (num_channels != 1)
>-    return false;
>-  if (!speech_encoder)
>-    return false;
>-  if (num_channels != speech_encoder->NumChannels())
>-    return false;
>-  if (sid_frame_interval_ms <
>-      static_cast<int>(speech_encoder->Max10MsFramesInAPacket() * 10))
>-    return false;
>-  if (num_cng_coefficients > WEBRTC_CNG_MAX_LPC_ORDER ||
>-      num_cng_coefficients <= 0)
>-    return false;
>-  return true;
>-}
>-
>-AudioEncoderCng::AudioEncoderCng(Config&& config)
>+class AudioEncoderCng final : public AudioEncoder {
>+ public:
>+  explicit AudioEncoderCng(AudioEncoderCngConfig&& config);
>+  ~AudioEncoderCng() override;
>+
>+  // Not copyable or moveable.
>+  AudioEncoderCng(const AudioEncoderCng&) = delete;
>+  AudioEncoderCng(AudioEncoderCng&&) = delete;
>+  AudioEncoderCng& operator=(const AudioEncoderCng&) = delete;
>+  AudioEncoderCng& operator=(AudioEncoderCng&&) = delete;
>+
>+  int SampleRateHz() const override;
>+  size_t NumChannels() const override;
>+  int RtpTimestampRateHz() const override;
>+  size_t Num10MsFramesInNextPacket() const override;
>+  size_t Max10MsFramesInAPacket() const override;
>+  int GetTargetBitrate() const override;
>+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
>+                         rtc::ArrayView<const int16_t> audio,
>+                         rtc::Buffer* encoded) override;
>+  void Reset() override;
>+  bool SetFec(bool enable) override;
>+  bool SetDtx(bool enable) override;
>+  bool SetApplication(Application application) override;
>+  void SetMaxPlaybackRate(int frequency_hz) override;
>+  rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
>+      override;
>+  void OnReceivedUplinkPacketLossFraction(
>+      float uplink_packet_loss_fraction) override;
>+  void OnReceivedUplinkRecoverablePacketLossFraction(
>+      float uplink_recoverable_packet_loss_fraction) override;
>+  void OnReceivedUplinkBandwidth(
>+      int target_audio_bitrate_bps,
>+      absl::optional<int64_t> bwe_period_ms) override;
>+
>+ private:
>+  EncodedInfo EncodePassive(size_t frames_to_encode, rtc::Buffer* encoded);
>+  EncodedInfo EncodeActive(size_t frames_to_encode, rtc::Buffer* encoded);
>+  size_t SamplesPer10msFrame() const;
>+
>+  std::unique_ptr<AudioEncoder> speech_encoder_;
>+  const int cng_payload_type_;
>+  const int num_cng_coefficients_;
>+  const int sid_frame_interval_ms_;
>+  std::vector<int16_t> speech_buffer_;
>+  std::vector<uint32_t> rtp_timestamps_;
>+  bool last_frame_active_;
>+  std::unique_ptr<Vad> vad_;
>+  std::unique_ptr<ComfortNoiseEncoder> cng_encoder_;
>+};
>+
>+AudioEncoderCng::AudioEncoderCng(AudioEncoderCngConfig&& config)
>     : speech_encoder_((static_cast<void>([&] {
>                          RTC_CHECK(config.IsOk()) << "Invalid configuration.";
>                        }()),
>@@ -261,4 +293,31 @@ size_t AudioEncoderCng::SamplesPer10msFrame() const {
>   return rtc::CheckedDivExact(10 * SampleRateHz(), 1000);
> }
> 
>+}  // namespace
>+
>+AudioEncoderCngConfig::AudioEncoderCngConfig() = default;
>+AudioEncoderCngConfig::AudioEncoderCngConfig(AudioEncoderCngConfig&&) = default;
>+AudioEncoderCngConfig::~AudioEncoderCngConfig() = default;
>+
>+bool AudioEncoderCngConfig::IsOk() const {
>+  if (num_channels != 1)
>+    return false;
>+  if (!speech_encoder)
>+    return false;
>+  if (num_channels != speech_encoder->NumChannels())
>+    return false;
>+  if (sid_frame_interval_ms <
>+      static_cast<int>(speech_encoder->Max10MsFramesInAPacket() * 10))
>+    return false;
>+  if (num_cng_coefficients > WEBRTC_CNG_MAX_LPC_ORDER ||
>+      num_cng_coefficients <= 0)
>+    return false;
>+  return true;
>+}
>+
>+std::unique_ptr<AudioEncoder> CreateComfortNoiseEncoder(
>+    AudioEncoderCngConfig&& config) {
>+  return absl::make_unique<AudioEncoderCng>(std::move(config));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
>index e4c65077651c0536ec23e0c9cebc99595fa474c3..2ef32364c7ea9df4ec16874b112005599f6208cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
>@@ -11,86 +11,38 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
> #define MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
> 
>+#include <stddef.h>
> #include <memory>
>-#include <vector>
> 
> #include "api/audio_codecs/audio_encoder.h"
> #include "common_audio/vad/include/vad.h"
>-#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
>-#include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
>-class Vad;
>-
>-class AudioEncoderCng final : public AudioEncoder {
>- public:
>-  struct Config {
>-    Config();
>-    Config(Config&&);
>-    ~Config();
>-    bool IsOk() const;
>-
>-    size_t num_channels = 1;
>-    int payload_type = 13;
>-    std::unique_ptr<AudioEncoder> speech_encoder;
>-    Vad::Aggressiveness vad_mode = Vad::kVadNormal;
>-    int sid_frame_interval_ms = 100;
>-    int num_cng_coefficients = 8;
>-    // The Vad pointer is mainly for testing. If a NULL pointer is passed, the
>-    // AudioEncoderCng creates (and destroys) a Vad object internally. If an
>-    // object is passed, the AudioEncoderCng assumes ownership of the Vad
>-    // object.
>-    Vad* vad = nullptr;
>-  };
>-
>-  explicit AudioEncoderCng(Config&& config);
>-  ~AudioEncoderCng() override;
>-
>-  int SampleRateHz() const override;
>-  size_t NumChannels() const override;
>-  int RtpTimestampRateHz() const override;
>-  size_t Num10MsFramesInNextPacket() const override;
>-  size_t Max10MsFramesInAPacket() const override;
>-  int GetTargetBitrate() const override;
>-  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
>-                         rtc::ArrayView<const int16_t> audio,
>-                         rtc::Buffer* encoded) override;
>-  void Reset() override;
>-  bool SetFec(bool enable) override;
>-  bool SetDtx(bool enable) override;
>-  bool SetApplication(Application application) override;
>-  void SetMaxPlaybackRate(int frequency_hz) override;
>-  rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
>-      override;
>-  void OnReceivedUplinkPacketLossFraction(
>-      float uplink_packet_loss_fraction) override;
>-  void OnReceivedUplinkRecoverablePacketLossFraction(
>-      float uplink_recoverable_packet_loss_fraction) override;
>-  void OnReceivedUplinkBandwidth(
>-      int target_audio_bitrate_bps,
>-      absl::optional<int64_t> bwe_period_ms) override;
>-
>- private:
>-  EncodedInfo EncodePassive(size_t frames_to_encode,
>-                            rtc::Buffer* encoded);
>-  EncodedInfo EncodeActive(size_t frames_to_encode,
>-                           rtc::Buffer* encoded);
>-  size_t SamplesPer10msFrame() const;
>-
>-  std::unique_ptr<AudioEncoder> speech_encoder_;
>-  const int cng_payload_type_;
>-  const int num_cng_coefficients_;
>-  const int sid_frame_interval_ms_;
>-  std::vector<int16_t> speech_buffer_;
>-  std::vector<uint32_t> rtp_timestamps_;
>-  bool last_frame_active_;
>-  std::unique_ptr<Vad> vad_;
>-  std::unique_ptr<ComfortNoiseEncoder> cng_encoder_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCng);
>+struct AudioEncoderCngConfig {
>+  // Moveable, not copyable.
>+  AudioEncoderCngConfig();
>+  AudioEncoderCngConfig(AudioEncoderCngConfig&&);
>+  ~AudioEncoderCngConfig();
>+
>+  bool IsOk() const;
>+
>+  size_t num_channels = 1;
>+  int payload_type = 13;
>+  std::unique_ptr<AudioEncoder> speech_encoder;
>+  Vad::Aggressiveness vad_mode = Vad::kVadNormal;
>+  int sid_frame_interval_ms = 100;
>+  int num_cng_coefficients = 8;
>+  // The Vad pointer is mainly for testing. If a NULL pointer is passed, the
>+  // AudioEncoderCng creates (and destroys) a Vad object internally. If an
>+  // object is passed, the AudioEncoderCng assumes ownership of the Vad
>+  // object.
>+  Vad* vad = nullptr;
> };
> 
>+std::unique_ptr<AudioEncoder> CreateComfortNoiseEncoder(
>+    AudioEncoderCngConfig&& config);
>+
> }  // namespace webrtc
> 
> #endif  // MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
>index a76dcbda2989f9e1f07c3a257852e6c1f9aeb980..e3655b464a5cef88a6dbe0630fd8ceb364371b27 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
>@@ -50,8 +50,8 @@ class AudioEncoderCngTest : public ::testing::Test {
>     cng_.reset();
>   }
> 
>-  AudioEncoderCng::Config MakeCngConfig() {
>-    AudioEncoderCng::Config config;
>+  AudioEncoderCngConfig MakeCngConfig() {
>+    AudioEncoderCngConfig config;
>     config.speech_encoder = std::move(mock_encoder_owner_);
>     EXPECT_TRUE(config.speech_encoder);
> 
>@@ -63,7 +63,7 @@ class AudioEncoderCngTest : public ::testing::Test {
>     return config;
>   }
> 
>-  void CreateCng(AudioEncoderCng::Config&& config) {
>+  void CreateCng(AudioEncoderCngConfig&& config) {
>     num_audio_samples_10ms_ = static_cast<size_t>(10 * sample_rate_hz_ / 1000);
>     ASSERT_LE(num_audio_samples_10ms_, kMaxNumSamples);
>     if (config.speech_encoder) {
>@@ -75,7 +75,7 @@ class AudioEncoderCngTest : public ::testing::Test {
>       EXPECT_CALL(*mock_encoder_, Max10MsFramesInAPacket())
>           .WillOnce(Return(1u));
>     }
>-    cng_.reset(new AudioEncoderCng(std::move(config)));
>+    cng_ = CreateComfortNoiseEncoder(std::move(config));
>   }
> 
>   void Encode() {
>@@ -193,7 +193,7 @@ class AudioEncoderCngTest : public ::testing::Test {
>     return encoded_info_.payload_type != kCngPayloadType;
>   }
> 
>-  std::unique_ptr<AudioEncoderCng> cng_;
>+  std::unique_ptr<AudioEncoder> cng_;
>   std::unique_ptr<MockAudioEncoder> mock_encoder_owner_;
>   MockAudioEncoder* mock_encoder_;
>   MockVad* mock_vad_;  // Ownership is transferred to |cng_|.
>@@ -432,7 +432,7 @@ class AudioEncoderCngDeathTest : public AudioEncoderCngTest {
>   // deleted.
>   void TearDown() override { cng_.reset(); }
> 
>-  AudioEncoderCng::Config MakeCngConfig() {
>+  AudioEncoderCngConfig MakeCngConfig() {
>     // Don't provide a Vad mock object, since it would leak when the test dies.
>     auto config = AudioEncoderCngTest::MakeCngConfig();
>     config.vad = nullptr;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
>index a07b093b3c7095bbdd4582726a63c9ed7f41aea7..f18fb28e98807b3012dbddc06206093dbe03a598 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
>@@ -13,6 +13,7 @@
> #include <algorithm>
> 
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
>index 3f8fadcd71b4dc7774d9b859e9c46745199bb7a6..6ff75298afa16a41b76e12eeebfd74739ab57302 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
> #define MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
> 
>+#include <stdint.h>
> #include <cstddef>
> 
> #include "api/array_view.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
>index 25f495f39d8187eba22c5ae1b2e04accafe9c5b8..d580a0509b4738d74b8f4dfd868f317fba13e0ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
> 
>+#include <utility>
>+
> #include "modules/audio_coding/codecs/g711/g711_interface.h"
> #include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
>index 29e4fa6593b93b7cbae26b26ce2132d31b64c050..9a01b8aff4ebf8d15c4cdeb1835e6d3408fd31ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
>@@ -11,7 +11,12 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
> #define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+#include <vector>
>+
> #include "api/audio_codecs/audio_decoder.h"
>+#include "rtc_base/buffer.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
>index c14287ed487e9dbb6f9ba13d23137e536fc82b14..dce16358ef3518fccc286ae21d04d92b92f4d64a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
>@@ -10,10 +10,9 @@
> 
> #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
> 
>-#include <algorithm>
>-#include <limits>
>+#include <cstdint>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "common_types.h"
> #include "modules/audio_coding/codecs/g711/g711_interface.h"
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
>index ea4a721b12052565d9819f000417b029cc6767c2..4de55a0bb50fdf7631efe7d029fb2b79486914f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
>@@ -11,6 +11,7 @@
> #include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
> 
> #include <string.h>
>+#include <utility>
> 
> #include "modules/audio_coding/codecs/g722/g722_interface.h"
> #include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
>index cb96c3c17ef81702649fc888ba3326de8d52f1d0..e63d5903090522985ad19fa4a64d02121359bb6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
>@@ -10,10 +10,9 @@
> 
> #include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
> 
>-#include <algorithm>
>+#include <cstdint>
> 
>-#include <limits>
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "common_types.h"
> #include "modules/audio_coding/codecs/g722/g722_interface.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
>index 9e58ce03ca474bb0c9cd2f0c623591a75ae41885..57b5abbe23df7fc2a2a197fa880a6c1d839a8c00 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
> 
>+#include <memory>
> #include <utility>
> 
> #include "modules/audio_coding/codecs/ilbc/ilbc.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
>index edb65d09c6d2fdacd78448b1d499a6814daf84bf..fcb2074dc95336018807da6d0c9dbe0c70085053 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
>@@ -11,7 +11,12 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+#include <vector>
>+
> #include "api/audio_codecs/audio_decoder.h"
>+#include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
> 
> typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>index dae956b30c0498ee040adad3d98b69755613a011..35e6fc64a62abac634a6f213a758e9628999d532 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>@@ -11,8 +11,9 @@
> #include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
> 
> #include <algorithm>
>-#include <limits>
>-#include "common_types.h"  // NOLINT(build/include)
>+#include <cstdint>
>+
>+#include "common_types.h"
> #include "modules/audio_coding/codecs/ilbc/ilbc.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt
>deleted file mode 100644
>index 99c6092116b7ff500ecf69cd398d74c81e9166bd..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt
>+++ /dev/null
>@@ -1,73 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
>-
>-#
>-# This script can be used to verify the bit exactness of iLBC fixed-point version 1.0.6
>-#
>-
>-INP=../../../../../../resources/audio_coding
>-EXEP=../../../../../../../out/Release
>-OUTP=./GeneratedFiles
>-mkdir ./GeneratedFiles
>-
>-$EXEP/iLBCtest 20 $INP/F00.INP $OUTP/F00.BIT20 $OUTP/F00.OUT20 $INP/clean.chn
>-$EXEP/iLBCtest 20 $INP/F01.INP $OUTP/F01.BIT20 $OUTP/F01.OUT20 $INP/clean.chn
>-$EXEP/iLBCtest 20 $INP/F02.INP $OUTP/F02.BIT20 $OUTP/F02.OUT20 $INP/clean.chn
>-$EXEP/iLBCtest 20 $INP/F03.INP $OUTP/F03.BIT20 $OUTP/F03.OUT20 $INP/clean.chn
>-$EXEP/iLBCtest 20 $INP/F04.INP $OUTP/F04.BIT20 $OUTP/F04.OUT20 $INP/clean.chn
>-$EXEP/iLBCtest 20 $INP/F05.INP $OUTP/F05.BIT20 $OUTP/F05.OUT20 $INP/clean.chn
>-$EXEP/iLBCtest 20 $INP/F06.INP $OUTP/F06.BIT20 $OUTP/F06.OUT20 $INP/clean.chn
>-
>-$EXEP/iLBCtest 30 $INP/F00.INP $OUTP/F00.BIT30 $OUTP/F00.OUT30 $INP/clean.chn
>-$EXEP/iLBCtest 30 $INP/F01.INP $OUTP/F01.BIT30 $OUTP/F01.OUT30 $INP/clean.chn
>-$EXEP/iLBCtest 30 $INP/F02.INP $OUTP/F02.BIT30 $OUTP/F02.OUT30 $INP/clean.chn
>-$EXEP/iLBCtest 30 $INP/F03.INP $OUTP/F03.BIT30 $OUTP/F03.OUT30 $INP/clean.chn
>-$EXEP/iLBCtest 30 $INP/F04.INP $OUTP/F04.BIT30 $OUTP/F04.OUT30 $INP/clean.chn
>-$EXEP/iLBCtest 30 $INP/F05.INP $OUTP/F05.BIT30 $OUTP/F05.OUT30 $INP/clean.chn
>-$EXEP/iLBCtest 30 $INP/F06.INP $OUTP/F06.BIT30 $OUTP/F06.OUT30 $INP/clean.chn
>-
>-$EXEP/iLBCtest 20 $INP/F00.INP $OUTP/F00.BIT20 $OUTP/F00_tlm10.OUT20 $INP/tlm10.chn
>-$EXEP/iLBCtest 20 $INP/F01.INP $OUTP/F01.BIT20 $OUTP/F01_tlm10.OUT20 $INP/tlm10.chn
>-$EXEP/iLBCtest 20 $INP/F02.INP $OUTP/F02.BIT20 $OUTP/F02_tlm10.OUT20 $INP/tlm10.chn
>-$EXEP/iLBCtest 30 $INP/F00.INP $OUTP/F00.BIT30 $OUTP/F00_tlm10.OUT30 $INP/tlm10.chn
>-$EXEP/iLBCtest 30 $INP/F01.INP $OUTP/F01.BIT30 $OUTP/F01_tlm10.OUT30 $INP/tlm10.chn
>-$EXEP/iLBCtest 30 $INP/F02.INP $OUTP/F02.BIT30 $OUTP/F02_tlm10.OUT30 $INP/tlm10.chn
>-
>-
>-diff $OUTP/F00.BIT20 $INP/F00.BIT20
>-diff $OUTP/F01.BIT20 $INP/F01.BIT20
>-diff $OUTP/F02.BIT20 $INP/F02.BIT20
>-diff $OUTP/F03.BIT20 $INP/F03.BIT20
>-diff $OUTP/F04.BIT20 $INP/F04.BIT20
>-diff $OUTP/F05.BIT20 $INP/F05.BIT20
>-diff $OUTP/F06.BIT20 $INP/F06.BIT20
>-diff $OUTP/F00.OUT20 $INP/F00.OUT20
>-diff $OUTP/F01.OUT20 $INP/F01.OUT20
>-diff $OUTP/F02.OUT20 $INP/F02.OUT20
>-diff $OUTP/F03.OUT20 $INP/F03.OUT20
>-diff $OUTP/F04.OUT20 $INP/F04.OUT20
>-diff $OUTP/F05.OUT20 $INP/F05.OUT20
>-diff $OUTP/F06.OUT20 $INP/F06.OUT20
>-
>-diff $OUTP/F00.BIT30 $INP/F00.BIT30
>-diff $OUTP/F01.BIT30 $INP/F01.BIT30
>-diff $OUTP/F02.BIT30 $INP/F02.BIT30
>-diff $OUTP/F03.BIT30 $INP/F03.BIT30
>-diff $OUTP/F04.BIT30 $INP/F04.BIT30
>-diff $OUTP/F05.BIT30 $INP/F05.BIT30
>-diff $OUTP/F06.BIT30 $INP/F06.BIT30
>-diff $OUTP/F00.OUT30 $INP/F00.OUT30
>-diff $OUTP/F01.OUT30 $INP/F01.OUT30
>-diff $OUTP/F02.OUT30 $INP/F02.OUT30
>-diff $OUTP/F03.OUT30 $INP/F03.OUT30
>-diff $OUTP/F04.OUT30 $INP/F04.OUT30
>-diff $OUTP/F05.OUT30 $INP/F05.OUT30
>-diff $OUTP/F06.OUT30 $INP/F06.OUT30
>-
>-diff $OUTP/F00_tlm10.OUT20 $INP/F00_tlm10.OUT20
>-diff $OUTP/F01_tlm10.OUT20 $INP/F01_tlm10.OUT20
>-diff $OUTP/F02_tlm10.OUT20 $INP/F02_tlm10.OUT20
>-diff $OUTP/F00_tlm10.OUT30 $INP/F00_tlm10.OUT30
>-diff $OUTP/F01_tlm10.OUT30 $INP/F01_tlm10.OUT30
>-diff $OUTP/F02_tlm10.OUT30 $INP/F02_tlm10.OUT30
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/ChannelFiles.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/ChannelFiles.txt
>deleted file mode 100644
>index 05f7410141e8399f2b6b5bbe13f262f5d2886e80..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/ChannelFiles.txt
>+++ /dev/null
>@@ -1,3 +0,0 @@
>-bottlenecks.txt
>-lowrates.txt
>-tworates.txt
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/InputFiles.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/InputFiles.txt
>deleted file mode 100644
>index f26b7afb6ca859a5209c6ccf3799d5803be42a31..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/InputFiles.txt
>+++ /dev/null
>@@ -1,31 +0,0 @@
>-DTMF_16kHz_long.pcm
>-DTMF_16kHz_short.pcm
>-F00.INP
>-F01.INP
>-F02.INP
>-F03.INP
>-F04.INP
>-F05.INP
>-F06.INP
>-longtest.pcm
>-ltest_speech_clean.pcm
>-ltest_music.pcm
>-ltest_speech_noisy.pcm
>-misc2.pcm
>-purenb.pcm
>-sawsweep_380_60.pcm
>-sinesweep.pcm
>-sinesweep_half.pcm
>-speechmusic.pcm
>-speechmusic_nb.pcm
>-speechoffice0dB.pcm
>-speech_and_misc_NB.pcm
>-speech_and_misc_WB.pcm
>-testM4.pcm
>-testM4D_rev.pcm  
>-testM4D.pcm  
>-testfile.pcm
>-tone_cisco.pcm
>-tone_cisco_long.pcm
>-wb_contspeech.pcm
>-wb_speech_office25db.pcm
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/InputFilesFew.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/InputFilesFew.txt
>deleted file mode 100644
>index 08bbde30d7337b2836dc53a0053a3a309ce2c490..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/InputFilesFew.txt
>+++ /dev/null
>@@ -1,6 +0,0 @@
>-DTMF_16kHz_short.pcm
>-ltest_speech_noisy.pcm
>-misc2.pcm
>-sinesweep.pcm
>-speechmusic.pcm
>-tone_cisco.pcm
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/ListOfTestCases.xls b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/ListOfTestCases.xls
>deleted file mode 100644
>index f0889ef4eddb53b5295e4248059e3e345223f25b..0000000000000000000000000000000000000000
>GIT binary patch
>literal 0
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>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/diffiSAC.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/diffiSAC.txt
>deleted file mode 100644
>index 96b87c066b3ac7a5bad928d6606e36bccea8e95b..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/diffiSAC.txt
>+++ /dev/null
>@@ -1,481 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
>-
>-diff ../dataqa350/i30_1DTMF_16kHz_long.pcm ../dataqa351/i30_1DTMF_16kHz_long.pcm
>-diff ../dataqa350/i60_1DTMF_16kHz_long.pcm ../dataqa351/i60_1DTMF_16kHz_long.pcm
>-diff ../dataqa350/i30_2DTMF_16kHz_long.pcm ../dataqa351/i30_2DTMF_16kHz_long.pcm
>-diff ../dataqa350/i60_2DTMF_16kHz_long.pcm ../dataqa351/i60_2DTMF_16kHz_long.pcm
>-diff ../dataqa350/i30_3DTMF_16kHz_long.pcm ../dataqa351/i30_3DTMF_16kHz_long.pcm
>-diff ../dataqa350/i60_3DTMF_16kHz_long.pcm ../dataqa351/i60_3DTMF_16kHz_long.pcm
>-diff ../dataqa350/i30_4DTMF_16kHz_long.pcm ../dataqa351/i30_4DTMF_16kHz_long.pcm
>-diff ../dataqa350/i60_4DTMF_16kHz_long.pcm ../dataqa351/i60_4DTMF_16kHz_long.pcm
>-diff ../dataqa350/i30_5DTMF_16kHz_long.pcm ../dataqa351/i30_5DTMF_16kHz_long.pcm
>-diff ../dataqa350/i60_5DTMF_16kHz_long.pcm ../dataqa351/i60_5DTMF_16kHz_long.pcm
>-diff ../dataqa350/i30_6DTMF_16kHz_long.pcm ../dataqa351/i30_6DTMF_16kHz_long.pcm
>-diff ../dataqa350/i60_6DTMF_16kHz_long.pcm ../dataqa351/i60_6DTMF_16kHz_long.pcm
>-diff ../dataqa350/a1DTMF_16kHz_long.pcm ../dataqa351/a1DTMF_16kHz_long.pcm
>-diff ../dataqa350/a2DTMF_16kHz_long.pcm ../dataqa351/a2DTMF_16kHz_long.pcm
>-diff ../dataqa350/a3DTMF_16kHz_long.pcm ../dataqa351/a3DTMF_16kHz_long.pcm
>-diff ../dataqa350/i30_7DTMF_16kHz_short.pcm ../dataqa351/i30_7DTMF_16kHz_short.pcm
>-diff ../dataqa350/i60_7DTMF_16kHz_short.pcm ../dataqa351/i60_7DTMF_16kHz_short.pcm
>-diff ../dataqa350/i30_8DTMF_16kHz_short.pcm ../dataqa351/i30_8DTMF_16kHz_short.pcm
>-diff ../dataqa350/i60_8DTMF_16kHz_short.pcm ../dataqa351/i60_8DTMF_16kHz_short.pcm
>-diff ../dataqa350/i30_9DTMF_16kHz_short.pcm ../dataqa351/i30_9DTMF_16kHz_short.pcm
>-diff ../dataqa350/i60_9DTMF_16kHz_short.pcm ../dataqa351/i60_9DTMF_16kHz_short.pcm
>-diff ../dataqa350/i30_10DTMF_16kHz_short.pcm ../dataqa351/i30_10DTMF_16kHz_short.pcm
>-diff ../dataqa350/i60_10DTMF_16kHz_short.pcm ../dataqa351/i60_10DTMF_16kHz_short.pcm
>-diff ../dataqa350/i30_11DTMF_16kHz_short.pcm ../dataqa351/i30_11DTMF_16kHz_short.pcm
>-diff ../dataqa350/i60_11DTMF_16kHz_short.pcm ../dataqa351/i60_11DTMF_16kHz_short.pcm
>-diff ../dataqa350/i30_12DTMF_16kHz_short.pcm ../dataqa351/i30_12DTMF_16kHz_short.pcm
>-diff ../dataqa350/i60_12DTMF_16kHz_short.pcm ../dataqa351/i60_12DTMF_16kHz_short.pcm
>-diff ../dataqa350/a4DTMF_16kHz_short.pcm ../dataqa351/a4DTMF_16kHz_short.pcm
>-diff ../dataqa350/a5DTMF_16kHz_short.pcm ../dataqa351/a5DTMF_16kHz_short.pcm
>-diff ../dataqa350/a6DTMF_16kHz_short.pcm ../dataqa351/a6DTMF_16kHz_short.pcm
>-diff ../dataqa350/i30_13F00.INP ../dataqa350/i30_13F00.INP
>-diff ../dataqa350/i60_13F00.INP ../dataqa350/i60_13F00.INP
>-diff ../dataqa350/i30_14F00.INP ../dataqa350/i30_14F00.INP
>-diff ../dataqa350/i60_14F00.INP ../dataqa350/i60_14F00.INP
>-diff ../dataqa350/i30_15F00.INP ../dataqa350/i30_15F00.INP
>-diff ../dataqa350/i60_15F00.INP ../dataqa350/i60_15F00.INP
>-diff ../dataqa350/i30_16F00.INP ../dataqa350/i30_16F00.INP
>-diff ../dataqa350/i60_16F00.INP ../dataqa350/i60_16F00.INP
>-diff ../dataqa350/i30_17F00.INP ../dataqa350/i30_17F00.INP
>-diff ../dataqa350/i60_17F00.INP ../dataqa350/i60_17F00.INP
>-diff ../dataqa350/i30_18F00.INP ../dataqa350/i30_18F00.INP
>-diff ../dataqa350/i60_18F00.INP ../dataqa350/i60_18F00.INP
>-diff ../dataqa350/a7F00.INP ../dataqa350/a7F00.INP
>-diff ../dataqa350/a8F00.INP ../dataqa350/a8F00.INP
>-diff ../dataqa350/a9F00.INP ../dataqa350/a9F00.INP
>-diff ../dataqa350/i30_19F01.INP ../dataqa350/i30_19F01.INP
>-diff ../dataqa350/i60_19F01.INP ../dataqa350/i60_19F01.INP
>-diff ../dataqa350/i30_20F01.INP ../dataqa350/i30_20F01.INP
>-diff ../dataqa350/i60_20F01.INP ../dataqa350/i60_20F01.INP
>-diff ../dataqa350/i30_21F01.INP ../dataqa350/i30_21F01.INP
>-diff ../dataqa350/i60_21F01.INP ../dataqa350/i60_21F01.INP
>-diff ../dataqa350/i30_22F01.INP ../dataqa350/i30_22F01.INP
>-diff ../dataqa350/i60_22F01.INP ../dataqa350/i60_22F01.INP
>-diff ../dataqa350/i30_23F01.INP ../dataqa350/i30_23F01.INP
>-diff ../dataqa350/i60_23F01.INP ../dataqa350/i60_23F01.INP
>-diff ../dataqa350/i30_24F01.INP ../dataqa350/i30_24F01.INP
>-diff ../dataqa350/i60_24F01.INP ../dataqa350/i60_24F01.INP
>-diff ../dataqa350/a10F01.INP ../dataqa350/a10F01.INP
>-diff ../dataqa350/a11F01.INP ../dataqa350/a11F01.INP
>-diff ../dataqa350/a12F01.INP ../dataqa350/a12F01.INP
>-diff ../dataqa350/i30_25F02.INP ../dataqa350/i30_25F02.INP
>-diff ../dataqa350/i60_25F02.INP ../dataqa350/i60_25F02.INP
>-diff ../dataqa350/i30_26F02.INP ../dataqa350/i30_26F02.INP
>-diff ../dataqa350/i60_26F02.INP ../dataqa350/i60_26F02.INP
>-diff ../dataqa350/i30_27F02.INP ../dataqa350/i30_27F02.INP
>-diff ../dataqa350/i60_27F02.INP ../dataqa350/i60_27F02.INP
>-diff ../dataqa350/i30_28F02.INP ../dataqa350/i30_28F02.INP
>-diff ../dataqa350/i60_28F02.INP ../dataqa350/i60_28F02.INP
>-diff ../dataqa350/i30_29F02.INP ../dataqa350/i30_29F02.INP
>-diff ../dataqa350/i60_29F02.INP ../dataqa350/i60_29F02.INP
>-diff ../dataqa350/i30_30F02.INP ../dataqa350/i30_30F02.INP
>-diff ../dataqa350/i60_30F02.INP ../dataqa350/i60_30F02.INP
>-diff ../dataqa350/a13F02.INP ../dataqa350/a13F02.INP
>-diff ../dataqa350/a14F02.INP ../dataqa350/a14F02.INP
>-diff ../dataqa350/a15F02.INP ../dataqa350/a15F02.INP
>-diff ../dataqa350/i30_31F03.INP ../dataqa350/i30_31F03.INP
>-diff ../dataqa350/i60_31F03.INP ../dataqa350/i60_31F03.INP
>-diff ../dataqa350/i30_32F03.INP ../dataqa350/i30_32F03.INP
>-diff ../dataqa350/i60_32F03.INP ../dataqa350/i60_32F03.INP
>-diff ../dataqa350/i30_33F03.INP ../dataqa350/i30_33F03.INP
>-diff ../dataqa350/i60_33F03.INP ../dataqa350/i60_33F03.INP
>-diff ../dataqa350/i30_34F03.INP ../dataqa350/i30_34F03.INP
>-diff ../dataqa350/i60_34F03.INP ../dataqa350/i60_34F03.INP
>-diff ../dataqa350/i30_35F03.INP ../dataqa350/i30_35F03.INP
>-diff ../dataqa350/i60_35F03.INP ../dataqa350/i60_35F03.INP
>-diff ../dataqa350/i30_36F03.INP ../dataqa350/i30_36F03.INP
>-diff ../dataqa350/i60_36F03.INP ../dataqa350/i60_36F03.INP
>-diff ../dataqa350/a16F03.INP ../dataqa350/a16F03.INP
>-diff ../dataqa350/a17F03.INP ../dataqa350/a17F03.INP
>-diff ../dataqa350/a18F03.INP ../dataqa350/a18F03.INP
>-diff ../dataqa350/i30_37F04.INP ../dataqa350/i30_37F04.INP
>-diff ../dataqa350/i60_37F04.INP ../dataqa350/i60_37F04.INP
>-diff ../dataqa350/i30_38F04.INP ../dataqa350/i30_38F04.INP
>-diff ../dataqa350/i60_38F04.INP ../dataqa350/i60_38F04.INP
>-diff ../dataqa350/i30_39F04.INP ../dataqa350/i30_39F04.INP
>-diff ../dataqa350/i60_39F04.INP ../dataqa350/i60_39F04.INP
>-diff ../dataqa350/i30_40F04.INP ../dataqa350/i30_40F04.INP
>-diff ../dataqa350/i60_40F04.INP ../dataqa350/i60_40F04.INP
>-diff ../dataqa350/i30_41F04.INP ../dataqa350/i30_41F04.INP
>-diff ../dataqa350/i60_41F04.INP ../dataqa350/i60_41F04.INP
>-diff ../dataqa350/i30_42F04.INP ../dataqa350/i30_42F04.INP
>-diff ../dataqa350/i60_42F04.INP ../dataqa350/i60_42F04.INP
>-diff ../dataqa350/a19F04.INP ../dataqa350/a19F04.INP
>-diff ../dataqa350/a20F04.INP ../dataqa350/a20F04.INP
>-diff ../dataqa350/a21F04.INP ../dataqa350/a21F04.INP
>-diff ../dataqa350/i30_43F05.INP ../dataqa350/i30_43F05.INP
>-diff ../dataqa350/i60_43F05.INP ../dataqa350/i60_43F05.INP
>-diff ../dataqa350/i30_44F05.INP ../dataqa350/i30_44F05.INP
>-diff ../dataqa350/i60_44F05.INP ../dataqa350/i60_44F05.INP
>-diff ../dataqa350/i30_45F05.INP ../dataqa350/i30_45F05.INP
>-diff ../dataqa350/i60_45F05.INP ../dataqa350/i60_45F05.INP
>-diff ../dataqa350/i30_46F05.INP ../dataqa350/i30_46F05.INP
>-diff ../dataqa350/i60_46F05.INP ../dataqa350/i60_46F05.INP
>-diff ../dataqa350/i30_47F05.INP ../dataqa350/i30_47F05.INP
>-diff ../dataqa350/i60_47F05.INP ../dataqa350/i60_47F05.INP
>-diff ../dataqa350/i30_48F05.INP ../dataqa350/i30_48F05.INP
>-diff ../dataqa350/i60_48F05.INP ../dataqa350/i60_48F05.INP
>-diff ../dataqa350/a22F05.INP ../dataqa350/a22F05.INP
>-diff ../dataqa350/a23F05.INP ../dataqa350/a23F05.INP
>-diff ../dataqa350/a24F05.INP ../dataqa350/a24F05.INP
>-diff ../dataqa350/i30_49F06.INP ../dataqa350/i30_49F06.INP
>-diff ../dataqa350/i60_49F06.INP ../dataqa350/i60_49F06.INP
>-diff ../dataqa350/i30_50F06.INP ../dataqa350/i30_50F06.INP
>-diff ../dataqa350/i60_50F06.INP ../dataqa350/i60_50F06.INP
>-diff ../dataqa350/i30_51F06.INP ../dataqa350/i30_51F06.INP
>-diff ../dataqa350/i60_51F06.INP ../dataqa350/i60_51F06.INP
>-diff ../dataqa350/i30_52F06.INP ../dataqa350/i30_52F06.INP
>-diff ../dataqa350/i60_52F06.INP ../dataqa350/i60_52F06.INP
>-diff ../dataqa350/i30_53F06.INP ../dataqa350/i30_53F06.INP
>-diff ../dataqa350/i60_53F06.INP ../dataqa350/i60_53F06.INP
>-diff ../dataqa350/i30_54F06.INP ../dataqa350/i30_54F06.INP
>-diff ../dataqa350/i60_54F06.INP ../dataqa350/i60_54F06.INP
>-diff ../dataqa350/a25F06.INP ../dataqa350/a25F06.INP
>-diff ../dataqa350/a26F06.INP ../dataqa350/a26F06.INP
>-diff ../dataqa350/a27F06.INP ../dataqa350/a27F06.INP
>-diff ../dataqa350/i30_55longtest.pcm ../dataqa351/i30_55longtest.pcm
>-diff ../dataqa350/i60_55longtest.pcm ../dataqa351/i60_55longtest.pcm
>-diff ../dataqa350/i30_56longtest.pcm ../dataqa351/i30_56longtest.pcm
>-diff ../dataqa350/i60_56longtest.pcm ../dataqa351/i60_56longtest.pcm
>-diff ../dataqa350/i30_57longtest.pcm ../dataqa351/i30_57longtest.pcm
>-diff ../dataqa350/i60_57longtest.pcm ../dataqa351/i60_57longtest.pcm
>-diff ../dataqa350/i30_58longtest.pcm ../dataqa351/i30_58longtest.pcm
>-diff ../dataqa350/i60_58longtest.pcm ../dataqa351/i60_58longtest.pcm
>-diff ../dataqa350/i30_59longtest.pcm ../dataqa351/i30_59longtest.pcm
>-diff ../dataqa350/i60_59longtest.pcm ../dataqa351/i60_59longtest.pcm
>-diff ../dataqa350/i30_60longtest.pcm ../dataqa351/i30_60longtest.pcm
>-diff ../dataqa350/i60_60longtest.pcm ../dataqa351/i60_60longtest.pcm
>-diff ../dataqa350/a28longtest.pcm ../dataqa351/a28longtest.pcm
>-diff ../dataqa350/a29longtest.pcm ../dataqa351/a29longtest.pcm
>-diff ../dataqa350/a30longtest.pcm ../dataqa351/a30longtest.pcm
>-diff ../dataqa350/i30_61ltest_speech_clean.pcm ../dataqa351/i30_61ltest_speech_clean.pcm
>-diff ../dataqa350/i60_61ltest_speech_clean.pcm ../dataqa351/i60_61ltest_speech_clean.pcm
>-diff ../dataqa350/i30_62ltest_speech_clean.pcm ../dataqa351/i30_62ltest_speech_clean.pcm
>-diff ../dataqa350/i60_62ltest_speech_clean.pcm ../dataqa351/i60_62ltest_speech_clean.pcm
>-diff ../dataqa350/i30_63ltest_speech_clean.pcm ../dataqa351/i30_63ltest_speech_clean.pcm
>-diff ../dataqa350/i60_63ltest_speech_clean.pcm ../dataqa351/i60_63ltest_speech_clean.pcm
>-diff ../dataqa350/i30_64ltest_speech_clean.pcm ../dataqa351/i30_64ltest_speech_clean.pcm
>-diff ../dataqa350/i60_64ltest_speech_clean.pcm ../dataqa351/i60_64ltest_speech_clean.pcm
>-diff ../dataqa350/i30_65ltest_speech_clean.pcm ../dataqa351/i30_65ltest_speech_clean.pcm
>-diff ../dataqa350/i60_65ltest_speech_clean.pcm ../dataqa351/i60_65ltest_speech_clean.pcm
>-diff ../dataqa350/i30_66ltest_speech_clean.pcm ../dataqa351/i30_66ltest_speech_clean.pcm
>-diff ../dataqa350/i60_66ltest_speech_clean.pcm ../dataqa351/i60_66ltest_speech_clean.pcm
>-diff ../dataqa350/a31ltest_speech_clean.pcm ../dataqa351/a31ltest_speech_clean.pcm
>-diff ../dataqa350/a32ltest_speech_clean.pcm ../dataqa351/a32ltest_speech_clean.pcm
>-diff ../dataqa350/a33ltest_speech_clean.pcm ../dataqa351/a33ltest_speech_clean.pcm
>-diff ../dataqa350/i30_67ltest_music.pcm ../dataqa351/i30_67ltest_music.pcm
>-diff ../dataqa350/i60_67ltest_music.pcm ../dataqa351/i60_67ltest_music.pcm
>-diff ../dataqa350/i30_68ltest_music.pcm ../dataqa351/i30_68ltest_music.pcm
>-diff ../dataqa350/i60_68ltest_music.pcm ../dataqa351/i60_68ltest_music.pcm
>-diff ../dataqa350/i30_69ltest_music.pcm ../dataqa351/i30_69ltest_music.pcm
>-diff ../dataqa350/i60_69ltest_music.pcm ../dataqa351/i60_69ltest_music.pcm
>-diff ../dataqa350/i30_70ltest_music.pcm ../dataqa351/i30_70ltest_music.pcm
>-diff ../dataqa350/i60_70ltest_music.pcm ../dataqa351/i60_70ltest_music.pcm
>-diff ../dataqa350/i30_71ltest_music.pcm ../dataqa351/i30_71ltest_music.pcm
>-diff ../dataqa350/i60_71ltest_music.pcm ../dataqa351/i60_71ltest_music.pcm
>-diff ../dataqa350/i30_72ltest_music.pcm ../dataqa351/i30_72ltest_music.pcm
>-diff ../dataqa350/i60_72ltest_music.pcm ../dataqa351/i60_72ltest_music.pcm
>-diff ../dataqa350/a34ltest_music.pcm ../dataqa351/a34ltest_music.pcm
>-diff ../dataqa350/a35ltest_music.pcm ../dataqa351/a35ltest_music.pcm
>-diff ../dataqa350/a36ltest_music.pcm ../dataqa351/a36ltest_music.pcm
>-diff ../dataqa350/i30_73ltest_speech_noisy.pcm ../dataqa351/i30_73ltest_speech_noisy.pcm
>-diff ../dataqa350/i60_73ltest_speech_noisy.pcm ../dataqa351/i60_73ltest_speech_noisy.pcm
>-diff ../dataqa350/i30_74ltest_speech_noisy.pcm ../dataqa351/i30_74ltest_speech_noisy.pcm
>-diff ../dataqa350/i60_74ltest_speech_noisy.pcm ../dataqa351/i60_74ltest_speech_noisy.pcm
>-diff ../dataqa350/i30_75ltest_speech_noisy.pcm ../dataqa351/i30_75ltest_speech_noisy.pcm
>-diff ../dataqa350/i60_75ltest_speech_noisy.pcm ../dataqa351/i60_75ltest_speech_noisy.pcm
>-diff ../dataqa350/i30_76ltest_speech_noisy.pcm ../dataqa351/i30_76ltest_speech_noisy.pcm
>-diff ../dataqa350/i60_76ltest_speech_noisy.pcm ../dataqa351/i60_76ltest_speech_noisy.pcm
>-diff ../dataqa350/i30_77ltest_speech_noisy.pcm ../dataqa351/i30_77ltest_speech_noisy.pcm
>-diff ../dataqa350/i60_77ltest_speech_noisy.pcm ../dataqa351/i60_77ltest_speech_noisy.pcm
>-diff ../dataqa350/i30_78ltest_speech_noisy.pcm ../dataqa351/i30_78ltest_speech_noisy.pcm
>-diff ../dataqa350/i60_78ltest_speech_noisy.pcm ../dataqa351/i60_78ltest_speech_noisy.pcm
>-diff ../dataqa350/a37ltest_speech_noisy.pcm ../dataqa351/a37ltest_speech_noisy.pcm
>-diff ../dataqa350/a38ltest_speech_noisy.pcm ../dataqa351/a38ltest_speech_noisy.pcm
>-diff ../dataqa350/a39ltest_speech_noisy.pcm ../dataqa351/a39ltest_speech_noisy.pcm
>-diff ../dataqa350/i30_79misc2.pcm ../dataqa351/i30_79misc2.pcm
>-diff ../dataqa350/i60_79misc2.pcm ../dataqa351/i60_79misc2.pcm
>-diff ../dataqa350/i30_80misc2.pcm ../dataqa351/i30_80misc2.pcm
>-diff ../dataqa350/i60_80misc2.pcm ../dataqa351/i60_80misc2.pcm
>-diff ../dataqa350/i30_81misc2.pcm ../dataqa351/i30_81misc2.pcm
>-diff ../dataqa350/i60_81misc2.pcm ../dataqa351/i60_81misc2.pcm
>-diff ../dataqa350/i30_82misc2.pcm ../dataqa351/i30_82misc2.pcm
>-diff ../dataqa350/i60_82misc2.pcm ../dataqa351/i60_82misc2.pcm
>-diff ../dataqa350/i30_83misc2.pcm ../dataqa351/i30_83misc2.pcm
>-diff ../dataqa350/i60_83misc2.pcm ../dataqa351/i60_83misc2.pcm
>-diff ../dataqa350/i30_84misc2.pcm ../dataqa351/i30_84misc2.pcm
>-diff ../dataqa350/i60_84misc2.pcm ../dataqa351/i60_84misc2.pcm
>-diff ../dataqa350/a40misc2.pcm ../dataqa351/a40misc2.pcm
>-diff ../dataqa350/a41misc2.pcm ../dataqa351/a41misc2.pcm
>-diff ../dataqa350/a42misc2.pcm ../dataqa351/a42misc2.pcm
>-diff ../dataqa350/i30_85purenb.pcm ../dataqa351/i30_85purenb.pcm
>-diff ../dataqa350/i60_85purenb.pcm ../dataqa351/i60_85purenb.pcm
>-diff ../dataqa350/i30_86purenb.pcm ../dataqa351/i30_86purenb.pcm
>-diff ../dataqa350/i60_86purenb.pcm ../dataqa351/i60_86purenb.pcm
>-diff ../dataqa350/i30_87purenb.pcm ../dataqa351/i30_87purenb.pcm
>-diff ../dataqa350/i60_87purenb.pcm ../dataqa351/i60_87purenb.pcm
>-diff ../dataqa350/i30_88purenb.pcm ../dataqa351/i30_88purenb.pcm
>-diff ../dataqa350/i60_88purenb.pcm ../dataqa351/i60_88purenb.pcm
>-diff ../dataqa350/i30_89purenb.pcm ../dataqa351/i30_89purenb.pcm
>-diff ../dataqa350/i60_89purenb.pcm ../dataqa351/i60_89purenb.pcm
>-diff ../dataqa350/i30_90purenb.pcm ../dataqa351/i30_90purenb.pcm
>-diff ../dataqa350/i60_90purenb.pcm ../dataqa351/i60_90purenb.pcm
>-diff ../dataqa350/a43purenb.pcm ../dataqa351/a43purenb.pcm
>-diff ../dataqa350/a44purenb.pcm ../dataqa351/a44purenb.pcm
>-diff ../dataqa350/a45purenb.pcm ../dataqa351/a45purenb.pcm
>-diff ../dataqa350/i30_91sawsweep_380_60.pcm ../dataqa351/i30_91sawsweep_380_60.pcm
>-diff ../dataqa350/i60_91sawsweep_380_60.pcm ../dataqa351/i60_91sawsweep_380_60.pcm
>-diff ../dataqa350/i30_92sawsweep_380_60.pcm ../dataqa351/i30_92sawsweep_380_60.pcm
>-diff ../dataqa350/i60_92sawsweep_380_60.pcm ../dataqa351/i60_92sawsweep_380_60.pcm
>-diff ../dataqa350/i30_93sawsweep_380_60.pcm ../dataqa351/i30_93sawsweep_380_60.pcm
>-diff ../dataqa350/i60_93sawsweep_380_60.pcm ../dataqa351/i60_93sawsweep_380_60.pcm
>-diff ../dataqa350/i30_94sawsweep_380_60.pcm ../dataqa351/i30_94sawsweep_380_60.pcm
>-diff ../dataqa350/i60_94sawsweep_380_60.pcm ../dataqa351/i60_94sawsweep_380_60.pcm
>-diff ../dataqa350/i30_95sawsweep_380_60.pcm ../dataqa351/i30_95sawsweep_380_60.pcm
>-diff ../dataqa350/i60_95sawsweep_380_60.pcm ../dataqa351/i60_95sawsweep_380_60.pcm
>-diff ../dataqa350/i30_96sawsweep_380_60.pcm ../dataqa351/i30_96sawsweep_380_60.pcm
>-diff ../dataqa350/i60_96sawsweep_380_60.pcm ../dataqa351/i60_96sawsweep_380_60.pcm
>-diff ../dataqa350/a46sawsweep_380_60.pcm ../dataqa351/a46sawsweep_380_60.pcm
>-diff ../dataqa350/a47sawsweep_380_60.pcm ../dataqa351/a47sawsweep_380_60.pcm
>-diff ../dataqa350/a48sawsweep_380_60.pcm ../dataqa351/a48sawsweep_380_60.pcm
>-diff ../dataqa350/i30_97sinesweep.pcm ../dataqa351/i30_97sinesweep.pcm
>-diff ../dataqa350/i60_97sinesweep.pcm ../dataqa351/i60_97sinesweep.pcm
>-diff ../dataqa350/i30_98sinesweep.pcm ../dataqa351/i30_98sinesweep.pcm
>-diff ../dataqa350/i60_98sinesweep.pcm ../dataqa351/i60_98sinesweep.pcm
>-diff ../dataqa350/i30_99sinesweep.pcm ../dataqa351/i30_99sinesweep.pcm
>-diff ../dataqa350/i60_99sinesweep.pcm ../dataqa351/i60_99sinesweep.pcm
>-diff ../dataqa350/i30_100sinesweep.pcm ../dataqa351/i30_100sinesweep.pcm
>-diff ../dataqa350/i60_100sinesweep.pcm ../dataqa351/i60_100sinesweep.pcm
>-diff ../dataqa350/i30_101sinesweep.pcm ../dataqa351/i30_101sinesweep.pcm
>-diff ../dataqa350/i60_101sinesweep.pcm ../dataqa351/i60_101sinesweep.pcm
>-diff ../dataqa350/i30_102sinesweep.pcm ../dataqa351/i30_102sinesweep.pcm
>-diff ../dataqa350/i60_102sinesweep.pcm ../dataqa351/i60_102sinesweep.pcm
>-diff ../dataqa350/a49sinesweep.pcm ../dataqa351/a49sinesweep.pcm
>-diff ../dataqa350/a50sinesweep.pcm ../dataqa351/a50sinesweep.pcm
>-diff ../dataqa350/a51sinesweep.pcm ../dataqa351/a51sinesweep.pcm
>-diff ../dataqa350/i30_103sinesweep_half.pcm ../dataqa351/i30_103sinesweep_half.pcm
>-diff ../dataqa350/i60_103sinesweep_half.pcm ../dataqa351/i60_103sinesweep_half.pcm
>-diff ../dataqa350/i30_104sinesweep_half.pcm ../dataqa351/i30_104sinesweep_half.pcm
>-diff ../dataqa350/i60_104sinesweep_half.pcm ../dataqa351/i60_104sinesweep_half.pcm
>-diff ../dataqa350/i30_105sinesweep_half.pcm ../dataqa351/i30_105sinesweep_half.pcm
>-diff ../dataqa350/i60_105sinesweep_half.pcm ../dataqa351/i60_105sinesweep_half.pcm
>-diff ../dataqa350/i30_106sinesweep_half.pcm ../dataqa351/i30_106sinesweep_half.pcm
>-diff ../dataqa350/i60_106sinesweep_half.pcm ../dataqa351/i60_106sinesweep_half.pcm
>-diff ../dataqa350/i30_107sinesweep_half.pcm ../dataqa351/i30_107sinesweep_half.pcm
>-diff ../dataqa350/i60_107sinesweep_half.pcm ../dataqa351/i60_107sinesweep_half.pcm
>-diff ../dataqa350/i30_108sinesweep_half.pcm ../dataqa351/i30_108sinesweep_half.pcm
>-diff ../dataqa350/i60_108sinesweep_half.pcm ../dataqa351/i60_108sinesweep_half.pcm
>-diff ../dataqa350/a52sinesweep_half.pcm ../dataqa351/a52sinesweep_half.pcm
>-diff ../dataqa350/a53sinesweep_half.pcm ../dataqa351/a53sinesweep_half.pcm
>-diff ../dataqa350/a54sinesweep_half.pcm ../dataqa351/a54sinesweep_half.pcm
>-diff ../dataqa350/i30_109speechmusic.pcm ../dataqa351/i30_109speechmusic.pcm
>-diff ../dataqa350/i60_109speechmusic.pcm ../dataqa351/i60_109speechmusic.pcm
>-diff ../dataqa350/i30_110speechmusic.pcm ../dataqa351/i30_110speechmusic.pcm
>-diff ../dataqa350/i60_110speechmusic.pcm ../dataqa351/i60_110speechmusic.pcm
>-diff ../dataqa350/i30_111speechmusic.pcm ../dataqa351/i30_111speechmusic.pcm
>-diff ../dataqa350/i60_111speechmusic.pcm ../dataqa351/i60_111speechmusic.pcm
>-diff ../dataqa350/i30_112speechmusic.pcm ../dataqa351/i30_112speechmusic.pcm
>-diff ../dataqa350/i60_112speechmusic.pcm ../dataqa351/i60_112speechmusic.pcm
>-diff ../dataqa350/i30_113speechmusic.pcm ../dataqa351/i30_113speechmusic.pcm
>-diff ../dataqa350/i60_113speechmusic.pcm ../dataqa351/i60_113speechmusic.pcm
>-diff ../dataqa350/i30_114speechmusic.pcm ../dataqa351/i30_114speechmusic.pcm
>-diff ../dataqa350/i60_114speechmusic.pcm ../dataqa351/i60_114speechmusic.pcm
>-diff ../dataqa350/a55speechmusic.pcm ../dataqa351/a55speechmusic.pcm
>-diff ../dataqa350/a56speechmusic.pcm ../dataqa351/a56speechmusic.pcm
>-diff ../dataqa350/a57speechmusic.pcm ../dataqa351/a57speechmusic.pcm
>-diff ../dataqa350/i30_115speechmusic_nb.pcm ../dataqa351/i30_115speechmusic_nb.pcm
>-diff ../dataqa350/i60_115speechmusic_nb.pcm ../dataqa351/i60_115speechmusic_nb.pcm
>-diff ../dataqa350/i30_116speechmusic_nb.pcm ../dataqa351/i30_116speechmusic_nb.pcm
>-diff ../dataqa350/i60_116speechmusic_nb.pcm ../dataqa351/i60_116speechmusic_nb.pcm
>-diff ../dataqa350/i30_117speechmusic_nb.pcm ../dataqa351/i30_117speechmusic_nb.pcm
>-diff ../dataqa350/i60_117speechmusic_nb.pcm ../dataqa351/i60_117speechmusic_nb.pcm
>-diff ../dataqa350/i30_118speechmusic_nb.pcm ../dataqa351/i30_118speechmusic_nb.pcm
>-diff ../dataqa350/i60_118speechmusic_nb.pcm ../dataqa351/i60_118speechmusic_nb.pcm
>-diff ../dataqa350/i30_119speechmusic_nb.pcm ../dataqa351/i30_119speechmusic_nb.pcm
>-diff ../dataqa350/i60_119speechmusic_nb.pcm ../dataqa351/i60_119speechmusic_nb.pcm
>-diff ../dataqa350/i30_120speechmusic_nb.pcm ../dataqa351/i30_120speechmusic_nb.pcm
>-diff ../dataqa350/i60_120speechmusic_nb.pcm ../dataqa351/i60_120speechmusic_nb.pcm
>-diff ../dataqa350/a58speechmusic_nb.pcm ../dataqa351/a58speechmusic_nb.pcm
>-diff ../dataqa350/a59speechmusic_nb.pcm ../dataqa351/a59speechmusic_nb.pcm
>-diff ../dataqa350/a60speechmusic_nb.pcm ../dataqa351/a60speechmusic_nb.pcm
>-diff ../dataqa350/i30_121speechoffice0dB.pcm ../dataqa351/i30_121speechoffice0dB.pcm
>-diff ../dataqa350/i60_121speechoffice0dB.pcm ../dataqa351/i60_121speechoffice0dB.pcm
>-diff ../dataqa350/i30_122speechoffice0dB.pcm ../dataqa351/i30_122speechoffice0dB.pcm
>-diff ../dataqa350/i60_122speechoffice0dB.pcm ../dataqa351/i60_122speechoffice0dB.pcm
>-diff ../dataqa350/i30_123speechoffice0dB.pcm ../dataqa351/i30_123speechoffice0dB.pcm
>-diff ../dataqa350/i60_123speechoffice0dB.pcm ../dataqa351/i60_123speechoffice0dB.pcm
>-diff ../dataqa350/i30_124speechoffice0dB.pcm ../dataqa351/i30_124speechoffice0dB.pcm
>-diff ../dataqa350/i60_124speechoffice0dB.pcm ../dataqa351/i60_124speechoffice0dB.pcm
>-diff ../dataqa350/i30_125speechoffice0dB.pcm ../dataqa351/i30_125speechoffice0dB.pcm
>-diff ../dataqa350/i60_125speechoffice0dB.pcm ../dataqa351/i60_125speechoffice0dB.pcm
>-diff ../dataqa350/i30_126speechoffice0dB.pcm ../dataqa351/i30_126speechoffice0dB.pcm
>-diff ../dataqa350/i60_126speechoffice0dB.pcm ../dataqa351/i60_126speechoffice0dB.pcm
>-diff ../dataqa350/a61speechoffice0dB.pcm ../dataqa351/a61speechoffice0dB.pcm
>-diff ../dataqa350/a62speechoffice0dB.pcm ../dataqa351/a62speechoffice0dB.pcm
>-diff ../dataqa350/a63speechoffice0dB.pcm ../dataqa351/a63speechoffice0dB.pcm
>-diff ../dataqa350/i30_127speech_and_misc_NB.pcm ../dataqa351/i30_127speech_and_misc_NB.pcm
>-diff ../dataqa350/i60_127speech_and_misc_NB.pcm ../dataqa351/i60_127speech_and_misc_NB.pcm
>-diff ../dataqa350/i30_128speech_and_misc_NB.pcm ../dataqa351/i30_128speech_and_misc_NB.pcm
>-diff ../dataqa350/i60_128speech_and_misc_NB.pcm ../dataqa351/i60_128speech_and_misc_NB.pcm
>-diff ../dataqa350/i30_129speech_and_misc_NB.pcm ../dataqa351/i30_129speech_and_misc_NB.pcm
>-diff ../dataqa350/i60_129speech_and_misc_NB.pcm ../dataqa351/i60_129speech_and_misc_NB.pcm
>-diff ../dataqa350/i30_130speech_and_misc_NB.pcm ../dataqa351/i30_130speech_and_misc_NB.pcm
>-diff ../dataqa350/i60_130speech_and_misc_NB.pcm ../dataqa351/i60_130speech_and_misc_NB.pcm
>-diff ../dataqa350/i30_131speech_and_misc_NB.pcm ../dataqa351/i30_131speech_and_misc_NB.pcm
>-diff ../dataqa350/i60_131speech_and_misc_NB.pcm ../dataqa351/i60_131speech_and_misc_NB.pcm
>-diff ../dataqa350/i30_132speech_and_misc_NB.pcm ../dataqa351/i30_132speech_and_misc_NB.pcm
>-diff ../dataqa350/i60_132speech_and_misc_NB.pcm ../dataqa351/i60_132speech_and_misc_NB.pcm
>-diff ../dataqa350/a64speech_and_misc_NB.pcm ../dataqa351/a64speech_and_misc_NB.pcm
>-diff ../dataqa350/a65speech_and_misc_NB.pcm ../dataqa351/a65speech_and_misc_NB.pcm
>-diff ../dataqa350/a66speech_and_misc_NB.pcm ../dataqa351/a66speech_and_misc_NB.pcm
>-diff ../dataqa350/i30_133speech_and_misc_WB.pcm ../dataqa351/i30_133speech_and_misc_WB.pcm
>-diff ../dataqa350/i60_133speech_and_misc_WB.pcm ../dataqa351/i60_133speech_and_misc_WB.pcm
>-diff ../dataqa350/i30_134speech_and_misc_WB.pcm ../dataqa351/i30_134speech_and_misc_WB.pcm
>-diff ../dataqa350/i60_134speech_and_misc_WB.pcm ../dataqa351/i60_134speech_and_misc_WB.pcm
>-diff ../dataqa350/i30_135speech_and_misc_WB.pcm ../dataqa351/i30_135speech_and_misc_WB.pcm
>-diff ../dataqa350/i60_135speech_and_misc_WB.pcm ../dataqa351/i60_135speech_and_misc_WB.pcm
>-diff ../dataqa350/i30_136speech_and_misc_WB.pcm ../dataqa351/i30_136speech_and_misc_WB.pcm
>-diff ../dataqa350/i60_136speech_and_misc_WB.pcm ../dataqa351/i60_136speech_and_misc_WB.pcm
>-diff ../dataqa350/i30_137speech_and_misc_WB.pcm ../dataqa351/i30_137speech_and_misc_WB.pcm
>-diff ../dataqa350/i60_137speech_and_misc_WB.pcm ../dataqa351/i60_137speech_and_misc_WB.pcm
>-diff ../dataqa350/i30_138speech_and_misc_WB.pcm ../dataqa351/i30_138speech_and_misc_WB.pcm
>-diff ../dataqa350/i60_138speech_and_misc_WB.pcm ../dataqa351/i60_138speech_and_misc_WB.pcm
>-diff ../dataqa350/a67speech_and_misc_WB.pcm ../dataqa351/a67speech_and_misc_WB.pcm
>-diff ../dataqa350/a68speech_and_misc_WB.pcm ../dataqa351/a68speech_and_misc_WB.pcm
>-diff ../dataqa350/a69speech_and_misc_WB.pcm ../dataqa351/a69speech_and_misc_WB.pcm
>-diff ../dataqa350/i30_139testM4.pcm ../dataqa351/i30_139testM4.pcm
>-diff ../dataqa350/i60_139testM4.pcm ../dataqa351/i60_139testM4.pcm
>-diff ../dataqa350/i30_140testM4.pcm ../dataqa351/i30_140testM4.pcm
>-diff ../dataqa350/i60_140testM4.pcm ../dataqa351/i60_140testM4.pcm
>-diff ../dataqa350/i30_141testM4.pcm ../dataqa351/i30_141testM4.pcm
>-diff ../dataqa350/i60_141testM4.pcm ../dataqa351/i60_141testM4.pcm
>-diff ../dataqa350/i30_142testM4.pcm ../dataqa351/i30_142testM4.pcm
>-diff ../dataqa350/i60_142testM4.pcm ../dataqa351/i60_142testM4.pcm
>-diff ../dataqa350/i30_143testM4.pcm ../dataqa351/i30_143testM4.pcm
>-diff ../dataqa350/i60_143testM4.pcm ../dataqa351/i60_143testM4.pcm
>-diff ../dataqa350/i30_144testM4.pcm ../dataqa351/i30_144testM4.pcm
>-diff ../dataqa350/i60_144testM4.pcm ../dataqa351/i60_144testM4.pcm
>-diff ../dataqa350/a70testM4.pcm ../dataqa351/a70testM4.pcm
>-diff ../dataqa350/a71testM4.pcm ../dataqa351/a71testM4.pcm
>-diff ../dataqa350/a72testM4.pcm ../dataqa351/a72testM4.pcm
>-diff ../dataqa350/i30_145testM4D_rev.pcm ../dataqa351/i30_145testM4D_rev.pcm
>-diff ../dataqa350/i60_145testM4D_rev.pcm ../dataqa351/i60_145testM4D_rev.pcm
>-diff ../dataqa350/i30_146testM4D_rev.pcm ../dataqa351/i30_146testM4D_rev.pcm
>-diff ../dataqa350/i60_146testM4D_rev.pcm ../dataqa351/i60_146testM4D_rev.pcm
>-diff ../dataqa350/i30_147testM4D_rev.pcm ../dataqa351/i30_147testM4D_rev.pcm
>-diff ../dataqa350/i60_147testM4D_rev.pcm ../dataqa351/i60_147testM4D_rev.pcm
>-diff ../dataqa350/i30_148testM4D_rev.pcm ../dataqa351/i30_148testM4D_rev.pcm
>-diff ../dataqa350/i60_148testM4D_rev.pcm ../dataqa351/i60_148testM4D_rev.pcm
>-diff ../dataqa350/i30_149testM4D_rev.pcm ../dataqa351/i30_149testM4D_rev.pcm
>-diff ../dataqa350/i60_149testM4D_rev.pcm ../dataqa351/i60_149testM4D_rev.pcm
>-diff ../dataqa350/i30_150testM4D_rev.pcm ../dataqa351/i30_150testM4D_rev.pcm
>-diff ../dataqa350/i60_150testM4D_rev.pcm ../dataqa351/i60_150testM4D_rev.pcm
>-diff ../dataqa350/a73testM4D_rev.pcm ../dataqa351/a73testM4D_rev.pcm
>-diff ../dataqa350/a74testM4D_rev.pcm ../dataqa351/a74testM4D_rev.pcm
>-diff ../dataqa350/a75testM4D_rev.pcm ../dataqa351/a75testM4D_rev.pcm
>-diff ../dataqa350/i30_151testM4D.pcm ../dataqa351/i30_151testM4D.pcm
>-diff ../dataqa350/i60_151testM4D.pcm ../dataqa351/i60_151testM4D.pcm
>-diff ../dataqa350/i30_152testM4D.pcm ../dataqa351/i30_152testM4D.pcm
>-diff ../dataqa350/i60_152testM4D.pcm ../dataqa351/i60_152testM4D.pcm
>-diff ../dataqa350/i30_153testM4D.pcm ../dataqa351/i30_153testM4D.pcm
>-diff ../dataqa350/i60_153testM4D.pcm ../dataqa351/i60_153testM4D.pcm
>-diff ../dataqa350/i30_154testM4D.pcm ../dataqa351/i30_154testM4D.pcm
>-diff ../dataqa350/i60_154testM4D.pcm ../dataqa351/i60_154testM4D.pcm
>-diff ../dataqa350/i30_155testM4D.pcm ../dataqa351/i30_155testM4D.pcm
>-diff ../dataqa350/i60_155testM4D.pcm ../dataqa351/i60_155testM4D.pcm
>-diff ../dataqa350/i30_156testM4D.pcm ../dataqa351/i30_156testM4D.pcm
>-diff ../dataqa350/i60_156testM4D.pcm ../dataqa351/i60_156testM4D.pcm
>-diff ../dataqa350/a76testM4D.pcm ../dataqa351/a76testM4D.pcm
>-diff ../dataqa350/a77testM4D.pcm ../dataqa351/a77testM4D.pcm
>-diff ../dataqa350/a78testM4D.pcm ../dataqa351/a78testM4D.pcm
>-diff ../dataqa350/i30_157testfile.pcm ../dataqa351/i30_157testfile.pcm
>-diff ../dataqa350/i60_157testfile.pcm ../dataqa351/i60_157testfile.pcm
>-diff ../dataqa350/i30_158testfile.pcm ../dataqa351/i30_158testfile.pcm
>-diff ../dataqa350/i60_158testfile.pcm ../dataqa351/i60_158testfile.pcm
>-diff ../dataqa350/i30_159testfile.pcm ../dataqa351/i30_159testfile.pcm
>-diff ../dataqa350/i60_159testfile.pcm ../dataqa351/i60_159testfile.pcm
>-diff ../dataqa350/i30_160testfile.pcm ../dataqa351/i30_160testfile.pcm
>-diff ../dataqa350/i60_160testfile.pcm ../dataqa351/i60_160testfile.pcm
>-diff ../dataqa350/i30_161testfile.pcm ../dataqa351/i30_161testfile.pcm
>-diff ../dataqa350/i60_161testfile.pcm ../dataqa351/i60_161testfile.pcm
>-diff ../dataqa350/i30_162testfile.pcm ../dataqa351/i30_162testfile.pcm
>-diff ../dataqa350/i60_162testfile.pcm ../dataqa351/i60_162testfile.pcm
>-diff ../dataqa350/a79testfile.pcm ../dataqa351/a79testfile.pcm
>-diff ../dataqa350/a80testfile.pcm ../dataqa351/a80testfile.pcm
>-diff ../dataqa350/a81testfile.pcm ../dataqa351/a81testfile.pcm
>-diff ../dataqa350/i30_163tone_cisco.pcm ../dataqa351/i30_163tone_cisco.pcm
>-diff ../dataqa350/i60_163tone_cisco.pcm ../dataqa351/i60_163tone_cisco.pcm
>-diff ../dataqa350/i30_164tone_cisco.pcm ../dataqa351/i30_164tone_cisco.pcm
>-diff ../dataqa350/i60_164tone_cisco.pcm ../dataqa351/i60_164tone_cisco.pcm
>-diff ../dataqa350/i30_165tone_cisco.pcm ../dataqa351/i30_165tone_cisco.pcm
>-diff ../dataqa350/i60_165tone_cisco.pcm ../dataqa351/i60_165tone_cisco.pcm
>-diff ../dataqa350/i30_166tone_cisco.pcm ../dataqa351/i30_166tone_cisco.pcm
>-diff ../dataqa350/i60_166tone_cisco.pcm ../dataqa351/i60_166tone_cisco.pcm
>-diff ../dataqa350/i30_167tone_cisco.pcm ../dataqa351/i30_167tone_cisco.pcm
>-diff ../dataqa350/i60_167tone_cisco.pcm ../dataqa351/i60_167tone_cisco.pcm
>-diff ../dataqa350/i30_168tone_cisco.pcm ../dataqa351/i30_168tone_cisco.pcm
>-diff ../dataqa350/i60_168tone_cisco.pcm ../dataqa351/i60_168tone_cisco.pcm
>-diff ../dataqa350/a82tone_cisco.pcm ../dataqa351/a82tone_cisco.pcm
>-diff ../dataqa350/a83tone_cisco.pcm ../dataqa351/a83tone_cisco.pcm
>-diff ../dataqa350/a84tone_cisco.pcm ../dataqa351/a84tone_cisco.pcm
>-diff ../dataqa350/i30_169tone_cisco_long.pcm ../dataqa351/i30_169tone_cisco_long.pcm
>-diff ../dataqa350/i60_169tone_cisco_long.pcm ../dataqa351/i60_169tone_cisco_long.pcm
>-diff ../dataqa350/i30_170tone_cisco_long.pcm ../dataqa351/i30_170tone_cisco_long.pcm
>-diff ../dataqa350/i60_170tone_cisco_long.pcm ../dataqa351/i60_170tone_cisco_long.pcm
>-diff ../dataqa350/i30_171tone_cisco_long.pcm ../dataqa351/i30_171tone_cisco_long.pcm
>-diff ../dataqa350/i60_171tone_cisco_long.pcm ../dataqa351/i60_171tone_cisco_long.pcm
>-diff ../dataqa350/i30_172tone_cisco_long.pcm ../dataqa351/i30_172tone_cisco_long.pcm
>-diff ../dataqa350/i60_172tone_cisco_long.pcm ../dataqa351/i60_172tone_cisco_long.pcm
>-diff ../dataqa350/i30_173tone_cisco_long.pcm ../dataqa351/i30_173tone_cisco_long.pcm
>-diff ../dataqa350/i60_173tone_cisco_long.pcm ../dataqa351/i60_173tone_cisco_long.pcm
>-diff ../dataqa350/i30_174tone_cisco_long.pcm ../dataqa351/i30_174tone_cisco_long.pcm
>-diff ../dataqa350/i60_174tone_cisco_long.pcm ../dataqa351/i60_174tone_cisco_long.pcm
>-diff ../dataqa350/a85tone_cisco_long.pcm ../dataqa351/a85tone_cisco_long.pcm
>-diff ../dataqa350/a86tone_cisco_long.pcm ../dataqa351/a86tone_cisco_long.pcm
>-diff ../dataqa350/a87tone_cisco_long.pcm ../dataqa351/a87tone_cisco_long.pcm
>-diff ../dataqa350/i30_175wb_contspeech.pcm ../dataqa351/i30_175wb_contspeech.pcm
>-diff ../dataqa350/i60_175wb_contspeech.pcm ../dataqa351/i60_175wb_contspeech.pcm
>-diff ../dataqa350/i30_176wb_contspeech.pcm ../dataqa351/i30_176wb_contspeech.pcm
>-diff ../dataqa350/i60_176wb_contspeech.pcm ../dataqa351/i60_176wb_contspeech.pcm
>-diff ../dataqa350/i30_177wb_contspeech.pcm ../dataqa351/i30_177wb_contspeech.pcm
>-diff ../dataqa350/i60_177wb_contspeech.pcm ../dataqa351/i60_177wb_contspeech.pcm
>-diff ../dataqa350/i30_178wb_contspeech.pcm ../dataqa351/i30_178wb_contspeech.pcm
>-diff ../dataqa350/i60_178wb_contspeech.pcm ../dataqa351/i60_178wb_contspeech.pcm
>-diff ../dataqa350/i30_179wb_contspeech.pcm ../dataqa351/i30_179wb_contspeech.pcm
>-diff ../dataqa350/i60_179wb_contspeech.pcm ../dataqa351/i60_179wb_contspeech.pcm
>-diff ../dataqa350/i30_180wb_contspeech.pcm ../dataqa351/i30_180wb_contspeech.pcm
>-diff ../dataqa350/i60_180wb_contspeech.pcm ../dataqa351/i60_180wb_contspeech.pcm
>-diff ../dataqa350/a88wb_contspeech.pcm ../dataqa351/a88wb_contspeech.pcm
>-diff ../dataqa350/a89wb_contspeech.pcm ../dataqa351/a89wb_contspeech.pcm
>-diff ../dataqa350/a90wb_contspeech.pcm ../dataqa351/a90wb_contspeech.pcm
>-diff ../dataqa350/i30_181wb_speech_office25db.pcm ../dataqa351/i30_181wb_speech_office25db.pcm
>-diff ../dataqa350/i60_181wb_speech_office25db.pcm ../dataqa351/i60_181wb_speech_office25db.pcm
>-diff ../dataqa350/i30_182wb_speech_office25db.pcm ../dataqa351/i30_182wb_speech_office25db.pcm
>-diff ../dataqa350/i60_182wb_speech_office25db.pcm ../dataqa351/i60_182wb_speech_office25db.pcm
>-diff ../dataqa350/i30_183wb_speech_office25db.pcm ../dataqa351/i30_183wb_speech_office25db.pcm
>-diff ../dataqa350/i60_183wb_speech_office25db.pcm ../dataqa351/i60_183wb_speech_office25db.pcm
>-diff ../dataqa350/i30_184wb_speech_office25db.pcm ../dataqa351/i30_184wb_speech_office25db.pcm
>-diff ../dataqa350/i60_184wb_speech_office25db.pcm ../dataqa351/i60_184wb_speech_office25db.pcm
>-diff ../dataqa350/i30_185wb_speech_office25db.pcm ../dataqa351/i30_185wb_speech_office25db.pcm
>-diff ../dataqa350/i60_185wb_speech_office25db.pcm ../dataqa351/i60_185wb_speech_office25db.pcm
>-diff ../dataqa350/i30_186wb_speech_office25db.pcm ../dataqa351/i30_186wb_speech_office25db.pcm
>-diff ../dataqa350/i60_186wb_speech_office25db.pcm ../dataqa351/i60_186wb_speech_office25db.pcm
>-diff ../dataqa350/a91wb_speech_office25db.pcm ../dataqa351/a91wb_speech_office25db.pcm
>-diff ../dataqa350/a92wb_speech_office25db.pcm ../dataqa351/a92wb_speech_office25db.pcm
>-diff ../dataqa350/a93wb_speech_office25db.pcm ../dataqa351/a93wb_speech_office25db.pcm
>-diff ../dataqa350/a30_1DTMF_16kHz_short.pcm ../dataqa351/a30_1DTMF_16kHz_short.pcm
>-diff ../dataqa350/a60_1DTMF_16kHz_short.pcm ../dataqa351/a60_1DTMF_16kHz_short.pcm
>-diff ../dataqa350/a30_2ltest_speech_noisy.pcm ../dataqa351/a30_2ltest_speech_noisy.pcm
>-diff ../dataqa350/a60_2ltest_speech_noisy.pcm ../dataqa351/a60_2ltest_speech_noisy.pcm
>-diff ../dataqa350/a30_3misc2.pcm ../dataqa351/a30_3misc2.pcm
>-diff ../dataqa350/a60_3misc2.pcm ../dataqa351/a60_3misc2.pcm
>-diff ../dataqa350/a30_4sinesweep.pcm ../dataqa351/a30_4sinesweep.pcm
>-diff ../dataqa350/a60_4sinesweep.pcm ../dataqa351/a60_4sinesweep.pcm
>-diff ../dataqa350/a30_5speechmusic.pcm ../dataqa351/a30_5speechmusic.pcm
>-diff ../dataqa350/a60_5speechmusic.pcm ../dataqa351/a60_5speechmusic.pcm
>-diff ../dataqa350/a30_6tone_cisco.pcm ../dataqa351/a30_6tone_cisco.pcm
>-diff ../dataqa350/a60_6tone_cisco.pcm ../dataqa351/a60_6tone_cisco.pcm
>-diff ../dataqa350/a60_7tone_cisco.pcm ../dataqa351/a60_7tone_cisco.pcm
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/diffiSACPLC.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/diffiSACPLC.txt
>deleted file mode 100644
>index 9e3629b2ca72a490e31c5b472dd91b3d4d2a8367..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/diffiSACPLC.txt
>+++ /dev/null
>@@ -1,20 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
>-
>-LOGFILE=logplc.txt
>-echo "START PLC TEST" > $LOGFILE
>-
>-OUTDIR1=../dataqaplc_0
>-OUTDIR2=../dataqaplc_1
>-
>-diff $OUTDIR1/outplc1.pcm $OUTDIR2/outplc1.pcm 
>-diff $OUTDIR1/outplc2.pcm $OUTDIR2/outplc2.pcm 
>-diff $OUTDIR1/outplc3.pcm $OUTDIR2/outplc3.pcm 
>-diff $OUTDIR1/outplc4.pcm $OUTDIR2/outplc4.pcm 
>-diff $OUTDIR1/outplc5.pcm $OUTDIR2/outplc5.pcm 
>-diff $OUTDIR1/outplc6.pcm $OUTDIR2/outplc6.pcm 
>-
>-echo DONE!
>-
>-
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACLongtest.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACLongtest.txt
>deleted file mode 100644
>index eeffc0c955edd07ac9f0eb0548cae100b6e03971..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACLongtest.txt
>+++ /dev/null
>@@ -1,61 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
>-
>-LOGFILE=logNormal.txt
>-echo "START ISAC TEST" > $LOGFILE
>-echo >> $LOGFILE
>-
>-ISAC=../Release/kenny.exe
>-ISACFIXFLOAT=../Release/testFixFloat.exe
>-
>-INFILES=$(cat InputFiles.txt)
>-SUBSET=$(cat InputFilesFew.txt)
>-CHANNELFILES=$(cat ChannelFiles.txt)
>-CHANNELLIST=($(cat ChannelFiles.txt))
>-INDIR=../data/orig
>-OUTDIR=../dataqa
>-mkdir -p $OUTDIR
>-
>-TARGETRATE=(10000 15000 20000 25000 30000 32000)
>-#echo ${CHANNELFILES[1]}
>-
>-index1=0
>-index2=0
>-
>-for file in $INFILES # loop over all input files
>-  do
>-  
>-  for rate in ${TARGETRATE[*]}
>-	do
>-	let "index1=index1+1"
>-	$ISAC -I $rate -FL 30 $INDIR/"$file" $OUTDIR/i30_$index1"$file" >> $LOGFILE
>-	$ISAC -I $rate -FL 60 $INDIR/"$file" $OUTDIR/i60_$index1"$file" >> $LOGFILE
>-  done
>-  for channel in $CHANNELFILES
>-	do
>-	let "index2=index2+1"
>-	$ISAC $INDIR/$channel $INDIR/"$file" $OUTDIR/a$index2"$file" >> $LOGFILE
>-  done
>-
>-done
>-
>-index1=0
>-
>-for file in $SUBSET # loop over the subset of input files
>-  do
>-	let "index1=index1+1"
>-	$ISAC $INDIR/${CHANNELLIST[0]} -FL 30 -FIXED_FL $INDIR/"$file" $OUTDIR/a30_$index1"$file" >> $LOGFILE
>-	$ISAC $INDIR/${CHANNELLIST[0]} -FL 60 -FIXED_FL $INDIR/"$file" $OUTDIR/a60_$index1"$file" >> $LOGFILE
>-done
>-
>-let "index1=index1+1"	
>- $ISAC $INDIR/${CHANNELLIST[0]} -INITRATE 25000 -FL 30 $INDIR/"$file" $OUTDIR/a60_$index1"$file" >> $LOGFILE
>-
>-# Run fault test
>-
>-#./runiSACfault.txt
>-
>-echo DONE!
>-
>-
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACNB.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACNB.txt
>deleted file mode 100644
>index 605595cc04fca61924b9ba7cc7caefb96d02a038..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACNB.txt
>+++ /dev/null
>@@ -1,45 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
>-
>-LOGFILE=logNB.txt
>-echo "START NARROWBAND TEST" > $LOGFILE
>-echo >> $LOGFILE
>-
>-ISAC=../Release/kenny.exe
>-ISACFIXFLOAT=../Release/testFixFloat.exe
>-
>-INFILES=$(cat InputFiles.txt)
>-SUBSET=$(cat InputFilesFew.txt)
>-CHANNELFILES=$(cat ChannelFiles.txt)
>-CHANNELLIST=($(cat ChannelFiles.txt))
>-INDIR=../data/orig
>-OUTDIR=../dataqaNB
>-mkdir -p $OUTDIR
>-
>-TARGETRATE=(10000 15000 20000 25000 30000 32000)
>-#echo ${CHANNELFILES[1]}
>-
>-index1=0
>-index2=0
>-
>-# Narrowband Interfaces
>-
>-for file in $SUBSET # loop over all input files
>-  do
>-  for rate in ${TARGETRATE[*]}
>-	do
>-	let "index1=index1+1"
>- 	$ISAC $rate -FL 30 -NB 1 $INDIR/"$file" $OUTDIR/nb130_$index1"$file" >> $LOGFILE
>-	$ISAC $rate -FL 60 -NB 1 $INDIR/"$file" $OUTDIR/nb160_$index1"$file" >> $LOGFILE
>-	$ISAC $rate -FL 30 -NB 2 $INDIR/"$file" $OUTDIR/nb230_$index1"$file" >> $LOGFILE
>-	$ISAC $rate -FL 60 -NB 2 $INDIR/"$file" $OUTDIR/nb260_$index1"$file" >> $LOGFILE
>-	$ISAC $rate -FL 30 -NB 2 -PL 10 $INDIR/"$file" $OUTDIR/nb2plc30_$index1"$file" >> $LOGFILE
>-	$ISAC $rate -FL 60 -NB 2 -PL 10 $INDIR/"$file" $OUTDIR/nb2plc60_$index1"$file" >> $LOGFILE
>-  done
>-
>-done
>-
>-echo DONE!
>-
>-
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACPLC.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACPLC.txt
>deleted file mode 100644
>index 6bee6f7c3f3238610728e274921f62878a458e96..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACPLC.txt
>+++ /dev/null
>@@ -1,23 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
>-
>-LOGFILE=logplc.txt
>-echo "START PLC TEST" > $LOGFILE
>-
>-ISAC=../Release/kenny.exe
>-
>-INDIR=../data/orig
>-OUTDIR=../dataqaplc_0
>-mkdir -p $OUTDIR
>-
>-$ISAC 12000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc1.pcm 
>-$ISAC 20000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc2.pcm 
>-$ISAC 32000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc3.pcm 
>-$ISAC 12000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc4.pcm 
>-$ISAC 20000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc5.pcm 
>-$ISAC 32000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc6.pcm 
>-
>-echo DONE!
>-
>-
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACRate.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACRate.txt
>deleted file mode 100644
>index d8403e099d650fb75eb368274bbe139f9d4982f2..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACRate.txt
>+++ /dev/null
>@@ -1,23 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
>-
>-LOGG=loggRate.txt
>-OUTDIR=../dataqaRate
>-mkdir -p $OUTDIR
>-
>-../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_1.pcm > $LOGG
>-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_2.pcm >> $LOGG
>-../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_3.pcm >> $LOGG
>-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_4.pcm >> $LOGG
>-../Release/kenny.exe 13000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_5.pcm >> $LOGG
>-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 60 -MAXRATE 32000 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_6.pcm >> $LOGG
>-../Release/kenny.exe 13000 -INIT_RATE 32000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_7.pcm >> $LOGG
>-
>-../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_11.pcm >> $LOGG
>-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/longspeech.pcm $OUTDIR/out_napi_12.pcm >> $LOGG
>-../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_13.pcm >> $LOGG
>-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/longspeech.pcm $OUTDIR/out_napi_14.pcm >> $LOGG
>-../Release/kenny.exe 13000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_15.pcm >> $LOGG
>-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 60 -MAXRATE 32000 ../data/orig/longspeech.pcm $OUTDIR/out_napi_16.pcm >> $LOGG
>-../Release/kenny.exe 13000 -INIT_RATE 32000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_17.pcm >> $LOGG
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACfault.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACfault.txt
>deleted file mode 100644
>index f4d9478fd49bff5103cd42ac459ea1c8ba4e8738..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACfault.txt
>+++ /dev/null
>@@ -1,40 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
>-
>-LOGFILE=logfault.txt
>-echo "START FAULT TEST" > $LOGFILE
>-
>-ISAC=../Release/kenny.exe
>-ISACFIXFLOAT=../Release/testFixFloat.exe
>-
>-INFILES=$(cat InputFiles.txt)
>-SUBSET=$(cat InputFilesFew.txt)
>-CHANNELFILES=$(cat ChannelFiles.txt)
>-CHANNELLIST=($(cat ChannelFiles.txt))
>-INDIR=../data/orig
>-OUTDIR=../dataqaft
>-mkdir -p $OUTDIR
>-
>-TARGETRATE=(10000 15000 20000 25000 30000 32000)
>-FAULTTEST=(1 2 3 4 5 6 7 9)
>-
>-index1=0
>-
>-file=wb_contspeech.pcm
>-
>-# Fault test
>-for testnr in ${FAULTTEST[*]}
>-   do
>-	$ISAC 32000 -F $testnr $INDIR/"$file" $OUTDIR/ft$testnr"$file" >> $LOGFILE
>-done
>-
>-# Fault test number 10, error in bitstream
>- $ISAC 32000 -F 10 $INDIR/"$file" $OUTDIR/ft10_"$file" >> $LOGFILE
>- $ISAC 32000 -F 10 -PL 10 $INDIR/"$file" $OUTDIR/ft10plc_"$file" >> $LOGFILE
>- $ISAC 32000 -F 10 -NB 1 $INDIR/"$file" $OUTDIR/ft10nb1_"$file" >> $LOGFILE
>- $ISAC 32000 -F 10 -NB 2 -PL 10 $INDIR/"$file" $OUTDIR/ft10nb2_"$file" >> $LOGFILE
>-
>-echo DONE!
>-
>-
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACfixfloat.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACfixfloat.txt
>deleted file mode 100644
>index c9e02df2e9f33660255e62e4f6a62c0aee115615..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/QA/runiSACfixfloat.txt
>+++ /dev/null
>@@ -1,47 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
>-
>-LOGFILE=logfxfl.txt
>-echo "START FIX-FLOAT TEST" > $LOGFILE
>-
>-
>-ISACFIXFLOAT=../testFixFloat.exe
>-
>-INFILES=$(cat InputFiles.txt)
>-SUBSET=$(cat InputFilesFew.txt)
>-CHANNELFILES=$(cat ChannelFiles.txt)
>-CHANNELLIST=($(cat ChannelFiles.txt))
>-INDIR=../data/orig
>-OUTDIR=../dataqafxfl
>-mkdir -p $OUTDIR
>-
>-index1=0
>-
>-for file in $INFILES # loop over all input files
>-  do
>-  
>-  for channel in $CHANNELFILES
>-	do
>-	let "index1=index1+1"
>-
>-	$ISACFIXFLOAT $INDIR/$channel -m 1 -PLC $INDIR/"$file" $OUTDIR/flfx$index1"$file" >> $LOGFILE
>-	$ISACFIXFLOAT $INDIR/$channel -m 2 -PLC $INDIR/"$file" $OUTDIR/fxfl$index1"$file" >> $LOGFILE
>-  done
>-
>-done
>-
>-index1=0
>-
>-for file in $SUBSET # loop over the subset of input files
>-  do
>-	let "index1=index1+1"
>-	$ISACFIXFLOAT $INDIR/$channel -m 1 -NB 1 $INDIR/"$file" $OUTDIR/flfxnb1_$index1"$file" >> $LOGFILE
>-	$ISACFIXFLOAT $INDIR/$channel -m 2 -NB 1 $INDIR/"$file" $OUTDIR/fxflnb1_$index1"$file" >> $LOGFILE
>-	$ISACFIXFLOAT $INDIR/$channel -m 1 -NB 2 -PLC $INDIR/"$file" $OUTDIR/flfxnb2_$index1"$file" >> $LOGFILE
>-	$ISACFIXFLOAT $INDIR/$channel -m 2 -NB 2 -PLC $INDIR/"$file" $OUTDIR/fxflnb2_$index1"$file" >> $LOGFILE
>-done
>-
>-echo DONE!
>-
>-
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
>index 28767afc633c09eec75ad08e2cbf71698b411640..6692a519ca97c302d7af5a0deb3bd5586928ece6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
>@@ -96,7 +96,7 @@ static void FindInvArSpec(const int16_t* ARCoefQ12,
>                           const int32_t gainQ10,
>                           int32_t* CurveQ16) {
>   int32_t CorrQ11[AR_ORDER + 1];
>-  int32_t sum, tmpGain;
>+  int64_t sum, tmpGain;
>   int32_t diffQ16[FRAMESAMPLES / 8];
>   const int16_t* CS_ptrQ9;
>   int k, n;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
>deleted file mode 100644
>index 3f05224a0ecc256ae116a7e8e9744e7a0050da6a..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
>+++ /dev/null
>@@ -1,433 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
>-
>-
>-
>-if  [ "$1" = "x64" ] || [ "$2" = "x64" ] || [ "$#" -eq 0 ]
>-    then
>-    PLATFORM=_X64
>-    ISAC=../x64/Release/ReleaseTest-API_2005.exe
>-elif [ "$1" = "LINUX" ] || [ "$2" = "LINUX" ]
>-    then
>-    PLATFORM=_linux
>-    ISAC=../ReleaseTest-API/isacswtest
>-else
>-    PLATFORM=_2005
>-    ISAC=../win32/Release/ReleaseTest-API_2005.exe
>-fi
>-
>-if  [ "$#" -eq 0 ] || [ "$1" = "all" ] || [ "$1" = "wb" ]
>-    then
>-    LOGFILE=logNormal"$PLATFORM".txt
>-    echo "START ISAC WB TEST" > $LOGFILE
>-    echo >> $LOGFILE
>-
>-    INFILES=$(cat InputFiles.txt)
>-    SUBSET=$(cat InputFilesFew.txt)
>-    CHANNELFILES=$(cat ChannelFiles.txt)
>-    CHANNELLIST=($(cat ChannelFiles.txt))
>-    INDIR=../data/orig
>-    OUTDIR=../dataqa"$PLATFORM"
>-    mkdir -p $OUTDIR
>-    rm -f $OUTDIR/*
>-    
>-    idx=0
>-    RATE=10000
>-    FRAMESIZE=30
>-    
>-
>-    for file in $INFILES # loop over all input files
>-      do
>-      
>-      echo "Input file: " $file
>-      echo "-----------------------------------"
>-      echo "Instantaneous with RATE " $RATE ", and Frame-size " $FRAMESIZE
>-      $ISAC -I -B $RATE -FL $FRAMESIZE -FS 16 $INDIR/"$file" $OUTDIR/i_"$FRAMESIZE"_"$RATE"_"$file" >> $LOGFILE
>-      echo
>-      
>-      name="${CHANNELLIST[$idx]}"
>-      echo "Adaptive with channel file: " $name 
>-      
>-      $ISAC -B $INDIR/${CHANNELLIST[$idx]} -FS 16 $INDIR/"$file" $OUTDIR/a_${name%.*}_"$file" >> $LOGFILE
>-      
>-      echo
>-      echo
>-      
>-#     alternate between 30 & 60 ms.
>-      if [ $FRAMESIZE -eq 30 ]
>-	  then
>-	  FRAMESIZE=60
>-      else
>-	  FRAMESIZE=30
>-      fi
>-      
>-#     rate between 10000 to 32000 bits/sec
>-      if [ $RATE -le 30000 ]
>-	  then
>-	  let "RATE=RATE+2000"
>-      else
>-	  let "RATE=10000"
>-      fi
>-      
>-#     there are only three channel file
>-      if [ $idx -ge 2 ]; then
>-	  idx=0
>-      else
>-	  let "idx=idx+1"
>-      fi
>-      
>-    done
>-
>-    idx=0
>-    
>-#   loop over the subset of input files
>-    for file in $SUBSET 
>-      do
>-      
>-      if [ $idx -eq 0 ]; then
>-	  $ISAC -B $INDIR/${CHANNELLIST[0]} -FL 30 -FIXED_FL -FS 16 $INDIR/"$file" $OUTDIR/a30_"$file" >> $LOGFILE
>-	  idx=1
>-      else
>-	  $ISAC -B $INDIR/${CHANNELLIST[0]} -FL 60 -FIXED_FL -FS 16 $INDIR/"$file" $OUTDIR/a60_"$file" >> $LOGFILE
>-	  idx=0
>-      fi
>-    done
>-
>-    $ISAC -B $INDIR/${CHANNELLIST[0]} -INITRATE 25000 -FL 30 -FS 16 $INDIR/"$file" $OUTDIR/a60_Init25kbps_"$file" >> $LOGFILE
>-
>-    echo
>-    echo WIDEBAND DONE!
>-    echo
>-    echo
>-fi
>-
>-if  [ "$#" -eq 0 ] || [ "$1" = "all" ] || [ "$1" = "swb" ]
>-    then
>-
>-    LOGFILE=logNormal_SWB"$PLATFORM".txt
>-    echo "START ISAC SWB TEST" > $LOGFILE
>-    echo >> $LOGFILE
>-
>-    echo STARTING TO TEST SUPER-WIDEBAND
>-    
>-    INFILES=$(cat InputFilesSWB.txt)
>-    INDIR=../data/origswb
>-    OUTDIR=../dataqaswb"$PLATFORM"
>-    mkdir -p $OUTDIR
>-    rm -f $OUTDIR/*
>-    
>-    for file in $INFILES
>-      do
>-      echo
>-      echo "Input file: " $file
>-      echo "--------------------------------"
>-      for RATE in 12000 20000 32000 38000 45000 50000 56000  
>-	do
>-	
>-	echo "Rate " $RATE
>-	$ISAC -I -B $RATE -FL 30 -FS 32 $INDIR/"$file" $OUTDIR/swb_"$RATE"_"$file" >> $LOGFILE
>-	echo
>-	
>-      done
>-  
>-    done
>-fi
>-
>-if  [ "$#" -eq 0 ] || [ "$1" = "all" ] || [ "$1" = "API" ]
>-    then
>-
>-    LOGFILE_API=logNormal_API"$PLATFORM".txt
>-    echo
>-    echo
>-    echo "START ISAC API TEST" > $LOGFILE_API
>-    echo >> $LOGFILE_API
>-    idx=1
>-    echo "                            Test Enforcement of frame-size"
>-    echo "========================================================================================"
>-    mkdir -p ../FrameSizeLim"$PLATFORM"
>-    rm -f ../FrameSizeLim"$PLATFORM"/*
>-    echo
>-    echo "-- No enforcement; BN 10000"
>-    echo
>-    $ISAC -B 10000 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
>-	../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo "-- Now Enforce 30 ms frame size with the same bottleneck"
>-    echo "There should not be any 60 ms frame"
>-    echo
>-    $ISAC -B 10000 -FL 30 -FIXED_FL -FS 16 ../data/orig/speech_and_misc_WB.pcm \
>-	../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo "-- No enforcement; BN 32000"
>-    echo
>-    $ISAC -B 32000 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
>-	../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo "-- Now Enforce 60 ms frame size with the same bottleneck"
>-    echo "There should not be any 30 ms frame"
>-    echo 
>-    $ISAC -B 32000 -FL 60 -FIXED_FL -FS 16 ../data/orig/speech_and_misc_WB.pcm \
>-	../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo
>-    echo
>-    echo
>-
>-    echo "                           Test Limiting of Payload Size and Rate"
>-    echo "========================================================================================"
>-    mkdir -p ../PayloadLim"$PLATFORM"
>-    rm -f ../PayloadLim"$PLATFORM"/*
>-    echo
>-    echo
>-    echo "-- No Limit, frame-size 60 ms, WIDEBAND"
>-    echo 
>-    $ISAC -I -B 32000 -FL 60 -FS 16                ../data/orig/speech_and_misc_WB.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo "-- Payload-size limit of 250, frame-size 60 ms, WIDEBAND"
>-    echo 
>-    $ISAC -I -B 32000 -FL 60 -FS 16 -MAX 250       ../data/orig/speech_and_misc_WB.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo "-- Rate limit of 33 kbps for 60 ms frame-size"
>-    echo
>-    $ISAC -I -B 32000 -FL 60 -FS 16 -MAXRATE 33000 ../data/orig/speech_and_misc_WB.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo 
>-    echo "________________________________________________________"
>-    echo
>-    echo
>-
>-    echo "-- No Limit, frame-size 30 ms, WIDEBAND"
>-    echo 
>-    $ISAC -I -B 32000 -FL 30 -FS 16                ../data/orig/speech_and_misc_WB.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo    
>-    echo "-- Payload-size limit of 130, frame-size 30 ms, WIDEBAND"
>-    echo 
>-    $ISAC -I -B 32000 -FL 30 -FS 16 -MAX 130       ../data/orig/speech_and_misc_WB.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo "-- Rate limit of 33 kbps for 30 ms frame-size, wideband"
>-    echo
>-    $ISAC -I -B 32000 -FL 30 -FS 16 -MAXRATE 33000 ../data/orig/speech_and_misc_WB.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo "________________________________________________________"
>-    echo
>-    echo
>-
>-    echo "-- No limit for 32 kbps, 30 ms, SUPER-WIDEBAND"
>-    echo
>-    $ISAC -I -B 32000 -FL 30 -FS 32          ../data/origswb/jstest_32.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo "-- Payload limit of 130 bytes for 32 kbps, 30 ms, SUPER-WIDEBAND"
>-    echo
>-    $ISAC -I -B 32000 -FL 30 -FS 32 -MAX 130 ../data/origswb/jstest_32.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo "________________________________________________________"
>-    echo
>-    echo
>-
>-    echo "-- No limit, Rate 45 kbps, 30 ms, SUPER-WIDEBAND, 12 kHz"
>-    echo
>-    $ISAC -I -B 45000 -FL 30 -FS 32               ../data/origswb/jstest_32.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo "-- Rate limit of 46 kbps for 42 kbps, 30 ms, SUPER-WIDEBAND, 12 kHz"
>-    echo
>-    $ISAC -I -B 45000 -FL 30 -FS 32 -MAXRATE 46000 ../data/origswb/jstest_32.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo "-- Payload limit of 170 bytes for 45 kbps, 30 ms, SUPER-WIDEBAND, 12 kHz"
>-    echo
>-    $ISAC -I -B 45000 -FL 30 -FS 32 -MAX 170       ../data/origswb/jstest_32.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo "________________________________________________________"
>-    echo
>-    echo
>-  
>-    echo "-- No limit for 56 kbps, 30 ms, SUPER-WIDEBAND, 16 kHz"
>-    echo
>-    $ISAC -I -B 56000 -FL 30 -FS 32                ../data/origswb/jstest_32.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo "-- Payload limit of 200 bytes for 56 kbps 30 ms, SUPER-WIDEBAND, 16 kHz"
>-    echo
>-    $ISAC -I -B 56000 -FL 30 -FS 32 -MAX 200       ../data/origswb/jstest_32.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    echo "-- Rate limit of 57 kbps for 56 kbps 30 ms, SUPER-WIDEBAND, 16 kHz"
>-    echo
>-    $ISAC -I -B 56000 -FL 30 -FS 32 -MAXRATE 57000 ../data/origswb/jstest_32.pcm \
>-	../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo "________________________________________________________"
>-    echo
>-    echo
>-    echo
>-    echo
>-    echo
>-    
>-    echo "                                    Test Trans-Coding"
>-    echo "========================================================================================"
>-    mkdir -p ../Transcoding"$PLATFORM"
>-    rm -f ../Transcoding"$PLATFORM"/*
>-    echo
>-    echo
>-    echo "-- 20 kbps, 30 ms, WIDEBAND"
>-    echo
>-    $ISAC -I -B 20000 -FL 30 -FS 16          ../data/orig/speech_and_misc_WB.pcm \
>-	../Transcoding"$PLATFORM"/APITest_refTrans20WB.pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    
>-    echo "-- 32 kbps trans-coding to 20 kbps, 30 ms, WIDEBAND"
>-    echo
>-    $ISAC -I -B 32000 -FL 30 -FS 16  -T 20000  ../Transcoding"$PLATFORM"/APITest_32T20.pcm \
>-	../data/orig/speech_and_misc_WB.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo "________________________________________________________"
>-
>-    echo
>-    echo
>-    echo "-- 38 kbps, 30 ms, SUPER-WIDEBAND"
>-    echo
>-    $ISAC -I -B 38000 -FL 30 -FS 32          ../data/origswb/jstest_32.pcm \
>-	../Transcoding"$PLATFORM"/APITest_refTrans38.pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    
>-    echo "-- 45 kbps trans-coding to 38 kbps, 30 ms, SUPER-WIDEBAND"
>-    echo
>-    $ISAC -I -B 45000 -FL 30 -FS 32  -T 38000  ../Transcoding"$PLATFORM"/APITest_45T38.pcm \
>-	../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>- 
>-    echo
>-    echo
>-    echo "-- 20 kbps, 30 ms, SUPER-WIDEBAND"
>-    echo
>-    $ISAC -I -B 20000 -FL 30 -FS 32          ../data/origswb/jstest_32.pcm \
>-	../Transcoding"$PLATFORM"/APITest_refTrans20SWB.pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-
>-    echo
>-    echo
>-    
>-    echo "-- 45 kbps trans-coding to 20 kbps, 30 ms, SUPER-WIDEBAND"
>-    echo
>-    $ISAC -I -B 45000 -FL 30 -FS 32  -T 20000  ../Transcoding"$PLATFORM"/APITest_45T20.pcm \
>-	../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo "________________________________________________________"
>-    echo
>-    echo
>-    echo "-- 50 kbps, 30 ms, SUPER-WIDEBAND"
>-    echo
>-    $ISAC -I -B 50000 -FL 30 -FS 32          ../data/origswb/jstest_32.pcm \
>-	../Transcoding"$PLATFORM"/APITest_refTrans50.pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    
>-    echo "-- 56 kbps trans-coding to 50 kbps, 30 ms, SUPER-WIDEBAND"
>-    echo
>-    $ISAC -I -B 56000 -FL 30 -FS 32  -T 50000  ../Transcoding"$PLATFORM"/APITest_56T50.pcm \
>-	../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    
>-    echo "-- 56 kbps trans-coding to 20 kbps, 30 ms, SUPER-WIDEBAND"
>-    echo
>-    $ISAC -I -B 56000 -FL 30 -FS 32  -T 20000 ../Transcoding"$PLATFORM"/APITest_56T20.pcm \
>-	../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo "________________________________________________________"
>-    echo
>-    echo
>-    echo 
>-    echo
>-    echo
>-    
>-    echo "                                         Test FEC"
>-    echo "========================================================================================"
>-    mkdir -p ../FEC"$PLATFORM"
>-    rm -f ../FEC"$PLATFORM"/*
>-    echo
>-    echo
>-    echo "-- 32 kbps with transcoding to 20kbps, 30 ms, WIDEBAND, 10% packet loss"
>-    $ISAC -I -B 32000 -FL 30 -FS 16 -PL 10 -T 20000 ../FEC"$PLATFORM"/APITest_PL10_WB30_T20.pcm \
>-	../data/orig/speech_and_misc_WB.pcm ../FEC"$PLATFORM"/APITest_PL10_WB30.pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    
>-    echo "-- 32 kbps, 60 ms, WIDEBAND, 10% packet loss"
>-    $ISAC -I -B 32000 -FL 60 -FS 16 -PL 10 ../data/orig/speech_and_misc_WB.pcm \
>-	../FEC"$PLATFORM"/APITest_PL10_WB60.pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-    
>-    echo "-- 32 kbps with transcoding to 20 kbps, 30 ms, SUPER-WIDEBAND, 10% packet loss"
>-    $ISAC -I -B 32000 -FL 30 -FS 32 -PL 10 -T 20000 ../FEC"$PLATFORM"/APITest_PL10_SWB_8kHz_T20.pcm \
>-	../data/origswb/jstest_32.pcm ../FEC"$PLATFORM"/APITest_PL10_SWB_8kHz.pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-
>-    echo "-- 45 kbps with Trascoding to 38 kbps, 30 ms, SUPER-WIDEBAND, 10% packet loss"
>-    $ISAC -I -B 45000 -FL 30 -FS 32 -PL 10 -T 38000 ../FEC"$PLATFORM"/APITest_PL10_SWB_12kHz_T38.pcm \
>-	../data/origswb/jstest_32.pcm ../FEC"$PLATFORM"/APITest_PL10_SWB_12kHz.pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-
>-    echo "-- 56 kbps with transcoding to 50 kbps, 30 ms, SUPER-WIDEBAND, 10% packet loss"
>-    $ISAC -I -B 56000 -FL 30 -FS 32 -PL 10 -T 50000 ../FEC"$PLATFORM"/APITest_PL10_SWB_16kHz_T50.pcm \
>-	../data/origswb/jstest_32.pcm ../FEC"$PLATFORM"/APITest_PL10_SWB_16kHz.pcm >> $LOGFILE_API
>-    let "idx=idx+1"
>-    echo
>-    echo
>-fi
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACfault.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACfault.txt
>deleted file mode 100644
>index 63829a4b9834ec32f24f8bd9f783eabb5bdd29ba..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACfault.txt
>+++ /dev/null
>@@ -1,80 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character????? ?????? 
>-if  [ "$1" = "x64" ] || [ "$#" -eq 0 ]
>-    then
>-    PLATFORM=_X64
>-    ISAC=../x64/Release/ReleaseTest-API_2005.exe
>-elif [ "$1" = "2005" ]
>-	then
>-    PLATFORM=_2005
>-    ISAC=../win32/Release/ReleaseTest-API_2005.exe
>-elif [ "$1" == "LINUX" ]
>-	then
>-    PLATFORM=_linux
>-    ISAC=../ReleaseTest-API/isacswtest
>-else
>-	echo Unknown Platform
>-	exit 2
>-fi
>-
>-LOGFILE=logfault$PLATFORM.txt
>-echo "START FAULT TEST" > $LOGFILE
>-
>-
>-INFILES=$(cat InputFiles.txt)
>-SUBSET=$(cat InputFilesFew.txt)
>-CHANNELFILES=$(cat ChannelFiles.txt)
>-CHANNELLIST=($(cat ChannelFiles.txt))
>-INDIR=../data/orig
>-INDIRSWB=../data/origswb
>-OUTDIR=../dataqaft$PLATFORM
>-mkdir -p $OUTDIR
>-
>-#maximum Target rate for different bandwidth
>-TARGETRATE=( 32000 32000 44000 56000 )
>-SAMPFREQ=( 16 32 32 32 )
>-FAULTTEST=(1 2 3 4 5 6 7 9)
>-
>-index1=0
>-
>-file_wb=../data/orig/16kHz.pcm
>-file_swb=../data/origswb/32kHz.pcm
>-
>-for idx in 0 1 2 3 
>-  do
>-# Fault test
>-  echo
>-  echo "Sampling Frequency " ${SAMPFREQ[idx]} "kHz, Rate " ${TARGETRATE[idx]} "bps."
>-  echo "---------------------------------------------------"
>-  if [ ${SAMPFREQ[idx]} -eq 16 ]; then
>-    file=$file_wb
>-  else
>-    file=$file_swb
>-  fi
>-
>-  for testnr in ${FAULTTEST[*]}
>-    do
>-    echo "Running Fault Test " $testnr
>-    $ISAC -I -B "${TARGETRATE[idx]}" -F $testnr -FS "${SAMPFREQ[idx]}" "$file" \
>-	$OUTDIR/ft"$testnr"_"${TARGETRATE[idx]}"_"${SAMPFREQ[idx]}".pcm >> LOGFILE
>-    echo
>-    
>-  done
>-  
>-# Fault test number 10, error in bitstream
>-  echo "Running Fault Test 10"
>-  $ISAC -I -B "${TARGETRATE[idx]}" -F 10        -FS "${SAMPFREQ[idx]}" "$file" \
>-    $OUTDIR/ft10_"${TARGETRATE[idx]}"_"${SAMPFREQ[idx]}".pcm >> LOGFILE
>-  echo
>-  echo "Running Fault Test 10 with packetloss"
>-  $ISAC -I -B "${TARGETRATE[idx]}" -F 10 -PL 10 -FS "${SAMPFREQ[idx]}" "$file" \
>-    $OUTDIR/ft10plc_"${TARGETRATE[idx]}"_"${SAMPFREQ[idx]}".pcm >> LOGFILE
>-  echo
>-done
>-
>-echo 
>-echo
>-echo DONE!
>-  
>-
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACfixfloat.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACfixfloat.txt
>deleted file mode 100644
>index 4cda78e4fea6b53a79ff7409935e7641284b9278..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACfixfloat.txt
>+++ /dev/null
>@@ -1,47 +0,0 @@
>-#!/bin/bash
>-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
>-
>-LOGFILE=logfxfl.txt
>-echo "START FIX-FLOAT TEST" > $LOGFILE
>-
>-
>-ISACFIXFLOAT=../../../fix/test/testFixFloat.exe
>-
>-INFILES=$(cat InputFiles.txt)
>-SUBSET=$(cat InputFilesFew.txt)
>-CHANNELFILES=$(cat ChannelFiles.txt)
>-CHANNELLIST=($(cat ChannelFiles.txt))
>-INDIR=../data/orig
>-OUTDIR=../dataqafxfl
>-mkdir -p $OUTDIR
>-
>-index1=0
>-
>-for file in $INFILES # loop over all input files
>-  do
>-  
>-  for channel in $CHANNELFILES
>-	do
>-	let "index1=index1+1"
>-
>-	$ISACFIXFLOAT $INDIR/$channel -m 1 -PLC $INDIR/"$file" $OUTDIR/flfx$index1"$file" >> $LOGFILE
>-	$ISACFIXFLOAT $INDIR/$channel -m 2 -PLC $INDIR/"$file" $OUTDIR/fxfl$index1"$file" >> $LOGFILE
>-  done
>-
>-done
>-
>-index1=0
>-
>-for file in $SUBSET # loop over the subset of input files
>-  do
>-	let "index1=index1+1"
>-	$ISACFIXFLOAT $INDIR/$channel -m 1 -NB 1 $INDIR/"$file" $OUTDIR/flfxnb1_$index1"$file" >> $LOGFILE
>-	$ISACFIXFLOAT $INDIR/$channel -m 2 -NB 1 $INDIR/"$file" $OUTDIR/fxflnb1_$index1"$file" >> $LOGFILE
>-	$ISACFIXFLOAT $INDIR/$channel -m 1 -NB 2 -PLC $INDIR/"$file" $OUTDIR/flfxnb2_$index1"$file" >> $LOGFILE
>-	$ISACFIXFLOAT $INDIR/$channel -m 2 -NB 2 -PLC $INDIR/"$file" $OUTDIR/fxflnb2_$index1"$file" >> $LOGFILE
>-done
>-
>-echo DONE!
>-
>-
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
>index 0bf3b19a3b17cb346bccd887762a277b2ec1e405..d9efc211da3dac6800b21b5d979db91bd6de6831 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
>@@ -14,6 +14,8 @@
> #include <memory>
> #include <utility>
> 
>+#include "rtc_base/checks.h"
>+
> namespace webrtc {
> 
> LegacyEncodedAudioFrame::LegacyEncodedAudioFrame(AudioDecoder* decoder,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
>index 05d4fe489c91de0a3c06bddb6cffff4fb205c85c..41b08f7898e2e8fbf964ccb31fe21349135ec010 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
>@@ -11,10 +11,14 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
> #define MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
> #include "api/audio_codecs/audio_decoder.h"
>+#include "rtc_base/buffer.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
>index 302b71403196aade86bb33fb1d4d8f49c778d26e..357cb1a20da799dd0288dd1ffe8446c75958ba61 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
>@@ -10,8 +10,11 @@
> 
> #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
> 
>+#include <memory>
> #include <utility>
> 
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
>index 70aa40bdc46b01e88b9ff0aa0f5fcec27f7e00ec..8043425243cd9ed564b704ce332331093946eb40 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
>@@ -11,8 +11,13 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
> #define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+#include <vector>
>+
> #include "api/audio_codecs/audio_decoder.h"
> #include "modules/audio_coding/codecs/opus/opus_interface.h"
>+#include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
>index e6240e69826a140f86818cc737203c0816c855de..1a88acf3c03526fb61e2bf460954254e41266c5b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
>@@ -15,7 +15,8 @@
> #include <utility>
> 
> #include "absl/memory/memory.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
>+#include "common_types.h"
> #include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
> #include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
> #include "modules/audio_coding/codecs/opus/opus_interface.h"
>@@ -214,8 +215,80 @@ int GetBitrateBps(const AudioEncoderOpusConfig& config) {
>   return *config.bitrate_bps;
> }
> 
>+bool IsValidPacketLossRate(int value) {
>+  return value >= 0 && value <= 100;
>+}
>+
>+float ToFraction(int percent) {
>+  return static_cast<float>(percent) / 100;
>+}
>+
>+float GetMinPacketLossRate() {
>+  constexpr char kPacketLossFieldTrial[] = "WebRTC-Audio-OpusMinPacketLossRate";
>+  const bool use_opus_min_packet_loss_rate =
>+      webrtc::field_trial::IsEnabled(kPacketLossFieldTrial);
>+  if (use_opus_min_packet_loss_rate) {
>+    const std::string field_trial_string =
>+        webrtc::field_trial::FindFullName(kPacketLossFieldTrial);
>+    constexpr int kDefaultMinPacketLossRate = 1;
>+    int value = kDefaultMinPacketLossRate;
>+    if (sscanf(field_trial_string.c_str(), "Enabled-%d", &value) == 1 &&
>+        !IsValidPacketLossRate(value)) {
>+      RTC_LOG(LS_WARNING) << "Invalid parameter for " << kPacketLossFieldTrial
>+                          << ", using default value: "
>+                          << kDefaultMinPacketLossRate;
>+      value = kDefaultMinPacketLossRate;
>+    }
>+    return ToFraction(value);
>+  }
>+  return 0.0;
>+}
>+
>+std::unique_ptr<AudioEncoderOpusImpl::NewPacketLossRateOptimizer>
>+GetNewPacketLossRateOptimizer() {
>+  constexpr char kPacketLossOptimizationName[] =
>+      "WebRTC-Audio-NewOpusPacketLossRateOptimization";
>+  const bool use_new_packet_loss_optimization =
>+      webrtc::field_trial::IsEnabled(kPacketLossOptimizationName);
>+  if (use_new_packet_loss_optimization) {
>+    const std::string field_trial_string =
>+        webrtc::field_trial::FindFullName(kPacketLossOptimizationName);
>+    int min_rate;
>+    int max_rate;
>+    float slope;
>+    if (sscanf(field_trial_string.c_str(), "Enabled-%d-%d-%f", &min_rate,
>+               &max_rate, &slope) == 3 &&
>+        IsValidPacketLossRate(min_rate) && IsValidPacketLossRate(max_rate)) {
>+      return absl::make_unique<
>+          AudioEncoderOpusImpl::NewPacketLossRateOptimizer>(
>+          ToFraction(min_rate), ToFraction(max_rate), slope);
>+    }
>+    RTC_LOG(LS_WARNING) << "Invalid parameters for "
>+                        << kPacketLossOptimizationName
>+                        << ", using default values.";
>+    return absl::make_unique<
>+        AudioEncoderOpusImpl::NewPacketLossRateOptimizer>();
>+  }
>+  return nullptr;
>+}
>+
> }  // namespace
> 
>+AudioEncoderOpusImpl::NewPacketLossRateOptimizer::NewPacketLossRateOptimizer(
>+    float min_packet_loss_rate,
>+    float max_packet_loss_rate,
>+    float slope)
>+    : min_packet_loss_rate_(min_packet_loss_rate),
>+      max_packet_loss_rate_(max_packet_loss_rate),
>+      slope_(slope) {}
>+
>+float AudioEncoderOpusImpl::NewPacketLossRateOptimizer::OptimizePacketLossRate(
>+    float packet_loss_rate) const {
>+  packet_loss_rate = slope_ * packet_loss_rate;
>+  return std::min(std::max(packet_loss_rate, min_packet_loss_rate_),
>+                  max_packet_loss_rate_);
>+}
>+
> void AudioEncoderOpusImpl::AppendSupportedEncoders(
>     std::vector<AudioCodecSpec>* specs) {
>   const SdpAudioFormat fmt = {
>@@ -244,7 +317,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
> 
> absl::optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
>     const SdpAudioFormat& format) {
>-  if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
>+  if (absl::EqualsIgnoreCase(format.name, GetPayloadName()) &&
>       format.clockrate_hz == 48000 && format.num_channels == 2) {
>     const size_t num_channels = GetChannelCount(format);
>     const int bitrate =
>@@ -276,7 +349,7 @@ AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
> 
> absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
>     const SdpAudioFormat& format) {
>-  if (STR_CASE_CMP(format.name.c_str(), "opus") != 0 ||
>+  if (!absl::EqualsIgnoreCase(format.name, "opus") ||
>       format.clockrate_hz != 48000 || format.num_channels != 2) {
>     return absl::nullopt;
>   }
>@@ -398,10 +471,14 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(
>     : payload_type_(payload_type),
>       send_side_bwe_with_overhead_(
>           webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
>+      use_link_capacity_for_adaptation_(webrtc::field_trial::IsEnabled(
>+          "WebRTC-Audio-LinkCapacityAdaptation")),
>       adjust_bandwidth_(
>           webrtc::field_trial::IsEnabled("WebRTC-AdjustOpusBandwidth")),
>       bitrate_changed_(true),
>       packet_loss_rate_(0.0),
>+      min_packet_loss_rate_(GetMinPacketLossRate()),
>+      new_packet_loss_optimizer_(GetNewPacketLossRateOptimizer()),
>       inst_(nullptr),
>       packet_loss_fraction_smoother_(new PacketLossFractionSmoother()),
>       audio_network_adaptor_creator_(audio_network_adaptor_creator),
>@@ -414,6 +491,7 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(
>   RTC_CHECK(config.payload_type == -1 || config.payload_type == payload_type);
> 
>   RTC_CHECK(RecreateEncoderInstance(config));
>+  SetProjectedPacketLossRate(packet_loss_rate_);
> }
> 
> AudioEncoderOpusImpl::AudioEncoderOpusImpl(const CodecInst& codec_inst)
>@@ -529,7 +607,8 @@ void AudioEncoderOpusImpl::OnReceivedUplinkRecoverablePacketLossFraction(
> 
> void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
>     int target_audio_bitrate_bps,
>-    absl::optional<int64_t> bwe_period_ms) {
>+    absl::optional<int64_t> bwe_period_ms,
>+    absl::optional<int64_t> link_capacity_allocation_bps) {
>   if (audio_network_adaptor_) {
>     audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
>     // We give smoothed bitrate allocation to audio network adaptor as
>@@ -547,6 +626,9 @@ void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
>       bitrate_smoother_->SetTimeConstantMs(*bwe_period_ms * 4);
>     bitrate_smoother_->AddSample(target_audio_bitrate_bps);
> 
>+    if (link_capacity_allocation_bps)
>+      link_capacity_allocation_bps_ = link_capacity_allocation_bps;
>+
>     ApplyAudioNetworkAdaptor();
>   } else if (send_side_bwe_with_overhead_) {
>     if (!overhead_bytes_per_packet_) {
>@@ -565,6 +647,18 @@ void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
>     SetTargetBitrate(target_audio_bitrate_bps);
>   }
> }
>+void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
>+    int target_audio_bitrate_bps,
>+    absl::optional<int64_t> bwe_period_ms) {
>+  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, bwe_period_ms,
>+                            absl::nullopt);
>+}
>+
>+void AudioEncoderOpusImpl::OnReceivedUplinkAllocation(
>+    BitrateAllocationUpdate update) {
>+  OnReceivedUplinkBandwidth(update.target_bitrate.bps(), update.bwe_period.ms(),
>+                            update.link_capacity.bps());
>+}
> 
> void AudioEncoderOpusImpl::OnReceivedRtt(int rtt_ms) {
>   if (!audio_network_adaptor_)
>@@ -738,9 +832,14 @@ void AudioEncoderOpusImpl::SetNumChannelsToEncode(
> }
> 
> void AudioEncoderOpusImpl::SetProjectedPacketLossRate(float fraction) {
>-  float opt_loss_rate = OptimizePacketLossRate(fraction, packet_loss_rate_);
>-  if (packet_loss_rate_ != opt_loss_rate) {
>-    packet_loss_rate_ = opt_loss_rate;
>+  if (new_packet_loss_optimizer_) {
>+    fraction = new_packet_loss_optimizer_->OptimizePacketLossRate(fraction);
>+  } else {
>+    fraction = OptimizePacketLossRate(fraction, packet_loss_rate_);
>+    fraction = std::max(fraction, min_packet_loss_rate_);
>+  }
>+  if (packet_loss_rate_ != fraction) {
>+    packet_loss_rate_ = fraction;
>     RTC_CHECK_EQ(
>         0, WebRtcOpus_SetPacketLossRate(
>                inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
>@@ -794,14 +893,20 @@ AudioEncoderOpusImpl::DefaultAudioNetworkAdaptorCreator(
> 
> void AudioEncoderOpusImpl::MaybeUpdateUplinkBandwidth() {
>   if (audio_network_adaptor_) {
>-    int64_t now_ms = rtc::TimeMillis();
>-    if (!bitrate_smoother_last_update_time_ ||
>-        now_ms - *bitrate_smoother_last_update_time_ >=
>-            config_.uplink_bandwidth_update_interval_ms) {
>-      absl::optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
>-      if (smoothed_bitrate)
>-        audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
>-      bitrate_smoother_last_update_time_ = now_ms;
>+    if (use_link_capacity_for_adaptation_ && link_capacity_allocation_bps_) {
>+      audio_network_adaptor_->SetUplinkBandwidth(
>+          *link_capacity_allocation_bps_);
>+    } else {
>+      int64_t now_ms = rtc::TimeMillis();
>+      if (!bitrate_smoother_last_update_time_ ||
>+          now_ms - *bitrate_smoother_last_update_time_ >=
>+              config_.uplink_bandwidth_update_interval_ms) {
>+        absl::optional<float> smoothed_bitrate =
>+            bitrate_smoother_->GetAverage();
>+        if (smoothed_bitrate)
>+          audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
>+        bitrate_smoother_last_update_time_ = now_ms;
>+      }
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
>index ea4b2650bbbe73646a39b1d78de5d62aa5e29ec4..150423fd3ee7aa7ec4a73cf4eebb8889e17816cb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
>@@ -34,6 +34,26 @@ struct CodecInst;
> 
> class AudioEncoderOpusImpl final : public AudioEncoder {
>  public:
>+  class NewPacketLossRateOptimizer {
>+   public:
>+    NewPacketLossRateOptimizer(float min_packet_loss_rate = 0.01,
>+                               float max_packet_loss_rate = 0.2,
>+                               float slope = 1.0);
>+
>+    float OptimizePacketLossRate(float packet_loss_rate) const;
>+
>+    // Getters for testing.
>+    float min_packet_loss_rate() const { return min_packet_loss_rate_; };
>+    float max_packet_loss_rate() const { return max_packet_loss_rate_; };
>+    float slope() const { return slope_; };
>+
>+   private:
>+    const float min_packet_loss_rate_;
>+    const float max_packet_loss_rate_;
>+    const float slope_;
>+    RTC_DISALLOW_COPY_AND_ASSIGN(NewPacketLossRateOptimizer);
>+  };
>+
>   static AudioEncoderOpusConfig CreateConfig(const CodecInst& codec_inst);
> 
>   // Returns empty if the current bitrate falls within the hysteresis window,
>@@ -99,6 +119,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
>   void OnReceivedUplinkBandwidth(
>       int target_audio_bitrate_bps,
>       absl::optional<int64_t> bwe_period_ms) override;
>+  void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
>   void OnReceivedRtt(int rtt_ms) override;
>   void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
>   void SetReceiverFrameLengthRange(int min_frame_length_ms,
>@@ -110,6 +131,9 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
> 
>   // Getters for testing.
>   float packet_loss_rate() const { return packet_loss_rate_; }
>+  NewPacketLossRateOptimizer* new_packet_loss_optimizer() const {
>+    return new_packet_loss_optimizer_.get();
>+  }
>   AudioEncoderOpusConfig::ApplicationMode application() const {
>     return config_.application;
>   }
>@@ -141,6 +165,11 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
>   void SetNumChannelsToEncode(size_t num_channels_to_encode);
>   void SetProjectedPacketLossRate(float fraction);
> 
>+  void OnReceivedUplinkBandwidth(
>+      int target_audio_bitrate_bps,
>+      absl::optional<int64_t> bwe_period_ms,
>+      absl::optional<int64_t> link_capacity_allocation);
>+
>   // TODO(minyue): remove "override" when we can deprecate
>   // |AudioEncoder::SetTargetBitrate|.
>   void SetTargetBitrate(int target_bps) override;
>@@ -155,9 +184,12 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
>   AudioEncoderOpusConfig config_;
>   const int payload_type_;
>   const bool send_side_bwe_with_overhead_;
>+  const bool use_link_capacity_for_adaptation_;
>   const bool adjust_bandwidth_;
>   bool bitrate_changed_;
>   float packet_loss_rate_;
>+  const float min_packet_loss_rate_;
>+  const std::unique_ptr<NewPacketLossRateOptimizer> new_packet_loss_optimizer_;
>   std::vector<int16_t> input_buffer_;
>   OpusEncInst* inst_;
>   uint32_t first_timestamp_in_buffer_;
>@@ -170,6 +202,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
>   absl::optional<size_t> overhead_bytes_per_packet_;
>   const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
>   absl::optional<int64_t> bitrate_smoother_last_update_time_;
>+  absl::optional<int64_t> link_capacity_allocation_bps_;
>   int consecutive_dtx_frames_;
> 
>   friend struct AudioEncoderOpus;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
>index 7a6d5fda4bbc16141ab84a935cd7334ed79b77de..d5c2c84f1ff713a1bcb61701b1558626a6b216ce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
>@@ -272,6 +272,51 @@ TEST(AudioEncoderOpusTest, PacketLossRateOptimized) {
>   // clang-format on
> }
> 
>+TEST(AudioEncoderOpusTest, PacketLossRateLowerBounded) {
>+  test::ScopedFieldTrials override_field_trials(
>+      "WebRTC-Audio-OpusMinPacketLossRate/Enabled-5/");
>+  auto states = CreateCodec(1);
>+  auto I = [](float a, float b) { return IntervalSteps(a, b, 10); };
>+  constexpr float eps = 1e-8f;
>+
>+  // clang-format off
>+  TestSetPacketLossRate(states.get(), I(0.00f      , 0.01f - eps), 0.05f);
>+  TestSetPacketLossRate(states.get(), I(0.01f + eps, 0.06f - eps), 0.05f);
>+  TestSetPacketLossRate(states.get(), I(0.06f + eps, 0.11f - eps), 0.05f);
>+  TestSetPacketLossRate(states.get(), I(0.11f + eps, 0.22f - eps), 0.10f);
>+  TestSetPacketLossRate(states.get(), I(0.22f + eps, 1.00f      ), 0.20f);
>+
>+  TestSetPacketLossRate(states.get(), I(1.00f      , 0.18f + eps), 0.20f);
>+  TestSetPacketLossRate(states.get(), I(0.18f - eps, 0.09f + eps), 0.10f);
>+  TestSetPacketLossRate(states.get(), I(0.09f - eps, 0.04f + eps), 0.05f);
>+  TestSetPacketLossRate(states.get(), I(0.04f - eps, 0.01f + eps), 0.05f);
>+  TestSetPacketLossRate(states.get(), I(0.01f - eps, 0.00f      ), 0.05f);
>+  // clang-format on
>+}
>+
>+TEST(AudioEncoderOpusTest, NewPacketLossRateOptimization) {
>+  {
>+    test::ScopedFieldTrials override_field_trials(
>+        "WebRTC-Audio-NewOpusPacketLossRateOptimization/Enabled-5-15-0.5/");
>+    auto states = CreateCodec(1);
>+
>+    TestSetPacketLossRate(states.get(), {0.00f}, 0.05f);
>+    TestSetPacketLossRate(states.get(), {0.12f}, 0.06f);
>+    TestSetPacketLossRate(states.get(), {0.22f}, 0.11f);
>+    TestSetPacketLossRate(states.get(), {0.50f}, 0.15f);
>+  }
>+  {
>+    test::ScopedFieldTrials override_field_trials(
>+        "WebRTC-Audio-NewOpusPacketLossRateOptimization/Enabled/");
>+    auto states = CreateCodec(1);
>+
>+    TestSetPacketLossRate(states.get(), {0.00f}, 0.01f);
>+    TestSetPacketLossRate(states.get(), {0.12f}, 0.12f);
>+    TestSetPacketLossRate(states.get(), {0.22f}, 0.20f);
>+    TestSetPacketLossRate(states.get(), {0.50f}, 0.20f);
>+  }
>+}
>+
> TEST(AudioEncoderOpusTest, SetReceiverFrameLengthRange) {
>   auto states = CreateCodec(2);
>   // Before calling to |SetReceiverFrameLengthRange|,
>@@ -446,6 +491,57 @@ TEST(AudioEncoderOpusTest, BitrateBounded) {
>   EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
> }
> 
>+TEST(AudioEncoderOpusTest, MinPacketLossRate) {
>+  constexpr float kDefaultMinPacketLossRate = 0.01;
>+  {
>+    test::ScopedFieldTrials override_field_trials(
>+        "WebRTC-Audio-OpusMinPacketLossRate/Enabled/");
>+    auto states = CreateCodec(1);
>+    EXPECT_EQ(kDefaultMinPacketLossRate, states->encoder->packet_loss_rate());
>+  }
>+  {
>+    test::ScopedFieldTrials override_field_trials(
>+        "WebRTC-Audio-OpusMinPacketLossRate/Enabled-200/");
>+    auto states = CreateCodec(1);
>+    EXPECT_EQ(kDefaultMinPacketLossRate, states->encoder->packet_loss_rate());
>+  }
>+  {
>+    test::ScopedFieldTrials override_field_trials(
>+        "WebRTC-Audio-OpusMinPacketLossRate/Enabled-50/");
>+    constexpr float kMinPacketLossRate = 0.5;
>+    auto states = CreateCodec(1);
>+    EXPECT_EQ(kMinPacketLossRate, states->encoder->packet_loss_rate());
>+  }
>+}
>+
>+TEST(AudioEncoderOpusTest, NewPacketLossRateOptimizer) {
>+  {
>+    auto states = CreateCodec(1);
>+    auto optimizer = states->encoder->new_packet_loss_optimizer();
>+    EXPECT_EQ(nullptr, optimizer);
>+  }
>+  {
>+    test::ScopedFieldTrials override_field_trials(
>+        "WebRTC-Audio-NewOpusPacketLossRateOptimization/Enabled/");
>+    auto states = CreateCodec(1);
>+    auto optimizer = states->encoder->new_packet_loss_optimizer();
>+    ASSERT_NE(nullptr, optimizer);
>+    EXPECT_FLOAT_EQ(0.01, optimizer->min_packet_loss_rate());
>+    EXPECT_FLOAT_EQ(0.20, optimizer->max_packet_loss_rate());
>+    EXPECT_FLOAT_EQ(1.00, optimizer->slope());
>+  }
>+  {
>+    test::ScopedFieldTrials override_field_trials(
>+        "WebRTC-Audio-NewOpusPacketLossRateOptimization/Enabled-2-50-0.7/");
>+    auto states = CreateCodec(1);
>+    auto optimizer = states->encoder->new_packet_loss_optimizer();
>+    ASSERT_NE(nullptr, optimizer);
>+    EXPECT_FLOAT_EQ(0.02, optimizer->min_packet_loss_rate());
>+    EXPECT_FLOAT_EQ(0.50, optimizer->max_packet_loss_rate());
>+    EXPECT_FLOAT_EQ(0.70, optimizer->slope());
>+  }
>+}
>+
> // Verifies that the complexity adaptation in the config works as intended.
> TEST(AudioEncoderOpusTest, ConfigComplexityAdaptation) {
>   AudioEncoderOpusConfig config;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
>index b07624dffc5c4e32c8363ee0a6f87910504a0560..1dd2ff289ee10280963c3db5de58545886c60d49 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
> 
>+#include <utility>
>+
> #include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
> #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
>index 7d234224e3cd00daa8e81b8ad3374c1793185939..9b478d806f7ac605d1a94933168cb0d251fa6141 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
>@@ -11,7 +11,12 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
> #define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+#include <vector>
>+
> #include "api/audio_codecs/audio_decoder.h"
>+#include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
>index 831daedc63f9d1f62bd33cff133ee8d0f8ea3326..106ab16794025183ada40181b6d4654095471b04 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
>@@ -10,12 +10,9 @@
> 
> #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
> 
>-#include <algorithm>
>-
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "common_types.h"
> #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/numerics/safe_conversions.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
>index 6d0fc2d015cc58868b6f8b01dedaf8fd118966c5..8f8bba527be45f90b9708c8a1c0b9c3e0ebba662 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
>@@ -10,6 +10,9 @@
> 
> #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
> 
>+#include <stdint.h>
>+#include <initializer_list>
>+
> namespace webrtc {
> 
> void Pcm16BAppendSupportedCodecSpecs(std::vector<AudioCodecSpec>* specs) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h
>index 980a9964088642dac8922de816236bb450ea6071..3fae717ff31d9c4c1ae3019097a555ad9c4678a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h
>@@ -13,7 +13,7 @@
> 
> #include <vector>
> 
>-#include "api/audio_codecs/audio_decoder_factory.h"
>+#include "api/audio_codecs/audio_format.h"
> 
> namespace webrtc {
> void Pcm16BAppendSupportedCodecSpecs(std::vector<AudioCodecSpec>* specs);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
>index 2601f26e44fddacdda18f4ab0edb7344170db2c6..124e81198179cd3cc05824b674d9772c5dc227a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
>@@ -11,8 +11,8 @@
> #include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
> 
> #include <string.h>
>-
> #include <utility>
>+#include <vector>
> 
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
>index 492ee3afb471763b87d2777d605830696db2b279..5a68876b898c143d56a1213f8bf460fa30728e59 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
>@@ -11,9 +11,12 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
> #define MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
>-#include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module.h
>index b9f22288fd9fce32b5f1e5f979be47f1cdc500a3..f9fdba5f511b33d0ab902aa9b2ec0dbd081a58b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module.h
>@@ -154,40 +154,6 @@ class AudioCodingModule {
>   //   Sender
>   //
> 
>-  ///////////////////////////////////////////////////////////////////////////
>-  // int32_t RegisterSendCodec()
>-  // Registers a codec, specified by |send_codec|, as sending codec.
>-  // This API can be called multiple of times to register Codec. The last codec
>-  // registered overwrites the previous ones.
>-  // The API can also be used to change payload type for CNG and RED, which are
>-  // registered by default to default payload types.
>-  // Note that registering CNG and RED won't overwrite speech codecs.
>-  // This API can be called to set/change the send payload-type, frame-size
>-  // or encoding rate (if applicable for the codec).
>-  //
>-  // Note: If a stereo codec is registered as send codec, VAD/DTX will
>-  // automatically be turned off, since it is not supported for stereo sending.
>-  //
>-  // Note: If a secondary encoder is already registered, and the new send-codec
>-  // has a sampling rate that does not match the secondary encoder, the
>-  // secondary encoder will be unregistered.
>-  //
>-  // Input:
>-  //   -send_codec         : Parameters of the codec to be registered, c.f.
>-  //                         common_types.h for the definition of
>-  //                         CodecInst.
>-  //
>-  // Return value:
>-  //   -1 if failed to initialize,
>-  //    0 if succeeded.
>-  //
>-  virtual int32_t RegisterSendCodec(const CodecInst& send_codec) = 0;
>-
>-  // Registers |external_speech_encoder| as encoder. The new encoder will
>-  // replace any previously registered speech encoder (internal or external).
>-  virtual void RegisterExternalSendCodec(
>-      AudioEncoder* external_speech_encoder) = 0;
>-
>   // |modifier| is called exactly once with one argument: a pointer to the
>   // unique_ptr that holds the current encoder (which is null if there is no
>   // current encoder). For the duration of the call, |modifier| has exclusive
>@@ -257,71 +223,6 @@ class AudioCodingModule {
>   //
>   virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0;
> 
>-  ///////////////////////////////////////////////////////////////////////////
>-  // (RED) Redundant Coding
>-  //
>-
>-  ///////////////////////////////////////////////////////////////////////////
>-  // int32_t SetREDStatus()
>-  // configure RED status i.e. on/off.
>-  //
>-  // RFC 2198 describes a solution which has a single payload type which
>-  // signifies a packet with redundancy. That packet then becomes a container,
>-  // encapsulating multiple payloads into a single RTP packet.
>-  // Such a scheme is flexible, since any amount of redundancy may be
>-  // encapsulated within a single packet.  There is, however, a small overhead
>-  // since each encapsulated payload must be preceded by a header indicating
>-  // the type of data enclosed.
>-  //
>-  // Input:
>-  //   -enable_red         : if true RED is enabled, otherwise RED is
>-  //                         disabled.
>-  //
>-  // Return value:
>-  //   -1 if failed to set RED status,
>-  //    0 if succeeded.
>-  //
>-  virtual int32_t SetREDStatus(bool enable_red) = 0;
>-
>-  ///////////////////////////////////////////////////////////////////////////
>-  // bool REDStatus()
>-  // Get RED status
>-  //
>-  // Return value:
>-  //   true if RED is enabled,
>-  //   false if RED is disabled.
>-  //
>-  virtual bool REDStatus() const = 0;
>-
>-  ///////////////////////////////////////////////////////////////////////////
>-  // (FEC) Forward Error Correction (codec internal)
>-  //
>-
>-  ///////////////////////////////////////////////////////////////////////////
>-  // int32_t SetCodecFEC()
>-  // Configures codec internal FEC status i.e. on/off. No effects on codecs that
>-  // do not provide internal FEC.
>-  //
>-  // Input:
>-  //   -enable_fec         : if true FEC will be enabled otherwise the FEC is
>-  //                         disabled.
>-  //
>-  // Return value:
>-  //   -1 if failed, or the codec does not support FEC
>-  //    0 if succeeded.
>-  //
>-  virtual int SetCodecFEC(bool enable_codec_fec) = 0;
>-
>-  ///////////////////////////////////////////////////////////////////////////
>-  // bool CodecFEC()
>-  // Gets status of codec internal FEC.
>-  //
>-  // Return value:
>-  //   true if FEC is enabled,
>-  //   false if FEC is disabled.
>-  //
>-  virtual bool CodecFEC() const = 0;
>-
>   ///////////////////////////////////////////////////////////////////////////
>   // int SetPacketLossRate()
>   // Sets expected packet loss rate for encoding. Some encoders provide packet
>@@ -343,55 +244,6 @@ class AudioCodingModule {
>   //   (VAD) Voice Activity Detection
>   //
> 
>-  ///////////////////////////////////////////////////////////////////////////
>-  // int32_t SetVAD()
>-  // If DTX is enabled & the codec does not have internal DTX/VAD
>-  // WebRtc VAD will be automatically enabled and |enable_vad| is ignored.
>-  //
>-  // If DTX is disabled but VAD is enabled no DTX packets are send,
>-  // regardless of whether the codec has internal DTX/VAD or not. In this
>-  // case, WebRtc VAD is running to label frames as active/in-active.
>-  //
>-  // NOTE! VAD/DTX is not supported when sending stereo.
>-  //
>-  // Inputs:
>-  //   -enable_dtx         : if true DTX is enabled,
>-  //                         otherwise DTX is disabled.
>-  //   -enable_vad         : if true VAD is enabled,
>-  //                         otherwise VAD is disabled.
>-  //   -vad_mode           : determines the aggressiveness of VAD. A more
>-  //                         aggressive mode results in more frames labeled
>-  //                         as in-active, c.f. definition of
>-  //                         ACMVADMode in audio_coding_module_typedefs.h
>-  //                         for valid values.
>-  //
>-  // Return value:
>-  //   -1 if failed to set up VAD/DTX,
>-  //    0 if succeeded.
>-  //
>-  virtual int32_t SetVAD(const bool enable_dtx = true,
>-                         const bool enable_vad = false,
>-                         const ACMVADMode vad_mode = VADNormal) = 0;
>-
>-  ///////////////////////////////////////////////////////////////////////////
>-  // int32_t VAD()
>-  // Get VAD status.
>-  //
>-  // Outputs:
>-  //   -dtx_enabled        : is set to true if DTX is enabled, otherwise
>-  //                         is set to false.
>-  //   -vad_enabled        : is set to true if VAD is enabled, otherwise
>-  //                         is set to false.
>-  //   -vad_mode            : is set to the current aggressiveness of VAD.
>-  //
>-  // Return value:
>-  //   -1 if fails to retrieve the setting of DTX/VAD,
>-  //    0 if succeeded.
>-  //
>-  virtual int32_t VAD(bool* dtx_enabled,
>-                      bool* vad_enabled,
>-                      ACMVADMode* vad_mode) const = 0;
>-
>   ///////////////////////////////////////////////////////////////////////////
>   // int32_t RegisterVADCallback()
>   // Call this method to register a callback function which is called
>@@ -455,29 +307,6 @@ class AudioCodingModule {
>   virtual bool RegisterReceiveCodec(int rtp_payload_type,
>                                     const SdpAudioFormat& audio_format) = 0;
> 
>-  ///////////////////////////////////////////////////////////////////////////
>-  // int32_t RegisterReceiveCodec()
>-  // Register possible decoders, can be called multiple times for
>-  // codecs, CNG-NB, CNG-WB, CNG-SWB, AVT and RED.
>-  //
>-  // Input:
>-  //   -receive_codec      : parameters of the codec to be registered, c.f.
>-  //                         common_types.h for the definition of
>-  //                         CodecInst.
>-  //
>-  // Return value:
>-  //   -1 if failed to register the codec
>-  //    0 if the codec registered successfully.
>-  //
>-  virtual int RegisterReceiveCodec(const CodecInst& receive_codec) = 0;
>-
>-  // Register a decoder; call repeatedly to register multiple decoders. |df| is
>-  // a decoder factory that returns an iSAC decoder; it will be called once if
>-  // the decoder being registered is iSAC.
>-  virtual int RegisterReceiveCodec(
>-      const CodecInst& receive_codec,
>-      rtc::FunctionView<std::unique_ptr<AudioDecoder>()> isac_factory) = 0;
>-
>   // Registers an external decoder. The name is only used to provide information
>   // back to the caller about the decoder. Hence, the name is arbitrary, and may
>   // be empty.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
>index cd4351b07ff1b943450904390873296e673354e4..bafff72e5ad4b6acf01ea283de7c061561a5f360 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
>@@ -13,6 +13,8 @@
> 
> #include <map>
> 
>+#include "rtc_base/deprecation.h"
>+
> namespace webrtc {
> 
> ///////////////////////////////////////////////////////////////////////////
>@@ -43,6 +45,84 @@ enum OpusApplicationMode {
>   kAudio = 1,
> };
> 
>+// Statistics for calls to AudioCodingModule::PlayoutData10Ms().
>+struct AudioDecodingCallStats {
>+  AudioDecodingCallStats()
>+      : calls_to_silence_generator(0),
>+        calls_to_neteq(0),
>+        decoded_normal(0),
>+        decoded_plc(0),
>+        decoded_cng(0),
>+        decoded_plc_cng(0),
>+        decoded_muted_output(0) {}
>+
>+  int calls_to_silence_generator;  // Number of calls where silence generated,
>+                                   // and NetEq was disengaged from decoding.
>+  int calls_to_neteq;              // Number of calls to NetEq.
>+  int decoded_normal;  // Number of calls where audio RTP packet decoded.
>+  int decoded_plc;     // Number of calls resulted in PLC.
>+  int decoded_cng;  // Number of calls where comfort noise generated due to DTX.
>+  int decoded_plc_cng;       // Number of calls resulted where PLC faded to CNG.
>+  int decoded_muted_output;  // Number of calls returning a muted state output.
>+};
>+
>+// NETEQ statistics.
>+struct NetworkStatistics {
>+  // current jitter buffer size in ms
>+  uint16_t currentBufferSize;
>+  // preferred (optimal) buffer size in ms
>+  uint16_t preferredBufferSize;
>+  // adding extra delay due to "peaky jitter"
>+  bool jitterPeaksFound;
>+  // Stats below correspond to similarly-named fields in the WebRTC stats spec.
>+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
>+  uint64_t totalSamplesReceived;
>+  uint64_t concealedSamples;
>+  uint64_t concealmentEvents;
>+  uint64_t jitterBufferDelayMs;
>+  // Stats below DO NOT correspond directly to anything in the WebRTC stats
>+  // Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
>+  uint16_t currentPacketLossRate;
>+  // Late loss rate; fraction between 0 and 1, scaled to Q14.
>+  union {
>+    RTC_DEPRECATED uint16_t currentDiscardRate;
>+  };
>+  // fraction (of original stream) of synthesized audio inserted through
>+  // expansion (in Q14)
>+  uint16_t currentExpandRate;
>+  // fraction (of original stream) of synthesized speech inserted through
>+  // expansion (in Q14)
>+  uint16_t currentSpeechExpandRate;
>+  // fraction of synthesized speech inserted through pre-emptive expansion
>+  // (in Q14)
>+  uint16_t currentPreemptiveRate;
>+  // fraction of data removed through acceleration (in Q14)
>+  uint16_t currentAccelerateRate;
>+  // fraction of data coming from secondary decoding (in Q14)
>+  uint16_t currentSecondaryDecodedRate;
>+  // Fraction of secondary data, including FEC and RED, that is discarded (in
>+  // Q14). Discarding of secondary data can be caused by the reception of the
>+  // primary data, obsoleting the secondary data. It can also be caused by early
>+  // or late arrival of secondary data.
>+  uint16_t currentSecondaryDiscardedRate;
>+  // clock-drift in parts-per-million (negative or positive)
>+  int32_t clockDriftPPM;
>+  // average packet waiting time in the jitter buffer (ms)
>+  int meanWaitingTimeMs;
>+  // median packet waiting time in the jitter buffer (ms)
>+  int medianWaitingTimeMs;
>+  // min packet waiting time in the jitter buffer (ms)
>+  int minWaitingTimeMs;
>+  // max packet waiting time in the jitter buffer (ms)
>+  int maxWaitingTimeMs;
>+  // added samples in off mode due to packet loss
>+  size_t addedSamples;
>+  // count of the number of buffer flushes
>+  uint64_t packetBufferFlushes;
>+  // number of samples expanded due to delayed packets
>+  uint64_t delayedPacketOutageSamples;
>+};
>+
> }  // namespace webrtc
> 
> #endif  // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.cc
>index 18350b0a78971064654497bfda761ea1d30ffaf4..6161a8f91bf294fc4ccc52df4fc23bfadb3e6c89 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.cc
>@@ -10,7 +10,10 @@
> 
> #include "modules/audio_coding/neteq/accelerate.h"
> 
>-#include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include <assert.h>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_coding/neteq/audio_multi_vector.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.h
>index 560956895b7df63cd665dc4ba141aa933956068c..1a3af42b2e9f3c4c88b91a708f446d141e86e258 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.h
>@@ -11,13 +11,15 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
> #define MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
> 
>-#include "modules/audio_coding/neteq/audio_multi_vector.h"
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "modules/audio_coding/neteq/time_stretch.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
>-// Forward declarations.
>+class AudioMultiVector;
> class BackgroundNoise;
> 
> // This class implements the Accelerate operation. Most of the work is done
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.h
>index 86f82820ca7a77bd66d0c640d195066dce9ebb26..a2dd3c3cb10cfd4ccb58bb2a88d099778bbd7202 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.h
>@@ -11,8 +11,8 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
> #define MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
> 
>-#include <string.h>  // Access to size_t.
>-
>+#include <stdint.h>
>+#include <string.h>
> #include <vector>
> 
> #include "api/array_view.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.h
>index 825a3bceac9be0f3b5ed60a27f568e773f293d30..d0db332db65de848d302f2064b302ff11b103a6a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.h
>@@ -11,7 +11,8 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
> #define MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
> 
>-#include <string.h>  // Access to size_t.
>+#include <string.h>
>+#include <cstdint>
> #include <memory>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.h
>index 58eecaa705ce8252dc87bb7754c23c7e2292ace4..84d7eb9d745af36a47c4767e9cfad530db14eeef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.h
>@@ -14,13 +14,12 @@
> #include <string.h>  // size_t
> #include <memory>
> 
>-#include "modules/audio_coding/neteq/audio_multi_vector.h"
>-#include "modules/audio_coding/neteq/include/neteq.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
> // Forward declarations.
>+class AudioMultiVector;
> class PostDecodeVad;
> 
> // This class handles estimation of background noise parameters.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
>index 6e8da0a01b95fb0cb96f03e93fcfba2a60c5f216..2f966185362f94eee9cbb609fe88c0301ff9dbcf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
>@@ -10,7 +10,8 @@
> 
> #include "modules/audio_coding/neteq/buffer_level_filter.h"
> 
>-#include <algorithm>  // Provide access to std::max.
>+#include <stdint.h>
>+#include <algorithm>
> 
> #include "rtc_base/numerics/safe_conversions.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.cc
>index b341acd5a50d6e58b8dd2365af5c6d6bda724689..cb2b74dbf2b19a0af8f153c68e74e4f7805f89c1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.cc
>@@ -11,11 +11,18 @@
> #include "modules/audio_coding/neteq/comfort_noise.h"
> 
> #include <assert.h>
>+#include <cstdint>
>+#include <memory>
> 
>-#include "api/audio_codecs/audio_decoder.h"
>+#include "api/array_view.h"
>+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
>+#include "modules/audio_coding/neteq/audio_multi_vector.h"
>+#include "modules/audio_coding/neteq/audio_vector.h"
> #include "modules/audio_coding/neteq/decoder_database.h"
> #include "modules/audio_coding/neteq/dsp_helper.h"
> #include "modules/audio_coding/neteq/sync_buffer.h"
>+#include "rtc_base/buffer.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.h
>index 5169124040f677c04fedca12a0419dbc38320191..3a9bfde7aab5b4e61813d5d69ea4f9060383bbeb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.h
>@@ -11,12 +11,14 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
> #define MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
> 
>-#include "modules/audio_coding/neteq/audio_multi_vector.h"
>+#include <stddef.h>
>+
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
> // Forward declarations.
>+class AudioMultiVector;
> class DecoderDatabase;
> class SyncBuffer;
> struct Packet;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/cross_correlation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/cross_correlation.h
>index a747772cfc4352bb28da5265f039ca347f44e1b6..9ce8be83aecf0f52deab7cdb2b63e73c9cc66726 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/cross_correlation.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/cross_correlation.h
>@@ -11,7 +11,8 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
> #define MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include <stddef.h>
>+#include <stdint.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.cc
>index 349fdab9f15d0973e9f8c393edd56f43f88fb712..83c2b3b90c569eab3cc11d491971ceb06932d3f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.cc
>@@ -11,8 +11,8 @@
> #include "modules/audio_coding/neteq/decision_logic.h"
> 
> #include <assert.h>
>-#include <algorithm>
>-#include <limits>
>+#include <stdio.h>
>+#include <string>
> 
> #include "modules/audio_coding/neteq/buffer_level_filter.h"
> #include "modules/audio_coding/neteq/decoder_database.h"
>@@ -20,8 +20,9 @@
> #include "modules/audio_coding/neteq/expand.h"
> #include "modules/audio_coding/neteq/packet_buffer.h"
> #include "modules/audio_coding/neteq/sync_buffer.h"
>-#include "modules/include/module_common_types.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> #include "system_wrappers/include/field_trial.h"
> 
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.h
>index 39761da0702b419b7d55573f332e7697d13f4246..2a533594b3382f471ce088a86f223e185f8b6f04 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.h
>@@ -12,7 +12,6 @@
> #define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
> 
> #include "modules/audio_coding/neteq/defines.h"
>-#include "modules/audio_coding/neteq/include/neteq.h"
> #include "modules/audio_coding/neteq/tick_timer.h"
> #include "rtc_base/constructormagic.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
>index 08720d1768f456d00081b30087e0f105aee15758..183b9c79c9356331f07d3f6806e91b8957c907ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
>@@ -31,7 +31,7 @@ TEST(DecisionLogic, CreateAndDestroy) {
>   TickTimer tick_timer;
>   PacketBuffer packet_buffer(10, &tick_timer);
>   DelayPeakDetector delay_peak_detector(&tick_timer);
>-  DelayManager delay_manager(240, &delay_peak_detector, &tick_timer);
>+  DelayManager delay_manager(240, 0, &delay_peak_detector, &tick_timer);
>   BufferLevelFilter buffer_level_filter;
>   DecisionLogic* logic = DecisionLogic::Create(
>       fs_hz, output_size_samples, false, &decoder_database, packet_buffer,
>@@ -48,7 +48,7 @@ TEST(DecisionLogic, PostponeDecodingAfterExpansionSettings) {
>   TickTimer tick_timer;
>   PacketBuffer packet_buffer(10, &tick_timer);
>   DelayPeakDetector delay_peak_detector(&tick_timer);
>-  DelayManager delay_manager(240, &delay_peak_detector, &tick_timer);
>+  DelayManager delay_manager(240, 0, &delay_peak_detector, &tick_timer);
>   BufferLevelFilter buffer_level_filter;
>   {
>     test::ScopedFieldTrials field_trial(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.cc
>index 1fd8c03ca24a63ba55590aacac34a05c03f5c6ba..0890beb9342d682fdce6ccbdb98c04571c183749 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.cc
>@@ -10,9 +10,15 @@
> 
> #include "modules/audio_coding/neteq/decoder_database.h"
> 
>-#include <utility>  // pair
>+#include <stddef.h>
>+#include <cstdint>
>+#include <list>
>+#include <type_traits>
>+#include <utility>
> 
>+#include "absl/strings/match.h"
> #include "api/audio_codecs/audio_decoder.h"
>+#include "common_types.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/strings/audio_format_to_string.h"
>@@ -101,7 +107,7 @@ AudioDecoder* DecoderDatabase::DecoderInfo::GetDecoder() const {
> }
> 
> bool DecoderDatabase::DecoderInfo::IsType(const char* name) const {
>-  return STR_CASE_CMP(audio_format_.name.c_str(), name) == 0;
>+  return absl::EqualsIgnoreCase(audio_format_.name, name);
> }
> 
> bool DecoderDatabase::DecoderInfo::IsType(const std::string& name) const {
>@@ -110,7 +116,7 @@ bool DecoderDatabase::DecoderInfo::IsType(const std::string& name) const {
> 
> absl::optional<DecoderDatabase::DecoderInfo::CngDecoder>
> DecoderDatabase::DecoderInfo::CngDecoder::Create(const SdpAudioFormat& format) {
>-  if (STR_CASE_CMP(format.name.c_str(), "CN") == 0) {
>+  if (absl::EqualsIgnoreCase(format.name, "CN")) {
>     // CN has a 1:1 RTP clock rate to sample rate ratio.
>     const int sample_rate_hz = format.clockrate_hz;
>     RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
>@@ -123,11 +129,11 @@ DecoderDatabase::DecoderInfo::CngDecoder::Create(const SdpAudioFormat& format) {
> 
> DecoderDatabase::DecoderInfo::Subtype
> DecoderDatabase::DecoderInfo::SubtypeFromFormat(const SdpAudioFormat& format) {
>-  if (STR_CASE_CMP(format.name.c_str(), "CN") == 0) {
>+  if (absl::EqualsIgnoreCase(format.name, "CN")) {
>     return Subtype::kComfortNoise;
>-  } else if (STR_CASE_CMP(format.name.c_str(), "telephone-event") == 0) {
>+  } else if (absl::EqualsIgnoreCase(format.name, "telephone-event")) {
>     return Subtype::kDtmf;
>-  } else if (STR_CASE_CMP(format.name.c_str(), "red") == 0) {
>+  } else if (absl::EqualsIgnoreCase(format.name, "red")) {
>     return Subtype::kRed;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.cc
>index e5eb592efa1347565b62b0b19da75f71a3200dd3..67e6a13c1d769526d1718224a877c2e642dc93a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.cc
>@@ -11,14 +11,15 @@
> #include "modules/audio_coding/neteq/delay_manager.h"
> 
> #include <assert.h>
>-#include <math.h>
>-
>-#include <algorithm>  // max, min
>+#include <stdio.h>
>+#include <stdlib.h>
>+#include <algorithm>
> #include <numeric>
>+#include <string>
> 
>-#include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "modules/audio_coding/neteq/delay_peak_detector.h"
>-#include "modules/include/module_common_types.h"
>+#include "modules/include/module_common_types_public.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "system_wrappers/include/field_trial.h"
>@@ -61,6 +62,7 @@ absl::optional<int> GetForcedLimitProbability() {
> namespace webrtc {
> 
> DelayManager::DelayManager(size_t max_packets_in_buffer,
>+                           int base_min_target_delay_ms,
>                            DelayPeakDetector* peak_detector,
>                            const TickTimer* tick_timer)
>     : first_packet_received_(false),
>@@ -68,13 +70,14 @@ DelayManager::DelayManager(size_t max_packets_in_buffer,
>       iat_vector_(kMaxIat + 1, 0),
>       iat_factor_(0),
>       tick_timer_(tick_timer),
>+      base_min_target_delay_ms_(base_min_target_delay_ms),
>       base_target_level_(4),                   // In Q0 domain.
>       target_level_(base_target_level_ << 8),  // In Q8 domain.
>       packet_len_ms_(0),
>       streaming_mode_(false),
>       last_seq_no_(0),
>       last_timestamp_(0),
>-      minimum_delay_ms_(0),
>+      minimum_delay_ms_(base_min_target_delay_ms_),
>       maximum_delay_ms_(target_level_),
>       iat_cumulative_sum_(0),
>       max_iat_cumulative_sum_(0),
>@@ -84,6 +87,8 @@ DelayManager::DelayManager(size_t max_packets_in_buffer,
>           field_trial::IsEnabled("WebRTC-Audio-NetEqFramelengthExperiment")),
>       forced_limit_probability_(GetForcedLimitProbability()) {
>   assert(peak_detector);  // Should never be NULL.
>+  RTC_DCHECK_GE(base_min_target_delay_ms_, 0);
>+  RTC_DCHECK_LE(minimum_delay_ms_, maximum_delay_ms_);
> 
>   Reset();
> }
>@@ -484,7 +489,7 @@ bool DelayManager::SetMinimumDelay(int delay_ms) {
>            static_cast<int>(3 * max_packets_in_buffer_ * packet_len_ms_ / 4))) {
>     return false;
>   }
>-  minimum_delay_ms_ = delay_ms;
>+  minimum_delay_ms_ = std::max(delay_ms, base_min_target_delay_ms_);
>   return true;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.h
>index cd5fc09031691f259fde43e1c023f0c30acbf1e5..2c8081b075fe41357ea6c64dbe2091e05d481fd8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.h
>@@ -31,9 +31,11 @@ class DelayManager {
> 
>   // Create a DelayManager object. Notify the delay manager that the packet
>   // buffer can hold no more than |max_packets_in_buffer| packets (i.e., this
>-  // is the number of packet slots in the buffer). Supply a PeakDetector
>-  // object to the DelayManager.
>+  // is the number of packet slots in the buffer) and that the target delay
>+  // should be greater than or equal to |base_min_target_delay_ms|. Supply a
>+  // PeakDetector object to the DelayManager.
>   DelayManager(size_t max_packets_in_buffer,
>+               int base_min_target_delay_ms,
>                DelayPeakDetector* peak_detector,
>                const TickTimer* tick_timer);
> 
>@@ -144,6 +146,8 @@ class DelayManager {
>   IATVector iat_vector_;                // Histogram of inter-arrival times.
>   int iat_factor_;  // Forgetting factor for updating the IAT histogram (Q15).
>   const TickTimer* tick_timer_;
>+  const int base_min_target_delay_ms_;  // Lower bound for target_level_ and
>+                                        // minimum_delay_ms_.
>   // Time elapsed since last packet.
>   std::unique_ptr<TickTimer::Stopwatch> packet_iat_stopwatch_;
>   int base_target_level_;  // Currently preferred buffer level before peak
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
>index e4e865fe46be30028032c7d546c93c82aadc1a10..6281a156799893c025155ea55e675fbe7e9c6915 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
>@@ -27,6 +27,7 @@ using ::testing::_;
> class DelayManagerTest : public ::testing::Test {
>  protected:
>   static const int kMaxNumberOfPackets = 240;
>+  static const int kMinDelayMs = 0;
>   static const int kTimeStepMs = 10;
>   static const int kFs = 8000;
>   static const int kFrameSizeMs = 20;
>@@ -56,7 +57,8 @@ void DelayManagerTest::SetUp() {
> 
> void DelayManagerTest::RecreateDelayManager() {
>   EXPECT_CALL(detector_, Reset()).Times(1);
>-  dm_.reset(new DelayManager(kMaxNumberOfPackets, &detector_, &tick_timer_));
>+  dm_.reset(new DelayManager(kMaxNumberOfPackets, kMinDelayMs, &detector_,
>+                             &tick_timer_));
> }
> 
> void DelayManagerTest::SetPacketAudioLength(int lengt_ms) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
>index eb9f6d530d5d651aa56721de7fecb01c207edaa3..893ce3e8fd24448cea66c7ea43de6b706c72467d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
>@@ -10,10 +10,9 @@
> 
> #include "modules/audio_coding/neteq/delay_peak_detector.h"
> 
>-#include <algorithm>  // max
>+#include <algorithm>
> 
> #include "rtc_base/checks.h"
>-#include "rtc_base/numerics/safe_conversions.h"
> #include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector.h
>index 9defca5ad0216478bc7edc03d4df998d4d8fc415..272d50ed509fce23b56268bd5df17d6e7aec6b3c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector.h
>@@ -11,8 +11,8 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_DELAY_PEAK_DETECTOR_H_
> #define MODULES_AUDIO_CODING_NETEQ_DELAY_PEAK_DETECTOR_H_
> 
>-#include <string.h>  // size_t
>-
>+#include <stdint.h>
>+#include <string.h>
> #include <list>
> #include <memory>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.h
>index efa2f9ce06447a8de2f2e34879506f580a68abcb..83794612c5fdf61a4e5d34a17ba9ba9a3fb98596 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.h
>@@ -11,9 +11,11 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
> #define MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
> 
>-#include <string.h>  // Access to size_t.
>+#include <stdint.h>
>+#include <string.h>
> 
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
>+#include "modules/audio_coding/neteq/audio_vector.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
>index 6de512772f2ea50d8b3eb285dd7f45a6c80ea01a..24b14ece416a1b46b3dd6754ff83eadf5a7ec4f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
>@@ -11,8 +11,9 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
> #define MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <list>
>-#include <string>  // size_t
> 
> #include "rtc_base/constructormagic.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
>index 6fdb95aacad67bf133cec71471e696703c328ab9..6c412e364d87d4cf747968459672b34eb0ccf924 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
>@@ -30,6 +30,7 @@
> 
> #include "modules/audio_coding/neteq/dtmf_tone_generator.h"
> 
>+#include "modules/audio_coding/neteq/audio_vector.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
>index a773ff353d3b30df6948dc7cad748d30f9326a53..22e166ef0c43f9bb80127421733de30163363d00 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
>@@ -11,6 +11,9 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
> #define MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "rtc_base/constructormagic.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.cc
>index 5f671ada6b6203526e1f283bc096f9213c56a54c..4a06d09c1c61a7261437a9a77cd178f3c59e004f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.cc
>@@ -17,6 +17,7 @@
> #include <limits>     // numeric_limits<T>
> 
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "modules/audio_coding/neteq/background_noise.h"
> #include "modules/audio_coding/neteq/cross_correlation.h"
> #include "modules/audio_coding/neteq/dsp_helper.h"
>@@ -322,8 +323,7 @@ void Expand::SetParametersForNormalAfterExpand() {
>   current_lag_index_ = 0;
>   lag_index_direction_ = 0;
>   stop_muting_ = true;  // Do not mute signal any more.
>-  statistics_->LogDelayedPacketOutageEvent(
>-      rtc::dchecked_cast<int>(expand_duration_samples_) / (fs_hz_ / 1000));
>+  statistics_->LogDelayedPacketOutageEvent(expand_duration_samples_, fs_hz_);
> }
> 
> void Expand::SetParametersForMergeAfterExpand() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.h
>index 4cfe7b9d8cbeec1e7990bb4279a64016be149600..30c34a2c300df162429531a0ed32ed92001ae650 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.h
>@@ -14,12 +14,13 @@
> #include <assert.h>
> #include <memory>
> 
>-#include "modules/audio_coding/neteq/audio_multi_vector.h"
>+#include "modules/audio_coding/neteq/audio_vector.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
> // Forward declarations.
>+class AudioMultiVector;
> class BackgroundNoise;
> class RandomVector;
> class StatisticsCalculator;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.h
>index 00907d4e5a841b4bfac592d72c96e788dc696652..bd079c6ab7750b0f3c1c5fa653780318aed765de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.h
>@@ -10,6 +10,7 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
> #define MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
> 
>+#include <stdint.h>
> #include <memory>
> #include <string>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_unittest.cc
>index b4e6466e9045002bff808bdaceefa726b7d0ea56..09914daca82295c88dadee8190413186832b5a1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_unittest.cc
>@@ -51,14 +51,16 @@ TEST(Expand, CreateUsingFactory) {
> namespace {
> class FakeStatisticsCalculator : public StatisticsCalculator {
>  public:
>-  void LogDelayedPacketOutageEvent(int outage_duration_ms) override {
>-    last_outage_duration_ms_ = outage_duration_ms;
>+  void LogDelayedPacketOutageEvent(int num_samples, int fs_hz) override {
>+    last_outage_duration_samples_ = num_samples;
>   }
> 
>-  int last_outage_duration_ms() const { return last_outage_duration_ms_; }
>+  int last_outage_duration_samples() const {
>+    return last_outage_duration_samples_;
>+  }
> 
>  private:
>-  int last_outage_duration_ms_ = 0;
>+  int last_outage_duration_samples_ = 0;
> };
> 
> // This is the same size that is given to the SyncBuffer object in NetEq.
>@@ -120,13 +122,12 @@ TEST_F(ExpandTest, DelayedPacketOutage) {
>     EXPECT_EQ(0, expand_.Process(&output));
>     EXPECT_GT(output.Size(), 0u);
>     sum_output_len_samples += output.Size();
>-    EXPECT_EQ(0, statistics_.last_outage_duration_ms());
>+    EXPECT_EQ(0, statistics_.last_outage_duration_samples());
>   }
>   expand_.SetParametersForNormalAfterExpand();
>   // Convert |sum_output_len_samples| to milliseconds.
>-  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
>-                                   (test_sample_rate_hz_ / 1000)),
>-            statistics_.last_outage_duration_ms());
>+  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples),
>+            statistics_.last_outage_duration_samples());
> }
> 
> // This test is similar to DelayedPacketOutage, but ends by calling
>@@ -140,10 +141,10 @@ TEST_F(ExpandTest, LostPacketOutage) {
>     EXPECT_EQ(0, expand_.Process(&output));
>     EXPECT_GT(output.Size(), 0u);
>     sum_output_len_samples += output.Size();
>-    EXPECT_EQ(0, statistics_.last_outage_duration_ms());
>+    EXPECT_EQ(0, statistics_.last_outage_duration_samples());
>   }
>   expand_.SetParametersForMergeAfterExpand();
>-  EXPECT_EQ(0, statistics_.last_outage_duration_ms());
>+  EXPECT_EQ(0, statistics_.last_outage_duration_samples());
> }
> 
> // This test is similar to the DelayedPacketOutage test above, but with the
>@@ -161,13 +162,12 @@ TEST_F(ExpandTest, CheckOutageStatsAfterReset) {
>       expand_.Reset();
>       sum_output_len_samples = 0;
>     }
>-    EXPECT_EQ(0, statistics_.last_outage_duration_ms());
>+    EXPECT_EQ(0, statistics_.last_outage_duration_samples());
>   }
>   expand_.SetParametersForNormalAfterExpand();
>   // Convert |sum_output_len_samples| to milliseconds.
>-  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples /
>-                                   (test_sample_rate_hz_ / 1000)),
>-            statistics_.last_outage_duration_ms());
>+  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples),
>+            statistics_.last_outage_duration_samples());
> }
> 
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/include/neteq.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/include/neteq.h
>index 530975f3b8e7d7c53bce375401fb25fb5745317e..2820fd84787d5636f11b170c92a5b37058d19e59 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/include/neteq.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/include/neteq.h
>@@ -72,6 +72,7 @@ struct NetEqLifetimeStatistics {
>   uint64_t jitter_buffer_delay_ms = 0;
>   // Below stat is not part of the spec.
>   uint64_t voice_concealed_samples = 0;
>+  uint64_t delayed_packet_outage_samples = 0;
> };
> 
> // Metrics that describe the operations performed in NetEq, and the internal
>@@ -112,6 +113,7 @@ class NetEq {
>     bool enable_post_decode_vad = false;
>     size_t max_packets_in_buffer = 50;
>     int max_delay_ms = 2000;
>+    int min_delay_ms = 0;
>     bool enable_fast_accelerate = false;
>     bool enable_muted_state = false;
>     absl::optional<AudioCodecPairId> codec_pair_id;
>@@ -231,13 +233,6 @@ class NetEq {
>   // statistics are never reset.
>   virtual NetEqOperationsAndState GetOperationsAndState() const = 0;
> 
>-  // Writes the current RTCP statistics to |stats|. The statistics are reset
>-  // and a new report period is started with the call.
>-  virtual void GetRtcpStatistics(RtcpStatistics* stats) = 0;
>-
>-  // Same as RtcpStatistics(), but does not reset anything.
>-  virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats) = 0;
>-
>   // Enables post-decode VAD. When enabled, GetAudio() will return
>   // kOutputVADPassive when the signal contains no speech.
>   virtual void EnableVad() = 0;
>@@ -266,10 +261,6 @@ class NetEq {
>   // Flushes both the packet buffer and the sync buffer.
>   virtual void FlushBuffers() = 0;
> 
>-  // Current usage of packet-buffer and it's limits.
>-  virtual void PacketBufferStatistics(int* current_num_packets,
>-                                      int* max_num_packets) const = 0;
>-
>   // Enables NACK and sets the maximum size of the NACK list, which should be
>   // positive and no larger than Nack::kNackListSizeLimit. If NACK is already
>   // enabled then the maximum NACK list size is modified accordingly.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
>index 9b2ed498d9dbfcc241ae14f7a75209601cade097..206cea7f10e7f569a45c486390c9b8748a5c5b05 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
>@@ -20,9 +20,13 @@ namespace webrtc {
> class MockDelayManager : public DelayManager {
>  public:
>   MockDelayManager(size_t max_packets_in_buffer,
>+                   int base_min_target_delay_ms,
>                    DelayPeakDetector* peak_detector,
>                    const TickTimer* tick_timer)
>-      : DelayManager(max_packets_in_buffer, peak_detector, tick_timer) {}
>+      : DelayManager(max_packets_in_buffer,
>+                     base_min_target_delay_ms,
>+                     peak_detector,
>+                     tick_timer) {}
>   virtual ~MockDelayManager() { Die(); }
>   MOCK_METHOD0(Die, void());
>   MOCK_CONST_METHOD0(iat_vector, const IATVector&());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.cc
>index c62cdf88e4dc13d84fe3011bbe6435ba49c8f7c4..e3ecfeaed27073c66ca2ae2aa0d2e1bd846db1e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.cc
>@@ -10,9 +10,9 @@
> 
> #include "modules/audio_coding/neteq/nack_tracker.h"
> 
>-#include <assert.h>  // For assert.
>-
>-#include <algorithm>  // For std::max.
>+#include <assert.h>
>+#include <cstdint>
>+#include <utility>
> 
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.h
>index 1936a944ccb52be74e5982b8dde36cf7a31ac1e3..d7c6b084222cfc2b0c4fde77ee44cd5862276903 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.h
>@@ -11,11 +11,12 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
> #define MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <map>
> #include <vector>
> 
>-#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
>-#include "modules/include/module_common_types.h"
>+#include "modules/include/module_common_types_public.h"
> #include "rtc_base/gtest_prod_util.h"
> 
> //
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq.cc
>index cf1c6aa665364553f30406b76aa430bc6f219e20..0e6147e232cb1470185bdf9872a708d8555ef6f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq.cc
>@@ -10,8 +10,6 @@
> 
> #include "modules/audio_coding/neteq/include/neteq.h"
> 
>-#include <memory>
>-
> #include "modules/audio_coding/neteq/neteq_impl.h"
> #include "rtc_base/strings/string_builder.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.cc
>index f428be1c3bc8250dcd16cbfe086a6294276ef45f..2a025f304e537ce11800b7c9160032e19d224547 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.cc
>@@ -11,13 +11,16 @@
> #include "modules/audio_coding/neteq/neteq_impl.h"
> 
> #include <assert.h>
>-
> #include <algorithm>
>+#include <cstdint>
>+#include <cstring>
>+#include <list>
> #include <utility>
> #include <vector>
> 
> #include "api/audio_codecs/audio_decoder.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
> #include "modules/audio_coding/neteq/accelerate.h"
> #include "modules/audio_coding/neteq/background_noise.h"
> #include "modules/audio_coding/neteq/buffer_level_filter.h"
>@@ -40,15 +43,14 @@
> #include "modules/audio_coding/neteq/red_payload_splitter.h"
> #include "modules/audio_coding/neteq/sync_buffer.h"
> #include "modules/audio_coding/neteq/tick_timer.h"
>+#include "modules/audio_coding/neteq/time_stretch.h"
> #include "modules/audio_coding/neteq/timestamp_scaler.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/sanitizer.h"
> #include "rtc_base/strings/audio_format_to_string.h"
>-#include "rtc_base/system/fallthrough.h"
> #include "rtc_base/trace_event.h"
>-#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
>@@ -61,6 +63,7 @@ NetEqImpl::Dependencies::Dependencies(
>           new DecoderDatabase(decoder_factory, config.codec_pair_id)),
>       delay_peak_detector(new DelayPeakDetector(tick_timer.get())),
>       delay_manager(new DelayManager(config.max_packets_in_buffer,
>+                                     config.min_delay_ms,
>                                      delay_peak_detector.get(),
>                                      tick_timer.get())),
>       dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)),
>@@ -385,20 +388,6 @@ NetEqOperationsAndState NetEqImpl::GetOperationsAndState() const {
>   return result;
> }
> 
>-void NetEqImpl::GetRtcpStatistics(RtcpStatistics* stats) {
>-  rtc::CritScope lock(&crit_sect_);
>-  if (stats) {
>-    rtcp_.GetStatistics(false, stats);
>-  }
>-}
>-
>-void NetEqImpl::GetRtcpStatisticsNoReset(RtcpStatistics* stats) {
>-  rtc::CritScope lock(&crit_sect_);
>-  if (stats) {
>-    rtcp_.GetStatistics(true, stats);
>-  }
>-}
>-
> void NetEqImpl::EnableVad() {
>   rtc::CritScope lock(&crit_sect_);
>   assert(vad_.get());
>@@ -473,12 +462,6 @@ void NetEqImpl::FlushBuffers() {
>   first_packet_ = true;
> }
> 
>-void NetEqImpl::PacketBufferStatistics(int* current_num_packets,
>-                                       int* max_num_packets) const {
>-  rtc::CritScope lock(&crit_sect_);
>-  packet_buffer_->BufferStat(current_num_packets, max_num_packets);
>-}
>-
> void NetEqImpl::EnableNack(size_t max_nack_list_size) {
>   rtc::CritScope lock(&crit_sect_);
>   if (!nack_enabled_) {
>@@ -574,8 +557,6 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
>     // Note: |first_packet_| will be cleared further down in this method, once
>     // the packet has been successfully inserted into the packet buffer.
> 
>-    rtcp_.Init(rtp_header.sequenceNumber);
>-
>     // Flush the packet buffer and DTMF buffer.
>     packet_buffer_->Flush();
>     dtmf_buffer_->Flush();
>@@ -590,9 +571,6 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
>     timestamp_ = main_timestamp;
>   }
> 
>-  // Update RTCP statistics, only for regular packets.
>-  rtcp_.Update(rtp_header, receive_timestamp);
>-
>   if (nack_enabled_) {
>     RTC_DCHECK(nack_);
>     if (update_sample_rate_and_channels) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.h
>index 8ef97ce0af376f941ead4b1ec3eaf1d8896ec9f3..525ae615d856efd42ea168378b9224c0362e1761 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.h
>@@ -17,12 +17,11 @@
> #include "absl/types/optional.h"
> #include "api/audio/audio_frame.h"
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
>-#include "modules/audio_coding/neteq/defines.h"
>+#include "modules/audio_coding/neteq/defines.h"  // Modes, Operations
> #include "modules/audio_coding/neteq/expand_uma_logger.h"
> #include "modules/audio_coding/neteq/include/neteq.h"
>-#include "modules/audio_coding/neteq/packet.h"  // Declare PacketList.
>+#include "modules/audio_coding/neteq/packet.h"
> #include "modules/audio_coding/neteq/random_vector.h"
>-#include "modules/audio_coding/neteq/rtcp.h"
> #include "modules/audio_coding/neteq/statistics_calculator.h"
> #include "modules/audio_coding/neteq/tick_timer.h"
> #include "rtc_base/constructormagic.h"
>@@ -170,17 +169,10 @@ class NetEqImpl : public webrtc::NetEq {
>   // after the call.
>   int NetworkStatistics(NetEqNetworkStatistics* stats) override;
> 
>-  // Writes the current RTCP statistics to |stats|. The statistics are reset
>-  // and a new report period is started with the call.
>-  void GetRtcpStatistics(RtcpStatistics* stats) override;
>-
>   NetEqLifetimeStatistics GetLifetimeStatistics() const override;
> 
>   NetEqOperationsAndState GetOperationsAndState() const override;
> 
>-  // Same as RtcpStatistics(), but does not reset anything.
>-  void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
>-
>   // Enables post-decode VAD. When enabled, GetAudio() will return
>   // kOutputVADPassive when the signal contains no speech.
>   void EnableVad() override;
>@@ -200,9 +192,6 @@ class NetEqImpl : public webrtc::NetEq {
>   // Flushes both the packet buffer and the sync buffer.
>   void FlushBuffers() override;
> 
>-  void PacketBufferStatistics(int* current_num_packets,
>-                              int* max_num_packets) const override;
>-
>   void EnableNack(size_t max_nack_list_size) override;
> 
>   void DisableNack() override;
>@@ -395,7 +384,6 @@ class NetEqImpl : public webrtc::NetEq {
>       RTC_GUARDED_BY(crit_sect_);
>   RandomVector random_vector_ RTC_GUARDED_BY(crit_sect_);
>   std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(crit_sect_);
>-  Rtcp rtcp_ RTC_GUARDED_BY(crit_sect_);
>   StatisticsCalculator stats_ RTC_GUARDED_BY(crit_sect_);
>   int fs_hz_ RTC_GUARDED_BY(crit_sect_);
>   int fs_mult_ RTC_GUARDED_BY(crit_sect_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
>index b772dfa71de119fbcc0406eb4bed46160e47f5e8..0e087c847f18fdfc35fd28f322f5fab25952a030 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
>@@ -92,7 +92,8 @@ class NetEqImplTest : public ::testing::Test {
> 
>     if (use_mock_delay_manager_) {
>       std::unique_ptr<MockDelayManager> mock(new MockDelayManager(
>-          config_.max_packets_in_buffer, delay_peak_detector_, tick_timer_));
>+          config_.max_packets_in_buffer, config_.min_delay_ms,
>+          delay_peak_detector_, tick_timer_));
>       mock_delay_manager_ = mock.get();
>       EXPECT_CALL(*mock_delay_manager_, set_streaming_mode(false)).Times(1);
>       deps.delay_manager = std::move(mock);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
>index 3ee0b5bf693f539fffe746dd449f479da2f2f44f..e8b50237703148a33aefa834c13a03fc85f6efdf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
>@@ -22,12 +22,12 @@
> 
> #include "api/audio/audio_frame.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
> #include "modules/audio_coding/neteq/tools/audio_loop.h"
> #include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
> #include "modules/audio_coding/neteq/tools/neteq_test.h"
> #include "modules/audio_coding/neteq/tools/rtp_file_source.h"
>+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
> #include "rtc_base/ignore_wundef.h"
> #include "rtc_base/messagedigest.h"
> #include "rtc_base/numerics/safe_conversions.h"
>@@ -52,7 +52,7 @@ RTC_PUSH_IGNORING_WUNDEF()
> RTC_POP_IGNORING_WUNDEF()
> #endif
> 
>-DEFINE_bool(gen_ref, false, "Generate reference files.");
>+WEBRTC_DEFINE_bool(gen_ref, false, "Generate reference files.");
> 
> namespace webrtc {
> 
>@@ -234,7 +234,14 @@ void ResultSink::VerifyChecksum(const std::string& checksum) {
>   buffer.resize(digest_->Size());
>   digest_->Finish(&buffer[0], buffer.size());
>   const std::string result = rtc::hex_encode(&buffer[0], digest_->Size());
>-  EXPECT_EQ(checksum, result);
>+  if (checksum.size() == result.size()) {
>+    EXPECT_EQ(checksum, result);
>+  } else {
>+    // Check result is one the '|'-separated checksums.
>+    EXPECT_NE(checksum.find(result), std::string::npos)
>+        << result << " should be one of these:\n"
>+        << checksum;
>+  }
> }
> 
> class NetEqDecodingTest : public ::testing::Test {
>@@ -258,7 +265,6 @@ class NetEqDecodingTest : public ::testing::Test {
>   void DecodeAndCompare(const std::string& rtp_file,
>                         const std::string& output_checksum,
>                         const std::string& network_stats_checksum,
>-                        const std::string& rtcp_stats_checksum,
>                         bool gen_ref);
> 
>   static void PopulateRtpInfo(int frame_index,
>@@ -366,7 +372,6 @@ void NetEqDecodingTest::DecodeAndCompare(
>     const std::string& rtp_file,
>     const std::string& output_checksum,
>     const std::string& network_stats_checksum,
>-    const std::string& rtcp_stats_checksum,
>     bool gen_ref) {
>   OpenInputFile(rtp_file);
> 
>@@ -378,10 +383,6 @@ void NetEqDecodingTest::DecodeAndCompare(
>       gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : "";
>   ResultSink network_stats(stat_out_file);
> 
>-  std::string rtcp_out_file =
>-      gen_ref ? webrtc::test::OutputPath() + "neteq_rtcp_stats.dat" : "";
>-  ResultSink rtcp_stats(rtcp_out_file);
>-
>   packet_ = rtp_source_->NextPacket();
>   int i = 0;
>   uint64_t last_concealed_samples = 0;
>@@ -418,11 +419,6 @@ void NetEqDecodingTest::DecodeAndCompare(
>       EXPECT_NEAR(
>           (delta_concealed_samples << 14) / delta_total_samples_received,
>           current_network_stats.expand_rate, (2 << 14) / 100.0);
>-
>-      // Process RTCPstat.
>-      RtcpStatistics current_rtcp_stats;
>-      neteq_->GetRtcpStatistics(&current_rtcp_stats);
>-      ASSERT_NO_FATAL_FAILURE(rtcp_stats.AddResult(current_rtcp_stats));
>     }
>   }
> 
>@@ -430,8 +426,6 @@ void NetEqDecodingTest::DecodeAndCompare(
>   output.VerifyChecksum(output_checksum);
>   SCOPED_TRACE("Check network stats.");
>   network_stats.VerifyChecksum(network_stats_checksum);
>-  SCOPED_TRACE("Check rtcp stats.");
>-  rtcp_stats.VerifyChecksum(rtcp_stats_checksum);
> }
> 
> void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
>@@ -481,14 +475,8 @@ TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
>                        "4b2370f5c794741d2a46be5c7935c66ef3fb53e9",
>                        "4b2370f5c794741d2a46be5c7935c66ef3fb53e9");
> 
>-  const std::string rtcp_stats_checksum =
>-      PlatformChecksum("b8880bf9fed2487efbddcb8d94b9937a29ae521d",
>-                       "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4", "not used",
>-                       "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
>-                       "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
>-
>   DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
>-                   rtcp_stats_checksum, FLAG_gen_ref);
>+                   FLAG_gen_ref);
> }
> 
> #if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
>@@ -501,12 +489,13 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
>   const std::string input_rtp_file =
>       webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
> 
>-  const std::string output_checksum =
>-      PlatformChecksum("14a63b3c7b925c82296be4bafc71bec85f2915c2",
>-                       "b7b7ed802b0e18ee416973bf3b9ae98599b0181d",
>-                       "5876e52dda90d5ca433c3726555b907b97c86374",
>-                       "14a63b3c7b925c82296be4bafc71bec85f2915c2",
>-                       "14a63b3c7b925c82296be4bafc71bec85f2915c2");
>+  // Checksum depends on libopus being compiled with or without SSE.
>+  const std::string maybe_sse =
>+      "14a63b3c7b925c82296be4bafc71bec85f2915c2|"
>+      "2c05677daa968d6c68b92adf4affb7cd9bb4d363";
>+  const std::string output_checksum = PlatformChecksum(
>+      maybe_sse, "b7b7ed802b0e18ee416973bf3b9ae98599b0181d",
>+      "5876e52dda90d5ca433c3726555b907b97c86374", maybe_sse, maybe_sse);
> 
>   const std::string network_stats_checksum =
>       PlatformChecksum("adb3272498e436d1c019cbfd71610e9510c54497",
>@@ -515,15 +504,8 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
>                        "adb3272498e436d1c019cbfd71610e9510c54497",
>                        "adb3272498e436d1c019cbfd71610e9510c54497");
> 
>-  const std::string rtcp_stats_checksum =
>-      PlatformChecksum("e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>-                       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>-                       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>-                       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>-                       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
>-
>   DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
>-                   rtcp_stats_checksum, FLAG_gen_ref);
>+                   FLAG_gen_ref);
> }
> 
> #if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
>@@ -536,21 +518,18 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
>   const std::string input_rtp_file =
>       webrtc::test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp");
> 
>-  const std::string output_checksum =
>-      PlatformChecksum("713af6c92881f5aab1285765ee6680da9d1c06ce",
>-                       "3ec991b96872123f1554c03c543ca5d518431e46",
>-                       "da9f9a2d94e0c2d67342fad4965d7b91cda50b25",
>-                       "713af6c92881f5aab1285765ee6680da9d1c06ce",
>-                       "713af6c92881f5aab1285765ee6680da9d1c06ce");
>+  const std::string maybe_sse =
>+      "713af6c92881f5aab1285765ee6680da9d1c06ce|"
>+      "2ac10c4e79aeedd0df2863b079da5848b40f00b5";
>+  const std::string output_checksum = PlatformChecksum(
>+      maybe_sse, "3ec991b96872123f1554c03c543ca5d518431e46",
>+      "da9f9a2d94e0c2d67342fad4965d7b91cda50b25", maybe_sse, maybe_sse);
> 
>   const std::string network_stats_checksum =
>       "bab58dc587d956f326056d7340c96eb9d2d3cc21";
> 
>-  const std::string rtcp_stats_checksum =
>-      "ac27a7f305efb58b39bf123dccee25dee5758e63";
>-
>   DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
>-                   rtcp_stats_checksum, FLAG_gen_ref);
>+                   FLAG_gen_ref);
> }
> 
> // Use fax mode to avoid time-scaling. This is to simplify the testing of
>@@ -1740,7 +1719,7 @@ TEST(NetEqNoTimeStretchingMode, RunTest) {
>       {8, kRtpExtensionVideoTiming}};
>   std::unique_ptr<NetEqInput> input(new NetEqRtpDumpInput(
>       webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"),
>-      rtp_ext_map));
>+      rtp_ext_map, absl::nullopt /*No SSRC filter*/));
>   std::unique_ptr<TimeLimitedNetEqInput> input_time_limit(
>       new TimeLimitedNetEqInput(std::move(input), 20000));
>   std::unique_ptr<AudioSink> output(new VoidAudioSink);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.h
>index b13f16e50dc818d1c0861ecc1fdc239c9877a92e..80451b5007813fa71396b871be55cdd813528570 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.h
>@@ -11,11 +11,9 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
> #define MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
> 
>+#include <stdint.h>
> #include <string.h>  // Access to size_t.
> 
>-#include <vector>
>-
>-#include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "modules/audio_coding/neteq/defines.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
>@@ -24,6 +22,7 @@
> namespace webrtc {
> 
> // Forward declarations.
>+class AudioMultiVector;
> class BackgroundNoise;
> class DecoderDatabase;
> class Expand;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet.h
>index 45c56e1b686b353bdfe947e57c710eb8c00e7cd9..358d8fa1abb8f8b9e097fe175545ba869ef1fdfc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet.h
>@@ -11,12 +11,14 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_H_
> #define MODULES_AUDIO_CODING_NETEQ_PACKET_H_
> 
>+#include <stdint.h>
> #include <list>
> #include <memory>
> 
> #include "api/audio_codecs/audio_decoder.h"
> #include "modules/audio_coding/neteq/tick_timer.h"
> #include "rtc_base/buffer.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.cc
>index eba4d3e68b66d6e571e9c06e45573817af7da978..343763b4e4d813fa6105c3ab99e6fea7060662fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.cc
>@@ -14,12 +14,17 @@
> 
> #include "modules/audio_coding/neteq/packet_buffer.h"
> 
>-#include <algorithm>  // find_if()
>+#include <algorithm>
>+#include <list>
>+#include <memory>
>+#include <type_traits>
>+#include <utility>
> 
> #include "api/audio_codecs/audio_decoder.h"
> #include "modules/audio_coding/neteq/decoder_database.h"
> #include "modules/audio_coding/neteq/statistics_calculator.h"
> #include "modules/audio_coding/neteq/tick_timer.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
> namespace webrtc {
>@@ -294,9 +299,4 @@ bool PacketBuffer::ContainsDtxOrCngPacket(
>   return false;
> }
> 
>-void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const {
>-  *num_packets = static_cast<int>(buffer_.size());
>-  *max_num_packets = static_cast<int>(max_number_of_packets_);
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.h
>index 7e34c1e9fce0b8003d43786625640c3c22f505ce..0f5cd7f8ed5dacd05c7051706b6d5f005fab4de0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.h
>@@ -14,7 +14,7 @@
> #include "absl/types/optional.h"
> #include "modules/audio_coding/neteq/decoder_database.h"
> #include "modules/audio_coding/neteq/packet.h"
>-#include "modules/include/module_common_types.h"
>+#include "modules/include/module_common_types_public.h"  // IsNewerTimestamp
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -125,8 +125,6 @@ class PacketBuffer {
>   virtual bool ContainsDtxOrCngPacket(
>       const DecoderDatabase* decoder_database) const;
> 
>-  virtual void BufferStat(int* num_packets, int* max_num_packets) const;
>-
>   // Static method returning true if |timestamp| is older than |timestamp_limit|
>   // but less than |horizon_samples| behind |timestamp_limit|. For instance,
>   // with timestamp_limit = 100 and horizon_samples = 10, a timestamp in the
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.h
>index ad4f0823bb7a5d72c8111e0758141f34f76c996d..27d69a6eeb2280afb834e683b9f2da1a4b8d3112 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.h
>@@ -11,13 +11,11 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
> #define MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
> 
>-#include <string>  // size_t
>+#include <stddef.h>
>+#include <stdint.h>
> 
> #include "api/audio_codecs/audio_decoder.h"
> #include "common_audio/vad/include/webrtc_vad.h"
>-#include "common_types.h"  // NOLINT(build/include)  // NULL
>-#include "modules/audio_coding/neteq/defines.h"
>-#include "modules/audio_coding/neteq/packet.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
>index 6159a9cb15d70e67086dff1f7ef0c025c751f393..cad8d6a50f6ccd2a50da2594bba79b0741df722f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
>@@ -10,9 +10,11 @@
> 
> #include "modules/audio_coding/neteq/preemptive_expand.h"
> 
>-#include <algorithm>  // min, max
>+#include <algorithm>
> 
>-#include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "api/array_view.h"
>+#include "modules/audio_coding/neteq/audio_multi_vector.h"
>+#include "modules/audio_coding/neteq/time_stretch.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.h
>index ace648f15cce257109976bbec88dc87b0a125f53..0f7b3bc8acf18bd3ce9e6fceddeffc8aeef5a3f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.h
>@@ -11,13 +11,15 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
> #define MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
> 
>-#include "modules/audio_coding/neteq/audio_multi_vector.h"
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "modules/audio_coding/neteq/time_stretch.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
>-// Forward declarations.
>+class AudioMultiVector;
> class BackgroundNoise;
> 
> // This class implements the PreemptiveExpand operation. Most of the work is
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.cc
>index f5435e8eab4ad1efb96d3a039f9676dcb829fc70..2dfe8386ff9a94eea97ee27fa7436574ceb7f021 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.cc
>@@ -11,10 +11,15 @@
> #include "modules/audio_coding/neteq/red_payload_splitter.h"
> 
> #include <assert.h>
>+#include <stddef.h>
>+#include <cstdint>
>+#include <list>
>+#include <utility>
> #include <vector>
> 
> #include "modules/audio_coding/neteq/decoder_database.h"
>-#include "rtc_base/checks.h"
>+#include "modules/audio_coding/neteq/packet.h"
>+#include "rtc_base/buffer.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.h
>index 5e239ddc708260e66ff4ea361c5cc26117564b46..55063e741769e0c031b92ba0f0753a90fed0f76f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.h
>@@ -16,7 +16,6 @@
> 
> namespace webrtc {
> 
>-// Forward declarations.
> class DecoderDatabase;
> 
> // This class handles splitting of RED payloads into smaller parts.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.cc
>deleted file mode 100644
>index 551eb5f75b401ffb2f95828b282d8fdd5afca9c2..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.cc
>+++ /dev/null
>@@ -1,94 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_coding/neteq/rtcp.h"
>-
>-#include <stdlib.h>
>-#include <string.h>
>-
>-#include <algorithm>
>-
>-namespace webrtc {
>-
>-void Rtcp::Init(uint16_t start_sequence_number) {
>-  cycles_ = 0;
>-  max_seq_no_ = start_sequence_number;
>-  base_seq_no_ = start_sequence_number;
>-  received_packets_ = 0;
>-  received_packets_prior_ = 0;
>-  expected_prior_ = 0;
>-  jitter_ = 0;
>-  transit_ = 0;
>-}
>-
>-void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) {
>-  // Update number of received packets, and largest packet number received.
>-  received_packets_++;
>-  int16_t sn_diff = rtp_header.sequenceNumber - max_seq_no_;
>-  if (sn_diff >= 0) {
>-    if (rtp_header.sequenceNumber < max_seq_no_) {
>-      // Wrap-around detected.
>-      cycles_++;
>-    }
>-    max_seq_no_ = rtp_header.sequenceNumber;
>-  }
>-
>-  // Calculate jitter according to RFC 3550, and update previous timestamps.
>-  // Note that the value in |jitter_| is in Q4.
>-  if (received_packets_ > 1) {
>-    int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_);
>-    int64_t jitter_diff = (std::abs(int64_t{ts_diff}) << 4) - jitter_;
>-    // Calculate 15 * jitter_ / 16 + jitter_diff / 16 (with proper rounding).
>-    jitter_ = jitter_ + ((jitter_diff + 8) >> 4);
>-    RTC_DCHECK_GE(jitter_, 0);
>-  }
>-  transit_ = rtp_header.timestamp - receive_timestamp;
>-}
>-
>-void Rtcp::GetStatistics(bool no_reset, RtcpStatistics* stats) {
>-  // Extended highest sequence number received.
>-  stats->extended_highest_sequence_number =
>-      (static_cast<int>(cycles_) << 16) + max_seq_no_;
>-
>-  // Calculate expected number of packets and compare it with the number of
>-  // packets that were actually received. The cumulative number of lost packets
>-  // can be extracted.
>-  uint32_t expected_packets =
>-      stats->extended_highest_sequence_number - base_seq_no_ + 1;
>-  if (received_packets_ == 0) {
>-    // No packets received, assume none lost.
>-    stats->packets_lost = 0;
>-  } else if (expected_packets > received_packets_) {
>-    stats->packets_lost = expected_packets - received_packets_;
>-    if (stats->packets_lost > 0xFFFFFF) {
>-      stats->packets_lost = 0xFFFFFF;
>-    }
>-  } else {
>-    stats->packets_lost = 0;
>-  }
>-
>-  // Fraction lost since last report.
>-  uint32_t expected_since_last = expected_packets - expected_prior_;
>-  uint32_t received_since_last = received_packets_ - received_packets_prior_;
>-  if (!no_reset) {
>-    expected_prior_ = expected_packets;
>-    received_packets_prior_ = received_packets_;
>-  }
>-  int32_t lost = expected_since_last - received_since_last;
>-  if (expected_since_last == 0 || lost <= 0 || received_packets_ == 0) {
>-    stats->fraction_lost = 0;
>-  } else {
>-    stats->fraction_lost = std::min(0xFFU, (lost << 8) / expected_since_last);
>-  }
>-
>-  stats->jitter = jitter_ >> 4;  // Scaling from Q4.
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.h
>deleted file mode 100644
>index b1de7eb7f487f27eb8bc696ce810698398498f09..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.h
>+++ /dev/null
>@@ -1,55 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_CODING_NETEQ_RTCP_H_
>-#define MODULES_AUDIO_CODING_NETEQ_RTCP_H_
>-
>-#include "modules/audio_coding/neteq/include/neteq.h"
>-#include "rtc_base/constructormagic.h"
>-
>-namespace webrtc {
>-
>-// Forward declaration.
>-struct RTPHeader;
>-
>-class Rtcp {
>- public:
>-  Rtcp() { Init(0); }
>-
>-  ~Rtcp() {}
>-
>-  // Resets the RTCP statistics, and sets the first received sequence number.
>-  void Init(uint16_t start_sequence_number);
>-
>-  // Updates the RTCP statistics with a new received packet.
>-  void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp);
>-
>-  // Returns the current RTCP statistics. If |no_reset| is true, the statistics
>-  // are not reset, otherwise they are.
>-  void GetStatistics(bool no_reset, RtcpStatistics* stats);
>-
>- private:
>-  uint16_t cycles_;       // The number of wrap-arounds for the sequence number.
>-  uint16_t max_seq_no_;   // The maximum sequence number received. Starts over
>-                          // from 0 after wrap-around.
>-  uint16_t base_seq_no_;  // The sequence number of the first received packet.
>-  uint32_t received_packets_;  // The number of packets that have been received.
>-  uint32_t received_packets_prior_;  // Number of packets received when last
>-                                     // report was generated.
>-  uint32_t expected_prior_;  // Expected number of packets, at the time of the
>-                             // last report.
>-  int64_t jitter_;           // Current jitter value in Q4.
>-  int32_t transit_;          // Clock difference for previous packet.
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(Rtcp);
>-};
>-
>-}  // namespace webrtc
>-#endif  // MODULES_AUDIO_CODING_NETEQ_RTCP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
>index c0e7a402aac00aa72a5f79dd34868f263c9f851d..50521fb03d364ea080d457047ac70f1a3879a666 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
>@@ -126,7 +126,10 @@ StatisticsCalculator::StatisticsCalculator()
>           100),
>       excess_buffer_delay_("WebRTC.Audio.AverageExcessBufferDelayMs",
>                            60000,  // 60 seconds report interval.
>-                           1000) {}
>+                           1000),
>+      buffer_full_counter_("WebRTC.Audio.JitterBufferFullPerMinute",
>+                           60000,  // 60 seconds report interval.
>+                           100) {}
> 
> StatisticsCalculator::~StatisticsCalculator() = default;
> 
>@@ -229,6 +232,7 @@ void StatisticsCalculator::IncreaseCounter(size_t num_samples, int fs_hz) {
>       rtc::CheckedDivExact(static_cast<int>(1000 * num_samples), fs_hz);
>   delayed_packet_outage_counter_.AdvanceClock(time_step_ms);
>   excess_buffer_delay_.AdvanceClock(time_step_ms);
>+  buffer_full_counter_.AdvanceClock(time_step_ms);
>   timestamps_since_last_report_ += static_cast<uint32_t>(num_samples);
>   if (timestamps_since_last_report_ >
>       static_cast<uint32_t>(fs_hz * kMaxReportPeriod)) {
>@@ -250,13 +254,17 @@ void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {
> 
> void StatisticsCalculator::FlushedPacketBuffer() {
>   operations_and_state_.packet_buffer_flushes++;
>+  buffer_full_counter_.RegisterSample();
> }
> 
>-void StatisticsCalculator::LogDelayedPacketOutageEvent(int outage_duration_ms) {
>+void StatisticsCalculator::LogDelayedPacketOutageEvent(int num_samples,
>+                                                       int fs_hz) {
>+  int outage_duration_ms = num_samples / (fs_hz / 1000);
>   RTC_HISTOGRAM_COUNTS("WebRTC.Audio.DelayedPacketOutageEventMs",
>                        outage_duration_ms, 1 /* min */, 2000 /* max */,
>                        100 /* bucket count */);
>   delayed_packet_outage_counter_.RegisterSample();
>+  lifetime_stats_.delayed_packet_outage_samples += num_samples;
> }
> 
> void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.h
>index 9e5dacfd920a63998026c3bcb96bed8f33eb04a6..49b74a04b3b4a6d77ce05e5704fc86c2bc4968ac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.h
>@@ -86,10 +86,10 @@ class StatisticsCalculator {
>   // Rerport that the packet buffer was flushed.
>   void FlushedPacketBuffer();
> 
>-  // Logs a delayed packet outage event of |outage_duration_ms|. A delayed
>-  // packet outage event is defined as an expand period caused not by an actual
>-  // packet loss, but by a delayed packet.
>-  virtual void LogDelayedPacketOutageEvent(int outage_duration_ms);
>+  // Logs a delayed packet outage event of |num_samples| expanded at a sample
>+  // rate of |fs_hz|. A delayed packet outage event is defined as an expand
>+  // period caused not by an actual packet loss, but by a delayed packet.
>+  virtual void LogDelayedPacketOutageEvent(int num_samples, int fs_hz);
> 
>   // Returns the current network statistics in |stats|. The current sample rate
>   // is |fs_hz|, the total number of samples in packet buffer and sync buffer
>@@ -199,6 +199,7 @@ class StatisticsCalculator {
>   size_t discarded_secondary_packets_;
>   PeriodicUmaCount delayed_packet_outage_counter_;
>   PeriodicUmaAverage excess_buffer_delay_;
>+  PeriodicUmaCount buffer_full_counter_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(StatisticsCalculator);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.h
>index 72e320c61f6f9017c8b2d540980249e80eeac092..d645e9163e9105c98f1b900b09caef416fffca73 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.h
>@@ -11,8 +11,13 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
> #define MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+#include <vector>
>+
> #include "api/audio/audio_frame.h"
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
>+#include "modules/audio_coding/neteq/audio_vector.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
>index ad61235cf28966daab491e6ddb7761af82850c4c..6f103452980de52f097452fb496ac32ff95c4f9b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
>@@ -25,7 +25,7 @@ namespace {
> static const int kInputSampleRateKhz = 8;
> static const int kOutputSampleRateKhz = 8;
> 
>-DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
>+WEBRTC_DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
> 
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
>index 94984b87e6f1239201924afa75dbf1016d08e325..651b0ca71a68ed83410a14de9d739b880553566d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
>@@ -21,7 +21,7 @@ static const int kIsacBlockDurationMs = 30;
> static const int kIsacInputSamplingKhz = 16;
> static const int kIsacOutputSamplingKhz = 16;
> 
>-DEFINE_int(bit_rate_kbps, 32, "Target bit rate (kbps).");
>+WEBRTC_DEFINE_int(bit_rate_kbps, 32, "Target bit rate (kbps).");
> 
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
>index 6861e4c55aafd6c44647fdbc04b2ad925086fc62..f4a36363baa95bea7eb03ca6d2a53d90eaa66d60 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
>@@ -22,24 +22,26 @@ namespace {
> static const int kOpusBlockDurationMs = 20;
> static const int kOpusSamplingKhz = 48;
> 
>-DEFINE_int(bit_rate_kbps, 32, "Target bit rate (kbps).");
>+WEBRTC_DEFINE_int(bit_rate_kbps, 32, "Target bit rate (kbps).");
> 
>-DEFINE_int(complexity,
>-           10,
>-           "Complexity: 0 ~ 10 -- defined as in Opus"
>-           "specification.");
>+WEBRTC_DEFINE_int(complexity,
>+                  10,
>+                  "Complexity: 0 ~ 10 -- defined as in Opus"
>+                  "specification.");
> 
>-DEFINE_int(maxplaybackrate, 48000, "Maximum playback rate (Hz).");
>+WEBRTC_DEFINE_int(maxplaybackrate, 48000, "Maximum playback rate (Hz).");
> 
>-DEFINE_int(application, 0, "Application mode: 0 -- VOIP, 1 -- Audio.");
>+WEBRTC_DEFINE_int(application, 0, "Application mode: 0 -- VOIP, 1 -- Audio.");
> 
>-DEFINE_int(reported_loss_rate, 10, "Reported percentile of packet loss.");
>+WEBRTC_DEFINE_int(reported_loss_rate,
>+                  10,
>+                  "Reported percentile of packet loss.");
> 
>-DEFINE_bool(fec, false, "Enable FEC for encoding (-nofec to disable).");
>+WEBRTC_DEFINE_bool(fec, false, "Enable FEC for encoding (-nofec to disable).");
> 
>-DEFINE_bool(dtx, false, "Enable DTX for encoding (-nodtx to disable).");
>+WEBRTC_DEFINE_bool(dtx, false, "Enable DTX for encoding (-nodtx to disable).");
> 
>-DEFINE_int(sub_packets, 1, "Number of sub packets to repacketize.");
>+WEBRTC_DEFINE_int(sub_packets, 1, "Number of sub packets to repacketize.");
> 
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
>index 8872b94a0639315c2e9fd966b1d5743941b6df8d..9c53919d54077ef3064da88cb942fef1d499f99f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
>@@ -26,7 +26,7 @@ namespace {
> static const int kInputSampleRateKhz = 48;
> static const int kOutputSampleRateKhz = 48;
> 
>-DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
>+WEBRTC_DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
> 
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
>index 54ff849739a3902f3c974ab62e0f58361a90f92d..85f22671eac47ada99fe389253b162f1d458d33c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
>@@ -25,7 +25,7 @@ namespace {
> static const int kInputSampleRateKhz = 8;
> static const int kOutputSampleRateKhz = 8;
> 
>-DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
>+WEBRTC_DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
> 
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
>index c486ac60bd7c89ce417229c520cc9f4a603ab9f6..70777a2d0214921483addcf20ad59d3c91baf3dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
>@@ -14,13 +14,12 @@
> 
> #include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
> #include "rtc_base/flags.h"
>-#include "test/testsupport/fileutils.h"
> 
> // Define command line flags.
>-DEFINE_int(runtime_ms, 10000, "Simulated runtime in ms.");
>-DEFINE_int(lossrate, 10, "Packet lossrate; drop every N packets.");
>-DEFINE_float(drift, 0.1f, "Clockdrift factor.");
>-DEFINE_bool(help, false, "Print this message.");
>+WEBRTC_DEFINE_int(runtime_ms, 10000, "Simulated runtime in ms.");
>+WEBRTC_DEFINE_int(lossrate, 10, "Packet lossrate; drop every N packets.");
>+WEBRTC_DEFINE_float(drift, 0.1f, "Clockdrift factor.");
>+WEBRTC_DEFINE_bool(help, false, "Print this message.");
> 
> int main(int argc, char* argv[]) {
>   std::string program_name = argv[0];
>@@ -33,7 +32,6 @@ int main(int argc, char* argv[]) {
>       "  --lossrate=N           drop every N packets; default is 10\n"
>       "  --drift=F              clockdrift factor between 0.0 and 1.0; "
>       "default is 0.1\n";
>-  webrtc::test::SetExecutablePath(argv[0]);
>   if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
>       argc != 1) {
>     printf("%s", usage.c_str());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tick_timer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tick_timer.h
>index 520099e346624b5575293585dc9d7ca7d3cc4d60..02f083e02eb979cc77affaa458166112622fa542 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tick_timer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tick_timer.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
> #define MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
>index 07d945edce0712d7f51e4ece41c9f9bad8e2bcba..b0461bb92d84ef34082b1c4b16d149dddff9f813 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/audio_coding/neteq/timestamp_scaler.h"
> 
>+#include "api/audio_codecs/audio_format.h"
> #include "modules/audio_coding/neteq/decoder_database.h"
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
>index 21c5f9ef0223f7b10e8a775b0fe007ba2f53ea96..9107e5e5ef4fc024c0e80521f4191d652fde3298 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
>@@ -19,11 +19,8 @@ namespace webrtc {
> namespace test {
> 
> NetEqEventLogInput::NetEqEventLogInput(const std::string& file_name,
>-                                       const RtpHeaderExtensionMap& hdr_ext_map)
>-    : source_(RtcEventLogSource::Create(file_name)) {
>-  for (const auto& ext_pair : hdr_ext_map) {
>-    source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
>-  }
>+                                       absl::optional<uint32_t> ssrc_filter)
>+    : source_(RtcEventLogSource::Create(file_name, ssrc_filter)) {
>   LoadNextPacket();
>   AdvanceOutputEvent();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h
>index 86cf9f2f02a92b6ef38c447bfd89fcff854cbd0b..e04df1684f1b871d3ccccee77e8fb4950e3b2348 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h
>@@ -28,7 +28,7 @@ class RtcEventLogSource;
> class NetEqEventLogInput final : public NetEqPacketSourceInput {
>  public:
>   NetEqEventLogInput(const std::string& file_name,
>-                     const RtpHeaderExtensionMap& hdr_ext_map);
>+                     absl::optional<uint32_t> ssrc_filter);
> 
>   absl::optional<int64_t> NextOutputEventTime() const override;
>   void AdvanceOutputEvent() override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
>index a86cf6afa5901784f71640630adde6a98a924f6f..ee02288c86d66e5f3dd06b1495cd8e62456662fb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
>@@ -58,15 +58,10 @@ std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::PopPacket() {
>   return packet_data;
> }
> 
>-void NetEqPacketSourceInput::SelectSsrc(uint32_t ssrc) {
>-  source()->SelectSsrc(ssrc);
>-  if (packet_ && packet_->header().ssrc != ssrc)
>-    LoadNextPacket();
>-}
>-
> NetEqRtpDumpInput::NetEqRtpDumpInput(const std::string& file_name,
>-                                     const RtpHeaderExtensionMap& hdr_ext_map)
>-    : source_(RtpFileSource::Create(file_name)) {
>+                                     const RtpHeaderExtensionMap& hdr_ext_map,
>+                                     absl::optional<uint32_t> ssrc_filter)
>+    : source_(RtpFileSource::Create(file_name, ssrc_filter)) {
>   for (const auto& ext_pair : hdr_ext_map) {
>     source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
>index b32c5340851f66255a55d66d86ee8f80e0956e84..8633d1f5dd1cf37ec671b8724e2cac28944e011b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
>@@ -12,8 +12,10 @@
> #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
> 
> #include <map>
>+#include <memory>
> #include <string>
> 
>+#include "absl/types/optional.h"
> #include "modules/audio_coding/neteq/tools/neteq_input.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> 
>@@ -32,7 +34,6 @@ class NetEqPacketSourceInput : public NetEqInput {
>   std::unique_ptr<PacketData> PopPacket() override;
>   absl::optional<RTPHeader> NextHeader() const override;
>   bool ended() const override { return !next_output_event_ms_; }
>-  void SelectSsrc(uint32_t);
> 
>  protected:
>   virtual PacketSource* source() = 0;
>@@ -48,7 +49,8 @@ class NetEqPacketSourceInput : public NetEqInput {
> class NetEqRtpDumpInput final : public NetEqPacketSourceInput {
>  public:
>   NetEqRtpDumpInput(const std::string& file_name,
>-                    const RtpHeaderExtensionMap& hdr_ext_map);
>+                    const RtpHeaderExtensionMap& hdr_ext_map,
>+                    absl::optional<uint32_t> ssrc_filter);
> 
>   absl::optional<int64_t> NextOutputEventTime() const override;
>   void AdvanceOutputEvent() override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
>index faca8959d4d4f5c7cf6a13e98ca32141b2b1f76e..2ee6779c539e1b9e67bb7c5fb071e251f5d09bd8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
>@@ -47,42 +47,47 @@ static bool ValidateFilename(const std::string& value, bool write) {
>   return true;
> }
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     in_filename,
>     DefaultInFilename().c_str(),
>     "Filename for input audio (specify sample rate with --input_sample_rate, "
>     "and channels with --channels).");
> 
>-DEFINE_int(input_sample_rate, 16000, "Sample rate of input file in Hz.");
>+WEBRTC_DEFINE_int(input_sample_rate, 16000, "Sample rate of input file in Hz.");
> 
>-DEFINE_int(channels, 1, "Number of channels in input audio.");
>+WEBRTC_DEFINE_int(channels, 1, "Number of channels in input audio.");
> 
>-DEFINE_string(out_filename,
>-              DefaultOutFilename().c_str(),
>-              "Name of output audio file.");
>+WEBRTC_DEFINE_string(out_filename,
>+                     DefaultOutFilename().c_str(),
>+                     "Name of output audio file.");
> 
>-DEFINE_int(runtime_ms, 10000, "Simulated runtime (milliseconds).");
>+WEBRTC_DEFINE_int(runtime_ms, 10000, "Simulated runtime (milliseconds).");
> 
>-DEFINE_int(packet_loss_rate, 10, "Percentile of packet loss.");
>+WEBRTC_DEFINE_int(packet_loss_rate, 10, "Percentile of packet loss.");
> 
>-DEFINE_int(random_loss_mode,
>-           kUniformLoss,
>-           "Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot "
>-           "loss, 3--fixed loss.");
>+WEBRTC_DEFINE_int(
>+    random_loss_mode,
>+    kUniformLoss,
>+    "Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot "
>+    "loss, 3--fixed loss.");
> 
>-DEFINE_int(burst_length,
>-           30,
>-           "Burst length in milliseconds, only valid for Gilbert Elliot loss.");
>+WEBRTC_DEFINE_int(
>+    burst_length,
>+    30,
>+    "Burst length in milliseconds, only valid for Gilbert Elliot loss.");
> 
>-DEFINE_float(drift_factor, 0.0, "Time drift factor.");
>+WEBRTC_DEFINE_float(drift_factor, 0.0, "Time drift factor.");
> 
>-DEFINE_int(preload_packets, 0, "Preload the buffer with this many packets.");
>+WEBRTC_DEFINE_int(preload_packets,
>+                  0,
>+                  "Preload the buffer with this many packets.");
> 
>-DEFINE_string(loss_events,
>-              "",
>-              "List of loss events time and duration separated by comma: "
>-              "<first_event_time> <first_event_duration>, <second_event_time> "
>-              "<second_event_duration>, ...");
>+WEBRTC_DEFINE_string(
>+    loss_events,
>+    "",
>+    "List of loss events time and duration separated by comma: "
>+    "<first_event_time> <first_event_duration>, <second_event_time> "
>+    "<second_event_duration>, ...");
> 
> // ProbTrans00Solver() is to calculate the transition probability from no-loss
> // state to itself in a modified Gilbert Elliot packet loss model. The result is
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
>index 25e8cd8816e120ba7b3d852f7a71e07f7bc9f686..c2726eb4704b873be27e0b71cd122c8e936524a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
>@@ -17,17 +17,17 @@
> #include "system_wrappers/include/field_trial.h"
> #include "test/field_trial.h"
> 
>-DEFINE_bool(codec_map,
>-            false,
>-            "Prints the mapping between RTP payload type and "
>-            "codec");
>-DEFINE_string(
>+WEBRTC_DEFINE_bool(codec_map,
>+                   false,
>+                   "Prints the mapping between RTP payload type and "
>+                   "codec");
>+WEBRTC_DEFINE_string(
>     force_fieldtrials,
>     "",
>     "Field trials control experimental feature code which can be forced. "
>     "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
>     " will assign the group Enable to field trial WebRTC-FooFeature.");
>-DEFINE_bool(help, false, "Prints this message");
>+WEBRTC_DEFINE_bool(help, false, "Prints this message");
> 
> int main(int argc, char* argv[]) {
>   webrtc::test::NetEqTestFactory factory;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc
>index 51077bc6b8b9235712bf2f16598cb354a617d93e..aa956ce8abcd311828a3337773f4ccc12d42deda 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc
>@@ -91,47 +91,60 @@ static bool ValidateExtensionId(int value) {
> }
> 
> // Define command line flags.
>-DEFINE_int(pcmu, 0, "RTP payload type for PCM-u");
>-DEFINE_int(pcma, 8, "RTP payload type for PCM-a");
>-DEFINE_int(ilbc, 102, "RTP payload type for iLBC");
>-DEFINE_int(isac, 103, "RTP payload type for iSAC");
>-DEFINE_int(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)");
>-DEFINE_int(opus, 111, "RTP payload type for Opus");
>-DEFINE_int(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)");
>-DEFINE_int(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)");
>-DEFINE_int(pcm16b_swb32, 95, "RTP payload type for PCM16b-swb32 (32 kHz)");
>-DEFINE_int(pcm16b_swb48, 96, "RTP payload type for PCM16b-swb48 (48 kHz)");
>-DEFINE_int(g722, 9, "RTP payload type for G.722");
>-DEFINE_int(avt, 106, "RTP payload type for AVT/DTMF (8 kHz)");
>-DEFINE_int(avt_16, 114, "RTP payload type for AVT/DTMF (16 kHz)");
>-DEFINE_int(avt_32, 115, "RTP payload type for AVT/DTMF (32 kHz)");
>-DEFINE_int(avt_48, 116, "RTP payload type for AVT/DTMF (48 kHz)");
>-DEFINE_int(red, 117, "RTP payload type for redundant audio (RED)");
>-DEFINE_int(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)");
>-DEFINE_int(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)");
>-DEFINE_int(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)");
>-DEFINE_int(cn_swb48, 100, "RTP payload type for comfort noise (48 kHz)");
>-DEFINE_string(replacement_audio_file,
>-              "",
>-              "A PCM file that will be used to populate "
>-              "dummy"
>-              " RTP packets");
>-DEFINE_string(ssrc,
>-              "",
>-              "Only use packets with this SSRC (decimal or hex, the latter "
>-              "starting with 0x)");
>-DEFINE_int(audio_level, 1, "Extension ID for audio level (RFC 6464)");
>-DEFINE_int(abs_send_time, 3, "Extension ID for absolute sender time");
>-DEFINE_int(transport_seq_no, 5, "Extension ID for transport sequence number");
>-DEFINE_int(video_content_type, 7, "Extension ID for video content type");
>-DEFINE_int(video_timing, 8, "Extension ID for video timing");
>-DEFINE_bool(matlabplot,
>-            false,
>-            "Generates a matlab script for plotting the delay profile");
>-DEFINE_bool(pythonplot,
>-            false,
>-            "Generates a python script for plotting the delay profile");
>-DEFINE_bool(concealment_events, false, "Prints concealment events");
>+WEBRTC_DEFINE_int(pcmu, 0, "RTP payload type for PCM-u");
>+WEBRTC_DEFINE_int(pcma, 8, "RTP payload type for PCM-a");
>+WEBRTC_DEFINE_int(ilbc, 102, "RTP payload type for iLBC");
>+WEBRTC_DEFINE_int(isac, 103, "RTP payload type for iSAC");
>+WEBRTC_DEFINE_int(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)");
>+WEBRTC_DEFINE_int(opus, 111, "RTP payload type for Opus");
>+WEBRTC_DEFINE_int(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)");
>+WEBRTC_DEFINE_int(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)");
>+WEBRTC_DEFINE_int(pcm16b_swb32,
>+                  95,
>+                  "RTP payload type for PCM16b-swb32 (32 kHz)");
>+WEBRTC_DEFINE_int(pcm16b_swb48,
>+                  96,
>+                  "RTP payload type for PCM16b-swb48 (48 kHz)");
>+WEBRTC_DEFINE_int(g722, 9, "RTP payload type for G.722");
>+WEBRTC_DEFINE_int(avt, 106, "RTP payload type for AVT/DTMF (8 kHz)");
>+WEBRTC_DEFINE_int(avt_16, 114, "RTP payload type for AVT/DTMF (16 kHz)");
>+WEBRTC_DEFINE_int(avt_32, 115, "RTP payload type for AVT/DTMF (32 kHz)");
>+WEBRTC_DEFINE_int(avt_48, 116, "RTP payload type for AVT/DTMF (48 kHz)");
>+WEBRTC_DEFINE_int(red, 117, "RTP payload type for redundant audio (RED)");
>+WEBRTC_DEFINE_int(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)");
>+WEBRTC_DEFINE_int(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)");
>+WEBRTC_DEFINE_int(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)");
>+WEBRTC_DEFINE_int(cn_swb48, 100, "RTP payload type for comfort noise (48 kHz)");
>+WEBRTC_DEFINE_string(replacement_audio_file,
>+                     "",
>+                     "A PCM file that will be used to populate "
>+                     "dummy"
>+                     " RTP packets");
>+WEBRTC_DEFINE_string(
>+    ssrc,
>+    "",
>+    "Only use packets with this SSRC (decimal or hex, the latter "
>+    "starting with 0x)");
>+WEBRTC_DEFINE_int(audio_level, 1, "Extension ID for audio level (RFC 6464)");
>+WEBRTC_DEFINE_int(abs_send_time, 3, "Extension ID for absolute sender time");
>+WEBRTC_DEFINE_int(transport_seq_no,
>+                  5,
>+                  "Extension ID for transport sequence number");
>+WEBRTC_DEFINE_int(video_content_type, 7, "Extension ID for video content type");
>+WEBRTC_DEFINE_int(video_timing, 8, "Extension ID for video timing");
>+WEBRTC_DEFINE_bool(matlabplot,
>+                   false,
>+                   "Generates a matlab script for plotting the delay profile");
>+WEBRTC_DEFINE_bool(pythonplot,
>+                   false,
>+                   "Generates a python script for plotting the delay profile");
>+WEBRTC_DEFINE_bool(concealment_events, false, "Prints concealment events");
>+WEBRTC_DEFINE_int(max_nr_packets_in_buffer,
>+                  50,
>+                  "Maximum allowed number of packets in the buffer");
>+WEBRTC_DEFINE_bool(enable_fast_accelerate,
>+                   false,
>+                   "Enables jitter buffer fast accelerate");
> 
> // Maps a codec type to a printable name string.
> std::string CodecName(NetEqDecoder codec) {
>@@ -309,25 +322,27 @@ std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTest(
>       {FLAG_video_content_type, kRtpExtensionVideoContentType},
>       {FLAG_video_timing, kRtpExtensionVideoTiming}};
> 
>+  absl::optional<uint32_t> ssrc_filter;
>+  // Check if an SSRC value was provided.
>+  if (strlen(FLAG_ssrc) > 0) {
>+    uint32_t ssrc;
>+    RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc)) << "Flag verification has failed.";
>+    ssrc_filter = ssrc;
>+  }
>+
>   std::unique_ptr<NetEqInput> input;
>   if (RtpFileSource::ValidRtpDump(input_file_name) ||
>       RtpFileSource::ValidPcap(input_file_name)) {
>-    input.reset(new NetEqRtpDumpInput(input_file_name, rtp_ext_map));
>+    input.reset(
>+        new NetEqRtpDumpInput(input_file_name, rtp_ext_map, ssrc_filter));
>   } else {
>-    input.reset(new NetEqEventLogInput(input_file_name, rtp_ext_map));
>+    input.reset(new NetEqEventLogInput(input_file_name, ssrc_filter));
>   }
> 
>   std::cout << "Input file: " << input_file_name << std::endl;
>   RTC_CHECK(input) << "Cannot open input file";
>   RTC_CHECK(!input->ended()) << "Input file is empty";
> 
>-  // Check if an SSRC value was provided.
>-  if (strlen(FLAG_ssrc) > 0) {
>-    uint32_t ssrc;
>-    RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc)) << "Flag verification has failed.";
>-    static_cast<NetEqPacketSourceInput*>(input.get())->SelectSsrc(ssrc);
>-  }
>-
>   // Check the sample rate.
>   absl::optional<int> sample_rate_hz;
>   std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc;
>@@ -457,6 +472,8 @@ std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTest(
>   callbacks.simulation_ended_callback = stats_plotter_.get();
>   NetEq::Config config;
>   config.sample_rate_hz = *sample_rate_hz;
>+  config.max_packets_in_buffer = FLAG_max_nr_packets_in_buffer;
>+  config.enable_fast_accelerate = FLAG_enable_fast_accelerate;
>   return absl::make_unique<NetEqTest>(config, codecs, ext_codecs_,
>                                       std::move(input), std::move(output),
>                                       callbacks);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.cc
>index b1a9b64193bf75b79a70c76e20f2012e89f5308a..0e7951b6e3c343ec22b970c1d61279c0872b6c51 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.cc
>@@ -49,6 +49,20 @@ Packet::Packet(uint8_t* packet_memory,
>   valid_header_ = ParseHeader(parser);
> }
> 
>+Packet::Packet(const RTPHeader& header,
>+               size_t virtual_packet_length_bytes,
>+               size_t virtual_payload_length_bytes,
>+               double time_ms)
>+    : header_(header),
>+      payload_memory_(),
>+      payload_(NULL),
>+      packet_length_bytes_(0),
>+      payload_length_bytes_(0),
>+      virtual_packet_length_bytes_(virtual_packet_length_bytes),
>+      virtual_payload_length_bytes_(virtual_payload_length_bytes),
>+      time_ms_(time_ms),
>+      valid_header_(true) {}
>+
> Packet::Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms)
>     : payload_memory_(packet_memory),
>       payload_(NULL),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.h
>index 623c5cb6359a99f5c93a39d23a07e411cba91ace..39137aae43a0aec59234007542b108fcfa7b883d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.h
>@@ -50,7 +50,17 @@ class Packet {
>          double time_ms,
>          const RtpHeaderParser& parser);
> 
>-  // The following two constructors are the same as above, but without a
>+  // Same as above, but creates the packet from an already parsed RTPHeader.
>+  // This is typically used when reading RTP dump files that only contain the
>+  // RTP headers, and no payload. The |virtual_packet_length_bytes| tells what
>+  // size the packet had on wire, including the now discarded payload,
>+  // The |virtual_payload_length_bytes| tells the size of the payload.
>+  Packet(const RTPHeader& header,
>+         size_t virtual_packet_length_bytes,
>+         size_t virtual_payload_length_bytes,
>+         double time_ms);
>+
>+  // The following constructors are the same as the first two, but without a
>   // parser. Note that when the object is constructed using any of these
>   // methods, the header will be parsed using a default RtpHeaderParser object.
>   // In particular, RTP header extensions won't be parsed.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.cc
>index 30bf431835552395e48cf789aaa3c42dcf7c45f1..598ae6edd408d3e3ec09f129dd5aca1ebb12ddbb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.cc
>@@ -13,7 +13,7 @@
> namespace webrtc {
> namespace test {
> 
>-PacketSource::PacketSource() : use_ssrc_filter_(false), ssrc_(0) {}
>+PacketSource::PacketSource() = default;
> 
> PacketSource::~PacketSource() = default;
> 
>@@ -21,10 +21,5 @@ void PacketSource::FilterOutPayloadType(uint8_t payload_type) {
>   filter_.set(payload_type, true);
> }
> 
>-void PacketSource::SelectSsrc(uint32_t ssrc) {
>-  use_ssrc_filter_ = true;
>-  ssrc_ = ssrc;
>-}
>-
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.h
>index fb689e33d4e4852c981440a2ae2180b82515804d..cb86a98999de06bfee85c568fe98cb6ff23e2082 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.h
>@@ -32,13 +32,8 @@ class PacketSource {
> 
>   virtual void FilterOutPayloadType(uint8_t payload_type);
> 
>-  virtual void SelectSsrc(uint32_t ssrc);
>-
>  protected:
>   std::bitset<128> filter_;  // Payload type is 7 bits in the RFC.
>-  // If SSRC filtering discards all packet that do not match the SSRC.
>-  bool use_ssrc_filter_;  // True when SSRC filtering is active.
>-  uint32_t ssrc_;  // The selected SSRC. All other SSRCs will be discarded.
> 
>  private:
>   RTC_DISALLOW_COPY_AND_ASSIGN(PacketSource);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
>index 9e435c7cf88fc2b9d6d866de480889f186b3f83f..582c2f2f13551f987baaf91aaea2ec8976ff2d4e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
>@@ -13,94 +13,115 @@
> #include <string.h>
> #include <iostream>
> #include <limits>
>+#include <utility>
> 
>+#include "logging/rtc_event_log/rtc_event_processor.h"
> #include "modules/audio_coding/neteq/tools/packet.h"
>-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
> namespace test {
> 
>-RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) {
>+namespace {
>+bool ShouldSkipStream(ParsedRtcEventLogNew::MediaType media_type,
>+                      uint32_t ssrc,
>+                      absl::optional<uint32_t> ssrc_filter) {
>+  if (media_type != ParsedRtcEventLogNew::MediaType::AUDIO)
>+    return true;
>+  if (ssrc_filter.has_value() && ssrc != *ssrc_filter)
>+    return true;
>+  return false;
>+}
>+}  // namespace
>+
>+RtcEventLogSource* RtcEventLogSource::Create(
>+    const std::string& file_name,
>+    absl::optional<uint32_t> ssrc_filter) {
>   RtcEventLogSource* source = new RtcEventLogSource();
>-  RTC_CHECK(source->OpenFile(file_name));
>+  RTC_CHECK(source->OpenFile(file_name, ssrc_filter));
>   return source;
> }
> 
> RtcEventLogSource::~RtcEventLogSource() {}
> 
>-bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
>-                                                   uint8_t id) {
>-  RTC_CHECK(parser_.get());
>-  return parser_->RegisterRtpHeaderExtension(type, id);
>-}
>-
> std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
>-  for (; rtp_packet_index_ < parsed_stream_.GetNumberOfEvents();
>-       rtp_packet_index_++) {
>-    if (parsed_stream_.GetEventType(rtp_packet_index_) ==
>-        ParsedRtcEventLogNew::EventType::RTP_EVENT) {
>-      PacketDirection direction;
>-      size_t header_length;
>-      size_t packet_length;
>-      uint64_t timestamp_us = parsed_stream_.GetTimestamp(rtp_packet_index_);
>-      parsed_stream_.GetRtpHeader(rtp_packet_index_, &direction, nullptr,
>-                                  &header_length, &packet_length, nullptr);
>-
>-      if (direction != kIncomingPacket) {
>-        continue;
>-      }
>-
>-      uint8_t* packet_header = new uint8_t[header_length];
>-      parsed_stream_.GetRtpHeader(rtp_packet_index_, nullptr, packet_header,
>-                                  nullptr, nullptr, nullptr);
>-      std::unique_ptr<Packet> packet(
>-          new Packet(packet_header, header_length, packet_length,
>-                     static_cast<double>(timestamp_us) / 1000, *parser_.get()));
>-
>-      if (!packet->valid_header()) {
>-        std::cout << "Warning: Packet with index " << rtp_packet_index_
>-                  << " has an invalid header and will be ignored." << std::endl;
>-        continue;
>-      }
>-
>-      if (parsed_stream_.GetMediaType(packet->header().ssrc, direction) !=
>-          ParsedRtcEventLogNew::MediaType::AUDIO) {
>-        continue;
>-      }
>-
>-      // Check if the packet should not be filtered out.
>-      if (!filter_.test(packet->header().payloadType) &&
>-          !(use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
>-        ++rtp_packet_index_;
>-        return packet;
>-      }
>-    }
>-  }
>-  return nullptr;
>+  if (rtp_packet_index_ >= rtp_packets_.size())
>+    return nullptr;
>+
>+  std::unique_ptr<Packet> packet = std::move(rtp_packets_[rtp_packet_index_++]);
>+  return packet;
> }
> 
> int64_t RtcEventLogSource::NextAudioOutputEventMs() {
>-  while (audio_output_index_ < parsed_stream_.GetNumberOfEvents()) {
>-    if (parsed_stream_.GetEventType(audio_output_index_) ==
>-        ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT) {
>-      LoggedAudioPlayoutEvent playout_event =
>-          parsed_stream_.GetAudioPlayout(audio_output_index_);
>-      if (!(use_ssrc_filter_ && playout_event.ssrc != ssrc_)) {
>-        audio_output_index_++;
>-        return playout_event.timestamp_us / 1000;
>-      }
>+  if (audio_output_index_ >= audio_outputs_.size())
>+    return std::numeric_limits<int64_t>::max();
>+
>+  int64_t output_time_ms = audio_outputs_[audio_output_index_++];
>+  return output_time_ms;
>+}
>+
>+RtcEventLogSource::RtcEventLogSource() : PacketSource() {}
>+
>+bool RtcEventLogSource::OpenFile(const std::string& file_name,
>+                                 absl::optional<uint32_t> ssrc_filter) {
>+  ParsedRtcEventLogNew parsed_log;
>+  if (!parsed_log.ParseFile(file_name))
>+    return false;
>+
>+  const auto first_log_end_time_us =
>+      parsed_log.stop_log_events().empty()
>+          ? std::numeric_limits<int64_t>::max()
>+          : parsed_log.stop_log_events().front().log_time_us();
>+
>+  auto handle_rtp_packet =
>+      [this,
>+       first_log_end_time_us](const webrtc::LoggedRtpPacketIncoming& incoming) {
>+        if (!filter_.test(incoming.rtp.header.payloadType) &&
>+            incoming.log_time_us() < first_log_end_time_us) {
>+          rtp_packets_.emplace_back(absl::make_unique<Packet>(
>+              incoming.rtp.header, incoming.rtp.total_length,
>+              incoming.rtp.total_length - incoming.rtp.header_length,
>+              static_cast<double>(incoming.log_time_ms())));
>+        }
>+      };
>+
>+  auto handle_audio_playout =
>+      [this, first_log_end_time_us](
>+          const webrtc::LoggedAudioPlayoutEvent& audio_playout) {
>+        if (audio_playout.log_time_us() < first_log_end_time_us) {
>+          audio_outputs_.emplace_back(audio_playout.log_time_ms());
>+        }
>+      };
>+
>+  // This wouldn't be needed if we knew that there was at most one audio stream.
>+  webrtc::RtcEventProcessor event_processor;
>+  for (const auto& rtp_packets : parsed_log.incoming_rtp_packets_by_ssrc()) {
>+    ParsedRtcEventLogNew::MediaType media_type =
>+        parsed_log.GetMediaType(rtp_packets.ssrc, webrtc::kIncomingPacket);
>+    if (ShouldSkipStream(media_type, rtp_packets.ssrc, ssrc_filter)) {
>+      continue;
>     }
>-    audio_output_index_++;
>+    auto rtp_view = absl::make_unique<
>+        webrtc::ProcessableEventList<webrtc::LoggedRtpPacketIncoming>>(
>+        rtp_packets.incoming_packets.begin(),
>+        rtp_packets.incoming_packets.end(), handle_rtp_packet);
>+    event_processor.AddEvents(std::move(rtp_view));
>+  }
>+
>+  for (const auto& audio_playouts : parsed_log.audio_playout_events()) {
>+    if (ssrc_filter.has_value() && audio_playouts.first != *ssrc_filter)
>+      continue;
>+    auto audio_view = absl::make_unique<
>+        webrtc::ProcessableEventList<webrtc::LoggedAudioPlayoutEvent>>(
>+        audio_playouts.second.begin(), audio_playouts.second.end(),
>+        handle_audio_playout);
>+    event_processor.AddEvents(std::move(audio_view));
>   }
>-  return std::numeric_limits<int64_t>::max();
>-}
> 
>-RtcEventLogSource::RtcEventLogSource()
>-    : PacketSource(), parser_(RtpHeaderParser::Create()) {}
>+  // Fills in rtp_packets_ and audio_outputs_.
>+  event_processor.ProcessEventsInOrder();
> 
>-bool RtcEventLogSource::OpenFile(const std::string& file_name) {
>-  return parsed_stream_.ParseFile(file_name);
>+  return true;
> }
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
>index db4eb19e11f369db904470cbb528eb254586e1ac..6b761f7dbbd262d6fcedaa58aff205af890e6df3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
>@@ -13,7 +13,9 @@
> 
> #include <memory>
> #include <string>
>+#include <vector>
> 
>+#include "absl/types/optional.h"
> #include "logging/rtc_event_log/rtc_event_log_parser_new.h"
> #include "modules/audio_coding/neteq/tools/packet_source.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>@@ -31,13 +33,11 @@ class RtcEventLogSource : public PacketSource {
>  public:
>   // Creates an RtcEventLogSource reading from |file_name|. If the file cannot
>   // be opened, or has the wrong format, NULL will be returned.
>-  static RtcEventLogSource* Create(const std::string& file_name);
>+  static RtcEventLogSource* Create(const std::string& file_name,
>+                                   absl::optional<uint32_t> ssrc_filter);
> 
>   virtual ~RtcEventLogSource();
> 
>-  // Registers an RTP header extension and binds it to |id|.
>-  virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
>-
>   std::unique_ptr<Packet> NextPacket() override;
> 
>   // Returns the timestamp of the next audio output event, in milliseconds. The
>@@ -48,14 +48,14 @@ class RtcEventLogSource : public PacketSource {
>  private:
>   RtcEventLogSource();
> 
>-  bool OpenFile(const std::string& file_name);
>+  bool OpenFile(const std::string& file_name,
>+                absl::optional<uint32_t> ssrc_filter);
> 
>+  std::vector<std::unique_ptr<Packet>> rtp_packets_;
>   size_t rtp_packet_index_ = 0;
>+  std::vector<int64_t> audio_outputs_;
>   size_t audio_output_index_ = 0;
> 
>-  ParsedRtcEventLogNew parsed_stream_;
>-  std::unique_ptr<RtpHeaderParser> parser_;
>-
>   RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
>index f9390381cd38d0d12ec113a313565eadf70484fe..9d3041ea668dca1debdc320cba5463630a248d8c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
>@@ -19,16 +19,16 @@
> #include "rtc_base/flags.h"
> 
> // Define command line flags.
>-DEFINE_int(red, 117, "RTP payload type for RED");
>-DEFINE_int(audio_level,
>-           -1,
>-           "Extension ID for audio level (RFC 6464); "
>-           "-1 not to print audio level");
>-DEFINE_int(abs_send_time,
>-           -1,
>-           "Extension ID for absolute sender time; "
>-           "-1 not to print absolute send time");
>-DEFINE_bool(help, false, "Print this message");
>+WEBRTC_DEFINE_int(red, 117, "RTP payload type for RED");
>+WEBRTC_DEFINE_int(audio_level,
>+                  -1,
>+                  "Extension ID for audio level (RFC 6464); "
>+                  "-1 not to print audio level");
>+WEBRTC_DEFINE_int(abs_send_time,
>+                  -1,
>+                  "Extension ID for absolute sender time; "
>+                  "-1 not to print absolute send time");
>+WEBRTC_DEFINE_bool(help, false, "Print this message");
> 
> int main(int argc, char* argv[]) {
>   std::string program_name = argv[0];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_encode.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_encode.cc
>index 5065ca16757d06fa525e6e3e77ad41d7cf21b298..14c6e58a9d644a731fc05ee84aa9f0c33cd84af6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_encode.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_encode.cc
>@@ -40,20 +40,24 @@ namespace test {
> namespace {
> 
> // Define command line flags.
>-DEFINE_bool(list_codecs, false, "Enumerate all codecs");
>-DEFINE_string(codec, "opus", "Codec to use");
>-DEFINE_int(frame_len, 0, "Frame length in ms; 0 indicates codec default value");
>-DEFINE_int(bitrate, 0, "Bitrate in kbps; 0 indicates codec default value");
>-DEFINE_int(payload_type,
>-           -1,
>-           "RTP payload type; -1 indicates codec default value");
>-DEFINE_int(cng_payload_type,
>-           -1,
>-           "RTP payload type for CNG; -1 indicates default value");
>-DEFINE_int(ssrc, 0, "SSRC to write to the RTP header");
>-DEFINE_bool(dtx, false, "Use DTX/CNG");
>-DEFINE_int(sample_rate, 48000, "Sample rate of the input file");
>-DEFINE_bool(help, false, "Print this message");
>+WEBRTC_DEFINE_bool(list_codecs, false, "Enumerate all codecs");
>+WEBRTC_DEFINE_string(codec, "opus", "Codec to use");
>+WEBRTC_DEFINE_int(frame_len,
>+                  0,
>+                  "Frame length in ms; 0 indicates codec default value");
>+WEBRTC_DEFINE_int(bitrate,
>+                  0,
>+                  "Bitrate in kbps; 0 indicates codec default value");
>+WEBRTC_DEFINE_int(payload_type,
>+                  -1,
>+                  "RTP payload type; -1 indicates codec default value");
>+WEBRTC_DEFINE_int(cng_payload_type,
>+                  -1,
>+                  "RTP payload type for CNG; -1 indicates default value");
>+WEBRTC_DEFINE_int(ssrc, 0, "SSRC to write to the RTP header");
>+WEBRTC_DEFINE_bool(dtx, false, "Use DTX/CNG");
>+WEBRTC_DEFINE_int(sample_rate, 48000, "Sample rate of the input file");
>+WEBRTC_DEFINE_bool(help, false, "Print this message");
> 
> // Add new codecs here, and to the map below.
> enum class CodecType {
>@@ -242,8 +246,8 @@ std::unique_ptr<AudioEncoder> CreateEncoder(CodecType codec_type,
>   return nullptr;
> }
> 
>-AudioEncoderCng::Config GetCngConfig(int sample_rate_hz) {
>-  AudioEncoderCng::Config cng_config;
>+AudioEncoderCngConfig GetCngConfig(int sample_rate_hz) {
>+  AudioEncoderCngConfig cng_config;
>   const auto default_payload_type = [&] {
>     switch (sample_rate_hz) {
>       case 8000:
>@@ -309,10 +313,10 @@ int RunRtpEncode(int argc, char* argv[]) {
> 
>   // Create an external VAD/CNG encoder if needed.
>   if (FLAG_dtx && !codec_it->second.internal_dtx) {
>-    AudioEncoderCng::Config cng_config = GetCngConfig(codec->SampleRateHz());
>+    AudioEncoderCngConfig cng_config = GetCngConfig(codec->SampleRateHz());
>     RTC_DCHECK(codec);
>     cng_config.speech_encoder = std::move(codec);
>-    codec = absl::make_unique<AudioEncoderCng>(std::move(cng_config));
>+    codec = CreateComfortNoiseEncoder(std::move(cng_config));
>   }
>   RTC_DCHECK(codec);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>index 7bf7d52083cc65c4018e1372ea4ada90e9fc3bb6..eda2b3e31fcbd4a069a15edd026442c4896d1495 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>@@ -26,8 +26,9 @@
> namespace webrtc {
> namespace test {
> 
>-RtpFileSource* RtpFileSource::Create(const std::string& file_name) {
>-  RtpFileSource* source = new RtpFileSource();
>+RtpFileSource* RtpFileSource::Create(const std::string& file_name,
>+                                     absl::optional<uint32_t> ssrc_filter) {
>+  RtpFileSource* source = new RtpFileSource(ssrc_filter);
>   RTC_CHECK(source->OpenFile(file_name));
>   return source;
> }
>@@ -72,7 +73,7 @@ std::unique_ptr<Packet> RtpFileSource::NextPacket() {
>       continue;
>     }
>     if (filter_.test(packet->header().payloadType) ||
>-        (use_ssrc_filter_ && packet->header().ssrc != ssrc_)) {
>+        (ssrc_filter_ && packet->header().ssrc != *ssrc_filter_)) {
>       // This payload type should be filtered out. Continue to the next packet.
>       continue;
>     }
>@@ -80,8 +81,10 @@ std::unique_ptr<Packet> RtpFileSource::NextPacket() {
>   }
> }
> 
>-RtpFileSource::RtpFileSource()
>-    : PacketSource(), parser_(RtpHeaderParser::Create()) {}
>+RtpFileSource::RtpFileSource(absl::optional<uint32_t> ssrc_filter)
>+    : PacketSource(),
>+      parser_(RtpHeaderParser::Create()),
>+      ssrc_filter_(ssrc_filter) {}
> 
> bool RtpFileSource::OpenFile(const std::string& file_name) {
>   rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
>@@ -89,7 +92,7 @@ bool RtpFileSource::OpenFile(const std::string& file_name) {
>     return true;
>   rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name));
>   if (!rtp_reader_) {
>-    RTC_FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
>+    FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
>                "that .pcapng is not supported.";
>   }
>   return true;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
>index 013333b89a369fa222f47cc2890f1f79aee74fa1..02ab897566190b335215cc683f9609b9bf720e8b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
>@@ -16,6 +16,7 @@
> #include <memory>
> #include <string>
> 
>+#include "absl/types/optional.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/neteq/tools/packet_source.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>@@ -33,7 +34,9 @@ class RtpFileSource : public PacketSource {
>  public:
>   // Creates an RtpFileSource reading from |file_name|. If the file cannot be
>   // opened, or has the wrong format, NULL will be returned.
>-  static RtpFileSource* Create(const std::string& file_name);
>+  static RtpFileSource* Create(
>+      const std::string& file_name,
>+      absl::optional<uint32_t> ssrc_filter = absl::nullopt);
> 
>   // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
>   static bool ValidRtpDump(const std::string& file_name);
>@@ -51,12 +54,13 @@ class RtpFileSource : public PacketSource {
>   static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
>   static const size_t kPacketHeaderSize = 8;
> 
>-  RtpFileSource();
>+  explicit RtpFileSource(absl::optional<uint32_t> ssrc_filter);
> 
>   bool OpenFile(const std::string& file_name);
> 
>   std::unique_ptr<RtpFileReader> rtp_reader_;
>   std::unique_ptr<RtpHeaderParser> parser_;
>+  const absl::optional<uint32_t> ssrc_filter_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc
>index 3c49443036c39f7322ef1e8e637c32d616819789..92a7a8d2150afb44e17fcc6d3e14905f67884a93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc
>@@ -23,7 +23,7 @@ namespace webrtc {
> namespace test {
> namespace {
> 
>-DEFINE_bool(help, false, "Print help message");
>+WEBRTC_DEFINE_bool(help, false, "Print help message");
> 
> constexpr size_t kRtpDumpHeaderLength = 8;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
>index c6f3f8c9f2562d492e48dde354654853c61d25eb..240836688cf0cfae2331af85826732dc6e29d8f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
>@@ -14,9 +14,9 @@
> #include <stdlib.h>
> #include <memory>
> 
>+#include "absl/strings/match.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
> #include "modules/audio_coding/test/utility.h"
>@@ -248,11 +248,11 @@ void EncodeDecodeTest::Perform() {
> 
>   for (int n = 0; n < numCodecs; n++) {
>     EXPECT_EQ(0, acm->Codec(n, &sendCodecTmp));
>-    if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
>+    if (absl::EqualsIgnoreCase(sendCodecTmp.plname, "telephone-event")) {
>       numPars[n] = 0;
>-    } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
>+    } else if (absl::EqualsIgnoreCase(sendCodecTmp.plname, "cn")) {
>       numPars[n] = 0;
>-    } else if (STR_CASE_CMP(sendCodecTmp.plname, "red") == 0) {
>+    } else if (absl::EqualsIgnoreCase(sendCodecTmp.plname, "red")) {
>       numPars[n] = 0;
>     } else if (sendCodecTmp.channels == 2) {
>       numPars[n] = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.cc
>index c56fed95dab96dc552c019b2bd22a785cca60e8a..8bb3971373da7f0721081a721b75d374e494961a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.cc
>@@ -173,14 +173,14 @@ void TestRedFec::RegisterSendCodec(
>   auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format,
>                                                     absl::nullopt);
>   EXPECT_NE(encoder, nullptr);
>-  if (STR_CASE_CMP(codec_format.name.c_str(), "opus") != 0) {
>+  if (!absl::EqualsIgnoreCase(codec_format.name, "opus")) {
>     if (vad_mode.has_value()) {
>-      AudioEncoderCng::Config config;
>+      AudioEncoderCngConfig config;
>       config.speech_encoder = std::move(encoder);
>       config.num_channels = 1;
>       config.payload_type = cn_payload_type;
>       config.vad_mode = vad_mode.value();
>-      encoder = absl::make_unique<AudioEncoderCng>(std::move(config));
>+      encoder = CreateComfortNoiseEncoder(std::move(config));
>       EXPECT_EQ(true,
>                 other_acm->RegisterReceiveCodec(
>                     cn_payload_type, {"CN", codec_format.clockrate_hz, 1}));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.cc
>index fd69fd262622e3143651da87f4f251bb8d8336a1..bf8e1894d0e41b936322dd5f7db4f51bd0042e22 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.cc
>@@ -12,9 +12,9 @@
> 
> #include <string>
> 
>+#include "absl/strings/match.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
> #include "modules/audio_coding/test/utility.h"
>@@ -626,14 +626,14 @@ void TestStereo::RegisterSendCodec(char side,
>   ASSERT_TRUE(my_acm != NULL);
> 
>   auto encoder_factory = CreateBuiltinAudioEncoderFactory();
>-  const int clockrate_hz = STR_CASE_CMP(codec_name, "g722") == 0
>+  const int clockrate_hz = absl::EqualsIgnoreCase(codec_name, "g722")
>                                ? sampling_freq_hz / 2
>                                : sampling_freq_hz;
>   const std::string ptime = rtc::ToString(rtc::CheckedDivExact(
>       pack_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
>   SdpAudioFormat::Parameters params = {{"ptime", ptime}};
>   RTC_CHECK(channels == 1 || channels == 2);
>-  if (STR_CASE_CMP(codec_name, "opus") == 0) {
>+  if (absl::EqualsIgnoreCase(codec_name, "opus")) {
>     if (channels == 2) {
>       params["stereo"] = "1";
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.cc
>index 4f02edaa83744c193b0e742ec22f78738e3e8eba..09c69f9c6bbe566096fa7bf2caa41fed5f71aec8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.cc
>@@ -12,6 +12,7 @@
> 
> #include <string>
> 
>+#include "absl/strings/match.h"
> #include "api/audio_codecs/audio_decoder_factory_template.h"
> #include "api/audio_codecs/audio_encoder_factory_template.h"
> #include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
>@@ -81,13 +82,13 @@ bool TestVadDtx::RegisterCodec(const SdpAudioFormat& codec_format,
>   auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format,
>                                                     absl::nullopt);
>   if (vad_mode.has_value() &&
>-      STR_CASE_CMP(codec_format.name.c_str(), "opus") != 0) {
>-    AudioEncoderCng::Config config;
>+      !absl::EqualsIgnoreCase(codec_format.name, "opus")) {
>+    AudioEncoderCngConfig config;
>     config.speech_encoder = std::move(encoder);
>     config.num_channels = 1;
>     config.payload_type = cn_payload_type;
>     config.vad_mode = vad_mode.value();
>-    encoder = absl::make_unique<AudioEncoderCng>(std::move(config));
>+    encoder = CreateComfortNoiseEncoder(std::move(config));
>     added_comfort_noise = true;
>   }
>   channel_->SetIsStereo(encoder->NumChannels() > 1);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.cc
>index a130ae3e66fdac8b03101690a0f094dff1f6b6bb..c332fe08f11183694b1a89eb89d1b4a31c28b6c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.cc
>@@ -23,6 +23,7 @@
> #include <time.h>
> #endif
> 
>+#include "absl/strings/match.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/isac/audio_encoder_isac_float.h"
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
>@@ -92,12 +93,12 @@ void ISACTest::Setup() {
>   for (codecCntr = 0; codecCntr < AudioCodingModule::NumberOfCodecs();
>        codecCntr++) {
>     EXPECT_EQ(0, AudioCodingModule::Codec(codecCntr, &codecParam));
>-    if (!STR_CASE_CMP(codecParam.plname, "ISAC") &&
>+    if (absl::EqualsIgnoreCase(codecParam.plname, "ISAC") &&
>         codecParam.plfreq == 16000) {
>       memcpy(&_paramISAC16kHz, &codecParam, sizeof(CodecInst));
>       _idISAC16kHz = codecCntr;
>     }
>-    if (!STR_CASE_CMP(codecParam.plname, "ISAC") &&
>+    if (absl::EqualsIgnoreCase(codecParam.plname, "ISAC") &&
>         codecParam.plfreq == 32000) {
>       memcpy(&_paramISAC32kHz, &codecParam, sizeof(CodecInst));
>       _idISAC32kHz = codecCntr;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.cc
>index 83c25b537e45c18270cfc33e86947564257f3f5d..53f807790ef3377ab20ef4700a913d90acb24d89 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.cc
>@@ -15,7 +15,7 @@
> #include <stdlib.h>
> #include <string.h>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/strings/match.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
> #include "test/gtest.h"
> 
>@@ -268,7 +268,7 @@ bool FixedPayloadTypeCodec(const char* payloadName) {
>       "G722", "QCELP", "CN",  "MPA",  "G728", "G729"};
> 
>   for (int n = 0; n < NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE; n++) {
>-    if (!STR_CASE_CMP(payloadName, fixPayloadTypeCodecs[n])) {
>+    if (absl::EqualsIgnoreCase(payloadName, fixPayloadTypeCodecs[n])) {
>       return true;
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/BUILD.gn
>index b690050981194395d7a075895197b71bd785fa67..67495aca77556e75593a1331d2e5f5838d444f8a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/BUILD.gn
>@@ -24,6 +24,16 @@ config("audio_device_warnings_config") {
>   }
> }
> 
>+rtc_source_set("audio_device_default") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "include/audio_device_default.h",
>+  ]
>+  deps = [
>+    ":audio_device_api",
>+  ]
>+}
>+
> rtc_source_set("audio_device") {
>   visibility = [ "*" ]
>   public_deps = [
>@@ -203,8 +213,8 @@ rtc_source_set("audio_device_impl") {
>   deps = [
>     ":audio_device_api",
>     ":audio_device_buffer",
>+    ":audio_device_default",
>     ":audio_device_generic",
>-    "../..:webrtc_common",
>     "../../api:array_view",
>     "../../common_audio",
>     "../../common_audio:common_audio_c",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_unittest.cc
>index b0cf42edec53a31595f9db47952e404f0db0a680..6e09139c79b68850f8d922ba945c687852478d49 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_unittest.cc
>@@ -25,10 +25,10 @@
> #include "modules/audio_device/include/mock_audio_transport.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/format_macros.h"
> #include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/timeutils.h"
>-#include "system_wrappers/include/event_wrapper.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>@@ -365,7 +365,7 @@ class MockAudioTransportAndroid : public test::MockAudioTransport {
> 
>   // Set default actions of the mock object. We are delegating to fake
>   // implementations (of AudioStreamInterface) here.
>-  void HandleCallbacks(EventWrapper* test_is_done,
>+  void HandleCallbacks(rtc::Event* test_is_done,
>                        AudioStreamInterface* audio_stream,
>                        int num_callbacks) {
>     test_is_done_ = test_is_done;
>@@ -448,7 +448,7 @@ class MockAudioTransportAndroid : public test::MockAudioTransport {
>   bool rec_mode() const { return type_ & kRecording; }
> 
>  private:
>-  EventWrapper* test_is_done_;
>+  rtc::Event* test_is_done_;
>   size_t num_callbacks_;
>   int type_;
>   size_t play_count_;
>@@ -460,7 +460,7 @@ class MockAudioTransportAndroid : public test::MockAudioTransport {
> // AudioDeviceTest test fixture.
> class AudioDeviceTest : public ::testing::Test {
>  protected:
>-  AudioDeviceTest() : test_is_done_(EventWrapper::Create()) {
>+  AudioDeviceTest() {
>     // One-time initialization of JVM and application context. Ensures that we
>     // can do calls between C++ and Java. Initializes both Java and OpenSL ES
>     // implementations.
>@@ -638,7 +638,7 @@ class AudioDeviceTest : public ::testing::Test {
>     return volume;
>   }
> 
>-  std::unique_ptr<EventWrapper> test_is_done_;
>+  rtc::Event test_is_done_;
>   rtc::scoped_refptr<AudioDeviceModule> audio_device_;
>   AudioParameters playout_parameters_;
>   AudioParameters record_parameters_;
>@@ -867,14 +867,14 @@ TEST_F(AudioDeviceTest, StopRecordingRequiresInitToRestart) {
> // audio samples to play out using the NeedMorePlayData callback.
> TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
>   MockAudioTransportAndroid mock(kPlayout);
>-  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
>   EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
>                                      kBytesPerSample, playout_channels(),
>                                      playout_sample_rate(), NotNull(), _, _, _))
>       .Times(AtLeast(kNumCallbacks));
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartPlayout();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopPlayout();
> }
> 
>@@ -884,7 +884,7 @@ TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
> // delay estimates as well (argument #6).
> TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
>   MockAudioTransportAndroid mock(kRecording);
>-  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
>   EXPECT_CALL(
>       mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
>                                     kBytesPerSample, record_channels(),
>@@ -893,7 +893,7 @@ TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
> 
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartRecording();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopRecording();
> }
> 
>@@ -901,7 +901,7 @@ TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
> // active in both directions.
> TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
>   MockAudioTransportAndroid mock(kPlayout | kRecording);
>-  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
>   EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
>                                      kBytesPerSample, playout_channels(),
>                                      playout_sample_rate(), NotNull(), _, _, _))
>@@ -914,7 +914,7 @@ TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartPlayout();
>   StartRecording();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopRecording();
>   StopPlayout();
> }
>@@ -930,12 +930,11 @@ TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
>   std::string file_name = GetFileName(playout_sample_rate());
>   std::unique_ptr<FileAudioStream> file_audio_stream(
>       new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
>-  mock.HandleCallbacks(test_is_done_.get(), file_audio_stream.get(),
>-                       num_callbacks);
>+  mock.HandleCallbacks(&test_is_done_, file_audio_stream.get(), num_callbacks);
>   // SetMaxPlayoutVolume();
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartPlayout();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopPlayout();
> }
> 
>@@ -964,13 +963,13 @@ TEST_F(AudioDeviceTest, DISABLED_RunPlayoutAndRecordingInFullDuplex) {
>   NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
>   std::unique_ptr<FifoAudioStream> fifo_audio_stream(
>       new FifoAudioStream(playout_frames_per_10ms_buffer()));
>-  mock.HandleCallbacks(test_is_done_.get(), fifo_audio_stream.get(),
>+  mock.HandleCallbacks(&test_is_done_, fifo_audio_stream.get(),
>                        kFullDuplexTimeInSec * kNumCallbacksPerSecond);
>   SetMaxPlayoutVolume();
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartRecording();
>   StartPlayout();
>-  test_is_done_->Wait(
>+  test_is_done_.Wait(
>       std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
>   StopPlayout();
>   StopRecording();
>@@ -997,14 +996,14 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
>   NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
>   std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
>       new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
>-  mock.HandleCallbacks(test_is_done_.get(), latency_audio_stream.get(),
>+  mock.HandleCallbacks(&test_is_done_, latency_audio_stream.get(),
>                        kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   SetMaxPlayoutVolume();
>   DisableBuiltInAECIfAvailable();
>   StartRecording();
>   StartPlayout();
>-  test_is_done_->Wait(
>+  test_is_done_.Wait(
>       std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
>   StopPlayout();
>   StopRecording();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.cc
>index 5c45780426b78f914a9b195161568c47d81d87a5..8f920cfff8ea9c34a365562b968aa0e7b7c74f04 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.cc
>@@ -8,19 +8,16 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <algorithm>
>+#include <string.h>
> #include <cmath>
>-
>-#include "modules/audio_device/audio_device_buffer.h"
>+#include <cstddef>
>+#include <cstdint>
> 
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>-#include "modules/audio_device/audio_device_config.h"
>-#include "rtc_base/arraysize.h"
>+#include "modules/audio_device/audio_device_buffer.h"
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/format_macros.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/metrics.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.h
>index 7f0cf83baf7d7ea7d6e7bc7dc0331735ebbbe1e3..bbe2969d43697558b65874e3501e7eda607990af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.h
>@@ -11,9 +11,11 @@
> #ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
> #define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <atomic>
> 
>-#include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/task_queue.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_data_observer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_data_observer.cc
>index 723d1c8bbdc74c9a9c6bd2856a412a9980818b8b..994c2efdf33210a1bf46ff72ed37596484b5e2a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_data_observer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_data_observer.cc
>@@ -9,6 +9,8 @@
>  */
> 
> #include "modules/audio_device/include/audio_device_data_observer.h"
>+
>+#include "modules/audio_device/include/audio_device_defines.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/refcountedobject.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_generic.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_generic.h
>index be9c072c412ce84b1bc74a493305d1b3989afb30..7d3c83e119603bb6ebde9225a9857243587bdf47 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_generic.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_generic.h
>@@ -11,8 +11,11 @@
> #ifndef AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H_
> #define AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H_
> 
>+#include <stdint.h>
>+
> #include "modules/audio_device/audio_device_buffer.h"
> #include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.cc
>index 91d6208fe1fd1e8e85470a68ef52e9590a642198..31d5b5ecfc51e9a8ca53b2c3aee89865983e06f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.cc
>@@ -10,12 +10,14 @@
> 
> #include "modules/audio_device/audio_device_impl.h"
> 
>-#include "modules/audio_device/audio_device_config.h"
>+#include <stddef.h>
>+
>+#include "modules/audio_device/audio_device_config.h"  // IWYU pragma: keep
> #include "modules/audio_device/audio_device_generic.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/refcount.h"
> #include "rtc_base/refcountedobject.h"
>+#include "rtc_base/scoped_ref_ptr.h"
> #include "system_wrappers/include/metrics.h"
> 
> #if defined(_WIN32)
>@@ -47,10 +49,10 @@
> #include "modules/audio_device/mac/audio_device_mac.h"
> #endif
> #if defined(WEBRTC_DUMMY_FILE_DEVICES)
>+#include "modules/audio_device/dummy/file_audio_device.h"
> #include "modules/audio_device/dummy/file_audio_device_factory.h"
> #endif
> #include "modules/audio_device/dummy/audio_device_dummy.h"
>-#include "modules/audio_device/dummy/file_audio_device.h"
> 
> #define CHECKinitialized_() \
>   {                         \
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.h
>index 481cdf32e00dca4ac9f766df25b0edfa1f1187d6..afe53b3240328d941769858e345898cd7c123819 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.h
>@@ -13,12 +13,11 @@
> 
> #if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "modules/audio_device/audio_device_buffer.h"
> #include "modules/audio_device/include/audio_device.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/criticalsection.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_unittest.cc
>index a82bc4a33ba3a7dbb46fc91b33bfee7b9565719d..1aa28e5d7c1230a3706a7976f0c6819117a99e28 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_unittest.cc
>@@ -509,7 +509,7 @@ class MockAudioTransport : public test::MockAudioTransport {
> class AudioDeviceTest
>     : public ::testing::TestWithParam<webrtc::AudioDeviceModule::AudioLayer> {
>  protected:
>-  AudioDeviceTest() : audio_layer_(GetParam()), event_(false, false) {
>+  AudioDeviceTest() : audio_layer_(GetParam()) {
> // TODO(webrtc:9778): Re-enable on THREAD_SANITIZER?
> #if !defined(ADDRESS_SANITIZER) && !defined(MEMORY_SANITIZER) && \
>     !defined(WEBRTC_DUMMY_AUDIO_BUILD) && !defined(THREAD_SANITIZER)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/audio_device_dummy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/audio_device_dummy.h
>index d709f776919fc7dc3a4ed893c9ce5b6fdd7c7458..2a2541098ea86869aee2dc31d9d2921d1f2fc81c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/audio_device_dummy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/audio_device_dummy.h
>@@ -11,9 +11,12 @@
> #ifndef AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H_
> #define AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H_
> 
>-#include <stdio.h>
>+#include <stdint.h>
> 
>+#include "modules/audio_device/audio_device_buffer.h"
> #include "modules/audio_device/audio_device_generic.h"
>+#include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.cc
>index b9aeaa15065a9887007ce5e222fd71129201faf4..2848eeae350895789b01c6153236dd6213d62a1b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.cc
>@@ -9,9 +9,13 @@
>  */
> 
> #include "modules/audio_device/dummy/file_audio_device.h"
>+
>+#include <string.h>
>+
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/platform_thread.h"
>+#include "rtc_base/timeutils.h"
> #include "system_wrappers/include/sleep.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.h
>index f07690e7e033790202bd904dda6d86f972bc292b..210c8f7b8be6026fb020a332f9fb6b1b06366265 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.h
>@@ -26,7 +26,6 @@ class PlatformThread;
> }  // namespace rtc
> 
> namespace webrtc {
>-class EventWrapper;
> 
> // This is a fake audio device which plays audio from a file as its microphone
> // and plays out into a file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc
>index 6b38d8bc055ae8530d1e5285baf705d22aa0b8fb..027b13bf5c5ce1bbb7ee76bfc1fa014e7ffaf54c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc
>@@ -10,8 +10,8 @@
> 
> #include "modules/audio_device/dummy/file_audio_device_factory.h"
> 
>+#include <stdio.h>
> #include <cstdlib>
>-#include <cstring>
> 
> #include "modules/audio_device/dummy/file_audio_device.h"
> #include "rtc_base/logging.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device_factory.h
>index 9cd5b3d62566f6b2db2870ab39a15f5c78f816f1..72f4ab2b384ed9407e167688c440f6df8a189fac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device_factory.h
>@@ -11,7 +11,7 @@
> #ifndef AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H_
> #define AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H_
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.cc
>index 4af344aef3786a30027307585b6171e9145d234e..b4f3c371ace369cdcfd45bb932616480830bd401 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.cc
>@@ -10,6 +10,9 @@
> 
> #include "modules/audio_device/fine_audio_buffer.h"
> 
>+#include <cstdint>
>+#include <cstring>
>+
> #include "modules/audio_device/audio_device_buffer.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_data_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_data_observer.h
>index 3a4ee99518aa926b1cc215f0693b330fc9761bb1..959cfa4662c696b926fff17c6945c15387b1e036 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_data_observer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_data_observer.h
>@@ -11,6 +11,9 @@
> #ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DATA_OBSERVER_H_
> #define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DATA_OBSERVER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "modules/audio_device/include/audio_device.h"
> #include "rtc_base/scoped_ref_ptr.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.cc
>index 7d55c47529a296f98d38e9fee9ac0882a1943c6c..8aea1d245e226be1d77a36df3bf35502e716c36e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.cc
>@@ -8,8 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> #include <algorithm>
>+#include <cstdint>
>+#include <cstdlib>
> #include <memory>
> #include <string>
>+#include <type_traits>
> #include <utility>
> #include <vector>
> 
>@@ -28,6 +31,7 @@
> #include "rtc_base/random.h"
> #include "rtc_base/refcountedobject.h"
> #include "rtc_base/thread.h"
>+#include "rtc_base/thread_annotations.h"
> #include "rtc_base/timeutils.h"
> 
> namespace webrtc {
>@@ -58,8 +62,6 @@ class TestAudioDeviceModuleImpl
>         audio_callback_(nullptr),
>         rendering_(false),
>         capturing_(false),
>-        done_rendering_(true, true),
>-        done_capturing_(true, true),
>         stop_thread_(false) {
>     auto good_sample_rate = [](int sr) {
>       return sr == 8000 || sr == 16000 || sr == 32000 || sr == 44100 ||
>@@ -108,14 +110,12 @@ class TestAudioDeviceModuleImpl
>     rtc::CritScope cs(&lock_);
>     RTC_CHECK(renderer_);
>     rendering_ = true;
>-    done_rendering_.Reset();
>     return 0;
>   }
> 
>   int32_t StopPlayout() {
>     rtc::CritScope cs(&lock_);
>     rendering_ = false;
>-    done_rendering_.Set();
>     return 0;
>   }
> 
>@@ -123,14 +123,12 @@ class TestAudioDeviceModuleImpl
>     rtc::CritScope cs(&lock_);
>     RTC_CHECK(capturer_);
>     capturing_ = true;
>-    done_capturing_.Reset();
>     return 0;
>   }
> 
>   int32_t StopRecording() {
>     rtc::CritScope cs(&lock_);
>     capturing_ = false;
>-    done_capturing_.Set();
>     return 0;
>   }
> 
>@@ -172,9 +170,10 @@ class TestAudioDeviceModuleImpl
>           uint32_t new_mic_level = 0;
>           if (recording_buffer_.size() > 0) {
>             audio_callback_->RecordedDataIsAvailable(
>-                recording_buffer_.data(), recording_buffer_.size(), 2,
>-                capturer_->NumChannels(), capturer_->SamplingFrequency(), 0, 0,
>-                0, false, new_mic_level);
>+                recording_buffer_.data(),
>+                recording_buffer_.size() / capturer_->NumChannels(),
>+                2 * capturer_->NumChannels(), capturer_->NumChannels(),
>+                capturer_->SamplingFrequency(), 0, 0, 0, false, new_mic_level);
>           }
>           if (!keep_capturing) {
>             capturing_ = false;
>@@ -187,9 +186,10 @@ class TestAudioDeviceModuleImpl
>           int64_t ntp_time_ms = -1;
>           const int sampling_frequency = renderer_->SamplingFrequency();
>           audio_callback_->NeedMorePlayData(
>-              SamplesPerFrame(sampling_frequency), 2, renderer_->NumChannels(),
>-              sampling_frequency, playout_buffer_.data(), samples_out,
>-              &elapsed_time_ms, &ntp_time_ms);
>+              SamplesPerFrame(sampling_frequency), 2 * renderer_->NumChannels(),
>+              renderer_->NumChannels(), sampling_frequency,
>+              playout_buffer_.data(), samples_out, &elapsed_time_ms,
>+              &ntp_time_ms);
>           const bool keep_rendering =
>               renderer_->Render(rtc::ArrayView<const int16_t>(
>                   playout_buffer_.data(), samples_out));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.h
>index 7d034dce8c70ecdb629c2c652b8e6b214bda6a4a..93f0b13601fb790f42548b541f3adbd403eaba4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.h
>@@ -10,14 +10,18 @@
> #ifndef MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
> #define MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> #include <string>
>-#include <vector>
> 
>+#include "api/array_view.h"
> #include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/event.h"
> #include "rtc_base/platform_file.h"
>+#include "rtc_base/scoped_ref_ptr.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
>index 47b873d4a99b71a0ed6b66116e9ceb89dd43a2d2..19ff9c9233236a8ae502aeb0240119358d3c2a4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
>@@ -22,11 +22,11 @@
> #include "modules/audio_device/ios/audio_device_ios.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/format_macros.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/timeutils.h"
>-#include "system_wrappers/include/event_wrapper.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>@@ -372,7 +372,7 @@ class MockAudioTransportIOS : public test::MockAudioTransport {
> 
>   // Set default actions of the mock object. We are delegating to fake
>   // implementations (of AudioStreamInterface) here.
>-  void HandleCallbacks(EventWrapper* test_is_done,
>+  void HandleCallbacks(rtc::Event* test_is_done,
>                        AudioStreamInterface* audio_stream,
>                        size_t num_callbacks) {
>     test_is_done_ = test_is_done;
>@@ -461,7 +461,7 @@ class MockAudioTransportIOS : public test::MockAudioTransport {
>   bool rec_mode() const { return type_ & kRecording; }
> 
>  private:
>-  EventWrapper* test_is_done_;
>+  rtc::Event* test_is_done_;
>   size_t num_callbacks_;
>   int type_;
>   size_t play_count_;
>@@ -472,7 +472,7 @@ class MockAudioTransportIOS : public test::MockAudioTransport {
> // AudioDeviceTest test fixture.
> class AudioDeviceTest : public ::testing::Test {
>  protected:
>-  AudioDeviceTest() : test_is_done_(EventWrapper::Create()) {
>+  AudioDeviceTest() {
>     old_sev_ = rtc::LogMessage::GetLogToDebug();
>     // Set suitable logging level here. Change to rtc::LS_INFO for more verbose
>     // output. See webrtc/rtc_base/logging.h for complete list of options.
>@@ -572,7 +572,7 @@ class AudioDeviceTest : public ::testing::Test {
>     EXPECT_FALSE(audio_device()->Recording());
>   }
> 
>-  std::unique_ptr<EventWrapper> test_is_done_;
>+  rtc::Event test_is_done_;
>   rtc::scoped_refptr<AudioDeviceModule> audio_device_;
>   AudioParameters playout_parameters_;
>   AudioParameters record_parameters_;
>@@ -662,7 +662,7 @@ TEST_F(AudioDeviceTest, DISABLED_StartPlayoutOnTwoInstances) {
>   // has been done successfully and that there is no conflict with the already
>   // playing first ADM.
>   MockAudioTransportIOS mock2(kPlayout);
>-  mock2.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  mock2.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
>   EXPECT_CALL(
>       mock2, NeedMorePlayData(playout_frames_per_10ms_buffer(), kBytesPerSample,
>                               playout_channels(), playout_sample_rate(),
>@@ -671,7 +671,7 @@ TEST_F(AudioDeviceTest, DISABLED_StartPlayoutOnTwoInstances) {
>   EXPECT_EQ(0, second_audio_device->RegisterAudioCallback(&mock2));
>   EXPECT_EQ(0, second_audio_device->StartPlayout());
>   EXPECT_TRUE(second_audio_device->Playing());
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   EXPECT_EQ(0, second_audio_device->StopPlayout());
>   EXPECT_FALSE(second_audio_device->Playing());
>   EXPECT_FALSE(second_audio_device->PlayoutIsInitialized());
>@@ -683,14 +683,14 @@ TEST_F(AudioDeviceTest, DISABLED_StartPlayoutOnTwoInstances) {
> // audio samples to play out using the NeedMorePlayData callback.
> TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
>   MockAudioTransportIOS mock(kPlayout);
>-  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
>   EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
>                                      kBytesPerSample, playout_channels(),
>                                      playout_sample_rate(), NotNull(), _, _, _))
>       .Times(AtLeast(kNumCallbacks));
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartPlayout();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopPlayout();
> }
> 
>@@ -698,7 +698,7 @@ TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
> // audio samples via the RecordedDataIsAvailable callback.
> TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
>   MockAudioTransportIOS mock(kRecording);
>-  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
>   EXPECT_CALL(mock,
>               RecordedDataIsAvailable(
>                   NotNull(), record_frames_per_10ms_buffer(), kBytesPerSample,
>@@ -708,7 +708,7 @@ TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
> 
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartRecording();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopRecording();
> }
> 
>@@ -716,7 +716,7 @@ TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
> // active in both directions.
> TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
>   MockAudioTransportIOS mock(kPlayout | kRecording);
>-  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
>   EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
>                                      kBytesPerSample, playout_channels(),
>                                      playout_sample_rate(), NotNull(), _, _, _))
>@@ -730,7 +730,7 @@ TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartPlayout();
>   StartRecording();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopRecording();
>   StopPlayout();
> }
>@@ -746,12 +746,11 @@ TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
>   std::string file_name = GetFileName(playout_sample_rate());
>   std::unique_ptr<FileAudioStream> file_audio_stream(
>       new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
>-  mock.HandleCallbacks(test_is_done_.get(), file_audio_stream.get(),
>-                       num_callbacks);
>+  mock.HandleCallbacks(&test_is_done_, file_audio_stream.get(), num_callbacks);
>   // SetMaxPlayoutVolume();
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartPlayout();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopPlayout();
> }
> 
>@@ -780,14 +779,13 @@ TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
>   NiceMock<MockAudioTransportIOS> mock(kPlayout | kRecording);
>   std::unique_ptr<FifoAudioStream> fifo_audio_stream(
>       new FifoAudioStream(playout_frames_per_10ms_buffer()));
>-  mock.HandleCallbacks(test_is_done_.get(), fifo_audio_stream.get(),
>-                       kFullDuplexTimeInSec * kNumCallbacksPerSecond);
>+  mock.HandleCallbacks(
>+      &test_is_done_, fifo_audio_stream.get(), kFullDuplexTimeInSec * kNumCallbacksPerSecond);
>   // SetMaxPlayoutVolume();
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartRecording();
>   StartPlayout();
>-  test_is_done_->Wait(
>-      std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
>+  test_is_done_.Wait(std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
>   StopPlayout();
>   StopRecording();
>   EXPECT_LE(fifo_audio_stream->average_size(), 10u);
>@@ -809,15 +807,15 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
>   NiceMock<MockAudioTransportIOS> mock(kPlayout | kRecording);
>   std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
>       new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
>-  mock.HandleCallbacks(test_is_done_.get(), latency_audio_stream.get(),
>+  mock.HandleCallbacks(&test_is_done_,
>+                       latency_audio_stream.get(),
>                        kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   // SetMaxPlayoutVolume();
>   // DisableBuiltInAECIfAvailable();
>   StartRecording();
>   StartPlayout();
>-  test_is_done_->Wait(
>-      std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
>+  test_is_done_.Wait(std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
>   StopPlayout();
>   StopRecording();
>   // Verify that the correct number of transmitted impulses are detected.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
>index 3d15565b8dc481400ac1a4e02ec6078e8c109887..d60dcafdced16cbca7ed3923f342cf2c13f7b416 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
>@@ -29,7 +29,6 @@ typedef webrtc::adm_linux_alsa::AlsaSymbolTable WebRTCAlsaSymbolTable;
> WebRTCAlsaSymbolTable* GetAlsaSymbolTable();
> 
> namespace webrtc {
>-class EventWrapper;
> 
> class AudioDeviceLinuxALSA : public AudioDeviceGeneric {
>  public:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
>index 2157af328c6d3b237420aaef759d2cec87fcb2c3..3706d9176ecd80fe549683313a071c06eb383b9d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
>@@ -8,11 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/audio_device/audio_device_config.h"
>+#include <string.h>
>+
> #include "modules/audio_device/linux/audio_device_pulse_linux.h"
>+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "system_wrappers/include/event_wrapper.h"
> 
> WebRTCPulseSymbolTable* GetPulseSymbolTable() {
>   static WebRTCPulseSymbolTable* pulse_symbol_table =
>@@ -31,10 +32,6 @@ namespace webrtc {
> 
> AudioDeviceLinuxPulse::AudioDeviceLinuxPulse()
>     : _ptrAudioBuffer(NULL),
>-      _timeEventRec(*EventWrapper::Create()),
>-      _timeEventPlay(*EventWrapper::Create()),
>-      _recStartEvent(*EventWrapper::Create()),
>-      _playStartEvent(*EventWrapper::Create()),
>       _inputDeviceIndex(0),
>       _outputDeviceIndex(0),
>       _inputDeviceIsSpecified(false),
>@@ -111,11 +108,6 @@ AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse() {
>     delete[] _recDeviceName;
>     _recDeviceName = NULL;
>   }
>-
>-  delete &_recStartEvent;
>-  delete &_playStartEvent;
>-  delete &_timeEventRec;
>-  delete &_timeEventPlay;
> }
> 
> void AudioDeviceLinuxPulse::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
>@@ -1065,7 +1057,7 @@ int32_t AudioDeviceLinuxPulse::StartRecording() {
> 
>   // The audio thread will signal when recording has started.
>   _timeEventRec.Set();
>-  if (kEventTimeout == _recStartEvent.Wait(10000)) {
>+  if (!_recStartEvent.Wait(10000)) {
>     {
>       rtc::CritScope lock(&_critSect);
>       _startRec = false;
>@@ -1180,7 +1172,7 @@ int32_t AudioDeviceLinuxPulse::StartPlayout() {
> 
>   // The audio thread will signal when playout has started.
>   _timeEventPlay.Set();
>-  if (kEventTimeout == _playStartEvent.Wait(10000)) {
>+  if (!_playStartEvent.Wait(10000)) {
>     {
>       rtc::CritScope lock(&_critSect);
>       _startPlay = false;
>@@ -1994,14 +1986,8 @@ bool AudioDeviceLinuxPulse::RecThreadFunc(void* pThis) {
> }
> 
> bool AudioDeviceLinuxPulse::PlayThreadProcess() {
>-  switch (_timeEventPlay.Wait(1000)) {
>-    case kEventSignaled:
>-      break;
>-    case kEventError:
>-      RTC_LOG(LS_WARNING) << "EventWrapper::Wait() failed";
>-      return true;
>-    case kEventTimeout:
>-      return true;
>+  if (!_timeEventPlay.Wait(1000)) {
>+    return true;
>   }
> 
>   rtc::CritScope lock(&_critSect);
>@@ -2168,14 +2154,8 @@ bool AudioDeviceLinuxPulse::PlayThreadProcess() {
> }
> 
> bool AudioDeviceLinuxPulse::RecThreadProcess() {
>-  switch (_timeEventRec.Wait(1000)) {
>-    case kEventSignaled:
>-      break;
>-    case kEventError:
>-      RTC_LOG(LS_WARNING) << "EventWrapper::Wait() failed";
>-      return true;
>-    case kEventTimeout:
>-      return true;
>+  if (!_timeEventRec.Wait(1000)) {
>+    return true;
>   }
> 
>   rtc::CritScope lock(&_critSect);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
>index 1655b18117fb24e31738eb9cb246a82c72b6d611..4de87fc3e6e3e4ed6e2154ecc197209c7728807f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
>@@ -13,10 +13,16 @@
> 
> #include <memory>
> 
>+#include "modules/audio_device/audio_device_buffer.h"
> #include "modules/audio_device/audio_device_generic.h"
>+#include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
> #include "modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
>+#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/platform_thread.h"
>+#include "rtc_base/thread_annotations.h"
> #include "rtc_base/thread_checker.h"
> 
> #if defined(WEBRTC_USE_X11)
>@@ -24,6 +30,8 @@
> #endif
> 
> #include <pulse/pulseaudio.h>
>+#include <stddef.h>
>+#include <stdint.h>
> 
> // We define this flag if it's missing from our headers, because we want to be
> // able to compile against old headers but still use PA_STREAM_ADJUST_LATENCY
>@@ -96,7 +104,6 @@ typedef webrtc::adm_linux_pulse::PulseAudioSymbolTable WebRTCPulseSymbolTable;
> WebRTCPulseSymbolTable* GetPulseSymbolTable();
> 
> namespace webrtc {
>-class EventWrapper;
> 
> class AudioDeviceLinuxPulse : public AudioDeviceGeneric {
>  public:
>@@ -255,10 +262,10 @@ class AudioDeviceLinuxPulse : public AudioDeviceGeneric {
>   AudioDeviceBuffer* _ptrAudioBuffer;
> 
>   rtc::CriticalSection _critSect;
>-  EventWrapper& _timeEventRec;
>-  EventWrapper& _timeEventPlay;
>-  EventWrapper& _recStartEvent;
>-  EventWrapper& _playStartEvent;
>+  rtc::Event _timeEventRec;
>+  rtc::Event _timeEventPlay;
>+  rtc::Event _recStartEvent;
>+  rtc::Event _playStartEvent;
> 
>   // TODO(pbos): Remove unique_ptr and use directly without resetting.
>   std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
>index 5ba923bb911c1d222449e51c8060d7f874188844..22e10407de2ff13f0aa6306462e6a4e97028ee58 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
>@@ -8,8 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <stddef.h>
>+
> #include "modules/audio_device/linux/audio_device_pulse_linux.h"
> #include "modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
>+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
>+#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
>index c5460b6237af872252307b2a7fc148180fc6203e..679d1b4b7c26d236e52306d8ad9d1afd85afa927 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
>@@ -11,13 +11,11 @@
> #ifndef AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
> #define AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
> 
>-#include "modules/audio_device/include/audio_device.h"
>-#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
>-#include "rtc_base/thread_checker.h"
>-
> #include <pulse/pulseaudio.h>
> #include <stdint.h>
> 
>+#include "rtc_base/thread_checker.h"
>+
> #ifndef UINT32_MAX
> #define UINT32_MAX ((uint32_t)-1)
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc
>index 063ac85736ddc4d1015f2186351ff02d2797f3fd..9daf7bddf7bf73c59de28fe353564c1b8f335f23 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc
>@@ -16,7 +16,6 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/platform_thread.h"
> #include "rtc_base/system/arch.h"
>-#include "system_wrappers/include/event_wrapper.h"
> 
> #include <ApplicationServices/ApplicationServices.h>
> #include <libkern/OSAtomic.h>  // OSAtomicCompareAndSwap()
>@@ -115,8 +114,6 @@ void AudioDeviceMac::logCAMsg(const rtc::LoggingSeverity sev,
> 
> AudioDeviceMac::AudioDeviceMac()
>     : _ptrAudioBuffer(NULL),
>-      _stopEventRec(*EventWrapper::Create()),
>-      _stopEvent(*EventWrapper::Create()),
>       _mixerManager(),
>       _inputDeviceIndex(0),
>       _outputDeviceIndex(0),
>@@ -153,9 +150,6 @@ AudioDeviceMac::AudioDeviceMac()
>       prev_key_state_() {
>   RTC_LOG(LS_INFO) << __FUNCTION__ << " created";
> 
>-  RTC_DCHECK(&_stopEvent != NULL);
>-  RTC_DCHECK(&_stopEventRec != NULL);
>-
>   memset(_renderConvertData, 0, sizeof(_renderConvertData));
>   memset(&_outStreamFormat, 0, sizeof(AudioStreamBasicDescription));
>   memset(&_outDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
>@@ -204,8 +198,6 @@ AudioDeviceMac::~AudioDeviceMac() {
>     RTC_LOG(LS_ERROR) << "semaphore_destroy() error: " << kernErr;
>   }
> 
>-  delete &_stopEvent;
>-  delete &_stopEventRec;
> }
> 
> // ============================================================================
>@@ -1338,7 +1330,7 @@ int32_t AudioDeviceMac::StopRecording() {
>       _recording = false;
>       _doStopRec = true;  // Signal to io proc to stop audio device
>       _critSect.Leave();  // Cannot be under lock, risk of deadlock
>-      if (kEventTimeout == _stopEventRec.Wait(2000)) {
>+      if (!_stopEventRec.Wait(2000)) {
>         rtc::CritScope critScoped(&_critSect);
>         RTC_LOG(LS_WARNING) << "Timed out stopping the capture IOProc."
>                             << "We may have failed to detect a device removal.";
>@@ -1366,7 +1358,7 @@ int32_t AudioDeviceMac::StopRecording() {
>       _recording = false;
>       _doStop = true;     // Signal to io proc to stop audio device
>       _critSect.Leave();  // Cannot be under lock, risk of deadlock
>-      if (kEventTimeout == _stopEvent.Wait(2000)) {
>+      if (!_stopEvent.Wait(2000)) {
>         rtc::CritScope critScoped(&_critSect);
>         RTC_LOG(LS_WARNING) << "Timed out stopping the shared IOProc."
>                             << "We may have failed to detect a device removal.";
>@@ -1474,7 +1466,7 @@ int32_t AudioDeviceMac::StopPlayout() {
>     _playing = false;
>     _doStop = true;     // Signal to io proc to stop audio device
>     _critSect.Leave();  // Cannot be under lock, risk of deadlock
>-    if (kEventTimeout == _stopEvent.Wait(2000)) {
>+    if (!_stopEvent.Wait(2000)) {
>       rtc::CritScope critScoped(&_critSect);
>       RTC_LOG(LS_WARNING) << "Timed out stopping the render IOProc."
>                           << "We may have failed to detect a device removal.";
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.h
>index 5cbdae28371a2930e0735a7423d45b66ea4cc135..5a7076065334cf754d756a04eb8170ea6bc8837c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.h
>@@ -16,9 +16,9 @@
> #include "modules/audio_device/audio_device_generic.h"
> #include "modules/audio_device/mac/audio_mixer_manager_mac.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/thread_annotations.h"
>-#include "system_wrappers/include/event_wrapper.h"
> 
> #include <AudioToolbox/AudioConverter.h>
> #include <CoreAudio/CoreAudio.h>
>@@ -31,7 +31,6 @@ class PlatformThread;
> }  // namespace rtc
> 
> namespace webrtc {
>-class EventWrapper;
> 
> const uint32_t N_REC_SAMPLES_PER_SEC = 48000;
> const uint32_t N_PLAY_SAMPLES_PER_SEC = 48000;
>@@ -252,8 +251,8 @@ class AudioDeviceMac : public AudioDeviceGeneric {
> 
>   rtc::CriticalSection _critSect;
> 
>-  EventWrapper& _stopEventRec;
>-  EventWrapper& _stopEvent;
>+  rtc::Event _stopEventRec;
>+  rtc::Event _stopEvent;
> 
>   // TODO(pbos): Replace with direct members, just start/stop, no need to
>   // recreate the thread.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/BUILD.gn
>index 0bb3ae137aeedddfd99c740f113f09b0a451c9f2..f935e83c5ba23f9efedae5542213b7646b8b5f83 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/BUILD.gn
>@@ -37,7 +37,6 @@ rtc_static_library("audio_mixer_impl") {
> 
>   deps = [
>     ":audio_frame_manipulator",
>-    "../..:webrtc_common",
>     "../../api:array_view",
>     "../../api/audio:audio_frame_api",
>     "../../api/audio:audio_mixer_api",
>@@ -48,6 +47,7 @@ rtc_static_library("audio_mixer_impl") {
>     "../../system_wrappers",
>     "../../system_wrappers:metrics",
>     "../audio_processing",
>+    "../audio_processing:api",
>     "../audio_processing:apm_logging",
>     "../audio_processing:audio_frame_view",
>     "../audio_processing/agc2:fixed_digital",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl.h
>index e8994a674bc605d856497386e1df4d58ad84014b..8edd3b8bdc4d3e4b60036175a845aeee66890721 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl.h
>@@ -17,6 +17,7 @@
> #include "api/audio/audio_mixer.h"
> #include "modules/audio_mixer/frame_combiner.h"
> #include "modules/audio_mixer/output_rate_calculator.h"
>+#include "rtc_base/criticalsection.h"
> #include "rtc_base/race_checker.h"
> #include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/thread_annotations.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_test.cc
>index 2004aeef463ab0ab902130f732406b99c0ecdc7b..b60787755d836e0282d0bad4d173f5f785281054 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_test.cc
>@@ -19,24 +19,25 @@
> #include "rtc_base/flags.h"
> #include "rtc_base/strings/string_builder.h"
> 
>-DEFINE_bool(help, false, "Prints this message");
>-DEFINE_int(sampling_rate,
>-           16000,
>-           "Rate at which to mix (all input streams must have this rate)");
>+WEBRTC_DEFINE_bool(help, false, "Prints this message");
>+WEBRTC_DEFINE_int(
>+    sampling_rate,
>+    16000,
>+    "Rate at which to mix (all input streams must have this rate)");
> 
>-DEFINE_bool(
>+WEBRTC_DEFINE_bool(
>     stereo,
>     false,
>     "Enable stereo (interleaved). Inputs need not be as this parameter.");
> 
>-DEFINE_bool(limiter, true, "Enable limiter.");
>-DEFINE_string(output_file,
>-              "mixed_file.wav",
>-              "File in which to store the mixed result.");
>-DEFINE_string(input_file_1, "", "First input. Default none.");
>-DEFINE_string(input_file_2, "", "Second input. Default none.");
>-DEFINE_string(input_file_3, "", "Third input. Default none.");
>-DEFINE_string(input_file_4, "", "Fourth input. Default none.");
>+WEBRTC_DEFINE_bool(limiter, true, "Enable limiter.");
>+WEBRTC_DEFINE_string(output_file,
>+                     "mixed_file.wav",
>+                     "File in which to store the mixed result.");
>+WEBRTC_DEFINE_string(input_file_1, "", "First input. Default none.");
>+WEBRTC_DEFINE_string(input_file_2, "", "Second input. Default none.");
>+WEBRTC_DEFINE_string(input_file_3, "", "Third input. Default none.");
>+WEBRTC_DEFINE_string(input_file_4, "", "Fourth input. Default none.");
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.cc
>index cff2282e33fe698241da9aa3b204c299128ed2fb..d3493f5c3dd437fdf4432669451267f7d8fbcf5a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.cc
>@@ -92,10 +92,10 @@ std::array<OneChannelBuffer, kMaximumAmountOfChannels> MixToFloatFrame(
>   return mixing_buffer;
> }
> 
>-void RunLimiter(AudioFrameView<float> mixing_buffer_view,
>-                FixedGainController* limiter) {
>+void RunLimiter(AudioFrameView<float> mixing_buffer_view, Limiter* limiter) {
>   const size_t sample_rate = mixing_buffer_view.samples_per_channel() * 1000 /
>                              AudioMixerImpl::kFrameDurationInMs;
>+  // TODO(alessiob): Avoid calling SetSampleRate every time.
>   limiter->SetSampleRate(sample_rate);
>   limiter->Process(mixing_buffer_view);
> }
>@@ -117,10 +117,8 @@ void InterleaveToAudioFrame(AudioFrameView<const float> mixing_buffer_view,
> 
> FrameCombiner::FrameCombiner(bool use_limiter)
>     : data_dumper_(new ApmDataDumper(0)),
>-      limiter_(data_dumper_.get(), "AudioMixer"),
>-      use_limiter_(use_limiter) {
>-  limiter_.SetGain(0.f);
>-}
>+      limiter_(static_cast<size_t>(48000), data_dumper_.get(), "AudioMixer"),
>+      use_limiter_(use_limiter) {}
> 
> FrameCombiner::~FrameCombiner() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.h
>index 70ac02775899070a40c752710f57f066f1d735a5..1c1cd53eff3115ee12ec186fefeb037fdc054e06 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.h
>@@ -15,11 +15,10 @@
> #include <vector>
> 
> #include "api/audio/audio_frame.h"
>-#include "modules/audio_processing/agc2/fixed_gain_controller.h"
>+#include "modules/audio_processing/agc2/limiter.h"
> 
> namespace webrtc {
> class ApmDataDumper;
>-class FixedGainController;
> 
> class FrameCombiner {
>  public:
>@@ -45,7 +44,7 @@ class FrameCombiner {
>                       size_t number_of_streams) const;
> 
>   std::unique_ptr<ApmDataDumper> data_dumper_;
>-  FixedGainController limiter_;
>+  Limiter limiter_;
>   const bool use_limiter_;
>   mutable int uma_logging_counter_ = 0;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/BUILD.gn
>index 61b4d495ed2f9161823cc7c53e38ad5d31468ea1..66b07df199e67dacf9245742beda30d144297847 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/BUILD.gn
>@@ -26,6 +26,41 @@ config("apm_debug_dump") {
>   }
> }
> 
>+rtc_static_library("config") {
>+  visibility = [ ":*" ]
>+  sources = [
>+    "include/config.cc",
>+    "include/config.h",
>+  ]
>+  deps = [
>+    "../../rtc_base:macromagic",
>+    "../../rtc_base/system:rtc_export",
>+  ]
>+}
>+
>+rtc_source_set("api") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "include/audio_processing.cc",
>+    "include/audio_processing.h",
>+  ]
>+  deps = [
>+    ":audio_frame_view",
>+    ":audio_generator_interface",
>+    ":audio_processing_statistics",
>+    ":config",
>+    ":gain_control_interface",
>+    "../../api/audio:aec3_config",
>+    "../../api/audio:echo_control",
>+    "../../rtc_base:deprecation",
>+    "../../rtc_base:macromagic",
>+    "../../rtc_base:ptr_util",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base/system:rtc_export",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
> rtc_static_library("audio_processing") {
>   visibility = [ "*" ]
>   configs += [ ":apm_debug_dump" ]
>@@ -55,10 +90,6 @@ rtc_static_library("audio_processing") {
>     "gain_controller2.h",
>     "include/aec_dump.cc",
>     "include/aec_dump.h",
>-    "include/audio_processing.cc",
>-    "include/audio_processing.h",
>-    "include/config.cc",
>-    "include/config.h",
>     "level_estimator_impl.cc",
>     "level_estimator_impl.h",
>     "low_cut_filter.cc",
>@@ -95,11 +126,13 @@ rtc_static_library("audio_processing") {
> 
>   defines = []
>   deps = [
>+    ":api",
>     ":apm_logging",
>     ":audio_frame_view",
>     ":audio_generator_interface",
>     ":audio_processing_c",
>     ":audio_processing_statistics",
>+    ":config",
>     ":gain_control_interface",
>     "../..:webrtc_common",
>     "../../api:array_view",
>@@ -115,6 +148,7 @@ rtc_static_library("audio_processing") {
>     "../../rtc_base:safe_minmax",
>     "../../rtc_base:sanitizer",
>     "../../rtc_base/system:arch",
>+    "../../rtc_base/system:rtc_export",
>     "../../system_wrappers:cpu_features_api",
>     "../../system_wrappers:field_trial",
>     "../../system_wrappers:metrics",
>@@ -162,6 +196,7 @@ rtc_source_set("audio_processing_statistics") {
>     "include/audio_processing_statistics.h",
>   ]
>   deps = [
>+    "../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -294,6 +329,7 @@ rtc_source_set("apm_logging") {
>   deps = [
>     "../../api:array_view",
>     "../../common_audio:common_audio",
>+    "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:stringutils",
>   ]
>@@ -307,6 +343,7 @@ if (rtc_include_tests) {
>       "include/mock_audio_processing.h",
>     ]
>     deps = [
>+      ":api",
>       ":audio_processing",
>       ":audio_processing_statistics",
>       "../../test:test_support",
>@@ -358,10 +395,12 @@ if (rtc_include_tests) {
> 
>     deps = [
>       ":analog_mic_simulation",
>+      ":api",
>       ":apm_logging",
>       ":audio_frame_view",
>       ":audio_processing",
>       ":audioproc_test_utils",
>+      ":config",
>       ":file_audio_generator_unittests",
>       ":mocks",
>       "../..:webrtc_common",
>@@ -375,6 +414,7 @@ if (rtc_include_tests) {
>       "../../rtc_base:protobuf_utils",
>       "../../rtc_base:rtc_base",
>       "../../rtc_base:rtc_base_approved",
>+      "../../rtc_base:rtc_base_tests_utils",
>       "../../rtc_base:safe_minmax",
>       "../../rtc_base/system:arch",
>       "../../rtc_base/system:file_wrapper",
>@@ -391,6 +431,8 @@ if (rtc_include_tests) {
>       "agc2:biquad_filter_unittests",
>       "agc2:fixed_digital_unittests",
>       "agc2:noise_estimator_unittests",
>+      "agc2:rnn_vad_with_level_unittests",
>+      "agc2:test_utils",
>       "agc2/rnn_vad:unittests",
>       "test/conversational_speech:unittest",
>       "utility:block_mean_calculator_unittest",
>@@ -418,7 +460,6 @@ if (rtc_include_tests) {
>         ":audioproc_unittest_proto",
>         ":runtime_settings_protobuf_utils",
>         "../../api/audio:audio_frame_api",
>-        "../../rtc_base:rtc_base_tests_utils",
>         "../../rtc_base:rtc_task_queue",
>         "aec_dump",
>         "aec_dump:aec_dump_unittests",
>@@ -482,6 +523,7 @@ if (rtc_include_tests) {
>     ]
> 
>     deps = [
>+      ":api",
>       ":audio_generator_factory",
>       ":audio_processing",
>       ":file_audio_generator",
>@@ -512,6 +554,7 @@ if (rtc_include_tests) {
>   if (rtc_enable_protobuf) {
>     rtc_source_set("audioproc_f_impl") {
>       testonly = true
>+      configs += [ ":apm_debug_dump" ]
>       sources = [
>         "test/aec_dump_based_simulator.cc",
>         "test/aec_dump_based_simulator.h",
>@@ -525,11 +568,14 @@ if (rtc_include_tests) {
> 
>       deps = [
>         ":analog_mic_simulation",
>+        ":api",
>+        ":apm_logging",
>         ":audio_processing",
>         ":audioproc_debug_proto",
>         ":audioproc_protobuf_utils",
>         ":audioproc_test_utils",
>         ":runtime_settings_protobuf_utils",
>+        "../../api/audio:aec3_config_json",
>         "../../api/audio:aec3_factory",
>         "../../common_audio:common_audio",
>         "../../rtc_base:checks",
>@@ -544,6 +590,7 @@ if (rtc_include_tests) {
>         "aec_dump:aec_dump_impl",
>         "//testing/gtest",
>         "//third_party/abseil-cpp/absl/memory",
>+        "//third_party/abseil-cpp/absl/strings",
>         "//third_party/abseil-cpp/absl/types:optional",
>       ]
>     }  # audioproc_f_impl
>@@ -553,6 +600,7 @@ if (rtc_include_tests) {
>         "test/audioproc_float_main.cc",
>       ]
>       deps = [
>+        ":api",
>         ":audio_processing",
>         "../../api:audioproc_f_api",
>         "../../rtc_base:rtc_base_approved",
>@@ -578,6 +626,7 @@ if (rtc_include_tests) {
>     ]
> 
>     deps = [
>+      ":api",
>       ":audio_processing",
>       "../../api:array_view",
>       "../../api/audio:audio_frame_api",
>@@ -662,6 +711,7 @@ if (rtc_include_tests) {
>       ]
> 
>       deps = [
>+        ":api",
>         ":audio_processing",
>         ":audioproc_debug_proto",
>         ":audioproc_protobuf_utils",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.cc
>index 5a8cf8f58b601f1ab25e499e32a7d1a37ea7b87f..62b8ad0f7c0709e5c34b7f646385c007b9a089cb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.cc
>@@ -21,6 +21,7 @@
> #include <algorithm>
> 
> #include "rtc_base/checks.h"
>+
> extern "C" {
> #include "common_audio/ring_buffer.h"
> }
>@@ -29,7 +30,6 @@ extern "C" {
> #include "modules/audio_processing/aec/aec_core_optimized_methods.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "modules/audio_processing/utility/delay_estimator_wrapper.h"
>-#include "rtc_base/checks.h"
> #include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "system_wrappers/include/metrics.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_resampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_resampler.cc
>index 2851c0b6dd575d6968b76669db25ee42005772f6..210c2bebe0e2113ad0e808d24f23b9bc808608ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_resampler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_resampler.cc
>@@ -14,7 +14,6 @@
> 
> #include "modules/audio_processing/aec/aec_resampler.h"
> 
>-#include <math.h>
> #include <stdlib.h>
> #include <string.h>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_resampler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_resampler.h
>index 130f7ec7c76647d1ce836fc232a41776feeaedeb..a112c434d0bc91ba1a9d56ebf1ae31b3618f0feb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_resampler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_resampler.h
>@@ -11,6 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
> #define MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
> 
>+#include <stddef.h>
>+
> #include "modules/audio_processing/aec/aec_core.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/BUILD.gn
>index e631732a4539ab1362f15b998bd5c11cef82732c..189bcfd71255339bc7dbf77f0f370cebb31f4e43 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/BUILD.gn
>@@ -20,16 +20,21 @@ rtc_static_library("aec3") {
>     "aec3_fft.h",
>     "aec_state.cc",
>     "aec_state.h",
>+    "api_call_jitter_metrics.cc",
>+    "api_call_jitter_metrics.h",
>     "block_delay_buffer.cc",
>     "block_delay_buffer.h",
>     "block_framer.cc",
>     "block_framer.h",
>     "block_processor.cc",
>     "block_processor.h",
>+    "block_processor2.cc",
>     "block_processor_metrics.cc",
>     "block_processor_metrics.h",
>     "cascaded_biquad_filter.cc",
>     "cascaded_biquad_filter.h",
>+    "clockdrift_detector.cc",
>+    "clockdrift_detector.h",
>     "comfort_noise_generator.cc",
>     "comfort_noise_generator.h",
>     "decimator.cc",
>@@ -76,10 +81,14 @@ rtc_static_library("aec3") {
>     "render_buffer.h",
>     "render_delay_buffer.cc",
>     "render_delay_buffer.h",
>+    "render_delay_buffer2.cc",
>     "render_delay_controller.cc",
>     "render_delay_controller.h",
>+    "render_delay_controller2.cc",
>     "render_delay_controller_metrics.cc",
>     "render_delay_controller_metrics.h",
>+    "render_reverb_model.cc",
>+    "render_reverb_model.h",
>     "render_signal_analyzer.cc",
>     "render_signal_analyzer.h",
>     "residual_echo_estimator.cc",
>@@ -96,6 +105,8 @@ rtc_static_library("aec3") {
>     "reverb_model_fallback.h",
>     "shadow_filter_update_gain.cc",
>     "shadow_filter_update_gain.h",
>+    "signal_dependent_erle_estimator.cc",
>+    "signal_dependent_erle_estimator.h",
>     "skew_estimator.cc",
>     "skew_estimator.h",
>     "stationarity_estimator.cc",
>@@ -183,11 +194,13 @@ if (rtc_include_tests) {
>         "adaptive_fir_filter_unittest.cc",
>         "aec3_fft_unittest.cc",
>         "aec_state_unittest.cc",
>+        "api_call_jitter_metrics_unittest.cc",
>         "block_delay_buffer_unittest.cc",
>         "block_framer_unittest.cc",
>         "block_processor_metrics_unittest.cc",
>         "block_processor_unittest.cc",
>         "cascaded_biquad_filter_unittest.cc",
>+        "clockdrift_detector_unittest.cc",
>         "comfort_noise_generator_unittest.cc",
>         "decimator_unittest.cc",
>         "echo_canceller3_unittest.cc",
>@@ -211,6 +224,7 @@ if (rtc_include_tests) {
>         "residual_echo_estimator_unittest.cc",
>         "reverb_model_estimator_unittest.cc",
>         "shadow_filter_update_gain_unittest.cc",
>+        "signal_dependent_erle_estimator_unittest.cc",
>         "skew_estimator_unittest.cc",
>         "subtractor_unittest.cc",
>         "suppression_filter_unittest.cc",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
>index 9a1e811fcfe3cf52ea9bb9868f91f927d8de7688..3ab1ebcc4b9debcd919eaabb2db3fd08c3a3d950 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
>@@ -24,7 +24,7 @@
> 
> #include "modules/audio_processing/aec3/fft_data.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
>@@ -417,12 +417,21 @@ void ApplyFilter_SSE2(const RenderBuffer& render_buffer,
> 
> }  // namespace aec3
> 
>+namespace {
>+
>+bool EnablePartialFilterReset() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3PartialFilterResetKillSwitch");
>+}
>+
>+}  // namespace
>+
> AdaptiveFirFilter::AdaptiveFirFilter(size_t max_size_partitions,
>                                      size_t initial_size_partitions,
>                                      size_t size_change_duration_blocks,
>                                      Aec3Optimization optimization,
>                                      ApmDataDumper* data_dumper)
>     : data_dumper_(data_dumper),
>+      use_partial_filter_reset_(EnablePartialFilterReset()),
>       fft_(),
>       optimization_(optimization),
>       max_size_partitions_(max_size_partitions),
>@@ -455,20 +464,22 @@ AdaptiveFirFilter::~AdaptiveFirFilter() = default;
> void AdaptiveFirFilter::HandleEchoPathChange() {
>   size_t current_h_size = h_.size();
>   h_.resize(GetTimeDomainLength(max_size_partitions_));
>-  std::fill(h_.begin(), h_.end(), 0.f);
>+  const size_t begin_coeffficient =
>+      use_partial_filter_reset_ ? current_h_size : 0;
>+  std::fill(h_.begin() + begin_coeffficient, h_.end(), 0.f);
>   h_.resize(current_h_size);
> 
>   size_t current_size_partitions = H_.size();
>   H_.resize(max_size_partitions_);
>-  for (auto& H_j : H_) {
>-    H_j.Clear();
>-  }
>-  H_.resize(current_size_partitions);
>-
>   H2_.resize(max_size_partitions_);
>-  for (auto& H2_k : H2_) {
>-    H2_k.fill(0.f);
>+
>+  const size_t begin_partition =
>+      use_partial_filter_reset_ ? current_size_partitions : 0;
>+  for (size_t k = begin_partition; k < max_size_partitions_; ++k) {
>+    H_[k].Clear();
>+    H2_[k].fill(0.f);
>   }
>+  H_.resize(current_size_partitions);
>   H2_.resize(current_size_partitions);
> 
>   erl_.fill(0.f);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.h
>index 5dfb4660fa21c0375cbba5e2c910549bfa0078b4..7c832a6cd23a405c46f8b985a5619e236080af48 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.h
>@@ -11,8 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_ADAPTIVE_FIR_FILTER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_ADAPTIVE_FIR_FILTER_H_
> 
>+#include <stddef.h>
> #include <array>
>-#include <memory>
> #include <vector>
> 
> #include "api/array_view.h"
>@@ -164,6 +164,7 @@ class AdaptiveFirFilter {
>   void UpdateSize();
> 
>   ApmDataDumper* const data_dumper_;
>+  const bool use_partial_filter_reset_;
>   const Aec3Fft fft_;
>   const Aec3Optimization optimization_;
>   const size_t max_size_partitions_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.cc
>index c374a3517d60e37a7991d4ee964b100ed7001d64..aeb848a5700a64820ddccd4f33ff93305b561f4f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.cc
>@@ -10,10 +10,11 @@
> 
> #include "modules/audio_processing/aec3/aec3_common.h"
> 
>-#include "system_wrappers/include/cpu_features_wrapper.h"
>+#include <stdint.h>
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/system/arch.h"
>+#include "system_wrappers/include/cpu_features_wrapper.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.cc
>index e29434d3002ddbf21443292bdfe19a7328fab10b..1832101855242030ae0a9be625e6bcdd1d639e42 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.cc
>@@ -12,6 +12,7 @@
> 
> #include <algorithm>
> #include <functional>
>+#include <iterator>
> 
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.h
>index b70022255d9564d332fdc2a46a91ad91aa3e7361..6cd649af8f4c96a421334377cba9556f44315ee2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.h
>@@ -17,6 +17,7 @@
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/fft_data.h"
> #include "modules/audio_processing/utility/ooura_fft.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.cc
>index 5e58a1c279c031abf629ff360934858e1e13485b..d5f256b1c1256e9fef27490d5080f292ebbd8aab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.cc
>@@ -11,7 +11,7 @@
> #include "modules/audio_processing/aec3/aec_state.h"
> 
> #include <math.h>
>-
>+#include <algorithm>
> #include <numeric>
> #include <vector>
> 
>@@ -30,6 +30,22 @@ bool EnableErleResetsAtGainChanges() {
>   return !field_trial::IsEnabled("WebRTC-Aec3ResetErleAtGainChangesKillSwitch");
> }
> 
>+bool UseLegacyFilterQualityState() {
>+  return field_trial::IsEnabled("WebRTC-Aec3FilterQualityStateKillSwitch");
>+}
>+
>+bool EnableLegacySaturationBehavior() {
>+  return field_trial::IsEnabled("WebRTC-Aec3NewSaturationBehaviorKillSwitch");
>+}
>+
>+bool UseSuppressionGainLimiter() {
>+  return field_trial::IsEnabled("WebRTC-Aec3GainLimiterDeactivationKillSwitch");
>+}
>+bool EnableErleUpdatesDuringReverb() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3EnableErleUpdatesDuringReverbKillSwitch");
>+}
>+
> constexpr size_t kBlocksSinceConvergencedFilterInit = 10000;
> constexpr size_t kBlocksSinceConsistentEstimateInit = 10000;
> 
>@@ -52,18 +68,10 @@ void AecState::GetResidualEchoScaling(
> }
> 
> absl::optional<float> AecState::ErleUncertainty() const {
>-  bool filter_has_had_time_to_converge;
>-  if (config_.filter.conservative_initial_phase) {
>-    filter_has_had_time_to_converge =
>-        strong_not_saturated_render_blocks_ >= 1.5f * kNumBlocksPerSecond;
>-  } else {
>-    filter_has_had_time_to_converge =
>-        strong_not_saturated_render_blocks_ >= 0.8f * kNumBlocksPerSecond;
>-  }
>-
>-  if (!filter_has_had_time_to_converge) {
>+  if (SaturatedEcho() && use_legacy_saturation_behavior_) {
>     return 1.f;
>   }
>+
>   return absl::nullopt;
> }
> 
>@@ -71,22 +79,24 @@ AecState::AecState(const EchoCanceller3Config& config)
>     : data_dumper_(
>           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>       config_(config),
>+      use_legacy_saturation_behavior_(EnableLegacySaturationBehavior()),
>+      enable_erle_resets_at_gain_changes_(EnableErleResetsAtGainChanges()),
>+      enable_erle_updates_during_reverb_(EnableErleUpdatesDuringReverb()),
>+      use_legacy_filter_quality_(UseLegacyFilterQualityState()),
>+      use_suppressor_gain_limiter_(UseSuppressionGainLimiter()),
>       initial_state_(config_),
>       delay_state_(config_),
>       transparent_state_(config_),
>       filter_quality_state_(config_),
>-      saturation_detector_(config_),
>+      legacy_filter_quality_state_(config_),
>+      legacy_saturation_detector_(config_),
>       erl_estimator_(2 * kNumBlocksPerSecond),
>-      erle_estimator_(2 * kNumBlocksPerSecond,
>-                      config_.erle.min,
>-                      config_.erle.max_l,
>-                      config_.erle.max_h),
>+      erle_estimator_(2 * kNumBlocksPerSecond, config_),
>       suppression_gain_limiter_(config_),
>       filter_analyzer_(config_),
>       echo_audibility_(
>           config_.echo_audibility.use_stationarity_properties_at_init),
>-      reverb_model_estimator_(config_),
>-      enable_erle_resets_at_gain_changes_(EnableErleResetsAtGainChanges()) {}
>+      reverb_model_estimator_(config_) {}
> 
> AecState::~AecState() = default;
> 
>@@ -97,13 +107,21 @@ void AecState::HandleEchoPathChange(
>     capture_signal_saturation_ = false;
>     strong_not_saturated_render_blocks_ = 0;
>     blocks_with_active_render_ = 0;
>-    suppression_gain_limiter_.Reset();
>+    if (use_suppressor_gain_limiter_) {
>+      suppression_gain_limiter_.Reset();
>+    }
>     initial_state_.Reset();
>     transparent_state_.Reset();
>-    saturation_detector_.Reset();
>+    if (use_legacy_saturation_behavior_) {
>+      legacy_saturation_detector_.Reset();
>+    }
>     erle_estimator_.Reset(true);
>     erl_estimator_.Reset();
>-    filter_quality_state_.Reset();
>+    if (use_legacy_filter_quality_) {
>+      legacy_filter_quality_state_.Reset();
>+    } else {
>+      filter_quality_state_.Reset();
>+    }
>   };
> 
>   // TODO(peah): Refine the reset scheme according to the type of gain and
>@@ -155,35 +173,56 @@ void AecState::Update(
>   strong_not_saturated_render_blocks_ +=
>       active_render && !SaturatedCapture() ? 1 : 0;
> 
>-  // Update the limit on the echo suppr ession after an echo path change to
>-  // avoid an initial echo burst.
>-  suppression_gain_limiter_.Update(render_buffer.GetRenderActivity(),
>-                                   TransparentMode());
>-  if (subtractor_output_analyzer_.ConvergedFilter()) {
>-    suppression_gain_limiter_.Deactivate();
>+  if (use_suppressor_gain_limiter_) {
>+    // Update the limit on the echo suppression after an echo path change to
>+    // avoid an initial echo burst.
>+    suppression_gain_limiter_.Update(render_buffer.GetRenderActivity(),
>+                                     TransparentMode());
>+
>+    if (subtractor_output_analyzer_.ConvergedFilter()) {
>+      suppression_gain_limiter_.Deactivate();
>+    }
>   }
> 
>+  std::array<float, kFftLengthBy2Plus1> X2_reverb;
>+  render_reverb_.Apply(
>+      render_buffer.GetSpectrumBuffer(), delay_state_.DirectPathFilterDelay(),
>+      config_.ep_strength.reverb_based_on_render ? ReverbDecay() : 0.f,
>+      X2_reverb);
>+
>   if (config_.echo_audibility.use_stationary_properties) {
>     // Update the echo audibility evaluator.
>-    echo_audibility_.Update(
>-        render_buffer, delay_state_.DirectPathFilterDelay(),
>-        delay_state_.ExternalDelayReported(),
>-        config_.ep_strength.reverb_based_on_render ? ReverbDecay() : 0.f);
>+    echo_audibility_.Update(render_buffer,
>+                            render_reverb_.GetReverbContributionPowerSpectrum(),
>+                            delay_state_.DirectPathFilterDelay(),
>+                            delay_state_.ExternalDelayReported());
>   }
> 
>   // Update the ERL and ERLE measures.
>   if (initial_state_.TransitionTriggered()) {
>     erle_estimator_.Reset(false);
>   }
>+
>   const auto& X2 = render_buffer.Spectrum(delay_state_.DirectPathFilterDelay());
>-  erle_estimator_.Update(X2, Y2, E2_main,
>+  const auto& X2_input_erle =
>+      enable_erle_updates_during_reverb_ ? X2_reverb : X2;
>+
>+  erle_estimator_.Update(render_buffer, adaptive_filter_frequency_response,
>+                         X2_input_erle, Y2, E2_main,
>                          subtractor_output_analyzer_.ConvergedFilter(),
>                          config_.erle.onset_detection);
>+
>   erl_estimator_.Update(subtractor_output_analyzer_.ConvergedFilter(), X2, Y2);
> 
>   // Detect and flag echo saturation.
>-  saturation_detector_.Update(aligned_render_block, SaturatedCapture(),
>-                              EchoPathGain());
>+  if (use_legacy_saturation_behavior_) {
>+    legacy_saturation_detector_.Update(aligned_render_block, SaturatedCapture(),
>+                                       EchoPathGain());
>+  } else {
>+    saturation_detector_.Update(aligned_render_block, SaturatedCapture(),
>+                                UsableLinearEstimate(), subtractor_output,
>+                                EchoPathGain());
>+  }
> 
>   // Update the decision on whether to use the initial state parameter set.
>   initial_state_.Update(active_render, SaturatedCapture());
>@@ -196,11 +235,17 @@ void AecState::Update(
>                             active_render, SaturatedCapture());
> 
>   // Analyze the quality of the filter.
>-  filter_quality_state_.Update(saturation_detector_.SaturatedEcho(),
>-                               active_render, SaturatedCapture(),
>-                               TransparentMode(), external_delay,
>-                               subtractor_output_analyzer_.ConvergedFilter(),
>-                               subtractor_output_analyzer_.DivergedFilter());
>+  if (use_legacy_filter_quality_) {
>+    legacy_filter_quality_state_.Update(
>+        SaturatedEcho(), active_render, SaturatedCapture(), TransparentMode(),
>+        external_delay, subtractor_output_analyzer_.ConvergedFilter(),
>+        subtractor_output_analyzer_.DivergedFilter());
>+  } else {
>+    filter_quality_state_.Update(active_render, TransparentMode(),
>+                                 SaturatedCapture(),
>+                                 filter_analyzer_.Consistent(), external_delay,
>+                                 subtractor_output_analyzer_.ConvergedFilter());
>+  }
> 
>   // Update the reverb estimate.
>   const bool stationary_block =
>@@ -227,8 +272,7 @@ void AecState::Update(
>   data_dumper_->DumpRaw("aec3_initial_state",
>                         initial_state_.InitialStateActive());
>   data_dumper_->DumpRaw("aec3_capture_saturation", SaturatedCapture());
>-  data_dumper_->DumpRaw("aec3_echo_saturation",
>-                        saturation_detector_.SaturatedEcho());
>+  data_dumper_->DumpRaw("aec3_echo_saturation", SaturatedEcho());
>   data_dumper_->DumpRaw("aec3_converged_filter",
>                         subtractor_output_analyzer_.ConvergedFilter());
>   data_dumper_->DumpRaw("aec3_diverged_filter",
>@@ -382,6 +426,51 @@ void AecState::TransparentMode::Update(int filter_delay_blocks,
> }
> 
> AecState::FilteringQualityAnalyzer::FilteringQualityAnalyzer(
>+    const EchoCanceller3Config& config) {}
>+
>+void AecState::FilteringQualityAnalyzer::Reset() {
>+  usable_linear_estimate_ = false;
>+  filter_update_blocks_since_reset_ = 0;
>+}
>+
>+void AecState::FilteringQualityAnalyzer::Update(
>+    bool active_render,
>+    bool transparent_mode,
>+    bool saturated_capture,
>+    bool consistent_estimate_,
>+    const absl::optional<DelayEstimate>& external_delay,
>+    bool converged_filter) {
>+  // Update blocks counter.
>+  const bool filter_update = active_render && !saturated_capture;
>+  filter_update_blocks_since_reset_ += filter_update ? 1 : 0;
>+  filter_update_blocks_since_start_ += filter_update ? 1 : 0;
>+
>+  // Store convergence flag when observed.
>+  convergence_seen_ = convergence_seen_ || converged_filter;
>+
>+  // Verify requirements for achieving a decent filter. The requirements for
>+  // filter adaptation at call startup are more restrictive than after an
>+  // in-call reset.
>+  const bool sufficient_data_to_converge_at_startup =
>+      filter_update_blocks_since_start_ > kNumBlocksPerSecond * 0.4f;
>+  const bool sufficient_data_to_converge_at_reset =
>+      sufficient_data_to_converge_at_startup &&
>+      filter_update_blocks_since_reset_ > kNumBlocksPerSecond * 0.2f;
>+
>+  // The linear filter can only be used it has had time to converge.
>+  usable_linear_estimate_ = sufficient_data_to_converge_at_startup &&
>+                            sufficient_data_to_converge_at_reset;
>+
>+  // The linear filter can only be used if an external delay or convergence have
>+  // been identified
>+  usable_linear_estimate_ =
>+      usable_linear_estimate_ && (external_delay || convergence_seen_);
>+
>+  // If transparent mode is on, deactivate usign the linear filter.
>+  usable_linear_estimate_ = usable_linear_estimate_ && !transparent_mode;
>+}
>+
>+AecState::LegacyFilteringQualityAnalyzer::LegacyFilteringQualityAnalyzer(
>     const EchoCanceller3Config& config)
>     : conservative_initial_phase_(config.filter.conservative_initial_phase),
>       required_blocks_for_convergence_(
>@@ -390,7 +479,7 @@ AecState::FilteringQualityAnalyzer::FilteringQualityAnalyzer(
>           config.echo_removal_control.linear_and_stable_echo_path),
>       non_converged_sequence_size_(kBlocksSinceConvergencedFilterInit) {}
> 
>-void AecState::FilteringQualityAnalyzer::Reset() {
>+void AecState::LegacyFilteringQualityAnalyzer::Reset() {
>   usable_linear_estimate_ = false;
>   strong_not_saturated_render_blocks_ = 0;
>   if (linear_and_stable_echo_path_) {
>@@ -402,7 +491,7 @@ void AecState::FilteringQualityAnalyzer::Reset() {
>   recent_convergence_ = true;
> }
> 
>-void AecState::FilteringQualityAnalyzer::Update(
>+void AecState::LegacyFilteringQualityAnalyzer::Update(
>     bool saturated_echo,
>     bool active_render,
>     bool saturated_capture,
>@@ -454,18 +543,41 @@ void AecState::FilteringQualityAnalyzer::Update(
>   }
> }
> 
>-AecState::SaturationDetector::SaturationDetector(
>+void AecState::SaturationDetector::Update(
>+    rtc::ArrayView<const float> x,
>+    bool saturated_capture,
>+    bool usable_linear_estimate,
>+    const SubtractorOutput& subtractor_output,
>+    float echo_path_gain) {
>+  saturated_echo_ = saturated_capture;
>+  if (usable_linear_estimate) {
>+    constexpr float kSaturationThreshold = 20000.f;
>+    saturated_echo_ =
>+        saturated_echo_ &&
>+        (subtractor_output.s_main_max_abs > kSaturationThreshold ||
>+         subtractor_output.s_shadow_max_abs > kSaturationThreshold);
>+  } else {
>+    const float max_sample = fabs(*std::max_element(
>+        x.begin(), x.end(), [](float a, float b) { return a * a < b * b; }));
>+
>+    const float kMargin = 10.f;
>+    float peak_echo_amplitude = max_sample * echo_path_gain * kMargin;
>+    saturated_echo_ = saturated_echo_ && peak_echo_amplitude > 32000;
>+  }
>+}
>+
>+AecState::LegacySaturationDetector::LegacySaturationDetector(
>     const EchoCanceller3Config& config)
>     : echo_can_saturate_(config.ep_strength.echo_can_saturate),
>       not_saturated_sequence_size_(1000) {}
> 
>-void AecState::SaturationDetector::Reset() {
>+void AecState::LegacySaturationDetector::Reset() {
>   not_saturated_sequence_size_ = 0;
> }
> 
>-void AecState::SaturationDetector::Update(rtc::ArrayView<const float> x,
>-                                          bool saturated_capture,
>-                                          float echo_path_gain) {
>+void AecState::LegacySaturationDetector::Update(rtc::ArrayView<const float> x,
>+                                                bool saturated_capture,
>+                                                float echo_path_gain) {
>   if (!echo_can_saturate_) {
>     saturated_echo_ = false;
>     return;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.h
>index ca476428a54d84acd22406087dfbe232d64b45fa..c9d9cf3f3a0bd28c4ab6f6622772c7858b2157ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.h
>@@ -11,9 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
> 
>-#include <math.h>
>-
>-#include <algorithm>
>+#include <stddef.h>
>+#include <array>
> #include <memory>
> #include <vector>
> 
>@@ -28,6 +27,7 @@
> #include "modules/audio_processing/aec3/erle_estimator.h"
> #include "modules/audio_processing/aec3/filter_analyzer.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
>+#include "modules/audio_processing/aec3/render_reverb_model.h"
> #include "modules/audio_processing/aec3/reverb_model_estimator.h"
> #include "modules/audio_processing/aec3/subtractor_output.h"
> #include "modules/audio_processing/aec3/subtractor_output_analyzer.h"
>@@ -46,11 +46,17 @@ class AecState {
>   // Returns whether the echo subtractor can be used to determine the residual
>   // echo.
>   bool UsableLinearEstimate() const {
>+    if (use_legacy_filter_quality_) {
>+      return legacy_filter_quality_state_.LinearFilterUsable();
>+    }
>     return filter_quality_state_.LinearFilterUsable();
>   }
> 
>   // Returns whether the echo subtractor output should be used as output.
>   bool UseLinearFilterOutput() const {
>+    if (use_legacy_filter_quality_) {
>+      return legacy_filter_quality_state_.LinearFilterUsable();
>+    }
>     return filter_quality_state_.LinearFilterUsable();
>   }
> 
>@@ -99,7 +105,11 @@ class AecState {
>   bool SaturatedCapture() const { return capture_signal_saturation_; }
> 
>   // Returns whether the echo signal is saturated.
>-  bool SaturatedEcho() const { return saturation_detector_.SaturatedEcho(); }
>+  bool SaturatedEcho() const {
>+    return use_legacy_saturation_behavior_
>+               ? legacy_saturation_detector_.SaturatedEcho()
>+               : saturation_detector_.SaturatedEcho();
>+  }
> 
>   // Updates the capture signal saturation.
>   void UpdateCaptureSaturation(bool capture_signal_saturation) {
>@@ -122,7 +132,11 @@ class AecState {
> 
>   // Returns the upper limit for the echo suppression gain.
>   float SuppressionGainLimit() const {
>-    return suppression_gain_limiter_.Limit();
>+    if (use_suppressor_gain_limiter_) {
>+      return suppression_gain_limiter_.Limit();
>+    } else {
>+      return 1.f;
>+    }
>   }
> 
>   // Returns whether the suppression gain limiter is active.
>@@ -153,13 +167,14 @@ class AecState {
>   }
> 
>  private:
>-  void UpdateSuppressorGainLimit(bool render_activity);
>-  bool DetectEchoSaturation(rtc::ArrayView<const float> x,
>-                            float echo_path_gain);
>-
>   static int instance_count_;
>   std::unique_ptr<ApmDataDumper> data_dumper_;
>   const EchoCanceller3Config config_;
>+  const bool use_legacy_saturation_behavior_;
>+  const bool enable_erle_resets_at_gain_changes_;
>+  const bool enable_erle_updates_during_reverb_;
>+  const bool use_legacy_filter_quality_;
>+  const bool use_suppressor_gain_limiter_;
> 
>   // Class for controlling the transition from the intial state, which in turn
>   // controls when the filter parameters for the initial state should be used.
>@@ -255,7 +270,37 @@ class AecState {
>   // suppressor.
>   class FilteringQualityAnalyzer {
>    public:
>-    explicit FilteringQualityAnalyzer(const EchoCanceller3Config& config);
>+    FilteringQualityAnalyzer(const EchoCanceller3Config& config);
>+
>+    // Returns whether the the linear filter can be used for the echo
>+    // canceller output.
>+    bool LinearFilterUsable() const { return usable_linear_estimate_; }
>+
>+    // Resets the state of the analyzer.
>+    void Reset();
>+
>+    // Updates the analysis based on new data.
>+    void Update(bool active_render,
>+                bool transparent_mode,
>+                bool saturated_capture,
>+                bool consistent_estimate_,
>+                const absl::optional<DelayEstimate>& external_delay,
>+                bool converged_filter);
>+
>+   private:
>+    bool usable_linear_estimate_ = false;
>+    size_t filter_update_blocks_since_reset_ = 0;
>+    size_t filter_update_blocks_since_start_ = 0;
>+    bool convergence_seen_ = false;
>+  } filter_quality_state_;
>+
>+  // Class containing the legacy functionality for analyzing how well the linear
>+  // filter is, and can be expected to perform on the current signals. The
>+  // purpose of this is for using to select the echo suppression functionality
>+  // as well as the input to the echo suppressor.
>+  class LegacyFilteringQualityAnalyzer {
>+   public:
>+    explicit LegacyFilteringQualityAnalyzer(const EchoCanceller3Config& config);
> 
>     // Returns whether the the linear filter is can be used for the echo
>     // canceller output.
>@@ -284,14 +329,32 @@ class AecState {
>     size_t active_non_converged_sequence_size_ = 0;
>     bool recent_convergence_during_activity_ = false;
>     bool recent_convergence_ = false;
>-  } filter_quality_state_;
>+  } legacy_filter_quality_state_;
> 
>-  // Class for detecting whether the echo is to be considered to be saturated.
>-  // The purpose of this is to allow customized behavior in the echo suppressor
>-  // for when the echo is saturated.
>+  // Class for detecting whether the echo is to be considered to be
>+  // saturated.
>   class SaturationDetector {
>    public:
>-    explicit SaturationDetector(const EchoCanceller3Config& config);
>+    // Returns whether the echo is to be considered saturated.
>+    bool SaturatedEcho() const { return saturated_echo_; };
>+
>+    // Updates the detection decision based on new data.
>+    void Update(rtc::ArrayView<const float> x,
>+                bool saturated_capture,
>+                bool usable_linear_estimate,
>+                const SubtractorOutput& subtractor_output,
>+                float echo_path_gain);
>+
>+   private:
>+    bool saturated_echo_ = false;
>+  } saturation_detector_;
>+
>+  // Legacy class for detecting whether the echo is to be considered to be
>+  // saturated. This is kept as a fallback solution to use instead of the class
>+  // SaturationDetector,
>+  class LegacySaturationDetector {
>+   public:
>+    explicit LegacySaturationDetector(const EchoCanceller3Config& config);
> 
>     // Returns whether the echo is to be considered saturated.
>     bool SaturatedEcho() const { return saturated_echo_; };
>@@ -308,7 +371,7 @@ class AecState {
>     const bool echo_can_saturate_;
>     size_t not_saturated_sequence_size_;
>     bool saturated_echo_ = false;
>-  } saturation_detector_;
>+  } legacy_saturation_detector_;
> 
>   ErlEstimator erl_estimator_;
>   ErleEstimator erle_estimator_;
>@@ -321,8 +384,8 @@ class AecState {
>   absl::optional<DelayEstimate> external_delay_;
>   EchoAudibility echo_audibility_;
>   ReverbModelEstimator reverb_model_estimator_;
>+  RenderReverbModel render_reverb_;
>   SubtractorOutputAnalyzer subtractor_output_analyzer_;
>-  bool enable_erle_resets_at_gain_changes_ = true;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state_unittest.cc
>index bf32a0b60720ecee71f32ef34a35b15374fe706d..a331006378162318b6c1c6e71a9332ceeaa0cce5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state_unittest.cc
>@@ -25,7 +25,7 @@ TEST(AecState, NormalUsage) {
>   absl::optional<DelayEstimate> delay_estimate =
>       DelayEstimate(DelayEstimate::Quality::kRefined, 10);
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
>   std::array<float, kFftLengthBy2Plus1> E2_main = {};
>   std::array<float, kFftLengthBy2Plus1> Y2 = {};
>   std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>@@ -56,7 +56,7 @@ TEST(AecState, NormalUsage) {
>   std::fill(x[0].begin(), x[0].end(), 101.f);
>   for (int k = 0; k < 3000; ++k) {
>     render_delay_buffer->Insert(x);
>-    output.UpdatePowers(y);
>+    output.ComputeMetrics(y);
>     state.Update(delay_estimate, converged_filter_frequency_response,
>                  impulse_response, *render_delay_buffer->GetRenderBuffer(),
>                  E2_main, Y2, output, y);
>@@ -65,7 +65,7 @@ TEST(AecState, NormalUsage) {
> 
>   // Verify that linear AEC usability becomes false after an echo path change is
>   // reported
>-  output.UpdatePowers(y);
>+  output.ComputeMetrics(y);
>   state.HandleEchoPathChange(EchoPathVariability(
>       false, EchoPathVariability::DelayAdjustment::kBufferReadjustment, false));
>   state.Update(delay_estimate, converged_filter_frequency_response,
>@@ -76,7 +76,7 @@ TEST(AecState, NormalUsage) {
>   // Verify that the active render detection works as intended.
>   std::fill(x[0].begin(), x[0].end(), 101.f);
>   render_delay_buffer->Insert(x);
>-  output.UpdatePowers(y);
>+  output.ComputeMetrics(y);
>   state.HandleEchoPathChange(EchoPathVariability(
>       true, EchoPathVariability::DelayAdjustment::kNewDetectedDelay, false));
>   state.Update(delay_estimate, converged_filter_frequency_response,
>@@ -86,7 +86,7 @@ TEST(AecState, NormalUsage) {
> 
>   for (int k = 0; k < 1000; ++k) {
>     render_delay_buffer->Insert(x);
>-    output.UpdatePowers(y);
>+    output.ComputeMetrics(y);
>     state.Update(delay_estimate, converged_filter_frequency_response,
>                  impulse_response, *render_delay_buffer->GetRenderBuffer(),
>                  E2_main, Y2, output, y);
>@@ -110,7 +110,7 @@ TEST(AecState, NormalUsage) {
> 
>   Y2.fill(10.f * 10000.f * 10000.f);
>   for (size_t k = 0; k < 1000; ++k) {
>-    output.UpdatePowers(y);
>+    output.ComputeMetrics(y);
>     state.Update(delay_estimate, converged_filter_frequency_response,
>                  impulse_response, *render_delay_buffer->GetRenderBuffer(),
>                  E2_main, Y2, output, y);
>@@ -128,7 +128,7 @@ TEST(AecState, NormalUsage) {
>   E2_main.fill(1.f * 10000.f * 10000.f);
>   Y2.fill(10.f * E2_main[0]);
>   for (size_t k = 0; k < 1000; ++k) {
>-    output.UpdatePowers(y);
>+    output.ComputeMetrics(y);
>     state.Update(delay_estimate, converged_filter_frequency_response,
>                  impulse_response, *render_delay_buffer->GetRenderBuffer(),
>                  E2_main, Y2, output, y);
>@@ -152,7 +152,7 @@ TEST(AecState, NormalUsage) {
>   E2_main.fill(1.f * 10000.f * 10000.f);
>   Y2.fill(5.f * E2_main[0]);
>   for (size_t k = 0; k < 1000; ++k) {
>-    output.UpdatePowers(y);
>+    output.ComputeMetrics(y);
>     state.Update(delay_estimate, converged_filter_frequency_response,
>                  impulse_response, *render_delay_buffer->GetRenderBuffer(),
>                  E2_main, Y2, output, y);
>@@ -179,7 +179,7 @@ TEST(AecState, ConvergedFilterDelay) {
>   EchoCanceller3Config config;
>   AecState state(config);
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
>   absl::optional<DelayEstimate> delay_estimate;
>   std::array<float, kFftLengthBy2Plus1> E2_main;
>   std::array<float, kFftLengthBy2Plus1> Y2;
>@@ -208,7 +208,7 @@ TEST(AecState, ConvergedFilterDelay) {
>     impulse_response[k * kBlockSize + 1] = 1.f;
> 
>     state.HandleEchoPathChange(echo_path_variability);
>-    output.UpdatePowers(y);
>+    output.ComputeMetrics(y);
>     state.Update(delay_estimate, frequency_response, impulse_response,
>                  *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, output,
>                  y);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/api_call_jitter_metrics.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/api_call_jitter_metrics.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..45f56a5dce6fab1371f244e89e2c025f7a440c98
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/api_call_jitter_metrics.cc
>@@ -0,0 +1,121 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/api_call_jitter_metrics.h"
>+
>+#include <algorithm>
>+#include <limits>
>+
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "system_wrappers/include/metrics.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+bool TimeToReportMetrics(int frames_since_last_report) {
>+  constexpr int kNumFramesPerSecond = 100;
>+  constexpr int kReportingIntervalFrames = 10 * kNumFramesPerSecond;
>+  return frames_since_last_report == kReportingIntervalFrames;
>+}
>+
>+}  // namespace
>+
>+ApiCallJitterMetrics::Jitter::Jitter()
>+    : max_(0), min_(std::numeric_limits<int>::max()) {}
>+
>+void ApiCallJitterMetrics::Jitter::Update(int num_api_calls_in_a_row) {
>+  min_ = std::min(min_, num_api_calls_in_a_row);
>+  max_ = std::max(max_, num_api_calls_in_a_row);
>+}
>+
>+void ApiCallJitterMetrics::Jitter::Reset() {
>+  min_ = std::numeric_limits<int>::max();
>+  max_ = 0;
>+}
>+
>+void ApiCallJitterMetrics::Reset() {
>+  render_jitter_.Reset();
>+  capture_jitter_.Reset();
>+  num_api_calls_in_a_row_ = 0;
>+  frames_since_last_report_ = 0;
>+  last_call_was_render_ = false;
>+  proper_call_observed_ = false;
>+}
>+
>+void ApiCallJitterMetrics::ReportRenderCall() {
>+  if (!last_call_was_render_) {
>+    // If the previous call was a capture and a proper call has been observed
>+    // (containing both render and capture data), storing the last number of
>+    // capture calls into the metrics.
>+    if (proper_call_observed_) {
>+      capture_jitter_.Update(num_api_calls_in_a_row_);
>+    }
>+
>+    // Reset the call counter to start counting render calls.
>+    num_api_calls_in_a_row_ = 0;
>+  }
>+  ++num_api_calls_in_a_row_;
>+  last_call_was_render_ = true;
>+}
>+
>+void ApiCallJitterMetrics::ReportCaptureCall() {
>+  if (last_call_was_render_) {
>+    // If the previous call was a render and a proper call has been observed
>+    // (containing both render and capture data), storing the last number of
>+    // render calls into the metrics.
>+    if (proper_call_observed_) {
>+      render_jitter_.Update(num_api_calls_in_a_row_);
>+    }
>+    // Reset the call counter to start counting capture calls.
>+    num_api_calls_in_a_row_ = 0;
>+
>+    // If this statement is reached, at least one render and one capture call
>+    // have been observed.
>+    proper_call_observed_ = true;
>+  }
>+  ++num_api_calls_in_a_row_;
>+  last_call_was_render_ = false;
>+
>+  // Only report and update jitter metrics for when a proper call, containing
>+  // both render and capture data, has been observed.
>+  if (proper_call_observed_ &&
>+      TimeToReportMetrics(++frames_since_last_report_)) {
>+    // Report jitter, where the base basic unit is frames.
>+    constexpr int kMaxJitterToReport = 50;
>+
>+    // Report max and min jitter for render and capture, in units of 20 ms.
>+    RTC_HISTOGRAM_COUNTS_LINEAR(
>+        "WebRTC.Audio.EchoCanceller.MaxRenderJitter",
>+        std::min(kMaxJitterToReport, render_jitter().max()), 1,
>+        kMaxJitterToReport, kMaxJitterToReport);
>+    RTC_HISTOGRAM_COUNTS_LINEAR(
>+        "WebRTC.Audio.EchoCanceller.MinRenderJitter",
>+        std::min(kMaxJitterToReport, render_jitter().min()), 1,
>+        kMaxJitterToReport, kMaxJitterToReport);
>+
>+    RTC_HISTOGRAM_COUNTS_LINEAR(
>+        "WebRTC.Audio.EchoCanceller.MaxCaptureJitter",
>+        std::min(kMaxJitterToReport, capture_jitter().max()), 1,
>+        kMaxJitterToReport, kMaxJitterToReport);
>+    RTC_HISTOGRAM_COUNTS_LINEAR(
>+        "WebRTC.Audio.EchoCanceller.MinCaptureJitter",
>+        std::min(kMaxJitterToReport, capture_jitter().min()), 1,
>+        kMaxJitterToReport, kMaxJitterToReport);
>+
>+    frames_since_last_report_ = 0;
>+    Reset();
>+  }
>+}
>+
>+bool ApiCallJitterMetrics::WillReportMetricsAtNextCapture() const {
>+  return TimeToReportMetrics(frames_since_last_report_ + 1);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..dd1fa82e9324707292e87ee2f375cf85e717f3db
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/api_call_jitter_metrics.h
>@@ -0,0 +1,60 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_API_CALL_JITTER_METRICS_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_API_CALL_JITTER_METRICS_H_
>+
>+namespace webrtc {
>+
>+// Stores data for reporting metrics on the API call jitter.
>+class ApiCallJitterMetrics {
>+ public:
>+  class Jitter {
>+   public:
>+    Jitter();
>+    void Update(int num_api_calls_in_a_row);
>+    void Reset();
>+
>+    int min() const { return min_; }
>+    int max() const { return max_; }
>+
>+   private:
>+    int max_;
>+    int min_;
>+  };
>+
>+  ApiCallJitterMetrics() { Reset(); }
>+
>+  // Update metrics for render API call.
>+  void ReportRenderCall();
>+
>+  // Update and periodically report metrics for capture API call.
>+  void ReportCaptureCall();
>+
>+  // Methods used only for testing.
>+  const Jitter& render_jitter() const { return render_jitter_; }
>+  const Jitter& capture_jitter() const { return capture_jitter_; }
>+  bool WillReportMetricsAtNextCapture() const;
>+
>+ private:
>+  void Reset();
>+
>+  Jitter render_jitter_;
>+  Jitter capture_jitter_;
>+
>+  int num_api_calls_in_a_row_ = 0;
>+  int frames_since_last_report_ = 0;
>+  bool last_call_was_render_ = false;
>+  bool proper_call_observed_ = false;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_API_CALL_JITTER_METRICS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/api_call_jitter_metrics_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/api_call_jitter_metrics_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..86608aa3e1093f97bc370dc9aa6b91bbff4933d4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/api_call_jitter_metrics_unittest.cc
>@@ -0,0 +1,109 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/api_call_jitter_metrics.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+// Verify constant jitter.
>+TEST(ApiCallJitterMetrics, ConstantJitter) {
>+  for (int jitter = 1; jitter < 20; ++jitter) {
>+    ApiCallJitterMetrics metrics;
>+    for (size_t k = 0; k < 30 * kNumBlocksPerSecond; ++k) {
>+      for (int j = 0; j < jitter; ++j) {
>+        metrics.ReportRenderCall();
>+      }
>+
>+      for (int j = 0; j < jitter; ++j) {
>+        metrics.ReportCaptureCall();
>+
>+        if (metrics.WillReportMetricsAtNextCapture()) {
>+          EXPECT_EQ(jitter, metrics.render_jitter().min());
>+          EXPECT_EQ(jitter, metrics.render_jitter().max());
>+          EXPECT_EQ(jitter, metrics.capture_jitter().min());
>+          EXPECT_EQ(jitter, metrics.capture_jitter().max());
>+        }
>+      }
>+    }
>+  }
>+}
>+
>+// Verify peaky jitter for the render.
>+TEST(ApiCallJitterMetrics, JitterPeakRender) {
>+  constexpr int kMinJitter = 2;
>+  constexpr int kJitterPeak = 10;
>+  constexpr int kPeakInterval = 100;
>+
>+  ApiCallJitterMetrics metrics;
>+  int render_surplus = 0;
>+
>+  for (size_t k = 0; k < 30 * kNumBlocksPerSecond; ++k) {
>+    const int num_render_calls =
>+        k % kPeakInterval == 0 ? kJitterPeak : kMinJitter;
>+    for (int j = 0; j < num_render_calls; ++j) {
>+      metrics.ReportRenderCall();
>+      ++render_surplus;
>+    }
>+
>+    ASSERT_LE(kMinJitter, render_surplus);
>+    const int num_capture_calls =
>+        render_surplus == kMinJitter ? kMinJitter : kMinJitter + 1;
>+    for (int j = 0; j < num_capture_calls; ++j) {
>+      metrics.ReportCaptureCall();
>+
>+      if (metrics.WillReportMetricsAtNextCapture()) {
>+        EXPECT_EQ(kMinJitter, metrics.render_jitter().min());
>+        EXPECT_EQ(kJitterPeak, metrics.render_jitter().max());
>+        EXPECT_EQ(kMinJitter, metrics.capture_jitter().min());
>+        EXPECT_EQ(kMinJitter + 1, metrics.capture_jitter().max());
>+      }
>+      --render_surplus;
>+    }
>+  }
>+}
>+
>+// Verify peaky jitter for the capture.
>+TEST(ApiCallJitterMetrics, JitterPeakCapture) {
>+  constexpr int kMinJitter = 2;
>+  constexpr int kJitterPeak = 10;
>+  constexpr int kPeakInterval = 100;
>+
>+  ApiCallJitterMetrics metrics;
>+  int capture_surplus = kMinJitter;
>+
>+  for (size_t k = 0; k < 30 * kNumBlocksPerSecond; ++k) {
>+    ASSERT_LE(kMinJitter, capture_surplus);
>+    const int num_render_calls =
>+        capture_surplus == kMinJitter ? kMinJitter : kMinJitter + 1;
>+    for (int j = 0; j < num_render_calls; ++j) {
>+      metrics.ReportRenderCall();
>+      --capture_surplus;
>+    }
>+
>+    const int num_capture_calls =
>+        k % kPeakInterval == 0 ? kJitterPeak : kMinJitter;
>+    for (int j = 0; j < num_capture_calls; ++j) {
>+      metrics.ReportCaptureCall();
>+
>+      if (metrics.WillReportMetricsAtNextCapture()) {
>+        EXPECT_EQ(kMinJitter, metrics.render_jitter().min());
>+        EXPECT_EQ(kMinJitter + 1, metrics.render_jitter().max());
>+        EXPECT_EQ(kMinJitter, metrics.capture_jitter().min());
>+        EXPECT_EQ(kJitterPeak, metrics.capture_jitter().max());
>+      }
>+      ++capture_surplus;
>+    }
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_delay_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_delay_buffer.h
>index 6e5fd5c09d0d46e25082d56805ac2dcf1eb4d088..624e9139a3888c90a086d25fc255a166ee389ced 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_delay_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_delay_buffer.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
> 
>+#include <stddef.h>
> #include <vector>
> 
> #include "modules/audio_processing/audio_buffer.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_framer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_framer.cc
>index 3160624515abae99c92486b7ce308cf828c382db..ca7667c24f4626f34cb5978e2671039c9d37d744 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_framer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_framer.cc
>@@ -12,6 +12,7 @@
> 
> #include <algorithm>
> 
>+#include "modules/audio_processing/aec3/aec3_common.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.cc
>index 1bb69584cc3fbf7932c1a672217d1f5e0d569a78..ef25e7c23bc754edcc604888aa03c6f2c9b71fc8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.cc
>@@ -9,12 +9,16 @@
>  */
> #include "modules/audio_processing/aec3/block_processor.h"
> 
>+#include <utility>
>+
> #include "absl/types/optional.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/block_processor_metrics.h"
>+#include "modules/audio_processing/aec3/delay_estimate.h"
> #include "modules/audio_processing/aec3/echo_path_variability.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
> 
>@@ -104,7 +108,7 @@ void BlockProcessorImpl::ProcessCapture(
>     if (!capture_properly_started_) {
>       capture_properly_started_ = true;
>       render_buffer_->Reset();
>-      delay_controller_->Reset();
>+      delay_controller_->Reset(true);
>     }
>   } else {
>     // If no render data has yet arrived, do not process the capture signal.
>@@ -119,7 +123,7 @@ void BlockProcessorImpl::ProcessCapture(
>       render_properly_started_) {
>     echo_path_variability.delay_change =
>         EchoPathVariability::DelayAdjustment::kBufferFlush;
>-    delay_controller_->Reset();
>+    delay_controller_->Reset(true);
>     RTC_LOG(LS_WARNING) << "Reset due to render buffer overrun at block  "
>                         << capture_call_counter_;
>   }
>@@ -135,11 +139,11 @@ void BlockProcessorImpl::ProcessCapture(
>         estimated_delay_->quality == DelayEstimate::Quality::kRefined) {
>       echo_path_variability.delay_change =
>           EchoPathVariability::DelayAdjustment::kDelayReset;
>-      delay_controller_->Reset();
>+      delay_controller_->Reset(true);
>       capture_properly_started_ = false;
>       render_properly_started_ = false;
> 
>-      RTC_LOG(LS_WARNING) << "Reset due to render buffer underrrun at block "
>+      RTC_LOG(LS_WARNING) << "Reset due to render buffer underrun at block "
>                           << capture_call_counter_;
>     }
>   } else if (render_event_ == RenderDelayBuffer::BufferingEvent::kApiCallSkew) {
>@@ -147,7 +151,7 @@ void BlockProcessorImpl::ProcessCapture(
>     // echo.
>     echo_path_variability.delay_change =
>         EchoPathVariability::DelayAdjustment::kDelayReset;
>-    delay_controller_->Reset();
>+    delay_controller_->Reset(true);
>     capture_properly_started_ = false;
>     render_properly_started_ = false;
>     RTC_LOG(LS_WARNING) << "Reset due to render buffer api skew at block "
>@@ -180,7 +184,7 @@ void BlockProcessorImpl::ProcessCapture(
>       if (estimated_delay_->quality == DelayEstimate::Quality::kRefined) {
>         echo_path_variability.delay_change =
>             EchoPathVariability::DelayAdjustment::kDelayReset;
>-        delay_controller_->Reset();
>+        delay_controller_->Reset(true);
>         render_buffer_->Reset();
>         capture_properly_started_ = false;
>         render_properly_started_ = false;
>@@ -190,6 +194,8 @@ void BlockProcessorImpl::ProcessCapture(
>     }
>   }
> 
>+  echo_path_variability.clock_drift = delay_controller_->HasClockdrift();
>+
>   // Remove the echo from the capture signal.
>   echo_remover_->ProcessCapture(
>       echo_path_variability, capture_signal_saturation, estimated_delay_,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.h
>index a3967ea7e4a41e577144e9c206b15f38a984b79f..5f7ec7c17c3c86b7892b30f3b1cb85fb0e1db655 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.h
>@@ -11,9 +11,12 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
> 
>+#include <stddef.h>
> #include <memory>
> #include <vector>
> 
>+#include "api/audio/echo_canceller3_config.h"
>+#include "api/audio/echo_control.h"
> #include "modules/audio_processing/aec3/echo_remover.h"
> #include "modules/audio_processing/aec3/render_delay_buffer.h"
> #include "modules/audio_processing/aec3/render_delay_controller.h"
>@@ -23,19 +26,33 @@ namespace webrtc {
> // Class for performing echo cancellation on 64 sample blocks of audio data.
> class BlockProcessor {
>  public:
>+  // Create a block processor with the legacy render buffering.
>   static BlockProcessor* Create(const EchoCanceller3Config& config,
>                                 int sample_rate_hz);
>+  // Create a block processor with the new render buffering.
>+  static BlockProcessor* Create2(const EchoCanceller3Config& config,
>+                                 int sample_rate_hz);
>   // Only used for testing purposes.
>   static BlockProcessor* Create(
>       const EchoCanceller3Config& config,
>       int sample_rate_hz,
>       std::unique_ptr<RenderDelayBuffer> render_buffer);
>+  static BlockProcessor* Create2(
>+      const EchoCanceller3Config& config,
>+      int sample_rate_hz,
>+      std::unique_ptr<RenderDelayBuffer> render_buffer);
>   static BlockProcessor* Create(
>       const EchoCanceller3Config& config,
>       int sample_rate_hz,
>       std::unique_ptr<RenderDelayBuffer> render_buffer,
>       std::unique_ptr<RenderDelayController> delay_controller,
>       std::unique_ptr<EchoRemover> echo_remover);
>+  static BlockProcessor* Create2(
>+      const EchoCanceller3Config& config,
>+      int sample_rate_hz,
>+      std::unique_ptr<RenderDelayBuffer> render_buffer,
>+      std::unique_ptr<RenderDelayController> delay_controller,
>+      std::unique_ptr<EchoRemover> echo_remover);
> 
>   virtual ~BlockProcessor() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor2.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..30bd3ee5ca2952ca227e41b61c402056340b1d4f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor2.cc
>@@ -0,0 +1,256 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include <stddef.h>
>+#include <memory>
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "api/audio/echo_control.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/block_processor.h"
>+#include "modules/audio_processing/aec3/block_processor_metrics.h"
>+#include "modules/audio_processing/aec3/delay_estimate.h"
>+#include "modules/audio_processing/aec3/echo_path_variability.h"
>+#include "modules/audio_processing/aec3/echo_remover.h"
>+#include "modules/audio_processing/aec3/render_delay_buffer.h"
>+#include "modules/audio_processing/aec3/render_delay_controller.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/atomicops.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+enum class BlockProcessorApiCall { kCapture, kRender };
>+
>+class BlockProcessorImpl2 final : public BlockProcessor {
>+ public:
>+  BlockProcessorImpl2(const EchoCanceller3Config& config,
>+                      int sample_rate_hz,
>+                      std::unique_ptr<RenderDelayBuffer> render_buffer,
>+                      std::unique_ptr<RenderDelayController> delay_controller,
>+                      std::unique_ptr<EchoRemover> echo_remover);
>+
>+  BlockProcessorImpl2() = delete;
>+
>+  ~BlockProcessorImpl2() override;
>+
>+  void ProcessCapture(bool echo_path_gain_change,
>+                      bool capture_signal_saturation,
>+                      std::vector<std::vector<float>>* capture_block) override;
>+
>+  void BufferRender(const std::vector<std::vector<float>>& block) override;
>+
>+  void UpdateEchoLeakageStatus(bool leakage_detected) override;
>+
>+  void GetMetrics(EchoControl::Metrics* metrics) const override;
>+
>+  void SetAudioBufferDelay(size_t delay_ms) override;
>+
>+ private:
>+  static int instance_count_;
>+  std::unique_ptr<ApmDataDumper> data_dumper_;
>+  const EchoCanceller3Config config_;
>+  bool capture_properly_started_ = false;
>+  bool render_properly_started_ = false;
>+  const size_t sample_rate_hz_;
>+  std::unique_ptr<RenderDelayBuffer> render_buffer_;
>+  std::unique_ptr<RenderDelayController> delay_controller_;
>+  std::unique_ptr<EchoRemover> echo_remover_;
>+  BlockProcessorMetrics metrics_;
>+  RenderDelayBuffer::BufferingEvent render_event_;
>+  size_t capture_call_counter_ = 0;
>+  absl::optional<DelayEstimate> estimated_delay_;
>+  absl::optional<int> echo_remover_delay_;
>+};
>+
>+int BlockProcessorImpl2::instance_count_ = 0;
>+
>+BlockProcessorImpl2::BlockProcessorImpl2(
>+    const EchoCanceller3Config& config,
>+    int sample_rate_hz,
>+    std::unique_ptr<RenderDelayBuffer> render_buffer,
>+    std::unique_ptr<RenderDelayController> delay_controller,
>+    std::unique_ptr<EchoRemover> echo_remover)
>+    : data_dumper_(
>+          new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>+      config_(config),
>+      sample_rate_hz_(sample_rate_hz),
>+      render_buffer_(std::move(render_buffer)),
>+      delay_controller_(std::move(delay_controller)),
>+      echo_remover_(std::move(echo_remover)),
>+      render_event_(RenderDelayBuffer::BufferingEvent::kNone) {
>+  RTC_DCHECK(ValidFullBandRate(sample_rate_hz_));
>+}
>+
>+BlockProcessorImpl2::~BlockProcessorImpl2() = default;
>+
>+void BlockProcessorImpl2::ProcessCapture(
>+    bool echo_path_gain_change,
>+    bool capture_signal_saturation,
>+    std::vector<std::vector<float>>* capture_block) {
>+  RTC_DCHECK(capture_block);
>+  RTC_DCHECK_EQ(NumBandsForRate(sample_rate_hz_), capture_block->size());
>+  RTC_DCHECK_EQ(kBlockSize, (*capture_block)[0].size());
>+
>+  capture_call_counter_++;
>+
>+  data_dumper_->DumpRaw("aec3_processblock_call_order",
>+                        static_cast<int>(BlockProcessorApiCall::kCapture));
>+  data_dumper_->DumpWav("aec3_processblock_capture_input", kBlockSize,
>+                        &(*capture_block)[0][0],
>+                        LowestBandRate(sample_rate_hz_), 1);
>+
>+  if (render_properly_started_) {
>+    if (!capture_properly_started_) {
>+      capture_properly_started_ = true;
>+      render_buffer_->Reset();
>+      delay_controller_->Reset(true);
>+    }
>+  } else {
>+    // If no render data has yet arrived, do not process the capture signal.
>+    return;
>+  }
>+
>+  EchoPathVariability echo_path_variability(
>+      echo_path_gain_change, EchoPathVariability::DelayAdjustment::kNone,
>+      false);
>+
>+  if (render_event_ == RenderDelayBuffer::BufferingEvent::kRenderOverrun &&
>+      render_properly_started_) {
>+    echo_path_variability.delay_change =
>+        EchoPathVariability::DelayAdjustment::kBufferFlush;
>+    delay_controller_->Reset(true);
>+    RTC_LOG(LS_WARNING) << "Reset due to render buffer overrun at block  "
>+                        << capture_call_counter_;
>+  }
>+  render_event_ = RenderDelayBuffer::BufferingEvent::kNone;
>+
>+  // Update the render buffers with any newly arrived render blocks and prepare
>+  // the render buffers for reading the render data corresponding to the current
>+  // capture block.
>+  RenderDelayBuffer::BufferingEvent buffer_event =
>+      render_buffer_->PrepareCaptureProcessing();
>+  // Reset the delay controller at render buffer underrun.
>+  if (buffer_event == RenderDelayBuffer::BufferingEvent::kRenderUnderrun) {
>+    delay_controller_->Reset(false);
>+  }
>+
>+  data_dumper_->DumpWav("aec3_processblock_capture_input2", kBlockSize,
>+                        &(*capture_block)[0][0],
>+                        LowestBandRate(sample_rate_hz_), 1);
>+
>+  // Compute and and apply the render delay required to achieve proper signal
>+  // alignment.
>+  estimated_delay_ = delay_controller_->GetDelay(
>+      render_buffer_->GetDownsampledRenderBuffer(), render_buffer_->Delay(),
>+      echo_remover_delay_, (*capture_block)[0]);
>+
>+  if (estimated_delay_) {
>+    bool delay_change = render_buffer_->SetDelay(estimated_delay_->delay);
>+    if (delay_change) {
>+      RTC_LOG(LS_WARNING) << "Delay changed to " << estimated_delay_->delay
>+                          << " at block " << capture_call_counter_;
>+      echo_path_variability.delay_change =
>+          EchoPathVariability::DelayAdjustment::kNewDetectedDelay;
>+    }
>+  }
>+
>+  echo_path_variability.clock_drift = delay_controller_->HasClockdrift();
>+
>+  // Remove the echo from the capture signal.
>+  echo_remover_->ProcessCapture(
>+      echo_path_variability, capture_signal_saturation, estimated_delay_,
>+      render_buffer_->GetRenderBuffer(), capture_block);
>+
>+  // Check to see if a refined delay estimate has been obtained from the echo
>+  // remover.
>+  echo_remover_delay_ = echo_remover_->Delay();
>+
>+  // Update the metrics.
>+  metrics_.UpdateCapture(false);
>+}
>+
>+void BlockProcessorImpl2::BufferRender(
>+    const std::vector<std::vector<float>>& block) {
>+  RTC_DCHECK_EQ(NumBandsForRate(sample_rate_hz_), block.size());
>+  RTC_DCHECK_EQ(kBlockSize, block[0].size());
>+  data_dumper_->DumpRaw("aec3_processblock_call_order",
>+                        static_cast<int>(BlockProcessorApiCall::kRender));
>+  data_dumper_->DumpWav("aec3_processblock_render_input", kBlockSize,
>+                        &block[0][0], LowestBandRate(sample_rate_hz_), 1);
>+  data_dumper_->DumpWav("aec3_processblock_render_input2", kBlockSize,
>+                        &block[0][0], LowestBandRate(sample_rate_hz_), 1);
>+
>+  render_event_ = render_buffer_->Insert(block);
>+
>+  metrics_.UpdateRender(render_event_ !=
>+                        RenderDelayBuffer::BufferingEvent::kNone);
>+
>+  render_properly_started_ = true;
>+  delay_controller_->LogRenderCall();
>+}
>+
>+void BlockProcessorImpl2::UpdateEchoLeakageStatus(bool leakage_detected) {
>+  echo_remover_->UpdateEchoLeakageStatus(leakage_detected);
>+}
>+
>+void BlockProcessorImpl2::GetMetrics(EchoControl::Metrics* metrics) const {
>+  echo_remover_->GetMetrics(metrics);
>+  const int block_size_ms = sample_rate_hz_ == 8000 ? 8 : 4;
>+  absl::optional<size_t> delay = render_buffer_->Delay();
>+  metrics->delay_ms = delay ? static_cast<int>(*delay) * block_size_ms : 0;
>+}
>+
>+void BlockProcessorImpl2::SetAudioBufferDelay(size_t delay_ms) {
>+  render_buffer_->SetAudioBufferDelay(delay_ms);
>+}
>+
>+}  // namespace
>+
>+BlockProcessor* BlockProcessor::Create2(const EchoCanceller3Config& config,
>+                                        int sample_rate_hz) {
>+  std::unique_ptr<RenderDelayBuffer> render_buffer(
>+      RenderDelayBuffer::Create2(config, NumBandsForRate(sample_rate_hz)));
>+  std::unique_ptr<RenderDelayController> delay_controller(
>+      RenderDelayController::Create2(config, sample_rate_hz));
>+  std::unique_ptr<EchoRemover> echo_remover(
>+      EchoRemover::Create(config, sample_rate_hz));
>+  return Create2(config, sample_rate_hz, std::move(render_buffer),
>+                 std::move(delay_controller), std::move(echo_remover));
>+}
>+
>+BlockProcessor* BlockProcessor::Create2(
>+    const EchoCanceller3Config& config,
>+    int sample_rate_hz,
>+    std::unique_ptr<RenderDelayBuffer> render_buffer) {
>+  std::unique_ptr<RenderDelayController> delay_controller(
>+      RenderDelayController::Create2(config, sample_rate_hz));
>+  std::unique_ptr<EchoRemover> echo_remover(
>+      EchoRemover::Create(config, sample_rate_hz));
>+  return Create2(config, sample_rate_hz, std::move(render_buffer),
>+                 std::move(delay_controller), std::move(echo_remover));
>+}
>+
>+BlockProcessor* BlockProcessor::Create2(
>+    const EchoCanceller3Config& config,
>+    int sample_rate_hz,
>+    std::unique_ptr<RenderDelayBuffer> render_buffer,
>+    std::unique_ptr<RenderDelayController> delay_controller,
>+    std::unique_ptr<EchoRemover> echo_remover) {
>+  return new BlockProcessorImpl2(
>+      config, sample_rate_hz, std::move(render_buffer),
>+      std::move(delay_controller), std::move(echo_remover));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_metrics.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_metrics.cc
>index c8bdda73898ec0601d5669761f77b515141f7ac1..deac1fcd221a05e420e51eade373b77b90e02046 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_metrics.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_metrics.cc
>@@ -11,6 +11,7 @@
> #include "modules/audio_processing/aec3/block_processor_metrics.h"
> 
> #include "modules/audio_processing/aec3/aec3_common.h"
>+#include "rtc_base/checks.h"
> #include "system_wrappers/include/metrics.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_unittest.cc
>index 71457868b24adbdbcd621bcdaad3f323b8fda986..8aba5b5661e5d594f6f65a27860e5a5192bffdda 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_unittest.cc
>@@ -37,7 +37,7 @@ using testing::_;
> // methods are callable.
> void RunBasicSetupAndApiCallTest(int sample_rate_hz, int num_iterations) {
>   std::unique_ptr<BlockProcessor> block_processor(
>-      BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz));
>+      BlockProcessor::Create2(EchoCanceller3Config(), sample_rate_hz));
>   std::vector<std::vector<float>> block(NumBandsForRate(sample_rate_hz),
>                                         std::vector<float>(kBlockSize, 1000.f));
> 
>@@ -51,7 +51,7 @@ void RunBasicSetupAndApiCallTest(int sample_rate_hz, int num_iterations) {
> #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
> void RunRenderBlockSizeVerificationTest(int sample_rate_hz) {
>   std::unique_ptr<BlockProcessor> block_processor(
>-      BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz));
>+      BlockProcessor::Create2(EchoCanceller3Config(), sample_rate_hz));
>   std::vector<std::vector<float>> block(
>       NumBandsForRate(sample_rate_hz), std::vector<float>(kBlockSize - 1, 0.f));
> 
>@@ -60,7 +60,7 @@ void RunRenderBlockSizeVerificationTest(int sample_rate_hz) {
> 
> void RunCaptureBlockSizeVerificationTest(int sample_rate_hz) {
>   std::unique_ptr<BlockProcessor> block_processor(
>-      BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz));
>+      BlockProcessor::Create2(EchoCanceller3Config(), sample_rate_hz));
>   std::vector<std::vector<float>> block(
>       NumBandsForRate(sample_rate_hz), std::vector<float>(kBlockSize - 1, 0.f));
> 
>@@ -72,7 +72,7 @@ void RunRenderNumBandsVerificationTest(int sample_rate_hz) {
>                                      ? NumBandsForRate(sample_rate_hz) + 1
>                                      : 1;
>   std::unique_ptr<BlockProcessor> block_processor(
>-      BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz));
>+      BlockProcessor::Create2(EchoCanceller3Config(), sample_rate_hz));
>   std::vector<std::vector<float>> block(wrong_num_bands,
>                                         std::vector<float>(kBlockSize, 0.f));
> 
>@@ -84,7 +84,7 @@ void RunCaptureNumBandsVerificationTest(int sample_rate_hz) {
>                                      ? NumBandsForRate(sample_rate_hz) + 1
>                                      : 1;
>   std::unique_ptr<BlockProcessor> block_processor(
>-      BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz));
>+      BlockProcessor::Create2(EchoCanceller3Config(), sample_rate_hz));
>   std::vector<std::vector<float>> block(wrong_num_bands,
>                                         std::vector<float>(kBlockSize, 0.f));
> 
>@@ -124,7 +124,7 @@ TEST(BlockProcessor, DISABLED_DelayControllerIntegration) {
>     EXPECT_CALL(*render_delay_buffer_mock, Delay())
>         .Times(kNumBlocks + 1)
>         .WillRepeatedly(Return(0));
>-    std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
>+    std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create2(
>         EchoCanceller3Config(), rate, std::move(render_delay_buffer_mock)));
> 
>     std::vector<std::vector<float>> render_block(
>@@ -173,7 +173,7 @@ TEST(BlockProcessor, DISABLED_SubmoduleIntegration) {
>     EXPECT_CALL(*echo_remover_mock, UpdateEchoLeakageStatus(_))
>         .Times(kNumBlocks);
> 
>-    std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create(
>+    std::unique_ptr<BlockProcessor> block_processor(BlockProcessor::Create2(
>         EchoCanceller3Config(), rate, std::move(render_delay_buffer_mock),
>         std::move(render_delay_controller_mock), std::move(echo_remover_mock)));
> 
>@@ -239,7 +239,7 @@ TEST(BlockProcessor, VerifyCaptureNumBandsCheck) {
> // Verifiers that the verification for null ProcessCapture input works.
> TEST(BlockProcessor, NullProcessCaptureParameter) {
>   EXPECT_DEATH(std::unique_ptr<BlockProcessor>(
>-                   BlockProcessor::Create(EchoCanceller3Config(), 8000))
>+                   BlockProcessor::Create2(EchoCanceller3Config(), 8000))
>                    ->ProcessCapture(false, false, nullptr),
>                "");
> }
>@@ -249,7 +249,7 @@ TEST(BlockProcessor, NullProcessCaptureParameter) {
> // tests on test bots has been fixed.
> TEST(BlockProcessor, DISABLED_WrongSampleRate) {
>   EXPECT_DEATH(std::unique_ptr<BlockProcessor>(
>-                   BlockProcessor::Create(EchoCanceller3Config(), 8001)),
>+                   BlockProcessor::Create2(EchoCanceller3Config(), 8001)),
>                "");
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.cc
>index 333d2267a418cfb1419a6a2a6bfe8bb55463dfde..5dfd7c54e2b4e4cf77a2600839c38272d073a93e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.cc
>@@ -9,6 +9,8 @@
>  */
> #include "modules/audio_processing/aec3/cascaded_biquad_filter.h"
> 
>+#include <algorithm>
>+
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h
>index 1e09fa6e5a4204761bc2411cc9b3a87a8aed9948..2a3b6d65c5a0f7510c060ff08eb5ab0005f00575 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_CASCADED_BIQUAD_FILTER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_CASCADED_BIQUAD_FILTER_H_
> 
>+#include <stddef.h>
> #include <complex>
> #include <vector>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/clockdrift_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/clockdrift_detector.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..2c49b795c43da5363fe286715f61b4837123c4d7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/clockdrift_detector.cc
>@@ -0,0 +1,61 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "modules/audio_processing/aec3/clockdrift_detector.h"
>+
>+namespace webrtc {
>+
>+ClockdriftDetector::ClockdriftDetector()
>+    : level_(Level::kNone), stability_counter_(0) {
>+  delay_history_.fill(0);
>+}
>+
>+ClockdriftDetector::~ClockdriftDetector() = default;
>+
>+void ClockdriftDetector::Update(int delay_estimate) {
>+  if (delay_estimate == delay_history_[0]) {
>+    // Reset clockdrift level if delay estimate is stable for 7500 blocks (30
>+    // seconds).
>+    if (++stability_counter_ > 7500)
>+      level_ = Level::kNone;
>+    return;
>+  }
>+
>+  stability_counter_ = 0;
>+  const int d1 = delay_history_[0] - delay_estimate;
>+  const int d2 = delay_history_[1] - delay_estimate;
>+  const int d3 = delay_history_[2] - delay_estimate;
>+
>+  // Patterns recognized as positive clockdrift:
>+  // [x-3], x-2, x-1, x.
>+  // [x-3], x-1, x-2, x.
>+  const bool probable_drift_up =
>+      (d1 == -1 && d2 == -2) || (d1 == -2 && d2 == -1);
>+  const bool drift_up = probable_drift_up && d3 == -3;
>+
>+  // Patterns recognized as negative clockdrift:
>+  // [x+3], x+2, x+1, x.
>+  // [x+3], x+1, x+2, x.
>+  const bool probable_drift_down = (d1 == 1 && d2 == 2) || (d1 == 2 && d2 == 1);
>+  const bool drift_down = probable_drift_down && d3 == 3;
>+
>+  // Set clockdrift level.
>+  if (drift_up || drift_down) {
>+    level_ = Level::kVerified;
>+  } else if ((probable_drift_up || probable_drift_down) &&
>+             level_ == Level::kNone) {
>+    level_ = Level::kProbable;
>+  }
>+
>+  // Shift delay history one step.
>+  delay_history_[2] = delay_history_[1];
>+  delay_history_[1] = delay_history_[0];
>+  delay_history_[0] = delay_estimate;
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/clockdrift_detector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/clockdrift_detector.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..22528c948922a80cdb94fc364a40a78a8abb2fe7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/clockdrift_detector.h
>@@ -0,0 +1,38 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_CLOCKDRIFT_DETECTOR_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_CLOCKDRIFT_DETECTOR_H_
>+
>+#include <array>
>+
>+namespace webrtc {
>+
>+class ApmDataDumper;
>+struct DownsampledRenderBuffer;
>+struct EchoCanceller3Config;
>+
>+// Detects clockdrift by analyzing the estimated delay.
>+class ClockdriftDetector {
>+ public:
>+  enum class Level { kNone, kProbable, kVerified, kNumCategories };
>+  ClockdriftDetector();
>+  ~ClockdriftDetector();
>+  void Update(int delay_estimate);
>+  Level ClockdriftLevel() const { return level_; }
>+
>+ private:
>+  std::array<int, 3> delay_history_;
>+  Level level_;
>+  size_t stability_counter_;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_CLOCKDRIFT_DETECTOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/clockdrift_detector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/clockdrift_detector_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..0f98b01d3aae3ebe167c9d9a4528ab43f8065d7f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/clockdrift_detector_unittest.cc
>@@ -0,0 +1,57 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/clockdrift_detector.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+TEST(ClockdriftDetector, ClockdriftDetector) {
>+  ClockdriftDetector c;
>+  // No clockdrift at start.
>+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kNone);
>+
>+  // Monotonically increasing delay.
>+  for (int i = 0; i < 100; i++)
>+    c.Update(1000);
>+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kNone);
>+  for (int i = 0; i < 100; i++)
>+    c.Update(1001);
>+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kNone);
>+  for (int i = 0; i < 100; i++)
>+    c.Update(1002);
>+  // Probable clockdrift.
>+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kProbable);
>+  for (int i = 0; i < 100; i++)
>+    c.Update(1003);
>+  // Verified clockdrift.
>+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kVerified);
>+
>+  // Stable delay.
>+  for (int i = 0; i < 10000; i++)
>+    c.Update(1003);
>+  // No clockdrift.
>+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kNone);
>+
>+  // Decreasing delay.
>+  for (int i = 0; i < 100; i++)
>+    c.Update(1001);
>+  for (int i = 0; i < 100; i++)
>+    c.Update(999);
>+  // Probable clockdrift.
>+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kProbable);
>+  for (int i = 0; i < 100; i++)
>+    c.Update(1000);
>+  for (int i = 0; i < 100; i++)
>+    c.Update(998);
>+  // Verified clockdrift.
>+  EXPECT_TRUE(c.ClockdriftLevel() == ClockdriftDetector::Level::kVerified);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.cc
>index 55faf4b5291a175bbe4bdab351614cd07beecf51..766e65863981cafba09dd51085def6f26c1f89f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.cc
>@@ -16,13 +16,16 @@
> #if defined(WEBRTC_ARCH_X86_FAMILY)
> #include <emmintrin.h>
> #endif
>-#include <math.h>
> #include <algorithm>
> #include <array>
>+#include <cmath>
>+#include <cstdint>
> #include <functional>
> #include <numeric>
> 
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "modules/audio_processing/aec3/vector_math.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>@@ -38,64 +41,10 @@ void TableRandomValue(int16_t* vector, int16_t vector_length, uint32_t* seed) {
> 
> }  // namespace
> 
>-namespace aec3 {
>-
>-#if defined(WEBRTC_ARCH_X86_FAMILY)
>-
>-void EstimateComfortNoise_SSE2(const std::array<float, kFftLengthBy2Plus1>& N2,
>-                               uint32_t* seed,
>-                               FftData* lower_band_noise,
>-                               FftData* upper_band_noise) {
>-  FftData* N_low = lower_band_noise;
>-  FftData* N_high = upper_band_noise;
>-
>-  // Compute square root spectrum.
>-  std::array<float, kFftLengthBy2Plus1> N;
>-  for (size_t k = 0; k < kFftLengthBy2; k += 4) {
>-    __m128 v = _mm_loadu_ps(&N2[k]);
>-    v = _mm_sqrt_ps(v);
>-    _mm_storeu_ps(&N[k], v);
>-  }
>-
>-  N[kFftLengthBy2] = sqrtf(N2[kFftLengthBy2]);
>-
>-  // Compute the noise level for the upper bands.
>-  constexpr float kOneByNumBands = 1.f / (kFftLengthBy2Plus1 / 2 + 1);
>-  constexpr int kFftLengthBy2Plus1By2 = kFftLengthBy2Plus1 / 2;
>-  const float high_band_noise_level =
>-      std::accumulate(N.begin() + kFftLengthBy2Plus1By2, N.end(), 0.f) *
>-      kOneByNumBands;
>-
>-  // Generate complex noise.
>-  std::array<int16_t, kFftLengthBy2 - 1> random_values_int;
>-  TableRandomValue(random_values_int.data(), random_values_int.size(), seed);
>-
>-  std::array<float, kFftLengthBy2 - 1> sin;
>-  std::array<float, kFftLengthBy2 - 1> cos;
>-  constexpr float kScale = 6.28318530717959f / 32768.0f;
>-  std::transform(random_values_int.begin(), random_values_int.end(),
>-                 sin.begin(), [&](int16_t a) { return -sinf(kScale * a); });
>-  std::transform(random_values_int.begin(), random_values_int.end(),
>-                 cos.begin(), [&](int16_t a) { return cosf(kScale * a); });
>-
>-  // Form low-frequency noise via spectral shaping.
>-  N_low->re[0] = N_low->re[kFftLengthBy2] = N_high->re[0] =
>-      N_high->re[kFftLengthBy2] = 0.f;
>-  std::transform(cos.begin(), cos.end(), N.begin() + 1, N_low->re.begin() + 1,
>-                 std::multiplies<float>());
>-  std::transform(sin.begin(), sin.end(), N.begin() + 1, N_low->im.begin() + 1,
>-                 std::multiplies<float>());
>-
>-  // Form the high-frequency noise via simple levelling.
>-  std::transform(cos.begin(), cos.end(), N_high->re.begin() + 1,
>-                 [&](float a) { return high_band_noise_level * a; });
>-  std::transform(sin.begin(), sin.end(), N_high->im.begin() + 1,
>-                 [&](float a) { return high_band_noise_level * a; });
>-}
>-
>-#endif
>+namespace {
> 
>-void EstimateComfortNoise(const std::array<float, kFftLengthBy2Plus1>& N2,
>+void GenerateComfortNoise(Aec3Optimization optimization,
>+                          const std::array<float, kFftLengthBy2Plus1>& N2,
>                           uint32_t* seed,
>                           FftData* lower_band_noise,
>                           FftData* upper_band_noise) {
>@@ -104,8 +53,8 @@ void EstimateComfortNoise(const std::array<float, kFftLengthBy2Plus1>& N2,
> 
>   // Compute square root spectrum.
>   std::array<float, kFftLengthBy2Plus1> N;
>-  std::transform(N2.begin(), N2.end(), N.begin(),
>-                 [](float a) { return sqrtf(a); });
>+  std::copy(N2.begin(), N2.end(), N.begin());
>+  aec3::VectorMath(optimization).Sqrt(N);
> 
>   // Compute the noise level for the upper bands.
>   constexpr float kOneByNumBands = 1.f / (kFftLengthBy2Plus1 / 2 + 1);
>@@ -118,13 +67,21 @@ void EstimateComfortNoise(const std::array<float, kFftLengthBy2Plus1>& N2,
>   std::array<int16_t, kFftLengthBy2 - 1> random_values_int;
>   TableRandomValue(random_values_int.data(), random_values_int.size(), seed);
> 
>+  // The analysis and synthesis windowing cause loss of power when
>+  // cross-fading the noise where frames are completely uncorrelated
>+  // (generated with random phase), hence the factor sqrt(2).
>+  // This is not the case for the speech signal where the input is overlapping
>+  // (strong correlation).
>   std::array<float, kFftLengthBy2 - 1> sin;
>   std::array<float, kFftLengthBy2 - 1> cos;
>   constexpr float kScale = 6.28318530717959f / 32768.0f;
>+  constexpr float kSqrt2 = 1.4142135623f;
>   std::transform(random_values_int.begin(), random_values_int.end(),
>-                 sin.begin(), [&](int16_t a) { return -sinf(kScale * a); });
>+                 sin.begin(),
>+                 [&](int16_t a) { return -sinf(kScale * a) * kSqrt2; });
>   std::transform(random_values_int.begin(), random_values_int.end(),
>-                 cos.begin(), [&](int16_t a) { return cosf(kScale * a); });
>+                 cos.begin(),
>+                 [&](int16_t a) { return cosf(kScale * a) * kSqrt2; });
> 
>   // Form low-frequency noise via spectral shaping.
>   N_low->re[0] = N_low->re[kFftLengthBy2] = N_high->re[0] =
>@@ -141,7 +98,7 @@ void EstimateComfortNoise(const std::array<float, kFftLengthBy2Plus1>& N2,
>                  [&](float a) { return high_band_noise_level * a; });
> }
> 
>-}  // namespace aec3
>+}  // namespace
> 
> ComfortNoiseGenerator::ComfortNoiseGenerator(Aec3Optimization optimization)
>     : optimization_(optimization),
>@@ -205,17 +162,8 @@ void ComfortNoiseGenerator::Compute(
>   const std::array<float, kFftLengthBy2Plus1>& N2 =
>       N2_initial_ ? *N2_initial_ : N2_;
> 
>-  switch (optimization_) {
>-#if defined(WEBRTC_ARCH_X86_FAMILY)
>-    case Aec3Optimization::kSse2:
>-      aec3::EstimateComfortNoise_SSE2(N2, &seed_, lower_band_noise,
>-                                      upper_band_noise);
>-      break;
>-#endif
>-    default:
>-      aec3::EstimateComfortNoise(N2, &seed_, lower_band_noise,
>-                                 upper_band_noise);
>-  }
>+  GenerateComfortNoise(optimization_, N2, &seed_, lower_band_noise,
>+                       upper_band_noise);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.h
>index 6a479896ecb976fb30abd3a3a69719c47ac164b9..3be386bab1e7687cc0f4af25cb8e47c2de15333f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_COMFORT_NOISE_GENERATOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_COMFORT_NOISE_GENERATOR_H_
> 
>+#include <stdint.h>
> #include <array>
> #include <memory>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc
>index 77a09edcd6a038efdfb88f9d5356d8de19e87e44..10ba69603681fd30b5edcf13104e782eab53c074 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc
>@@ -52,45 +52,6 @@ TEST(ComfortNoiseGenerator, NullUpperBandNoise) {
> 
> #endif
> 
>-#if defined(WEBRTC_ARCH_X86_FAMILY)
>-// Verifies that the optimized methods are bitexact to their reference
>-// counterparts.
>-TEST(ComfortNoiseGenerator, TestOptimizations) {
>-  if (WebRtc_GetCPUInfo(kSSE2) != 0) {
>-    Random random_generator(42U);
>-    uint32_t seed = 42;
>-    uint32_t seed_SSE2 = 42;
>-    std::array<float, kFftLengthBy2Plus1> N2;
>-    FftData lower_band_noise;
>-    FftData upper_band_noise;
>-    FftData lower_band_noise_SSE2;
>-    FftData upper_band_noise_SSE2;
>-    for (int k = 0; k < 10; ++k) {
>-      for (size_t j = 0; j < N2.size(); ++j) {
>-        N2[j] = random_generator.Rand<float>() * 1000.f;
>-      }
>-
>-      EstimateComfortNoise(N2, &seed, &lower_band_noise, &upper_band_noise);
>-      EstimateComfortNoise_SSE2(N2, &seed_SSE2, &lower_band_noise_SSE2,
>-                                &upper_band_noise_SSE2);
>-      for (size_t j = 0; j < lower_band_noise.re.size(); ++j) {
>-        EXPECT_NEAR(lower_band_noise.re[j], lower_band_noise_SSE2.re[j],
>-                    0.00001f);
>-        EXPECT_NEAR(upper_band_noise.re[j], upper_band_noise_SSE2.re[j],
>-                    0.00001f);
>-      }
>-      for (size_t j = 1; j < lower_band_noise.re.size() - 1; ++j) {
>-        EXPECT_NEAR(lower_band_noise.im[j], lower_band_noise_SSE2.im[j],
>-                    0.00001f);
>-        EXPECT_NEAR(upper_band_noise.im[j], upper_band_noise_SSE2.im[j],
>-                    0.00001f);
>-      }
>-    }
>-  }
>-}
>-
>-#endif
>-
> TEST(ComfortNoiseGenerator, CorrectLevel) {
>   ComfortNoiseGenerator cng(DetectOptimization());
>   AecState aec_state(EchoCanceller3Config{});
>@@ -113,8 +74,8 @@ TEST(ComfortNoiseGenerator, CorrectLevel) {
>   for (int k = 0; k < 10000; ++k) {
>     cng.Compute(aec_state, N2, &n_lower, &n_upper);
>   }
>-  EXPECT_NEAR(N2[0], Power(n_lower), N2[0] / 10.f);
>-  EXPECT_NEAR(N2[0], Power(n_upper), N2[0] / 10.f);
>+  EXPECT_NEAR(2.f * N2[0], Power(n_lower), N2[0] / 10.f);
>+  EXPECT_NEAR(2.f * N2[0], Power(n_upper), N2[0] / 10.f);
> }
> 
> }  // namespace aec3
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.cc
>index d9faa62dc446976a13d3ea06f0b3294ddf3acd67..bd03237ca080fb5341452a1faa39e160edeeee07 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.cc
>@@ -9,6 +9,10 @@
>  */
> #include "modules/audio_processing/aec3/decimator.h"
> 
>+#include <array>
>+#include <vector>
>+
>+#include "modules/audio_processing/aec3/aec3_common.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc
>index df0af6e0d8600160d2b574a969c3883aa9f59081..c105911aa8834df2049d281c84577fdfa558ec23 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/audio_processing/aec3/downsampled_render_buffer.h"
> 
>+#include <algorithm>
>+
> namespace webrtc {
> 
> DownsampledRenderBuffer::DownsampledRenderBuffer(size_t downsampled_buffer_size)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h
>index 943949666a069f0e447ba93257804dcdaed8dccd..c91ea3b836f60e85273b94abb68108efbf452746 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h
>@@ -11,9 +11,9 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
> 
>+#include <stddef.h>
> #include <vector>
> 
>-#include "modules/audio_processing/aec3/aec3_common.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.cc
>index fa123de351e5449daaaba66f899a8e79a2108eea..e857a7e45a08476f028c7cfe3804f8f8d328763d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.cc
>@@ -12,8 +12,10 @@
> 
> #include <algorithm>
> #include <cmath>
>+#include <utility>
>+#include <vector>
> 
>-#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "api/array_view.h"
> #include "modules/audio_processing/aec3/matrix_buffer.h"
> #include "modules/audio_processing/aec3/stationarity_estimator.h"
> #include "modules/audio_processing/aec3/vector_buffer.h"
>@@ -27,16 +29,18 @@ EchoAudibility::EchoAudibility(bool use_render_stationarity_at_init)
> 
> EchoAudibility::~EchoAudibility() = default;
> 
>-void EchoAudibility::Update(const RenderBuffer& render_buffer,
>-                            int delay_blocks,
>-                            bool external_delay_seen,
>-                            float reverb_decay) {
>+void EchoAudibility::Update(
>+    const RenderBuffer& render_buffer,
>+    rtc::ArrayView<const float> render_reverb_contribution_spectrum,
>+    int delay_blocks,
>+    bool external_delay_seen) {
>   UpdateRenderNoiseEstimator(render_buffer.GetSpectrumBuffer(),
>                              render_buffer.GetBlockBuffer(),
>                              external_delay_seen);
> 
>   if (external_delay_seen || use_render_stationarity_at_init_) {
>-    UpdateRenderStationarityFlags(render_buffer, delay_blocks, reverb_decay);
>+    UpdateRenderStationarityFlags(
>+        render_buffer, render_reverb_contribution_spectrum, delay_blocks);
>   }
> }
> 
>@@ -48,8 +52,8 @@ void EchoAudibility::Reset() {
> 
> void EchoAudibility::UpdateRenderStationarityFlags(
>     const RenderBuffer& render_buffer,
>-    int delay_blocks,
>-    float reverb_decay) {
>+    rtc::ArrayView<const float> render_reverb_contribution_spectrum,
>+    int delay_blocks) {
>   const VectorBuffer& spectrum_buffer = render_buffer.GetSpectrumBuffer();
>   int idx_at_delay =
>       spectrum_buffer.OffsetIndex(spectrum_buffer.read, delay_blocks);
>@@ -57,8 +61,9 @@ void EchoAudibility::UpdateRenderStationarityFlags(
>   int num_lookahead = render_buffer.Headroom() - delay_blocks + 1;
>   num_lookahead = std::max(0, num_lookahead);
> 
>-  render_stationarity_.UpdateStationarityFlags(spectrum_buffer, idx_at_delay,
>-                                               num_lookahead, reverb_decay);
>+  render_stationarity_.UpdateStationarityFlags(
>+      spectrum_buffer, render_reverb_contribution_spectrum, idx_at_delay,
>+      num_lookahead);
> }
> 
> void EchoAudibility::UpdateRenderNoiseEstimator(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.h
>index 3b252f3ea2b7e5cbd9065376c2769887a76298b7..b903ca0cf1d79efba8e4b7485392f7500f755221 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.h
>@@ -11,11 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_AUDIBILITY_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_ECHO_AUDIBILITY_H_
> 
>-#include <algorithm>
>-#include <array>
>-#include <limits>
>-#include <memory>
>-#include <vector>
>+#include <stddef.h>
> 
> #include "absl/types/optional.h"
> #include "api/array_view.h"
>@@ -27,8 +23,6 @@
> 
> namespace webrtc {
> 
>-class ApmDataDumper;
>-
> class EchoAudibility {
>  public:
>   explicit EchoAudibility(bool use_render_stationarity_at_init);
>@@ -36,9 +30,9 @@ class EchoAudibility {
> 
>   // Feed new render data to the echo audibility estimator.
>   void Update(const RenderBuffer& render_buffer,
>+              rtc::ArrayView<const float> render_reverb_contribution_spectrum,
>               int delay_blocks,
>-              bool external_delay_seen,
>-              float reverb_decay);
>+              bool external_delay_seen);
>   // Get the residual echo scaling.
>   void GetResidualEchoScaling(bool filter_has_had_time_to_converge,
>                               rtc::ArrayView<float> residual_scaling) const {
>@@ -62,9 +56,10 @@ class EchoAudibility {
>   void Reset();
> 
>   // Updates the render stationarity flags for the current frame.
>-  void UpdateRenderStationarityFlags(const RenderBuffer& render_buffer,
>-                                     int delay_blocks,
>-                                     float reverb_decay);
>+  void UpdateRenderStationarityFlags(
>+      const RenderBuffer& render_buffer,
>+      rtc::ArrayView<const float> render_reverb_contribution_spectrum,
>+      int delay_blocks);
> 
>   // Updates the noise estimator with the new render data since the previous
>   // call to this method.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.cc
>index e5219c72ab7fae9a9e0bfe2e8e5ff453076c9740..f05edb15c3e56af98ca6bfca90262316f86cb39b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.cc
>@@ -9,9 +9,12 @@
>  */
> #include "modules/audio_processing/aec3/echo_canceller3.h"
> 
>+#include <algorithm>
>+#include <utility>
>+
>+#include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
>-#include "rtc_base/logging.h"
> #include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
>@@ -41,25 +44,44 @@ bool EnableReverbModelling() {
>   return !field_trial::IsEnabled("WebRTC-Aec3ReverbModellingKillSwitch");
> }
> 
>-bool EnableSuppressorNearendAveraging() {
>-  return !field_trial::IsEnabled(
>-      "WebRTC-Aec3SuppressorNearendAveragingKillSwitch");
>+bool EnableUnityInitialRampupGain() {
>+  return field_trial::IsEnabled("WebRTC-Aec3EnableUnityInitialRampupGain");
> }
> 
>-bool EnableSlowFilterAdaptation() {
>-  return !field_trial::IsEnabled("WebRTC-Aec3SlowFilterAdaptationKillSwitch");
>+bool EnableUnityNonZeroRampupGain() {
>+  return field_trial::IsEnabled("WebRTC-Aec3EnableUnityNonZeroRampupGain");
> }
> 
>-bool EnableShadowFilterJumpstart() {
>-  return !field_trial::IsEnabled("WebRTC-Aec3ShadowFilterJumpstartKillSwitch");
>+bool EnableLongReverb() {
>+  return field_trial::IsEnabled("WebRTC-Aec3ShortReverbKillSwitch");
> }
> 
>-bool EnableUnityInitialRampupGain() {
>-  return field_trial::IsEnabled("WebRTC-Aec3EnableUnityInitialRampupGain");
>+bool EnableNewFilterParams() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3NewFilterParamsKillSwitch");
> }
> 
>-bool EnableUnityNonZeroRampupGain() {
>-  return field_trial::IsEnabled("WebRTC-Aec3EnableUnityNonZeroRampupGain");
>+bool EnableLegacyDominantNearend() {
>+  return field_trial::IsEnabled("WebRTC-Aec3EnableLegacyDominantNearend");
>+}
>+
>+bool UseLegacyNormalSuppressorTuning() {
>+  return field_trial::IsEnabled("WebRTC-Aec3UseLegacyNormalSuppressorTuning");
>+}
>+
>+bool ActivateStationarityProperties() {
>+  return field_trial::IsEnabled("WebRTC-Aec3UseStationarityProperties");
>+}
>+
>+bool ActivateStationarityPropertiesAtInit() {
>+  return field_trial::IsEnabled("WebRTC-Aec3UseStationarityPropertiesAtInit");
>+}
>+
>+bool EnableNewRenderBuffering() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3NewRenderBufferingKillSwitch");
>+}
>+
>+bool UseEarlyDelayDetection() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3EarlyDelayDetectionKillSwitch");
> }
> 
> // Method for adjusting config parameter dependencies..
>@@ -71,68 +93,27 @@ EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
>     adjusted_cfg.ep_strength.default_len = 0.f;
>   }
> 
>-  // Use customized parameters when the system has clock-drift.
>-  if (config.echo_removal_control.has_clock_drift) {
>-    RTC_LOG(LS_WARNING)
>-        << "Customizing parameters to work well for the clock-drift case.";
>-    if (config.ep_strength.bounded_erl) {
>-      adjusted_cfg.ep_strength.default_len = 0.85f;
>-      adjusted_cfg.ep_strength.lf = 0.01f;
>-      adjusted_cfg.ep_strength.mf = 0.01f;
>-      adjusted_cfg.ep_strength.hf = 0.01f;
>-      adjusted_cfg.echo_model.render_pre_window_size = 1;
>-      adjusted_cfg.echo_model.render_post_window_size = 1;
>-      adjusted_cfg.echo_model.nonlinear_hold = 3;
>-      adjusted_cfg.echo_model.nonlinear_release = 0.001f;
>-    } else {
>-      adjusted_cfg.ep_strength.bounded_erl = true;
>-      adjusted_cfg.delay.down_sampling_factor = 2;
>-      adjusted_cfg.ep_strength.default_len = 0.8f;
>-      adjusted_cfg.ep_strength.lf = 0.01f;
>-      adjusted_cfg.ep_strength.mf = 0.01f;
>-      adjusted_cfg.ep_strength.hf = 0.01f;
>-      adjusted_cfg.filter.main = {30, 0.1f, 0.8f, 0.001f, 20075344.f};
>-      adjusted_cfg.filter.shadow = {30, 0.7f, 20075344.f};
>-      adjusted_cfg.filter.main_initial = {30, 0.1f, 1.5f, 0.001f, 20075344.f};
>-      adjusted_cfg.filter.shadow_initial = {30, 0.9f, 20075344.f};
>-      adjusted_cfg.echo_model.render_pre_window_size = 2;
>-      adjusted_cfg.echo_model.render_post_window_size = 2;
>-      adjusted_cfg.echo_model.nonlinear_hold = 3;
>-      adjusted_cfg.echo_model.nonlinear_release = 0.6f;
>-    }
>-  }
>-
>   if (UseShortDelayEstimatorWindow()) {
>     adjusted_cfg.delay.num_filters =
>         std::min(adjusted_cfg.delay.num_filters, static_cast<size_t>(5));
>   }
> 
>-  if (EnableReverbBasedOnRender() == false) {
>-    adjusted_cfg.ep_strength.reverb_based_on_render = false;
>+  bool use_new_render_buffering =
>+      EnableNewRenderBuffering() && config.buffering.use_new_render_buffering;
>+  // Old render buffering needs one more filter to cover the same delay.
>+  if (!use_new_render_buffering) {
>+    adjusted_cfg.delay.num_filters += 1;
>   }
> 
>-  if (!EnableSuppressorNearendAveraging()) {
>-    adjusted_cfg.suppressor.nearend_average_blocks = 1;
>+  if (EnableReverbBasedOnRender() == false) {
>+    adjusted_cfg.ep_strength.reverb_based_on_render = false;
>   }
> 
>-  if (!EnableSlowFilterAdaptation()) {
>-    if (!EnableShadowFilterJumpstart()) {
>-      adjusted_cfg.filter.main.leakage_converged = 0.005f;
>-      adjusted_cfg.filter.main.leakage_diverged = 0.1f;
>-    }
>-    adjusted_cfg.filter.main_initial.leakage_converged = 0.05f;
>-    adjusted_cfg.filter.main_initial.leakage_diverged = 5.f;
>-  }
>-
>-  if (!EnableShadowFilterJumpstart()) {
>-    if (EnableSlowFilterAdaptation()) {
>-      adjusted_cfg.filter.main.leakage_converged = 0.0005f;
>-      adjusted_cfg.filter.main.leakage_diverged = 0.01f;
>-    } else {
>-      adjusted_cfg.filter.main.leakage_converged = 0.005f;
>-      adjusted_cfg.filter.main.leakage_diverged = 0.1f;
>-    }
>-    adjusted_cfg.filter.main.error_floor = 0.001f;
>+  if (!EnableNewFilterParams()) {
>+    adjusted_cfg.filter.main.leakage_diverged = 0.01f;
>+    adjusted_cfg.filter.main.error_floor = 0.1f;
>+    adjusted_cfg.filter.main.error_ceil = 1E10f;
>+    adjusted_cfg.filter.main_initial.error_ceil = 1E10f;
>   }
> 
>   if (EnableUnityInitialRampupGain() &&
>@@ -147,6 +128,42 @@ EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
>     adjusted_cfg.echo_removal_control.gain_rampup.first_non_zero_gain = 1.f;
>   }
> 
>+  if (EnableLongReverb()) {
>+    adjusted_cfg.ep_strength.default_len = 0.88f;
>+  }
>+
>+  if (EnableLegacyDominantNearend()) {
>+    adjusted_cfg.suppressor.nearend_tuning =
>+        EchoCanceller3Config::Suppressor::Tuning(
>+            EchoCanceller3Config::Suppressor::MaskingThresholds(.2f, .3f, .3f),
>+            EchoCanceller3Config::Suppressor::MaskingThresholds(.07f, .1f, .3f),
>+            2.0f, 0.25f);
>+  }
>+
>+  if (UseLegacyNormalSuppressorTuning()) {
>+    adjusted_cfg.suppressor.normal_tuning =
>+        EchoCanceller3Config::Suppressor::Tuning(
>+            EchoCanceller3Config::Suppressor::MaskingThresholds(.2f, .3f, .3f),
>+            EchoCanceller3Config::Suppressor::MaskingThresholds(.07f, .1f, .3f),
>+            2.0f, 0.25f);
>+
>+    adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold = 10.f;
>+    adjusted_cfg.suppressor.dominant_nearend_detection.snr_threshold = 10.f;
>+    adjusted_cfg.suppressor.dominant_nearend_detection.hold_duration = 25;
>+  }
>+
>+  if (ActivateStationarityProperties()) {
>+    adjusted_cfg.echo_audibility.use_stationary_properties = true;
>+  }
>+
>+  if (ActivateStationarityPropertiesAtInit()) {
>+    adjusted_cfg.echo_audibility.use_stationarity_properties_at_init = true;
>+  }
>+
>+  if (!UseEarlyDelayDetection()) {
>+    adjusted_cfg.delay.delay_selection_thresholds = {25, 25};
>+  }
>+
>   return adjusted_cfg;
> }
> 
>@@ -330,12 +347,16 @@ int EchoCanceller3::instance_count_ = 0;
> EchoCanceller3::EchoCanceller3(const EchoCanceller3Config& config,
>                                int sample_rate_hz,
>                                bool use_highpass_filter)
>-    : EchoCanceller3(
>-          AdjustConfig(config),
>-          sample_rate_hz,
>-          use_highpass_filter,
>-          std::unique_ptr<BlockProcessor>(
>-              BlockProcessor::Create(AdjustConfig(config), sample_rate_hz))) {}
>+    : EchoCanceller3(AdjustConfig(config),
>+                     sample_rate_hz,
>+                     use_highpass_filter,
>+                     std::unique_ptr<BlockProcessor>(
>+                         EnableNewRenderBuffering() &&
>+                                 config.buffering.use_new_render_buffering
>+                             ? BlockProcessor::Create2(AdjustConfig(config),
>+                                                       sample_rate_hz)
>+                             : BlockProcessor::Create(AdjustConfig(config),
>+                                                      sample_rate_hz))) {}
> EchoCanceller3::EchoCanceller3(const EchoCanceller3Config& config,
>                                int sample_rate_hz,
>                                bool use_highpass_filter,
>@@ -425,6 +446,10 @@ void EchoCanceller3::ProcessCapture(AudioBuffer* capture, bool level_change) {
>   data_dumper_->DumpRaw("aec3_call_order",
>                         static_cast<int>(EchoCanceller3ApiCall::kCapture));
> 
>+  // Report capture call in the metrics and periodically update API call
>+  // metrics.
>+  api_call_metrics_.ReportCaptureCall();
>+
>   // Optionally delay the capture signal.
>   if (config_.delay.fixed_capture_delay_samples > 0) {
>     block_delay_buffer_.DelaySignal(capture);
>@@ -479,6 +504,9 @@ void EchoCanceller3::EmptyRenderQueue() {
>   bool frame_to_buffer =
>       render_transfer_queue_.Remove(&render_queue_output_frame_);
>   while (frame_to_buffer) {
>+    // Report render call in the metrics.
>+    api_call_metrics_.ReportRenderCall();
>+
>     BufferRenderFrameContent(&render_queue_output_frame_, 0, &render_blocker_,
>                              block_processor_.get(), &block_, &sub_frame_view_);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.h
>index f5520ba3ace5c3492a6b2a26896c883255f8855a..671d27167675c66c99c3e21e2572c6baf734e4f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.h
>@@ -11,7 +11,14 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_
> 
>+#include <stddef.h>
>+#include <memory>
>+#include <vector>
>+
>+#include "api/array_view.h"
> #include "api/audio/echo_canceller3_config.h"
>+#include "api/audio/echo_control.h"
>+#include "modules/audio_processing/aec3/api_call_jitter_metrics.h"
> #include "modules/audio_processing/aec3/block_delay_buffer.h"
> #include "modules/audio_processing/aec3/block_framer.h"
> #include "modules/audio_processing/aec3/block_processor.h"
>@@ -19,9 +26,11 @@
> #include "modules/audio_processing/aec3/frame_blocker.h"
> #include "modules/audio_processing/audio_buffer.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/race_checker.h"
> #include "rtc_base/swap_queue.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
> 
>@@ -132,6 +141,7 @@ class EchoCanceller3 : public EchoControl {
>   std::vector<rtc::ArrayView<float>> sub_frame_view_
>       RTC_GUARDED_BY(capture_race_checker_);
>   BlockDelayBuffer block_delay_buffer_ RTC_GUARDED_BY(capture_race_checker_);
>+  ApiCallJitterMetrics api_call_metrics_ RTC_GUARDED_BY(capture_race_checker_);
> 
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoCanceller3);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc
>index 638ddc44084e85f2ac670c9ef52023ece4f6a36f..6069ed6be697dbc8315170303763f6f3c5f98e95 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc
>@@ -9,31 +9,21 @@
>  */
> #include "modules/audio_processing/aec3/echo_path_delay_estimator.h"
> 
>-#include <algorithm>
> #include <array>
> 
> #include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
>-#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
>-namespace {
>-size_t GetDownSamplingFactor(const EchoCanceller3Config& config) {
>-  // Do not use down sampling factor 8 if kill switch is triggered.
>-  return (config.delay.down_sampling_factor == 8 &&
>-          field_trial::IsEnabled("WebRTC-Aec3DownSamplingFactor8KillSwitch"))
>-             ? 4
>-             : config.delay.down_sampling_factor;
>-}
>-}  // namespace
> 
> EchoPathDelayEstimator::EchoPathDelayEstimator(
>     ApmDataDumper* data_dumper,
>     const EchoCanceller3Config& config)
>     : data_dumper_(data_dumper),
>-      down_sampling_factor_(GetDownSamplingFactor(config)),
>+      down_sampling_factor_(config.delay.down_sampling_factor),
>       sub_block_size_(down_sampling_factor_ != 0
>                           ? kBlockSize / down_sampling_factor_
>                           : kBlockSize),
>@@ -45,7 +35,7 @@ EchoPathDelayEstimator::EchoPathDelayEstimator(
>           kMatchedFilterWindowSizeSubBlocks,
>           config.delay.num_filters,
>           kMatchedFilterAlignmentShiftSizeSubBlocks,
>-          GetDownSamplingFactor(config) == 8
>+          config.delay.down_sampling_factor == 8
>               ? config.render_levels.poor_excitation_render_limit_ds8
>               : config.render_levels.poor_excitation_render_limit,
>           config.delay.delay_estimate_smoothing,
>@@ -59,13 +49,8 @@ EchoPathDelayEstimator::EchoPathDelayEstimator(
> 
> EchoPathDelayEstimator::~EchoPathDelayEstimator() = default;
> 
>-void EchoPathDelayEstimator::Reset(bool soft_reset) {
>-  if (!soft_reset) {
>-    matched_filter_lag_aggregator_.Reset();
>-  }
>-  matched_filter_.Reset();
>-  old_aggregated_lag_ = absl::nullopt;
>-  consistent_estimate_counter_ = 0;
>+void EchoPathDelayEstimator::Reset(bool reset_delay_confidence) {
>+  Reset(true, reset_delay_confidence);
> }
> 
> absl::optional<DelayEstimate> EchoPathDelayEstimator::EstimateDelay(
>@@ -88,6 +73,12 @@ absl::optional<DelayEstimate> EchoPathDelayEstimator::EstimateDelay(
>       matched_filter_lag_aggregator_.Aggregate(
>           matched_filter_.GetLagEstimates());
> 
>+  // Run clockdrift detection.
>+  if (aggregated_matched_filter_lag &&
>+      (*aggregated_matched_filter_lag).quality ==
>+          DelayEstimate::Quality::kRefined)
>+    clockdrift_detector_.Update((*aggregated_matched_filter_lag).delay);
>+
>   // TODO(peah): Move this logging outside of this class once EchoCanceller3
>   // development is done.
>   data_dumper_->DumpRaw(
>@@ -112,10 +103,19 @@ absl::optional<DelayEstimate> EchoPathDelayEstimator::EstimateDelay(
>   old_aggregated_lag_ = aggregated_matched_filter_lag;
>   constexpr size_t kNumBlocksPerSecondBy2 = kNumBlocksPerSecond / 2;
>   if (consistent_estimate_counter_ > kNumBlocksPerSecondBy2) {
>-    Reset(true);
>+    Reset(false, false);
>   }
> 
>   return aggregated_matched_filter_lag;
> }
> 
>+void EchoPathDelayEstimator::Reset(bool reset_lag_aggregator,
>+                                   bool reset_delay_confidence) {
>+  if (reset_lag_aggregator) {
>+    matched_filter_lag_aggregator_.Reset(reset_delay_confidence);
>+  }
>+  matched_filter_.Reset();
>+  old_aggregated_lag_ = absl::nullopt;
>+  consistent_estimate_counter_ = 0;
>+}
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
>index cea9abcb515169cd7bc1fdd3f771e50394b65a70..1f14735d1a51afb0d4dfb3ae6866b92a8280872e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
>@@ -11,13 +11,13 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_ECHO_PATH_DELAY_ESTIMATOR_H_
> 
>-#include <vector>
>+#include <stddef.h>
> 
> #include "absl/types/optional.h"
>-#include "api/audio/echo_canceller3_config.h"
>+#include "api/array_view.h"
>+#include "modules/audio_processing/aec3/clockdrift_detector.h"
> #include "modules/audio_processing/aec3/decimator.h"
> #include "modules/audio_processing/aec3/delay_estimate.h"
>-#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
> #include "modules/audio_processing/aec3/matched_filter.h"
> #include "modules/audio_processing/aec3/matched_filter_lag_aggregator.h"
> #include "rtc_base/constructormagic.h"
>@@ -25,6 +25,8 @@
> namespace webrtc {
> 
> class ApmDataDumper;
>+struct DownsampledRenderBuffer;
>+struct EchoCanceller3Config;
> 
> // Estimates the delay of the echo path.
> class EchoPathDelayEstimator {
>@@ -33,9 +35,9 @@ class EchoPathDelayEstimator {
>                          const EchoCanceller3Config& config);
>   ~EchoPathDelayEstimator();
> 
>-  // Resets the estimation. If the soft-reset is specified, only  the matched
>-  // filters are reset.
>-  void Reset(bool soft_reset);
>+  // Resets the estimation. If the delay confidence is reset, the reset behavior
>+  // is as if the call is restarted.
>+  void Reset(bool reset_delay_confidence);
> 
>   // Produce a delay estimate if such is avaliable.
>   absl::optional<DelayEstimate> EstimateDelay(
>@@ -48,6 +50,11 @@ class EchoPathDelayEstimator {
>                                         down_sampling_factor_);
>   }
> 
>+  // Returns the level of detected clockdrift.
>+  ClockdriftDetector::Level Clockdrift() const {
>+    return clockdrift_detector_.ClockdriftLevel();
>+  }
>+
>  private:
>   ApmDataDumper* const data_dumper_;
>   const size_t down_sampling_factor_;
>@@ -57,6 +64,10 @@ class EchoPathDelayEstimator {
>   MatchedFilterLagAggregator matched_filter_lag_aggregator_;
>   absl::optional<DelayEstimate> old_aggregated_lag_;
>   size_t consistent_estimate_counter_ = 0;
>+  ClockdriftDetector clockdrift_detector_;
>+
>+  // Internal reset method with more granularity.
>+  void Reset(bool reset_lag_aggregator, bool reset_delay_confidence);
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(EchoPathDelayEstimator);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
>index 311a4a22d5f8ee1ec764771050dd29ea1f640e3e..a4e3133bb5baae18b365ef3758c4dbd611cf0563 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
>@@ -39,7 +39,7 @@ TEST(EchoPathDelayEstimator, BasicApiCalls) {
>   ApmDataDumper data_dumper(0);
>   EchoCanceller3Config config;
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
>   EchoPathDelayEstimator estimator(&data_dumper, config);
>   std::vector<std::vector<float>> render(3, std::vector<float>(kBlockSize));
>   std::vector<float> capture(kBlockSize);
>@@ -64,12 +64,9 @@ TEST(EchoPathDelayEstimator, DelayEstimation) {
>     config.delay.num_filters = 10;
>     for (size_t delay_samples : {30, 64, 150, 200, 800, 4000}) {
>       SCOPED_TRACE(ProduceDebugText(delay_samples, down_sampling_factor));
>-
>-      config.delay.api_call_jitter_blocks = 5;
>       std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-          RenderDelayBuffer::Create(config, 3));
>-      DelayBuffer<float> signal_delay_buffer(
>-          delay_samples + 2 * config.delay.api_call_jitter_blocks * 64);
>+          RenderDelayBuffer::Create2(config, 3));
>+      DelayBuffer<float> signal_delay_buffer(delay_samples);
>       EchoPathDelayEstimator estimator(&data_dumper, config);
> 
>       absl::optional<DelayEstimate> estimated_delay_samples;
>@@ -97,9 +94,7 @@ TEST(EchoPathDelayEstimator, DelayEstimation) {
>         // domain.
>         size_t delay_ds = delay_samples / down_sampling_factor;
>         size_t estimated_delay_ds =
>-            (estimated_delay_samples->delay -
>-             (config.delay.api_call_jitter_blocks + 1) * 64) /
>-            down_sampling_factor;
>+            estimated_delay_samples->delay / down_sampling_factor;
>         EXPECT_NEAR(delay_ds, estimated_delay_ds, 1);
>       } else {
>         ADD_FAILURE();
>@@ -118,7 +113,7 @@ TEST(EchoPathDelayEstimator, NoDelayEstimatesForLowLevelRenderSignals) {
>   ApmDataDumper data_dumper(0);
>   EchoPathDelayEstimator estimator(&data_dumper, config);
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(EchoCanceller3Config(), 3));
>+      RenderDelayBuffer::Create2(EchoCanceller3Config(), 3));
>   for (size_t k = 0; k < 100; ++k) {
>     RandomizeSampleVector(&random_generator, render[0]);
>     for (auto& render_k : render[0]) {
>@@ -142,7 +137,7 @@ TEST(EchoPathDelayEstimator, DISABLED_WrongRenderBlockSize) {
>   EchoCanceller3Config config;
>   EchoPathDelayEstimator estimator(&data_dumper, config);
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
>   std::vector<float> capture(kBlockSize);
>   EXPECT_DEATH(estimator.EstimateDelay(
>                    render_delay_buffer->GetDownsampledRenderBuffer(), capture),
>@@ -157,7 +152,7 @@ TEST(EchoPathDelayEstimator, WrongCaptureBlockSize) {
>   EchoCanceller3Config config;
>   EchoPathDelayEstimator estimator(&data_dumper, config);
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
>   std::vector<float> capture(std::vector<float>(kBlockSize - 1));
>   EXPECT_DEATH(estimator.EstimateDelay(
>                    render_delay_buffer->GetDownsampledRenderBuffer(), capture),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.cc
>index acdaf3104b9fae28099a02d1d6686c4e5f273156..cfb73951992cd9784d3328f5698bfb9ca7e243de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.cc
>@@ -10,26 +10,29 @@
> #include "modules/audio_processing/aec3/echo_remover.h"
> 
> #include <math.h>
>+#include <stddef.h>
> #include <algorithm>
>+#include <array>
> #include <memory>
>-#include <numeric>
>-#include <string>
> 
> #include "api/array_view.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/aec3_fft.h"
> #include "modules/audio_processing/aec3/aec_state.h"
> #include "modules/audio_processing/aec3/comfort_noise_generator.h"
> #include "modules/audio_processing/aec3/echo_path_variability.h"
> #include "modules/audio_processing/aec3/echo_remover_metrics.h"
> #include "modules/audio_processing/aec3/fft_data.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
>-#include "modules/audio_processing/aec3/render_delay_buffer.h"
>+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
> #include "modules/audio_processing/aec3/residual_echo_estimator.h"
> #include "modules/audio_processing/aec3/subtractor.h"
>+#include "modules/audio_processing/aec3/subtractor_output.h"
> #include "modules/audio_processing/aec3/suppression_filter.h"
> #include "modules/audio_processing/aec3/suppression_gain.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/field_trial.h"
>@@ -175,7 +178,7 @@ EchoRemoverImpl::EchoRemoverImpl(const EchoCanceller3Config& config,
>       subtractor_(config, data_dumper_.get(), optimization_),
>       suppression_gain_(config_, optimization_, sample_rate_hz),
>       cng_(optimization_),
>-      suppression_filter_(sample_rate_hz_),
>+      suppression_filter_(optimization_, sample_rate_hz_),
>       render_signal_analyzer_(config_),
>       residual_echo_estimator_(config_),
>       aec_state_(config_) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc
>index a04026b4f5f55bfb4e42a98b3063dd437707c366..da7a224e863a64630b8cea29801097f9dce69681 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc
>@@ -11,9 +11,11 @@
> #include "modules/audio_processing/aec3/echo_remover_metrics.h"
> 
> #include <math.h>
>+#include <stddef.h>
> #include <algorithm>
> #include <numeric>
> 
>+#include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_minmax.h"
> #include "system_wrappers/include/metrics.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.h
>index 17b803a1cf9888bd0b49b87c755cbcda3ef0133d..0707a5f031d9ae619b13d1f152e6902ab43fad76 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.h
>@@ -11,6 +11,9 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_METRICS_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_METRICS_H_
> 
>+#include <array>
>+
>+#include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/aec_state.h"
> #include "rtc_base/constructormagic.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_unittest.cc
>index da03f4c8e8db7df563a1a28d023a0a3333dea917..8bf76c4060d62045b888b764a54b08cb97c6d96e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_unittest.cc
>@@ -48,7 +48,7 @@ TEST(EchoRemover, BasicApiCalls) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>     std::unique_ptr<EchoRemover> remover(
>         EchoRemover::Create(EchoCanceller3Config(), rate));
>-    std::unique_ptr<RenderDelayBuffer> render_buffer(RenderDelayBuffer::Create(
>+    std::unique_ptr<RenderDelayBuffer> render_buffer(RenderDelayBuffer::Create2(
>         EchoCanceller3Config(), NumBandsForRate(rate)));
> 
>     std::vector<std::vector<float>> render(NumBandsForRate(rate),
>@@ -89,7 +89,7 @@ TEST(EchoRemover, WrongCaptureBlockSize) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>     std::unique_ptr<EchoRemover> remover(
>         EchoRemover::Create(EchoCanceller3Config(), rate));
>-    std::unique_ptr<RenderDelayBuffer> render_buffer(RenderDelayBuffer::Create(
>+    std::unique_ptr<RenderDelayBuffer> render_buffer(RenderDelayBuffer::Create2(
>         EchoCanceller3Config(), NumBandsForRate(rate)));
>     std::vector<std::vector<float>> capture(
>         NumBandsForRate(rate), std::vector<float>(kBlockSize - 1, 0.f));
>@@ -111,7 +111,7 @@ TEST(EchoRemover, DISABLED_WrongCaptureNumBands) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>     std::unique_ptr<EchoRemover> remover(
>         EchoRemover::Create(EchoCanceller3Config(), rate));
>-    std::unique_ptr<RenderDelayBuffer> render_buffer(RenderDelayBuffer::Create(
>+    std::unique_ptr<RenderDelayBuffer> render_buffer(RenderDelayBuffer::Create2(
>         EchoCanceller3Config(), NumBandsForRate(rate)));
>     std::vector<std::vector<float>> capture(
>         NumBandsForRate(rate == 48000 ? 16000 : rate + 16000),
>@@ -131,7 +131,7 @@ TEST(EchoRemover, NullCapture) {
>   std::unique_ptr<EchoRemover> remover(
>       EchoRemover::Create(EchoCanceller3Config(), 8000));
>   std::unique_ptr<RenderDelayBuffer> render_buffer(
>-      RenderDelayBuffer::Create(EchoCanceller3Config(), 3));
>+      RenderDelayBuffer::Create2(EchoCanceller3Config(), 3));
>   EchoPathVariability echo_path_variability(
>       false, EchoPathVariability::DelayAdjustment::kNone, false);
>   EXPECT_DEATH(
>@@ -161,7 +161,7 @@ TEST(EchoRemover, BasicEchoRemoval) {
>       config.delay.min_echo_path_delay_blocks = 0;
>       std::unique_ptr<EchoRemover> remover(EchoRemover::Create(config, rate));
>       std::unique_ptr<RenderDelayBuffer> render_buffer(
>-          RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>+          RenderDelayBuffer::Create2(config, NumBandsForRate(rate)));
>       render_buffer->SetDelay(delay_samples / kBlockSize);
> 
>       std::vector<std::unique_ptr<DelayBuffer<float>>> delay_buffers(x.size());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erl_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erl_estimator.cc
>index 2da9cd888350406bf78902806fe5649689e44e10..85b1e022da2f87266e0e8572a741cc84b11c86c1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erl_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erl_estimator.cc
>@@ -13,6 +13,8 @@
> #include <algorithm>
> #include <numeric>
> 
>+#include "rtc_base/checks.h"
>+
> namespace webrtc {
> 
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erl_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erl_estimator.h
>index dacd546a6de732f625b599bd6d04031c90b02331..29718c37836e8a8bd156279371d8272f00c2db34 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erl_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erl_estimator.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_ERL_ESTIMATOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_ERL_ESTIMATOR_H_
> 
>+#include <stddef.h>
> #include <array>
> 
> #include "api/array_view.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.cc
>index 711085f96f0894c7c5c29c86e722e977bb6b4f5d..656a9c7fdf530661e526f55c07de3b9ea680ed93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.cc
>@@ -10,19 +10,18 @@
> 
> #include "modules/audio_processing/aec3/erle_estimator.h"
> 
>-#include "api/array_view.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>-#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
> ErleEstimator::ErleEstimator(size_t startup_phase_length_blocks_,
>-                             float min_erle,
>-                             float max_erle_lf,
>-                             float max_erle_hf)
>+                             const EchoCanceller3Config& config)
>     : startup_phase_length_blocks__(startup_phase_length_blocks_),
>-      fullband_erle_estimator_(min_erle, max_erle_lf),
>-      subband_erle_estimator_(min_erle, max_erle_lf, max_erle_hf) {
>+      use_signal_dependent_erle_(config.erle.num_sections > 1),
>+      fullband_erle_estimator_(config.erle.min, config.erle.max_l),
>+      subband_erle_estimator_(config),
>+      signal_dependent_erle_estimator_(config) {
>   Reset(true);
> }
> 
>@@ -31,20 +30,25 @@ ErleEstimator::~ErleEstimator() = default;
> void ErleEstimator::Reset(bool delay_change) {
>   fullband_erle_estimator_.Reset();
>   subband_erle_estimator_.Reset();
>+  signal_dependent_erle_estimator_.Reset();
>   if (delay_change) {
>     blocks_since_reset_ = 0;
>   }
> }
> 
>-void ErleEstimator::Update(rtc::ArrayView<const float> render_spectrum,
>-                           rtc::ArrayView<const float> capture_spectrum,
>-                           rtc::ArrayView<const float> subtractor_spectrum,
>-                           bool converged_filter,
>-                           bool onset_detection) {
>-  RTC_DCHECK_EQ(kFftLengthBy2Plus1, render_spectrum.size());
>+void ErleEstimator::Update(
>+    const RenderBuffer& render_buffer,
>+    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+        filter_frequency_response,
>+    rtc::ArrayView<const float> reverb_render_spectrum,
>+    rtc::ArrayView<const float> capture_spectrum,
>+    rtc::ArrayView<const float> subtractor_spectrum,
>+    bool converged_filter,
>+    bool onset_detection) {
>+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, reverb_render_spectrum.size());
>   RTC_DCHECK_EQ(kFftLengthBy2Plus1, capture_spectrum.size());
>   RTC_DCHECK_EQ(kFftLengthBy2Plus1, subtractor_spectrum.size());
>-  const auto& X2 = render_spectrum;
>+  const auto& X2_reverb = reverb_render_spectrum;
>   const auto& Y2 = capture_spectrum;
>   const auto& E2 = subtractor_spectrum;
> 
>@@ -52,14 +56,23 @@ void ErleEstimator::Update(rtc::ArrayView<const float> render_spectrum,
>     return;
>   }
> 
>-  subband_erle_estimator_.Update(X2, Y2, E2, converged_filter, onset_detection);
>-  fullband_erle_estimator_.Update(X2, Y2, E2, converged_filter);
>+  subband_erle_estimator_.Update(X2_reverb, Y2, E2, converged_filter,
>+                                 onset_detection);
>+
>+  if (use_signal_dependent_erle_) {
>+    signal_dependent_erle_estimator_.Update(
>+        render_buffer, filter_frequency_response, X2_reverb, Y2, E2,
>+        subband_erle_estimator_.Erle(), converged_filter);
>+  }
>+
>+  fullband_erle_estimator_.Update(X2_reverb, Y2, E2, converged_filter);
> }
> 
> void ErleEstimator::Dump(
>     const std::unique_ptr<ApmDataDumper>& data_dumper) const {
>   fullband_erle_estimator_.Dump(data_dumper);
>   subband_erle_estimator_.Dump(data_dumper);
>+  signal_dependent_erle_estimator_.Dump(data_dumper);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.h
>index 8ca605ff71d58eb8679caa6f0d36ec0ea2566d3c..8036c2198b4e32bf4e8da1202685b9f5a50d4b2f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.h
>@@ -11,13 +11,17 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_ERLE_ESTIMATOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_ERLE_ESTIMATOR_H_
> 
>+#include <stddef.h>
> #include <array>
> #include <memory>
> 
> #include "absl/types/optional.h"
> #include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/fullband_erle_estimator.h"
>+#include "modules/audio_processing/aec3/render_buffer.h"
>+#include "modules/audio_processing/aec3/signal_dependent_erle_estimator.h"
> #include "modules/audio_processing/aec3/subband_erle_estimator.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> 
>@@ -28,16 +32,17 @@ namespace webrtc {
> class ErleEstimator {
>  public:
>   ErleEstimator(size_t startup_phase_length_blocks_,
>-                float min_erle,
>-                float max_erle_lf,
>-                float max_erle_hf);
>+                const EchoCanceller3Config& config);
>   ~ErleEstimator();
> 
>   // Resets the fullband ERLE estimator and the subbands ERLE estimators.
>   void Reset(bool delay_change);
> 
>   // Updates the ERLE estimates.
>-  void Update(rtc::ArrayView<const float> render_spectrum,
>+  void Update(const RenderBuffer& render_buffer,
>+              const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+                  filter_frequency_response,
>+              rtc::ArrayView<const float> reverb_render_spectrum,
>               rtc::ArrayView<const float> capture_spectrum,
>               rtc::ArrayView<const float> subtractor_spectrum,
>               bool converged_filter,
>@@ -45,11 +50,12 @@ class ErleEstimator {
> 
>   // Returns the most recent subband ERLE estimates.
>   const std::array<float, kFftLengthBy2Plus1>& Erle() const {
>-    return subband_erle_estimator_.Erle();
>+    return use_signal_dependent_erle_ ? signal_dependent_erle_estimator_.Erle()
>+                                      : subband_erle_estimator_.Erle();
>   }
>   // Returns the subband ERLE that are estimated during onsets. Used
>   // for logging/testing.
>-  const std::array<float, kFftLengthBy2Plus1>& ErleOnsets() const {
>+  rtc::ArrayView<const float> ErleOnsets() const {
>     return subband_erle_estimator_.ErleOnsets();
>   }
> 
>@@ -70,8 +76,10 @@ class ErleEstimator {
> 
>  private:
>   const size_t startup_phase_length_blocks__;
>+  const bool use_signal_dependent_erle_;
>   FullBandErleEstimator fullband_erle_estimator_;
>   SubbandErleEstimator subband_erle_estimator_;
>+  SignalDependentErleEstimator signal_dependent_erle_estimator_;
>   size_t blocks_since_reset_ = 0;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator_unittest.cc
>index 2cb050af3b3da112ae2f975884d50d1eddb358eb..59a747159337f368960e497e30d9c35bbae56f94 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator_unittest.cc
>@@ -12,6 +12,9 @@
> 
> #include "api/array_view.h"
> #include "modules/audio_processing/aec3/erle_estimator.h"
>+#include "modules/audio_processing/aec3/render_delay_buffer.h"
>+#include "modules/audio_processing/aec3/vector_buffer.h"
>+#include "rtc_base/random.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -19,11 +22,9 @@ namespace webrtc {
> namespace {
> 
> constexpr int kLowFrequencyLimit = kFftLengthBy2 / 2;
>-constexpr float kMaxErleLf = 8.f;
>-constexpr float kMaxErleHf = 1.5f;
>-constexpr float kMinErle = 1.0f;
> constexpr float kTrueErle = 10.f;
> constexpr float kTrueErleOnsets = 1.0f;
>+constexpr float kEchoPathGain = 3.f;
> 
> void VerifyErleBands(rtc::ArrayView<const float> erle,
>                      float reference_lf,
>@@ -44,80 +45,157 @@ void VerifyErle(rtc::ArrayView<const float> erle,
>   EXPECT_NEAR(reference_lf, erle_time_domain, 0.5);
> }
> 
>-void FormFarendFrame(std::array<float, kFftLengthBy2Plus1>* X2,
>+void FormFarendTimeFrame(rtc::ArrayView<float> x) {
>+  const std::array<float, kBlockSize> frame = {
>+      7459.88, 17209.6, 17383,   20768.9, 16816.7, 18386.3, 4492.83, 9675.85,
>+      6665.52, 14808.6, 9342.3,  7483.28, 19261.7, 4145.98, 1622.18, 13475.2,
>+      7166.32, 6856.61, 21937,   7263.14, 9569.07, 14919,   8413.32, 7551.89,
>+      7848.65, 6011.27, 13080.6, 15865.2, 12656,   17459.6, 4263.93, 4503.03,
>+      9311.79, 21095.8, 12657.9, 13906.6, 19267.2, 11338.1, 16828.9, 11501.6,
>+      11405,   15031.4, 14541.6, 19765.5, 18346.3, 19350.2, 3157.47, 18095.8,
>+      1743.68, 21328.2, 19727.5, 7295.16, 10332.4, 11055.5, 20107.4, 14708.4,
>+      12416.2, 16434,   2454.69, 9840.8,  6867.23, 1615.75, 6059.9,  8394.19};
>+  RTC_DCHECK_GE(x.size(), frame.size());
>+  std::copy(frame.begin(), frame.end(), x.begin());
>+}
>+
>+void FormFarendFrame(const RenderBuffer& render_buffer,
>+                     std::array<float, kFftLengthBy2Plus1>* X2,
>                      std::array<float, kFftLengthBy2Plus1>* E2,
>                      std::array<float, kFftLengthBy2Plus1>* Y2,
>                      float erle) {
>-  X2->fill(500 * 1000.f * 1000.f);
>-  E2->fill(1000.f * 1000.f);
>-  Y2->fill(erle * (*E2)[0]);
>-}
>+  const auto& spectrum_buffer = render_buffer.GetSpectrumBuffer();
>+  const auto& X2_from_buffer = spectrum_buffer.buffer[spectrum_buffer.write];
>+  std::copy(X2_from_buffer.begin(), X2_from_buffer.end(), X2->begin());
>+  std::transform(X2->begin(), X2->end(), Y2->begin(),
>+                 [](float a) { return a * kEchoPathGain * kEchoPathGain; });
>+  std::transform(Y2->begin(), Y2->end(), E2->begin(),
>+                 [erle](float a) { return a / erle; });
>+
>+}  // namespace
> 
>-void FormNearendFrame(std::array<float, kFftLengthBy2Plus1>* X2,
>+void FormNearendFrame(rtc::ArrayView<float> x,
>+                      std::array<float, kFftLengthBy2Plus1>* X2,
>                       std::array<float, kFftLengthBy2Plus1>* E2,
>                       std::array<float, kFftLengthBy2Plus1>* Y2) {
>+  x[0] = 0.f;
>   X2->fill(0.f);
>   Y2->fill(500.f * 1000.f * 1000.f);
>   E2->fill((*Y2)[0]);
> }
> 
>+void GetFilterFreq(std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+                       filter_frequency_response,
>+                   size_t delay_headroom_blocks) {
>+  RTC_DCHECK_GE(filter_frequency_response.size(), delay_headroom_blocks);
>+  for (auto& block_freq_resp : filter_frequency_response) {
>+    block_freq_resp.fill(0.f);
>+  }
>+
>+  for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
>+    filter_frequency_response[delay_headroom_blocks][k] = kEchoPathGain;
>+  }
>+}
>+
> }  // namespace
> 
> TEST(ErleEstimator, VerifyErleIncreaseAndHold) {
>   std::array<float, kFftLengthBy2Plus1> X2;
>   std::array<float, kFftLengthBy2Plus1> E2;
>   std::array<float, kFftLengthBy2Plus1> Y2;
>+  EchoCanceller3Config config;
>+  std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>+  std::vector<std::array<float, kFftLengthBy2Plus1>> filter_frequency_response(
>+      config.filter.main.length_blocks);
>+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>+      RenderDelayBuffer::Create2(config, 3));
> 
>-  ErleEstimator estimator(0, kMinErle, kMaxErleLf, kMaxErleHf);
>+  GetFilterFreq(filter_frequency_response, config.delay.delay_headroom_blocks);
> 
>-  // Verifies that the ERLE estimate is properly increased to higher values.
>-  FormFarendFrame(&X2, &E2, &Y2, kTrueErle);
>+  ErleEstimator estimator(0, config);
> 
>+  FormFarendTimeFrame(x[0]);
>+  render_delay_buffer->Insert(x);
>+  render_delay_buffer->PrepareCaptureProcessing();
>+  // Verifies that the ERLE estimate is properly increased to higher values.
>+  FormFarendFrame(*render_delay_buffer->GetRenderBuffer(), &X2, &E2, &Y2,
>+                  kTrueErle);
>   for (size_t k = 0; k < 200; ++k) {
>-    estimator.Update(X2, Y2, E2, true, true);
>+    render_delay_buffer->Insert(x);
>+    render_delay_buffer->PrepareCaptureProcessing();
>+    estimator.Update(*render_delay_buffer->GetRenderBuffer(),
>+                     filter_frequency_response, X2, Y2, E2, true, true);
>   }
>   VerifyErle(estimator.Erle(), std::pow(2.f, estimator.FullbandErleLog2()),
>-             kMaxErleLf, kMaxErleHf);
>+             config.erle.max_l, config.erle.max_h);
> 
>-  FormNearendFrame(&X2, &E2, &Y2);
>+  FormNearendFrame(x[0], &X2, &E2, &Y2);
>   // Verifies that the ERLE is not immediately decreased during nearend
>   // activity.
>   for (size_t k = 0; k < 50; ++k) {
>-    estimator.Update(X2, Y2, E2, true, true);
>+    render_delay_buffer->Insert(x);
>+    render_delay_buffer->PrepareCaptureProcessing();
>+    estimator.Update(*render_delay_buffer->GetRenderBuffer(),
>+                     filter_frequency_response, X2, Y2, E2, true, true);
>   }
>   VerifyErle(estimator.Erle(), std::pow(2.f, estimator.FullbandErleLog2()),
>-             kMaxErleLf, kMaxErleHf);
>+             config.erle.max_l, config.erle.max_h);
> }
> 
> TEST(ErleEstimator, VerifyErleTrackingOnOnsets) {
>   std::array<float, kFftLengthBy2Plus1> X2;
>   std::array<float, kFftLengthBy2Plus1> E2;
>   std::array<float, kFftLengthBy2Plus1> Y2;
>+  EchoCanceller3Config config;
>+  std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>+  std::vector<std::array<float, kFftLengthBy2Plus1>> filter_frequency_response(
>+      config.filter.main.length_blocks);
>+
>+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>+      RenderDelayBuffer::Create2(config, 3));
>+
>+  GetFilterFreq(filter_frequency_response, config.delay.delay_headroom_blocks);
>+
>+  ErleEstimator estimator(0, config);
> 
>-  ErleEstimator estimator(0, kMinErle, kMaxErleLf, kMaxErleHf);
>+  FormFarendTimeFrame(x[0]);
>+  render_delay_buffer->Insert(x);
>+  render_delay_buffer->PrepareCaptureProcessing();
> 
>   for (size_t burst = 0; burst < 20; ++burst) {
>-    FormFarendFrame(&X2, &E2, &Y2, kTrueErleOnsets);
>+    FormFarendFrame(*render_delay_buffer->GetRenderBuffer(), &X2, &E2, &Y2,
>+                    kTrueErleOnsets);
>     for (size_t k = 0; k < 10; ++k) {
>-      estimator.Update(X2, Y2, E2, true, true);
>+      render_delay_buffer->Insert(x);
>+      render_delay_buffer->PrepareCaptureProcessing();
>+      estimator.Update(*render_delay_buffer->GetRenderBuffer(),
>+                       filter_frequency_response, X2, Y2, E2, true, true);
>     }
>-    FormFarendFrame(&X2, &E2, &Y2, kTrueErle);
>+    FormFarendFrame(*render_delay_buffer->GetRenderBuffer(), &X2, &E2, &Y2,
>+                    kTrueErle);
>     for (size_t k = 0; k < 200; ++k) {
>-      estimator.Update(X2, Y2, E2, true, true);
>+      render_delay_buffer->Insert(x);
>+      render_delay_buffer->PrepareCaptureProcessing();
>+      estimator.Update(*render_delay_buffer->GetRenderBuffer(),
>+                       filter_frequency_response, X2, Y2, E2, true, true);
>     }
>-    FormNearendFrame(&X2, &E2, &Y2);
>+    FormNearendFrame(x[0], &X2, &E2, &Y2);
>     for (size_t k = 0; k < 300; ++k) {
>-      estimator.Update(X2, Y2, E2, true, true);
>+      render_delay_buffer->Insert(x);
>+      render_delay_buffer->PrepareCaptureProcessing();
>+      estimator.Update(*render_delay_buffer->GetRenderBuffer(),
>+                       filter_frequency_response, X2, Y2, E2, true, true);
>     }
>   }
>-  VerifyErleBands(estimator.ErleOnsets(), kMinErle, kMinErle);
>-  FormNearendFrame(&X2, &E2, &Y2);
>+  VerifyErleBands(estimator.ErleOnsets(), config.erle.min, config.erle.min);
>+  FormNearendFrame(x[0], &X2, &E2, &Y2);
>   for (size_t k = 0; k < 1000; k++) {
>-    estimator.Update(X2, Y2, E2, true, true);
>+    estimator.Update(*render_delay_buffer->GetRenderBuffer(),
>+                     filter_frequency_response, X2, Y2, E2, true, true);
>   }
>   // Verifies that during ne activity, Erle converges to the Erle for onsets.
>   VerifyErle(estimator.Erle(), std::pow(2.f, estimator.FullbandErleLog2()),
>-             kMinErle, kMinErle);
>+             config.erle.min, config.erle.min);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_buffer.h
>index 47ede41825cfc1b6355f9b095f152f64b23d403b..9f81a910a8fb7094f87d0515aa2baf30e1d8040b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_buffer.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_FFT_BUFFER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_FFT_BUFFER_H_
> 
>+#include <stddef.h>
> #include <vector>
> 
> #include "modules/audio_processing/aec3/fft_data.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.cc
>index 790cb1e8b52d6bff187b4bdd402019cf7fa1ed68..5b890d74746c9de1bccfc5f2ae3134803bc31bd5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.cc
>@@ -16,6 +16,7 @@
> #include <numeric>
> 
> #include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/render_buffer.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.h
>index 47e953366621dc23893279f5d7eb45cca666988c..99a0e2597377234ae219081aa8d33e10f978552a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.h
>@@ -11,20 +11,20 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_FILTER_ANALYZER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_FILTER_ANALYZER_H_
> 
>+#include <stddef.h>
> #include <array>
>+#include <memory>
> #include <vector>
> 
>-#include "absl/types/optional.h"
> #include "api/array_view.h"
> #include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>-#include "modules/audio_processing/aec3/cascaded_biquad_filter.h"
>-#include "modules/audio_processing/aec3/render_buffer.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
> class ApmDataDumper;
>+class RenderBuffer;
> 
> // Class for analyzing the properties of an adaptive filter.
> class FilterAnalyzer {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/frame_blocker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/frame_blocker.cc
>index 0a0c0e2faea5993086376342aa8df3de69c9bed9..ca122e5ebbdee8a79aab77bca0de599166586b16 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/frame_blocker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/frame_blocker.cc
>@@ -10,8 +10,7 @@
> 
> #include "modules/audio_processing/aec3/frame_blocker.h"
> 
>-#include <algorithm>
>-
>+#include "modules/audio_processing/aec3/aec3_common.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc
>index db9be7c104e969e50b2811dac6033174d1bbfaa5..7893b97b3a27aee6673a43a09489a137f6d5663f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fullband_erle_estimator.cc
>@@ -18,6 +18,7 @@
> #include "api/array_view.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_minmax.h"
> 
> namespace webrtc {
>@@ -25,7 +26,7 @@ namespace webrtc {
> namespace {
> constexpr float kEpsilon = 1e-3f;
> constexpr float kX2BandEnergyThreshold = 44015068.0f;
>-constexpr int kErleHold = 100;
>+constexpr int kBlocksToHoldErle = 100;
> constexpr int kPointsToAccumulate = 6;
> }  // namespace
> 
>@@ -54,7 +55,7 @@ void FullBandErleEstimator::Update(rtc::ArrayView<const float> X2,
>       const float Y2_sum = std::accumulate(Y2.begin(), Y2.end(), 0.0f);
>       const float E2_sum = std::accumulate(E2.begin(), E2.end(), 0.0f);
>       if (instantaneous_erle_.Update(Y2_sum, E2_sum)) {
>-        hold_counter_time_domain_ = kErleHold;
>+        hold_counter_time_domain_ = kBlocksToHoldErle;
>         erle_time_domain_log2_ +=
>             0.1f * ((instantaneous_erle_.GetInstErleLog2().value()) -
>                     erle_time_domain_log2_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc
>index 4f48cd47480b2b60f29738c00d65c69eeb1fa05a..ef87d142db1fc519b8ba85896477117cafea224c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc
>@@ -13,7 +13,12 @@
> #include <algorithm>
> #include <functional>
> 
>+#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/echo_path_variability.h"
>+#include "modules/audio_processing/aec3/fft_data.h"
>+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
>+#include "modules/audio_processing/aec3/subtractor_output.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
>@@ -127,7 +132,10 @@ void MainFilterUpdateGain::Compute(
>                  H_error_increase.begin(), std::multiplies<float>());
>   std::transform(H_error_.begin(), H_error_.end(), H_error_increase.begin(),
>                  H_error_.begin(), [&](float a, float b) {
>-                   return std::max(a + b, current_config_.error_floor);
>+                   float error = a + b;
>+                   error = std::max(error, current_config_.error_floor);
>+                   error = std::min(error, current_config_.error_ceil);
>+                   return error;
>                  });
> 
>   data_dumper_->DumpRaw("aec3_main_gain_H_error", H_error_);
>@@ -153,6 +161,9 @@ void MainFilterUpdateGain::UpdateCurrentConfig() {
>       current_config_.error_floor =
>           average(old_target_config_.error_floor, target_config_.error_floor,
>                   change_factor);
>+      current_config_.error_ceil =
>+          average(old_target_config_.error_ceil, target_config_.error_ceil,
>+                  change_factor);
>       current_config_.noise_gate =
>           average(old_target_config_.noise_gate, target_config_.noise_gate,
>                   change_factor);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.h
>index 525b52279c46ae6b0ee8323efb4e4c15a8ae075b..892ff689a470d30445af62814044c21a6aecd5f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.h
>@@ -11,20 +11,22 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_MAIN_FILTER_UPDATE_GAIN_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_MAIN_FILTER_UPDATE_GAIN_H_
> 
>+#include <stddef.h>
>+#include <array>
> #include <memory>
>-#include <vector>
> 
> #include "api/audio/echo_canceller3_config.h"
>-#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>-#include "modules/audio_processing/aec3/echo_path_variability.h"
>-#include "modules/audio_processing/aec3/render_signal_analyzer.h"
>-#include "modules/audio_processing/aec3/subtractor_output.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
>+class AdaptiveFirFilter;
> class ApmDataDumper;
>+struct EchoPathVariability;
>+struct FftData;
>+class RenderSignalAnalyzer;
>+struct SubtractorOutput;
> 
> // Provides functionality for  computing the adaptive gain for the main filter.
> class MainFilterUpdateGain {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc
>index 8c44ae064e2122e9e128a7ef5d89ecd49e681ce9..093b194736f86dcaf35c9a3875067c58a2b24f53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc
>@@ -66,7 +66,7 @@ void RunFilterUpdateTest(int num_blocks_to_process,
>   config.delay.min_echo_path_delay_blocks = 0;
>   config.delay.default_delay = 1;
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
>   AecState aec_state(config);
>   RenderSignalAnalyzer render_signal_analyzer(config);
>   absl::optional<DelayEstimate> delay_estimate;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.cc
>index 6744bd5812a0c75e3dc2534e94e60071819454b8..757219d52c22b22f93b9ed92d08291831b4501c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.cc
>@@ -19,10 +19,14 @@
> #include <emmintrin.h>
> #endif
> #include <algorithm>
>+#include <cstddef>
>+#include <initializer_list>
>+#include <iterator>
> #include <numeric>
> 
>-#include "api/audio/echo_canceller3_config.h"
>+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.h
>index 2ef48286d79ee6229642b21a080d943315c0f608..2a65339a4cbfefe3d1b02b187fa1d1b0c28084d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.h
>@@ -11,18 +11,19 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_
> 
>-#include <array>
>-#include <memory>
>+#include <stddef.h>
> #include <vector>
> 
>-#include "absl/types/optional.h"
> #include "api/array_view.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>-#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/system/arch.h"
> 
> namespace webrtc {
>+
>+class ApmDataDumper;
>+struct DownsampledRenderBuffer;
>+
> namespace aec3 {
> 
> #if defined(WEBRTC_HAS_NEON)
>@@ -65,7 +66,6 @@ void MatchedFilterCore(size_t x_start_index,
> 
> }  // namespace aec3
> 
>-class ApmDataDumper;
> 
> // Produces recursively updated cross-correlation estimates for several signal
> // shifts where the intra-shift spacing is uniform.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
>index 7a03e60d8bfd94428b4a3507dc92f1078bda4071..f59525eee9e901ec3b2a4402889eaf12ab828775 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
>@@ -9,7 +9,11 @@
>  */
> #include "modules/audio_processing/aec3/matched_filter_lag_aggregator.h"
> 
>+#include <algorithm>
>+#include <iterator>
>+
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>@@ -27,11 +31,13 @@ MatchedFilterLagAggregator::MatchedFilterLagAggregator(
> 
> MatchedFilterLagAggregator::~MatchedFilterLagAggregator() = default;
> 
>-void MatchedFilterLagAggregator::Reset() {
>+void MatchedFilterLagAggregator::Reset(bool hard_reset) {
>   std::fill(histogram_.begin(), histogram_.end(), 0);
>   histogram_data_.fill(0);
>   histogram_data_index_ = 0;
>-  significant_candidate_found_ = false;
>+  if (hard_reset) {
>+    significant_candidate_found_ = false;
>+  }
> }
> 
> absl::optional<DelayEstimate> MatchedFilterLagAggregator::Aggregate(
>@@ -77,9 +83,13 @@ absl::optional<DelayEstimate> MatchedFilterLagAggregator::Aggregate(
>     if (histogram_[candidate] > thresholds_.converged ||
>         (histogram_[candidate] > thresholds_.initial &&
>          !significant_candidate_found_)) {
>-      return DelayEstimate(DelayEstimate::Quality::kRefined, candidate);
>+      DelayEstimate::Quality quality = significant_candidate_found_
>+                                           ? DelayEstimate::Quality::kRefined
>+                                           : DelayEstimate::Quality::kCoarse;
>+      return DelayEstimate(quality, candidate);
>     }
>   }
>+
>   return absl::nullopt;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
>index c57051aa685f3374a408e9d51c2df6f71a9c38d8..d7f34aed60734bf4152bbc376c26d5b7f9230688 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
>@@ -34,7 +34,7 @@ class MatchedFilterLagAggregator {
>   ~MatchedFilterLagAggregator();
> 
>   // Resets the aggregator.
>-  void Reset();
>+  void Reset(bool hard_reset);
> 
>   // Aggregates the provided lag estimates.
>   absl::optional<DelayEstimate> Aggregate(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_unittest.cc
>index 16b603a13fe37daec690e9c939ba5d2ecf46814f..0c1711894b44f214599f4ac11ec5d7afe4304f17 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_unittest.cc
>@@ -165,7 +165,7 @@ TEST(MatchedFilter, LagEstimation) {
>                            config.delay.delay_candidate_detection_threshold);
> 
>       std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-          RenderDelayBuffer::Create(config, 3));
>+          RenderDelayBuffer::Create2(config, 3));
> 
>       // Analyze the correlation between render and capture.
>       for (size_t k = 0; k < (600 + delay_samples / sub_block_size); ++k) {
>@@ -261,7 +261,7 @@ TEST(MatchedFilter, LagNotReliableForUncorrelatedRenderAndCapture) {
>     std::fill(capture.begin(), capture.end(), 0.f);
>     ApmDataDumper data_dumper(0);
>     std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-        RenderDelayBuffer::Create(config, 3));
>+        RenderDelayBuffer::Create2(config, 3));
>     MatchedFilter filter(&data_dumper, DetectOptimization(), sub_block_size,
>                          kWindowSizeSubBlocks, kNumMatchedFilters,
>                          kAlignmentShiftSubBlocks, 150,
>@@ -306,7 +306,7 @@ TEST(MatchedFilter, LagNotUpdatedForLowLevelRender) {
>                          config.delay.delay_estimate_smoothing,
>                          config.delay.delay_candidate_detection_threshold);
>     std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-        RenderDelayBuffer::Create(config, 3));
>+        RenderDelayBuffer::Create2(EchoCanceller3Config(), 3));
>     Decimator capture_decimator(down_sampling_factor);
> 
>     // Analyze the correlation between render and capture.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matrix_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matrix_buffer.cc
>index f95e7f46902e1f05e31a045aa7b9274223ca5b8a..bd6daea95c905ca9e178f8b5622c8b0ac98fa7be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matrix_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matrix_buffer.cc
>@@ -10,7 +10,7 @@
> 
> #include "modules/audio_processing/aec3/matrix_buffer.h"
> 
>-#include "modules/audio_processing/aec3/aec3_common.h"
>+#include <algorithm>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matrix_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matrix_buffer.h
>index 64aac0a4f65e577bb1ebfd78af6e22b2d1e4482e..cae3759f92084a3d9e1c807604da8f39a8bf218e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matrix_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matrix_buffer.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_MATRIX_BUFFER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_MATRIX_BUFFER_H_
> 
>+#include <stddef.h>
> #include <vector>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
>index ed5971cda0d36ccb616f60eaf26aa6d82c7f69c2..5f652e192f97a1d8eddef05597ba4598292c9cd0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
>@@ -25,7 +25,7 @@ class MockRenderDelayController : public RenderDelayController {
>   MockRenderDelayController();
>   virtual ~MockRenderDelayController();
> 
>-  MOCK_METHOD0(Reset, void());
>+  MOCK_METHOD1(Reset, void(bool reset_delay_statistics));
>   MOCK_METHOD0(LogRenderCall, void());
>   MOCK_METHOD4(GetDelay,
>                absl::optional<DelayEstimate>(
>@@ -33,6 +33,7 @@ class MockRenderDelayController : public RenderDelayController {
>                    size_t render_delay_buffer_delay,
>                    const absl::optional<int>& echo_remover_delay,
>                    rtc::ArrayView<const float> capture));
>+  MOCK_CONST_METHOD0(HasClockdrift, bool());
> };
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.cc
>index e9d64e6b32632660f5a0453352234194ef876135..7a81ee89ea67e0dc8e39626cb6259193a4738f0f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.cc
>@@ -14,6 +14,8 @@
> #include <algorithm>
> #include <functional>
> 
>+#include "rtc_base/checks.h"
>+
> namespace webrtc {
> namespace aec3 {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.h
>index 94497d782c8d9d2aa40c09b348237ceea0bd5e6a..0f855beffb0f50e2d396803748e42d560153187a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
> 
>+#include <stddef.h>
> #include <vector>
> 
> #include "api/array_view.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.h
>index dd67268efd71ba8a98fed2cd902f16fe7d392594..4c7c60cf857d7d83d49ae1d32432ddd5234e2de5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.h
>@@ -11,14 +11,17 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_BUFFER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_RENDER_BUFFER_H_
> 
>+#include <stddef.h>
> #include <array>
>-#include <memory>
>+#include <vector>
> 
> #include "api/array_view.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/fft_buffer.h"
> #include "modules/audio_processing/aec3/fft_data.h"
> #include "modules/audio_processing/aec3/matrix_buffer.h"
> #include "modules/audio_processing/aec3/vector_buffer.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
>index 37a2378738cc5be3060d3c4560c5afd644a63213..1ec27790a842db95ed979da7a4a2559b94719c8f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
>@@ -10,17 +10,21 @@
> 
> #include "modules/audio_processing/aec3/render_delay_buffer.h"
> 
>-#include <string.h>
>+#include <stdlib.h>
> #include <algorithm>
>+#include <memory>
> #include <numeric>
> 
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/aec3_fft.h"
>-#include "modules/audio_processing/aec3/block_processor.h"
> #include "modules/audio_processing/aec3/decimator.h"
> #include "modules/audio_processing/aec3/fft_buffer.h"
> #include "modules/audio_processing/aec3/fft_data.h"
> #include "modules/audio_processing/aec3/matrix_buffer.h"
>+#include "modules/audio_processing/aec3/vector_buffer.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
>@@ -35,14 +39,6 @@ bool EnableZeroExternalDelayHeadroom() {
>       "WebRTC-Aec3ZeroExternalDelayHeadroomKillSwitch");
> }
> 
>-size_t GetDownSamplingFactor(const EchoCanceller3Config& config) {
>-  // Do not use down sampling factor 8 if kill switch is triggered.
>-  return (config.delay.down_sampling_factor == 8 &&
>-          field_trial::IsEnabled("WebRTC-Aec3DownSamplingFactor8KillSwitch"))
>-             ? 4
>-             : config.delay.down_sampling_factor;
>-}
>-
> class RenderDelayBufferImpl final : public RenderDelayBuffer {
>  public:
>   RenderDelayBufferImpl(const EchoCanceller3Config& config, size_t num_bands);
>@@ -177,7 +173,7 @@ RenderDelayBufferImpl::RenderDelayBufferImpl(const EchoCanceller3Config& config,
>           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>       optimization_(DetectOptimization()),
>       config_(config),
>-      down_sampling_factor_(GetDownSamplingFactor(config)),
>+      down_sampling_factor_(config.delay.down_sampling_factor),
>       use_zero_external_delay_headroom_(EnableZeroExternalDelayHeadroom()),
>       sub_block_size_(static_cast<int>(down_sampling_factor_ > 0
>                                            ? kBlockSize / down_sampling_factor_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.h
>index a6d6874e3e4696b105de04a402c3327141384a3f..bd242f743f9b2a240ddb2c6d76359c842eb33b9a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.h
>@@ -12,15 +12,10 @@
> #define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
> 
> #include <stddef.h>
>-#include <array>
> #include <vector>
> 
>-#include "absl/types/optional.h"
>-#include "api/array_view.h"
> #include "api/audio/echo_canceller3_config.h"
>-#include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/downsampled_render_buffer.h"
>-#include "modules/audio_processing/aec3/fft_data.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
> 
> namespace webrtc {
>@@ -33,12 +28,13 @@ class RenderDelayBuffer {
>     kNone,
>     kRenderUnderrun,
>     kRenderOverrun,
>-    kApiCallSkew,
>-    kRenderDataLost
>+    kApiCallSkew
>   };
> 
>   static RenderDelayBuffer* Create(const EchoCanceller3Config& config,
>                                    size_t num_bands);
>+  static RenderDelayBuffer* Create2(const EchoCanceller3Config& config,
>+                                    size_t num_bands);
>   virtual ~RenderDelayBuffer() = default;
> 
>   // Resets the buffer alignment.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer2.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..6992c5b898765f2529d99a355a881cda5f56db95
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer2.cc
>@@ -0,0 +1,453 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <string.h>
>+#include <algorithm>
>+#include <memory>
>+#include <numeric>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/aec3_fft.h"
>+#include "modules/audio_processing/aec3/decimator.h"
>+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
>+#include "modules/audio_processing/aec3/fft_buffer.h"
>+#include "modules/audio_processing/aec3/fft_data.h"
>+#include "modules/audio_processing/aec3/matrix_buffer.h"
>+#include "modules/audio_processing/aec3/render_buffer.h"
>+#include "modules/audio_processing/aec3/render_delay_buffer.h"
>+#include "modules/audio_processing/aec3/vector_buffer.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/atomicops.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+class RenderDelayBufferImpl2 final : public RenderDelayBuffer {
>+ public:
>+  RenderDelayBufferImpl2(const EchoCanceller3Config& config, size_t num_bands);
>+  RenderDelayBufferImpl2() = delete;
>+  ~RenderDelayBufferImpl2() override;
>+
>+  void Reset() override;
>+  BufferingEvent Insert(const std::vector<std::vector<float>>& block) override;
>+  BufferingEvent PrepareCaptureProcessing() override;
>+  bool SetDelay(size_t delay) override;
>+  size_t Delay() const override { return ComputeDelay(); }
>+  size_t MaxDelay() const override {
>+    return blocks_.buffer.size() - 1 - buffer_headroom_;
>+  }
>+  RenderBuffer* GetRenderBuffer() override { return &echo_remover_buffer_; }
>+
>+  const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const override {
>+    return low_rate_;
>+  }
>+
>+  int BufferLatency() const;
>+  bool CausalDelay(size_t delay) const override;
>+  void SetAudioBufferDelay(size_t delay_ms) override;
>+
>+ private:
>+  static int instance_count_;
>+  std::unique_ptr<ApmDataDumper> data_dumper_;
>+  const Aec3Optimization optimization_;
>+  const EchoCanceller3Config config_;
>+  size_t down_sampling_factor_;
>+  const int sub_block_size_;
>+  MatrixBuffer blocks_;
>+  VectorBuffer spectra_;
>+  FftBuffer ffts_;
>+  absl::optional<size_t> delay_;
>+  RenderBuffer echo_remover_buffer_;
>+  DownsampledRenderBuffer low_rate_;
>+  Decimator render_decimator_;
>+  const Aec3Fft fft_;
>+  std::vector<float> render_ds_;
>+  const int buffer_headroom_;
>+  bool last_call_was_render_ = false;
>+  int num_api_calls_in_a_row_ = 0;
>+  int max_observed_jitter_ = 1;
>+  size_t capture_call_counter_ = 0;
>+  size_t render_call_counter_ = 0;
>+  bool render_activity_ = false;
>+  size_t render_activity_counter_ = 0;
>+  absl::optional<size_t> external_audio_buffer_delay_;
>+  bool external_audio_buffer_delay_verified_after_reset_ = false;
>+  size_t min_latency_blocks_ = 0;
>+  size_t excess_render_detection_counter_ = 0;
>+  size_t num_bands_;
>+
>+  int MapDelayToTotalDelay(size_t delay) const;
>+  int ComputeDelay() const;
>+  void ApplyTotalDelay(int delay);
>+  void InsertBlock(const std::vector<std::vector<float>>& block,
>+                   int previous_write);
>+  bool DetectActiveRender(rtc::ArrayView<const float> x) const;
>+  bool DetectExcessRenderBlocks();
>+  void IncrementWriteIndices();
>+  void IncrementLowRateReadIndices();
>+  void IncrementReadIndices();
>+  bool RenderOverrun();
>+  bool RenderUnderrun();
>+};
>+
>+int RenderDelayBufferImpl2::instance_count_ = 0;
>+
>+RenderDelayBufferImpl2::RenderDelayBufferImpl2(
>+    const EchoCanceller3Config& config,
>+    size_t num_bands)
>+    : data_dumper_(
>+          new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>+      optimization_(DetectOptimization()),
>+      config_(config),
>+      down_sampling_factor_(config.delay.down_sampling_factor),
>+      sub_block_size_(static_cast<int>(down_sampling_factor_ > 0
>+                                           ? kBlockSize / down_sampling_factor_
>+                                           : kBlockSize)),
>+      blocks_(GetRenderDelayBufferSize(down_sampling_factor_,
>+                                       config.delay.num_filters,
>+                                       config.filter.main.length_blocks),
>+              num_bands,
>+              kBlockSize),
>+      spectra_(blocks_.buffer.size(), kFftLengthBy2Plus1),
>+      ffts_(blocks_.buffer.size()),
>+      delay_(config_.delay.default_delay),
>+      echo_remover_buffer_(&blocks_, &spectra_, &ffts_),
>+      low_rate_(GetDownSampledBufferSize(down_sampling_factor_,
>+                                         config.delay.num_filters)),
>+      render_decimator_(down_sampling_factor_),
>+      fft_(),
>+      render_ds_(sub_block_size_, 0.f),
>+      buffer_headroom_(config.filter.main.length_blocks),
>+      num_bands_(num_bands) {
>+  RTC_DCHECK_EQ(blocks_.buffer.size(), ffts_.buffer.size());
>+  RTC_DCHECK_EQ(spectra_.buffer.size(), ffts_.buffer.size());
>+
>+  Reset();
>+}
>+
>+RenderDelayBufferImpl2::~RenderDelayBufferImpl2() = default;
>+
>+// Resets the buffer delays and clears the reported delays.
>+void RenderDelayBufferImpl2::Reset() {
>+  last_call_was_render_ = false;
>+  num_api_calls_in_a_row_ = 1;
>+  min_latency_blocks_ = 0;
>+  excess_render_detection_counter_ = 0;
>+
>+  // Initialize the read index to one sub-block before the write index.
>+  low_rate_.read = low_rate_.OffsetIndex(low_rate_.write, sub_block_size_);
>+
>+  // Check for any external audio buffer delay and whether it is feasible.
>+  if (external_audio_buffer_delay_) {
>+    const size_t headroom = 2;
>+    size_t audio_buffer_delay_to_set;
>+    // Minimum delay is 1 (like the low-rate render buffer).
>+    if (*external_audio_buffer_delay_ <= headroom) {
>+      audio_buffer_delay_to_set = 1;
>+    } else {
>+      audio_buffer_delay_to_set = *external_audio_buffer_delay_ - headroom;
>+    }
>+
>+    audio_buffer_delay_to_set = std::min(audio_buffer_delay_to_set, MaxDelay());
>+
>+    // When an external delay estimate is available, use that delay as the
>+    // initial render buffer delay.
>+    ApplyTotalDelay(audio_buffer_delay_to_set);
>+    delay_ = ComputeDelay();
>+
>+    external_audio_buffer_delay_verified_after_reset_ = false;
>+  } else {
>+    // If an external delay estimate is not available, use that delay as the
>+    // initial delay. Set the render buffer delays to the default delay.
>+    ApplyTotalDelay(config_.delay.default_delay);
>+
>+    // Unset the delays which are set by SetDelay.
>+    delay_ = absl::nullopt;
>+  }
>+}
>+
>+// Inserts a new block into the render buffers.
>+RenderDelayBuffer::BufferingEvent RenderDelayBufferImpl2::Insert(
>+    const std::vector<std::vector<float>>& block) {
>+  ++render_call_counter_;
>+  if (delay_) {
>+    if (!last_call_was_render_) {
>+      last_call_was_render_ = true;
>+      num_api_calls_in_a_row_ = 1;
>+    } else {
>+      if (++num_api_calls_in_a_row_ > max_observed_jitter_) {
>+        max_observed_jitter_ = num_api_calls_in_a_row_;
>+        RTC_LOG(LS_WARNING)
>+            << "New max number api jitter observed at render block "
>+            << render_call_counter_ << ":  " << num_api_calls_in_a_row_
>+            << " blocks";
>+      }
>+    }
>+  }
>+
>+  // Increase the write indices to where the new blocks should be written.
>+  const int previous_write = blocks_.write;
>+  IncrementWriteIndices();
>+
>+  // Allow overrun and do a reset when render overrun occurrs due to more render
>+  // data being inserted than capture data is received.
>+  BufferingEvent event =
>+      RenderOverrun() ? BufferingEvent::kRenderOverrun : BufferingEvent::kNone;
>+
>+  // Detect and update render activity.
>+  if (!render_activity_) {
>+    render_activity_counter_ += DetectActiveRender(block[0]) ? 1 : 0;
>+    render_activity_ = render_activity_counter_ >= 20;
>+  }
>+
>+  // Insert the new render block into the specified position.
>+  InsertBlock(block, previous_write);
>+
>+  if (event != BufferingEvent::kNone) {
>+    Reset();
>+  }
>+
>+  return event;
>+}
>+
>+// Prepares the render buffers for processing another capture block.
>+RenderDelayBuffer::BufferingEvent
>+RenderDelayBufferImpl2::PrepareCaptureProcessing() {
>+  RenderDelayBuffer::BufferingEvent event = BufferingEvent::kNone;
>+  ++capture_call_counter_;
>+
>+  if (delay_) {
>+    if (last_call_was_render_) {
>+      last_call_was_render_ = false;
>+      num_api_calls_in_a_row_ = 1;
>+    } else {
>+      if (++num_api_calls_in_a_row_ > max_observed_jitter_) {
>+        max_observed_jitter_ = num_api_calls_in_a_row_;
>+        RTC_LOG(LS_WARNING)
>+            << "New max number api jitter observed at capture block "
>+            << capture_call_counter_ << ":  " << num_api_calls_in_a_row_
>+            << " blocks";
>+      }
>+    }
>+  }
>+
>+  if (DetectExcessRenderBlocks()) {
>+    // Too many render blocks compared to capture blocks. Risk of delay ending
>+    // up before the filter used by the delay estimator.
>+    RTC_LOG(LS_WARNING) << "Excess render blocks detected at block "
>+                        << capture_call_counter_;
>+    Reset();
>+    event = BufferingEvent::kRenderOverrun;
>+  } else if (RenderUnderrun()) {
>+    // Don't increment the read indices of the low rate buffer if there is a
>+    // render underrun.
>+    RTC_LOG(LS_WARNING) << "Render buffer underrun detected at block "
>+                        << capture_call_counter_;
>+    IncrementReadIndices();
>+    // Incrementing the buffer index without increasing the low rate buffer
>+    // index means that the delay is reduced by one.
>+    if (delay_ && *delay_ > 0)
>+      delay_ = *delay_ - 1;
>+    event = BufferingEvent::kRenderUnderrun;
>+  } else {
>+    // Increment the read indices in the render buffers to point to the most
>+    // recent block to use in the capture processing.
>+    IncrementLowRateReadIndices();
>+    IncrementReadIndices();
>+  }
>+
>+  echo_remover_buffer_.SetRenderActivity(render_activity_);
>+  if (render_activity_) {
>+    render_activity_counter_ = 0;
>+    render_activity_ = false;
>+  }
>+
>+  return event;
>+}
>+
>+// Sets the delay and returns a bool indicating whether the delay was changed.
>+bool RenderDelayBufferImpl2::SetDelay(size_t delay) {
>+  if (!external_audio_buffer_delay_verified_after_reset_ &&
>+      external_audio_buffer_delay_ && delay_) {
>+    int difference = static_cast<int>(delay) - static_cast<int>(*delay_);
>+    RTC_LOG(LS_WARNING) << "Mismatch between first estimated delay after reset "
>+                           "and externally reported audio buffer delay: "
>+                        << difference << " blocks";
>+    external_audio_buffer_delay_verified_after_reset_ = true;
>+  }
>+  if (delay_ && *delay_ == delay) {
>+    return false;
>+  }
>+  delay_ = delay;
>+
>+  // Compute the total delay and limit the delay to the allowed range.
>+  int total_delay = MapDelayToTotalDelay(*delay_);
>+  total_delay =
>+      std::min(MaxDelay(), static_cast<size_t>(std::max(total_delay, 0)));
>+
>+  // Apply the delay to the buffers.
>+  ApplyTotalDelay(total_delay);
>+  return true;
>+}
>+
>+// Returns whether the specified delay is causal.
>+bool RenderDelayBufferImpl2::CausalDelay(size_t delay) const {
>+  // TODO(gustaf): Remove this from RenderDelayBuffer.
>+  return true;
>+}
>+
>+void RenderDelayBufferImpl2::SetAudioBufferDelay(size_t delay_ms) {
>+  if (!external_audio_buffer_delay_) {
>+    RTC_LOG(LS_WARNING)
>+        << "Receiving a first externally reported audio buffer delay of "
>+        << delay_ms << " ms.";
>+  }
>+
>+  // Convert delay from milliseconds to blocks (rounded down).
>+  external_audio_buffer_delay_ = delay_ms >> ((num_bands_ == 1) ? 1 : 2);
>+}
>+
>+// Maps the externally computed delay to the delay used internally.
>+int RenderDelayBufferImpl2::MapDelayToTotalDelay(
>+    size_t external_delay_blocks) const {
>+  const int latency_blocks = BufferLatency();
>+  return latency_blocks + static_cast<int>(external_delay_blocks);
>+}
>+
>+// Returns the delay (not including call jitter).
>+int RenderDelayBufferImpl2::ComputeDelay() const {
>+  const int latency_blocks = BufferLatency();
>+  int internal_delay = spectra_.read >= spectra_.write
>+                           ? spectra_.read - spectra_.write
>+                           : spectra_.size + spectra_.read - spectra_.write;
>+
>+  return internal_delay - latency_blocks;
>+}
>+
>+// Set the read indices according to the delay.
>+void RenderDelayBufferImpl2::ApplyTotalDelay(int delay) {
>+  RTC_LOG(LS_WARNING) << "Applying total delay of " << delay << " blocks.";
>+  blocks_.read = blocks_.OffsetIndex(blocks_.write, -delay);
>+  spectra_.read = spectra_.OffsetIndex(spectra_.write, delay);
>+  ffts_.read = ffts_.OffsetIndex(ffts_.write, delay);
>+}
>+
>+// Inserts a block into the render buffers.
>+void RenderDelayBufferImpl2::InsertBlock(
>+    const std::vector<std::vector<float>>& block,
>+    int previous_write) {
>+  auto& b = blocks_;
>+  auto& lr = low_rate_;
>+  auto& ds = render_ds_;
>+  auto& f = ffts_;
>+  auto& s = spectra_;
>+  RTC_DCHECK_EQ(block.size(), b.buffer[b.write].size());
>+  for (size_t k = 0; k < block.size(); ++k) {
>+    RTC_DCHECK_EQ(block[k].size(), b.buffer[b.write][k].size());
>+    std::copy(block[k].begin(), block[k].end(), b.buffer[b.write][k].begin());
>+  }
>+
>+  data_dumper_->DumpWav("aec3_render_decimator_input", block[0].size(),
>+                        block[0].data(), 16000, 1);
>+  render_decimator_.Decimate(block[0], ds);
>+  data_dumper_->DumpWav("aec3_render_decimator_output", ds.size(), ds.data(),
>+                        16000 / down_sampling_factor_, 1);
>+  std::copy(ds.rbegin(), ds.rend(), lr.buffer.begin() + lr.write);
>+  fft_.PaddedFft(block[0], b.buffer[previous_write][0], &f.buffer[f.write]);
>+  f.buffer[f.write].Spectrum(optimization_, s.buffer[s.write]);
>+}
>+
>+bool RenderDelayBufferImpl2::DetectActiveRender(
>+    rtc::ArrayView<const float> x) const {
>+  const float x_energy = std::inner_product(x.begin(), x.end(), x.begin(), 0.f);
>+  return x_energy > (config_.render_levels.active_render_limit *
>+                     config_.render_levels.active_render_limit) *
>+                        kFftLengthBy2;
>+}
>+
>+bool RenderDelayBufferImpl2::DetectExcessRenderBlocks() {
>+  bool excess_render_detected = false;
>+  const size_t latency_blocks = static_cast<size_t>(BufferLatency());
>+  // The recently seen minimum latency in blocks. Should be close to 0.
>+  min_latency_blocks_ = std::min(min_latency_blocks_, latency_blocks);
>+  // After processing a configurable number of blocks the minimum latency is
>+  // checked.
>+  if (++excess_render_detection_counter_ >=
>+      config_.buffering.excess_render_detection_interval_blocks) {
>+    // If the minimum latency is not lower than the threshold there have been
>+    // more render than capture frames.
>+    excess_render_detected = min_latency_blocks_ >
>+                             config_.buffering.max_allowed_excess_render_blocks;
>+    // Reset the counter and let the minimum latency be the current latency.
>+    min_latency_blocks_ = latency_blocks;
>+    excess_render_detection_counter_ = 0;
>+  }
>+
>+  data_dumper_->DumpRaw("aec3_latency_blocks", latency_blocks);
>+  data_dumper_->DumpRaw("aec3_min_latency_blocks", min_latency_blocks_);
>+  data_dumper_->DumpRaw("aec3_excess_render_detected", excess_render_detected);
>+  return excess_render_detected;
>+}
>+
>+// Computes the latency in the buffer (the number of unread sub-blocks).
>+int RenderDelayBufferImpl2::BufferLatency() const {
>+  const DownsampledRenderBuffer& l = low_rate_;
>+  int latency_samples = (l.buffer.size() + l.read - l.write) % l.buffer.size();
>+  int latency_blocks = latency_samples / sub_block_size_;
>+  return latency_blocks;
>+}
>+
>+// Increments the write indices for the render buffers.
>+void RenderDelayBufferImpl2::IncrementWriteIndices() {
>+  low_rate_.UpdateWriteIndex(-sub_block_size_);
>+  blocks_.IncWriteIndex();
>+  spectra_.DecWriteIndex();
>+  ffts_.DecWriteIndex();
>+}
>+
>+// Increments the read indices of the low rate render buffers.
>+void RenderDelayBufferImpl2::IncrementLowRateReadIndices() {
>+  low_rate_.UpdateReadIndex(-sub_block_size_);
>+}
>+
>+// Increments the read indices for the render buffers.
>+void RenderDelayBufferImpl2::IncrementReadIndices() {
>+  if (blocks_.read != blocks_.write) {
>+    blocks_.IncReadIndex();
>+    spectra_.DecReadIndex();
>+    ffts_.DecReadIndex();
>+  }
>+}
>+
>+// Checks for a render buffer overrun.
>+bool RenderDelayBufferImpl2::RenderOverrun() {
>+  return low_rate_.read == low_rate_.write || blocks_.read == blocks_.write;
>+}
>+
>+// Checks for a render buffer underrun.
>+bool RenderDelayBufferImpl2::RenderUnderrun() {
>+  return low_rate_.read == low_rate_.write;
>+}
>+
>+}  // namespace
>+
>+RenderDelayBuffer* RenderDelayBuffer::Create2(
>+    const EchoCanceller3Config& config,
>+    size_t num_bands) {
>+  return new RenderDelayBufferImpl2(config, num_bands);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
>index ee895972619f40416d9cff2bc291a089871f41c6..d1530c6142c4e90b3d219ada7df3dff818bb5aaf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
>@@ -38,7 +38,7 @@ TEST(RenderDelayBuffer, BufferOverflow) {
>   for (auto rate : {8000, 16000, 32000, 48000}) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>     std::unique_ptr<RenderDelayBuffer> delay_buffer(
>-        RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>+        RenderDelayBuffer::Create2(config, NumBandsForRate(rate)));
>     std::vector<std::vector<float>> block_to_insert(
>         NumBandsForRate(rate), std::vector<float>(kBlockSize, 0.f));
>     for (size_t k = 0; k < 10; ++k) {
>@@ -62,7 +62,7 @@ TEST(RenderDelayBuffer, BufferOverflow) {
> TEST(RenderDelayBuffer, AvailableBlock) {
>   constexpr size_t kNumBands = 1;
>   std::unique_ptr<RenderDelayBuffer> delay_buffer(
>-      RenderDelayBuffer::Create(EchoCanceller3Config(), kNumBands));
>+      RenderDelayBuffer::Create2(EchoCanceller3Config(), kNumBands));
>   std::vector<std::vector<float>> input_block(
>       kNumBands, std::vector<float>(kBlockSize, 1.f));
>   EXPECT_EQ(RenderDelayBuffer::BufferingEvent::kNone,
>@@ -74,7 +74,7 @@ TEST(RenderDelayBuffer, AvailableBlock) {
> TEST(RenderDelayBuffer, SetDelay) {
>   EchoCanceller3Config config;
>   std::unique_ptr<RenderDelayBuffer> delay_buffer(
>-      RenderDelayBuffer::Create(config, 1));
>+      RenderDelayBuffer::Create2(config, 1));
>   ASSERT_TRUE(delay_buffer->Delay());
>   delay_buffer->Reset();
>   size_t initial_internal_delay = config.delay.min_echo_path_delay_blocks +
>@@ -93,7 +93,7 @@ TEST(RenderDelayBuffer, SetDelay) {
> // tests on test bots has been fixed.
> TEST(RenderDelayBuffer, DISABLED_WrongDelay) {
>   std::unique_ptr<RenderDelayBuffer> delay_buffer(
>-      RenderDelayBuffer::Create(EchoCanceller3Config(), 3));
>+      RenderDelayBuffer::Create2(EchoCanceller3Config(), 3));
>   EXPECT_DEATH(delay_buffer->SetDelay(21), "");
> }
> 
>@@ -101,7 +101,7 @@ TEST(RenderDelayBuffer, DISABLED_WrongDelay) {
> TEST(RenderDelayBuffer, WrongNumberOfBands) {
>   for (auto rate : {16000, 32000, 48000}) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>-    std::unique_ptr<RenderDelayBuffer> delay_buffer(RenderDelayBuffer::Create(
>+    std::unique_ptr<RenderDelayBuffer> delay_buffer(RenderDelayBuffer::Create2(
>         EchoCanceller3Config(), NumBandsForRate(rate)));
>     std::vector<std::vector<float>> block_to_insert(
>         NumBandsForRate(rate < 48000 ? rate + 16000 : 16000),
>@@ -115,7 +115,7 @@ TEST(RenderDelayBuffer, WrongBlockLength) {
>   for (auto rate : {8000, 16000, 32000, 48000}) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>     std::unique_ptr<RenderDelayBuffer> delay_buffer(
>-        RenderDelayBuffer::Create(EchoCanceller3Config(), 3));
>+        RenderDelayBuffer::Create2(EchoCanceller3Config(), 3));
>     std::vector<std::vector<float>> block_to_insert(
>         NumBandsForRate(rate), std::vector<float>(kBlockSize - 1, 0.f));
>     EXPECT_DEATH(delay_buffer->Insert(block_to_insert), "");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
>index f336c392b4ee7039623668221d84c805fb6a9588..c4665eaa224dec1edd4325b22826104d4157a98c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
>@@ -9,10 +9,9 @@
>  */
> #include "modules/audio_processing/aec3/render_delay_controller.h"
> 
>+#include <stdlib.h>
> #include <algorithm>
> #include <memory>
>-#include <numeric>
>-#include <string>
> #include <vector>
> 
> #include "api/audio/echo_canceller3_config.h"
>@@ -20,7 +19,9 @@
> #include "modules/audio_processing/aec3/echo_path_delay_estimator.h"
> #include "modules/audio_processing/aec3/render_delay_controller_metrics.h"
> #include "modules/audio_processing/aec3/skew_estimator.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/field_trial.h"
>@@ -44,6 +45,10 @@ bool UseOffsetBlocks() {
>   return field_trial::IsEnabled("WebRTC-Aec3UseOffsetBlocks");
> }
> 
>+bool UseEarlyDelayDetection() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3EarlyDelayDetectionKillSwitch");
>+}
>+
> constexpr int kSkewHistorySizeLog2 = 8;
> 
> class RenderDelayControllerImpl final : public RenderDelayController {
>@@ -52,17 +57,19 @@ class RenderDelayControllerImpl final : public RenderDelayController {
>                             int non_causal_offset,
>                             int sample_rate_hz);
>   ~RenderDelayControllerImpl() override;
>-  void Reset() override;
>+  void Reset(bool reset_delay_confidence) override;
>   void LogRenderCall() override;
>   absl::optional<DelayEstimate> GetDelay(
>       const DownsampledRenderBuffer& render_buffer,
>       size_t render_delay_buffer_delay,
>       const absl::optional<int>& echo_remover_delay,
>       rtc::ArrayView<const float> capture) override;
>+  bool HasClockdrift() const override;
> 
>  private:
>   static int instance_count_;
>   std::unique_ptr<ApmDataDumper> data_dumper_;
>+  const bool use_early_delay_detection_;
>   const int delay_headroom_blocks_;
>   const int hysteresis_limit_1_blocks_;
>   const int hysteresis_limit_2_blocks_;
>@@ -81,6 +88,7 @@ class RenderDelayControllerImpl final : public RenderDelayController {
>   size_t capture_call_counter_ = 0;
>   int delay_change_counter_ = 0;
>   size_t soft_reset_counter_ = 0;
>+  DelayEstimate::Quality last_delay_estimate_quality_;
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderDelayControllerImpl);
> };
> 
>@@ -129,6 +137,7 @@ RenderDelayControllerImpl::RenderDelayControllerImpl(
>     int sample_rate_hz)
>     : data_dumper_(
>           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>+      use_early_delay_detection_(UseEarlyDelayDetection()),
>       delay_headroom_blocks_(
>           static_cast<int>(config.delay.delay_headroom_blocks)),
>       hysteresis_limit_1_blocks_(
>@@ -139,7 +148,8 @@ RenderDelayControllerImpl::RenderDelayControllerImpl(
>       use_offset_blocks_(UseOffsetBlocks()),
>       delay_estimator_(data_dumper_.get(), config),
>       delay_buf_(kBlockSize * non_causal_offset, 0.f),
>-      skew_estimator_(kSkewHistorySizeLog2) {
>+      skew_estimator_(kSkewHistorySizeLog2),
>+      last_delay_estimate_quality_(DelayEstimate::Quality::kCoarse) {
>   RTC_DCHECK(ValidFullBandRate(sample_rate_hz));
>   delay_estimator_.LogDelayEstimationProperties(sample_rate_hz,
>                                                 delay_buf_.size());
>@@ -147,16 +157,19 @@ RenderDelayControllerImpl::RenderDelayControllerImpl(
> 
> RenderDelayControllerImpl::~RenderDelayControllerImpl() = default;
> 
>-void RenderDelayControllerImpl::Reset() {
>+void RenderDelayControllerImpl::Reset(bool reset_delay_confidence) {
>   delay_ = absl::nullopt;
>   delay_samples_ = absl::nullopt;
>   skew_ = absl::nullopt;
>   previous_offset_blocks_ = 0;
>   std::fill(delay_buf_.begin(), delay_buf_.end(), 0.f);
>-  delay_estimator_.Reset(false);
>+  delay_estimator_.Reset(reset_delay_confidence);
>   skew_estimator_.Reset();
>   delay_change_counter_ = 0;
>   soft_reset_counter_ = 0;
>+  if (reset_delay_confidence) {
>+    last_delay_estimate_quality_ = DelayEstimate::Quality::kCoarse;
>+  }
> }
> 
> void RenderDelayControllerImpl::LogRenderCall() {
>@@ -194,9 +207,6 @@ absl::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
>   absl::optional<int> skew = skew_estimator_.GetSkewFromCapture();
> 
>   if (delay_samples) {
>-    // TODO(peah): Refactor the rest of the code to assume a kRefined estimate
>-    // quality.
>-    RTC_DCHECK(DelayEstimate::Quality::kRefined == delay_samples->quality);
>     if (!delay_samples_ || delay_samples->delay != delay_samples_->delay) {
>       delay_change_counter_ = 0;
>     }
>@@ -239,7 +249,7 @@ absl::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
>     } else if (soft_reset_counter_ > 10 * kNumBlocksPerSecond) {
>       // Soft reset the delay estimator if there is a significant offset
>       // detected.
>-      delay_estimator_.Reset(true);
>+      delay_estimator_.Reset(false);
>       soft_reset_counter_ = 0;
>     }
>   }
>@@ -264,14 +274,20 @@ absl::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
> 
>   if (delay_samples_) {
>     // Compute the render delay buffer delay.
>-    delay_ = ComputeBufferDelay(
>-        delay_, delay_headroom_blocks_, hysteresis_limit_1_blocks_,
>-        hysteresis_limit_2_blocks_, offset_blocks, *delay_samples_);
>+    const bool use_hysteresis =
>+        last_delay_estimate_quality_ == DelayEstimate::Quality::kRefined &&
>+        delay_samples_->quality == DelayEstimate::Quality::kRefined;
>+    delay_ = ComputeBufferDelay(delay_, delay_headroom_blocks_,
>+                                use_hysteresis ? hysteresis_limit_1_blocks_ : 0,
>+                                use_hysteresis ? hysteresis_limit_2_blocks_ : 0,
>+                                offset_blocks, *delay_samples_);
>+    last_delay_estimate_quality_ = delay_samples_->quality;
>   }
> 
>   metrics_.Update(delay_samples_ ? absl::optional<size_t>(delay_samples_->delay)
>                                  : absl::nullopt,
>-                  delay_ ? delay_->delay : 0, skew_shift);
>+                  delay_ ? delay_->delay : 0, skew_shift,
>+                  delay_estimator_.Clockdrift());
> 
>   data_dumper_->DumpRaw("aec3_render_delay_controller_delay",
>                         delay_samples ? delay_samples->delay : 0);
>@@ -287,6 +303,10 @@ absl::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
>   return delay_;
> }
> 
>+bool RenderDelayControllerImpl::HasClockdrift() const {
>+  return delay_estimator_.Clockdrift() != ClockdriftDetector::Level::kNone;
>+}
>+
> }  // namespace
> 
> RenderDelayController* RenderDelayController::Create(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.h
>index ddd95482fcce18b8bba92da9f4698525ea1c0894..b46ed892527f792881ef548c3b219698e2ceb8ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.h
>@@ -27,10 +27,13 @@ class RenderDelayController {
>   static RenderDelayController* Create(const EchoCanceller3Config& config,
>                                        int non_causal_offset,
>                                        int sample_rate_hz);
>+  static RenderDelayController* Create2(const EchoCanceller3Config& config,
>+                                        int sample_rate_hz);
>   virtual ~RenderDelayController() = default;
> 
>-  // Resets the delay controller.
>-  virtual void Reset() = 0;
>+  // Resets the delay controller. If the delay confidence is reset, the reset
>+  // behavior is as if the call is restarted.
>+  virtual void Reset(bool reset_delay_confidence) = 0;
> 
>   // Logs a render call.
>   virtual void LogRenderCall() = 0;
>@@ -41,6 +44,9 @@ class RenderDelayController {
>       size_t render_delay_buffer_delay,
>       const absl::optional<int>& echo_remover_delay,
>       rtc::ArrayView<const float> capture) = 0;
>+
>+  // Returns true if clockdrift has been detected.
>+  virtual bool HasClockdrift() const = 0;
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller2.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..00daf8f2af71c2ddabb374acadfedb796998e426
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller2.cc
>@@ -0,0 +1,218 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include <stddef.h>
>+#include <algorithm>
>+#include <memory>
>+
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/delay_estimate.h"
>+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
>+#include "modules/audio_processing/aec3/echo_path_delay_estimator.h"
>+#include "modules/audio_processing/aec3/render_delay_controller.h"
>+#include "modules/audio_processing/aec3/render_delay_controller_metrics.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/atomicops.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/constructormagic.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+bool UseEarlyDelayDetection() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3EarlyDelayDetectionKillSwitch");
>+}
>+
>+class RenderDelayControllerImpl2 final : public RenderDelayController {
>+ public:
>+  RenderDelayControllerImpl2(const EchoCanceller3Config& config,
>+                             int sample_rate_hz);
>+  ~RenderDelayControllerImpl2() override;
>+  void Reset(bool reset_delay_confidence) override;
>+  void LogRenderCall() override;
>+  absl::optional<DelayEstimate> GetDelay(
>+      const DownsampledRenderBuffer& render_buffer,
>+      size_t render_delay_buffer_delay,
>+      const absl::optional<int>& echo_remover_delay,
>+      rtc::ArrayView<const float> capture) override;
>+  bool HasClockdrift() const override;
>+
>+ private:
>+  static int instance_count_;
>+  std::unique_ptr<ApmDataDumper> data_dumper_;
>+  const bool use_early_delay_detection_;
>+  const int delay_headroom_blocks_;
>+  const int hysteresis_limit_1_blocks_;
>+  const int hysteresis_limit_2_blocks_;
>+  absl::optional<DelayEstimate> delay_;
>+  EchoPathDelayEstimator delay_estimator_;
>+  RenderDelayControllerMetrics metrics_;
>+  absl::optional<DelayEstimate> delay_samples_;
>+  size_t capture_call_counter_ = 0;
>+  int delay_change_counter_ = 0;
>+  DelayEstimate::Quality last_delay_estimate_quality_;
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderDelayControllerImpl2);
>+};
>+
>+DelayEstimate ComputeBufferDelay(
>+    const absl::optional<DelayEstimate>& current_delay,
>+    int delay_headroom_blocks,
>+    int hysteresis_limit_1_blocks,
>+    int hysteresis_limit_2_blocks,
>+    DelayEstimate estimated_delay) {
>+  // The below division is not exact and the truncation is intended.
>+  const int echo_path_delay_blocks = estimated_delay.delay >> kBlockSizeLog2;
>+
>+  // Compute the buffer delay increase required to achieve the desired latency.
>+  size_t new_delay_blocks =
>+      std::max(echo_path_delay_blocks - delay_headroom_blocks, 0);
>+
>+  // Add hysteresis.
>+  if (current_delay) {
>+    size_t current_delay_blocks = current_delay->delay;
>+    if (new_delay_blocks > current_delay_blocks) {
>+      if (new_delay_blocks <=
>+          current_delay_blocks + hysteresis_limit_1_blocks) {
>+        new_delay_blocks = current_delay_blocks;
>+      }
>+    } else if (new_delay_blocks < current_delay_blocks) {
>+      size_t hysteresis_limit = std::max(
>+          static_cast<int>(current_delay_blocks) - hysteresis_limit_2_blocks,
>+          0);
>+      if (new_delay_blocks >= hysteresis_limit) {
>+        new_delay_blocks = current_delay_blocks;
>+      }
>+    }
>+  }
>+
>+  DelayEstimate new_delay = estimated_delay;
>+  new_delay.delay = new_delay_blocks;
>+  return new_delay;
>+}
>+
>+int RenderDelayControllerImpl2::instance_count_ = 0;
>+
>+RenderDelayControllerImpl2::RenderDelayControllerImpl2(
>+    const EchoCanceller3Config& config,
>+    int sample_rate_hz)
>+    : data_dumper_(
>+          new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>+      use_early_delay_detection_(UseEarlyDelayDetection()),
>+      delay_headroom_blocks_(
>+          static_cast<int>(config.delay.delay_headroom_blocks)),
>+      hysteresis_limit_1_blocks_(
>+          static_cast<int>(config.delay.hysteresis_limit_1_blocks)),
>+      hysteresis_limit_2_blocks_(
>+          static_cast<int>(config.delay.hysteresis_limit_2_blocks)),
>+      delay_estimator_(data_dumper_.get(), config),
>+      last_delay_estimate_quality_(DelayEstimate::Quality::kCoarse) {
>+  RTC_DCHECK(ValidFullBandRate(sample_rate_hz));
>+  delay_estimator_.LogDelayEstimationProperties(sample_rate_hz, 0);
>+}
>+
>+RenderDelayControllerImpl2::~RenderDelayControllerImpl2() = default;
>+
>+void RenderDelayControllerImpl2::Reset(bool reset_delay_confidence) {
>+  delay_ = absl::nullopt;
>+  delay_samples_ = absl::nullopt;
>+  delay_estimator_.Reset(reset_delay_confidence);
>+  delay_change_counter_ = 0;
>+  if (reset_delay_confidence) {
>+    last_delay_estimate_quality_ = DelayEstimate::Quality::kCoarse;
>+  }
>+}
>+
>+void RenderDelayControllerImpl2::LogRenderCall() {}
>+
>+absl::optional<DelayEstimate> RenderDelayControllerImpl2::GetDelay(
>+    const DownsampledRenderBuffer& render_buffer,
>+    size_t render_delay_buffer_delay,
>+    const absl::optional<int>& echo_remover_delay,
>+    rtc::ArrayView<const float> capture) {
>+  RTC_DCHECK_EQ(kBlockSize, capture.size());
>+  ++capture_call_counter_;
>+
>+  auto delay_samples = delay_estimator_.EstimateDelay(render_buffer, capture);
>+
>+  // Overrule the delay estimator delay if the echo remover reports a delay.
>+  if (echo_remover_delay) {
>+    int total_echo_remover_delay_samples =
>+        (render_delay_buffer_delay + *echo_remover_delay) * kBlockSize;
>+    delay_samples = DelayEstimate(DelayEstimate::Quality::kRefined,
>+                                  total_echo_remover_delay_samples);
>+  }
>+
>+  if (delay_samples) {
>+    if (!delay_samples_ || delay_samples->delay != delay_samples_->delay) {
>+      delay_change_counter_ = 0;
>+    }
>+    if (delay_samples_) {
>+      delay_samples_->blocks_since_last_change =
>+          delay_samples_->delay == delay_samples->delay
>+              ? delay_samples_->blocks_since_last_change + 1
>+              : 0;
>+      delay_samples_->blocks_since_last_update = 0;
>+      delay_samples_->delay = delay_samples->delay;
>+      delay_samples_->quality = delay_samples->quality;
>+    } else {
>+      delay_samples_ = delay_samples;
>+    }
>+  } else {
>+    if (delay_samples_) {
>+      ++delay_samples_->blocks_since_last_change;
>+      ++delay_samples_->blocks_since_last_update;
>+    }
>+  }
>+
>+  if (delay_change_counter_ < 2 * kNumBlocksPerSecond) {
>+    ++delay_change_counter_;
>+  }
>+
>+  if (delay_samples_) {
>+    // Compute the render delay buffer delay.
>+    const bool use_hysteresis =
>+        last_delay_estimate_quality_ == DelayEstimate::Quality::kRefined &&
>+        delay_samples_->quality == DelayEstimate::Quality::kRefined;
>+    delay_ = ComputeBufferDelay(delay_, delay_headroom_blocks_,
>+                                use_hysteresis ? hysteresis_limit_1_blocks_ : 0,
>+                                use_hysteresis ? hysteresis_limit_2_blocks_ : 0,
>+                                *delay_samples_);
>+    last_delay_estimate_quality_ = delay_samples_->quality;
>+  }
>+
>+  metrics_.Update(delay_samples_ ? absl::optional<size_t>(delay_samples_->delay)
>+                                 : absl::nullopt,
>+                  delay_ ? delay_->delay : 0, 0, delay_estimator_.Clockdrift());
>+
>+  data_dumper_->DumpRaw("aec3_render_delay_controller_delay",
>+                        delay_samples ? delay_samples->delay : 0);
>+  data_dumper_->DumpRaw("aec3_render_delay_controller_buffer_delay",
>+                        delay_ ? delay_->delay : 0);
>+
>+  return delay_;
>+}
>+
>+bool RenderDelayControllerImpl2::HasClockdrift() const {
>+  return delay_estimator_.Clockdrift() != ClockdriftDetector::Level::kNone;
>+}
>+
>+}  // namespace
>+
>+RenderDelayController* RenderDelayController::Create2(
>+    const EchoCanceller3Config& config,
>+    int sample_rate_hz) {
>+  return new RenderDelayControllerImpl2(config, sample_rate_hz);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
>index 09db33968340e17567053764337cb9fe24c5d646..582e03348297ec39656f76953a2b4f35e6ae8035 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
>@@ -13,6 +13,7 @@
> #include <algorithm>
> 
> #include "modules/audio_processing/aec3/aec3_common.h"
>+#include "rtc_base/checks.h"
> #include "system_wrappers/include/metrics.h"
> 
> namespace webrtc {
>@@ -45,7 +46,8 @@ RenderDelayControllerMetrics::RenderDelayControllerMetrics() = default;
> void RenderDelayControllerMetrics::Update(
>     absl::optional<size_t> delay_samples,
>     size_t buffer_delay_blocks,
>-    absl::optional<int> skew_shift_blocks) {
>+    absl::optional<int> skew_shift_blocks,
>+    ClockdriftDetector::Level clockdrift) {
>   ++call_counter_;
> 
>   if (!initial_update) {
>@@ -114,6 +116,10 @@ void RenderDelayControllerMetrics::Update(
>         static_cast<int>(delay_changes),
>         static_cast<int>(DelayChangesCategory::kNumCategories));
> 
>+    RTC_HISTOGRAM_ENUMERATION(
>+        "WebRTC.Audio.EchoCanceller.Clockdrift", static_cast<int>(clockdrift),
>+        static_cast<int>(ClockdriftDetector::Level::kNumCategories));
>+
>     metrics_reported_ = true;
>     call_counter_ = 0;
>     ResetMetrics();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
>index 1cfe899b4885cb304b27aff5971eca9627057e2f..22cc202e73a52d316b128ea482bd6dddc3fb0066 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
>@@ -11,7 +11,10 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
> 
>+#include <stddef.h>
>+
> #include "absl/types/optional.h"
>+#include "modules/audio_processing/aec3/clockdrift_detector.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -24,7 +27,8 @@ class RenderDelayControllerMetrics {
>   // Updates the metric with new data.
>   void Update(absl::optional<size_t> delay_samples,
>               size_t buffer_delay_blocks,
>-              absl::optional<int> skew_shift_blocks);
>+              absl::optional<int> skew_shift_blocks,
>+              ClockdriftDetector::Level clockdrift);
> 
>   // Returns true if the metrics have just been reported, otherwise false.
>   bool MetricsReported() { return metrics_reported_; }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
>index e867de4df428280ddc6c1d4fee49040b13001d08..216b0e220d4b265777bd34bca920fd956bcbb3bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
>@@ -22,10 +22,12 @@ TEST(RenderDelayControllerMetrics, NormalUsage) {
> 
>   for (int j = 0; j < 3; ++j) {
>     for (int k = 0; k < kMetricsReportingIntervalBlocks - 1; ++k) {
>-      metrics.Update(absl::nullopt, 0, absl::nullopt);
>+      metrics.Update(absl::nullopt, 0, absl::nullopt,
>+                     ClockdriftDetector::Level::kNone);
>       EXPECT_FALSE(metrics.MetricsReported());
>     }
>-    metrics.Update(absl::nullopt, 0, absl::nullopt);
>+    metrics.Update(absl::nullopt, 0, absl::nullopt,
>+                   ClockdriftDetector::Level::kNone);
>     EXPECT_TRUE(metrics.MetricsReported());
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_unittest.cc
>index 93c64998e9bd40833584d937f26db112efbae507..e9f02d324c13d34ed4dc2a444e08ae33bc70d674 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_unittest.cc
>@@ -57,10 +57,9 @@ TEST(RenderDelayController, NoRenderSignal) {
>       for (auto rate : {8000, 16000, 32000, 48000}) {
>         SCOPED_TRACE(ProduceDebugText(rate));
>         std::unique_ptr<RenderDelayBuffer> delay_buffer(
>-            RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>+            RenderDelayBuffer::Create2(config, NumBandsForRate(rate)));
>         std::unique_ptr<RenderDelayController> delay_controller(
>-            RenderDelayController::Create(
>-                config, RenderDelayBuffer::DelayEstimatorOffset(config), rate));
>+            RenderDelayController::Create2(config, rate));
>         for (size_t k = 0; k < 100; ++k) {
>           auto delay = delay_controller->GetDelay(
>               delay_buffer->GetDownsampledRenderBuffer(), delay_buffer->Delay(),
>@@ -87,11 +86,9 @@ TEST(RenderDelayController, BasicApiCalls) {
>         std::vector<std::vector<float>> render_block(
>             NumBandsForRate(rate), std::vector<float>(kBlockSize, 0.f));
>         std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-            RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>+            RenderDelayBuffer::Create2(config, NumBandsForRate(rate)));
>         std::unique_ptr<RenderDelayController> delay_controller(
>-            RenderDelayController::Create(
>-                EchoCanceller3Config(),
>-                RenderDelayBuffer::DelayEstimatorOffset(config), rate));
>+            RenderDelayController::Create2(EchoCanceller3Config(), rate));
>         for (size_t k = 0; k < 10; ++k) {
>           render_delay_buffer->Insert(render_block);
>           render_delay_buffer->PrepareCaptureProcessing();
>@@ -128,11 +125,9 @@ TEST(RenderDelayController, Alignment) {
>           absl::optional<DelayEstimate> delay_blocks;
>           SCOPED_TRACE(ProduceDebugText(rate, delay_samples));
>           std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-              RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>+              RenderDelayBuffer::Create2(config, NumBandsForRate(rate)));
>           std::unique_ptr<RenderDelayController> delay_controller(
>-              RenderDelayController::Create(
>-                  config, RenderDelayBuffer::DelayEstimatorOffset(config),
>-                  rate));
>+              RenderDelayController::Create2(config, rate));
>           DelayBuffer<float> signal_delay_buffer(delay_samples);
>           for (size_t k = 0; k < (400 + delay_samples / kBlockSize); ++k) {
>             RandomizeSampleVector(&random_generator, render_block[0]);
>@@ -179,11 +174,9 @@ TEST(RenderDelayController, NonCausalAlignment) {
>           absl::optional<DelayEstimate> delay_blocks;
>           SCOPED_TRACE(ProduceDebugText(rate, -delay_samples));
>           std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-              RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>+              RenderDelayBuffer::Create2(config, NumBandsForRate(rate)));
>           std::unique_ptr<RenderDelayController> delay_controller(
>-              RenderDelayController::Create(
>-                  EchoCanceller3Config(),
>-                  RenderDelayBuffer::DelayEstimatorOffset(config), rate));
>+              RenderDelayController::Create2(EchoCanceller3Config(), rate));
>           DelayBuffer<float> signal_delay_buffer(-delay_samples);
>           for (int k = 0;
>                k < (400 - delay_samples / static_cast<int>(kBlockSize)); ++k) {
>@@ -223,11 +216,9 @@ TEST(RenderDelayController, AlignmentWithJitter) {
>           absl::optional<DelayEstimate> delay_blocks;
>           SCOPED_TRACE(ProduceDebugText(rate, delay_samples));
>           std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-              RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>+              RenderDelayBuffer::Create2(config, NumBandsForRate(rate)));
>           std::unique_ptr<RenderDelayController> delay_controller(
>-              RenderDelayController::Create(
>-                  config, RenderDelayBuffer::DelayEstimatorOffset(config),
>-                  rate));
>+              RenderDelayController::Create2(config, rate));
>           DelayBuffer<float> signal_delay_buffer(delay_samples);
>           for (size_t j = 0; j < (1000 + delay_samples / kBlockSize) /
>                                          config.delay.api_call_jitter_blocks +
>@@ -280,11 +271,10 @@ TEST(RenderDelayController, InitialHeadroom) {
>       for (auto rate : {8000, 16000, 32000, 48000}) {
>         SCOPED_TRACE(ProduceDebugText(rate));
>         std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-            RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>+            RenderDelayBuffer::Create2(config, NumBandsForRate(rate)));
> 
>         std::unique_ptr<RenderDelayController> delay_controller(
>-            RenderDelayController::Create(
>-                config, RenderDelayBuffer::DelayEstimatorOffset(config), rate));
>+            RenderDelayController::Create2(config, rate));
>       }
>     }
>   }
>@@ -300,12 +290,10 @@ TEST(RenderDelayController, WrongCaptureSize) {
>   for (auto rate : {8000, 16000, 32000, 48000}) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>     std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-        RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>+        RenderDelayBuffer::Create2(config, NumBandsForRate(rate)));
>     EXPECT_DEATH(
>         std::unique_ptr<RenderDelayController>(
>-            RenderDelayController::Create(
>-                EchoCanceller3Config(),
>-                RenderDelayBuffer::DelayEstimatorOffset(config), rate))
>+            RenderDelayController::Create2(EchoCanceller3Config(), rate))
>             ->GetDelay(render_delay_buffer->GetDownsampledRenderBuffer(),
>                        render_delay_buffer->Delay(), echo_remover_delay, block),
>         "");
>@@ -320,11 +308,10 @@ TEST(RenderDelayController, DISABLED_WrongSampleRate) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>     EchoCanceller3Config config;
>     std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-        RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>+        RenderDelayBuffer::Create2(config, NumBandsForRate(rate)));
>     EXPECT_DEATH(
>-        std::unique_ptr<RenderDelayController>(RenderDelayController::Create(
>-            EchoCanceller3Config(),
>-            RenderDelayBuffer::DelayEstimatorOffset(config), rate)),
>+        std::unique_ptr<RenderDelayController>(
>+            RenderDelayController::Create2(EchoCanceller3Config(), rate)),
>         "");
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_reverb_model.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_reverb_model.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..8ad54c0cc345f65d56df2e7f04fc399dfc56c32f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_reverb_model.cc
>@@ -0,0 +1,44 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/render_reverb_model.h"
>+
>+#include <algorithm>
>+
>+#include "api/array_view.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+RenderReverbModel::RenderReverbModel() {
>+  Reset();
>+}
>+
>+RenderReverbModel::~RenderReverbModel() = default;
>+
>+void RenderReverbModel::Reset() {
>+  render_reverb_.Reset();
>+}
>+
>+void RenderReverbModel::Apply(const VectorBuffer& spectrum_buffer,
>+                              int delay_blocks,
>+                              float reverb_decay,
>+                              rtc::ArrayView<float> reverb_power_spectrum) {
>+  int idx_at_delay =
>+      spectrum_buffer.OffsetIndex(spectrum_buffer.read, delay_blocks);
>+  int idx_past = spectrum_buffer.IncIndex(idx_at_delay);
>+  const auto& X2 = spectrum_buffer.buffer[idx_at_delay];
>+  RTC_DCHECK_EQ(X2.size(), reverb_power_spectrum.size());
>+  std::copy(X2.begin(), X2.end(), reverb_power_spectrum.begin());
>+  render_reverb_.AddReverbNoFreqShaping(spectrum_buffer.buffer[idx_past], 1.0f,
>+                                        reverb_decay, reverb_power_spectrum);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_reverb_model.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_reverb_model.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..d404a69ecd2f20c075979a89e54ce7c315449124
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_reverb_model.h
>@@ -0,0 +1,49 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_REVERB_MODEL_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_REVERB_MODEL_H_
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/aec3/reverb_model.h"
>+#include "modules/audio_processing/aec3/vector_buffer.h"
>+
>+namespace webrtc {
>+
>+// The RenderReverbModel class applies an exponential reverberant model over the
>+// render spectrum.
>+class RenderReverbModel {
>+ public:
>+  RenderReverbModel();
>+  ~RenderReverbModel();
>+
>+  // Resets the state.
>+  void Reset();
>+
>+  // Applies the reverberation model over the render spectrum. It also returns
>+  // the reverberation render power spectrum in the array reverb_power_spectrum.
>+  void Apply(const VectorBuffer& spectrum_buffer,
>+             int delay_blocks,
>+             float reverb_decay,
>+             rtc::ArrayView<float> reverb_power_spectrum);
>+
>+  // Gets the reverberation spectrum that was added to the render spectrum for
>+  // computing the reverberation render spectrum.
>+  rtc::ArrayView<const float> GetReverbContributionPowerSpectrum() const {
>+    return render_reverb_.GetPowerSpectrum();
>+  }
>+
>+ private:
>+  ReverbModel render_reverb_;
>+};
>+
>+}  // namespace webrtc.
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_RENDER_REVERB_MODEL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.cc
>index 50c34ce43f61a502edf2813cc07b044068f4141c..33b04bf70c9ff80d3e903911e793121f3c99d55c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.cc
>@@ -12,7 +12,10 @@
> 
> #include <math.h>
> #include <algorithm>
>+#include <utility>
>+#include <vector>
> 
>+#include "api/array_view.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
>index c603c921447b905cd0f42e5101389056683fbd6e..8a44232cf9a0d31c7fd5a2d6056b47e03ade3856 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
>@@ -11,13 +11,15 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
> 
>+#include <algorithm>
> #include <array>
>-#include <memory>
>+#include <cstddef>
> 
> #include "absl/types/optional.h"
> #include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
>index f9b195583875f02968900314a326fb662d28e9ac..a993f8f8acf89cd9b0235c38044571fc3259987c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
>@@ -59,7 +59,7 @@ TEST(RenderSignalAnalyzer, NoFalseDetectionOfNarrowBands) {
>   std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>   std::array<float, kBlockSize> x_old;
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(EchoCanceller3Config(), 3));
>+      RenderDelayBuffer::Create2(EchoCanceller3Config(), 3));
>   std::array<float, kFftLengthBy2Plus1> mask;
>   x_old.fill(0.f);
> 
>@@ -93,7 +93,7 @@ TEST(RenderSignalAnalyzer, NarrowBandDetection) {
>   EchoCanceller3Config config;
>   config.delay.min_echo_path_delay_blocks = 0;
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
> 
>   std::array<float, kFftLengthBy2Plus1> mask;
>   x_old.fill(0.f);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.cc
>index 2e3ad9fea727e29ea4cfc1706f293dff678d809a..627dd900c4e8950e0dc687bed54d3c5d6d2dbb4a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.cc
>@@ -11,9 +11,11 @@
> 
> #include "modules/audio_processing/aec3/residual_echo_estimator.h"
> 
>-#include <numeric>
>+#include <stddef.h>
>+#include <algorithm>
> #include <vector>
> 
>+#include "api/array_view.h"
> #include "modules/audio_processing/aec3/reverb_model.h"
> #include "modules/audio_processing/aec3/reverb_model_fallback.h"
> #include "rtc_base/checks.h"
>@@ -105,9 +107,15 @@ void ResidualEchoEstimator::Estimate(
> 
>   // Estimate the residual echo power.
>   if (aec_state.UsableLinearEstimate()) {
>-    RTC_DCHECK(!aec_state.SaturatedEcho());
>     LinearEstimate(S2_linear, aec_state.Erle(), aec_state.ErleUncertainty(),
>                    R2);
>+
>+    // When there is saturated echo, assume the same spectral content as is
>+    // present in the micropone signal.
>+    if (aec_state.SaturatedEcho()) {
>+      std::copy(Y2.begin(), Y2.end(), R2->begin());
>+    }
>+
>     // Adds the estimated unmodelled echo power to the residual echo power
>     // estimate.
>     if (echo_reverb_) {
>@@ -151,10 +159,10 @@ void ResidualEchoEstimator::Estimate(
>     }
>     NonLinearEstimate(echo_path_gain, X2, Y2, R2);
> 
>-    // If the echo is saturated, estimate the echo power as the maximum echo
>-    // power with a leakage factor.
>+    // When there is saturated echo, assume the same spectral content as is
>+    // present in the micropone signal.
>     if (aec_state.SaturatedEcho()) {
>-      R2->fill((*std::max_element(R2->begin(), R2->end())) * 100.f);
>+      std::copy(Y2.begin(), Y2.end(), R2->begin());
>     }
> 
>     if (!(aec_state.TransparentMode() && soft_transparent_mode_)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.h
>index 6dcf24f090790b724e8e2e4742cf33ca14213cae..52885a58d85a3382422cdf9ce22ef203ba129fbb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.h
>@@ -11,18 +11,18 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_RESIDUAL_ECHO_ESTIMATOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_RESIDUAL_ECHO_ESTIMATOR_H_
> 
>-#include <algorithm>
> #include <array>
> #include <memory>
>-#include <vector>
> 
>-#include "api/array_view.h"
>+#include "absl/types/optional.h"
> #include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/aec_state.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
> #include "modules/audio_processing/aec3/reverb_model.h"
> #include "modules/audio_processing/aec3/reverb_model_fallback.h"
>+#include "modules/audio_processing/aec3/vector_buffer.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -39,7 +39,7 @@ class ResidualEchoEstimator {
>                 std::array<float, kFftLengthBy2Plus1>* R2);
> 
>   // Returns the reverberant power spectrum contributions to the echo residual.
>-  const std::array<float, kFftLengthBy2Plus1>& GetReverbPowerSpectrum() const {
>+  rtc::ArrayView<const float> GetReverbPowerSpectrum() const {
>     if (echo_reverb_) {
>       return echo_reverb_->GetPowerSpectrum();
>     } else {
>@@ -66,7 +66,6 @@ class ResidualEchoEstimator {
>                          const std::array<float, kFftLengthBy2Plus1>& Y2,
>                          std::array<float, kFftLengthBy2Plus1>* R2);
> 
>-
>   // Estimates the echo generating signal power as gated maximal power over a
>   // time window.
>   void EchoGeneratingPower(const VectorBuffer& spectrum_buffer,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
>index 832d8ca1d33b6576489ae21b81025a9b663c3207..2e73a7e6116101fcc6bcca17f2c7081a511c9af1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
>@@ -27,7 +27,7 @@ TEST(ResidualEchoEstimator, NullResidualEchoPowerOutput) {
>   EchoCanceller3Config config;
>   AecState aec_state(config);
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
>   std::vector<std::array<float, kFftLengthBy2Plus1>> H2;
>   std::array<float, kFftLengthBy2Plus1> S2_linear;
>   std::array<float, kFftLengthBy2Plus1> Y2;
>@@ -48,7 +48,7 @@ TEST(ResidualEchoEstimator, DISABLED_BasicTest) {
>   ResidualEchoEstimator estimator(config);
>   AecState aec_state(config);
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
> 
>   std::array<float, kFftLengthBy2Plus1> E2_main;
>   std::array<float, kFftLengthBy2Plus1> E2_shadow;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.cc
>index f80afa2dfcba0963fe19c209d3203d177c0c8c59..95fd13af596269ae151afeaf874146a173c3a17f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.cc
>@@ -10,11 +10,13 @@
> 
> #include "modules/audio_processing/aec3/reverb_decay_estimator.h"
> 
>+#include <stddef.h>
> #include <algorithm>
> #include <cmath>
> #include <numeric>
> 
> #include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
> #include "system_wrappers/include/field_trial.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.h
>index 67a84ab8fcbf533d9c7af0da8e9928883a60a6d6..4c8d0c643462feae77eda9a45e5e1a175e061628 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.h
>@@ -16,12 +16,12 @@
> 
> #include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/audio/echo_canceller3_config.h"
>-#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"  // kMaxAdaptiveFilter...
> 
> namespace webrtc {
> 
> class ApmDataDumper;
>+struct EchoCanceller3Config;
> 
> // Class for estimating the decay of the late reverb.
> class ReverbDecayEstimator {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.cc
>index 0d82515ec4b25688d7354be132a82208a965dc6b..d2103d442e4b45ff84e16a76d604c8a04649b65e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.cc
>@@ -10,14 +10,12 @@
> 
> #include "modules/audio_processing/aec3/reverb_frequency_response.h"
> 
>+#include <stddef.h>
> #include <algorithm>
> #include <array>
>-#include <cmath>
>-#include <memory>
> #include <numeric>
> 
> #include "api/array_view.h"
>-#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "rtc_base/checks.h"
> #include "system_wrappers/include/field_trial.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.h
>index 23485e009fbe30eb6d327361be6a14073081b02c..eb63b8e52e854bcca9f184eefa6518739cd8b8a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.h
>@@ -11,13 +11,12 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_FREQUENCY_RESPONSE_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_REVERB_FREQUENCY_RESPONSE_H_
> 
>-#include <memory>
>+#include <array>
> #include <vector>
> 
> #include "absl/types/optional.h"
> #include "api/array_view.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>-#include "modules/audio_processing/logging/apm_data_dumper.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.cc
>index 0ca248fc75431ea63377c24ed693c6bec6b907e6..f0a24c0249823485d997361b7566db1926363444 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.cc
>@@ -10,13 +10,11 @@
> 
> #include "modules/audio_processing/aec3/reverb_model.h"
> 
>-#include <math.h>
>-
>+#include <stddef.h>
> #include <algorithm>
> #include <functional>
> 
> #include "api/array_view.h"
>-#include "modules/audio_processing/aec3/aec3_common.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.h
>index d3087a72d91023a53d7ca59e6e5fd054929738d5..56e2266e569e7447a1fb0fc2f6279820d997034a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.h
>@@ -11,6 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_H_
> 
>+#include <array>
>+
> #include "api/array_view.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> 
>@@ -50,9 +52,7 @@ class ReverbModel {
>       float reverb_decay);
> 
>   // Returns the current power spectrum reverberation contributions.
>-  const std::array<float, kFftLengthBy2Plus1>& GetPowerSpectrum() const {
>-    return reverb_;
>-  }
>+  rtc::ArrayView<const float> GetPowerSpectrum() const { return reverb_; }
> 
>  private:
>   // Updates the reverberation contributions.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator.h
>index b6a3591e091970b90c8ce012c01d7ca34967ca0c..1112f93a719354baafc3c98844cc0b941a5acf10 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator.h
>@@ -11,11 +11,13 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_ESTIMATOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_ESTIMATOR_H_
> 
>+#include <array>
> #include <vector>
> 
> #include "absl/types/optional.h"
> #include "api/array_view.h"
> #include "api/audio/echo_canceller3_config.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"  // kFftLengthBy2Plus1
> #include "modules/audio_processing/aec3/reverb_decay_estimator.h"
> #include "modules/audio_processing/aec3/reverb_frequency_response.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_fallback.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_fallback.h
>index 1b2a953ea6417c2bee806c637de6f12e473b5451..1bd2b594e06aac0ba509d6ecf5192501c8c6f1d0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_fallback.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_fallback.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_FALLBACK_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_FALLBACK_H_
> 
>+#include <stddef.h>
> #include <array>
> #include <vector>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h
>index a92bc3b8b7fa017a88a3eb94e0b75d2b26bbb8e6..05e632fa78b87313e2ef5d1db7fd774c5a167056 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h
>@@ -11,11 +11,13 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_SHADOW_FILTER_UPDATE_GAIN_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_SHADOW_FILTER_UPDATE_GAIN_H_
> 
>+#include <stddef.h>
>+#include <array>
>+
> #include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>-#include "modules/audio_processing/aec3/render_buffer.h"
>+#include "modules/audio_processing/aec3/fft_data.h"
> #include "modules/audio_processing/aec3/render_signal_analyzer.h"
>-#include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain_unittest.cc
>index c040bbff91f2b7bbbefa58d5412cf300004fe2de..017c679588b1b725de763860e609cdbabdc6c0c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain_unittest.cc
>@@ -53,7 +53,7 @@ void RunFilterUpdateTest(int num_blocks_to_process,
>   config.delay.min_echo_path_delay_blocks = 0;
>   config.delay.default_delay = 1;
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
> 
>   std::array<float, kBlockSize> x_old;
>   x_old.fill(0.f);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..32b36ab215a5471c1cb90d390515af738d9650c9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.cc
>@@ -0,0 +1,368 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/signal_dependent_erle_estimator.h"
>+
>+#include <algorithm>
>+#include <functional>
>+#include <numeric>
>+
>+#include "modules/audio_processing/aec3/vector_buffer.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+constexpr std::array<size_t, SignalDependentErleEstimator::kSubbands + 1>
>+    kBandBoundaries = {1, 8, 16, 24, 32, 48, kFftLengthBy2Plus1};
>+
>+std::array<size_t, kFftLengthBy2Plus1> FormSubbandMap() {
>+  std::array<size_t, kFftLengthBy2Plus1> map_band_to_subband;
>+  size_t subband = 1;
>+  for (size_t k = 0; k < map_band_to_subband.size(); ++k) {
>+    RTC_DCHECK_LT(subband, kBandBoundaries.size());
>+    if (k >= kBandBoundaries[subband]) {
>+      subband++;
>+      RTC_DCHECK_LT(k, kBandBoundaries[subband]);
>+    }
>+    map_band_to_subband[k] = subband - 1;
>+  }
>+  return map_band_to_subband;
>+}
>+
>+// Defines the size in blocks of the sections that are used for dividing the
>+// linear filter. The sections are split in a non-linear manner so that lower
>+// sections that typically represent the direct path have a larger resolution
>+// than the higher sections which typically represent more reverberant acoustic
>+// paths.
>+std::vector<size_t> DefineFilterSectionSizes(size_t delay_headroom_blocks,
>+                                             size_t num_blocks,
>+                                             size_t num_sections) {
>+  size_t filter_length_blocks = num_blocks - delay_headroom_blocks;
>+  std::vector<size_t> section_sizes(num_sections);
>+  size_t remaining_blocks = filter_length_blocks;
>+  size_t remaining_sections = num_sections;
>+  size_t estimator_size = 2;
>+  size_t idx = 0;
>+  while (remaining_sections > 1 &&
>+         remaining_blocks > estimator_size * remaining_sections) {
>+    RTC_DCHECK_LT(idx, section_sizes.size());
>+    section_sizes[idx] = estimator_size;
>+    remaining_blocks -= estimator_size;
>+    remaining_sections--;
>+    estimator_size *= 2;
>+    idx++;
>+  }
>+
>+  size_t last_groups_size = remaining_blocks / remaining_sections;
>+  for (; idx < num_sections; idx++) {
>+    section_sizes[idx] = last_groups_size;
>+  }
>+  section_sizes[num_sections - 1] +=
>+      remaining_blocks - last_groups_size * remaining_sections;
>+  return section_sizes;
>+}
>+
>+// Forms the limits in blocks for each filter section. Those sections
>+// are used for analyzing the echo estimates and investigating which
>+// linear filter sections contribute most to the echo estimate energy.
>+std::vector<size_t> SetSectionsBoundaries(size_t delay_headroom_blocks,
>+                                          size_t num_blocks,
>+                                          size_t num_sections) {
>+  std::vector<size_t> estimator_boundaries_blocks(num_sections + 1);
>+  if (estimator_boundaries_blocks.size() == 2) {
>+    estimator_boundaries_blocks[0] = 0;
>+    estimator_boundaries_blocks[1] = num_blocks;
>+    return estimator_boundaries_blocks;
>+  }
>+  RTC_DCHECK_GT(estimator_boundaries_blocks.size(), 2);
>+  const std::vector<size_t> section_sizes =
>+      DefineFilterSectionSizes(delay_headroom_blocks, num_blocks,
>+                               estimator_boundaries_blocks.size() - 1);
>+
>+  size_t idx = 0;
>+  size_t current_size_block = 0;
>+  RTC_DCHECK_EQ(section_sizes.size() + 1, estimator_boundaries_blocks.size());
>+  estimator_boundaries_blocks[0] = delay_headroom_blocks;
>+  for (size_t k = delay_headroom_blocks; k < num_blocks; ++k) {
>+    current_size_block++;
>+    if (current_size_block >= section_sizes[idx]) {
>+      idx = idx + 1;
>+      if (idx == section_sizes.size()) {
>+        break;
>+      }
>+      estimator_boundaries_blocks[idx] = k + 1;
>+      current_size_block = 0;
>+    }
>+  }
>+  estimator_boundaries_blocks[section_sizes.size()] = num_blocks;
>+  return estimator_boundaries_blocks;
>+}
>+
>+std::array<float, SignalDependentErleEstimator::kSubbands>
>+SetMaxErleSubbands(float max_erle_l, float max_erle_h, size_t limit_subband_l) {
>+  std::array<float, SignalDependentErleEstimator::kSubbands> max_erle;
>+  std::fill(max_erle.begin(), max_erle.begin() + limit_subband_l, max_erle_l);
>+  std::fill(max_erle.begin() + limit_subband_l, max_erle.end(), max_erle_h);
>+  return max_erle;
>+}
>+
>+}  // namespace
>+
>+SignalDependentErleEstimator::SignalDependentErleEstimator(
>+    const EchoCanceller3Config& config)
>+    : min_erle_(config.erle.min),
>+      num_sections_(config.erle.num_sections),
>+      num_blocks_(config.filter.main.length_blocks),
>+      delay_headroom_blocks_(config.delay.delay_headroom_blocks),
>+      band_to_subband_(FormSubbandMap()),
>+      max_erle_(SetMaxErleSubbands(config.erle.max_l,
>+                                   config.erle.max_h,
>+                                   band_to_subband_[kFftLengthBy2 / 2])),
>+      section_boundaries_blocks_(SetSectionsBoundaries(delay_headroom_blocks_,
>+                                                       num_blocks_,
>+                                                       num_sections_)),
>+      S2_section_accum_(num_sections_),
>+      erle_estimators_(num_sections_),
>+      correction_factors_(num_sections_) {
>+  RTC_DCHECK_LE(num_sections_, num_blocks_);
>+  RTC_DCHECK_GE(num_sections_, 1);
>+
>+  Reset();
>+}
>+
>+SignalDependentErleEstimator::~SignalDependentErleEstimator() = default;
>+
>+void SignalDependentErleEstimator::Reset() {
>+  erle_.fill(min_erle_);
>+  for (auto& erle : erle_estimators_) {
>+    erle.fill(min_erle_);
>+  }
>+  erle_ref_.fill(min_erle_);
>+  for (auto& factor : correction_factors_) {
>+    factor.fill(1.0f);
>+  }
>+  num_updates_.fill(0);
>+}
>+
>+// Updates the Erle estimate by analyzing the current input signals. It takes
>+// the render buffer and the filter frequency response in order to do an
>+// estimation of the number of sections of the linear filter that are needed
>+// for getting the majority of the energy in the echo estimate. Based on that
>+// number of sections, it updates the erle estimation by introducing a
>+// correction factor to the erle that is given as an input to this method.
>+void SignalDependentErleEstimator::Update(
>+    const RenderBuffer& render_buffer,
>+    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+        filter_frequency_response,
>+    rtc::ArrayView<const float> X2,
>+    rtc::ArrayView<const float> Y2,
>+    rtc::ArrayView<const float> E2,
>+    rtc::ArrayView<const float> average_erle,
>+    bool converged_filter) {
>+  RTC_DCHECK_GT(num_sections_, 1);
>+
>+  // Gets the number of filter sections that are needed for achieving 90 %
>+  // of the power spectrum energy of the echo estimate.
>+  std::array<size_t, kFftLengthBy2Plus1> n_active_sections;
>+  ComputeNumberOfActiveFilterSections(render_buffer, filter_frequency_response,
>+                                      n_active_sections);
>+
>+  if (converged_filter) {
>+    // Updates the correction factor that is used for correcting the erle and
>+    // adapt it to the particular characteristics of the input signal.
>+    UpdateCorrectionFactors(X2, Y2, E2, n_active_sections);
>+  }
>+
>+  // Applies the correction factor to the input erle for getting a more refined
>+  // erle estimation for the current input signal.
>+  for (size_t k = 0; k < kFftLengthBy2; ++k) {
>+    float correction_factor =
>+        correction_factors_[n_active_sections[k]][band_to_subband_[k]];
>+    erle_[k] = rtc::SafeClamp(average_erle[k] * correction_factor, min_erle_,
>+                              max_erle_[band_to_subband_[k]]);
>+  }
>+}
>+
>+void SignalDependentErleEstimator::Dump(
>+    const std::unique_ptr<ApmDataDumper>& data_dumper) const {
>+  for (auto& erle : erle_estimators_) {
>+    data_dumper->DumpRaw("aec3_all_erle", erle);
>+  }
>+  data_dumper->DumpRaw("aec3_ref_erle", erle_ref_);
>+  for (auto& factor : correction_factors_) {
>+    data_dumper->DumpRaw("aec3_erle_correction_factor", factor);
>+  }
>+  data_dumper->DumpRaw("aec3_erle", erle_);
>+}
>+
>+// Estimates for each band the smallest number of sections in the filter that
>+// together constitute 90% of the estimated echo energy.
>+void SignalDependentErleEstimator::ComputeNumberOfActiveFilterSections(
>+    const RenderBuffer& render_buffer,
>+    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+        filter_frequency_response,
>+    rtc::ArrayView<size_t> n_active_filter_sections) {
>+  RTC_DCHECK_GT(num_sections_, 1);
>+  // Computes an approximation of the power spectrum if the filter would have
>+  // been limited to a certain number of filter sections.
>+  ComputeEchoEstimatePerFilterSection(render_buffer, filter_frequency_response);
>+  // For each band, computes the number of filter sections that are needed for
>+  // achieving the 90 % energy in the echo estimate.
>+  ComputeActiveFilterSections(n_active_filter_sections);
>+}
>+
>+void SignalDependentErleEstimator::UpdateCorrectionFactors(
>+    rtc::ArrayView<const float> X2,
>+    rtc::ArrayView<const float> Y2,
>+    rtc::ArrayView<const float> E2,
>+    rtc::ArrayView<const size_t> n_active_sections) {
>+  constexpr float kX2BandEnergyThreshold = 44015068.0f;
>+  constexpr float kSmthConstantDecreases = 0.1f;
>+  constexpr float kSmthConstantIncreases = kSmthConstantDecreases / 2.f;
>+  auto subband_powers = [](rtc::ArrayView<const float> power_spectrum,
>+                           rtc::ArrayView<float> power_spectrum_subbands) {
>+    for (size_t subband = 0; subband < kSubbands; ++subband) {
>+      RTC_DCHECK_LE(kBandBoundaries[subband + 1], power_spectrum.size());
>+      power_spectrum_subbands[subband] = std::accumulate(
>+          power_spectrum.begin() + kBandBoundaries[subband],
>+          power_spectrum.begin() + kBandBoundaries[subband + 1], 0.f);
>+    }
>+  };
>+
>+  std::array<float, kSubbands> X2_subbands, E2_subbands, Y2_subbands;
>+  subband_powers(X2, X2_subbands);
>+  subband_powers(E2, E2_subbands);
>+  subband_powers(Y2, Y2_subbands);
>+  std::array<size_t, kSubbands> idx_subbands;
>+  for (size_t subband = 0; subband < kSubbands; ++subband) {
>+    // When aggregating the number of active sections in the filter for
>+    // different bands we choose to take the minimum of all of them. As an
>+    // example, if for one of the bands it is the direct path its main
>+    // contributor to the final echo estimate, we consider the direct path is
>+    // as well the main contributor for the subband that contains that
>+    // particular band. That aggregate number of sections will be later used as
>+    // the identifier of the erle estimator that needs to be updated.
>+    RTC_DCHECK_LE(kBandBoundaries[subband + 1], n_active_sections.size());
>+    idx_subbands[subband] = *std::min_element(
>+        n_active_sections.begin() + kBandBoundaries[subband],
>+        n_active_sections.begin() + kBandBoundaries[subband + 1]);
>+  }
>+
>+  std::array<float, kSubbands> new_erle;
>+  std::array<bool, kSubbands> is_erle_updated;
>+  is_erle_updated.fill(false);
>+  new_erle.fill(0.f);
>+  for (size_t subband = 0; subband < kSubbands; ++subband) {
>+    if (X2_subbands[subband] > kX2BandEnergyThreshold &&
>+        E2_subbands[subband] > 0) {
>+      new_erle[subband] = Y2_subbands[subband] / E2_subbands[subband];
>+      RTC_DCHECK_GT(new_erle[subband], 0);
>+      is_erle_updated[subband] = true;
>+      ++num_updates_[subband];
>+    }
>+  }
>+
>+  for (size_t subband = 0; subband < kSubbands; ++subband) {
>+    const size_t idx = idx_subbands[subband];
>+    RTC_DCHECK_LT(idx, erle_estimators_.size());
>+    float alpha = new_erle[subband] > erle_estimators_[idx][subband]
>+                      ? kSmthConstantIncreases
>+                      : kSmthConstantDecreases;
>+    alpha = static_cast<float>(is_erle_updated[subband]) * alpha;
>+    erle_estimators_[idx][subband] +=
>+        alpha * (new_erle[subband] - erle_estimators_[idx][subband]);
>+    erle_estimators_[idx][subband] = rtc::SafeClamp(
>+        erle_estimators_[idx][subband], min_erle_, max_erle_[subband]);
>+  }
>+
>+  for (size_t subband = 0; subband < kSubbands; ++subband) {
>+    float alpha = new_erle[subband] > erle_ref_[subband]
>+                      ? kSmthConstantIncreases
>+                      : kSmthConstantDecreases;
>+    alpha = static_cast<float>(is_erle_updated[subband]) * alpha;
>+    erle_ref_[subband] += alpha * (new_erle[subband] - erle_ref_[subband]);
>+    erle_ref_[subband] =
>+        rtc::SafeClamp(erle_ref_[subband], min_erle_, max_erle_[subband]);
>+  }
>+
>+  for (size_t subband = 0; subband < kSubbands; ++subband) {
>+    constexpr int kNumUpdateThr = 50;
>+    if (is_erle_updated[subband] && num_updates_[subband] > kNumUpdateThr) {
>+      const size_t idx = idx_subbands[subband];
>+      RTC_DCHECK_GT(erle_ref_[subband], 0.f);
>+      // Computes the ratio between the erle that is updated using all the
>+      // points and the erle that is updated only on signals that share the
>+      // same number of active filter sections.
>+      float new_correction_factor =
>+          erle_estimators_[idx][subband] / erle_ref_[subband];
>+
>+      correction_factors_[idx][subband] +=
>+          0.1f * (new_correction_factor - correction_factors_[idx][subband]);
>+    }
>+  }
>+}
>+
>+void SignalDependentErleEstimator::ComputeEchoEstimatePerFilterSection(
>+    const RenderBuffer& render_buffer,
>+    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+        filter_frequency_response) {
>+  const VectorBuffer& spectrum_render_buffer =
>+      render_buffer.GetSpectrumBuffer();
>+
>+  RTC_DCHECK_EQ(S2_section_accum_.size() + 1,
>+                section_boundaries_blocks_.size());
>+  size_t idx_render = render_buffer.Position();
>+  idx_render = spectrum_render_buffer.OffsetIndex(
>+      idx_render, section_boundaries_blocks_[0]);
>+
>+  for (size_t section = 0; section < num_sections_; ++section) {
>+    std::array<float, kFftLengthBy2Plus1> X2_section;
>+    std::array<float, kFftLengthBy2Plus1> H2_section;
>+    X2_section.fill(0.f);
>+    H2_section.fill(0.f);
>+    for (size_t block = section_boundaries_blocks_[section];
>+         block < section_boundaries_blocks_[section + 1]; ++block) {
>+      std::transform(X2_section.begin(), X2_section.end(),
>+                     spectrum_render_buffer.buffer[idx_render].begin(),
>+                     X2_section.begin(), std::plus<float>());
>+      std::transform(H2_section.begin(), H2_section.end(),
>+                     filter_frequency_response[block].begin(),
>+                     H2_section.begin(), std::plus<float>());
>+      idx_render = spectrum_render_buffer.IncIndex(idx_render);
>+    }
>+
>+    std::transform(X2_section.begin(), X2_section.end(), H2_section.begin(),
>+                   S2_section_accum_[section].begin(),
>+                   std::multiplies<float>());
>+  }
>+
>+  for (size_t section = 1; section < num_sections_; ++section) {
>+    std::transform(S2_section_accum_[section - 1].begin(),
>+                   S2_section_accum_[section - 1].end(),
>+                   S2_section_accum_[section].begin(),
>+                   S2_section_accum_[section].begin(), std::plus<float>());
>+  }
>+}
>+
>+void SignalDependentErleEstimator::ComputeActiveFilterSections(
>+    rtc::ArrayView<size_t> number_active_filter_sections) const {
>+  std::fill(number_active_filter_sections.begin(),
>+            number_active_filter_sections.end(), 0);
>+  for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
>+    size_t section = num_sections_;
>+    float target = 0.9f * S2_section_accum_[num_sections_ - 1][k];
>+    while (section > 0 && S2_section_accum_[section - 1][k] >= target) {
>+      number_active_filter_sections[k] = --section;
>+    }
>+  }
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..d8b56c2b20c382dd219c0dd27ad6535cbe495cd5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator.h
>@@ -0,0 +1,93 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SIGNAL_DEPENDENT_ERLE_ESTIMATOR_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_SIGNAL_DEPENDENT_ERLE_ESTIMATOR_H_
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/render_buffer.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+
>+namespace webrtc {
>+
>+// This class estimates the dependency of the Erle to the input signal. By
>+// looking at the input signal, an estimation on whether the current echo
>+// estimate is due to the direct path or to a more reverberant one is performed.
>+// Once that estimation is done, it is possible to refine the average Erle that
>+// this class receive as an input.
>+class SignalDependentErleEstimator {
>+ public:
>+  explicit SignalDependentErleEstimator(const EchoCanceller3Config& config);
>+
>+  ~SignalDependentErleEstimator();
>+
>+  void Reset();
>+
>+  // Returns the Erle per frequency subband.
>+  const std::array<float, kFftLengthBy2Plus1>& Erle() const { return erle_; }
>+
>+  // Updates the Erle estimate. The Erle that is passed as an input is required
>+  // to be an estimation of the average Erle achieved by the linear filter.
>+  void Update(const RenderBuffer& render_buffer,
>+              const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+                  filter_frequency_response,
>+              rtc::ArrayView<const float> X2,
>+              rtc::ArrayView<const float> Y2,
>+              rtc::ArrayView<const float> E2,
>+              rtc::ArrayView<const float> average_erle,
>+              bool converged_filter);
>+
>+  void Dump(const std::unique_ptr<ApmDataDumper>& data_dumper) const;
>+
>+  static constexpr size_t kSubbands = 6;
>+
>+ private:
>+  void ComputeNumberOfActiveFilterSections(
>+      const RenderBuffer& render_buffer,
>+      const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+          filter_frequency_response,
>+      rtc::ArrayView<size_t> n_active_filter_sections);
>+
>+  void UpdateCorrectionFactors(rtc::ArrayView<const float> X2,
>+                               rtc::ArrayView<const float> Y2,
>+                               rtc::ArrayView<const float> E2,
>+                               rtc::ArrayView<const size_t> n_active_sections);
>+
>+  void ComputeEchoEstimatePerFilterSection(
>+      const RenderBuffer& render_buffer,
>+      const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+          filter_frequency_response);
>+
>+  void ComputeActiveFilterSections(
>+      rtc::ArrayView<size_t> number_active_filter_sections) const;
>+
>+  const float min_erle_;
>+  const size_t num_sections_;
>+  const size_t num_blocks_;
>+  const size_t delay_headroom_blocks_;
>+  const std::array<size_t, kFftLengthBy2Plus1> band_to_subband_;
>+  const std::array<float, kSubbands> max_erle_;
>+  const std::vector<size_t> section_boundaries_blocks_;
>+  std::array<float, kFftLengthBy2Plus1> erle_;
>+  std::vector<std::array<float, kFftLengthBy2Plus1>> S2_section_accum_;
>+  std::vector<std::array<float, kSubbands>> erle_estimators_;
>+  std::array<float, kSubbands> erle_ref_;
>+  std::vector<std::array<float, kSubbands>> correction_factors_;
>+  std::array<int, kSubbands> num_updates_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SIGNAL_DEPENDENT_ERLE_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..aec605f7abf0285d03552b2c11fb3a5322698ac3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/signal_dependent_erle_estimator_unittest.cc
>@@ -0,0 +1,155 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/signal_dependent_erle_estimator.h"
>+
>+#include <algorithm>
>+#include <iostream>
>+#include <string>
>+
>+#include "api/audio/echo_canceller3_config.h"
>+#include "modules/audio_processing/aec3/render_buffer.h"
>+#include "modules/audio_processing/aec3/render_delay_buffer.h"
>+#include "rtc_base/strings/string_builder.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+void GetActiveFrame(rtc::ArrayView<float> x) {
>+  const std::array<float, kBlockSize> frame = {
>+      7459.88, 17209.6, 17383,   20768.9, 16816.7, 18386.3, 4492.83, 9675.85,
>+      6665.52, 14808.6, 9342.3,  7483.28, 19261.7, 4145.98, 1622.18, 13475.2,
>+      7166.32, 6856.61, 21937,   7263.14, 9569.07, 14919,   8413.32, 7551.89,
>+      7848.65, 6011.27, 13080.6, 15865.2, 12656,   17459.6, 4263.93, 4503.03,
>+      9311.79, 21095.8, 12657.9, 13906.6, 19267.2, 11338.1, 16828.9, 11501.6,
>+      11405,   15031.4, 14541.6, 19765.5, 18346.3, 19350.2, 3157.47, 18095.8,
>+      1743.68, 21328.2, 19727.5, 7295.16, 10332.4, 11055.5, 20107.4, 14708.4,
>+      12416.2, 16434,   2454.69, 9840.8,  6867.23, 1615.75, 6059.9,  8394.19};
>+  RTC_DCHECK_GE(x.size(), frame.size());
>+  std::copy(frame.begin(), frame.end(), x.begin());
>+}
>+
>+class TestInputs {
>+ public:
>+  explicit TestInputs(const EchoCanceller3Config& cfg);
>+  ~TestInputs();
>+  const RenderBuffer& GetRenderBuffer() { return *render_buffer_; }
>+  rtc::ArrayView<const float> GetX2() { return X2_; }
>+  rtc::ArrayView<const float> GetY2() { return Y2_; }
>+  rtc::ArrayView<const float> GetE2() { return E2_; }
>+  std::vector<std::array<float, kFftLengthBy2Plus1>> GetH2() { return H2_; }
>+  void Update();
>+
>+ private:
>+  void UpdateCurrentPowerSpectra();
>+  int n_ = 0;
>+  std::unique_ptr<RenderDelayBuffer> render_delay_buffer_;
>+  RenderBuffer* render_buffer_;
>+  std::array<float, kFftLengthBy2Plus1> X2_;
>+  std::array<float, kFftLengthBy2Plus1> Y2_;
>+  std::array<float, kFftLengthBy2Plus1> E2_;
>+  std::vector<std::array<float, kFftLengthBy2Plus1>> H2_;
>+  std::vector<std::vector<float>> x_;
>+};
>+
>+TestInputs::TestInputs(const EchoCanceller3Config& cfg)
>+    : render_delay_buffer_(RenderDelayBuffer::Create2(cfg, 1)),
>+      H2_(cfg.filter.main.length_blocks),
>+      x_(1, std::vector<float>(kBlockSize, 0.f)) {
>+  render_delay_buffer_->SetDelay(4);
>+  render_buffer_ = render_delay_buffer_->GetRenderBuffer();
>+  for (auto& H : H2_) {
>+    H.fill(0.f);
>+  }
>+  H2_[0].fill(1.0f);
>+}
>+
>+TestInputs::~TestInputs() = default;
>+
>+void TestInputs::Update() {
>+  if (n_ % 2 == 0) {
>+    std::fill(x_[0].begin(), x_[0].end(), 0.f);
>+  } else {
>+    GetActiveFrame(x_[0]);
>+  }
>+
>+  render_delay_buffer_->Insert(x_);
>+  render_delay_buffer_->PrepareCaptureProcessing();
>+  UpdateCurrentPowerSpectra();
>+  ++n_;
>+}
>+
>+void TestInputs::UpdateCurrentPowerSpectra() {
>+  const VectorBuffer& spectrum_render_buffer =
>+      render_buffer_->GetSpectrumBuffer();
>+  size_t idx = render_buffer_->Position();
>+  size_t prev_idx = spectrum_render_buffer.OffsetIndex(idx, 1);
>+  auto& X2 = spectrum_render_buffer.buffer[idx];
>+  auto& X2_prev = spectrum_render_buffer.buffer[prev_idx];
>+  std::copy(X2.begin(), X2.end(), X2_.begin());
>+  RTC_DCHECK_EQ(X2.size(), Y2_.size());
>+  for (size_t k = 0; k < X2.size(); ++k) {
>+    E2_[k] = 0.01f * X2_prev[k];
>+    Y2_[k] = X2[k] + E2_[k];
>+  }
>+}
>+
>+}  // namespace
>+
>+TEST(SignalDependentErleEstimator, SweepSettings) {
>+  EchoCanceller3Config cfg;
>+  size_t max_length_blocks = 50;
>+  for (size_t blocks = 0; blocks < max_length_blocks; blocks = blocks + 10) {
>+    for (size_t delay_headroom = 0; delay_headroom < 5; ++delay_headroom) {
>+      for (size_t num_sections = 2; num_sections < max_length_blocks;
>+           ++num_sections) {
>+        cfg.filter.main.length_blocks = blocks;
>+        cfg.filter.main_initial.length_blocks =
>+            std::min(cfg.filter.main_initial.length_blocks, blocks);
>+        cfg.delay.delay_headroom_blocks = delay_headroom;
>+        cfg.erle.num_sections = num_sections;
>+        if (EchoCanceller3Config::Validate(&cfg)) {
>+          SignalDependentErleEstimator s(cfg);
>+          std::array<float, kFftLengthBy2Plus1> average_erle;
>+          average_erle.fill(cfg.erle.max_l);
>+          TestInputs inputs(cfg);
>+          for (size_t n = 0; n < 10; ++n) {
>+            inputs.Update();
>+            s.Update(inputs.GetRenderBuffer(), inputs.GetH2(), inputs.GetX2(),
>+                     inputs.GetY2(), inputs.GetE2(), average_erle, true);
>+          }
>+        }
>+      }
>+    }
>+  }
>+}
>+
>+TEST(SignalDependentErleEstimator, LongerRun) {
>+  EchoCanceller3Config cfg;
>+  cfg.filter.main.length_blocks = 2;
>+  cfg.filter.main_initial.length_blocks = 1;
>+  cfg.delay.delay_headroom_blocks = 0;
>+  cfg.delay.hysteresis_limit_1_blocks = 0;
>+  cfg.erle.num_sections = 2;
>+  EXPECT_EQ(EchoCanceller3Config::Validate(&cfg), true);
>+  std::array<float, kFftLengthBy2Plus1> average_erle;
>+  average_erle.fill(cfg.erle.max_l);
>+  SignalDependentErleEstimator s(cfg);
>+  TestInputs inputs(cfg);
>+  for (size_t n = 0; n < 200; ++n) {
>+    inputs.Update();
>+    s.Update(inputs.GetRenderBuffer(), inputs.GetH2(), inputs.GetX2(),
>+             inputs.GetY2(), inputs.GetE2(), average_erle, true);
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.cc
>index 310e4e9dd1da22c541ba3a0708b4a44fef4b05fb..a2099fcd01301c2b95848368a7341fcd6d9e7735 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.cc
>@@ -10,7 +10,6 @@
> #include "modules/audio_processing/aec3/skew_estimator.h"
> 
> #include <algorithm>
>-#include <numeric>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.h
>index ff2526025a87faafafe67dd09c30b9f747c535bc..b0946d833e543a872a6ab23c7ddf028d9ecca651 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_SKEW_ESTIMATOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_SKEW_ESTIMATOR_H_
> 
>+#include <stddef.h>
> #include <vector>
> 
> #include "absl/types/optional.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.cc
>index efeabf1117f2098f434f728997657a64529e88c4..25100bfd4af56e745c95cfd6d86676df3c7c2081 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.cc
>@@ -14,6 +14,7 @@
> #include <array>
> #include <vector>
> 
>+#include "api/array_view.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/vector_buffer.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>@@ -40,7 +41,6 @@ void StationarityEstimator::Reset() {
>   noise_.Reset();
>   hangovers_.fill(0);
>   stationarity_flags_.fill(false);
>-  render_reverb_.Reset();
> }
> 
> // Update just the noise estimator. Usefull until the delay is known
>@@ -54,9 +54,9 @@ void StationarityEstimator::UpdateNoiseEstimator(
> 
> void StationarityEstimator::UpdateStationarityFlags(
>     const VectorBuffer& spectrum_buffer,
>+    rtc::ArrayView<const float> render_reverb_contribution_spectrum,
>     int idx_current,
>-    int num_lookahead,
>-    float reverb_decay) {
>+    int num_lookahead) {
>   std::array<int, kWindowLength> indexes;
>   int num_lookahead_bounded = std::min(num_lookahead, kWindowLength - 1);
>   int idx = idx_current;
>@@ -79,12 +79,9 @@ void StationarityEstimator::UpdateStationarityFlags(
>       spectrum_buffer.DecIndex(indexes[kWindowLength - 1]),
>       spectrum_buffer.OffsetIndex(idx_current, -(num_lookahead_bounded + 1)));
> 
>-  int idx_past = spectrum_buffer.IncIndex(idx_current);
>-  render_reverb_.UpdateReverbContributionsNoFreqShaping(
>-      spectrum_buffer.buffer[idx_past], 1.0f, reverb_decay);
>   for (size_t k = 0; k < stationarity_flags_.size(); ++k) {
>     stationarity_flags_[k] = EstimateBandStationarity(
>-        spectrum_buffer, render_reverb_.GetPowerSpectrum(), indexes, k);
>+        spectrum_buffer, render_reverb_contribution_spectrum, indexes, k);
>   }
>   UpdateHangover();
>   SmoothStationaryPerFreq();
>@@ -102,7 +99,7 @@ bool StationarityEstimator::IsBlockStationary() const {
> 
> bool StationarityEstimator::EstimateBandStationarity(
>     const VectorBuffer& spectrum_buffer,
>-    const std::array<float, kFftLengthBy2Plus1>& reverb,
>+    rtc::ArrayView<const float> reverb,
>     const std::array<int, kWindowLength>& indexes,
>     size_t band) const {
>   constexpr float kThrStationarity = 10.f;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.h
>index e2c5a6253471f8486f11bee7c6193774c54ae6d9..704859a1fe783c59c1b8eef79461c550f19e400e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.h
>@@ -11,18 +11,19 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_STATIONARITY_ESTIMATOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_STATIONARITY_ESTIMATOR_H_
> 
>+#include <stddef.h>
> #include <array>
> #include <memory>
>-#include <vector>
> 
> #include "api/array_view.h"
>-#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"  // kFftLengthBy2Plus1...
> #include "modules/audio_processing/aec3/reverb_model.h"
>-#include "modules/audio_processing/aec3/vector_buffer.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
> class ApmDataDumper;
>+struct VectorBuffer;
> 
> class StationarityEstimator {
>  public:
>@@ -37,10 +38,11 @@ class StationarityEstimator {
> 
>   // Update the flag indicating whether this current frame is stationary. For
>   // getting a more robust estimation, it looks at future and/or past frames.
>-  void UpdateStationarityFlags(const VectorBuffer& spectrum_buffer,
>-                               int idx_current,
>-                               int num_lookahead,
>-                               float reverb_decay);
>+  void UpdateStationarityFlags(
>+      const VectorBuffer& spectrum_buffer,
>+      rtc::ArrayView<const float> render_reverb_contribution_spectrum,
>+      int idx_current,
>+      int num_lookahead);
> 
>   // Returns true if the current band is stationary.
>   bool IsBandStationary(size_t band) const {
>@@ -57,11 +59,10 @@ class StationarityEstimator {
> 
>   // Get an estimation of the stationarity for the current band by looking
>   // at the past/present/future available data.
>-  bool EstimateBandStationarity(
>-      const VectorBuffer& spectrum_buffer,
>-      const std::array<float, kFftLengthBy2Plus1>& reverb,
>-      const std::array<int, kWindowLength>& indexes,
>-      size_t band) const;
>+  bool EstimateBandStationarity(const VectorBuffer& spectrum_buffer,
>+                                rtc::ArrayView<const float> reverb,
>+                                const std::array<int, kWindowLength>& indexes,
>+                                size_t band) const;
> 
>   // True if all bands at the current point are stationary.
>   bool AreAllBandsStationary();
>@@ -111,7 +112,6 @@ class StationarityEstimator {
>   NoiseSpectrum noise_;
>   std::array<int, kFftLengthBy2Plus1> hangovers_;
>   std::array<bool, kFftLengthBy2Plus1> stationarity_flags_;
>-  ReverbModel render_reverb_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subband_erle_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subband_erle_estimator.cc
>index d8cb7a76315f26624a106540d75e69a39cecd725..9453e5739f0d6a39143088739b311875b35792b7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subband_erle_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subband_erle_estimator.cc
>@@ -11,35 +11,38 @@
> #include "modules/audio_processing/aec3/subband_erle_estimator.h"
> 
> #include <algorithm>
>-#include <memory>
>+#include <functional>
> 
>-#include "absl/types/optional.h"
>-#include "api/array_view.h"
>-#include "modules/audio_processing/aec3/aec3_common.h"
>-#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_minmax.h"
> #include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
> namespace {
>-constexpr int kPointsToAccumulate = 6;
>+
> constexpr float kX2BandEnergyThreshold = 44015068.0f;
>-constexpr int kErleHold = 100;
>-constexpr int kBlocksForOnsetDetection = kErleHold + 150;
>+constexpr int kBlocksToHoldErle = 100;
>+constexpr int kBlocksForOnsetDetection = kBlocksToHoldErle + 150;
>+constexpr int kPointsToAccumulate = 6;
> 
> bool EnableAdaptErleOnLowRender() {
>   return !field_trial::IsEnabled("WebRTC-Aec3AdaptErleOnLowRenderKillSwitch");
> }
> 
>+std::array<float, kFftLengthBy2Plus1> SetMaxErleBands(float max_erle_l,
>+                                                      float max_erle_h) {
>+  std::array<float, kFftLengthBy2Plus1> max_erle;
>+  std::fill(max_erle.begin(), max_erle.begin() + kFftLengthBy2 / 2, max_erle_l);
>+  std::fill(max_erle.begin() + kFftLengthBy2 / 2, max_erle.end(), max_erle_h);
>+  return max_erle;
>+}
>+
> }  // namespace
> 
>-SubbandErleEstimator::SubbandErleEstimator(float min_erle,
>-                                           float max_erle_lf,
>-                                           float max_erle_hf)
>-    : min_erle_(min_erle),
>-      max_erle_lf_(max_erle_lf),
>-      max_erle_hf_(max_erle_hf),
>+SubbandErleEstimator::SubbandErleEstimator(const EchoCanceller3Config& config)
>+    : min_erle_(config.erle.min),
>+      max_erle_(SetMaxErleBands(config.erle.max_l, config.erle.max_h)),
>       adapt_on_low_render_(EnableAdaptErleOnLowRender()) {
>   Reset();
> }
>@@ -49,8 +52,9 @@ SubbandErleEstimator::~SubbandErleEstimator() = default;
> void SubbandErleEstimator::Reset() {
>   erle_.fill(min_erle_);
>   erle_onsets_.fill(min_erle_);
>-  hold_counters_.fill(0);
>   coming_onset_.fill(true);
>+  hold_counters_.fill(0);
>+  ResetAccumulatedSpectra();
> }
> 
> void SubbandErleEstimator::Update(rtc::ArrayView<const float> X2,
>@@ -62,10 +66,8 @@ void SubbandErleEstimator::Update(rtc::ArrayView<const float> X2,
>     // Note that the use of the converged_filter flag already imposed
>     // a minimum of the erle that can be estimated as that flag would
>     // be false if the filter is performing poorly.
>-    constexpr size_t kFftLengthBy4 = kFftLengthBy2 / 2;
>-    UpdateBands(X2, Y2, E2, 1, kFftLengthBy4, max_erle_lf_, onset_detection);
>-    UpdateBands(X2, Y2, E2, kFftLengthBy4, kFftLengthBy2, max_erle_hf_,
>-                onset_detection);
>+    UpdateAccumulatedSpectra(X2, Y2, E2);
>+    UpdateBands(onset_detection);
>   }
> 
>   if (onset_detection) {
>@@ -78,53 +80,45 @@ void SubbandErleEstimator::Update(rtc::ArrayView<const float> X2,
> 
> void SubbandErleEstimator::Dump(
>     const std::unique_ptr<ApmDataDumper>& data_dumper) const {
>-  data_dumper->DumpRaw("aec3_erle", Erle());
>   data_dumper->DumpRaw("aec3_erle_onset", ErleOnsets());
> }
> 
>-void SubbandErleEstimator::UpdateBands(rtc::ArrayView<const float> X2,
>-                                       rtc::ArrayView<const float> Y2,
>-                                       rtc::ArrayView<const float> E2,
>-                                       size_t start,
>-                                       size_t stop,
>-                                       float max_erle,
>-                                       bool onset_detection) {
>-  auto erle_band_update = [](float erle_band, float new_erle,
>-                             bool low_render_energy, float alpha_inc,
>-                             float alpha_dec, float min_erle, float max_erle) {
>-    if (new_erle < erle_band && low_render_energy) {
>-      // Decreases are not allowed if low render energy signals were used for
>-      // the erle computation.
>-      return erle_band;
>+void SubbandErleEstimator::UpdateBands(bool onset_detection) {
>+  std::array<float, kFftLengthBy2> new_erle;
>+  std::array<bool, kFftLengthBy2> is_erle_updated;
>+  is_erle_updated.fill(false);
>+
>+  for (size_t k = 1; k < kFftLengthBy2; ++k) {
>+    if (accum_spectra_.num_points_[k] == kPointsToAccumulate &&
>+        accum_spectra_.E2_[k] > 0.f) {
>+      new_erle[k] = accum_spectra_.Y2_[k] / accum_spectra_.E2_[k];
>+      is_erle_updated[k] = true;
>     }
>-    float alpha = new_erle > erle_band ? alpha_inc : alpha_dec;
>-    float erle_band_out = erle_band;
>-    erle_band_out = erle_band + alpha * (new_erle - erle_band);
>-    erle_band_out = rtc::SafeClamp(erle_band_out, min_erle, max_erle);
>-    return erle_band_out;
>-  };
>-
>-  for (size_t k = start; k < stop; ++k) {
>-    if (adapt_on_low_render_ || X2[k] > kX2BandEnergyThreshold) {
>-      bool low_render_energy = false;
>-      absl::optional<float> new_erle = instantaneous_erle_.Update(
>-          X2[k], Y2[k], E2[k], k, &low_render_energy);
>-      if (new_erle) {
>-        RTC_DCHECK(adapt_on_low_render_ || !low_render_energy);
>-        if (onset_detection && !low_render_energy) {
>-          if (coming_onset_[k]) {
>-            coming_onset_[k] = false;
>-            erle_onsets_[k] = erle_band_update(
>-                erle_onsets_[k], new_erle.value(), low_render_energy, 0.15f,
>-                0.3f, min_erle_, max_erle);
>-          }
>-          hold_counters_[k] = kBlocksForOnsetDetection;
>+  }
>+
>+  if (onset_detection) {
>+    for (size_t k = 1; k < kFftLengthBy2; ++k) {
>+      if (is_erle_updated[k] && !accum_spectra_.low_render_energy_[k]) {
>+        if (coming_onset_[k]) {
>+          coming_onset_[k] = false;
>+          float alpha = new_erle[k] < erle_onsets_[k] ? 0.3f : 0.15f;
>+          erle_onsets_[k] = rtc::SafeClamp(
>+              erle_onsets_[k] + alpha * (new_erle[k] - erle_onsets_[k]),
>+              min_erle_, max_erle_[k]);
>         }
>+        hold_counters_[k] = kBlocksForOnsetDetection;
>+      }
>+    }
>+  }
> 
>-        erle_[k] =
>-            erle_band_update(erle_[k], new_erle.value(), low_render_energy,
>-                             0.05f, 0.1f, min_erle_, max_erle);
>+  for (size_t k = 1; k < kFftLengthBy2; ++k) {
>+    if (is_erle_updated[k]) {
>+      float alpha = 0.05f;
>+      if (new_erle[k] < erle_[k]) {
>+        alpha = accum_spectra_.low_render_energy_[k] ? 0.f : 0.1f;
>       }
>+      erle_[k] = rtc::SafeClamp(erle_[k] + alpha * (new_erle[k] - erle_[k]),
>+                                min_erle_, max_erle_[k]);
>     }
>   }
> }
>@@ -132,7 +126,7 @@ void SubbandErleEstimator::UpdateBands(rtc::ArrayView<const float> X2,
> void SubbandErleEstimator::DecreaseErlePerBandForLowRenderSignals() {
>   for (size_t k = 1; k < kFftLengthBy2; ++k) {
>     hold_counters_[k]--;
>-    if (hold_counters_[k] <= (kBlocksForOnsetDetection - kErleHold)) {
>+    if (hold_counters_[k] <= (kBlocksForOnsetDetection - kBlocksToHoldErle)) {
>       if (erle_[k] > erle_onsets_[k]) {
>         erle_[k] = std::max(erle_onsets_[k], 0.97f * erle_[k]);
>         RTC_DCHECK_LE(min_erle_, erle_[k]);
>@@ -145,43 +139,55 @@ void SubbandErleEstimator::DecreaseErlePerBandForLowRenderSignals() {
>   }
> }
> 
>-SubbandErleEstimator::ErleInstantaneous::ErleInstantaneous() {
>-  Reset();
>+void SubbandErleEstimator::ResetAccumulatedSpectra() {
>+  accum_spectra_.Y2_.fill(0.f);
>+  accum_spectra_.E2_.fill(0.f);
>+  accum_spectra_.num_points_.fill(0);
>+  accum_spectra_.low_render_energy_.fill(false);
> }
> 
>-SubbandErleEstimator::ErleInstantaneous::~ErleInstantaneous() = default;
>-
>-absl::optional<float> SubbandErleEstimator::ErleInstantaneous::Update(
>-    float X2,
>-    float Y2,
>-    float E2,
>-    size_t band,
>-    bool* low_render_energy) {
>-  absl::optional<float> erle_instantaneous = absl::nullopt;
>-  RTC_DCHECK_LT(band, kFftLengthBy2Plus1);
>-  Y2_acum_[band] += Y2;
>-  E2_acum_[band] += E2;
>-  low_render_energy_[band] =
>-      low_render_energy_[band] || X2 < kX2BandEnergyThreshold;
>-  if (++num_points_[band] == kPointsToAccumulate) {
>-    if (E2_acum_[band]) {
>-      erle_instantaneous = Y2_acum_[band] / E2_acum_[band];
>+void SubbandErleEstimator::UpdateAccumulatedSpectra(
>+    rtc::ArrayView<const float> X2,
>+    rtc::ArrayView<const float> Y2,
>+    rtc::ArrayView<const float> E2) {
>+  auto& st = accum_spectra_;
>+  if (adapt_on_low_render_) {
>+    if (st.num_points_[0] == kPointsToAccumulate) {
>+      st.num_points_[0] = 0;
>+      st.Y2_.fill(0.f);
>+      st.E2_.fill(0.f);
>+      st.low_render_energy_.fill(false);
>+    }
>+    std::transform(Y2.begin(), Y2.end(), st.Y2_.begin(), st.Y2_.begin(),
>+                   std::plus<float>());
>+    std::transform(E2.begin(), E2.end(), st.E2_.begin(), st.E2_.begin(),
>+                   std::plus<float>());
>+
>+    for (size_t k = 0; k < X2.size(); ++k) {
>+      st.low_render_energy_[k] =
>+          st.low_render_energy_[k] || X2[k] < kX2BandEnergyThreshold;
>+    }
>+    st.num_points_[0]++;
>+    st.num_points_.fill(st.num_points_[0]);
>+
>+  } else {
>+    // The update is always done using high render energy signals and
>+    // therefore the field accum_spectra_.low_render_energy_ does not need to
>+    // be modified.
>+    for (size_t k = 0; k < X2.size(); ++k) {
>+      if (X2[k] > kX2BandEnergyThreshold) {
>+        if (st.num_points_[k] == kPointsToAccumulate) {
>+          st.Y2_[k] = 0.f;
>+          st.E2_[k] = 0.f;
>+          st.num_points_[k] = 0;
>+        }
>+        st.Y2_[k] += Y2[k];
>+        st.E2_[k] += E2[k];
>+        st.num_points_[k]++;
>+      }
>+      RTC_DCHECK_EQ(st.low_render_energy_[k], false);
>     }
>-    *low_render_energy = low_render_energy_[band];
>-    num_points_[band] = 0;
>-    Y2_acum_[band] = 0.f;
>-    E2_acum_[band] = 0.f;
>-    low_render_energy_[band] = false;
>   }
>-
>-  return erle_instantaneous;
>-}
>-
>-void SubbandErleEstimator::ErleInstantaneous::Reset() {
>-  Y2_acum_.fill(0.f);
>-  E2_acum_.fill(0.f);
>-  low_render_energy_.fill(false);
>-  num_points_.fill(0);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subband_erle_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subband_erle_estimator.h
>index aa5e5ccb2486362553218a9c5990f37c03dcd6b8..b9862dbc6def1f314a87ed44effe2633ad47ba75 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subband_erle_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subband_erle_estimator.h
>@@ -11,11 +11,13 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBBAND_ERLE_ESTIMATOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_SUBBAND_ERLE_ESTIMATOR_H_
> 
>+#include <stddef.h>
> #include <array>
> #include <memory>
>+#include <vector>
> 
>-#include "absl/types/optional.h"
> #include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> 
>@@ -24,7 +26,7 @@ namespace webrtc {
> // Estimates the echo return loss enhancement for each frequency subband.
> class SubbandErleEstimator {
>  public:
>-  SubbandErleEstimator(float min_erle, float max_erle_lf, float max_erle_hf);
>+  explicit SubbandErleEstimator(const EchoCanceller3Config& config);
>   ~SubbandErleEstimator();
> 
>   // Resets the ERLE estimator.
>@@ -41,55 +43,35 @@ class SubbandErleEstimator {
>   const std::array<float, kFftLengthBy2Plus1>& Erle() const { return erle_; }
> 
>   // Returns the ERLE estimate at onsets.
>-  const std::array<float, kFftLengthBy2Plus1>& ErleOnsets() const {
>-    return erle_onsets_;
>-  }
>+  rtc::ArrayView<const float> ErleOnsets() const { return erle_onsets_; }
> 
>   void Dump(const std::unique_ptr<ApmDataDumper>& data_dumper) const;
> 
>  private:
>-  void UpdateBands(rtc::ArrayView<const float> X2,
>-                   rtc::ArrayView<const float> Y2,
>-                   rtc::ArrayView<const float> E2,
>-                   size_t start,
>-                   size_t stop,
>-                   float max_erle,
>-                   bool onset_detection);
>-  void DecreaseErlePerBandForLowRenderSignals();
>-
>-  class ErleInstantaneous {
>-   public:
>-    ErleInstantaneous();
>-    ~ErleInstantaneous();
>-    // Updates the ERLE for a band with a new block. Returns absl::nullopt
>-    // if not enough points were accumulated for doing the estimation,
>-    // otherwise, it returns the ERLE. When the ERLE is returned, the
>-    // low_render_energy flag contains information on whether the estimation was
>-    // done using low level render signals.
>-    absl::optional<float> Update(float X2,
>-                                 float Y2,
>-                                 float E2,
>-                                 size_t band,
>-                                 bool* low_render_energy);
>-    // Resets the ERLE estimator to its initial state.
>-    void Reset();
>-
>-   private:
>-    std::array<float, kFftLengthBy2Plus1> Y2_acum_;
>-    std::array<float, kFftLengthBy2Plus1> E2_acum_;
>+  struct AccumulatedSpectra {
>+    std::array<float, kFftLengthBy2Plus1> Y2_;
>+    std::array<float, kFftLengthBy2Plus1> E2_;
>     std::array<bool, kFftLengthBy2Plus1> low_render_energy_;
>     std::array<int, kFftLengthBy2Plus1> num_points_;
>   };
> 
>-  ErleInstantaneous instantaneous_erle_;
>+  void UpdateAccumulatedSpectra(rtc::ArrayView<const float> X2,
>+                                rtc::ArrayView<const float> Y2,
>+                                rtc::ArrayView<const float> E2);
>+
>+  void ResetAccumulatedSpectra();
>+
>+  void UpdateBands(bool onset_detection);
>+  void DecreaseErlePerBandForLowRenderSignals();
>+
>+  const float min_erle_;
>+  const std::array<float, kFftLengthBy2Plus1> max_erle_;
>+  const bool adapt_on_low_render_;
>+  AccumulatedSpectra accum_spectra_;
>   std::array<float, kFftLengthBy2Plus1> erle_;
>   std::array<float, kFftLengthBy2Plus1> erle_onsets_;
>   std::array<bool, kFftLengthBy2Plus1> coming_onset_;
>   std::array<int, kFftLengthBy2Plus1> hold_counters_;
>-  const float min_erle_;
>-  const float max_erle_lf_;
>-  const float max_erle_hf_;
>-  const bool adapt_on_low_render_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.cc
>index 609e8ac7ed1b927c72809b91421af7c263854b8d..90069c7a749fc5bb1523f778f49d79a578b7d106 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.cc
>@@ -11,12 +11,12 @@
> #include "modules/audio_processing/aec3/subtractor.h"
> 
> #include <algorithm>
>-#include <numeric>
>+#include <utility>
> 
> #include "api/array_view.h"
>+#include "modules/audio_processing/aec3/fft_data.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_minmax.h"
> #include "system_wrappers/include/field_trial.h"
> 
>@@ -191,7 +191,7 @@ void Subtractor::Process(const RenderBuffer& render_buffer,
>                   adaptation_during_saturation_, &shadow_saturation);
> 
>   // Compute the signal powers in the subtractor output.
>-  output->UpdatePowers(y);
>+  output->ComputeMetrics(y);
> 
>   // Adjust the filter if needed.
>   bool main_filter_adjusted = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.h
>index c92a971f1178373ba0a362c4ca4fb667bc97dbbd..bec014deec957d3ccdee4722ce1fbdd786b20406 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.h
>@@ -11,11 +11,13 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
> 
>-#include <algorithm>
>+#include <math.h>
>+#include <stddef.h>
> #include <array>
> #include <vector>
>-#include "math.h"
> 
>+#include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/adaptive_fir_filter.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/aec3_fft.h"
>@@ -23,10 +25,11 @@
> #include "modules/audio_processing/aec3/echo_path_variability.h"
> #include "modules/audio_processing/aec3/main_filter_update_gain.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
>+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
> #include "modules/audio_processing/aec3/shadow_filter_update_gain.h"
> #include "modules/audio_processing/aec3/subtractor_output.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>-#include "modules/audio_processing/utility/ooura_fft.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.cc
>index affa4a3a06fa24db1cf3e52b27c3438212b0ed79..922cc3d1b3f3e413e5f27f0c47d75eb25c97aade 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.cc
>@@ -33,7 +33,7 @@ void SubtractorOutput::Reset() {
>   y2 = 0.f;
> }
> 
>-void SubtractorOutput::UpdatePowers(rtc::ArrayView<const float> y) {
>+void SubtractorOutput::ComputeMetrics(rtc::ArrayView<const float> y) {
>   const auto sum_of_squares = [](float a, float b) { return a + b * b; };
>   y2 = std::accumulate(y.begin(), y.end(), 0.f, sum_of_squares);
>   e2_main = std::accumulate(e_main.begin(), e_main.end(), 0.f, sum_of_squares);
>@@ -42,6 +42,14 @@ void SubtractorOutput::UpdatePowers(rtc::ArrayView<const float> y) {
>   s2_main = std::accumulate(s_main.begin(), s_main.end(), 0.f, sum_of_squares);
>   s2_shadow =
>       std::accumulate(s_shadow.begin(), s_shadow.end(), 0.f, sum_of_squares);
>+
>+  s_main_max_abs = *std::max_element(s_main.begin(), s_main.end());
>+  s_main_max_abs = std::max(s_main_max_abs,
>+                            -(*std::min_element(s_main.begin(), s_main.end())));
>+
>+  s_shadow_max_abs = *std::max_element(s_shadow.begin(), s_shadow.end());
>+  s_shadow_max_abs = std::max(
>+      s_shadow_max_abs, -(*std::min_element(s_shadow.begin(), s_shadow.end())));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.h
>index 89727e9396f122baf36bdf895abd4be478ef88b3..5f6fd3ed71207e419b476887f74739b6aaa8fa87 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.h
>@@ -36,12 +36,14 @@ struct SubtractorOutput {
>   float e2_main = 0.f;
>   float e2_shadow = 0.f;
>   float y2 = 0.f;
>+  float s_main_max_abs = 0.f;
>+  float s_shadow_max_abs = 0.f;
> 
>   // Reset the struct content.
>   void Reset();
> 
>   // Updates the powers of the signals.
>-  void UpdatePowers(rtc::ArrayView<const float> y);
>+  void ComputeMetrics(rtc::ArrayView<const float> y);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.cc
>index 3cacb4563014a365a9a1e279f96c02ac64f1a1fb..9374b8099f47b6b43b17582cf43f8d53ca3f863c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.cc
>@@ -10,9 +10,9 @@
> 
> #include "modules/audio_processing/aec3/subtractor_output_analyzer.h"
> 
>-#include <array>
>-#include <numeric>
>+#include <algorithm>
> 
>+#include "modules/audio_processing/aec3/aec3_common.h"
> #include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.h
>index b59a68e5e60b19451eaa502381cae2d5c23f184a..0e23ad5dfaf39b0bb40d2b5a75ddddab07308a10 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.h
>@@ -11,7 +11,6 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_ANALYZER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_ANALYZER_H_
> 
>-#include "api/array_view.h"
> #include "modules/audio_processing/aec3/subtractor_output.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_unittest.cc
>index ba744a6a8396537e219e86190202a2dc140feb78..8d14cc1a072527887774bdd57728e355a3068083 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_unittest.cc
>@@ -44,7 +44,7 @@ float RunSubtractorTest(int num_blocks_to_process,
>   config.delay.min_echo_path_delay_blocks = 0;
>   config.delay.default_delay = 1;
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
>   RenderSignalAnalyzer render_signal_analyzer(config);
>   Random random_generator(42U);
>   Aec3Fft fft;
>@@ -127,7 +127,7 @@ TEST(Subtractor, DISABLED_NullOutput) {
>   EchoCanceller3Config config;
>   Subtractor subtractor(config, &data_dumper, DetectOptimization());
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
>   RenderSignalAnalyzer render_signal_analyzer(config);
>   std::vector<float> y(kBlockSize, 0.f);
> 
>@@ -143,7 +143,7 @@ TEST(Subtractor, WrongCaptureSize) {
>   EchoCanceller3Config config;
>   Subtractor subtractor(config, &data_dumper, DetectOptimization());
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
>   RenderSignalAnalyzer render_signal_analyzer(config);
>   std::vector<float> y(kBlockSize - 1, 0.f);
>   SubtractorOutput output;
>@@ -210,21 +210,4 @@ TEST(Subtractor, NonConvergenceOnUncorrelatedSignals) {
>   }
> }
> 
>-// Verifies that the subtractor is properly reset when there is an echo path
>-// change.
>-TEST(Subtractor, EchoPathChangeReset) {
>-  std::vector<int> blocks_with_echo_path_changes;
>-  blocks_with_echo_path_changes.push_back(99);
>-  for (size_t filter_length_blocks : {12, 20, 30}) {
>-    for (size_t delay_samples : {0, 64, 150, 200, 301}) {
>-      SCOPED_TRACE(ProduceDebugText(delay_samples, filter_length_blocks));
>-
>-      float echo_to_nearend_power = RunSubtractorTest(
>-          100, delay_samples, filter_length_blocks, filter_length_blocks, false,
>-          blocks_with_echo_path_changes);
>-      EXPECT_NEAR(1.f, echo_to_nearend_power, 0.0000001f);
>-    }
>-  }
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.cc
>index 87e3008dc76aad8b26c709a44651ff3d6703792c..6fe296c219c632408d4797cb42a9967bf504ba4e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.cc
>@@ -10,13 +10,14 @@
> 
> #include "modules/audio_processing/aec3/suppression_filter.h"
> 
>-#include <math.h>
> #include <algorithm>
>+#include <cmath>
> #include <cstring>
> #include <functional>
>-#include <numeric>
>+#include <iterator>
> 
>-#include "modules/audio_processing/utility/ooura_fft.h"
>+#include "modules/audio_processing/aec3/vector_math.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_minmax.h"
> 
> namespace webrtc {
>@@ -59,8 +60,10 @@ const float kSqrtHanning[kFftLength] = {
> 
> }  // namespace
> 
>-SuppressionFilter::SuppressionFilter(int sample_rate_hz)
>-    : sample_rate_hz_(sample_rate_hz),
>+SuppressionFilter::SuppressionFilter(Aec3Optimization optimization,
>+                                     int sample_rate_hz)
>+    : optimization_(optimization),
>+      sample_rate_hz_(sample_rate_hz),
>       fft_(),
>       e_output_old_(NumBandsForRate(sample_rate_hz_)) {
>   RTC_DCHECK(ValidFullBandRate(sample_rate_hz_));
>@@ -90,18 +93,17 @@ void SuppressionFilter::ApplyGain(
>   std::transform(suppression_gain.begin(), suppression_gain.end(), E.im.begin(),
>                  E.im.begin(), std::multiplies<float>());
> 
>-  // Compute and add the comfort noise.
>-  std::array<float, kFftLengthBy2Plus1> scaled_comfort_noise;
>+  // Comfort noise gain is sqrt(1-g^2), where g is the suppression gain.
>+  std::array<float, kFftLengthBy2Plus1> noise_gain;
>   std::transform(suppression_gain.begin(), suppression_gain.end(),
>-                 comfort_noise.re.begin(), scaled_comfort_noise.begin(),
>-                 [](float a, float b) { return std::max(1.f - a, 0.f) * b; });
>-  std::transform(scaled_comfort_noise.begin(), scaled_comfort_noise.end(),
>-                 E.re.begin(), E.re.begin(), std::plus<float>());
>-  std::transform(suppression_gain.begin(), suppression_gain.end(),
>-                 comfort_noise.im.begin(), scaled_comfort_noise.begin(),
>-                 [](float a, float b) { return std::max(1.f - a, 0.f) * b; });
>-  std::transform(scaled_comfort_noise.begin(), scaled_comfort_noise.end(),
>-                 E.im.begin(), E.im.begin(), std::plus<float>());
>+                 noise_gain.begin(), [](float g) { return 1.f - g * g; });
>+  aec3::VectorMath(optimization_).Sqrt(noise_gain);
>+
>+  // Scale and add the comfort noise.
>+  for (size_t k = 0; k < kFftLengthBy2Plus1; k++) {
>+    E.re[k] += noise_gain[k] * comfort_noise.re[k];
>+    E.im[k] += noise_gain[k] * comfort_noise.im[k];
>+  }
> 
>   // Synthesis filterbank.
>   std::array<float, kFftLength> e_extended;
>@@ -135,7 +137,7 @@ void SuppressionFilter::ApplyGain(
> 
>     // Scale and apply the noise to the signals.
>     const float high_bands_noise_scaling =
>-        0.4f * std::max(1.f - high_bands_gain, 0.f);
>+        0.4f * std::sqrt(1.f - high_bands_gain * high_bands_gain);
> 
>     std::transform(
>         (*e)[1].begin(), (*e)[1].end(), time_domain_high_band_noise.begin(),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.h
>index 237408d7f9dc3e598ee38edcd894282ed7d4b206..edd1290014d2a4ab62373a2a7955a908383f3d09 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.h
>@@ -16,13 +16,15 @@
> 
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/aec3_fft.h"
>+#include "modules/audio_processing/aec3/fft_data.h"
>+#include "modules/audio_processing/utility/ooura_fft.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
> class SuppressionFilter {
>  public:
>-  explicit SuppressionFilter(int sample_rate_hz);
>+  SuppressionFilter(Aec3Optimization optimization, int sample_rate_hz);
>   ~SuppressionFilter();
>   void ApplyGain(const FftData& comfort_noise,
>                  const FftData& comfort_noise_high_bands,
>@@ -32,6 +34,7 @@ class SuppressionFilter {
>                  std::vector<std::vector<float>>* e);
> 
>  private:
>+  const Aec3Optimization optimization_;
>   const int sample_rate_hz_;
>   const OouraFft ooura_fft_;
>   const Aec3Fft fft_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter_unittest.cc
>index eaa608eed5d7faa05b346b3bd7ec9afc54a13e4a..9e4ff7c28b4c82310304b2321f0172b675e8dfc5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter_unittest.cc
>@@ -45,21 +45,21 @@ TEST(SuppressionFilter, NullOutput) {
>   FftData E;
>   std::array<float, kFftLengthBy2Plus1> gain;
> 
>-  EXPECT_DEATH(SuppressionFilter(16000).ApplyGain(cn, cn_high_bands, gain, 1.0f,
>-                                                  E, nullptr),
>+  EXPECT_DEATH(SuppressionFilter(Aec3Optimization::kNone, 16000)
>+                   .ApplyGain(cn, cn_high_bands, gain, 1.0f, E, nullptr),
>                "");
> }
> 
> // Verifies the check for allowed sample rate.
> TEST(SuppressionFilter, ProperSampleRate) {
>-  EXPECT_DEATH(SuppressionFilter(16001), "");
>+  EXPECT_DEATH(SuppressionFilter(Aec3Optimization::kNone, 16001), "");
> }
> 
> #endif
> 
> // Verifies that no comfort noise is added when the gain is 1.
> TEST(SuppressionFilter, ComfortNoiseInUnityGain) {
>-  SuppressionFilter filter(48000);
>+  SuppressionFilter filter(Aec3Optimization::kNone, 48000);
>   FftData cn;
>   FftData cn_high_bands;
>   std::array<float, kFftLengthBy2Plus1> gain;
>@@ -89,7 +89,7 @@ TEST(SuppressionFilter, ComfortNoiseInUnityGain) {
> 
> // Verifies that the suppressor is able to suppress a signal.
> TEST(SuppressionFilter, SignalSuppression) {
>-  SuppressionFilter filter(48000);
>+  SuppressionFilter filter(Aec3Optimization::kNone, 48000);
>   FftData cn;
>   FftData cn_high_bands;
>   std::array<float, kFftLengthBy2> e_old_;
>@@ -131,7 +131,7 @@ TEST(SuppressionFilter, SignalSuppression) {
> // Verifies that the suppressor is able to pass through a desired signal while
> // applying suppressing for some frequencies.
> TEST(SuppressionFilter, SignalTransparency) {
>-  SuppressionFilter filter(48000);
>+  SuppressionFilter filter(Aec3Optimization::kNone, 48000);
>   FftData cn;
>   std::array<float, kFftLengthBy2> e_old_;
>   Aec3Fft fft;
>@@ -171,7 +171,7 @@ TEST(SuppressionFilter, SignalTransparency) {
> 
> // Verifies that the suppressor delay.
> TEST(SuppressionFilter, Delay) {
>-  SuppressionFilter filter(48000);
>+  SuppressionFilter filter(Aec3Optimization::kNone, 48000);
>   FftData cn;
>   FftData cn_high_bands;
>   std::array<float, kFftLengthBy2> e_old_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.cc
>index 613256673069a56ad976dd9abd3a55d0e4f98555..88cfc0a01e4e0d18385c59bfc6180c7ce9773681 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.cc
>@@ -11,8 +11,8 @@
> #include "modules/audio_processing/aec3/suppression_gain.h"
> 
> #include <math.h>
>+#include <stddef.h>
> #include <algorithm>
>-#include <functional>
> #include <numeric>
> 
> #include "modules/audio_processing/aec3/moving_average.h"
>@@ -20,15 +20,10 @@
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
>-#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> namespace {
> 
>-bool EnableNewSuppression() {
>-  return !field_trial::IsEnabled("WebRTC-Aec3NewSuppressionKillSwitch");
>-}
>-
> // Adjust the gains according to the presence of known external filters.
> void AdjustForExternalFilters(std::array<float, kFftLengthBy2Plus1>* gain) {
>   // Limit the low frequency gains to avoid the impact of the high-pass filter
>@@ -83,57 +78,6 @@ void WeightEchoForAudibility(const EchoCanceller3Config& config,
>   weigh(threshold, normalizer, 7, kFftLengthBy2Plus1, echo, weighted_echo);
> }
> 
>-// Computes the gain to reduce the echo to a non audible level.
>-void GainToNoAudibleEchoFallback(
>-    const EchoCanceller3Config& config,
>-    bool low_noise_render,
>-    bool saturated_echo,
>-    bool linear_echo_estimate,
>-    const std::array<float, kFftLengthBy2Plus1>& nearend,
>-    const std::array<float, kFftLengthBy2Plus1>& weighted_echo,
>-    const std::array<float, kFftLengthBy2Plus1>& masker,
>-    const std::array<float, kFftLengthBy2Plus1>& min_gain,
>-    const std::array<float, kFftLengthBy2Plus1>& max_gain,
>-    const std::array<float, kFftLengthBy2Plus1>& one_by_weighted_echo,
>-    std::array<float, kFftLengthBy2Plus1>* gain) {
>-  float nearend_masking_margin = 0.f;
>-  if (linear_echo_estimate) {
>-    nearend_masking_margin =
>-        low_noise_render
>-            ? config.gain_mask.m9
>-            : (saturated_echo ? config.gain_mask.m2 : config.gain_mask.m3);
>-  } else {
>-    nearend_masking_margin = config.gain_mask.m7;
>-  }
>-
>-  RTC_DCHECK_LE(0.f, nearend_masking_margin);
>-  RTC_DCHECK_GT(1.f, nearend_masking_margin);
>-
>-  const float masker_margin =
>-      linear_echo_estimate ? config.gain_mask.m0 : config.gain_mask.m8;
>-
>-  for (size_t k = 0; k < gain->size(); ++k) {
>-    // TODO(devicentepena): Experiment by removing the reverberation estimation
>-    // from the nearend signal before computing the gains.
>-    const float unity_gain_masker = std::max(nearend[k], masker[k]);
>-    RTC_DCHECK_LE(0.f, nearend_masking_margin * unity_gain_masker);
>-    if (weighted_echo[k] <= nearend_masking_margin * unity_gain_masker ||
>-        unity_gain_masker <= 0.f) {
>-      (*gain)[k] = 1.f;
>-    } else {
>-      RTC_DCHECK_LT(0.f, unity_gain_masker);
>-      (*gain)[k] =
>-          std::max(0.f, (1.f - config.gain_mask.gain_curve_slope *
>-                                   weighted_echo[k] / unity_gain_masker) *
>-                            config.gain_mask.gain_curve_offset);
>-      (*gain)[k] = std::max(masker_margin * masker[k] * one_by_weighted_echo[k],
>-                            (*gain)[k]);
>-    }
>-
>-    (*gain)[k] = std::min(std::max((*gain)[k], min_gain[k]), max_gain[k]);
>-  }
>-}
>-
> // TODO(peah): Make adaptive to take the actual filter error into account.
> constexpr size_t kUpperAccurateBandPlus1 = 29;
> 
>@@ -321,30 +265,9 @@ void SuppressionGain::LowerBandGain(
>   std::array<float, kFftLengthBy2Plus1> max_gain;
>   GetMaxGain(max_gain);
> 
>-  // Iteratively compute the gain required to attenuate the echo to a non
>-  // noticeable level.
>-
>-  if (enable_new_suppression_) {
>     GainToNoAudibleEcho(nearend, weighted_residual_echo, comfort_noise,
>                         min_gain, max_gain, gain);
>     AdjustForExternalFilters(gain);
>-  } else {
>-    const bool linear_echo_estimate = aec_state.UsableLinearEstimate();
>-    std::array<float, kFftLengthBy2Plus1> masker;
>-    std::array<float, kFftLengthBy2Plus1> one_by_weighted_echo;
>-    std::transform(weighted_residual_echo.begin(), weighted_residual_echo.end(),
>-                   one_by_weighted_echo.begin(),
>-                   [](float e) { return e > 0.f ? 1.f / e : 1.f; });
>-    gain->fill(0.f);
>-    for (int k = 0; k < 2; ++k) {
>-      std::copy(comfort_noise.begin(), comfort_noise.end(), masker.begin());
>-      GainToNoAudibleEchoFallback(config_, low_noise_render, saturated_echo,
>-                                  linear_echo_estimate, nearend,
>-                                  weighted_residual_echo, masker, min_gain,
>-                                  max_gain, one_by_weighted_echo, gain);
>-      AdjustForExternalFilters(gain);
>-    }
>-  }
> 
>   // Adjust the gain for frequencies which have not yet converged.
>   AdjustNonConvergedFrequencies(gain);
>@@ -372,7 +295,6 @@ SuppressionGain::SuppressionGain(const EchoCanceller3Config& config,
>       config_(config),
>       state_change_duration_blocks_(
>           static_cast<int>(config_.filter.config_change_duration_blocks)),
>-      enable_new_suppression_(EnableNewSuppression()),
>       moving_average_(kFftLengthBy2Plus1,
>                       config.suppressor.nearend_average_blocks),
>       nearend_params_(config_.suppressor.nearend_tuning),
>@@ -416,7 +338,7 @@ void SuppressionGain::GetGain(
> 
>   // Update the state selection.
>   dominant_nearend_detector_.Update(nearend_spectrum, residual_echo_spectrum,
>-                                    comfort_noise_spectrum);
>+                                    comfort_noise_spectrum, initial_state_);
> 
>   // Compute gain for the lower band.
>   bool low_noise_render = low_render_detector_.Detect(render);
>@@ -475,14 +397,17 @@ bool SuppressionGain::LowNoiseRenderDetector::Detect(
> SuppressionGain::DominantNearendDetector::DominantNearendDetector(
>     const EchoCanceller3Config::Suppressor::DominantNearendDetection config)
>     : enr_threshold_(config.enr_threshold),
>+      enr_exit_threshold_(config.enr_exit_threshold),
>       snr_threshold_(config.snr_threshold),
>       hold_duration_(config.hold_duration),
>-      trigger_threshold_(config.trigger_threshold) {}
>+      trigger_threshold_(config.trigger_threshold),
>+      use_during_initial_phase_(config.use_during_initial_phase) {}
> 
> void SuppressionGain::DominantNearendDetector::Update(
>     rtc::ArrayView<const float> nearend_spectrum,
>     rtc::ArrayView<const float> residual_echo_spectrum,
>-    rtc::ArrayView<const float> comfort_noise_spectrum) {
>+    rtc::ArrayView<const float> comfort_noise_spectrum,
>+    bool initial_state) {
>   auto low_frequency_energy = [](rtc::ArrayView<const float> spectrum) {
>     RTC_DCHECK_LE(16, spectrum.size());
>     return std::accumulate(spectrum.begin() + 1, spectrum.begin() + 16, 0.f);
>@@ -493,7 +418,8 @@ void SuppressionGain::DominantNearendDetector::Update(
> 
>   // Detect strong active nearend if the nearend is sufficiently stronger than
>   // the echo and the nearend noise.
>-  if (ne_sum > enr_threshold_ * echo_sum &&
>+  if ((!initial_state || use_during_initial_phase_) &&
>+      ne_sum > enr_threshold_ * echo_sum &&
>       ne_sum > snr_threshold_ * noise_sum) {
>     if (++trigger_counter_ >= trigger_threshold_) {
>       // After a period of strong active nearend activity, flag nearend mode.
>@@ -505,6 +431,12 @@ void SuppressionGain::DominantNearendDetector::Update(
>     trigger_counter_ = std::max(0, trigger_counter_ - 1);
>   }
> 
>+  // Exit nearend-state early at strong echo.
>+  if (ne_sum < enr_exit_threshold_ * echo_sum &&
>+      echo_sum > snr_threshold_ * noise_sum) {
>+    hold_counter_ = 0;
>+  }
>+
>   // Remain in any nearend mode for a certain duration.
>   hold_counter_ = std::max(0, hold_counter_ - 1);
>   nearend_state_ = hold_counter_ > 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.h
>index 4eb8581a862a6dcca3202e0c06e6063d7237f9fd..e74cd9632b123aadb1042ba4b46bb871f7d2e8e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.h
>@@ -12,13 +12,18 @@
> #define MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_GAIN_H_
> 
> #include <array>
>+#include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
> #include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/aec_state.h"
>+#include "modules/audio_processing/aec3/fft_data.h"
> #include "modules/audio_processing/aec3/moving_average.h"
> #include "modules/audio_processing/aec3/render_signal_analyzer.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -102,13 +107,16 @@ class SuppressionGain {
>     // Updates the state selection based on latest spectral estimates.
>     void Update(rtc::ArrayView<const float> nearend_spectrum,
>                 rtc::ArrayView<const float> residual_echo_spectrum,
>-                rtc::ArrayView<const float> comfort_noise_spectrum);
>+                rtc::ArrayView<const float> comfort_noise_spectrum,
>+                bool initial_state);
> 
>    private:
>     const float enr_threshold_;
>+    const float enr_exit_threshold_;
>     const float snr_threshold_;
>     const int hold_duration_;
>     const int trigger_threshold_;
>+    const bool use_during_initial_phase_;
> 
>     bool nearend_state_ = false;
>     int trigger_counter_ = 0;
>@@ -137,7 +145,6 @@ class SuppressionGain {
>   LowNoiseRenderDetector low_render_detector_;
>   bool initial_state_ = true;
>   int initial_state_change_counter_ = 0;
>-  const bool enable_new_suppression_;
>   aec3::MovingAverage moving_average_;
>   const GainParameters nearend_params_;
>   const GainParameters normal_params_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_limiter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_limiter.cc
>index e3d7a660bf0879193155d009e8c822dc79adb7d2..6625a779c244a889fa6aed3b30ee27093faa4d66 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_limiter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_limiter.cc
>@@ -14,6 +14,7 @@
> #include <algorithm>
> 
> #include "modules/audio_processing/aec3/aec3_common.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_unittest.cc
>index 1ff96ca6e0552d8cd3c35083ea80bd18cf637e66..651fd3674bc13c96ee3b210f931673e012f0c3ba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_unittest.cc
>@@ -77,7 +77,7 @@ TEST(SuppressionGain, BasicGainComputation) {
>   ApmDataDumper data_dumper(42);
>   Subtractor subtractor(config, &data_dumper, DetectOptimization());
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>-      RenderDelayBuffer::Create(config, 3));
>+      RenderDelayBuffer::Create2(config, 3));
>   absl::optional<DelayEstimate> delay_estimate;
> 
>   // Ensure that a strong noise is detected to mask any echoes.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.cc
>index f491168d9a141ae1789770ea01696c74233b4be4..0682885c0c7108881cc4269074bb6aa30cb039a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.cc
>@@ -10,7 +10,7 @@
> 
> #include "modules/audio_processing/aec3/vector_buffer.h"
> 
>-#include "modules/audio_processing/aec3/aec3_common.h"
>+#include <algorithm>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.h
>index c7d4f68b27c3886f39bac4b9198eca06d0a53288..4c0257ccea4cd0e89b5367381acc87973483e5c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_VECTOR_BUFFER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_VECTOR_BUFFER_H_
> 
>+#include <stddef.h>
> #include <vector>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/BUILD.gn
>index e5fee3e95f7154c9a2076f7c63a15ae0ebb1cb03..d4262ca485835dad51497a35c951386b9ef2c7f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/BUILD.gn
>@@ -17,6 +17,7 @@ rtc_source_set("aec_dump") {
>   deps = [
>     "../",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/system:rtc_export",
>   ]
> }
> 
>@@ -42,6 +43,7 @@ rtc_source_set("mock_aec_dump_unittests") {
> 
>   deps = [
>     ":mock_aec_dump",
>+    "..:api",
>     "../",
>     "../../../rtc_base:rtc_base_approved",
>     "//testing/gtest",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
>index e3f00f67a2e28e0bcec4a150661d1e50229a80e4..1e55d594ee31340f2965831369040b176cec7115 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
>@@ -16,6 +16,7 @@
> 
> #include "modules/audio_processing/include/aec_dump.h"
> #include "rtc_base/platform_file.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace rtc {
> class TaskQueue;
>@@ -23,7 +24,7 @@ class TaskQueue;
> 
> namespace webrtc {
> 
>-class AecDumpFactory {
>+class RTC_EXPORT AecDumpFactory {
>  public:
>   // The |worker_queue| may not be null and must outlive the created
>   // AecDump instance. |max_log_size_bytes == -1| means the log size
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc
>index 2732934140684886959eeea5a5669019562881a6..9020f2b9dca14e635b8341f1d64692b100bf8ae5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc
>@@ -65,7 +65,7 @@ AecDumpImpl::AecDumpImpl(std::unique_ptr<FileWrapper> debug_file,
> 
> AecDumpImpl::~AecDumpImpl() {
>   // Block until all tasks have finished running.
>-  rtc::Event thread_sync_event(false /* manual_reset */, false);
>+  rtc::Event thread_sync_event;
>   worker_queue_->PostTask([&thread_sync_event] { thread_sync_event.Set(); });
>   // Wait until the event has been signaled with .Set(). By then all
>   // pending tasks will have finished.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.cc
>index 0e56b50c5ef306b89bb6e28a83fcd051d7d403a2..67b70bfb36670a4398ec8d20019c43e80fe21541 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.cc
>@@ -12,16 +12,15 @@
> 
> #include <stddef.h>
> #include <stdlib.h>
>+#include <string.h>
> 
> extern "C" {
> #include "common_audio/ring_buffer.h"
> #include "common_audio/signal_processing/include/real_fft.h"
> }
>+#include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "modules/audio_processing/aecm/echo_control_mobile.h"
> #include "modules/audio_processing/utility/delay_estimator_wrapper.h"
>-extern "C" {
>-#include "system_wrappers/include/cpu_features_wrapper.h"
>-}
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
>index c947563b2805e2e9771241f3bd1e543c6af2f848..257dfbf56e4e81634ca881691349a16dcfb687be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
>@@ -14,10 +14,12 @@
> #include <stdio.h>
> #endif
> #include <stdlib.h>
>+#include <string.h>
> 
> extern "C" {
> #include "common_audio/ring_buffer.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "modules/audio_processing/aecm/aecm_defines.h"
> }
> #include "modules/audio_processing/aecm/aecm_core.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.cc
>index 12c8cfb7fbd1731d782ffc4d9312d4e488d976d2..c24db0dd5225be2821437af84f59da43d82ecc67 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.cc
>@@ -12,8 +12,6 @@
> 
> #include <cmath>
> #include <cstdlib>
>-
>-#include <algorithm>
> #include <vector>
> 
> #include "modules/audio_processing/agc/loudness_histogram.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.h
>index 5d34c212f3abff98ee6f3aafbdf5d801d97d2d84..abd68d5e311d81fa8d2a24b5c14e3d54f32abeb2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.h
>@@ -17,7 +17,6 @@
> 
> namespace webrtc {
> 
>-class AudioFrame;
> class LoudnessHistogram;
> 
> class Agc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
>index dc6d45121cdd6dbfd0f9fcc018870cafb1efec22..5c4deeccbf29c5e93ab01745bd643d3b7a5880ae 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/audio_processing/agc/agc_manager_direct.h"
> 
>+#include <algorithm>
> #include <cmath>
> 
> #ifdef WEBRTC_AGC_DEBUG_DUMP
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.cc
>index 0ed5850e72d567533a32badaf32f4485a4fb13b7..cd57b82bcc72465a911a7f367c1941584239eaaf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.cc
>@@ -10,8 +10,8 @@
> 
> #include "modules/audio_processing/agc/loudness_histogram.h"
> 
>+#include <string.h>
> #include <cmath>
>-#include <cstring>
> 
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.h
>index d8ad7511764a5aa1846541f8cbeba0a8d896d739..b210be96e3b91eaa282d1df539377cfabd39dcf2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.h
>@@ -11,8 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_
> #define MODULES_AUDIO_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_
> 
>-#include <string.h>
>-
>+#include <stdint.h>
> #include <memory>
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/BUILD.gn
>index 1865fde3e91e4fabf779cae6f180c326ad95c65b..5431a150c23c2b80a98551ade63247e70ec6afb3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/BUILD.gn
>@@ -27,6 +27,7 @@ rtc_source_set("level_estimation_agc") {
>     ":gain_applier",
>     ":noise_level_estimator",
>     ":rnn_vad_with_level",
>+    "..:api",
>     "..:apm_logging",
>     "..:audio_frame_view",
>     "../../../api:array_view",
>@@ -58,6 +59,7 @@ rtc_source_set("adaptive_digital") {
>     ":gain_applier",
>     ":noise_level_estimator",
>     ":rnn_vad_with_level",
>+    "..:api",
>     "..:apm_logging",
>     "..:audio_frame_view",
>     "../../../api:array_view",
>@@ -96,12 +98,10 @@ rtc_source_set("fixed_digital") {
>   sources = [
>     "fixed_digital_level_estimator.cc",
>     "fixed_digital_level_estimator.h",
>-    "fixed_gain_controller.cc",
>-    "fixed_gain_controller.h",
>-    "gain_curve_applier.cc",
>-    "gain_curve_applier.h",
>     "interpolated_gain_curve.cc",
>     "interpolated_gain_curve.h",
>+    "limiter.cc",
>+    "limiter.h",
>   ]
> 
>   configs += [ "..:apm_debug_dump" ]
>@@ -128,6 +128,7 @@ rtc_source_set("gain_applier") {
>   deps = [
>     ":common",
>     "..:audio_frame_view",
>+    "../../../api:array_view",
>     "../../../rtc_base:safe_minmax",
>   ]
> }
>@@ -191,8 +192,8 @@ rtc_source_set("adaptive_digital_unittests") {
>     "../../../api:array_view",
>     "../../../common_audio",
>     "../../../rtc_base:checks",
>+    "../../../rtc_base:gunit_helpers",
>     "../../../rtc_base:rtc_base_approved",
>-    "../../../rtc_base:rtc_base_tests_utils",
>   ]
> }
> 
>@@ -203,7 +204,7 @@ rtc_source_set("biquad_filter_unittests") {
>   ]
>   deps = [
>     ":biquad_filter",
>-    "../../../rtc_base:rtc_base_tests_utils",
>+    "../../../rtc_base:gunit_helpers",
>   ]
> }
> 
>@@ -216,11 +217,10 @@ rtc_source_set("fixed_digital_unittests") {
>     "compute_interpolated_gain_curve.cc",
>     "compute_interpolated_gain_curve.h",
>     "fixed_digital_level_estimator_unittest.cc",
>-    "fixed_gain_controller_unittest.cc",
>-    "gain_curve_applier_unittest.cc",
>     "interpolated_gain_curve_unittest.cc",
>-    "limiter.cc",
>-    "limiter.h",
>+    "limiter_db_gain_curve.cc",
>+    "limiter_db_gain_curve.h",
>+    "limiter_db_gain_curve_unittest.cc",
>     "limiter_unittest.cc",
>   ]
>   deps = [
>@@ -232,8 +232,8 @@ rtc_source_set("fixed_digital_unittests") {
>     "../../../api:array_view",
>     "../../../common_audio",
>     "../../../rtc_base:checks",
>+    "../../../rtc_base:gunit_helpers",
>     "../../../rtc_base:rtc_base_approved",
>-    "../../../rtc_base:rtc_base_tests_utils",
>     "../../../system_wrappers:metrics",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
>@@ -254,14 +254,29 @@ rtc_source_set("noise_estimator_unittests") {
>     "..:audio_frame_view",
>     "../../../api:array_view",
>     "../../../rtc_base:checks",
>+    "../../../rtc_base:gunit_helpers",
>     "../../../rtc_base:rtc_base_approved",
>-    "../../../rtc_base:rtc_base_tests_utils",
>+  ]
>+}
>+
>+rtc_source_set("rnn_vad_with_level_unittests") {
>+  testonly = true
>+  sources = [
>+    "vad_with_level_unittest.cc",
>+  ]
>+  deps = [
>+    ":rnn_vad_with_level",
>+    "..:audio_frame_view",
>+    "../../../rtc_base:gunit_helpers",
>   ]
> }
> 
> rtc_source_set("test_utils") {
>   testonly = true
>-  visibility = [ ":*" ]
>+  visibility = [
>+    ":*",
>+    "..:audio_processing_unittests",
>+  ]
>   sources = [
>     "agc2_testing_common.cc",
>     "agc2_testing_common.h",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.cc
>index c7346c6042997e009d90db2cccd518804c513958..a5d36089c484964dbcfb5df00c0a25acd6f25501 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.cc
>@@ -10,12 +10,10 @@
> 
> #include "modules/audio_processing/agc2/adaptive_agc.h"
> 
>-#include <algorithm>
>-#include <numeric>
>-
> #include "common_audio/include/audio_util.h"
> #include "modules/audio_processing/agc2/vad_with_level.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>@@ -27,6 +25,19 @@ AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper)
>   RTC_DCHECK(apm_data_dumper);
> }
> 
>+AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper,
>+                         const AudioProcessing::Config::GainController2& config)
>+    : speech_level_estimator_(
>+          apm_data_dumper,
>+          config.adaptive_digital.level_estimator,
>+          config.adaptive_digital.use_saturation_protector,
>+          config.adaptive_digital.extra_saturation_margin_db),
>+      gain_applier_(apm_data_dumper),
>+      apm_data_dumper_(apm_data_dumper),
>+      noise_level_estimator_(apm_data_dumper) {
>+  RTC_DCHECK(apm_data_dumper);
>+}
>+
> AdaptiveAgc::~AdaptiveAgc() = default;
> 
> void AdaptiveAgc::Process(AudioFrameView<float> float_frame,
>@@ -37,9 +48,9 @@ void AdaptiveAgc::Process(AudioFrameView<float> float_frame,
>                             signal_with_levels.vad_result.speech_probability);
>   apm_data_dumper_->DumpRaw("agc2_vad_rms_dbfs",
>                             signal_with_levels.vad_result.speech_rms_dbfs);
>-
>   apm_data_dumper_->DumpRaw("agc2_vad_peak_dbfs",
>                             signal_with_levels.vad_result.speech_peak_dbfs);
>+
>   speech_level_estimator_.UpdateEstimation(signal_with_levels.vad_result);
> 
>   signal_with_levels.input_level_dbfs =
>@@ -61,7 +72,6 @@ void AdaptiveAgc::Process(AudioFrameView<float> float_frame,
> 
>   // The gain applier applies the gain.
>   gain_applier_.Process(signal_with_levels);
>-  ;
> }
> 
> void AdaptiveAgc::Reset() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.h
>index 792b2bc916de5ab9c6facd99961fb47b8171e22e..16c0082ed8166d2e4a873041913b18ed4414abe9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.h
>@@ -11,13 +11,12 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
> 
>-#include <memory>
>-
> #include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
> #include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
> #include "modules/audio_processing/agc2/noise_level_estimator.h"
> #include "modules/audio_processing/agc2/vad_with_level.h"
> #include "modules/audio_processing/include/audio_frame_view.h"
>+#include "modules/audio_processing/include/audio_processing.h"
> 
> namespace webrtc {
> class ApmDataDumper;
>@@ -25,6 +24,8 @@ class ApmDataDumper;
> class AdaptiveAgc {
>  public:
>   explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
>+  AdaptiveAgc(ApmDataDumper* apm_data_dumper,
>+              const AudioProcessing::Config::GainController2& config);
>   ~AdaptiveAgc();
> 
>   void Process(AudioFrameView<float> float_frame, float last_audio_level);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
>index d4560caf1a9d8471a7e26e8127297df51393d829..6ece83b239240fa8e11abe4c0544a9e7ce5721ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
>@@ -15,6 +15,7 @@
> #include "common_audio/include/audio_util.h"
> #include "modules/audio_processing/agc2/agc2_common.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_minmax.h"
> #include "system_wrappers/include/metrics.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
>index b670f4b4692056c40aab7ff8e8e97d8a7fdbb807..8640324b593ffb126f15d54b21792efe4155e50c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
>@@ -12,13 +12,27 @@
> 
> #include "modules/audio_processing/agc2/agc2_common.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_minmax.h"
> 
> namespace webrtc {
> 
> AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
>     ApmDataDumper* apm_data_dumper)
>-    : saturation_protector_(apm_data_dumper),
>+    : level_estimator_(
>+          AudioProcessing::Config::GainController2::LevelEstimator::kRms),
>+      use_saturation_protector_(true),
>+      saturation_protector_(apm_data_dumper),
>+      apm_data_dumper_(apm_data_dumper) {}
>+
>+AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
>+    ApmDataDumper* apm_data_dumper,
>+    AudioProcessing::Config::GainController2::LevelEstimator level_estimator,
>+    bool use_saturation_protector,
>+    float extra_saturation_margin_db)
>+    : level_estimator_(level_estimator),
>+      use_saturation_protector_(use_saturation_protector),
>+      saturation_protector_(apm_data_dumper, extra_saturation_margin_db),
>       apm_data_dumper_(apm_data_dumper) {}
> 
> void AdaptiveModeLevelEstimator::UpdateEstimation(
>@@ -42,20 +56,38 @@ void AdaptiveModeLevelEstimator::UpdateEstimation(
> 
>   const float leak_factor = buffer_is_full ? kFullBufferLeakFactor : 1.f;
> 
>+  // Read speech level estimation.
>+  float speech_level_dbfs = 0.f;
>+  using LevelEstimatorType =
>+      AudioProcessing::Config::GainController2::LevelEstimator;
>+  switch (level_estimator_) {
>+    case LevelEstimatorType::kRms:
>+      speech_level_dbfs = vad_data.speech_rms_dbfs;
>+      break;
>+    case LevelEstimatorType::kPeak:
>+      speech_level_dbfs = vad_data.speech_peak_dbfs;
>+      break;
>+  }
>+
>+  // Update speech level estimation.
>   estimate_numerator_ = estimate_numerator_ * leak_factor +
>-                        vad_data.speech_rms_dbfs * vad_data.speech_probability;
>+                        speech_level_dbfs * vad_data.speech_probability;
>   estimate_denominator_ =
>       estimate_denominator_ * leak_factor + vad_data.speech_probability;
>-
>   last_estimate_with_offset_dbfs_ = estimate_numerator_ / estimate_denominator_;
> 
>-  saturation_protector_.UpdateMargin(vad_data, last_estimate_with_offset_dbfs_);
>-  DebugDumpEstimate();
>+  if (use_saturation_protector_) {
>+    saturation_protector_.UpdateMargin(vad_data,
>+                                       last_estimate_with_offset_dbfs_);
>+    DebugDumpEstimate();
>+  }
> }
> 
> float AdaptiveModeLevelEstimator::LatestLevelEstimate() const {
>   return rtc::SafeClamp<float>(
>-      last_estimate_with_offset_dbfs_ + saturation_protector_.LastMargin(),
>+      last_estimate_with_offset_dbfs_ +
>+          (use_saturation_protector_ ? saturation_protector_.LastMargin()
>+                                     : 0.f),
>       -90.f, 30.f);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.h
>index 4d4180c4806870004120aa3c921ee755e8b1b619..63b9de2aec44a124dea725a3a6ffc4f601709afb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.h
>@@ -11,8 +11,12 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
> 
>+#include <stddef.h>
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"  // kFullBufferSizeMs...
> #include "modules/audio_processing/agc2/saturation_protector.h"
> #include "modules/audio_processing/agc2/vad_with_level.h"
>+#include "modules/audio_processing/include/audio_processing.h"
> 
> namespace webrtc {
> class ApmDataDumper;
>@@ -20,6 +24,11 @@ class ApmDataDumper;
> class AdaptiveModeLevelEstimator {
>  public:
>   explicit AdaptiveModeLevelEstimator(ApmDataDumper* apm_data_dumper);
>+  AdaptiveModeLevelEstimator(
>+      ApmDataDumper* apm_data_dumper,
>+      AudioProcessing::Config::GainController2::LevelEstimator level_estimator,
>+      bool use_saturation_protector,
>+      float extra_saturation_margin_db);
>   void UpdateEstimation(const VadWithLevel::LevelAndProbability& vad_data);
>   float LatestLevelEstimate() const;
>   void Reset();
>@@ -30,6 +39,9 @@ class AdaptiveModeLevelEstimator {
>  private:
>   void DebugDumpEstimate();
> 
>+  const AudioProcessing::Config::GainController2::LevelEstimator
>+      level_estimator_;
>+  const bool use_saturation_protector_;
>   size_t buffer_size_ms_ = 0;
>   float last_estimate_with_offset_dbfs_ = kInitialSpeechLevelEstimateDbfs;
>   float estimate_numerator_ = 0.f;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc
>index 4cee9633f7778daeda05f10fcf0649bfe2844641..b7c64373fc8a00a8291046e71741c133f8739e1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc
>@@ -10,6 +10,10 @@
> 
> #include "modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h"
> 
>+#include <cmath>
>+#include <vector>
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
> #include "modules/audio_processing/include/audio_frame_view.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h
>index 484b12893325a01ececbe6be6cae8742d0a8c557..6d12339888c2e8400513c3bf4e02202b6bc1826b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h
>@@ -11,10 +11,12 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_AGC_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_AGC_H_
> 
>-#include <vector>
>+#include <stddef.h>
>+#include <stdint.h>
> 
> #include "modules/audio_processing/agc/agc.h"
> #include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
>+#include "modules/audio_processing/agc2/saturation_protector.h"
> #include "modules/audio_processing/agc2/vad_with_level.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_common.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_common.cc
>index 5da353fa8fd138feb4ff7b23c54c61c92124497b..af943df78fbb828f83c0a0cf6af1728ab2505a64 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_common.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_common.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/audio_processing/agc2/agc2_common.h"
> 
>+#include <stdio.h>
> #include <string>
> 
> #include "system_wrappers/include/field_trial.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_common.h
>index 71d33e594da23bdf249c4b6b7cbb6fc0ffff07b0..55dd648db49ab161a1f9850ed4a87e689c61e3d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_common.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_common.h
>@@ -13,8 +13,6 @@
> 
> #include <stddef.h>
> 
>-#include <cmath>
>-
> namespace webrtc {
> 
> constexpr float kMinFloatS16Value = -32768.f;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.cc
>index 9858d505846591f880e94db8174a321131a1f3f4..da8557c19082c1ca18091b3b6f484ae5ebb46ef3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/audio_processing/agc2/biquad_filter.h"
> 
>+#include <stddef.h>
>+
> namespace webrtc {
> 
> // Transposed direct form I implementation of a bi-quad filter applied to an
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.h
>index 284930c5952831a0c7f12550136508b5504e9a1e..3d78c07607888922760fc05b681ac5ea5caac0ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.h
>@@ -11,6 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
> 
>+#include <algorithm>
>+
> #include "api/array_view.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/constructormagic.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
>index f395bceffa1bc77cd6379d06d619dfb44b9bee12..bc92613b697e6f1a40624c2d352aaae510b0c824 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
>@@ -19,23 +19,24 @@
> 
> #include "modules/audio_processing/agc2/agc2_common.h"
> #include "modules/audio_processing/agc2/agc2_testing_common.h"
>-#include "modules/audio_processing/agc2/limiter.h"
>+#include "modules/audio_processing/agc2/limiter_db_gain_curve.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
> namespace {
> 
> std::pair<double, double> ComputeLinearApproximationParams(
>-    const Limiter* limiter,
>+    const LimiterDbGainCurve* limiter,
>     const double x) {
>   const double m = limiter->GetGainFirstDerivativeLinear(x);
>   const double q = limiter->GetGainLinear(x) - m * x;
>   return {m, q};
> }
> 
>-double ComputeAreaUnderPiecewiseLinearApproximation(const Limiter* limiter,
>-                                                    const double x0,
>-                                                    const double x1) {
>+double ComputeAreaUnderPiecewiseLinearApproximation(
>+    const LimiterDbGainCurve* limiter,
>+    const double x0,
>+    const double x1) {
>   RTC_CHECK_LT(x0, x1);
> 
>   // Linear approximation in x0 and x1.
>@@ -60,7 +61,7 @@ double ComputeAreaUnderPiecewiseLinearApproximation(const Limiter* limiter,
> // Computes the approximation error in the limiter region for a given interval.
> // The error is computed as the difference between the areas beneath the limiter
> // curve to approximate and its linear under-approximation.
>-double LimiterUnderApproximationNegativeError(const Limiter* limiter,
>+double LimiterUnderApproximationNegativeError(const LimiterDbGainCurve* limiter,
>                                               const double x0,
>                                               const double x1) {
>   const double area_limiter = limiter->GetGainIntegralLinear(x0, x1);
>@@ -77,7 +78,7 @@ double LimiterUnderApproximationNegativeError(const Limiter* limiter,
> // are assigned by halving intervals (starting with the whole beyond-knee region
> // as a single interval). However, even if sub-optimal, this algorithm works
> // well in practice and it is efficiently implemented using priority queues.
>-std::vector<double> SampleLimiterRegion(const Limiter* limiter) {
>+std::vector<double> SampleLimiterRegion(const LimiterDbGainCurve* limiter) {
>   static_assert(kInterpolatedGainCurveBeyondKneePoints > 2, "");
> 
>   struct Interval {
>@@ -131,7 +132,7 @@ std::vector<double> SampleLimiterRegion(const Limiter* limiter) {
> // Compute the parameters to over-approximate the knee region via linear
> // interpolation. Over-approximating is saturation-safe since the knee region is
> // convex.
>-void PrecomputeKneeApproxParams(const Limiter* limiter,
>+void PrecomputeKneeApproxParams(const LimiterDbGainCurve* limiter,
>                                 test::InterpolatedParameters* parameters) {
>   static_assert(kInterpolatedGainCurveKneePoints > 2, "");
>   // Get |kInterpolatedGainCurveKneePoints| - 1 equally spaced points.
>@@ -165,7 +166,7 @@ void PrecomputeKneeApproxParams(const Limiter* limiter,
> // interpolation and greedy sampling. Under-approximating is saturation-safe
> // since the beyond-knee region is concave.
> void PrecomputeBeyondKneeApproxParams(
>-    const Limiter* limiter,
>+    const LimiterDbGainCurve* limiter,
>     test::InterpolatedParameters* parameters) {
>   // Find points on which the linear pieces are tangent to the gain curve.
>   const auto samples = SampleLimiterRegion(limiter);
>@@ -216,7 +217,7 @@ namespace test {
> 
> InterpolatedParameters ComputeInterpolatedGainCurveApproximationParams() {
>   InterpolatedParameters parameters;
>-  Limiter limiter;
>+  LimiterDbGainCurve limiter;
>   parameters.computed_approximation_params_x.fill(0.0f);
>   parameters.computed_approximation_params_m.fill(0.0f);
>   parameters.computed_approximation_params_q.fill(0.0f);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
>index 39cc764185c1d8fb135d4cb843e9366613d83cc7..971f4f62b7901588b7406ccd8d5cbdec0520b2a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
>@@ -13,6 +13,7 @@
> #include <algorithm>
> #include <cmath>
> 
>+#include "api/array_view.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.cc
>index 0d7e3a61b15d30ac545067d767695fe2e7c0e57e..ef908dc35828494e4d6eadf92c08bc5f08cef3df 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.cc
>@@ -10,13 +10,9 @@
> 
> #include "modules/audio_processing/agc2/fixed_gain_controller.h"
> 
>-#include <algorithm>
>-#include <cmath>
>-
> #include "api/array_view.h"
> #include "common_audio/include/audio_util.h"
> #include "modules/audio_processing/agc2/agc2_common.h"
>-#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>@@ -39,7 +35,7 @@ FixedGainController::FixedGainController(ApmDataDumper* apm_data_dumper)
> FixedGainController::FixedGainController(ApmDataDumper* apm_data_dumper,
>                                          std::string histogram_name_prefix)
>     : apm_data_dumper_(apm_data_dumper),
>-      gain_curve_applier_(48000, apm_data_dumper_, histogram_name_prefix) {
>+      limiter_(48000, apm_data_dumper_, histogram_name_prefix) {
>   // Do update histograms.xml when adding name prefixes.
>   RTC_DCHECK(histogram_name_prefix == "" || histogram_name_prefix == "Test" ||
>              histogram_name_prefix == "AudioMixer" ||
>@@ -61,12 +57,12 @@ void FixedGainController::SetGain(float gain_to_apply_db) {
>   // Reset the gain curve applier to quickly react on abrupt level changes
>   // caused by large changes of the applied gain.
>   if (previous_applied_gained != gain_to_apply_) {
>-    gain_curve_applier_.Reset();
>+    limiter_.Reset();
>   }
> }
> 
> void FixedGainController::SetSampleRate(size_t sample_rate_hz) {
>-  gain_curve_applier_.SetSampleRate(sample_rate_hz);
>+  limiter_.SetSampleRate(sample_rate_hz);
> }
> 
> void FixedGainController::Process(AudioFrameView<float> signal) {
>@@ -84,7 +80,7 @@ void FixedGainController::Process(AudioFrameView<float> signal) {
>   }
> 
>   // Use the limiter.
>-  gain_curve_applier_.Process(signal);
>+  limiter_.Process(signal);
> 
>   // Dump data for debug.
>   const auto channel_view = signal.channel(0);
>@@ -100,6 +96,6 @@ void FixedGainController::Process(AudioFrameView<float> signal) {
> }
> 
> float FixedGainController::LastAudioLevel() const {
>-  return gain_curve_applier_.LastAudioLevel();
>+  return limiter_.LastAudioLevel();
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.h
>deleted file mode 100644
>index ff6ab811729494ed86dbd8626a6319bba61a4f1d..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.h
>+++ /dev/null
>@@ -1,42 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_AGC2_FIXED_GAIN_CONTROLLER_H_
>-#define MODULES_AUDIO_PROCESSING_AGC2_FIXED_GAIN_CONTROLLER_H_
>-
>-#include "modules/audio_processing/agc2/gain_curve_applier.h"
>-#include "modules/audio_processing/include/audio_frame_view.h"
>-
>-namespace webrtc {
>-class ApmDataDumper;
>-
>-class FixedGainController {
>- public:
>-  explicit FixedGainController(ApmDataDumper* apm_data_dumper);
>-  FixedGainController(ApmDataDumper* apm_data_dumper,
>-                      std::string histogram_name_prefix);
>-
>-  void Process(AudioFrameView<float> signal);
>-
>-  // Rate and gain may be changed at any time (but not concurrently
>-  // with any other method call).
>-  void SetGain(float gain_to_apply_db);
>-  void SetSampleRate(size_t sample_rate_hz);
>-  float LastAudioLevel() const;
>-
>- private:
>-  float gain_to_apply_ = 1.f;
>-  ApmDataDumper* apm_data_dumper_ = nullptr;
>-  GainCurveApplier gain_curve_applier_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_AGC2_FIXED_GAIN_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller_unittest.cc
>deleted file mode 100644
>index db1732aa47648ad190e626f696cfe49386d952a4..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller_unittest.cc
>+++ /dev/null
>@@ -1,258 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/agc2/fixed_gain_controller.h"
>-
>-#include "absl/memory/memory.h"
>-#include "api/array_view.h"
>-#include "modules/audio_processing/agc2/agc2_testing_common.h"
>-#include "modules/audio_processing/agc2/vector_float_frame.h"
>-#include "modules/audio_processing/logging/apm_data_dumper.h"
>-#include "rtc_base/gunit.h"
>-#include "system_wrappers/include/metrics.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-constexpr float kInputLevelLinear = 15000.f;
>-
>-constexpr float kGainToApplyDb = 15.f;
>-
>-float RunFixedGainControllerWithConstantInput(FixedGainController* fixed_gc,
>-                                              const float input_level,
>-                                              const size_t num_frames,
>-                                              const int sample_rate) {
>-  // Give time to the level etimator to converge.
>-  for (size_t i = 0; i < num_frames; ++i) {
>-    VectorFloatFrame vectors_with_float_frame(
>-        1, rtc::CheckedDivExact(sample_rate, 100), input_level);
>-    fixed_gc->Process(vectors_with_float_frame.float_frame_view());
>-  }
>-
>-  // Process the last frame with constant input level.
>-  VectorFloatFrame vectors_with_float_frame_last(
>-      1, rtc::CheckedDivExact(sample_rate, 100), input_level);
>-  fixed_gc->Process(vectors_with_float_frame_last.float_frame_view());
>-
>-  // Return the last sample from the last processed frame.
>-  const auto channel =
>-      vectors_with_float_frame_last.float_frame_view().channel(0);
>-  return channel[channel.size() - 1];
>-}
>-
>-std::unique_ptr<ApmDataDumper> GetApmDataDumper() {
>-  return absl::make_unique<ApmDataDumper>(0);
>-}
>-
>-std::unique_ptr<FixedGainController> CreateFixedGainController(
>-    float gain_to_apply,
>-    size_t rate,
>-    std::string histogram_name_prefix,
>-    ApmDataDumper* test_data_dumper) {
>-  std::unique_ptr<FixedGainController> fgc =
>-      absl::make_unique<FixedGainController>(test_data_dumper,
>-                                             histogram_name_prefix);
>-  fgc->SetGain(gain_to_apply);
>-  fgc->SetSampleRate(rate);
>-  return fgc;
>-}
>-
>-std::unique_ptr<FixedGainController> CreateFixedGainController(
>-    float gain_to_apply,
>-    size_t rate,
>-    ApmDataDumper* test_data_dumper) {
>-  return CreateFixedGainController(gain_to_apply, rate, "", test_data_dumper);
>-}
>-
>-}  // namespace
>-
>-TEST(AutomaticGainController2FixedDigital, CreateUse) {
>-  const int kSampleRate = 44000;
>-  auto test_data_dumper = GetApmDataDumper();
>-  std::unique_ptr<FixedGainController> fixed_gc = CreateFixedGainController(
>-      kGainToApplyDb, kSampleRate, test_data_dumper.get());
>-  VectorFloatFrame vectors_with_float_frame(
>-      1, rtc::CheckedDivExact(kSampleRate, 100), kInputLevelLinear);
>-  auto float_frame = vectors_with_float_frame.float_frame_view();
>-  fixed_gc->Process(float_frame);
>-  const auto channel = float_frame.channel(0);
>-  EXPECT_LT(kInputLevelLinear, channel[0]);
>-}
>-
>-TEST(AutomaticGainController2FixedDigital, CheckSaturationBehaviorWithLimiter) {
>-  const float kInputLevel = 32767.f;
>-  const size_t kNumFrames = 5;
>-  const size_t kSampleRate = 42000;
>-
>-  auto test_data_dumper = GetApmDataDumper();
>-
>-  const auto gains_no_saturation =
>-      test::LinSpace(0.1, test::kLimiterMaxInputLevelDbFs - 0.01, 10);
>-  for (const auto gain_db : gains_no_saturation) {
>-    // Since |test::kLimiterMaxInputLevelDbFs| > |gain_db|, the
>-    // limiter will not saturate the signal.
>-    std::unique_ptr<FixedGainController> fixed_gc_no_saturation =
>-        CreateFixedGainController(gain_db, kSampleRate, test_data_dumper.get());
>-
>-    // Saturation not expected.
>-    SCOPED_TRACE(std::to_string(gain_db));
>-    EXPECT_LT(
>-        RunFixedGainControllerWithConstantInput(
>-            fixed_gc_no_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>-        32767.f);
>-  }
>-
>-  const auto gains_saturation =
>-      test::LinSpace(test::kLimiterMaxInputLevelDbFs + 0.01, 10, 10);
>-  for (const auto gain_db : gains_saturation) {
>-    // Since |test::kLimiterMaxInputLevelDbFs| < |gain|, the limiter
>-    // will saturate the signal.
>-    std::unique_ptr<FixedGainController> fixed_gc_saturation =
>-        CreateFixedGainController(gain_db, kSampleRate, test_data_dumper.get());
>-
>-    // Saturation expected.
>-    SCOPED_TRACE(std::to_string(gain_db));
>-    EXPECT_FLOAT_EQ(
>-        RunFixedGainControllerWithConstantInput(
>-            fixed_gc_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>-        32767.f);
>-  }
>-}
>-
>-TEST(AutomaticGainController2FixedDigital,
>-     CheckSaturationBehaviorWithLimiterSingleSample) {
>-  const float kInputLevel = 32767.f;
>-  const size_t kNumFrames = 5;
>-  const size_t kSampleRate = 8000;
>-
>-  auto test_data_dumper = GetApmDataDumper();
>-
>-  const auto gains_no_saturation =
>-      test::LinSpace(0.1, test::kLimiterMaxInputLevelDbFs - 0.01, 10);
>-  for (const auto gain_db : gains_no_saturation) {
>-    // Since |gain| > |test::kLimiterMaxInputLevelDbFs|, the limiter will
>-    // not saturate the signal.
>-    std::unique_ptr<FixedGainController> fixed_gc_no_saturation =
>-        CreateFixedGainController(gain_db, kSampleRate, test_data_dumper.get());
>-
>-    // Saturation not expected.
>-    SCOPED_TRACE(std::to_string(gain_db));
>-    EXPECT_LT(
>-        RunFixedGainControllerWithConstantInput(
>-            fixed_gc_no_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>-        32767.f);
>-  }
>-
>-  const auto gains_saturation =
>-      test::LinSpace(test::kLimiterMaxInputLevelDbFs + 0.01, 10, 10);
>-  for (const auto gain_db : gains_saturation) {
>-    // Singe |gain| < |test::kLimiterMaxInputLevelDbFs|, the limiter will
>-    // saturate the signal.
>-    std::unique_ptr<FixedGainController> fixed_gc_saturation =
>-        CreateFixedGainController(gain_db, kSampleRate, test_data_dumper.get());
>-
>-    // Saturation expected.
>-    SCOPED_TRACE(std::to_string(gain_db));
>-    EXPECT_FLOAT_EQ(
>-        RunFixedGainControllerWithConstantInput(
>-            fixed_gc_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>-        32767.f);
>-  }
>-}
>-
>-TEST(AutomaticGainController2FixedDigital, GainShouldChangeOnSetGain) {
>-  constexpr float kInputLevel = 1000.f;
>-  constexpr size_t kNumFrames = 5;
>-  constexpr size_t kSampleRate = 8000;
>-  constexpr float kGainDbNoChange = 0.f;
>-  constexpr float kGainDbFactor10 = 20.f;
>-
>-  auto test_data_dumper = GetApmDataDumper();
>-  std::unique_ptr<FixedGainController> fixed_gc_no_saturation =
>-      CreateFixedGainController(kGainDbNoChange, kSampleRate,
>-                                test_data_dumper.get());
>-
>-  // Signal level is unchanged with 0 db gain.
>-  EXPECT_FLOAT_EQ(
>-      RunFixedGainControllerWithConstantInput(
>-          fixed_gc_no_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>-      kInputLevel);
>-
>-  fixed_gc_no_saturation->SetGain(kGainDbFactor10);
>-
>-  // +20db should increase signal by a factor of 10.
>-  EXPECT_FLOAT_EQ(
>-      RunFixedGainControllerWithConstantInput(
>-          fixed_gc_no_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>-      kInputLevel * 10);
>-}
>-
>-TEST(AutomaticGainController2FixedDigital,
>-     SetGainShouldBeFastAndTimeInvariant) {
>-  // Number of frames required for the fixed gain controller to adapt on the
>-  // input signal when the gain changes.
>-  constexpr size_t kNumFrames = 5;
>-
>-  constexpr float kInputLevel = 1000.f;
>-  constexpr size_t kSampleRate = 8000;
>-  constexpr float kGainDbLow = 0.f;
>-  constexpr float kGainDbHigh = 40.f;
>-  static_assert(kGainDbLow < kGainDbHigh, "");
>-
>-  auto test_data_dumper = GetApmDataDumper();
>-  std::unique_ptr<FixedGainController> fixed_gc = CreateFixedGainController(
>-      kGainDbLow, kSampleRate, test_data_dumper.get());
>-
>-  fixed_gc->SetGain(kGainDbLow);
>-  const float output_level_pre = RunFixedGainControllerWithConstantInput(
>-      fixed_gc.get(), kInputLevel, kNumFrames, kSampleRate);
>-
>-  fixed_gc->SetGain(kGainDbHigh);
>-  RunFixedGainControllerWithConstantInput(fixed_gc.get(), kInputLevel,
>-                                          kNumFrames, kSampleRate);
>-
>-  fixed_gc->SetGain(kGainDbLow);
>-  const float output_level_post = RunFixedGainControllerWithConstantInput(
>-      fixed_gc.get(), kInputLevel, kNumFrames, kSampleRate);
>-
>-  EXPECT_EQ(output_level_pre, output_level_post);
>-}
>-
>-TEST(AutomaticGainController2FixedDigital, RegionHistogramIsUpdated) {
>-  constexpr size_t kSampleRate = 8000;
>-  constexpr float kGainDb = 0.f;
>-  constexpr float kInputLevel = 1000.f;
>-  constexpr size_t kNumFrames = 5;
>-
>-  metrics::Reset();
>-
>-  auto test_data_dumper = GetApmDataDumper();
>-  std::unique_ptr<FixedGainController> fixed_gc_no_saturation =
>-      CreateFixedGainController(kGainDb, kSampleRate, "Test",
>-                                test_data_dumper.get());
>-
>-  static_cast<void>(RunFixedGainControllerWithConstantInput(
>-      fixed_gc_no_saturation.get(), kInputLevel, kNumFrames, kSampleRate));
>-
>-  // Destroying FixedGainController should cause the last limiter region to be
>-  // logged.
>-  fixed_gc_no_saturation.reset();
>-
>-  EXPECT_EQ(1, metrics::NumSamples(
>-                   "WebRTC.Audio.Test.FixedDigitalGainCurveRegion.Identity"));
>-  EXPECT_EQ(0, metrics::NumSamples(
>-                   "WebRTC.Audio.Test.FixedDigitalGainCurveRegion.Knee"));
>-  EXPECT_EQ(0, metrics::NumSamples(
>-                   "WebRTC.Audio.Test.FixedDigitalGainCurveRegion.Limiter"));
>-  EXPECT_EQ(0, metrics::NumSamples(
>-                   "WebRTC.Audio.Test.FixedDigitalGainCurveRegion.Saturation"));
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.cc
>index 38eb1de03f9589bbd7fed044b830e88fc28c1470..8c437177e3cbfa6be506a6bdf5676a70f89b1cf7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/audio_processing/agc2/gain_applier.h"
> 
>+#include "api/array_view.h"
> #include "modules/audio_processing/agc2/agc2_common.h"
> #include "rtc_base/numerics/safe_minmax.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.h
>index e2567b19dbbdc9a3be18db4146c6a2ab009db9ca..7f9f00e07600e2c818489e2c611e7a326bc30038 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.h
>@@ -11,6 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
> 
>+#include <stddef.h>
>+
> #include "modules/audio_processing/include/audio_frame_view.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.cc
>deleted file mode 100644
>index 1eca21b98e71817872a0543a0f09e88591bdab5e..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.cc
>+++ /dev/null
>@@ -1,141 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/agc2/gain_curve_applier.h"
>-
>-#include <algorithm>
>-#include <array>
>-#include <cmath>
>-
>-#include "api/array_view.h"
>-#include "modules/audio_processing/logging/apm_data_dumper.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-// This constant affects the way scaling factors are interpolated for the first
>-// sub-frame of a frame. Only in the case in which the first sub-frame has an
>-// estimated level which is greater than the that of the previous analyzed
>-// sub-frame, linear interpolation is replaced with a power function which
>-// reduces the chances of over-shooting (and hence saturation), however reducing
>-// the fixed gain effectiveness.
>-constexpr float kAttackFirstSubframeInterpolationPower = 8.f;
>-
>-void InterpolateFirstSubframe(float last_factor,
>-                              float current_factor,
>-                              rtc::ArrayView<float> subframe) {
>-  const auto n = subframe.size();
>-  constexpr auto p = kAttackFirstSubframeInterpolationPower;
>-  for (size_t i = 0; i < n; ++i) {
>-    subframe[i] = std::pow(1.f - i / n, p) * (last_factor - current_factor) +
>-                  current_factor;
>-  }
>-}
>-
>-void ComputePerSampleSubframeFactors(
>-    const std::array<float, kSubFramesInFrame + 1>& scaling_factors,
>-    size_t samples_per_channel,
>-    rtc::ArrayView<float> per_sample_scaling_factors) {
>-  const size_t num_subframes = scaling_factors.size() - 1;
>-  const size_t subframe_size =
>-      rtc::CheckedDivExact(samples_per_channel, num_subframes);
>-
>-  // Handle first sub-frame differently in case of attack.
>-  const bool is_attack = scaling_factors[0] > scaling_factors[1];
>-  if (is_attack) {
>-    InterpolateFirstSubframe(
>-        scaling_factors[0], scaling_factors[1],
>-        rtc::ArrayView<float>(
>-            per_sample_scaling_factors.subview(0, subframe_size)));
>-  }
>-
>-  for (size_t i = is_attack ? 1 : 0; i < num_subframes; ++i) {
>-    const size_t subframe_start = i * subframe_size;
>-    const float scaling_start = scaling_factors[i];
>-    const float scaling_end = scaling_factors[i + 1];
>-    const float scaling_diff = (scaling_end - scaling_start) / subframe_size;
>-    for (size_t j = 0; j < subframe_size; ++j) {
>-      per_sample_scaling_factors[subframe_start + j] =
>-          scaling_start + scaling_diff * j;
>-    }
>-  }
>-}
>-
>-void ScaleSamples(rtc::ArrayView<const float> per_sample_scaling_factors,
>-                  AudioFrameView<float> signal) {
>-  const size_t samples_per_channel = signal.samples_per_channel();
>-  RTC_DCHECK_EQ(samples_per_channel, per_sample_scaling_factors.size());
>-  for (size_t i = 0; i < signal.num_channels(); ++i) {
>-    auto channel = signal.channel(i);
>-    for (size_t j = 0; j < samples_per_channel; ++j) {
>-      channel[j] *= per_sample_scaling_factors[j];
>-    }
>-  }
>-}
>-
>-}  // namespace
>-
>-GainCurveApplier::GainCurveApplier(size_t sample_rate_hz,
>-                                   ApmDataDumper* apm_data_dumper,
>-                                   std::string histogram_name)
>-    : interp_gain_curve_(apm_data_dumper, histogram_name),
>-      level_estimator_(sample_rate_hz, apm_data_dumper),
>-      apm_data_dumper_(apm_data_dumper) {}
>-
>-GainCurveApplier::~GainCurveApplier() = default;
>-
>-void GainCurveApplier::Process(AudioFrameView<float> signal) {
>-  const auto level_estimate = level_estimator_.ComputeLevel(signal);
>-
>-  RTC_DCHECK_EQ(level_estimate.size() + 1, scaling_factors_.size());
>-  scaling_factors_[0] = last_scaling_factor_;
>-  std::transform(level_estimate.begin(), level_estimate.end(),
>-                 scaling_factors_.begin() + 1, [this](float x) {
>-                   return interp_gain_curve_.LookUpGainToApply(x);
>-                 });
>-
>-  const size_t samples_per_channel = signal.samples_per_channel();
>-  RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
>-
>-  auto per_sample_scaling_factors = rtc::ArrayView<float>(
>-      &per_sample_scaling_factors_[0], samples_per_channel);
>-  ComputePerSampleSubframeFactors(scaling_factors_, samples_per_channel,
>-                                  per_sample_scaling_factors);
>-  ScaleSamples(per_sample_scaling_factors, signal);
>-
>-  last_scaling_factor_ = scaling_factors_.back();
>-
>-  // Dump data for debug.
>-  apm_data_dumper_->DumpRaw("agc2_gain_curve_applier_scaling_factors",
>-                            samples_per_channel,
>-                            per_sample_scaling_factors_.data());
>-}
>-
>-InterpolatedGainCurve::Stats GainCurveApplier::GetGainCurveStats() const {
>-  return interp_gain_curve_.get_stats();
>-}
>-
>-void GainCurveApplier::SetSampleRate(size_t sample_rate_hz) {
>-  level_estimator_.SetSampleRate(sample_rate_hz);
>-  // Check that per_sample_scaling_factors_ is large enough.
>-  RTC_DCHECK_LE(sample_rate_hz,
>-                kMaximalNumberOfSamplesPerChannel * 1000 / kFrameDurationMs);
>-}
>-
>-void GainCurveApplier::Reset() {
>-  level_estimator_.Reset();
>-}
>-
>-float GainCurveApplier::LastAudioLevel() const {
>-  return level_estimator_.LastAudioLevel();
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.h
>deleted file mode 100644
>index e0be19e06944c83c12ed695cbc50072dd8eba34f..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.h
>+++ /dev/null
>@@ -1,63 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_AGC2_GAIN_CURVE_APPLIER_H_
>-#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_CURVE_APPLIER_H_
>-
>-#include <vector>
>-
>-#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
>-#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
>-#include "modules/audio_processing/include/audio_frame_view.h"
>-#include "rtc_base/constructormagic.h"
>-
>-namespace webrtc {
>-class ApmDataDumper;
>-
>-class GainCurveApplier {
>- public:
>-  GainCurveApplier(size_t sample_rate_hz,
>-                   ApmDataDumper* apm_data_dumper,
>-                   std::string histogram_name_prefix);
>-
>-  ~GainCurveApplier();
>-
>-  void Process(AudioFrameView<float> signal);
>-  InterpolatedGainCurve::Stats GetGainCurveStats() const;
>-
>-  // Supported rates must be
>-  // * supported by FixedDigitalLevelEstimator
>-  // * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
>-  //   so that samples_per_channel fit in the
>-  //   per_sample_scaling_factors_ array.
>-  void SetSampleRate(size_t sample_rate_hz);
>-
>-  // Resets the internal state.
>-  void Reset();
>-
>-  float LastAudioLevel() const;
>-
>- private:
>-  const InterpolatedGainCurve interp_gain_curve_;
>-  FixedDigitalLevelEstimator level_estimator_;
>-  ApmDataDumper* const apm_data_dumper_ = nullptr;
>-
>-  // Work array containing the sub-frame scaling factors to be interpolated.
>-  std::array<float, kSubFramesInFrame + 1> scaling_factors_ = {};
>-  std::array<float, kMaximalNumberOfSamplesPerChannel>
>-      per_sample_scaling_factors_ = {};
>-  float last_scaling_factor_ = 1.f;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(GainCurveApplier);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_AGC2_GAIN_CURVE_APPLIER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier_unittest.cc
>deleted file mode 100644
>index 0f75f62c407cbb3d99048f4082bbc7c49a1ff0c8..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier_unittest.cc
>+++ /dev/null
>@@ -1,60 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/agc2/gain_curve_applier.h"
>-
>-#include "common_audio/include/audio_util.h"
>-#include "modules/audio_processing/agc2/agc2_common.h"
>-#include "modules/audio_processing/agc2/agc2_testing_common.h"
>-#include "modules/audio_processing/agc2/vector_float_frame.h"
>-#include "modules/audio_processing/logging/apm_data_dumper.h"
>-#include "rtc_base/gunit.h"
>-
>-namespace webrtc {
>-
>-TEST(GainCurveApplier, GainCurveApplierShouldConstructAndRun) {
>-  const int sample_rate_hz = 48000;
>-  ApmDataDumper apm_data_dumper(0);
>-
>-  GainCurveApplier gain_curve_applier(sample_rate_hz, &apm_data_dumper, "");
>-
>-  VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
>-                                            kMaxAbsFloatS16Value);
>-  gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
>-}
>-
>-TEST(GainCurveApplier, OutputVolumeAboveThreshold) {
>-  const int sample_rate_hz = 48000;
>-  const float input_level =
>-      (kMaxAbsFloatS16Value + DbfsToFloatS16(test::kLimiterMaxInputLevelDbFs)) /
>-      2.f;
>-  ApmDataDumper apm_data_dumper(0);
>-
>-  GainCurveApplier gain_curve_applier(sample_rate_hz, &apm_data_dumper, "");
>-
>-  // Give the level estimator time to adapt.
>-  for (int i = 0; i < 5; ++i) {
>-    VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
>-                                              input_level);
>-    gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
>-  }
>-
>-  VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
>-                                            input_level);
>-  gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
>-  rtc::ArrayView<const float> channel =
>-      vectors_with_float_frame.float_frame_view().channel(0);
>-
>-  for (const auto& sample : channel) {
>-    EXPECT_LT(0.9f * kMaxAbsFloatS16Value, sample);
>-  }
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc
>index 73e6a8ecc36f1560c82509797ffd55875f8bd0f7..f5d6b47169c2226d4d1c2dd125b396d00b792c57 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc
>@@ -10,10 +10,12 @@
> 
> #include "modules/audio_processing/agc2/interpolated_gain_curve.h"
> 
>+#include <algorithm>
>+#include <iterator>
>+
> #include "modules/audio_processing/agc2/agc2_common.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc
>index dd696313ae266574a4b8531ee9fbb96450cc2b0a..a8e0f2361c91a1213067807e04958d01f58f7879 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include <array>
>+#include <type_traits>
> #include <vector>
> 
> #include "api/array_view.h"
>@@ -16,7 +17,7 @@
> #include "modules/audio_processing/agc2/agc2_common.h"
> #include "modules/audio_processing/agc2/compute_interpolated_gain_curve.h"
> #include "modules/audio_processing/agc2/interpolated_gain_curve.h"
>-#include "modules/audio_processing/agc2/limiter.h"
>+#include "modules/audio_processing/agc2/limiter_db_gain_curve.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/gunit.h"
>@@ -27,7 +28,8 @@ namespace {
> constexpr double kLevelEpsilon = 1e-2 * kMaxAbsFloatS16Value;
> constexpr float kInterpolatedGainCurveTolerance = 1.f / 32768.f;
> ApmDataDumper apm_data_dumper(0);
>-const Limiter limiter;
>+static_assert(std::is_trivially_destructible<LimiterDbGainCurve>::value, "");
>+const LimiterDbGainCurve limiter;
> 
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.cc
>index d2b9877be11ed4a361de93fa6dad73520aa00cdf..1589f07404d990039ddb93fe063c41dbea5ecea5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.cc
>@@ -10,128 +10,141 @@
> 
> #include "modules/audio_processing/agc2/limiter.h"
> 
>+#include <algorithm>
>+#include <array>
> #include <cmath>
> 
>-#include "common_audio/include/audio_util.h"
>+#include "api/array_view.h"
> #include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_minmax.h"
> 
> namespace webrtc {
> namespace {
> 
>-double ComputeKneeStart(double max_input_level_db,
>-                        double knee_smoothness_db,
>-                        double compression_ratio) {
>-  RTC_CHECK_LT((compression_ratio - 1.0) * knee_smoothness_db /
>-                   (2.0 * compression_ratio),
>-               max_input_level_db);
>-  return -knee_smoothness_db / 2.0 -
>-         max_input_level_db / (compression_ratio - 1.0);
>+// This constant affects the way scaling factors are interpolated for the first
>+// sub-frame of a frame. Only in the case in which the first sub-frame has an
>+// estimated level which is greater than the that of the previous analyzed
>+// sub-frame, linear interpolation is replaced with a power function which
>+// reduces the chances of over-shooting (and hence saturation), however reducing
>+// the fixed gain effectiveness.
>+constexpr float kAttackFirstSubframeInterpolationPower = 8.f;
>+
>+void InterpolateFirstSubframe(float last_factor,
>+                              float current_factor,
>+                              rtc::ArrayView<float> subframe) {
>+  const auto n = subframe.size();
>+  constexpr auto p = kAttackFirstSubframeInterpolationPower;
>+  for (size_t i = 0; i < n; ++i) {
>+    subframe[i] = std::pow(1.f - i / n, p) * (last_factor - current_factor) +
>+                  current_factor;
>+  }
> }
> 
>-std::array<double, 3> ComputeKneeRegionPolynomial(double knee_start_dbfs,
>-                                                  double knee_smoothness_db,
>-                                                  double compression_ratio) {
>-  const double a = (1.0 - compression_ratio) /
>-                   (2.0 * knee_smoothness_db * compression_ratio);
>-  const double b = 1.0 - 2.0 * a * knee_start_dbfs;
>-  const double c = a * knee_start_dbfs * knee_start_dbfs;
>-  return {{a, b, c}};
>-}
>+void ComputePerSampleSubframeFactors(
>+    const std::array<float, kSubFramesInFrame + 1>& scaling_factors,
>+    size_t samples_per_channel,
>+    rtc::ArrayView<float> per_sample_scaling_factors) {
>+  const size_t num_subframes = scaling_factors.size() - 1;
>+  const size_t subframe_size =
>+      rtc::CheckedDivExact(samples_per_channel, num_subframes);
>+
>+  // Handle first sub-frame differently in case of attack.
>+  const bool is_attack = scaling_factors[0] > scaling_factors[1];
>+  if (is_attack) {
>+    InterpolateFirstSubframe(
>+        scaling_factors[0], scaling_factors[1],
>+        rtc::ArrayView<float>(
>+            per_sample_scaling_factors.subview(0, subframe_size)));
>+  }
> 
>-double ComputeLimiterD1(double max_input_level_db, double compression_ratio) {
>-  return (std::pow(10.0, -max_input_level_db / (20.0 * compression_ratio)) *
>-          (1.0 - compression_ratio) / compression_ratio) /
>-         kMaxAbsFloatS16Value;
>+  for (size_t i = is_attack ? 1 : 0; i < num_subframes; ++i) {
>+    const size_t subframe_start = i * subframe_size;
>+    const float scaling_start = scaling_factors[i];
>+    const float scaling_end = scaling_factors[i + 1];
>+    const float scaling_diff = (scaling_end - scaling_start) / subframe_size;
>+    for (size_t j = 0; j < subframe_size; ++j) {
>+      per_sample_scaling_factors[subframe_start + j] =
>+          scaling_start + scaling_diff * j;
>+    }
>+  }
> }
> 
>-constexpr double ComputeLimiterD2(double compression_ratio) {
>-  return (1.0 - 2.0 * compression_ratio) / compression_ratio;
>+void ScaleSamples(rtc::ArrayView<const float> per_sample_scaling_factors,
>+                  AudioFrameView<float> signal) {
>+  const size_t samples_per_channel = signal.samples_per_channel();
>+  RTC_DCHECK_EQ(samples_per_channel, per_sample_scaling_factors.size());
>+  for (size_t i = 0; i < signal.num_channels(); ++i) {
>+    auto channel = signal.channel(i);
>+    for (size_t j = 0; j < samples_per_channel; ++j) {
>+      channel[j] = rtc::SafeClamp(channel[j] * per_sample_scaling_factors[j],
>+                                  kMinFloatS16Value, kMaxFloatS16Value);
>+    }
>+  }
> }
> 
>-double ComputeLimiterI2(double max_input_level_db,
>-                        double compression_ratio,
>-                        double gain_curve_limiter_i1) {
>-  RTC_CHECK_NE(gain_curve_limiter_i1, 0.f);
>-  return std::pow(10.0, -max_input_level_db / (20.0 * compression_ratio)) /
>-         gain_curve_limiter_i1 /
>-         std::pow(kMaxAbsFloatS16Value, gain_curve_limiter_i1 - 1);
>+void CheckLimiterSampleRate(size_t sample_rate_hz) {
>+  // Check that per_sample_scaling_factors_ is large enough.
>+  RTC_DCHECK_LE(sample_rate_hz,
>+                kMaximalNumberOfSamplesPerChannel * 1000 / kFrameDurationMs);
> }
> 
> }  // namespace
> 
>-Limiter::Limiter()
>-    : max_input_level_linear_(DbfsToFloatS16(max_input_level_db_)),
>-      knee_start_dbfs_(ComputeKneeStart(max_input_level_db_,
>-                                        knee_smoothness_db_,
>-                                        compression_ratio_)),
>-      knee_start_linear_(DbfsToFloatS16(knee_start_dbfs_)),
>-      limiter_start_dbfs_(knee_start_dbfs_ + knee_smoothness_db_),
>-      limiter_start_linear_(DbfsToFloatS16(limiter_start_dbfs_)),
>-      knee_region_polynomial_(ComputeKneeRegionPolynomial(knee_start_dbfs_,
>-                                                          knee_smoothness_db_,
>-                                                          compression_ratio_)),
>-      gain_curve_limiter_d1_(
>-          ComputeLimiterD1(max_input_level_db_, compression_ratio_)),
>-      gain_curve_limiter_d2_(ComputeLimiterD2(compression_ratio_)),
>-      gain_curve_limiter_i1_(1.0 / compression_ratio_),
>-      gain_curve_limiter_i2_(ComputeLimiterI2(max_input_level_db_,
>-                                              compression_ratio_,
>-                                              gain_curve_limiter_i1_)) {
>-  static_assert(knee_smoothness_db_ > 0.0f, "");
>-  static_assert(compression_ratio_ > 1.0f, "");
>-  RTC_CHECK_GE(max_input_level_db_, knee_start_dbfs_ + knee_smoothness_db_);
>+Limiter::Limiter(size_t sample_rate_hz,
>+                 ApmDataDumper* apm_data_dumper,
>+                 std::string histogram_name)
>+    : interp_gain_curve_(apm_data_dumper, histogram_name),
>+      level_estimator_(sample_rate_hz, apm_data_dumper),
>+      apm_data_dumper_(apm_data_dumper) {
>+  CheckLimiterSampleRate(sample_rate_hz);
> }
> 
>-constexpr double Limiter::max_input_level_db_;
>-constexpr double Limiter::knee_smoothness_db_;
>-constexpr double Limiter::compression_ratio_;
>+Limiter::~Limiter() = default;
> 
>-double Limiter::GetOutputLevelDbfs(double input_level_dbfs) const {
>-  if (input_level_dbfs < knee_start_dbfs_) {
>-    return input_level_dbfs;
>-  } else if (input_level_dbfs < limiter_start_dbfs_) {
>-    return GetKneeRegionOutputLevelDbfs(input_level_dbfs);
>-  }
>-  return GetCompressorRegionOutputLevelDbfs(input_level_dbfs);
>-}
>+void Limiter::Process(AudioFrameView<float> signal) {
>+  const auto level_estimate = level_estimator_.ComputeLevel(signal);
> 
>-double Limiter::GetGainLinear(double input_level_linear) const {
>-  if (input_level_linear < knee_start_linear_) {
>-    return 1.0;
>-  }
>-  return DbfsToFloatS16(
>-             GetOutputLevelDbfs(FloatS16ToDbfs(input_level_linear))) /
>-         input_level_linear;
>+  RTC_DCHECK_EQ(level_estimate.size() + 1, scaling_factors_.size());
>+  scaling_factors_[0] = last_scaling_factor_;
>+  std::transform(level_estimate.begin(), level_estimate.end(),
>+                 scaling_factors_.begin() + 1, [this](float x) {
>+                   return interp_gain_curve_.LookUpGainToApply(x);
>+                 });
>+
>+  const size_t samples_per_channel = signal.samples_per_channel();
>+  RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
>+
>+  auto per_sample_scaling_factors = rtc::ArrayView<float>(
>+      &per_sample_scaling_factors_[0], samples_per_channel);
>+  ComputePerSampleSubframeFactors(scaling_factors_, samples_per_channel,
>+                                  per_sample_scaling_factors);
>+  ScaleSamples(per_sample_scaling_factors, signal);
>+
>+  last_scaling_factor_ = scaling_factors_.back();
>+
>+  // Dump data for debug.
>+  apm_data_dumper_->DumpRaw("agc2_gain_curve_applier_scaling_factors",
>+                            samples_per_channel,
>+                            per_sample_scaling_factors_.data());
> }
> 
>-// Computes the first derivative of GetGainLinear() in |x|.
>-double Limiter::GetGainFirstDerivativeLinear(double x) const {
>-  // Beyond-knee region only.
>-  RTC_CHECK_GE(x, limiter_start_linear_ - 1e-7 * kMaxAbsFloatS16Value);
>-  return gain_curve_limiter_d1_ *
>-         std::pow(x / kMaxAbsFloatS16Value, gain_curve_limiter_d2_);
>+InterpolatedGainCurve::Stats Limiter::GetGainCurveStats() const {
>+  return interp_gain_curve_.get_stats();
> }
> 
>-// Computes the integral of GetGainLinear() in the range [x0, x1].
>-double Limiter::GetGainIntegralLinear(double x0, double x1) const {
>-  RTC_CHECK_LE(x0, x1);                     // Valid interval.
>-  RTC_CHECK_GE(x0, limiter_start_linear_);  // Beyond-knee region only.
>-  auto limiter_integral = [this](const double& x) {
>-    return gain_curve_limiter_i2_ * std::pow(x, gain_curve_limiter_i1_);
>-  };
>-  return limiter_integral(x1) - limiter_integral(x0);
>+void Limiter::SetSampleRate(size_t sample_rate_hz) {
>+  CheckLimiterSampleRate(sample_rate_hz);
>+  level_estimator_.SetSampleRate(sample_rate_hz);
> }
> 
>-double Limiter::GetKneeRegionOutputLevelDbfs(double input_level_dbfs) const {
>-  return knee_region_polynomial_[0] * input_level_dbfs * input_level_dbfs +
>-         knee_region_polynomial_[1] * input_level_dbfs +
>-         knee_region_polynomial_[2];
>+void Limiter::Reset() {
>+  level_estimator_.Reset();
> }
> 
>-double Limiter::GetCompressorRegionOutputLevelDbfs(
>-    double input_level_dbfs) const {
>-  return (input_level_dbfs - max_input_level_db_) / compression_ratio_;
>+float Limiter::LastAudioLevel() const {
>+  return level_estimator_.LastAudioLevel();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.h
>index f350baeef105e3b9acff472bf947872f80eb1db8..1e0ab71a183786dd1616bd9e03086d2df7ae0369 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.h
>@@ -11,62 +11,52 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
> 
>-#include <array>
>+#include <string>
>+#include <vector>
> 
>-#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
>+#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+#include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>+class ApmDataDumper;
> 
>-// A class for computing gain curve parameters. The gain curve is
>-// defined by constants kLimiterMaxInputLevelDbFs, kLimiterKneeSmoothnessDb,
>-// kLimiterCompressionRatio. The curve consints of one linear part,
>-// one quadratic polynomial part and another linear part. The
>-// constants define the parameters of the parts.
> class Limiter {
>  public:
>-  Limiter();
>+  Limiter(size_t sample_rate_hz,
>+          ApmDataDumper* apm_data_dumper,
>+          std::string histogram_name_prefix);
>+  Limiter(const Limiter& limiter) = delete;
>+  Limiter& operator=(const Limiter& limiter) = delete;
>+  ~Limiter();
> 
>-  double max_input_level_db() const { return max_input_level_db_; }
>-  double max_input_level_linear() const { return max_input_level_linear_; }
>-  double knee_start_linear() const { return knee_start_linear_; }
>-  double limiter_start_linear() const { return limiter_start_linear_; }
>+  // Applies limiter and hard-clipping to |signal|.
>+  void Process(AudioFrameView<float> signal);
>+  InterpolatedGainCurve::Stats GetGainCurveStats() const;
> 
>-  // These methods can be marked 'constexpr' in C++ 14.
>-  double GetOutputLevelDbfs(double input_level_dbfs) const;
>-  double GetGainLinear(double input_level_linear) const;
>-  double GetGainFirstDerivativeLinear(double x) const;
>-  double GetGainIntegralLinear(double x0, double x1) const;
>+  // Supported rates must be
>+  // * supported by FixedDigitalLevelEstimator
>+  // * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
>+  //   so that samples_per_channel fit in the
>+  //   per_sample_scaling_factors_ array.
>+  void SetSampleRate(size_t sample_rate_hz);
> 
>- private:
>-  double GetKneeRegionOutputLevelDbfs(double input_level_dbfs) const;
>-  double GetCompressorRegionOutputLevelDbfs(double input_level_dbfs) const;
>-
>-  static constexpr double max_input_level_db_ = test::kLimiterMaxInputLevelDbFs;
>-  static constexpr double knee_smoothness_db_ = test::kLimiterKneeSmoothnessDb;
>-  static constexpr double compression_ratio_ = test::kLimiterCompressionRatio;
>-
>-  const double max_input_level_linear_;
>-
>-  // Do not modify signal with level <= knee_start_dbfs_.
>-  const double knee_start_dbfs_;
>-  const double knee_start_linear_;
>+  // Resets the internal state.
>+  void Reset();
> 
>-  // The upper end of the knee region, which is between knee_start_dbfs_ and
>-  // limiter_start_dbfs_.
>-  const double limiter_start_dbfs_;
>-  const double limiter_start_linear_;
>+  float LastAudioLevel() const;
> 
>-  // Coefficients {a, b, c} of the knee region polynomial
>-  // ax^2 + bx + c in the DB scale.
>-  const std::array<double, 3> knee_region_polynomial_;
>-
>-  // Parameters for the computation of the first derivative of GetGainLinear().
>-  const double gain_curve_limiter_d1_;
>-  const double gain_curve_limiter_d2_;
>-
>-  // Parameters for the computation of the integral of GetGainLinear().
>-  const double gain_curve_limiter_i1_;
>-  const double gain_curve_limiter_i2_;
>+ private:
>+  const InterpolatedGainCurve interp_gain_curve_;
>+  FixedDigitalLevelEstimator level_estimator_;
>+  ApmDataDumper* const apm_data_dumper_ = nullptr;
>+
>+  // Work array containing the sub-frame scaling factors to be interpolated.
>+  std::array<float, kSubFramesInFrame + 1> scaling_factors_ = {};
>+  std::array<float, kMaximalNumberOfSamplesPerChannel>
>+      per_sample_scaling_factors_ = {};
>+  float last_scaling_factor_ = 1.f;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_db_gain_curve.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_db_gain_curve.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..d55ed5df58c1aa5dc1834f6019c011322d2e37c2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_db_gain_curve.cc
>@@ -0,0 +1,138 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/limiter_db_gain_curve.h"
>+
>+#include <cmath>
>+
>+#include "common_audio/include/audio_util.h"
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+double ComputeKneeStart(double max_input_level_db,
>+                        double knee_smoothness_db,
>+                        double compression_ratio) {
>+  RTC_CHECK_LT((compression_ratio - 1.0) * knee_smoothness_db /
>+                   (2.0 * compression_ratio),
>+               max_input_level_db);
>+  return -knee_smoothness_db / 2.0 -
>+         max_input_level_db / (compression_ratio - 1.0);
>+}
>+
>+std::array<double, 3> ComputeKneeRegionPolynomial(double knee_start_dbfs,
>+                                                  double knee_smoothness_db,
>+                                                  double compression_ratio) {
>+  const double a = (1.0 - compression_ratio) /
>+                   (2.0 * knee_smoothness_db * compression_ratio);
>+  const double b = 1.0 - 2.0 * a * knee_start_dbfs;
>+  const double c = a * knee_start_dbfs * knee_start_dbfs;
>+  return {{a, b, c}};
>+}
>+
>+double ComputeLimiterD1(double max_input_level_db, double compression_ratio) {
>+  return (std::pow(10.0, -max_input_level_db / (20.0 * compression_ratio)) *
>+          (1.0 - compression_ratio) / compression_ratio) /
>+         kMaxAbsFloatS16Value;
>+}
>+
>+constexpr double ComputeLimiterD2(double compression_ratio) {
>+  return (1.0 - 2.0 * compression_ratio) / compression_ratio;
>+}
>+
>+double ComputeLimiterI2(double max_input_level_db,
>+                        double compression_ratio,
>+                        double gain_curve_limiter_i1) {
>+  RTC_CHECK_NE(gain_curve_limiter_i1, 0.f);
>+  return std::pow(10.0, -max_input_level_db / (20.0 * compression_ratio)) /
>+         gain_curve_limiter_i1 /
>+         std::pow(kMaxAbsFloatS16Value, gain_curve_limiter_i1 - 1);
>+}
>+
>+}  // namespace
>+
>+LimiterDbGainCurve::LimiterDbGainCurve()
>+    : max_input_level_linear_(DbfsToFloatS16(max_input_level_db_)),
>+      knee_start_dbfs_(ComputeKneeStart(max_input_level_db_,
>+                                        knee_smoothness_db_,
>+                                        compression_ratio_)),
>+      knee_start_linear_(DbfsToFloatS16(knee_start_dbfs_)),
>+      limiter_start_dbfs_(knee_start_dbfs_ + knee_smoothness_db_),
>+      limiter_start_linear_(DbfsToFloatS16(limiter_start_dbfs_)),
>+      knee_region_polynomial_(ComputeKneeRegionPolynomial(knee_start_dbfs_,
>+                                                          knee_smoothness_db_,
>+                                                          compression_ratio_)),
>+      gain_curve_limiter_d1_(
>+          ComputeLimiterD1(max_input_level_db_, compression_ratio_)),
>+      gain_curve_limiter_d2_(ComputeLimiterD2(compression_ratio_)),
>+      gain_curve_limiter_i1_(1.0 / compression_ratio_),
>+      gain_curve_limiter_i2_(ComputeLimiterI2(max_input_level_db_,
>+                                              compression_ratio_,
>+                                              gain_curve_limiter_i1_)) {
>+  static_assert(knee_smoothness_db_ > 0.0f, "");
>+  static_assert(compression_ratio_ > 1.0f, "");
>+  RTC_CHECK_GE(max_input_level_db_, knee_start_dbfs_ + knee_smoothness_db_);
>+}
>+
>+constexpr double LimiterDbGainCurve::max_input_level_db_;
>+constexpr double LimiterDbGainCurve::knee_smoothness_db_;
>+constexpr double LimiterDbGainCurve::compression_ratio_;
>+
>+double LimiterDbGainCurve::GetOutputLevelDbfs(double input_level_dbfs) const {
>+  if (input_level_dbfs < knee_start_dbfs_) {
>+    return input_level_dbfs;
>+  } else if (input_level_dbfs < limiter_start_dbfs_) {
>+    return GetKneeRegionOutputLevelDbfs(input_level_dbfs);
>+  }
>+  return GetCompressorRegionOutputLevelDbfs(input_level_dbfs);
>+}
>+
>+double LimiterDbGainCurve::GetGainLinear(double input_level_linear) const {
>+  if (input_level_linear < knee_start_linear_) {
>+    return 1.0;
>+  }
>+  return DbfsToFloatS16(
>+             GetOutputLevelDbfs(FloatS16ToDbfs(input_level_linear))) /
>+         input_level_linear;
>+}
>+
>+// Computes the first derivative of GetGainLinear() in |x|.
>+double LimiterDbGainCurve::GetGainFirstDerivativeLinear(double x) const {
>+  // Beyond-knee region only.
>+  RTC_CHECK_GE(x, limiter_start_linear_ - 1e-7 * kMaxAbsFloatS16Value);
>+  return gain_curve_limiter_d1_ *
>+         std::pow(x / kMaxAbsFloatS16Value, gain_curve_limiter_d2_);
>+}
>+
>+// Computes the integral of GetGainLinear() in the range [x0, x1].
>+double LimiterDbGainCurve::GetGainIntegralLinear(double x0, double x1) const {
>+  RTC_CHECK_LE(x0, x1);                     // Valid interval.
>+  RTC_CHECK_GE(x0, limiter_start_linear_);  // Beyond-knee region only.
>+  auto limiter_integral = [this](const double& x) {
>+    return gain_curve_limiter_i2_ * std::pow(x, gain_curve_limiter_i1_);
>+  };
>+  return limiter_integral(x1) - limiter_integral(x0);
>+}
>+
>+double LimiterDbGainCurve::GetKneeRegionOutputLevelDbfs(
>+    double input_level_dbfs) const {
>+  return knee_region_polynomial_[0] * input_level_dbfs * input_level_dbfs +
>+         knee_region_polynomial_[1] * input_level_dbfs +
>+         knee_region_polynomial_[2];
>+}
>+
>+double LimiterDbGainCurve::GetCompressorRegionOutputLevelDbfs(
>+    double input_level_dbfs) const {
>+  return (input_level_dbfs - max_input_level_db_) / compression_ratio_;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_db_gain_curve.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_db_gain_curve.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..9086e26739b7cfc20fe6a4c353a4d7bc5aa06885
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_db_gain_curve.h
>@@ -0,0 +1,76 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_LIMITER_DB_GAIN_CURVE_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_DB_GAIN_CURVE_H_
>+
>+#include <array>
>+
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+
>+namespace webrtc {
>+
>+// A class for computing a limiter gain curve (in dB scale) given a set of
>+// hard-coded parameters (namely, kLimiterDbGainCurveMaxInputLevelDbFs,
>+// kLimiterDbGainCurveKneeSmoothnessDb, and
>+// kLimiterDbGainCurveCompressionRatio). The generated curve consists of four
>+// regions: identity (linear), knee (quadratic polynomial), compression
>+// (linear), saturation (linear). The aforementioned constants are used to shape
>+// the different regions.
>+class LimiterDbGainCurve {
>+ public:
>+  LimiterDbGainCurve();
>+
>+  double max_input_level_db() const { return max_input_level_db_; }
>+  double max_input_level_linear() const { return max_input_level_linear_; }
>+  double knee_start_linear() const { return knee_start_linear_; }
>+  double limiter_start_linear() const { return limiter_start_linear_; }
>+
>+  // These methods can be marked 'constexpr' in C++ 14.
>+  double GetOutputLevelDbfs(double input_level_dbfs) const;
>+  double GetGainLinear(double input_level_linear) const;
>+  double GetGainFirstDerivativeLinear(double x) const;
>+  double GetGainIntegralLinear(double x0, double x1) const;
>+
>+ private:
>+  double GetKneeRegionOutputLevelDbfs(double input_level_dbfs) const;
>+  double GetCompressorRegionOutputLevelDbfs(double input_level_dbfs) const;
>+
>+  static constexpr double max_input_level_db_ = test::kLimiterMaxInputLevelDbFs;
>+  static constexpr double knee_smoothness_db_ = test::kLimiterKneeSmoothnessDb;
>+  static constexpr double compression_ratio_ = test::kLimiterCompressionRatio;
>+
>+  const double max_input_level_linear_;
>+
>+  // Do not modify signal with level <= knee_start_dbfs_.
>+  const double knee_start_dbfs_;
>+  const double knee_start_linear_;
>+
>+  // The upper end of the knee region, which is between knee_start_dbfs_ and
>+  // limiter_start_dbfs_.
>+  const double limiter_start_dbfs_;
>+  const double limiter_start_linear_;
>+
>+  // Coefficients {a, b, c} of the knee region polynomial
>+  // ax^2 + bx + c in the DB scale.
>+  const std::array<double, 3> knee_region_polynomial_;
>+
>+  // Parameters for the computation of the first derivative of GetGainLinear().
>+  const double gain_curve_limiter_d1_;
>+  const double gain_curve_limiter_d2_;
>+
>+  // Parameters for the computation of the integral of GetGainLinear().
>+  const double gain_curve_limiter_i1_;
>+  const double gain_curve_limiter_i2_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_LIMITER_DB_GAIN_CURVE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_db_gain_curve_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_db_gain_curve_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..049c8d568ea50f0caef27952f1d7bfa5ed0d54b6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_db_gain_curve_unittest.cc
>@@ -0,0 +1,60 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/limiter_db_gain_curve.h"
>+
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+
>+TEST(FixedDigitalGainController2Limiter, ConstructDestruct) {
>+  LimiterDbGainCurve l;
>+}
>+
>+TEST(FixedDigitalGainController2Limiter, GainCurveShouldBeMonotone) {
>+  LimiterDbGainCurve l;
>+  float last_output_level = 0.f;
>+  bool has_last_output_level = false;
>+  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
>+    const float current_output_level = l.GetOutputLevelDbfs(level);
>+    if (!has_last_output_level) {
>+      last_output_level = current_output_level;
>+      has_last_output_level = true;
>+    }
>+    EXPECT_LE(last_output_level, current_output_level);
>+    last_output_level = current_output_level;
>+  }
>+}
>+
>+TEST(FixedDigitalGainController2Limiter, GainCurveShouldBeContinuous) {
>+  LimiterDbGainCurve l;
>+  float last_output_level = 0.f;
>+  bool has_last_output_level = false;
>+  constexpr float kMaxDelta = 0.5f;
>+  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
>+    const float current_output_level = l.GetOutputLevelDbfs(level);
>+    if (!has_last_output_level) {
>+      last_output_level = current_output_level;
>+      has_last_output_level = true;
>+    }
>+    EXPECT_LE(current_output_level, last_output_level + kMaxDelta);
>+    last_output_level = current_output_level;
>+  }
>+}
>+
>+TEST(FixedDigitalGainController2Limiter, OutputGainShouldBeLessThanFullScale) {
>+  LimiterDbGainCurve l;
>+  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
>+    const float current_output_level = l.GetOutputLevelDbfs(level);
>+    EXPECT_LE(current_output_level, 0.f);
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_unittest.cc
>index 707981263475aa3b9f0eb35189175985bb3a7932..e662a7fc890458e38b58f7812f70299387fdae2d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_unittest.cc
>@@ -10,50 +10,50 @@
> 
> #include "modules/audio_processing/agc2/limiter.h"
> 
>+#include "common_audio/include/audio_util.h"
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+#include "modules/audio_processing/agc2/vector_float_frame.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/gunit.h"
> 
> namespace webrtc {
> 
>-TEST(FixedDigitalGainController2Limiter, ConstructDestruct) {
>-  Limiter l;
>-}
>+TEST(Limiter, LimiterShouldConstructAndRun) {
>+  const int sample_rate_hz = 48000;
>+  ApmDataDumper apm_data_dumper(0);
> 
>-TEST(FixedDigitalGainController2Limiter, GainCurveShouldBeMonotone) {
>-  Limiter l;
>-  float last_output_level = 0.f;
>-  bool has_last_output_level = false;
>-  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
>-    const float current_output_level = l.GetOutputLevelDbfs(level);
>-    if (!has_last_output_level) {
>-      last_output_level = current_output_level;
>-      has_last_output_level = true;
>-    }
>-    EXPECT_LE(last_output_level, current_output_level);
>-    last_output_level = current_output_level;
>-  }
>+  Limiter limiter(sample_rate_hz, &apm_data_dumper, "");
>+
>+  VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
>+                                            kMaxAbsFloatS16Value);
>+  limiter.Process(vectors_with_float_frame.float_frame_view());
> }
> 
>-TEST(FixedDigitalGainController2Limiter, GainCurveShouldBeContinuous) {
>-  Limiter l;
>-  float last_output_level = 0.f;
>-  bool has_last_output_level = false;
>-  constexpr float kMaxDelta = 0.5f;
>-  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
>-    const float current_output_level = l.GetOutputLevelDbfs(level);
>-    if (!has_last_output_level) {
>-      last_output_level = current_output_level;
>-      has_last_output_level = true;
>-    }
>-    EXPECT_LE(current_output_level, last_output_level + kMaxDelta);
>-    last_output_level = current_output_level;
>+TEST(Limiter, OutputVolumeAboveThreshold) {
>+  const int sample_rate_hz = 48000;
>+  const float input_level =
>+      (kMaxAbsFloatS16Value + DbfsToFloatS16(test::kLimiterMaxInputLevelDbFs)) /
>+      2.f;
>+  ApmDataDumper apm_data_dumper(0);
>+
>+  Limiter limiter(sample_rate_hz, &apm_data_dumper, "");
>+
>+  // Give the level estimator time to adapt.
>+  for (int i = 0; i < 5; ++i) {
>+    VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
>+                                              input_level);
>+    limiter.Process(vectors_with_float_frame.float_frame_view());
>   }
>-}
> 
>-TEST(FixedDigitalGainController2Limiter, OutputGainShouldBeLessThanFullScale) {
>-  Limiter l;
>-  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
>-    const float current_output_level = l.GetOutputLevelDbfs(level);
>-    EXPECT_LE(current_output_level, 0.f);
>+  VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
>+                                            input_level);
>+  limiter.Process(vectors_with_float_frame.float_frame_view());
>+  rtc::ArrayView<const float> channel =
>+      vectors_with_float_frame.float_frame_view().channel(0);
>+
>+  for (const auto& sample : channel) {
>+    EXPECT_LT(0.9f * kMaxAbsFloatS16Value, sample);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.cc
>index d9aaf1f9bd3f6689de7fad047b9334a3f0f706d2..6e43672ce0316f2762ce960e62607bc85b972c6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.cc
>@@ -10,13 +10,15 @@
> 
> #include "modules/audio_processing/agc2/noise_level_estimator.h"
> 
>-#include <math.h>
>-
>+#include <stddef.h>
> #include <algorithm>
>+#include <cmath>
> #include <numeric>
> 
>+#include "api/array_view.h"
> #include "common_audio/include/audio_util.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.cc
>index 9e08126e892e3752c69ab3fc8120ce64d51a2d17..5735faf3d2cbd364d0d50c8b5ad0b4abb0d01cd2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.cc
>@@ -16,6 +16,7 @@
> #include "api/array_view.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
>index 8ab56736a2f2b8b919f43f6415bff14b9024262a..8f472a55ed649d28121b9ea4c2bc15745c096eb9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
> 
>+#include <array>
>+
> #include "modules/audio_processing/agc2/rnn_vad/lp_residual.h"
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h
>index 1f63885c4e6d8774c2c44dc01125f598a753a402..ce5cce1857cd417bcbea4d79d011944cb86e5e4a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h
>@@ -11,12 +11,12 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
> 
>-#include <memory>
> #include <vector>
> 
> #include "api/array_view.h"
> #include "modules/audio_processing/agc2/biquad_filter.h"
> #include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "modules/audio_processing/agc2/rnn_vad/pitch_info.h"
> #include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
> #include "modules/audio_processing/agc2/rnn_vad/sequence_buffer.h"
> #include "modules/audio_processing/agc2/rnn_vad/spectral_features.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.cc
>index 1017400a2824d1bba2c462c0673348bc14a5baa0..a1c5dac47774e40b189708b6545d5f9af901ee25 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/audio_processing/agc2/rnn_vad/fft_util.h"
> 
>+#include <stddef.h>
> #include <cmath>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h
>index bffafd291f936e79a4961b1f7dddbb4b5e157f21..cddedca5d614ec94a0f36544b73d9309e4e8dd6a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h
>@@ -11,6 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_LP_RESIDUAL_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_LP_RESIDUAL_H_
> 
>+#include <stddef.h>
>+
> #include "api/array_view.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
>index 1f8859d5b04d453506ba185e7cc90ee8d4af9af3..aa0b751d28a1abd5e4d70cd5e2d0c91e644f0827 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
>@@ -10,6 +10,11 @@
> 
> #include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
> 
>+#include <array>
>+#include <cstddef>
>+
>+#include "rtc_base/checks.h"
>+
> namespace webrtc {
> namespace rnn_vad {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
>index b7b44d293769c0af932416922b75ed6aae002534..32ee8c00df849b5642170a080304c51df3c67fd0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
>@@ -10,10 +10,12 @@
> 
> #include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
> 
>+#include <stdlib.h>
> #include <algorithm>
> #include <cmath>
>+#include <complex>
>+#include <cstddef>
> #include <numeric>
>-#include <utility>
> 
> #include "modules/audio_processing/agc2/rnn_vad/common.h"
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
>index 75f7f17a42fc2968f3c287ee146d26731ae58693..bb747bb03e4c5d89c122dfc824477693d38ffb5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
> 
>+#include <stddef.h>
> #include <array>
> 
> #include "api/array_view.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.h
>index b3a3b9c485c172f6276cb49a2bb54d17a03c8b15..a7d057d576959001a44f7d3b0a7969960974c2d1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.h
>@@ -11,6 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_H_
> 
>+#include <stddef.h>
>+#include <sys/types.h>
> #include <array>
> 
> #include "api/array_view.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc
>index 5fba0bfac877c4dc6dfaf84268706c630bc679bf..b66dfd684d7443c83669f237c561d3ace9f05866 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc
>@@ -25,22 +25,22 @@ namespace rnn_vad {
> namespace test {
> namespace {
> 
>-DEFINE_string(i, "", "Path to the input wav file");
>+WEBRTC_DEFINE_string(i, "", "Path to the input wav file");
> std::string InputWavFile() {
>   return static_cast<std::string>(FLAG_i);
> }
> 
>-DEFINE_string(f, "", "Path to the output features file");
>+WEBRTC_DEFINE_string(f, "", "Path to the output features file");
> std::string OutputFeaturesFile() {
>   return static_cast<std::string>(FLAG_f);
> }
> 
>-DEFINE_string(o, "", "Path to the output VAD probabilities file");
>+WEBRTC_DEFINE_string(o, "", "Path to the output VAD probabilities file");
> std::string OutputVadProbsFile() {
>   return static_cast<std::string>(FLAG_o);
> }
> 
>-DEFINE_bool(help, false, "Prints this message");
>+WEBRTC_DEFINE_bool(help, false, "Prints this message");
> 
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h
>index bedd7ab3d7d1ada5824f176a7b55a9797b3b75b8..5c33dcdd245616b89131d0b7fc36be731aba4f7e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h
>@@ -13,6 +13,7 @@
> 
> #include <array>
> #include <complex>
>+#include <cstddef>
> #include <vector>
> 
> #include "api/array_view.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
>index 9f4e21850f5f76e9c0fb6b71aa28f4f9d1adae12..74211fe814da8871969a3ab0600e5a7552a8c83f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
>@@ -12,6 +12,7 @@
> 
> #include <algorithm>
> #include <cmath>
>+#include <cstddef>
> 
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
>index 45bb382078e65727342d84633abb658d0d8919d8..edfd18cc85063b28c72bfcd6a72c9ce8b63391dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
> 
>+#include <stddef.h>
> #include <array>
> #include <complex>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.cc
>index 0895583b83962ff360ae6013e3a6bbdf66acd7ed..94a52eaacaad867770a475dbc1bb33abf2a86088 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.cc
>@@ -11,6 +11,7 @@
> #include "modules/audio_processing/agc2/saturation_protector.h"
> 
> #include <algorithm>
>+#include <iterator>
> 
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/numerics/safe_minmax.h"
>@@ -56,9 +57,14 @@ float SaturationProtector::PeakEnveloper::Query() const {
> }
> 
> SaturationProtector::SaturationProtector(ApmDataDumper* apm_data_dumper)
>+    : SaturationProtector(apm_data_dumper, GetExtraSaturationMarginOffsetDb()) {
>+}
>+
>+SaturationProtector::SaturationProtector(ApmDataDumper* apm_data_dumper,
>+                                         float extra_saturation_margin_db)
>     : apm_data_dumper_(apm_data_dumper),
>       last_margin_(GetInitialSaturationMarginDb()),
>-      extra_saturation_margin_db_(GetExtraSaturationMarginOffsetDb()) {}
>+      extra_saturation_margin_db_(extra_saturation_margin_db) {}
> 
> void SaturationProtector::UpdateMargin(
>     const VadWithLevel::LevelAndProbability& vad_data,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.h
>index 1705f6ac5fbbbb0a21ff1cee43bce975feb5d682..e637469070eab6032a1f66e5fb5d9059c85ac1f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.h
>@@ -24,6 +24,9 @@ class SaturationProtector {
>  public:
>   explicit SaturationProtector(ApmDataDumper* apm_data_dumper);
> 
>+  SaturationProtector(ApmDataDumper* apm_data_dumper,
>+                      float extra_saturation_margin_db);
>+
>   // Update and return margin estimate. This method should be called
>   // whenever a frame is reliably classified as 'speech'.
>   //
>@@ -60,7 +63,7 @@ class SaturationProtector {
> 
>   float last_margin_;
>   PeakEnveloper peak_enveloper_;
>-  float extra_saturation_margin_db_;
>+  const float extra_saturation_margin_db_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier.cc
>index 0ec34148b9353a884e1c1a208f043ad95fd636af..8778c494265797a4ff88b620e7804b944af53f5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier.cc
>@@ -18,7 +18,7 @@
> #include "modules/audio_processing/agc2/down_sampler.h"
> #include "modules/audio_processing/agc2/noise_spectrum_estimator.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>-#include "rtc_base/constructormagic.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.cc
>index decfacd0c18037608357cb7d153d6dc9ccd81312..52970dfe677c3d5e71efc2fd2e87ab4ec4a35203 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.cc
>@@ -11,10 +11,12 @@
> #include "modules/audio_processing/agc2/vad_with_level.h"
> 
> #include <algorithm>
>+#include <array>
>+#include <cmath>
> 
>+#include "api/array_view.h"
> #include "common_audio/include/audio_util.h"
> #include "modules/audio_processing/agc2/rnn_vad/common.h"
>-#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.h
>index 67a00ced6c1060966991db4b8ec83d59cda0a825..b0ad868d4b97c4a6b5360ece1363aa056be0516b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.h
>@@ -11,7 +11,6 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC2_VAD_WITH_LEVEL_H_
> #define MODULES_AUDIO_PROCESSING_AGC2_VAD_WITH_LEVEL_H_
> 
>-#include "api/array_view.h"
> #include "common_audio/resampler/include/push_resampler.h"
> #include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
> #include "modules/audio_processing/agc2/rnn_vad/rnn.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..f9aee62ba9aab410af1be2d019587fdc2cf5639a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level_unittest.cc
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/vad_with_level.h"
>+
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+TEST(AutomaticGainController2VadWithLevelEstimator,
>+     PeakLevelGreaterThanRmsLevel) {
>+  constexpr size_t kSampleRateHz = 8000;
>+
>+  // 10 ms input frame, constant except for one peak value.
>+  // Handcrafted so that the average is lower than the peak value.
>+  std::array<float, kSampleRateHz / 100> frame;
>+  frame.fill(1000.f);
>+  frame[10] = 2000.f;
>+  float* const channel0 = frame.data();
>+  AudioFrameView<float> frame_view(&channel0, 1, frame.size());
>+
>+  // Compute audio frame levels (the VAD result is ignored).
>+  VadWithLevel vad_with_level;
>+  auto levels_and_vad_prob = vad_with_level.AnalyzeFrame(frame_view);
>+
>+  // Compare peak and RMS levels.
>+  EXPECT_LT(levels_and_vad_prob.speech_rms_dbfs,
>+            levels_and_vad_prob.speech_peak_dbfs);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.cc
>index f163f5a07ded289233fe8faa8ca29dda041cc6a5..0c38a4fe82b854de906f06e675fc694842c9b61b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.cc
>@@ -10,11 +10,13 @@
> 
> #include "modules/audio_processing/audio_buffer.h"
> 
>+#include <string.h>
>+#include <cstdint>
>+
> #include "common_audio/channel_buffer.h"
> #include "common_audio/include/audio_util.h"
> #include "common_audio/resampler/push_sinc_resampler.h"
>-#include "common_audio/signal_processing/include/signal_processing_library.h"
>-#include "modules/audio_processing/common.h"
>+#include "modules/audio_processing/splitting_filter.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.h
>index ade3eec13db2a17377d4cebf074ba772ccfac55a..469646e8db4b68d9fd9748289f45a3f0ddd366d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.h
>@@ -11,18 +11,21 @@
> #ifndef MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
> #define MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> #include <vector>
> 
> #include "api/audio/audio_frame.h"
> #include "common_audio/channel_buffer.h"
> #include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/audio_processing/splitting_filter.h"
>+#include "rtc_base/gtest_prod_util.h"
> 
> namespace webrtc {
> 
>-class PushSincResampler;
> class IFChannelBuffer;
>+class PushSincResampler;
>+class SplittingFilter;
> 
> enum Band { kBand0To8kHz = 0, kBand8To16kHz = 1, kBand16To24kHz = 2 };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.cc
>index b9153fa08b92040a66b88ab2bfaab209c95ecb72..2937c0680bf5ac699b2c28ff968419688381f1eb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.cc
>@@ -10,15 +10,16 @@
> 
> #include "modules/audio_processing/audio_processing_impl.h"
> 
>-#include <math.h>
> #include <algorithm>
>+#include <cstdint>
> #include <string>
>+#include <type_traits>
>+#include <utility>
> 
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
> #include "common_audio/audio_converter.h"
>-#include "common_audio/channel_buffer.h"
> #include "common_audio/include/audio_util.h"
>-#include "common_audio/signal_processing/include/signal_processing_library.h"
>-#include "modules/audio_processing/aec/aec_core.h"
> #include "modules/audio_processing/agc/agc_manager_direct.h"
> #include "modules/audio_processing/agc2/gain_applier.h"
> #include "modules/audio_processing/audio_buffer.h"
>@@ -28,6 +29,7 @@
> #include "modules/audio_processing/gain_control_for_experimental_agc.h"
> #include "modules/audio_processing/gain_control_impl.h"
> #include "modules/audio_processing/gain_controller2.h"
>+#include "modules/audio_processing/include/audio_frame_view.h"
> #include "modules/audio_processing/level_estimator_impl.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "modules/audio_processing/low_cut_filter.h"
>@@ -37,10 +39,9 @@
> #include "modules/audio_processing/voice_detection_impl.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/platform_file.h"
> #include "rtc_base/refcountedobject.h"
>-#include "rtc_base/system/arch.h"
> #include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/metrics.h"
>@@ -115,29 +116,6 @@ static const size_t kMaxAllowedValuesOfSamplesPerFrame = 480;
> // TODO(peah): Decrease this once we properly handle hugely unbalanced
> // reverse and forward call numbers.
> static const size_t kMaxNumFramesToBuffer = 100;
>-
>-class HighPassFilterImpl : public HighPassFilter {
>- public:
>-  explicit HighPassFilterImpl(AudioProcessingImpl* apm) : apm_(apm) {}
>-  ~HighPassFilterImpl() override = default;
>-
>-  // HighPassFilter implementation.
>-  int Enable(bool enable) override {
>-    apm_->MutateConfig([enable](AudioProcessing::Config* config) {
>-      config->high_pass_filter.enabled = enable;
>-    });
>-
>-    return AudioProcessing::kNoError;
>-  }
>-
>-  bool is_enabled() const override {
>-    return apm_->GetConfig().high_pass_filter.enabled;
>-  }
>-
>- private:
>-  AudioProcessingImpl* apm_;
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(HighPassFilterImpl);
>-};
> }  // namespace
> 
> // Throughout webrtc, it's assumed that success is represented by zero.
>@@ -249,8 +227,6 @@ bool AudioProcessingImpl::ApmSubmoduleStates::LowCutFilteringRequired() const {
> struct AudioProcessingImpl::ApmPublicSubmodules {
>   ApmPublicSubmodules() {}
>   // Accessed externally of APM without any lock acquired.
>-  std::unique_ptr<EchoCancellationImpl> echo_cancellation;
>-  std::unique_ptr<EchoControlMobileImpl> echo_control_mobile;
>   std::unique_ptr<GainControlImpl> gain_control;
>   std::unique_ptr<LevelEstimatorImpl> level_estimator;
>   std::unique_ptr<NoiseSuppressionImpl> noise_suppression;
>@@ -276,11 +252,14 @@ struct AudioProcessingImpl::ApmPrivateSubmodules {
>   std::unique_ptr<GainController2> gain_controller2;
>   std::unique_ptr<LowCutFilter> low_cut_filter;
>   rtc::scoped_refptr<EchoDetector> echo_detector;
>+  std::unique_ptr<EchoCancellationImpl> echo_cancellation;
>   std::unique_ptr<EchoControl> echo_controller;
>+  std::unique_ptr<EchoControlMobileImpl> echo_control_mobile;
>   std::unique_ptr<CustomProcessing> capture_post_processor;
>   std::unique_ptr<CustomProcessing> render_pre_processor;
>   std::unique_ptr<GainApplier> pre_amplifier;
>   std::unique_ptr<CustomAudioAnalyzer> capture_analyzer;
>+  std::unique_ptr<LevelEstimatorImpl> output_level_estimator;
> };
> 
> AudioProcessingBuilder::AudioProcessingBuilder() = default;
>@@ -352,7 +331,6 @@ AudioProcessingImpl::AudioProcessingImpl(
>       render_runtime_settings_(kRuntimeSettingQueueSize),
>       capture_runtime_settings_enqueuer_(&capture_runtime_settings_),
>       render_runtime_settings_enqueuer_(&render_runtime_settings_),
>-      high_pass_filter_impl_(new HighPassFilterImpl(this)),
>       echo_control_factory_(std::move(echo_control_factory)),
>       submodule_states_(!!capture_post_processor,
>                         !!render_pre_processor,
>@@ -390,10 +368,6 @@ AudioProcessingImpl::AudioProcessingImpl(
>     capture_nonlocked_.echo_controller_enabled =
>         static_cast<bool>(echo_control_factory_);
> 
>-    public_submodules_->echo_cancellation.reset(
>-        new EchoCancellationImpl(&crit_render_, &crit_capture_));
>-    public_submodules_->echo_control_mobile.reset(
>-        new EchoControlMobileImpl(&crit_render_, &crit_capture_));
>     public_submodules_->gain_control.reset(
>         new GainControlImpl(&crit_render_, &crit_capture_));
>     public_submodules_->level_estimator.reset(
>@@ -412,6 +386,8 @@ AudioProcessingImpl::AudioProcessingImpl(
>           new rtc::RefCountedObject<ResidualEchoDetector>();
>     }
> 
>+    private_submodules_->echo_cancellation.reset(new EchoCancellationImpl());
>+    private_submodules_->echo_control_mobile.reset(new EchoControlMobileImpl());
>     // TODO(alessiob): Move the injected gain controller once injection is
>     // implemented.
>     private_submodules_->gain_controller2.reset(new GainController2());
>@@ -530,16 +506,16 @@ int AudioProcessingImpl::InitializeLocked() {
>                       formats_.api_format.output_stream().num_channels(),
>                       formats_.api_format.output_stream().num_frames()));
> 
>-  public_submodules_->echo_cancellation->Initialize(
>+  private_submodules_->echo_cancellation->Initialize(
>       proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(),
>       num_proc_channels());
>   AllocateRenderQueue();
> 
>-  int success = public_submodules_->echo_cancellation->enable_metrics(true);
>+  int success = private_submodules_->echo_cancellation->enable_metrics(true);
>   RTC_DCHECK_EQ(0, success);
>-  success = public_submodules_->echo_cancellation->enable_delay_logging(true);
>+  success = private_submodules_->echo_cancellation->enable_delay_logging(true);
>   RTC_DCHECK_EQ(0, success);
>-  public_submodules_->echo_control_mobile->Initialize(
>+  private_submodules_->echo_control_mobile->Initialize(
>       proc_split_sample_rate_hz(), num_reverse_channels(),
>       num_output_channels());
> 
>@@ -662,18 +638,18 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
> }
> 
> void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
>-  config_ = config;
>-
>   // Run in a single-threaded manner when applying the settings.
>   rtc::CritScope cs_render(&crit_render_);
>   rtc::CritScope cs_capture(&crit_capture_);
> 
>-  public_submodules_->echo_cancellation->Enable(
>+  config_ = config;
>+
>+  private_submodules_->echo_cancellation->Enable(
>       config_.echo_canceller.enabled && !config_.echo_canceller.mobile_mode);
>-  public_submodules_->echo_control_mobile->Enable(
>+  private_submodules_->echo_control_mobile->Enable(
>       config_.echo_canceller.enabled && config_.echo_canceller.mobile_mode);
> 
>-  public_submodules_->echo_cancellation->set_suppression_level(
>+  private_submodules_->echo_cancellation->set_suppression_level(
>       config.echo_canceller.legacy_moderate_suppression_level
>           ? EchoCancellationImpl::SuppressionLevel::kModerateSuppression
>           : EchoCancellationImpl::SuppressionLevel::kHighSuppression);
>@@ -698,6 +674,13 @@ void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
>                    << config_.gain_controller2.enabled;
>   RTC_LOG(LS_INFO) << "Pre-amplifier activated: "
>                    << config_.pre_amplifier.enabled;
>+
>+  if (config_.level_estimation.enabled &&
>+      !private_submodules_->output_level_estimator) {
>+    private_submodules_->output_level_estimator.reset(
>+        new LevelEstimatorImpl(&crit_capture_));
>+    private_submodules_->output_level_estimator->Enable(true);
>+  }
> }
> 
> void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {
>@@ -705,7 +688,7 @@ void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {
>   rtc::CritScope cs_render(&crit_render_);
>   rtc::CritScope cs_capture(&crit_capture_);
> 
>-  public_submodules_->echo_cancellation->SetExtraOptions(config);
>+  private_submodules_->echo_cancellation->SetExtraOptions(config);
> 
>   if (capture_.transient_suppressor_enabled !=
>       config.Get<ExperimentalNs>().enabled) {
>@@ -1082,12 +1065,12 @@ void AudioProcessingImpl::AllocateRenderQueue() {
> void AudioProcessingImpl::EmptyQueuedRenderAudio() {
>   rtc::CritScope cs_capture(&crit_capture_);
>   while (aec_render_signal_queue_->Remove(&aec_capture_queue_buffer_)) {
>-    public_submodules_->echo_cancellation->ProcessRenderAudio(
>+    private_submodules_->echo_cancellation->ProcessRenderAudio(
>         aec_capture_queue_buffer_);
>   }
> 
>   while (aecm_render_signal_queue_->Remove(&aecm_capture_queue_buffer_)) {
>-    public_submodules_->echo_control_mobile->ProcessRenderAudio(
>+    private_submodules_->echo_control_mobile->ProcessRenderAudio(
>         aecm_capture_queue_buffer_);
>   }
> 
>@@ -1109,9 +1092,6 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
>     // Acquire the capture lock in order to safely call the function
>     // that retrieves the render side data. This function accesses APM
>     // getters that need the capture lock held when being called.
>-    // The lock needs to be released as
>-    // public_submodules_->echo_control_mobile->is_enabled() acquires this lock
>-    // as well.
>     rtc::CritScope cs_capture(&crit_capture_);
>     EmptyQueuedRenderAudio();
>   }
>@@ -1180,8 +1160,8 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
>   // Ensure that not both the AEC and AECM are active at the same time.
>   // TODO(peah): Simplify once the public API Enable functions for these
>   // are moved to APM.
>-  RTC_DCHECK(!(public_submodules_->echo_cancellation->is_enabled() &&
>-               public_submodules_->echo_control_mobile->is_enabled()));
>+  RTC_DCHECK(!(private_submodules_->echo_cancellation->is_enabled() &&
>+               private_submodules_->echo_control_mobile->is_enabled()));
> 
>   MaybeUpdateHistograms();
> 
>@@ -1265,7 +1245,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
> 
>   // Ensure that the stream delay was set before the call to the
>   // AEC ProcessCaptureAudio function.
>-  if (public_submodules_->echo_cancellation->is_enabled() &&
>+  if (private_submodules_->echo_cancellation->is_enabled() &&
>       !private_submodules_->echo_controller && !was_stream_delay_set()) {
>     return AudioProcessing::kStreamParameterNotSetError;
>   }
>@@ -1281,11 +1261,11 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
>     private_submodules_->echo_controller->ProcessCapture(
>         capture_buffer, capture_.echo_path_gain_change);
>   } else {
>-    RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(
>+    RETURN_ON_ERR(private_submodules_->echo_cancellation->ProcessCaptureAudio(
>         capture_buffer, stream_delay_ms()));
>   }
> 
>-  if (public_submodules_->echo_control_mobile->is_enabled() &&
>+  if (private_submodules_->echo_control_mobile->is_enabled() &&
>       public_submodules_->noise_suppression->is_enabled()) {
>     capture_buffer->CopyLowPassToReference();
>   }
>@@ -1293,14 +1273,14 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
> 
>   // Ensure that the stream delay was set before the call to the
>   // AECM ProcessCaptureAudio function.
>-  if (public_submodules_->echo_control_mobile->is_enabled() &&
>+  if (private_submodules_->echo_control_mobile->is_enabled() &&
>       !was_stream_delay_set()) {
>     return AudioProcessing::kStreamParameterNotSetError;
>   }
> 
>   if (!(private_submodules_->echo_controller ||
>-        public_submodules_->echo_cancellation->is_enabled())) {
>-    RETURN_ON_ERR(public_submodules_->echo_control_mobile->ProcessCaptureAudio(
>+        private_submodules_->echo_cancellation->is_enabled())) {
>+    RETURN_ON_ERR(private_submodules_->echo_control_mobile->ProcessCaptureAudio(
>         capture_buffer, stream_delay_ms()));
>   }
> 
>@@ -1315,7 +1295,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
>   }
>   RETURN_ON_ERR(public_submodules_->gain_control->ProcessCaptureAudio(
>       capture_buffer,
>-      public_submodules_->echo_cancellation->stream_has_echo()));
>+      private_submodules_->echo_cancellation->stream_has_echo()));
> 
>   if (submodule_states_.CaptureMultiBandProcessingActive() &&
>       SampleRateSupportsMultiBand(
>@@ -1364,6 +1344,13 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
> 
>   // The level estimator operates on the recombined data.
>   public_submodules_->level_estimator->ProcessStream(capture_buffer);
>+  if (config_.level_estimation.enabled) {
>+    private_submodules_->output_level_estimator->ProcessStream(capture_buffer);
>+    capture_.stats.output_rms_dbfs =
>+        private_submodules_->output_level_estimator->RMS();
>+  } else {
>+    capture_.stats.output_rms_dbfs = absl::nullopt;
>+  }
> 
>   capture_output_rms_.Analyze(rtc::ArrayView<const int16_t>(
>       capture_buffer->channels_const()[0],
>@@ -1613,112 +1600,52 @@ void AudioProcessingImpl::DetachPlayoutAudioGenerator() {
>   // Delete audio generator, if one is attached.
> }
> 
>-AudioProcessing::AudioProcessingStatistics::AudioProcessingStatistics() {
>-  residual_echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);
>-  echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);
>-  echo_return_loss_enhancement.Set(-100.0f, -100.0f, -100.0f, -100.0f);
>-  a_nlp.Set(-100.0f, -100.0f, -100.0f, -100.0f);
>-}
>-
>-AudioProcessing::AudioProcessingStatistics::AudioProcessingStatistics(
>-    const AudioProcessingStatistics& other) = default;
>-
>-AudioProcessing::AudioProcessingStatistics::~AudioProcessingStatistics() =
>-    default;
>-
>-// TODO(ivoc): Remove this when GetStatistics() becomes pure virtual.
>-AudioProcessing::AudioProcessingStatistics AudioProcessing::GetStatistics()
>-    const {
>-  return AudioProcessingStatistics();
>-}
>-
>-// TODO(ivoc): Remove this when GetStatistics() becomes pure virtual.
>-AudioProcessingStats AudioProcessing::GetStatistics(
>+AudioProcessingStats AudioProcessingImpl::GetStatistics(
>     bool has_remote_tracks) const {
>-  return AudioProcessingStats();
>-}
>-
>-AudioProcessing::AudioProcessingStatistics AudioProcessingImpl::GetStatistics()
>-    const {
>-  AudioProcessingStatistics stats;
>+  rtc::CritScope cs_capture(&crit_capture_);
>+  if (!has_remote_tracks) {
>+    return capture_.stats;
>+  }
>+  AudioProcessingStats stats = capture_.stats;
>   EchoCancellationImpl::Metrics metrics;
>   if (private_submodules_->echo_controller) {
>-    rtc::CritScope cs_capture(&crit_capture_);
>     auto ec_metrics = private_submodules_->echo_controller->GetMetrics();
>-    float erl = static_cast<float>(ec_metrics.echo_return_loss);
>-    float erle = static_cast<float>(ec_metrics.echo_return_loss_enhancement);
>-    // Instant value will also be used for min, max and average.
>-    stats.echo_return_loss.Set(erl, erl, erl, erl);
>-    stats.echo_return_loss_enhancement.Set(erle, erle, erle, erle);
>-  } else if (public_submodules_->echo_cancellation->GetMetrics(&metrics) ==
>+    stats.echo_return_loss = ec_metrics.echo_return_loss;
>+    stats.echo_return_loss_enhancement =
>+        ec_metrics.echo_return_loss_enhancement;
>+    stats.delay_ms = ec_metrics.delay_ms;
>+  } else if (private_submodules_->echo_cancellation->GetMetrics(&metrics) ==
>              Error::kNoError) {
>-    stats.a_nlp.Set(metrics.a_nlp);
>-    stats.divergent_filter_fraction = metrics.divergent_filter_fraction;
>-    stats.echo_return_loss.Set(metrics.echo_return_loss);
>-    stats.echo_return_loss_enhancement.Set(
>-        metrics.echo_return_loss_enhancement);
>-    stats.residual_echo_return_loss.Set(metrics.residual_echo_return_loss);
>+    if (metrics.divergent_filter_fraction != -1.0f) {
>+      stats.divergent_filter_fraction =
>+          absl::optional<double>(metrics.divergent_filter_fraction);
>+    }
>+    if (metrics.echo_return_loss.instant != -100) {
>+      stats.echo_return_loss =
>+          absl::optional<double>(metrics.echo_return_loss.instant);
>+    }
>+    if (metrics.echo_return_loss_enhancement.instant != -100) {
>+      stats.echo_return_loss_enhancement =
>+          absl::optional<double>(metrics.echo_return_loss_enhancement.instant);
>+    }
>   }
>-  {
>-    rtc::CritScope cs_capture(&crit_capture_);
>+  if (config_.residual_echo_detector.enabled) {
>     RTC_DCHECK(private_submodules_->echo_detector);
>     auto ed_metrics = private_submodules_->echo_detector->GetMetrics();
>     stats.residual_echo_likelihood = ed_metrics.echo_likelihood;
>     stats.residual_echo_likelihood_recent_max =
>         ed_metrics.echo_likelihood_recent_max;
>   }
>-  public_submodules_->echo_cancellation->GetDelayMetrics(
>-      &stats.delay_median, &stats.delay_standard_deviation,
>-      &stats.fraction_poor_delays);
>-  return stats;
>-}
>-
>-AudioProcessingStats AudioProcessingImpl::GetStatistics(
>-    bool has_remote_tracks) const {
>-  AudioProcessingStats stats;
>-  if (has_remote_tracks) {
>-    EchoCancellationImpl::Metrics metrics;
>-    if (private_submodules_->echo_controller) {
>-      rtc::CritScope cs_capture(&crit_capture_);
>-      auto ec_metrics = private_submodules_->echo_controller->GetMetrics();
>-      stats.echo_return_loss = ec_metrics.echo_return_loss;
>-      stats.echo_return_loss_enhancement =
>-          ec_metrics.echo_return_loss_enhancement;
>-      stats.delay_ms = ec_metrics.delay_ms;
>-    } else if (public_submodules_->echo_cancellation->GetMetrics(&metrics) ==
>-               Error::kNoError) {
>-      if (metrics.divergent_filter_fraction != -1.0f) {
>-        stats.divergent_filter_fraction =
>-            absl::optional<double>(metrics.divergent_filter_fraction);
>-      }
>-      if (metrics.echo_return_loss.instant != -100) {
>-        stats.echo_return_loss =
>-            absl::optional<double>(metrics.echo_return_loss.instant);
>-      }
>-      if (metrics.echo_return_loss_enhancement.instant != -100) {
>-        stats.echo_return_loss_enhancement = absl::optional<double>(
>-            metrics.echo_return_loss_enhancement.instant);
>-      }
>+  int delay_median, delay_std;
>+  float fraction_poor_delays;
>+  if (private_submodules_->echo_cancellation->GetDelayMetrics(
>+          &delay_median, &delay_std, &fraction_poor_delays) ==
>+      Error::kNoError) {
>+    if (delay_median >= 0) {
>+      stats.delay_median_ms = absl::optional<int32_t>(delay_median);
>     }
>-    if (config_.residual_echo_detector.enabled) {
>-      rtc::CritScope cs_capture(&crit_capture_);
>-      RTC_DCHECK(private_submodules_->echo_detector);
>-      auto ed_metrics = private_submodules_->echo_detector->GetMetrics();
>-      stats.residual_echo_likelihood = ed_metrics.echo_likelihood;
>-      stats.residual_echo_likelihood_recent_max =
>-          ed_metrics.echo_likelihood_recent_max;
>-    }
>-    int delay_median, delay_std;
>-    float fraction_poor_delays;
>-    if (public_submodules_->echo_cancellation->GetDelayMetrics(
>-            &delay_median, &delay_std, &fraction_poor_delays) ==
>-        Error::kNoError) {
>-      if (delay_median >= 0) {
>-        stats.delay_median_ms = absl::optional<int32_t>(delay_median);
>-      }
>-      if (delay_std >= 0) {
>-        stats.delay_standard_deviation_ms = absl::optional<int32_t>(delay_std);
>-      }
>+    if (delay_std >= 0) {
>+      stats.delay_standard_deviation_ms = absl::optional<int32_t>(delay_std);
>     }
>   }
>   return stats;
>@@ -1731,10 +1658,6 @@ GainControl* AudioProcessingImpl::gain_control() const {
>   return public_submodules_->gain_control.get();
> }
> 
>-HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
>-  return high_pass_filter_impl_.get();
>-}
>-
> LevelEstimator* AudioProcessingImpl::level_estimator() const {
>   return public_submodules_->level_estimator.get();
> }
>@@ -1764,8 +1687,8 @@ AudioProcessing::Config AudioProcessingImpl::GetConfig() const {
> bool AudioProcessingImpl::UpdateActiveSubmoduleStates() {
>   return submodule_states_.Update(
>       config_.high_pass_filter.enabled,
>-      public_submodules_->echo_cancellation->is_enabled(),
>-      public_submodules_->echo_control_mobile->is_enabled(),
>+      private_submodules_->echo_cancellation->is_enabled(),
>+      private_submodules_->echo_control_mobile->is_enabled(),
>       config_.residual_echo_detector.enabled,
>       public_submodules_->noise_suppression->is_enabled(),
>       public_submodules_->gain_control->is_enabled(),
>@@ -1852,15 +1775,15 @@ void AudioProcessingImpl::InitializePreProcessor() {
> void AudioProcessingImpl::MaybeUpdateHistograms() {
>   static const int kMinDiffDelayMs = 60;
> 
>-  if (public_submodules_->echo_cancellation->is_enabled()) {
>+  if (private_submodules_->echo_cancellation->is_enabled()) {
>     // Activate delay_jumps_ counters if we know echo_cancellation is running.
>     // If a stream has echo we know that the echo_cancellation is in process.
>     if (capture_.stream_delay_jumps == -1 &&
>-        public_submodules_->echo_cancellation->stream_has_echo()) {
>+        private_submodules_->echo_cancellation->stream_has_echo()) {
>       capture_.stream_delay_jumps = 0;
>     }
>     if (capture_.aec_system_delay_jumps == -1 &&
>-        public_submodules_->echo_cancellation->stream_has_echo()) {
>+        private_submodules_->echo_cancellation->stream_has_echo()) {
>       capture_.aec_system_delay_jumps = 0;
>     }
> 
>@@ -1883,7 +1806,7 @@ void AudioProcessingImpl::MaybeUpdateHistograms() {
>         rtc::CheckedDivExact(capture_nonlocked_.split_rate, 1000);
>     RTC_DCHECK_LT(0, samples_per_ms);
>     const int aec_system_delay_ms =
>-        public_submodules_->echo_cancellation->GetSystemDelayInSamples() /
>+        private_submodules_->echo_cancellation->GetSystemDelayInSamples() /
>         samples_per_ms;
>     const int diff_aec_system_delay_ms =
>         aec_system_delay_ms - capture_.last_aec_system_delay_ms;
>@@ -1927,7 +1850,7 @@ void AudioProcessingImpl::WriteAecDumpConfigMessage(bool forced) {
>     return;
>   }
>   std::string experiments_description =
>-      public_submodules_->echo_cancellation->GetExperimentsDescription();
>+      private_submodules_->echo_cancellation->GetExperimentsDescription();
>   // TODO(peah): Add semicolon-separated concatenations of experiment
>   // descriptions for other submodules.
>   if (constants_.agc_clipped_level_min != kClippedLevelMin) {
>@@ -1942,22 +1865,22 @@ void AudioProcessingImpl::WriteAecDumpConfigMessage(bool forced) {
> 
>   InternalAPMConfig apm_config;
> 
>-  apm_config.aec_enabled = public_submodules_->echo_cancellation->is_enabled();
>+  apm_config.aec_enabled = private_submodules_->echo_cancellation->is_enabled();
>   apm_config.aec_delay_agnostic_enabled =
>-      public_submodules_->echo_cancellation->is_delay_agnostic_enabled();
>+      private_submodules_->echo_cancellation->is_delay_agnostic_enabled();
>   apm_config.aec_drift_compensation_enabled =
>-      public_submodules_->echo_cancellation->is_drift_compensation_enabled();
>+      private_submodules_->echo_cancellation->is_drift_compensation_enabled();
>   apm_config.aec_extended_filter_enabled =
>-      public_submodules_->echo_cancellation->is_extended_filter_enabled();
>+      private_submodules_->echo_cancellation->is_extended_filter_enabled();
>   apm_config.aec_suppression_level = static_cast<int>(
>-      public_submodules_->echo_cancellation->suppression_level());
>+      private_submodules_->echo_cancellation->suppression_level());
> 
>   apm_config.aecm_enabled =
>-      public_submodules_->echo_control_mobile->is_enabled();
>+      private_submodules_->echo_control_mobile->is_enabled();
>   apm_config.aecm_comfort_noise_enabled =
>-      public_submodules_->echo_control_mobile->is_comfort_noise_enabled();
>-  apm_config.aecm_routing_mode =
>-      static_cast<int>(public_submodules_->echo_control_mobile->routing_mode());
>+      private_submodules_->echo_control_mobile->is_comfort_noise_enabled();
>+  apm_config.aecm_routing_mode = static_cast<int>(
>+      private_submodules_->echo_control_mobile->routing_mode());
> 
>   apm_config.agc_enabled = public_submodules_->gain_control->is_enabled();
>   apm_config.agc_mode =
>@@ -2032,7 +1955,7 @@ void AudioProcessingImpl::RecordAudioProcessingState() {
>   AecDump::AudioProcessingState audio_proc_state;
>   audio_proc_state.delay = capture_nonlocked_.stream_delay_ms;
>   audio_proc_state.drift =
>-      public_submodules_->echo_cancellation->stream_drift_samples();
>+      private_submodules_->echo_cancellation->stream_drift_samples();
>   audio_proc_state.level = gain_control()->stream_analog_level();
>   audio_proc_state.keypress = capture_.key_pressed;
>   aec_dump_->AddAudioProcessingState(audio_proc_state);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.h
>index 50cb82bf13ae1752df518e44197ac0eb9fb7a7b8..2f946c5e13d89204df74621beb42c48048d2b8e8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.h
>@@ -18,6 +18,7 @@
> #include "modules/audio_processing/audio_buffer.h"
> #include "modules/audio_processing/include/aec_dump.h"
> #include "modules/audio_processing/include/audio_processing.h"
>+#include "modules/audio_processing/include/audio_processing_statistics.h"
> #include "modules/audio_processing/render_queue_item_verifier.h"
> #include "modules/audio_processing/rms_level.h"
> #include "rtc_base/criticalsection.h"
>@@ -109,7 +110,6 @@ class AudioProcessingImpl : public AudioProcessing {
>   bool was_stream_delay_set() const override
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
> 
>-  AudioProcessingStatistics GetStatistics() const override;
>   AudioProcessingStats GetStatistics(bool has_remote_tracks) const override;
> 
>   // Methods returning pointers to APM submodules.
>@@ -118,8 +118,6 @@ class AudioProcessingImpl : public AudioProcessing {
>   // created only once in a single-treaded manner
>   // during APM creation).
>   GainControl* gain_control() const override;
>-  // TODO(peah): Deprecate this API call.
>-  HighPassFilter* high_pass_filter() const override;
>   LevelEstimator* level_estimator() const override;
>   NoiseSuppression* noise_suppression() const override;
>   VoiceDetection* voice_detection() const override;
>@@ -164,9 +162,6 @@ class AudioProcessingImpl : public AudioProcessing {
>   RuntimeSettingEnqueuer capture_runtime_settings_enqueuer_;
>   RuntimeSettingEnqueuer render_runtime_settings_enqueuer_;
> 
>-  // Submodule interface implementations.
>-  std::unique_ptr<HighPassFilter> high_pass_filter_impl_;
>-
>   // EchoControl factory.
>   std::unique_ptr<EchoControlFactory> echo_control_factory_;
> 
>@@ -396,6 +391,7 @@ class AudioProcessingImpl : public AudioProcessing {
>     bool echo_path_gain_change;
>     int prev_analog_mic_level;
>     float prev_pre_amp_gain;
>+    AudioProcessingStats stats;
>   } capture_ RTC_GUARDED_BY(crit_capture_);
> 
>   struct ApmCaptureNonLockedState {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
>index 0e0a33f5700693f134274d7737afa3c5440891b7..9685ef95e9cf571ad60c2c8001f73e81f4359157 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
>@@ -487,10 +487,7 @@ void PopulateAudioFrame(AudioFrame* frame,
> }
> 
> AudioProcessingImplLockTest::AudioProcessingImplLockTest()
>-    : test_complete_(false, false),
>-      render_call_event_(false, false),
>-      capture_call_event_(false, false),
>-      render_thread_(RenderProcessorThreadFunc, this, "render"),
>+    : render_thread_(RenderProcessorThreadFunc, this, "render"),
>       capture_thread_(CaptureProcessorThreadFunc, this, "capture"),
>       stats_thread_(StatsProcessorThreadFunc, this, "stats"),
>       apm_(AudioProcessingBuilder().Create()),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
>index 95f736c22f168bfdb87267839d7b042ea3fe9916..84cb5743072706b01f88cc21418e6e3e17cc9d80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
>@@ -18,11 +18,11 @@
> #include "api/array_view.h"
> #include "modules/audio_processing/test/test_utils.h"
> #include "rtc_base/atomicops.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/platform_thread.h"
> #include "rtc_base/random.h"
> #include "system_wrappers/include/clock.h"
>-#include "system_wrappers/include/event_wrapper.h"
> #include "test/gtest.h"
> #include "test/testsupport/perf_test.h"
> 
>@@ -391,8 +391,7 @@ class TimedThreadApiProcessor {
> class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
>  public:
>   CallSimulator()
>-      : test_complete_(EventWrapper::Create()),
>-        render_thread_(
>+      : render_thread_(
>             new rtc::PlatformThread(RenderProcessorThreadFunc, this, "render")),
>         capture_thread_(new rtc::PlatformThread(CaptureProcessorThreadFunc,
>                                                 this,
>@@ -401,10 +400,10 @@ class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
>         simulation_config_(static_cast<SimulationConfig>(GetParam())) {}
> 
>   // Run the call simulation with a timeout.
>-  EventTypeWrapper Run() {
>+  bool Run() {
>     StartThreads();
> 
>-    EventTypeWrapper result = test_complete_->Wait(kTestTimeout);
>+    bool result = test_complete_.Wait(kTestTimeout);
> 
>     StopThreads();
> 
>@@ -420,7 +419,7 @@ class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
>   // done.
>   bool MaybeEndTest() {
>     if (frame_counters_.BothCountersExceedeThreshold(kMinNumFramesToProcess)) {
>-      test_complete_->Set();
>+      test_complete_.Set();
>       return true;
>     }
>     return false;
>@@ -570,7 +569,7 @@ class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
>   }
> 
>   // Event handler for the test.
>-  const std::unique_ptr<EventWrapper> test_complete_;
>+  rtc::Event test_complete_;
> 
>   // Thread related variables.
>   std::unique_ptr<rtc::PlatformThread> render_thread_;
>@@ -619,7 +618,7 @@ const float CallSimulator::kCaptureInputFloatLevel = 0.03125f;
> // TODO(peah): Reactivate once issue 7712 has been resolved.
> TEST_P(CallSimulator, DISABLED_ApiCallDurationTest) {
>   // Run test and verify that it did not time out.
>-  EXPECT_EQ(kEventSignaled, Run());
>+  EXPECT_TRUE(Run());
> }
> 
> INSTANTIATE_TEST_CASE_P(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_unittest.cc
>index 3f0decc957a634214f3d393f42c4518819b79b81..6809ab9dcd9e3140e02906e47c48a2410ad14c2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_unittest.cc
>@@ -1278,7 +1278,6 @@ TEST_F(ApmTest, AllProcessingDisabledByDefault) {
>   EXPECT_FALSE(config.echo_canceller.enabled);
>   EXPECT_FALSE(config.high_pass_filter.enabled);
>   EXPECT_FALSE(apm_->gain_control()->is_enabled());
>-  EXPECT_FALSE(apm_->high_pass_filter()->is_enabled());
>   EXPECT_FALSE(apm_->level_estimator()->is_enabled());
>   EXPECT_FALSE(apm_->noise_suppression()->is_enabled());
>   EXPECT_FALSE(apm_->voice_detection()->is_enabled());
>@@ -2802,4 +2801,42 @@ TEST(MAYBE_ApmStatistics, AECMEnabledTest) {
>   EXPECT_FALSE(stats.delay_median_ms);
>   EXPECT_FALSE(stats.delay_standard_deviation_ms);
> }
>+
>+TEST(ApmStatistics, ReportOutputRmsDbfs) {
>+  ProcessingConfig processing_config = {
>+      {{32000, 1}, {32000, 1}, {32000, 1}, {32000, 1}}};
>+  AudioProcessing::Config config;
>+
>+  // Set up an audioframe.
>+  AudioFrame frame;
>+  frame.num_channels_ = 1;
>+  SetFrameSampleRate(&frame, AudioProcessing::NativeRate::kSampleRate48kHz);
>+
>+  // Fill the audio frame with a sawtooth pattern.
>+  int16_t* ptr = frame.mutable_data();
>+  for (size_t i = 0; i < frame.kMaxDataSizeSamples; i++) {
>+    ptr[i] = 10000 * ((i % 3) - 1);
>+  }
>+
>+  std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().Create());
>+  apm->Initialize(processing_config);
>+
>+  // If not enabled, no metric should be reported.
>+  EXPECT_EQ(apm->ProcessStream(&frame), 0);
>+  EXPECT_FALSE(apm->GetStatistics(false).output_rms_dbfs);
>+
>+  // If enabled, metrics should be reported.
>+  config.level_estimation.enabled = true;
>+  apm->ApplyConfig(config);
>+  EXPECT_EQ(apm->ProcessStream(&frame), 0);
>+  auto stats = apm->GetStatistics(false);
>+  EXPECT_TRUE(stats.output_rms_dbfs);
>+  EXPECT_GE(*stats.output_rms_dbfs, 0);
>+
>+  // If re-disabled, the value is again not reported.
>+  config.level_estimation.enabled = false;
>+  apm->ApplyConfig(config);
>+  EXPECT_EQ(apm->ProcessStream(&frame), 0);
>+  EXPECT_FALSE(apm->GetStatistics(false).output_rms_dbfs);
>+}
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_bit_exact_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_bit_exact_unittest.cc
>index 7f60c4b5d83881d46e7fa2a63d18cf01a68b4ecc..af0741ee178496a1319e9ca4b531735d5b72b707 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_bit_exact_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_bit_exact_unittest.cc
>@@ -74,9 +74,7 @@ void RunBitexactnessTest(
>     EchoCancellationImpl::SuppressionLevel suppression_level,
>     bool stream_has_echo_reference,
>     const rtc::ArrayView<const float>& output_reference) {
>-  rtc::CriticalSection crit_render;
>-  rtc::CriticalSection crit_capture;
>-  EchoCancellationImpl echo_canceller(&crit_render, &crit_capture);
>+  EchoCancellationImpl echo_canceller;
>   SetupComponent(sample_rate_hz, suppression_level, drift_compensation_enabled,
>                  &echo_canceller);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.cc
>index e6fe7691c4903d9279ab01be53796ffc724697e2..73fe51b752f66d2c0f8a251c29ee6af14aa34286 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.cc
>@@ -10,11 +10,13 @@
> 
> #include "modules/audio_processing/echo_cancellation_impl.h"
> 
>+#include <stdint.h>
> #include <string.h>
> 
> #include "modules/audio_processing/aec/aec_core.h"
> #include "modules/audio_processing/aec/echo_cancellation.h"
> #include "modules/audio_processing/audio_buffer.h"
>+#include "modules/audio_processing/include/config.h"
> #include "rtc_base/checks.h"
> #include "system_wrappers/include/field_trial.h"
> 
>@@ -103,11 +105,8 @@ class EchoCancellationImpl::Canceller {
>   void* state_;
> };
> 
>-EchoCancellationImpl::EchoCancellationImpl(rtc::CriticalSection* crit_render,
>-                                           rtc::CriticalSection* crit_capture)
>-    : crit_render_(crit_render),
>-      crit_capture_(crit_capture),
>-      drift_compensation_enabled_(false),
>+EchoCancellationImpl::EchoCancellationImpl()
>+    : drift_compensation_enabled_(false),
>       metrics_enabled_(true),
>       suppression_level_(kHighSuppression),
>       stream_drift_samples_(0),
>@@ -116,16 +115,12 @@ EchoCancellationImpl::EchoCancellationImpl(rtc::CriticalSection* crit_render,
>       delay_logging_enabled_(true),
>       extended_filter_enabled_(false),
>       delay_agnostic_enabled_(false),
>-      enforce_zero_stream_delay_(EnforceZeroStreamDelay()) {
>-  RTC_DCHECK(crit_render);
>-  RTC_DCHECK(crit_capture);
>-}
>+      enforce_zero_stream_delay_(EnforceZeroStreamDelay()) {}
> 
> EchoCancellationImpl::~EchoCancellationImpl() = default;
> 
> void EchoCancellationImpl::ProcessRenderAudio(
>     rtc::ArrayView<const float> packed_render_audio) {
>-  rtc::CritScope cs_capture(crit_capture_);
>   if (!enabled_) {
>     return;
>   }
>@@ -149,7 +144,6 @@ void EchoCancellationImpl::ProcessRenderAudio(
> 
> int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio,
>                                               int stream_delay_ms) {
>-  rtc::CritScope cs_capture(crit_capture_);
>   if (!enabled_) {
>     return AudioProcessing::kNoError;
>   }
>@@ -206,10 +200,6 @@ int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio,
> }
> 
> int EchoCancellationImpl::Enable(bool enable) {
>-  // Run in a single-threaded manner.
>-  rtc::CritScope cs_render(crit_render_);
>-  rtc::CritScope cs_capture(crit_capture_);
>-
>   if (enable && !enabled_) {
>     enabled_ = enable;  // Must be set before Initialize() is called.
> 
>@@ -227,68 +217,52 @@ int EchoCancellationImpl::Enable(bool enable) {
> }
> 
> bool EchoCancellationImpl::is_enabled() const {
>-  rtc::CritScope cs(crit_capture_);
>   return enabled_;
> }
> 
> int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
>-  {
>-    if (MapSetting(level) == -1) {
>-      return AudioProcessing::kBadParameterError;
>-    }
>-    rtc::CritScope cs(crit_capture_);
>-    suppression_level_ = level;
>+  if (MapSetting(level) == -1) {
>+    return AudioProcessing::kBadParameterError;
>   }
>+  suppression_level_ = level;
>   return Configure();
> }
> 
> EchoCancellationImpl::SuppressionLevel EchoCancellationImpl::suppression_level()
>     const {
>-  rtc::CritScope cs(crit_capture_);
>   return suppression_level_;
> }
> 
> int EchoCancellationImpl::enable_drift_compensation(bool enable) {
>-  {
>-    rtc::CritScope cs(crit_capture_);
>-    drift_compensation_enabled_ = enable;
>-  }
>+  drift_compensation_enabled_ = enable;
>   return Configure();
> }
> 
> bool EchoCancellationImpl::is_drift_compensation_enabled() const {
>-  rtc::CritScope cs(crit_capture_);
>   return drift_compensation_enabled_;
> }
> 
> void EchoCancellationImpl::set_stream_drift_samples(int drift) {
>-  rtc::CritScope cs(crit_capture_);
>   was_stream_drift_set_ = true;
>   stream_drift_samples_ = drift;
> }
> 
> int EchoCancellationImpl::stream_drift_samples() const {
>-  rtc::CritScope cs(crit_capture_);
>   return stream_drift_samples_;
> }
> 
> int EchoCancellationImpl::enable_metrics(bool enable) {
>-  {
>-    rtc::CritScope cs(crit_capture_);
>-    metrics_enabled_ = enable;
>-  }
>+  metrics_enabled_ = enable;
>   return Configure();
> }
> 
> bool EchoCancellationImpl::are_metrics_enabled() const {
>-  rtc::CritScope cs(crit_capture_);
>   return enabled_ && metrics_enabled_;
> }
> 
> // TODO(ajm): we currently just use the metrics from the first AEC. Think more
> //            aboue the best way to extend this to multi-channel.
> int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
>-  rtc::CritScope cs(crit_capture_);
>   if (metrics == NULL) {
>     return AudioProcessing::kNullPointerError;
>   }
>@@ -331,46 +305,36 @@ int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
> }
> 
> bool EchoCancellationImpl::stream_has_echo() const {
>-  rtc::CritScope cs(crit_capture_);
>   return stream_has_echo_;
> }
> 
> int EchoCancellationImpl::enable_delay_logging(bool enable) {
>-  {
>-    rtc::CritScope cs(crit_capture_);
>-    delay_logging_enabled_ = enable;
>-  }
>+  delay_logging_enabled_ = enable;
>   return Configure();
> }
> 
> bool EchoCancellationImpl::is_delay_logging_enabled() const {
>-  rtc::CritScope cs(crit_capture_);
>   return enabled_ && delay_logging_enabled_;
> }
> 
> bool EchoCancellationImpl::is_delay_agnostic_enabled() const {
>-  rtc::CritScope cs(crit_capture_);
>   return delay_agnostic_enabled_;
> }
> 
> std::string EchoCancellationImpl::GetExperimentsDescription() {
>-  rtc::CritScope cs(crit_capture_);
>   return refined_adaptive_filter_enabled_ ? "RefinedAdaptiveFilter;" : "";
> }
> 
> bool EchoCancellationImpl::is_refined_adaptive_filter_enabled() const {
>-  rtc::CritScope cs(crit_capture_);
>   return refined_adaptive_filter_enabled_;
> }
> 
> bool EchoCancellationImpl::is_extended_filter_enabled() const {
>-  rtc::CritScope cs(crit_capture_);
>   return extended_filter_enabled_;
> }
> 
> // TODO(bjornv): How should we handle the multi-channel case?
> int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
>-  rtc::CritScope cs(crit_capture_);
>   float fraction_poor_delays = 0;
>   return GetDelayMetrics(median, std, &fraction_poor_delays);
> }
>@@ -378,7 +342,6 @@ int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
> int EchoCancellationImpl::GetDelayMetrics(int* median,
>                                           int* std,
>                                           float* fraction_poor_delays) {
>-  rtc::CritScope cs(crit_capture_);
>   if (median == NULL) {
>     return AudioProcessing::kNullPointerError;
>   }
>@@ -400,7 +363,6 @@ int EchoCancellationImpl::GetDelayMetrics(int* median,
> }
> 
> struct AecCore* EchoCancellationImpl::aec_core() const {
>-  rtc::CritScope cs(crit_capture_);
>   if (!enabled_) {
>     return NULL;
>   }
>@@ -411,9 +373,6 @@ void EchoCancellationImpl::Initialize(int sample_rate_hz,
>                                       size_t num_reverse_channels,
>                                       size_t num_output_channels,
>                                       size_t num_proc_channels) {
>-  rtc::CritScope cs_render(crit_render_);
>-  rtc::CritScope cs_capture(crit_capture_);
>-
>   stream_properties_.reset(
>       new StreamProperties(sample_rate_hz, num_reverse_channels,
>                            num_output_channels, num_proc_channels));
>@@ -442,7 +401,6 @@ void EchoCancellationImpl::Initialize(int sample_rate_hz,
> }
> 
> int EchoCancellationImpl::GetSystemDelayInSamples() const {
>-  rtc::CritScope cs(crit_capture_);
>   RTC_DCHECK(enabled_);
>   // Report the delay for the first AEC component.
>   return WebRtcAec_system_delay(WebRtcAec_aec_core(cancellers_[0]->state()));
>@@ -471,7 +429,6 @@ void EchoCancellationImpl::PackRenderAudioBuffer(
> 
> void EchoCancellationImpl::SetExtraOptions(const webrtc::Config& config) {
>   {
>-    rtc::CritScope cs(crit_capture_);
>     extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled;
>     delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled;
>     refined_adaptive_filter_enabled_ =
>@@ -481,8 +438,6 @@ void EchoCancellationImpl::SetExtraOptions(const webrtc::Config& config) {
> }
> 
> int EchoCancellationImpl::Configure() {
>-  rtc::CritScope cs_render(crit_render_);
>-  rtc::CritScope cs_capture(crit_capture_);
>   AecConfig config;
>   config.metricsMode = metrics_enabled_;
>   config.nlpMode = MapSetting(suppression_level_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.h
>index 091a7b5116a2ad7e618a9cfe1847413e7f442ba2..a8b43a8cd62b8ec151b80de0b0ae64913677ffc1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.h
>@@ -11,12 +11,16 @@
> #ifndef MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
> #define MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
> 
>+#include <stddef.h>
> #include <memory>
>+#include <string>
> #include <vector>
> 
>+#include "api/array_view.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
> 
>@@ -28,8 +32,7 @@ class AudioBuffer;
> // for PC and IP phone applications.
> class EchoCancellationImpl {
>  public:
>-  EchoCancellationImpl(rtc::CriticalSection* crit_render,
>-                       rtc::CriticalSection* crit_capture);
>+  explicit EchoCancellationImpl();
>   ~EchoCancellationImpl();
> 
>   void ProcessRenderAudio(rtc::ArrayView<const float> packed_render_audio);
>@@ -80,17 +83,23 @@ class EchoCancellationImpl {
>   // P_a:    Internal signal power at the point before the AEC's non-linear
>   //         processor.
>   struct Metrics {
>+    struct Statistic {
>+      int instant = 0;  // Instantaneous value.
>+      int average = 0;  // Long-term average.
>+      int maximum = 0;  // Long-term maximum.
>+      int minimum = 0;  // Long-term minimum.
>+    };
>     // RERL = ERL + ERLE
>-    AudioProcessing::Statistic residual_echo_return_loss;
>+    Statistic residual_echo_return_loss;
> 
>     // ERL = 10log_10(P_far / P_echo)
>-    AudioProcessing::Statistic echo_return_loss;
>+    Statistic echo_return_loss;
> 
>     // ERLE = 10log_10(P_echo / P_out)
>-    AudioProcessing::Statistic echo_return_loss_enhancement;
>+    Statistic echo_return_loss_enhancement;
> 
>     // (Pre non-linear processing suppression) A_NLP = 10log_10(P_echo / P_a)
>-    AudioProcessing::Statistic a_nlp;
>+    Statistic a_nlp;
> 
>     // Fraction of time that the AEC linear filter is divergent, in a 1-second
>     // non-overlapped aggregation window.
>@@ -150,28 +159,23 @@ class EchoCancellationImpl {
>   void AllocateRenderQueue();
>   int Configure();
> 
>-  rtc::CriticalSection* const crit_render_ RTC_ACQUIRED_BEFORE(crit_capture_);
>-  rtc::CriticalSection* const crit_capture_;
>-
>   bool enabled_ = false;
>-  bool drift_compensation_enabled_ RTC_GUARDED_BY(crit_capture_);
>-  bool metrics_enabled_ RTC_GUARDED_BY(crit_capture_);
>-  SuppressionLevel suppression_level_ RTC_GUARDED_BY(crit_capture_);
>-  int stream_drift_samples_ RTC_GUARDED_BY(crit_capture_);
>-  bool was_stream_drift_set_ RTC_GUARDED_BY(crit_capture_);
>-  bool stream_has_echo_ RTC_GUARDED_BY(crit_capture_);
>-  bool delay_logging_enabled_ RTC_GUARDED_BY(crit_capture_);
>-  bool extended_filter_enabled_ RTC_GUARDED_BY(crit_capture_);
>-  bool delay_agnostic_enabled_ RTC_GUARDED_BY(crit_capture_);
>-  bool refined_adaptive_filter_enabled_ RTC_GUARDED_BY(crit_capture_) = false;
>+  bool drift_compensation_enabled_;
>+  bool metrics_enabled_;
>+  SuppressionLevel suppression_level_;
>+  int stream_drift_samples_;
>+  bool was_stream_drift_set_;
>+  bool stream_has_echo_;
>+  bool delay_logging_enabled_;
>+  bool extended_filter_enabled_;
>+  bool delay_agnostic_enabled_;
>+  bool refined_adaptive_filter_enabled_ = false;
> 
>   // Only active on Chrome OS devices.
>-  const bool enforce_zero_stream_delay_ RTC_GUARDED_BY(crit_capture_);
>+  const bool enforce_zero_stream_delay_;
> 
>   std::vector<std::unique_ptr<Canceller>> cancellers_;
>   std::unique_ptr<StreamProperties> stream_properties_;
>-
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoCancellationImpl);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
>index c50c52f857bf7351b6b81515c7b97bd5758dda5e..9d81c1df3b76b23a19d0aaa106dcc52f0c4d0cdb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl_unittest.cc
>@@ -18,9 +18,7 @@
> 
> namespace webrtc {
> TEST(EchoCancellationInternalTest, ExtendedFilter) {
>-  rtc::CriticalSection crit_render;
>-  rtc::CriticalSection crit_capture;
>-  EchoCancellationImpl echo_canceller(&crit_render, &crit_capture);
>+  EchoCancellationImpl echo_canceller;
>   echo_canceller.Initialize(AudioProcessing::kSampleRate32kHz, 2, 2, 2);
> 
>   EXPECT_TRUE(echo_canceller.aec_core() == nullptr);
>@@ -51,9 +49,7 @@ TEST(EchoCancellationInternalTest, ExtendedFilter) {
> }
> 
> TEST(EchoCancellationInternalTest, DelayAgnostic) {
>-  rtc::CriticalSection crit_render;
>-  rtc::CriticalSection crit_capture;
>-  EchoCancellationImpl echo_canceller(&crit_render, &crit_capture);
>+  EchoCancellationImpl echo_canceller;
>   echo_canceller.Initialize(AudioProcessing::kSampleRate32kHz, 1, 1, 1);
> 
>   EXPECT_TRUE(echo_canceller.aec_core() == NULL);
>@@ -85,9 +81,7 @@ TEST(EchoCancellationInternalTest, DelayAgnostic) {
> }
> 
> TEST(EchoCancellationInternalTest, InterfaceConfiguration) {
>-  rtc::CriticalSection crit_render;
>-  rtc::CriticalSection crit_capture;
>-  EchoCancellationImpl echo_canceller(&crit_render, &crit_capture);
>+  EchoCancellationImpl echo_canceller;
>   echo_canceller.Initialize(AudioProcessing::kSampleRate16kHz, 1, 1, 1);
> 
>   EXPECT_EQ(0, echo_canceller.enable_drift_compensation(true));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_bit_exact_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_bit_exact_unittest.cc
>index 8b8369260a55651dec20e891d9639e92af17b51e..7844daf5225266acfe0dbcc461d9c8ef0c7e0e96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_bit_exact_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_bit_exact_unittest.cc
>@@ -64,9 +64,7 @@ void RunBitexactnessTest(int sample_rate_hz,
>                          EchoControlMobileImpl::RoutingMode routing_mode,
>                          bool comfort_noise_enabled,
>                          const rtc::ArrayView<const float>& output_reference) {
>-  rtc::CriticalSection crit_render;
>-  rtc::CriticalSection crit_capture;
>-  EchoControlMobileImpl echo_control_mobile(&crit_render, &crit_capture);
>+  EchoControlMobileImpl echo_control_mobile;
>   SetupComponent(sample_rate_hz, routing_mode, comfort_noise_enabled,
>                  &echo_control_mobile);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
>index bd125c6e0de09a457eb25ba9ada59dd2e3e5ea71..b9fbf42052797dffaa3d639e1620268b08560801 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
>@@ -11,11 +11,13 @@
> #include "modules/audio_processing/echo_control_mobile_impl.h"
> 
> #include <string.h>
>+#include <cstdint>
> 
> #include "modules/audio_processing/aecm/echo_control_mobile.h"
> #include "modules/audio_processing/audio_buffer.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/logging.h"
> 
> namespace webrtc {
> 
>@@ -55,10 +57,6 @@ AudioProcessing::Error MapError(int err) {
> }
> }  // namespace
> 
>-size_t EchoControlMobileImpl::echo_path_size_bytes() {
>-  return WebRtcAecm_echo_path_size_bytes();
>-}
>-
> struct EchoControlMobileImpl::StreamProperties {
>   StreamProperties() = delete;
>   StreamProperties(int sample_rate_hz,
>@@ -90,17 +88,10 @@ class EchoControlMobileImpl::Canceller {
>     return state_;
>   }
> 
>-  void Initialize(int sample_rate_hz,
>-                  unsigned char* external_echo_path,
>-                  size_t echo_path_size_bytes) {
>+  void Initialize(int sample_rate_hz) {
>     RTC_DCHECK(state_);
>     int error = WebRtcAecm_Init(state_, sample_rate_hz);
>     RTC_DCHECK_EQ(AudioProcessing::kNoError, error);
>-    if (external_echo_path != NULL) {
>-      error = WebRtcAecm_InitEchoPath(state_, external_echo_path,
>-                                      echo_path_size_bytes);
>-      RTC_DCHECK_EQ(AudioProcessing::kNoError, error);
>-    }
>   }
> 
>  private:
>@@ -108,27 +99,13 @@ class EchoControlMobileImpl::Canceller {
>   RTC_DISALLOW_COPY_AND_ASSIGN(Canceller);
> };
> 
>-EchoControlMobileImpl::EchoControlMobileImpl(rtc::CriticalSection* crit_render,
>-                                             rtc::CriticalSection* crit_capture)
>-    : crit_render_(crit_render),
>-      crit_capture_(crit_capture),
>-      routing_mode_(kSpeakerphone),
>-      comfort_noise_enabled_(false),
>-      external_echo_path_(NULL) {
>-  RTC_DCHECK(crit_render);
>-  RTC_DCHECK(crit_capture);
>-}
>+EchoControlMobileImpl::EchoControlMobileImpl()
>+    : routing_mode_(kSpeakerphone), comfort_noise_enabled_(false) {}
> 
>-EchoControlMobileImpl::~EchoControlMobileImpl() {
>-  if (external_echo_path_ != NULL) {
>-    delete[] external_echo_path_;
>-    external_echo_path_ = NULL;
>-  }
>-}
>+EchoControlMobileImpl::~EchoControlMobileImpl() {}
> 
> void EchoControlMobileImpl::ProcessRenderAudio(
>     rtc::ArrayView<const int16_t> packed_render_audio) {
>-  rtc::CritScope cs_capture(crit_capture_);
>   if (!enabled_) {
>     return;
>   }
>@@ -181,7 +158,6 @@ size_t EchoControlMobileImpl::NumCancellersRequired(
> 
> int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio,
>                                                int stream_delay_ms) {
>-  rtc::CritScope cs_capture(crit_capture_);
>   if (!enabled_) {
>     return AudioProcessing::kNoError;
>   }
>@@ -228,8 +204,6 @@ int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio,
> 
> int EchoControlMobileImpl::Enable(bool enable) {
>   // Ensure AEC and AECM are not both enabled.
>-  rtc::CritScope cs_render(crit_render_);
>-  rtc::CritScope cs_capture(crit_capture_);
>   RTC_DCHECK(stream_properties_);
> 
>   if (enable &&
>@@ -252,7 +226,6 @@ int EchoControlMobileImpl::Enable(bool enable) {
> }
> 
> bool EchoControlMobileImpl::is_enabled() const {
>-  rtc::CritScope cs(crit_capture_);
>   return enabled_;
> }
> 
>@@ -260,90 +233,26 @@ int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) {
>   if (MapSetting(mode) == -1) {
>     return AudioProcessing::kBadParameterError;
>   }
>-
>-  {
>-    rtc::CritScope cs(crit_capture_);
>     routing_mode_ = mode;
>-  }
>   return Configure();
> }
> 
> EchoControlMobileImpl::RoutingMode EchoControlMobileImpl::routing_mode() const {
>-  rtc::CritScope cs(crit_capture_);
>   return routing_mode_;
> }
> 
> int EchoControlMobileImpl::enable_comfort_noise(bool enable) {
>-  {
>-    rtc::CritScope cs(crit_capture_);
>     comfort_noise_enabled_ = enable;
>-  }
>   return Configure();
> }
> 
> bool EchoControlMobileImpl::is_comfort_noise_enabled() const {
>-  rtc::CritScope cs(crit_capture_);
>   return comfort_noise_enabled_;
> }
> 
>-int EchoControlMobileImpl::SetEchoPath(const void* echo_path,
>-                                       size_t size_bytes) {
>-  {
>-    rtc::CritScope cs_render(crit_render_);
>-    rtc::CritScope cs_capture(crit_capture_);
>-    if (echo_path == NULL) {
>-      return AudioProcessing::kNullPointerError;
>-    }
>-    if (size_bytes != echo_path_size_bytes()) {
>-      // Size mismatch
>-      return AudioProcessing::kBadParameterError;
>-    }
>-
>-    if (external_echo_path_ == NULL) {
>-      external_echo_path_ = new unsigned char[size_bytes];
>-    }
>-    memcpy(external_echo_path_, echo_path, size_bytes);
>-  }
>-
>-  // TODO(peah): Simplify once the Enable function has been removed from
>-  // the public APM API.
>-  RTC_DCHECK(stream_properties_);
>-  Initialize(stream_properties_->sample_rate_hz,
>-             stream_properties_->num_reverse_channels,
>-             stream_properties_->num_output_channels);
>-  return AudioProcessing::kNoError;
>-}
>-
>-int EchoControlMobileImpl::GetEchoPath(void* echo_path,
>-                                       size_t size_bytes) const {
>-  rtc::CritScope cs(crit_capture_);
>-  if (echo_path == NULL) {
>-    return AudioProcessing::kNullPointerError;
>-  }
>-  if (size_bytes != echo_path_size_bytes()) {
>-    // Size mismatch
>-    return AudioProcessing::kBadParameterError;
>-  }
>-  if (!enabled_) {
>-    return AudioProcessing::kNotEnabledError;
>-  }
>-
>-  // Get the echo path from the first channel
>-  int32_t err =
>-      WebRtcAecm_GetEchoPath(cancellers_[0]->state(), echo_path, size_bytes);
>-  if (err != 0) {
>-    return MapError(err);
>-  }
>-
>-  return AudioProcessing::kNoError;
>-}
>-
> void EchoControlMobileImpl::Initialize(int sample_rate_hz,
>                                        size_t num_reverse_channels,
>                                        size_t num_output_channels) {
>-  rtc::CritScope cs_render(crit_render_);
>-  rtc::CritScope cs_capture(crit_capture_);
>-
>   stream_properties_.reset(new StreamProperties(
>       sample_rate_hz, num_reverse_channels, num_output_channels));
> 
>@@ -363,16 +272,12 @@ void EchoControlMobileImpl::Initialize(int sample_rate_hz,
>     if (!canceller) {
>       canceller.reset(new Canceller());
>     }
>-    canceller->Initialize(sample_rate_hz, external_echo_path_,
>-                          echo_path_size_bytes());
>+    canceller->Initialize(sample_rate_hz);
>   }
>-
>   Configure();
> }
> 
> int EchoControlMobileImpl::Configure() {
>-  rtc::CritScope cs_render(crit_render_);
>-  rtc::CritScope cs_capture(crit_capture_);
>   AecmConfig config;
>   config.cngMode = comfort_noise_enabled_;
>   config.echoMode = MapSetting(routing_mode_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.h
>index a5b66c8311b80447a91d557a749ffe702727abbb..d5a8b0b445155e655e42b3702f1f37523f7a2f0e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.h
>@@ -11,14 +11,12 @@
> #ifndef MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
> #define MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> #include <vector>
> 
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/audio_processing/render_queue_item_verifier.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/criticalsection.h"
>-#include "rtc_base/swap_queue.h"
>+#include "api/array_view.h"
> 
> namespace webrtc {
> 
>@@ -28,8 +26,7 @@ class AudioBuffer;
> // robust option intended for use on mobile devices.
> class EchoControlMobileImpl {
>  public:
>-  EchoControlMobileImpl(rtc::CriticalSection* crit_render,
>-                        rtc::CriticalSection* crit_capture);
>+  EchoControlMobileImpl();
> 
>   ~EchoControlMobileImpl();
> 
>@@ -57,26 +54,6 @@ class EchoControlMobileImpl {
>   int enable_comfort_noise(bool enable);
>   bool is_comfort_noise_enabled() const;
> 
>-  // A typical use case is to initialize the component with an echo path from a
>-  // previous call. The echo path is retrieved using |GetEchoPath()|, typically
>-  // at the end of a call. The data can then be stored for later use as an
>-  // initializer before the next call, using |SetEchoPath()|.
>-  //
>-  // Controlling the echo path this way requires the data |size_bytes| to match
>-  // the internal echo path size. This size can be acquired using
>-  // |echo_path_size_bytes()|. |SetEchoPath()| causes an entire reset, worth
>-  // noting if it is to be called during an ongoing call.
>-  //
>-  // It is possible that version incompatibilities may result in a stored echo
>-  // path of the incorrect size. In this case, the stored path should be
>-  // discarded.
>-  int SetEchoPath(const void* echo_path, size_t size_bytes);
>-  int GetEchoPath(void* echo_path, size_t size_bytes) const;
>-
>-  // The returned path size is guaranteed not to change for the lifetime of
>-  // the application.
>-  static size_t echo_path_size_bytes();
>-
>   void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
>   int ProcessCaptureAudio(AudioBuffer* audio, int stream_delay_ms);
> 
>@@ -98,20 +75,13 @@ class EchoControlMobileImpl {
> 
>   int Configure();
> 
>-  rtc::CriticalSection* const crit_render_ RTC_ACQUIRED_BEFORE(crit_capture_);
>-  rtc::CriticalSection* const crit_capture_;
>-
>   bool enabled_ = false;
> 
>-  RoutingMode routing_mode_ RTC_GUARDED_BY(crit_capture_);
>-  bool comfort_noise_enabled_ RTC_GUARDED_BY(crit_capture_);
>-  unsigned char* external_echo_path_ RTC_GUARDED_BY(crit_render_)
>-      RTC_GUARDED_BY(crit_capture_);
>+  RoutingMode routing_mode_;
>+  bool comfort_noise_enabled_;
> 
>   std::vector<std::unique_ptr<Canceller>> cancellers_;
>   std::unique_ptr<StreamProperties> stream_properties_;
>-
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoControlMobileImpl);
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_unittest.cc
>index d7e470ca120f9cadda7feb81a21fe49570905312..f0e60483d411a3e4e43b5e5b88aa7616ab438cd3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_unittest.cc
>@@ -18,9 +18,7 @@
> 
> namespace webrtc {
> TEST(EchoControlMobileTest, InterfaceConfiguration) {
>-  rtc::CriticalSection crit_render;
>-  rtc::CriticalSection crit_capture;
>-  EchoControlMobileImpl aecm(&crit_render, &crit_capture);
>+  EchoControlMobileImpl aecm;
>   aecm.Initialize(AudioProcessing::kSampleRate16kHz, 2, 2);
> 
>   // Turn AECM on
>@@ -46,28 +44,6 @@ TEST(EchoControlMobileTest, InterfaceConfiguration) {
>   EXPECT_EQ(0, aecm.enable_comfort_noise(true));
>   EXPECT_TRUE(aecm.is_comfort_noise_enabled());
> 
>-  // Set and get echo path
>-  const size_t echo_path_size = aecm.echo_path_size_bytes();
>-  std::vector<uint8_t> echo_path_in(echo_path_size);
>-  std::vector<uint8_t> echo_path_out(echo_path_size);
>-  EXPECT_EQ(AudioProcessing::kNullPointerError,
>-            aecm.SetEchoPath(nullptr, echo_path_size));
>-  EXPECT_EQ(AudioProcessing::kNullPointerError,
>-            aecm.GetEchoPath(nullptr, echo_path_size));
>-  EXPECT_EQ(AudioProcessing::kBadParameterError,
>-            aecm.GetEchoPath(echo_path_out.data(), 1));
>-  EXPECT_EQ(0, aecm.GetEchoPath(echo_path_out.data(), echo_path_size));
>-  for (size_t i = 0; i < echo_path_size; i++) {
>-    echo_path_in[i] = echo_path_out[i] + 1;
>-  }
>-  EXPECT_EQ(AudioProcessing::kBadParameterError,
>-            aecm.SetEchoPath(echo_path_in.data(), 1));
>-  EXPECT_EQ(0, aecm.SetEchoPath(echo_path_in.data(), echo_path_size));
>-  EXPECT_EQ(0, aecm.GetEchoPath(echo_path_out.data(), echo_path_size));
>-  for (size_t i = 0; i < echo_path_size; i++) {
>-    EXPECT_EQ(echo_path_in[i], echo_path_out[i]);
>-  }
>-
>   // Turn AECM off
>   EXPECT_EQ(0, aecm.Enable(false));
>   EXPECT_FALSE(aecm.is_enabled());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.h
>index 9f5cdfa95364359d6d3ae3d56f6590e1434b4548..c52311f863b48b6278ff330f183adf5d2dbbe522 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_CIRCULAR_BUFFER_H_
> #define MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_CIRCULAR_BUFFER_H_
> 
>+#include <stddef.h>
> #include <vector>
> 
> #include "absl/types/optional.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc
>index 4ab856c2c33e570940faeb98c0f7f388d359840e..1479d58b5430c98ba707dc263a2f2cd9b8bd116d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc
>@@ -13,7 +13,6 @@
> #include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
>-#include "rtc_base/checks.h"
> #include "rtc_base/criticalsection.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.cc
>index 685a27ff6681209ab2a4b842bc712e2b494dbb90..d5118ba073d9940cfcbab353c88cefec72ad6863 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.cc
>@@ -10,10 +10,14 @@
> 
> #include "modules/audio_processing/gain_control_impl.h"
> 
>+#include <cstdint>
>+
> #include "absl/types/optional.h"
> #include "modules/audio_processing/agc/legacy/gain_control.h"
> #include "modules/audio_processing/audio_buffer.h"
>+#include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.h
>index 959422fb44c20ea139378157d6a121ac30a67b6e..c21d9113262d475b83d57cb296e225e9afd7747f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.h
>@@ -11,14 +11,16 @@
> #ifndef MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
> #define MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> #include <vector>
> 
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/audio_processing/render_queue_item_verifier.h"
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "modules/audio_processing/include/gain_control.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>-#include "rtc_base/swap_queue.h"
> #include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.cc
>index 29af962e745fb50631722241b9d949299c7f1e7d..2a327449df2e164282bd06c28d3ce9812df762d7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/audio_processing/gain_controller2.h"
> 
>+#include "common_audio/include/audio_util.h"
> #include "modules/audio_processing/audio_buffer.h"
> #include "modules/audio_processing/include/audio_frame_view.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>@@ -24,8 +25,10 @@ int GainController2::instance_count_ = 0;
> GainController2::GainController2()
>     : data_dumper_(
>           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>-      fixed_gain_controller_(data_dumper_.get()),
>-      adaptive_agc_(data_dumper_.get()) {}
>+      gain_applier_(/*hard_clip_samples=*/false,
>+                    /*initial_gain_factor=*/0.f),
>+      adaptive_agc_(new AdaptiveAgc(data_dumper_.get())),
>+      limiter_(static_cast<size_t>(48000), data_dumper_.get(), "Agc2") {}
> 
> GainController2::~GainController2() = default;
> 
>@@ -34,7 +37,7 @@ void GainController2::Initialize(int sample_rate_hz) {
>              sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
>              sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
>              sample_rate_hz == AudioProcessing::kSampleRate48kHz);
>-  fixed_gain_controller_.SetSampleRate(sample_rate_hz);
>+  limiter_.SetSampleRate(sample_rate_hz);
>   data_dumper_->InitiateNewSetOfRecordings();
>   data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz);
> }
>@@ -42,37 +45,71 @@ void GainController2::Initialize(int sample_rate_hz) {
> void GainController2::Process(AudioBuffer* audio) {
>   AudioFrameView<float> float_frame(audio->channels_f(), audio->num_channels(),
>                                     audio->num_frames());
>-  if (adaptive_digital_mode_) {
>-    adaptive_agc_.Process(float_frame, fixed_gain_controller_.LastAudioLevel());
>+  // Apply fixed gain first, then the adaptive one.
>+  gain_applier_.ApplyGain(float_frame);
>+  if (config_.adaptive_digital.enabled) {
>+    adaptive_agc_->Process(float_frame, limiter_.LastAudioLevel());
>   }
>-  fixed_gain_controller_.Process(float_frame);
>+  limiter_.Process(float_frame);
> }
> 
> void GainController2::NotifyAnalogLevel(int level) {
>-  if (analog_level_ != level && adaptive_digital_mode_) {
>-    adaptive_agc_.Reset();
>+  if (analog_level_ != level && config_.adaptive_digital.enabled) {
>+    adaptive_agc_->Reset();
>   }
>   analog_level_ = level;
> }
> 
> void GainController2::ApplyConfig(
>     const AudioProcessing::Config::GainController2& config) {
>-  RTC_DCHECK(Validate(config));
>+  RTC_DCHECK(Validate(config))
>+      << " the invalid config was " << ToString(config);
>+
>   config_ = config;
>-  fixed_gain_controller_.SetGain(config_.fixed_gain_db);
>-  adaptive_digital_mode_ = config_.adaptive_digital_mode;
>+  if (config.fixed_digital.gain_db != config_.fixed_digital.gain_db) {
>+    // Reset the limiter to quickly react on abrupt level changes caused by
>+    // large changes of the fixed gain.
>+    limiter_.Reset();
>+  }
>+  gain_applier_.SetGainFactor(DbToRatio(config_.fixed_digital.gain_db));
>+  adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get(), config_));
> }
> 
> bool GainController2::Validate(
>     const AudioProcessing::Config::GainController2& config) {
>-  return config.fixed_gain_db >= 0.f;
>+  return config.fixed_digital.gain_db >= 0.f &&
>+         config.fixed_digital.gain_db < 50.f &&
>+         config.adaptive_digital.extra_saturation_margin_db >= 0.f &&
>+         config.adaptive_digital.extra_saturation_margin_db <= 100.f;
> }
> 
> std::string GainController2::ToString(
>     const AudioProcessing::Config::GainController2& config) {
>   rtc::StringBuilder ss;
>-  ss << "{enabled: " << (config.enabled ? "true" : "false") << ", "
>-     << "fixed_gain_dB: " << config.fixed_gain_db << "}";
>+  std::string adaptive_digital_level_estimator;
>+  using LevelEstimatorType =
>+      AudioProcessing::Config::GainController2::LevelEstimator;
>+  switch (config.adaptive_digital.level_estimator) {
>+    case LevelEstimatorType::kRms:
>+      adaptive_digital_level_estimator = "RMS";
>+      break;
>+    case LevelEstimatorType::kPeak:
>+      adaptive_digital_level_estimator = "peak";
>+      break;
>+  }
>+  // clang-format off
>+  // clang formatting doesn't respect custom nested style.
>+  ss << "{"
>+     << "enabled: " << (config.enabled ? "true" : "false") << ", "
>+     << "fixed_digital: {gain_db: " << config.fixed_digital.gain_db << "}, "
>+     << "adaptive_digital: {"
>+      << "enabled: "
>+        << (config.adaptive_digital.enabled ? "true" : "false") << ", "
>+      << "level_estimator: " << adaptive_digital_level_estimator << ", "
>+      << "extra_saturation_margin_db:"
>+        << config.adaptive_digital.extra_saturation_margin_db << "}"
>+      << "}";
>+  // clang-format on
>   return ss.Release();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.h
>index b4727d0e777695e70fd24d49ceb7ad8fa7979a9c..3a11810bb59d9539566d8c75e13000e09811c8a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.h
>@@ -15,7 +15,8 @@
> #include <string>
> 
> #include "modules/audio_processing/agc2/adaptive_agc.h"
>-#include "modules/audio_processing/agc2/fixed_gain_controller.h"
>+#include "modules/audio_processing/agc2/gain_applier.h"
>+#include "modules/audio_processing/agc2/limiter.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/constructormagic.h"
> 
>@@ -43,11 +44,11 @@ class GainController2 {
>  private:
>   static int instance_count_;
>   std::unique_ptr<ApmDataDumper> data_dumper_;
>-  FixedGainController fixed_gain_controller_;
>   AudioProcessing::Config::GainController2 config_;
>-  AdaptiveAgc adaptive_agc_;
>+  GainApplier gain_applier_;
>+  std::unique_ptr<AdaptiveAgc> adaptive_agc_;
>+  Limiter limiter_;
>   int analog_level_ = -1;
>-  bool adaptive_digital_mode_ = true;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(GainController2);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2_unittest.cc
>index 5bedccd64a8bea76cca79a3663ac50980fdaf5c4..27d540a1988e9e8969fcdb08c75ea02bfce4fd72 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2_unittest.cc
>@@ -10,20 +10,20 @@
> 
> #include <algorithm>
> 
>+#include "absl/memory/memory.h"
> #include "api/array_view.h"
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
> #include "modules/audio_processing/audio_buffer.h"
> #include "modules/audio_processing/gain_controller2.h"
>+#include "modules/audio_processing/test/audio_buffer_tools.h"
>+#include "modules/audio_processing/test/bitexactness_tools.h"
> #include "rtc_base/checks.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> namespace test {
>-
> namespace {
> 
>-constexpr size_t kFrameSizeMs = 10u;
>-constexpr size_t kStereo = 2u;
>-
> void SetAudioBufferSamples(float value, AudioBuffer* ab) {
>   // Sets all the samples in |ab| to |value|.
>   for (size_t k = 0; k < ab->num_channels(); ++k) {
>@@ -32,6 +32,75 @@ void SetAudioBufferSamples(float value, AudioBuffer* ab) {
>   }
> }
> 
>+float RunAgc2WithConstantInput(GainController2* agc2,
>+                               float input_level,
>+                               size_t num_frames,
>+                               int sample_rate) {
>+  const int num_samples = rtc::CheckedDivExact(sample_rate, 100);
>+  AudioBuffer ab(num_samples, 1, num_samples, 1, num_samples);
>+
>+  // Give time to the level estimator to converge.
>+  for (size_t i = 0; i < num_frames + 1; ++i) {
>+    SetAudioBufferSamples(input_level, &ab);
>+    agc2->Process(&ab);
>+  }
>+
>+  // Return the last sample from the last processed frame.
>+  return ab.channels_f()[0][num_samples - 1];
>+}
>+
>+AudioProcessing::Config::GainController2 CreateAgc2FixedDigitalModeConfig(
>+    float fixed_gain_db) {
>+  AudioProcessing::Config::GainController2 config;
>+  config.adaptive_digital.enabled = false;
>+  config.fixed_digital.gain_db = fixed_gain_db;
>+  // TODO(alessiob): Check why ASSERT_TRUE() below does not compile.
>+  EXPECT_TRUE(GainController2::Validate(config));
>+  return config;
>+}
>+
>+std::unique_ptr<GainController2> CreateAgc2FixedDigitalMode(
>+    float fixed_gain_db,
>+    size_t sample_rate_hz) {
>+  auto agc2 = absl::make_unique<GainController2>();
>+  agc2->ApplyConfig(CreateAgc2FixedDigitalModeConfig(fixed_gain_db));
>+  agc2->Initialize(sample_rate_hz);
>+  return agc2;
>+}
>+
>+float GainAfterProcessingFile(GainController2* gain_controller) {
>+  // Set up an AudioBuffer to be filled from the speech file.
>+  constexpr size_t kStereo = 2u;
>+  const StreamConfig capture_config(AudioProcessing::kSampleRate48kHz, kStereo,
>+                                    false);
>+  AudioBuffer ab(capture_config.num_frames(), capture_config.num_channels(),
>+                 capture_config.num_frames(), capture_config.num_channels(),
>+                 capture_config.num_frames());
>+  test::InputAudioFile capture_file(
>+      test::GetApmCaptureTestVectorFileName(AudioProcessing::kSampleRate48kHz));
>+  std::vector<float> capture_input(capture_config.num_frames() *
>+                                   capture_config.num_channels());
>+
>+  // The file should contain at least this many frames. Every iteration, we put
>+  // a frame through the gain controller.
>+  const int kNumFramesToProcess = 100;
>+  for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
>+    ReadFloatSamplesFromStereoFile(capture_config.num_frames(),
>+                                   capture_config.num_channels(), &capture_file,
>+                                   capture_input);
>+
>+    test::CopyVectorToAudioBuffer(capture_config, capture_input, &ab);
>+    gain_controller->Process(&ab);
>+  }
>+
>+  // Send in a last frame with values constant 1 (It's low enough to detect high
>+  // gain, and for ease of computation). The applied gain is the result.
>+  constexpr float sample_value = 1.f;
>+  SetAudioBufferSamples(sample_value, &ab);
>+  gain_controller->Process(&ab);
>+  return ab.channels_f()[0][0];
>+}
>+
> }  // namespace
> 
> TEST(GainController2, CreateApplyConfig) {
>@@ -44,47 +113,173 @@ TEST(GainController2, CreateApplyConfig) {
>   gain_controller2->ApplyConfig(config);
> 
>   // Check that attenuation is not allowed.
>-  config.fixed_gain_db = -5.f;
>+  config.fixed_digital.gain_db = -5.f;
>   EXPECT_FALSE(GainController2::Validate(config));
> 
>   // Check that valid configurations are applied.
>-  for (const float& fixed_gain_db : {0.f, 5.f, 10.f, 50.f}) {
>-    config.fixed_gain_db = fixed_gain_db;
>+  for (const float& fixed_gain_db : {0.f, 5.f, 10.f, 40.f}) {
>+    config.fixed_digital.gain_db = fixed_gain_db;
>     EXPECT_TRUE(GainController2::Validate(config));
>     gain_controller2->ApplyConfig(config);
>   }
> }
> 
> TEST(GainController2, ToString) {
>-  // Tests GainController2::ToString().
>+  // Tests GainController2::ToString(). Only test the enabled property.
>   AudioProcessing::Config::GainController2 config;
>-  config.fixed_gain_db = 5.f;
> 
>   config.enabled = false;
>-  EXPECT_EQ("{enabled: false, fixed_gain_dB: 5}",
>-            GainController2::ToString(config));
>+  EXPECT_EQ("{enabled: false", GainController2::ToString(config).substr(0, 15));
> 
>   config.enabled = true;
>-  EXPECT_EQ("{enabled: true, fixed_gain_dB: 5}",
>-            GainController2::ToString(config));
>+  EXPECT_EQ("{enabled: true", GainController2::ToString(config).substr(0, 14));
> }
> 
>-TEST(GainController2, Usage) {
>-  // Tests GainController2::Process() on an AudioBuffer instance.
>-  std::unique_ptr<GainController2> gain_controller2(new GainController2());
>-  gain_controller2->Initialize(AudioProcessing::kSampleRate48kHz);
>-  const size_t num_frames = rtc::CheckedDivExact<size_t>(
>-      kFrameSizeMs * AudioProcessing::kSampleRate48kHz, 1000);
>-  AudioBuffer ab(num_frames, kStereo, num_frames, kStereo, num_frames);
>-  constexpr float sample_value = 1000.f;
>-  SetAudioBufferSamples(sample_value, &ab);
>+TEST(GainController2FixedDigital, GainShouldChangeOnSetGain) {
>+  constexpr float kInputLevel = 1000.f;
>+  constexpr size_t kNumFrames = 5;
>+  constexpr size_t kSampleRateHz = 8000;
>+  constexpr float kGain0Db = 0.f;
>+  constexpr float kGain20Db = 20.f;
>+
>+  auto agc2_fixed = CreateAgc2FixedDigitalMode(kGain0Db, kSampleRateHz);
>+
>+  // Signal level is unchanged with 0 db gain.
>+  EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(agc2_fixed.get(), kInputLevel,
>+                                           kNumFrames, kSampleRateHz),
>+                  kInputLevel);
>+
>+  // +20 db should increase signal by a factor of 10.
>+  agc2_fixed->ApplyConfig(CreateAgc2FixedDigitalModeConfig(kGain20Db));
>+  EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(agc2_fixed.get(), kInputLevel,
>+                                           kNumFrames, kSampleRateHz),
>+                  kInputLevel * 10);
>+}
>+
>+TEST(GainController2FixedDigital, ChangeFixedGainShouldBeFastAndTimeInvariant) {
>+  // Number of frames required for the fixed gain controller to adapt on the
>+  // input signal when the gain changes.
>+  constexpr size_t kNumFrames = 5;
>+
>+  constexpr float kInputLevel = 1000.f;
>+  constexpr size_t kSampleRateHz = 8000;
>+  constexpr float kGainDbLow = 0.f;
>+  constexpr float kGainDbHigh = 25.f;
>+  static_assert(kGainDbLow < kGainDbHigh, "");
>+
>+  auto agc2_fixed = CreateAgc2FixedDigitalMode(kGainDbLow, kSampleRateHz);
>+
>+  // Start with a lower gain.
>+  const float output_level_pre = RunAgc2WithConstantInput(
>+      agc2_fixed.get(), kInputLevel, kNumFrames, kSampleRateHz);
>+
>+  // Increase gain.
>+  agc2_fixed->ApplyConfig(CreateAgc2FixedDigitalModeConfig(kGainDbHigh));
>+  static_cast<void>(RunAgc2WithConstantInput(agc2_fixed.get(), kInputLevel,
>+                                             kNumFrames, kSampleRateHz));
>+
>+  // Back to the lower gain.
>+  agc2_fixed->ApplyConfig(CreateAgc2FixedDigitalModeConfig(kGainDbLow));
>+  const float output_level_post = RunAgc2WithConstantInput(
>+      agc2_fixed.get(), kInputLevel, kNumFrames, kSampleRateHz);
>+
>+  EXPECT_EQ(output_level_pre, output_level_post);
>+}
>+
>+struct FixedDigitalTestParams {
>+  FixedDigitalTestParams(float gain_db_min,
>+                         float gain_db_max,
>+                         size_t sample_rate,
>+                         bool saturation_expected)
>+      : gain_db_min(gain_db_min),
>+        gain_db_max(gain_db_max),
>+        sample_rate(sample_rate),
>+        saturation_expected(saturation_expected) {}
>+  float gain_db_min;
>+  float gain_db_max;
>+  size_t sample_rate;
>+  bool saturation_expected;
>+};
>+
>+class FixedDigitalTest
>+    : public testing::Test,
>+      public testing::WithParamInterface<FixedDigitalTestParams> {};
>+
>+TEST_P(FixedDigitalTest, CheckSaturationBehaviorWithLimiter) {
>+  const float kInputLevel = 32767.f;
>+  const size_t kNumFrames = 5;
>+
>+  const auto params = GetParam();
>+
>+  const auto gains_db =
>+      test::LinSpace(params.gain_db_min, params.gain_db_max, 10);
>+  for (const auto gain_db : gains_db) {
>+    SCOPED_TRACE(std::to_string(gain_db));
>+    auto agc2_fixed = CreateAgc2FixedDigitalMode(gain_db, params.sample_rate);
>+    const float processed_sample = RunAgc2WithConstantInput(
>+        agc2_fixed.get(), kInputLevel, kNumFrames, params.sample_rate);
>+    if (params.saturation_expected) {
>+      EXPECT_FLOAT_EQ(processed_sample, 32767.f);
>+    } else {
>+      EXPECT_LT(processed_sample, 32767.f);
>+    }
>+  }
>+}
>+
>+static_assert(test::kLimiterMaxInputLevelDbFs < 10, "");
>+INSTANTIATE_TEST_CASE_P(
>+    GainController2,
>+    FixedDigitalTest,
>+    ::testing::Values(
>+        // When gain < |test::kLimiterMaxInputLevelDbFs|, the limiter will not
>+        // saturate the signal (at any sample rate).
>+        FixedDigitalTestParams(0.1f,
>+                               test::kLimiterMaxInputLevelDbFs - 0.01f,
>+                               8000,
>+                               false),
>+        FixedDigitalTestParams(0.1,
>+                               test::kLimiterMaxInputLevelDbFs - 0.01f,
>+                               48000,
>+                               false),
>+        // When gain > |test::kLimiterMaxInputLevelDbFs|, the limiter will
>+        // saturate the signal (at any sample rate).
>+        FixedDigitalTestParams(test::kLimiterMaxInputLevelDbFs + 0.01f,
>+                               10.f,
>+                               8000,
>+                               true),
>+        FixedDigitalTestParams(test::kLimiterMaxInputLevelDbFs + 0.01f,
>+                               10.f,
>+                               48000,
>+                               true)));
>+
>+TEST(GainController2, UsageSaturationMargin) {
>+  GainController2 gain_controller2;
>+  gain_controller2.Initialize(AudioProcessing::kSampleRate48kHz);
>+
>   AudioProcessing::Config::GainController2 config;
>+  // Check that samples are not amplified as much when extra margin is
>+  // high. They should not be amplified at all, but only after convergence. GC2
>+  // starts with a gain, and it takes time until it's down to 0 dB.
>+  config.fixed_digital.gain_db = 0.f;
>+  config.adaptive_digital.enabled = true;
>+  config.adaptive_digital.extra_saturation_margin_db = 50.f;
>+  gain_controller2.ApplyConfig(config);
> 
>-  // Check that samples are amplified when the fixed gain is greater than 0 dB.
>-  config.fixed_gain_db = 5.f;
>-  gain_controller2->ApplyConfig(config);
>-  gain_controller2->Process(&ab);
>-  EXPECT_LT(sample_value, ab.channels_f()[0][0]);
>+  EXPECT_LT(GainAfterProcessingFile(&gain_controller2), 2.f);
>+}
>+
>+TEST(GainController2, UsageNoSaturationMargin) {
>+  GainController2 gain_controller2;
>+  gain_controller2.Initialize(AudioProcessing::kSampleRate48kHz);
>+
>+  AudioProcessing::Config::GainController2 config;
>+  // Check that some gain is applied if there is no margin.
>+  config.fixed_digital.gain_db = 0.f;
>+  config.adaptive_digital.enabled = true;
>+  config.adaptive_digital.extra_saturation_margin_db = 0.f;
>+  gain_controller2.ApplyConfig(config);
>+
>+  EXPECT_GT(GainAfterProcessingFile(&gain_controller2), 2.f);
> }
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.h
>index 313e9d7b1de0b62e1abb6ca254ded0414cfa8b72..b734adfbbbb36a777e3ec9b6bb99f4505fdcf20f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.h
>@@ -11,14 +11,13 @@
> #ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
> #define MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
> 
>-#include <memory>
>+#include <stdint.h>
> #include <string>
>-#include <vector>
> 
>-#include "api/array_view.h"
> #include "api/audio/audio_frame.h"
> #include "modules/audio_processing/include/audio_frame_view.h"
> #include "modules/audio_processing/include/audio_processing.h"
>+#include "rtc_base/deprecation.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.cc
>index 75eedafc292387f2e9915f318d002ec1520afe97..27cd8824fec305c5b31f5a335ef8d63e6af32e86 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.cc
>@@ -10,8 +10,6 @@
> 
> #include "modules/audio_processing/include/audio_processing.h"
> 
>-#include "rtc_base/checks.h"
>-
> namespace webrtc {
> 
> void CustomProcessing::SetRuntimeSetting(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.h
>index 26880e37760264ddaac9404ee80e7479ec12634d..df51313229da28c1391878ac39a8e94343ab155f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.h
>@@ -34,6 +34,7 @@
> #include "rtc_base/platform_file.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -48,7 +49,6 @@ class ProcessingConfig;
> 
> class EchoDetector;
> class GainControl;
>-class HighPassFilter;
> class LevelEstimator;
> class NoiseSuppression;
> class CustomAudioAnalyzer;
>@@ -240,7 +240,6 @@ class AudioProcessing : public rtc::RefCountInterface {
>   // by changing the default values in the AudioProcessing::Config struct.
>   // The config is applied by passing the struct to the ApplyConfig method.
>   struct Config {
>-    // TODO(bugs.webrtc.org/9535): Currently unused. Use this to determine AEC.
>     struct EchoCanceller {
>       bool enabled = false;
>       bool mobile_mode = false;
>@@ -271,11 +270,24 @@ class AudioProcessing : public rtc::RefCountInterface {
>     // first applies a fixed gain. The adaptive digital AGC can be turned off by
>     // setting |adaptive_digital_mode=false|.
>     struct GainController2 {
>+      enum LevelEstimator { kRms, kPeak };
>       bool enabled = false;
>-      bool adaptive_digital_mode = true;
>-      float fixed_gain_db = 0.f;
>+      struct {
>+        float gain_db = 0.f;
>+      } fixed_digital;
>+      struct {
>+        bool enabled = false;
>+        LevelEstimator level_estimator = kRms;
>+        bool use_saturation_protector = true;
>+        float extra_saturation_margin_db = 2.f;
>+      } adaptive_digital;
>     } gain_controller2;
> 
>+    // Enables reporting of |output_rms_dbfs| in webrtc::AudioProcessingStats.
>+    struct LevelEstimation {
>+      bool enabled = false;
>+    } level_estimation;
>+
>     // Explicit copy assignment implementation to avoid issues with memory
>     // sanitizer complaints in case of self-assignment.
>     // TODO(peah): Add buildflag to ensure that this is only included for memory
>@@ -520,85 +532,17 @@ class AudioProcessing : public rtc::RefCountInterface {
>   // specific member variables are reset.
>   virtual void UpdateHistogramsOnCallEnd() = 0;
> 
>-  // TODO(ivoc): Remove when the calling code no longer uses the old Statistics
>-  //             API.
>-  struct Statistic {
>-    int instant = 0;  // Instantaneous value.
>-    int average = 0;  // Long-term average.
>-    int maximum = 0;  // Long-term maximum.
>-    int minimum = 0;  // Long-term minimum.
>-  };
>-
>-  struct Stat {
>-    void Set(const Statistic& other) {
>-      Set(other.instant, other.average, other.maximum, other.minimum);
>-    }
>-    void Set(float instant, float average, float maximum, float minimum) {
>-      instant_ = instant;
>-      average_ = average;
>-      maximum_ = maximum;
>-      minimum_ = minimum;
>-    }
>-    float instant() const { return instant_; }
>-    float average() const { return average_; }
>-    float maximum() const { return maximum_; }
>-    float minimum() const { return minimum_; }
>-
>-   private:
>-    float instant_ = 0.0f;  // Instantaneous value.
>-    float average_ = 0.0f;  // Long-term average.
>-    float maximum_ = 0.0f;  // Long-term maximum.
>-    float minimum_ = 0.0f;  // Long-term minimum.
>-  };
>-
>-  struct AudioProcessingStatistics {
>-    AudioProcessingStatistics();
>-    AudioProcessingStatistics(const AudioProcessingStatistics& other);
>-    ~AudioProcessingStatistics();
>-
>-    // AEC Statistics.
>-    // RERL = ERL + ERLE
>-    Stat residual_echo_return_loss;
>-    // ERL = 10log_10(P_far / P_echo)
>-    Stat echo_return_loss;
>-    // ERLE = 10log_10(P_echo / P_out)
>-    Stat echo_return_loss_enhancement;
>-    // (Pre non-linear processing suppression) A_NLP = 10log_10(P_echo / P_a)
>-    Stat a_nlp;
>-    // Fraction of time that the AEC linear filter is divergent, in a 1-second
>-    // non-overlapped aggregation window.
>-    float divergent_filter_fraction = -1.0f;
>-
>-    // The delay metrics consists of the delay median and standard deviation. It
>-    // also consists of the fraction of delay estimates that can make the echo
>-    // cancellation perform poorly. The values are aggregated until the first
>-    // call to |GetStatistics()| and afterwards aggregated and updated every
>-    // second. Note that if there are several clients pulling metrics from
>-    // |GetStatistics()| during a session the first call from any of them will
>-    // change to one second aggregation window for all.
>-    int delay_median = -1;
>-    int delay_standard_deviation = -1;
>-    float fraction_poor_delays = -1.0f;
>-
>-    // Residual echo detector likelihood.
>-    float residual_echo_likelihood = -1.0f;
>-    // Maximum residual echo likelihood from the last time period.
>-    float residual_echo_likelihood_recent_max = -1.0f;
>-  };
>-
>-  // TODO(ivoc): Make this pure virtual when all subclasses have been updated.
>-  virtual AudioProcessingStatistics GetStatistics() const;
>-
>-  // This returns the stats as optionals and it will replace the regular
>-  // GetStatistics.
>-  virtual AudioProcessingStats GetStatistics(bool has_remote_tracks) const;
>+  // Get audio processing statistics. The |has_remote_tracks| argument should be
>+  // set if there are active remote tracks (this would usually be true during
>+  // a call). If there are no remote tracks some of the stats will not be set by
>+  // AudioProcessing, because they only make sense if there is at least one
>+  // remote track.
>+  virtual AudioProcessingStats GetStatistics(bool has_remote_tracks) const = 0;
> 
>   // These provide access to the component interfaces and should never return
>   // NULL. The pointers will be valid for the lifetime of the APM instance.
>   // The memory for these objects is entirely managed internally.
>   virtual GainControl* gain_control() const = 0;
>-  // TODO(peah): Deprecate this API call.
>-  virtual HighPassFilter* high_pass_filter() const = 0;
>   virtual LevelEstimator* level_estimator() const = 0;
>   virtual NoiseSuppression* noise_suppression() const = 0;
>   virtual VoiceDetection* voice_detection() const = 0;
>@@ -648,7 +592,7 @@ class AudioProcessing : public rtc::RefCountInterface {
>   static const int kChunkSizeMs = 10;
> };
> 
>-class AudioProcessingBuilder {
>+class RTC_EXPORT AudioProcessingBuilder {
>  public:
>   AudioProcessingBuilder();
>   ~AudioProcessingBuilder();
>@@ -788,17 +732,6 @@ class ProcessingConfig {
>   StreamConfig streams[StreamName::kNumStreamNames];
> };
> 
>-// TODO(peah): Remove this interface.
>-// A filtering component which removes DC offset and low-frequency noise.
>-// Recommended to be enabled on the client-side.
>-class HighPassFilter {
>- public:
>-  virtual int Enable(bool enable) = 0;
>-  virtual bool is_enabled() const = 0;
>-
>-  virtual ~HighPassFilter() {}
>-};
>-
> // An estimation component used to retrieve level metrics.
> class LevelEstimator {
>  public:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing_statistics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing_statistics.h
>index 237d23c5bce8b2bdadd7cba7f4298d1d6e93739c..683db052e6c6b7e69e87035840e1cc6f292c7129 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing_statistics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing_statistics.h
>@@ -11,16 +11,27 @@
> #ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
> #define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
> 
>+#include <stdint.h>
>+
> #include "absl/types/optional.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> // This version of the stats uses Optionals, it will replace the regular
> // AudioProcessingStatistics struct.
>-struct AudioProcessingStats {
>+struct RTC_EXPORT AudioProcessingStats {
>   AudioProcessingStats();
>   AudioProcessingStats(const AudioProcessingStats& other);
>   ~AudioProcessingStats();
> 
>+  // The root mean square (RMS) level in dBFS (decibels from digital
>+  // full-scale) of the last capture frame, after processing. It is
>+  // constrained to [-127, 0].
>+  // The computation follows: https://tools.ietf.org/html/rfc6465
>+  // with the intent that it can provide the RTP audio level indication.
>+  // Only reported if level estimation is enabled in AudioProcessing::Config.
>+  absl::optional<int> output_rms_dbfs;
>+
>   // AEC Statistics.
>   // ERL = 10log_10(P_far / P_echo)
>   absl::optional<double> echo_return_loss;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/config.h
>index 398aab61cffe63dd52f0581d980ac0f67f144da9..e77d3fd16d7b41963de37fa79bf61083dce92d37 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/config.h
>@@ -14,6 +14,7 @@
> #include <map>
> 
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -57,7 +58,7 @@ enum class ConfigOptionID {
> //    config.Set<Algo1_CostFunction>(new SqrCost());
> //
> // Note: This class is thread-compatible (like STL containers).
>-class Config {
>+class RTC_EXPORT Config {
>  public:
>   // Returns the option if set or a default constructed one.
>   // Callers that access options too often are encouraged to cache the result.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/mock_audio_processing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/mock_audio_processing.h
>index 4a1fe626bd5d58b0b85e19e8d1b242f00341d048..f00a16d5e2d2610b171b40c62a13dfb82cac34a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/mock_audio_processing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/mock_audio_processing.h
>@@ -42,13 +42,6 @@ class MockGainControl : public GainControl {
>   MOCK_CONST_METHOD0(stream_is_saturated, bool());
> };
> 
>-class MockHighPassFilter : public HighPassFilter {
>- public:
>-  virtual ~MockHighPassFilter() {}
>-  MOCK_METHOD1(Enable, int(bool enable));
>-  MOCK_CONST_METHOD0(is_enabled, bool());
>-};
>-
> class MockLevelEstimator : public LevelEstimator {
>  public:
>   virtual ~MockLevelEstimator() {}
>@@ -114,7 +107,6 @@ class MockAudioProcessing : public testing::NiceMock<AudioProcessing> {
>  public:
>   MockAudioProcessing()
>       : gain_control_(new testing::NiceMock<MockGainControl>()),
>-        high_pass_filter_(new testing::NiceMock<MockHighPassFilter>()),
>         level_estimator_(new testing::NiceMock<MockLevelEstimator>()),
>         noise_suppression_(new testing::NiceMock<MockNoiseSuppression>()),
>         voice_detection_(new testing::NiceMock<MockVoiceDetection>()) {}
>@@ -180,12 +172,8 @@ class MockAudioProcessing : public testing::NiceMock<AudioProcessing> {
>   MOCK_METHOD0(DetachPlayoutAudioGenerator, void());
> 
>   MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
>-  MOCK_CONST_METHOD0(GetStatistics, AudioProcessingStatistics());
>   MOCK_CONST_METHOD1(GetStatistics, AudioProcessingStats(bool));
>   virtual MockGainControl* gain_control() const { return gain_control_.get(); }
>-  virtual MockHighPassFilter* high_pass_filter() const {
>-    return high_pass_filter_.get();
>-  }
>   virtual MockLevelEstimator* level_estimator() const {
>     return level_estimator_.get();
>   }
>@@ -200,7 +188,6 @@ class MockAudioProcessing : public testing::NiceMock<AudioProcessing> {
> 
>  private:
>   std::unique_ptr<MockGainControl> gain_control_;
>-  std::unique_ptr<MockHighPassFilter> high_pass_filter_;
>   std::unique_ptr<MockLevelEstimator> level_estimator_;
>   std::unique_ptr<MockNoiseSuppression> noise_suppression_;
>   std::unique_ptr<MockVoiceDetection> voice_detection_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_estimator_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_estimator_impl.cc
>index c937f8452534cea733b0c1f622858e7ea5ebe372..5b49b35fdcc5653824e084ae544425dda0f0722e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_estimator_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_estimator_impl.cc
>@@ -10,9 +10,13 @@
> 
> #include "modules/audio_processing/level_estimator_impl.h"
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "api/array_view.h"
> #include "modules/audio_processing/audio_buffer.h"
> #include "modules/audio_processing/rms_level.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
>index e2e8602b989b366bc2a62f67681b508ee53fac5f..cc879c8b08fdc5d7d6cbff9949166e2ad6894125 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
>@@ -11,7 +11,6 @@
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> 
> #include "rtc_base/strings/string_builder.h"
>-#include "rtc_base/stringutils.h"
> 
> // Check to verify that the define is properly set.
> #if !defined(WEBRTC_APM_DEBUG_DUMP) || \
>@@ -20,16 +19,30 @@
> #endif
> 
> namespace webrtc {
>-
> namespace {
> 
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-std::string FormFileName(const char* name,
>+
>+#if defined(WEBRTC_WIN)
>+constexpr char kPathDelimiter = '\\';
>+#else
>+constexpr char kPathDelimiter = '/';
>+#endif
>+
>+std::string FormFileName(const char* output_dir,
>+                         const char* name,
>                          int instance_index,
>                          int reinit_index,
>                          const std::string& suffix) {
>   char buf[1024];
>   rtc::SimpleStringBuilder ss(buf);
>+  const size_t output_dir_size = strlen(output_dir);
>+  if (output_dir_size > 0) {
>+    ss << output_dir;
>+    if (output_dir[output_dir_size - 1] != kPathDelimiter) {
>+      ss << kPathDelimiter;
>+    }
>+  }
>   ss << name << "_" << instance_index << "-" << reinit_index << suffix;
>   return ss.str();
> }
>@@ -41,18 +54,22 @@ std::string FormFileName(const char* name,
> ApmDataDumper::ApmDataDumper(int instance_index)
>     : instance_index_(instance_index) {}
> #else
>-ApmDataDumper::ApmDataDumper(int instance_index) {}
>+ApmDataDumper::ApmDataDumper(int instance_index){};
> #endif
> 
>-ApmDataDumper::~ApmDataDumper() {}
>+ApmDataDumper::~ApmDataDumper() = default;
> 
> #if WEBRTC_APM_DEBUG_DUMP == 1
>+bool ApmDataDumper::recording_activated_ = false;
>+char ApmDataDumper::output_dir_[] = "";
>+
> FILE* ApmDataDumper::GetRawFile(const char* name) {
>-  std::string filename =
>-      FormFileName(name, instance_index_, recording_set_index_, ".dat");
>+  std::string filename = FormFileName(output_dir_, name, instance_index_,
>+                                      recording_set_index_, ".dat");
>   auto& f = raw_files_[filename];
>   if (!f) {
>     f.reset(fopen(filename.c_str(), "wb"));
>+    RTC_CHECK(f.get()) << "Cannot write to " << filename << ".";
>   }
>   return f.get();
> }
>@@ -60,15 +77,14 @@ FILE* ApmDataDumper::GetRawFile(const char* name) {
> WavWriter* ApmDataDumper::GetWavFile(const char* name,
>                                      int sample_rate_hz,
>                                      int num_channels) {
>-  std::string filename =
>-      FormFileName(name, instance_index_, recording_set_index_, ".wav");
>+  std::string filename = FormFileName(output_dir_, name, instance_index_,
>+                                      recording_set_index_, ".wav");
>   auto& f = wav_files_[filename];
>   if (!f) {
>     f.reset(new WavWriter(filename.c_str(), sample_rate_hz, num_channels));
>   }
>   return f.get();
> }
>-
> #endif
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.h
>index d045027b354165a02419ba9d44b768f6313efc2a..b541ae84b9b86a7494aef73d797079419963bc1f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.h
>@@ -11,14 +11,20 @@
> #ifndef MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
> #define MODULES_AUDIO_PROCESSING_LOGGING_APM_DATA_DUMPER_H_
> 
>+#include <stdint.h>
> #include <stdio.h>
>+#include <string.h>
> 
>-#include <memory>
> #include <string>
>+#if WEBRTC_APM_DEBUG_DUMP == 1
> #include <unordered_map>
>+#endif
> 
> #include "api/array_view.h"
>+#if WEBRTC_APM_DEBUG_DUMP == 1
> #include "common_audio/wav_file.h"
>+#include "rtc_base/checks.h"
>+#endif
> #include "rtc_base/constructormagic.h"
> 
> // Check to verify that the define is properly set.
>@@ -47,6 +53,21 @@ class ApmDataDumper {
> 
>   ~ApmDataDumper();
> 
>+  // Activates or deactivate the dumping functionality.
>+  static void SetActivated(bool activated) {
>+#if WEBRTC_APM_DEBUG_DUMP == 1
>+    recording_activated_ = activated;
>+#endif
>+  }
>+
>+  // Set an optional output directory.
>+  static void SetOutputDirectory(const std::string& output_dir) {
>+#if WEBRTC_APM_DEBUG_DUMP == 1
>+    RTC_CHECK_LT(output_dir.size(), kOutputDirMaxLength);
>+    strncpy(output_dir_, output_dir.c_str(), output_dir.size());
>+#endif
>+  }
>+
>   // Reinitializes the data dumping such that new versions
>   // of all files being dumped to are created.
>   void InitiateNewSetOfRecordings() {
>@@ -59,115 +80,155 @@ class ApmDataDumper {
>   // various formats.
>   void DumpRaw(const char* name, double v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    FILE* file = GetRawFile(name);
>-    fwrite(&v, sizeof(v), 1, file);
>+    if (recording_activated_) {
>+      FILE* file = GetRawFile(name);
>+      fwrite(&v, sizeof(v), 1, file);
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, size_t v_length, const double* v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    FILE* file = GetRawFile(name);
>-    fwrite(v, sizeof(v[0]), v_length, file);
>+    if (recording_activated_) {
>+      FILE* file = GetRawFile(name);
>+      fwrite(v, sizeof(v[0]), v_length, file);
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, rtc::ArrayView<const double> v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    DumpRaw(name, v.size(), v.data());
>+    if (recording_activated_) {
>+      DumpRaw(name, v.size(), v.data());
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, float v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    FILE* file = GetRawFile(name);
>-    fwrite(&v, sizeof(v), 1, file);
>+    if (recording_activated_) {
>+      FILE* file = GetRawFile(name);
>+      fwrite(&v, sizeof(v), 1, file);
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, size_t v_length, const float* v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    FILE* file = GetRawFile(name);
>-    fwrite(v, sizeof(v[0]), v_length, file);
>+    if (recording_activated_) {
>+      FILE* file = GetRawFile(name);
>+      fwrite(v, sizeof(v[0]), v_length, file);
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, rtc::ArrayView<const float> v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    DumpRaw(name, v.size(), v.data());
>+    if (recording_activated_) {
>+      DumpRaw(name, v.size(), v.data());
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, bool v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    DumpRaw(name, static_cast<int16_t>(v));
>+    if (recording_activated_) {
>+      DumpRaw(name, static_cast<int16_t>(v));
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, size_t v_length, const bool* v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    FILE* file = GetRawFile(name);
>-    for (size_t k = 0; k < v_length; ++k) {
>-      int16_t value = static_cast<int16_t>(v[k]);
>-      fwrite(&value, sizeof(value), 1, file);
>+    if (recording_activated_) {
>+      FILE* file = GetRawFile(name);
>+      for (size_t k = 0; k < v_length; ++k) {
>+        int16_t value = static_cast<int16_t>(v[k]);
>+        fwrite(&value, sizeof(value), 1, file);
>+      }
>     }
> #endif
>   }
> 
>   void DumpRaw(const char* name, rtc::ArrayView<const bool> v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    DumpRaw(name, v.size(), v.data());
>+    if (recording_activated_) {
>+      DumpRaw(name, v.size(), v.data());
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, int16_t v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    FILE* file = GetRawFile(name);
>-    fwrite(&v, sizeof(v), 1, file);
>+    if (recording_activated_) {
>+      FILE* file = GetRawFile(name);
>+      fwrite(&v, sizeof(v), 1, file);
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, size_t v_length, const int16_t* v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    FILE* file = GetRawFile(name);
>-    fwrite(v, sizeof(v[0]), v_length, file);
>+    if (recording_activated_) {
>+      FILE* file = GetRawFile(name);
>+      fwrite(v, sizeof(v[0]), v_length, file);
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, rtc::ArrayView<const int16_t> v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    DumpRaw(name, v.size(), v.data());
>+    if (recording_activated_) {
>+      DumpRaw(name, v.size(), v.data());
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, int32_t v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    FILE* file = GetRawFile(name);
>-    fwrite(&v, sizeof(v), 1, file);
>+    if (recording_activated_) {
>+      FILE* file = GetRawFile(name);
>+      fwrite(&v, sizeof(v), 1, file);
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, size_t v_length, const int32_t* v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    FILE* file = GetRawFile(name);
>-    fwrite(v, sizeof(v[0]), v_length, file);
>+    if (recording_activated_) {
>+      FILE* file = GetRawFile(name);
>+      fwrite(v, sizeof(v[0]), v_length, file);
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, size_t v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    FILE* file = GetRawFile(name);
>-    fwrite(&v, sizeof(v), 1, file);
>+    if (recording_activated_) {
>+      FILE* file = GetRawFile(name);
>+      fwrite(&v, sizeof(v), 1, file);
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, size_t v_length, const size_t* v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    FILE* file = GetRawFile(name);
>-    fwrite(v, sizeof(v[0]), v_length, file);
>+    if (recording_activated_) {
>+      FILE* file = GetRawFile(name);
>+      fwrite(v, sizeof(v[0]), v_length, file);
>+    }
> #endif
>   }
> 
>   void DumpRaw(const char* name, rtc::ArrayView<const int32_t> v) {
>+#if WEBRTC_APM_DEBUG_DUMP == 1
>+    if (recording_activated_) {
>+      DumpRaw(name, v.size(), v.data());
>+    }
>+#endif
>+  }
>+
>+  void DumpRaw(const char* name, rtc::ArrayView<const size_t> v) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>     DumpRaw(name, v.size(), v.data());
> #endif
>@@ -179,8 +240,10 @@ class ApmDataDumper {
>                int sample_rate_hz,
>                int num_channels) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels);
>-    file->WriteSamples(v, v_length);
>+    if (recording_activated_) {
>+      WavWriter* file = GetWavFile(name, sample_rate_hz, num_channels);
>+      file->WriteSamples(v, v_length);
>+    }
> #endif
>   }
> 
>@@ -189,12 +252,17 @@ class ApmDataDumper {
>                int sample_rate_hz,
>                int num_channels) {
> #if WEBRTC_APM_DEBUG_DUMP == 1
>-    DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
>+    if (recording_activated_) {
>+      DumpWav(name, v.size(), v.data(), sample_rate_hz, num_channels);
>+    }
> #endif
>   }
> 
>  private:
> #if WEBRTC_APM_DEBUG_DUMP == 1
>+  static bool recording_activated_;
>+  static constexpr size_t kOutputDirMaxLength = 1024;
>+  static char output_dir_[kOutputDirMaxLength];
>   const int instance_index_;
>   int recording_set_index_ = 0;
>   std::unordered_map<std::string, std::unique_ptr<FILE, RawFileCloseFunctor>>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/low_cut_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/low_cut_filter.cc
>index 5245c68d02a419e6f6560550e6ebd09de1ffdbb4..581d6bc42a88030f6af20071318dad9ad76807a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/low_cut_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/low_cut_filter.cc
>@@ -10,8 +10,13 @@
> 
> #include "modules/audio_processing/low_cut_filter.h"
> 
>+#include <stdint.h>
>+#include <cstring>
>+
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "modules/audio_processing/audio_buffer.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_impl.cc
>index 15d404323c36f88eff1f1618e0421a0f65aac1cf..d8d9e32340b1ccf12eea7f7ccbd3575f142ddd1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_impl.cc
>@@ -11,9 +11,11 @@
> #include "modules/audio_processing/noise_suppression_impl.h"
> 
> #include "modules/audio_processing/audio_buffer.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #if defined(WEBRTC_NS_FLOAT)
> #include "modules/audio_processing/ns/noise_suppression.h"
>+
> #define NS_CREATE WebRtcNs_Create
> #define NS_FREE WebRtcNs_Free
> #define NS_INIT WebRtcNs_Init
>@@ -21,6 +23,7 @@
> typedef NsHandle NsState;
> #elif defined(WEBRTC_NS_FIXED)
> #include "modules/audio_processing/ns/noise_suppression_x.h"
>+
> #define NS_CREATE WebRtcNsx_Create
> #define NS_FREE WebRtcNsx_Free
> #define NS_INIT WebRtcNsx_Init
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.cc
>index e80501358eaaafe241118c7cbfbc8248f9d466e1..3454214fd739d6dd36f0aafc0b8f2503ad5ba9a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.cc
>@@ -13,9 +13,11 @@
> #include <algorithm>
> #include <numeric>
> 
>+#include "absl/types/optional.h"
> #include "modules/audio_processing/audio_buffer.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/metrics.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.h
>index 9aa549a57c1e336a4ec6d793437e3bfaf3eb4870..e6b5849ead8bde88dcc2f717c306b6b20b266fdc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.h
>@@ -11,6 +11,9 @@
> #ifndef MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
> #define MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "absl/types/optional.h"
> #include "api/array_view.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
>index 65f3c87497c85cda635a4f26e86806c5004ec8f3..df9f4976be27d0ae03c8053b0052cd0dbe37eca0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
>@@ -18,12 +18,14 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>+#include "api/audio/echo_canceller3_config_json.h"
> #include "api/audio/echo_canceller3_factory.h"
> #include "common_audio/include/audio_util.h"
> #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
> #include "modules/audio_processing/echo_cancellation_impl.h"
> #include "modules/audio_processing/echo_control_mobile_impl.h"
> #include "modules/audio_processing/include/audio_processing.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "modules/audio_processing/test/fake_recording_device.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>@@ -34,603 +36,32 @@
> namespace webrtc {
> namespace test {
> namespace {
>-
>-// Prints out the currently used AEC3 parameter values in JSON format.
>-void PrintAec3ParameterValues(const EchoCanceller3Config& cfg) {
>-  std::cout << "{";
>-  std::cout << "\"delay\": {";
>-  std::cout << "\"default_delay\": " << cfg.delay.default_delay << ",";
>-  std::cout << "\"down_sampling_factor\": " << cfg.delay.down_sampling_factor
>-            << ",";
>-  std::cout << "\"num_filters\": " << cfg.delay.num_filters << ",";
>-  std::cout << "\"api_call_jitter_blocks\": "
>-            << cfg.delay.api_call_jitter_blocks << ",";
>-  std::cout << "\"min_echo_path_delay_blocks\": "
>-            << cfg.delay.min_echo_path_delay_blocks << ",";
>-  std::cout << "\"delay_headroom_blocks\": " << cfg.delay.delay_headroom_blocks
>-            << ",";
>-  std::cout << "\"hysteresis_limit_1_blocks\": "
>-            << cfg.delay.hysteresis_limit_1_blocks << ",";
>-  std::cout << "\"hysteresis_limit_2_blocks\": "
>-            << cfg.delay.hysteresis_limit_2_blocks << ",";
>-  std::cout << "\"skew_hysteresis_blocks\": "
>-            << cfg.delay.skew_hysteresis_blocks << ",";
>-  std::cout << "\"fixed_capture_delay_samples\": "
>-            << cfg.delay.fixed_capture_delay_samples << ",";
>-  std::cout << "\"delay_estimate_smoothing\": "
>-            << cfg.delay.delay_estimate_smoothing << ",";
>-  std::cout << "\"delay_candidate_detection_threshold\": "
>-            << cfg.delay.delay_candidate_detection_threshold << ",";
>-
>-  std::cout << "\"delay_selection_thresholds\": {";
>-  std::cout << "\"initial\": " << cfg.delay.delay_selection_thresholds.initial
>-            << ",";
>-  std::cout << "\"converged\": "
>-            << cfg.delay.delay_selection_thresholds.converged;
>-  std::cout << "}";
>-
>-  std::cout << "},";
>-
>-  std::cout << "\"filter\": {";
>-  std::cout << "\"main\": [";
>-  std::cout << cfg.filter.main.length_blocks << ",";
>-  std::cout << cfg.filter.main.leakage_converged << ",";
>-  std::cout << cfg.filter.main.leakage_diverged << ",";
>-  std::cout << cfg.filter.main.error_floor << ",";
>-  std::cout << cfg.filter.main.noise_gate;
>-  std::cout << "],";
>-
>-  std::cout << "\"shadow\": [";
>-  std::cout << cfg.filter.shadow.length_blocks << ",";
>-  std::cout << cfg.filter.shadow.rate << ",";
>-  std::cout << cfg.filter.shadow.noise_gate;
>-  std::cout << "],";
>-
>-  std::cout << "\"main_initial\": [";
>-  std::cout << cfg.filter.main_initial.length_blocks << ",";
>-  std::cout << cfg.filter.main_initial.leakage_converged << ",";
>-  std::cout << cfg.filter.main_initial.leakage_diverged << ",";
>-  std::cout << cfg.filter.main_initial.error_floor << ",";
>-  std::cout << cfg.filter.main_initial.noise_gate;
>-  std::cout << "],";
>-
>-  std::cout << "\"shadow_initial\": [";
>-  std::cout << cfg.filter.shadow_initial.length_blocks << ",";
>-  std::cout << cfg.filter.shadow_initial.rate << ",";
>-  std::cout << cfg.filter.shadow_initial.noise_gate;
>-  std::cout << "],";
>-
>-  std::cout << "\"config_change_duration_blocks\": "
>-            << cfg.filter.config_change_duration_blocks << ",";
>-  std::cout << "\"initial_state_seconds\": " << cfg.filter.initial_state_seconds
>-            << ",";
>-  std::cout << "\"conservative_initial_phase\": "
>-            << (cfg.filter.conservative_initial_phase ? "true" : "false")
>-            << ",";
>-  std::cout << "\"enable_shadow_filter_output_usage\": "
>-            << (cfg.filter.enable_shadow_filter_output_usage ? "true"
>-                                                             : "false");
>-
>-  std::cout << "},";
>-
>-  std::cout << "\"erle\": {";
>-  std::cout << "\"min\": " << cfg.erle.min << ",";
>-  std::cout << "\"max_l\": " << cfg.erle.max_l << ",";
>-  std::cout << "\"max_h\": " << cfg.erle.max_h << ",";
>-  std::cout << "\"onset_detection\": "
>-            << (cfg.erle.onset_detection ? "true" : "false");
>-  std::cout << "},";
>-
>-  std::cout << "\"ep_strength\": {";
>-  std::cout << "\"lf\": " << cfg.ep_strength.lf << ",";
>-  std::cout << "\"mf\": " << cfg.ep_strength.mf << ",";
>-  std::cout << "\"hf\": " << cfg.ep_strength.hf << ",";
>-  std::cout << "\"default_len\": " << cfg.ep_strength.default_len << ",";
>-  std::cout << "\"reverb_based_on_render\": "
>-            << (cfg.ep_strength.reverb_based_on_render ? "true" : "false")
>-            << ",";
>-  std::cout << "\"echo_can_saturate\": "
>-            << (cfg.ep_strength.echo_can_saturate ? "true" : "false") << ",";
>-  std::cout << "\"bounded_erl\": "
>-            << (cfg.ep_strength.bounded_erl ? "true" : "false");
>-
>-  std::cout << "},";
>-
>-  std::cout << "\"gain_mask\": {";
>-  std::cout << "\"m0\": " << cfg.gain_mask.m0 << ",";
>-  std::cout << "\"m1\": " << cfg.gain_mask.m1 << ",";
>-  std::cout << "\"m2\": " << cfg.gain_mask.m2 << ",";
>-  std::cout << "\"m3\": " << cfg.gain_mask.m3 << ",";
>-  std::cout << "\"m5\": " << cfg.gain_mask.m5 << ",";
>-  std::cout << "\"m6\": " << cfg.gain_mask.m6 << ",";
>-  std::cout << "\"m7\": " << cfg.gain_mask.m7 << ",";
>-  std::cout << "\"m8\": " << cfg.gain_mask.m8 << ",";
>-  std::cout << "\"m9\": " << cfg.gain_mask.m9 << ",";
>-  std::cout << "\"gain_curve_offset\": " << cfg.gain_mask.gain_curve_offset
>-            << ",";
>-  std::cout << "\"gain_curve_slope\": " << cfg.gain_mask.gain_curve_slope
>-            << ",";
>-  std::cout << "\"temporal_masking_lf\": " << cfg.gain_mask.temporal_masking_lf
>-            << ",";
>-  std::cout << "\"temporal_masking_hf\": " << cfg.gain_mask.temporal_masking_hf
>-            << ",";
>-  std::cout << "\"temporal_masking_lf_bands\": "
>-            << cfg.gain_mask.temporal_masking_lf_bands;
>-  std::cout << "},";
>-
>-  std::cout << "\"echo_audibility\": {";
>-  std::cout << "\"low_render_limit\": " << cfg.echo_audibility.low_render_limit
>-            << ",";
>-  std::cout << "\"normal_render_limit\": "
>-            << cfg.echo_audibility.normal_render_limit << ",";
>-  std::cout << "\"floor_power\": " << cfg.echo_audibility.floor_power << ",";
>-  std::cout << "\"audibility_threshold_lf\": "
>-            << cfg.echo_audibility.audibility_threshold_lf << ",";
>-  std::cout << "\"audibility_threshold_mf\": "
>-            << cfg.echo_audibility.audibility_threshold_mf << ",";
>-  std::cout << "\"audibility_threshold_hf\": "
>-            << cfg.echo_audibility.audibility_threshold_hf << ",";
>-  std::cout << "\"use_stationary_properties\": "
>-            << (cfg.echo_audibility.use_stationary_properties ? "true"
>-                                                              : "false")
>-            << ",";
>-  std::cout << "\"use_stationarity_properties_at_init\": "
>-            << (cfg.echo_audibility.use_stationarity_properties_at_init
>-                    ? "true"
>-                    : "false");
>-  std::cout << "},";
>-
>-  std::cout << "\"render_levels\": {";
>-  std::cout << "\"active_render_limit\": "
>-            << cfg.render_levels.active_render_limit << ",";
>-  std::cout << "\"poor_excitation_render_limit\": "
>-            << cfg.render_levels.poor_excitation_render_limit << ",";
>-  std::cout << "\"poor_excitation_render_limit_ds8\": "
>-            << cfg.render_levels.poor_excitation_render_limit_ds8;
>-  std::cout << "},";
>-
>-  std::cout << "\"echo_removal_control\": {";
>-  std::cout << "\"gain_rampup\": {";
>-  std::cout << "\"initial_gain\": "
>-            << cfg.echo_removal_control.gain_rampup.initial_gain << ",";
>-  std::cout << "\"first_non_zero_gain\": "
>-            << cfg.echo_removal_control.gain_rampup.first_non_zero_gain << ",";
>-  std::cout << "\"non_zero_gain_blocks\": "
>-            << cfg.echo_removal_control.gain_rampup.non_zero_gain_blocks << ",";
>-  std::cout << "\"full_gain_blocks\": "
>-            << cfg.echo_removal_control.gain_rampup.full_gain_blocks;
>-  std::cout << "},";
>-  std::cout << "\"has_clock_drift\": "
>-            << (cfg.echo_removal_control.has_clock_drift ? "true" : "false")
>-            << ",";
>-  std::cout << "\"linear_and_stable_echo_path\": "
>-            << (cfg.echo_removal_control.linear_and_stable_echo_path ? "true"
>-                                                                     : "false");
>-
>-  std::cout << "},";
>-
>-  std::cout << "\"echo_model\": {";
>-  std::cout << "\"noise_floor_hold\": " << cfg.echo_model.noise_floor_hold
>-            << ",";
>-  std::cout << "\"min_noise_floor_power\": "
>-            << cfg.echo_model.min_noise_floor_power << ",";
>-  std::cout << "\"stationary_gate_slope\": "
>-            << cfg.echo_model.stationary_gate_slope << ",";
>-  std::cout << "\"noise_gate_power\": " << cfg.echo_model.noise_gate_power
>-            << ",";
>-  std::cout << "\"noise_gate_slope\": " << cfg.echo_model.noise_gate_slope
>-            << ",";
>-  std::cout << "\"render_pre_window_size\": "
>-            << cfg.echo_model.render_pre_window_size << ",";
>-  std::cout << "\"render_post_window_size\": "
>-            << cfg.echo_model.render_post_window_size << ",";
>-  std::cout << "\"render_pre_window_size_init\": "
>-            << cfg.echo_model.render_pre_window_size_init << ",";
>-  std::cout << "\"render_post_window_size_init\": "
>-            << cfg.echo_model.render_post_window_size_init << ",";
>-  std::cout << "\"nonlinear_hold\": " << cfg.echo_model.nonlinear_hold << ",";
>-  std::cout << "\"nonlinear_release\": " << cfg.echo_model.nonlinear_release;
>-  std::cout << "},";
>-
>-  std::cout << "\"suppressor\": {";
>-  std::cout << "\"nearend_average_blocks\": "
>-            << cfg.suppressor.nearend_average_blocks << ",";
>-  std::cout << "\"normal_tuning\": {";
>-  std::cout << "\"mask_lf\": [";
>-  std::cout << cfg.suppressor.normal_tuning.mask_lf.enr_transparent << ",";
>-  std::cout << cfg.suppressor.normal_tuning.mask_lf.enr_suppress << ",";
>-  std::cout << cfg.suppressor.normal_tuning.mask_lf.emr_transparent;
>-  std::cout << "],";
>-  std::cout << "\"mask_hf\": [";
>-  std::cout << cfg.suppressor.normal_tuning.mask_hf.enr_transparent << ",";
>-  std::cout << cfg.suppressor.normal_tuning.mask_hf.enr_suppress << ",";
>-  std::cout << cfg.suppressor.normal_tuning.mask_hf.emr_transparent;
>-  std::cout << "],";
>-  std::cout << "\"max_inc_factor\": "
>-            << cfg.suppressor.normal_tuning.max_inc_factor << ",";
>-  std::cout << "\"max_dec_factor_lf\": "
>-            << cfg.suppressor.normal_tuning.max_dec_factor_lf;
>-  std::cout << "},";
>-  std::cout << "\"nearend_tuning\": {";
>-  std::cout << "\"mask_lf\": [";
>-  std::cout << cfg.suppressor.nearend_tuning.mask_lf.enr_transparent << ",";
>-  std::cout << cfg.suppressor.nearend_tuning.mask_lf.enr_suppress << ",";
>-  std::cout << cfg.suppressor.nearend_tuning.mask_lf.emr_transparent;
>-  std::cout << "],";
>-  std::cout << "\"mask_hf\": [";
>-  std::cout << cfg.suppressor.nearend_tuning.mask_hf.enr_transparent << ",";
>-  std::cout << cfg.suppressor.nearend_tuning.mask_hf.enr_suppress << ",";
>-  std::cout << cfg.suppressor.nearend_tuning.mask_hf.emr_transparent;
>-  std::cout << "],";
>-  std::cout << "\"max_inc_factor\": "
>-            << cfg.suppressor.nearend_tuning.max_inc_factor << ",";
>-  std::cout << "\"max_dec_factor_lf\": "
>-            << cfg.suppressor.nearend_tuning.max_dec_factor_lf;
>-  std::cout << "},";
>-  std::cout << "\"dominant_nearend_detection\": {";
>-  std::cout << "\"enr_threshold\": "
>-            << cfg.suppressor.dominant_nearend_detection.enr_threshold << ",";
>-  std::cout << "\"snr_threshold\": "
>-            << cfg.suppressor.dominant_nearend_detection.snr_threshold << ",";
>-  std::cout << "\"hold_duration\": "
>-            << cfg.suppressor.dominant_nearend_detection.hold_duration << ",";
>-  std::cout << "\"trigger_threshold\": "
>-            << cfg.suppressor.dominant_nearend_detection.trigger_threshold;
>-  std::cout << "},";
>-  std::cout << "\"high_bands_suppression\": {";
>-  std::cout << "\"enr_threshold\": "
>-            << cfg.suppressor.high_bands_suppression.enr_threshold << ",";
>-  std::cout << "\"max_gain_during_echo\": "
>-            << cfg.suppressor.high_bands_suppression.max_gain_during_echo;
>-  std::cout << "},";
>-  std::cout << "\"floor_first_increase\": "
>-            << cfg.suppressor.floor_first_increase << ",";
>-  std::cout << "\"enforce_transparent\": "
>-            << (cfg.suppressor.enforce_transparent ? "true" : "false") << ",";
>-  std::cout << "\"enforce_empty_higher_bands\": "
>-            << (cfg.suppressor.enforce_empty_higher_bands ? "true" : "false");
>-  std::cout << "}";
>-  std::cout << "}";
>-  std::cout << std::endl;
>+// Helper for reading JSON from a file and parsing it to an AEC3 configuration.
>+EchoCanceller3Config ReadAec3ConfigFromJsonFile(const std::string& filename) {
>+  std::string json_string;
>+  std::string s;
>+  std::ifstream f(filename.c_str());
>+  if (f.fail()) {
>+    std::cout << "Failed to open the file " << filename << std::endl;
>+    RTC_CHECK(false);
>+  }
>+  while (std::getline(f, s)) {
>+    json_string += s;
>+  }
>+
>+  bool parsing_successful;
>+  EchoCanceller3Config cfg;
>+  Aec3ConfigFromJsonString(json_string, &cfg, &parsing_successful);
>+  if (!parsing_successful) {
>+    std::cout << "Parsing of json string failed: " << std::endl
>+              << json_string << std::endl;
>+    RTC_CHECK(false);
>+  }
>+  RTC_CHECK(EchoCanceller3Config::Validate(&cfg));
>+
>+  return cfg;
> }
> 
>-// Class for parsing the AEC3 parameters from a JSON file and producing a config
>-// struct.
>-class Aec3ParametersParser {
>- public:
>-  static EchoCanceller3Config Parse(const std::string& filename) {
>-    return Aec3ParametersParser().ParseFile(filename);
>-  }
>-
>- private:
>-  Aec3ParametersParser() = default;
>-
>-  void ReadParam(const Json::Value& root,
>-                 std::string param_name,
>-                 bool* param) const {
>-    RTC_CHECK(param);
>-    bool v;
>-    if (rtc::GetBoolFromJsonObject(root, param_name, &v)) {
>-      *param = v;
>-    }
>-  }
>-
>-  void ReadParam(const Json::Value& root,
>-                 std::string param_name,
>-                 size_t* param) const {
>-    RTC_CHECK(param);
>-    int v;
>-    if (rtc::GetIntFromJsonObject(root, param_name, &v)) {
>-      *param = v;
>-    }
>-  }
>-
>-  void ReadParam(const Json::Value& root,
>-                 std::string param_name,
>-                 int* param) const {
>-    RTC_CHECK(param);
>-    int v;
>-    if (rtc::GetIntFromJsonObject(root, param_name, &v)) {
>-      *param = v;
>-    }
>-  }
>-
>-  void ReadParam(const Json::Value& root,
>-                 std::string param_name,
>-                 float* param) const {
>-    RTC_CHECK(param);
>-    double v;
>-    if (rtc::GetDoubleFromJsonObject(root, param_name, &v)) {
>-      *param = static_cast<float>(v);
>-    }
>-  }
>-
>-  void ReadParam(const Json::Value& root,
>-                 std::string param_name,
>-                 EchoCanceller3Config::Filter::MainConfiguration* param) const {
>-    RTC_CHECK(param);
>-    Json::Value json_array;
>-    if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
>-      std::vector<double> v;
>-      rtc::JsonArrayToDoubleVector(json_array, &v);
>-      if (v.size() != 5) {
>-        std::cout << "Incorrect array size for " << param_name << std::endl;
>-        RTC_CHECK(false);
>-      }
>-      param->length_blocks = static_cast<size_t>(v[0]);
>-      param->leakage_converged = static_cast<float>(v[1]);
>-      param->leakage_diverged = static_cast<float>(v[2]);
>-      param->error_floor = static_cast<float>(v[3]);
>-      param->noise_gate = static_cast<float>(v[4]);
>-    }
>-  }
>-
>-  void ReadParam(
>-      const Json::Value& root,
>-      std::string param_name,
>-      EchoCanceller3Config::Filter::ShadowConfiguration* param) const {
>-    RTC_CHECK(param);
>-    Json::Value json_array;
>-    if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
>-      std::vector<double> v;
>-      rtc::JsonArrayToDoubleVector(json_array, &v);
>-      if (v.size() != 3) {
>-        std::cout << "Incorrect array size for " << param_name << std::endl;
>-        RTC_CHECK(false);
>-      }
>-      param->length_blocks = static_cast<size_t>(v[0]);
>-      param->rate = static_cast<float>(v[1]);
>-      param->noise_gate = static_cast<float>(v[2]);
>-    }
>-  }
>-
>-  void ReadParam(
>-      const Json::Value& root,
>-      std::string param_name,
>-      EchoCanceller3Config::Suppressor::MaskingThresholds* param) const {
>-    RTC_CHECK(param);
>-    Json::Value json_array;
>-    if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
>-      std::vector<double> v;
>-      rtc::JsonArrayToDoubleVector(json_array, &v);
>-      if (v.size() != 3) {
>-        std::cout << "Incorrect array size for " << param_name << std::endl;
>-        RTC_CHECK(false);
>-      }
>-      param->enr_transparent = static_cast<float>(v[0]);
>-      param->enr_suppress = static_cast<float>(v[1]);
>-      param->emr_transparent = static_cast<float>(v[2]);
>-    }
>-  }
>-
>-  EchoCanceller3Config ParseFile(const std::string& filename) const {
>-    EchoCanceller3Config cfg;
>-    Json::Value root;
>-    std::string s;
>-    std::string json_string;
>-    std::ifstream f(filename.c_str());
>-
>-    if (f.fail()) {
>-      std::cout << "Failed to open the file " << filename << std::endl;
>-      RTC_CHECK(false);
>-    }
>-
>-    while (std::getline(f, s)) {
>-      json_string += s;
>-    }
>-    bool success = Json::Reader().parse(json_string, root);
>-    if (!success) {
>-      std::cout << "Incorrect JSON format:" << std::endl;
>-      std::cout << json_string << std::endl;
>-      RTC_CHECK(false);
>-    }
>-
>-    Json::Value section;
>-    if (rtc::GetValueFromJsonObject(root, "delay", &section)) {
>-      ReadParam(section, "default_delay", &cfg.delay.default_delay);
>-      ReadParam(section, "down_sampling_factor",
>-                &cfg.delay.down_sampling_factor);
>-      ReadParam(section, "num_filters", &cfg.delay.num_filters);
>-      ReadParam(section, "api_call_jitter_blocks",
>-                &cfg.delay.api_call_jitter_blocks);
>-      ReadParam(section, "min_echo_path_delay_blocks",
>-                &cfg.delay.min_echo_path_delay_blocks);
>-      ReadParam(section, "delay_headroom_blocks",
>-                &cfg.delay.delay_headroom_blocks);
>-      ReadParam(section, "hysteresis_limit_1_blocks",
>-                &cfg.delay.hysteresis_limit_1_blocks);
>-      ReadParam(section, "hysteresis_limit_2_blocks",
>-                &cfg.delay.hysteresis_limit_2_blocks);
>-      ReadParam(section, "skew_hysteresis_blocks",
>-                &cfg.delay.skew_hysteresis_blocks);
>-      ReadParam(section, "fixed_capture_delay_samples",
>-                &cfg.delay.fixed_capture_delay_samples);
>-      ReadParam(section, "delay_estimate_smoothing",
>-                &cfg.delay.delay_estimate_smoothing);
>-      ReadParam(section, "delay_candidate_detection_threshold",
>-                &cfg.delay.delay_candidate_detection_threshold);
>-
>-      Json::Value subsection;
>-      if (rtc::GetValueFromJsonObject(section, "delay_selection_thresholds",
>-                                      &subsection)) {
>-        ReadParam(subsection, "initial",
>-                  &cfg.delay.delay_selection_thresholds.initial);
>-        ReadParam(subsection, "converged",
>-                  &cfg.delay.delay_selection_thresholds.converged);
>-      }
>-    }
>-
>-    if (rtc::GetValueFromJsonObject(root, "filter", &section)) {
>-      ReadParam(section, "main", &cfg.filter.main);
>-      ReadParam(section, "shadow", &cfg.filter.shadow);
>-      ReadParam(section, "main_initial", &cfg.filter.main_initial);
>-      ReadParam(section, "shadow_initial", &cfg.filter.shadow_initial);
>-      ReadParam(section, "config_change_duration_blocks",
>-                &cfg.filter.config_change_duration_blocks);
>-      ReadParam(section, "initial_state_seconds",
>-                &cfg.filter.initial_state_seconds);
>-      ReadParam(section, "conservative_initial_phase",
>-                &cfg.filter.conservative_initial_phase);
>-      ReadParam(section, "enable_shadow_filter_output_usage",
>-                &cfg.filter.enable_shadow_filter_output_usage);
>-    }
>-
>-    if (rtc::GetValueFromJsonObject(root, "erle", &section)) {
>-      ReadParam(section, "min", &cfg.erle.min);
>-      ReadParam(section, "max_l", &cfg.erle.max_l);
>-      ReadParam(section, "max_h", &cfg.erle.max_h);
>-      ReadParam(section, "onset_detection", &cfg.erle.onset_detection);
>-    }
>-
>-    if (rtc::GetValueFromJsonObject(root, "ep_strength", &section)) {
>-      ReadParam(section, "lf", &cfg.ep_strength.lf);
>-      ReadParam(section, "mf", &cfg.ep_strength.mf);
>-      ReadParam(section, "hf", &cfg.ep_strength.hf);
>-      ReadParam(section, "default_len", &cfg.ep_strength.default_len);
>-      ReadParam(section, "reverb_based_on_render",
>-                &cfg.ep_strength.reverb_based_on_render);
>-      ReadParam(section, "echo_can_saturate",
>-                &cfg.ep_strength.echo_can_saturate);
>-      ReadParam(section, "bounded_erl", &cfg.ep_strength.bounded_erl);
>-    }
>-
>-    if (rtc::GetValueFromJsonObject(root, "gain_mask", &section)) {
>-      ReadParam(section, "m1", &cfg.gain_mask.m1);
>-      ReadParam(section, "m2", &cfg.gain_mask.m2);
>-      ReadParam(section, "m3", &cfg.gain_mask.m3);
>-      ReadParam(section, "m5", &cfg.gain_mask.m5);
>-      ReadParam(section, "m6", &cfg.gain_mask.m6);
>-      ReadParam(section, "m7", &cfg.gain_mask.m7);
>-      ReadParam(section, "m8", &cfg.gain_mask.m8);
>-      ReadParam(section, "m9", &cfg.gain_mask.m9);
>-
>-      ReadParam(section, "gain_curve_offset", &cfg.gain_mask.gain_curve_offset);
>-      ReadParam(section, "gain_curve_slope", &cfg.gain_mask.gain_curve_slope);
>-      ReadParam(section, "temporal_masking_lf",
>-                &cfg.gain_mask.temporal_masking_lf);
>-      ReadParam(section, "temporal_masking_hf",
>-                &cfg.gain_mask.temporal_masking_hf);
>-      ReadParam(section, "temporal_masking_lf_bands",
>-                &cfg.gain_mask.temporal_masking_lf_bands);
>-    }
>-
>-    if (rtc::GetValueFromJsonObject(root, "echo_audibility", &section)) {
>-      ReadParam(section, "low_render_limit",
>-                &cfg.echo_audibility.low_render_limit);
>-      ReadParam(section, "normal_render_limit",
>-                &cfg.echo_audibility.normal_render_limit);
>-
>-      ReadParam(section, "floor_power", &cfg.echo_audibility.floor_power);
>-      ReadParam(section, "audibility_threshold_lf",
>-                &cfg.echo_audibility.audibility_threshold_lf);
>-      ReadParam(section, "audibility_threshold_mf",
>-                &cfg.echo_audibility.audibility_threshold_mf);
>-      ReadParam(section, "audibility_threshold_hf",
>-                &cfg.echo_audibility.audibility_threshold_hf);
>-      ReadParam(section, "use_stationary_properties",
>-                &cfg.echo_audibility.use_stationary_properties);
>-      ReadParam(section, "use_stationary_properties_at_init",
>-                &cfg.echo_audibility.use_stationarity_properties_at_init);
>-    }
>-
>-    if (rtc::GetValueFromJsonObject(root, "echo_removal_control", &section)) {
>-      Json::Value subsection;
>-      if (rtc::GetValueFromJsonObject(section, "gain_rampup", &subsection)) {
>-        ReadParam(subsection, "initial_gain",
>-                  &cfg.echo_removal_control.gain_rampup.initial_gain);
>-        ReadParam(subsection, "first_non_zero_gain",
>-                  &cfg.echo_removal_control.gain_rampup.first_non_zero_gain);
>-        ReadParam(subsection, "non_zero_gain_blocks",
>-                  &cfg.echo_removal_control.gain_rampup.non_zero_gain_blocks);
>-        ReadParam(subsection, "full_gain_blocks",
>-                  &cfg.echo_removal_control.gain_rampup.full_gain_blocks);
>-      }
>-      ReadParam(section, "has_clock_drift",
>-                &cfg.echo_removal_control.has_clock_drift);
>-      ReadParam(section, "linear_and_stable_echo_path",
>-                &cfg.echo_removal_control.linear_and_stable_echo_path);
>-    }
>-
>-    if (rtc::GetValueFromJsonObject(root, "echo_model", &section)) {
>-      Json::Value subsection;
>-      ReadParam(section, "noise_floor_hold", &cfg.echo_model.noise_floor_hold);
>-      ReadParam(section, "min_noise_floor_power",
>-                &cfg.echo_model.min_noise_floor_power);
>-      ReadParam(section, "stationary_gate_slope",
>-                &cfg.echo_model.stationary_gate_slope);
>-      ReadParam(section, "noise_gate_power", &cfg.echo_model.noise_gate_power);
>-      ReadParam(section, "noise_gate_slope", &cfg.echo_model.noise_gate_slope);
>-      ReadParam(section, "render_pre_window_size",
>-                &cfg.echo_model.render_pre_window_size);
>-      ReadParam(section, "render_post_window_size",
>-                &cfg.echo_model.render_post_window_size);
>-      ReadParam(section, "render_pre_window_size_init",
>-                &cfg.echo_model.render_pre_window_size_init);
>-      ReadParam(section, "render_post_window_size_init",
>-                &cfg.echo_model.render_post_window_size_init);
>-      ReadParam(section, "nonlinear_hold", &cfg.echo_model.nonlinear_hold);
>-      ReadParam(section, "nonlinear_release",
>-                &cfg.echo_model.nonlinear_release);
>-    }
>-
>-    Json::Value subsection;
>-    if (rtc::GetValueFromJsonObject(root, "suppressor", &section)) {
>-      ReadParam(section, "nearend_average_blocks",
>-                &cfg.suppressor.nearend_average_blocks);
>-
>-      if (rtc::GetValueFromJsonObject(section, "normal_tuning", &subsection)) {
>-        ReadParam(subsection, "mask_lf", &cfg.suppressor.normal_tuning.mask_lf);
>-        ReadParam(subsection, "mask_hf", &cfg.suppressor.normal_tuning.mask_hf);
>-        ReadParam(subsection, "max_inc_factor",
>-                  &cfg.suppressor.normal_tuning.max_inc_factor);
>-        ReadParam(subsection, "max_dec_factor_lf",
>-                  &cfg.suppressor.normal_tuning.max_dec_factor_lf);
>-      }
>-
>-      if (rtc::GetValueFromJsonObject(section, "nearend_tuning", &subsection)) {
>-        ReadParam(subsection, "mask_lf",
>-                  &cfg.suppressor.nearend_tuning.mask_lf);
>-        ReadParam(subsection, "mask_hf",
>-                  &cfg.suppressor.nearend_tuning.mask_hf);
>-        ReadParam(subsection, "max_inc_factor",
>-                  &cfg.suppressor.nearend_tuning.max_inc_factor);
>-        ReadParam(subsection, "max_dec_factor_lf",
>-                  &cfg.suppressor.nearend_tuning.max_dec_factor_lf);
>-      }
>-
>-      if (rtc::GetValueFromJsonObject(section, "dominant_nearend_detection",
>-                                      &subsection)) {
>-        ReadParam(subsection, "enr_threshold",
>-                  &cfg.suppressor.dominant_nearend_detection.enr_threshold);
>-        ReadParam(subsection, "snr_threshold",
>-                  &cfg.suppressor.dominant_nearend_detection.snr_threshold);
>-        ReadParam(subsection, "hold_duration",
>-                  &cfg.suppressor.dominant_nearend_detection.hold_duration);
>-        ReadParam(subsection, "trigger_threshold",
>-                  &cfg.suppressor.dominant_nearend_detection.trigger_threshold);
>-      }
>-
>-      if (rtc::GetValueFromJsonObject(section, "high_bands_suppression",
>-                                      &subsection)) {
>-        ReadParam(subsection, "enr_threshold",
>-                  &cfg.suppressor.high_bands_suppression.enr_threshold);
>-        ReadParam(subsection, "max_gain_during_echo",
>-                  &cfg.suppressor.high_bands_suppression.max_gain_during_echo);
>-      }
>-
>-      ReadParam(section, "floor_first_increase",
>-                &cfg.suppressor.floor_first_increase);
>-      ReadParam(section, "enforce_transparent",
>-                &cfg.suppressor.enforce_transparent);
>-      ReadParam(section, "enforce_empty_higher_bands",
>-                &cfg.suppressor.enforce_empty_higher_bands);
>-    }
>-
>-    std::cout << std::endl;
>-    return cfg;
>-  }
>-};
>-
> void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) {
>   RTC_CHECK_EQ(src.num_channels_, dest->num_channels());
>   RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames());
>@@ -695,6 +126,13 @@ AudioProcessingSimulator::AudioProcessingSimulator(
>           settings.initial_mic_level,
>           settings_.simulate_mic_gain ? *settings.simulated_mic_kind : 0),
>       worker_queue_("file_writer_task_queue") {
>+  RTC_CHECK(!settings_.dump_internal_data || WEBRTC_APM_DEBUG_DUMP == 1);
>+  ApmDataDumper::SetActivated(settings_.dump_internal_data);
>+  if (settings_.dump_internal_data_output_dir.has_value()) {
>+    ApmDataDumper::SetOutputDirectory(
>+        settings_.dump_internal_data_output_dir.value());
>+  }
>+
>   if (settings_.ed_graph_output_filename &&
>       !settings_.ed_graph_output_filename->empty()) {
>     residual_echo_likelihood_graph_writer_.open(
>@@ -776,9 +214,9 @@ void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
>   }
> 
>   if (residual_echo_likelihood_graph_writer_.is_open()) {
>-    auto stats = ap_->GetStatistics();
>-    residual_echo_likelihood_graph_writer_ << stats.residual_echo_likelihood
>-                                           << ", ";
>+    auto stats = ap_->GetStatistics(true /*has_remote_tracks*/);
>+    residual_echo_likelihood_graph_writer_
>+        << stats.residual_echo_likelihood.value_or(-1.f) << ", ";
>   }
> 
>   ++num_process_stream_calls_;
>@@ -918,10 +356,13 @@ void AudioProcessingSimulator::CreateAudioProcessor() {
>   }
>   if (settings_.use_agc2) {
>     apm_config.gain_controller2.enabled = *settings_.use_agc2;
>-    apm_config.gain_controller2.fixed_gain_db = settings_.agc2_fixed_gain_db;
>+    apm_config.gain_controller2.fixed_digital.gain_db =
>+        settings_.agc2_fixed_gain_db;
>     if (settings_.agc2_use_adaptive_gain) {
>-      apm_config.gain_controller2.adaptive_digital_mode =
>+      apm_config.gain_controller2.adaptive_digital.enabled =
>           *settings_.agc2_use_adaptive_gain;
>+      apm_config.gain_controller2.adaptive_digital.level_estimator =
>+          settings_.agc2_adaptive_level_estimator;
>     }
>   }
>   if (settings_.use_pre_amplifier) {
>@@ -944,7 +385,7 @@ void AudioProcessingSimulator::CreateAudioProcessor() {
>       if (settings_.use_verbose_logging) {
>         std::cout << "Reading AEC3 Parameters from JSON input." << std::endl;
>       }
>-      cfg = Aec3ParametersParser::Parse(*settings_.aec3_settings_filename);
>+      cfg = ReadAec3ConfigFromJsonFile(*settings_.aec3_settings_filename);
>     }
>     echo_control_factory.reset(new EchoCanceller3Factory(cfg));
> 
>@@ -952,7 +393,7 @@ void AudioProcessingSimulator::CreateAudioProcessor() {
>       if (!settings_.use_quiet_output) {
>         std::cout << "AEC3 settings:" << std::endl;
>       }
>-      PrintAec3ParameterValues(cfg);
>+      std::cout << Aec3ConfigToJsonString(cfg) << std::endl;
>     }
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.h
>index 4e580b4fced9548d58ac2d55443f446fbb53e7c2..4b67537b49a5b00b6a57d2610924522e2fb826cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.h
>@@ -75,6 +75,8 @@ struct SimulationSettings {
>   absl::optional<int> agc_compression_gain;
>   absl::optional<bool> agc2_use_adaptive_gain;
>   float agc2_fixed_gain_db;
>+  AudioProcessing::Config::GainController2::LevelEstimator
>+      agc2_adaptive_level_estimator;
>   float pre_amplifier_gain_factor;
>   absl::optional<int> vad_likelihood;
>   absl::optional<int> ns_level;
>@@ -92,6 +94,8 @@ struct SimulationSettings {
>   bool fixed_interface = false;
>   bool store_intermediate_output = false;
>   bool print_aec3_parameter_values = false;
>+  bool dump_internal_data = false;
>+  absl::optional<std::string> dump_internal_data_output_dir;
>   absl::optional<std::string> custom_call_order_filename;
>   absl::optional<std::string> aec3_settings_filename;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_impl.cc
>index 3a448877d819ad11601173b0957ecb3dadb12c51..0ff2806e6d5decb06d687700b6d23c4320c156e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_impl.cc
>@@ -14,13 +14,17 @@
> #include <memory>
> #include <string>
> #include <utility>
>+#include <vector>
> 
>+#include "absl/strings/string_view.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/test/aec_dump_based_simulator.h"
> #include "modules/audio_processing/test/audio_processing_simulator.h"
> #include "modules/audio_processing/test/audioproc_float_impl.h"
> #include "modules/audio_processing/test/wav_based_simulator.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/flags.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
> namespace test {
>@@ -37,155 +41,185 @@ const char kUsageDescription[] =
>     "processing module, either based on wav files or "
>     "protobuf debug dump recordings.\n";
> 
>-DEFINE_string(dump_input, "", "Aec dump input filename");
>-DEFINE_string(dump_output, "", "Aec dump output filename");
>-DEFINE_string(i, "", "Forward stream input wav filename");
>-DEFINE_string(o, "", "Forward stream output wav filename");
>-DEFINE_string(ri, "", "Reverse stream input wav filename");
>-DEFINE_string(ro, "", "Reverse stream output wav filename");
>-DEFINE_string(artificial_nearend, "", "Artificial nearend wav filename");
>-DEFINE_int(output_num_channels,
>-           kParameterNotSpecifiedValue,
>-           "Number of forward stream output channels");
>-DEFINE_int(reverse_output_num_channels,
>-           kParameterNotSpecifiedValue,
>-           "Number of Reverse stream output channels");
>-DEFINE_int(output_sample_rate_hz,
>-           kParameterNotSpecifiedValue,
>-           "Forward stream output sample rate in Hz");
>-DEFINE_int(reverse_output_sample_rate_hz,
>-           kParameterNotSpecifiedValue,
>-           "Reverse stream output sample rate in Hz");
>-DEFINE_bool(fixed_interface,
>-            false,
>-            "Use the fixed interface when operating on wav files");
>-DEFINE_int(aec,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the echo canceller");
>-DEFINE_int(aecm,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the mobile echo controller");
>-DEFINE_int(ed,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate (0) the residual echo detector");
>-DEFINE_string(ed_graph, "", "Output filename for graph of echo likelihood");
>-DEFINE_int(agc,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the AGC");
>-DEFINE_int(agc2,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the AGC2");
>-DEFINE_int(pre_amplifier,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the pre amplifier");
>-DEFINE_int(hpf,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the high-pass filter");
>-DEFINE_int(ns,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the noise suppressor");
>-DEFINE_int(ts,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the transient suppressor");
>-DEFINE_int(vad,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the voice activity detector");
>-DEFINE_int(le,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the level estimator");
>-DEFINE_bool(all_default,
>-            false,
>-            "Activate all of the default components (will be overridden by any "
>-            "other settings)");
>-DEFINE_int(aec_suppression_level,
>-           kParameterNotSpecifiedValue,
>-           "Set the aec suppression level (0-2)");
>-DEFINE_int(delay_agnostic,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the AEC delay agnostic mode");
>-DEFINE_int(extended_filter,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the AEC extended filter mode");
>-DEFINE_int(aec3,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the experimental AEC mode AEC3");
>-DEFINE_int(experimental_agc,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the experimental AGC");
>-DEFINE_int(experimental_agc_disable_digital_adaptive,
>-           kParameterNotSpecifiedValue,
>-           "Force-deactivate (1) digital adaptation in "
>-           "experimental AGC. Digital adaptation is active by default (0).");
>-DEFINE_int(experimental_agc_analyze_before_aec,
>-           kParameterNotSpecifiedValue,
>-           "Make level estimation happen before AEC"
>-           " in the experimental AGC. After AEC is the default (0)");
>-DEFINE_int(
>+WEBRTC_DEFINE_string(dump_input, "", "Aec dump input filename");
>+WEBRTC_DEFINE_string(dump_output, "", "Aec dump output filename");
>+WEBRTC_DEFINE_string(i, "", "Forward stream input wav filename");
>+WEBRTC_DEFINE_string(o, "", "Forward stream output wav filename");
>+WEBRTC_DEFINE_string(ri, "", "Reverse stream input wav filename");
>+WEBRTC_DEFINE_string(ro, "", "Reverse stream output wav filename");
>+WEBRTC_DEFINE_string(artificial_nearend, "", "Artificial nearend wav filename");
>+WEBRTC_DEFINE_int(output_num_channels,
>+                  kParameterNotSpecifiedValue,
>+                  "Number of forward stream output channels");
>+WEBRTC_DEFINE_int(reverse_output_num_channels,
>+                  kParameterNotSpecifiedValue,
>+                  "Number of Reverse stream output channels");
>+WEBRTC_DEFINE_int(output_sample_rate_hz,
>+                  kParameterNotSpecifiedValue,
>+                  "Forward stream output sample rate in Hz");
>+WEBRTC_DEFINE_int(reverse_output_sample_rate_hz,
>+                  kParameterNotSpecifiedValue,
>+                  "Reverse stream output sample rate in Hz");
>+WEBRTC_DEFINE_bool(fixed_interface,
>+                   false,
>+                   "Use the fixed interface when operating on wav files");
>+WEBRTC_DEFINE_int(aec,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the echo canceller");
>+WEBRTC_DEFINE_int(aecm,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the mobile echo controller");
>+WEBRTC_DEFINE_int(ed,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate (0) the residual echo detector");
>+WEBRTC_DEFINE_string(ed_graph,
>+                     "",
>+                     "Output filename for graph of echo likelihood");
>+WEBRTC_DEFINE_int(agc,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the AGC");
>+WEBRTC_DEFINE_int(agc2,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the AGC2");
>+WEBRTC_DEFINE_int(pre_amplifier,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the pre amplifier");
>+WEBRTC_DEFINE_int(hpf,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the high-pass filter");
>+WEBRTC_DEFINE_int(ns,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the noise suppressor");
>+WEBRTC_DEFINE_int(ts,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the transient suppressor");
>+WEBRTC_DEFINE_int(vad,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the voice activity detector");
>+WEBRTC_DEFINE_int(le,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the level estimator");
>+WEBRTC_DEFINE_bool(
>+    all_default,
>+    false,
>+    "Activate all of the default components (will be overridden by any "
>+    "other settings)");
>+WEBRTC_DEFINE_int(aec_suppression_level,
>+                  kParameterNotSpecifiedValue,
>+                  "Set the aec suppression level (0-2)");
>+WEBRTC_DEFINE_int(delay_agnostic,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the AEC delay agnostic mode");
>+WEBRTC_DEFINE_int(extended_filter,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the AEC extended filter mode");
>+WEBRTC_DEFINE_int(
>+    aec3,
>+    kParameterNotSpecifiedValue,
>+    "Activate (1) or deactivate(0) the experimental AEC mode AEC3");
>+WEBRTC_DEFINE_int(experimental_agc,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the experimental AGC");
>+WEBRTC_DEFINE_int(
>+    experimental_agc_disable_digital_adaptive,
>+    kParameterNotSpecifiedValue,
>+    "Force-deactivate (1) digital adaptation in "
>+    "experimental AGC. Digital adaptation is active by default (0).");
>+WEBRTC_DEFINE_int(experimental_agc_analyze_before_aec,
>+                  kParameterNotSpecifiedValue,
>+                  "Make level estimation happen before AEC"
>+                  " in the experimental AGC. After AEC is the default (0)");
>+WEBRTC_DEFINE_int(
>     experimental_agc_agc2_level_estimator,
>     kParameterNotSpecifiedValue,
>     "AGC2 level estimation"
>     " in the experimental AGC. AGC1 level estimation is the default (0)");
>-DEFINE_int(
>+WEBRTC_DEFINE_int(
>     refined_adaptive_filter,
>     kParameterNotSpecifiedValue,
>     "Activate (1) or deactivate(0) the refined adaptive filter functionality");
>-DEFINE_int(agc_mode, kParameterNotSpecifiedValue, "Specify the AGC mode (0-2)");
>-DEFINE_int(agc_target_level,
>-           kParameterNotSpecifiedValue,
>-           "Specify the AGC target level (0-31)");
>-DEFINE_int(agc_limiter,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the level estimator");
>-DEFINE_int(agc_compression_gain,
>-           kParameterNotSpecifiedValue,
>-           "Specify the AGC compression gain (0-90)");
>-DEFINE_float(agc2_enable_adaptive_gain,
>-             kParameterNotSpecifiedValue,
>-             "Activate (1) or deactivate(0) the AGC2 adaptive gain");
>-DEFINE_float(agc2_fixed_gain_db, 0.f, "AGC2 fixed gain (dB) to apply");
>-DEFINE_float(pre_amplifier_gain_factor,
>-             1.f,
>-             "Pre-amplifier gain factor (linear) to apply");
>-DEFINE_int(vad_likelihood,
>-           kParameterNotSpecifiedValue,
>-           "Specify the VAD likelihood (0-3)");
>-DEFINE_int(ns_level, kParameterNotSpecifiedValue, "Specify the NS level (0-3)");
>-DEFINE_int(stream_delay,
>-           kParameterNotSpecifiedValue,
>-           "Specify the stream delay in ms to use");
>-DEFINE_int(use_stream_delay,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) reporting the stream delay");
>-DEFINE_int(stream_drift_samples,
>-           kParameterNotSpecifiedValue,
>-           "Specify the number of stream drift samples to use");
>-DEFINE_int(initial_mic_level, 100, "Initial mic level (0-255)");
>-DEFINE_int(simulate_mic_gain,
>-           0,
>-           "Activate (1) or deactivate(0) the analog mic gain simulation");
>-DEFINE_int(simulated_mic_kind,
>-           kParameterNotSpecifiedValue,
>-           "Specify which microphone kind to use for microphone simulation");
>-DEFINE_bool(performance_report, false, "Report the APM performance ");
>-DEFINE_bool(verbose, false, "Produce verbose output");
>-DEFINE_bool(quiet, false, "Avoid producing information about the progress.");
>-DEFINE_bool(bitexactness_report,
>-            false,
>-            "Report bitexactness for aec dump result reproduction");
>-DEFINE_bool(discard_settings_in_aecdump,
>-            false,
>-            "Discard any config settings specified in the aec dump");
>-DEFINE_bool(store_intermediate_output,
>-            false,
>-            "Creates new output files after each init");
>-DEFINE_string(custom_call_order_file, "", "Custom process API call order file");
>-DEFINE_bool(print_aec3_parameter_values,
>-            false,
>-            "Print parameter values used in AEC3 in JSON-format");
>-DEFINE_string(aec3_settings,
>-              "",
>-              "File in JSON-format with custom AEC3 settings");
>-DEFINE_bool(help, false, "Print this message");
>+WEBRTC_DEFINE_int(agc_mode,
>+                  kParameterNotSpecifiedValue,
>+                  "Specify the AGC mode (0-2)");
>+WEBRTC_DEFINE_int(agc_target_level,
>+                  kParameterNotSpecifiedValue,
>+                  "Specify the AGC target level (0-31)");
>+WEBRTC_DEFINE_int(agc_limiter,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) the level estimator");
>+WEBRTC_DEFINE_int(agc_compression_gain,
>+                  kParameterNotSpecifiedValue,
>+                  "Specify the AGC compression gain (0-90)");
>+WEBRTC_DEFINE_float(agc2_enable_adaptive_gain,
>+                    kParameterNotSpecifiedValue,
>+                    "Activate (1) or deactivate(0) the AGC2 adaptive gain");
>+WEBRTC_DEFINE_float(agc2_fixed_gain_db, 0.f, "AGC2 fixed gain (dB) to apply");
>+
>+std::vector<std::string> GetAgc2AdaptiveLevelEstimatorNames() {
>+  return {"RMS", "peak"};
>+}
>+WEBRTC_DEFINE_string(
>+    agc2_adaptive_level_estimator,
>+    "RMS",
>+    "AGC2 adaptive digital level estimator to use [RMS, peak]");
>+
>+WEBRTC_DEFINE_float(pre_amplifier_gain_factor,
>+                    1.f,
>+                    "Pre-amplifier gain factor (linear) to apply");
>+WEBRTC_DEFINE_int(vad_likelihood,
>+                  kParameterNotSpecifiedValue,
>+                  "Specify the VAD likelihood (0-3)");
>+WEBRTC_DEFINE_int(ns_level,
>+                  kParameterNotSpecifiedValue,
>+                  "Specify the NS level (0-3)");
>+WEBRTC_DEFINE_int(stream_delay,
>+                  kParameterNotSpecifiedValue,
>+                  "Specify the stream delay in ms to use");
>+WEBRTC_DEFINE_int(use_stream_delay,
>+                  kParameterNotSpecifiedValue,
>+                  "Activate (1) or deactivate(0) reporting the stream delay");
>+WEBRTC_DEFINE_int(stream_drift_samples,
>+                  kParameterNotSpecifiedValue,
>+                  "Specify the number of stream drift samples to use");
>+WEBRTC_DEFINE_int(initial_mic_level, 100, "Initial mic level (0-255)");
>+WEBRTC_DEFINE_int(
>+    simulate_mic_gain,
>+    0,
>+    "Activate (1) or deactivate(0) the analog mic gain simulation");
>+WEBRTC_DEFINE_int(
>+    simulated_mic_kind,
>+    kParameterNotSpecifiedValue,
>+    "Specify which microphone kind to use for microphone simulation");
>+WEBRTC_DEFINE_bool(performance_report, false, "Report the APM performance ");
>+WEBRTC_DEFINE_bool(verbose, false, "Produce verbose output");
>+WEBRTC_DEFINE_bool(quiet,
>+                   false,
>+                   "Avoid producing information about the progress.");
>+WEBRTC_DEFINE_bool(bitexactness_report,
>+                   false,
>+                   "Report bitexactness for aec dump result reproduction");
>+WEBRTC_DEFINE_bool(discard_settings_in_aecdump,
>+                   false,
>+                   "Discard any config settings specified in the aec dump");
>+WEBRTC_DEFINE_bool(store_intermediate_output,
>+                   false,
>+                   "Creates new output files after each init");
>+WEBRTC_DEFINE_string(custom_call_order_file,
>+                     "",
>+                     "Custom process API call order file");
>+WEBRTC_DEFINE_bool(print_aec3_parameter_values,
>+                   false,
>+                   "Print parameter values used in AEC3 in JSON-format");
>+WEBRTC_DEFINE_string(aec3_settings,
>+                     "",
>+                     "File in JSON-format with custom AEC3 settings");
>+WEBRTC_DEFINE_bool(dump_data,
>+                   false,
>+                   "Dump internal data during the call (requires build flag)");
>+WEBRTC_DEFINE_string(dump_data_output_dir,
>+                     "",
>+                     "Internal data dump output directory");
>+WEBRTC_DEFINE_bool(help, false, "Print this message");
> 
> void SetSettingIfSpecified(const std::string& value,
>                            absl::optional<std::string>* parameter) {
>@@ -208,6 +242,27 @@ void SetSettingIfFlagSet(int32_t flag, absl::optional<bool>* parameter) {
>   }
> }
> 
>+AudioProcessing::Config::GainController2::LevelEstimator
>+MapAgc2AdaptiveLevelEstimator(absl::string_view name) {
>+  if (name.compare("RMS") == 0) {
>+    return AudioProcessing::Config::GainController2::LevelEstimator::kRms;
>+  }
>+  if (name.compare("peak") == 0) {
>+    return AudioProcessing::Config::GainController2::LevelEstimator::kPeak;
>+  }
>+  auto concat_strings =
>+      [](const std::vector<std::string>& strings) -> std::string {
>+    rtc::StringBuilder ss;
>+    for (const auto& s : strings) {
>+      ss << " " << s;
>+    }
>+    return ss.Release();
>+  };
>+  RTC_CHECK(false)
>+      << "Invalid value for agc2_adaptive_level_estimator, valid options:"
>+      << concat_strings(GetAgc2AdaptiveLevelEstimatorNames()) << ".";
>+}
>+
> SimulationSettings CreateSettings() {
>   SimulationSettings settings;
>   if (FLAG_all_default) {
>@@ -275,6 +330,8 @@ SimulationSettings CreateSettings() {
>   SetSettingIfFlagSet(FLAG_agc2_enable_adaptive_gain,
>                       &settings.agc2_use_adaptive_gain);
>   settings.agc2_fixed_gain_db = FLAG_agc2_fixed_gain_db;
>+  settings.agc2_adaptive_level_estimator =
>+      MapAgc2AdaptiveLevelEstimator(FLAG_agc2_adaptive_level_estimator);
>   settings.pre_amplifier_gain_factor = FLAG_pre_amplifier_gain_factor;
>   SetSettingIfSpecified(FLAG_vad_likelihood, &settings.vad_likelihood);
>   SetSettingIfSpecified(FLAG_ns_level, &settings.ns_level);
>@@ -296,6 +353,9 @@ SimulationSettings CreateSettings() {
>   settings.fixed_interface = FLAG_fixed_interface;
>   settings.store_intermediate_output = FLAG_store_intermediate_output;
>   settings.print_aec3_parameter_values = FLAG_print_aec3_parameter_values;
>+  settings.dump_internal_data = FLAG_dump_data;
>+  SetSettingIfSpecified(FLAG_dump_data_output_dir,
>+                        &settings.dump_internal_data_output_dir);
> 
>   return settings;
> }
>@@ -446,6 +506,15 @@ void PerformBasicParameterSanityChecks(const SimulationSettings& settings) {
>       settings.artificial_nearend_filename &&
>           !valid_wav_name(*settings.artificial_nearend_filename),
>       "Error: --artifical_nearend must be a valid .wav file name.\n");
>+
>+  ReportConditionalErrorAndExit(
>+      WEBRTC_APM_DEBUG_DUMP == 0 && settings.dump_internal_data,
>+      "Error: --dump_data cannot be set without proper build support.\n");
>+
>+  ReportConditionalErrorAndExit(
>+      !settings.dump_internal_data &&
>+          settings.dump_internal_data_output_dir.has_value(),
>+      "Error: --dump_data_output_dir cannot be set without --dump_data.\n");
> }
> 
> }  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
>index 721ebc7ee5669f552a1eec96b4211a822a658d74..551781b8bc90a6bfad530aa3817e58f9a794b9cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
>@@ -51,6 +51,7 @@ rtc_static_library("lib") {
>     "../../../../common_audio",
>     "../../../../rtc_base:checks",
>     "../../../../rtc_base:rtc_base_approved",
>+    "../../../../test:fileutils",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
>   visibility = [ ":*" ]  # Only targets in this file can depend on this.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator.cc
>index 741a1ca86ba210bf58290c82805a175d2ab932e5..fa561cf8e05516a8c0faccbec5778bacb4b7a53d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator.cc
>@@ -32,10 +32,10 @@ const char kUsageDescription[] =
>     "Command-line tool to generate multiple-end audio tracks to simulate "
>     "conversational speech with two or more participants.\n";
> 
>-DEFINE_string(i, "", "Directory containing the speech turn wav files");
>-DEFINE_string(t, "", "Path to the timing text file");
>-DEFINE_string(o, "", "Output wav files destination path");
>-DEFINE_bool(help, false, "Prints this message");
>+WEBRTC_DEFINE_string(i, "", "Directory containing the speech turn wav files");
>+WEBRTC_DEFINE_string(t, "", "Path to the timing text file");
>+WEBRTC_DEFINE_string(o, "", "Output wav files destination path");
>+WEBRTC_DEFINE_bool(help, false, "Prints this message");
> 
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
>index eb8e3bed3c510181ccc46938ad4b36caf1f174d4..a45e3bbfa72fbee5c8a659b065a6a8f8f30d295c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
>@@ -12,7 +12,6 @@
> 
> #include "modules/audio_processing/test/conversational_speech/mock_wavreader.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/pathutils.h"
> #include "test/gmock.h"
> 
> namespace webrtc {
>@@ -39,9 +38,10 @@ MockWavReaderFactory::~MockWavReaderFactory() = default;
> std::unique_ptr<WavReaderInterface> MockWavReaderFactory::CreateMock(
>     const std::string& filepath) {
>   // Search the parameters corresponding to filepath.
>-  const rtc::Pathname audiotrack_file_path(filepath);
>-  const auto it =
>-      audiotrack_names_params_.find(audiotrack_file_path.filename());
>+  size_t delimiter = filepath.find_last_of("/\\");  // Either windows or posix
>+  std::string filename =
>+      filepath.substr(delimiter == std::string::npos ? 0 : delimiter + 1);
>+  const auto it = audiotrack_names_params_.find(filename);
> 
>   // If not found, use default parameters.
>   if (it == audiotrack_names_params_.end()) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
>index d633d90e9c426a2765f6aae219c5bc92064649c8..4e2f54d1509579d3d53fa11522d6a4dac27717fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
>@@ -14,7 +14,7 @@
> #include <iterator>
> 
> #include "rtc_base/logging.h"
>-#include "rtc_base/pathutils.h"
>+#include "test/testsupport/fileutils.h"
> 
> namespace webrtc {
> namespace test {
>@@ -50,13 +50,13 @@ bool MultiEndCall::CreateAudioTrackReaders() {
>     if (it != audiotrack_readers_.end())
>       continue;
> 
>-    // Instance Pathname to retrieve the full path to the audiotrack file.
>-    const rtc::Pathname audiotrack_file_path(audiotracks_path_,
>-                                             turn.audiotrack_file_name);
>+    const std::string audiotrack_file_path =
>+        test::JoinFilename(audiotracks_path_, turn.audiotrack_file_name);
> 
>     // Map the audiotrack file name to a new instance of WavReaderInterface.
>     std::unique_ptr<WavReaderInterface> wavreader =
>-        wavreader_abstract_factory_->Create(audiotrack_file_path.pathname());
>+        wavreader_abstract_factory_->Create(
>+            test::JoinFilename(audiotracks_path_, turn.audiotrack_file_name));
> 
>     if (sample_rate_hz_ == 0) {
>       sample_rate_hz_ = wavreader->SampleRate();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.cc
>index c2fb7808e08e6102571558fba61f5572bcfe9c48..c400499fd34103006c4d8a20e08aa73ef0940684 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.cc
>@@ -25,7 +25,7 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/pathutils.h"
>+#include "test/testsupport/fileutils.h"
> 
> namespace webrtc {
> namespace test {
>@@ -46,21 +46,20 @@ InitSpeakerOutputFilePaths(const std::set<std::string>& speaker_names,
> 
>   // Add near-end and far-end output paths into the map.
>   for (const auto& speaker_name : speaker_names) {
>-    const rtc::Pathname near_end_path(output_path,
>-                                      "s_" + speaker_name + "-near_end.wav");
>+    const std::string near_end_path =
>+        test::JoinFilename(output_path, "s_" + speaker_name + "-near_end.wav");
>     RTC_LOG(LS_VERBOSE) << "The near-end audio track will be created in "
>-                        << near_end_path.pathname() << ".";
>+                        << near_end_path << ".";
> 
>-    const rtc::Pathname far_end_path(output_path,
>-                                     "s_" + speaker_name + "-far_end.wav");
>+    const std::string far_end_path =
>+        test::JoinFilename(output_path, "s_" + speaker_name + "-far_end.wav");
>     RTC_LOG(LS_VERBOSE) << "The far-end audio track will be created in "
>-                        << far_end_path.pathname() << ".";
>+                        << far_end_path << ".";
> 
>     // Add to map.
>     speaker_output_file_paths_map->emplace(
>         std::piecewise_construct, std::forward_as_tuple(speaker_name),
>-        std::forward_as_tuple(near_end_path.pathname(),
>-                              far_end_path.pathname()));
>+        std::forward_as_tuple(near_end_path, far_end_path));
>   }
> 
>   return speaker_output_file_paths_map;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc
>index a6184b5f9fda011fa7d68ae4c7a7ef526716517e..4b6ada27b6a6f82c17e86f5077fc88128ff6a71f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/apm_vad.cc
>@@ -24,9 +24,9 @@ constexpr int kMaxSampleRate = 48000;
> constexpr size_t kMaxFrameLen =
>     kAudioFrameLengthMilliseconds * kMaxSampleRate / 1000;
> 
>-DEFINE_string(i, "", "Input wav file");
>-DEFINE_string(o_probs, "", "VAD probabilities output file");
>-DEFINE_string(o_rms, "", "VAD output file");
>+WEBRTC_DEFINE_string(i, "", "Input wav file");
>+WEBRTC_DEFINE_string(o_probs, "", "VAD probabilities output file");
>+WEBRTC_DEFINE_string(o_rms, "", "VAD output file");
> 
> int main(int argc, char* argv[]) {
>   if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true))
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
>index 98cf84c888f7b5c13f84c0e3fbe4eb2592bc06ac..35a2c11aeb2279cdfb3e3fd1cafc3157eb6278e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
>@@ -26,13 +26,13 @@ constexpr size_t kMaxFrameLen = kMaxFrameLenMs * kMaxSampleRate / 1000;
> 
> const double kOneDbReduction = DbToRatio(-1.0);
> 
>-DEFINE_string(i, "", "Input wav file");
>-DEFINE_string(oc, "", "Config output file");
>-DEFINE_string(ol, "", "Levels output file");
>-DEFINE_float(a, 5.f, "Attack (ms)");
>-DEFINE_float(d, 20.f, "Decay (ms)");
>-DEFINE_int(f, 10, "Frame length (ms)");
>-DEFINE_bool(help, false, "prints this message");
>+WEBRTC_DEFINE_string(i, "", "Input wav file");
>+WEBRTC_DEFINE_string(oc, "", "Config output file");
>+WEBRTC_DEFINE_string(ol, "", "Levels output file");
>+WEBRTC_DEFINE_float(a, 5.f, "Attack (ms)");
>+WEBRTC_DEFINE_float(d, 20.f, "Decay (ms)");
>+WEBRTC_DEFINE_int(f, 10, "Frame length (ms)");
>+WEBRTC_DEFINE_bool(help, false, "prints this message");
> 
> int main(int argc, char* argv[]) {
>   if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
>index 191cb1e9fcfd152f9e0943768f0f5b36dfe72598..8a134ed185d9ce9603ad7d1bc559c4afc3ba3526 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
>@@ -27,8 +27,8 @@ constexpr size_t kMaxFrameLen =
> 
> constexpr uint8_t kBitmaskBuffSize = 8;
> 
>-DEFINE_string(i, "", "Input wav file");
>-DEFINE_string(o, "", "VAD output file");
>+WEBRTC_DEFINE_string(i, "", "Input wav file");
>+WEBRTC_DEFINE_string(o, "", "VAD output file");
> 
> int main(int argc, char* argv[]) {
>   if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true))
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.cc
>index a199bb0c1b9d3e1c07bcea1287534d1ade091752..83810bfe3c276eb0022d0b8348fe070881f6a4a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.cc
>@@ -10,7 +10,7 @@
> 
> #include "modules/audio_processing/transient/moving_moments.h"
> 
>-#include <cmath>
>+#include <algorithm>
> 
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.cc
>index c3bf282b0e1ff17d1ae7b48789b15120b57f211c..8997d4c09206faae3a5c4ca41d420e9eeef58acd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.cc
>@@ -13,12 +13,12 @@
> #include <float.h>
> #include <math.h>
> #include <string.h>
>-
> #include <algorithm>
> 
> #include "modules/audio_processing/transient/common.h"
> #include "modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h"
> #include "modules/audio_processing/transient/moving_moments.h"
>+#include "modules/audio_processing/transient/wpd_node.h"
> #include "modules/audio_processing/transient/wpd_tree.h"
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.h
>index 3267b3a6cdc4a5a666b7b4ed1b87845feaca829a..23b88f82b10cf2e6ec37728b11631c7a2e175ac1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_DETECTOR_H_
> #define MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_DETECTOR_H_
> 
>+#include <stddef.h>
> #include <deque>
> #include <memory>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppression_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
>index 9e7ecd578652e46b4462f97739c6a4ce650ab342..e15f69c04201550ea1622e843d8b7a252b951101 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
>@@ -23,26 +23,28 @@
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
> 
>-DEFINE_string(in_file_name, "", "PCM file that contains the signal.");
>-DEFINE_string(detection_file_name,
>-              "",
>-              "PCM file that contains the detection signal.");
>-DEFINE_string(reference_file_name,
>-              "",
>-              "PCM file that contains the reference signal.");
>-
>-DEFINE_int(chunk_size_ms,
>-           10,
>-           "Time between each chunk of samples in milliseconds.");
>-
>-DEFINE_int(sample_rate_hz, 16000, "Sampling frequency of the signal in Hertz.");
>-DEFINE_int(detection_rate_hz,
>-           0,
>-           "Sampling frequency of the detection signal in Hertz.");
>-
>-DEFINE_int(num_channels, 1, "Number of channels.");
>-
>-DEFINE_bool(help, false, "Print this message.");
>+WEBRTC_DEFINE_string(in_file_name, "", "PCM file that contains the signal.");
>+WEBRTC_DEFINE_string(detection_file_name,
>+                     "",
>+                     "PCM file that contains the detection signal.");
>+WEBRTC_DEFINE_string(reference_file_name,
>+                     "",
>+                     "PCM file that contains the reference signal.");
>+
>+WEBRTC_DEFINE_int(chunk_size_ms,
>+                  10,
>+                  "Time between each chunk of samples in milliseconds.");
>+
>+WEBRTC_DEFINE_int(sample_rate_hz,
>+                  16000,
>+                  "Sampling frequency of the signal in Hertz.");
>+WEBRTC_DEFINE_int(detection_rate_hz,
>+                  0,
>+                  "Sampling frequency of the detection signal in Hertz.");
>+
>+WEBRTC_DEFINE_int(num_channels, 1, "Number of channels.");
>+
>+WEBRTC_DEFINE_bool(help, false, "Print this message.");
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.h
>index 9ae3fc660a881e2dfc51f5f79b73a46abef819a0..ae51966a3d4ebb850b60cd468c53e2cff2041db0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.h
>@@ -11,9 +11,9 @@
> #ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
> #define MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
> 
>-#include <deque>
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
>-#include <set>
> 
> #include "rtc_base/gtest_prod_util.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.cc
>index 72f4d76891f386de3dc12707b79d16141ed07d85..c8aa6158817afeeceb9c249b0737e07ab7de6c87 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.cc
>@@ -10,10 +10,8 @@
> 
> #include "modules/audio_processing/transient/wpd_tree.h"
> 
>-#include <math.h>
> #include <string.h>
> 
>-#include "modules/audio_processing/transient/dyadic_decimator.h"
> #include "modules/audio_processing/transient/wpd_node.h"
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.h
>index 707a89d44ac16fdb24abdf99e9c15897ad6461a7..b62135dafa8d7ced899ac1a67023dc1c7e4261a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_WPD_TREE_H_
> #define MODULES_AUDIO_PROCESSING_TRANSIENT_WPD_TREE_H_
> 
>+#include <stddef.h>
> #include <memory>
> 
> #include "modules/audio_processing/transient/wpd_node.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.h
>index 70fd9033661092133ae3f736f9320c981899a2b7..d8fb3592c9644d4d2ab68c7c9e418efa684357f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.h
>@@ -11,9 +11,11 @@
> #ifndef MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
> #define MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
> 
>+#include "rtc_base/system/rtc_export.h"
>+
> namespace webrtc {
> 
>-class TypingDetection {
>+class RTC_EXPORT TypingDetection {
>  public:
>   TypingDetection();
>   virtual ~TypingDetection();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.cc
>index c3333cea5beb000a9f52263c4f1f78c992b407d8..8628bd39f0aecc2191c3d4a252112d4d3422f4a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.cc
>@@ -23,8 +23,6 @@
> 
> #include "modules/audio_processing/utility/ooura_fft.h"
> 
>-#include <math.h>
>-
> #include "modules/audio_processing/utility/ooura_fft_tables_common.h"
> #include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc
>index 9b5d0f37a4497c7476ada387bfd7bb86fc2dc86c..0e4a44beccf41e6203e6f05811aaf3f9aca52649 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc
>@@ -8,10 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/audio_processing/utility/ooura_fft.h"
>-
> #include <emmintrin.h>
>+#include <xmmintrin.h>
> 
>+#include "modules/audio_processing/utility/ooura_fft.h"
> #include "modules/audio_processing/utility/ooura_fft_tables_common.h"
> #include "modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h"
> #include "rtc_base/system/arch.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm.cc
>index cd8a1a8a375c7edb89a85302ca0b75771e688ab3..3b8764c4d02a9bb31733aae1ac112119d2ce56a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm.cc
>@@ -11,7 +11,6 @@
> #include "modules/audio_processing/vad/gmm.h"
> 
> #include <math.h>
>-#include <stdlib.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
>index 025ef2059d912b36b8e87f36b210734ef47a4c93..68e60dc66ae5176b29acd0477d1cd40544e3eee6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
>@@ -10,7 +10,6 @@
> 
> #include "modules/audio_processing/vad/pitch_based_vad.h"
> 
>-#include <math.h>
> #include <string.h>
> 
> #include "modules/audio_processing/vad/common.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.h
>index 4d327652c680172752b9f67310024b156e050355..22bc0f2263901007e2afda78aaa0c66ba44e6ceb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.h
>@@ -18,7 +18,6 @@
> 
> namespace webrtc {
> 
>-class AudioFrame;
> class VadCircularBuffer;
> 
> // Computes the probability of the input audio frame to be active given
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.cc
>index b9967d7aea5bf43944662c4361c5e5c825be08fd..4156d7e7952a9f1ae4e01078b43ccd60c9c55da5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.cc
>@@ -10,7 +10,6 @@
> 
> #include "modules/audio_processing/vad/pole_zero_filter.h"
> 
>-#include <stdlib.h>
> #include <string.h>
> #include <algorithm>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.cc
>index 19a5282c3f71584b840fd3c85e0ba954fb01bd22..1397668eb4bcbd2432fd03c72856a29d64819d3d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.cc
>@@ -12,7 +12,7 @@
> 
> #include <string.h>
> 
>-#include "audio/utility/audio_frame_operations.h"
>+#include "common_audio/vad/include/webrtc_vad.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.h
>index 79650fbe8c6cae1c3aded93acc12ae9ffd03159f..3dff4163c22e81d325ea2bec5a0b90eb52029f22 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.h
>@@ -11,13 +11,14 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC_STANDALONE_VAD_H_
> #define MODULES_AUDIO_PROCESSING_AGC_STANDALONE_VAD_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "common_audio/vad/include/webrtc_vad.h"
> #include "modules/audio_processing/vad/common.h"
> 
> namespace webrtc {
> 
>-class AudioFrame;
>-
> class StandaloneVad {
>  public:
>   static StandaloneVad* Create();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.h
>index e34091bdb6abd1e8a3b31c159f1f024e370d3b16..9be3467ef8523086473bb2046818ed24113fc8e3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.h
>@@ -11,13 +11,14 @@
> #ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
> #define MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> 
>-#include "modules/audio_processing/vad/common.h"
>+#include "modules/audio_processing/vad/common.h"  // AudioFeatures, kSampleR...
> 
> namespace webrtc {
> 
>-class AudioFrame;
> class PoleZeroFilter;
> 
> class VadAudioProc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.h
>index e424ac1a28278e5452e0253bc5e5977aa07493ce..d140fe2aa3a4140361cda106551d77a9679c694a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.h
>@@ -11,6 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_
> #define MODULES_AUDIO_PROCESSING_VAD_VOICE_ACTIVITY_DETECTOR_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> #include <vector>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.cc
>index 9280be1ef9d0c24f5eea8f18500ac880e2eaa12f..c55ca4aee8771ab10b98ba80b1f58e44b55f9a1e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.cc
>@@ -10,8 +10,10 @@
> 
> #include "modules/audio_processing/voice_detection_impl.h"
> 
>+#include "api/audio/audio_frame.h"
> #include "common_audio/vad/include/webrtc_vad.h"
> #include "modules/audio_processing/audio_buffer.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.h
>index 4b724bdd25647ec3f1ae524cc57ffef04248dab5..c4384736799e01083d4358dace2ec511817edf79 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.h
>@@ -11,11 +11,13 @@
> #ifndef MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
> #define MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
> 
>+#include <stddef.h>
> #include <memory>
> 
> #include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
> 
>@@ -42,6 +44,7 @@ class VoiceDetectionImpl : public VoiceDetection {
> 
>  private:
>   class Vad;
>+
>   rtc::CriticalSection* const crit_;
>   bool enabled_ RTC_GUARDED_BY(crit_) = false;
>   bool stream_has_voice_ RTC_GUARDED_BY(crit_) = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/BUILD.gn
>index 72046f0a015e02a8272f9430294bad0e4887addd..b501cb4457f2fe5f40d910bb5c06eae3ebd60322 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/BUILD.gn
>@@ -15,6 +15,8 @@ rtc_static_library("bitrate_controller") {
>     "bitrate_controller_impl.cc",
>     "bitrate_controller_impl.h",
>     "include/bitrate_controller.h",
>+    "loss_based_bandwidth_estimation.cc",
>+    "loss_based_bandwidth_estimation.h",
>     "send_side_bandwidth_estimation.cc",
>     "send_side_bandwidth_estimation.h",
>   ]
>@@ -31,10 +33,16 @@ rtc_static_library("bitrate_controller") {
> 
>   deps = [
>     "..:module_api",
>+    "../../api/transport:network_control",
>+    "../../api/units:data_rate",
>+    "../../api/units:time_delta",
>+    "../../api/units:timestamp",
>     "../../logging:rtc_event_bwe",
>     "../../logging:rtc_event_log_api",
>     "../../rtc_base:checks",
>+    "../../rtc_base:deprecation",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base/experiments:field_trial_parser",
>     "../../system_wrappers",
>     "../../system_wrappers:field_trial",
>     "../../system_wrappers:metrics",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
>index 93045b73a36bb344a365d7bcaebf8b38ad223791..57c85f6f6fc57fe1217d48c6c0ff67769201f6bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
>@@ -21,10 +21,10 @@
> 
> namespace webrtc {
> namespace {
>-absl::optional<DataRate> ToOptionalDataRate(int start_bitrate_bps) {
>-  if (start_bitrate_bps == -1)
>-    return absl::nullopt;
>-  return DataRate::bps(start_bitrate_bps);
>+absl::optional<DataRate> ToOptionalDataRate(int send_bitrate_bps) {
>+  if (send_bitrate_bps > 0)
>+    return DataRate::bps(send_bitrate_bps);
>+  return absl::nullopt;
> }
> DataRate MaxRate(int max_bitrate_bps) {
>   if (max_bitrate_bps == -1)
>@@ -153,14 +153,12 @@ void BitrateControllerImpl::OnDelayBasedBweResult(
>     rtc::CritScope cs(&critsect_);
>     if (result.probe) {
>       bandwidth_estimation_.SetSendBitrate(
>-          DataRate::bps(result.target_bitrate_bps),
>-          Timestamp::ms(clock_->TimeInMilliseconds()));
>+          result.target_bitrate, Timestamp::ms(clock_->TimeInMilliseconds()));
>     }
>     // Since SetSendBitrate now resets the delay-based estimate, we have to call
>     // UpdateDelayBasedEstimate after SetSendBitrate.
>     bandwidth_estimation_.UpdateDelayBasedEstimate(
>-        Timestamp::ms(clock_->TimeInMilliseconds()),
>-        DataRate::bps(result.target_bitrate_bps));
>+        Timestamp::ms(clock_->TimeInMilliseconds()), result.target_bitrate);
>   }
>   MaybeTriggerOnNetworkChanged();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
>index 96f8a1ef266eb9e62ecf0f801ccc31d08c7eda8b..cf281214aa3cfb4685e4be8f3965323b9c90ce00 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
>@@ -169,7 +169,7 @@ TEST_F(BitrateControllerTest, OneBitrateObserverOneRtcpObserver) {
>   EXPECT_EQ(300000, bitrate_observer_.last_bitrate_);
> 
>   // Test that a low delay-based estimate limits the combined estimate.
>-  webrtc::DelayBasedBwe::Result result(false, 280000);
>+  webrtc::DelayBasedBwe::Result result(false, webrtc::DataRate::kbps(280));
>   controller_->OnDelayBasedBweResult(result);
>   EXPECT_EQ(280000, bitrate_observer_.last_bitrate_);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/include/bitrate_controller.h
>index f67600d19083784e467d3fd22f727d3f0e332311..3e11fa76a3241de6089e8e2a3410c0c865f09d61 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/include/bitrate_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/include/bitrate_controller.h
>@@ -15,15 +15,17 @@
> #ifndef MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
> #define MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
> 
>-#include <map>
>+#include <stddef.h>
>+#include <stdint.h>
> 
> #include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
> #include "modules/include/module.h"
>-#include "modules/pacing/paced_sender.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "rtc_base/deprecation.h"
> 
> namespace webrtc {
> 
>+class Clock;
> class RtcEventLog;
> 
> // Deprecated
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..5d7f8aa2c36d7fd66d562aacb86de6ce7666ce47
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.cc
>@@ -0,0 +1,215 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/bitrate_controller/loss_based_bandwidth_estimation.h"
>+
>+#include <algorithm>
>+#include <string>
>+#include <vector>
>+
>+#include "api/units/data_rate.h"
>+#include "api/units/time_delta.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+const char kBweLossBasedControl[] = "WebRTC-Bwe-LossBasedControl";
>+
>+// Increase slower when RTT is high.
>+double GetIncreaseFactor(const LossBasedControlConfig& config, TimeDelta rtt) {
>+  // Clamp the RTT
>+  if (rtt < config.increase_low_rtt) {
>+    rtt = config.increase_low_rtt;
>+  } else if (rtt > config.increase_high_rtt) {
>+    rtt = config.increase_high_rtt;
>+  }
>+  auto rtt_range = config.increase_high_rtt.Get() - config.increase_low_rtt;
>+  if (rtt_range <= TimeDelta::Zero()) {
>+    RTC_DCHECK(false);  // Only on misconfiguration.
>+    return config.min_increase_factor;
>+  }
>+  auto rtt_offset = rtt - config.increase_low_rtt;
>+  auto relative_offset = std::max(0.0, std::min(rtt_offset / rtt_range, 1.0));
>+  auto factor_range = config.max_increase_factor - config.min_increase_factor;
>+  return config.min_increase_factor + (1 - relative_offset) * factor_range;
>+}
>+
>+double LossFromBitrate(DataRate bitrate,
>+                       DataRate loss_bandwidth_balance,
>+                       double exponent) {
>+  if (loss_bandwidth_balance >= bitrate)
>+    return 1.0;
>+  return pow(loss_bandwidth_balance / bitrate, exponent);
>+}
>+
>+DataRate BitrateFromLoss(double loss,
>+                         DataRate loss_bandwidth_balance,
>+                         double exponent) {
>+  if (exponent <= 0) {
>+    RTC_DCHECK(false);
>+    return DataRate::Infinity();
>+  }
>+  if (loss < 1e-5)
>+    return DataRate::Infinity();
>+  return loss_bandwidth_balance * pow(loss, -1.0 / exponent);
>+}
>+
>+double ExponentialUpdate(TimeDelta window, TimeDelta interval) {
>+  // Use the convention that exponential window length (which is really
>+  // infinite) is the time it takes to dampen to 1/e.
>+  if (window <= TimeDelta::Zero()) {
>+    RTC_DCHECK(false);
>+    return 1.0f;
>+  }
>+  return 1.0f - exp(interval / window * -1.0);
>+}
>+
>+}  // namespace
>+
>+LossBasedControlConfig::LossBasedControlConfig()
>+    : enabled(field_trial::IsEnabled(kBweLossBasedControl)),
>+      min_increase_factor("min_incr", 1.02),
>+      max_increase_factor("max_incr", 1.08),
>+      increase_low_rtt("incr_low_rtt", TimeDelta::ms(200)),
>+      increase_high_rtt("incr_high_rtt", TimeDelta::ms(800)),
>+      decrease_factor("decr", 0.99),
>+      loss_window("loss_win", TimeDelta::ms(800)),
>+      loss_max_window("loss_max_win", TimeDelta::ms(800)),
>+      acknowledged_rate_max_window("ackrate_max_win", TimeDelta::ms(800)),
>+      increase_offset("incr_offset", DataRate::bps(1000)),
>+      loss_bandwidth_balance_increase("balance_incr", DataRate::kbps(0.5)),
>+      loss_bandwidth_balance_decrease("balance_decr", DataRate::kbps(4)),
>+      loss_bandwidth_balance_exponent("exponent", 0.5),
>+      allow_resets("resets", false),
>+      decrease_interval("decr_intvl", TimeDelta::ms(300)),
>+      loss_report_timeout("timeout", TimeDelta::ms(6000)) {
>+  std::string trial_string = field_trial::FindFullName(kBweLossBasedControl);
>+  ParseFieldTrial(
>+      {&min_increase_factor, &max_increase_factor, &increase_low_rtt,
>+       &increase_high_rtt, &decrease_factor, &loss_window, &loss_max_window,
>+       &acknowledged_rate_max_window, &increase_offset,
>+       &loss_bandwidth_balance_increase, &loss_bandwidth_balance_decrease,
>+       &loss_bandwidth_balance_exponent, &allow_resets, &decrease_interval,
>+       &loss_report_timeout},
>+      trial_string);
>+}
>+LossBasedControlConfig::LossBasedControlConfig(const LossBasedControlConfig&) =
>+    default;
>+LossBasedControlConfig::~LossBasedControlConfig() = default;
>+
>+LossBasedBandwidthEstimation::LossBasedBandwidthEstimation()
>+    : config_(LossBasedControlConfig()),
>+      average_loss_(0),
>+      average_loss_max_(0),
>+      loss_based_bitrate_(DataRate::Zero()),
>+      acknowledged_bitrate_max_(DataRate::Zero()),
>+      acknowledged_bitrate_last_update_(Timestamp::MinusInfinity()),
>+      time_last_decrease_(Timestamp::MinusInfinity()),
>+      has_decreased_since_last_loss_report_(false),
>+      last_loss_packet_report_(Timestamp::MinusInfinity()),
>+      last_loss_ratio_(0) {}
>+
>+void LossBasedBandwidthEstimation::UpdateLossStatistics(
>+    const std::vector<PacketResult>& packet_results,
>+    Timestamp at_time) {
>+  if (packet_results.empty()) {
>+    RTC_DCHECK(false);
>+    return;
>+  }
>+  int loss_count = 0;
>+  for (auto pkt : packet_results) {
>+    loss_count += pkt.receive_time.IsInfinite() ? 1 : 0;
>+  }
>+  last_loss_ratio_ = static_cast<double>(loss_count) / packet_results.size();
>+  const TimeDelta time_passed = last_loss_packet_report_.IsFinite()
>+                                    ? at_time - last_loss_packet_report_
>+                                    : TimeDelta::seconds(1);
>+  last_loss_packet_report_ = at_time;
>+  has_decreased_since_last_loss_report_ = false;
>+
>+  average_loss_ += ExponentialUpdate(config_.loss_window, time_passed) *
>+                   (last_loss_ratio_ - average_loss_);
>+  if (average_loss_ > average_loss_max_) {
>+    average_loss_max_ = average_loss_;
>+  } else {
>+    average_loss_max_ +=
>+        ExponentialUpdate(config_.loss_max_window, time_passed) *
>+        (average_loss_ - average_loss_max_);
>+  }
>+}
>+
>+void LossBasedBandwidthEstimation::UpdateAcknowledgedBitrate(
>+    DataRate acknowledged_bitrate,
>+    Timestamp at_time) {
>+  const TimeDelta time_passed =
>+      acknowledged_bitrate_last_update_.IsFinite()
>+          ? at_time - acknowledged_bitrate_last_update_
>+          : TimeDelta::seconds(1);
>+  acknowledged_bitrate_last_update_ = at_time;
>+  if (acknowledged_bitrate > acknowledged_bitrate_max_) {
>+    acknowledged_bitrate_max_ = acknowledged_bitrate;
>+  } else {
>+    acknowledged_bitrate_max_ +=
>+        ExponentialUpdate(config_.acknowledged_rate_max_window, time_passed) *
>+        (acknowledged_bitrate - acknowledged_bitrate_max_);
>+  }
>+}
>+
>+void LossBasedBandwidthEstimation::Update(Timestamp at_time,
>+                                          DataRate min_bitrate,
>+                                          TimeDelta last_round_trip_time) {
>+  // Only increase if loss has been low for some time.
>+  const double loss_estimate_for_increase = average_loss_max_;
>+  // Avoid multiple decreases from averaging over one loss spike.
>+  const double loss_estimate_for_decrease =
>+      std::min(average_loss_, last_loss_ratio_);
>+
>+  const double loss_increase_threshold = LossFromBitrate(
>+      loss_based_bitrate_, config_.loss_bandwidth_balance_increase,
>+      config_.loss_bandwidth_balance_exponent);
>+  const double loss_decrease_threshold = LossFromBitrate(
>+      loss_based_bitrate_, config_.loss_bandwidth_balance_decrease,
>+      config_.loss_bandwidth_balance_exponent);
>+  const bool allow_decrease =
>+      !has_decreased_since_last_loss_report_ &&
>+      (at_time - time_last_decrease_ >=
>+       last_round_trip_time + config_.decrease_interval);
>+
>+  if (loss_estimate_for_increase < loss_increase_threshold) {
>+    // Increase bitrate by RTT-adaptive ratio.
>+    DataRate new_increased_bitrate =
>+        min_bitrate * GetIncreaseFactor(config_, last_round_trip_time) +
>+        config_.increase_offset;
>+    // The bitrate that would make the loss "just high enough".
>+    const DataRate new_increased_bitrate_cap = BitrateFromLoss(
>+        loss_estimate_for_increase, config_.loss_bandwidth_balance_increase,
>+        config_.loss_bandwidth_balance_exponent);
>+    new_increased_bitrate =
>+        std::min(new_increased_bitrate, new_increased_bitrate_cap);
>+    loss_based_bitrate_ = std::max(new_increased_bitrate, loss_based_bitrate_);
>+  } else if (loss_estimate_for_decrease > loss_decrease_threshold &&
>+             allow_decrease) {
>+    DataRate new_decreased_bitrate =
>+        config_.decrease_factor * acknowledged_bitrate_max_;
>+    // The bitrate that would make the loss "just acceptable".
>+    const DataRate new_decreased_bitrate_floor = BitrateFromLoss(
>+        loss_estimate_for_decrease, config_.loss_bandwidth_balance_decrease,
>+        config_.loss_bandwidth_balance_exponent);
>+    new_decreased_bitrate =
>+        std::max(new_decreased_bitrate, new_decreased_bitrate_floor);
>+    if (new_decreased_bitrate < loss_based_bitrate_) {
>+      time_last_decrease_ = at_time;
>+      has_decreased_since_last_loss_report_ = true;
>+      loss_based_bitrate_ = new_decreased_bitrate;
>+    }
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..0f560769ba43c5d823fd56fbbc089e627c95e6f9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/loss_based_bandwidth_estimation.h
>@@ -0,0 +1,80 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_BITRATE_CONTROLLER_LOSS_BASED_BANDWIDTH_ESTIMATION_H_
>+#define MODULES_BITRATE_CONTROLLER_LOSS_BASED_BANDWIDTH_ESTIMATION_H_
>+
>+#include <vector>
>+
>+#include "api/units/data_rate.h"
>+#include "api/units/time_delta.h"
>+#include "api/units/timestamp.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "rtc_base/experiments/field_trial_parser.h"
>+
>+namespace webrtc {
>+
>+struct LossBasedControlConfig {
>+  LossBasedControlConfig();
>+  LossBasedControlConfig(const LossBasedControlConfig&);
>+  LossBasedControlConfig& operator=(const LossBasedControlConfig&) = default;
>+  ~LossBasedControlConfig();
>+  bool enabled;
>+  FieldTrialParameter<double> min_increase_factor;
>+  FieldTrialParameter<double> max_increase_factor;
>+  FieldTrialParameter<TimeDelta> increase_low_rtt;
>+  FieldTrialParameter<TimeDelta> increase_high_rtt;
>+  FieldTrialParameter<double> decrease_factor;
>+  FieldTrialParameter<TimeDelta> loss_window;
>+  FieldTrialParameter<TimeDelta> loss_max_window;
>+  FieldTrialParameter<TimeDelta> acknowledged_rate_max_window;
>+  FieldTrialParameter<DataRate> increase_offset;
>+  FieldTrialParameter<DataRate> loss_bandwidth_balance_increase;
>+  FieldTrialParameter<DataRate> loss_bandwidth_balance_decrease;
>+  FieldTrialParameter<double> loss_bandwidth_balance_exponent;
>+  FieldTrialParameter<bool> allow_resets;
>+  FieldTrialParameter<TimeDelta> decrease_interval;
>+  FieldTrialParameter<TimeDelta> loss_report_timeout;
>+};
>+
>+class LossBasedBandwidthEstimation {
>+ public:
>+  LossBasedBandwidthEstimation();
>+  void Update(Timestamp at_time,
>+              DataRate min_bitrate,
>+              TimeDelta last_round_trip_time);
>+  void UpdateAcknowledgedBitrate(DataRate acknowledged_bitrate,
>+                                 Timestamp at_time);
>+  void MaybeReset(DataRate bitrate) {
>+    if (config_.allow_resets)
>+      loss_based_bitrate_ = bitrate;
>+  }
>+  void SetInitialBitrate(DataRate bitrate) { loss_based_bitrate_ = bitrate; }
>+  bool Enabled() const { return config_.enabled; }
>+  void UpdateLossStatistics(const std::vector<PacketResult>& packet_results,
>+                            Timestamp at_time);
>+  DataRate GetEstimate() const { return loss_based_bitrate_; }
>+
>+ private:
>+  LossBasedControlConfig config_;
>+  double average_loss_;
>+  double average_loss_max_;
>+  DataRate loss_based_bitrate_;
>+  DataRate acknowledged_bitrate_max_;
>+  Timestamp acknowledged_bitrate_last_update_;
>+  Timestamp time_last_decrease_;
>+  bool has_decreased_since_last_loss_report_;
>+  Timestamp last_loss_packet_report_;
>+  double last_loss_ratio_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_BITRATE_CONTROLLER_LOSS_BASED_BANDWIDTH_ESTIMATION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
>index cc108a95b08fcb9f59376421e062f6e4155b82d1..394c14747b7acb2ad014a54005d48a557f03b74d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
>@@ -11,12 +11,12 @@
> #include "modules/bitrate_controller/send_side_bandwidth_estimation.h"
> 
> #include <algorithm>
>-#include <cmath>
> #include <cstdio>
> #include <limits>
> #include <string>
> 
> #include "absl/memory/memory.h"
>+#include "logging/rtc_event_log/events/rtc_event.h"
> #include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>@@ -104,6 +104,83 @@ bool ReadBweLossExperimentParameters(float* low_loss_threshold,
> }
> }  // namespace
> 
>+LinkCapacityTracker::LinkCapacityTracker()
>+    : tracking_rate("rate", TimeDelta::seconds(10)) {
>+  ParseFieldTrial({&tracking_rate},
>+                  field_trial::FindFullName("WebRTC-Bwe-LinkCapacity"));
>+}
>+
>+LinkCapacityTracker::~LinkCapacityTracker() {}
>+
>+void LinkCapacityTracker::OnOveruse(DataRate acknowledged_rate,
>+                                    Timestamp at_time) {
>+  capacity_estimate_bps_ =
>+      std::min(capacity_estimate_bps_, acknowledged_rate.bps<double>());
>+  last_link_capacity_update_ = at_time;
>+}
>+
>+void LinkCapacityTracker::OnStartingRate(DataRate start_rate) {
>+  if (last_link_capacity_update_.IsInfinite())
>+    capacity_estimate_bps_ = start_rate.bps<double>();
>+}
>+
>+void LinkCapacityTracker::OnRateUpdate(DataRate acknowledged,
>+                                       Timestamp at_time) {
>+  if (acknowledged.bps() > capacity_estimate_bps_) {
>+    TimeDelta delta = at_time - last_link_capacity_update_;
>+    double alpha = delta.IsFinite() ? exp(-(delta / tracking_rate.Get())) : 0;
>+    capacity_estimate_bps_ = alpha * capacity_estimate_bps_ +
>+                             (1 - alpha) * acknowledged.bps<double>();
>+  }
>+  last_link_capacity_update_ = at_time;
>+}
>+
>+void LinkCapacityTracker::OnRttBackoff(DataRate backoff_rate,
>+                                       Timestamp at_time) {
>+  capacity_estimate_bps_ =
>+      std::min(capacity_estimate_bps_, backoff_rate.bps<double>());
>+  last_link_capacity_update_ = at_time;
>+}
>+
>+DataRate LinkCapacityTracker::estimate() const {
>+  return DataRate::bps(capacity_estimate_bps_);
>+}
>+
>+RttBasedBackoff::RttBasedBackoff()
>+    : rtt_limit_("limit", TimeDelta::PlusInfinity()),
>+      drop_fraction_("fraction", 0.5),
>+      drop_interval_("interval", TimeDelta::ms(300)),
>+      persist_on_route_change_("persist"),
>+      // By initializing this to plus infinity, we make sure that we never
>+      // trigger rtt backoff unless packet feedback is enabled.
>+      last_propagation_rtt_update_(Timestamp::PlusInfinity()),
>+      last_propagation_rtt_(TimeDelta::Zero()) {
>+  ParseFieldTrial({&rtt_limit_, &drop_fraction_, &drop_interval_,
>+                   &persist_on_route_change_},
>+                  field_trial::FindFullName("WebRTC-Bwe-MaxRttLimit"));
>+}
>+
>+void RttBasedBackoff::OnRouteChange() {
>+  if (!persist_on_route_change_) {
>+    last_propagation_rtt_update_ = Timestamp::PlusInfinity();
>+    last_propagation_rtt_ = TimeDelta::Zero();
>+  }
>+}
>+
>+void RttBasedBackoff::UpdatePropagationRtt(Timestamp at_time,
>+                                           TimeDelta propagation_rtt) {
>+  last_propagation_rtt_update_ = at_time;
>+  last_propagation_rtt_ = propagation_rtt;
>+}
>+
>+TimeDelta RttBasedBackoff::RttLowerBound(Timestamp at_time) const {
>+  // TODO(srte): Use time since last unacknowledged packet for this.
>+  TimeDelta time_since_rtt = at_time - last_propagation_rtt_update_;
>+  return time_since_rtt + last_propagation_rtt_;
>+}
>+
>+RttBasedBackoff::~RttBasedBackoff() = default;
>+
> SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
>     : lost_packets_since_last_loss_update_(0),
>       expected_packets_since_last_loss_update_(0),
>@@ -151,14 +228,44 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
> 
> SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
> 
>+void SendSideBandwidthEstimation::OnRouteChange() {
>+  lost_packets_since_last_loss_update_ = 0;
>+  expected_packets_since_last_loss_update_ = 0;
>+  current_bitrate_ = DataRate::Zero();
>+  min_bitrate_configured_ =
>+      DataRate::bps(congestion_controller::GetMinBitrateBps());
>+  max_bitrate_configured_ = kDefaultMaxBitrate;
>+  last_low_bitrate_log_ = Timestamp::MinusInfinity();
>+  has_decreased_since_last_fraction_loss_ = false;
>+  last_loss_feedback_ = Timestamp::MinusInfinity();
>+  last_loss_packet_report_ = Timestamp::MinusInfinity();
>+  last_timeout_ = Timestamp::MinusInfinity();
>+  last_fraction_loss_ = 0;
>+  last_logged_fraction_loss_ = 0;
>+  last_round_trip_time_ = TimeDelta::Zero();
>+  bwe_incoming_ = DataRate::Zero();
>+  delay_based_bitrate_ = DataRate::Zero();
>+  time_last_decrease_ = Timestamp::MinusInfinity();
>+  first_report_time_ = Timestamp::MinusInfinity();
>+  initially_lost_packets_ = 0;
>+  bitrate_at_2_seconds_ = DataRate::Zero();
>+  uma_update_state_ = kNoUpdate;
>+  uma_rtt_state_ = kNoUpdate;
>+  last_rtc_event_log_ = Timestamp::MinusInfinity();
>+
>+  rtt_backoff_.OnRouteChange();
>+}
>+
> void SendSideBandwidthEstimation::SetBitrates(
>     absl::optional<DataRate> send_bitrate,
>     DataRate min_bitrate,
>     DataRate max_bitrate,
>     Timestamp at_time) {
>   SetMinMaxBitrate(min_bitrate, max_bitrate);
>-  if (send_bitrate)
>+  if (send_bitrate) {
>+    link_capacity_.OnStartingRate(*send_bitrate);
>     SetSendBitrate(*send_bitrate, at_time);
>+  }
> }
> 
> void SendSideBandwidthEstimation::SetSendBitrate(DataRate bitrate,
>@@ -166,6 +273,9 @@ void SendSideBandwidthEstimation::SetSendBitrate(DataRate bitrate,
>   RTC_DCHECK(bitrate > DataRate::Zero());
>   // Reset to avoid being capped by the estimate.
>   delay_based_bitrate_ = DataRate::Zero();
>+  if (loss_based_bandwidth_estimation_.Enabled()) {
>+    loss_based_bandwidth_estimation_.MaybeReset(bitrate);
>+  }
>   CapBitrateToThresholds(at_time, bitrate);
>   // Clear last sent bitrate history so the new value can be used directly
>   // and not capped.
>@@ -195,6 +305,10 @@ void SendSideBandwidthEstimation::CurrentEstimate(int* bitrate,
>   *rtt = last_round_trip_time_.ms<int64_t>();
> }
> 
>+DataRate SendSideBandwidthEstimation::GetEstimatedLinkCapacity() const {
>+  return link_capacity_.estimate();
>+}
>+
> void SendSideBandwidthEstimation::UpdateReceiverEstimate(Timestamp at_time,
>                                                          DataRate bandwidth) {
>   bwe_incoming_ = bandwidth;
>@@ -203,10 +317,33 @@ void SendSideBandwidthEstimation::UpdateReceiverEstimate(Timestamp at_time,
> 
> void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(Timestamp at_time,
>                                                            DataRate bitrate) {
>+  if (acknowledged_rate_) {
>+    if (bitrate < delay_based_bitrate_) {
>+      link_capacity_.OnOveruse(*acknowledged_rate_, at_time);
>+    }
>+  }
>   delay_based_bitrate_ = bitrate;
>   CapBitrateToThresholds(at_time, current_bitrate_);
> }
> 
>+void SendSideBandwidthEstimation::SetAcknowledgedRate(
>+    absl::optional<DataRate> acknowledged_rate,
>+    Timestamp at_time) {
>+  acknowledged_rate_ = acknowledged_rate;
>+  if (acknowledged_rate && loss_based_bandwidth_estimation_.Enabled()) {
>+    loss_based_bandwidth_estimation_.UpdateAcknowledgedBitrate(
>+        *acknowledged_rate, at_time);
>+  }
>+}
>+
>+void SendSideBandwidthEstimation::IncomingPacketFeedbackVector(
>+    const TransportPacketsFeedback& report) {
>+  if (loss_based_bandwidth_estimation_.Enabled()) {
>+    loss_based_bandwidth_estimation_.UpdateLossStatistics(
>+        report.packet_feedbacks, report.feedback_time);
>+  }
>+}
>+
> void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
>                                                       TimeDelta rtt,
>                                                       int number_of_packets,
>@@ -295,11 +432,24 @@ void SendSideBandwidthEstimation::UpdateRtt(TimeDelta rtt, Timestamp at_time) {
> 
> void SendSideBandwidthEstimation::UpdateEstimate(Timestamp at_time) {
>   DataRate new_bitrate = current_bitrate_;
>+  if (rtt_backoff_.RttLowerBound(at_time) > rtt_backoff_.rtt_limit_) {
>+    if (at_time - time_last_decrease_ >= rtt_backoff_.drop_interval_) {
>+      time_last_decrease_ = at_time;
>+      new_bitrate = current_bitrate_ * rtt_backoff_.drop_fraction_;
>+      link_capacity_.OnRttBackoff(new_bitrate, at_time);
>+    }
>+    CapBitrateToThresholds(at_time, new_bitrate);
>+    return;
>+  }
>+
>   // We trust the REMB and/or delay-based estimate during the first 2 seconds if
>   // we haven't had any packet loss reported, to allow startup bitrate probing.
>   if (last_fraction_loss_ == 0 && IsInStartPhase(at_time)) {
>     new_bitrate = std::max(bwe_incoming_, new_bitrate);
>     new_bitrate = std::max(delay_based_bitrate_, new_bitrate);
>+    if (loss_based_bandwidth_estimation_.Enabled()) {
>+      loss_based_bandwidth_estimation_.SetInitialBitrate(new_bitrate);
>+    }
> 
>     if (new_bitrate != current_bitrate_) {
>       min_bitrate_history_.clear();
>@@ -314,6 +464,15 @@ void SendSideBandwidthEstimation::UpdateEstimate(Timestamp at_time) {
>     CapBitrateToThresholds(at_time, current_bitrate_);
>     return;
>   }
>+
>+  if (loss_based_bandwidth_estimation_.Enabled()) {
>+    loss_based_bandwidth_estimation_.Update(
>+        at_time, min_bitrate_history_.front().second, last_round_trip_time_);
>+    new_bitrate = MaybeRampupOrBackoff(new_bitrate, at_time);
>+    CapBitrateToThresholds(at_time, new_bitrate);
>+    return;
>+  }
>+
>   TimeDelta time_since_loss_packet_report = at_time - last_loss_packet_report_;
>   TimeDelta time_since_loss_feedback = at_time - last_loss_feedback_;
>   if (time_since_loss_packet_report < 1.2 * kMaxRtcpFeedbackInterval) {
>@@ -328,7 +487,7 @@ void SendSideBandwidthEstimation::UpdateEstimate(Timestamp at_time) {
>       // Note that by remembering the bitrate over the last second one can
>       // rampup up one second faster than if only allowed to start ramping
>       // at 8% per second rate now. E.g.:
>-      //   If sending a constant 100kbps it can rampup immediatly to 108kbps
>+      //   If sending a constant 100kbps it can rampup immediately to 108kbps
>       //   whenever a receiver report is received with lower packet loss.
>       //   If instead one would do: current_bitrate_ *= 1.08^(delta time),
>       //   it would take over one second since the lower packet loss to achieve
>@@ -382,6 +541,12 @@ void SendSideBandwidthEstimation::UpdateEstimate(Timestamp at_time) {
>   CapBitrateToThresholds(at_time, new_bitrate);
> }
> 
>+void SendSideBandwidthEstimation::UpdatePropagationRtt(
>+    Timestamp at_time,
>+    TimeDelta propagation_rtt) {
>+  rtt_backoff_.UpdatePropagationRtt(at_time, propagation_rtt);
>+}
>+
> bool SendSideBandwidthEstimation::IsInStartPhase(Timestamp at_time) const {
>   return first_report_time_.IsInfinite() ||
>          at_time - first_report_time_ < kStartPhase;
>@@ -407,6 +572,35 @@ void SendSideBandwidthEstimation::UpdateMinHistory(Timestamp at_time) {
>   min_bitrate_history_.push_back(std::make_pair(at_time, current_bitrate_));
> }
> 
>+DataRate SendSideBandwidthEstimation::MaybeRampupOrBackoff(DataRate new_bitrate,
>+                                                           Timestamp at_time) {
>+  // TODO(crodbro): reuse this code in UpdateEstimate instead of current
>+  // inlining of very similar functionality.
>+  const TimeDelta time_since_loss_packet_report =
>+      at_time - last_loss_packet_report_;
>+  const TimeDelta time_since_loss_feedback = at_time - last_loss_feedback_;
>+  if (time_since_loss_packet_report < 1.2 * kMaxRtcpFeedbackInterval) {
>+    new_bitrate = min_bitrate_history_.front().second * 1.08;
>+    new_bitrate += DataRate::bps(1000);
>+  } else if (time_since_loss_feedback >
>+                 kFeedbackTimeoutIntervals * kMaxRtcpFeedbackInterval &&
>+             (last_timeout_.IsInfinite() ||
>+              at_time - last_timeout_ > kTimeoutInterval)) {
>+    if (in_timeout_experiment_) {
>+      RTC_LOG(LS_WARNING) << "Feedback timed out ("
>+                          << ToString(time_since_loss_feedback)
>+                          << "), reducing bitrate.";
>+      new_bitrate = new_bitrate * 0.8;
>+      // Reset accumulators since we've already acted on missing feedback and
>+      // shouldn't to act again on these old lost packets.
>+      lost_packets_since_last_loss_update_ = 0;
>+      expected_packets_since_last_loss_update_ = 0;
>+      last_timeout_ = at_time;
>+    }
>+  }
>+  return new_bitrate;
>+}
>+
> void SendSideBandwidthEstimation::CapBitrateToThresholds(Timestamp at_time,
>                                                          DataRate bitrate) {
>   if (bwe_incoming_ > DataRate::Zero() && bitrate > bwe_incoming_) {
>@@ -416,6 +610,10 @@ void SendSideBandwidthEstimation::CapBitrateToThresholds(Timestamp at_time,
>       bitrate > delay_based_bitrate_) {
>     bitrate = delay_based_bitrate_;
>   }
>+  if (loss_based_bandwidth_estimation_.Enabled() &&
>+      loss_based_bandwidth_estimation_.GetEstimate() > DataRate::Zero()) {
>+    bitrate = std::min(bitrate, loss_based_bandwidth_estimation_.GetEstimate());
>+  }
>   if (bitrate > max_bitrate_configured_) {
>     bitrate = max_bitrate_configured_;
>   }
>@@ -441,5 +639,10 @@ void SendSideBandwidthEstimation::CapBitrateToThresholds(Timestamp at_time,
>     last_rtc_event_log_ = at_time;
>   }
>   current_bitrate_ = bitrate;
>+
>+  if (acknowledged_rate_) {
>+    link_capacity_.OnRateUpdate(std::min(current_bitrate_, *acknowledged_rate_),
>+                                at_time);
>+  }
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
>index 2c8b4ee3d5aba0463329d7c24b8200323b6e49b7..b016faba2ee3f987951452faa1ea79048de15678 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
>@@ -13,27 +13,71 @@
> #ifndef MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
> #define MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
> 
>+#include <stdint.h>
> #include <deque>
> #include <utility>
> #include <vector>
> 
> #include "absl/types/optional.h"
>+#include "api/transport/network_types.h"
>+#include "api/units/data_rate.h"
>+#include "api/units/time_delta.h"
>+#include "api/units/timestamp.h"
>+#include "modules/bitrate_controller/loss_based_bandwidth_estimation.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "rtc_base/experiments/field_trial_parser.h"
> 
> namespace webrtc {
> 
> class RtcEventLog;
> 
>+class LinkCapacityTracker {
>+ public:
>+  LinkCapacityTracker();
>+  ~LinkCapacityTracker();
>+  void OnOveruse(DataRate acknowledged_rate, Timestamp at_time);
>+  void OnStartingRate(DataRate start_rate);
>+  void OnRateUpdate(DataRate acknowledged, Timestamp at_time);
>+  void OnRttBackoff(DataRate backoff_rate, Timestamp at_time);
>+  DataRate estimate() const;
>+
>+ private:
>+  FieldTrialParameter<TimeDelta> tracking_rate;
>+  double capacity_estimate_bps_ = 0;
>+  Timestamp last_link_capacity_update_ = Timestamp::MinusInfinity();
>+};
>+
>+class RttBasedBackoff {
>+ public:
>+  RttBasedBackoff();
>+  ~RttBasedBackoff();
>+  void OnRouteChange();
>+  void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt);
>+  TimeDelta RttLowerBound(Timestamp at_time) const;
>+
>+  FieldTrialParameter<TimeDelta> rtt_limit_;
>+  FieldTrialParameter<double> drop_fraction_;
>+  FieldTrialParameter<TimeDelta> drop_interval_;
>+  FieldTrialFlag persist_on_route_change_;
>+
>+ public:
>+  Timestamp last_propagation_rtt_update_;
>+  TimeDelta last_propagation_rtt_;
>+};
>+
> class SendSideBandwidthEstimation {
>  public:
>   SendSideBandwidthEstimation() = delete;
>   explicit SendSideBandwidthEstimation(RtcEventLog* event_log);
>-  virtual ~SendSideBandwidthEstimation();
>+  ~SendSideBandwidthEstimation();
> 
>+  void OnRouteChange();
>   void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
>-
>+  DataRate GetEstimatedLinkCapacity() const;
>   // Call periodically to update estimate.
>   void UpdateEstimate(Timestamp at_time);
>+  void OnSentPacket(SentPacket sent_packet);
>+  void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt);
> 
>   // Call when we receive a RTCP message with TMMBR or REMB.
>   void UpdateReceiverEstimate(Timestamp at_time, DataRate bandwidth);
>@@ -62,6 +106,9 @@ class SendSideBandwidthEstimation {
>   void SetSendBitrate(DataRate bitrate, Timestamp at_time);
>   void SetMinMaxBitrate(DataRate min_bitrate, DataRate max_bitrate);
>   int GetMinBitrate() const;
>+  void SetAcknowledgedRate(absl::optional<DataRate> acknowledged_rate,
>+                           Timestamp at_time);
>+  void IncomingPacketFeedbackVector(const TransportPacketsFeedback& report);
> 
>  private:
>   enum UmaState { kNoUpdate, kFirstDone, kDone };
>@@ -75,16 +122,22 @@ class SendSideBandwidthEstimation {
>   // min bitrate used during last kBweIncreaseIntervalMs.
>   void UpdateMinHistory(Timestamp at_time);
> 
>+  DataRate MaybeRampupOrBackoff(DataRate new_bitrate, Timestamp at_time);
>+
>   // Cap |bitrate| to [min_bitrate_configured_, max_bitrate_configured_] and
>   // set |current_bitrate_| to the capped value and updates the event log.
>   void CapBitrateToThresholds(Timestamp at_time, DataRate bitrate);
> 
>+  RttBasedBackoff rtt_backoff_;
>+  LinkCapacityTracker link_capacity_;
>+
>   std::deque<std::pair<Timestamp, DataRate> > min_bitrate_history_;
> 
>   // incoming filters
>   int lost_packets_since_last_loss_update_;
>   int expected_packets_since_last_loss_update_;
> 
>+  absl::optional<DataRate> acknowledged_rate_;
>   DataRate current_bitrate_;
>   DataRate min_bitrate_configured_;
>   DataRate max_bitrate_configured_;
>@@ -113,6 +166,7 @@ class SendSideBandwidthEstimation {
>   float low_loss_threshold_;
>   float high_loss_threshold_;
>   DataRate bitrate_threshold_;
>+  LossBasedBandwidthEstimation loss_based_bandwidth_estimation_;
> };
> }  // namespace webrtc
> #endif  // MODULES_BITRATE_CONTROLLER_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
>index becc616cf31561b3b4e869158e3e0d48246b74af..ccfbd8550327e86fa1757248f7409b5be1b46fb7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
>@@ -23,7 +23,7 @@ MATCHER(LossBasedBweUpdateWithBitrateOnly, "") {
>     return false;
>   }
>   auto bwe_event = static_cast<RtcEventBweUpdateLossBased*>(arg);
>-  return bwe_event->bitrate_bps_ > 0 && bwe_event->fraction_loss_ == 0;
>+  return bwe_event->bitrate_bps() > 0 && bwe_event->fraction_loss() == 0;
> }
> 
> MATCHER(LossBasedBweUpdateWithBitrateAndLossFraction, "") {
>@@ -31,7 +31,7 @@ MATCHER(LossBasedBweUpdateWithBitrateAndLossFraction, "") {
>     return false;
>   }
>   auto bwe_event = static_cast<RtcEventBweUpdateLossBased*>(arg);
>-  return bwe_event->bitrate_bps_ > 0 && bwe_event->fraction_loss_ > 0;
>+  return bwe_event->bitrate_bps() > 0 && bwe_event->fraction_loss() > 0;
> }
> 
> void TestProbing(bool use_delay_based) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/BUILD.gn
>index 2fcb71212f5247cdab800b22571f0186a05e98f7..73d81f16ec27cb021ab9f16f73d9898c6d977385 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/BUILD.gn
>@@ -20,8 +20,6 @@ rtc_static_library("congestion_controller") {
>   visibility = [ "*" ]
>   configs += [ ":bwe_test_logging" ]
>   sources = [
>-    "congestion_window_pushback_controller.cc",
>-    "congestion_window_pushback_controller.h",
>     "include/network_changed_observer.h",
>     "include/receive_side_congestion_controller.h",
>     "include/send_side_congestion_controller.h",
>@@ -43,6 +41,7 @@ rtc_static_library("congestion_controller") {
>     "../../rtc_base:checks",
>     "../../rtc_base:ptr_util",
>     "../../rtc_base:rate_limiter",
>+    "../../rtc_base/network:sent_packet",
>     "../../system_wrappers",
>     "../../system_wrappers:field_trial",
>     "../bitrate_controller",
>@@ -52,6 +51,7 @@ rtc_static_library("congestion_controller") {
>     "goog_cc:delay_based_bwe",
>     "goog_cc:estimators",
>     "goog_cc:probe_controller",
>+    "goog_cc:pushback_controller",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
> 
>@@ -98,7 +98,6 @@ if (rtc_include_tests) {
>     testonly = true
> 
>     sources = [
>-      "congestion_window_pushback_controller_unittest.cc",
>       "receive_side_congestion_controller_unittest.cc",
>       "send_side_congestion_controller_unittest.cc",
>       "transport_feedback_adapter_unittest.cc",
>@@ -111,7 +110,7 @@ if (rtc_include_tests) {
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base",
>       "../../rtc_base:rtc_base_approved",
>-      "../../rtc_base:rtc_base_tests_utils",
>+      "../../rtc_base/network:sent_packet",
>       "../../system_wrappers",
>       "../../test:field_trial",
>       "../../test:test_support",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.cc
>index b82ded42202cc4740eaf993c05aa7e743cb8ac0f..863ce4bf98d9692292cf8fea490c27c4b63044f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.cc
>@@ -401,16 +401,15 @@ bool BbrNetworkController::IsProbingForMoreBandwidth() const {
> 
> NetworkControlUpdate BbrNetworkController::OnTransportPacketsFeedback(
>     TransportPacketsFeedback msg) {
>+  if (msg.packet_feedbacks.empty())
>+    return NetworkControlUpdate();
>+
>   Timestamp feedback_recv_time = msg.feedback_time;
>-  absl::optional<SentPacket> last_sent_packet =
>-      msg.PacketsWithFeedback().back().sent_packet;
>-  if (!last_sent_packet.has_value()) {
>-    RTC_LOG(LS_WARNING) << "Last ack packet not in history, no RTT update";
>-  } else {
>-    Timestamp send_time = last_sent_packet->send_time;
>-    TimeDelta send_delta = feedback_recv_time - send_time;
>-    rtt_stats_.UpdateRtt(send_delta, TimeDelta::Zero(), feedback_recv_time);
>-  }
>+  SentPacket last_sent_packet = msg.PacketsWithFeedback().back().sent_packet;
>+
>+  Timestamp send_time = last_sent_packet.send_time;
>+  TimeDelta send_delta = feedback_recv_time - send_time;
>+  rtt_stats_.UpdateRtt(send_delta, TimeDelta::Zero(), feedback_recv_time);
> 
>   const DataSize total_data_acked_before = sampler_->total_data_acked();
> 
>@@ -431,7 +430,7 @@ NetworkControlUpdate BbrNetworkController::OnTransportPacketsFeedback(
>   // Input the new data into the BBR model of the connection.
>   if (!acked_packets.empty()) {
>     int64_t last_acked_packet =
>-        acked_packets.rbegin()->sent_packet->sequence_number;
>+        acked_packets.rbegin()->sent_packet.sequence_number;
> 
>     is_round_start = UpdateRoundTripCounter(last_acked_packet);
>     min_rtt_expired =
>@@ -472,7 +471,7 @@ NetworkControlUpdate BbrNetworkController::OnTransportPacketsFeedback(
>   DataSize data_acked = sampler_->total_data_acked() - total_data_acked_before;
>   DataSize data_lost = DataSize::Zero();
>   for (const PacketResult& packet : lost_packets) {
>-    data_lost += packet.sent_packet->size;
>+    data_lost += packet.sent_packet.size;
>   }
> 
>   // After the model is updated, recalculate the pacing rate and congestion
>@@ -483,7 +482,7 @@ NetworkControlUpdate BbrNetworkController::OnTransportPacketsFeedback(
>   // Cleanup internal state.
>   if (!acked_packets.empty()) {
>     sampler_->RemoveObsoletePackets(
>-        acked_packets.back().sent_packet->sequence_number);
>+        acked_packets.back().sent_packet.sequence_number);
>   }
>   return CreateRateUpdate(msg.feedback_time);
> }
>@@ -550,7 +549,7 @@ void BbrNetworkController::EnterProbeBandwidthMode(Timestamp now) {
> void BbrNetworkController::DiscardLostPackets(
>     const std::vector<PacketResult>& lost_packets) {
>   for (const PacketResult& packet : lost_packets) {
>-    sampler_->OnPacketLost(packet.sent_packet->sequence_number);
>+    sampler_->OnPacketLost(packet.sent_packet.sequence_number);
>   }
> }
> 
>@@ -569,8 +568,8 @@ bool BbrNetworkController::UpdateBandwidthAndMinRtt(
>     const std::vector<PacketResult>& acked_packets) {
>   TimeDelta sample_rtt = TimeDelta::PlusInfinity();
>   for (const auto& packet : acked_packets) {
>-    BandwidthSample bandwidth_sample = sampler_->OnPacketAcknowledged(
>-        now, packet.sent_packet->sequence_number);
>+    BandwidthSample bandwidth_sample =
>+        sampler_->OnPacketAcknowledged(now, packet.sent_packet.sequence_number);
>     last_sample_is_app_limited_ = bandwidth_sample.is_app_limited;
>     if (!bandwidth_sample.rtt.IsZero()) {
>       sample_rtt = std::min(sample_rtt, bandwidth_sample.rtt);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.cc
>deleted file mode 100644
>index daa51bddd30977992d8ad0e9f8529377be6d3ae4..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.cc
>+++ /dev/null
>@@ -1,93 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <string>
>-#include <algorithm>
>-
>-#include "modules/congestion_controller/congestion_window_pushback_controller.h"
>-#include "system_wrappers/include/field_trial.h"
>-
>-namespace webrtc {
>-
>-// When CongestionWindowPushback is enabled, the pacer is oblivious to
>-// the congestion window. The relation between outstanding data and
>-// the congestion window affects encoder allocations directly.
>-// This experiment is build on top of congestion window experiment.
>-const char kCongestionPushbackExperiment[] = "WebRTC-CongestionWindowPushback";
>-const uint32_t kDefaultMinPushbackTargetBitrateBps = 30000;
>-
>-bool ReadCongestionWindowPushbackExperimentParameter(
>-    uint32_t* min_pushback_target_bitrate_bps) {
>-  RTC_DCHECK(min_pushback_target_bitrate_bps);
>-  std::string experiment_string =
>-      webrtc::field_trial::FindFullName(kCongestionPushbackExperiment);
>-  int parsed_values = sscanf(experiment_string.c_str(), "Enabled-%" PRIu32,
>-                             min_pushback_target_bitrate_bps);
>-  if (parsed_values == 1) {
>-    RTC_CHECK_GE(*min_pushback_target_bitrate_bps, 0)
>-        << "Min pushback target bitrate must be greater than or equal to 0.";
>-    return true;
>-  }
>-  return false;
>-}
>-
>-CongestionWindowPushbackController::CongestionWindowPushbackController() {
>-  if (!ReadCongestionWindowPushbackExperimentParameter(
>-          &min_pushback_target_bitrate_bps_)) {
>-    min_pushback_target_bitrate_bps_ = kDefaultMinPushbackTargetBitrateBps;
>-  }
>-}
>-
>-void CongestionWindowPushbackController::UpdateOutstandingData(
>-    size_t outstanding_bytes) {
>-  outstanding_bytes_ = outstanding_bytes;
>-}
>-
>-void CongestionWindowPushbackController::UpdateMaxOutstandingData(
>-    size_t max_outstanding_bytes) {
>-  DataSize data_window = DataSize::bytes(max_outstanding_bytes);
>-  if (current_data_window_) {
>-    data_window = (data_window + current_data_window_.value()) / 2;
>-  }
>-  current_data_window_ = data_window;
>-}
>-
>-void CongestionWindowPushbackController::SetDataWindow(DataSize data_window) {
>-  current_data_window_ = data_window;
>-}
>-
>-uint32_t CongestionWindowPushbackController::UpdateTargetBitrate(
>-    uint32_t bitrate_bps) {
>-  if (!current_data_window_ || current_data_window_->IsZero())
>-    return bitrate_bps;
>-  double fill_ratio =
>-      outstanding_bytes_ / static_cast<double>(current_data_window_->bytes());
>-  if (fill_ratio > 1.5) {
>-    encoding_rate_ratio_ *= 0.9;
>-  } else if (fill_ratio > 1) {
>-    encoding_rate_ratio_ *= 0.95;
>-  } else if (fill_ratio < 0.1) {
>-    encoding_rate_ratio_ = 1.0;
>-  } else {
>-    encoding_rate_ratio_ *= 1.05;
>-    encoding_rate_ratio_ = std::min(encoding_rate_ratio_, 1.0);
>-  }
>-  uint32_t adjusted_target_bitrate_bps =
>-      static_cast<uint32_t>(bitrate_bps * encoding_rate_ratio_);
>-
>-  // Do not adjust below the minimum pushback bitrate but do obey if the
>-  // original estimate is below it.
>-  bitrate_bps = adjusted_target_bitrate_bps < min_pushback_target_bitrate_bps_
>-                    ? std::min(bitrate_bps, min_pushback_target_bitrate_bps_)
>-                    : adjusted_target_bitrate_bps;
>-  return bitrate_bps;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.h
>deleted file mode 100644
>index f608590583bdb811f846a43ca9e3588745392488..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.h
>+++ /dev/null
>@@ -1,43 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_CONGESTION_CONTROLLER_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
>-#define MODULES_CONGESTION_CONTROLLER_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
>-
>-#include "api/transport/network_types.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "rtc_base/criticalsection.h"
>-#include "rtc_base/format_macros.h"
>-
>-namespace webrtc {
>-
>-// This class enables pushback from congestion window directly to video encoder.
>-// When the congestion window is filling up, the video encoder target bitrate
>-// will be reduced accordingly to accommodate the network changes. To avoid
>-// pausing video too frequently, a minimum encoder target bitrate threshold is
>-// used to prevent video pause due to a full congestion window.
>-class CongestionWindowPushbackController {
>- public:
>-  CongestionWindowPushbackController();
>-  void UpdateOutstandingData(size_t outstanding_bytes);
>-  void UpdateMaxOutstandingData(size_t max_outstanding_bytes);
>-  uint32_t UpdateTargetBitrate(uint32_t bitrate_bps);
>-  void SetDataWindow(DataSize data_window);
>-
>- private:
>-  absl::optional<DataSize> current_data_window_;
>-  size_t outstanding_bytes_ = 0;
>-  uint32_t min_pushback_target_bitrate_bps_;
>-  double encoding_rate_ratio_ = 1.0;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_CONGESTION_CONTROLLER_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller_unittest.cc
>deleted file mode 100644
>index d9d6a79d62ffbc60bdd8a558d5e01b9524b54f1d..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller_unittest.cc
>+++ /dev/null
>@@ -1,65 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/congestion_controller/congestion_window_pushback_controller.h"
>-#include "test/gmock.h"
>-#include "test/gtest.h"
>-
>-using testing::_;
>-
>-namespace webrtc {
>-namespace test {
>-
>-class CongestionWindowPushbackControllerTest : public ::testing::Test {
>- protected:
>-  CongestionWindowPushbackController cwnd_controller_;
>-};
>-
>-TEST_F(CongestionWindowPushbackControllerTest, FullCongestionWindow) {
>-  cwnd_controller_.UpdateOutstandingData(100000);
>-  cwnd_controller_.UpdateMaxOutstandingData(50000);
>-
>-  uint32_t bitrate_bps = 80000;
>-  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>-  EXPECT_EQ(72000u, bitrate_bps);
>-
>-  cwnd_controller_.UpdateMaxOutstandingData(50000);
>-  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>-  EXPECT_EQ(static_cast<uint32_t>(72000 * 0.9 * 0.9), bitrate_bps);
>-}
>-
>-TEST_F(CongestionWindowPushbackControllerTest, NormalCongestionWindow) {
>-  cwnd_controller_.UpdateOutstandingData(100000);
>-  cwnd_controller_.SetDataWindow(DataSize::bytes(200000));
>-
>-  uint32_t bitrate_bps = 80000;
>-  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>-  EXPECT_EQ(80000u, bitrate_bps);
>-
>-  cwnd_controller_.UpdateMaxOutstandingData(20000);
>-  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>-  EXPECT_EQ(80000u, bitrate_bps);
>-}
>-
>-TEST_F(CongestionWindowPushbackControllerTest, LowBitrate) {
>-  cwnd_controller_.UpdateOutstandingData(100000);
>-  cwnd_controller_.SetDataWindow(DataSize::bytes(50000));
>-
>-  uint32_t bitrate_bps = 35000;
>-  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>-  EXPECT_EQ(static_cast<uint32_t>(35000 * 0.9), bitrate_bps);
>-
>-  cwnd_controller_.UpdateMaxOutstandingData(20000);
>-  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>-  EXPECT_EQ(30000u, bitrate_bps);
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/BUILD.gn
>index b12690b817e13cccb5e0063c937323ca75b0aec2..38169bbb4d029a3e85706ca6b9c4d79b7a745e3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/BUILD.gn
>@@ -19,10 +19,8 @@ config("bwe_test_logging") {
> rtc_static_library("goog_cc") {
>   configs += [ ":bwe_test_logging" ]
>   sources = [
>-    "goog_cc_factory.cc",
>     "goog_cc_network_control.cc",
>     "goog_cc_network_control.h",
>-    "include/goog_cc_factory.h",
>   ]
> 
>   deps = [
>@@ -30,6 +28,7 @@ rtc_static_library("goog_cc") {
>     ":delay_based_bwe",
>     ":estimators",
>     ":probe_controller",
>+    ":pushback_controller",
>     "../..:module_api",
>     "../../..:webrtc_common",
>     "../../../api/transport:network_control",
>@@ -49,6 +48,20 @@ rtc_static_library("goog_cc") {
>   ]
> }
> 
>+rtc_source_set("pushback_controller") {
>+  sources = [
>+    "congestion_window_pushback_controller.cc",
>+    "congestion_window_pushback_controller.h",
>+  ]
>+  deps = [
>+    "../../../api/transport:network_control",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../system_wrappers:field_trial",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
> rtc_source_set("alr_detector") {
>   sources = [
>     "alr_detector.cc",
>@@ -84,6 +97,7 @@ rtc_source_set("estimators") {
>   ]
> 
>   deps = [
>+    "../../../api/units:data_rate",
>     "../../../logging:rtc_event_bwe",
>     "../../../logging:rtc_event_log_api",
>     "../../../rtc_base:checks",
>@@ -107,6 +121,7 @@ rtc_source_set("delay_based_bwe") {
> 
>   deps = [
>     ":estimators",
>+    "../../../api/transport:network_control",
>     "../../../logging:rtc_event_bwe",
>     "../../../logging:rtc_event_log_api",
>     "../../../rtc_base:checks",
>@@ -115,7 +130,9 @@ rtc_source_set("delay_based_bwe") {
>     "../../../system_wrappers:metrics",
>     "../../pacing",
>     "../../remote_bitrate_estimator",
>+    "../../rtp_rtcp:rtp_rtcp_format",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -149,6 +166,7 @@ if (rtc_include_tests) {
>       ":estimators",
>       ":goog_cc",
>       "..:test_controller_printer",
>+      "../../../api/transport:goog_cc",
>     ]
>   }
>   rtc_source_set("goog_cc_unittests") {
>@@ -157,6 +175,7 @@ if (rtc_include_tests) {
>     sources = [
>       "acknowledged_bitrate_estimator_unittest.cc",
>       "alr_detector_unittest.cc",
>+      "congestion_window_pushback_controller_unittest.cc",
>       "delay_based_bwe_unittest.cc",
>       "delay_based_bwe_unittest_helper.cc",
>       "delay_based_bwe_unittest_helper.h",
>@@ -175,6 +194,8 @@ if (rtc_include_tests) {
>       ":estimators",
>       ":goog_cc",
>       ":probe_controller",
>+      ":pushback_controller",
>+      "../../../api/transport:goog_cc",
>       "../../../api/transport:network_control",
>       "../../../api/transport:network_control_test",
>       "../../../logging:mocks",
>@@ -194,4 +215,39 @@ if (rtc_include_tests) {
>       "//third_party/abseil-cpp/absl/memory",
>     ]
>   }
>+  rtc_source_set("goog_cc_slow_tests") {
>+    testonly = true
>+
>+    sources = [
>+      "goog_cc_network_control_slowtest.cc",
>+    ]
>+    if (!build_with_chromium && is_clang) {
>+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+    }
>+    deps = [
>+      ":alr_detector",
>+      ":delay_based_bwe",
>+      ":estimators",
>+      ":goog_cc",
>+      ":probe_controller",
>+      ":pushback_controller",
>+      "../../../api/transport:goog_cc",
>+      "../../../api/transport:network_control",
>+      "../../../api/transport:network_control_test",
>+      "../../../logging:mocks",
>+      "../../../rtc_base:checks",
>+      "../../../rtc_base:rtc_base_approved",
>+      "../../../rtc_base/experiments:alr_experiment",
>+      "../../../system_wrappers",
>+      "../../../system_wrappers:field_trial",
>+      "../../../test:field_trial",
>+      "../../../test:test_support",
>+      "../../../test/scenario",
>+      "../../pacing",
>+      "../../remote_bitrate_estimator",
>+      "../../rtp_rtcp:rtp_rtcp_format",
>+      "//testing/gmock",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
>index 0722b2218bd597f7873f8a178766de6975579735..939da4cc33ceafd0b6b62d44f7fc6ce642ab7c8b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
>@@ -10,11 +10,15 @@
> 
> #include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
> 
>+#include <stddef.h>
>+#include <algorithm>
> #include <utility>
> 
> #include "absl/memory/memory.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
>@@ -31,7 +35,9 @@ AcknowledgedBitrateEstimator::~AcknowledgedBitrateEstimator() {}
> 
> AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
>     std::unique_ptr<BitrateEstimator> bitrate_estimator)
>-    : bitrate_estimator_(std::move(bitrate_estimator)) {}
>+    : account_for_unacknowledged_traffic_(
>+          field_trial::IsEnabled("WebRTC-Bwe-AccountForUnacked")),
>+      bitrate_estimator_(std::move(bitrate_estimator)) {}
> 
> void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
>     const std::vector<PacketFeedback>& packet_feedback_vector) {
>@@ -41,18 +47,41 @@ void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
>   for (const auto& packet : packet_feedback_vector) {
>     if (IsInSendTimeHistory(packet)) {
>       MaybeExpectFastRateChange(packet.send_time_ms);
>-      bitrate_estimator_->Update(packet.arrival_time_ms,
>-                                 rtc::dchecked_cast<int>(packet.payload_size));
>+      int acknowledged_estimate = rtc::dchecked_cast<int>(packet.payload_size);
>+      if (account_for_unacknowledged_traffic_)
>+        acknowledged_estimate += packet.unacknowledged_data;
>+      bitrate_estimator_->Update(packet.arrival_time_ms, acknowledged_estimate);
>     }
>   }
> }
> 
> absl::optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
>   auto estimated_bitrate = bitrate_estimator_->bitrate_bps();
>-  return estimated_bitrate
>-             ? *estimated_bitrate +
>-                   allocated_bitrate_without_feedback_bps_.value_or(0)
>-             : estimated_bitrate;
>+  // If we account for unacknowledged traffic, we should not add the allocated
>+  // bitrate for unallocated stream as we expect it to be included already.
>+  if (account_for_unacknowledged_traffic_) {
>+    return estimated_bitrate;
>+  } else {
>+    return estimated_bitrate
>+               ? *estimated_bitrate + allocated_bitrate_without_feedback_bps_
>+               : estimated_bitrate;
>+  }
>+}
>+
>+absl::optional<uint32_t> AcknowledgedBitrateEstimator::PeekBps() const {
>+  return bitrate_estimator_->PeekBps();
>+}
>+
>+absl::optional<DataRate> AcknowledgedBitrateEstimator::bitrate() const {
>+  if (bitrate_bps())
>+    return DataRate::bps(*bitrate_bps());
>+  return absl::nullopt;
>+}
>+
>+absl::optional<DataRate> AcknowledgedBitrateEstimator::PeekRate() const {
>+  if (PeekBps())
>+    return DataRate::bps(*PeekBps());
>+  return absl::nullopt;
> }
> 
> void AcknowledgedBitrateEstimator::SetAlrEndedTimeMs(
>@@ -62,7 +91,7 @@ void AcknowledgedBitrateEstimator::SetAlrEndedTimeMs(
> 
> void AcknowledgedBitrateEstimator::SetAllocatedBitrateWithoutFeedback(
>     uint32_t bitrate_bps) {
>-  allocated_bitrate_without_feedback_bps_.emplace(bitrate_bps);
>+  allocated_bitrate_without_feedback_bps_ = bitrate_bps;
> }
> 
> void AcknowledgedBitrateEstimator::MaybeExpectFastRateChange(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
>index ccb9718b2509e6ac783fe6935473963af2942ea7..645c0b913a7e04e6b7a0d51600b73388e9d2a4d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
>@@ -17,6 +17,8 @@
> #include "absl/types/optional.h"
> #include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
> 
>+#include "api/units/data_rate.h"
>+
> namespace webrtc {
> 
> struct PacketFeedback;
>@@ -32,14 +34,18 @@ class AcknowledgedBitrateEstimator {
>   void IncomingPacketFeedbackVector(
>       const std::vector<PacketFeedback>& packet_feedback_vector);
>   absl::optional<uint32_t> bitrate_bps() const;
>+  absl::optional<uint32_t> PeekBps() const;
>+  absl::optional<DataRate> bitrate() const;
>+  absl::optional<DataRate> PeekRate() const;
>   void SetAlrEndedTimeMs(int64_t alr_ended_time_ms);
>   void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps);
> 
>  private:
>   void MaybeExpectFastRateChange(int64_t packet_arrival_time_ms);
>+  const bool account_for_unacknowledged_traffic_;
>   absl::optional<int64_t> alr_ended_time_ms_;
>   std::unique_ptr<BitrateEstimator> bitrate_estimator_;
>-  absl::optional<uint32_t> allocated_bitrate_without_feedback_bps_;
>+  uint32_t allocated_bitrate_without_feedback_bps_ = 0;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.cc
>index ed8f3a66f4128c8c4bef81a4ee7564389bcdbf7f..db669428f022123c551f7704f7fa54d68a6d4e76 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.cc
>@@ -10,20 +10,17 @@
> 
> #include "modules/congestion_controller/goog_cc/alr_detector.h"
> 
>-#include <algorithm>
>+#include <cstdint>
> #include <cstdio>
>-#include <string>
> 
> #include "absl/memory/memory.h"
>+#include "logging/rtc_event_log/events/rtc_event.h"
> #include "logging/rtc_event_log/events/rtc_event_alr_state.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/experiments/alr_experiment.h"
>-#include "rtc_base/format_macros.h"
>-#include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/timeutils.h"
>-#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> AlrDetector::AlrDetector() : AlrDetector(nullptr) {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.h
>index 874354bc580d9f20ced28c1f1f80bfab6b987366..c30ba1d0932d17b51ce3ea83cfb24d843f84f483 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.h
>@@ -11,10 +11,11 @@
> #ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
> #define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "absl/types/optional.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/pacing/interval_budget.h"
>-#include "rtc_base/rate_statistics.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
>index 3a3d002c23be081e511d12d2dbd7ee027177f8df..425268514d8b3a34510a8c7fc3f95d36094570fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/congestion_controller/goog_cc/alr_detector.h"
> 
>+#include "rtc_base/checks.h"
> #include "rtc_base/experiments/alr_experiment.h"
> #include "test/field_trial.h"
> #include "test/gtest.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
>index c776827802c9ec058702b68db890c59b701417bf..0a403927429fc9a207b03fef0219476de1210d6d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
>@@ -10,11 +10,11 @@
> 
> #include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
> 
>+#include <stdio.h>
> #include <cmath>
> #include <string>
> 
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
>-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/field_trial.h"
> 
>@@ -132,6 +132,12 @@ absl::optional<uint32_t> BitrateEstimator::bitrate_bps() const {
>   return bitrate_estimate_ * 1000;
> }
> 
>+absl::optional<uint32_t> BitrateEstimator::PeekBps() const {
>+  if (current_window_ms_ > 0)
>+    return sum_ * 8000 / current_window_ms_;
>+  return absl::nullopt;
>+}
>+
> void BitrateEstimator::ExpectFastRateChange() {
>   // By setting the bitrate-estimate variance to a higher value we allow the
>   // bitrate to change fast for the next few samples.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h
>index 610fa214e7a900c9992b8c22158563d29c3978c2..d3df2b510472d0a81511a8c7819c4d6e0ef9dfba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
> #define MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
> 
>-#include <vector>
>+#include <stdint.h>
> 
> #include "absl/types/optional.h"
> 
>@@ -29,6 +29,7 @@ class BitrateEstimator {
>   virtual void Update(int64_t now_ms, int bytes);
> 
>   virtual absl::optional<uint32_t> bitrate_bps() const;
>+  absl::optional<uint32_t> PeekBps() const;
> 
>   virtual void ExpectFastRateChange();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..1e59cb5b4dfe102144e0e6b398e51ecd66a629c1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc
>@@ -0,0 +1,99 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <algorithm>
>+#include <string>
>+
>+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/format_macros.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+
>+// When CongestionWindowPushback is enabled, the pacer is oblivious to
>+// the congestion window. The relation between outstanding data and
>+// the congestion window affects encoder allocations directly.
>+// This experiment is build on top of congestion window experiment.
>+const char kCongestionPushbackExperiment[] = "WebRTC-CongestionWindowPushback";
>+const uint32_t kDefaultMinPushbackTargetBitrateBps = 30000;
>+
>+bool ReadCongestionWindowPushbackExperimentParameter(
>+    uint32_t* min_pushback_target_bitrate_bps) {
>+  RTC_DCHECK(min_pushback_target_bitrate_bps);
>+  std::string experiment_string =
>+      webrtc::field_trial::FindFullName(kCongestionPushbackExperiment);
>+  int parsed_values = sscanf(experiment_string.c_str(), "Enabled-%" PRIu32,
>+                             min_pushback_target_bitrate_bps);
>+  if (parsed_values == 1) {
>+    RTC_CHECK_GE(*min_pushback_target_bitrate_bps, 0)
>+        << "Min pushback target bitrate must be greater than or equal to 0.";
>+    return true;
>+  }
>+  return false;
>+}
>+
>+CongestionWindowPushbackController::CongestionWindowPushbackController() {
>+  if (!ReadCongestionWindowPushbackExperimentParameter(
>+          &min_pushback_target_bitrate_bps_)) {
>+    min_pushback_target_bitrate_bps_ = kDefaultMinPushbackTargetBitrateBps;
>+  }
>+}
>+
>+CongestionWindowPushbackController::CongestionWindowPushbackController(
>+    uint32_t min_pushback_target_bitrate_bps)
>+    : min_pushback_target_bitrate_bps_(min_pushback_target_bitrate_bps) {}
>+
>+void CongestionWindowPushbackController::UpdateOutstandingData(
>+    size_t outstanding_bytes) {
>+  outstanding_bytes_ = outstanding_bytes;
>+}
>+
>+void CongestionWindowPushbackController::UpdateMaxOutstandingData(
>+    size_t max_outstanding_bytes) {
>+  DataSize data_window = DataSize::bytes(max_outstanding_bytes);
>+  if (current_data_window_) {
>+    data_window = (data_window + current_data_window_.value()) / 2;
>+  }
>+  current_data_window_ = data_window;
>+}
>+
>+void CongestionWindowPushbackController::SetDataWindow(DataSize data_window) {
>+  current_data_window_ = data_window;
>+}
>+
>+uint32_t CongestionWindowPushbackController::UpdateTargetBitrate(
>+    uint32_t bitrate_bps) {
>+  if (!current_data_window_ || current_data_window_->IsZero())
>+    return bitrate_bps;
>+  double fill_ratio =
>+      outstanding_bytes_ / static_cast<double>(current_data_window_->bytes());
>+  if (fill_ratio > 1.5) {
>+    encoding_rate_ratio_ *= 0.9;
>+  } else if (fill_ratio > 1) {
>+    encoding_rate_ratio_ *= 0.95;
>+  } else if (fill_ratio < 0.1) {
>+    encoding_rate_ratio_ = 1.0;
>+  } else {
>+    encoding_rate_ratio_ *= 1.05;
>+    encoding_rate_ratio_ = std::min(encoding_rate_ratio_, 1.0);
>+  }
>+  uint32_t adjusted_target_bitrate_bps =
>+      static_cast<uint32_t>(bitrate_bps * encoding_rate_ratio_);
>+
>+  // Do not adjust below the minimum pushback bitrate but do obey if the
>+  // original estimate is below it.
>+  bitrate_bps = adjusted_target_bitrate_bps < min_pushback_target_bitrate_bps_
>+                    ? std::min(bitrate_bps, min_pushback_target_bitrate_bps_)
>+                    : adjusted_target_bitrate_bps;
>+  return bitrate_bps;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..7ef0974cce3459611d90a74b2ad6969cff7aaf38
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h
>@@ -0,0 +1,42 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
>+
>+#include "api/transport/network_types.h"
>+
>+namespace webrtc {
>+
>+// This class enables pushback from congestion window directly to video encoder.
>+// When the congestion window is filling up, the video encoder target bitrate
>+// will be reduced accordingly to accommodate the network changes. To avoid
>+// pausing video too frequently, a minimum encoder target bitrate threshold is
>+// used to prevent video pause due to a full congestion window.
>+class CongestionWindowPushbackController {
>+ public:
>+  CongestionWindowPushbackController();
>+  explicit CongestionWindowPushbackController(
>+      uint32_t min_pushback_target_bitrate_bps);
>+  void UpdateOutstandingData(size_t outstanding_bytes);
>+  void UpdateMaxOutstandingData(size_t max_outstanding_bytes);
>+  uint32_t UpdateTargetBitrate(uint32_t bitrate_bps);
>+  void SetDataWindow(DataSize data_window);
>+
>+ private:
>+  absl::optional<DataSize> current_data_window_;
>+  size_t outstanding_bytes_ = 0;
>+  uint32_t min_pushback_target_bitrate_bps_;
>+  double encoding_rate_ratio_ = 1.0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..30617eebb52cce983b20b8ed4f798b25c866c6a9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller_unittest.cc
>@@ -0,0 +1,65 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+
>+using testing::_;
>+
>+namespace webrtc {
>+namespace test {
>+
>+class CongestionWindowPushbackControllerTest : public ::testing::Test {
>+ protected:
>+  CongestionWindowPushbackController cwnd_controller_;
>+};
>+
>+TEST_F(CongestionWindowPushbackControllerTest, FullCongestionWindow) {
>+  cwnd_controller_.UpdateOutstandingData(100000);
>+  cwnd_controller_.UpdateMaxOutstandingData(50000);
>+
>+  uint32_t bitrate_bps = 80000;
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(72000u, bitrate_bps);
>+
>+  cwnd_controller_.UpdateMaxOutstandingData(50000);
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(static_cast<uint32_t>(72000 * 0.9 * 0.9), bitrate_bps);
>+}
>+
>+TEST_F(CongestionWindowPushbackControllerTest, NormalCongestionWindow) {
>+  cwnd_controller_.UpdateOutstandingData(100000);
>+  cwnd_controller_.SetDataWindow(DataSize::bytes(200000));
>+
>+  uint32_t bitrate_bps = 80000;
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(80000u, bitrate_bps);
>+
>+  cwnd_controller_.UpdateMaxOutstandingData(20000);
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(80000u, bitrate_bps);
>+}
>+
>+TEST_F(CongestionWindowPushbackControllerTest, LowBitrate) {
>+  cwnd_controller_.UpdateOutstandingData(100000);
>+  cwnd_controller_.SetDataWindow(DataSize::bytes(50000));
>+
>+  uint32_t bitrate_bps = 35000;
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(static_cast<uint32_t>(35000 * 0.9), bitrate_bps);
>+
>+  cwnd_controller_.UpdateMaxOutstandingData(20000);
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(30000u, bitrate_bps);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
>index c21966cf7f5444de87126a2dc68eabb350a34f11..4894c9863b6836113c5e13bbac39d4892ce709e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
>@@ -11,26 +11,25 @@
> #include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
> 
> #include <algorithm>
>-#include <cmath>
>+#include <cstdint>
> #include <cstdio>
> #include <string>
> 
> #include "absl/memory/memory.h"
>+#include "api/transport/network_types.h"  // For PacedPacketInfo
>+#include "logging/rtc_event_log/events/rtc_event.h"
> #include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/congestion_controller/goog_cc/trendline_estimator.h"
>-#include "modules/pacing/paced_sender.h"
>-#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/field_trial.h"
> #include "system_wrappers/include/metrics.h"
> 
>+namespace webrtc {
> namespace {
>-static const int64_t kStreamTimeOutMs = 2000;
>+constexpr TimeDelta kStreamTimeOut = TimeDelta::Seconds<2>();
> constexpr int kTimestampGroupLengthMs = 5;
> constexpr int kAbsSendTimeFraction = 18;
> constexpr int kAbsSendTimeInterArrivalUpshift = 8;
>@@ -63,24 +62,22 @@ size_t ReadTrendlineFilterWindowSize() {
>       return window_size;
>     RTC_LOG(WARNING) << "Window size must be greater than 1.";
>   }
>-  RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweTrendlineFilter "
>-                         "experiment from field trial string. Using default.";
>+  RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweWindowSizeInPackets"
>+                         " experiment from field trial string. Using default.";
>   return kDefaultTrendlineWindowSize;
> }
> }  // namespace
> 
>-namespace webrtc {
>-
> DelayBasedBwe::Result::Result()
>     : updated(false),
>       probe(false),
>-      target_bitrate_bps(0),
>+      target_bitrate(DataRate::Zero()),
>       recovered_from_overuse(false) {}
> 
>-DelayBasedBwe::Result::Result(bool probe, uint32_t target_bitrate_bps)
>+DelayBasedBwe::Result::Result(bool probe, DataRate target_bitrate)
>     : updated(true),
>       probe(probe),
>-      target_bitrate_bps(target_bitrate_bps),
>+      target_bitrate(target_bitrate),
>       recovered_from_overuse(false) {}
> 
> DelayBasedBwe::Result::~Result() {}
>@@ -89,9 +86,8 @@ DelayBasedBwe::DelayBasedBwe(RtcEventLog* event_log)
>     : event_log_(event_log),
>       inter_arrival_(),
>       delay_detector_(),
>-      last_seen_packet_ms_(-1),
>+      last_seen_packet_(Timestamp::MinusInfinity()),
>       uma_recorded_(false),
>-      probe_bitrate_estimator_(event_log),
>       trendline_window_size_(
>           webrtc::field_trial::IsEnabled(kBweWindowSizeInPacketsExperiment)
>               ? ReadTrendlineFilterWindowSize()
>@@ -99,7 +95,7 @@ DelayBasedBwe::DelayBasedBwe(RtcEventLog* event_log)
>       trendline_smoothing_coeff_(kDefaultTrendlineSmoothingCoeff),
>       trendline_threshold_gain_(kDefaultTrendlineThresholdGain),
>       consecutive_delayed_feedbacks_(0),
>-      prev_bitrate_(0),
>+      prev_bitrate_(DataRate::Zero()),
>       prev_state_(BandwidthUsage::kBwNormal) {
>   RTC_LOG(LS_INFO)
>       << "Using Trendline filter for delay change estimation with window size "
>@@ -113,8 +109,9 @@ DelayBasedBwe::~DelayBasedBwe() {}
> 
> DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
>     const std::vector<PacketFeedback>& packet_feedback_vector,
>-    absl::optional<uint32_t> acked_bitrate_bps,
>-    int64_t at_time_ms) {
>+    absl::optional<DataRate> acked_bitrate,
>+    absl::optional<DataRate> probe_bitrate,
>+    Timestamp at_time) {
>   RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
>                             packet_feedback_vector.end(),
>                             PacketFeedbackComparator()));
>@@ -141,7 +138,7 @@ DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
>     if (packet_feedback.send_time_ms < 0)
>       continue;
>     delayed_feedback = false;
>-    IncomingPacketFeedback(packet_feedback, at_time_ms);
>+    IncomingPacketFeedback(packet_feedback, at_time);
>     if (prev_detector_state == BandwidthUsage::kBwUnderusing &&
>         delay_detector_->State() == BandwidthUsage::kBwNormal) {
>       recovered_from_overuse = true;
>@@ -150,43 +147,51 @@ DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
>   }
> 
>   if (delayed_feedback) {
>-    ++consecutive_delayed_feedbacks_;
>-    if (consecutive_delayed_feedbacks_ >= kMaxConsecutiveFailedLookups) {
>-      consecutive_delayed_feedbacks_ = 0;
>-      return OnLongFeedbackDelay(packet_feedback_vector.back().arrival_time_ms);
>-    }
>+    Timestamp arrival_time = Timestamp::PlusInfinity();
>+    if (packet_feedback_vector.back().arrival_time_ms > 0)
>+      arrival_time =
>+          Timestamp::ms(packet_feedback_vector.back().arrival_time_ms);
>+    return OnDelayedFeedback(arrival_time);
>+
>   } else {
>     consecutive_delayed_feedbacks_ = 0;
>-    return MaybeUpdateEstimate(acked_bitrate_bps, recovered_from_overuse,
>-                               at_time_ms);
>+    return MaybeUpdateEstimate(acked_bitrate, probe_bitrate,
>+                               recovered_from_overuse, at_time);
>+  }
>+  return Result();
>+}
>+
>+DelayBasedBwe::Result DelayBasedBwe::OnDelayedFeedback(Timestamp receive_time) {
>+  ++consecutive_delayed_feedbacks_;
>+  if (consecutive_delayed_feedbacks_ >= kMaxConsecutiveFailedLookups) {
>+    consecutive_delayed_feedbacks_ = 0;
>+    return OnLongFeedbackDelay(receive_time);
>   }
>   return Result();
> }
> 
> DelayBasedBwe::Result DelayBasedBwe::OnLongFeedbackDelay(
>-    int64_t arrival_time_ms) {
>+    Timestamp arrival_time) {
>   // Estimate should always be valid since a start bitrate always is set in the
>   // Call constructor. An alternative would be to return an empty Result here,
>   // or to estimate the throughput based on the feedback we received.
>   RTC_DCHECK(rate_control_.ValidEstimate());
>-  rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2,
>-                            arrival_time_ms);
>+  rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, arrival_time);
>   Result result;
>   result.updated = true;
>   result.probe = false;
>-  result.target_bitrate_bps = rate_control_.LatestEstimate();
>+  result.target_bitrate = rate_control_.LatestEstimate();
>   RTC_LOG(LS_WARNING) << "Long feedback delay detected, reducing BWE to "
>-                      << result.target_bitrate_bps;
>+                      << ToString(result.target_bitrate);
>   return result;
> }
> 
> void DelayBasedBwe::IncomingPacketFeedback(
>     const PacketFeedback& packet_feedback,
>-    int64_t at_time_ms) {
>-  int64_t now_ms = at_time_ms;
>+    Timestamp at_time) {
>   // Reset if the stream has timed out.
>-  if (last_seen_packet_ms_ == -1 ||
>-      now_ms - last_seen_packet_ms_ > kStreamTimeOutMs) {
>+  if (last_seen_packet_.IsInfinite() ||
>+      at_time - last_seen_packet_ > kStreamTimeOut) {
>     inter_arrival_.reset(
>         new InterArrival((kTimestampGroupLengthMs << kInterArrivalShift) / 1000,
>                          kTimestampToMs, true));
>@@ -194,7 +199,7 @@ void DelayBasedBwe::IncomingPacketFeedback(
>                                                  trendline_smoothing_coeff_,
>                                                  trendline_threshold_gain_));
>   }
>-  last_seen_packet_ms_ = now_ms;
>+  last_seen_packet_ = at_time;
> 
>   uint32_t send_time_24bits =
>       static_cast<uint32_t>(
>@@ -211,115 +216,109 @@ void DelayBasedBwe::IncomingPacketFeedback(
>   int64_t t_delta = 0;
>   int size_delta = 0;
>   if (inter_arrival_->ComputeDeltas(timestamp, packet_feedback.arrival_time_ms,
>-                                    now_ms, packet_feedback.payload_size,
>+                                    at_time.ms(), packet_feedback.payload_size,
>                                     &ts_delta, &t_delta, &size_delta)) {
>     double ts_delta_ms = (1000.0 * ts_delta) / (1 << kInterArrivalShift);
>     delay_detector_->Update(t_delta, ts_delta_ms,
>                             packet_feedback.arrival_time_ms);
>   }
>-  if (packet_feedback.pacing_info.probe_cluster_id !=
>-      PacedPacketInfo::kNotAProbe) {
>-    probe_bitrate_estimator_.HandleProbeAndEstimateBitrate(packet_feedback);
>-  }
> }
> 
> DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
>-    absl::optional<uint32_t> acked_bitrate_bps,
>+    absl::optional<DataRate> acked_bitrate,
>+    absl::optional<DataRate> probe_bitrate,
>     bool recovered_from_overuse,
>-    int64_t at_time_ms) {
>+    Timestamp at_time) {
>   Result result;
>-  int64_t now_ms = at_time_ms;
> 
>-  absl::optional<int> probe_bitrate_bps =
>-      probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps();
>   // Currently overusing the bandwidth.
>   if (delay_detector_->State() == BandwidthUsage::kBwOverusing) {
>-    if (acked_bitrate_bps &&
>-        rate_control_.TimeToReduceFurther(now_ms, *acked_bitrate_bps)) {
>+    if (acked_bitrate &&
>+        rate_control_.TimeToReduceFurther(at_time, *acked_bitrate)) {
>       result.updated =
>-          UpdateEstimate(now_ms, acked_bitrate_bps, &result.target_bitrate_bps);
>-    } else if (!acked_bitrate_bps && rate_control_.ValidEstimate() &&
>-               rate_control_.InitialTimeToReduceFurther(now_ms)) {
>+          UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
>+    } else if (!acked_bitrate && rate_control_.ValidEstimate() &&
>+               rate_control_.InitialTimeToReduceFurther(at_time)) {
>       // Overusing before we have a measured acknowledged bitrate. Reduce send
>       // rate by 50% every 200 ms.
>       // TODO(tschumim): Improve this and/or the acknowledged bitrate estimator
>       // so that we (almost) always have a bitrate estimate.
>-      rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, now_ms);
>+      rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, at_time);
>       result.updated = true;
>       result.probe = false;
>-      result.target_bitrate_bps = rate_control_.LatestEstimate();
>+      result.target_bitrate = rate_control_.LatestEstimate();
>     }
>   } else {
>-    if (probe_bitrate_bps) {
>+    if (probe_bitrate) {
>       result.probe = true;
>       result.updated = true;
>-      result.target_bitrate_bps = *probe_bitrate_bps;
>-      rate_control_.SetEstimate(*probe_bitrate_bps, now_ms);
>+      result.target_bitrate = *probe_bitrate;
>+      rate_control_.SetEstimate(*probe_bitrate, at_time);
>     } else {
>       result.updated =
>-          UpdateEstimate(now_ms, acked_bitrate_bps, &result.target_bitrate_bps);
>+          UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
>       result.recovered_from_overuse = recovered_from_overuse;
>     }
>   }
>   BandwidthUsage detector_state = delay_detector_->State();
>-  if ((result.updated && prev_bitrate_ != result.target_bitrate_bps) ||
>+  if ((result.updated && prev_bitrate_ != result.target_bitrate) ||
>       detector_state != prev_state_) {
>-    uint32_t bitrate_bps =
>-        result.updated ? result.target_bitrate_bps : prev_bitrate_;
>+    DataRate bitrate = result.updated ? result.target_bitrate : prev_bitrate_;
> 
>-    BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", now_ms, bitrate_bps);
>+    BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", at_time.ms(), bitrate.bps());
> 
>     if (event_log_) {
>       event_log_->Log(absl::make_unique<RtcEventBweUpdateDelayBased>(
>-          bitrate_bps, detector_state));
>+          bitrate.bps(), detector_state));
>     }
> 
>-    prev_bitrate_ = bitrate_bps;
>+    prev_bitrate_ = bitrate;
>     prev_state_ = detector_state;
>   }
>   return result;
> }
> 
>-bool DelayBasedBwe::UpdateEstimate(int64_t now_ms,
>-                                   absl::optional<uint32_t> acked_bitrate_bps,
>-                                   uint32_t* target_bitrate_bps) {
>-  const RateControlInput input(delay_detector_->State(), acked_bitrate_bps);
>-  *target_bitrate_bps = rate_control_.Update(&input, now_ms);
>+bool DelayBasedBwe::UpdateEstimate(Timestamp at_time,
>+                                   absl::optional<DataRate> acked_bitrate,
>+                                   DataRate* target_rate) {
>+  const RateControlInput input(delay_detector_->State(), acked_bitrate);
>+  *target_rate = rate_control_.Update(&input, at_time);
>   return rate_control_.ValidEstimate();
> }
> 
>-void DelayBasedBwe::OnRttUpdate(int64_t avg_rtt_ms) {
>-  rate_control_.SetRtt(avg_rtt_ms);
>+void DelayBasedBwe::OnRttUpdate(TimeDelta avg_rtt) {
>+  rate_control_.SetRtt(avg_rtt);
> }
> 
> bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs,
>-                                   uint32_t* bitrate_bps) const {
>+                                   DataRate* bitrate) const {
>   // Currently accessed from both the process thread (see
>   // ModuleRtpRtcpImpl::Process()) and the configuration thread (see
>   // Call::GetStats()). Should in the future only be accessed from a single
>   // thread.
>   RTC_DCHECK(ssrcs);
>-  RTC_DCHECK(bitrate_bps);
>+  RTC_DCHECK(bitrate);
>   if (!rate_control_.ValidEstimate())
>     return false;
> 
>   *ssrcs = {kFixedSsrc};
>-  *bitrate_bps = rate_control_.LatestEstimate();
>+  *bitrate = rate_control_.LatestEstimate();
>   return true;
> }
> 
>-void DelayBasedBwe::SetStartBitrate(int start_bitrate_bps) {
>-  RTC_LOG(LS_INFO) << "BWE Setting start bitrate to: " << start_bitrate_bps;
>-  rate_control_.SetStartBitrate(start_bitrate_bps);
>+void DelayBasedBwe::SetStartBitrate(DataRate start_bitrate) {
>+  RTC_LOG(LS_INFO) << "BWE Setting start bitrate to: "
>+                   << ToString(start_bitrate);
>+  rate_control_.SetStartBitrate(start_bitrate);
> }
> 
>-void DelayBasedBwe::SetMinBitrate(int min_bitrate_bps) {
>+void DelayBasedBwe::SetMinBitrate(DataRate min_bitrate) {
>   // Called from both the configuration thread and the network thread. Shouldn't
>   // be called from the network thread in the future.
>-  rate_control_.SetMinBitrate(min_bitrate_bps);
>+  rate_control_.SetMinBitrate(min_bitrate);
> }
> 
>-int64_t DelayBasedBwe::GetExpectedBwePeriodMs() const {
>-  return rate_control_.GetExpectedBandwidthPeriodMs();
>+TimeDelta DelayBasedBwe::GetExpectedBwePeriod() const {
>+  return rate_control_.GetExpectedBandwidthPeriod();
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h
>index 42164baad4ac71c20e9768432c1adff6d44ddb47..616c5b0896bdf2fe03490b748ea0ce3419e8dac6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h
>@@ -11,16 +11,18 @@
> #ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
> #define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
>-#include <utility>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
> #include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
> #include "modules/remote_bitrate_estimator/aimd_rate_control.h"
>-#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
>+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/remote_bitrate_estimator/inter_arrival.h"
>-#include "rtc_base/checks.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"  // For PacketFeedback
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/race_checker.h"
> 
>@@ -31,11 +33,11 @@ class DelayBasedBwe {
>  public:
>   struct Result {
>     Result();
>-    Result(bool probe, uint32_t target_bitrate_bps);
>+    Result(bool probe, DataRate target_bitrate);
>     ~Result();
>     bool updated;
>     bool probe;
>-    uint32_t target_bitrate_bps;
>+    DataRate target_bitrate = DataRate::Zero();
>     bool recovered_from_overuse;
>   };
> 
>@@ -44,42 +46,43 @@ class DelayBasedBwe {
> 
>   Result IncomingPacketFeedbackVector(
>       const std::vector<PacketFeedback>& packet_feedback_vector,
>-      absl::optional<uint32_t> acked_bitrate_bps,
>-      int64_t at_time_ms);
>-  void OnRttUpdate(int64_t avg_rtt_ms);
>-  bool LatestEstimate(std::vector<uint32_t>* ssrcs,
>-                      uint32_t* bitrate_bps) const;
>-  void SetStartBitrate(int start_bitrate_bps);
>-  void SetMinBitrate(int min_bitrate_bps);
>-  int64_t GetExpectedBwePeriodMs() const;
>+      absl::optional<DataRate> acked_bitrate,
>+      absl::optional<DataRate> probe_bitrate,
>+      Timestamp at_time);
>+  Result OnDelayedFeedback(Timestamp receive_time);
>+  void OnRttUpdate(TimeDelta avg_rtt);
>+  bool LatestEstimate(std::vector<uint32_t>* ssrcs, DataRate* bitrate) const;
>+  void SetStartBitrate(DataRate start_bitrate);
>+  void SetMinBitrate(DataRate min_bitrate);
>+  TimeDelta GetExpectedBwePeriod() const;
> 
>  private:
>   friend class GoogCcStatePrinter;
>   void IncomingPacketFeedback(const PacketFeedback& packet_feedback,
>-                              int64_t at_time_ms);
>-  Result OnLongFeedbackDelay(int64_t arrival_time_ms);
>-  Result MaybeUpdateEstimate(absl::optional<uint32_t> acked_bitrate_bps,
>+                              Timestamp at_time);
>+  Result OnLongFeedbackDelay(Timestamp arrival_time);
>+  Result MaybeUpdateEstimate(absl::optional<DataRate> acked_bitrate,
>+                             absl::optional<DataRate> probe_bitrate,
>                              bool request_probe,
>-                             int64_t at_time_ms);
>+                             Timestamp at_time);
>   // Updates the current remote rate estimate and returns true if a valid
>   // estimate exists.
>-  bool UpdateEstimate(int64_t now_ms,
>-                      absl::optional<uint32_t> acked_bitrate_bps,
>-                      uint32_t* target_bitrate_bps);
>+  bool UpdateEstimate(Timestamp now,
>+                      absl::optional<DataRate> acked_bitrate,
>+                      DataRate* target_bitrate);
> 
>   rtc::RaceChecker network_race_;
>   RtcEventLog* const event_log_;
>   std::unique_ptr<InterArrival> inter_arrival_;
>   std::unique_ptr<DelayIncreaseDetectorInterface> delay_detector_;
>-  int64_t last_seen_packet_ms_;
>+  Timestamp last_seen_packet_;
>   bool uma_recorded_;
>   AimdRateControl rate_control_;
>-  ProbeBitrateEstimator probe_bitrate_estimator_;
>   size_t trendline_window_size_;
>   double trendline_smoothing_coeff_;
>   double trendline_threshold_gain_;
>   int consecutive_delayed_feedbacks_;
>-  uint32_t prev_bitrate_;
>+  DataRate prev_bitrate_;
>   BandwidthUsage prev_state_;
> 
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DelayBasedBwe);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
>index b2add3e6e94a8f502780076bc377c92677f123c6..7de11b16840f00afad8e70afd59f732e4079b230 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
>@@ -25,13 +25,13 @@ constexpr int kNumProbesCluster1 = 8;
> const PacedPacketInfo kPacingInfo0(0, kNumProbesCluster0, 2000);
> const PacedPacketInfo kPacingInfo1(1, kNumProbesCluster1, 4000);
> constexpr float kTargetUtilizationFraction = 0.95f;
>-constexpr int64_t kDummyTimestamp = 1000;
>+constexpr Timestamp kDummyTimestamp = Timestamp::Seconds<1000>();
> }  // namespace
> 
> TEST_F(DelayBasedBweTest, NoCrashEmptyFeedback) {
>   std::vector<PacketFeedback> packet_feedback_vector;
>   bitrate_estimator_->IncomingPacketFeedbackVector(
>-      packet_feedback_vector, absl::nullopt, kDummyTimestamp);
>+      packet_feedback_vector, absl::nullopt, absl::nullopt, kDummyTimestamp);
> }
> 
> TEST_F(DelayBasedBweTest, NoCrashOnlyLostFeedback) {
>@@ -43,7 +43,7 @@ TEST_F(DelayBasedBweTest, NoCrashOnlyLostFeedback) {
>                                                   PacketFeedback::kNoSendTime,
>                                                   1, 1500, PacedPacketInfo()));
>   bitrate_estimator_->IncomingPacketFeedbackVector(
>-      packet_feedback_vector, absl::nullopt, kDummyTimestamp);
>+      packet_feedback_vector, absl::nullopt, absl::nullopt, kDummyTimestamp);
> }
> 
> TEST_F(DelayBasedBweTest, ProbeDetection) {
>@@ -144,12 +144,12 @@ TEST_F(DelayBasedBweTest, ProbeDetectionSlowerArrivalHighBitrate) {
> }
> 
> TEST_F(DelayBasedBweTest, GetExpectedBwePeriodMs) {
>-  int64_t default_interval_ms = bitrate_estimator_->GetExpectedBwePeriodMs();
>-  EXPECT_GT(default_interval_ms, 0);
>+  auto default_interval = bitrate_estimator_->GetExpectedBwePeriod();
>+  EXPECT_GT(default_interval.ms(), 0);
>   CapacityDropTestHelper(1, true, 333, 0);
>-  int64_t interval_ms = bitrate_estimator_->GetExpectedBwePeriodMs();
>-  EXPECT_GT(interval_ms, 0);
>-  EXPECT_NE(interval_ms, default_interval_ms);
>+  auto interval = bitrate_estimator_->GetExpectedBwePeriod();
>+  EXPECT_GT(interval.ms(), 0);
>+  EXPECT_NE(interval.ms(), default_interval.ms());
> }
> 
> TEST_F(DelayBasedBweTest, InitialBehavior) {
>@@ -199,20 +199,20 @@ TEST_F(DelayBasedBweTest, TestLongTimeoutAndWrap) {
> }
> 
> TEST_F(DelayBasedBweTest, TestInitialOveruse) {
>-  const uint32_t kStartBitrate = 300e3;
>-  const uint32_t kInitialCapacityBps = 200e3;
>+  const DataRate kStartBitrate = DataRate::kbps(300);
>+  const DataRate kInitialCapacity = DataRate::kbps(200);
>   const uint32_t kDummySsrc = 0;
>   // High FPS to ensure that we send a lot of packets in a short time.
>   const int kFps = 90;
> 
>-  stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate));
>-  stream_generator_->set_capacity_bps(kInitialCapacityBps);
>+  stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate.bps()));
>+  stream_generator_->set_capacity_bps(kInitialCapacity.bps());
> 
>   // Needed to initialize the AimdRateControl.
>   bitrate_estimator_->SetStartBitrate(kStartBitrate);
> 
>   // Produce 30 frames (in 1/3 second) and give them to the estimator.
>-  uint32_t bitrate_bps = kStartBitrate;
>+  int64_t bitrate_bps = kStartBitrate.bps();
>   bool seen_overuse = false;
>   for (int i = 0; i < 30; ++i) {
>     bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
>@@ -222,7 +222,8 @@ TEST_F(DelayBasedBweTest, TestInitialOveruse) {
>     EXPECT_FALSE(acknowledged_bitrate_estimator_->bitrate_bps().has_value());
>     if (overuse) {
>       EXPECT_TRUE(bitrate_observer_.updated());
>-      EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate / 2, 15000);
>+      EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate.bps() / 2,
>+                  15000);
>       bitrate_bps = bitrate_observer_.latest_bitrate();
>       seen_overuse = true;
>       break;
>@@ -232,7 +233,8 @@ TEST_F(DelayBasedBweTest, TestInitialOveruse) {
>     }
>   }
>   EXPECT_TRUE(seen_overuse);
>-  EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate / 2, 15000);
>+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate.bps() / 2,
>+              15000);
> }
> 
> class DelayBasedBweTestWithBackoffTimeoutExperiment : public DelayBasedBweTest {
>@@ -243,20 +245,20 @@ class DelayBasedBweTestWithBackoffTimeoutExperiment : public DelayBasedBweTest {
> 
> // This test subsumes and improves DelayBasedBweTest.TestInitialOveruse above.
> TEST_F(DelayBasedBweTestWithBackoffTimeoutExperiment, TestInitialOveruse) {
>-  const uint32_t kStartBitrate = 300e3;
>-  const uint32_t kInitialCapacityBps = 200e3;
>+  const DataRate kStartBitrate = DataRate::kbps(300);
>+  const DataRate kInitialCapacity = DataRate::kbps(200);
>   const uint32_t kDummySsrc = 0;
>   // High FPS to ensure that we send a lot of packets in a short time.
>   const int kFps = 90;
> 
>-  stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate));
>-  stream_generator_->set_capacity_bps(kInitialCapacityBps);
>+  stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate.bps()));
>+  stream_generator_->set_capacity_bps(kInitialCapacity.bps());
> 
>   // Needed to initialize the AimdRateControl.
>   bitrate_estimator_->SetStartBitrate(kStartBitrate);
> 
>   // Produce 30 frames (in 1/3 second) and give them to the estimator.
>-  uint32_t bitrate_bps = kStartBitrate;
>+  int64_t bitrate_bps = kStartBitrate.bps();
>   bool seen_overuse = false;
>   for (int frames = 0; frames < 30 && !seen_overuse; ++frames) {
>     bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
>@@ -266,7 +268,8 @@ TEST_F(DelayBasedBweTestWithBackoffTimeoutExperiment, TestInitialOveruse) {
>     EXPECT_FALSE(acknowledged_bitrate_estimator_->bitrate_bps().has_value());
>     if (overuse) {
>       EXPECT_TRUE(bitrate_observer_.updated());
>-      EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate / 2, 15000);
>+      EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate.bps() / 2,
>+                  15000);
>       bitrate_bps = bitrate_observer_.latest_bitrate();
>       seen_overuse = true;
>     } else if (bitrate_observer_.updated()) {
>@@ -282,8 +285,8 @@ TEST_F(DelayBasedBweTestWithBackoffTimeoutExperiment, TestInitialOveruse) {
>     EXPECT_FALSE(overuse);
>     if (bitrate_observer_.updated()) {
>       bitrate_bps = bitrate_observer_.latest_bitrate();
>-      EXPECT_GE(bitrate_bps, kStartBitrate / 2 - 15000);
>-      EXPECT_LE(bitrate_bps, kInitialCapacityBps + 15000);
>+      EXPECT_GE(bitrate_bps, kStartBitrate.bps() / 2 - 15000);
>+      EXPECT_LE(bitrate_bps, kInitialCapacity.bps() + 15000);
>       bitrate_observer_.Reset();
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
>index 2a00801eb1c6cd686432479a72987c4eba351812..47288e5fbc64fcad3a924a2276994cf604005032 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
>@@ -154,6 +154,7 @@ DelayBasedBweTest::DelayBasedBweTest()
>       clock_(100000000),
>       acknowledged_bitrate_estimator_(
>           absl::make_unique<AcknowledgedBitrateEstimator>()),
>+      probe_bitrate_estimator_(new ProbeBitrateEstimator(nullptr)),
>       bitrate_estimator_(new DelayBasedBwe(nullptr)),
>       stream_generator_(new test::StreamGenerator(1e6,  // Capacity.
>                                                   clock_.TimeInMicroseconds())),
>@@ -166,6 +167,7 @@ DelayBasedBweTest::DelayBasedBweTest(const std::string& field_trial_string)
>       clock_(100000000),
>       acknowledged_bitrate_estimator_(
>           absl::make_unique<AcknowledgedBitrateEstimator>()),
>+      probe_bitrate_estimator_(new ProbeBitrateEstimator(nullptr)),
>       bitrate_estimator_(new DelayBasedBwe(nullptr)),
>       stream_generator_(new test::StreamGenerator(1e6,  // Capacity.
>                                                   clock_.TimeInMicroseconds())),
>@@ -198,15 +200,20 @@ void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
>                         sequence_number, payload_size, pacing_info);
>   std::vector<PacketFeedback> packets;
>   packets.push_back(packet);
>+  if (packet.send_time_ms != PacketFeedback::kNoSendTime &&
>+      packet.pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe)
>+    probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
>+
>   acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
>   DelayBasedBwe::Result result =
>       bitrate_estimator_->IncomingPacketFeedbackVector(
>-          packets, acknowledged_bitrate_estimator_->bitrate_bps(),
>-          clock_.TimeInMilliseconds());
>+          packets, acknowledged_bitrate_estimator_->bitrate(),
>+          probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
>+          Timestamp::ms(clock_.TimeInMilliseconds()));
>   const uint32_t kDummySsrc = 0;
>   if (result.updated) {
>     bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
>-                                              result.target_bitrate_bps);
>+                                              result.target_bitrate.bps());
>   }
> }
> 
>@@ -232,18 +239,23 @@ bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
>   for (auto& packet : packets) {
>     RTC_CHECK_GE(packet.arrival_time_ms + arrival_time_offset_ms_, 0);
>     packet.arrival_time_ms += arrival_time_offset_ms_;
>+
>+    if (packet.send_time_ms != PacketFeedback::kNoSendTime &&
>+        packet.pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe)
>+      probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
>   }
> 
>   acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
>   DelayBasedBwe::Result result =
>       bitrate_estimator_->IncomingPacketFeedbackVector(
>-          packets, acknowledged_bitrate_estimator_->bitrate_bps(),
>-          clock_.TimeInMilliseconds());
>+          packets, acknowledged_bitrate_estimator_->bitrate(),
>+          probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
>+          Timestamp::ms(clock_.TimeInMilliseconds()));
>   const uint32_t kDummySsrc = 0;
>   if (result.updated) {
>     bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
>-                                              result.target_bitrate_bps);
>-    if (!first_update_ && result.target_bitrate_bps < bitrate_bps)
>+                                              result.target_bitrate.bps());
>+    if (!first_update_ && result.target_bitrate.bps() < bitrate_bps)
>       overuse = true;
>     first_update_ = false;
>   }
>@@ -289,14 +301,14 @@ void DelayBasedBweTest::InitialBehaviorTestHelper(
>   const int kFramerate = 50;  // 50 fps to avoid rounding errors.
>   const int kFrameIntervalMs = 1000 / kFramerate;
>   const PacedPacketInfo kPacingInfo(0, 5, 5000);
>-  uint32_t bitrate_bps = 0;
>+  DataRate bitrate = DataRate::Zero();
>   int64_t send_time_ms = 0;
>   uint16_t sequence_number = 0;
>   std::vector<uint32_t> ssrcs;
>-  EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
>+  EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
>   EXPECT_EQ(0u, ssrcs.size());
>   clock_.AdvanceTimeMilliseconds(1000);
>-  EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
>+  EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
>   EXPECT_FALSE(bitrate_observer_.updated());
>   bitrate_observer_.Reset();
>   clock_.AdvanceTimeMilliseconds(1000);
>@@ -308,7 +320,7 @@ void DelayBasedBweTest::InitialBehaviorTestHelper(
>         i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();
> 
>     if (i == kNumInitialPackets) {
>-      EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
>+      EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
>       EXPECT_EQ(0u, ssrcs.size());
>       EXPECT_FALSE(bitrate_observer_.updated());
>       bitrate_observer_.Reset();
>@@ -318,13 +330,14 @@ void DelayBasedBweTest::InitialBehaviorTestHelper(
>     clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
>     send_time_ms += kFrameIntervalMs;
>   }
>-  EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
>+  EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
>   ASSERT_EQ(1u, ssrcs.size());
>   EXPECT_EQ(kDefaultSsrc, ssrcs.front());
>-  EXPECT_NEAR(expected_converge_bitrate, bitrate_bps, kAcceptedBitrateErrorBps);
>+  EXPECT_NEAR(expected_converge_bitrate, bitrate.bps(),
>+              kAcceptedBitrateErrorBps);
>   EXPECT_TRUE(bitrate_observer_.updated());
>   bitrate_observer_.Reset();
>-  EXPECT_EQ(bitrate_observer_.latest_bitrate(), bitrate_bps);
>+  EXPECT_EQ(bitrate_observer_.latest_bitrate(), bitrate.bps());
> }
> 
> void DelayBasedBweTest::RateIncreaseReorderingTestHelper(
>@@ -509,7 +522,7 @@ void DelayBasedBweTest::TestWrappingHelper(int silence_time_s) {
>     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
>     send_time_ms += kFrameIntervalMs;
>   }
>-  uint32_t bitrate_before = 0;
>+  DataRate bitrate_before = DataRate::Zero();
>   std::vector<uint32_t> ssrcs;
>   bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_before);
> 
>@@ -522,7 +535,7 @@ void DelayBasedBweTest::TestWrappingHelper(int silence_time_s) {
>     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
>     send_time_ms += kFrameIntervalMs;
>   }
>-  uint32_t bitrate_after = 0;
>+  DataRate bitrate_after = DataRate::Zero();
>   bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
>   EXPECT_LT(bitrate_after, bitrate_before);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
>index 7ba18c367527bb726354e4c7412443e4ed4066b1..444e0cd67df4fb4685e339fc5328a51f28585190 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
>@@ -172,6 +172,7 @@ class DelayBasedBweTest : public ::testing::Test {
>   SimulatedClock clock_;  // Time at the receiver.
>   test::TestBitrateObserver bitrate_observer_;
>   std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator_;
>+  const std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_;
>   std::unique_ptr<DelayBasedBwe> bitrate_estimator_;
>   std::unique_ptr<test::StreamGenerator> stream_generator_;
>   int64_t arrival_time_offset_ms_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_factory.cc
>deleted file mode 100644
>index 2e0ccc17ca4b3a3b8186bfe63a16e96ad519ad16..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_factory.cc
>+++ /dev/null
>@@ -1,43 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/congestion_controller/goog_cc/include/goog_cc_factory.h"
>-
>-#include "absl/memory/memory.h"
>-#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
>-namespace webrtc {
>-GoogCcNetworkControllerFactory::GoogCcNetworkControllerFactory(
>-    RtcEventLog* event_log)
>-    : event_log_(event_log) {}
>-
>-std::unique_ptr<NetworkControllerInterface>
>-GoogCcNetworkControllerFactory::Create(NetworkControllerConfig config) {
>-  return absl::make_unique<GoogCcNetworkController>(event_log_, config, false);
>-}
>-
>-TimeDelta GoogCcNetworkControllerFactory::GetProcessInterval() const {
>-  const int64_t kUpdateIntervalMs = 25;
>-  return TimeDelta::ms(kUpdateIntervalMs);
>-}
>-
>-GoogCcFeedbackNetworkControllerFactory::GoogCcFeedbackNetworkControllerFactory(
>-    RtcEventLog* event_log)
>-    : event_log_(event_log) {}
>-
>-std::unique_ptr<NetworkControllerInterface>
>-GoogCcFeedbackNetworkControllerFactory::Create(NetworkControllerConfig config) {
>-  return absl::make_unique<GoogCcNetworkController>(event_log_, config, true);
>-}
>-
>-TimeDelta GoogCcFeedbackNetworkControllerFactory::GetProcessInterval() const {
>-  const int64_t kUpdateIntervalMs = 25;
>-  return TimeDelta::ms(kUpdateIntervalMs);
>-}
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
>index 4ab9694f37bc037e4c95e26ac439b5f2c15a1f6d..4376953b7cc1cf4935c22ab2a87de90efc0e9e36 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
>@@ -22,7 +22,6 @@
> #include "absl/memory/memory.h"
> #include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
> #include "modules/congestion_controller/goog_cc/alr_detector.h"
>-#include "modules/congestion_controller/goog_cc/include/goog_cc_factory.h"
> #include "modules/congestion_controller/goog_cc/probe_controller.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
>@@ -36,6 +35,11 @@ namespace webrtc {
> namespace {
> 
> const char kCwndExperiment[] = "WebRTC-CwndExperiment";
>+// When CongestionWindowPushback is enabled, the pacer is oblivious to
>+// the congestion window. The relation between outstanding data and
>+// the congestion window affects encoder allocations directly.
>+const char kCongestionPushbackExperiment[] = "WebRTC-CongestionWindowPushback";
>+
> const int64_t kDefaultAcceptedQueueMs = 250;
> 
> // From RTCPSender video report interval.
>@@ -48,6 +52,18 @@ constexpr TimeDelta kLossUpdateInterval = TimeDelta::Millis<1000>();
> // overshoots from the encoder.
> const float kDefaultPaceMultiplier = 2.5f;
> 
>+bool IsCongestionWindowPushbackExperimentEnabled() {
>+  return webrtc::field_trial::IsEnabled(kCongestionPushbackExperiment) &&
>+         webrtc::field_trial::IsEnabled(kCwndExperiment);
>+}
>+
>+std::unique_ptr<CongestionWindowPushbackController>
>+MaybeInitalizeCongestionWindowPushbackController() {
>+  return IsCongestionWindowPushbackExperimentEnabled()
>+             ? absl::make_unique<CongestionWindowPushbackController>()
>+             : nullptr;
>+}
>+
> bool CwndExperimentEnabled() {
>   std::string experiment_string =
>       webrtc::field_trial::FindFullName(kCwndExperiment);
>@@ -90,13 +106,10 @@ std::vector<PacketFeedback> ReceivedPacketsFeedbackAsRtp(
>     if (fb.receive_time.IsFinite()) {
>       PacketFeedback pf(fb.receive_time.ms(), 0);
>       pf.creation_time_ms = report.feedback_time.ms();
>-      if (fb.sent_packet.has_value()) {
>-        pf.payload_size = fb.sent_packet->size.bytes();
>-        pf.pacing_info = fb.sent_packet->pacing_info;
>-        pf.send_time_ms = fb.sent_packet->send_time.ms();
>-      } else {
>-        pf.send_time_ms = PacketFeedback::kNoSendTime;
>-      }
>+      pf.payload_size = fb.sent_packet.size.bytes();
>+      pf.pacing_info = fb.sent_packet.pacing_info;
>+      pf.send_time_ms = fb.sent_packet.send_time.ms();
>+      pf.unacknowledged_data = fb.sent_packet.prior_unacked_data.bytes();
>       packet_feedback_vector.push_back(pf);
>     }
>   }
>@@ -114,21 +127,27 @@ int64_t GetBpsOrDefault(const absl::optional<DataRate>& rate,
> 
> }  // namespace
> 
>-
> GoogCcNetworkController::GoogCcNetworkController(RtcEventLog* event_log,
>                                                  NetworkControllerConfig config,
>                                                  bool feedback_only)
>     : event_log_(event_log),
>       packet_feedback_only_(feedback_only),
>+      safe_reset_on_route_change_("Enabled"),
>+      safe_reset_acknowledged_rate_("ack"),
>+      use_stable_bandwidth_estimate_(
>+          field_trial::IsEnabled("WebRTC-Bwe-StableBandwidthEstimate")),
>       probe_controller_(new ProbeController()),
>+      congestion_window_pushback_controller_(
>+          MaybeInitalizeCongestionWindowPushbackController()),
>       bandwidth_estimation_(
>           absl::make_unique<SendSideBandwidthEstimation>(event_log_)),
>       alr_detector_(absl::make_unique<AlrDetector>()),
>+      probe_bitrate_estimator_(new ProbeBitrateEstimator(event_log)),
>       delay_based_bwe_(new DelayBasedBwe(event_log_)),
>       acknowledged_bitrate_estimator_(
>           absl::make_unique<AcknowledgedBitrateEstimator>()),
>       initial_config_(config),
>-      last_bandwidth_(*config.constraints.starting_rate),
>+      last_target_rate_(*config.constraints.starting_rate),
>       pacing_factor_(config.stream_based_config.pacing_factor.value_or(
>           kDefaultPaceMultiplier)),
>       min_pacing_rate_(config.stream_based_config.min_pacing_rate.value_or(
>@@ -138,7 +157,12 @@ GoogCcNetworkController::GoogCcNetworkController(RtcEventLog* event_log,
>       max_total_allocated_bitrate_(DataRate::Zero()),
>       in_cwnd_experiment_(CwndExperimentEnabled()),
>       accepted_queue_ms_(kDefaultAcceptedQueueMs) {
>-  delay_based_bwe_->SetMinBitrate(congestion_controller::GetMinBitrateBps());
>+  RTC_DCHECK(config.constraints.at_time.IsFinite());
>+  ParseFieldTrial(
>+      {&safe_reset_on_route_change_, &safe_reset_acknowledged_rate_},
>+      field_trial::FindFullName("WebRTC-Bwe-SafeResetOnRouteChange"));
>+
>+  delay_based_bwe_->SetMinBitrate(congestion_controller::GetMinBitrate());
>   if (in_cwnd_experiment_ &&
>       !ReadCwndExperimentParameter(&accepted_queue_ms_)) {
>     RTC_LOG(LS_WARNING) << "Failed to parse parameters for CwndExperiment "
>@@ -165,16 +189,39 @@ NetworkControlUpdate GoogCcNetworkController::OnNetworkRouteChange(
> 
>   ClampBitrates(&start_bitrate_bps, &min_bitrate_bps, &max_bitrate_bps);
> 
>-  bandwidth_estimation_ =
>-      absl::make_unique<SendSideBandwidthEstimation>(event_log_);
>+  if (safe_reset_on_route_change_) {
>+    absl::optional<uint32_t> estimated_bitrate_bps;
>+    if (safe_reset_acknowledged_rate_) {
>+      estimated_bitrate_bps = acknowledged_bitrate_estimator_->bitrate_bps();
>+      if (!estimated_bitrate_bps)
>+        estimated_bitrate_bps = acknowledged_bitrate_estimator_->PeekBps();
>+    } else {
>+      int32_t target_bitrate_bps;
>+      uint8_t fraction_loss;
>+      int64_t rtt_ms;
>+      bandwidth_estimation_->CurrentEstimate(&target_bitrate_bps,
>+                                             &fraction_loss, &rtt_ms);
>+      estimated_bitrate_bps = target_bitrate_bps;
>+    }
>+    if (estimated_bitrate_bps && (!msg.constraints.starting_rate ||
>+                                  estimated_bitrate_bps < start_bitrate_bps)) {
>+      start_bitrate_bps = *estimated_bitrate_bps;
>+      msg.constraints.starting_rate = DataRate::bps(start_bitrate_bps);
>+    }
>+  }
>+
>+  acknowledged_bitrate_estimator_.reset(new AcknowledgedBitrateEstimator());
>+  probe_bitrate_estimator_.reset(new ProbeBitrateEstimator(event_log_));
>+  delay_based_bwe_.reset(new DelayBasedBwe(event_log_));
>+  if (msg.constraints.starting_rate)
>+    delay_based_bwe_->SetStartBitrate(*msg.constraints.starting_rate);
>+  // TODO(srte): Use original values instead of converted.
>+  delay_based_bwe_->SetMinBitrate(DataRate::bps(min_bitrate_bps));
>+  bandwidth_estimation_->OnRouteChange();
>   bandwidth_estimation_->SetBitrates(
>       msg.constraints.starting_rate, DataRate::bps(min_bitrate_bps),
>       msg.constraints.max_data_rate.value_or(DataRate::Infinity()),
>       msg.at_time);
>-  delay_based_bwe_.reset(new DelayBasedBwe(event_log_));
>-  acknowledged_bitrate_estimator_.reset(new AcknowledgedBitrateEstimator());
>-  delay_based_bwe_->SetStartBitrate(start_bitrate_bps);
>-  delay_based_bwe_->SetMinBitrate(min_bitrate_bps);
> 
>   probe_controller_->Reset(msg.at_time.ms());
>   NetworkControlUpdate update;
>@@ -239,7 +286,7 @@ NetworkControlUpdate GoogCcNetworkController::OnRoundTripTimeUpdate(
>   if (packet_feedback_only_)
>     return NetworkControlUpdate();
>   if (msg.smoothed) {
>-    delay_based_bwe_->OnRttUpdate(msg.round_trip_time.ms());
>+    delay_based_bwe_->OnRttUpdate(msg.round_trip_time);
>   } else {
>     bandwidth_estimation_->UpdateRtt(msg.round_trip_time, msg.receive_time);
>   }
>@@ -250,7 +297,22 @@ NetworkControlUpdate GoogCcNetworkController::OnSentPacket(
>     SentPacket sent_packet) {
>   alr_detector_->OnBytesSent(sent_packet.size.bytes(),
>                              sent_packet.send_time.ms());
>-  return NetworkControlUpdate();
>+  if (!first_packet_sent_) {
>+    first_packet_sent_ = true;
>+    // Initialize feedback time to send time to allow estimation of RTT until
>+    // first feedback is received.
>+    bandwidth_estimation_->UpdatePropagationRtt(sent_packet.send_time,
>+                                                TimeDelta::Zero());
>+  }
>+  if (congestion_window_pushback_controller_) {
>+    congestion_window_pushback_controller_->UpdateOutstandingData(
>+        sent_packet.data_in_flight.bytes());
>+    NetworkControlUpdate update;
>+    MaybeTriggerOnNetworkChanged(&update, sent_packet.send_time);
>+    return update;
>+  } else {
>+    return NetworkControlUpdate();
>+  }
> }
> 
> NetworkControlUpdate GoogCcNetworkController::OnStreamsConfig(
>@@ -312,9 +374,9 @@ GoogCcNetworkController::UpdateBitrateConstraints(
>       starting_rate, DataRate::bps(min_bitrate_bps),
>       constraints.max_data_rate.value_or(DataRate::Infinity()),
>       constraints.at_time);
>-  if (start_bitrate_bps > 0)
>-    delay_based_bwe_->SetStartBitrate(start_bitrate_bps);
>-  delay_based_bwe_->SetMinBitrate(min_bitrate_bps);
>+  if (starting_rate)
>+    delay_based_bwe_->SetStartBitrate(*starting_rate);
>+  delay_based_bwe_->SetMinBitrate(DataRate::bps(min_bitrate_bps));
>   return probes;
> }
> 
>@@ -331,38 +393,61 @@ NetworkControlUpdate GoogCcNetworkController::OnTransportLossReport(
> 
> NetworkControlUpdate GoogCcNetworkController::OnTransportPacketsFeedback(
>     TransportPacketsFeedback report) {
>-  TimeDelta feedback_max_rtt = TimeDelta::MinusInfinity();
>+  if (report.packet_feedbacks.empty()) {
>+    DelayBasedBwe::Result result = delay_based_bwe_->OnDelayedFeedback(
>+        report.sendless_arrival_times.back());
>+    NetworkControlUpdate update;
>+    if (result.updated) {
>+      bandwidth_estimation_->UpdateDelayBasedEstimate(report.feedback_time,
>+                                                      result.target_bitrate);
>+      MaybeTriggerOnNetworkChanged(&update, report.feedback_time);
>+    }
>+    return update;
>+  }
>+
>+  if (congestion_window_pushback_controller_) {
>+    congestion_window_pushback_controller_->UpdateOutstandingData(
>+        report.data_in_flight.bytes());
>+  }
>+  TimeDelta max_feedback_rtt = TimeDelta::MinusInfinity();
>+  TimeDelta min_propagation_rtt = TimeDelta::PlusInfinity();
>   Timestamp max_recv_time = Timestamp::MinusInfinity();
>-  for (const auto& packet_feedback : report.ReceivedWithSendInfo()) {
>-    TimeDelta rtt =
>-        report.feedback_time - packet_feedback.sent_packet->send_time;
>-    // max() is used to account for feedback being delayed by the
>-    // receiver.
>-    feedback_max_rtt = std::max(feedback_max_rtt, rtt);
>-    max_recv_time = std::max(max_recv_time, packet_feedback.receive_time);
>+
>+  std::vector<PacketResult> feedbacks = report.ReceivedWithSendInfo();
>+  for (const auto& feedback : feedbacks)
>+    max_recv_time = std::max(max_recv_time, feedback.receive_time);
>+
>+  for (const auto& feedback : feedbacks) {
>+    TimeDelta feedback_rtt =
>+        report.feedback_time - feedback.sent_packet.send_time;
>+    TimeDelta min_pending_time = feedback.receive_time - max_recv_time;
>+    TimeDelta propagation_rtt = feedback_rtt - min_pending_time;
>+    max_feedback_rtt = std::max(max_feedback_rtt, feedback_rtt);
>+    min_propagation_rtt = std::min(min_propagation_rtt, propagation_rtt);
>   }
>-  absl::optional<int64_t> min_feedback_max_rtt_ms;
>-  if (feedback_max_rtt.IsFinite()) {
>-    feedback_max_rtts_.push_back(feedback_max_rtt.ms());
>+
>+  if (max_feedback_rtt.IsFinite()) {
>+    feedback_max_rtts_.push_back(max_feedback_rtt.ms());
>     const size_t kMaxFeedbackRttWindow = 32;
>     if (feedback_max_rtts_.size() > kMaxFeedbackRttWindow)
>       feedback_max_rtts_.pop_front();
>-    min_feedback_max_rtt_ms.emplace(*std::min_element(
>-        feedback_max_rtts_.begin(), feedback_max_rtts_.end()));
>+    // TODO(srte): Use time since last unacknowledged packet.
>+    bandwidth_estimation_->UpdatePropagationRtt(report.feedback_time,
>+                                                min_propagation_rtt);
>   }
>   if (packet_feedback_only_) {
>     if (!feedback_max_rtts_.empty()) {
>       int64_t sum_rtt_ms = std::accumulate(feedback_max_rtts_.begin(),
>                                            feedback_max_rtts_.end(), 0);
>       int64_t mean_rtt_ms = sum_rtt_ms / feedback_max_rtts_.size();
>-      delay_based_bwe_->OnRttUpdate(mean_rtt_ms);
>+      delay_based_bwe_->OnRttUpdate(TimeDelta::ms(mean_rtt_ms));
>     }
> 
>     TimeDelta feedback_min_rtt = TimeDelta::PlusInfinity();
>-    for (const auto& packet_feedback : report.ReceivedWithSendInfo()) {
>+    for (const auto& packet_feedback : feedbacks) {
>       TimeDelta pending_time = packet_feedback.receive_time - max_recv_time;
>       TimeDelta rtt = report.feedback_time -
>-                      packet_feedback.sent_packet->send_time - pending_time;
>+                      packet_feedback.sent_packet.send_time - pending_time;
>       // Value used for predicting NACK round trip time in FEC controller.
>       feedback_min_rtt = std::min(rtt, feedback_min_rtt);
>     }
>@@ -400,22 +485,33 @@ NetworkControlUpdate GoogCcNetworkController::OnTransportPacketsFeedback(
>   previously_in_alr = alr_start_time.has_value();
>   acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
>       received_feedback_vector);
>-  auto acknowledged_bitrate = acknowledged_bitrate_estimator_->bitrate_bps();
>+  auto acknowledged_bitrate = acknowledged_bitrate_estimator_->bitrate();
>+  bandwidth_estimation_->SetAcknowledgedRate(acknowledged_bitrate,
>+                                             report.feedback_time);
>+  bandwidth_estimation_->IncomingPacketFeedbackVector(report);
>+  for (const auto& feedback : received_feedback_vector) {
>+    if (feedback.pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe) {
>+      probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(feedback);
>+    }
>+  }
>+  absl::optional<DataRate> probe_bitrate =
>+      probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate();
>+
>   DelayBasedBwe::Result result;
>   result = delay_based_bwe_->IncomingPacketFeedbackVector(
>-      received_feedback_vector, acknowledged_bitrate,
>-      report.feedback_time.ms());
>+      received_feedback_vector, acknowledged_bitrate, probe_bitrate,
>+      report.feedback_time);
> 
>   NetworkControlUpdate update;
>   if (result.updated) {
>     if (result.probe) {
>-      bandwidth_estimation_->SetSendBitrate(
>-          DataRate::bps(result.target_bitrate_bps), report.feedback_time);
>+      bandwidth_estimation_->SetSendBitrate(result.target_bitrate,
>+                                            report.feedback_time);
>     }
>     // Since SetSendBitrate now resets the delay-based estimate, we have to call
>     // UpdateDelayBasedEstimate after SetSendBitrate.
>-    bandwidth_estimation_->UpdateDelayBasedEstimate(
>-        report.feedback_time, DataRate::bps(result.target_bitrate_bps));
>+    bandwidth_estimation_->UpdateDelayBasedEstimate(report.feedback_time,
>+                                                    result.target_bitrate);
>     // Update the estimate in the ProbeController, in case we want to probe.
>     MaybeTriggerOnNetworkChanged(&update, report.feedback_time);
>   }
>@@ -428,11 +524,14 @@ NetworkControlUpdate GoogCcNetworkController::OnTransportPacketsFeedback(
> 
>   // No valid RTT could be because send-side BWE isn't used, in which case
>   // we don't try to limit the outstanding packets.
>-  if (in_cwnd_experiment_ && min_feedback_max_rtt_ms) {
>+  if (in_cwnd_experiment_ && max_feedback_rtt.IsFinite()) {
>+    int64_t min_feedback_max_rtt_ms =
>+        *std::min_element(feedback_max_rtts_.begin(), feedback_max_rtts_.end());
>+
>     const DataSize kMinCwnd = DataSize::bytes(2 * 1500);
>     TimeDelta time_window =
>-        TimeDelta::ms(*min_feedback_max_rtt_ms + accepted_queue_ms_);
>-    DataSize data_window = last_bandwidth_ * time_window;
>+        TimeDelta::ms(min_feedback_max_rtt_ms + accepted_queue_ms_);
>+    DataSize data_window = last_target_rate_ * time_window;
>     if (current_data_window_) {
>       data_window =
>           std::max(kMinCwnd, (data_window + current_data_window_.value()) / 2);
>@@ -440,10 +539,13 @@ NetworkControlUpdate GoogCcNetworkController::OnTransportPacketsFeedback(
>       data_window = std::max(kMinCwnd, data_window);
>     }
>     current_data_window_ = data_window;
>-    RTC_LOG(LS_INFO) << "Feedback rtt: " << *min_feedback_max_rtt_ms
>-                     << " Bitrate: " << last_bandwidth_.bps();
>   }
>-  update.congestion_window = current_data_window_;
>+  if (congestion_window_pushback_controller_ && current_data_window_) {
>+    congestion_window_pushback_controller_->SetDataWindow(
>+        *current_data_window_);
>+  } else {
>+    update.congestion_window = current_data_window_;
>+  }
> 
>   return update;
> }
>@@ -460,7 +562,7 @@ NetworkControlUpdate GoogCcNetworkController::GetNetworkState(
>       last_estimated_fraction_loss_ / 255.0;
>   update.target_rate->network_estimate.round_trip_time = rtt;
>   update.target_rate->network_estimate.bwe_period =
>-      TimeDelta::ms(delay_based_bwe_->GetExpectedBwePeriodMs());
>+      delay_based_bwe_->GetExpectedBwePeriod();
>   update.target_rate->at_time = at_time;
>   update.target_rate->target_rate = bandwidth;
>   update.pacer_config = GetPacingRates(at_time);
>@@ -496,27 +598,38 @@ void GoogCcNetworkController::MaybeTriggerOnNetworkChanged(
> 
>     alr_detector_->SetEstimatedBitrate(estimated_bitrate_bps);
> 
>-    DataRate bandwidth = DataRate::bps(estimated_bitrate_bps);
>-    last_bandwidth_ = bandwidth;
>+    last_target_rate_ = DataRate::bps(estimated_bitrate_bps);
>+    DataRate bandwidth = use_stable_bandwidth_estimate_
>+                             ? bandwidth_estimation_->GetEstimatedLinkCapacity()
>+                             : last_target_rate_;
> 
>-    TimeDelta bwe_period =
>-        TimeDelta::ms(delay_based_bwe_->GetExpectedBwePeriodMs());
>+    TimeDelta bwe_period = delay_based_bwe_->GetExpectedBwePeriod();
> 
>-    TargetTransferRate target_rate;
>-    target_rate.at_time = at_time;
>     // Set the target rate to the full estimated bandwidth since the estimation
>     // for legacy reasons includes target rate constraints.
>-    target_rate.target_rate = bandwidth;
>+    DataRate target_rate = last_target_rate_;
>+    if (congestion_window_pushback_controller_) {
>+      int64_t pushback_rate =
>+          congestion_window_pushback_controller_->UpdateTargetBitrate(
>+              target_rate.bps());
>+      pushback_rate = std::max<int64_t>(bandwidth_estimation_->GetMinBitrate(),
>+                                        pushback_rate);
>+      target_rate = DataRate::bps(pushback_rate);
>+    }
>+
>+    TargetTransferRate target_rate_msg;
>+    target_rate_msg.at_time = at_time;
>+    target_rate_msg.target_rate = target_rate;
>+    target_rate_msg.network_estimate.at_time = at_time;
>+    target_rate_msg.network_estimate.round_trip_time = TimeDelta::ms(rtt_ms);
>+    target_rate_msg.network_estimate.bandwidth = bandwidth;
>+    target_rate_msg.network_estimate.loss_rate_ratio = fraction_loss / 255.0f;
>+    target_rate_msg.network_estimate.bwe_period = bwe_period;
> 
>-    target_rate.network_estimate.at_time = at_time;
>-    target_rate.network_estimate.round_trip_time = TimeDelta::ms(rtt_ms);
>-    target_rate.network_estimate.bandwidth = bandwidth;
>-    target_rate.network_estimate.loss_rate_ratio = fraction_loss / 255.0f;
>-    target_rate.network_estimate.bwe_period = bwe_period;
>-    update->target_rate = target_rate;
>+    update->target_rate = target_rate_msg;
> 
>-    auto probes =
>-        probe_controller_->SetEstimatedBitrate(bandwidth.bps(), at_time.ms());
>+    auto probes = probe_controller_->SetEstimatedBitrate(
>+        last_target_rate_.bps(), at_time.ms());
>     update->probe_cluster_configs.insert(update->probe_cluster_configs.end(),
>                                          probes.begin(), probes.end());
>     update->pacer_config = GetPacingRates(at_time);
>@@ -525,8 +638,8 @@ void GoogCcNetworkController::MaybeTriggerOnNetworkChanged(
> 
> PacerConfig GoogCcNetworkController::GetPacingRates(Timestamp at_time) const {
>   DataRate pacing_rate =
>-      std::max(min_pacing_rate_, last_bandwidth_) * pacing_factor_;
>-  DataRate padding_rate = std::min(max_padding_rate_, last_bandwidth_);
>+      std::max(min_pacing_rate_, last_target_rate_) * pacing_factor_;
>+  DataRate padding_rate = std::min(max_padding_rate_, last_target_rate_);
>   PacerConfig msg;
>   msg.at_time = at_time;
>   msg.time_window = TimeDelta::seconds(1);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h
>index f7efda396277e4e633c3da1f80509ecac9be9f47..a62f59bb44bc0e0e5c3b8a12ab244e05098b49a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h
>@@ -22,6 +22,7 @@
> #include "modules/bitrate_controller/send_side_bandwidth_estimation.h"
> #include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
> #include "modules/congestion_controller/goog_cc/alr_detector.h"
>+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
> #include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
> #include "modules/congestion_controller/goog_cc/probe_controller.h"
> #include "rtc_base/constructormagic.h"
>@@ -63,24 +64,31 @@ class GoogCcNetworkController : public NetworkControllerInterface {
> 
>   RtcEventLog* const event_log_;
>   const bool packet_feedback_only_;
>+  FieldTrialFlag safe_reset_on_route_change_;
>+  FieldTrialFlag safe_reset_acknowledged_rate_;
>+  const bool use_stable_bandwidth_estimate_;
> 
>   const std::unique_ptr<ProbeController> probe_controller_;
>+  const std::unique_ptr<CongestionWindowPushbackController>
>+      congestion_window_pushback_controller_;
> 
>   std::unique_ptr<SendSideBandwidthEstimation> bandwidth_estimation_;
>   std::unique_ptr<AlrDetector> alr_detector_;
>+  std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_;
>   std::unique_ptr<DelayBasedBwe> delay_based_bwe_;
>   std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator_;
> 
>   absl::optional<NetworkControllerConfig> initial_config_;
> 
>+  bool first_packet_sent_ = false;
>+
>   Timestamp next_loss_update_ = Timestamp::MinusInfinity();
>   int lost_packets_since_last_loss_update_ = 0;
>   int expected_packets_since_last_loss_update_ = 0;
> 
>   std::deque<int64_t> feedback_max_rtts_;
> 
>-  DataRate last_bandwidth_;
>-  absl::optional<TargetTransferRate> last_target_rate_;
>+  DataRate last_target_rate_;
> 
>   int32_t last_estimated_bitrate_bps_ = 0;
>   uint8_t last_estimated_fraction_loss_ = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_slowtest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_slowtest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..cfa0d1c096dda3f6a35a6f8c4aa60b3ed5bcece7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_slowtest.cc
>@@ -0,0 +1,129 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/transport/goog_cc_factory.h"
>+#include "test/field_trial.h"
>+#include "test/gtest.h"
>+#include "test/scenario/scenario.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+TEST(GoogCcNetworkControllerTest, MaintainsLowRateInSafeResetTrial) {
>+  const DataRate kLinkCapacity = DataRate::kbps(200);
>+  const DataRate kStartRate = DataRate::kbps(300);
>+
>+  ScopedFieldTrials trial("WebRTC-Bwe-SafeResetOnRouteChange/Enabled/");
>+  Scenario s("googcc_unit/safe_reset_low", true);
>+  auto* send_net = s.CreateSimulationNode([&](NetworkNodeConfig* c) {
>+    c->simulation.bandwidth = kLinkCapacity;
>+    c->simulation.delay = TimeDelta::ms(10);
>+  });
>+  // TODO(srte): replace with SimulatedTimeClient when it supports probing.
>+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
>+    c->transport.cc = TransportControllerConfig::CongestionController::kGoogCc;
>+    c->transport.rates.start_rate = kStartRate;
>+  });
>+  auto* route = s.CreateRoutes(client, {send_net},
>+                               s.CreateClient("return", CallClientConfig()),
>+                               {s.CreateSimulationNode(NetworkNodeConfig())});
>+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
>+  // Trigger reroute message, but keep transport unchanged.
>+  s.ChangeRoute(route->forward(), {send_net});
>+
>+  // Allow the controller to stabilize.
>+  s.RunFor(TimeDelta::ms(500));
>+  EXPECT_NEAR(client->send_bandwidth().kbps(), kLinkCapacity.kbps(), 50);
>+  s.ChangeRoute(route->forward(), {send_net});
>+  // Allow new settings to propagate.
>+  s.RunFor(TimeDelta::ms(100));
>+  // Under the trial, the target should be unchanged for low rates.
>+  EXPECT_NEAR(client->send_bandwidth().kbps(), kLinkCapacity.kbps(), 50);
>+}
>+
>+TEST(GoogCcNetworkControllerTest, CutsHighRateInSafeResetTrial) {
>+  const DataRate kLinkCapacity = DataRate::kbps(1000);
>+  const DataRate kStartRate = DataRate::kbps(300);
>+
>+  ScopedFieldTrials trial("WebRTC-Bwe-SafeResetOnRouteChange/Enabled/");
>+  Scenario s("googcc_unit/safe_reset_high_cut", true);
>+  auto send_net = s.CreateSimulationNode([&](NetworkNodeConfig* c) {
>+    c->simulation.bandwidth = kLinkCapacity;
>+    c->simulation.delay = TimeDelta::ms(50);
>+  });
>+  // TODO(srte): replace with SimulatedTimeClient when it supports probing.
>+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
>+    c->transport.cc = TransportControllerConfig::CongestionController::kGoogCc;
>+    c->transport.rates.start_rate = kStartRate;
>+  });
>+  auto* route = s.CreateRoutes(client, {send_net},
>+                               s.CreateClient("return", CallClientConfig()),
>+                               {s.CreateSimulationNode(NetworkNodeConfig())});
>+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
>+
>+  s.ChangeRoute(route->forward(), {send_net});
>+  // Allow the controller to stabilize.
>+  s.RunFor(TimeDelta::ms(500));
>+  EXPECT_NEAR(client->send_bandwidth().kbps(), kLinkCapacity.kbps(), 300);
>+  s.ChangeRoute(route->forward(), {send_net});
>+  // Allow new settings to propagate.
>+  s.RunFor(TimeDelta::ms(50));
>+  // Under the trial, the target should be reset from high values.
>+  EXPECT_NEAR(client->send_bandwidth().kbps(), kStartRate.kbps(), 30);
>+}
>+
>+#ifdef WEBRTC_LINUX  // bugs.webrtc.org/10036
>+#define MAYBE_DetectsHighRateInSafeResetTrial \
>+  DISABLED_DetectsHighRateInSafeResetTrial
>+#else
>+#define MAYBE_DetectsHighRateInSafeResetTrial DetectsHighRateInSafeResetTrial
>+#endif
>+TEST(GoogCcNetworkControllerTest, MAYBE_DetectsHighRateInSafeResetTrial) {
>+  ScopedFieldTrials trial("WebRTC-Bwe-SafeResetOnRouteChange/Enabled/");
>+  const DataRate kInitialLinkCapacity = DataRate::kbps(200);
>+  const DataRate kNewLinkCapacity = DataRate::kbps(800);
>+  const DataRate kStartRate = DataRate::kbps(300);
>+
>+  Scenario s("googcc_unit/safe_reset_high_detect", true);
>+  auto* initial_net = s.CreateSimulationNode([&](NetworkNodeConfig* c) {
>+    c->simulation.bandwidth = kInitialLinkCapacity;
>+    c->simulation.delay = TimeDelta::ms(50);
>+  });
>+  auto* new_net = s.CreateSimulationNode([&](NetworkNodeConfig* c) {
>+    c->simulation.bandwidth = kNewLinkCapacity;
>+    c->simulation.delay = TimeDelta::ms(50);
>+  });
>+  // TODO(srte): replace with SimulatedTimeClient when it supports probing.
>+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
>+    c->transport.cc = TransportControllerConfig::CongestionController::kGoogCc;
>+    c->transport.rates.start_rate = kStartRate;
>+  });
>+  auto* route = s.CreateRoutes(client, {initial_net},
>+                               s.CreateClient("return", CallClientConfig()),
>+                               {s.CreateSimulationNode(NetworkNodeConfig())});
>+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
>+  s.ChangeRoute(route->forward(), {initial_net});
>+
>+  // Allow the controller to stabilize.
>+  s.RunFor(TimeDelta::ms(1000));
>+  EXPECT_NEAR(client->send_bandwidth().kbps(), kInitialLinkCapacity.kbps(), 50);
>+  s.ChangeRoute(route->forward(), {new_net});
>+  // Allow new settings to propagate.
>+  s.RunFor(TimeDelta::ms(100));
>+  // Under the field trial, the target rate should be unchanged since it's lower
>+  // than the starting rate.
>+  EXPECT_NEAR(client->send_bandwidth().kbps(), kInitialLinkCapacity.kbps(), 50);
>+  // However, probing should have made us detect the higher rate.
>+  s.RunFor(TimeDelta::ms(2000));
>+  EXPECT_NEAR(client->send_bandwidth().kbps(), kNewLinkCapacity.kbps(), 200);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
>index e3f3abf2b11ebe1164fc3fd6caec49f9ef8c4c37..fecd1b32d464bcab04aaf37acbf0e49b40a13abc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
>@@ -8,12 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "api/transport/goog_cc_factory.h"
> #include "api/transport/test/network_control_tester.h"
> #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
>-#include "modules/congestion_controller/goog_cc/include/goog_cc_factory.h"
>-#include "test/scenario/scenario.h"
>-
>+#include "test/field_trial.h"
> #include "test/gtest.h"
>+#include "test/scenario/scenario.h"
> 
> using testing::Field;
> using testing::Matcher;
>@@ -84,9 +84,9 @@ class GoogCcNetworkControllerTest : public ::testing::Test {
>                             PacedPacketInfo pacing_info) {
>     PacketResult packet_result;
>     packet_result.sent_packet = SentPacket();
>-    packet_result.sent_packet->send_time = Timestamp::ms(send_time_ms);
>-    packet_result.sent_packet->size = DataSize::bytes(payload_size);
>-    packet_result.sent_packet->pacing_info = pacing_info;
>+    packet_result.sent_packet.send_time = Timestamp::ms(send_time_ms);
>+    packet_result.sent_packet.size = DataSize::bytes(payload_size);
>+    packet_result.sent_packet.pacing_info = pacing_info;
>     packet_result.receive_time = Timestamp::ms(arrival_time_ms);
>     return packet_result;
>   }
>@@ -122,7 +122,7 @@ class GoogCcNetworkControllerTest : public ::testing::Test {
>           CreateResult(current_time_.ms() + delay_buildup, current_time_.ms(),
>                        kPayloadSize, PacedPacketInfo());
>       delay_buildup += delay;
>-      controller_->OnSentPacket(*packet.sent_packet);
>+      controller_->OnSentPacket(packet.sent_packet);
>       TransportPacketsFeedback feedback;
>       feedback.feedback_time = packet.receive_time;
>       feedback.packet_feedbacks.push_back(packet);
>@@ -213,16 +213,13 @@ TEST_F(GoogCcNetworkControllerTest, LongFeedbackDelays) {
>     packets.push_back(
>         CreateResult(i * 100 + 40, 2 * i * 100 + 40, 1500, kPacingInfo1));
> 
>+    TransportPacketsFeedback feedback;
>     for (PacketResult& packet : packets) {
>-      controller_->OnSentPacket(*packet.sent_packet);
>-      // Simulate packet timeout
>-      packet.sent_packet = absl::nullopt;
>+      controller_->OnSentPacket(packet.sent_packet);
>+      feedback.sendless_arrival_times.push_back(packet.receive_time);
>     }
> 
>-    TransportPacketsFeedback feedback;
>     feedback.feedback_time = packets[0].receive_time;
>-    feedback.packet_feedbacks = packets;
>-
>     AdvanceTimeMilliseconds(kFeedbackTimeoutMs);
>     SentPacket later_packet;
>     later_packet.send_time = Timestamp::ms(kFeedbackTimeoutMs + i * 200 + 40);
>@@ -246,7 +243,7 @@ TEST_F(GoogCcNetworkControllerTest, LongFeedbackDelays) {
>     packets.push_back(CreateResult(140, 240, 1500, kPacingInfo1));
> 
>     for (const PacketResult& packet : packets)
>-      controller_->OnSentPacket(*packet.sent_packet);
>+      controller_->OnSentPacket(packet.sent_packet);
> 
>     TransportPacketsFeedback feedback;
>     feedback.feedback_time = packets[0].receive_time;
>@@ -308,8 +305,51 @@ TEST_F(GoogCcNetworkControllerTest,
>               20);
> }
> 
>-TEST_F(GoogCcNetworkControllerTest, ScenarioQuickTest) {
>-  Scenario s("googcc_unit/scenario_quick", false);
>+TEST_F(GoogCcNetworkControllerTest, LimitsToMinRateIfRttIsHighInTrial) {
>+  // The field trial limits maximum RTT to 2 seconds, higher RTT means that the
>+  // controller backs off until it reaches the minimum configured bitrate. This
>+  // allows the RTT to recover faster than the regular control mechanism would
>+  // achieve.
>+  ScopedFieldTrials trial("WebRTC-Bwe-MaxRttLimit/limit:2s/");
>+  // In the test case, we limit the capacity and add a cross traffic packet
>+  // burst that blocks media from being sent. This causes the RTT to quickly
>+  // increase above the threshold in the trial.
>+  const DataRate kLinkCapacity = DataRate::kbps(100);
>+  const DataRate kMinRate = DataRate::kbps(20);
>+  const TimeDelta kBufferBloatDuration = TimeDelta::seconds(10);
>+  Scenario s("googcc_unit/limit_trial", false);
>+  NetworkNodeConfig net_conf;
>+  auto send_net = s.CreateSimulationNode([=](NetworkNodeConfig* c) {
>+    c->simulation.bandwidth = kLinkCapacity;
>+    c->simulation.delay = TimeDelta::ms(100);
>+    c->update_frequency = TimeDelta::ms(5);
>+  });
>+  auto ret_net = s.CreateSimulationNode([](NetworkNodeConfig* c) {
>+    c->simulation.delay = TimeDelta::ms(100);
>+    c->update_frequency = TimeDelta::ms(5);
>+  });
>+  SimulatedTimeClientConfig config;
>+  config.transport.cc =
>+      TransportControllerConfig::CongestionController::kGoogCc;
>+  config.transport.rates.min_rate = kMinRate;
>+  config.transport.rates.start_rate = kLinkCapacity;
>+  SimulatedTimeClient* client = s.CreateSimulatedTimeClient(
>+      "send", config, {PacketStreamConfig()}, {send_net}, {ret_net});
>+  // Run for a few seconds to allow the controller to stabilize.
>+  s.RunFor(TimeDelta::seconds(10));
>+  const DataSize kBloatPacketSize = DataSize::bytes(1000);
>+  const int kBloatPacketCount =
>+      static_cast<int>(kBufferBloatDuration * kLinkCapacity / kBloatPacketSize);
>+  // This will cause the RTT to be large for a while.
>+  s.TriggerPacketBurst({send_net}, kBloatPacketCount, kBloatPacketSize.bytes());
>+  // Wait to allow the high RTT to be detected and acted upon.
>+  s.RunFor(TimeDelta::seconds(4));
>+  // By now the target rate should have dropped to the minimum configured rate.
>+  EXPECT_NEAR(client->target_rate_kbps(), kMinRate.kbps(), 1);
>+}
>+
>+TEST_F(GoogCcNetworkControllerTest, UpdatesTargetRateBasedOnLinkCapacity) {
>+  Scenario s("googcc_unit/target_capacity", false);
>   SimulatedTimeClientConfig config;
>   config.transport.cc =
>       TransportControllerConfig::CongestionController::kGoogCcFeedback;
>@@ -359,5 +399,61 @@ TEST_F(GoogCcNetworkControllerTest, ScenarioQuickTest) {
>   EXPECT_NEAR(client->target_rate_kbps(), 90, 20);
> }
> 
>+TEST_F(GoogCcNetworkControllerTest, DefaultEstimateVariesInSteadyState) {
>+  ScopedFieldTrials trial("WebRTC-Bwe-StableBandwidthEstimate/Disabled/");
>+  Scenario s("googcc_unit/no_stable_varies", false);
>+  SimulatedTimeClientConfig config;
>+  config.transport.cc =
>+      TransportControllerConfig::CongestionController::kGoogCcFeedback;
>+  NetworkNodeConfig net_conf;
>+  net_conf.simulation.bandwidth = DataRate::kbps(500);
>+  net_conf.simulation.delay = TimeDelta::ms(100);
>+  net_conf.update_frequency = TimeDelta::ms(5);
>+  auto send_net = s.CreateSimulationNode(net_conf);
>+  auto ret_net = s.CreateSimulationNode(net_conf);
>+  SimulatedTimeClient* client = s.CreateSimulatedTimeClient(
>+      "send", config, {PacketStreamConfig()}, {send_net}, {ret_net});
>+  // Run for a while to allow the estimate to stabilize.
>+  s.RunFor(TimeDelta::seconds(20));
>+  DataRate min_estimate = DataRate::PlusInfinity();
>+  DataRate max_estimate = DataRate::MinusInfinity();
>+  // Measure variation in steady state.
>+  for (int i = 0; i < 20; ++i) {
>+    min_estimate = std::min(min_estimate, client->link_capacity());
>+    max_estimate = std::max(max_estimate, client->link_capacity());
>+    s.RunFor(TimeDelta::seconds(1));
>+  }
>+  // We should expect drops by at least 15% (default backoff.)
>+  EXPECT_LT(min_estimate / max_estimate, 0.85);
>+}
>+
>+TEST_F(GoogCcNetworkControllerTest, StableEstimateDoesNotVaryInSteadyState) {
>+  ScopedFieldTrials trial("WebRTC-Bwe-StableBandwidthEstimate/Enabled/");
>+  Scenario s("googcc_unit/stable_is_stable", false);
>+  SimulatedTimeClientConfig config;
>+  config.transport.cc =
>+      TransportControllerConfig::CongestionController::kGoogCcFeedback;
>+  NetworkNodeConfig net_conf;
>+  net_conf.simulation.bandwidth = DataRate::kbps(500);
>+  net_conf.simulation.delay = TimeDelta::ms(100);
>+  net_conf.update_frequency = TimeDelta::ms(5);
>+  auto send_net = s.CreateSimulationNode(net_conf);
>+  auto ret_net = s.CreateSimulationNode(net_conf);
>+  SimulatedTimeClient* client = s.CreateSimulatedTimeClient(
>+      "send", config, {PacketStreamConfig()}, {send_net}, {ret_net});
>+  // Run for a while to allow the estimate to stabilize.
>+  s.RunFor(TimeDelta::seconds(20));
>+  DataRate min_estimate = DataRate::PlusInfinity();
>+  DataRate max_estimate = DataRate::MinusInfinity();
>+  // Measure variation in steady state.
>+  for (int i = 0; i < 20; ++i) {
>+    min_estimate = std::min(min_estimate, client->link_capacity());
>+    max_estimate = std::max(max_estimate, client->link_capacity());
>+    s.RunFor(TimeDelta::seconds(1));
>+  }
>+  // We expect no variation under the trial in steady state.
>+  EXPECT_GT(min_estimate / max_estimate, 0.95);
>+}
>+
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/include/goog_cc_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/include/goog_cc_factory.h
>deleted file mode 100644
>index 0d91fff65fa96d8a5cac251fabe8032660464ad5..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/include/goog_cc_factory.h
>+++ /dev/null
>@@ -1,48 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_INCLUDE_GOOG_CC_FACTORY_H_
>-#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_INCLUDE_GOOG_CC_FACTORY_H_
>-#include <memory>
>-
>-#include "api/transport/network_control.h"
>-
>-namespace webrtc {
>-class Clock;
>-class RtcEventLog;
>-
>-class GoogCcNetworkControllerFactory
>-    : public NetworkControllerFactoryInterface {
>- public:
>-  explicit GoogCcNetworkControllerFactory(RtcEventLog*);
>-  std::unique_ptr<NetworkControllerInterface> Create(
>-      NetworkControllerConfig config) override;
>-  TimeDelta GetProcessInterval() const override;
>-
>- private:
>-  RtcEventLog* const event_log_;
>-};
>-
>-// Factory to create packet feedback only GoogCC, this can be used for
>-// connections providing packet receive time feedback but no other reports.
>-class GoogCcFeedbackNetworkControllerFactory
>-    : public NetworkControllerFactoryInterface {
>- public:
>-  explicit GoogCcFeedbackNetworkControllerFactory(RtcEventLog*);
>-  std::unique_ptr<NetworkControllerInterface> Create(
>-      NetworkControllerConfig config) override;
>-  TimeDelta GetProcessInterval() const override;
>-
>- private:
>-  RtcEventLog* const event_log_;
>-};
>-}  // namespace webrtc
>-
>-#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_INCLUDE_GOOG_CC_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.cc
>index 0ad7728302ec20878f9aa504a5becd8c59be612f..45d2fe32114edea59fe41fedb78c2d99d7aead74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.cc
>@@ -10,12 +10,10 @@
> 
> #include "modules/congestion_controller/goog_cc/median_slope_estimator.h"
> 
>-#include <algorithm>
> #include <vector>
> 
>-#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
>-#include "rtc_base/logging.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
>index 0f0d17ea48db06221e4cbdf17f53b77ef3cb5753..7ba2fb5a13bc94d284ee657ece482801886db7d7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
>@@ -170,11 +170,13 @@ int ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
>   return *estimated_bitrate_bps_;
> }
> 
>-absl::optional<int>
>-ProbeBitrateEstimator::FetchAndResetLastEstimatedBitrateBps() {
>+absl::optional<DataRate>
>+ProbeBitrateEstimator::FetchAndResetLastEstimatedBitrate() {
>   absl::optional<int> estimated_bitrate_bps = estimated_bitrate_bps_;
>   estimated_bitrate_bps_.reset();
>-  return estimated_bitrate_bps;
>+  if (estimated_bitrate_bps)
>+    return DataRate::bps(*estimated_bitrate_bps);
>+  return absl::nullopt;
> }
> 
> void ProbeBitrateEstimator::EraseOldClusters(int64_t timestamp_ms) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
>index ed9252137fea1afe3fcfb6a77ca4a441588b6bea..ae83ed380feb875c6c94ea425ab96d8aecd998aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
>@@ -28,7 +28,7 @@ class ProbeBitrateEstimator {
>   // Returns the estimated bitrate if the probe completes a valid cluster.
>   int HandleProbeAndEstimateBitrate(const PacketFeedback& packet_feedback);
> 
>-  absl::optional<int> FetchAndResetLastEstimatedBitrateBps();
>+  absl::optional<DataRate> FetchAndResetLastEstimatedBitrate();
> 
>  private:
>   struct AggregatedCluster {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
>index f8a40f839f385a3005b4a665772292d020ab8555..fcf555924537ab86ad93de34831ebb66d82c0390 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
>@@ -202,7 +202,7 @@ TEST_F(TestProbeBitrateEstimator, IgnoreSizeFirstReceivePacket) {
> }
> 
> TEST_F(TestProbeBitrateEstimator, NoLastEstimatedBitrateBps) {
>-  EXPECT_FALSE(probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps());
>+  EXPECT_FALSE(probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrate());
> }
> 
> TEST_F(TestProbeBitrateEstimator, FetchLastEstimatedBitrateBps) {
>@@ -211,11 +211,11 @@ TEST_F(TestProbeBitrateEstimator, FetchLastEstimatedBitrateBps) {
>   AddPacketFeedback(0, 1000, 20, 30);
>   AddPacketFeedback(0, 1000, 30, 40);
> 
>-  auto estimated_bitrate_bps =
>-      probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps();
>-  EXPECT_TRUE(estimated_bitrate_bps);
>-  EXPECT_NEAR(*estimated_bitrate_bps, 800000, 10);
>-  EXPECT_FALSE(probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps());
>+  auto estimated_bitrate =
>+      probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrate();
>+  EXPECT_TRUE(estimated_bitrate);
>+  EXPECT_NEAR(estimated_bitrate->bps(), 800000, 10);
>+  EXPECT_FALSE(probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrate());
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.cc
>index c2a8e023d7d40eb78ee52c9243b51f607ce4b904..7bbe8e7138a4e627679f6c4856b2e841e37f2d7d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.cc
>@@ -146,6 +146,7 @@ std::vector<ProbeClusterConfig> ProbeController::OnMaxTotalAllocatedBitrate(
>     max_total_allocated_bitrate_ = max_total_allocated_bitrate;
>     return InitiateProbing(at_time_ms, {max_total_allocated_bitrate}, false);
>   }
>+  max_total_allocated_bitrate_ = max_total_allocated_bitrate;
>   return std::vector<ProbeClusterConfig>();
> }
> 
>@@ -260,6 +261,16 @@ std::vector<ProbeClusterConfig> ProbeController::RequestProbe(
>   return std::vector<ProbeClusterConfig>();
> }
> 
>+std::vector<ProbeClusterConfig> ProbeController::InitiateCapacityProbing(
>+    int64_t bitrate_bps,
>+    int64_t at_time_ms) {
>+  if (state_ != State::kWaitingForProbingResult) {
>+    RTC_DCHECK(network_available_);
>+    return InitiateProbing(at_time_ms, {2 * bitrate_bps}, true);
>+  }
>+  return std::vector<ProbeClusterConfig>();
>+}
>+
> void ProbeController::Reset(int64_t at_time_ms) {
>   network_available_ = true;
>   state_ = State::kInit;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.h
>index 6b6d4ae8b4b52a815aa7e8e4ee92161b7e5d66b3..4ddd77f75c37b6dbd9b53893d3c0b5e93944c7f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.h
>@@ -60,6 +60,9 @@ class ProbeController {
>   RTC_WARN_UNUSED_RESULT std::vector<ProbeClusterConfig> RequestProbe(
>       int64_t at_time_ms);
> 
>+  RTC_WARN_UNUSED_RESULT std::vector<ProbeClusterConfig>
>+  InitiateCapacityProbing(int64_t bitrate_bps, int64_t at_time_ms);
>+
>   // Resets the ProbeController to a state equivalent to as if it was just
>   // created EXCEPT for |enable_periodic_alr_probing_|.
>   void Reset(int64_t at_time_ms);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
>index 3264b37e6ddc0e4638b57819b0de87ed0cd8768d..59aad511c008b080cdef5c5e2c3b0543bfcbcb53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
>@@ -8,6 +8,9 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> #include "modules/congestion_controller/goog_cc/test/goog_cc_printer.h"
>+
>+#include <math.h>
>+
> #include "modules/congestion_controller/goog_cc/trendline_estimator.h"
> 
> namespace webrtc {
>@@ -33,9 +36,11 @@ void GoogCcStatePrinter::PrintValues(FILE* out) {
>   RTC_CHECK(controller_);
>   auto* detector = controller_->delay_based_bwe_->delay_detector_.get();
>   auto* trendline_estimator = reinterpret_cast<TrendlineEstimator*>(detector);
>-  fprintf(out, "%i %i %i %.6lf %.6lf %.6lf",
>+  fprintf(out, "%i %f %i %.6lf %.6lf %.6lf",
>           controller_->delay_based_bwe_->rate_control_.rate_control_state_,
>-          controller_->delay_based_bwe_->rate_control_.rate_control_region_,
>+          controller_->delay_based_bwe_->rate_control_
>+                  .link_capacity_estimate_kbps_.value_or(NAN) *
>+              1000 / 8,
>           controller_->alr_detector_->alr_started_time_ms_.has_value(),
>           trendline_estimator->prev_trend_,
>           trendline_estimator->prev_modified_trend_,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h
>index 4dfc059b479a9475abeb961f539b58cde040d2c4..5d4f2c1840f4c97a83075f29cc9dcc6c6e557af2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h
>@@ -12,8 +12,8 @@
> 
> #include <memory>
> 
>+#include "api/transport/goog_cc_factory.h"
> #include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
>-#include "modules/congestion_controller/goog_cc/include/goog_cc_factory.h"
> #include "modules/congestion_controller/test/controller_printer.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.h
>index a7d22a950efad02dc387569a7eb2400ae81bfc97..1826d812480e4720701b86961a56e51041a1908f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.h
>@@ -12,11 +12,11 @@
> 
> #include <stddef.h>
> #include <stdint.h>
>-
> #include <deque>
> #include <utility>
> 
> #include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
>+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
>index ba26272e6fb47b4c20df53a51f8f9e83ac16144f..746e590a59f41e7abc4452f3c35e13b9cb9997a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
>@@ -113,6 +113,9 @@ class SendSideCongestionController
> 
>   void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) override;
> 
>+  void EnableCongestionWindowPushback(int64_t accepted_queue_ms,
>+                                      uint32_t min_pushback_target_bitrate_bps);
>+
>  private:
>   void MaybeTriggerOnNetworkChanged();
> 
>@@ -136,7 +139,7 @@ class SendSideCongestionController
>       RTC_GUARDED_BY(probe_lock_);
> 
>   const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
>-  TransportFeedbackAdapter transport_feedback_adapter_;
>+  LegacyTransportFeedbackAdapter transport_feedback_adapter_;
>   rtc::CriticalSection network_state_lock_;
>   uint32_t last_reported_bitrate_bps_ RTC_GUARDED_BY(network_state_lock_);
>   uint8_t last_reported_fraction_loss_ RTC_GUARDED_BY(network_state_lock_);
>@@ -149,6 +152,8 @@ class SendSideCongestionController
>   bool pacer_paused_;
>   rtc::CriticalSection bwe_lock_;
>   int min_bitrate_bps_ RTC_GUARDED_BY(bwe_lock_);
>+  std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_
>+      RTC_GUARDED_BY(bwe_lock_);
>   std::unique_ptr<DelayBasedBwe> delay_based_bwe_ RTC_GUARDED_BY(bwe_lock_);
>   bool in_cwnd_experiment_;
>   int64_t accepted_queue_ms_;
>@@ -161,7 +166,7 @@ class SendSideCongestionController
>   bool pacer_pushback_experiment_ = false;
>   float encoding_rate_ = 1.0;
> 
>-  const std::unique_ptr<CongestionWindowPushbackController>
>+  std::unique_ptr<CongestionWindowPushbackController>
>       congestion_window_pushback_controller_;
> 
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SendSideCongestionController);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/monitor_interval.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/monitor_interval.cc
>index d430b7f1790a7a38febd40adc6152a59f546554d..281b9224f0c3beb0503f35336157ff05104c48e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/monitor_interval.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/monitor_interval.cc
>@@ -31,25 +31,24 @@ PccMonitorInterval::PccMonitorInterval(const PccMonitorInterval& other) =
> void PccMonitorInterval::OnPacketsFeedback(
>     const std::vector<PacketResult>& packets_results) {
>   for (const PacketResult& packet_result : packets_results) {
>-    if (!packet_result.sent_packet.has_value() ||
>-        packet_result.sent_packet->send_time <= start_time_) {
>+    if (packet_result.sent_packet.send_time <= start_time_) {
>       continue;
>     }
>     // Here we assume that if some packets are reordered with packets sent
>     // after the end of the monitor interval, then they are lost. (Otherwise
>     // it is not clear how long should we wait for packets feedback to arrive).
>-    if (packet_result.sent_packet->send_time >
>+    if (packet_result.sent_packet.send_time >
>         start_time_ + interval_duration_) {
>       feedback_collection_done_ = true;
>       return;
>     }
>     if (packet_result.receive_time.IsInfinite()) {
>-      lost_packets_sent_time_.push_back(packet_result.sent_packet->send_time);
>+      lost_packets_sent_time_.push_back(packet_result.sent_packet.send_time);
>     } else {
>       received_packets_.push_back(
>-          {packet_result.receive_time - packet_result.sent_packet->send_time,
>-           packet_result.sent_packet->send_time});
>-      received_packets_size_ += packet_result.sent_packet->size;
>+          {packet_result.receive_time - packet_result.sent_packet.send_time,
>+           packet_result.sent_packet.send_time});
>+      received_packets_size_ += packet_result.sent_packet.size;
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/pcc_network_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/pcc_network_controller.cc
>index 98cda7409bd1fb4227149419c0527a8b8b3449b5..47ad33eac227bd053221589166e73633a06d6674 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/pcc_network_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/pcc_network_controller.cc
>@@ -148,7 +148,7 @@ NetworkControlUpdate PccNetworkController::OnSentPacket(SentPacket msg) {
>   if (IsTimeoutExpired(msg.send_time)) {
>     DataSize received_size = DataSize::Zero();
>     for (size_t i = 1; i < last_received_packets_.size(); ++i) {
>-      received_size += last_received_packets_[i].sent_packet->size;
>+      received_size += last_received_packets_[i].sent_packet.size;
>     }
>     TimeDelta sending_time = TimeDelta::Zero();
>     if (last_received_packets_.size() > 0)
>@@ -166,7 +166,7 @@ NetworkControlUpdate PccNetworkController::OnSentPacket(SentPacket msg) {
>       msg.send_time - start_time_ >= kStartupDuration) {
>     DataSize received_size = DataSize::Zero();
>     for (size_t i = 1; i < last_received_packets_.size(); ++i) {
>-      received_size += last_received_packets_[i].sent_packet->size;
>+      received_size += last_received_packets_[i].sent_packet.size;
>     }
>     TimeDelta sending_time = TimeDelta::Zero();
>     if (last_received_packets_.size() > 0)
>@@ -260,6 +260,8 @@ bool PccNetworkController::IsFeedbackCollectionDone() const {
> 
> NetworkControlUpdate PccNetworkController::OnTransportPacketsFeedback(
>     TransportPacketsFeedback msg) {
>+  if (msg.packet_feedbacks.empty())
>+    return NetworkControlUpdate();
>   // Save packets to last_received_packets_ array.
>   for (const PacketResult& packet_result : msg.ReceivedWithSendInfo()) {
>     last_received_packets_.push_back(packet_result);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/rtt_tracker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/rtt_tracker.cc
>index a0545759f9f1208ed9161aada4d6a56d4ffe55b1..533a573a7ac8d4bb4077a67d3791633d3ae04856 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/rtt_tracker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/pcc/rtt_tracker.cc
>@@ -23,12 +23,11 @@ void RttTracker::OnPacketsFeedback(
>     Timestamp feedback_received_time) {
>   TimeDelta packet_rtt = TimeDelta::MinusInfinity();
>   for (const PacketResult& packet_result : packet_feedbacks) {
>-    if (!packet_result.sent_packet.has_value() ||
>-        packet_result.receive_time.IsInfinite())
>+    if (packet_result.receive_time.IsInfinite())
>       continue;
>     packet_rtt = std::max<TimeDelta>(
>         packet_rtt,
>-        feedback_received_time - packet_result.sent_packet->send_time);
>+        feedback_received_time - packet_result.sent_packet.send_time);
>   }
>   if (packet_rtt.IsFinite())
>     rtt_estimate_ = (1 - alpha_) * rtt_estimate_ + alpha_ * packet_rtt;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/BUILD.gn
>index 134fb940fdd6528e65ba332d8294b175d1074542..bffe1f893eb54ba5bacdc980545ba58cca8d678c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/BUILD.gn
>@@ -21,8 +21,6 @@ rtc_static_library("congestion_controller") {
>   configs += [ ":bwe_test_logging" ]
>   sources = [
>     "include/send_side_congestion_controller.h",
>-    "pacer_controller.cc",
>-    "pacer_controller.h",
>     "send_side_congestion_controller.cc",
>   ]
> 
>@@ -32,10 +30,12 @@ rtc_static_library("congestion_controller") {
>   }
> 
>   deps = [
>+    ":control_handler",
>     ":transport_feedback",
>     "../:congestion_controller",
>     "../..:module_api",
>     "../../..:webrtc_common",
>+    "../../../api/transport:goog_cc",
>     "../../../api/transport:network_control",
>     "../../../rtc_base:checks",
>     "../../../rtc_base:rate_limiter",
>@@ -43,12 +43,12 @@ rtc_static_library("congestion_controller") {
>     "../../../rtc_base:safe_minmax",
>     "../../../rtc_base:sequenced_task_checker",
>     "../../../rtc_base/experiments:congestion_controller_experiment",
>+    "../../../rtc_base/network:sent_packet",
>     "../../../system_wrappers",
>     "../../../system_wrappers:field_trial",
>     "../../pacing",
>     "../../remote_bitrate_estimator",
>     "../../rtp_rtcp:rtp_rtcp_format",
>-    "../goog_cc",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
> 
>@@ -57,6 +57,38 @@ rtc_static_library("congestion_controller") {
>   }
> }
> 
>+rtc_source_set("control_handler") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "control_handler.cc",
>+    "control_handler.h",
>+  ]
>+
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+
>+  deps = [
>+    "../:congestion_controller",
>+    "../../../api/transport:network_control",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rate_limiter",
>+    "../../../rtc_base:safe_minmax",
>+    "../../../rtc_base:sequenced_task_checker",
>+    "../../../rtc_base/experiments:congestion_controller_experiment",
>+    "../../../system_wrappers",
>+    "../../../system_wrappers:field_trial",
>+    "../../pacing",
>+    "../../remote_bitrate_estimator",
>+    "../../rtp_rtcp:rtp_rtcp_format",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+
>+  if (!build_with_mozilla) {
>+    deps += [ "../../../rtc_base:rtc_base" ]
>+  }
>+}
> rtc_static_library("transport_feedback") {
>   visibility = [ "*" ]
>   sources = [
>@@ -69,8 +101,10 @@ rtc_static_library("transport_feedback") {
>   deps = [
>     "../..:module_api",
>     "../../../api/transport:network_control",
>+    "../../../api/units:data_size",
>     "../../../rtc_base:checks",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/network:sent_packet",
>     "../../../system_wrappers",
>     "../../rtp_rtcp:rtp_rtcp_format",
>   ]
>@@ -97,7 +131,7 @@ if (rtc_include_tests) {
>       "../../../rtc_base:checks",
>       "../../../rtc_base:rtc_base",
>       "../../../rtc_base:rtc_base_approved",
>-      "../../../rtc_base:rtc_base_tests_utils",
>+      "../../../rtc_base/network:sent_packet",
>       "../../../system_wrappers",
>       "../../../test:field_trial",
>       "../../../test:test_support",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/control_handler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/control_handler.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..e6cadb720e9c0b1cca5a26f5e22325f947e5d696
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/control_handler.cc
>@@ -0,0 +1,169 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/rtp/control_handler.h"
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+// When PacerPushbackExperiment is enabled, build-up in the pacer due to
>+// the congestion window and/or data spikes reduces encoder allocations.
>+bool IsPacerPushbackExperimentEnabled() {
>+  return field_trial::IsEnabled("WebRTC-PacerPushbackExperiment");
>+}
>+
>+// By default, pacer emergency stops encoder when buffer reaches a high level.
>+bool IsPacerEmergencyStopDisabled() {
>+  return field_trial::IsEnabled("WebRTC-DisablePacerEmergencyStop");
>+}
>+
>+}  // namespace
>+CongestionControlHandler::CongestionControlHandler(
>+    NetworkChangedObserver* observer,
>+    PacedSender* pacer)
>+    : observer_(observer),
>+      pacer_(pacer),
>+      pacer_pushback_experiment_(IsPacerPushbackExperimentEnabled()),
>+      disable_pacer_emergency_stop_(IsPacerEmergencyStopDisabled()) {
>+  sequenced_checker_.Detach();
>+}
>+
>+CongestionControlHandler::~CongestionControlHandler() {}
>+
>+void CongestionControlHandler::PostUpdates(NetworkControlUpdate update) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  if (update.congestion_window) {
>+    if (update.congestion_window->IsFinite())
>+      pacer_->SetCongestionWindow(update.congestion_window->bytes());
>+    else
>+      pacer_->SetCongestionWindow(PacedSender::kNoCongestionWindow);
>+  }
>+  if (update.pacer_config) {
>+    pacer_->SetPacingRates(update.pacer_config->data_rate().bps(),
>+                           update.pacer_config->pad_rate().bps());
>+  }
>+  for (const auto& probe : update.probe_cluster_configs) {
>+    int64_t bitrate_bps = probe.target_data_rate.bps();
>+    pacer_->CreateProbeCluster(bitrate_bps);
>+  }
>+  if (update.target_rate) {
>+    current_target_rate_msg_ = *update.target_rate;
>+    OnNetworkInvalidation();
>+  }
>+}
>+
>+void CongestionControlHandler::OnNetworkAvailability(NetworkAvailability msg) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  if (network_available_ != msg.network_available) {
>+    network_available_ = msg.network_available;
>+    pacer_->UpdateOutstandingData(0);
>+    SetPacerState(!msg.network_available);
>+    OnNetworkInvalidation();
>+  }
>+}
>+
>+void CongestionControlHandler::OnOutstandingData(DataSize in_flight_data) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  pacer_->UpdateOutstandingData(in_flight_data.bytes());
>+  OnNetworkInvalidation();
>+}
>+
>+void CongestionControlHandler::OnPacerQueueUpdate(
>+    TimeDelta expected_queue_time) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  pacer_expected_queue_ms_ = expected_queue_time.ms();
>+  OnNetworkInvalidation();
>+}
>+
>+void CongestionControlHandler::SetPacerState(bool paused) {
>+  if (paused && !pacer_paused_)
>+    pacer_->Pause();
>+  else if (!paused && pacer_paused_)
>+    pacer_->Resume();
>+  pacer_paused_ = paused;
>+}
>+
>+void CongestionControlHandler::OnNetworkInvalidation() {
>+  if (!current_target_rate_msg_.has_value())
>+    return;
>+
>+  uint32_t target_bitrate_bps = current_target_rate_msg_->target_rate.bps();
>+  int64_t rtt_ms =
>+      current_target_rate_msg_->network_estimate.round_trip_time.ms();
>+  float loss_rate_ratio =
>+      current_target_rate_msg_->network_estimate.loss_rate_ratio;
>+
>+  int loss_ratio_255 = loss_rate_ratio * 255;
>+  uint8_t fraction_loss =
>+      rtc::dchecked_cast<uint8_t>(rtc::SafeClamp(loss_ratio_255, 0, 255));
>+
>+  int64_t probing_interval_ms =
>+      current_target_rate_msg_->network_estimate.bwe_period.ms();
>+
>+  if (!network_available_) {
>+    target_bitrate_bps = 0;
>+  } else if (pacer_pushback_experiment_) {
>+    int64_t queue_length_ms = pacer_expected_queue_ms_;
>+
>+    if (queue_length_ms == 0) {
>+      encoding_rate_ratio_ = 1.0;
>+    } else if (queue_length_ms > 50) {
>+      double encoding_ratio = 1.0 - queue_length_ms / 1000.0;
>+      encoding_rate_ratio_ = std::min(encoding_rate_ratio_, encoding_ratio);
>+      encoding_rate_ratio_ = std::max(encoding_rate_ratio_, 0.0);
>+    }
>+
>+    target_bitrate_bps *= encoding_rate_ratio_;
>+    target_bitrate_bps = target_bitrate_bps < 50000 ? 0 : target_bitrate_bps;
>+  } else if (!disable_pacer_emergency_stop_) {
>+    target_bitrate_bps = IsSendQueueFull() ? 0 : target_bitrate_bps;
>+  }
>+
>+  if (HasNetworkParametersToReportChanged(target_bitrate_bps, fraction_loss,
>+                                          rtt_ms)) {
>+    observer_->OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt_ms,
>+                                probing_interval_ms);
>+  }
>+}
>+bool CongestionControlHandler::HasNetworkParametersToReportChanged(
>+    int64_t target_bitrate_bps,
>+    uint8_t fraction_loss,
>+    int64_t rtt_ms) {
>+  bool changed = last_reported_target_bitrate_bps_ != target_bitrate_bps ||
>+                 (target_bitrate_bps > 0 &&
>+                  (last_reported_fraction_loss_ != fraction_loss ||
>+                   last_reported_rtt_ms_ != rtt_ms));
>+  if (changed &&
>+      (last_reported_target_bitrate_bps_ == 0 || target_bitrate_bps == 0)) {
>+    RTC_LOG(LS_INFO) << "Bitrate estimate state changed, BWE: "
>+                     << target_bitrate_bps << " bps.";
>+  }
>+  last_reported_target_bitrate_bps_ = target_bitrate_bps;
>+  last_reported_fraction_loss_ = fraction_loss;
>+  last_reported_rtt_ms_ = rtt_ms;
>+  return changed;
>+}
>+
>+bool CongestionControlHandler::IsSendQueueFull() const {
>+  return pacer_expected_queue_ms_ > PacedSender::kMaxQueueLengthMs;
>+}
>+
>+absl::optional<TargetTransferRate>
>+CongestionControlHandler::last_transfer_rate() {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  return current_target_rate_msg_;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/control_handler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/control_handler.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..456c2a9e27128406f5bb8040a56dd44cd8905600
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/control_handler.h
>@@ -0,0 +1,72 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_
>+#define MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_
>+
>+#include <algorithm>
>+#include <memory>
>+
>+#include "api/transport/network_control.h"
>+#include "modules/congestion_controller/include/network_changed_observer.h"
>+#include "modules/pacing/paced_sender.h"
>+#include "rtc_base/sequenced_task_checker.h"
>+
>+namespace webrtc {
>+// This is used to observe the network controller state and route calls to
>+// the proper handler. It also keeps cached values for safe asynchronous use.
>+// This makes sure that things running on the worker queue can't access state
>+// in SendSideCongestionController, which would risk causing data race on
>+// destruction unless members are properly ordered.
>+class CongestionControlHandler {
>+ public:
>+  CongestionControlHandler(NetworkChangedObserver* observer,
>+                           PacedSender* pacer);
>+  ~CongestionControlHandler();
>+
>+  void PostUpdates(NetworkControlUpdate update);
>+
>+  void OnNetworkAvailability(NetworkAvailability msg);
>+  void OnOutstandingData(DataSize in_flight_data);
>+  void OnPacerQueueUpdate(TimeDelta expected_queue_time);
>+
>+  absl::optional<TargetTransferRate> last_transfer_rate();
>+
>+ private:
>+  void SetPacerState(bool paused);
>+  void OnNetworkInvalidation();
>+  bool IsSendQueueFull() const;
>+  bool HasNetworkParametersToReportChanged(int64_t bitrate_bps,
>+                                           float loss_rate_ratio,
>+                                           TimeDelta rtt);
>+
>+  bool HasNetworkParametersToReportChanged(int64_t bitrate_bps,
>+                                           uint8_t fraction_loss,
>+                                           int64_t rtt);
>+  NetworkChangedObserver* observer_ = nullptr;
>+  PacedSender* const pacer_;
>+
>+  absl::optional<TargetTransferRate> current_target_rate_msg_;
>+  bool network_available_ = true;
>+  bool pacer_paused_ = false;
>+  int64_t last_reported_target_bitrate_bps_ = 0;
>+  uint8_t last_reported_fraction_loss_ = 0;
>+  int64_t last_reported_rtt_ms_ = 0;
>+  const bool pacer_pushback_experiment_;
>+  const bool disable_pacer_emergency_stop_;
>+  uint32_t min_pushback_target_bitrate_bps_;
>+  int64_t pacer_expected_queue_ms_ = 0;
>+  double encoding_rate_ratio_ = 1.0;
>+
>+  rtc::SequencedTaskChecker sequenced_checker_;
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(CongestionControlHandler);
>+};
>+}  // namespace webrtc
>+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/include/send_side_congestion_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/include/send_side_congestion_controller.h
>index efe4d2ff47756b441f52d64f149bf8a2ecac2854..55465cb75f88f695cec986697dfebddeb911f364 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/include/send_side_congestion_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/include/send_side_congestion_controller.h
>@@ -21,7 +21,7 @@
> #include "api/transport/network_types.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/congestion_controller/include/send_side_congestion_controller_interface.h"
>-#include "modules/congestion_controller/rtp/pacer_controller.h"
>+#include "modules/congestion_controller/rtp/control_handler.h"
> #include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
> #include "modules/include/module.h"
> #include "modules/include/module_common_types.h"
>@@ -40,18 +40,11 @@ struct SentPacket;
> namespace webrtc {
> 
> class Clock;
>-class RateLimiter;
> class RtcEventLog;
> 
> namespace webrtc_cc {
> 
> namespace send_side_cc_internal {
>-// This is used to observe the network controller state and route calls to
>-// the proper handler. It also keeps cached values for safe asynchronous use.
>-// This makes sure that things running on the worker queue can't access state
>-// in SendSideCongestionController, which would risk causing data race on
>-// destruction unless members are properly ordered.
>-class ControlHandler;
> 
> // TODO(srte): Make sure the PeriodicTask implementation is reusable and move it
> // to task_queue.h.
>@@ -182,10 +175,7 @@ class SendSideCongestionController
>   const std::unique_ptr<NetworkControllerFactoryInterface>
>       controller_factory_fallback_ RTC_GUARDED_BY(task_queue_);
> 
>-  const std::unique_ptr<PacerController> pacer_controller_
>-      RTC_GUARDED_BY(task_queue_);
>-
>-  std::unique_ptr<send_side_cc_internal::ControlHandler> control_handler_
>+  std::unique_ptr<CongestionControlHandler> control_handler_
>       RTC_GUARDED_BY(task_queue_);
> 
>   std::unique_ptr<NetworkControllerInterface> controller_
>@@ -201,6 +191,7 @@ class SendSideCongestionController
>   NetworkControllerConfig initial_config_ RTC_GUARDED_BY(task_queue_);
>   StreamsConfig streams_config_ RTC_GUARDED_BY(task_queue_);
> 
>+  const bool reset_feedback_on_route_change_;
>   const bool send_side_bwe_with_overhead_;
>   // Transport overhead is written by OnNetworkRouteChanged and read by
>   // AddPacket.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.cc
>deleted file mode 100644
>index af062e881eeef20800be0077bcc977433337cd58..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.cc
>+++ /dev/null
>@@ -1,70 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/congestion_controller/rtp/pacer_controller.h"
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-
>-namespace webrtc {
>-
>-PacerController::PacerController(PacedSender* pacer) : pacer_(pacer) {
>-  sequenced_checker_.Detach();
>-}
>-
>-PacerController::~PacerController() = default;
>-
>-void PacerController::OnCongestionWindow(DataSize congestion_window) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  if (congestion_window.IsFinite())
>-    pacer_->SetCongestionWindow(congestion_window.bytes());
>-  else
>-    pacer_->SetCongestionWindow(PacedSender::kNoCongestionWindow);
>-}
>-
>-void PacerController::OnNetworkAvailability(NetworkAvailability msg) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  network_available_ = msg.network_available;
>-  pacer_->UpdateOutstandingData(0);
>-  SetPacerState(!msg.network_available);
>-}
>-
>-void PacerController::OnNetworkRouteChange(NetworkRouteChange) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  pacer_->UpdateOutstandingData(0);
>-}
>-
>-void PacerController::OnPacerConfig(PacerConfig msg) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  DataRate pacing_rate = msg.data_window / msg.time_window;
>-  DataRate padding_rate = msg.pad_window / msg.time_window;
>-  pacer_->SetPacingRates(pacing_rate.bps(), padding_rate.bps());
>-}
>-
>-void PacerController::OnProbeClusterConfig(ProbeClusterConfig config) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  int64_t bitrate_bps = config.target_data_rate.bps();
>-  pacer_->CreateProbeCluster(bitrate_bps);
>-}
>-
>-void PacerController::OnOutstandingData(DataSize in_flight_data) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  pacer_->UpdateOutstandingData(in_flight_data.bytes());
>-}
>-
>-void PacerController::SetPacerState(bool paused) {
>-  if (paused && !pacer_paused_)
>-    pacer_->Pause();
>-  else if (!paused && pacer_paused_)
>-    pacer_->Resume();
>-  pacer_paused_ = paused;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.h
>deleted file mode 100644
>index b778b784890355491474aeeee56312be37498103..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.h
>+++ /dev/null
>@@ -1,50 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_CONGESTION_CONTROLLER_RTP_PACER_CONTROLLER_H_
>-#define MODULES_CONGESTION_CONTROLLER_RTP_PACER_CONTROLLER_H_
>-
>-#include <memory>
>-
>-#include "api/transport/network_types.h"
>-#include "modules/pacing/paced_sender.h"
>-#include "rtc_base/sequenced_task_checker.h"
>-
>-namespace webrtc {
>-class Clock;
>-
>-// Wrapper class to control pacer using task queues. Note that this class is
>-// only designed to be used from a single task queue and has no built in
>-// concurrency safety.
>-// TODO(srte): Integrate this interface directly into PacedSender.
>-class PacerController {
>- public:
>-  explicit PacerController(PacedSender* pacer);
>-  ~PacerController();
>-  void OnCongestionWindow(DataSize msg);
>-  void OnNetworkAvailability(NetworkAvailability msg);
>-  void OnNetworkRouteChange(NetworkRouteChange msg);
>-  void OnOutstandingData(DataSize in_flight_data);
>-  void OnPacerConfig(PacerConfig msg);
>-  void OnProbeClusterConfig(ProbeClusterConfig msg);
>-
>- private:
>-  void SetPacerState(bool paused);
>-  PacedSender* const pacer_;
>-
>-  absl::optional<PacerConfig> current_pacer_config_;
>-  bool pacer_paused_ = false;
>-  bool network_available_ = true;
>-
>-  rtc::SequencedTaskChecker sequenced_checker_;
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PacerController);
>-};
>-}  // namespace webrtc
>-#endif  // MODULES_CONGESTION_CONTROLLER_RTP_PACER_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller.cc
>index 266a5dc7400cbd11079294f59809cad87bc51584..982380f457aed3c84fe3dd097b1f54dbb77615e0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller.cc
>@@ -15,19 +15,19 @@
> #include <memory>
> #include <vector>
> #include "absl/memory/memory.h"
>+#include "api/transport/goog_cc_factory.h"
> #include "api/transport/network_types.h"
>-#include "modules/congestion_controller/congestion_window_pushback_controller.h"
>-#include "modules/congestion_controller/goog_cc/include/goog_cc_factory.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/format_macros.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/network/sent_packet.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/numerics/safe_minmax.h"
> #include "rtc_base/rate_limiter.h"
> #include "rtc_base/sequenced_task_checker.h"
>-#include "rtc_base/socket.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/field_trial.h"
> 
>@@ -37,70 +37,8 @@ namespace webrtc {
> namespace webrtc_cc {
> namespace {
> using send_side_cc_internal::PeriodicTask;
>-
>-const char kCwndExperiment[] = "WebRTC-CwndExperiment";
>-
>-// When CongestionWindowPushback is enabled, the pacer is oblivious to
>-// the congestion window. The relation between outstanding data and
>-// the congestion window affects encoder allocations directly.
>-const char kCongestionPushbackExperiment[] = "WebRTC-CongestionWindowPushback";
>-
>-// When PacerPushbackExperiment is enabled, build-up in the pacer due to
>-// the congestion window and/or data spikes reduces encoder allocations.
>-const char kPacerPushbackExperiment[] = "WebRTC-PacerPushbackExperiment";
> const int64_t PacerQueueUpdateIntervalMs = 25;
> 
>-bool IsPacerPushbackExperimentEnabled() {
>-  return webrtc::field_trial::IsEnabled(kPacerPushbackExperiment);
>-}
>-
>-bool IsCongestionWindowPushbackExperimentEnabled() {
>-  return webrtc::field_trial::IsEnabled(kCongestionPushbackExperiment) &&
>-         webrtc::field_trial::IsEnabled(kCwndExperiment);
>-}
>-
>-std::unique_ptr<CongestionWindowPushbackController>
>-MaybeInitalizeCongestionWindowPushbackController() {
>-  return IsCongestionWindowPushbackExperimentEnabled()
>-             ? absl::make_unique<CongestionWindowPushbackController>()
>-             : nullptr;
>-}
>-
>-void SortPacketFeedbackVector(std::vector<webrtc::PacketFeedback>* input) {
>-  std::sort(input->begin(), input->end(), PacketFeedbackComparator());
>-}
>-
>-PacketResult NetworkPacketFeedbackFromRtpPacketFeedback(
>-    const webrtc::PacketFeedback& pf) {
>-  PacketResult feedback;
>-  if (pf.arrival_time_ms == webrtc::PacketFeedback::kNotReceived)
>-    feedback.receive_time = Timestamp::PlusInfinity();
>-  else
>-    feedback.receive_time = Timestamp::ms(pf.arrival_time_ms);
>-  if (pf.send_time_ms != webrtc::PacketFeedback::kNoSendTime) {
>-    feedback.sent_packet = SentPacket();
>-    feedback.sent_packet->sequence_number = pf.long_sequence_number;
>-    feedback.sent_packet->send_time = Timestamp::ms(pf.send_time_ms);
>-    feedback.sent_packet->size = DataSize::bytes(pf.payload_size);
>-    feedback.sent_packet->pacing_info = pf.pacing_info;
>-  }
>-  return feedback;
>-}
>-
>-std::vector<PacketResult> PacketResultsFromRtpFeedbackVector(
>-    const std::vector<PacketFeedback>& feedback_vector) {
>-  RTC_DCHECK(std::is_sorted(feedback_vector.begin(), feedback_vector.end(),
>-                            PacketFeedbackComparator()));
>-
>-  std::vector<PacketResult> packet_feedbacks;
>-  packet_feedbacks.reserve(feedback_vector.size());
>-  for (const PacketFeedback& rtp_feedback : feedback_vector) {
>-    auto feedback = NetworkPacketFeedbackFromRtpPacketFeedback(rtp_feedback);
>-    packet_feedbacks.push_back(feedback);
>-  }
>-  return packet_feedbacks;
>-}
>-
> TargetRateConstraints ConvertConstraints(int min_bitrate_bps,
>                                          int max_bitrate_bps,
>                                          int start_bitrate_bps,
>@@ -166,177 +104,6 @@ static PeriodicTask* StartPeriodicTask(rtc::TaskQueue* task_queue,
> 
> }  // namespace
> 
>-namespace send_side_cc_internal {
>-class ControlHandler {
>- public:
>-  ControlHandler(NetworkChangedObserver* observer,
>-                 PacerController* pacer_controller,
>-                 const Clock* clock);
>-
>-  void PostUpdates(NetworkControlUpdate update);
>-
>-  void OnNetworkAvailability(NetworkAvailability msg);
>-  void OnOutstandingData(DataSize in_flight_data);
>-  void OnPacerQueueUpdate(TimeDelta expected_queue_time);
>-
>-  absl::optional<TargetTransferRate> last_transfer_rate();
>-
>- private:
>-  void OnNetworkInvalidation();
>-  bool GetNetworkParameters(int32_t* estimated_bitrate_bps,
>-                            uint8_t* fraction_loss,
>-                            int64_t* rtt_ms);
>-  bool IsSendQueueFull() const;
>-  bool HasNetworkParametersToReportChanged(int64_t bitrate_bps,
>-                                           uint8_t fraction_loss,
>-                                           int64_t rtt);
>-  NetworkChangedObserver* observer_ = nullptr;
>-  PacerController* pacer_controller_;
>-
>-  absl::optional<TargetTransferRate> current_target_rate_msg_;
>-  bool network_available_ = true;
>-  int64_t last_reported_target_bitrate_bps_ = 0;
>-  uint8_t last_reported_fraction_loss_ = 0;
>-  int64_t last_reported_rtt_ms_ = 0;
>-  const bool pacer_pushback_experiment_ = false;
>-  uint32_t min_pushback_target_bitrate_bps_;
>-  int64_t pacer_expected_queue_ms_ = 0;
>-  double encoding_rate_ratio_ = 1.0;
>-  const std::unique_ptr<CongestionWindowPushbackController>
>-      congestion_window_pushback_controller_;
>-
>-  rtc::SequencedTaskChecker sequenced_checker_;
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ControlHandler);
>-};
>-
>-ControlHandler::ControlHandler(NetworkChangedObserver* observer,
>-                               PacerController* pacer_controller,
>-                               const Clock* clock)
>-    : observer_(observer),
>-      pacer_controller_(pacer_controller),
>-      pacer_pushback_experiment_(IsPacerPushbackExperimentEnabled()),
>-      congestion_window_pushback_controller_(
>-          MaybeInitalizeCongestionWindowPushbackController()) {
>-  sequenced_checker_.Detach();
>-}
>-
>-void ControlHandler::PostUpdates(NetworkControlUpdate update) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  if (update.congestion_window) {
>-    if (congestion_window_pushback_controller_) {
>-      congestion_window_pushback_controller_->SetDataWindow(
>-          update.congestion_window.value());
>-    } else {
>-      pacer_controller_->OnCongestionWindow(*update.congestion_window);
>-    }
>-  }
>-  if (update.pacer_config) {
>-    pacer_controller_->OnPacerConfig(*update.pacer_config);
>-  }
>-  for (const auto& probe : update.probe_cluster_configs) {
>-    pacer_controller_->OnProbeClusterConfig(probe);
>-  }
>-  if (update.target_rate) {
>-    current_target_rate_msg_ = *update.target_rate;
>-    OnNetworkInvalidation();
>-  }
>-}
>-
>-void ControlHandler::OnNetworkAvailability(NetworkAvailability msg) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  network_available_ = msg.network_available;
>-  OnNetworkInvalidation();
>-}
>-
>-void ControlHandler::OnOutstandingData(DataSize in_flight_data) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  if (congestion_window_pushback_controller_) {
>-    congestion_window_pushback_controller_->UpdateOutstandingData(
>-        in_flight_data.bytes());
>-  }
>-  OnNetworkInvalidation();
>-}
>-
>-void ControlHandler::OnPacerQueueUpdate(TimeDelta expected_queue_time) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  pacer_expected_queue_ms_ = expected_queue_time.ms();
>-  OnNetworkInvalidation();
>-}
>-
>-void ControlHandler::OnNetworkInvalidation() {
>-  if (!current_target_rate_msg_.has_value())
>-    return;
>-
>-  uint32_t target_bitrate_bps = current_target_rate_msg_->target_rate.bps();
>-  int64_t rtt_ms =
>-      current_target_rate_msg_->network_estimate.round_trip_time.ms();
>-  float loss_rate_ratio =
>-      current_target_rate_msg_->network_estimate.loss_rate_ratio;
>-
>-  int loss_ratio_255 = loss_rate_ratio * 255;
>-  uint8_t fraction_loss =
>-      rtc::dchecked_cast<uint8_t>(rtc::SafeClamp(loss_ratio_255, 0, 255));
>-
>-  int64_t probing_interval_ms =
>-      current_target_rate_msg_->network_estimate.bwe_period.ms();
>-
>-  if (!network_available_) {
>-    target_bitrate_bps = 0;
>-  } else if (congestion_window_pushback_controller_) {
>-    target_bitrate_bps =
>-        congestion_window_pushback_controller_->UpdateTargetBitrate(
>-            target_bitrate_bps);
>-  } else if (!pacer_pushback_experiment_) {
>-    target_bitrate_bps = IsSendQueueFull() ? 0 : target_bitrate_bps;
>-  } else {
>-    int64_t queue_length_ms = pacer_expected_queue_ms_;
>-
>-    if (queue_length_ms == 0) {
>-      encoding_rate_ratio_ = 1.0;
>-    } else if (queue_length_ms > 50) {
>-      double encoding_ratio = 1.0 - queue_length_ms / 1000.0;
>-      encoding_rate_ratio_ = std::min(encoding_rate_ratio_, encoding_ratio);
>-      encoding_rate_ratio_ = std::max(encoding_rate_ratio_, 0.0);
>-    }
>-
>-    target_bitrate_bps *= encoding_rate_ratio_;
>-    target_bitrate_bps = target_bitrate_bps < 50000 ? 0 : target_bitrate_bps;
>-  }
>-  if (HasNetworkParametersToReportChanged(target_bitrate_bps, fraction_loss,
>-                                          rtt_ms)) {
>-    observer_->OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt_ms,
>-                                probing_interval_ms);
>-  }
>-}
>-bool ControlHandler::HasNetworkParametersToReportChanged(
>-    int64_t target_bitrate_bps,
>-    uint8_t fraction_loss,
>-    int64_t rtt_ms) {
>-  bool changed = last_reported_target_bitrate_bps_ != target_bitrate_bps ||
>-                 (target_bitrate_bps > 0 &&
>-                  (last_reported_fraction_loss_ != fraction_loss ||
>-                   last_reported_rtt_ms_ != rtt_ms));
>-  if (changed &&
>-      (last_reported_target_bitrate_bps_ == 0 || target_bitrate_bps == 0)) {
>-    RTC_LOG(LS_INFO) << "Bitrate estimate state changed, BWE: "
>-                     << target_bitrate_bps << " bps.";
>-  }
>-  last_reported_target_bitrate_bps_ = target_bitrate_bps;
>-  last_reported_fraction_loss_ = fraction_loss;
>-  last_reported_rtt_ms_ = rtt_ms;
>-  return changed;
>-}
>-
>-bool ControlHandler::IsSendQueueFull() const {
>-  return pacer_expected_queue_ms_ > PacedSender::kMaxQueueLengthMs;
>-}
>-
>-absl::optional<TargetTransferRate> ControlHandler::last_transfer_rate() {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  return current_target_rate_msg_;
>-}
>-}  // namespace send_side_cc_internal
>-
> SendSideCongestionController::SendSideCongestionController(
>     const Clock* clock,
>     rtc::TaskQueue* task_queue,
>@@ -352,10 +119,11 @@ SendSideCongestionController::SendSideCongestionController(
>       controller_factory_with_feedback_(controller_factory),
>       controller_factory_fallback_(
>           absl::make_unique<GoogCcNetworkControllerFactory>(event_log)),
>-      pacer_controller_(absl::make_unique<PacerController>(pacer_)),
>       process_interval_(controller_factory_fallback_->GetProcessInterval()),
>       last_report_block_time_(Timestamp::ms(clock_->TimeInMilliseconds())),
>       observer_(nullptr),
>+      reset_feedback_on_route_change_(
>+          !field_trial::IsEnabled("WebRTC-Bwe-NoFeedbackReset")),
>       send_side_bwe_with_overhead_(
>           webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
>       transport_overhead_bytes_per_packet_(0),
>@@ -398,8 +166,8 @@ void SendSideCongestionController::MaybeRecreateControllers() {
>   if (!network_available_ || !observer_)
>     return;
>   if (!control_handler_) {
>-    control_handler_ = absl::make_unique<send_side_cc_internal::ControlHandler>(
>-        observer_, pacer_controller_.get(), clock_);
>+    control_handler_ =
>+        absl::make_unique<CongestionControlHandler>(observer_, pacer_);
>   }
> 
>   initial_config_.constraints.at_time =
>@@ -489,8 +257,9 @@ void SendSideCongestionController::OnNetworkRouteChanged(
>     int start_bitrate_bps,
>     int min_bitrate_bps,
>     int max_bitrate_bps) {
>-  transport_feedback_adapter_.SetNetworkIds(network_route.local_network_id,
>-                                            network_route.remote_network_id);
>+  if (reset_feedback_on_route_change_)
>+    transport_feedback_adapter_.SetNetworkIds(network_route.local_network_id,
>+                                              network_route.remote_network_id);
>   transport_overhead_bytes_per_packet_ = network_route.packet_overhead;
> 
>   NetworkRouteChange msg;
>@@ -505,7 +274,7 @@ void SendSideCongestionController::OnNetworkRouteChanged(
>     } else {
>       UpdateInitialConstraints(msg.constraints);
>     }
>-    pacer_controller_->OnNetworkRouteChange(msg);
>+    pacer_->UpdateOutstandingData(0);
>   });
> }
> 
>@@ -572,7 +341,6 @@ void SendSideCongestionController::SignalNetworkState(NetworkState state) {
>     network_available_ = msg.network_available;
>     if (controller_) {
>       control_handler_->PostUpdates(controller_->OnNetworkAvailability(msg));
>-      pacer_controller_->OnNetworkAvailability(msg);
>       control_handler_->OnNetworkAvailability(msg);
>     } else {
>       MaybeCreateControllers();
>@@ -582,27 +350,16 @@ void SendSideCongestionController::SignalNetworkState(NetworkState state) {
> 
> void SendSideCongestionController::OnSentPacket(
>     const rtc::SentPacket& sent_packet) {
>-  // We're not interested in packets without an id, which may be stun packets,
>-  // etc, sent on the same transport.
>-  if (sent_packet.packet_id == -1)
>-    return;
>-  transport_feedback_adapter_.OnSentPacket(sent_packet.packet_id,
>-                                           sent_packet.send_time_ms);
>-  MaybeUpdateOutstandingData();
>-  auto packet = transport_feedback_adapter_.GetPacket(sent_packet.packet_id);
>-  if (packet.has_value()) {
>-    SentPacket msg;
>-    msg.size = DataSize::bytes(packet->payload_size);
>-    msg.send_time = Timestamp::ms(packet->send_time_ms);
>-    msg.sequence_number = packet->long_sequence_number;
>-    msg.data_in_flight =
>-        DataSize::bytes(transport_feedback_adapter_.GetOutstandingBytes());
>-    task_queue_->PostTask([this, msg]() {
>+  absl::optional<SentPacket> packet_msg =
>+      transport_feedback_adapter_.ProcessSentPacket(sent_packet);
>+  if (packet_msg) {
>+    task_queue_->PostTask([this, packet_msg]() {
>       RTC_DCHECK_RUN_ON(task_queue_);
>       if (controller_)
>-        control_handler_->PostUpdates(controller_->OnSentPacket(msg));
>+        control_handler_->PostUpdates(controller_->OnSentPacket(*packet_msg));
>     });
>   }
>+  MaybeUpdateOutstandingData();
> }
> 
> void SendSideCongestionController::OnRttUpdate(int64_t avg_rtt_ms,
>@@ -687,39 +444,24 @@ void SendSideCongestionController::AddPacket(
> void SendSideCongestionController::OnTransportFeedback(
>     const rtcp::TransportFeedback& feedback) {
>   RTC_DCHECK_RUNS_SERIALIZED(&worker_race_);
>-  int64_t feedback_time_ms = clock_->TimeInMilliseconds();
> 
>-  DataSize prior_in_flight =
>-      DataSize::bytes(transport_feedback_adapter_.GetOutstandingBytes());
>-  transport_feedback_adapter_.OnTransportFeedback(feedback);
>-  MaybeUpdateOutstandingData();
>-
>-  std::vector<PacketFeedback> feedback_vector =
>-      transport_feedback_adapter_.GetTransportFeedbackVector();
>-  SortPacketFeedbackVector(&feedback_vector);
>-
>-  if (!feedback_vector.empty()) {
>-    TransportPacketsFeedback msg;
>-    msg.packet_feedbacks = PacketResultsFromRtpFeedbackVector(feedback_vector);
>-    msg.feedback_time = Timestamp::ms(feedback_time_ms);
>-    msg.prior_in_flight = prior_in_flight;
>-    msg.data_in_flight =
>-        DataSize::bytes(transport_feedback_adapter_.GetOutstandingBytes());
>-    task_queue_->PostTask([this, msg]() {
>+  absl::optional<TransportPacketsFeedback> feedback_msg =
>+      transport_feedback_adapter_.ProcessTransportFeedback(feedback);
>+  if (feedback_msg) {
>+    task_queue_->PostTask([this, feedback_msg]() {
>       RTC_DCHECK_RUN_ON(task_queue_);
>       if (controller_)
>         control_handler_->PostUpdates(
>-            controller_->OnTransportPacketsFeedback(msg));
>+            controller_->OnTransportPacketsFeedback(*feedback_msg));
>     });
>   }
>+  MaybeUpdateOutstandingData();
> }
> 
> void SendSideCongestionController::MaybeUpdateOutstandingData() {
>-  DataSize in_flight_data =
>-      DataSize::bytes(transport_feedback_adapter_.GetOutstandingBytes());
>+  DataSize in_flight_data = transport_feedback_adapter_.GetOutstandingData();
>   task_queue_->PostTask([this, in_flight_data]() {
>     RTC_DCHECK_RUN_ON(task_queue_);
>-    pacer_controller_->OnOutstandingData(in_flight_data);
>     if (control_handler_)
>       control_handler_->OnOutstandingData(in_flight_data);
>   });
>@@ -740,7 +482,7 @@ void SendSideCongestionController::PostPeriodicTasksForTest() {
> }
> 
> void SendSideCongestionController::WaitOnTasksForTest() {
>-  rtc::Event event(false, false);
>+  rtc::Event event;
>   task_queue_->PostTask([&event]() { event.Set(); });
>   event.Wait(rtc::Event::kForever);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc
>index d610afe13d0f3e2145187062788a693f6f4d7de6..6ed7a49553f1924378ab2e78000d225c1951f84c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc
>@@ -17,7 +17,7 @@
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
>-#include "rtc_base/socket.h"
>+#include "rtc_base/network/sent_packet.h"
> #include "system_wrappers/include/clock.h"
> #include "test/field_trial.h"
> #include "test/gmock.h"
>@@ -108,8 +108,11 @@ class SendSideCongestionControllerTest : public ::testing::Test {
>     controller_->AddPacket(ssrc, packet_feedback.sequence_number,
>                            packet_feedback.payload_size,
>                            packet_feedback.pacing_info);
>+    rtc::PacketInfo packet_info;
>+    packet_info.included_in_feedback = true;
>     controller_->OnSentPacket(rtc::SentPacket(packet_feedback.sequence_number,
>-                                              packet_feedback.send_time_ms));
>+                                              packet_feedback.send_time_ms,
>+                                              packet_info));
>   }
> 
>   // Allows us to track the target bitrate, without prescribing the exact
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.cc
>index 73682cd5b2426545caeb513468b4de3a9908c525..110dac399321248869e1a9f81502a6fd4132e27e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.cc
>@@ -15,6 +15,7 @@
> 
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
>@@ -41,8 +42,19 @@ void SendTimeHistory::AddAndRemoveOld(const PacketFeedback& packet) {
>   PacketFeedback packet_copy = packet;
>   packet_copy.long_sequence_number = unwrapped_seq_num;
>   history_.insert(std::make_pair(unwrapped_seq_num, packet_copy));
>-  if (packet.send_time_ms >= 0)
>+  if (packet.send_time_ms >= 0) {
>     AddPacketBytes(packet_copy);
>+    last_send_time_ms_ = std::max(last_send_time_ms_, packet.send_time_ms);
>+  }
>+}
>+
>+void SendTimeHistory::AddUntracked(size_t packet_size, int64_t send_time_ms) {
>+  if (send_time_ms < last_send_time_ms_) {
>+    RTC_LOG(LS_WARNING) << "ignoring untracked data for out of order packet.";
>+  }
>+  pending_untracked_size_ += packet_size;
>+  last_untracked_send_time_ms_ =
>+      std::max(last_untracked_send_time_ms_, send_time_ms);
> }
> 
> bool SendTimeHistory::OnSentPacket(uint16_t sequence_number,
>@@ -53,8 +65,17 @@ bool SendTimeHistory::OnSentPacket(uint16_t sequence_number,
>     return false;
>   bool packet_retransmit = it->second.send_time_ms >= 0;
>   it->second.send_time_ms = send_time_ms;
>+  last_send_time_ms_ = std::max(last_send_time_ms_, send_time_ms);
>   if (!packet_retransmit)
>     AddPacketBytes(it->second);
>+  if (pending_untracked_size_ > 0) {
>+    if (send_time_ms < last_untracked_send_time_ms_)
>+      RTC_LOG(LS_WARNING)
>+          << "appending acknowledged data for out of order packet. (Diff: "
>+          << last_untracked_send_time_ms_ - send_time_ms << " ms.)";
>+    it->second.unacknowledged_data += pending_untracked_size_;
>+    pending_untracked_size_ = 0;
>+  }
>   return true;
> }
> 
>@@ -90,13 +111,13 @@ bool SendTimeHistory::GetFeedback(PacketFeedback* packet_feedback,
>   return true;
> }
> 
>-size_t SendTimeHistory::GetOutstandingBytes(uint16_t local_net_id,
>-                                            uint16_t remote_net_id) const {
>+DataSize SendTimeHistory::GetOutstandingData(uint16_t local_net_id,
>+                                             uint16_t remote_net_id) const {
>   auto it = in_flight_bytes_.find({local_net_id, remote_net_id});
>   if (it != in_flight_bytes_.end()) {
>-    return it->second;
>+    return DataSize::bytes(it->second);
>   } else {
>-    return 0;
>+    return DataSize::Zero();
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.h
>index 7cc6ad76414880491178e55d6b664b1f1d817894..656b7e16a34b1dfdab877287d21179ffd1fc2b5f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.h
>@@ -14,6 +14,7 @@
> #include <map>
> #include <utility>
> 
>+#include "api/units/data_size.h"
> #include "modules/include/module_common_types.h"
> #include "rtc_base/constructormagic.h"
> 
>@@ -29,6 +30,8 @@ class SendTimeHistory {
>   // Cleanup old entries, then add new packet info with provided parameters.
>   void AddAndRemoveOld(const PacketFeedback& packet);
> 
>+  void AddUntracked(size_t packet_size, int64_t send_time_ms);
>+
>   // Updates packet info identified by |sequence_number| with |send_time_ms|.
>   // Return false if not found.
>   bool OnSentPacket(uint16_t sequence_number, int64_t send_time_ms);
>@@ -41,8 +44,8 @@ class SendTimeHistory {
>   // thus be non-null and have the sequence_number field set.
>   bool GetFeedback(PacketFeedback* packet_feedback, bool remove);
> 
>-  size_t GetOutstandingBytes(uint16_t local_net_id,
>-                             uint16_t remote_net_id) const;
>+  DataSize GetOutstandingData(uint16_t local_net_id,
>+                              uint16_t remote_net_id) const;
> 
>  private:
>   using RemoteAndLocalNetworkId = std::pair<uint16_t, uint16_t>;
>@@ -52,6 +55,9 @@ class SendTimeHistory {
>   void UpdateAckedSeqNum(int64_t acked_seq_num);
>   const Clock* const clock_;
>   const int64_t packet_age_limit_ms_;
>+  size_t pending_untracked_size_ = 0;
>+  int64_t last_send_time_ms_ = -1;
>+  int64_t last_untracked_send_time_ms_ = -1;
>   SequenceNumberUnwrapper seq_num_unwrapper_;
>   std::map<int64_t, PacketFeedback> history_;
>   absl::optional<int64_t> last_ack_seq_num_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
>index d1b4855e369d4d026a3c9d53f33ae3210745b784..73d95231843b74db4e65379e255e0bb6f241697e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
>@@ -19,8 +19,28 @@
> #include "rtc_base/numerics/mod_ops.h"
> 
> namespace webrtc {
>-namespace webrtc_cc {
>+namespace {
>+void SortPacketFeedbackVector(std::vector<webrtc::PacketFeedback>* input) {
>+  std::sort(input->begin(), input->end(), PacketFeedbackComparator());
>+}
> 
>+PacketResult NetworkPacketFeedbackFromRtpPacketFeedback(
>+    const webrtc::PacketFeedback& pf) {
>+  PacketResult feedback;
>+  if (pf.arrival_time_ms == webrtc::PacketFeedback::kNotReceived) {
>+    feedback.receive_time = Timestamp::PlusInfinity();
>+  } else {
>+    feedback.receive_time = Timestamp::ms(pf.arrival_time_ms);
>+  }
>+  feedback.sent_packet.sequence_number = pf.long_sequence_number;
>+  feedback.sent_packet.send_time = Timestamp::ms(pf.send_time_ms);
>+  feedback.sent_packet.size = DataSize::bytes(pf.payload_size);
>+  feedback.sent_packet.pacing_info = pf.pacing_info;
>+  feedback.sent_packet.prior_unacked_data =
>+      DataSize::bytes(pf.unacknowledged_data);
>+  return feedback;
>+}
>+}  // namespace
> const int64_t kNoTimestamp = -1;
> const int64_t kSendTimeHistoryWindowMs = 60000;
> const int64_t kBaseTimestampScaleFactor =
>@@ -77,16 +97,59 @@ void TransportFeedbackAdapter::AddPacket(uint32_t ssrc,
>   }
> }
> 
>-void TransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
>-                                            int64_t send_time_ms) {
>+absl::optional<SentPacket> TransportFeedbackAdapter::ProcessSentPacket(
>+    const rtc::SentPacket& sent_packet) {
>   rtc::CritScope cs(&lock_);
>-  send_time_history_.OnSentPacket(sequence_number, send_time_ms);
>+  // TODO(srte): Only use one way to indicate that packet feedback is used.
>+  if (sent_packet.info.included_in_feedback || sent_packet.packet_id != -1) {
>+    send_time_history_.OnSentPacket(sent_packet.packet_id,
>+                                    sent_packet.send_time_ms);
>+    absl::optional<PacketFeedback> packet =
>+        send_time_history_.GetPacket(sent_packet.packet_id);
>+    if (packet) {
>+      SentPacket msg;
>+      msg.size = DataSize::bytes(packet->payload_size);
>+      msg.send_time = Timestamp::ms(packet->send_time_ms);
>+      msg.sequence_number = packet->long_sequence_number;
>+      msg.prior_unacked_data = DataSize::bytes(packet->unacknowledged_data);
>+      msg.data_in_flight =
>+          send_time_history_.GetOutstandingData(local_net_id_, remote_net_id_);
>+      return msg;
>+    }
>+  } else if (sent_packet.info.included_in_allocation) {
>+    send_time_history_.AddUntracked(sent_packet.info.packet_size_bytes,
>+                                    sent_packet.send_time_ms);
>+  }
>+  return absl::nullopt;
> }
> 
>-absl::optional<PacketFeedback> TransportFeedbackAdapter::GetPacket(
>-    uint16_t sequence_number) const {
>-  rtc::CritScope cs(&lock_);
>-  return send_time_history_.GetPacket(sequence_number);
>+absl::optional<TransportPacketsFeedback>
>+TransportFeedbackAdapter::ProcessTransportFeedback(
>+    const rtcp::TransportFeedback& feedback) {
>+  int64_t feedback_time_ms = clock_->TimeInMilliseconds();
>+  DataSize prior_in_flight = GetOutstandingData();
>+  OnTransportFeedback(feedback);
>+  std::vector<PacketFeedback> feedback_vector = last_packet_feedback_vector_;
>+  if (feedback_vector.empty())
>+    return absl::nullopt;
>+
>+  SortPacketFeedbackVector(&feedback_vector);
>+  TransportPacketsFeedback msg;
>+  for (const PacketFeedback& rtp_feedback : feedback_vector) {
>+    if (rtp_feedback.send_time_ms != PacketFeedback::kNoSendTime) {
>+      auto feedback = NetworkPacketFeedbackFromRtpPacketFeedback(rtp_feedback);
>+      msg.packet_feedbacks.push_back(feedback);
>+    } else if (rtp_feedback.arrival_time_ms == PacketFeedback::kNotReceived) {
>+      msg.sendless_arrival_times.push_back(Timestamp::PlusInfinity());
>+    } else {
>+      msg.sendless_arrival_times.push_back(
>+          Timestamp::ms(rtp_feedback.arrival_time_ms));
>+    }
>+  }
>+  msg.feedback_time = Timestamp::ms(feedback_time_ms);
>+  msg.prior_in_flight = prior_in_flight;
>+  msg.data_in_flight = GetOutstandingData();
>+  return msg;
> }
> 
> void TransportFeedbackAdapter::SetNetworkIds(uint16_t local_id,
>@@ -96,6 +159,11 @@ void TransportFeedbackAdapter::SetNetworkIds(uint16_t local_id,
>   remote_net_id_ = remote_id;
> }
> 
>+DataSize TransportFeedbackAdapter::GetOutstandingData() const {
>+  rtc::CritScope cs(&lock_);
>+  return send_time_history_.GetOutstandingData(local_net_id_, remote_net_id_);
>+}
>+
> std::vector<PacketFeedback> TransportFeedbackAdapter::GetPacketFeedbackVector(
>     const rtcp::TransportFeedback& feedback) {
>   int64_t timestamp_us = feedback.GetBaseTimeUs();
>@@ -184,10 +252,4 @@ std::vector<PacketFeedback>
> TransportFeedbackAdapter::GetTransportFeedbackVector() const {
>   return last_packet_feedback_vector_;
> }
>-
>-size_t TransportFeedbackAdapter::GetOutstandingBytes() const {
>-  rtc::CritScope cs(&lock_);
>-  return send_time_history_.GetOutstandingBytes(local_net_id_, remote_net_id_);
>-}
>-}  // namespace webrtc_cc
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h
>index e2477ca31f3443faa1d2341b02b290e31a1b308f..1741d6d66a0c413ad05605bff95a0162da041069 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h
>@@ -17,6 +17,7 @@
> #include "api/transport/network_types.h"
> #include "modules/congestion_controller/rtp/send_time_history.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/network/sent_packet.h"
> #include "rtc_base/thread_annotations.h"
> #include "rtc_base/thread_checker.h"
> #include "system_wrappers/include/clock.h"
>@@ -29,7 +30,6 @@ namespace rtcp {
> class TransportFeedback;
> }  // namespace rtcp
> 
>-namespace webrtc_cc {
> class TransportFeedbackAdapter {
>  public:
>   explicit TransportFeedbackAdapter(const Clock* clock);
>@@ -42,22 +42,22 @@ class TransportFeedbackAdapter {
>                  uint16_t sequence_number,
>                  size_t length,
>                  const PacedPacketInfo& pacing_info);
>-  void OnSentPacket(uint16_t sequence_number, int64_t send_time_ms);
> 
>-  // TODO(holmer): This method should return DelayBasedBwe::Result so that we
>-  // can get rid of the dependency on BitrateController. Requires changes
>-  // to the CongestionController interface.
>-  void OnTransportFeedback(const rtcp::TransportFeedback& feedback);
>-  std::vector<PacketFeedback> GetTransportFeedbackVector() const;
>-  absl::optional<PacketFeedback> GetPacket(uint16_t sequence_number) const;
>+  absl::optional<SentPacket> ProcessSentPacket(
>+      const rtc::SentPacket& sent_packet);
> 
>-  void SetTransportOverhead(int transport_overhead_bytes_per_packet);
>+  absl::optional<TransportPacketsFeedback> ProcessTransportFeedback(
>+      const rtcp::TransportFeedback& feedback);
>+
>+  std::vector<PacketFeedback> GetTransportFeedbackVector() const;
> 
>   void SetNetworkIds(uint16_t local_id, uint16_t remote_id);
> 
>-  size_t GetOutstandingBytes() const;
>+  DataSize GetOutstandingData() const;
> 
>  private:
>+  void OnTransportFeedback(const rtcp::TransportFeedback& feedback);
>+
>   std::vector<PacketFeedback> GetPacketFeedbackVector(
>       const rtcp::TransportFeedback& feedback);
> 
>@@ -75,7 +75,6 @@ class TransportFeedbackAdapter {
>       RTC_GUARDED_BY(&observers_lock_);
> };
> 
>-}  // namespace webrtc_cc
> }  // namespace webrtc
> 
> #endif  // MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
>index 70087678a239da39b1716802f2e9353bfc90ff50..d1404df63f418d72137ebd4e6a6cc411795a2f2c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
>@@ -69,8 +69,9 @@ class TransportFeedbackAdapterTest : public ::testing::Test {
>     adapter_->AddPacket(kSsrc, packet_feedback.sequence_number,
>                         packet_feedback.payload_size,
>                         packet_feedback.pacing_info);
>-    adapter_->OnSentPacket(packet_feedback.sequence_number,
>-                           packet_feedback.send_time_ms);
>+    adapter_->ProcessSentPacket(rtc::SentPacket(packet_feedback.sequence_number,
>+                                                packet_feedback.send_time_ms,
>+                                                rtc::PacketInfo()));
>   }
> 
>   static constexpr uint32_t kSsrc = 8492;
>@@ -100,7 +101,7 @@ TEST_F(TransportFeedbackAdapterTest, ObserverSanity) {
>   }
> 
>   EXPECT_CALL(mock, OnPacketFeedbackVector(_)).Times(1);
>-  adapter_->OnTransportFeedback(feedback);
>+  adapter_->ProcessTransportFeedback(feedback);
> 
>   adapter_->DeRegisterPacketFeedbackObserver(&mock);
> 
>@@ -115,7 +116,7 @@ TEST_F(TransportFeedbackAdapterTest, ObserverSanity) {
>   EXPECT_TRUE(feedback.AddReceivedPacket(new_packet.sequence_number,
>                                          new_packet.arrival_time_ms * 1000));
>   EXPECT_CALL(mock, OnPacketFeedbackVector(_)).Times(0);
>-  adapter_->OnTransportFeedback(second_feedback);
>+  adapter_->ProcessTransportFeedback(second_feedback);
> }
> 
> #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>@@ -156,7 +157,7 @@ TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
> 
>   feedback.Build();
> 
>-  adapter_->OnTransportFeedback(feedback);
>+  adapter_->ProcessTransportFeedback(feedback);
>   ComparePacketFeedbackVectors(packets, adapter_->GetTransportFeedbackVector());
> }
> 
>@@ -189,7 +190,7 @@ TEST_F(TransportFeedbackAdapterTest, FeedbackVectorReportsUnreceived) {
> 
>   feedback.Build();
> 
>-  adapter_->OnTransportFeedback(feedback);
>+  adapter_->ProcessTransportFeedback(feedback);
>   ComparePacketFeedbackVectors(sent_packets,
>                                adapter_->GetTransportFeedbackVector());
> }
>@@ -233,7 +234,7 @@ TEST_F(TransportFeedbackAdapterTest, HandlesDroppedPackets) {
>     expected_packets[i].pacing_info = PacedPacketInfo();
>   }
> 
>-  adapter_->OnTransportFeedback(feedback);
>+  adapter_->ProcessTransportFeedback(feedback);
>   ComparePacketFeedbackVectors(expected_packets,
>                                adapter_->GetTransportFeedbackVector());
> }
>@@ -269,7 +270,7 @@ TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
>     std::vector<PacketFeedback> expected_packets;
>     expected_packets.push_back(packets[i]);
> 
>-    adapter_->OnTransportFeedback(*feedback.get());
>+    adapter_->ProcessTransportFeedback(*feedback.get());
>     ComparePacketFeedbackVectors(expected_packets,
>                                  adapter_->GetTransportFeedbackVector());
>   }
>@@ -298,7 +299,7 @@ TEST_F(TransportFeedbackAdapterTest, HandlesArrivalReordering) {
>   // Adapter keeps the packets ordered by sequence number (which is itself
>   // assigned by the order of transmission). Reordering by some other criteria,
>   // eg. arrival time, is up to the observers.
>-  adapter_->OnTransportFeedback(feedback);
>+  adapter_->ProcessTransportFeedback(feedback);
>   ComparePacketFeedbackVectors(packets, adapter_->GetTransportFeedbackVector());
> }
> 
>@@ -362,7 +363,7 @@ TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
>   std::vector<PacketFeedback> received_feedback;
> 
>   EXPECT_TRUE(feedback.get() != nullptr);
>-  adapter_->OnTransportFeedback(*feedback.get());
>+  adapter_->ProcessTransportFeedback(*feedback.get());
>   ComparePacketFeedbackVectors(sent_packets,
>                                adapter_->GetTransportFeedbackVector());
> 
>@@ -377,7 +378,7 @@ TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
>       rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size());
> 
>   EXPECT_TRUE(feedback.get() != nullptr);
>-  adapter_->OnTransportFeedback(*feedback.get());
>+  adapter_->ProcessTransportFeedback(*feedback.get());
>   {
>     std::vector<PacketFeedback> expected_packets;
>     expected_packets.push_back(packet_feedback);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
>index f8b3155bf9dc66d90d4157296edba75cffb2563c..147f103e7869771d26e99f3c5f1af67fd16c17d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
>@@ -18,16 +18,16 @@
> 
> #include "absl/memory/memory.h"
> #include "modules/bitrate_controller/include/bitrate_controller.h"
>-#include "modules/congestion_controller/congestion_window_pushback_controller.h"
> #include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
> #include "modules/congestion_controller/goog_cc/probe_controller.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/format_macros.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/network/sent_packet.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/rate_limiter.h"
>-#include "rtc_base/socket.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/field_trial.h"
> 
>@@ -143,6 +143,7 @@ SendSideCongestionController::SendSideCongestionController(
>       pause_pacer_(false),
>       pacer_paused_(false),
>       min_bitrate_bps_(congestion_controller::GetMinBitrateBps()),
>+      probe_bitrate_estimator_(new ProbeBitrateEstimator(event_log_)),
>       delay_based_bwe_(new DelayBasedBwe(event_log_)),
>       in_cwnd_experiment_(CwndExperimentEnabled()),
>       accepted_queue_ms_(kDefaultAcceptedQueueMs),
>@@ -153,7 +154,7 @@ SendSideCongestionController::SendSideCongestionController(
>       pacer_pushback_experiment_(IsPacerPushbackExperimentEnabled()),
>       congestion_window_pushback_controller_(
>           MaybeCreateCongestionWindowPushbackController()) {
>-  delay_based_bwe_->SetMinBitrate(min_bitrate_bps_);
>+  delay_based_bwe_->SetMinBitrate(DataRate::bps(min_bitrate_bps_));
>   if (in_cwnd_experiment_ &&
>       !ReadCwndExperimentParameter(&accepted_queue_ms_)) {
>     RTC_LOG(LS_WARNING) << "Failed to parse parameters for CwndExperiment "
>@@ -164,6 +165,23 @@ SendSideCongestionController::SendSideCongestionController(
> 
> SendSideCongestionController::~SendSideCongestionController() {}
> 
>+void SendSideCongestionController::EnableCongestionWindowPushback(
>+    int64_t accepted_queue_ms,
>+    uint32_t min_pushback_target_bitrate_bps) {
>+  RTC_DCHECK(!congestion_window_pushback_controller_)
>+      << "The congestion pushback is already enabled.";
>+  RTC_CHECK_GE(accepted_queue_ms, 0)
>+      << "Accepted must be greater than or equal to 0.";
>+  RTC_CHECK_GE(min_pushback_target_bitrate_bps, 0)
>+      << "Min pushback target bitrate must be greater than or equal to 0.";
>+
>+  in_cwnd_experiment_ = true;
>+  accepted_queue_ms_ = accepted_queue_ms;
>+  congestion_window_pushback_controller_ =
>+      absl::make_unique<CongestionWindowPushbackController>(
>+          min_pushback_target_bitrate_bps);
>+}
>+
> void SendSideCongestionController::RegisterPacketFeedbackObserver(
>     PacketFeedbackObserver* observer) {
>   transport_feedback_adapter_.RegisterPacketFeedbackObserver(observer);
>@@ -204,9 +222,9 @@ void SendSideCongestionController::SetBweBitrates(int min_bitrate_bps,
>   {
>     rtc::CritScope cs(&bwe_lock_);
>     if (start_bitrate_bps > 0)
>-      delay_based_bwe_->SetStartBitrate(start_bitrate_bps);
>+      delay_based_bwe_->SetStartBitrate(DataRate::bps(start_bitrate_bps));
>     min_bitrate_bps_ = min_bitrate_bps;
>-    delay_based_bwe_->SetMinBitrate(min_bitrate_bps_);
>+    delay_based_bwe_->SetMinBitrate(DataRate::bps(min_bitrate_bps_));
>   }
>   MaybeTriggerOnNetworkChanged();
> }
>@@ -241,10 +259,13 @@ void SendSideCongestionController::OnNetworkRouteChanged(
>     rtc::CritScope cs(&bwe_lock_);
>     transport_overhead_bytes_per_packet_ = network_route.packet_overhead;
>     min_bitrate_bps_ = min_bitrate_bps;
>+    probe_bitrate_estimator_.reset(new ProbeBitrateEstimator(event_log_));
>     delay_based_bwe_.reset(new DelayBasedBwe(event_log_));
>     acknowledged_bitrate_estimator_.reset(new AcknowledgedBitrateEstimator());
>-    delay_based_bwe_->SetStartBitrate(bitrate_bps);
>-    delay_based_bwe_->SetMinBitrate(min_bitrate_bps);
>+    if (bitrate_bps > 0) {
>+      delay_based_bwe_->SetStartBitrate(DataRate::bps(bitrate_bps));
>+    }
>+    delay_based_bwe_->SetMinBitrate(DataRate::bps(min_bitrate_bps));
>   }
>   {
>     rtc::CritScope cs(&probe_lock_);
>@@ -321,7 +342,7 @@ void SendSideCongestionController::OnSentPacket(
> void SendSideCongestionController::OnRttUpdate(int64_t avg_rtt_ms,
>                                                int64_t max_rtt_ms) {
>   rtc::CritScope cs(&bwe_lock_);
>-  delay_based_bwe_->OnRttUpdate(avg_rtt_ms);
>+  delay_based_bwe_->OnRttUpdate(TimeDelta::ms(avg_rtt_ms));
> }
> 
> int64_t SendSideCongestionController::TimeUntilNextProcess() {
>@@ -397,9 +418,16 @@ void SendSideCongestionController::OnTransportFeedback(
>   DelayBasedBwe::Result result;
>   {
>     rtc::CritScope cs(&bwe_lock_);
>+    for (const auto& packet : feedback_vector) {
>+      if (packet.send_time_ms != PacketFeedback::kNoSendTime &&
>+          packet.pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe) {
>+        probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
>+      }
>+    }
>     result = delay_based_bwe_->IncomingPacketFeedbackVector(
>-        feedback_vector, acknowledged_bitrate_estimator_->bitrate_bps(),
>-        clock_->TimeInMilliseconds());
>+        feedback_vector, acknowledged_bitrate_estimator_->bitrate(),
>+        probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
>+        Timestamp::ms(clock_->TimeInMilliseconds()));
>   }
>   if (result.updated) {
>     bitrate_controller_->OnDelayBasedBweResult(result);
>@@ -501,7 +529,7 @@ void SendSideCongestionController::MaybeTriggerOnNetworkChanged() {
>     int64_t probing_interval_ms;
>     {
>       rtc::CritScope cs(&bwe_lock_);
>-      probing_interval_ms = delay_based_bwe_->GetExpectedBwePeriodMs();
>+      probing_interval_ms = delay_based_bwe_->GetExpectedBwePeriod().ms();
>     }
>     {
>       rtc::CritScope cs(&observer_lock_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc
>index 3653b8c22f6cc7a7ac572045b7013baa6191dc0c..5f58f21cdf8946c4ac8fa885454606c4b7563ef9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc
>@@ -18,7 +18,7 @@
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
>-#include "rtc_base/socket.h"
>+#include "rtc_base/network/sent_packet.h"
> #include "system_wrappers/include/clock.h"
> #include "test/field_trial.h"
> #include "test/gmock.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
>index 3baa3ad5a53c9ae2d58956ca24a842231784e842..b8b620d008b302f3a51873a49bf1efddae572aa6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
>@@ -26,7 +26,8 @@ const int64_t kBaseTimestampScaleFactor =
>     rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
> const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
> 
>-TransportFeedbackAdapter::TransportFeedbackAdapter(const Clock* clock)
>+LegacyTransportFeedbackAdapter::LegacyTransportFeedbackAdapter(
>+    const Clock* clock)
>     : send_time_history_(clock, kSendTimeHistoryWindowMs),
>       clock_(clock),
>       current_offset_ms_(kNoTimestamp),
>@@ -34,11 +35,11 @@ TransportFeedbackAdapter::TransportFeedbackAdapter(const Clock* clock)
>       local_net_id_(0),
>       remote_net_id_(0) {}
> 
>-TransportFeedbackAdapter::~TransportFeedbackAdapter() {
>+LegacyTransportFeedbackAdapter::~LegacyTransportFeedbackAdapter() {
>   RTC_DCHECK(observers_.empty());
> }
> 
>-void TransportFeedbackAdapter::RegisterPacketFeedbackObserver(
>+void LegacyTransportFeedbackAdapter::RegisterPacketFeedbackObserver(
>     PacketFeedbackObserver* observer) {
>   rtc::CritScope cs(&observers_lock_);
>   RTC_DCHECK(observer);
>@@ -47,7 +48,7 @@ void TransportFeedbackAdapter::RegisterPacketFeedbackObserver(
>   observers_.push_back(observer);
> }
> 
>-void TransportFeedbackAdapter::DeRegisterPacketFeedbackObserver(
>+void LegacyTransportFeedbackAdapter::DeRegisterPacketFeedbackObserver(
>     PacketFeedbackObserver* observer) {
>   rtc::CritScope cs(&observers_lock_);
>   RTC_DCHECK(observer);
>@@ -56,10 +57,11 @@ void TransportFeedbackAdapter::DeRegisterPacketFeedbackObserver(
>   observers_.erase(it);
> }
> 
>-void TransportFeedbackAdapter::AddPacket(uint32_t ssrc,
>-                                         uint16_t sequence_number,
>-                                         size_t length,
>-                                         const PacedPacketInfo& pacing_info) {
>+void LegacyTransportFeedbackAdapter::AddPacket(
>+    uint32_t ssrc,
>+    uint16_t sequence_number,
>+    size_t length,
>+    const PacedPacketInfo& pacing_info) {
>   {
>     rtc::CritScope cs(&lock_);
>     const int64_t creation_time_ms = clock_->TimeInMilliseconds();
>@@ -76,20 +78,21 @@ void TransportFeedbackAdapter::AddPacket(uint32_t ssrc,
>   }
> }
> 
>-void TransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
>-                                            int64_t send_time_ms) {
>+void LegacyTransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
>+                                                  int64_t send_time_ms) {
>   rtc::CritScope cs(&lock_);
>   send_time_history_.OnSentPacket(sequence_number, send_time_ms);
> }
> 
>-void TransportFeedbackAdapter::SetNetworkIds(uint16_t local_id,
>-                                             uint16_t remote_id) {
>+void LegacyTransportFeedbackAdapter::SetNetworkIds(uint16_t local_id,
>+                                                   uint16_t remote_id) {
>   rtc::CritScope cs(&lock_);
>   local_net_id_ = local_id;
>   remote_net_id_ = remote_id;
> }
> 
>-std::vector<PacketFeedback> TransportFeedbackAdapter::GetPacketFeedbackVector(
>+std::vector<PacketFeedback>
>+LegacyTransportFeedbackAdapter::GetPacketFeedbackVector(
>     const rtcp::TransportFeedback& feedback) {
>   int64_t timestamp_us = feedback.GetBaseTimeUs();
>   int64_t now_ms = clock_->TimeInMilliseconds();
>@@ -177,7 +180,7 @@ std::vector<PacketFeedback> TransportFeedbackAdapter::GetPacketFeedbackVector(
>   return packet_feedback_vector;
> }
> 
>-void TransportFeedbackAdapter::OnTransportFeedback(
>+void LegacyTransportFeedbackAdapter::OnTransportFeedback(
>     const rtcp::TransportFeedback& feedback) {
>   last_packet_feedback_vector_ = GetPacketFeedbackVector(feedback);
>   {
>@@ -189,18 +192,19 @@ void TransportFeedbackAdapter::OnTransportFeedback(
> }
> 
> std::vector<PacketFeedback>
>-TransportFeedbackAdapter::GetTransportFeedbackVector() const {
>+LegacyTransportFeedbackAdapter::GetTransportFeedbackVector() const {
>   return last_packet_feedback_vector_;
> }
> 
>-absl::optional<int64_t> TransportFeedbackAdapter::GetMinFeedbackLoopRtt()
>+absl::optional<int64_t> LegacyTransportFeedbackAdapter::GetMinFeedbackLoopRtt()
>     const {
>   rtc::CritScope cs(&lock_);
>   return min_feedback_rtt_;
> }
> 
>-size_t TransportFeedbackAdapter::GetOutstandingBytes() const {
>+size_t LegacyTransportFeedbackAdapter::GetOutstandingBytes() const {
>   rtc::CritScope cs(&lock_);
>-  return send_time_history_.GetOutstandingBytes(local_net_id_, remote_net_id_);
>+  return send_time_history_.GetOutstandingData(local_net_id_, remote_net_id_)
>+      .bytes();
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.h
>index 206236ce8ba7d2d92d16960c7c761ce406a4d2a7..c6ac2ad9ad4bae9b2d254832e96535a07f60dfbc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.h
>@@ -29,10 +29,12 @@ namespace rtcp {
> class TransportFeedback;
> }  // namespace rtcp
> 
>-class TransportFeedbackAdapter {
>+// Deprecated, use version in
>+// modules/congeestion_controller/rtp/transport_feedback_adapter.h
>+class LegacyTransportFeedbackAdapter {
>  public:
>-  explicit TransportFeedbackAdapter(const Clock* clock);
>-  virtual ~TransportFeedbackAdapter();
>+  explicit LegacyTransportFeedbackAdapter(const Clock* clock);
>+  virtual ~LegacyTransportFeedbackAdapter();
> 
>   void RegisterPacketFeedbackObserver(PacketFeedbackObserver* observer);
>   void DeRegisterPacketFeedbackObserver(PacketFeedbackObserver* observer);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc
>index 39aa72e8167e7ab8c744c4752e2e47535fea6cb9..dc4eab13d651080c2c8bd2d25b1c59e182e8e60e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc
>@@ -52,7 +52,7 @@ class LegacyTransportFeedbackAdapterTest : public ::testing::Test {
>   virtual ~LegacyTransportFeedbackAdapterTest() {}
> 
>   virtual void SetUp() {
>-    adapter_.reset(new TransportFeedbackAdapter(&clock_));
>+    adapter_.reset(new LegacyTransportFeedbackAdapter(&clock_));
>   }
> 
>   virtual void TearDown() { adapter_.reset(); }
>@@ -75,7 +75,7 @@ class LegacyTransportFeedbackAdapterTest : public ::testing::Test {
>   static constexpr uint32_t kSsrc = 8492;
> 
>   SimulatedClock clock_;
>-  std::unique_ptr<TransportFeedbackAdapter> adapter_;
>+  std::unique_ptr<LegacyTransportFeedbackAdapter> adapter_;
> };
> 
> TEST_F(LegacyTransportFeedbackAdapterTest, ObserverSanity) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/BUILD.gn
>index 74de1747afd3c04b871597dc83b1ba20ee302866..390fc0a051b264058cd0c6c6febd9c7207a01544 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/BUILD.gn
>@@ -6,6 +6,7 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>+import("//build/config/linux/pkg_config.gni")
> import("//build/config/ui.gni")
> import("../../webrtc.gni")
> 
>@@ -28,8 +29,8 @@ rtc_static_library("primitives") {
>   ]
> 
>   deps = [
>-    "../..:webrtc_common",
>     "../../rtc_base:checks",
>+    "../../rtc_base/system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
> 
>@@ -95,7 +96,6 @@ if (rtc_include_tests) {
>       ":desktop_capture",
>       ":desktop_capture_mock",
>       ":primitives",
>-      "../..:webrtc_common",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>       "../../system_wrappers:cpu_features_api",
>@@ -164,6 +164,23 @@ if (rtc_include_tests) {
>   }
> }
> 
>+if (is_linux) {
>+  if (rtc_use_pipewire) {
>+    pkg_config("pipewire") {
>+      packages = [ "libpipewire-0.2" ]
>+
>+      defines = [ "WEBRTC_USE_PIPEWIRE" ]
>+    }
>+
>+    pkg_config("gio") {
>+      packages = [
>+        "gio-2.0",
>+        "gio-unix-2.0",
>+      ]
>+    }
>+  }
>+}
>+
> rtc_source_set("desktop_capture") {
>   visibility = [ "*" ]
>   public_deps = [
>@@ -321,39 +338,69 @@ rtc_static_library("desktop_capture_generic") {
>     ]
>   }
> 
>+  if (rtc_use_x11 || rtc_use_pipewire) {
>+    sources += [
>+      "mouse_cursor_monitor_linux.cc",
>+      "screen_capturer_linux.cc",
>+      "window_capturer_linux.cc",
>+    ]
>+
>+    if (build_with_mozilla) {
>+      sources += [ "app_capturer_linux.cc" ]
>+    }
>+  }
>+
>   if (rtc_use_x11) {
>     sources += [
>-      "mouse_cursor_monitor_x11.cc",
>-      "screen_capturer_x11.cc",
>-      "window_capturer_x11.cc",
>-      "window_finder_x11.cc",
>-      "window_finder_x11.h",
>-      "x11/shared_x_display.cc",
>-      "x11/shared_x_display.h",
>-      "x11/window_list_utils.cc",
>-      "x11/window_list_utils.h",
>-      "x11/x_atom_cache.cc",
>-      "x11/x_atom_cache.h",
>-      "x11/x_error_trap.cc",
>-      "x11/x_error_trap.h",
>-      "x11/x_server_pixel_buffer.cc",
>-      "x11/x_server_pixel_buffer.h",
>+      "linux/mouse_cursor_monitor_x11.cc",
>+      "linux/mouse_cursor_monitor_x11.h",
>+      "linux/screen_capturer_x11.cc",
>+      "linux/screen_capturer_x11.h",
>+      "linux/shared_x_display.cc",
>+      "linux/shared_x_display.h",
>+      "linux/window_capturer_x11.cc",
>+      "linux/window_capturer_x11.h",
>+      "linux/window_finder_x11.cc",
>+      "linux/window_finder_x11.h",
>+      "linux/window_list_utils.cc",
>+      "linux/window_list_utils.h",
>+      "linux/x_atom_cache.cc",
>+      "linux/x_atom_cache.h",
>+      "linux/x_error_trap.cc",
>+      "linux/x_error_trap.h",
>+      "linux/x_server_pixel_buffer.cc",
>+      "linux/x_server_pixel_buffer.h",
>     ]
>     configs += [ "//build/config/linux:x11" ]
> 
>     if (build_with_mozilla) {
>       sources += [
>-        "app_capturer_x11.cc",
>-        "app_capturer_x11.h",
>-        "x11/desktop_device_info_x11.cc",
>-        "x11/desktop_device_info_x11.h",
>-        "x11/shared_x_util.cc",
>-        "x11/shared_x_util.h",
>+        "linux/app_capturer_x11.cc",
>+        "linux/desktop_device_info_linux.cc",
>+        "linux/desktop_device_info_linux.h",
>+        "linux/shared_x_util.cc",
>+        "linux/shared_x_util.h",
>       ]
>     }
>   }
> 
>-  if (!is_win && !is_mac && !rtc_use_x11) {
>+  if (rtc_use_pipewire) {
>+    sources += [
>+      "linux/base_capturer_pipewire.cc",
>+      "linux/base_capturer_pipewire.h",
>+      "linux/screen_capturer_pipewire.cc",
>+      "linux/screen_capturer_pipewire.h",
>+      "linux/window_capturer_pipewire.cc",
>+      "linux/window_capturer_pipewire.h",
>+    ]
>+
>+    configs += [
>+      ":gio",
>+      ":pipewire",
>+    ]
>+  }
>+
>+  if (!is_win && !is_mac && !rtc_use_x11 && !rtc_use_pipewire) {
>     sources += [
>       "mouse_cursor_monitor_null.cc",
>       "screen_capturer_null.cc",
>@@ -370,12 +417,12 @@ rtc_static_library("desktop_capture_generic") {
> 
>   deps = [
>     ":primitives",
>-    "../..:webrtc_common",
>     "../../api:refcountedbase",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base",  # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
>     "../../rtc_base/synchronization:rw_lock_wrapper",
>     "../../rtc_base/system:arch",
>+    "../../rtc_base/system:rtc_export",
>     "../../system_wrappers",
>     "../../system_wrappers:cpu_features_api",
>     "../../system_wrappers:metrics",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame.h
>index 59dced40179bdd93d81c3b47ae73a5f7199c6a3c..6782398a6ab1e21d35e115bc4801a6acf32814c8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame.h
>@@ -12,6 +12,7 @@
> #define MODULES_DESKTOP_CAPTURE_CROPPED_DESKTOP_FRAME_H_
> 
> #include "modules/desktop_capture/desktop_frame.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -20,9 +21,9 @@ namespace webrtc {
> // |frame| should not be nullptr. |rect| is in |frame| coordinate, i.e.
> // |frame|->top_left() does not impact the area of |rect|.
> // Returns nullptr frame if |rect| is not contained by the bounds of |frame|.
>-std::unique_ptr<DesktopFrame> CreateCroppedDesktopFrame(
>-    std::unique_ptr<DesktopFrame> frame,
>-    const DesktopRect& rect);
>+std::unique_ptr<DesktopFrame> RTC_EXPORT
>+CreateCroppedDesktopFrame(std::unique_ptr<DesktopFrame> frame,
>+                          const DesktopRect& rect);
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer.h
>index c889801da3c5bbfeb5c849cb76e0fc5c86f4b70f..224198fa615110b14216dc50e3fcc03181af7ffd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer.h
>@@ -15,14 +15,15 @@
> 
> #include "modules/desktop_capture/desktop_capture_options.h"
> #include "modules/desktop_capture/desktop_capturer.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // WindowCapturer implementation that uses a screen capturer to capture the
> // whole screen and crops the video frame to the window area when the captured
> // window is on top.
>-class CroppingWindowCapturer : public DesktopCapturer,
>-                               public DesktopCapturer::Callback {
>+class RTC_EXPORT CroppingWindowCapturer : public DesktopCapturer,
>+                                          public DesktopCapturer::Callback {
>  public:
>   static std::unique_ptr<DesktopCapturer> CreateCapturer(
>       const DesktopCaptureOptions& options);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
>index 7dff7101f35d833c69331f66b7b7a6fb3c330089..fa5d15c92f6018f8ffbaf0c44ecdfc433a94df99 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
>@@ -17,14 +17,16 @@
> #include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/mouse_cursor_monitor.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // A wrapper for DesktopCapturer that also captures mouse using specified
> // MouseCursorMonitor and renders it on the generated streams.
>-class DesktopAndCursorComposer : public DesktopCapturer,
>-                                 public DesktopCapturer::Callback,
>-                                 public MouseCursorMonitor::Callback {
>+class RTC_EXPORT DesktopAndCursorComposer
>+    : public DesktopCapturer,
>+      public DesktopCapturer::Callback,
>+      public MouseCursorMonitor::Callback {
>  public:
>   // Creates a new blender that captures mouse cursor using
>   // MouseCursorMonitor::Create(options) and renders it into the frames
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_options.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_options.h
>index 60aa05c3fd3329d2747812f0f4838514e79d3f82..e9587a3f50c31b08535d8d7a05bfd469ff377671 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_options.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_options.h
>@@ -12,9 +12,10 @@
> 
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> #if defined(USE_X11)
>-#include "modules/desktop_capture/x11/shared_x_display.h"
>+#include "modules/desktop_capture/linux/shared_x_display.h"
> #endif
> 
> #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
>@@ -26,7 +27,7 @@ namespace webrtc {
> 
> // An object that stores initialization parameters for screen and window
> // capturers.
>-class DesktopCaptureOptions {
>+class RTC_EXPORT DesktopCaptureOptions {
>  public:
>   // Returns instance of DesktopCaptureOptions with default parameters. On Linux
>   // also initializes X window connection. x_display() will be set to null if
>@@ -114,6 +115,11 @@ class DesktopCaptureOptions {
>   }
> #endif
> 
>+#if defined(WEBRTC_USE_PIPEWIRE)
>+  bool allow_pipewire() const { return allow_pipewire_; }
>+  void set_allow_pipewire(bool allow) { allow_pipewire_ = allow; }
>+#endif
>+
>  private:
> #if defined(USE_X11)
>   rtc::scoped_refptr<SharedXDisplay> x_display_;
>@@ -137,6 +143,9 @@ class DesktopCaptureOptions {
> #endif
>   bool disable_effects_ = true;
>   bool detect_updated_region_ = false;
>+#if defined(WEBRTC_USE_PIPEWIRE)
>+  bool allow_pipewire_ = false;
>+#endif
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.cc
>index 166d97cc52af69977e3da1eb9b1b94c570b8105e..190f61e81755a326ffa046aeef3a03bc2792c460 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.cc
>@@ -8,6 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <cstring>
>+
> #include "modules/desktop_capture/desktop_capturer.h"
> 
> #include "modules/desktop_capture/desktop_capture_options.h"
>@@ -60,4 +62,17 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateScreenCapturer(
>   return capturer;
> }
> 
>+#if defined(WEBRTC_USE_PIPEWIRE) || defined(USE_X11)
>+bool DesktopCapturer::IsRunningUnderWayland() {
>+  const char* xdg_session_type = getenv("XDG_SESSION_TYPE");
>+  if (!xdg_session_type || strncmp(xdg_session_type, "wayland", 7) != 0)
>+    return false;
>+
>+  if (!(getenv("WAYLAND_DISPLAY")))
>+    return false;
>+
>+  return true;
>+}
>+#endif  // defined(WEBRTC_USE_PIPEWIRE) || defined(USE_X11)
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.h
>index 47196cfe491083d666930eb9556b39a1c3d7292f..f9dacc1f7a38477f7e3283eeb62d83a913abf0a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.h
>@@ -22,6 +22,7 @@
> #include "modules/desktop_capture/desktop_capture_types.h"
> #include "modules/desktop_capture/desktop_frame.h"
> #include "modules/desktop_capture/shared_memory.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -29,7 +30,7 @@ class DesktopCaptureOptions;
> class DesktopFrame;
> 
> // Abstract interface for screen and window capturers.
>-class DesktopCapturer {
>+class RTC_EXPORT DesktopCapturer {
>  public:
>   enum class Result {
>     // The frame was captured successfully.
>@@ -134,6 +135,10 @@ class DesktopCapturer {
>   static std::unique_ptr<DesktopCapturer> CreateScreenCapturer(
>       const DesktopCaptureOptions& options);
> 
>+#if defined(WEBRTC_USE_PIPEWIRE) || defined(USE_X11)
>+  static bool IsRunningUnderWayland();
>+#endif  // defined(WEBRTC_USE_PIPEWIRE) || defined(USE_X11)
>+
>  protected:
>   // CroppingWindowCapturer needs to create raw capturers without wrappers, so
>   // the following two functions are protected.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.h
>index 6f7edfcc4f7095e1ef2366e8088e4694b5389218..29b84b718372cf0d2275729484c5961408926438 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.h
>@@ -18,13 +18,14 @@
> #include "modules/desktop_capture/desktop_region.h"
> #include "modules/desktop_capture/shared_memory.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> const int kStandardDPI = 96;
> 
> // DesktopFrame represents a video frame captured from the screen.
>-class DesktopFrame {
>+class RTC_EXPORT DesktopFrame {
>  public:
>   // DesktopFrame objects always hold RGBA data.
>   static const int kBytesPerPixel = 4;
>@@ -135,7 +136,7 @@ class DesktopFrame {
> };
> 
> // A DesktopFrame that stores data in the heap.
>-class BasicDesktopFrame : public DesktopFrame {
>+class RTC_EXPORT BasicDesktopFrame : public DesktopFrame {
>  public:
>   explicit BasicDesktopFrame(DesktopSize size);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.h
>index 09ad55bb4bf156d3ca99f8136cbc5f6686fac398..623a4555f9f48d6c610831d2d0335a179bd6a7ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.h
>@@ -14,6 +14,7 @@
> #include <stdint.h>
> 
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -74,7 +75,7 @@ class DesktopSize {
> };
> 
> // Represents a rectangle on the screen.
>-class DesktopRect {
>+class RTC_EXPORT DesktopRect {
>  public:
>   static DesktopRect MakeSize(const DesktopSize& size) {
>     return DesktopRect(0, 0, size.width(), size.height());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.h
>index f4b9209e2dcaffc5d2127febff276d34037e3390..cbb2d8c48b5303da731a5b7e47e17f00dae77371 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.h
>@@ -16,6 +16,7 @@
> 
> #include "modules/desktop_capture/desktop_geometry.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -23,7 +24,7 @@ namespace webrtc {
> //
> // Internally each region is stored as a set of rows where each row contains one
> // or more rectangles aligned vertically.
>-class DesktopRegion {
>+class RTC_EXPORT DesktopRegion {
>  private:
>   // The following private types need to be declared first because they are used
>   // in the public Iterator.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fake_desktop_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fake_desktop_capturer.h
>index 82c053cbe098b991bba10963bbab13f57f485261..fd867e335da1923a226df62c50b13fc3d5249904 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fake_desktop_capturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fake_desktop_capturer.h
>@@ -17,6 +17,7 @@
> #include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_frame_generator.h"
> #include "modules/desktop_capture/shared_memory.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -31,7 +32,7 @@ namespace webrtc {
> // Double buffering is guaranteed by the FrameGenerator. FrameGenerator
> // implements in desktop_frame_generator.h guarantee double buffering, they
> // creates a new instance of DesktopFrame each time.
>-class FakeDesktopCapturer : public DesktopCapturer {
>+class RTC_EXPORT FakeDesktopCapturer : public DesktopCapturer {
>  public:
>   FakeDesktopCapturer();
>   ~FakeDesktopCapturer() override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/base_capturer_pipewire.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/base_capturer_pipewire.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..62d9994d32a525ee9e0fe59b2ede6ae3cc41310b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/base_capturer_pipewire.cc
>@@ -0,0 +1,843 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/base_capturer_pipewire.h"
>+
>+#include <gio/gunixfdlist.h>
>+#include <glib-object.h>
>+
>+#include <spa/param/format-utils.h>
>+#include <spa/param/props.h>
>+#include <spa/param/video/raw-utils.h>
>+#include <spa/support/type-map.h>
>+
>+#include <memory>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+const char kDesktopBusName[] = "org.freedesktop.portal.Desktop";
>+const char kDesktopObjectPath[] = "/org/freedesktop/portal/desktop";
>+const char kDesktopRequestObjectPath[] =
>+    "/org/freedesktop/portal/desktop/request";
>+const char kSessionInterfaceName[] = "org.freedesktop.portal.Session";
>+const char kRequestInterfaceName[] = "org.freedesktop.portal.Request";
>+const char kScreenCastInterfaceName[] = "org.freedesktop.portal.ScreenCast";
>+
>+const int kBytesPerPixel = 4;
>+
>+// static
>+void BaseCapturerPipeWire::OnStateChanged(void* data,
>+                                          pw_remote_state old_state,
>+                                          pw_remote_state state,
>+                                          const char* error_message) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(data);
>+  RTC_DCHECK(that);
>+
>+  switch (state) {
>+    case PW_REMOTE_STATE_ERROR:
>+      RTC_LOG(LS_ERROR) << "PipeWire remote state error: " << error_message;
>+      break;
>+    case PW_REMOTE_STATE_CONNECTED:
>+      RTC_LOG(LS_INFO) << "PipeWire remote state: connected.";
>+      that->CreateReceivingStream();
>+      break;
>+    case PW_REMOTE_STATE_CONNECTING:
>+      RTC_LOG(LS_INFO) << "PipeWire remote state: connecting.";
>+      break;
>+    case PW_REMOTE_STATE_UNCONNECTED:
>+      RTC_LOG(LS_INFO) << "PipeWire remote state: unconnected.";
>+      break;
>+  }
>+}
>+
>+// static
>+void BaseCapturerPipeWire::OnStreamStateChanged(void* data,
>+                                                pw_stream_state old_state,
>+                                                pw_stream_state state,
>+                                                const char* error_message) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(data);
>+  RTC_DCHECK(that);
>+
>+  switch (state) {
>+    case PW_STREAM_STATE_ERROR:
>+      RTC_LOG(LS_ERROR) << "PipeWire stream state error: " << error_message;
>+      break;
>+    case PW_STREAM_STATE_CONFIGURE:
>+      pw_stream_set_active(that->pw_stream_, true);
>+      break;
>+    case PW_STREAM_STATE_UNCONNECTED:
>+    case PW_STREAM_STATE_CONNECTING:
>+    case PW_STREAM_STATE_READY:
>+    case PW_STREAM_STATE_PAUSED:
>+    case PW_STREAM_STATE_STREAMING:
>+      break;
>+  }
>+}
>+
>+// static
>+void BaseCapturerPipeWire::OnStreamFormatChanged(void* data,
>+                                                 const struct spa_pod* format) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(data);
>+  RTC_DCHECK(that);
>+
>+  RTC_LOG(LS_INFO) << "PipeWire stream format changed.";
>+
>+  if (!format) {
>+    pw_stream_finish_format(that->pw_stream_, /*res=*/0, /*params=*/nullptr,
>+                            /*n_params=*/0);
>+    return;
>+  }
>+
>+  that->spa_video_format_ = new spa_video_info_raw();
>+  spa_format_video_raw_parse(format, that->spa_video_format_,
>+                             &that->pw_type_->format_video);
>+
>+  auto width = that->spa_video_format_->size.width;
>+  auto height = that->spa_video_format_->size.height;
>+  auto stride = SPA_ROUND_UP_N(width * kBytesPerPixel, 4);
>+  auto size = height * stride;
>+
>+  uint8_t buffer[1024] = {};
>+  auto builder = spa_pod_builder{buffer, sizeof(buffer)};
>+
>+  // Setup buffers and meta header for new format.
>+  const struct spa_pod* params[2];
>+  params[0] = reinterpret_cast<spa_pod*>(spa_pod_builder_object(
>+      &builder,
>+      // id to enumerate buffer requirements
>+      that->pw_core_type_->param.idBuffers,
>+      that->pw_core_type_->param_buffers.Buffers,
>+      // Size: specified as integer (i) and set to specified size
>+      ":", that->pw_core_type_->param_buffers.size, "i", size,
>+      // Stride: specified as integer (i) and set to specified stride
>+      ":", that->pw_core_type_->param_buffers.stride, "i", stride,
>+      // Buffers: specifies how many buffers we want to deal with, set as
>+      // integer (i) where preferred number is 8, then allowed number is defined
>+      // as range (r) from min and max values and it is undecided (u) to allow
>+      // negotiation
>+      ":", that->pw_core_type_->param_buffers.buffers, "iru", 8,
>+      SPA_POD_PROP_MIN_MAX(1, 32),
>+      // Align: memory alignment of the buffer, set as integer (i) to specified
>+      // value
>+      ":", that->pw_core_type_->param_buffers.align, "i", 16));
>+  params[1] = reinterpret_cast<spa_pod*>(spa_pod_builder_object(
>+      &builder,
>+      // id to enumerate supported metadata
>+      that->pw_core_type_->param.idMeta, that->pw_core_type_->param_meta.Meta,
>+      // Type: specified as id or enum (I)
>+      ":", that->pw_core_type_->param_meta.type, "I",
>+      that->pw_core_type_->meta.Header,
>+      // Size: size of the metadata, specified as integer (i)
>+      ":", that->pw_core_type_->param_meta.size, "i",
>+      sizeof(struct spa_meta_header)));
>+
>+  pw_stream_finish_format(that->pw_stream_, /*res=*/0, params, /*n_params=*/2);
>+}
>+
>+// static
>+void BaseCapturerPipeWire::OnStreamProcess(void* data) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(data);
>+  RTC_DCHECK(that);
>+
>+  pw_buffer* buf = nullptr;
>+
>+  if (!(buf = pw_stream_dequeue_buffer(that->pw_stream_))) {
>+    return;
>+  }
>+
>+  that->HandleBuffer(buf);
>+
>+  pw_stream_queue_buffer(that->pw_stream_, buf);
>+}
>+
>+BaseCapturerPipeWire::BaseCapturerPipeWire(CaptureSourceType source_type)
>+    : capture_source_type_(source_type) {}
>+
>+BaseCapturerPipeWire::~BaseCapturerPipeWire() {
>+  if (pw_main_loop_) {
>+    pw_thread_loop_stop(pw_main_loop_);
>+  }
>+
>+  if (pw_type_) {
>+    delete pw_type_;
>+  }
>+
>+  if (spa_video_format_) {
>+    delete spa_video_format_;
>+  }
>+
>+  if (pw_stream_) {
>+    pw_stream_destroy(pw_stream_);
>+  }
>+
>+  if (pw_remote_) {
>+    pw_remote_destroy(pw_remote_);
>+  }
>+
>+  if (pw_core_) {
>+    pw_core_destroy(pw_core_);
>+  }
>+
>+  if (pw_main_loop_) {
>+    pw_thread_loop_destroy(pw_main_loop_);
>+  }
>+
>+  if (pw_loop_) {
>+    pw_loop_destroy(pw_loop_);
>+  }
>+
>+  if (current_frame_) {
>+    free(current_frame_);
>+  }
>+
>+  if (start_request_signal_id_) {
>+    g_dbus_connection_signal_unsubscribe(connection_, start_request_signal_id_);
>+  }
>+  if (sources_request_signal_id_) {
>+    g_dbus_connection_signal_unsubscribe(connection_,
>+                                         sources_request_signal_id_);
>+  }
>+  if (session_request_signal_id_) {
>+    g_dbus_connection_signal_unsubscribe(connection_,
>+                                         session_request_signal_id_);
>+  }
>+
>+  if (session_handle_) {
>+    GDBusMessage* message = g_dbus_message_new_method_call(
>+        kDesktopBusName, session_handle_, kSessionInterfaceName, "Close");
>+    if (message) {
>+      GError* error = nullptr;
>+      g_dbus_connection_send_message(connection_, message,
>+                                     G_DBUS_SEND_MESSAGE_FLAGS_NONE,
>+                                     /*out_serial=*/nullptr, &error);
>+      if (error) {
>+        RTC_LOG(LS_ERROR) << "Failed to close the session: " << error->message;
>+        g_error_free(error);
>+      }
>+      g_object_unref(message);
>+    }
>+  }
>+
>+  g_free(start_handle_);
>+  g_free(sources_handle_);
>+  g_free(session_handle_);
>+  g_free(portal_handle_);
>+
>+  if (proxy_) {
>+    g_clear_object(&proxy_);
>+  }
>+}
>+
>+void BaseCapturerPipeWire::InitPortal() {
>+  g_dbus_proxy_new_for_bus(
>+      G_BUS_TYPE_SESSION, G_DBUS_PROXY_FLAGS_NONE, /*info=*/nullptr,
>+      kDesktopBusName, kDesktopObjectPath, kScreenCastInterfaceName,
>+      /*cancellable=*/nullptr,
>+      reinterpret_cast<GAsyncReadyCallback>(OnProxyRequested), this);
>+}
>+
>+void BaseCapturerPipeWire::InitPipeWire() {
>+  pw_init(/*argc=*/nullptr, /*argc=*/nullptr);
>+
>+  pw_loop_ = pw_loop_new(/*properties=*/nullptr);
>+  pw_main_loop_ = pw_thread_loop_new(pw_loop_, "pipewire-main-loop");
>+
>+  pw_core_ = pw_core_new(pw_loop_, /*properties=*/nullptr);
>+  pw_core_type_ = pw_core_get_type(pw_core_);
>+  pw_remote_ = pw_remote_new(pw_core_, nullptr, /*user_data_size=*/0);
>+
>+  InitPipeWireTypes();
>+
>+  // Initialize event handlers, remote end and stream-related.
>+  pw_remote_events_.version = PW_VERSION_REMOTE_EVENTS;
>+  pw_remote_events_.state_changed = &OnStateChanged;
>+
>+  pw_stream_events_.version = PW_VERSION_STREAM_EVENTS;
>+  pw_stream_events_.state_changed = &OnStreamStateChanged;
>+  pw_stream_events_.format_changed = &OnStreamFormatChanged;
>+  pw_stream_events_.process = &OnStreamProcess;
>+
>+  pw_remote_add_listener(pw_remote_, &spa_remote_listener_, &pw_remote_events_,
>+                         this);
>+  pw_remote_connect_fd(pw_remote_, pw_fd_);
>+
>+  if (pw_thread_loop_start(pw_main_loop_) < 0) {
>+    RTC_LOG(LS_ERROR) << "Failed to start main PipeWire loop";
>+    portal_init_failed_ = true;
>+  }
>+}
>+
>+void BaseCapturerPipeWire::InitPipeWireTypes() {
>+  spa_type_map* map = pw_core_type_->map;
>+  pw_type_ = new PipeWireType();
>+
>+  spa_type_media_type_map(map, &pw_type_->media_type);
>+  spa_type_media_subtype_map(map, &pw_type_->media_subtype);
>+  spa_type_format_video_map(map, &pw_type_->format_video);
>+  spa_type_video_format_map(map, &pw_type_->video_format);
>+}
>+
>+void BaseCapturerPipeWire::CreateReceivingStream() {
>+  spa_rectangle pwMinScreenBounds = spa_rectangle{1, 1};
>+  spa_rectangle pwScreenBounds =
>+      spa_rectangle{static_cast<uint32_t>(desktop_size_.width()),
>+                    static_cast<uint32_t>(desktop_size_.height())};
>+
>+  spa_fraction pwFrameRateMin = spa_fraction{0, 1};
>+  spa_fraction pwFrameRateMax = spa_fraction{60, 1};
>+
>+  pw_properties* reuseProps = pw_properties_new("pipewire.client.reuse", "1",
>+                                                /*end of varargs*/ nullptr);
>+  pw_stream_ = pw_stream_new(pw_remote_, "webrtc-consume-stream", reuseProps);
>+
>+  uint8_t buffer[1024] = {};
>+  const spa_pod* params[1];
>+  spa_pod_builder builder = spa_pod_builder{buffer, sizeof(buffer)};
>+  params[0] = reinterpret_cast<spa_pod*>(spa_pod_builder_object(
>+      &builder,
>+      // id to enumerate formats
>+      pw_core_type_->param.idEnumFormat, pw_core_type_->spa_format, "I",
>+      pw_type_->media_type.video, "I", pw_type_->media_subtype.raw,
>+      // Video format: specified as id or enum (I), preferred format is BGRx,
>+      // then allowed formats are enumerated (e) and the format is undecided (u)
>+      // to allow negotiation
>+      ":", pw_type_->format_video.format, "Ieu", pw_type_->video_format.BGRx,
>+      SPA_POD_PROP_ENUM(2, pw_type_->video_format.RGBx,
>+                        pw_type_->video_format.BGRx),
>+      // Video size: specified as rectangle (R), preferred size is specified as
>+      // first parameter, then allowed size is defined as range (r) from min and
>+      // max values and the format is undecided (u) to allow negotiation
>+      ":", pw_type_->format_video.size, "Rru", &pwScreenBounds, 2,
>+      &pwMinScreenBounds, &pwScreenBounds,
>+      // Frame rate: specified as fraction (F) and set to minimum frame rate
>+      // value
>+      ":", pw_type_->format_video.framerate, "F", &pwFrameRateMin,
>+      // Max frame rate: specified as fraction (F), preferred frame rate is set
>+      // to maximum value, then allowed frame rate is defined as range (r) from
>+      // min and max values and it is undecided (u) to allow negotiation
>+      ":", pw_type_->format_video.max_framerate, "Fru", &pwFrameRateMax, 2,
>+      &pwFrameRateMin, &pwFrameRateMax));
>+
>+  pw_stream_add_listener(pw_stream_, &spa_stream_listener_, &pw_stream_events_,
>+                         this);
>+  pw_stream_flags flags = static_cast<pw_stream_flags>(
>+      PW_STREAM_FLAG_AUTOCONNECT | PW_STREAM_FLAG_INACTIVE |
>+      PW_STREAM_FLAG_MAP_BUFFERS);
>+  if (pw_stream_connect(pw_stream_, PW_DIRECTION_INPUT, /*port_path=*/nullptr,
>+                        flags, params,
>+                        /*n_params=*/1) != 0) {
>+    RTC_LOG(LS_ERROR) << "Could not connect receiving stream.";
>+    portal_init_failed_ = true;
>+    return;
>+  }
>+}
>+
>+void BaseCapturerPipeWire::HandleBuffer(pw_buffer* buffer) {
>+  spa_buffer* spaBuffer = buffer->buffer;
>+  void* src = nullptr;
>+
>+  if (!(src = spaBuffer->datas[0].data)) {
>+    return;
>+  }
>+
>+  uint32_t maxSize = spaBuffer->datas[0].maxsize;
>+  int32_t srcStride = spaBuffer->datas[0].chunk->stride;
>+  if (srcStride != (desktop_size_.width() * kBytesPerPixel)) {
>+    RTC_LOG(LS_ERROR) << "Got buffer with stride different from screen stride: "
>+                      << srcStride
>+                      << " != " << (desktop_size_.width() * kBytesPerPixel);
>+    portal_init_failed_ = true;
>+    return;
>+  }
>+
>+  if (!current_frame_) {
>+    current_frame_ = static_cast<uint8_t*>(malloc(maxSize));
>+  }
>+  RTC_DCHECK(current_frame_ != nullptr);
>+
>+  // If both sides decided to go with the RGBx format we need to convert it to
>+  // BGRx to match color format expected by WebRTC.
>+  if (spa_video_format_->format == pw_type_->video_format.RGBx) {
>+    uint8_t* tempFrame = static_cast<uint8_t*>(malloc(maxSize));
>+    std::memcpy(tempFrame, src, maxSize);
>+    ConvertRGBxToBGRx(tempFrame, maxSize);
>+    std::memcpy(current_frame_, tempFrame, maxSize);
>+    free(tempFrame);
>+  } else {
>+    std::memcpy(current_frame_, src, maxSize);
>+  }
>+}
>+
>+void BaseCapturerPipeWire::ConvertRGBxToBGRx(uint8_t* frame, uint32_t size) {
>+  // Change color format for KDE KWin which uses RGBx and not BGRx
>+  for (uint32_t i = 0; i < size; i += 4) {
>+    uint8_t tempR = frame[i];
>+    uint8_t tempB = frame[i + 2];
>+    frame[i] = tempB;
>+    frame[i + 2] = tempR;
>+  }
>+}
>+
>+guint BaseCapturerPipeWire::SetupRequestResponseSignal(
>+    const gchar* object_path,
>+    GDBusSignalCallback callback) {
>+  return g_dbus_connection_signal_subscribe(
>+      connection_, kDesktopBusName, kRequestInterfaceName, "Response",
>+      object_path, /*arg0=*/nullptr, G_DBUS_SIGNAL_FLAGS_NO_MATCH_RULE,
>+      callback, this, /*user_data_free_func=*/nullptr);
>+}
>+
>+// static
>+void BaseCapturerPipeWire::OnProxyRequested(GObject* /*object*/,
>+                                            GAsyncResult* result,
>+                                            gpointer user_data) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(user_data);
>+  RTC_DCHECK(that);
>+
>+  GError* error = nullptr;
>+  that->proxy_ = g_dbus_proxy_new_finish(result, &error);
>+  if (!that->proxy_) {
>+    RTC_LOG(LS_ERROR) << "Failed to create a proxy for the screen cast portal: "
>+                      << error->message;
>+    g_error_free(error);
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+  that->connection_ = g_dbus_proxy_get_connection(that->proxy_);
>+
>+  RTC_LOG(LS_INFO) << "Created proxy for the screen cast portal.";
>+  that->SessionRequest();
>+}
>+
>+// static
>+gchar* BaseCapturerPipeWire::PrepareSignalHandle(GDBusConnection* connection,
>+                                                 const gchar* token) {
>+  gchar* sender = g_strdup(g_dbus_connection_get_unique_name(connection) + 1);
>+  for (int i = 0; sender[i]; i++) {
>+    if (sender[i] == '.') {
>+      sender[i] = '_';
>+    }
>+  }
>+
>+  gchar* handle = g_strconcat(kDesktopRequestObjectPath, "/", sender, "/",
>+                              token, /*end of varargs*/ nullptr);
>+  g_free(sender);
>+
>+  return handle;
>+}
>+
>+void BaseCapturerPipeWire::SessionRequest() {
>+  GVariantBuilder builder;
>+  gchar* variant_string;
>+
>+  g_variant_builder_init(&builder, G_VARIANT_TYPE_VARDICT);
>+  variant_string =
>+      g_strdup_printf("webrtc_session%d", g_random_int_range(0, G_MAXINT));
>+  g_variant_builder_add(&builder, "{sv}", "session_handle_token",
>+                        g_variant_new_string(variant_string));
>+  g_free(variant_string);
>+  variant_string = g_strdup_printf("webrtc%d", g_random_int_range(0, G_MAXINT));
>+  g_variant_builder_add(&builder, "{sv}", "handle_token",
>+                        g_variant_new_string(variant_string));
>+
>+  portal_handle_ = PrepareSignalHandle(connection_, variant_string);
>+  session_request_signal_id_ = SetupRequestResponseSignal(
>+      portal_handle_, OnSessionRequestResponseSignal);
>+  g_free(variant_string);
>+
>+  RTC_LOG(LS_INFO) << "Screen cast session requested.";
>+  g_dbus_proxy_call(
>+      proxy_, "CreateSession", g_variant_new("(a{sv})", &builder),
>+      G_DBUS_CALL_FLAGS_NONE, /*timeout=*/-1, /*cancellable=*/nullptr,
>+      reinterpret_cast<GAsyncReadyCallback>(OnSessionRequested), this);
>+}
>+
>+// static
>+void BaseCapturerPipeWire::OnSessionRequested(GDBusConnection* connection,
>+                                              GAsyncResult* result,
>+                                              gpointer user_data) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(user_data);
>+  RTC_DCHECK(that);
>+
>+  GError* error = nullptr;
>+  GVariant* variant = g_dbus_proxy_call_finish(that->proxy_, result, &error);
>+  if (!variant) {
>+    RTC_LOG(LS_ERROR) << "Failed to create a screen cast session: "
>+                      << error->message;
>+    g_error_free(error);
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+  RTC_LOG(LS_INFO) << "Initializing the screen cast session.";
>+
>+  gchar* handle = nullptr;
>+  g_variant_get_child(variant, 0, "o", &handle);
>+  g_variant_unref(variant);
>+  if (!handle) {
>+    RTC_LOG(LS_ERROR) << "Failed to initialize the screen cast session.";
>+    if (that->session_request_signal_id_) {
>+      g_dbus_connection_signal_unsubscribe(connection,
>+                                           that->session_request_signal_id_);
>+      that->session_request_signal_id_ = 0;
>+    }
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+
>+  g_free(handle);
>+
>+  RTC_LOG(LS_INFO) << "Subscribing to the screen cast session.";
>+}
>+
>+// static
>+void BaseCapturerPipeWire::OnSessionRequestResponseSignal(
>+    GDBusConnection* connection,
>+    const gchar* sender_name,
>+    const gchar* object_path,
>+    const gchar* interface_name,
>+    const gchar* signal_name,
>+    GVariant* parameters,
>+    gpointer user_data) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(user_data);
>+  RTC_DCHECK(that);
>+
>+  RTC_LOG(LS_INFO)
>+      << "Received response for the screen cast session subscription.";
>+
>+  guint32 portal_response;
>+  GVariant* response_data;
>+  g_variant_get(parameters, "(u@a{sv})", &portal_response, &response_data);
>+  g_variant_lookup(response_data, "session_handle", "s",
>+                   &that->session_handle_);
>+  g_variant_unref(response_data);
>+
>+  if (!that->session_handle_ || portal_response) {
>+    RTC_LOG(LS_ERROR)
>+        << "Failed to request the screen cast session subscription.";
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+
>+  that->SourcesRequest();
>+}
>+
>+void BaseCapturerPipeWire::SourcesRequest() {
>+  GVariantBuilder builder;
>+  gchar* variant_string;
>+
>+  g_variant_builder_init(&builder, G_VARIANT_TYPE_VARDICT);
>+  // We want to record monitor content.
>+  g_variant_builder_add(&builder, "{sv}", "types",
>+                        g_variant_new_uint32(capture_source_type_));
>+  // We don't want to allow selection of multiple sources.
>+  g_variant_builder_add(&builder, "{sv}", "multiple",
>+                        g_variant_new_boolean(false));
>+  variant_string = g_strdup_printf("webrtc%d", g_random_int_range(0, G_MAXINT));
>+  g_variant_builder_add(&builder, "{sv}", "handle_token",
>+                        g_variant_new_string(variant_string));
>+
>+  sources_handle_ = PrepareSignalHandle(connection_, variant_string);
>+  sources_request_signal_id_ = SetupRequestResponseSignal(
>+      sources_handle_, OnSourcesRequestResponseSignal);
>+  g_free(variant_string);
>+
>+  RTC_LOG(LS_INFO) << "Requesting sources from the screen cast session.";
>+  g_dbus_proxy_call(
>+      proxy_, "SelectSources",
>+      g_variant_new("(oa{sv})", session_handle_, &builder),
>+      G_DBUS_CALL_FLAGS_NONE, /*timeout=*/-1, /*cancellable=*/nullptr,
>+      reinterpret_cast<GAsyncReadyCallback>(OnSourcesRequested), this);
>+}
>+
>+// static
>+void BaseCapturerPipeWire::OnSourcesRequested(GDBusConnection* connection,
>+                                              GAsyncResult* result,
>+                                              gpointer user_data) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(user_data);
>+  RTC_DCHECK(that);
>+
>+  GError* error = nullptr;
>+  GVariant* variant = g_dbus_proxy_call_finish(that->proxy_, result, &error);
>+  if (!variant) {
>+    RTC_LOG(LS_ERROR) << "Failed to request the sources: " << error->message;
>+    g_error_free(error);
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+
>+  RTC_LOG(LS_INFO) << "Sources requested from the screen cast session.";
>+
>+  gchar* handle = nullptr;
>+  g_variant_get_child(variant, 0, "o", &handle);
>+  g_variant_unref(variant);
>+  if (!handle) {
>+    RTC_LOG(LS_ERROR) << "Failed to initialize the screen cast session.";
>+    if (that->sources_request_signal_id_) {
>+      g_dbus_connection_signal_unsubscribe(connection,
>+                                           that->sources_request_signal_id_);
>+      that->sources_request_signal_id_ = 0;
>+    }
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+
>+  g_free(handle);
>+
>+  RTC_LOG(LS_INFO) << "Subscribed to sources signal.";
>+}
>+
>+// static
>+void BaseCapturerPipeWire::OnSourcesRequestResponseSignal(
>+    GDBusConnection* connection,
>+    const gchar* sender_name,
>+    const gchar* object_path,
>+    const gchar* interface_name,
>+    const gchar* signal_name,
>+    GVariant* parameters,
>+    gpointer user_data) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(user_data);
>+  RTC_DCHECK(that);
>+
>+  RTC_LOG(LS_INFO) << "Received sources signal from session.";
>+
>+  guint32 portal_response;
>+  g_variant_get(parameters, "(u@a{sv})", &portal_response, nullptr);
>+  if (portal_response) {
>+    RTC_LOG(LS_ERROR)
>+        << "Failed to select sources for the screen cast session.";
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+
>+  that->StartRequest();
>+}
>+
>+void BaseCapturerPipeWire::StartRequest() {
>+  GVariantBuilder builder;
>+  gchar* variant_string;
>+
>+  g_variant_builder_init(&builder, G_VARIANT_TYPE_VARDICT);
>+  variant_string = g_strdup_printf("webrtc%d", g_random_int_range(0, G_MAXINT));
>+  g_variant_builder_add(&builder, "{sv}", "handle_token",
>+                        g_variant_new_string(variant_string));
>+
>+  start_handle_ = PrepareSignalHandle(connection_, variant_string);
>+  start_request_signal_id_ =
>+      SetupRequestResponseSignal(start_handle_, OnStartRequestResponseSignal);
>+  g_free(variant_string);
>+
>+  // "Identifier for the application window", this is Wayland, so not "x11:...".
>+  const gchar parent_window[] = "";
>+
>+  RTC_LOG(LS_INFO) << "Starting the screen cast session.";
>+  g_dbus_proxy_call(
>+      proxy_, "Start",
>+      g_variant_new("(osa{sv})", session_handle_, parent_window, &builder),
>+      G_DBUS_CALL_FLAGS_NONE, /*timeout=*/-1, /*cancellable=*/nullptr,
>+      reinterpret_cast<GAsyncReadyCallback>(OnStartRequested), this);
>+}
>+
>+// static
>+void BaseCapturerPipeWire::OnStartRequested(GDBusConnection* connection,
>+                                            GAsyncResult* result,
>+                                            gpointer user_data) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(user_data);
>+  RTC_DCHECK(that);
>+
>+  GError* error = nullptr;
>+  GVariant* variant = g_dbus_proxy_call_finish(that->proxy_, result, &error);
>+  if (!variant) {
>+    RTC_LOG(LS_ERROR) << "Failed to start the screen cast session: "
>+                      << error->message;
>+    g_error_free(error);
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+
>+  RTC_LOG(LS_INFO) << "Initializing the start of the screen cast session.";
>+
>+  gchar* handle = nullptr;
>+  g_variant_get_child(variant, 0, "o", &handle);
>+  g_variant_unref(variant);
>+  if (!handle) {
>+    RTC_LOG(LS_ERROR)
>+        << "Failed to initialize the start of the screen cast session.";
>+    if (that->start_request_signal_id_) {
>+      g_dbus_connection_signal_unsubscribe(connection,
>+                                           that->start_request_signal_id_);
>+      that->start_request_signal_id_ = 0;
>+    }
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+
>+  g_free(handle);
>+
>+  RTC_LOG(LS_INFO) << "Subscribed to the start signal.";
>+}
>+
>+// static
>+void BaseCapturerPipeWire::OnStartRequestResponseSignal(
>+    GDBusConnection* connection,
>+    const gchar* sender_name,
>+    const gchar* object_path,
>+    const gchar* interface_name,
>+    const gchar* signal_name,
>+    GVariant* parameters,
>+    gpointer user_data) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(user_data);
>+  RTC_DCHECK(that);
>+
>+  RTC_LOG(LS_INFO) << "Start signal received.";
>+  guint32 portal_response;
>+  GVariant* response_data;
>+  GVariantIter* iter = nullptr;
>+  g_variant_get(parameters, "(u@a{sv})", &portal_response, &response_data);
>+  if (portal_response || !response_data) {
>+    RTC_LOG(LS_ERROR) << "Failed to start the screen cast session.";
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+
>+  // Array of PipeWire streams. See
>+  // https://github.com/flatpak/xdg-desktop-portal/blob/master/data/org.freedesktop.portal.ScreenCast.xml
>+  // documentation for <method name="Start">.
>+  if (g_variant_lookup(response_data, "streams", "a(ua{sv})", &iter)) {
>+    GVariant* variant;
>+
>+    while (g_variant_iter_next(iter, "@(ua{sv})", &variant)) {
>+      guint32 stream_id;
>+      gint32 width;
>+      gint32 height;
>+      GVariant* options;
>+
>+      g_variant_get(variant, "(u@a{sv})", &stream_id, &options);
>+      RTC_DCHECK(options != nullptr);
>+
>+      g_variant_lookup(options, "size", "(ii)", &width, &height);
>+
>+      that->desktop_size_.set(width, height);
>+
>+      g_variant_unref(options);
>+      g_variant_unref(variant);
>+    }
>+  }
>+  g_variant_iter_free(iter);
>+  g_variant_unref(response_data);
>+
>+  that->OpenPipeWireRemote();
>+}
>+
>+void BaseCapturerPipeWire::OpenPipeWireRemote() {
>+  GVariantBuilder builder;
>+  g_variant_builder_init(&builder, G_VARIANT_TYPE_VARDICT);
>+
>+  RTC_LOG(LS_INFO) << "Opening the PipeWire remote.";
>+
>+  g_dbus_proxy_call_with_unix_fd_list(
>+      proxy_, "OpenPipeWireRemote",
>+      g_variant_new("(oa{sv})", session_handle_, &builder),
>+      G_DBUS_CALL_FLAGS_NONE, /*timeout=*/-1, /*fd_list=*/nullptr,
>+      /*cancellable=*/nullptr,
>+      reinterpret_cast<GAsyncReadyCallback>(OnOpenPipeWireRemoteRequested),
>+      this);
>+}
>+
>+// static
>+void BaseCapturerPipeWire::OnOpenPipeWireRemoteRequested(
>+    GDBusConnection* connection,
>+    GAsyncResult* result,
>+    gpointer user_data) {
>+  BaseCapturerPipeWire* that = static_cast<BaseCapturerPipeWire*>(user_data);
>+  RTC_DCHECK(that);
>+
>+  GError* error = nullptr;
>+  GUnixFDList* outlist = nullptr;
>+  GVariant* variant = g_dbus_proxy_call_with_unix_fd_list_finish(
>+      that->proxy_, &outlist, result, &error);
>+  if (!variant) {
>+    RTC_LOG(LS_ERROR) << "Failed to open the PipeWire remote: "
>+                      << error->message;
>+    g_error_free(error);
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+
>+  gint32 index;
>+  g_variant_get(variant, "(h)", &index);
>+
>+  if ((that->pw_fd_ = g_unix_fd_list_get(outlist, index, &error)) == -1) {
>+    RTC_LOG(LS_ERROR) << "Failed to get file descriptor from the list: "
>+                      << error->message;
>+    g_error_free(error);
>+    g_variant_unref(variant);
>+    that->portal_init_failed_ = true;
>+    return;
>+  }
>+
>+  g_variant_unref(variant);
>+  g_object_unref(outlist);
>+
>+  that->InitPipeWire();
>+  RTC_LOG(LS_INFO) << "PipeWire remote opened.";
>+}
>+
>+void BaseCapturerPipeWire::Start(Callback* callback) {
>+  RTC_DCHECK(!callback_);
>+  RTC_DCHECK(callback);
>+
>+  InitPortal();
>+
>+  callback_ = callback;
>+}
>+
>+void BaseCapturerPipeWire::CaptureFrame() {
>+  if (portal_init_failed_) {
>+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>+    return;
>+  }
>+
>+  if (!current_frame_) {
>+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
>+    return;
>+  }
>+
>+  std::unique_ptr<DesktopFrame> result(new BasicDesktopFrame(desktop_size_));
>+  result->CopyPixelsFrom(
>+      current_frame_, (desktop_size_.width() * kBytesPerPixel),
>+      DesktopRect::MakeWH(desktop_size_.width(), desktop_size_.height()));
>+  if (!result) {
>+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
>+    return;
>+  }
>+  callback_->OnCaptureResult(Result::SUCCESS, std::move(result));
>+}
>+
>+bool BaseCapturerPipeWire::GetSourceList(SourceList* sources) {
>+  RTC_DCHECK(sources->size() == 0);
>+  // List of available screens is already presented by the xdg-desktop-portal.
>+  // But we have to add an empty source as the code expects it.
>+  sources->push_back({0});
>+  return true;
>+}
>+
>+bool BaseCapturerPipeWire::SelectSource(SourceId id) {
>+  // Screen selection is handled by the xdg-desktop-portal.
>+  return true;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/base_capturer_pipewire.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/base_capturer_pipewire.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..56b101acbaa601b83f81c7b48890c79f3cb66755
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/base_capturer_pipewire.h
>@@ -0,0 +1,167 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_BASE_CAPTURER_PIPEWIRE_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_BASE_CAPTURER_PIPEWIRE_H_
>+
>+#include <gio/gio.h>
>+#define typeof __typeof__
>+#include <pipewire/pipewire.h>
>+#include <spa/param/video/format-utils.h>
>+
>+#include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+class PipeWireType {
>+ public:
>+  spa_type_media_type media_type;
>+  spa_type_media_subtype media_subtype;
>+  spa_type_format_video format_video;
>+  spa_type_video_format video_format;
>+};
>+
>+class BaseCapturerPipeWire : public DesktopCapturer {
>+ public:
>+  enum CaptureSourceType { Screen = 1, Window };
>+
>+  explicit BaseCapturerPipeWire(CaptureSourceType source_type);
>+  ~BaseCapturerPipeWire() override;
>+
>+  // DesktopCapturer interface.
>+  void Start(Callback* delegate) override;
>+  void CaptureFrame() override;
>+  bool GetSourceList(SourceList* sources) override;
>+  bool SelectSource(SourceId id) override;
>+
>+ private:
>+  // PipeWire types -->
>+  pw_core* pw_core_ = nullptr;
>+  pw_type* pw_core_type_ = nullptr;
>+  pw_stream* pw_stream_ = nullptr;
>+  pw_remote* pw_remote_ = nullptr;
>+  pw_loop* pw_loop_ = nullptr;
>+  pw_thread_loop* pw_main_loop_ = nullptr;
>+  PipeWireType* pw_type_ = nullptr;
>+
>+  spa_hook spa_stream_listener_ = {};
>+  spa_hook spa_remote_listener_ = {};
>+
>+  pw_stream_events pw_stream_events_ = {};
>+  pw_remote_events pw_remote_events_ = {};
>+
>+  spa_video_info_raw* spa_video_format_ = nullptr;
>+
>+  gint32 pw_fd_ = -1;
>+
>+  CaptureSourceType capture_source_type_ =
>+      BaseCapturerPipeWire::CaptureSourceType::Screen;
>+
>+  // <-- end of PipeWire types
>+
>+  GDBusConnection* connection_ = nullptr;
>+  GDBusProxy* proxy_ = nullptr;
>+  gchar* portal_handle_ = nullptr;
>+  gchar* session_handle_ = nullptr;
>+  gchar* sources_handle_ = nullptr;
>+  gchar* start_handle_ = nullptr;
>+  guint session_request_signal_id_ = 0;
>+  guint sources_request_signal_id_ = 0;
>+  guint start_request_signal_id_ = 0;
>+
>+  DesktopSize desktop_size_ = {};
>+  DesktopCaptureOptions options_ = {};
>+
>+  uint8_t* current_frame_ = nullptr;
>+  Callback* callback_ = nullptr;
>+
>+  bool portal_init_failed_ = false;
>+
>+  void InitPortal();
>+  void InitPipeWire();
>+  void InitPipeWireTypes();
>+
>+  void CreateReceivingStream();
>+  void HandleBuffer(pw_buffer* buffer);
>+
>+  void ConvertRGBxToBGRx(uint8_t* frame, uint32_t size);
>+
>+  static void OnStateChanged(void* data,
>+                             pw_remote_state old_state,
>+                             pw_remote_state state,
>+                             const char* error);
>+  static void OnStreamStateChanged(void* data,
>+                                   pw_stream_state old_state,
>+                                   pw_stream_state state,
>+                                   const char* error_message);
>+
>+  static void OnStreamFormatChanged(void* data, const struct spa_pod* format);
>+  static void OnStreamProcess(void* data);
>+  static void OnNewBuffer(void* data, uint32_t id);
>+
>+  guint SetupRequestResponseSignal(const gchar* object_path,
>+                                   GDBusSignalCallback callback);
>+
>+  static void OnProxyRequested(GObject* object,
>+                               GAsyncResult* result,
>+                               gpointer user_data);
>+
>+  static gchar* PrepareSignalHandle(GDBusConnection* connection,
>+                                    const gchar* token);
>+
>+  void SessionRequest();
>+  static void OnSessionRequested(GDBusConnection* connection,
>+                                 GAsyncResult* result,
>+                                 gpointer user_data);
>+  static void OnSessionRequestResponseSignal(GDBusConnection* connection,
>+                                             const gchar* sender_name,
>+                                             const gchar* object_path,
>+                                             const gchar* interface_name,
>+                                             const gchar* signal_name,
>+                                             GVariant* parameters,
>+                                             gpointer user_data);
>+
>+  void SourcesRequest();
>+  static void OnSourcesRequested(GDBusConnection* connection,
>+                                 GAsyncResult* result,
>+                                 gpointer user_data);
>+  static void OnSourcesRequestResponseSignal(GDBusConnection* connection,
>+                                             const gchar* sender_name,
>+                                             const gchar* object_path,
>+                                             const gchar* interface_name,
>+                                             const gchar* signal_name,
>+                                             GVariant* parameters,
>+                                             gpointer user_data);
>+
>+  void StartRequest();
>+  static void OnStartRequested(GDBusConnection* connection,
>+                               GAsyncResult* result,
>+                               gpointer user_data);
>+  static void OnStartRequestResponseSignal(GDBusConnection* connection,
>+                                           const gchar* sender_name,
>+                                           const gchar* object_path,
>+                                           const gchar* interface_name,
>+                                           const gchar* signal_name,
>+                                           GVariant* parameters,
>+                                           gpointer user_data);
>+
>+  void OpenPipeWireRemote();
>+  static void OnOpenPipeWireRemoteRequested(GDBusConnection* connection,
>+                                            GAsyncResult* result,
>+                                            gpointer user_data);
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(BaseCapturerPipeWire);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_BASE_CAPTURER_PIPEWIRE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/mouse_cursor_monitor_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/mouse_cursor_monitor_x11.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..f7503822bced8bbdbe47f3cd105abd8f1032f350
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/mouse_cursor_monitor_x11.cc
>@@ -0,0 +1,247 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/mouse_cursor_monitor_x11.h"
>+
>+#include <X11/Xutil.h>
>+#include <X11/extensions/Xfixes.h>
>+
>+#include <algorithm>
>+#include <memory>
>+
>+#include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capture_types.h"
>+#include "modules/desktop_capture/desktop_frame.h"
>+#include "modules/desktop_capture/linux/x_error_trap.h"
>+#include "modules/desktop_capture/mouse_cursor.h"
>+#include "modules/desktop_capture/mouse_cursor_monitor.h"
>+#include "rtc_base/logging.h"
>+
>+namespace {
>+
>+// WindowCapturer returns window IDs of X11 windows with WM_STATE attribute.
>+// These windows may not be immediate children of the root window, because
>+// window managers may re-parent them to add decorations. However,
>+// XQueryPointer() expects to be passed children of the root. This function
>+// searches up the list of the windows to find the root child that corresponds
>+// to |window|.
>+Window GetTopLevelWindow(Display* display, Window window) {
>+  while (true) {
>+    // If the window is in WithdrawnState then look at all of its children.
>+    ::Window root, parent;
>+    ::Window* children;
>+    unsigned int num_children;
>+    if (!XQueryTree(display, window, &root, &parent, &children,
>+                    &num_children)) {
>+      RTC_LOG(LS_ERROR) << "Failed to query for child windows although window"
>+                        << "does not have a valid WM_STATE.";
>+      return None;
>+    }
>+    if (children)
>+      XFree(children);
>+
>+    if (parent == root)
>+      break;
>+
>+    window = parent;
>+  }
>+
>+  return window;
>+}
>+
>+}  // namespace
>+
>+namespace webrtc {
>+
>+MouseCursorMonitorX11::MouseCursorMonitorX11(
>+    const DesktopCaptureOptions& options,
>+    Window window)
>+    : x_display_(options.x_display()),
>+      callback_(NULL),
>+      mode_(SHAPE_AND_POSITION),
>+      window_(window),
>+      have_xfixes_(false),
>+      xfixes_event_base_(-1),
>+      xfixes_error_base_(-1) {
>+  // Set a default initial cursor shape in case XFixes is not present.
>+  const int kSize = 5;
>+  std::unique_ptr<DesktopFrame> default_cursor(
>+      new BasicDesktopFrame(DesktopSize(kSize, kSize)));
>+  const uint8_t pixels[kSize * kSize] = {
>+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0xff,
>+      0x00, 0x00, 0xff, 0xff, 0xff, 0x00, 0x00, 0xff, 0xff,
>+      0xff, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
>+  uint8_t* ptr = default_cursor->data();
>+  for (int y = 0; y < kSize; ++y) {
>+    for (int x = 0; x < kSize; ++x) {
>+      *ptr++ = pixels[kSize * y + x];
>+      *ptr++ = pixels[kSize * y + x];
>+      *ptr++ = pixels[kSize * y + x];
>+      *ptr++ = 0xff;
>+    }
>+  }
>+  DesktopVector hotspot(2, 2);
>+  cursor_shape_.reset(new MouseCursor(default_cursor.release(), hotspot));
>+}
>+
>+MouseCursorMonitorX11::~MouseCursorMonitorX11() {
>+  if (have_xfixes_) {
>+    x_display_->RemoveEventHandler(xfixes_event_base_ + XFixesCursorNotify,
>+                                   this);
>+  }
>+}
>+
>+void MouseCursorMonitorX11::Init(Callback* callback, Mode mode) {
>+  // Init can be called only once per instance of MouseCursorMonitor.
>+  RTC_DCHECK(!callback_);
>+  RTC_DCHECK(callback);
>+
>+  callback_ = callback;
>+  mode_ = mode;
>+
>+  have_xfixes_ =
>+      XFixesQueryExtension(display(), &xfixes_event_base_, &xfixes_error_base_);
>+
>+  if (have_xfixes_) {
>+    // Register for changes to the cursor shape.
>+    XFixesSelectCursorInput(display(), window_, XFixesDisplayCursorNotifyMask);
>+    x_display_->AddEventHandler(xfixes_event_base_ + XFixesCursorNotify, this);
>+
>+    CaptureCursor();
>+  } else {
>+    RTC_LOG(LS_INFO) << "X server does not support XFixes.";
>+  }
>+}
>+
>+void MouseCursorMonitorX11::Capture() {
>+  RTC_DCHECK(callback_);
>+
>+  // Process X11 events in case XFixes has sent cursor notification.
>+  x_display_->ProcessPendingXEvents();
>+
>+  // cursor_shape_| is set only if we were notified of a cursor shape change.
>+  if (cursor_shape_.get())
>+    callback_->OnMouseCursor(cursor_shape_.release());
>+
>+  // Get cursor position if necessary.
>+  if (mode_ == SHAPE_AND_POSITION) {
>+    int root_x;
>+    int root_y;
>+    int win_x;
>+    int win_y;
>+    Window root_window;
>+    Window child_window;
>+    unsigned int mask;
>+
>+    XErrorTrap error_trap(display());
>+    Bool result = XQueryPointer(display(), window_, &root_window, &child_window,
>+                                &root_x, &root_y, &win_x, &win_y, &mask);
>+    CursorState state;
>+    if (!result || error_trap.GetLastErrorAndDisable() != 0) {
>+      state = OUTSIDE;
>+    } else {
>+      // In screen mode (window_ == root_window) the mouse is always inside.
>+      // XQueryPointer() sets |child_window| to None if the cursor is outside
>+      // |window_|.
>+      state =
>+          (window_ == root_window || child_window != None) ? INSIDE : OUTSIDE;
>+    }
>+
>+    // As the comments to GetTopLevelWindow() above indicate, in window capture,
>+    // the cursor position capture happens in |window_|, while the frame catpure
>+    // happens in |child_window|. These two windows are not alwyas same, as
>+    // window manager may add some decorations to the |window_|. So translate
>+    // the coordinate in |window_| to the coordinate space of |child_window|.
>+    if (window_ != root_window && state == INSIDE) {
>+      int translated_x, translated_y;
>+      Window unused;
>+      if (XTranslateCoordinates(display(), window_, child_window, win_x, win_y,
>+                                &translated_x, &translated_y, &unused)) {
>+        win_x = translated_x;
>+        win_y = translated_y;
>+      }
>+    }
>+
>+    // X11 always starts the coordinate from (0, 0), so we do not need to
>+    // translate here.
>+    callback_->OnMouseCursorPosition(DesktopVector(root_x, root_y));
>+  }
>+}
>+
>+bool MouseCursorMonitorX11::HandleXEvent(const XEvent& event) {
>+  if (have_xfixes_ && event.type == xfixes_event_base_ + XFixesCursorNotify) {
>+    const XFixesCursorNotifyEvent* cursor_event =
>+        reinterpret_cast<const XFixesCursorNotifyEvent*>(&event);
>+    if (cursor_event->subtype == XFixesDisplayCursorNotify) {
>+      CaptureCursor();
>+    }
>+    // Return false, even if the event has been handled, because there might be
>+    // other listeners for cursor notifications.
>+  }
>+  return false;
>+}
>+
>+void MouseCursorMonitorX11::CaptureCursor() {
>+  RTC_DCHECK(have_xfixes_);
>+
>+  XFixesCursorImage* img;
>+  {
>+    XErrorTrap error_trap(display());
>+    img = XFixesGetCursorImage(display());
>+    if (!img || error_trap.GetLastErrorAndDisable() != 0)
>+      return;
>+  }
>+
>+  std::unique_ptr<DesktopFrame> image(
>+      new BasicDesktopFrame(DesktopSize(img->width, img->height)));
>+
>+  uint64_t* src = reinterpret_cast<uint64_t*>(img->pixels);
>+  uint32_t* dst = reinterpret_cast<uint32_t*>(image->data());
>+  uint32_t* dst_end = dst + (img->width * img->height);
>+  while (dst < dst_end) {
>+    *dst++ = *src++;
>+  }
>+
>+  DesktopVector hotspot(std::min(img->width, img->xhot),
>+                        std::min(img->height, img->yhot));
>+
>+  XFree(img);
>+
>+  cursor_shape_.reset(new MouseCursor(image.release(), hotspot));
>+}
>+
>+// static
>+MouseCursorMonitor* MouseCursorMonitorX11::CreateForWindow(
>+    const DesktopCaptureOptions& options,
>+    WindowId window) {
>+  if (!options.x_display())
>+    return NULL;
>+  window = GetTopLevelWindow(options.x_display()->display(), window);
>+  if (window == None)
>+    return NULL;
>+  return new MouseCursorMonitorX11(options, window);
>+}
>+
>+MouseCursorMonitor* MouseCursorMonitorX11::CreateForScreen(
>+    const DesktopCaptureOptions& options,
>+    ScreenId screen) {
>+  if (!options.x_display())
>+    return NULL;
>+  return new MouseCursorMonitorX11(
>+      options, DefaultRootWindow(options.x_display()->display()));
>+}
>+
>+std::unique_ptr<MouseCursorMonitor> MouseCursorMonitorX11::Create(
>+    const DesktopCaptureOptions& options) {
>+  return std::unique_ptr<MouseCursorMonitor>(
>+      CreateForScreen(options, kFullDesktopScreenId));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/mouse_cursor_monitor_x11.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/mouse_cursor_monitor_x11.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..3e8e0ca78c324de451b6e94a114156d5a342eaac
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/mouse_cursor_monitor_x11.h
>@@ -0,0 +1,64 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_MOUSE_CURSOR_MONITOR_X11_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_MOUSE_CURSOR_MONITOR_X11_H_
>+
>+#include <X11/Xlib.h>
>+
>+#include <memory>
>+
>+#include "modules/desktop_capture/linux/shared_x_display.h"
>+#include "modules/desktop_capture/mouse_cursor_monitor.h"
>+
>+namespace webrtc {
>+
>+class MouseCursorMonitorX11 : public MouseCursorMonitor,
>+                              public SharedXDisplay::XEventHandler {
>+ public:
>+  MouseCursorMonitorX11(const DesktopCaptureOptions& options, Window window);
>+  ~MouseCursorMonitorX11() override;
>+
>+  static MouseCursorMonitor* CreateForWindow(
>+      const DesktopCaptureOptions& options,
>+      WindowId window);
>+  static MouseCursorMonitor* CreateForScreen(
>+      const DesktopCaptureOptions& options,
>+      ScreenId screen);
>+  static std::unique_ptr<MouseCursorMonitor> Create(
>+      const DesktopCaptureOptions& options);
>+
>+  void Init(Callback* callback, Mode mode) override;
>+  void Capture() override;
>+
>+ private:
>+  // SharedXDisplay::XEventHandler interface.
>+  bool HandleXEvent(const XEvent& event) override;
>+
>+  Display* display() { return x_display_->display(); }
>+
>+  // Captures current cursor shape and stores it in |cursor_shape_|.
>+  void CaptureCursor();
>+
>+  rtc::scoped_refptr<SharedXDisplay> x_display_;
>+  Callback* callback_;
>+  Mode mode_;
>+  Window window_;
>+
>+  bool have_xfixes_;
>+  int xfixes_event_base_;
>+  int xfixes_error_base_;
>+
>+  std::unique_ptr<MouseCursor> cursor_shape_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_MOUSE_CURSOR_MONITOR_X11_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_pipewire.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_pipewire.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..724d8537d31178df40ad324074c30f5cced0090a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_pipewire.cc
>@@ -0,0 +1,30 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/screen_capturer_pipewire.h"
>+
>+#include <memory>
>+
>+#include "absl/memory/memory.h"
>+
>+namespace webrtc {
>+
>+ScreenCapturerPipeWire::ScreenCapturerPipeWire()
>+    : BaseCapturerPipeWire(BaseCapturerPipeWire::CaptureSourceType::Screen) {}
>+ScreenCapturerPipeWire::~ScreenCapturerPipeWire() {}
>+
>+// static
>+std::unique_ptr<DesktopCapturer>
>+ScreenCapturerPipeWire::CreateRawScreenCapturer(
>+    const DesktopCaptureOptions& options) {
>+  return absl::make_unique<ScreenCapturerPipeWire>();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_pipewire.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_pipewire.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..66dcd680e06a320fa37acfbf0bafb72b5670eefa
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_pipewire.h
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_SCREEN_CAPTURER_PIPEWIRE_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_SCREEN_CAPTURER_PIPEWIRE_H_
>+
>+#include <memory>
>+
>+#include "modules/desktop_capture/linux/base_capturer_pipewire.h"
>+
>+namespace webrtc {
>+
>+class ScreenCapturerPipeWire : public BaseCapturerPipeWire {
>+ public:
>+  ScreenCapturerPipeWire();
>+  ~ScreenCapturerPipeWire() override;
>+
>+  static std::unique_ptr<DesktopCapturer> CreateRawScreenCapturer(
>+      const DesktopCaptureOptions& options);
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerPipeWire);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_SCREEN_CAPTURER_PIPEWIRE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_x11.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..9d6a085b65ccfaaf3f0e2853d797377c468a9a79
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_x11.cc
>@@ -0,0 +1,337 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/screen_capturer_x11.h"
>+
>+#include <string.h>
>+
>+#include <X11/Xlib.h>
>+#include <X11/Xutil.h>
>+#include <X11/extensions/Xdamage.h>
>+#include <X11/extensions/Xfixes.h>
>+
>+#include <memory>
>+#include <set>
>+#include <utility>
>+
>+#include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
>+#include "modules/desktop_capture/desktop_frame.h"
>+#include "modules/desktop_capture/linux/x_server_pixel_buffer.h"
>+#include "modules/desktop_capture/screen_capture_frame_queue.h"
>+#include "modules/desktop_capture/screen_capturer_helper.h"
>+#include "modules/desktop_capture/shared_desktop_frame.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/timeutils.h"
>+#include "rtc_base/trace_event.h"
>+
>+namespace webrtc {
>+
>+ScreenCapturerX11::ScreenCapturerX11() {
>+  helper_.SetLogGridSize(4);
>+}
>+
>+ScreenCapturerX11::~ScreenCapturerX11() {
>+  options_.x_display()->RemoveEventHandler(ConfigureNotify, this);
>+  if (use_damage_) {
>+    options_.x_display()->RemoveEventHandler(damage_event_base_ + XDamageNotify,
>+                                             this);
>+  }
>+  DeinitXlib();
>+}
>+
>+bool ScreenCapturerX11::Init(const DesktopCaptureOptions& options) {
>+  TRACE_EVENT0("webrtc", "ScreenCapturerX11::Init");
>+  options_ = options;
>+
>+  root_window_ = RootWindow(display(), DefaultScreen(display()));
>+  if (root_window_ == BadValue) {
>+    RTC_LOG(LS_ERROR) << "Unable to get the root window";
>+    DeinitXlib();
>+    return false;
>+  }
>+
>+  gc_ = XCreateGC(display(), root_window_, 0, NULL);
>+  if (gc_ == NULL) {
>+    RTC_LOG(LS_ERROR) << "Unable to get graphics context";
>+    DeinitXlib();
>+    return false;
>+  }
>+
>+  options_.x_display()->AddEventHandler(ConfigureNotify, this);
>+
>+  // Check for XFixes extension. This is required for cursor shape
>+  // notifications, and for our use of XDamage.
>+  if (XFixesQueryExtension(display(), &xfixes_event_base_,
>+                           &xfixes_error_base_)) {
>+    has_xfixes_ = true;
>+  } else {
>+    RTC_LOG(LS_INFO) << "X server does not support XFixes.";
>+  }
>+
>+  // Register for changes to the dimensions of the root window.
>+  XSelectInput(display(), root_window_, StructureNotifyMask);
>+
>+  if (!x_server_pixel_buffer_.Init(display(), DefaultRootWindow(display()))) {
>+    RTC_LOG(LS_ERROR) << "Failed to initialize pixel buffer.";
>+    return false;
>+  }
>+
>+  if (options_.use_update_notifications()) {
>+    InitXDamage();
>+  }
>+
>+  return true;
>+}
>+
>+void ScreenCapturerX11::InitXDamage() {
>+  // Our use of XDamage requires XFixes.
>+  if (!has_xfixes_) {
>+    return;
>+  }
>+
>+  // Check for XDamage extension.
>+  if (!XDamageQueryExtension(display(), &damage_event_base_,
>+                             &damage_error_base_)) {
>+    RTC_LOG(LS_INFO) << "X server does not support XDamage.";
>+    return;
>+  }
>+
>+  // TODO(lambroslambrou): Disable DAMAGE in situations where it is known
>+  // to fail, such as when Desktop Effects are enabled, with graphics
>+  // drivers (nVidia, ATI) that fail to report DAMAGE notifications
>+  // properly.
>+
>+  // Request notifications every time the screen becomes damaged.
>+  damage_handle_ =
>+      XDamageCreate(display(), root_window_, XDamageReportNonEmpty);
>+  if (!damage_handle_) {
>+    RTC_LOG(LS_ERROR) << "Unable to initialize XDamage.";
>+    return;
>+  }
>+
>+  // Create an XFixes server-side region to collate damage into.
>+  damage_region_ = XFixesCreateRegion(display(), 0, 0);
>+  if (!damage_region_) {
>+    XDamageDestroy(display(), damage_handle_);
>+    RTC_LOG(LS_ERROR) << "Unable to create XFixes region.";
>+    return;
>+  }
>+
>+  options_.x_display()->AddEventHandler(damage_event_base_ + XDamageNotify,
>+                                        this);
>+
>+  use_damage_ = true;
>+  RTC_LOG(LS_INFO) << "Using XDamage extension.";
>+}
>+
>+void ScreenCapturerX11::Start(Callback* callback) {
>+  RTC_DCHECK(!callback_);
>+  RTC_DCHECK(callback);
>+
>+  callback_ = callback;
>+}
>+
>+void ScreenCapturerX11::CaptureFrame() {
>+  TRACE_EVENT0("webrtc", "ScreenCapturerX11::CaptureFrame");
>+  int64_t capture_start_time_nanos = rtc::TimeNanos();
>+
>+  queue_.MoveToNextFrame();
>+  RTC_DCHECK(!queue_.current_frame() || !queue_.current_frame()->IsShared());
>+
>+  // Process XEvents for XDamage and cursor shape tracking.
>+  options_.x_display()->ProcessPendingXEvents();
>+
>+  // ProcessPendingXEvents() may call ScreenConfigurationChanged() which
>+  // reinitializes |x_server_pixel_buffer_|. Check if the pixel buffer is still
>+  // in a good shape.
>+  if (!x_server_pixel_buffer_.is_initialized()) {
>+    // We failed to initialize pixel buffer.
>+    RTC_LOG(LS_ERROR) << "Pixel buffer is not initialized.";
>+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>+    return;
>+  }
>+
>+  // If the current frame is from an older generation then allocate a new one.
>+  // Note that we can't reallocate other buffers at this point, since the caller
>+  // may still be reading from them.
>+  if (!queue_.current_frame()) {
>+    queue_.ReplaceCurrentFrame(
>+        SharedDesktopFrame::Wrap(std::unique_ptr<DesktopFrame>(
>+            new BasicDesktopFrame(x_server_pixel_buffer_.window_size()))));
>+  }
>+
>+  std::unique_ptr<DesktopFrame> result = CaptureScreen();
>+  if (!result) {
>+    RTC_LOG(LS_WARNING) << "Temporarily failed to capture screen.";
>+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
>+    return;
>+  }
>+
>+  last_invalid_region_ = result->updated_region();
>+  result->set_capture_time_ms((rtc::TimeNanos() - capture_start_time_nanos) /
>+                              rtc::kNumNanosecsPerMillisec);
>+  callback_->OnCaptureResult(Result::SUCCESS, std::move(result));
>+}
>+
>+bool ScreenCapturerX11::GetSourceList(SourceList* sources) {
>+  RTC_DCHECK(sources->size() == 0);
>+  // TODO(jiayl): implement screen enumeration.
>+  sources->push_back({0});
>+  return true;
>+}
>+
>+bool ScreenCapturerX11::SelectSource(SourceId id) {
>+  // TODO(jiayl): implement screen selection.
>+  return true;
>+}
>+
>+bool ScreenCapturerX11::HandleXEvent(const XEvent& event) {
>+  if (use_damage_ && (event.type == damage_event_base_ + XDamageNotify)) {
>+    const XDamageNotifyEvent* damage_event =
>+        reinterpret_cast<const XDamageNotifyEvent*>(&event);
>+    if (damage_event->damage != damage_handle_)
>+      return false;
>+    RTC_DCHECK(damage_event->level == XDamageReportNonEmpty);
>+    return true;
>+  } else if (event.type == ConfigureNotify) {
>+    ScreenConfigurationChanged();
>+    return true;
>+  }
>+  return false;
>+}
>+
>+std::unique_ptr<DesktopFrame> ScreenCapturerX11::CaptureScreen() {
>+  std::unique_ptr<SharedDesktopFrame> frame = queue_.current_frame()->Share();
>+  RTC_DCHECK(x_server_pixel_buffer_.window_size().equals(frame->size()));
>+
>+  // Pass the screen size to the helper, so it can clip the invalid region if it
>+  // expands that region to a grid.
>+  helper_.set_size_most_recent(frame->size());
>+
>+  // In the DAMAGE case, ensure the frame is up-to-date with the previous frame
>+  // if any.  If there isn't a previous frame, that means a screen-resolution
>+  // change occurred, and |invalid_rects| will be updated to include the whole
>+  // screen.
>+  if (use_damage_ && queue_.previous_frame())
>+    SynchronizeFrame();
>+
>+  DesktopRegion* updated_region = frame->mutable_updated_region();
>+
>+  x_server_pixel_buffer_.Synchronize();
>+  if (use_damage_ && queue_.previous_frame()) {
>+    // Atomically fetch and clear the damage region.
>+    XDamageSubtract(display(), damage_handle_, None, damage_region_);
>+    int rects_num = 0;
>+    XRectangle bounds;
>+    XRectangle* rects = XFixesFetchRegionAndBounds(display(), damage_region_,
>+                                                   &rects_num, &bounds);
>+    for (int i = 0; i < rects_num; ++i) {
>+      updated_region->AddRect(DesktopRect::MakeXYWH(
>+          rects[i].x, rects[i].y, rects[i].width, rects[i].height));
>+    }
>+    XFree(rects);
>+    helper_.InvalidateRegion(*updated_region);
>+
>+    // Capture the damaged portions of the desktop.
>+    helper_.TakeInvalidRegion(updated_region);
>+
>+    // Clip the damaged portions to the current screen size, just in case some
>+    // spurious XDamage notifications were received for a previous (larger)
>+    // screen size.
>+    updated_region->IntersectWith(
>+        DesktopRect::MakeSize(x_server_pixel_buffer_.window_size()));
>+
>+    for (DesktopRegion::Iterator it(*updated_region); !it.IsAtEnd();
>+         it.Advance()) {
>+      if (!x_server_pixel_buffer_.CaptureRect(it.rect(), frame.get()))
>+        return nullptr;
>+    }
>+  } else {
>+    // Doing full-screen polling, or this is the first capture after a
>+    // screen-resolution change.  In either case, need a full-screen capture.
>+    DesktopRect screen_rect = DesktopRect::MakeSize(frame->size());
>+    if (!x_server_pixel_buffer_.CaptureRect(screen_rect, frame.get()))
>+      return nullptr;
>+    updated_region->SetRect(screen_rect);
>+  }
>+
>+  return std::move(frame);
>+}
>+
>+void ScreenCapturerX11::ScreenConfigurationChanged() {
>+  TRACE_EVENT0("webrtc", "ScreenCapturerX11::ScreenConfigurationChanged");
>+  // Make sure the frame buffers will be reallocated.
>+  queue_.Reset();
>+
>+  helper_.ClearInvalidRegion();
>+  if (!x_server_pixel_buffer_.Init(display(), DefaultRootWindow(display()))) {
>+    RTC_LOG(LS_ERROR) << "Failed to initialize pixel buffer after screen "
>+                         "configuration change.";
>+  }
>+}
>+
>+void ScreenCapturerX11::SynchronizeFrame() {
>+  // Synchronize the current buffer with the previous one since we do not
>+  // capture the entire desktop. Note that encoder may be reading from the
>+  // previous buffer at this time so thread access complaints are false
>+  // positives.
>+
>+  // TODO(hclam): We can reduce the amount of copying here by subtracting
>+  // |capturer_helper_|s region from |last_invalid_region_|.
>+  // http://crbug.com/92354
>+  RTC_DCHECK(queue_.previous_frame());
>+
>+  DesktopFrame* current = queue_.current_frame();
>+  DesktopFrame* last = queue_.previous_frame();
>+  RTC_DCHECK(current != last);
>+  for (DesktopRegion::Iterator it(last_invalid_region_); !it.IsAtEnd();
>+       it.Advance()) {
>+    current->CopyPixelsFrom(*last, it.rect().top_left(), it.rect());
>+  }
>+}
>+
>+void ScreenCapturerX11::DeinitXlib() {
>+  if (gc_) {
>+    XFreeGC(display(), gc_);
>+    gc_ = nullptr;
>+  }
>+
>+  x_server_pixel_buffer_.Release();
>+
>+  if (display()) {
>+    if (damage_handle_) {
>+      XDamageDestroy(display(), damage_handle_);
>+      damage_handle_ = 0;
>+    }
>+
>+    if (damage_region_) {
>+      XFixesDestroyRegion(display(), damage_region_);
>+      damage_region_ = 0;
>+    }
>+  }
>+}
>+
>+// static
>+std::unique_ptr<DesktopCapturer> ScreenCapturerX11::CreateRawScreenCapturer(
>+    const DesktopCaptureOptions& options) {
>+  if (!options.x_display())
>+    return nullptr;
>+
>+  std::unique_ptr<ScreenCapturerX11> capturer(new ScreenCapturerX11());
>+  if (!capturer.get()->Init(options)) {
>+    return nullptr;
>+  }
>+
>+  return std::move(capturer);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_x11.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_x11.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..9f3b60c3d87e5b17dd5949777851fb9cc49b98e3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/screen_capturer_x11.h
>@@ -0,0 +1,123 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_SCREEN_CAPTURER_X11_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_SCREEN_CAPTURER_X11_H_
>+
>+#include <X11/Xlib.h>
>+#include <X11/extensions/Xdamage.h>
>+
>+#include <memory>
>+#include <set>
>+#include <utility>
>+
>+#include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
>+#include "modules/desktop_capture/desktop_frame.h"
>+#include "modules/desktop_capture/linux/x_server_pixel_buffer.h"
>+#include "modules/desktop_capture/screen_capture_frame_queue.h"
>+#include "modules/desktop_capture/screen_capturer_helper.h"
>+#include "modules/desktop_capture/shared_desktop_frame.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+// A class to perform video frame capturing for Linux on X11.
>+//
>+// If XDamage is used, this class sets DesktopFrame::updated_region() according
>+// to the areas reported by XDamage. Otherwise this class does not detect
>+// DesktopFrame::updated_region(), the field is always set to the entire frame
>+// rectangle. ScreenCapturerDifferWrapper should be used if that functionality
>+// is necessary.
>+class ScreenCapturerX11 : public DesktopCapturer,
>+                          public SharedXDisplay::XEventHandler {
>+ public:
>+  ScreenCapturerX11();
>+  ~ScreenCapturerX11() override;
>+
>+  static std::unique_ptr<DesktopCapturer> CreateRawScreenCapturer(
>+      const DesktopCaptureOptions& options);
>+
>+  // TODO(ajwong): Do we really want this to be synchronous?
>+  bool Init(const DesktopCaptureOptions& options);
>+
>+  // DesktopCapturer interface.
>+  void Start(Callback* delegate) override;
>+  void CaptureFrame() override;
>+  bool GetSourceList(SourceList* sources) override;
>+  bool SelectSource(SourceId id) override;
>+
>+ private:
>+  Display* display() { return options_.x_display()->display(); }
>+
>+  // SharedXDisplay::XEventHandler interface.
>+  bool HandleXEvent(const XEvent& event) override;
>+
>+  void InitXDamage();
>+
>+  // Capture screen pixels to the current buffer in the queue. In the DAMAGE
>+  // case, the ScreenCapturerHelper already holds the list of invalid rectangles
>+  // from HandleXEvent(). In the non-DAMAGE case, this captures the
>+  // whole screen, then calculates some invalid rectangles that include any
>+  // differences between this and the previous capture.
>+  std::unique_ptr<DesktopFrame> CaptureScreen();
>+
>+  // Called when the screen configuration is changed.
>+  void ScreenConfigurationChanged();
>+
>+  // Synchronize the current buffer with |last_buffer_|, by copying pixels from
>+  // the area of |last_invalid_rects|.
>+  // Note this only works on the assumption that kNumBuffers == 2, as
>+  // |last_invalid_rects| holds the differences from the previous buffer and
>+  // the one prior to that (which will then be the current buffer).
>+  void SynchronizeFrame();
>+
>+  void DeinitXlib();
>+
>+  DesktopCaptureOptions options_;
>+
>+  Callback* callback_ = nullptr;
>+
>+  // X11 graphics context.
>+  GC gc_ = nullptr;
>+  Window root_window_ = BadValue;
>+
>+  // XFixes.
>+  bool has_xfixes_ = false;
>+  int xfixes_event_base_ = -1;
>+  int xfixes_error_base_ = -1;
>+
>+  // XDamage information.
>+  bool use_damage_ = false;
>+  Damage damage_handle_ = 0;
>+  int damage_event_base_ = -1;
>+  int damage_error_base_ = -1;
>+  XserverRegion damage_region_ = 0;
>+
>+  // Access to the X Server's pixel buffer.
>+  XServerPixelBuffer x_server_pixel_buffer_;
>+
>+  // A thread-safe list of invalid rectangles, and the size of the most
>+  // recently captured screen.
>+  ScreenCapturerHelper helper_;
>+
>+  // Queue of the frames buffers.
>+  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
>+
>+  // Invalid region from the previous capture. This is used to synchronize the
>+  // current with the last buffer used.
>+  DesktopRegion last_invalid_region_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerX11);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_SCREEN_CAPTURER_X11_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/shared_x_display.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/shared_x_display.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..c475db6e78759b53abbee977f09b60e4056ac055
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/shared_x_display.cc
>@@ -0,0 +1,89 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/shared_x_display.h"
>+
>+#include <X11/Xlib.h>
>+
>+#include <algorithm>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+SharedXDisplay::SharedXDisplay(Display* display) : display_(display) {
>+  RTC_DCHECK(display_);
>+}
>+
>+SharedXDisplay::~SharedXDisplay() {
>+  RTC_DCHECK(event_handlers_.empty());
>+  XCloseDisplay(display_);
>+}
>+
>+// static
>+rtc::scoped_refptr<SharedXDisplay> SharedXDisplay::Create(
>+    const std::string& display_name) {
>+  Display* display =
>+      XOpenDisplay(display_name.empty() ? NULL : display_name.c_str());
>+  if (!display) {
>+    RTC_LOG(LS_ERROR) << "Unable to open display";
>+    return NULL;
>+  }
>+  return new SharedXDisplay(display);
>+}
>+
>+// static
>+rtc::scoped_refptr<SharedXDisplay> SharedXDisplay::CreateDefault() {
>+  return Create(std::string());
>+}
>+
>+void SharedXDisplay::AddEventHandler(int type, XEventHandler* handler) {
>+  event_handlers_[type].push_back(handler);
>+}
>+
>+void SharedXDisplay::RemoveEventHandler(int type, XEventHandler* handler) {
>+  EventHandlersMap::iterator handlers = event_handlers_.find(type);
>+  if (handlers == event_handlers_.end())
>+    return;
>+
>+  std::vector<XEventHandler*>::iterator new_end =
>+      std::remove(handlers->second.begin(), handlers->second.end(), handler);
>+  handlers->second.erase(new_end, handlers->second.end());
>+
>+  // Check if no handlers left for this event.
>+  if (handlers->second.empty())
>+    event_handlers_.erase(handlers);
>+}
>+
>+void SharedXDisplay::ProcessPendingXEvents() {
>+  // Hold reference to |this| to prevent it from being destroyed while
>+  // processing events.
>+  rtc::scoped_refptr<SharedXDisplay> self(this);
>+
>+  // Find the number of events that are outstanding "now."  We don't just loop
>+  // on XPending because we want to guarantee this terminates.
>+  int events_to_process = XPending(display());
>+  XEvent e;
>+
>+  for (int i = 0; i < events_to_process; i++) {
>+    XNextEvent(display(), &e);
>+    EventHandlersMap::iterator handlers = event_handlers_.find(e.type);
>+    if (handlers == event_handlers_.end())
>+      continue;
>+    for (std::vector<XEventHandler*>::iterator it = handlers->second.begin();
>+         it != handlers->second.end(); ++it) {
>+      if ((*it)->HandleXEvent(e))
>+        break;
>+    }
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/shared_x_display.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/shared_x_display.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..a14f41fab8a3b73241777405b1ac476188ea1722
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/shared_x_display.h
>@@ -0,0 +1,81 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_SHARED_X_DISPLAY_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_SHARED_X_DISPLAY_H_
>+
>+#include <map>
>+#include <vector>
>+
>+#include <string>
>+
>+#include "api/refcountedbase.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+
>+// Including Xlib.h will involve evil defines (Bool, Status, True, False), which
>+// easily conflict with other headers.
>+typedef struct _XDisplay Display;
>+typedef union _XEvent XEvent;
>+
>+namespace webrtc {
>+
>+// A ref-counted object to store XDisplay connection.
>+class SharedXDisplay : public rtc::RefCountedBase {
>+ public:
>+  class XEventHandler {
>+   public:
>+    virtual ~XEventHandler() {}
>+
>+    // Processes XEvent. Returns true if the event has been handled.
>+    virtual bool HandleXEvent(const XEvent& event) = 0;
>+  };
>+
>+  // Takes ownership of |display|.
>+  explicit SharedXDisplay(Display* display);
>+
>+  // Creates a new X11 Display for the |display_name|. NULL is returned if X11
>+  // connection failed. Equivalent to CreateDefault() when |display_name| is
>+  // empty.
>+  static rtc::scoped_refptr<SharedXDisplay> Create(
>+      const std::string& display_name);
>+
>+  // Creates X11 Display connection for the default display (e.g. specified in
>+  // DISPLAY). NULL is returned if X11 connection failed.
>+  static rtc::scoped_refptr<SharedXDisplay> CreateDefault();
>+
>+  Display* display() { return display_; }
>+
>+  // Adds a new event |handler| for XEvent's of |type|.
>+  void AddEventHandler(int type, XEventHandler* handler);
>+
>+  // Removes event |handler| added using |AddEventHandler|. Doesn't do anything
>+  // if |handler| is not registered.
>+  void RemoveEventHandler(int type, XEventHandler* handler);
>+
>+  // Processes pending XEvents, calling corresponding event handlers.
>+  void ProcessPendingXEvents();
>+
>+ protected:
>+  ~SharedXDisplay() override;
>+
>+ private:
>+  typedef std::map<int, std::vector<XEventHandler*> > EventHandlersMap;
>+
>+  Display* display_;
>+
>+  EventHandlersMap event_handlers_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(SharedXDisplay);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_SHARED_X_DISPLAY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_pipewire.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_pipewire.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..1c195aaf4a01423407c727cb602c54ed76e9d9ea
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_pipewire.cc
>@@ -0,0 +1,30 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/window_capturer_pipewire.h"
>+
>+#include <memory>
>+
>+#include "absl/memory/memory.h"
>+
>+namespace webrtc {
>+
>+WindowCapturerPipeWire::WindowCapturerPipeWire()
>+    : BaseCapturerPipeWire(BaseCapturerPipeWire::CaptureSourceType::Window) {}
>+WindowCapturerPipeWire::~WindowCapturerPipeWire() {}
>+
>+// static
>+std::unique_ptr<DesktopCapturer>
>+WindowCapturerPipeWire::CreateRawWindowCapturer(
>+    const DesktopCaptureOptions& options) {
>+  return absl::make_unique<WindowCapturerPipeWire>();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_pipewire.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_pipewire.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..7f184ef29992dfcb7884cefe32eba591d716f455
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_pipewire.h
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_CAPTURER_PIPEWIRE_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_CAPTURER_PIPEWIRE_H_
>+
>+#include <memory>
>+
>+#include "modules/desktop_capture/linux/base_capturer_pipewire.h"
>+
>+namespace webrtc {
>+
>+class WindowCapturerPipeWire : public BaseCapturerPipeWire {
>+ public:
>+  WindowCapturerPipeWire();
>+  ~WindowCapturerPipeWire() override;
>+
>+  static std::unique_ptr<DesktopCapturer> CreateRawWindowCapturer(
>+      const DesktopCaptureOptions& options);
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(WindowCapturerPipeWire);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_CAPTURER_PIPEWIRE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_x11.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..4ef7c5cb3882eebf4a325b01664f935d56b342b1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_x11.cc
>@@ -0,0 +1,244 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/window_capturer_x11.h"
>+
>+#include <X11/Xutil.h>
>+#include <X11/extensions/Xcomposite.h>
>+#include <X11/extensions/Xrender.h>
>+
>+#include <memory>
>+#include <string>
>+#include <utility>
>+
>+#include "modules/desktop_capture/desktop_frame.h"
>+#include "modules/desktop_capture/linux/shared_x_display.h"
>+#include "modules/desktop_capture/linux/window_finder_x11.h"
>+#include "modules/desktop_capture/linux/window_list_utils.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/trace_event.h"
>+
>+namespace webrtc {
>+
>+WindowCapturerX11::WindowCapturerX11(const DesktopCaptureOptions& options)
>+    : x_display_(options.x_display()),
>+      atom_cache_(display()),
>+      window_finder_(&atom_cache_) {
>+  int event_base, error_base, major_version, minor_version;
>+  if (XCompositeQueryExtension(display(), &event_base, &error_base) &&
>+      XCompositeQueryVersion(display(), &major_version, &minor_version) &&
>+      // XCompositeNameWindowPixmap() requires version 0.2
>+      (major_version > 0 || minor_version >= 2)) {
>+    has_composite_extension_ = true;
>+  } else {
>+    RTC_LOG(LS_INFO) << "Xcomposite extension not available or too old.";
>+  }
>+
>+  x_display_->AddEventHandler(ConfigureNotify, this);
>+}
>+
>+WindowCapturerX11::~WindowCapturerX11() {
>+  x_display_->RemoveEventHandler(ConfigureNotify, this);
>+}
>+
>+bool WindowCapturerX11::GetSourceList(SourceList* sources) {
>+  return GetWindowList(&atom_cache_, [this, sources](::Window window) {
>+    Source w;
>+    w.id = window;
>+    if (this->GetWindowTitle(window, &w.title)) {
>+      sources->push_back(w);
>+    }
>+    return true;
>+  });
>+}
>+
>+bool WindowCapturerX11::SelectSource(SourceId id) {
>+  if (!x_server_pixel_buffer_.Init(display(), id))
>+    return false;
>+
>+  // Tell the X server to send us window resizing events.
>+  XSelectInput(display(), id, StructureNotifyMask);
>+
>+  selected_window_ = id;
>+
>+  // In addition to needing X11 server-side support for Xcomposite, it actually
>+  // needs to be turned on for the window. If the user has modern
>+  // hardware/drivers but isn't using a compositing window manager, that won't
>+  // be the case. Here we automatically turn it on.
>+
>+  // Redirect drawing to an offscreen buffer (ie, turn on compositing). X11
>+  // remembers who has requested this and will turn it off for us when we exit.
>+  XCompositeRedirectWindow(display(), id, CompositeRedirectAutomatic);
>+
>+  return true;
>+}
>+
>+bool WindowCapturerX11::FocusOnSelectedSource() {
>+  if (!selected_window_)
>+    return false;
>+
>+  unsigned int num_children;
>+  ::Window* children;
>+  ::Window parent;
>+  ::Window root;
>+  // Find the root window to pass event to.
>+  int status = XQueryTree(display(), selected_window_, &root, &parent,
>+                          &children, &num_children);
>+  if (status == 0) {
>+    RTC_LOG(LS_ERROR) << "Failed to query for the root window.";
>+    return false;
>+  }
>+
>+  if (children)
>+    XFree(children);
>+
>+  XRaiseWindow(display(), selected_window_);
>+
>+  // Some window managers (e.g., metacity in GNOME) consider it illegal to
>+  // raise a window without also giving it input focus with
>+  // _NET_ACTIVE_WINDOW, so XRaiseWindow() on its own isn't enough.
>+  Atom atom = XInternAtom(display(), "_NET_ACTIVE_WINDOW", True);
>+  if (atom != None) {
>+    XEvent xev;
>+    xev.xclient.type = ClientMessage;
>+    xev.xclient.serial = 0;
>+    xev.xclient.send_event = True;
>+    xev.xclient.window = selected_window_;
>+    xev.xclient.message_type = atom;
>+
>+    // The format member is set to 8, 16, or 32 and specifies whether the
>+    // data should be viewed as a list of bytes, shorts, or longs.
>+    xev.xclient.format = 32;
>+
>+    memset(xev.xclient.data.l, 0, sizeof(xev.xclient.data.l));
>+
>+    XSendEvent(display(), root, False,
>+               SubstructureRedirectMask | SubstructureNotifyMask, &xev);
>+  }
>+  XFlush(display());
>+  return true;
>+}
>+
>+void WindowCapturerX11::Start(Callback* callback) {
>+  RTC_DCHECK(!callback_);
>+  RTC_DCHECK(callback);
>+
>+  callback_ = callback;
>+}
>+
>+void WindowCapturerX11::CaptureFrame() {
>+  TRACE_EVENT0("webrtc", "WindowCapturerX11::CaptureFrame");
>+
>+  if (!x_server_pixel_buffer_.IsWindowValid()) {
>+    RTC_LOG(LS_ERROR) << "The window is no longer valid.";
>+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>+    return;
>+  }
>+
>+  x_display_->ProcessPendingXEvents();
>+
>+  if (!has_composite_extension_) {
>+    // Without the Xcomposite extension we capture when the whole window is
>+    // visible on screen and not covered by any other window. This is not
>+    // something we want so instead, just bail out.
>+    RTC_LOG(LS_ERROR) << "No Xcomposite extension detected.";
>+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>+    return;
>+  }
>+
>+  if (GetWindowState(&atom_cache_, selected_window_) == IconicState) {
>+    // Window is in minimized. Return a 1x1 frame as same as OSX/Win does.
>+    std::unique_ptr<DesktopFrame> frame(
>+        new BasicDesktopFrame(DesktopSize(1, 1)));
>+    callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
>+    return;
>+  }
>+
>+  std::unique_ptr<DesktopFrame> frame(
>+      new BasicDesktopFrame(x_server_pixel_buffer_.window_size()));
>+
>+  x_server_pixel_buffer_.Synchronize();
>+  if (!x_server_pixel_buffer_.CaptureRect(DesktopRect::MakeSize(frame->size()),
>+                                          frame.get())) {
>+    RTC_LOG(LS_WARNING) << "Temporarily failed to capture winodw.";
>+    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
>+    return;
>+  }
>+
>+  frame->mutable_updated_region()->SetRect(
>+      DesktopRect::MakeSize(frame->size()));
>+  frame->set_top_left(x_server_pixel_buffer_.window_rect().top_left());
>+
>+  callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
>+}
>+
>+bool WindowCapturerX11::IsOccluded(const DesktopVector& pos) {
>+  return window_finder_.GetWindowUnderPoint(pos) !=
>+         static_cast<WindowId>(selected_window_);
>+}
>+
>+bool WindowCapturerX11::HandleXEvent(const XEvent& event) {
>+  if (event.type == ConfigureNotify) {
>+    XConfigureEvent xce = event.xconfigure;
>+    if (xce.window == selected_window_) {
>+      if (!DesktopRectFromXAttributes(xce).equals(
>+              x_server_pixel_buffer_.window_rect())) {
>+        if (!x_server_pixel_buffer_.Init(display(), selected_window_)) {
>+          RTC_LOG(LS_ERROR)
>+              << "Failed to initialize pixel buffer after resizing.";
>+        }
>+      }
>+    }
>+  }
>+
>+  // Always returns false, so other observers can still receive the events.
>+  return false;
>+}
>+
>+bool WindowCapturerX11::GetWindowTitle(::Window window, std::string* title) {
>+  int status;
>+  bool result = false;
>+  XTextProperty window_name;
>+  window_name.value = nullptr;
>+  if (window) {
>+    status = XGetWMName(display(), window, &window_name);
>+    if (status && window_name.value && window_name.nitems) {
>+      int cnt;
>+      char** list = nullptr;
>+      status =
>+          Xutf8TextPropertyToTextList(display(), &window_name, &list, &cnt);
>+      if (status >= Success && cnt && *list) {
>+        if (cnt > 1) {
>+          RTC_LOG(LS_INFO) << "Window has " << cnt
>+                           << " text properties, only using the first one.";
>+        }
>+        *title = *list;
>+        result = true;
>+      }
>+      if (list)
>+        XFreeStringList(list);
>+    }
>+    if (window_name.value)
>+      XFree(window_name.value);
>+  }
>+  return result;
>+}
>+
>+// static
>+std::unique_ptr<DesktopCapturer> WindowCapturerX11::CreateRawWindowCapturer(
>+    const DesktopCaptureOptions& options) {
>+  if (!options.x_display())
>+    return nullptr;
>+  return std::unique_ptr<DesktopCapturer>(new WindowCapturerX11(options));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_x11.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_x11.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..4cd2c97cb9fddea5a7875fde236495c17f5591b8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_capturer_x11.h
>@@ -0,0 +1,68 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_CAPTURER_X11_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_CAPTURER_X11_H_
>+
>+#include <memory>
>+#include <string>
>+
>+#include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
>+#include "modules/desktop_capture/linux/window_finder_x11.h"
>+#include "modules/desktop_capture/linux/x_atom_cache.h"
>+#include "modules/desktop_capture/linux/x_server_pixel_buffer.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+class WindowCapturerX11 : public DesktopCapturer,
>+                          public SharedXDisplay::XEventHandler {
>+ public:
>+  explicit WindowCapturerX11(const DesktopCaptureOptions& options);
>+  ~WindowCapturerX11() override;
>+
>+  static std::unique_ptr<DesktopCapturer> CreateRawWindowCapturer(
>+      const DesktopCaptureOptions& options);
>+
>+  // DesktopCapturer interface.
>+  void Start(Callback* callback) override;
>+  void CaptureFrame() override;
>+  bool GetSourceList(SourceList* sources) override;
>+  bool SelectSource(SourceId id) override;
>+  bool FocusOnSelectedSource() override;
>+  bool IsOccluded(const DesktopVector& pos) override;
>+
>+  // SharedXDisplay::XEventHandler interface.
>+  bool HandleXEvent(const XEvent& event) override;
>+
>+ private:
>+  Display* display() { return x_display_->display(); }
>+
>+  // Returns window title for the specified X |window|.
>+  bool GetWindowTitle(::Window window, std::string* title);
>+
>+  Callback* callback_ = nullptr;
>+
>+  rtc::scoped_refptr<SharedXDisplay> x_display_;
>+
>+  bool has_composite_extension_ = false;
>+
>+  ::Window selected_window_ = 0;
>+  XServerPixelBuffer x_server_pixel_buffer_;
>+  XAtomCache atom_cache_;
>+  WindowFinderX11 window_finder_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(WindowCapturerX11);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_CAPTURER_X11_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_finder_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_finder_x11.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..46fadc2c6fb43292dba613706bdc302fe9c819e3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_finder_x11.cc
>@@ -0,0 +1,51 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/window_finder_x11.h"
>+
>+#include <memory>
>+
>+#include "absl/memory/memory.h"
>+#include "modules/desktop_capture/linux/window_list_utils.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+WindowFinderX11::WindowFinderX11(XAtomCache* cache) : cache_(cache) {
>+  RTC_DCHECK(cache_);
>+}
>+
>+WindowFinderX11::~WindowFinderX11() = default;
>+
>+WindowId WindowFinderX11::GetWindowUnderPoint(DesktopVector point) {
>+  WindowId id = kNullWindowId;
>+  GetWindowList(cache_, [&id, this, point](::Window window) {
>+    DesktopRect rect;
>+    if (GetWindowRect(this->cache_->display(), window, &rect) &&
>+        rect.Contains(point)) {
>+      id = window;
>+      return false;
>+    }
>+    return true;
>+  });
>+  return id;
>+}
>+
>+// static
>+std::unique_ptr<WindowFinder> WindowFinder::Create(
>+    const WindowFinder::Options& options) {
>+  if (options.cache == nullptr) {
>+    return nullptr;
>+  }
>+
>+  return absl::make_unique<WindowFinderX11>(options.cache);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_finder_x11.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_finder_x11.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..d0bba8697b5f7544d4d6cccde6f12ce56a9ce488
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_finder_x11.h
>@@ -0,0 +1,35 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_FINDER_X11_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_FINDER_X11_H_
>+
>+#include "modules/desktop_capture/window_finder.h"
>+
>+namespace webrtc {
>+
>+class XAtomCache;
>+
>+// The implementation of WindowFinder for X11.
>+class WindowFinderX11 final : public WindowFinder {
>+ public:
>+  explicit WindowFinderX11(XAtomCache* cache);
>+  ~WindowFinderX11() override;
>+
>+  // WindowFinder implementation.
>+  WindowId GetWindowUnderPoint(DesktopVector point) override;
>+
>+ private:
>+  XAtomCache* const cache_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_FINDER_X11_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_list_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_list_utils.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..fd8f5f72cd5e5e5e0512909f797a30229de0dc96
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_list_utils.cc
>@@ -0,0 +1,246 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/window_list_utils.h"
>+
>+#include <X11/Xatom.h>
>+#include <X11/Xlib.h>
>+#include <X11/Xutil.h>
>+#include <string.h>
>+
>+#include <algorithm>
>+
>+#include "modules/desktop_capture/linux/x_error_trap.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+class DeferXFree {
>+ public:
>+  explicit DeferXFree(void* data) : data_(data) {}
>+  ~DeferXFree();
>+
>+ private:
>+  void* const data_;
>+};
>+
>+DeferXFree::~DeferXFree() {
>+  if (data_)
>+    XFree(data_);
>+}
>+
>+// Convenience wrapper for XGetWindowProperty() results.
>+template <class PropertyType>
>+class XWindowProperty {
>+ public:
>+  XWindowProperty(Display* display, Window window, Atom property) {
>+    const int kBitsPerByte = 8;
>+    Atom actual_type;
>+    int actual_format;
>+    unsigned long bytes_after;  // NOLINT: type required by XGetWindowProperty
>+    int status = XGetWindowProperty(
>+        display, window, property, 0L, ~0L, False, AnyPropertyType,
>+        &actual_type, &actual_format, &size_, &bytes_after, &data_);
>+    if (status != Success) {
>+      data_ = nullptr;
>+      return;
>+    }
>+    if (sizeof(PropertyType) * kBitsPerByte != actual_format) {
>+      size_ = 0;
>+      return;
>+    }
>+
>+    is_valid_ = true;
>+  }
>+
>+  ~XWindowProperty() {
>+    if (data_)
>+      XFree(data_);
>+  }
>+
>+  // True if we got properly value successfully.
>+  bool is_valid() const { return is_valid_; }
>+
>+  // Size and value of the property.
>+  size_t size() const { return size_; }
>+  const PropertyType* data() const {
>+    return reinterpret_cast<PropertyType*>(data_);
>+  }
>+  PropertyType* data() { return reinterpret_cast<PropertyType*>(data_); }
>+
>+ private:
>+  bool is_valid_ = false;
>+  unsigned long size_ = 0;  // NOLINT: type required by XGetWindowProperty
>+  unsigned char* data_ = nullptr;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(XWindowProperty);
>+};
>+
>+// Iterates through |window| hierarchy to find first visible window, i.e. one
>+// that has WM_STATE property set to NormalState.
>+// See http://tronche.com/gui/x/icccm/sec-4.html#s-4.1.3.1 .
>+::Window GetApplicationWindow(XAtomCache* cache, ::Window window) {
>+  int32_t state = GetWindowState(cache, window);
>+  if (state == NormalState) {
>+    // Window has WM_STATE==NormalState. Return it.
>+    return window;
>+  } else if (state == IconicState) {
>+    // Window is in minimized. Skip it.
>+    return 0;
>+  }
>+
>+  RTC_DCHECK_EQ(state, WithdrawnState);
>+  // If the window is in WithdrawnState then look at all of its children.
>+  ::Window root, parent;
>+  ::Window* children;
>+  unsigned int num_children;
>+  if (!XQueryTree(cache->display(), window, &root, &parent, &children,
>+                  &num_children)) {
>+    RTC_LOG(LS_ERROR) << "Failed to query for child windows although window"
>+                      << "does not have a valid WM_STATE.";
>+    return 0;
>+  }
>+  ::Window app_window = 0;
>+  for (unsigned int i = 0; i < num_children; ++i) {
>+    app_window = GetApplicationWindow(cache, children[i]);
>+    if (app_window)
>+      break;
>+  }
>+
>+  if (children)
>+    XFree(children);
>+  return app_window;
>+}
>+
>+// Returns true if the |window| is a desktop element.
>+bool IsDesktopElement(XAtomCache* cache, ::Window window) {
>+  RTC_DCHECK(cache);
>+  if (window == 0)
>+    return false;
>+
>+  // First look for _NET_WM_WINDOW_TYPE. The standard
>+  // (http://standards.freedesktop.org/wm-spec/latest/ar01s05.html#id2760306)
>+  // says this hint *should* be present on all windows, and we use the existence
>+  // of _NET_WM_WINDOW_TYPE_NORMAL in the property to indicate a window is not
>+  // a desktop element (that is, only "normal" windows should be shareable).
>+  XWindowProperty<uint32_t> window_type(cache->display(), window,
>+                                        cache->WindowType());
>+  if (window_type.is_valid() && window_type.size() > 0) {
>+    uint32_t* end = window_type.data() + window_type.size();
>+    bool is_normal =
>+        (end != std::find(window_type.data(), end, cache->WindowTypeNormal()));
>+    return !is_normal;
>+  }
>+
>+  // Fall back on using the hint.
>+  XClassHint class_hint;
>+  Status status = XGetClassHint(cache->display(), window, &class_hint);
>+  if (status == 0) {
>+    // No hints, assume this is a normal application window.
>+    return false;
>+  }
>+
>+  DeferXFree free_res_name(class_hint.res_name);
>+  DeferXFree free_res_class(class_hint.res_class);
>+  return strcmp("gnome-panel", class_hint.res_name) == 0 ||
>+         strcmp("desktop_window", class_hint.res_name) == 0;
>+}
>+
>+}  // namespace
>+
>+int32_t GetWindowState(XAtomCache* cache, ::Window window) {
>+  // Get WM_STATE property of the window.
>+  XWindowProperty<uint32_t> window_state(cache->display(), window,
>+                                         cache->WmState());
>+
>+  // WM_STATE is considered to be set to WithdrawnState when it missing.
>+  return window_state.is_valid() ? *window_state.data() : WithdrawnState;
>+}
>+
>+bool GetWindowList(XAtomCache* cache,
>+                   rtc::FunctionView<bool(::Window)> on_window) {
>+  RTC_DCHECK(cache);
>+  RTC_DCHECK(on_window);
>+  ::Display* const display = cache->display();
>+
>+  int failed_screens = 0;
>+  const int num_screens = XScreenCount(display);
>+  for (int screen = 0; screen < num_screens; screen++) {
>+    ::Window root_window = XRootWindow(display, screen);
>+    ::Window parent;
>+    ::Window* children;
>+    unsigned int num_children;
>+    {
>+      XErrorTrap error_trap(display);
>+      if (XQueryTree(display, root_window, &root_window, &parent, &children,
>+                     &num_children) == 0 ||
>+          error_trap.GetLastErrorAndDisable() != 0) {
>+        failed_screens++;
>+        RTC_LOG(LS_ERROR) << "Failed to query for child windows for screen "
>+                          << screen;
>+        continue;
>+      }
>+    }
>+
>+    DeferXFree free_children(children);
>+
>+    for (unsigned int i = 0; i < num_children; i++) {
>+      // Iterates in reverse order to return windows from front to back.
>+      ::Window app_window =
>+          GetApplicationWindow(cache, children[num_children - 1 - i]);
>+      if (app_window && !IsDesktopElement(cache, app_window)) {
>+        if (!on_window(app_window)) {
>+          return true;
>+        }
>+      }
>+    }
>+  }
>+
>+  return failed_screens < num_screens;
>+}
>+
>+bool GetWindowRect(::Display* display,
>+                   ::Window window,
>+                   DesktopRect* rect,
>+                   XWindowAttributes* attributes /* = nullptr */) {
>+  XWindowAttributes local_attributes;
>+  int offset_x;
>+  int offset_y;
>+  if (attributes == nullptr) {
>+    attributes = &local_attributes;
>+  }
>+
>+  {
>+    XErrorTrap error_trap(display);
>+    if (!XGetWindowAttributes(display, window, attributes) ||
>+        error_trap.GetLastErrorAndDisable() != 0) {
>+      return false;
>+    }
>+  }
>+  *rect = DesktopRectFromXAttributes(*attributes);
>+
>+  {
>+    XErrorTrap error_trap(display);
>+    ::Window child;
>+    if (!XTranslateCoordinates(display, window, attributes->root, -rect->left(),
>+                               -rect->top(), &offset_x, &offset_y, &child) ||
>+        error_trap.GetLastErrorAndDisable() != 0) {
>+      return false;
>+    }
>+  }
>+  rect->Translate(offset_x, offset_y);
>+  return true;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_list_utils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_list_utils.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..2057c99b84c23e4772fc56acc354afb4ee1eba81
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/window_list_utils.h
>@@ -0,0 +1,54 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_LIST_UTILS_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_LIST_UTILS_H_
>+
>+#include <X11/Xlib.h>
>+
>+#include "modules/desktop_capture/desktop_geometry.h"
>+#include "modules/desktop_capture/linux/x_atom_cache.h"
>+#include "rtc_base/function_view.h"
>+
>+namespace webrtc {
>+
>+// Synchronously iterates all on-screen windows in |cache|.display() in
>+// decreasing z-order and sends them one-by-one to |on_window| function before
>+// GetWindowList() returns. If |on_window| returns false, this function ignores
>+// other windows and returns immediately. GetWindowList() returns false if
>+// native APIs failed. If multiple screens are attached to the |display|, this
>+// function returns false only when native APIs failed on all screens. Menus,
>+// panels and minimized windows will be ignored.
>+bool GetWindowList(XAtomCache* cache,
>+                   rtc::FunctionView<bool(::Window)> on_window);
>+
>+// Returns WM_STATE property of the |window|. This function returns
>+// WithdrawnState if the |window| is missing.
>+int32_t GetWindowState(XAtomCache* cache, ::Window window);
>+
>+// Returns the rectangle of the |window| in the coordinates of |display|. This
>+// function returns false if native APIs failed. If |attributes| is provided, it
>+// will be filled with the attributes of |window|. The |rect| is in system
>+// coordinate, i.e. the primary monitor always starts from (0, 0).
>+bool GetWindowRect(::Display* display,
>+                   ::Window window,
>+                   DesktopRect* rect,
>+                   XWindowAttributes* attributes = nullptr);
>+
>+// Creates a DesktopRect from |attributes|.
>+template <typename T>
>+DesktopRect DesktopRectFromXAttributes(const T& attributes) {
>+  return DesktopRect::MakeXYWH(attributes.x, attributes.y, attributes.width,
>+                               attributes.height);
>+}
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_WINDOW_LIST_UTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_atom_cache.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_atom_cache.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..832e6166cb435cda87e3a9d1b46f4ea49aa6c0b6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_atom_cache.cc
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/x_atom_cache.h"
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+XAtomCache::XAtomCache(::Display* display) : display_(display) {
>+  RTC_DCHECK(display_);
>+}
>+
>+XAtomCache::~XAtomCache() = default;
>+
>+::Display* XAtomCache::display() const {
>+  return display_;
>+}
>+
>+Atom XAtomCache::WmState() {
>+  return CreateIfNotExist(&wm_state_, "WM_STATE");
>+}
>+
>+Atom XAtomCache::WindowType() {
>+  return CreateIfNotExist(&window_type_, "_NET_WM_WINDOW_TYPE");
>+}
>+
>+Atom XAtomCache::WindowTypeNormal() {
>+  return CreateIfNotExist(&window_type_normal_, "_NET_WM_WINDOW_TYPE_NORMAL");
>+}
>+
>+Atom XAtomCache::CreateIfNotExist(Atom* atom, const char* name) {
>+  RTC_DCHECK(atom);
>+  if (*atom == None) {
>+    *atom = XInternAtom(display(), name, True);
>+  }
>+  return *atom;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_atom_cache.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_atom_cache.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..71789f05124e10100bfa0d290883ba82f1c362ee
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_atom_cache.h
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X_ATOM_CACHE_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_X_ATOM_CACHE_H_
>+
>+#include <X11/Xatom.h>
>+#include <X11/Xlib.h>
>+
>+namespace webrtc {
>+
>+// A cache of Atom. Each Atom object is created on demand.
>+class XAtomCache final {
>+ public:
>+  explicit XAtomCache(::Display* display);
>+  ~XAtomCache();
>+
>+  ::Display* display() const;
>+
>+  Atom WmState();
>+  Atom WindowType();
>+  Atom WindowTypeNormal();
>+
>+ private:
>+  // If |*atom| is None, this function uses XInternAtom() to retrieve an Atom.
>+  Atom CreateIfNotExist(Atom* atom, const char* name);
>+
>+  ::Display* const display_;
>+  Atom wm_state_ = None;
>+  Atom window_type_ = None;
>+  Atom window_type_normal_ = None;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X_ATOM_CACHE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_error_trap.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_error_trap.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..bb5ca4512dc62a28253228a7f603113fcaa1d996
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_error_trap.cc
>@@ -0,0 +1,68 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/x_error_trap.h"
>+
>+#include <assert.h>
>+
>+#if defined(TOOLKIT_GTK)
>+#include <gdk/gdk.h>
>+#endif  // !defined(TOOLKIT_GTK)
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+#if !defined(TOOLKIT_GTK)
>+
>+// TODO(sergeyu): This code is not thread safe. Fix it. Bug 2202.
>+static bool g_xserver_error_trap_enabled = false;
>+static int g_last_xserver_error_code = 0;
>+
>+int XServerErrorHandler(Display* display, XErrorEvent* error_event) {
>+  assert(g_xserver_error_trap_enabled);
>+  g_last_xserver_error_code = error_event->error_code;
>+  return 0;
>+}
>+
>+#endif  // !defined(TOOLKIT_GTK)
>+
>+}  // namespace
>+
>+XErrorTrap::XErrorTrap(Display* display)
>+    : original_error_handler_(NULL), enabled_(true) {
>+#if defined(TOOLKIT_GTK)
>+  gdk_error_trap_push();
>+#else   // !defined(TOOLKIT_GTK)
>+  assert(!g_xserver_error_trap_enabled);
>+  original_error_handler_ = XSetErrorHandler(&XServerErrorHandler);
>+  g_xserver_error_trap_enabled = true;
>+  g_last_xserver_error_code = 0;
>+#endif  // !defined(TOOLKIT_GTK)
>+}
>+
>+int XErrorTrap::GetLastErrorAndDisable() {
>+  enabled_ = false;
>+#if defined(TOOLKIT_GTK)
>+  return gdk_error_trap_push();
>+#else   // !defined(TOOLKIT_GTK)
>+  assert(g_xserver_error_trap_enabled);
>+  XSetErrorHandler(original_error_handler_);
>+  g_xserver_error_trap_enabled = false;
>+  return g_last_xserver_error_code;
>+#endif  // !defined(TOOLKIT_GTK)
>+}
>+
>+XErrorTrap::~XErrorTrap() {
>+  if (enabled_)
>+    GetLastErrorAndDisable();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_error_trap.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_error_trap.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..a9d059a0bc3608e6d15e6e9674eaeb89e396bd3d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_error_trap.h
>@@ -0,0 +1,39 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X_ERROR_TRAP_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_X_ERROR_TRAP_H_
>+
>+#include <X11/Xlib.h>
>+
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+// Helper class that registers X Window error handler. Caller can use
>+// GetLastErrorAndDisable() to get the last error that was caught, if any.
>+class XErrorTrap {
>+ public:
>+  explicit XErrorTrap(Display* display);
>+  ~XErrorTrap();
>+
>+  // Returns last error and removes unregisters the error handler.
>+  int GetLastErrorAndDisable();
>+
>+ private:
>+  XErrorHandler original_error_handler_;
>+  bool enabled_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(XErrorTrap);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X_ERROR_TRAP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_server_pixel_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_server_pixel_buffer.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..19dd1398c9cb0f6c64009c575ef0fc5115c1db5b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_server_pixel_buffer.cc
>@@ -0,0 +1,351 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/linux/x_server_pixel_buffer.h"
>+
>+#include <string.h>
>+#include <sys/shm.h>
>+
>+#include "modules/desktop_capture/desktop_frame.h"
>+#include "modules/desktop_capture/linux/window_list_utils.h"
>+#include "modules/desktop_capture/linux/x_error_trap.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+// Returns the number of bits |mask| has to be shifted left so its last
>+// (most-significant) bit set becomes the most-significant bit of the word.
>+// When |mask| is 0 the function returns 31.
>+uint32_t MaskToShift(uint32_t mask) {
>+  int shift = 0;
>+  if ((mask & 0xffff0000u) == 0) {
>+    mask <<= 16;
>+    shift += 16;
>+  }
>+  if ((mask & 0xff000000u) == 0) {
>+    mask <<= 8;
>+    shift += 8;
>+  }
>+  if ((mask & 0xf0000000u) == 0) {
>+    mask <<= 4;
>+    shift += 4;
>+  }
>+  if ((mask & 0xc0000000u) == 0) {
>+    mask <<= 2;
>+    shift += 2;
>+  }
>+  if ((mask & 0x80000000u) == 0)
>+    shift += 1;
>+
>+  return shift;
>+}
>+
>+// Returns true if |image| is in RGB format.
>+bool IsXImageRGBFormat(XImage* image) {
>+  return image->bits_per_pixel == 32 && image->red_mask == 0xff0000 &&
>+         image->green_mask == 0xff00 && image->blue_mask == 0xff;
>+}
>+
>+// We expose two forms of blitting to handle variations in the pixel format.
>+// In FastBlit(), the operation is effectively a memcpy.
>+void FastBlit(XImage* x_image,
>+              uint8_t* src_pos,
>+              const DesktopRect& rect,
>+              DesktopFrame* frame) {
>+  int src_stride = x_image->bytes_per_line;
>+  int dst_x = rect.left(), dst_y = rect.top();
>+
>+  uint8_t* dst_pos = frame->data() + frame->stride() * dst_y;
>+  dst_pos += dst_x * DesktopFrame::kBytesPerPixel;
>+
>+  int height = rect.height();
>+  int row_bytes = rect.width() * DesktopFrame::kBytesPerPixel;
>+  for (int y = 0; y < height; ++y) {
>+    memcpy(dst_pos, src_pos, row_bytes);
>+    src_pos += src_stride;
>+    dst_pos += frame->stride();
>+  }
>+}
>+
>+void SlowBlit(XImage* x_image,
>+              uint8_t* src_pos,
>+              const DesktopRect& rect,
>+              DesktopFrame* frame) {
>+  int src_stride = x_image->bytes_per_line;
>+  int dst_x = rect.left(), dst_y = rect.top();
>+  int width = rect.width(), height = rect.height();
>+
>+  uint32_t red_mask = x_image->red_mask;
>+  uint32_t green_mask = x_image->red_mask;
>+  uint32_t blue_mask = x_image->blue_mask;
>+
>+  uint32_t red_shift = MaskToShift(red_mask);
>+  uint32_t green_shift = MaskToShift(green_mask);
>+  uint32_t blue_shift = MaskToShift(blue_mask);
>+
>+  int bits_per_pixel = x_image->bits_per_pixel;
>+
>+  uint8_t* dst_pos = frame->data() + frame->stride() * dst_y;
>+  dst_pos += dst_x * DesktopFrame::kBytesPerPixel;
>+  // TODO(hclam): Optimize, perhaps using MMX code or by converting to
>+  // YUV directly.
>+  // TODO(sergeyu): This code doesn't handle XImage byte order properly and
>+  // won't work with 24bpp images. Fix it.
>+  for (int y = 0; y < height; y++) {
>+    uint32_t* dst_pos_32 = reinterpret_cast<uint32_t*>(dst_pos);
>+    uint32_t* src_pos_32 = reinterpret_cast<uint32_t*>(src_pos);
>+    uint16_t* src_pos_16 = reinterpret_cast<uint16_t*>(src_pos);
>+    for (int x = 0; x < width; x++) {
>+      // Dereference through an appropriately-aligned pointer.
>+      uint32_t pixel;
>+      if (bits_per_pixel == 32) {
>+        pixel = src_pos_32[x];
>+      } else if (bits_per_pixel == 16) {
>+        pixel = src_pos_16[x];
>+      } else {
>+        pixel = src_pos[x];
>+      }
>+      uint32_t r = (pixel & red_mask) << red_shift;
>+      uint32_t g = (pixel & green_mask) << green_shift;
>+      uint32_t b = (pixel & blue_mask) << blue_shift;
>+      // Write as 32-bit RGB.
>+      dst_pos_32[x] =
>+          ((r >> 8) & 0xff0000) | ((g >> 16) & 0xff00) | ((b >> 24) & 0xff);
>+    }
>+    dst_pos += frame->stride();
>+    src_pos += src_stride;
>+  }
>+}
>+
>+}  // namespace
>+
>+XServerPixelBuffer::XServerPixelBuffer() {}
>+
>+XServerPixelBuffer::~XServerPixelBuffer() {
>+  Release();
>+}
>+
>+void XServerPixelBuffer::Release() {
>+  if (x_image_) {
>+    XDestroyImage(x_image_);
>+    x_image_ = nullptr;
>+  }
>+  if (x_shm_image_) {
>+    XDestroyImage(x_shm_image_);
>+    x_shm_image_ = nullptr;
>+  }
>+  if (shm_pixmap_) {
>+    XFreePixmap(display_, shm_pixmap_);
>+    shm_pixmap_ = 0;
>+  }
>+  if (shm_gc_) {
>+    XFreeGC(display_, shm_gc_);
>+    shm_gc_ = nullptr;
>+  }
>+
>+  ReleaseSharedMemorySegment();
>+
>+  window_ = 0;
>+}
>+
>+void XServerPixelBuffer::ReleaseSharedMemorySegment() {
>+  if (!shm_segment_info_)
>+    return;
>+  if (shm_segment_info_->shmaddr != nullptr)
>+    shmdt(shm_segment_info_->shmaddr);
>+  if (shm_segment_info_->shmid != -1)
>+    shmctl(shm_segment_info_->shmid, IPC_RMID, 0);
>+  delete shm_segment_info_;
>+  shm_segment_info_ = nullptr;
>+}
>+
>+bool XServerPixelBuffer::Init(Display* display, Window window) {
>+  Release();
>+  display_ = display;
>+
>+  XWindowAttributes attributes;
>+  if (!GetWindowRect(display_, window, &window_rect_, &attributes)) {
>+    return false;
>+  }
>+
>+  window_ = window;
>+  InitShm(attributes);
>+
>+  return true;
>+}
>+
>+void XServerPixelBuffer::InitShm(const XWindowAttributes& attributes) {
>+  Visual* default_visual = attributes.visual;
>+  int default_depth = attributes.depth;
>+
>+  int major, minor;
>+  Bool have_pixmaps;
>+  if (!XShmQueryVersion(display_, &major, &minor, &have_pixmaps)) {
>+    // Shared memory not supported. CaptureRect will use the XImage API instead.
>+    return;
>+  }
>+
>+  bool using_shm = false;
>+  shm_segment_info_ = new XShmSegmentInfo;
>+  shm_segment_info_->shmid = -1;
>+  shm_segment_info_->shmaddr = nullptr;
>+  shm_segment_info_->readOnly = False;
>+  x_shm_image_ = XShmCreateImage(display_, default_visual, default_depth,
>+                                 ZPixmap, 0, shm_segment_info_,
>+                                 window_rect_.width(), window_rect_.height());
>+  if (x_shm_image_) {
>+    shm_segment_info_->shmid =
>+        shmget(IPC_PRIVATE, x_shm_image_->bytes_per_line * x_shm_image_->height,
>+               IPC_CREAT | 0600);
>+    if (shm_segment_info_->shmid != -1) {
>+      void* shmat_result = shmat(shm_segment_info_->shmid, 0, 0);
>+      if (shmat_result != reinterpret_cast<void*>(-1)) {
>+        shm_segment_info_->shmaddr = reinterpret_cast<char*>(shmat_result);
>+        x_shm_image_->data = shm_segment_info_->shmaddr;
>+
>+        XErrorTrap error_trap(display_);
>+        using_shm = XShmAttach(display_, shm_segment_info_);
>+        XSync(display_, False);
>+        if (error_trap.GetLastErrorAndDisable() != 0)
>+          using_shm = false;
>+        if (using_shm) {
>+          RTC_LOG(LS_VERBOSE)
>+              << "Using X shared memory segment " << shm_segment_info_->shmid;
>+        }
>+      }
>+    } else {
>+      RTC_LOG(LS_WARNING) << "Failed to get shared memory segment. "
>+                             "Performance may be degraded.";
>+    }
>+  }
>+
>+  if (!using_shm) {
>+    RTC_LOG(LS_WARNING)
>+        << "Not using shared memory. Performance may be degraded.";
>+    ReleaseSharedMemorySegment();
>+    return;
>+  }
>+
>+  if (have_pixmaps)
>+    have_pixmaps = InitPixmaps(default_depth);
>+
>+  shmctl(shm_segment_info_->shmid, IPC_RMID, 0);
>+  shm_segment_info_->shmid = -1;
>+
>+  RTC_LOG(LS_VERBOSE) << "Using X shared memory extension v" << major << "."
>+                      << minor << " with" << (have_pixmaps ? "" : "out")
>+                      << " pixmaps.";
>+}
>+
>+bool XServerPixelBuffer::InitPixmaps(int depth) {
>+  if (XShmPixmapFormat(display_) != ZPixmap)
>+    return false;
>+
>+  {
>+    XErrorTrap error_trap(display_);
>+    shm_pixmap_ = XShmCreatePixmap(
>+        display_, window_, shm_segment_info_->shmaddr, shm_segment_info_,
>+        window_rect_.width(), window_rect_.height(), depth);
>+    XSync(display_, False);
>+    if (error_trap.GetLastErrorAndDisable() != 0) {
>+      // |shm_pixmap_| is not not valid because the request was not processed
>+      // by the X Server, so zero it.
>+      shm_pixmap_ = 0;
>+      return false;
>+    }
>+  }
>+
>+  {
>+    XErrorTrap error_trap(display_);
>+    XGCValues shm_gc_values;
>+    shm_gc_values.subwindow_mode = IncludeInferiors;
>+    shm_gc_values.graphics_exposures = False;
>+    shm_gc_ = XCreateGC(display_, window_,
>+                        GCSubwindowMode | GCGraphicsExposures, &shm_gc_values);
>+    XSync(display_, False);
>+    if (error_trap.GetLastErrorAndDisable() != 0) {
>+      XFreePixmap(display_, shm_pixmap_);
>+      shm_pixmap_ = 0;
>+      shm_gc_ = 0;  // See shm_pixmap_ comment above.
>+      return false;
>+    }
>+  }
>+
>+  return true;
>+}
>+
>+bool XServerPixelBuffer::IsWindowValid() const {
>+  XWindowAttributes attributes;
>+  {
>+    XErrorTrap error_trap(display_);
>+    if (!XGetWindowAttributes(display_, window_, &attributes) ||
>+        error_trap.GetLastErrorAndDisable() != 0) {
>+      return false;
>+    }
>+  }
>+  return true;
>+}
>+
>+void XServerPixelBuffer::Synchronize() {
>+  if (shm_segment_info_ && !shm_pixmap_) {
>+    // XShmGetImage can fail if the display is being reconfigured.
>+    XErrorTrap error_trap(display_);
>+    // XShmGetImage fails if the window is partially out of screen.
>+    xshm_get_image_succeeded_ =
>+        XShmGetImage(display_, window_, x_shm_image_, 0, 0, AllPlanes);
>+  }
>+}
>+
>+bool XServerPixelBuffer::CaptureRect(const DesktopRect& rect,
>+                                     DesktopFrame* frame) {
>+  RTC_DCHECK_LE(rect.right(), window_rect_.width());
>+  RTC_DCHECK_LE(rect.bottom(), window_rect_.height());
>+
>+  XImage* image;
>+  uint8_t* data;
>+
>+  if (shm_segment_info_ && (shm_pixmap_ || xshm_get_image_succeeded_)) {
>+    if (shm_pixmap_) {
>+      XCopyArea(display_, window_, shm_pixmap_, shm_gc_, rect.left(),
>+                rect.top(), rect.width(), rect.height(), rect.left(),
>+                rect.top());
>+      XSync(display_, False);
>+    }
>+
>+    image = x_shm_image_;
>+    data = reinterpret_cast<uint8_t*>(image->data) +
>+           rect.top() * image->bytes_per_line +
>+           rect.left() * image->bits_per_pixel / 8;
>+
>+  } else {
>+    if (x_image_)
>+      XDestroyImage(x_image_);
>+    x_image_ = XGetImage(display_, window_, rect.left(), rect.top(),
>+                         rect.width(), rect.height(), AllPlanes, ZPixmap);
>+    if (!x_image_)
>+      return false;
>+
>+    image = x_image_;
>+    data = reinterpret_cast<uint8_t*>(image->data);
>+  }
>+
>+  if (IsXImageRGBFormat(image)) {
>+    FastBlit(image, data, rect, frame);
>+  } else {
>+    SlowBlit(image, data, rect, frame);
>+  }
>+
>+  return true;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_server_pixel_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_server_pixel_buffer.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..a9398dac9968016981f3e3fea57a3e86e5bd49f2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/linux/x_server_pixel_buffer.h
>@@ -0,0 +1,84 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// Don't include this file in any .h files because it pulls in some X headers.
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_LINUX_X_SERVER_PIXEL_BUFFER_H_
>+#define MODULES_DESKTOP_CAPTURE_LINUX_X_SERVER_PIXEL_BUFFER_H_
>+
>+#include <X11/Xutil.h>
>+#include <X11/extensions/XShm.h>
>+
>+#include "modules/desktop_capture/desktop_geometry.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+class DesktopFrame;
>+
>+// A class to allow the X server's pixel buffer to be accessed as efficiently
>+// as possible.
>+class XServerPixelBuffer {
>+ public:
>+  XServerPixelBuffer();
>+  ~XServerPixelBuffer();
>+
>+  void Release();
>+
>+  // Allocate (or reallocate) the pixel buffer for |window|. Returns false in
>+  // case of an error (e.g. window doesn't exist).
>+  bool Init(Display* display, Window window);
>+
>+  bool is_initialized() { return window_ != 0; }
>+
>+  // Returns the size of the window the buffer was initialized for.
>+  DesktopSize window_size() { return window_rect_.size(); }
>+
>+  // Returns the rectangle of the window the buffer was initialized for.
>+  const DesktopRect& window_rect() { return window_rect_; }
>+
>+  // Returns true if the window can be found.
>+  bool IsWindowValid() const;
>+
>+  // If shared memory is being used without pixmaps, synchronize this pixel
>+  // buffer with the root window contents (otherwise, this is a no-op).
>+  // This is to avoid doing a full-screen capture for each individual
>+  // rectangle in the capture list, when it only needs to be done once at the
>+  // beginning.
>+  void Synchronize();
>+
>+  // Capture the specified rectangle and stores it in the |frame|. In the case
>+  // where the full-screen data is captured by Synchronize(), this simply
>+  // returns the pointer without doing any more work. The caller must ensure
>+  // that |rect| is not larger than window_size().
>+  bool CaptureRect(const DesktopRect& rect, DesktopFrame* frame);
>+
>+ private:
>+  void ReleaseSharedMemorySegment();
>+
>+  void InitShm(const XWindowAttributes& attributes);
>+  bool InitPixmaps(int depth);
>+
>+  Display* display_ = nullptr;
>+  Window window_ = 0;
>+  DesktopRect window_rect_;
>+  XImage* x_image_ = nullptr;
>+  XShmSegmentInfo* shm_segment_info_ = nullptr;
>+  XImage* x_shm_image_ = nullptr;
>+  Pixmap shm_pixmap_ = 0;
>+  GC shm_gc_ = nullptr;
>+  bool xshm_get_image_succeeded_ = false;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(XServerPixelBuffer);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_LINUX_X_SERVER_PIXEL_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc
>index d4364bedc2be5e3d6fdcb55db3fa73e1b24971aa..9a32f638d8fdb9f3abc1ec43442928021416b62a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc
>@@ -17,19 +17,12 @@
> 
> namespace webrtc {
> 
>-// The amount of time allowed for displays to reconfigure.
>-static const int64_t kDisplayConfigurationEventTimeoutMs = 10 * 1000;
>-
>-DesktopConfigurationMonitor::DesktopConfigurationMonitor()
>-    : display_configuration_capture_event_(EventWrapper::Create()) {
>+DesktopConfigurationMonitor::DesktopConfigurationMonitor() {
>   CGError err = CGDisplayRegisterReconfigurationCallback(
>       DesktopConfigurationMonitor::DisplaysReconfiguredCallback, this);
>-  if (err != kCGErrorSuccess) {
>+  if (err != kCGErrorSuccess)
>     RTC_LOG(LS_ERROR) << "CGDisplayRegisterReconfigurationCallback " << err;
>-    abort();
>-  }
>-  display_configuration_capture_event_->Set();
>-
>+  rtc::CritScope cs(&desktop_configuration_lock_);
>   desktop_configuration_ = MacDesktopConfiguration::GetCurrent(
>       MacDesktopConfiguration::TopLeftOrigin);
> }
>@@ -41,19 +34,13 @@ DesktopConfigurationMonitor::~DesktopConfigurationMonitor() {
>     RTC_LOG(LS_ERROR) << "CGDisplayRemoveReconfigurationCallback " << err;
> }
> 
>-void DesktopConfigurationMonitor::Lock() {
>-  if (!display_configuration_capture_event_->Wait(
>-          kDisplayConfigurationEventTimeoutMs)) {
>-    RTC_LOG_F(LS_ERROR) << "Event wait timed out.";
>-    abort();
>-  }
>-}
>-
>-void DesktopConfigurationMonitor::Unlock() {
>-  display_configuration_capture_event_->Set();
>+MacDesktopConfiguration DesktopConfigurationMonitor::desktop_configuration() {
>+  rtc::CritScope crit(&desktop_configuration_lock_);
>+  return desktop_configuration_;
> }
> 
> // static
>+// This method may be called on any system thread.
> void DesktopConfigurationMonitor::DisplaysReconfiguredCallback(
>     CGDirectDisplayID display,
>     CGDisplayChangeSummaryFlags flags,
>@@ -72,24 +59,15 @@ void DesktopConfigurationMonitor::DisplaysReconfigured(
>                    << flags;
> 
>   if (flags & kCGDisplayBeginConfigurationFlag) {
>-    if (reconfiguring_displays_.empty()) {
>-      // If this is the first display to start reconfiguring then wait on
>-      // |display_configuration_capture_event_| to block the capture thread
>-      // from accessing display memory until the reconfiguration completes.
>-      if (!display_configuration_capture_event_->Wait(
>-              kDisplayConfigurationEventTimeoutMs)) {
>-        RTC_LOG_F(LS_ERROR) << "Event wait timed out.";
>-        abort();
>-      }
>-    }
>     reconfiguring_displays_.insert(display);
>-  } else {
>-    reconfiguring_displays_.erase(display);
>-    if (reconfiguring_displays_.empty()) {
>-      desktop_configuration_ = MacDesktopConfiguration::GetCurrent(
>-          MacDesktopConfiguration::TopLeftOrigin);
>-      display_configuration_capture_event_->Set();
>-    }
>+    return;
>+  }
>+
>+  reconfiguring_displays_.erase(display);
>+  if (reconfiguring_displays_.empty()) {
>+    rtc::CritScope cs(&desktop_configuration_lock_);
>+    desktop_configuration_ = MacDesktopConfiguration::GetCurrent(
>+        MacDesktopConfiguration::TopLeftOrigin);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
>index 5215e5496ade8d2b012cb0c582f955bb4d02b634..cbf580b235301f6a37fd389efad9d75a1db7954e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
>@@ -19,25 +19,17 @@
> #include "api/refcountedbase.h"
> #include "modules/desktop_capture/mac/desktop_configuration.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/criticalsection.h"
> 
> namespace webrtc {
> 
>-class EventWrapper;
>-
> // The class provides functions to synchronize capturing and display
> // reconfiguring across threads, and the up-to-date MacDesktopConfiguration.
> class DesktopConfigurationMonitor : public rtc::RefCountedBase {
>  public:
>   DesktopConfigurationMonitor();
>-  // Acquires a lock on the current configuration.
>-  void Lock();
>-  // Releases the lock previously acquired.
>-  void Unlock();
>-  // Returns the current desktop configuration. Should only be called when the
>-  // lock has been acquired.
>-  const MacDesktopConfiguration& desktop_configuration() {
>-    return desktop_configuration_;
>-  }
>+  // Returns the current desktop configuration.
>+  MacDesktopConfiguration desktop_configuration();
> 
>  protected:
>   ~DesktopConfigurationMonitor() override;
>@@ -49,9 +41,10 @@ class DesktopConfigurationMonitor : public rtc::RefCountedBase {
>   void DisplaysReconfigured(CGDirectDisplayID display,
>                             CGDisplayChangeSummaryFlags flags);
> 
>+  rtc::CriticalSection desktop_configuration_lock_;
>+  MacDesktopConfiguration desktop_configuration_
>+      RTC_GUARDED_BY(&desktop_configuration_lock_);
>   std::set<CGDirectDisplayID> reconfiguring_displays_;
>-  MacDesktopConfiguration desktop_configuration_;
>-  std::unique_ptr<EventWrapper> display_configuration_capture_event_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(DesktopConfigurationMonitor);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/screen_capturer_mac.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/screen_capturer_mac.mm
>index 0d4fa072b5f6433af00822e2a376378f51f43f28..542d1c5f43fe74fa80d5fa52f39e704d9a438759 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/screen_capturer_mac.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/screen_capturer_mac.mm
>@@ -167,11 +167,7 @@ ScreenCapturerMac::~ScreenCapturerMac() {
> 
> bool ScreenCapturerMac::Init() {
>   TRACE_EVENT0("webrtc", "ScreenCapturerMac::Init");
>-
>-  desktop_config_monitor_->Lock();
>   desktop_config_ = desktop_config_monitor_->desktop_configuration();
>-  desktop_config_monitor_->Unlock();
>-
>   return true;
> }
> 
>@@ -207,7 +203,6 @@ void ScreenCapturerMac::CaptureFrame() {
>   queue_.MoveToNextFrame();
>   RTC_DCHECK(!queue_.current_frame() || !queue_.current_frame()->IsShared());
> 
>-  desktop_config_monitor_->Lock();
>   MacDesktopConfiguration new_config = desktop_config_monitor_->desktop_configuration();
>   if (!desktop_config_.Equals(new_config)) {
>     desktop_config_ = new_config;
>@@ -234,7 +229,6 @@ void ScreenCapturerMac::CaptureFrame() {
>   DesktopFrame* current_frame = queue_.current_frame();
> 
>   if (!CgBlit(*current_frame, region)) {
>-    desktop_config_monitor_->Unlock();
>     callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>     return;
>   }
>@@ -256,10 +250,6 @@ void ScreenCapturerMac::CaptureFrame() {
> 
>   helper_.set_size_most_recent(new_frame->size());
> 
>-  // Signal that we are done capturing data from the display framebuffer,
>-  // and accessing display structures.
>-  desktop_config_monitor_->Unlock();
>-
>   new_frame->set_capture_time_ms((rtc::TimeNanos() - capture_start_time_nanos) /
>                                  rtc::kNumNanosecsPerMillisec);
>   callback_->OnCaptureResult(Result::SUCCESS, std::move(new_frame));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_linux.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..def344ef408a6962706c1e94084d207daa687cbc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_linux.cc
>@@ -0,0 +1,51 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/mouse_cursor_monitor.h"
>+
>+#if defined(USE_X11)
>+#include "modules/desktop_capture/linux/mouse_cursor_monitor_x11.h"
>+#endif  // defined(USE_X11)
>+
>+namespace webrtc {
>+
>+// static
>+MouseCursorMonitor* MouseCursorMonitor::CreateForWindow(
>+    const DesktopCaptureOptions& options,
>+    WindowId window) {
>+#if defined(USE_X11)
>+  return MouseCursorMonitorX11::CreateForWindow(options, window);
>+#else
>+  return nullptr;
>+#endif  // defined(USE_X11)
>+}
>+
>+// static
>+MouseCursorMonitor* MouseCursorMonitor::CreateForScreen(
>+    const DesktopCaptureOptions& options,
>+    ScreenId screen) {
>+#if defined(USE_X11)
>+  return MouseCursorMonitorX11::CreateForScreen(options, screen);
>+#else
>+  return nullptr;
>+#endif  // defined(USE_X11)
>+}
>+
>+// static
>+std::unique_ptr<MouseCursorMonitor> MouseCursorMonitor::Create(
>+    const DesktopCaptureOptions& options) {
>+#if defined(USE_X11)
>+  return MouseCursorMonitorX11::Create(options);
>+#else
>+  return nullptr;
>+#endif  // defined(USE_X11)
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
>index de8876438f8a9860e9b540ce9e8fdd165c785fb7..3402a5ffbbcbe040dd153baf4f4cddcbfc6e3997 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
>@@ -117,10 +117,8 @@ void MouseCursorMonitorMac::Capture() {
> 
>   DesktopVector position(gc_position.x, gc_position.y);
> 
>-  configuration_monitor_->Lock();
>   MacDesktopConfiguration configuration =
>       configuration_monitor_->desktop_configuration();
>-  configuration_monitor_->Unlock();
>   float scale = GetScaleFactorAtPosition(configuration, position);
> 
>   CaptureImage(scale);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc
>deleted file mode 100644
>index 41749bd2841bb5f6aadb276bb429e50a484b6a4d..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc
>+++ /dev/null
>@@ -1,277 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-
>-#include "modules/desktop_capture/mouse_cursor_monitor.h"
>-
>-#include <X11/Xlib.h>
>-#include <X11/Xutil.h>
>-#include <X11/extensions/Xfixes.h>
>-
>-#include "modules/desktop_capture/desktop_capture_options.h"
>-#include "modules/desktop_capture/desktop_capture_types.h"
>-#include "modules/desktop_capture/desktop_frame.h"
>-#include "modules/desktop_capture/mouse_cursor.h"
>-#include "modules/desktop_capture/x11/x_error_trap.h"
>-#include "rtc_base/logging.h"
>-
>-namespace {
>-
>-// WindowCapturer returns window IDs of X11 windows with WM_STATE attribute.
>-// These windows may not be immediate children of the root window, because
>-// window managers may re-parent them to add decorations. However,
>-// XQueryPointer() expects to be passed children of the root. This function
>-// searches up the list of the windows to find the root child that corresponds
>-// to |window|.
>-Window GetTopLevelWindow(Display* display, Window window) {
>-  while (true) {
>-    // If the window is in WithdrawnState then look at all of its children.
>-    ::Window root, parent;
>-    ::Window* children;
>-    unsigned int num_children;
>-    if (!XQueryTree(display, window, &root, &parent, &children,
>-                    &num_children)) {
>-      RTC_LOG(LS_ERROR) << "Failed to query for child windows although window"
>-                        << "does not have a valid WM_STATE.";
>-      return None;
>-    }
>-    if (children)
>-      XFree(children);
>-
>-    if (parent == root)
>-      break;
>-
>-    window = parent;
>-  }
>-
>-  return window;
>-}
>-
>-}  // namespace
>-
>-namespace webrtc {
>-
>-class MouseCursorMonitorX11 : public MouseCursorMonitor,
>-                              public SharedXDisplay::XEventHandler {
>- public:
>-  MouseCursorMonitorX11(const DesktopCaptureOptions& options, Window window);
>-  ~MouseCursorMonitorX11() override;
>-
>-  void Init(Callback* callback, Mode mode) override;
>-  void Capture() override;
>-
>- private:
>-  // SharedXDisplay::XEventHandler interface.
>-  bool HandleXEvent(const XEvent& event) override;
>-
>-  Display* display() { return x_display_->display(); }
>-
>-  // Captures current cursor shape and stores it in |cursor_shape_|.
>-  void CaptureCursor();
>-
>-  rtc::scoped_refptr<SharedXDisplay> x_display_;
>-  Callback* callback_;
>-  Mode mode_;
>-  Window window_;
>-
>-  bool have_xfixes_;
>-  int xfixes_event_base_;
>-  int xfixes_error_base_;
>-
>-  std::unique_ptr<MouseCursor> cursor_shape_;
>-};
>-
>-MouseCursorMonitorX11::MouseCursorMonitorX11(
>-    const DesktopCaptureOptions& options,
>-    Window window)
>-    : x_display_(options.x_display()),
>-      callback_(NULL),
>-      mode_(SHAPE_AND_POSITION),
>-      window_(window),
>-      have_xfixes_(false),
>-      xfixes_event_base_(-1),
>-      xfixes_error_base_(-1) {
>-  // Set a default initial cursor shape in case XFixes is not present.
>-  const int kSize = 5;
>-  std::unique_ptr<DesktopFrame> default_cursor(
>-      new BasicDesktopFrame(DesktopSize(kSize, kSize)));
>-  const uint8_t pixels[kSize * kSize] = {
>-      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0xff,
>-      0x00, 0x00, 0xff, 0xff, 0xff, 0x00, 0x00, 0xff, 0xff,
>-      0xff, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
>-  uint8_t* ptr = default_cursor->data();
>-  for (int y = 0; y < kSize; ++y) {
>-    for (int x = 0; x < kSize; ++x) {
>-      *ptr++ = pixels[kSize * y + x];
>-      *ptr++ = pixels[kSize * y + x];
>-      *ptr++ = pixels[kSize * y + x];
>-      *ptr++ = 0xff;
>-    }
>-  }
>-  DesktopVector hotspot(2, 2);
>-  cursor_shape_.reset(new MouseCursor(default_cursor.release(), hotspot));
>-}
>-
>-MouseCursorMonitorX11::~MouseCursorMonitorX11() {
>-  if (have_xfixes_) {
>-    x_display_->RemoveEventHandler(xfixes_event_base_ + XFixesCursorNotify,
>-                                   this);
>-  }
>-}
>-
>-void MouseCursorMonitorX11::Init(Callback* callback, Mode mode) {
>-  // Init can be called only once per instance of MouseCursorMonitor.
>-  RTC_DCHECK(!callback_);
>-  RTC_DCHECK(callback);
>-
>-  callback_ = callback;
>-  mode_ = mode;
>-
>-  have_xfixes_ =
>-      XFixesQueryExtension(display(), &xfixes_event_base_, &xfixes_error_base_);
>-
>-  if (have_xfixes_) {
>-    // Register for changes to the cursor shape.
>-    XFixesSelectCursorInput(display(), window_, XFixesDisplayCursorNotifyMask);
>-    x_display_->AddEventHandler(xfixes_event_base_ + XFixesCursorNotify, this);
>-
>-    CaptureCursor();
>-  } else {
>-    RTC_LOG(LS_INFO) << "X server does not support XFixes.";
>-  }
>-}
>-
>-void MouseCursorMonitorX11::Capture() {
>-  RTC_DCHECK(callback_);
>-
>-  // Process X11 events in case XFixes has sent cursor notification.
>-  x_display_->ProcessPendingXEvents();
>-
>-  // cursor_shape_| is set only if we were notified of a cursor shape change.
>-  if (cursor_shape_.get())
>-    callback_->OnMouseCursor(cursor_shape_.release());
>-
>-  // Get cursor position if necessary.
>-  if (mode_ == SHAPE_AND_POSITION) {
>-    int root_x;
>-    int root_y;
>-    int win_x;
>-    int win_y;
>-    Window root_window;
>-    Window child_window;
>-    unsigned int mask;
>-
>-    XErrorTrap error_trap(display());
>-    Bool result = XQueryPointer(display(), window_, &root_window, &child_window,
>-                                &root_x, &root_y, &win_x, &win_y, &mask);
>-    CursorState state;
>-    if (!result || error_trap.GetLastErrorAndDisable() != 0) {
>-      state = OUTSIDE;
>-    } else {
>-      // In screen mode (window_ == root_window) the mouse is always inside.
>-      // XQueryPointer() sets |child_window| to None if the cursor is outside
>-      // |window_|.
>-      state =
>-          (window_ == root_window || child_window != None) ? INSIDE : OUTSIDE;
>-    }
>-
>-    // As the comments to GetTopLevelWindow() above indicate, in window capture,
>-    // the cursor position capture happens in |window_|, while the frame catpure
>-    // happens in |child_window|. These two windows are not alwyas same, as
>-    // window manager may add some decorations to the |window_|. So translate
>-    // the coordinate in |window_| to the coordinate space of |child_window|.
>-    if (window_ != root_window && state == INSIDE) {
>-      int translated_x, translated_y;
>-      Window unused;
>-      if (XTranslateCoordinates(display(), window_, child_window, win_x, win_y,
>-                                &translated_x, &translated_y, &unused)) {
>-        win_x = translated_x;
>-        win_y = translated_y;
>-      }
>-    }
>-
>-    // X11 always starts the coordinate from (0, 0), so we do not need to
>-    // translate here.
>-    callback_->OnMouseCursorPosition(DesktopVector(root_x, root_y));
>-  }
>-}
>-
>-bool MouseCursorMonitorX11::HandleXEvent(const XEvent& event) {
>-  if (have_xfixes_ && event.type == xfixes_event_base_ + XFixesCursorNotify) {
>-    const XFixesCursorNotifyEvent* cursor_event =
>-        reinterpret_cast<const XFixesCursorNotifyEvent*>(&event);
>-    if (cursor_event->subtype == XFixesDisplayCursorNotify) {
>-      CaptureCursor();
>-    }
>-    // Return false, even if the event has been handled, because there might be
>-    // other listeners for cursor notifications.
>-  }
>-  return false;
>-}
>-
>-void MouseCursorMonitorX11::CaptureCursor() {
>-  RTC_DCHECK(have_xfixes_);
>-
>-  XFixesCursorImage* img;
>-  {
>-    XErrorTrap error_trap(display());
>-    img = XFixesGetCursorImage(display());
>-    if (!img || error_trap.GetLastErrorAndDisable() != 0)
>-      return;
>-  }
>-
>-  std::unique_ptr<DesktopFrame> image(
>-      new BasicDesktopFrame(DesktopSize(img->width, img->height)));
>-
>-  // Xlib stores 32-bit data in longs, even if longs are 64-bits long.
>-  unsigned long* src = img->pixels;
>-  uint32_t* dst = reinterpret_cast<uint32_t*>(image->data());
>-  uint32_t* dst_end = dst + (img->width * img->height);
>-  while (dst < dst_end) {
>-    *dst++ = static_cast<uint32_t>(*src++);
>-  }
>-
>-  DesktopVector hotspot(std::min(img->width, img->xhot),
>-                        std::min(img->height, img->yhot));
>-
>-  XFree(img);
>-
>-  cursor_shape_.reset(new MouseCursor(image.release(), hotspot));
>-}
>-
>-// static
>-MouseCursorMonitor* MouseCursorMonitor::CreateForWindow(
>-    const DesktopCaptureOptions& options,
>-    WindowId window) {
>-  if (!options.x_display())
>-    return NULL;
>-  window = GetTopLevelWindow(options.x_display()->display(), window);
>-  if (window == None)
>-    return NULL;
>-  return new MouseCursorMonitorX11(options, window);
>-}
>-
>-MouseCursorMonitor* MouseCursorMonitor::CreateForScreen(
>-    const DesktopCaptureOptions& options,
>-    ScreenId screen) {
>-  if (!options.x_display())
>-    return NULL;
>-  return new MouseCursorMonitorX11(
>-      options, DefaultRootWindow(options.x_display()->display()));
>-}
>-
>-std::unique_ptr<MouseCursorMonitor> MouseCursorMonitor::Create(
>-    const DesktopCaptureOptions& options) {
>-  return std::unique_ptr<MouseCursorMonitor>(
>-      CreateForScreen(options, kFullDesktopScreenId));
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_linux.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..cf8a9dd0e0db4008e17f43aa2507e74910672b46
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_linux.cc
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
>+
>+#if defined(WEBRTC_USE_PIPEWIRE)
>+#include "modules/desktop_capture/linux/screen_capturer_pipewire.h"
>+#endif  // defined(WEBRTC_USE_PIPEWIRE)
>+
>+#if defined(USE_X11)
>+#include "modules/desktop_capture/linux/screen_capturer_x11.h"
>+#endif  // defined(USE_X11)
>+
>+namespace webrtc {
>+
>+// static
>+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
>+    const DesktopCaptureOptions& options) {
>+#if defined(WEBRTC_USE_PIPEWIRE)
>+  if (options.allow_pipewire() && DesktopCapturer::IsRunningUnderWayland()) {
>+    return ScreenCapturerPipeWire::CreateRawScreenCapturer(options);
>+  }
>+#endif  // defined(WEBRTC_USE_PIPEWIRE)
>+
>+#if defined(USE_X11)
>+  return ScreenCapturerX11::CreateRawScreenCapturer(options);
>+#endif  // defined(USE_X11)
>+
>+  return nullptr;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_x11.cc
>deleted file mode 100644
>index e95fccc2082a97042aaffb8ddd6c2434df497e70..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_x11.cc
>+++ /dev/null
>@@ -1,425 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <string.h>
>-
>-#include <memory>
>-#include <set>
>-#include <utility>
>-
>-#include <X11/Xlib.h>
>-#include <X11/Xutil.h>
>-#include <X11/extensions/Xdamage.h>
>-#include <X11/extensions/Xfixes.h>
>-
>-#include "modules/desktop_capture/desktop_capture_options.h"
>-#include "modules/desktop_capture/desktop_capturer.h"
>-#include "modules/desktop_capture/desktop_frame.h"
>-#include "modules/desktop_capture/screen_capture_frame_queue.h"
>-#include "modules/desktop_capture/screen_capturer_helper.h"
>-#include "modules/desktop_capture/shared_desktop_frame.h"
>-#include "modules/desktop_capture/x11/x_server_pixel_buffer.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/timeutils.h"
>-#include "rtc_base/trace_event.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-// A class to perform video frame capturing for Linux.
>-//
>-// If XDamage is used, this class sets DesktopFrame::updated_region() according
>-// to the areas reported by XDamage. Otherwise this class does not detect
>-// DesktopFrame::updated_region(), the field is always set to the entire frame
>-// rectangle. ScreenCapturerDifferWrapper should be used if that functionality
>-// is necessary.
>-class ScreenCapturerLinux : public DesktopCapturer,
>-                            public SharedXDisplay::XEventHandler {
>- public:
>-  ScreenCapturerLinux();
>-  ~ScreenCapturerLinux() override;
>-
>-  // TODO(ajwong): Do we really want this to be synchronous?
>-  bool Init(const DesktopCaptureOptions& options);
>-
>-  // DesktopCapturer interface.
>-  void Start(Callback* delegate) override;
>-  void CaptureFrame() override;
>-  bool GetSourceList(SourceList* sources) override;
>-  bool SelectSource(SourceId id) override;
>-
>- private:
>-  Display* display() { return options_.x_display()->display(); }
>-
>-  // SharedXDisplay::XEventHandler interface.
>-  bool HandleXEvent(const XEvent& event) override;
>-
>-  void InitXDamage();
>-
>-  // Capture screen pixels to the current buffer in the queue. In the DAMAGE
>-  // case, the ScreenCapturerHelper already holds the list of invalid rectangles
>-  // from HandleXEvent(). In the non-DAMAGE case, this captures the
>-  // whole screen, then calculates some invalid rectangles that include any
>-  // differences between this and the previous capture.
>-  std::unique_ptr<DesktopFrame> CaptureScreen();
>-
>-  // Called when the screen configuration is changed.
>-  void ScreenConfigurationChanged();
>-
>-  // Synchronize the current buffer with |last_buffer_|, by copying pixels from
>-  // the area of |last_invalid_rects|.
>-  // Note this only works on the assumption that kNumBuffers == 2, as
>-  // |last_invalid_rects| holds the differences from the previous buffer and
>-  // the one prior to that (which will then be the current buffer).
>-  void SynchronizeFrame();
>-
>-  void DeinitXlib();
>-
>-  DesktopCaptureOptions options_;
>-
>-  Callback* callback_ = nullptr;
>-
>-  // X11 graphics context.
>-  GC gc_ = nullptr;
>-  Window root_window_ = BadValue;
>-
>-  // XFixes.
>-  bool has_xfixes_ = false;
>-  int xfixes_event_base_ = -1;
>-  int xfixes_error_base_ = -1;
>-
>-  // XDamage information.
>-  bool use_damage_ = false;
>-  Damage damage_handle_ = 0;
>-  int damage_event_base_ = -1;
>-  int damage_error_base_ = -1;
>-  XserverRegion damage_region_ = 0;
>-
>-  // Access to the X Server's pixel buffer.
>-  XServerPixelBuffer x_server_pixel_buffer_;
>-
>-  // A thread-safe list of invalid rectangles, and the size of the most
>-  // recently captured screen.
>-  ScreenCapturerHelper helper_;
>-
>-  // Queue of the frames buffers.
>-  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
>-
>-  // Invalid region from the previous capture. This is used to synchronize the
>-  // current with the last buffer used.
>-  DesktopRegion last_invalid_region_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerLinux);
>-};
>-
>-ScreenCapturerLinux::ScreenCapturerLinux() {
>-  helper_.SetLogGridSize(4);
>-}
>-
>-ScreenCapturerLinux::~ScreenCapturerLinux() {
>-  options_.x_display()->RemoveEventHandler(ConfigureNotify, this);
>-  if (use_damage_) {
>-    options_.x_display()->RemoveEventHandler(damage_event_base_ + XDamageNotify,
>-                                             this);
>-  }
>-  DeinitXlib();
>-}
>-
>-bool ScreenCapturerLinux::Init(const DesktopCaptureOptions& options) {
>-  TRACE_EVENT0("webrtc", "ScreenCapturerLinux::Init");
>-  options_ = options;
>-
>-  root_window_ = RootWindow(display(), DefaultScreen(display()));
>-  if (root_window_ == BadValue) {
>-    RTC_LOG(LS_ERROR) << "Unable to get the root window";
>-    DeinitXlib();
>-    return false;
>-  }
>-
>-  gc_ = XCreateGC(display(), root_window_, 0, NULL);
>-  if (gc_ == NULL) {
>-    RTC_LOG(LS_ERROR) << "Unable to get graphics context";
>-    DeinitXlib();
>-    return false;
>-  }
>-
>-  options_.x_display()->AddEventHandler(ConfigureNotify, this);
>-
>-  // Check for XFixes extension. This is required for cursor shape
>-  // notifications, and for our use of XDamage.
>-  if (XFixesQueryExtension(display(), &xfixes_event_base_,
>-                           &xfixes_error_base_)) {
>-    has_xfixes_ = true;
>-  } else {
>-    RTC_LOG(LS_INFO) << "X server does not support XFixes.";
>-  }
>-
>-  // Register for changes to the dimensions of the root window.
>-  XSelectInput(display(), root_window_, StructureNotifyMask);
>-
>-  if (!x_server_pixel_buffer_.Init(display(), DefaultRootWindow(display()))) {
>-    RTC_LOG(LS_ERROR) << "Failed to initialize pixel buffer.";
>-    return false;
>-  }
>-
>-  if (options_.use_update_notifications()) {
>-    InitXDamage();
>-  }
>-
>-  return true;
>-}
>-
>-void ScreenCapturerLinux::InitXDamage() {
>-  // Our use of XDamage requires XFixes.
>-  if (!has_xfixes_) {
>-    return;
>-  }
>-
>-  // Check for XDamage extension.
>-  if (!XDamageQueryExtension(display(), &damage_event_base_,
>-                             &damage_error_base_)) {
>-    RTC_LOG(LS_INFO) << "X server does not support XDamage.";
>-    return;
>-  }
>-
>-  // TODO(lambroslambrou): Disable DAMAGE in situations where it is known
>-  // to fail, such as when Desktop Effects are enabled, with graphics
>-  // drivers (nVidia, ATI) that fail to report DAMAGE notifications
>-  // properly.
>-
>-  // Request notifications every time the screen becomes damaged.
>-  damage_handle_ =
>-      XDamageCreate(display(), root_window_, XDamageReportNonEmpty);
>-  if (!damage_handle_) {
>-    RTC_LOG(LS_ERROR) << "Unable to initialize XDamage.";
>-    return;
>-  }
>-
>-  // Create an XFixes server-side region to collate damage into.
>-  damage_region_ = XFixesCreateRegion(display(), 0, 0);
>-  if (!damage_region_) {
>-    XDamageDestroy(display(), damage_handle_);
>-    RTC_LOG(LS_ERROR) << "Unable to create XFixes region.";
>-    return;
>-  }
>-
>-  options_.x_display()->AddEventHandler(damage_event_base_ + XDamageNotify,
>-                                        this);
>-
>-  use_damage_ = true;
>-  RTC_LOG(LS_INFO) << "Using XDamage extension.";
>-}
>-
>-void ScreenCapturerLinux::Start(Callback* callback) {
>-  RTC_DCHECK(!callback_);
>-  RTC_DCHECK(callback);
>-
>-  callback_ = callback;
>-}
>-
>-void ScreenCapturerLinux::CaptureFrame() {
>-  TRACE_EVENT0("webrtc", "ScreenCapturerLinux::CaptureFrame");
>-  int64_t capture_start_time_nanos = rtc::TimeNanos();
>-
>-  queue_.MoveToNextFrame();
>-  RTC_DCHECK(!queue_.current_frame() || !queue_.current_frame()->IsShared());
>-
>-  // Process XEvents for XDamage and cursor shape tracking.
>-  options_.x_display()->ProcessPendingXEvents();
>-
>-  // ProcessPendingXEvents() may call ScreenConfigurationChanged() which
>-  // reinitializes |x_server_pixel_buffer_|. Check if the pixel buffer is still
>-  // in a good shape.
>-  if (!x_server_pixel_buffer_.is_initialized()) {
>-    // We failed to initialize pixel buffer.
>-    RTC_LOG(LS_ERROR) << "Pixel buffer is not initialized.";
>-    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>-    return;
>-  }
>-
>-  // If the current frame is from an older generation then allocate a new one.
>-  // Note that we can't reallocate other buffers at this point, since the caller
>-  // may still be reading from them.
>-  if (!queue_.current_frame()) {
>-    queue_.ReplaceCurrentFrame(
>-        SharedDesktopFrame::Wrap(std::unique_ptr<DesktopFrame>(
>-            new BasicDesktopFrame(x_server_pixel_buffer_.window_size()))));
>-  }
>-
>-  std::unique_ptr<DesktopFrame> result = CaptureScreen();
>-  if (!result) {
>-    RTC_LOG(LS_WARNING) << "Temporarily failed to capture screen.";
>-    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
>-    return;
>-  }
>-
>-  last_invalid_region_ = result->updated_region();
>-  result->set_capture_time_ms((rtc::TimeNanos() - capture_start_time_nanos) /
>-                              rtc::kNumNanosecsPerMillisec);
>-  callback_->OnCaptureResult(Result::SUCCESS, std::move(result));
>-}
>-
>-bool ScreenCapturerLinux::GetSourceList(SourceList* sources) {
>-  RTC_DCHECK(sources->size() == 0);
>-  // TODO(jiayl): implement screen enumeration.
>-  sources->push_back({0});
>-  return true;
>-}
>-
>-bool ScreenCapturerLinux::SelectSource(SourceId id) {
>-  // TODO(jiayl): implement screen selection.
>-  return true;
>-}
>-
>-bool ScreenCapturerLinux::HandleXEvent(const XEvent& event) {
>-  if (use_damage_ && (event.type == damage_event_base_ + XDamageNotify)) {
>-    const XDamageNotifyEvent* damage_event =
>-        reinterpret_cast<const XDamageNotifyEvent*>(&event);
>-    if (damage_event->damage != damage_handle_)
>-      return false;
>-    RTC_DCHECK(damage_event->level == XDamageReportNonEmpty);
>-    return true;
>-  } else if (event.type == ConfigureNotify) {
>-    ScreenConfigurationChanged();
>-    return true;
>-  }
>-  return false;
>-}
>-
>-std::unique_ptr<DesktopFrame> ScreenCapturerLinux::CaptureScreen() {
>-  std::unique_ptr<SharedDesktopFrame> frame = queue_.current_frame()->Share();
>-  RTC_DCHECK(x_server_pixel_buffer_.window_size().equals(frame->size()));
>-
>-  // Pass the screen size to the helper, so it can clip the invalid region if it
>-  // expands that region to a grid.
>-  helper_.set_size_most_recent(frame->size());
>-
>-  // In the DAMAGE case, ensure the frame is up-to-date with the previous frame
>-  // if any.  If there isn't a previous frame, that means a screen-resolution
>-  // change occurred, and |invalid_rects| will be updated to include the whole
>-  // screen.
>-  if (use_damage_ && queue_.previous_frame())
>-    SynchronizeFrame();
>-
>-  DesktopRegion* updated_region = frame->mutable_updated_region();
>-
>-  x_server_pixel_buffer_.Synchronize();
>-  if (use_damage_ && queue_.previous_frame()) {
>-    // Atomically fetch and clear the damage region.
>-    XDamageSubtract(display(), damage_handle_, None, damage_region_);
>-    int rects_num = 0;
>-    XRectangle bounds;
>-    XRectangle* rects = XFixesFetchRegionAndBounds(display(), damage_region_,
>-                                                   &rects_num, &bounds);
>-    for (int i = 0; i < rects_num; ++i) {
>-      updated_region->AddRect(DesktopRect::MakeXYWH(
>-          rects[i].x, rects[i].y, rects[i].width, rects[i].height));
>-    }
>-    XFree(rects);
>-    helper_.InvalidateRegion(*updated_region);
>-
>-    // Capture the damaged portions of the desktop.
>-    helper_.TakeInvalidRegion(updated_region);
>-
>-    // Clip the damaged portions to the current screen size, just in case some
>-    // spurious XDamage notifications were received for a previous (larger)
>-    // screen size.
>-    updated_region->IntersectWith(
>-        DesktopRect::MakeSize(x_server_pixel_buffer_.window_size()));
>-
>-    for (DesktopRegion::Iterator it(*updated_region); !it.IsAtEnd();
>-         it.Advance()) {
>-      if (!x_server_pixel_buffer_.CaptureRect(it.rect(), frame.get()))
>-        return nullptr;
>-    }
>-  } else {
>-    // Doing full-screen polling, or this is the first capture after a
>-    // screen-resolution change.  In either case, need a full-screen capture.
>-    DesktopRect screen_rect = DesktopRect::MakeSize(frame->size());
>-    if (!x_server_pixel_buffer_.CaptureRect(screen_rect, frame.get()))
>-      return nullptr;
>-    updated_region->SetRect(screen_rect);
>-  }
>-
>-  return std::move(frame);
>-}
>-
>-void ScreenCapturerLinux::ScreenConfigurationChanged() {
>-  TRACE_EVENT0("webrtc", "ScreenCapturerLinux::ScreenConfigurationChanged");
>-  // Make sure the frame buffers will be reallocated.
>-  queue_.Reset();
>-
>-  helper_.ClearInvalidRegion();
>-  if (!x_server_pixel_buffer_.Init(display(), DefaultRootWindow(display()))) {
>-    RTC_LOG(LS_ERROR) << "Failed to initialize pixel buffer after screen "
>-                         "configuration change.";
>-  }
>-}
>-
>-void ScreenCapturerLinux::SynchronizeFrame() {
>-  // Synchronize the current buffer with the previous one since we do not
>-  // capture the entire desktop. Note that encoder may be reading from the
>-  // previous buffer at this time so thread access complaints are false
>-  // positives.
>-
>-  // TODO(hclam): We can reduce the amount of copying here by subtracting
>-  // |capturer_helper_|s region from |last_invalid_region_|.
>-  // http://crbug.com/92354
>-  RTC_DCHECK(queue_.previous_frame());
>-
>-  DesktopFrame* current = queue_.current_frame();
>-  DesktopFrame* last = queue_.previous_frame();
>-  RTC_DCHECK(current != last);
>-  for (DesktopRegion::Iterator it(last_invalid_region_); !it.IsAtEnd();
>-       it.Advance()) {
>-    current->CopyPixelsFrom(*last, it.rect().top_left(), it.rect());
>-  }
>-}
>-
>-void ScreenCapturerLinux::DeinitXlib() {
>-  if (gc_) {
>-    XFreeGC(display(), gc_);
>-    gc_ = nullptr;
>-  }
>-
>-  x_server_pixel_buffer_.Release();
>-
>-  if (display()) {
>-    if (damage_handle_) {
>-      XDamageDestroy(display(), damage_handle_);
>-      damage_handle_ = 0;
>-    }
>-
>-    if (damage_region_) {
>-      XFixesDestroyRegion(display(), damage_region_);
>-      damage_region_ = 0;
>-    }
>-  }
>-}
>-
>-}  // namespace
>-
>-// static
>-std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
>-    const DesktopCaptureOptions& options) {
>-  if (!options.x_display())
>-    return nullptr;
>-
>-  std::unique_ptr<ScreenCapturerLinux> capturer(new ScreenCapturerLinux());
>-  if (!capturer.get()->Init(options)) {
>-    return nullptr;
>-  }
>-
>-  return std::move(capturer);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_linux.cc
>index 39ceca7dc61799c46b7322518e6651a356d51017..2ab5e74891c62af9074e6eda486397d67f82e725 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_linux.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_linux.cc
>@@ -14,9 +14,9 @@
> #include <memory>
> 
> #include "absl/memory/memory.h"
>+#include "modules/desktop_capture/linux/shared_x_display.h"
> #include "modules/desktop_capture/screen_drawer.h"
> #include "modules/desktop_capture/screen_drawer_lock_posix.h"
>-#include "modules/desktop_capture/x11/shared_x_display.h"
> #include "rtc_base/checks.h"
> #include "system_wrappers/include/sleep.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_unittest.cc
>index 5f0f3cc5733518d57abc9a86ed0d7cb4eb2fbd3a..d48fa89c0b5bc0a621eef6992a5887e2f1c4f9bb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_unittest.cc
>@@ -140,7 +140,12 @@ TEST(ScreenDrawerTest, DISABLED_DrawRectangles) {
>   SleepMs(10000);
> }
> 
>-TEST(ScreenDrawerTest, TwoScreenDrawerLocks) {
>+#if defined(THREAD_SANITIZER)  // bugs.webrtc.org/10019
>+#define MAYBE_TwoScreenDrawerLocks DISABLED_TwoScreenDrawerLocks
>+#else
>+#define MAYBE_TwoScreenDrawerLocks TwoScreenDrawerLocks
>+#endif
>+TEST(ScreenDrawerTest, MAYBE_TwoScreenDrawerLocks) {
> #if defined(WEBRTC_POSIX)
>   // ScreenDrawerLockPosix won't be able to unlink the named semaphore. So use a
>   // different semaphore name here to avoid deadlock.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_desktop_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_desktop_frame.h
>index f70508cee15b6906c24e6d136632e3b3a524c0fd..ea12c000f2a64528f5c08fc73c27d12e198bb91d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_desktop_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_desktop_frame.h
>@@ -16,12 +16,13 @@
> #include "rtc_base/refcount.h"
> #include "rtc_base/refcountedobject.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
> // SharedDesktopFrame is a DesktopFrame that may have multiple instances all
> // sharing the same buffer.
>-class SharedDesktopFrame : public DesktopFrame {
>+class RTC_EXPORT SharedDesktopFrame : public DesktopFrame {
>  public:
>   ~SharedDesktopFrame() override;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_linux.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..82359e50c2db19c748e417bceec6b588e4ed1038
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_linux.cc
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
>+
>+#if defined(WEBRTC_USE_PIPEWIRE)
>+#include "modules/desktop_capture/linux/window_capturer_pipewire.h"
>+#endif  // defined(WEBRTC_USE_PIPEWIRE)
>+
>+#if defined(USE_X11)
>+#include "modules/desktop_capture/linux/window_capturer_x11.h"
>+#endif  // defined(USE_X11)
>+
>+namespace webrtc {
>+
>+// static
>+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawWindowCapturer(
>+    const DesktopCaptureOptions& options) {
>+#if defined(WEBRTC_USE_PIPEWIRE)
>+  if (options.allow_pipewire() && DesktopCapturer::IsRunningUnderWayland()) {
>+    return WindowCapturerPipeWire::CreateRawWindowCapturer(options);
>+  }
>+#endif  // defined(WEBRTC_USE_PIPEWIRE)
>+
>+#if defined(USE_X11)
>+  return WindowCapturerX11::CreateRawWindowCapturer(options);
>+#endif  // defined(USE_X11)
>+
>+  return nullptr;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_mac.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_mac.mm
>index 95e622a77a1c3cbc1b90b26722b7d7710866c3ee..b0647f041b6d86085175de0e75b82420a1fb4a71 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_mac.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_mac.mm
>@@ -140,9 +140,7 @@ bool WindowCapturerMac::FocusOnSelectedSource() {
> bool WindowCapturerMac::IsOccluded(const DesktopVector& pos) {
>   DesktopVector sys_pos = pos;
>   if (configuration_monitor_) {
>-    configuration_monitor_->Lock();
>     auto configuration = configuration_monitor_->desktop_configuration();
>-    configuration_monitor_->Unlock();
>     sys_pos = pos.add(configuration.bounds.top_left());
>   }
>   return window_finder_.GetWindowUnderPoint(sys_pos) != window_id_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_x11.cc
>deleted file mode 100644
>index c4ca3ae5e8c40096a945e9f120ffc9bbba58465b..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_x11.cc
>+++ /dev/null
>@@ -1,288 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <string.h>
>-
>-#include <X11/Xutil.h>
>-#include <X11/extensions/Xcomposite.h>
>-#include <X11/extensions/Xrender.h>
>-
>-#include <utility>
>-
>-#include "modules/desktop_capture/desktop_capture_options.h"
>-#include "modules/desktop_capture/desktop_capturer.h"
>-#include "modules/desktop_capture/desktop_frame.h"
>-#include "modules/desktop_capture/window_finder_x11.h"
>-#include "modules/desktop_capture/x11/shared_x_display.h"
>-#include "modules/desktop_capture/x11/window_list_utils.h"
>-#include "modules/desktop_capture/x11/x_atom_cache.h"
>-#include "modules/desktop_capture/x11/x_server_pixel_buffer.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/scoped_ref_ptr.h"
>-#include "rtc_base/trace_event.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-
>-class WindowCapturerLinux : public DesktopCapturer,
>-                            public SharedXDisplay::XEventHandler {
>- public:
>-  WindowCapturerLinux(const DesktopCaptureOptions& options);
>-  ~WindowCapturerLinux() override;
>-
>-  // DesktopCapturer interface.
>-  void Start(Callback* callback) override;
>-  void CaptureFrame() override;
>-  bool GetSourceList(SourceList* sources) override;
>-  bool SelectSource(SourceId id) override;
>-  bool FocusOnSelectedSource() override;
>-  bool IsOccluded(const DesktopVector& pos) override;
>-
>-  // SharedXDisplay::XEventHandler interface.
>-  bool HandleXEvent(const XEvent& event) override;
>-
>- private:
>-  Display* display() { return x_display_->display(); }
>-
>-  // Returns window title for the specified X |window|.
>-  bool GetWindowTitle(::Window window, std::string* title);
>-
>-  Callback* callback_ = nullptr;
>-
>-  rtc::scoped_refptr<SharedXDisplay> x_display_;
>-
>-  bool has_composite_extension_ = false;
>-
>-  ::Window selected_window_ = 0;
>-  XServerPixelBuffer x_server_pixel_buffer_;
>-  XAtomCache atom_cache_;
>-  WindowFinderX11 window_finder_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(WindowCapturerLinux);
>-};
>-
>-WindowCapturerLinux::WindowCapturerLinux(const DesktopCaptureOptions& options)
>-    : x_display_(options.x_display()),
>-      atom_cache_(display()),
>-      window_finder_(&atom_cache_) {
>-  int event_base, error_base, major_version, minor_version;
>-  if (XCompositeQueryExtension(display(), &event_base, &error_base) &&
>-      XCompositeQueryVersion(display(), &major_version, &minor_version) &&
>-      // XCompositeNameWindowPixmap() requires version 0.2
>-      (major_version > 0 || minor_version >= 2)) {
>-    has_composite_extension_ = true;
>-  } else {
>-    RTC_LOG(LS_INFO) << "Xcomposite extension not available or too old.";
>-  }
>-
>-  x_display_->AddEventHandler(ConfigureNotify, this);
>-}
>-
>-WindowCapturerLinux::~WindowCapturerLinux() {
>-  x_display_->RemoveEventHandler(ConfigureNotify, this);
>-}
>-
>-bool WindowCapturerLinux::GetSourceList(SourceList* sources) {
>-  return GetWindowList(&atom_cache_, [this, sources](::Window window) {
>-    Source w;
>-    w.id = window;
>-    if (this->GetWindowTitle(window, &w.title)) {
>-      sources->push_back(w);
>-    }
>-    return true;
>-  });
>-}
>-
>-bool WindowCapturerLinux::SelectSource(SourceId id) {
>-  if (!x_server_pixel_buffer_.Init(display(), id))
>-    return false;
>-
>-  // Tell the X server to send us window resizing events.
>-  XSelectInput(display(), id, StructureNotifyMask);
>-
>-  selected_window_ = id;
>-
>-  // In addition to needing X11 server-side support for Xcomposite, it actually
>-  // needs to be turned on for the window. If the user has modern
>-  // hardware/drivers but isn't using a compositing window manager, that won't
>-  // be the case. Here we automatically turn it on.
>-
>-  // Redirect drawing to an offscreen buffer (ie, turn on compositing). X11
>-  // remembers who has requested this and will turn it off for us when we exit.
>-  XCompositeRedirectWindow(display(), id, CompositeRedirectAutomatic);
>-
>-  return true;
>-}
>-
>-bool WindowCapturerLinux::FocusOnSelectedSource() {
>-  if (!selected_window_)
>-    return false;
>-
>-  unsigned int num_children;
>-  ::Window* children;
>-  ::Window parent;
>-  ::Window root;
>-  // Find the root window to pass event to.
>-  int status = XQueryTree(display(), selected_window_, &root, &parent,
>-                          &children, &num_children);
>-  if (status == 0) {
>-    RTC_LOG(LS_ERROR) << "Failed to query for the root window.";
>-    return false;
>-  }
>-
>-  if (children)
>-    XFree(children);
>-
>-  XRaiseWindow(display(), selected_window_);
>-
>-  // Some window managers (e.g., metacity in GNOME) consider it illegal to
>-  // raise a window without also giving it input focus with
>-  // _NET_ACTIVE_WINDOW, so XRaiseWindow() on its own isn't enough.
>-  Atom atom = XInternAtom(display(), "_NET_ACTIVE_WINDOW", True);
>-  if (atom != None) {
>-    XEvent xev;
>-    xev.xclient.type = ClientMessage;
>-    xev.xclient.serial = 0;
>-    xev.xclient.send_event = True;
>-    xev.xclient.window = selected_window_;
>-    xev.xclient.message_type = atom;
>-
>-    // The format member is set to 8, 16, or 32 and specifies whether the
>-    // data should be viewed as a list of bytes, shorts, or longs.
>-    xev.xclient.format = 32;
>-
>-    memset(xev.xclient.data.l, 0, sizeof(xev.xclient.data.l));
>-
>-    XSendEvent(display(), root, False,
>-               SubstructureRedirectMask | SubstructureNotifyMask, &xev);
>-  }
>-  XFlush(display());
>-  return true;
>-}
>-
>-void WindowCapturerLinux::Start(Callback* callback) {
>-  RTC_DCHECK(!callback_);
>-  RTC_DCHECK(callback);
>-
>-  callback_ = callback;
>-}
>-
>-void WindowCapturerLinux::CaptureFrame() {
>-  TRACE_EVENT0("webrtc", "WindowCapturerLinux::CaptureFrame");
>-
>-  if (!x_server_pixel_buffer_.IsWindowValid()) {
>-    RTC_LOG(LS_ERROR) << "The window is no longer valid.";
>-    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>-    return;
>-  }
>-
>-  x_display_->ProcessPendingXEvents();
>-
>-  if (!has_composite_extension_) {
>-    // Without the Xcomposite extension we capture when the whole window is
>-    // visible on screen and not covered by any other window. This is not
>-    // something we want so instead, just bail out.
>-    RTC_LOG(LS_ERROR) << "No Xcomposite extension detected.";
>-    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>-    return;
>-  }
>-
>-  if (GetWindowState(&atom_cache_, selected_window_) == IconicState) {
>-    // Window is in minimized. Return a 1x1 frame as same as OSX/Win does.
>-    std::unique_ptr<DesktopFrame> frame(
>-        new BasicDesktopFrame(DesktopSize(1, 1)));
>-    callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
>-    return;
>-  }
>-
>-  std::unique_ptr<DesktopFrame> frame(
>-      new BasicDesktopFrame(x_server_pixel_buffer_.window_size()));
>-
>-  x_server_pixel_buffer_.Synchronize();
>-  if (!x_server_pixel_buffer_.CaptureRect(DesktopRect::MakeSize(frame->size()),
>-                                          frame.get())) {
>-    RTC_LOG(LS_WARNING) << "Temporarily failed to capture winodw.";
>-    callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
>-    return;
>-  }
>-
>-  frame->mutable_updated_region()->SetRect(
>-      DesktopRect::MakeSize(frame->size()));
>-  frame->set_top_left(x_server_pixel_buffer_.window_rect().top_left());
>-
>-  callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
>-}
>-
>-bool WindowCapturerLinux::IsOccluded(const DesktopVector& pos) {
>-  return window_finder_.GetWindowUnderPoint(pos) !=
>-         static_cast<WindowId>(selected_window_);
>-}
>-
>-bool WindowCapturerLinux::HandleXEvent(const XEvent& event) {
>-  if (event.type == ConfigureNotify) {
>-    XConfigureEvent xce = event.xconfigure;
>-    if (xce.window == selected_window_) {
>-      if (!DesktopRectFromXAttributes(xce).equals(
>-              x_server_pixel_buffer_.window_rect())) {
>-        if (!x_server_pixel_buffer_.Init(display(), selected_window_)) {
>-          RTC_LOG(LS_ERROR)
>-              << "Failed to initialize pixel buffer after resizing.";
>-        }
>-      }
>-    }
>-  }
>-
>-  // Always returns false, so other observers can still receive the events.
>-  return false;
>-}
>-
>-bool WindowCapturerLinux::GetWindowTitle(::Window window, std::string* title) {
>-  int status;
>-  bool result = false;
>-  XTextProperty window_name;
>-  window_name.value = nullptr;
>-  if (window) {
>-    status = XGetWMName(display(), window, &window_name);
>-    if (status && window_name.value && window_name.nitems) {
>-      int cnt;
>-      char** list = nullptr;
>-      status =
>-          Xutf8TextPropertyToTextList(display(), &window_name, &list, &cnt);
>-      if (status >= Success && cnt && *list) {
>-        if (cnt > 1) {
>-          RTC_LOG(LS_INFO) << "Window has " << cnt
>-                           << " text properties, only using the first one.";
>-        }
>-        *title = *list;
>-        result = true;
>-      }
>-      if (list)
>-        XFreeStringList(list);
>-    }
>-    if (window_name.value)
>-      XFree(window_name.value);
>-  }
>-  return result;
>-}
>-
>-}  // namespace
>-
>-// static
>-std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawWindowCapturer(
>-    const DesktopCaptureOptions& options) {
>-  if (!options.x_display())
>-    return nullptr;
>-  return std::unique_ptr<DesktopCapturer>(new WindowCapturerLinux(options));
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_unittest.cc
>index 88327aaac5483ad95199810177e875e2f6838120..df9067ef2cb597335732aefe491cea24a4e38b72 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_unittest.cc
>@@ -21,8 +21,8 @@
> 
> #if defined(USE_X11)
> #include "absl/memory/memory.h"
>-#include "modules/desktop_capture/x11/shared_x_display.h"
>-#include "modules/desktop_capture/x11/x_atom_cache.h"
>+#include "modules/desktop_capture/linux/shared_x_display.h"
>+#include "modules/desktop_capture/linux/x_atom_cache.h"
> #endif
> 
> #if defined(WEBRTC_WIN)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_x11.cc
>deleted file mode 100644
>index 03694326d2fd840fb4e932a290e8dada17aed52b..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_x11.cc
>+++ /dev/null
>@@ -1,49 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/desktop_capture/window_finder_x11.h"
>-
>-#include "absl/memory/memory.h"
>-#include "modules/desktop_capture/x11/window_list_utils.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-
>-WindowFinderX11::WindowFinderX11(XAtomCache* cache) : cache_(cache) {
>-  RTC_DCHECK(cache_);
>-}
>-
>-WindowFinderX11::~WindowFinderX11() = default;
>-
>-WindowId WindowFinderX11::GetWindowUnderPoint(DesktopVector point) {
>-  WindowId id = kNullWindowId;
>-  GetWindowList(cache_, [&id, this, point](::Window window) {
>-    DesktopRect rect;
>-    if (GetWindowRect(this->cache_->display(), window, &rect) &&
>-        rect.Contains(point)) {
>-      id = window;
>-      return false;
>-    }
>-    return true;
>-  });
>-  return id;
>-}
>-
>-// static
>-std::unique_ptr<WindowFinder> WindowFinder::Create(
>-    const WindowFinder::Options& options) {
>-  if (options.cache == nullptr) {
>-    return nullptr;
>-  }
>-
>-  return absl::make_unique<WindowFinderX11>(options.cache);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_x11.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_x11.h
>deleted file mode 100644
>index 38c30c4670bac82e3995c9e1177ca14cba6d7ed2..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_x11.h
>+++ /dev/null
>@@ -1,35 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_X11_H_
>-#define MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_X11_H_
>-
>-#include "modules/desktop_capture/window_finder.h"
>-
>-namespace webrtc {
>-
>-class XAtomCache;
>-
>-// The implementation of WindowFinder for X11.
>-class WindowFinderX11 final : public WindowFinder {
>- public:
>-  explicit WindowFinderX11(XAtomCache* cache);
>-  ~WindowFinderX11() override;
>-
>-  // WindowFinder implementation.
>-  WindowId GetWindowUnderPoint(DesktopVector point) override;
>-
>- private:
>-  XAtomCache* const cache_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_DESKTOP_CAPTURE_WINDOW_FINDER_X11_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/shared_x_display.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/shared_x_display.cc
>deleted file mode 100644
>index db6a64b81d9b751ba95a3caf2be9b2e8719aa1d1..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/shared_x_display.cc
>+++ /dev/null
>@@ -1,89 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/desktop_capture/x11/shared_x_display.h"
>-
>-#include <X11/Xlib.h>
>-
>-#include <algorithm>
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-
>-namespace webrtc {
>-
>-SharedXDisplay::SharedXDisplay(Display* display) : display_(display) {
>-  RTC_DCHECK(display_);
>-}
>-
>-SharedXDisplay::~SharedXDisplay() {
>-  RTC_DCHECK(event_handlers_.empty());
>-  XCloseDisplay(display_);
>-}
>-
>-// static
>-rtc::scoped_refptr<SharedXDisplay> SharedXDisplay::Create(
>-    const std::string& display_name) {
>-  Display* display =
>-      XOpenDisplay(display_name.empty() ? NULL : display_name.c_str());
>-  if (!display) {
>-    RTC_LOG(LS_ERROR) << "Unable to open display";
>-    return NULL;
>-  }
>-  return new SharedXDisplay(display);
>-}
>-
>-// static
>-rtc::scoped_refptr<SharedXDisplay> SharedXDisplay::CreateDefault() {
>-  return Create(std::string());
>-}
>-
>-void SharedXDisplay::AddEventHandler(int type, XEventHandler* handler) {
>-  event_handlers_[type].push_back(handler);
>-}
>-
>-void SharedXDisplay::RemoveEventHandler(int type, XEventHandler* handler) {
>-  EventHandlersMap::iterator handlers = event_handlers_.find(type);
>-  if (handlers == event_handlers_.end())
>-    return;
>-
>-  std::vector<XEventHandler*>::iterator new_end =
>-      std::remove(handlers->second.begin(), handlers->second.end(), handler);
>-  handlers->second.erase(new_end, handlers->second.end());
>-
>-  // Check if no handlers left for this event.
>-  if (handlers->second.empty())
>-    event_handlers_.erase(handlers);
>-}
>-
>-void SharedXDisplay::ProcessPendingXEvents() {
>-  // Hold reference to |this| to prevent it from being destroyed while
>-  // processing events.
>-  rtc::scoped_refptr<SharedXDisplay> self(this);
>-
>-  // Find the number of events that are outstanding "now."  We don't just loop
>-  // on XPending because we want to guarantee this terminates.
>-  int events_to_process = XPending(display());
>-  XEvent e;
>-
>-  for (int i = 0; i < events_to_process; i++) {
>-    XNextEvent(display(), &e);
>-    EventHandlersMap::iterator handlers = event_handlers_.find(e.type);
>-    if (handlers == event_handlers_.end())
>-      continue;
>-    for (std::vector<XEventHandler*>::iterator it = handlers->second.begin();
>-         it != handlers->second.end(); ++it) {
>-      if ((*it)->HandleXEvent(e))
>-        break;
>-    }
>-  }
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/shared_x_display.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/shared_x_display.h
>deleted file mode 100644
>index 4d93b3b175144d14c7d363b4c1b3c62787c48cef..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/shared_x_display.h
>+++ /dev/null
>@@ -1,81 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_DESKTOP_CAPTURE_X11_SHARED_X_DISPLAY_H_
>-#define MODULES_DESKTOP_CAPTURE_X11_SHARED_X_DISPLAY_H_
>-
>-#include <map>
>-#include <vector>
>-
>-#include <string>
>-
>-#include "api/refcountedbase.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/scoped_ref_ptr.h"
>-
>-// Including Xlib.h will involve evil defines (Bool, Status, True, False), which
>-// easily conflict with other headers.
>-typedef struct _XDisplay Display;
>-typedef union _XEvent XEvent;
>-
>-namespace webrtc {
>-
>-// A ref-counted object to store XDisplay connection.
>-class SharedXDisplay : public rtc::RefCountedBase {
>- public:
>-  class XEventHandler {
>-   public:
>-    virtual ~XEventHandler() {}
>-
>-    // Processes XEvent. Returns true if the event has been handled.
>-    virtual bool HandleXEvent(const XEvent& event) = 0;
>-  };
>-
>-  // Takes ownership of |display|.
>-  explicit SharedXDisplay(Display* display);
>-
>-  // Creates a new X11 Display for the |display_name|. NULL is returned if X11
>-  // connection failed. Equivalent to CreateDefault() when |display_name| is
>-  // empty.
>-  static rtc::scoped_refptr<SharedXDisplay> Create(
>-      const std::string& display_name);
>-
>-  // Creates X11 Display connection for the default display (e.g. specified in
>-  // DISPLAY). NULL is returned if X11 connection failed.
>-  static rtc::scoped_refptr<SharedXDisplay> CreateDefault();
>-
>-  Display* display() { return display_; }
>-
>-  // Adds a new event |handler| for XEvent's of |type|.
>-  void AddEventHandler(int type, XEventHandler* handler);
>-
>-  // Removes event |handler| added using |AddEventHandler|. Doesn't do anything
>-  // if |handler| is not registered.
>-  void RemoveEventHandler(int type, XEventHandler* handler);
>-
>-  // Processes pending XEvents, calling corresponding event handlers.
>-  void ProcessPendingXEvents();
>-
>- protected:
>-  ~SharedXDisplay() override;
>-
>- private:
>-  typedef std::map<int, std::vector<XEventHandler*> > EventHandlersMap;
>-
>-  Display* display_;
>-
>-  EventHandlersMap event_handlers_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(SharedXDisplay);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_DESKTOP_CAPTURE_X11_SHARED_X_DISPLAY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.cc
>deleted file mode 100644
>index fe070d887824c0af65a9f657917ee9286cbaca5c..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.cc
>+++ /dev/null
>@@ -1,246 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/desktop_capture/x11/window_list_utils.h"
>-
>-#include <X11/Xatom.h>
>-#include <X11/Xlib.h>
>-#include <X11/Xutil.h>
>-#include <string.h>
>-
>-#include <algorithm>
>-
>-#include "modules/desktop_capture/x11/x_error_trap.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/logging.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-
>-class DeferXFree {
>- public:
>-  explicit DeferXFree(void* data) : data_(data) {}
>-  ~DeferXFree();
>-
>- private:
>-  void* const data_;
>-};
>-
>-DeferXFree::~DeferXFree() {
>-  if (data_)
>-    XFree(data_);
>-}
>-
>-// Convenience wrapper for XGetWindowProperty() results.
>-template <class PropertyType>
>-class XWindowProperty {
>- public:
>-  XWindowProperty(Display* display, Window window, Atom property) {
>-    const int kBitsPerByte = 8;
>-    Atom actual_type;
>-    int actual_format;
>-    unsigned long bytes_after;  // NOLINT: type required by XGetWindowProperty
>-    int status = XGetWindowProperty(
>-        display, window, property, 0L, ~0L, False, AnyPropertyType,
>-        &actual_type, &actual_format, &size_, &bytes_after, &data_);
>-    if (status != Success) {
>-      data_ = nullptr;
>-      return;
>-    }
>-    if (sizeof(PropertyType) * kBitsPerByte != actual_format) {
>-      size_ = 0;
>-      return;
>-    }
>-
>-    is_valid_ = true;
>-  }
>-
>-  ~XWindowProperty() {
>-    if (data_)
>-      XFree(data_);
>-  }
>-
>-  // True if we got properly value successfully.
>-  bool is_valid() const { return is_valid_; }
>-
>-  // Size and value of the property.
>-  size_t size() const { return size_; }
>-  const PropertyType* data() const {
>-    return reinterpret_cast<PropertyType*>(data_);
>-  }
>-  PropertyType* data() { return reinterpret_cast<PropertyType*>(data_); }
>-
>- private:
>-  bool is_valid_ = false;
>-  unsigned long size_ = 0;  // NOLINT: type required by XGetWindowProperty
>-  unsigned char* data_ = nullptr;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(XWindowProperty);
>-};
>-
>-// Iterates through |window| hierarchy to find first visible window, i.e. one
>-// that has WM_STATE property set to NormalState.
>-// See http://tronche.com/gui/x/icccm/sec-4.html#s-4.1.3.1 .
>-::Window GetApplicationWindow(XAtomCache* cache, ::Window window) {
>-  int32_t state = GetWindowState(cache, window);
>-  if (state == NormalState) {
>-    // Window has WM_STATE==NormalState. Return it.
>-    return window;
>-  } else if (state == IconicState) {
>-    // Window is in minimized. Skip it.
>-    return 0;
>-  }
>-
>-  RTC_DCHECK_EQ(state, WithdrawnState);
>-  // If the window is in WithdrawnState then look at all of its children.
>-  ::Window root, parent;
>-  ::Window* children;
>-  unsigned int num_children;
>-  if (!XQueryTree(cache->display(), window, &root, &parent, &children,
>-                  &num_children)) {
>-    RTC_LOG(LS_ERROR) << "Failed to query for child windows although window"
>-                      << "does not have a valid WM_STATE.";
>-    return 0;
>-  }
>-  ::Window app_window = 0;
>-  for (unsigned int i = 0; i < num_children; ++i) {
>-    app_window = GetApplicationWindow(cache, children[i]);
>-    if (app_window)
>-      break;
>-  }
>-
>-  if (children)
>-    XFree(children);
>-  return app_window;
>-}
>-
>-// Returns true if the |window| is a desktop element.
>-bool IsDesktopElement(XAtomCache* cache, ::Window window) {
>-  RTC_DCHECK(cache);
>-  if (window == 0)
>-    return false;
>-
>-  // First look for _NET_WM_WINDOW_TYPE. The standard
>-  // (http://standards.freedesktop.org/wm-spec/latest/ar01s05.html#id2760306)
>-  // says this hint *should* be present on all windows, and we use the existence
>-  // of _NET_WM_WINDOW_TYPE_NORMAL in the property to indicate a window is not
>-  // a desktop element (that is, only "normal" windows should be shareable).
>-  XWindowProperty<uint32_t> window_type(cache->display(), window,
>-                                        cache->WindowType());
>-  if (window_type.is_valid() && window_type.size() > 0) {
>-    uint32_t* end = window_type.data() + window_type.size();
>-    bool is_normal =
>-        (end != std::find(window_type.data(), end, cache->WindowTypeNormal()));
>-    return !is_normal;
>-  }
>-
>-  // Fall back on using the hint.
>-  XClassHint class_hint;
>-  Status status = XGetClassHint(cache->display(), window, &class_hint);
>-  if (status == 0) {
>-    // No hints, assume this is a normal application window.
>-    return false;
>-  }
>-
>-  DeferXFree free_res_name(class_hint.res_name);
>-  DeferXFree free_res_class(class_hint.res_class);
>-  return strcmp("gnome-panel", class_hint.res_name) == 0 ||
>-         strcmp("desktop_window", class_hint.res_name) == 0;
>-}
>-
>-}  // namespace
>-
>-int32_t GetWindowState(XAtomCache* cache, ::Window window) {
>-  // Get WM_STATE property of the window.
>-  XWindowProperty<uint32_t> window_state(cache->display(), window,
>-                                         cache->WmState());
>-
>-  // WM_STATE is considered to be set to WithdrawnState when it missing.
>-  return window_state.is_valid() ? *window_state.data() : WithdrawnState;
>-}
>-
>-bool GetWindowList(XAtomCache* cache,
>-                   rtc::FunctionView<bool(::Window)> on_window) {
>-  RTC_DCHECK(cache);
>-  RTC_DCHECK(on_window);
>-  ::Display* const display = cache->display();
>-
>-  int failed_screens = 0;
>-  const int num_screens = XScreenCount(display);
>-  for (int screen = 0; screen < num_screens; screen++) {
>-    ::Window root_window = XRootWindow(display, screen);
>-    ::Window parent;
>-    ::Window* children;
>-    unsigned int num_children;
>-    {
>-      XErrorTrap error_trap(display);
>-      if (XQueryTree(display, root_window, &root_window, &parent, &children,
>-                     &num_children) == 0 ||
>-          error_trap.GetLastErrorAndDisable() != 0) {
>-        failed_screens++;
>-        RTC_LOG(LS_ERROR) << "Failed to query for child windows for screen "
>-                          << screen;
>-        continue;
>-      }
>-    }
>-
>-    DeferXFree free_children(children);
>-
>-    for (unsigned int i = 0; i < num_children; i++) {
>-      // Iterates in reverse order to return windows from front to back.
>-      ::Window app_window =
>-          GetApplicationWindow(cache, children[num_children - 1 - i]);
>-      if (app_window && !IsDesktopElement(cache, app_window)) {
>-        if (!on_window(app_window)) {
>-          return true;
>-        }
>-      }
>-    }
>-  }
>-
>-  return failed_screens < num_screens;
>-}
>-
>-bool GetWindowRect(::Display* display,
>-                   ::Window window,
>-                   DesktopRect* rect,
>-                   XWindowAttributes* attributes /* = nullptr */) {
>-  XWindowAttributes local_attributes;
>-  int offset_x;
>-  int offset_y;
>-  if (attributes == nullptr) {
>-    attributes = &local_attributes;
>-  }
>-
>-  {
>-    XErrorTrap error_trap(display);
>-    if (!XGetWindowAttributes(display, window, attributes) ||
>-        error_trap.GetLastErrorAndDisable() != 0) {
>-      return false;
>-    }
>-  }
>-  *rect = DesktopRectFromXAttributes(*attributes);
>-
>-  {
>-    XErrorTrap error_trap(display);
>-    ::Window child;
>-    if (!XTranslateCoordinates(display, window, attributes->root, -rect->left(),
>-                               -rect->top(), &offset_x, &offset_y, &child) ||
>-        error_trap.GetLastErrorAndDisable() != 0) {
>-      return false;
>-    }
>-  }
>-  rect->Translate(offset_x, offset_y);
>-  return true;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.h
>deleted file mode 100644
>index 72d2a70d2107ddda919db78d91130de2d4813e2e..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.h
>+++ /dev/null
>@@ -1,54 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_DESKTOP_CAPTURE_X11_WINDOW_LIST_UTILS_H_
>-#define MODULES_DESKTOP_CAPTURE_X11_WINDOW_LIST_UTILS_H_
>-
>-#include <X11/Xlib.h>
>-
>-#include "modules/desktop_capture/desktop_geometry.h"
>-#include "modules/desktop_capture/x11/x_atom_cache.h"
>-#include "rtc_base/function_view.h"
>-
>-namespace webrtc {
>-
>-// Synchronously iterates all on-screen windows in |cache|.display() in
>-// decreasing z-order and sends them one-by-one to |on_window| function before
>-// GetWindowList() returns. If |on_window| returns false, this function ignores
>-// other windows and returns immediately. GetWindowList() returns false if
>-// native APIs failed. If multiple screens are attached to the |display|, this
>-// function returns false only when native APIs failed on all screens. Menus,
>-// panels and minimized windows will be ignored.
>-bool GetWindowList(XAtomCache* cache,
>-                   rtc::FunctionView<bool(::Window)> on_window);
>-
>-// Returns WM_STATE property of the |window|. This function returns
>-// WithdrawnState if the |window| is missing.
>-int32_t GetWindowState(XAtomCache* cache, ::Window window);
>-
>-// Returns the rectangle of the |window| in the coordinates of |display|. This
>-// function returns false if native APIs failed. If |attributes| is provided, it
>-// will be filled with the attributes of |window|. The |rect| is in system
>-// coordinate, i.e. the primary monitor always starts from (0, 0).
>-bool GetWindowRect(::Display* display,
>-                   ::Window window,
>-                   DesktopRect* rect,
>-                   XWindowAttributes* attributes = nullptr);
>-
>-// Creates a DesktopRect from |attributes|.
>-template <typename T>
>-DesktopRect DesktopRectFromXAttributes(const T& attributes) {
>-  return DesktopRect::MakeXYWH(attributes.x, attributes.y, attributes.width,
>-                               attributes.height);
>-}
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_DESKTOP_CAPTURE_X11_WINDOW_LIST_UTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_atom_cache.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_atom_cache.cc
>deleted file mode 100644
>index fa3a1fdebddd5d91bb99a87a8d0dc6b16c3d58a7..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_atom_cache.cc
>+++ /dev/null
>@@ -1,47 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/desktop_capture/x11/x_atom_cache.h"
>-
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-
>-XAtomCache::XAtomCache(::Display* display) : display_(display) {
>-  RTC_DCHECK(display_);
>-}
>-
>-XAtomCache::~XAtomCache() = default;
>-
>-::Display* XAtomCache::display() const {
>-  return display_;
>-}
>-
>-Atom XAtomCache::WmState() {
>-  return CreateIfNotExist(&wm_state_, "WM_STATE");
>-}
>-
>-Atom XAtomCache::WindowType() {
>-  return CreateIfNotExist(&window_type_, "_NET_WM_WINDOW_TYPE");
>-}
>-
>-Atom XAtomCache::WindowTypeNormal() {
>-  return CreateIfNotExist(&window_type_normal_, "_NET_WM_WINDOW_TYPE_NORMAL");
>-}
>-
>-Atom XAtomCache::CreateIfNotExist(Atom* atom, const char* name) {
>-  RTC_DCHECK(atom);
>-  if (*atom == None) {
>-    *atom = XInternAtom(display(), name, True);
>-  }
>-  return *atom;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_atom_cache.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_atom_cache.h
>deleted file mode 100644
>index 5246c2bb5c9a86b8b0db1c66e95b6e2415d7e686..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_atom_cache.h
>+++ /dev/null
>@@ -1,43 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_DESKTOP_CAPTURE_X11_X_ATOM_CACHE_H_
>-#define MODULES_DESKTOP_CAPTURE_X11_X_ATOM_CACHE_H_
>-
>-#include <X11/Xatom.h>
>-#include <X11/Xlib.h>
>-
>-namespace webrtc {
>-
>-// A cache of Atom. Each Atom object is created on demand.
>-class XAtomCache final {
>- public:
>-  explicit XAtomCache(::Display* display);
>-  ~XAtomCache();
>-
>-  ::Display* display() const;
>-
>-  Atom WmState();
>-  Atom WindowType();
>-  Atom WindowTypeNormal();
>-
>- private:
>-  // If |*atom| is None, this function uses XInternAtom() to retrieve an Atom.
>-  Atom CreateIfNotExist(Atom* atom, const char* name);
>-
>-  ::Display* const display_;
>-  Atom wm_state_ = None;
>-  Atom window_type_ = None;
>-  Atom window_type_normal_ = None;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_DESKTOP_CAPTURE_X11_X_ATOM_CACHE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_error_trap.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_error_trap.cc
>deleted file mode 100644
>index 6559c3dbe9f5166564ae8abb608bc2ee277b39d7..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_error_trap.cc
>+++ /dev/null
>@@ -1,68 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/desktop_capture/x11/x_error_trap.h"
>-
>-#include <assert.h>
>-
>-#if defined(TOOLKIT_GTK)
>-#include <gdk/gdk.h>
>-#endif  // !defined(TOOLKIT_GTK)
>-
>-namespace webrtc {
>-
>-namespace {
>-
>-#if !defined(TOOLKIT_GTK)
>-
>-// TODO(sergeyu): This code is not thread safe. Fix it. Bug 2202.
>-static bool g_xserver_error_trap_enabled = false;
>-static int g_last_xserver_error_code = 0;
>-
>-int XServerErrorHandler(Display* display, XErrorEvent* error_event) {
>-  assert(g_xserver_error_trap_enabled);
>-  g_last_xserver_error_code = error_event->error_code;
>-  return 0;
>-}
>-
>-#endif  // !defined(TOOLKIT_GTK)
>-
>-}  // namespace
>-
>-XErrorTrap::XErrorTrap(Display* display)
>-    : original_error_handler_(NULL), enabled_(true) {
>-#if defined(TOOLKIT_GTK)
>-  gdk_error_trap_push();
>-#else   // !defined(TOOLKIT_GTK)
>-  assert(!g_xserver_error_trap_enabled);
>-  original_error_handler_ = XSetErrorHandler(&XServerErrorHandler);
>-  g_xserver_error_trap_enabled = true;
>-  g_last_xserver_error_code = 0;
>-#endif  // !defined(TOOLKIT_GTK)
>-}
>-
>-int XErrorTrap::GetLastErrorAndDisable() {
>-  enabled_ = false;
>-#if defined(TOOLKIT_GTK)
>-  return gdk_error_trap_push();
>-#else   // !defined(TOOLKIT_GTK)
>-  assert(g_xserver_error_trap_enabled);
>-  XSetErrorHandler(original_error_handler_);
>-  g_xserver_error_trap_enabled = false;
>-  return g_last_xserver_error_code;
>-#endif  // !defined(TOOLKIT_GTK)
>-}
>-
>-XErrorTrap::~XErrorTrap() {
>-  if (enabled_)
>-    GetLastErrorAndDisable();
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_error_trap.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_error_trap.h
>deleted file mode 100644
>index ea9ebf670b7ea10fe7528a6207a14f19a6fec90f..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_error_trap.h
>+++ /dev/null
>@@ -1,39 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_DESKTOP_CAPTURE_X11_X_ERROR_TRAP_H_
>-#define MODULES_DESKTOP_CAPTURE_X11_X_ERROR_TRAP_H_
>-
>-#include <X11/Xlib.h>
>-
>-#include "rtc_base/constructormagic.h"
>-
>-namespace webrtc {
>-
>-// Helper class that registers X Window error handler. Caller can use
>-// GetLastErrorAndDisable() to get the last error that was caught, if any.
>-class XErrorTrap {
>- public:
>-  explicit XErrorTrap(Display* display);
>-  ~XErrorTrap();
>-
>-  // Returns last error and removes unregisters the error handler.
>-  int GetLastErrorAndDisable();
>-
>- private:
>-  XErrorHandler original_error_handler_;
>-  bool enabled_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(XErrorTrap);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_DESKTOP_CAPTURE_X11_X_ERROR_TRAP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc
>deleted file mode 100644
>index e327d3fc7dde2a21283987ea4882213dd514d0cf..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc
>+++ /dev/null
>@@ -1,351 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/desktop_capture/x11/x_server_pixel_buffer.h"
>-
>-#include <string.h>
>-#include <sys/shm.h>
>-
>-#include "modules/desktop_capture/desktop_frame.h"
>-#include "modules/desktop_capture/x11/window_list_utils.h"
>-#include "modules/desktop_capture/x11/x_error_trap.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-
>-// Returns the number of bits |mask| has to be shifted left so its last
>-// (most-significant) bit set becomes the most-significant bit of the word.
>-// When |mask| is 0 the function returns 31.
>-uint32_t MaskToShift(uint32_t mask) {
>-  int shift = 0;
>-  if ((mask & 0xffff0000u) == 0) {
>-    mask <<= 16;
>-    shift += 16;
>-  }
>-  if ((mask & 0xff000000u) == 0) {
>-    mask <<= 8;
>-    shift += 8;
>-  }
>-  if ((mask & 0xf0000000u) == 0) {
>-    mask <<= 4;
>-    shift += 4;
>-  }
>-  if ((mask & 0xc0000000u) == 0) {
>-    mask <<= 2;
>-    shift += 2;
>-  }
>-  if ((mask & 0x80000000u) == 0)
>-    shift += 1;
>-
>-  return shift;
>-}
>-
>-// Returns true if |image| is in RGB format.
>-bool IsXImageRGBFormat(XImage* image) {
>-  return image->bits_per_pixel == 32 && image->red_mask == 0xff0000 &&
>-         image->green_mask == 0xff00 && image->blue_mask == 0xff;
>-}
>-
>-// We expose two forms of blitting to handle variations in the pixel format.
>-// In FastBlit(), the operation is effectively a memcpy.
>-void FastBlit(XImage* x_image,
>-              uint8_t* src_pos,
>-              const DesktopRect& rect,
>-              DesktopFrame* frame) {
>-  int src_stride = x_image->bytes_per_line;
>-  int dst_x = rect.left(), dst_y = rect.top();
>-
>-  uint8_t* dst_pos = frame->data() + frame->stride() * dst_y;
>-  dst_pos += dst_x * DesktopFrame::kBytesPerPixel;
>-
>-  int height = rect.height();
>-  int row_bytes = rect.width() * DesktopFrame::kBytesPerPixel;
>-  for (int y = 0; y < height; ++y) {
>-    memcpy(dst_pos, src_pos, row_bytes);
>-    src_pos += src_stride;
>-    dst_pos += frame->stride();
>-  }
>-}
>-
>-void SlowBlit(XImage* x_image,
>-              uint8_t* src_pos,
>-              const DesktopRect& rect,
>-              DesktopFrame* frame) {
>-  int src_stride = x_image->bytes_per_line;
>-  int dst_x = rect.left(), dst_y = rect.top();
>-  int width = rect.width(), height = rect.height();
>-
>-  uint32_t red_mask = x_image->red_mask;
>-  uint32_t green_mask = x_image->red_mask;
>-  uint32_t blue_mask = x_image->blue_mask;
>-
>-  uint32_t red_shift = MaskToShift(red_mask);
>-  uint32_t green_shift = MaskToShift(green_mask);
>-  uint32_t blue_shift = MaskToShift(blue_mask);
>-
>-  int bits_per_pixel = x_image->bits_per_pixel;
>-
>-  uint8_t* dst_pos = frame->data() + frame->stride() * dst_y;
>-  dst_pos += dst_x * DesktopFrame::kBytesPerPixel;
>-  // TODO(hclam): Optimize, perhaps using MMX code or by converting to
>-  // YUV directly.
>-  // TODO(sergeyu): This code doesn't handle XImage byte order properly and
>-  // won't work with 24bpp images. Fix it.
>-  for (int y = 0; y < height; y++) {
>-    uint32_t* dst_pos_32 = reinterpret_cast<uint32_t*>(dst_pos);
>-    uint32_t* src_pos_32 = reinterpret_cast<uint32_t*>(src_pos);
>-    uint16_t* src_pos_16 = reinterpret_cast<uint16_t*>(src_pos);
>-    for (int x = 0; x < width; x++) {
>-      // Dereference through an appropriately-aligned pointer.
>-      uint32_t pixel;
>-      if (bits_per_pixel == 32) {
>-        pixel = src_pos_32[x];
>-      } else if (bits_per_pixel == 16) {
>-        pixel = src_pos_16[x];
>-      } else {
>-        pixel = src_pos[x];
>-      }
>-      uint32_t r = (pixel & red_mask) << red_shift;
>-      uint32_t g = (pixel & green_mask) << green_shift;
>-      uint32_t b = (pixel & blue_mask) << blue_shift;
>-      // Write as 32-bit RGB.
>-      dst_pos_32[x] =
>-          ((r >> 8) & 0xff0000) | ((g >> 16) & 0xff00) | ((b >> 24) & 0xff);
>-    }
>-    dst_pos += frame->stride();
>-    src_pos += src_stride;
>-  }
>-}
>-
>-}  // namespace
>-
>-XServerPixelBuffer::XServerPixelBuffer() {}
>-
>-XServerPixelBuffer::~XServerPixelBuffer() {
>-  Release();
>-}
>-
>-void XServerPixelBuffer::Release() {
>-  if (x_image_) {
>-    XDestroyImage(x_image_);
>-    x_image_ = nullptr;
>-  }
>-  if (x_shm_image_) {
>-    XDestroyImage(x_shm_image_);
>-    x_shm_image_ = nullptr;
>-  }
>-  if (shm_pixmap_) {
>-    XFreePixmap(display_, shm_pixmap_);
>-    shm_pixmap_ = 0;
>-  }
>-  if (shm_gc_) {
>-    XFreeGC(display_, shm_gc_);
>-    shm_gc_ = nullptr;
>-  }
>-
>-  ReleaseSharedMemorySegment();
>-
>-  window_ = 0;
>-}
>-
>-void XServerPixelBuffer::ReleaseSharedMemorySegment() {
>-  if (!shm_segment_info_)
>-    return;
>-  if (shm_segment_info_->shmaddr != nullptr)
>-    shmdt(shm_segment_info_->shmaddr);
>-  if (shm_segment_info_->shmid != -1)
>-    shmctl(shm_segment_info_->shmid, IPC_RMID, 0);
>-  delete shm_segment_info_;
>-  shm_segment_info_ = nullptr;
>-}
>-
>-bool XServerPixelBuffer::Init(Display* display, Window window) {
>-  Release();
>-  display_ = display;
>-
>-  XWindowAttributes attributes;
>-  if (!GetWindowRect(display_, window, &window_rect_, &attributes)) {
>-    return false;
>-  }
>-
>-  window_ = window;
>-  InitShm(attributes);
>-
>-  return true;
>-}
>-
>-void XServerPixelBuffer::InitShm(const XWindowAttributes& attributes) {
>-  Visual* default_visual = attributes.visual;
>-  int default_depth = attributes.depth;
>-
>-  int major, minor;
>-  Bool have_pixmaps;
>-  if (!XShmQueryVersion(display_, &major, &minor, &have_pixmaps)) {
>-    // Shared memory not supported. CaptureRect will use the XImage API instead.
>-    return;
>-  }
>-
>-  bool using_shm = false;
>-  shm_segment_info_ = new XShmSegmentInfo;
>-  shm_segment_info_->shmid = -1;
>-  shm_segment_info_->shmaddr = nullptr;
>-  shm_segment_info_->readOnly = False;
>-  x_shm_image_ = XShmCreateImage(display_, default_visual, default_depth,
>-                                 ZPixmap, 0, shm_segment_info_,
>-                                 window_rect_.width(), window_rect_.height());
>-  if (x_shm_image_) {
>-    shm_segment_info_->shmid =
>-        shmget(IPC_PRIVATE, x_shm_image_->bytes_per_line * x_shm_image_->height,
>-               IPC_CREAT | 0600);
>-    if (shm_segment_info_->shmid != -1) {
>-      void* shmat_result = shmat(shm_segment_info_->shmid, 0, 0);
>-      if (shmat_result != reinterpret_cast<void*>(-1)) {
>-        shm_segment_info_->shmaddr = reinterpret_cast<char*>(shmat_result);
>-        x_shm_image_->data = shm_segment_info_->shmaddr;
>-
>-        XErrorTrap error_trap(display_);
>-        using_shm = XShmAttach(display_, shm_segment_info_);
>-        XSync(display_, False);
>-        if (error_trap.GetLastErrorAndDisable() != 0)
>-          using_shm = false;
>-        if (using_shm) {
>-          RTC_LOG(LS_VERBOSE)
>-              << "Using X shared memory segment " << shm_segment_info_->shmid;
>-        }
>-      }
>-    } else {
>-      RTC_LOG(LS_WARNING) << "Failed to get shared memory segment. "
>-                             "Performance may be degraded.";
>-    }
>-  }
>-
>-  if (!using_shm) {
>-    RTC_LOG(LS_WARNING)
>-        << "Not using shared memory. Performance may be degraded.";
>-    ReleaseSharedMemorySegment();
>-    return;
>-  }
>-
>-  if (have_pixmaps)
>-    have_pixmaps = InitPixmaps(default_depth);
>-
>-  shmctl(shm_segment_info_->shmid, IPC_RMID, 0);
>-  shm_segment_info_->shmid = -1;
>-
>-  RTC_LOG(LS_VERBOSE) << "Using X shared memory extension v" << major << "."
>-                      << minor << " with" << (have_pixmaps ? "" : "out")
>-                      << " pixmaps.";
>-}
>-
>-bool XServerPixelBuffer::InitPixmaps(int depth) {
>-  if (XShmPixmapFormat(display_) != ZPixmap)
>-    return false;
>-
>-  {
>-    XErrorTrap error_trap(display_);
>-    shm_pixmap_ = XShmCreatePixmap(
>-        display_, window_, shm_segment_info_->shmaddr, shm_segment_info_,
>-        window_rect_.width(), window_rect_.height(), depth);
>-    XSync(display_, False);
>-    if (error_trap.GetLastErrorAndDisable() != 0) {
>-      // |shm_pixmap_| is not not valid because the request was not processed
>-      // by the X Server, so zero it.
>-      shm_pixmap_ = 0;
>-      return false;
>-    }
>-  }
>-
>-  {
>-    XErrorTrap error_trap(display_);
>-    XGCValues shm_gc_values;
>-    shm_gc_values.subwindow_mode = IncludeInferiors;
>-    shm_gc_values.graphics_exposures = False;
>-    shm_gc_ = XCreateGC(display_, window_,
>-                        GCSubwindowMode | GCGraphicsExposures, &shm_gc_values);
>-    XSync(display_, False);
>-    if (error_trap.GetLastErrorAndDisable() != 0) {
>-      XFreePixmap(display_, shm_pixmap_);
>-      shm_pixmap_ = 0;
>-      shm_gc_ = 0;  // See shm_pixmap_ comment above.
>-      return false;
>-    }
>-  }
>-
>-  return true;
>-}
>-
>-bool XServerPixelBuffer::IsWindowValid() const {
>-  XWindowAttributes attributes;
>-  {
>-    XErrorTrap error_trap(display_);
>-    if (!XGetWindowAttributes(display_, window_, &attributes) ||
>-        error_trap.GetLastErrorAndDisable() != 0) {
>-      return false;
>-    }
>-  }
>-  return true;
>-}
>-
>-void XServerPixelBuffer::Synchronize() {
>-  if (shm_segment_info_ && !shm_pixmap_) {
>-    // XShmGetImage can fail if the display is being reconfigured.
>-    XErrorTrap error_trap(display_);
>-    // XShmGetImage fails if the window is partially out of screen.
>-    xshm_get_image_succeeded_ =
>-        XShmGetImage(display_, window_, x_shm_image_, 0, 0, AllPlanes);
>-  }
>-}
>-
>-bool XServerPixelBuffer::CaptureRect(const DesktopRect& rect,
>-                                     DesktopFrame* frame) {
>-  RTC_DCHECK_LE(rect.right(), window_rect_.width());
>-  RTC_DCHECK_LE(rect.bottom(), window_rect_.height());
>-
>-  XImage* image;
>-  uint8_t* data;
>-
>-  if (shm_segment_info_ && (shm_pixmap_ || xshm_get_image_succeeded_)) {
>-    if (shm_pixmap_) {
>-      XCopyArea(display_, window_, shm_pixmap_, shm_gc_, rect.left(),
>-                rect.top(), rect.width(), rect.height(), rect.left(),
>-                rect.top());
>-      XSync(display_, False);
>-    }
>-
>-    image = x_shm_image_;
>-    data = reinterpret_cast<uint8_t*>(image->data) +
>-           rect.top() * image->bytes_per_line +
>-           rect.left() * image->bits_per_pixel / 8;
>-
>-  } else {
>-    if (x_image_)
>-      XDestroyImage(x_image_);
>-    x_image_ = XGetImage(display_, window_, rect.left(), rect.top(),
>-                         rect.width(), rect.height(), AllPlanes, ZPixmap);
>-    if (!x_image_)
>-      return false;
>-
>-    image = x_image_;
>-    data = reinterpret_cast<uint8_t*>(image->data);
>-  }
>-
>-  if (IsXImageRGBFormat(image)) {
>-    FastBlit(image, data, rect, frame);
>-  } else {
>-    SlowBlit(image, data, rect, frame);
>-  }
>-
>-  return true;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h
>deleted file mode 100644
>index 7e20945f3d529cafee0be365604a4ff35eb99e44..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.h
>+++ /dev/null
>@@ -1,84 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// Don't include this file in any .h files because it pulls in some X headers.
>-
>-#ifndef MODULES_DESKTOP_CAPTURE_X11_X_SERVER_PIXEL_BUFFER_H_
>-#define MODULES_DESKTOP_CAPTURE_X11_X_SERVER_PIXEL_BUFFER_H_
>-
>-#include "modules/desktop_capture/desktop_geometry.h"
>-#include "rtc_base/constructormagic.h"
>-
>-#include <X11/Xutil.h>
>-#include <X11/extensions/XShm.h>
>-
>-namespace webrtc {
>-
>-class DesktopFrame;
>-
>-// A class to allow the X server's pixel buffer to be accessed as efficiently
>-// as possible.
>-class XServerPixelBuffer {
>- public:
>-  XServerPixelBuffer();
>-  ~XServerPixelBuffer();
>-
>-  void Release();
>-
>-  // Allocate (or reallocate) the pixel buffer for |window|. Returns false in
>-  // case of an error (e.g. window doesn't exist).
>-  bool Init(Display* display, Window window);
>-
>-  bool is_initialized() { return window_ != 0; }
>-
>-  // Returns the size of the window the buffer was initialized for.
>-  DesktopSize window_size() { return window_rect_.size(); }
>-
>-  // Returns the rectangle of the window the buffer was initialized for.
>-  const DesktopRect& window_rect() { return window_rect_; }
>-
>-  // Returns true if the window can be found.
>-  bool IsWindowValid() const;
>-
>-  // If shared memory is being used without pixmaps, synchronize this pixel
>-  // buffer with the root window contents (otherwise, this is a no-op).
>-  // This is to avoid doing a full-screen capture for each individual
>-  // rectangle in the capture list, when it only needs to be done once at the
>-  // beginning.
>-  void Synchronize();
>-
>-  // Capture the specified rectangle and stores it in the |frame|. In the case
>-  // where the full-screen data is captured by Synchronize(), this simply
>-  // returns the pointer without doing any more work. The caller must ensure
>-  // that |rect| is not larger than window_size().
>-  bool CaptureRect(const DesktopRect& rect, DesktopFrame* frame);
>-
>- private:
>-  void ReleaseSharedMemorySegment();
>-
>-  void InitShm(const XWindowAttributes& attributes);
>-  bool InitPixmaps(int depth);
>-
>-  Display* display_ = nullptr;
>-  Window window_ = 0;
>-  DesktopRect window_rect_;
>-  XImage* x_image_ = nullptr;
>-  XShmSegmentInfo* shm_segment_info_ = nullptr;
>-  XImage* x_shm_image_ = nullptr;
>-  Pixmap shm_pixmap_ = 0;
>-  GC shm_gc_ = nullptr;
>-  bool xshm_get_image_succeeded_ = false;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(XServerPixelBuffer);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_DESKTOP_CAPTURE_X11_X_SERVER_PIXEL_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.cc
>index 4ad5d1444e9c57a71215b5969037af372d5e25a7..80eba2e3051cd2d1fdf0aa51f62f5048da05ea01 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.cc
>@@ -11,6 +11,7 @@
> #include "modules/include/module_common_types.h"
> 
> #include <string.h>
>+#include <cstdint>
> #include <utility>
> 
> #include "rtc_base/numerics/safe_conversions.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.h
>index 98ff7673110cd039ec6cd32547ae718d9cbdda7c..e058cc84e66ff79545afe0e083b311246a89f3f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.h
>@@ -19,6 +19,7 @@
> #include "modules/include/module_common_types_public.h"
> #include "modules/include/module_fec_types.h"
> #include "modules/rtp_rtcp/source/rtp_video_header.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -33,7 +34,7 @@ struct WebRtcRTPHeader {
>   int64_t ntp_time_ms;
> };
> 
>-class RTPFragmentationHeader {
>+class RTC_EXPORT RTPFragmentationHeader {
>  public:
>   RTPFragmentationHeader();
>   RTPFragmentationHeader(const RTPFragmentationHeader&) = delete;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/BUILD.gn
>index a06233ad90131b407b019684ead81fd7330d0aa4..9bc61936f16822b7d8eab37a50fa55bf8fa365bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/BUILD.gn
>@@ -29,7 +29,6 @@ rtc_static_library("pacing") {
>   deps = [
>     ":interval_budget",
>     "..:module_api",
>-    "../../:webrtc_common",
>     "../../api/transport:network_control",
>     "../../logging:rtc_event_bwe",
>     "../../logging:rtc_event_log_api",
>@@ -56,7 +55,6 @@ rtc_source_set("interval_budget") {
>   ]
> 
>   deps = [
>-    "../../:webrtc_common",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>   ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.cc
>index cbf1d2b97efb9272eb259caeb0dc3cde93205fa0..4ff7c0d583dd1d708dd90034f1008e013ed1ab53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.cc
>@@ -13,9 +13,9 @@
> #include <algorithm>
> 
> #include "absl/memory/memory.h"
>+#include "logging/rtc_event_log/events/rtc_event.h"
> #include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
>-#include "modules/pacing/paced_sender.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
>@@ -96,8 +96,9 @@ void BitrateProber::CreateProbeCluster(int bitrate_bps, int64_t now_ms) {
>   ProbeCluster cluster;
>   cluster.time_created_ms = now_ms;
>   cluster.pace_info.probe_cluster_min_probes = kMinProbePacketsSent;
>-  cluster.pace_info.probe_cluster_min_bytes =
>-      bitrate_bps * kMinProbeDurationMs / 8000;
>+  cluster.pace_info.probe_cluster_min_bytes = static_cast<int32_t>(
>+      static_cast<int64_t>(bitrate_bps) * kMinProbeDurationMs / 8000);
>+  RTC_DCHECK_GE(cluster.pace_info.probe_cluster_min_bytes, 0);
>   cluster.pace_info.send_bitrate_bps = bitrate_bps;
>   cluster.pace_info.probe_cluster_id = next_cluster_id_++;
>   clusters_.push(cluster);
>@@ -126,9 +127,9 @@ int BitrateProber::TimeUntilNextProbe(int64_t now_ms) {
>   if (next_probe_time_ms_ >= 0) {
>     time_until_probe_ms = next_probe_time_ms_ - now_ms;
>     if (time_until_probe_ms < -kMaxProbeDelayMs) {
>-      RTC_LOG(LS_WARNING) << "Probe delay too high"
>-                          << " (next_ms:" << next_probe_time_ms_
>-                          << ", now_ms: " << now_ms << ")";
>+      RTC_DLOG(LS_WARNING) << "Probe delay too high"
>+                           << " (next_ms:" << next_probe_time_ms_
>+                           << ", now_ms: " << now_ms << ")";
>       return -1;
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.h
>index b2548c006de83f5e597d3652df48feb20c25c67c..bb98100dfac5c6c74966cec6058c723479ad5693 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.h
>@@ -11,10 +11,11 @@
> #ifndef MODULES_PACING_BITRATE_PROBER_H_
> #define MODULES_PACING_BITRATE_PROBER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <queue>
> 
> #include "api/transport/network_types.h"
>-#include "modules/include/module_common_types.h"
> 
> namespace webrtc {
> class RtcEventLog;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober_unittest.cc
>index dc596982db022b3a9334a1ecbb9a0385bcf10247..95d26593cbddb91fc6325f52f402479a8f903837 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober_unittest.cc
>@@ -147,6 +147,24 @@ TEST(BitrateProberTest, ScaleBytesUsedForProbing) {
>   EXPECT_FALSE(prober.IsProbing());
> }
> 
>+TEST(BitrateProberTest, HighBitrateProbing) {
>+  BitrateProber prober;
>+  constexpr int kBitrateBps = 1000000000;  // 1 Gbps.
>+  constexpr int kPacketSizeBytes = 1000;
>+  constexpr int kExpectedBytesSent = (kBitrateBps / 8000) * 15;
>+
>+  prober.CreateProbeCluster(kBitrateBps, 0);
>+  prober.OnIncomingPacket(kPacketSizeBytes);
>+  int bytes_sent = 0;
>+  while (bytes_sent < kExpectedBytesSent) {
>+    ASSERT_TRUE(prober.IsProbing());
>+    prober.ProbeSent(0, kPacketSizeBytes);
>+    bytes_sent += kPacketSizeBytes;
>+  }
>+
>+  EXPECT_FALSE(prober.IsProbing());
>+}
>+
> TEST(BitrateProberTest, ProbeClusterTimeout) {
>   BitrateProber prober;
>   constexpr int kBitrateBps = 300000;  // 300 kbps
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/interval_budget.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/interval_budget.h
>index 880fe784ba74f07ee812e94dfea9cad5ef3d5ce3..d09b06e34034ae243f6bf1a10c2fddc645cfc108 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/interval_budget.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/interval_budget.h
>@@ -11,7 +11,8 @@
> #ifndef MODULES_PACING_INTERVAL_BUDGET_H_
> #define MODULES_PACING_INTERVAL_BUDGET_H_
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include <stddef.h>
>+#include <stdint.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.cc
>index 8cc442c228e849fdfbf2dd6b8ebfb33efd17307e..7e0ac781e2b784cb7e9a68a1599260971cda3572 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.cc
>@@ -11,15 +11,11 @@
> #include "modules/pacing/paced_sender.h"
> 
> #include <algorithm>
>-#include <map>
>-#include <queue>
>-#include <set>
> #include <utility>
>-#include <vector>
> 
> #include "absl/memory/memory.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/congestion_controller/goog_cc/alr_detector.h"
>-#include "modules/include/module_common_types.h"
> #include "modules/pacing/bitrate_prober.h"
> #include "modules/pacing/interval_budget.h"
> #include "modules/utility/include/process_thread.h"
>@@ -58,9 +54,9 @@ PacedSender::PacedSender(const Clock* clock,
>       video_blocks_audio_(!field_trial::IsDisabled("WebRTC-Pacer-BlockAudio")),
>       last_timestamp_ms_(clock_->TimeInMilliseconds()),
>       paused_(false),
>-      media_budget_(absl::make_unique<IntervalBudget>(0)),
>-      padding_budget_(absl::make_unique<IntervalBudget>(0)),
>-      prober_(absl::make_unique<BitrateProber>(event_log)),
>+      media_budget_(0),
>+      padding_budget_(0),
>+      prober_(event_log),
>       probing_send_failure_(false),
>       estimated_bitrate_bps_(0),
>       min_send_bitrate_kbps_(0u),
>@@ -69,7 +65,7 @@ PacedSender::PacedSender(const Clock* clock,
>       time_last_process_us_(clock->TimeInMicroseconds()),
>       last_send_time_us_(clock->TimeInMicroseconds()),
>       first_sent_packet_ms_(-1),
>-      packets_(clock),
>+      packets_(clock->TimeInMicroseconds()),
>       packet_counter_(0),
>       pacing_factor_(kDefaultPaceMultiplier),
>       queue_time_limit(kMaxQueueLengthMs),
>@@ -84,7 +80,7 @@ PacedSender::~PacedSender() {}
> 
> void PacedSender::CreateProbeCluster(int bitrate_bps) {
>   rtc::CritScope cs(&critsect_);
>-  prober_->CreateProbeCluster(bitrate_bps, TimeMilliseconds());
>+  prober_.CreateProbeCluster(bitrate_bps, TimeMilliseconds());
> }
> 
> void PacedSender::Pause() {
>@@ -149,7 +145,7 @@ int64_t PacedSender::TimeMilliseconds() const {
> void PacedSender::SetProbingEnabled(bool enabled) {
>   rtc::CritScope cs(&critsect_);
>   RTC_CHECK_EQ(0, packet_counter_);
>-  prober_->SetEnabled(enabled);
>+  prober_.SetEnabled(enabled);
> }
> 
> void PacedSender::SetEstimatedBitrate(uint32_t bitrate_bps) {
>@@ -157,7 +153,7 @@ void PacedSender::SetEstimatedBitrate(uint32_t bitrate_bps) {
>     RTC_LOG(LS_ERROR) << "PacedSender is not designed to handle 0 bitrate.";
>   rtc::CritScope cs(&critsect_);
>   estimated_bitrate_bps_ = bitrate_bps;
>-  padding_budget_->set_target_rate_kbps(
>+  padding_budget_.set_target_rate_kbps(
>       std::min(estimated_bitrate_bps_ / 1000, max_padding_bitrate_kbps_));
>   pacing_bitrate_kbps_ =
>       std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
>@@ -173,7 +169,7 @@ void PacedSender::SetSendBitrateLimits(int min_send_bitrate_bps,
>       std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
>       pacing_factor_;
>   max_padding_bitrate_kbps_ = padding_bitrate / 1000;
>-  padding_budget_->set_target_rate_kbps(
>+  padding_budget_.set_target_rate_kbps(
>       std::min(estimated_bitrate_bps_ / 1000, max_padding_bitrate_kbps_));
> }
> 
>@@ -182,7 +178,7 @@ void PacedSender::SetPacingRates(uint32_t pacing_rate_bps,
>   rtc::CritScope cs(&critsect_);
>   RTC_DCHECK(pacing_rate_bps > 0);
>   pacing_bitrate_kbps_ = pacing_rate_bps / 1000;
>-  padding_budget_->set_target_rate_kbps(padding_rate_bps / 1000);
>+  padding_budget_.set_target_rate_kbps(padding_rate_bps / 1000);
> }
> 
> void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
>@@ -196,7 +192,7 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
>       << "SetPacingRate must be called before InsertPacket.";
> 
>   int64_t now_ms = TimeMilliseconds();
>-  prober_->OnIncomingPacket(bytes);
>+  prober_.OnIncomingPacket(bytes);
> 
>   if (capture_time_ms < 0)
>     capture_time_ms = now_ms;
>@@ -254,17 +250,15 @@ int64_t PacedSender::TimeUntilNextProcess() {
>   if (paused_)
>     return std::max<int64_t>(kPausedProcessIntervalMs - elapsed_time_ms, 0);
> 
>-  if (prober_->IsProbing()) {
>-    int64_t ret = prober_->TimeUntilNextProbe(TimeMilliseconds());
>+  if (prober_.IsProbing()) {
>+    int64_t ret = prober_.TimeUntilNextProbe(TimeMilliseconds());
>     if (ret > 0 || (ret == 0 && !probing_send_failure_))
>       return ret;
>   }
>   return std::max<int64_t>(kMinPacketLimitMs - elapsed_time_ms, 0);
> }
> 
>-void PacedSender::Process() {
>-  int64_t now_us = clock_->TimeInMicroseconds();
>-  rtc::CritScope cs(&critsect_);
>+int64_t PacedSender::UpdateTimeAndGetElapsedMs(int64_t now_us) {
>   int64_t elapsed_time_ms = (now_us - time_last_process_us_ + 500) / 1000;
>   time_last_process_us_ = now_us;
>   if (elapsed_time_ms > kMaxElapsedTimeMs) {
>@@ -273,6 +267,10 @@ void PacedSender::Process() {
>                         << kMaxElapsedTimeMs << " ms";
>     elapsed_time_ms = kMaxElapsedTimeMs;
>   }
>+  return elapsed_time_ms;
>+}
>+
>+bool PacedSender::ShouldSendKeepalive(int64_t now_us) const {
>   if (send_padding_if_silent_ || paused_ || Congested()) {
>     // We send a padding packet every 500 ms to ensure we won't get stuck in
>     // congested state due to no feedback being received.
>@@ -281,12 +279,25 @@ void PacedSender::Process() {
>       // We can not send padding unless a normal packet has first been sent. If
>       // we do, timestamps get messed up.
>       if (packet_counter_ > 0) {
>-        PacedPacketInfo pacing_info;
>-        size_t bytes_sent = SendPadding(1, pacing_info);
>-        alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
>+        return true;
>       }
>     }
>   }
>+  return false;
>+}
>+
>+void PacedSender::Process() {
>+  rtc::CritScope cs(&critsect_);
>+  int64_t now_us = clock_->TimeInMicroseconds();
>+  int64_t elapsed_time_ms = UpdateTimeAndGetElapsedMs(now_us);
>+  if (ShouldSendKeepalive(now_us)) {
>+    critsect_.Leave();
>+    size_t bytes_sent = packet_sender_->TimeToSendPadding(1, PacedPacketInfo());
>+    critsect_.Enter();
>+    OnPaddingSent(bytes_sent);
>+    alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
>+  }
>+
>   if (paused_)
>     return;
> 
>@@ -308,35 +319,39 @@ void PacedSender::Process() {
>       }
>     }
> 
>-    media_budget_->set_target_rate_kbps(target_bitrate_kbps);
>+    media_budget_.set_target_rate_kbps(target_bitrate_kbps);
>     UpdateBudgetWithElapsedTime(elapsed_time_ms);
>   }
> 
>-  bool is_probing = prober_->IsProbing();
>+  bool is_probing = prober_.IsProbing();
>   PacedPacketInfo pacing_info;
>   size_t bytes_sent = 0;
>   size_t recommended_probe_size = 0;
>   if (is_probing) {
>-    pacing_info = prober_->CurrentCluster();
>-    recommended_probe_size = prober_->RecommendedMinProbeSize();
>+    pacing_info = prober_.CurrentCluster();
>+    recommended_probe_size = prober_.RecommendedMinProbeSize();
>   }
>-  // The paused state is checked in the loop since SendPacket leaves the
>-  // critical section allowing the paused state to be changed from other code.
>+  // The paused state is checked in the loop since it leaves the critical
>+  // section allowing the paused state to be changed from other code.
>   while (!packets_.Empty() && !paused_) {
>-    // Since we need to release the lock in order to send, we first pop the
>-    // element from the priority queue but keep it in storage, so that we can
>-    // reinsert it if send fails.
>-    const RoundRobinPacketQueue::Packet& packet = packets_.BeginPop();
>+    const auto* packet = GetPendingPacket(pacing_info);
>+    if (packet == nullptr)
>+      break;
> 
>-    if (SendPacket(packet, pacing_info)) {
>-      bytes_sent += packet.bytes;
>+    critsect_.Leave();
>+    bool success = packet_sender_->TimeToSendPacket(
>+        packet->ssrc, packet->sequence_number, packet->capture_time_ms,
>+        packet->retransmission, pacing_info);
>+    critsect_.Enter();
>+    if (success) {
>+      bytes_sent += packet->bytes;
>       // Send succeeded, remove it from the queue.
>-      packets_.FinalizePop(packet);
>+      OnPacketSent(std::move(packet));
>       if (is_probing && bytes_sent > recommended_probe_size)
>         break;
>     } else {
>       // Send failed, put it back into the queue.
>-      packets_.CancelPop(packet);
>+      packets_.CancelPop(*packet);
>       break;
>     }
>   }
>@@ -347,16 +362,21 @@ void PacedSender::Process() {
>     if (packet_counter_ > 0) {
>       int padding_needed =
>           static_cast<int>(is_probing ? (recommended_probe_size - bytes_sent)
>-                                      : padding_budget_->bytes_remaining());
>+                                      : padding_budget_.bytes_remaining());
>       if (padding_needed > 0) {
>-        bytes_sent += SendPadding(padding_needed, pacing_info);
>+        critsect_.Leave();
>+        size_t padding_sent =
>+            packet_sender_->TimeToSendPadding(padding_needed, pacing_info);
>+        critsect_.Enter();
>+        bytes_sent += padding_sent;
>+        OnPaddingSent(padding_sent);
>       }
>     }
>   }
>   if (is_probing) {
>     probing_send_failure_ = bytes_sent == 0;
>     if (!probing_send_failure_)
>-      prober_->ProbeSent(TimeMilliseconds(), bytes_sent);
>+      prober_.ProbeSent(TimeMilliseconds(), bytes_sent);
>   }
>   alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
> }
>@@ -367,66 +387,58 @@ void PacedSender::ProcessThreadAttached(ProcessThread* process_thread) {
>   process_thread_ = process_thread;
> }
> 
>-bool PacedSender::SendPacket(const RoundRobinPacketQueue::Packet& packet,
>-                             const PacedPacketInfo& pacing_info) {
>-  RTC_DCHECK(!paused_);
>-  bool audio_packet = packet.priority == kHighPriority;
>+const RoundRobinPacketQueue::Packet* PacedSender::GetPendingPacket(
>+    const PacedPacketInfo& pacing_info) {
>+  // Since we need to release the lock in order to send, we first pop the
>+  // element from the priority queue but keep it in storage, so that we can
>+  // reinsert it if send fails.
>+  const RoundRobinPacketQueue::Packet* packet = &packets_.BeginPop();
>+  bool audio_packet = packet->priority == kHighPriority;
>   bool apply_pacing =
>       !audio_packet || account_for_audio_ || video_blocks_audio_;
>-  if (apply_pacing && (Congested() || (media_budget_->bytes_remaining() == 0 &&
>+  if (apply_pacing && (Congested() || (media_budget_.bytes_remaining() == 0 &&
>                                        pacing_info.probe_cluster_id ==
>                                            PacedPacketInfo::kNotAProbe))) {
>-    return false;
>+    packets_.CancelPop(*packet);
>+    return nullptr;
>   }
>+  return packet;
>+}
> 
>-  critsect_.Leave();
>-  const bool success = packet_sender_->TimeToSendPacket(
>-      packet.ssrc, packet.sequence_number, packet.capture_time_ms,
>-      packet.retransmission, pacing_info);
>-  critsect_.Enter();
>-
>-  if (success) {
>-    if (first_sent_packet_ms_ == -1)
>-      first_sent_packet_ms_ = TimeMilliseconds();
>-    if (!audio_packet || account_for_audio_) {
>-      // Update media bytes sent.
>-      // TODO(eladalon): TimeToSendPacket() can also return |true| in some
>-      // situations where nothing actually ended up being sent to the network,
>-      // and we probably don't want to update the budget in such cases.
>-      // https://bugs.chromium.org/p/webrtc/issues/detail?id=8052
>-      UpdateBudgetWithBytesSent(packet.bytes);
>-      last_send_time_us_ = clock_->TimeInMicroseconds();
>-    }
>+void PacedSender::OnPacketSent(const RoundRobinPacketQueue::Packet* packet) {
>+  if (first_sent_packet_ms_ == -1)
>+    first_sent_packet_ms_ = TimeMilliseconds();
>+  bool audio_packet = packet->priority == kHighPriority;
>+  if (!audio_packet || account_for_audio_) {
>+    // Update media bytes sent.
>+    // TODO(eladalon): TimeToSendPacket() can also return |true| in some
>+    // situations where nothing actually ended up being sent to the network,
>+    // and we probably don't want to update the budget in such cases.
>+    // https://bugs.chromium.org/p/webrtc/issues/detail?id=8052
>+    UpdateBudgetWithBytesSent(packet->bytes);
>+    last_send_time_us_ = clock_->TimeInMicroseconds();
>   }
>-
>-  return success;
>+  // Send succeeded, remove it from the queue.
>+  packets_.FinalizePop(*packet);
> }
> 
>-size_t PacedSender::SendPadding(size_t padding_needed,
>-                                const PacedPacketInfo& pacing_info) {
>-  RTC_DCHECK_GT(packet_counter_, 0);
>-  critsect_.Leave();
>-  size_t bytes_sent =
>-      packet_sender_->TimeToSendPadding(padding_needed, pacing_info);
>-  critsect_.Enter();
>-
>+void PacedSender::OnPaddingSent(size_t bytes_sent) {
>   if (bytes_sent > 0) {
>     UpdateBudgetWithBytesSent(bytes_sent);
>   }
>   last_send_time_us_ = clock_->TimeInMicroseconds();
>-  return bytes_sent;
> }
> 
> void PacedSender::UpdateBudgetWithElapsedTime(int64_t delta_time_ms) {
>   delta_time_ms = std::min(kMaxIntervalTimeMs, delta_time_ms);
>-  media_budget_->IncreaseBudget(delta_time_ms);
>-  padding_budget_->IncreaseBudget(delta_time_ms);
>+  media_budget_.IncreaseBudget(delta_time_ms);
>+  padding_budget_.IncreaseBudget(delta_time_ms);
> }
> 
> void PacedSender::UpdateBudgetWithBytesSent(size_t bytes_sent) {
>   outstanding_bytes_ += bytes_sent;
>-  media_budget_->UseBudget(bytes_sent);
>-  padding_budget_->UseBudget(bytes_sent);
>+  media_budget_.UseBudget(bytes_sent);
>+  padding_budget_.UseBudget(bytes_sent);
> }
> 
> void PacedSender::SetPacingFactor(float pacing_factor) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.h
>index f12917408a6b41684228fb040983d29b3f838120..4586d29cca97e76e05b7615edf07f2b2da7635a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.h
>@@ -11,20 +11,25 @@
> #ifndef MODULES_PACING_PACED_SENDER_H_
> #define MODULES_PACING_PACED_SENDER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> 
> #include "absl/types/optional.h"
>+#include "api/transport/network_types.h"
>+#include "modules/pacing/bitrate_prober.h"
>+#include "modules/pacing/interval_budget.h"
> #include "modules/pacing/pacer.h"
> #include "modules/pacing/round_robin_packet_queue.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/utility/include/process_thread.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
> class AlrDetector;
>-class BitrateProber;
> class Clock;
> class RtcEventLog;
>-class IntervalBudget;
> 
> class PacedSender : public Pacer {
>  public:
>@@ -138,19 +143,25 @@ class PacedSender : public Pacer {
>   void SetQueueTimeLimit(int limit_ms);
> 
>  private:
>+  int64_t UpdateTimeAndGetElapsedMs(int64_t now_us)
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>+  bool ShouldSendKeepalive(int64_t at_time_us) const
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>+
>   // Updates the number of bytes that can be sent for the next time interval.
>   void UpdateBudgetWithElapsedTime(int64_t delta_time_in_ms)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>   void UpdateBudgetWithBytesSent(size_t bytes)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
> 
>-  bool SendPacket(const RoundRobinPacketQueue::Packet& packet,
>-                  const PacedPacketInfo& cluster_info)
>+  const RoundRobinPacketQueue::Packet* GetPendingPacket(
>+      const PacedPacketInfo& pacing_info)
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>+  void OnPacketSent(const RoundRobinPacketQueue::Packet* packet)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>-  size_t SendPadding(size_t padding_needed, const PacedPacketInfo& cluster_info)
>+  void OnPaddingSent(size_t padding_sent)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
> 
>-  void OnBytesSent(size_t bytes_sent) RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>   bool Congested() const RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>   int64_t TimeMilliseconds() const RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
> 
>@@ -169,15 +180,13 @@ class PacedSender : public Pacer {
>   bool paused_ RTC_GUARDED_BY(critsect_);
>   // This is the media budget, keeping track of how many bits of media
>   // we can pace out during the current interval.
>-  const std::unique_ptr<IntervalBudget> media_budget_
>-      RTC_PT_GUARDED_BY(critsect_);
>+  IntervalBudget media_budget_ RTC_GUARDED_BY(critsect_);
>   // This is the padding budget, keeping track of how many bits of padding we're
>   // allowed to send out during the current interval. This budget will be
>   // utilized when there's no media to send.
>-  const std::unique_ptr<IntervalBudget> padding_budget_
>-      RTC_PT_GUARDED_BY(critsect_);
>+  IntervalBudget padding_budget_ RTC_GUARDED_BY(critsect_);
> 
>-  const std::unique_ptr<BitrateProber> prober_ RTC_PT_GUARDED_BY(critsect_);
>+  BitrateProber prober_ RTC_GUARDED_BY(critsect_);
>   bool probing_send_failure_ RTC_GUARDED_BY(critsect_);
>   // Actual configured bitrates (media_budget_ may temporarily be higher in
>   // order to meet pace time constraint).
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.cc
>index b27c73c2f0c7e7e76ba86fbeb803d36596c8c7a8..24be7eebf2a9e32db77286c687af41adf9f4a5e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.cc
>@@ -11,8 +11,10 @@
> #include "modules/pacing/packet_router.h"
> 
> #include <algorithm>
>+#include <cstdint>
> #include <limits>
> 
>+#include "absl/types/optional.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.h
>index f01e9ef493396541a6f883f2d84c8e68ae37b9a7..b7cc5ff196687da2bc8a490bf8f2836b1323d16d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.h
>@@ -11,16 +11,17 @@
> #ifndef MODULES_PACING_PACKET_ROUTER_H_
> #define MODULES_PACING_PACKET_ROUTER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <list>
> #include <vector>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "api/transport/network_types.h"
> #include "modules/pacing/paced_sender.h"
> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>-#include "rtc_base/race_checker.h"
> #include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.cc
>index e892ddb4ea2b5086481bd90213e560bdbaab5dd1..e61d68215aca8a7476babe34878b6de6046ab359 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.cc
>@@ -11,9 +11,10 @@
> #include "modules/pacing/round_robin_packet_queue.h"
> 
> #include <algorithm>
>+#include <cstdint>
>+#include <utility>
> 
> #include "rtc_base/checks.h"
>-#include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
> 
>@@ -53,8 +54,8 @@ RoundRobinPacketQueue::Stream::Stream() : bytes(0), ssrc(0) {}
> RoundRobinPacketQueue::Stream::Stream(const Stream& stream) = default;
> RoundRobinPacketQueue::Stream::~Stream() {}
> 
>-RoundRobinPacketQueue::RoundRobinPacketQueue(const Clock* clock)
>-    : time_last_updated_(clock->TimeInMilliseconds()) {}
>+RoundRobinPacketQueue::RoundRobinPacketQueue(int64_t start_time_us)
>+    : time_last_updated_ms_(start_time_us / 1000) {}
> 
> RoundRobinPacketQueue::~RoundRobinPacketQueue() {}
> 
>@@ -132,7 +133,7 @@ void RoundRobinPacketQueue::FinalizePop(const Packet& packet) {
>     // by subtracting it now we effectively remove the time spent in in the
>     // queue while in a paused state.
>     int64_t time_in_non_paused_state_ms =
>-        time_last_updated_ - packet.enqueue_time_ms - pause_time_sum_ms_;
>+        time_last_updated_ms_ - packet.enqueue_time_ms - pause_time_sum_ms_;
>     queue_time_sum_ms_ -= time_in_non_paused_state_ms;
> 
>     RTC_CHECK(packet.enqueue_time_it != enqueue_times_.end());
>@@ -189,11 +190,11 @@ int64_t RoundRobinPacketQueue::OldestEnqueueTimeMs() const {
> }
> 
> void RoundRobinPacketQueue::UpdateQueueTime(int64_t timestamp_ms) {
>-  RTC_CHECK_GE(timestamp_ms, time_last_updated_);
>-  if (timestamp_ms == time_last_updated_)
>+  RTC_CHECK_GE(timestamp_ms, time_last_updated_ms_);
>+  if (timestamp_ms == time_last_updated_ms_)
>     return;
> 
>-  int64_t delta_ms = timestamp_ms - time_last_updated_;
>+  int64_t delta_ms = timestamp_ms - time_last_updated_ms_;
> 
>   if (paused_) {
>     pause_time_sum_ms_ += delta_ms;
>@@ -201,7 +202,7 @@ void RoundRobinPacketQueue::UpdateQueueTime(int64_t timestamp_ms) {
>     queue_time_sum_ms_ += delta_ms * size_packets_;
>   }
> 
>-  time_last_updated_ = timestamp_ms;
>+  time_last_updated_ms_ = timestamp_ms;
> }
> 
> void RoundRobinPacketQueue::SetPauseState(bool paused, int64_t timestamp_ms) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.h
>index ee91e2efc0c93444fb6959551f9fcb10d07ccf38..74b855a483d55e157ba0a09f66204b2ac7998bd1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.h
>@@ -11,18 +11,22 @@
> #ifndef MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_
> #define MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <list>
> #include <map>
> #include <queue>
> #include <set>
> 
>+#include "absl/types/optional.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
> 
> class RoundRobinPacketQueue {
>  public:
>-  explicit RoundRobinPacketQueue(const Clock* clock);
>+  explicit RoundRobinPacketQueue(int64_t start_time_us);
>   ~RoundRobinPacketQueue();
> 
>   struct Packet {
>@@ -105,7 +109,7 @@ class RoundRobinPacketQueue {
>   // Just used to verify correctness.
>   bool IsSsrcScheduled(uint32_t ssrc) const;
> 
>-  int64_t time_last_updated_;
>+  int64_t time_last_updated_ms_;
>   absl::optional<Packet> pop_packet_;
>   absl::optional<Stream*> pop_stream_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/BUILD.gn
>index a8d6850749cc9081f93ccd93b5e923116fe6ad14..f6991ea69d403cfd7d50d1bca9e6163b837a70fd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/BUILD.gn
>@@ -42,7 +42,9 @@ rtc_static_library("remote_bitrate_estimator") {
>   deps = [
>     "../..:webrtc_common",
>     "../../api/units:data_rate",
>+    "../../api/units:timestamp",
>     "../../modules:module_api",
>+    "../../modules:module_api_public",
>     "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>@@ -227,6 +229,7 @@ if (rtc_include_tests) {
>       "../../rtc_base:rtc_base_approved",
>       "../../test:fileutils",
>       "../../test:test_main",
>+      "../../test:test_support",
>       "//testing/gtest",
>     ]
>     data = [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
>index a6e64368977f5ab06b004ba8a6e9d216e88f1997..b2ca8012889c6d953e3599efbd27cbeba5f743b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
>@@ -11,7 +11,6 @@
> #include "modules/remote_bitrate_estimator/aimd_rate_control.h"
> 
> #include <inttypes.h>
>-
> #include <algorithm>
> #include <cassert>
> #include <cmath>
>@@ -19,9 +18,7 @@
> #include <string>
> 
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>-#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
> #include "modules/remote_bitrate_estimator/overuse_detector.h"
>-#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_minmax.h"
>@@ -29,25 +26,24 @@
> 
> namespace webrtc {
> 
>-static const int64_t kDefaultRttMs = 200;
>-static const int64_t kMaxFeedbackIntervalMs = 1000;
>-static const float kDefaultBackoffFactor = 0.85f;
>-static const int64_t kDefaultInitialBackOffIntervalMs = 200;
>+constexpr TimeDelta kDefaultRtt = TimeDelta::Millis<200>();
>+constexpr double kDefaultBackoffFactor = 0.85;
>+constexpr TimeDelta kDefaultInitialBackOffInterval = TimeDelta::Millis<200>();
> 
> const char kBweBackOffFactorExperiment[] = "WebRTC-BweBackOffFactor";
> const char kBweInitialBackOffIntervalExperiment[] =
>     "WebRTC-BweInitialBackOffInterval";
> 
>-float ReadBackoffFactor() {
>+double ReadBackoffFactor() {
>   std::string experiment_string =
>       webrtc::field_trial::FindFullName(kBweBackOffFactorExperiment);
>-  float backoff_factor;
>+  double backoff_factor;
>   int parsed_values =
>-      sscanf(experiment_string.c_str(), "Enabled-%f", &backoff_factor);
>+      sscanf(experiment_string.c_str(), "Enabled-%lf", &backoff_factor);
>   if (parsed_values == 1) {
>-    if (backoff_factor >= 1.0f) {
>+    if (backoff_factor >= 1.0) {
>       RTC_LOG(WARNING) << "Back-off factor must be less than 1.";
>-    } else if (backoff_factor <= 0.0f) {
>+    } else if (backoff_factor <= 0.0) {
>       RTC_LOG(WARNING) << "Back-off factor must be greater than 0.";
>     } else {
>       return backoff_factor;
>@@ -58,7 +54,7 @@ float ReadBackoffFactor() {
>   return kDefaultBackoffFactor;
> }
> 
>-int64_t ReadInitialBackoffIntervalMs() {
>+TimeDelta ReadInitialBackoffInterval() {
>   std::string experiment_string =
>       webrtc::field_trial::FindFullName(kBweInitialBackOffIntervalExperiment);
>   int64_t backoff_interval;
>@@ -66,7 +62,7 @@ int64_t ReadInitialBackoffIntervalMs() {
>       sscanf(experiment_string.c_str(), "Enabled-%" SCNd64, &backoff_interval);
>   if (parsed_values == 1) {
>     if (10 <= backoff_interval && backoff_interval <= 200) {
>-      return backoff_interval;
>+      return TimeDelta::ms(backoff_interval);
>     }
>     RTC_LOG(WARNING)
>         << "Initial back-off interval must be between 10 and 200 ms.";
>@@ -74,334 +70,347 @@ int64_t ReadInitialBackoffIntervalMs() {
>   RTC_LOG(LS_WARNING) << "Failed to parse parameters for "
>                       << kBweInitialBackOffIntervalExperiment
>                       << " experiment. Using default.";
>-  return kDefaultInitialBackOffIntervalMs;
>+  return kDefaultInitialBackOffInterval;
> }
> 
> AimdRateControl::AimdRateControl()
>-    : min_configured_bitrate_bps_(congestion_controller::GetMinBitrateBps()),
>-      max_configured_bitrate_bps_(30000000),
>-      current_bitrate_bps_(max_configured_bitrate_bps_),
>-      latest_estimated_throughput_bps_(current_bitrate_bps_),
>-      avg_max_bitrate_kbps_(-1.0f),
>-      var_max_bitrate_kbps_(0.4f),
>+    : min_configured_bitrate_(congestion_controller::GetMinBitrate()),
>+      max_configured_bitrate_(DataRate::kbps(30000)),
>+      current_bitrate_(max_configured_bitrate_),
>+      latest_estimated_throughput_(current_bitrate_),
>+      link_capacity_estimate_kbps_(),
>+      var_link_capacity_estimate_kbps_(0.4),
>       rate_control_state_(kRcHold),
>-      rate_control_region_(kRcMaxUnknown),
>-      time_last_bitrate_change_(-1),
>-      time_last_bitrate_decrease_(-1),
>-      time_first_throughput_estimate_(-1),
>+      time_last_bitrate_change_(Timestamp::MinusInfinity()),
>+      time_last_bitrate_decrease_(Timestamp::MinusInfinity()),
>+      time_first_throughput_estimate_(Timestamp::MinusInfinity()),
>       bitrate_is_initialized_(false),
>       beta_(webrtc::field_trial::IsEnabled(kBweBackOffFactorExperiment)
>                 ? ReadBackoffFactor()
>                 : kDefaultBackoffFactor),
>-      rtt_(kDefaultRttMs),
>+      rtt_(kDefaultRtt),
>       in_experiment_(!AdaptiveThresholdExperimentIsDisabled()),
>       smoothing_experiment_(
>           webrtc::field_trial::IsEnabled("WebRTC-Audio-BandwidthSmoothing")),
>       in_initial_backoff_interval_experiment_(
>           webrtc::field_trial::IsEnabled(kBweInitialBackOffIntervalExperiment)),
>-      initial_backoff_interval_ms_(kDefaultInitialBackOffIntervalMs) {
>+      initial_backoff_interval_(kDefaultInitialBackOffInterval) {
>   if (in_initial_backoff_interval_experiment_) {
>-    initial_backoff_interval_ms_ = ReadInitialBackoffIntervalMs();
>+    initial_backoff_interval_ = ReadInitialBackoffInterval();
>     RTC_LOG(LS_INFO) << "Using aimd rate control with initial back-off interval"
>-                     << " " << initial_backoff_interval_ms_ << " ms.";
>+                     << " " << ToString(initial_backoff_interval_) << ".";
>   }
>   RTC_LOG(LS_INFO) << "Using aimd rate control with back off factor " << beta_;
> }
> 
> AimdRateControl::~AimdRateControl() {}
> 
>-void AimdRateControl::SetStartBitrate(int start_bitrate_bps) {
>-  current_bitrate_bps_ = start_bitrate_bps;
>-  latest_estimated_throughput_bps_ = current_bitrate_bps_;
>+void AimdRateControl::SetStartBitrate(DataRate start_bitrate) {
>+  current_bitrate_ = start_bitrate;
>+  latest_estimated_throughput_ = current_bitrate_;
>   bitrate_is_initialized_ = true;
> }
> 
>-void AimdRateControl::SetMinBitrate(int min_bitrate_bps) {
>-  min_configured_bitrate_bps_ = min_bitrate_bps;
>-  current_bitrate_bps_ = std::max<int>(min_bitrate_bps, current_bitrate_bps_);
>+void AimdRateControl::SetMinBitrate(DataRate min_bitrate) {
>+  min_configured_bitrate_ = min_bitrate;
>+  current_bitrate_ = std::max(min_bitrate, current_bitrate_);
> }
> 
> bool AimdRateControl::ValidEstimate() const {
>   return bitrate_is_initialized_;
> }
> 
>-int64_t AimdRateControl::GetFeedbackInterval() const {
>+TimeDelta AimdRateControl::GetFeedbackInterval() const {
>   // Estimate how often we can send RTCP if we allocate up to 5% of bandwidth
>   // to feedback.
>-  static const int kRtcpSize = 80;
>-  const int64_t interval = static_cast<int64_t>(
>-      kRtcpSize * 8.0 * 1000.0 / (0.05 * current_bitrate_bps_) + 0.5);
>-  const int64_t kMinFeedbackIntervalMs = 200;
>-  return rtc::SafeClamp(interval, kMinFeedbackIntervalMs,
>-                        kMaxFeedbackIntervalMs);
>+  const DataSize kRtcpSize = DataSize::bytes(80);
>+  const DataRate rtcp_bitrate = current_bitrate_ * 0.05;
>+  const TimeDelta interval = kRtcpSize / rtcp_bitrate;
>+  const TimeDelta kMinFeedbackInterval = TimeDelta::ms(200);
>+  const TimeDelta kMaxFeedbackInterval = TimeDelta::ms(1000);
>+  return interval.Clamped(kMinFeedbackInterval, kMaxFeedbackInterval);
> }
> 
>-bool AimdRateControl::TimeToReduceFurther(
>-    int64_t now_ms,
>-    uint32_t estimated_throughput_bps) const {
>-  const int64_t bitrate_reduction_interval =
>-      std::max<int64_t>(std::min<int64_t>(rtt_, 200), 10);
>-  if (now_ms - time_last_bitrate_change_ >= bitrate_reduction_interval) {
>+bool AimdRateControl::TimeToReduceFurther(Timestamp at_time,
>+                                          DataRate estimated_throughput) const {
>+  const TimeDelta bitrate_reduction_interval =
>+      rtt_.Clamped(TimeDelta::ms(10), TimeDelta::ms(200));
>+  if (at_time - time_last_bitrate_change_ >= bitrate_reduction_interval) {
>     return true;
>   }
>   if (ValidEstimate()) {
>     // TODO(terelius/holmer): Investigate consequences of increasing
>     // the threshold to 0.95 * LatestEstimate().
>-    const uint32_t threshold = static_cast<uint32_t>(0.5 * LatestEstimate());
>-    return estimated_throughput_bps < threshold;
>+    const DataRate threshold = 0.5 * LatestEstimate();
>+    return estimated_throughput < threshold;
>   }
>   return false;
> }
> 
>-bool AimdRateControl::InitialTimeToReduceFurther(int64_t now_ms) const {
>+bool AimdRateControl::InitialTimeToReduceFurther(Timestamp at_time) const {
>   if (!in_initial_backoff_interval_experiment_) {
>     return ValidEstimate() &&
>-           TimeToReduceFurther(now_ms, LatestEstimate() / 2 - 1);
>+           TimeToReduceFurther(at_time,
>+                               LatestEstimate() / 2 - DataRate::bps(1));
>   }
>   // TODO(terelius): We could use the RTT (clamped to suitable limits) instead
>   // of a fixed bitrate_reduction_interval.
>-  if (time_last_bitrate_decrease_ == -1 ||
>-      now_ms - time_last_bitrate_decrease_ >= initial_backoff_interval_ms_) {
>+  if (time_last_bitrate_decrease_.IsInfinite() ||
>+      at_time - time_last_bitrate_decrease_ >= initial_backoff_interval_) {
>     return true;
>   }
>   return false;
> }
> 
>-uint32_t AimdRateControl::LatestEstimate() const {
>-  return current_bitrate_bps_;
>+DataRate AimdRateControl::LatestEstimate() const {
>+  return current_bitrate_;
> }
> 
>-void AimdRateControl::SetRtt(int64_t rtt) {
>+void AimdRateControl::SetRtt(TimeDelta rtt) {
>   rtt_ = rtt;
> }
> 
>-uint32_t AimdRateControl::Update(const RateControlInput* input,
>-                                 int64_t now_ms) {
>+DataRate AimdRateControl::Update(const RateControlInput* input,
>+                                 Timestamp at_time) {
>   RTC_CHECK(input);
> 
>   // Set the initial bit rate value to what we're receiving the first half
>   // second.
>   // TODO(bugs.webrtc.org/9379): The comment above doesn't match to the code.
>   if (!bitrate_is_initialized_) {
>-    const int64_t kInitializationTimeMs = 5000;
>-    RTC_DCHECK_LE(kBitrateWindowMs, kInitializationTimeMs);
>-    if (time_first_throughput_estimate_ < 0) {
>-      if (input->estimated_throughput_bps)
>-        time_first_throughput_estimate_ = now_ms;
>-    } else if (now_ms - time_first_throughput_estimate_ >
>-                   kInitializationTimeMs &&
>-               input->estimated_throughput_bps) {
>-      current_bitrate_bps_ = *input->estimated_throughput_bps;
>+    const TimeDelta kInitializationTime = TimeDelta::seconds(5);
>+    RTC_DCHECK_LE(kBitrateWindowMs, kInitializationTime.ms());
>+    if (time_first_throughput_estimate_.IsInfinite()) {
>+      if (input->estimated_throughput)
>+        time_first_throughput_estimate_ = at_time;
>+    } else if (at_time - time_first_throughput_estimate_ >
>+                   kInitializationTime &&
>+               input->estimated_throughput) {
>+      current_bitrate_ = *input->estimated_throughput;
>       bitrate_is_initialized_ = true;
>     }
>   }
> 
>-  current_bitrate_bps_ = ChangeBitrate(current_bitrate_bps_, *input, now_ms);
>-  return current_bitrate_bps_;
>+  current_bitrate_ = ChangeBitrate(current_bitrate_, *input, at_time);
>+  return current_bitrate_;
> }
> 
>-void AimdRateControl::SetEstimate(int bitrate_bps, int64_t now_ms) {
>+void AimdRateControl::SetEstimate(DataRate bitrate, Timestamp at_time) {
>   bitrate_is_initialized_ = true;
>-  uint32_t prev_bitrate_bps = current_bitrate_bps_;
>-  current_bitrate_bps_ = ClampBitrate(bitrate_bps, bitrate_bps);
>-  time_last_bitrate_change_ = now_ms;
>-  if (current_bitrate_bps_ < prev_bitrate_bps) {
>-    time_last_bitrate_decrease_ = now_ms;
>+  DataRate prev_bitrate = current_bitrate_;
>+  current_bitrate_ = ClampBitrate(bitrate, bitrate);
>+  time_last_bitrate_change_ = at_time;
>+  if (current_bitrate_ < prev_bitrate) {
>+    time_last_bitrate_decrease_ = at_time;
>   }
> }
> 
>-int AimdRateControl::GetNearMaxIncreaseRateBps() const {
>-  RTC_DCHECK_GT(current_bitrate_bps_, 0);
>-  double bits_per_frame = static_cast<double>(current_bitrate_bps_) / 30.0;
>-  double packets_per_frame = std::ceil(bits_per_frame / (8.0 * 1200.0));
>-  double avg_packet_size_bits = bits_per_frame / packets_per_frame;
>+double AimdRateControl::GetNearMaxIncreaseRateBpsPerSecond() const {
>+  RTC_DCHECK(!current_bitrate_.IsZero());
>+  const TimeDelta kFrameInterval = TimeDelta::seconds(1) / 30;
>+  DataSize frame_size = current_bitrate_ * kFrameInterval;
>+  const DataSize kPacketSize = DataSize::bytes(1200);
>+  double packets_per_frame = std::ceil(frame_size / kPacketSize);
>+  DataSize avg_packet_size = frame_size / packets_per_frame;
> 
>   // Approximate the over-use estimator delay to 100 ms.
>-  const int64_t response_time = in_experiment_ ? (rtt_ + 100) * 2 : rtt_ + 100;
>-  constexpr double kMinIncreaseRateBps = 4000;
>-  return static_cast<int>(std::max(
>-      kMinIncreaseRateBps, (avg_packet_size_bits * 1000) / response_time));
>+  TimeDelta response_time = rtt_ + TimeDelta::ms(100);
>+  if (in_experiment_)
>+    response_time = response_time * 2;
>+  double increase_rate_bps_per_second =
>+      (avg_packet_size / response_time).bps<double>();
>+  double kMinIncreaseRateBpsPerSecond = 4000;
>+  return std::max(kMinIncreaseRateBpsPerSecond, increase_rate_bps_per_second);
> }
> 
>-int AimdRateControl::GetExpectedBandwidthPeriodMs() const {
>-  const int kMinPeriodMs = smoothing_experiment_ ? 500 : 2000;
>-  constexpr int kDefaultPeriodMs = 3000;
>-  constexpr int kMaxPeriodMs = 50000;
>+TimeDelta AimdRateControl::GetExpectedBandwidthPeriod() const {
>+  const TimeDelta kMinPeriod =
>+      smoothing_experiment_ ? TimeDelta::ms(500) : TimeDelta::seconds(2);
>+  const TimeDelta kDefaultPeriod = TimeDelta::seconds(3);
>+  const TimeDelta kMaxPeriod = TimeDelta::seconds(50);
> 
>-  int increase_rate = GetNearMaxIncreaseRateBps();
>+  double increase_rate_bps_per_second = GetNearMaxIncreaseRateBpsPerSecond();
>   if (!last_decrease_)
>-    return smoothing_experiment_ ? kMinPeriodMs : kDefaultPeriodMs;
>-
>-  return std::min(kMaxPeriodMs,
>-                  std::max<int>(1000 * static_cast<int64_t>(*last_decrease_) /
>-                                    increase_rate,
>-                                kMinPeriodMs));
>+    return smoothing_experiment_ ? kMinPeriod : kDefaultPeriod;
>+  double time_to_recover_decrease_seconds =
>+      last_decrease_->bps() / increase_rate_bps_per_second;
>+  TimeDelta period = TimeDelta::seconds(time_to_recover_decrease_seconds);
>+  return period.Clamped(kMinPeriod, kMaxPeriod);
> }
> 
>-uint32_t AimdRateControl::ChangeBitrate(uint32_t new_bitrate_bps,
>+DataRate AimdRateControl::ChangeBitrate(DataRate new_bitrate,
>                                         const RateControlInput& input,
>-                                        int64_t now_ms) {
>-  uint32_t estimated_throughput_bps =
>-      input.estimated_throughput_bps.value_or(latest_estimated_throughput_bps_);
>-  if (input.estimated_throughput_bps)
>-    latest_estimated_throughput_bps_ = *input.estimated_throughput_bps;
>+                                        Timestamp at_time) {
>+  DataRate estimated_throughput =
>+      input.estimated_throughput.value_or(latest_estimated_throughput_);
>+  if (input.estimated_throughput)
>+    latest_estimated_throughput_ = *input.estimated_throughput;
> 
>   // An over-use should always trigger us to reduce the bitrate, even though
>   // we have not yet established our first estimate. By acting on the over-use,
>   // we will end up with a valid estimate.
>   if (!bitrate_is_initialized_ &&
>       input.bw_state != BandwidthUsage::kBwOverusing)
>-    return current_bitrate_bps_;
>+    return current_bitrate_;
> 
>-  ChangeState(input, now_ms);
>+  ChangeState(input, at_time);
>   // Calculated here because it's used in multiple places.
>-  const float estimated_throughput_kbps = estimated_throughput_bps / 1000.0f;
>+  const double estimated_throughput_kbps = estimated_throughput.kbps<double>();
>+
>   // Calculate the max bit rate std dev given the normalized
>-  // variance and the current throughput bitrate.
>-  const float std_max_bit_rate =
>-      sqrt(var_max_bitrate_kbps_ * avg_max_bitrate_kbps_);
>+  // variance and the current throughput bitrate. The standard deviation will
>+  // only be used if link_capacity_estimate_kbps_ has a value.
>+  const double std_link_capacity_kbps =
>+      sqrt(var_link_capacity_estimate_kbps_ *
>+           link_capacity_estimate_kbps_.value_or(0));
>   switch (rate_control_state_) {
>     case kRcHold:
>       break;
> 
>     case kRcIncrease:
>-      if (avg_max_bitrate_kbps_ >= 0 &&
>-          estimated_throughput_kbps >
>-              avg_max_bitrate_kbps_ + 3 * std_max_bit_rate) {
>-        rate_control_region_ = kRcMaxUnknown;
>-
>-        avg_max_bitrate_kbps_ = -1.0;
>+      if (link_capacity_estimate_kbps_.has_value() &&
>+          estimated_throughput_kbps > link_capacity_estimate_kbps_.value() +
>+                                          3 * std_link_capacity_kbps) {
>+        // The link capacity appears to have changed. Forget the previous
>+        // estimate and use multiplicative increase to quickly discover new
>+        // capacity.
>+        link_capacity_estimate_kbps_.reset();
>       }
>-      if (rate_control_region_ == kRcNearMax) {
>-        uint32_t additive_increase_bps =
>-            AdditiveRateIncrease(now_ms, time_last_bitrate_change_);
>-        new_bitrate_bps += additive_increase_bps;
>+      if (link_capacity_estimate_kbps_.has_value()) {
>+        // The link_capacity_estimate_kbps_ is reset if the measured throughput
>+        // is too far from the estimate. We can therefore assume that our target
>+        // rate is reasonably close to link capacity and use additive increase.
>+        DataRate additive_increase =
>+            AdditiveRateIncrease(at_time, time_last_bitrate_change_);
>+        new_bitrate += additive_increase;
>       } else {
>-        uint32_t multiplicative_increase_bps = MultiplicativeRateIncrease(
>-            now_ms, time_last_bitrate_change_, new_bitrate_bps);
>-        new_bitrate_bps += multiplicative_increase_bps;
>+        // If we don't have an estimate of the link capacity, use faster ramp up
>+        // to discover the capacity.
>+        DataRate multiplicative_increase = MultiplicativeRateIncrease(
>+            at_time, time_last_bitrate_change_, new_bitrate);
>+        new_bitrate += multiplicative_increase;
>       }
> 
>-      time_last_bitrate_change_ = now_ms;
>+      time_last_bitrate_change_ = at_time;
>       break;
> 
>     case kRcDecrease:
>-      // Set bit rate to something slightly lower than max
>+      // Set bit rate to something slightly lower than the measured throughput
>       // to get rid of any self-induced delay.
>-      new_bitrate_bps =
>-          static_cast<uint32_t>(beta_ * estimated_throughput_bps + 0.5);
>-      if (new_bitrate_bps > current_bitrate_bps_) {
>+      new_bitrate = estimated_throughput * beta_;
>+      if (new_bitrate > current_bitrate_) {
>         // Avoid increasing the rate when over-using.
>-        if (rate_control_region_ != kRcMaxUnknown) {
>-          new_bitrate_bps = static_cast<uint32_t>(
>-              beta_ * avg_max_bitrate_kbps_ * 1000 + 0.5f);
>+        if (link_capacity_estimate_kbps_.has_value()) {
>+          new_bitrate =
>+              DataRate::kbps(beta_ * link_capacity_estimate_kbps_.value());
>         }
>-        new_bitrate_bps = std::min(new_bitrate_bps, current_bitrate_bps_);
>+        new_bitrate = std::min(new_bitrate, current_bitrate_);
>       }
>-      rate_control_region_ = kRcNearMax;
> 
>-      if (bitrate_is_initialized_ &&
>-          estimated_throughput_bps < current_bitrate_bps_) {
>-        constexpr float kDegradationFactor = 0.9f;
>+      if (bitrate_is_initialized_ && estimated_throughput < current_bitrate_) {
>+        constexpr double kDegradationFactor = 0.9;
>         if (smoothing_experiment_ &&
>-            new_bitrate_bps <
>-                kDegradationFactor * beta_ * current_bitrate_bps_) {
>+            new_bitrate < kDegradationFactor * beta_ * current_bitrate_) {
>           // If bitrate decreases more than a normal back off after overuse, it
>           // indicates a real network degradation. We do not let such a decrease
>           // to determine the bandwidth estimation period.
>           last_decrease_ = absl::nullopt;
>         } else {
>-          last_decrease_ = current_bitrate_bps_ - new_bitrate_bps;
>+          last_decrease_ = current_bitrate_ - new_bitrate;
>         }
>       }
>-      if (estimated_throughput_kbps <
>-          avg_max_bitrate_kbps_ - 3 * std_max_bit_rate) {
>-        avg_max_bitrate_kbps_ = -1.0f;
>+      if (link_capacity_estimate_kbps_.has_value() &&
>+          estimated_throughput_kbps < link_capacity_estimate_kbps_.value() -
>+                                          3 * std_link_capacity_kbps) {
>+        // The current throughput is far from the estimated link capacity. Clear
>+        // the estimate to allow an fast update in UpdateLinkCapacityEstimate.
>+        link_capacity_estimate_kbps_.reset();
>       }
> 
>       bitrate_is_initialized_ = true;
>-      UpdateMaxThroughputEstimate(estimated_throughput_kbps);
>+      UpdateLinkCapacityEstimate(estimated_throughput_kbps);
>       // Stay on hold until the pipes are cleared.
>       rate_control_state_ = kRcHold;
>-      time_last_bitrate_change_ = now_ms;
>-      time_last_bitrate_decrease_ = now_ms;
>+      time_last_bitrate_change_ = at_time;
>+      time_last_bitrate_decrease_ = at_time;
>       break;
> 
>     default:
>       assert(false);
>   }
>-  return ClampBitrate(new_bitrate_bps, estimated_throughput_bps);
>+  return ClampBitrate(new_bitrate, estimated_throughput);
> }
> 
>-uint32_t AimdRateControl::ClampBitrate(
>-    uint32_t new_bitrate_bps,
>-    uint32_t estimated_throughput_bps) const {
>+DataRate AimdRateControl::ClampBitrate(DataRate new_bitrate,
>+                                       DataRate estimated_throughput) const {
>   // Don't change the bit rate if the send side is too far off.
>   // We allow a bit more lag at very low rates to not too easily get stuck if
>   // the encoder produces uneven outputs.
>-  const uint32_t max_bitrate_bps =
>-      static_cast<uint32_t>(1.5f * estimated_throughput_bps) + 10000;
>-  if (new_bitrate_bps > current_bitrate_bps_ &&
>-      new_bitrate_bps > max_bitrate_bps) {
>-    new_bitrate_bps = std::max(current_bitrate_bps_, max_bitrate_bps);
>+  const DataRate max_bitrate = 1.5 * estimated_throughput + DataRate::kbps(10);
>+  if (new_bitrate > current_bitrate_ && new_bitrate > max_bitrate) {
>+    new_bitrate = std::max(current_bitrate_, max_bitrate);
>   }
>-  new_bitrate_bps = std::max(new_bitrate_bps, min_configured_bitrate_bps_);
>-  return new_bitrate_bps;
>+  new_bitrate = std::max(new_bitrate, min_configured_bitrate_);
>+  return new_bitrate;
> }
> 
>-uint32_t AimdRateControl::MultiplicativeRateIncrease(
>-    int64_t now_ms,
>-    int64_t last_ms,
>-    uint32_t current_bitrate_bps) const {
>+DataRate AimdRateControl::MultiplicativeRateIncrease(
>+    Timestamp at_time,
>+    Timestamp last_time,
>+    DataRate current_bitrate) const {
>   double alpha = 1.08;
>-  if (last_ms > -1) {
>-    auto time_since_last_update_ms =
>-        rtc::SafeMin<int64_t>(now_ms - last_ms, 1000);
>-    alpha = pow(alpha, time_since_last_update_ms / 1000.0);
>+  if (last_time.IsFinite()) {
>+    auto time_since_last_update = at_time - last_time;
>+    alpha = pow(alpha, std::min(time_since_last_update.seconds<double>(), 1.0));
>   }
>-  uint32_t multiplicative_increase_bps =
>-      std::max(current_bitrate_bps * (alpha - 1.0), 1000.0);
>-  return multiplicative_increase_bps;
>+  DataRate multiplicative_increase =
>+      std::max(current_bitrate * (alpha - 1.0), DataRate::bps(1000));
>+  return multiplicative_increase;
> }
> 
>-uint32_t AimdRateControl::AdditiveRateIncrease(int64_t now_ms,
>-                                               int64_t last_ms) const {
>-  return static_cast<uint32_t>((now_ms - last_ms) *
>-                               GetNearMaxIncreaseRateBps() / 1000);
>+DataRate AimdRateControl::AdditiveRateIncrease(Timestamp at_time,
>+                                               Timestamp last_time) const {
>+  double time_period_seconds = (at_time - last_time).seconds<double>();
>+  double data_rate_increase_bps =
>+      GetNearMaxIncreaseRateBpsPerSecond() * time_period_seconds;
>+  return DataRate::bps(data_rate_increase_bps);
> }
> 
>-void AimdRateControl::UpdateMaxThroughputEstimate(
>-    float estimated_throughput_kbps) {
>-  const float alpha = 0.05f;
>-  if (avg_max_bitrate_kbps_ == -1.0f) {
>-    avg_max_bitrate_kbps_ = estimated_throughput_kbps;
>+void AimdRateControl::UpdateLinkCapacityEstimate(
>+    double estimated_throughput_kbps) {
>+  const double alpha = 0.05;
>+  if (!link_capacity_estimate_kbps_.has_value()) {
>+    link_capacity_estimate_kbps_ = estimated_throughput_kbps;
>   } else {
>-    avg_max_bitrate_kbps_ =
>-        (1 - alpha) * avg_max_bitrate_kbps_ + alpha * estimated_throughput_kbps;
>+    link_capacity_estimate_kbps_ =
>+        (1 - alpha) * link_capacity_estimate_kbps_.value() +
>+        alpha * estimated_throughput_kbps;
>   }
>-  // Estimate the max bit rate variance and normalize the variance
>-  // with the average max bit rate.
>-  const float norm = std::max(avg_max_bitrate_kbps_, 1.0f);
>-  var_max_bitrate_kbps_ =
>-      (1 - alpha) * var_max_bitrate_kbps_ +
>-      alpha * (avg_max_bitrate_kbps_ - estimated_throughput_kbps) *
>-          (avg_max_bitrate_kbps_ - estimated_throughput_kbps) / norm;
>+  // Estimate the variance of the link capacity estimate and normalize the
>+  // variance with the link capacity estimate.
>+  const double norm = std::max(link_capacity_estimate_kbps_.value(), 1.0);
>+  var_link_capacity_estimate_kbps_ =
>+      (1 - alpha) * var_link_capacity_estimate_kbps_ +
>+      alpha *
>+          (link_capacity_estimate_kbps_.value() - estimated_throughput_kbps) *
>+          (link_capacity_estimate_kbps_.value() - estimated_throughput_kbps) /
>+          norm;
>   // 0.4 ~= 14 kbit/s at 500 kbit/s
>-  if (var_max_bitrate_kbps_ < 0.4f) {
>-    var_max_bitrate_kbps_ = 0.4f;
>+  if (var_link_capacity_estimate_kbps_ < 0.4) {
>+    var_link_capacity_estimate_kbps_ = 0.4;
>   }
>-  // 2.5f ~= 35 kbit/s at 500 kbit/s
>-  if (var_max_bitrate_kbps_ > 2.5f) {
>-    var_max_bitrate_kbps_ = 2.5f;
>+  // 2.5 ~= 35 kbit/s at 500 kbit/s
>+  if (var_link_capacity_estimate_kbps_ > 2.5) {
>+    var_link_capacity_estimate_kbps_ = 2.5;
>   }
> }
> 
> void AimdRateControl::ChangeState(const RateControlInput& input,
>-                                  int64_t now_ms) {
>+                                  Timestamp at_time) {
>   switch (input.bw_state) {
>     case BandwidthUsage::kBwNormal:
>       if (rate_control_state_ == kRcHold) {
>-        time_last_bitrate_change_ = now_ms;
>+        time_last_bitrate_change_ = at_time;
>         rate_control_state_ = kRcIncrease;
>       }
>       break;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
>index 2a3f71d98ae8a7558a152fd754eb817b41960946..adb959a14b9c38b058e5e061afb9edace5ffbdcc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
>@@ -11,8 +11,13 @@
> #ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_AIMD_RATE_CONTROL_H_
> #define MODULES_REMOTE_BITRATE_ESTIMATOR_AIMD_RATE_CONTROL_H_
> 
>+#include <stdint.h>
>+
>+#include "absl/types/optional.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>-#include "rtc_base/constructormagic.h"
>+
>+#include "api/units/data_rate.h"
>+#include "api/units/timestamp.h"
> 
> namespace webrtc {
> 
>@@ -30,28 +35,28 @@ class AimdRateControl {
>   // either if it has been explicitly set via SetStartBitrate/SetEstimate, or if
>   // we have measured a throughput.
>   bool ValidEstimate() const;
>-  void SetStartBitrate(int start_bitrate_bps);
>-  void SetMinBitrate(int min_bitrate_bps);
>-  int64_t GetFeedbackInterval() const;
>+  void SetStartBitrate(DataRate start_bitrate);
>+  void SetMinBitrate(DataRate min_bitrate);
>+  TimeDelta GetFeedbackInterval() const;
> 
>   // Returns true if the bitrate estimate hasn't been changed for more than
>   // an RTT, or if the estimated_throughput is less than half of the current
>   // estimate. Should be used to decide if we should reduce the rate further
>   // when over-using.
>-  bool TimeToReduceFurther(int64_t now_ms,
>-                           uint32_t estimated_throughput_bps) const;
>+  bool TimeToReduceFurther(Timestamp at_time,
>+                           DataRate estimated_throughput) const;
>   // As above. To be used if overusing before we have measured a throughput.
>-  bool InitialTimeToReduceFurther(int64_t now_ms) const;
>+  bool InitialTimeToReduceFurther(Timestamp at_time) const;
> 
>-  uint32_t LatestEstimate() const;
>-  void SetRtt(int64_t rtt);
>-  uint32_t Update(const RateControlInput* input, int64_t now_ms);
>-  void SetEstimate(int bitrate_bps, int64_t now_ms);
>+  DataRate LatestEstimate() const;
>+  void SetRtt(TimeDelta rtt);
>+  DataRate Update(const RateControlInput* input, Timestamp at_time);
>+  void SetEstimate(DataRate bitrate, Timestamp at_time);
> 
>   // Returns the increase rate when used bandwidth is near the link capacity.
>-  int GetNearMaxIncreaseRateBps() const;
>+  double GetNearMaxIncreaseRateBpsPerSecond() const;
>   // Returns the expected time between overuse signals (assuming steady state).
>-  int GetExpectedBandwidthPeriodMs() const;
>+  TimeDelta GetExpectedBandwidthPeriod() const;
> 
>  private:
>   friend class GoogCcStatePrinter;
>@@ -62,41 +67,40 @@ class AimdRateControl {
>   // in the "decrease" state the bitrate will be decreased to slightly below the
>   // current throughput. When in the "hold" state the bitrate will be kept
>   // constant to allow built up queues to drain.
>-  uint32_t ChangeBitrate(uint32_t current_bitrate,
>+  DataRate ChangeBitrate(DataRate current_bitrate,
>                          const RateControlInput& input,
>-                         int64_t now_ms);
>-  // Clamps new_bitrate_bps to within the configured min bitrate and a linear
>+                         Timestamp at_time);
>+  // Clamps new_bitrate to within the configured min bitrate and a linear
>   // function of the throughput, so that the new bitrate can't grow too
>   // large compared to the bitrate actually being received by the other end.
>-  uint32_t ClampBitrate(uint32_t new_bitrate_bps,
>-                        uint32_t estimated_throughput_bps) const;
>-  uint32_t MultiplicativeRateIncrease(int64_t now_ms,
>-                                      int64_t last_ms,
>-                                      uint32_t current_bitrate_bps) const;
>-  uint32_t AdditiveRateIncrease(int64_t now_ms, int64_t last_ms) const;
>-  void UpdateChangePeriod(int64_t now_ms);
>-  void UpdateMaxThroughputEstimate(float estimated_throughput_kbps);
>-  void ChangeState(const RateControlInput& input, int64_t now_ms);
>+  DataRate ClampBitrate(DataRate new_bitrate,
>+                        DataRate estimated_throughput) const;
>+  DataRate MultiplicativeRateIncrease(Timestamp at_time,
>+                                      Timestamp last_ms,
>+                                      DataRate current_bitrate) const;
>+  DataRate AdditiveRateIncrease(Timestamp at_time, Timestamp last_time) const;
>+  void UpdateChangePeriod(Timestamp at_time);
>+  void UpdateLinkCapacityEstimate(double estimated_throughput_kbps);
>+  void ChangeState(const RateControlInput& input, Timestamp at_time);
> 
>-  uint32_t min_configured_bitrate_bps_;
>-  uint32_t max_configured_bitrate_bps_;
>-  uint32_t current_bitrate_bps_;
>-  uint32_t latest_estimated_throughput_bps_;
>-  float avg_max_bitrate_kbps_;
>-  float var_max_bitrate_kbps_;
>+  DataRate min_configured_bitrate_;
>+  DataRate max_configured_bitrate_;
>+  DataRate current_bitrate_;
>+  DataRate latest_estimated_throughput_;
>+  absl::optional<double> link_capacity_estimate_kbps_;
>+  double var_link_capacity_estimate_kbps_;
>   RateControlState rate_control_state_;
>-  RateControlRegion rate_control_region_;
>-  int64_t time_last_bitrate_change_;
>-  int64_t time_last_bitrate_decrease_;
>-  int64_t time_first_throughput_estimate_;
>+  Timestamp time_last_bitrate_change_;
>+  Timestamp time_last_bitrate_decrease_;
>+  Timestamp time_first_throughput_estimate_;
>   bool bitrate_is_initialized_;
>-  float beta_;
>-  int64_t rtt_;
>+  double beta_;
>+  TimeDelta rtt_;
>   const bool in_experiment_;
>   const bool smoothing_experiment_;
>   const bool in_initial_backoff_interval_experiment_;
>-  int64_t initial_backoff_interval_ms_;
>-  absl::optional<int> last_decrease_;
>+  TimeDelta initial_backoff_interval_;
>+  absl::optional<DataRate> last_decrease_;
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
>index 5cde7cedaf492dbda5a3f3bb0f30d7e11e5e0072..4bec9e8ff500781d253f5cd3475a627b202d4b62 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
>@@ -40,13 +40,26 @@ AimdRateControlStates CreateAimdRateControlStates() {
>   states.simulated_clock.reset(new SimulatedClock(kClockInitialTime));
>   return states;
> }
>-
>+absl::optional<DataRate> OptionalRateFromOptionalBps(
>+    absl::optional<int> bitrate_bps) {
>+  if (bitrate_bps) {
>+    return DataRate::bps(*bitrate_bps);
>+  } else {
>+    return absl::nullopt;
>+  }
>+}
> void UpdateRateControl(const AimdRateControlStates& states,
>                        const BandwidthUsage& bandwidth_usage,
>                        absl::optional<uint32_t> throughput_estimate,
>                        int64_t now_ms) {
>-  RateControlInput input(bandwidth_usage, throughput_estimate);
>-  states.aimd_rate_control->Update(&input, now_ms);
>+  RateControlInput input(bandwidth_usage,
>+                         OptionalRateFromOptionalBps(throughput_estimate));
>+  states.aimd_rate_control->Update(&input, Timestamp::ms(now_ms));
>+}
>+void SetEstimate(const AimdRateControlStates& states, int bitrate_bps) {
>+  states.aimd_rate_control->SetEstimate(
>+      DataRate::bps(bitrate_bps),
>+      Timestamp::ms(states.simulated_clock->TimeInMilliseconds()));
> }
> 
> }  // namespace
>@@ -54,40 +67,40 @@ void UpdateRateControl(const AimdRateControlStates& states,
> TEST(AimdRateControlTest, MinNearMaxIncreaseRateOnLowBandwith) {
>   auto states = CreateAimdRateControlStates();
>   constexpr int kBitrate = 30000;
>-  states.aimd_rate_control->SetEstimate(
>-      kBitrate, states.simulated_clock->TimeInMilliseconds());
>-  EXPECT_EQ(4000, states.aimd_rate_control->GetNearMaxIncreaseRateBps());
>+  SetEstimate(states, kBitrate);
>+  EXPECT_EQ(4000,
>+            states.aimd_rate_control->GetNearMaxIncreaseRateBpsPerSecond());
> }
> 
> TEST(AimdRateControlTest, NearMaxIncreaseRateIs5kbpsOn90kbpsAnd200msRtt) {
>   auto states = CreateAimdRateControlStates();
>   constexpr int kBitrate = 90000;
>-  states.aimd_rate_control->SetEstimate(
>-      kBitrate, states.simulated_clock->TimeInMilliseconds());
>-  EXPECT_EQ(5000, states.aimd_rate_control->GetNearMaxIncreaseRateBps());
>+  SetEstimate(states, kBitrate);
>+  EXPECT_EQ(5000,
>+            states.aimd_rate_control->GetNearMaxIncreaseRateBpsPerSecond());
> }
> 
> TEST(AimdRateControlTest, NearMaxIncreaseRateIs5kbpsOn60kbpsAnd100msRtt) {
>   auto states = CreateAimdRateControlStates();
>   constexpr int kBitrate = 60000;
>-  states.aimd_rate_control->SetEstimate(
>-      kBitrate, states.simulated_clock->TimeInMilliseconds());
>-  states.aimd_rate_control->SetRtt(100);
>-  EXPECT_EQ(5000, states.aimd_rate_control->GetNearMaxIncreaseRateBps());
>+  SetEstimate(states, kBitrate);
>+  states.aimd_rate_control->SetRtt(TimeDelta::ms(100));
>+  EXPECT_EQ(5000,
>+            states.aimd_rate_control->GetNearMaxIncreaseRateBpsPerSecond());
> }
> 
> TEST(AimdRateControlTest, GetIncreaseRateAndBandwidthPeriod) {
>   // Smoothing experiment disabled
>   auto states = CreateAimdRateControlStates();
>   constexpr int kBitrate = 300000;
>-  states.aimd_rate_control->SetEstimate(
>-      kBitrate, states.simulated_clock->TimeInMilliseconds());
>+  SetEstimate(states, kBitrate);
>   UpdateRateControl(states, BandwidthUsage::kBwOverusing, kBitrate,
>                     states.simulated_clock->TimeInMilliseconds());
>-  EXPECT_NEAR(14000, states.aimd_rate_control->GetNearMaxIncreaseRateBps(),
>+  EXPECT_NEAR(14000,
>+              states.aimd_rate_control->GetNearMaxIncreaseRateBpsPerSecond(),
>               1000);
>   EXPECT_EQ(kDefaultPeriodMsNoSmoothingExp,
>-            states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
>+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
> }
> 
> TEST(AimdRateControlTest, GetIncreaseRateAndBandwidthPeriodSmoothingExp) {
>@@ -95,21 +108,20 @@ TEST(AimdRateControlTest, GetIncreaseRateAndBandwidthPeriodSmoothingExp) {
>   test::ScopedFieldTrials override_field_trials(kSmoothingExpFieldTrial);
>   auto states = CreateAimdRateControlStates();
>   constexpr int kBitrate = 300000;
>-  states.aimd_rate_control->SetEstimate(
>-      kBitrate, states.simulated_clock->TimeInMilliseconds());
>+  SetEstimate(states, kBitrate);
>   UpdateRateControl(states, BandwidthUsage::kBwOverusing, kBitrate,
>                     states.simulated_clock->TimeInMilliseconds());
>-  EXPECT_NEAR(14000, states.aimd_rate_control->GetNearMaxIncreaseRateBps(),
>+  EXPECT_NEAR(14000,
>+              states.aimd_rate_control->GetNearMaxIncreaseRateBpsPerSecond(),
>               1000);
>   EXPECT_EQ(kMinBwePeriodMsSmoothingExp,
>-            states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
>+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
> }
> 
> TEST(AimdRateControlTest, BweLimitedByAckedBitrate) {
>   auto states = CreateAimdRateControlStates();
>   constexpr int kAckedBitrate = 10000;
>-  states.aimd_rate_control->SetEstimate(
>-      kAckedBitrate, states.simulated_clock->TimeInMilliseconds());
>+  SetEstimate(states, kAckedBitrate);
>   while (states.simulated_clock->TimeInMilliseconds() - kClockInitialTime <
>          20000) {
>     UpdateRateControl(states, BandwidthUsage::kBwNormal, kAckedBitrate,
>@@ -118,14 +130,13 @@ TEST(AimdRateControlTest, BweLimitedByAckedBitrate) {
>   }
>   ASSERT_TRUE(states.aimd_rate_control->ValidEstimate());
>   EXPECT_EQ(static_cast<uint32_t>(1.5 * kAckedBitrate + 10000),
>-            states.aimd_rate_control->LatestEstimate());
>+            states.aimd_rate_control->LatestEstimate().bps());
> }
> 
> TEST(AimdRateControlTest, BweNotLimitedByDecreasingAckedBitrate) {
>   auto states = CreateAimdRateControlStates();
>   constexpr int kAckedBitrate = 100000;
>-  states.aimd_rate_control->SetEstimate(
>-      kAckedBitrate, states.simulated_clock->TimeInMilliseconds());
>+  SetEstimate(states, kAckedBitrate);
>   while (states.simulated_clock->TimeInMilliseconds() - kClockInitialTime <
>          20000) {
>     UpdateRateControl(states, BandwidthUsage::kBwNormal, kAckedBitrate,
>@@ -135,10 +146,10 @@ TEST(AimdRateControlTest, BweNotLimitedByDecreasingAckedBitrate) {
>   ASSERT_TRUE(states.aimd_rate_control->ValidEstimate());
>   // If the acked bitrate decreases the BWE shouldn't be reduced to 1.5x
>   // what's being acked, but also shouldn't get to increase more.
>-  uint32_t prev_estimate = states.aimd_rate_control->LatestEstimate();
>+  uint32_t prev_estimate = states.aimd_rate_control->LatestEstimate().bps();
>   UpdateRateControl(states, BandwidthUsage::kBwNormal, kAckedBitrate / 2,
>                     states.simulated_clock->TimeInMilliseconds());
>-  uint32_t new_estimate = states.aimd_rate_control->LatestEstimate();
>+  uint32_t new_estimate = states.aimd_rate_control->LatestEstimate().bps();
>   EXPECT_NEAR(new_estimate, static_cast<uint32_t>(1.5 * kAckedBitrate + 10000),
>               2000);
>   EXPECT_EQ(new_estimate, prev_estimate);
>@@ -147,35 +158,34 @@ TEST(AimdRateControlTest, BweNotLimitedByDecreasingAckedBitrate) {
> TEST(AimdRateControlTest, DefaultPeriodUntilFirstOveruse) {
>   // Smoothing experiment disabled
>   auto states = CreateAimdRateControlStates();
>-  states.aimd_rate_control->SetStartBitrate(300000);
>+  states.aimd_rate_control->SetStartBitrate(DataRate::kbps(300));
>   EXPECT_EQ(kDefaultPeriodMsNoSmoothingExp,
>-            states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
>+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
>   states.simulated_clock->AdvanceTimeMilliseconds(100);
>   UpdateRateControl(states, BandwidthUsage::kBwOverusing, 280000,
>                     states.simulated_clock->TimeInMilliseconds());
>   EXPECT_NE(kDefaultPeriodMsNoSmoothingExp,
>-            states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
>+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
> }
> 
> TEST(AimdRateControlTest, MinPeriodUntilFirstOveruseSmoothingExp) {
>   // Smoothing experiment enabled
>   test::ScopedFieldTrials override_field_trials(kSmoothingExpFieldTrial);
>   auto states = CreateAimdRateControlStates();
>-  states.aimd_rate_control->SetStartBitrate(300000);
>+  states.aimd_rate_control->SetStartBitrate(DataRate::kbps(300));
>   EXPECT_EQ(kMinBwePeriodMsSmoothingExp,
>-            states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
>+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
>   states.simulated_clock->AdvanceTimeMilliseconds(100);
>   UpdateRateControl(states, BandwidthUsage::kBwOverusing, 280000,
>                     states.simulated_clock->TimeInMilliseconds());
>   EXPECT_NE(kMinBwePeriodMsSmoothingExp,
>-            states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
>+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
> }
> 
> TEST(AimdRateControlTest, ExpectedPeriodAfter20kbpsDropAnd5kbpsIncrease) {
>   auto states = CreateAimdRateControlStates();
>   constexpr int kInitialBitrate = 110000;
>-  states.aimd_rate_control->SetEstimate(
>-      kInitialBitrate, states.simulated_clock->TimeInMilliseconds());
>+  SetEstimate(states, kInitialBitrate);
>   states.simulated_clock->AdvanceTimeMilliseconds(100);
>   // Make the bitrate drop by 20 kbps to get to 90 kbps.
>   // The rate increase at 90 kbps should be 5 kbps, so the period should be 4 s.
>@@ -183,8 +193,9 @@ TEST(AimdRateControlTest, ExpectedPeriodAfter20kbpsDropAnd5kbpsIncrease) {
>       (kInitialBitrate - 20000) / kFractionAfterOveruse;
>   UpdateRateControl(states, BandwidthUsage::kBwOverusing, kAckedBitrate,
>                     states.simulated_clock->TimeInMilliseconds());
>-  EXPECT_EQ(5000, states.aimd_rate_control->GetNearMaxIncreaseRateBps());
>-  EXPECT_EQ(4000, states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
>+  EXPECT_EQ(5000,
>+            states.aimd_rate_control->GetNearMaxIncreaseRateBpsPerSecond());
>+  EXPECT_EQ(4000, states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
> }
> 
> TEST(AimdRateControlTest, MinPeriodAfterLargeBitrateDecreaseSmoothingExp) {
>@@ -192,8 +203,7 @@ TEST(AimdRateControlTest, MinPeriodAfterLargeBitrateDecreaseSmoothingExp) {
>   test::ScopedFieldTrials override_field_trials(kSmoothingExpFieldTrial);
>   auto states = CreateAimdRateControlStates();
>   constexpr int kInitialBitrate = 110000;
>-  states.aimd_rate_control->SetEstimate(
>-      kInitialBitrate, states.simulated_clock->TimeInMilliseconds());
>+  SetEstimate(states, kInitialBitrate);
>   states.simulated_clock->AdvanceTimeMilliseconds(100);
>   // Make such a large drop in bitrate that should be treated as network
>   // degradation.
>@@ -201,20 +211,19 @@ TEST(AimdRateControlTest, MinPeriodAfterLargeBitrateDecreaseSmoothingExp) {
>   UpdateRateControl(states, BandwidthUsage::kBwOverusing, kAckedBitrate,
>                     states.simulated_clock->TimeInMilliseconds());
>   EXPECT_EQ(kMinBwePeriodMsSmoothingExp,
>-            states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
>+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
> }
> 
> TEST(AimdRateControlTest, BandwidthPeriodIsNotBelowMin) {
>   auto states = CreateAimdRateControlStates();
>   constexpr int kInitialBitrate = 10000;
>-  states.aimd_rate_control->SetEstimate(
>-      kInitialBitrate, states.simulated_clock->TimeInMilliseconds());
>+  SetEstimate(states, kInitialBitrate);
>   states.simulated_clock->AdvanceTimeMilliseconds(100);
>   // Make a small (1.5 kbps) bitrate drop to 8.5 kbps.
>   UpdateRateControl(states, BandwidthUsage::kBwOverusing, kInitialBitrate - 1,
>                     states.simulated_clock->TimeInMilliseconds());
>   EXPECT_EQ(kMinBwePeriodMsNoSmoothingExp,
>-            states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
>+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
> }
> 
> TEST(AimdRateControlTest, BandwidthPeriodIsNotAboveMaxSmoothingExp) {
>@@ -222,29 +231,27 @@ TEST(AimdRateControlTest, BandwidthPeriodIsNotAboveMaxSmoothingExp) {
>   test::ScopedFieldTrials override_field_trials(kSmoothingExpFieldTrial);
>   auto states = CreateAimdRateControlStates();
>   constexpr int kInitialBitrate = 50000000;
>-  states.aimd_rate_control->SetEstimate(
>-      kInitialBitrate, states.simulated_clock->TimeInMilliseconds());
>+  SetEstimate(states, kInitialBitrate);
>   states.simulated_clock->AdvanceTimeMilliseconds(100);
>   // Make a large (10 Mbps) bitrate drop to 10 kbps.
>   constexpr int kAckedBitrate = 40000000 / kFractionAfterOveruse;
>   UpdateRateControl(states, BandwidthUsage::kBwOverusing, kAckedBitrate,
>                     states.simulated_clock->TimeInMilliseconds());
>   EXPECT_EQ(kMaxBwePeriodMs,
>-            states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
>+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
> }
> 
> TEST(AimdRateControlTest, BandwidthPeriodIsNotAboveMaxNoSmoothingExp) {
>   auto states = CreateAimdRateControlStates();
>   constexpr int kInitialBitrate = 10010000;
>-  states.aimd_rate_control->SetEstimate(
>-      kInitialBitrate, states.simulated_clock->TimeInMilliseconds());
>+  SetEstimate(states, kInitialBitrate);
>   states.simulated_clock->AdvanceTimeMilliseconds(100);
>   // Make a large (10 Mbps) bitrate drop to 10 kbps.
>   constexpr int kAckedBitrate = 10000 / kFractionAfterOveruse;
>   UpdateRateControl(states, BandwidthUsage::kBwOverusing, kAckedBitrate,
>                     states.simulated_clock->TimeInMilliseconds());
>   EXPECT_EQ(kMaxBwePeriodMs,
>-            states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
>+            states.aimd_rate_control->GetExpectedBandwidthPeriod().ms());
> }
> 
> TEST(AimdRateControlTest, SendingRateBoundedWhenThroughputNotEstimated) {
>@@ -265,7 +272,7 @@ TEST(AimdRateControlTest, SendingRateBoundedWhenThroughputNotEstimated) {
>                       states.simulated_clock->TimeInMilliseconds());
>     states.simulated_clock->AdvanceTimeMilliseconds(100);
>   }
>-  EXPECT_LE(states.aimd_rate_control->LatestEstimate(),
>+  EXPECT_LE(states.aimd_rate_control->LatestEstimate().bps(),
>             kInitialBitrateBps * 1.5 + 10000);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_defines.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_defines.cc
>index 6cbe46834896771863a38a073d5d190c545b3b3c..91f3cd4050b78f0ff72d246f05fb38317f74a2e7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_defines.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_defines.cc
>@@ -34,8 +34,8 @@ DataRate GetMinBitrate() {
> 
> RateControlInput::RateControlInput(
>     BandwidthUsage bw_state,
>-    const absl::optional<uint32_t>& estimated_throughput_bps)
>-    : bw_state(bw_state), estimated_throughput_bps(estimated_throughput_bps) {}
>+    const absl::optional<DataRate>& estimated_throughput)
>+    : bw_state(bw_state), estimated_throughput(estimated_throughput) {}
> 
> RateControlInput::~RateControlInput() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
>index d9185de99cadb0f0c4434586ce64768b3f8bfd1c..5223a6b424b1a0057eeaecd2acc55a1a41f478b0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
>@@ -11,6 +11,8 @@
> #ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
> #define MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
> 
>+#include <stdint.h>
>+
> #include "absl/types/optional.h"
> #include "api/units/data_rate.h"
> 
>@@ -45,15 +47,13 @@ enum class BandwidthUsage {
> 
> enum RateControlState { kRcHold, kRcIncrease, kRcDecrease };
> 
>-enum RateControlRegion { kRcNearMax, kRcAboveMax, kRcMaxUnknown };
>-
> struct RateControlInput {
>   RateControlInput(BandwidthUsage bw_state,
>-                   const absl::optional<uint32_t>& estimated_throughput_bps);
>+                   const absl::optional<DataRate>& estimated_throughput);
>   ~RateControlInput();
> 
>   BandwidthUsage bw_state;
>-  absl::optional<uint32_t> estimated_throughput_bps;
>+  absl::optional<DataRate> estimated_throughput;
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
>index 3a0f579052f90f4d90499c516941718fa740dbcd..b8e683b89a6f2efc2f39580394c136309dd675ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
>@@ -10,10 +10,9 @@
> 
> #include "modules/remote_bitrate_estimator/inter_arrival.h"
> 
>-#include <algorithm>
> #include <cassert>
> 
>-#include "modules/include/module_common_types.h"
>+#include "modules/include/module_common_types_public.h"
> #include "rtc_base/logging.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
>index 776ca42a56ff2203262b0149a62958e2b741c1e6..f3dbe1e998829580047d951513cdd673d9b93c86 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
>@@ -12,15 +12,12 @@
> 
> #include <math.h>
> #include <stdio.h>
>-#include <stdlib.h>
>-
> #include <algorithm>
> #include <string>
> 
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_minmax.h"
> #include "system_wrappers/include/field_trial.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
>index f322c30496097fc15b63028351ade6e0efd7f283..61d1c3f707e3206a139dff33e07421601369c402 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
>@@ -10,9 +10,8 @@
> #ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_OVERUSE_DETECTOR_H_
> #define MODULES_REMOTE_BITRATE_ESTIMATOR_OVERUSE_DETECTOR_H_
> 
>-#include <list>
>+#include <stdint.h>
> 
>-#include "modules/include/module_common_types.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "rtc_base/constructormagic.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
>index 09de5c6e7c1c6279cebd135db1c41ce475c8dd66..206ceba84c2468e6f8359b6b9004e962f0ab5b70 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
>@@ -12,9 +12,7 @@
> 
> #include <assert.h>
> #include <math.h>
>-#include <stdlib.h>
> #include <string.h>
>-
> #include <algorithm>
> 
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h
>index 4c924938aa0b864c08188c6464aff528d26f5a4d..3d7bd16be47e7379923040dc5a868592ff761452 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h
>@@ -10,6 +10,7 @@
> #ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_OVERUSE_ESTIMATOR_H_
> #define MODULES_REMOTE_BITRATE_ESTIMATOR_OVERUSE_ESTIMATOR_H_
> 
>+#include <stdint.h>
> #include <deque>
> 
> #include "common_types.h"  // NOLINT(build/include)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
>index a7cfe4c90acc6bb0f00b0a7a242c93e896bd2f75..1ad35c7bf0db741bd6a7253f6858a7fa28dabb65 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
>@@ -22,6 +22,16 @@
> #include "system_wrappers/include/metrics.h"
> 
> namespace webrtc {
>+namespace {
>+absl::optional<DataRate> OptionalRateFromOptionalBps(
>+    absl::optional<int> bitrate_bps) {
>+  if (bitrate_bps) {
>+    return DataRate::bps(*bitrate_bps);
>+  } else {
>+    return absl::nullopt;
>+  }
>+}
>+}  // namespace
> 
> enum {
>   kTimestampGroupLengthMs = 5,
>@@ -188,7 +198,8 @@ RemoteBitrateEstimatorAbsSendTime::ProcessClusters(int64_t now_ms) {
>                        << " bps. Mean send delta: " << best_it->send_mean_ms
>                        << " ms, mean recv delta: " << best_it->recv_mean_ms
>                        << " ms, num probes: " << best_it->count;
>-      remote_rate_.SetEstimate(probe_bitrate_bps, now_ms);
>+      remote_rate_.SetEstimate(DataRate::bps(probe_bitrate_bps),
>+                               Timestamp::ms(now_ms));
>       return ProbeResult::kBitrateUpdated;
>     }
>   }
>@@ -205,7 +216,7 @@ bool RemoteBitrateEstimatorAbsSendTime::IsBitrateImproving(
>   bool initial_probe = !remote_rate_.ValidEstimate() && new_bitrate_bps > 0;
>   bool bitrate_above_estimate =
>       remote_rate_.ValidEstimate() &&
>-      new_bitrate_bps > static_cast<int>(remote_rate_.LatestEstimate());
>+      new_bitrate_bps > remote_rate_.LatestEstimate().bps<int>();
>   return initial_probe || bitrate_above_estimate;
> }
> 
>@@ -316,13 +327,14 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
>       // Check if it's time for a periodic update or if we should update because
>       // of an over-use.
>       if (last_update_ms_ == -1 ||
>-          now_ms - last_update_ms_ > remote_rate_.GetFeedbackInterval()) {
>+          now_ms - last_update_ms_ > remote_rate_.GetFeedbackInterval().ms()) {
>         update_estimate = true;
>       } else if (detector_.State() == BandwidthUsage::kBwOverusing) {
>         absl::optional<uint32_t> incoming_rate =
>             incoming_bitrate_.Rate(arrival_time_ms);
>         if (incoming_rate &&
>-            remote_rate_.TimeToReduceFurther(now_ms, *incoming_rate)) {
>+            remote_rate_.TimeToReduceFurther(Timestamp::ms(now_ms),
>+                                             DataRate::bps(*incoming_rate))) {
>           update_estimate = true;
>         }
>       }
>@@ -332,9 +344,11 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
>       // The first overuse should immediately trigger a new estimate.
>       // We also have to update the estimate immediately if we are overusing
>       // and the target bitrate is too high compared to what we are receiving.
>-      const RateControlInput input(detector_.State(),
>-                                   incoming_bitrate_.Rate(arrival_time_ms));
>-      target_bitrate_bps = remote_rate_.Update(&input, now_ms);
>+      const RateControlInput input(
>+          detector_.State(),
>+          OptionalRateFromOptionalBps(incoming_bitrate_.Rate(arrival_time_ms)));
>+      target_bitrate_bps =
>+          remote_rate_.Update(&input, Timestamp::ms(now_ms)).bps<uint32_t>();
>       update_estimate = remote_rate_.ValidEstimate();
>       ssrcs = Keys(ssrcs_);
>     }
>@@ -374,7 +388,7 @@ void RemoteBitrateEstimatorAbsSendTime::TimeoutStreams(int64_t now_ms) {
> void RemoteBitrateEstimatorAbsSendTime::OnRttUpdate(int64_t avg_rtt_ms,
>                                                     int64_t max_rtt_ms) {
>   rtc::CritScope lock(&crit_);
>-  remote_rate_.SetRtt(avg_rtt_ms);
>+  remote_rate_.SetRtt(TimeDelta::ms(avg_rtt_ms));
> }
> 
> void RemoteBitrateEstimatorAbsSendTime::RemoveStream(uint32_t ssrc) {
>@@ -399,7 +413,7 @@ bool RemoteBitrateEstimatorAbsSendTime::LatestEstimate(
>   if (ssrcs_.empty()) {
>     *bitrate_bps = 0;
>   } else {
>-    *bitrate_bps = remote_rate_.LatestEstimate();
>+    *bitrate_bps = remote_rate_.LatestEstimate().bps<uint32_t>();
>   }
>   return true;
> }
>@@ -408,6 +422,6 @@ void RemoteBitrateEstimatorAbsSendTime::SetMinBitrate(int min_bitrate_bps) {
>   // Called from both the configuration thread and the network thread. Shouldn't
>   // be called from the network thread in the future.
>   rtc::CritScope lock(&crit_);
>-  remote_rate_.SetMinBitrate(min_bitrate_bps);
>+  remote_rate_.SetMinBitrate(DataRate::bps(min_bitrate_bps));
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
>index 3d791f78487d9b4da877b0dcea01d5e91f5a6964..a267051944b10f19b8a10dedd522eb7b723e9da8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
>@@ -10,17 +10,33 @@
> 
> #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
> 
>+#include <assert.h>
>+#include <cstdint>
> #include <utility>
> 
>+#include "absl/types/optional.h"
>+#include "common_types.h"  // NOLINT(build/include)
> #include "modules/remote_bitrate_estimator/aimd_rate_control.h"
>+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/remote_bitrate_estimator/inter_arrival.h"
> #include "modules/remote_bitrate_estimator/overuse_detector.h"
> #include "modules/remote_bitrate_estimator/overuse_estimator.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/metrics.h"
> 
> namespace webrtc {
>+namespace {
>+absl::optional<DataRate> OptionalRateFromOptionalBps(
>+    absl::optional<int> bitrate_bps) {
>+  if (bitrate_bps) {
>+    return DataRate::bps(*bitrate_bps);
>+  } else {
>+    return absl::nullopt;
>+  }
>+}
>+}  // namespace
> 
> enum { kTimestampGroupLengthMs = 5 };
> static const double kTimestampToMs = 1.0 / 90.0;
>@@ -127,7 +143,8 @@ void RemoteBitrateEstimatorSingleStream::IncomingPacket(
>         incoming_bitrate_.Rate(now_ms);
>     if (incoming_bitrate_bps &&
>         (prior_state != BandwidthUsage::kBwOverusing ||
>-         GetRemoteRate()->TimeToReduceFurther(now_ms, *incoming_bitrate_bps))) {
>+         GetRemoteRate()->TimeToReduceFurther(
>+             Timestamp::ms(now_ms), DataRate::bps(*incoming_bitrate_bps)))) {
>       // The first overuse should immediately trigger a new estimate.
>       // We also have to update the estimate immediately if we are overusing
>       // and the target bitrate is too high compared to what we are receiving.
>@@ -181,10 +198,12 @@ void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t now_ms) {
>   }
>   AimdRateControl* remote_rate = GetRemoteRate();
> 
>-  const RateControlInput input(bw_state, incoming_bitrate_.Rate(now_ms));
>-  uint32_t target_bitrate = remote_rate->Update(&input, now_ms);
>+  const RateControlInput input(
>+      bw_state, OptionalRateFromOptionalBps(incoming_bitrate_.Rate(now_ms)));
>+  uint32_t target_bitrate =
>+      remote_rate->Update(&input, Timestamp::ms(now_ms)).bps<uint32_t>();
>   if (remote_rate->ValidEstimate()) {
>-    process_interval_ms_ = remote_rate->GetFeedbackInterval();
>+    process_interval_ms_ = remote_rate->GetFeedbackInterval().ms();
>     RTC_DCHECK_GT(process_interval_ms_, 0);
>     std::vector<uint32_t> ssrcs;
>     GetSsrcs(&ssrcs);
>@@ -196,7 +215,7 @@ void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t now_ms) {
> void RemoteBitrateEstimatorSingleStream::OnRttUpdate(int64_t avg_rtt_ms,
>                                                      int64_t max_rtt_ms) {
>   rtc::CritScope cs(&crit_sect_);
>-  GetRemoteRate()->SetRtt(avg_rtt_ms);
>+  GetRemoteRate()->SetRtt(TimeDelta::ms(avg_rtt_ms));
> }
> 
> void RemoteBitrateEstimatorSingleStream::RemoveStream(unsigned int ssrc) {
>@@ -220,7 +239,7 @@ bool RemoteBitrateEstimatorSingleStream::LatestEstimate(
>   if (ssrcs->empty())
>     *bitrate_bps = 0;
>   else
>-    *bitrate_bps = remote_rate_->LatestEstimate();
>+    *bitrate_bps = remote_rate_->LatestEstimate().bps<uint32_t>();
>   return true;
> }
> 
>@@ -243,7 +262,7 @@ AimdRateControl* RemoteBitrateEstimatorSingleStream::GetRemoteRate() {
> 
> void RemoteBitrateEstimatorSingleStream::SetMinBitrate(int min_bitrate_bps) {
>   rtc::CritScope cs(&crit_sect_);
>-  remote_rate_->SetMinBitrate(min_bitrate_bps);
>+  remote_rate_->SetMinBitrate(DataRate::bps(min_bitrate_bps));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
>index 18b435affdd094931a331f2f4fcf1f32dec6f61e..638f0d68027abc5508c1c8a01a69d6dfc00ff04c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
>@@ -11,6 +11,8 @@
> #ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_SINGLE_STREAM_H_
> #define MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_SINGLE_STREAM_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <map>
> #include <memory>
> #include <vector>
>@@ -20,9 +22,13 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/rate_statistics.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
> 
>+class Clock;
>+struct RTPHeader;
>+
> class RemoteBitrateEstimatorSingleStream : public RemoteBitrateEstimator {
>  public:
>   RemoteBitrateEstimatorSingleStream(RemoteBitrateObserver* observer,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
>index 120db13c7f2854d76313424ef9abb4c7b570df57..2a4ef0647373b665186db5f7831c354d0dca7669 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
>@@ -55,7 +55,7 @@ TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropThreeStreamsWrap) {
> }
> 
> TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropThirteenStreamsWrap) {
>-  CapacityDropTestHelper(13, true, 733, 0);
>+  CapacityDropTestHelper(13, true, 567, 0);
> }
> 
> TEST_F(RemoteBitrateEstimatorSingleTest, CapacityDropNineteenStreamsWrap) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
>index a4444375805b4be6e52688d4dbbe7dc0eaa246c4..289ff70ee2157ece315e0a2311c094b57e7040b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
>@@ -32,6 +32,7 @@ SendSideBweSender::SendSideBweSender(int kbps,
>                                                      &event_log_)),
>       acknowledged_bitrate_estimator_(
>           absl::make_unique<AcknowledgedBitrateEstimator>()),
>+      probe_bitrate_estimator_(new ProbeBitrateEstimator(nullptr)),
>       bwe_(new DelayBasedBwe(nullptr)),
>       feedback_observer_(bitrate_controller_.get()),
>       clock_(clock),
>@@ -44,7 +45,7 @@ SendSideBweSender::SendSideBweSender(int kbps,
>   bitrate_controller_->SetStartBitrate(1000 * kbps);
>   bitrate_controller_->SetMinMaxBitrate(1000 * kMinBitrateKbps,
>                                         1000 * kMaxBitrateKbps);
>-  bwe_->SetMinBitrate(1000 * kMinBitrateKbps);
>+  bwe_->SetMinBitrate(DataRate::kbps(kMinBitrateKbps));
> }
> 
> SendSideBweSender::~SendSideBweSender() {}
>@@ -72,16 +73,22 @@ void SendSideBweSender::GiveFeedback(const FeedbackPacket& feedback) {
> 
>   int64_t rtt_ms =
>       clock_->TimeInMilliseconds() - feedback.latest_send_time_ms();
>-  bwe_->OnRttUpdate(rtt_ms);
>+  bwe_->OnRttUpdate(TimeDelta::ms(rtt_ms));
>   BWE_TEST_LOGGING_PLOT(1, "RTT", clock_->TimeInMilliseconds(), rtt_ms);
> 
>   std::sort(packet_feedback_vector.begin(), packet_feedback_vector.end(),
>             PacketFeedbackComparator());
>   acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
>       packet_feedback_vector);
>+  for (PacketFeedback& packet : packet_feedback_vector) {
>+    if (packet.send_time_ms != PacketFeedback::kNoSendTime &&
>+        packet.pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe)
>+      probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
>+  }
>   DelayBasedBwe::Result result = bwe_->IncomingPacketFeedbackVector(
>-      packet_feedback_vector, acknowledged_bitrate_estimator_->bitrate_bps(),
>-      clock_->TimeInMilliseconds());
>+      packet_feedback_vector, acknowledged_bitrate_estimator_->bitrate(),
>+      probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
>+      Timestamp::ms(clock_->TimeInMilliseconds()));
>   if (result.updated)
>     bitrate_controller_->OnDelayBasedBweResult(result);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
>index 6e939c1f035706addcd35f20637181eb7794e50b..5b45e66fbca754b893847d4851c648330e47df5c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
>@@ -39,6 +39,7 @@ class SendSideBweSender : public BweSender, public RemoteBitrateObserver {
>  protected:
>   std::unique_ptr<BitrateController> bitrate_controller_;
>   std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator_;
>+  std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_;
>   std::unique_ptr<DelayBasedBwe> bwe_;
>   RtcpBandwidthObserver* feedback_observer_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
>index 0230db17dc01c2ddc159287e2320519a2a7b950c..d7cdb54d92879a42c87aa7bd6f5f2072ce8fad48 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
>@@ -24,28 +24,30 @@
> 
> namespace flags {
> 
>-DEFINE_string(extension_type,
>-              "abs",
>-              "Extension type, either abs for absolute send time or tsoffset "
>-              "for timestamp offset.");
>+WEBRTC_DEFINE_string(
>+    extension_type,
>+    "abs",
>+    "Extension type, either abs for absolute send time or tsoffset "
>+    "for timestamp offset.");
> std::string ExtensionType() {
>   return static_cast<std::string>(FLAG_extension_type);
> }
> 
>-DEFINE_int(extension_id, 3, "Extension id.");
>+WEBRTC_DEFINE_int(extension_id, 3, "Extension id.");
> int ExtensionId() {
>   return static_cast<int>(FLAG_extension_id);
> }
> 
>-DEFINE_string(input_file, "", "Input file.");
>+WEBRTC_DEFINE_string(input_file, "", "Input file.");
> std::string InputFile() {
>   return static_cast<std::string>(FLAG_input_file);
> }
> 
>-DEFINE_string(ssrc_filter,
>-              "",
>-              "Comma-separated list of SSRCs in hexadecimal which are to be "
>-              "used as input to the BWE (only applicable to pcap files).");
>+WEBRTC_DEFINE_string(
>+    ssrc_filter,
>+    "",
>+    "Comma-separated list of SSRCs in hexadecimal which are to be "
>+    "used as input to the BWE (only applicable to pcap files).");
> std::set<uint32_t> SsrcFilter() {
>   std::string ssrc_filter_string = static_cast<std::string>(FLAG_ssrc_filter);
>   if (ssrc_filter_string.empty())
>@@ -64,7 +66,7 @@ std::set<uint32_t> SsrcFilter() {
>   return ssrcs;
> }
> 
>-DEFINE_bool(help, false, "Print this message.");
>+WEBRTC_DEFINE_bool(help, false, "Print this message.");
> }  // namespace flags
> 
> bool ParseArgsAndSetupEstimator(int argc,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/BUILD.gn
>index 4b37290765273efc693c1a14aa7a1056923371ff..edb981bc67edc172d1ad858b59435a40a0b65835 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/BUILD.gn
>@@ -11,6 +11,7 @@ import("../../webrtc.gni")
> rtc_source_set("rtp_rtcp_format") {
>   visibility = [ "*" ]
>   public = [
>+    "include/rtcp_statistics.h",
>     "include/rtp_cvo.h",
>     "include/rtp_header_extension_map.h",
>     "include/rtp_rtcp_defines.h",
>@@ -85,6 +86,7 @@ rtc_source_set("rtp_rtcp_format") {
> 
>   deps = [
>     "..:module_api",
>+    "..:module_api_public",
>     "../..:webrtc_common",
>     "../../api:array_view",
>     "../../api:libjingle_peerconnection_api",
>@@ -96,6 +98,7 @@ rtc_source_set("rtp_rtcp_format") {
>     "../../rtc_base:deprecation",
>     "../../rtc_base:rtc_base_approved",
>     "../../system_wrappers",
>+    "../video_coding:codec_globals_headers",
>     "//third_party/abseil-cpp/absl/types:optional",
>     "//third_party/abseil-cpp/absl/types:variant",
>   ]
>@@ -186,7 +189,10 @@ rtc_static_library("rtp_rtcp") {
> 
>   deps = [
>     ":rtp_rtcp_format",
>+    ":rtp_video_header",
>     "..:module_api",
>+    "..:module_api_public",
>+    "..:module_fec_api",
>     "../..:webrtc_common",
>     "../../api:array_view",
>     "../../api:libjingle_peerconnection_api",
>@@ -217,9 +223,12 @@ rtc_static_library("rtp_rtcp") {
>     "../../system_wrappers:metrics",
>     "../audio_coding:audio_format_conversion",
>     "../remote_bitrate_estimator",
>+    "../video_coding:codec_globals_headers",
>     "//third_party/abseil-cpp/absl/container:inlined_vector",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>+    "//third_party/abseil-cpp/absl/types:variant",
>   ]
> }
> 
>@@ -264,6 +273,7 @@ rtc_source_set("rtp_video_header") {
>     "../../api/video:video_frame",
>     "../../modules/video_coding:codec_globals_headers",
>     "//third_party/abseil-cpp/absl/container:inlined_vector",
>+    "//third_party/abseil-cpp/absl/types:optional",
>     "//third_party/abseil-cpp/absl/types:variant",
>   ]
> }
>@@ -322,6 +332,7 @@ if (rtc_include_tests) {
>       ":rtp_rtcp",
>       "../../test:fileutils",
>       "../../test:test_main",
>+      "../../test:test_support",
>       "//testing/gtest",
>     ]
>   }  # test_packet_masks_metrics
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h
>index f006111e117b62fb3de994ae0c2089a2ae869258..f0ed576c87560793c39b3793338cb04b88ed15e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h
>@@ -11,16 +11,20 @@
> #ifndef MODULES_RTP_RTCP_INCLUDE_FLEXFEC_RECEIVER_H_
> #define MODULES_RTP_RTCP_INCLUDE_FLEXFEC_RECEIVER_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/include/ulpfec_receiver.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "rtc_base/sequenced_task_checker.h"
>-#include "system_wrappers/include/clock.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
> 
>+class Clock;
>+
> class FlexfecReceiver {
>  public:
>   FlexfecReceiver(uint32_t ssrc,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_sender.h
>index 12277e128f5554919ff29763d2f3adf83677d850..acee11764c0aaf15ab67234bb0288353fec4f574 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_sender.h
>@@ -18,17 +18,15 @@
> #include "api/array_view.h"
> #include "api/rtpparameters.h"
> #include "modules/include/module_common_types.h"
>-#include "modules/rtp_rtcp/include/flexfec_sender.h"
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
>-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "modules/rtp_rtcp/source/ulpfec_generator.h"
> #include "rtc_base/random.h"
>-#include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
> 
>+class Clock;
> class RtpPacketToSend;
> 
> // Note that this class is not thread safe, and thus requires external
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/receive_statistics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/receive_statistics.h
>index 24d6f81237a5b6eefbd264806650f0143cb994b4..c299ea69d66ec03bd481dc79e86c539734541db0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/receive_statistics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/receive_statistics.h
>@@ -12,11 +12,13 @@
> #define MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
> 
> #include <map>
>+#include <memory>
> #include <vector>
> 
> #include "call/rtp_packet_sink_interface.h"
> #include "modules/include/module.h"
> #include "modules/include/module_common_types.h"
>+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
> #include "rtc_base/deprecation.h"
>@@ -54,14 +56,14 @@ class ReceiveStatistics : public ReceiveStatisticsProvider,
>  public:
>   ~ReceiveStatistics() override = default;
> 
>-  static ReceiveStatistics* Create(Clock* clock);
>+  static ReceiveStatistics* Create(Clock* clock) {
>+    return Create(clock, nullptr, nullptr).release();
>+  }
> 
>-  // Updates the receive statistics with this packet.
>-  // TODO(bugs.webrtc.org/8016): Deprecated. Delete as soon as
>-  // downstream code is updated to use OnRtpPacket.
>-  RTC_DEPRECATED
>-  virtual void IncomingPacket(const RTPHeader& rtp_header,
>-                              size_t packet_length) = 0;
>+  static std::unique_ptr<ReceiveStatistics> Create(
>+      Clock* clock,
>+      RtcpStatisticsCallback* rtcp_callback,
>+      StreamDataCountersCallback* rtp_callback);
> 
>   // Increment counter for number of FEC packets received.
>   virtual void FecPacketReceived(const RtpPacketReceived& packet) = 0;
>@@ -74,14 +76,6 @@ class ReceiveStatistics : public ReceiveStatisticsProvider,
>   // Detect retransmissions, enabling updates of the retransmitted counters. The
>   // default is false.
>   virtual void EnableRetransmitDetection(uint32_t ssrc, bool enable) = 0;
>-
>-  // Called on new RTCP stats creation.
>-  virtual void RegisterRtcpStatisticsCallback(
>-      RtcpStatisticsCallback* callback) = 0;
>-
>-  // Called on new RTP stats creation.
>-  virtual void RegisterRtpStatisticsCallback(
>-      StreamDataCountersCallback* callback) = 0;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
>index 5195e8ab7e316c9edef7410cdaa65b1df1b04b1a..e6d269c4ddabb1b9b53ba88e35ef767542d53832 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
> #define MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
> 
>-#include <memory>
>+#include <stdint.h>
> 
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/numerics/moving_median_filter.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtcp_statistics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtcp_statistics.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..e1d576de2d86fa8e85226bbfb59f5a6b3874bf62
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtcp_statistics.h
>@@ -0,0 +1,36 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_
>+#define MODULES_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_
>+
>+#include <stdint.h>
>+
>+namespace webrtc {
>+
>+// Statistics for an RTCP channel
>+struct RtcpStatistics {
>+  uint8_t fraction_lost = 0;
>+  int32_t packets_lost = 0;  // Defined as a 24 bit signed integer in RTCP
>+  uint32_t extended_highest_sequence_number = 0;
>+  uint32_t jitter = 0;
>+};
>+
>+class RtcpStatisticsCallback {
>+ public:
>+  virtual ~RtcpStatisticsCallback() {}
>+
>+  virtual void StatisticsUpdated(const RtcpStatistics& statistics,
>+                                 uint32_t ssrc) = 0;
>+  virtual void CNameChanged(const char* cname, uint32_t ssrc) = 0;
>+};
>+
>+}  // namespace webrtc
>+#endif  // MODULES_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h
>index b8f27a13124532e5b42eec626063e9b31c98c82d..391c5beaa5a62bcb079d191231a625ba2a832ab8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_EXTENSION_MAP_H_
> #define MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_EXTENSION_MAP_H_
> 
>+#include <stdint.h>
> #include <string>
> 
> #include "api/array_view.h"
>@@ -26,6 +27,7 @@ class RtpHeaderExtensionMap {
>   static constexpr int kInvalidId = 0;
> 
>   RtpHeaderExtensionMap();
>+  explicit RtpHeaderExtensionMap(bool extmap_allow_mixed);
>   explicit RtpHeaderExtensionMap(rtc::ArrayView<const RtpExtension> extensions);
> 
>   template <typename Extension>
>@@ -53,18 +55,19 @@ class RtpHeaderExtensionMap {
>   }
>   int32_t Deregister(RTPExtensionType type);
> 
>-  bool IsMixedOneTwoByteHeaderSupported() const {
>-    return mixed_one_two_byte_header_supported_;
>-  }
>-  void SetMixedOneTwoByteHeaderSupported(bool supported) {
>-    mixed_one_two_byte_header_supported_ = supported;
>+  // Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285.
>+  // Set to true if it's allowed to mix one- and two-byte RTP header extensions
>+  // in the same stream.
>+  bool ExtmapAllowMixed() const { return extmap_allow_mixed_; }
>+  void SetExtmapAllowMixed(bool extmap_allow_mixed) {
>+    extmap_allow_mixed_ = extmap_allow_mixed;
>   }
> 
>  private:
>   bool Register(int id, RTPExtensionType type, const char* uri);
> 
>   uint8_t ids_[kRtpExtensionNumberOfExtensions];
>-  bool mixed_one_two_byte_header_supported_;
>+  bool extmap_allow_mixed_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
>index 2d84fc1e6903920f997df44b6df08863dbf057d8..85eab90a73acd52147a17cbe594445c326352376 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
>@@ -10,6 +10,7 @@
> #ifndef MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
> #define MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
> 
>+#include "api/rtpparameters.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> 
> namespace webrtc {
>@@ -36,8 +37,14 @@ class RtpHeaderParser {
>   virtual bool RegisterRtpHeaderExtension(RTPExtensionType type,
>                                           uint8_t id) = 0;
> 
>+  // Registers an RTP header extension.
>+  virtual bool RegisterRtpHeaderExtension(RtpExtension extension) = 0;
>+
>   // De-registers an RTP header extension.
>   virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0;
>+
>+  // De-registers an RTP header extension.
>+  virtual bool DeregisterRtpHeaderExtension(RtpExtension extension) = 0;
> };
> }  // namespace webrtc
> #endif  // MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
>index 9b29f7622e2ff134230ca056b1d8a65553d6e2c5..d136a5e6e9a3ba7623ccb3c4c732131212714d69 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
>@@ -18,9 +18,9 @@
> 
> #include "absl/types/optional.h"
> #include "api/video/video_bitrate_allocation.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/include/module.h"
> #include "modules/rtp_rtcp/include/flexfec_sender.h"
>+#include "modules/rtp_rtcp/include/receive_statistics.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/deprecation.h"
>@@ -28,6 +28,7 @@
> namespace webrtc {
> 
> // Forward declarations.
>+class FrameEncryptorInterface;
> class OverheadObserver;
> class RateLimiter;
> class ReceiveStatisticsProvider;
>@@ -92,11 +93,20 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
>     RateLimiter* retransmission_rate_limiter = nullptr;
>     OverheadObserver* overhead_observer = nullptr;
>     RtpKeepAliveConfig keepalive_config;
>-    RtcpIntervalConfig rtcp_interval_config;
>+
>+    int rtcp_report_interval_ms = 0;
> 
>     // Update network2 instead of pacer_exit field of video timing extension.
>     bool populate_network2_timestamp = false;
> 
>+    // E2EE Custom Video Frame Encryption
>+    FrameEncryptorInterface* frame_encryptor = nullptr;
>+    // Require all outgoing frames to be encrypted with a FrameEncryptor.
>+    bool require_frame_encryption = false;
>+
>+    // Corresponds to extmap-allow-mixed in SDP negotiation.
>+    bool extmap_allow_mixed = false;
>+
>    private:
>     RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
>   };
>@@ -136,6 +146,8 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
>   // Returns -1 on failure else 0.
>   virtual int32_t DeRegisterSendPayload(int8_t payload_type) = 0;
> 
>+  virtual void SetExtmapAllowMixed(bool extmap_allow_mixed) = 0;
>+
>   // (De)registers RTP header extension type and id.
>   // Returns -1 on failure else 0.
>   virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
>@@ -213,6 +225,10 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
>   // Returns current media sending status.
>   virtual bool SendingMedia() const = 0;
> 
>+  // Indicate that the packets sent by this module should be counted towards the
>+  // bitrate estimate since the stream participates in the bitrate allocation.
>+  virtual void SetAsPartOfAllocation(bool part_of_allocation) = 0;
>+
>   // Returns current bitrate in Kbit/s.
>   virtual void BitrateSent(uint32_t* total_rate,
>                            uint32_t* video_rate,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
>index f86b238b7daebe632f1fe4dad23a4483444a8c9c..d743f52f7e6df3ea268321200c33758516f29eeb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
>@@ -9,6 +9,14 @@
>  */
> 
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/source/rtp_packet.h"
>+
>+#include <ctype.h>
>+#include <string.h>
>+#include <algorithm>
>+#include <type_traits>
>+
>+#include "api/array_view.h"
> 
> namespace webrtc {
> 
>@@ -54,4 +62,80 @@ PayloadUnion::~PayloadUnion() = default;
> PayloadUnion& PayloadUnion::operator=(const PayloadUnion&) = default;
> PayloadUnion& PayloadUnion::operator=(PayloadUnion&&) = default;
> 
>+PacketFeedback::PacketFeedback(int64_t arrival_time_ms,
>+                               uint16_t sequence_number)
>+    : PacketFeedback(-1,
>+                     arrival_time_ms,
>+                     kNoSendTime,
>+                     sequence_number,
>+                     0,
>+                     0,
>+                     0,
>+                     PacedPacketInfo()) {}
>+
>+PacketFeedback::PacketFeedback(int64_t arrival_time_ms,
>+                               int64_t send_time_ms,
>+                               uint16_t sequence_number,
>+                               size_t payload_size,
>+                               const PacedPacketInfo& pacing_info)
>+    : PacketFeedback(-1,
>+                     arrival_time_ms,
>+                     send_time_ms,
>+                     sequence_number,
>+                     payload_size,
>+                     0,
>+                     0,
>+                     pacing_info) {}
>+
>+PacketFeedback::PacketFeedback(int64_t creation_time_ms,
>+                               uint16_t sequence_number,
>+                               size_t payload_size,
>+                               uint16_t local_net_id,
>+                               uint16_t remote_net_id,
>+                               const PacedPacketInfo& pacing_info)
>+    : PacketFeedback(creation_time_ms,
>+                     kNotReceived,
>+                     kNoSendTime,
>+                     sequence_number,
>+                     payload_size,
>+                     local_net_id,
>+                     remote_net_id,
>+                     pacing_info) {}
>+
>+PacketFeedback::PacketFeedback(int64_t creation_time_ms,
>+                               int64_t arrival_time_ms,
>+                               int64_t send_time_ms,
>+                               uint16_t sequence_number,
>+                               size_t payload_size,
>+                               uint16_t local_net_id,
>+                               uint16_t remote_net_id,
>+                               const PacedPacketInfo& pacing_info)
>+    : creation_time_ms(creation_time_ms),
>+      arrival_time_ms(arrival_time_ms),
>+      send_time_ms(send_time_ms),
>+      sequence_number(sequence_number),
>+      payload_size(payload_size),
>+      unacknowledged_data(0),
>+      local_net_id(local_net_id),
>+      remote_net_id(remote_net_id),
>+      pacing_info(pacing_info) {}
>+
>+PacketFeedback::PacketFeedback(const PacketFeedback&) = default;
>+PacketFeedback& PacketFeedback::operator=(const PacketFeedback&) = default;
>+PacketFeedback::~PacketFeedback() = default;
>+
>+bool PacketFeedback::operator==(const PacketFeedback& rhs) const {
>+  return arrival_time_ms == rhs.arrival_time_ms &&
>+         send_time_ms == rhs.send_time_ms &&
>+         sequence_number == rhs.sequence_number &&
>+         payload_size == rhs.payload_size && pacing_info == rhs.pacing_info;
>+}
>+
>+void RtpPacketCounter::AddPacket(const RtpPacket& packet) {
>+  ++packets;
>+  header_bytes += packet.headers_size();
>+  padding_bytes += packet.padding_size();
>+  payload_bytes += packet.payload_size();
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
>index 6503b4edcc3a3b88d7fe0ce657b03ef7b780ad6c..ab4fcaecc79a481071bc166dc79a52feb0a62556 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
>@@ -27,28 +27,20 @@
> #define IP_PACKET_SIZE 1500  // we assume ethernet
> 
> namespace webrtc {
>+class RtpPacket;
> namespace rtcp {
> class TransportFeedback;
> }
> 
> const int kVideoPayloadTypeFrequency = 90000;
>-// TODO(solenberg): RTP time stamp rate for RTCP is fixed at 8k, this is legacy
>-// and should be fixed.
>-// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6458
>+
>+// TODO(bugs.webrtc.org/6458): Remove this when all the depending projects are
>+// updated to correctly set rtp rate for RtcpSender.
> const int kBogusRtpRateForAudioRtcp = 8000;
> 
> // Minimum RTP header size in bytes.
> const uint8_t kRtpHeaderSize = 12;
> 
>-struct RtcpIntervalConfig final {
>-  RtcpIntervalConfig() = default;
>-  RtcpIntervalConfig(int64_t video_interval_ms, int64_t audio_interval_ms)
>-      : video_interval_ms(video_interval_ms),
>-        audio_interval_ms(audio_interval_ms) {}
>-  int64_t video_interval_ms = 1000;
>-  int64_t audio_interval_ms = 5000;
>-};
>-
> struct AudioPayload {
>   SdpAudioFormat format;
>   uint32_t rate;
>@@ -112,6 +104,7 @@ enum RTPExtensionType : int {
>   kRtpExtensionRepairedRtpStreamId,
>   kRtpExtensionMid,
>   kRtpExtensionGenericFrameDescriptor,
>+  kRtpExtensionColorSpace,
>   kRtpExtensionNumberOfExtensions  // Must be the last entity in the enum.
> };
> 
>@@ -222,15 +215,6 @@ struct RtpState {
>   bool media_has_been_sent;
> };
> 
>-class RtpData {
>- public:
>-  virtual ~RtpData() {}
>-
>-  virtual int32_t OnReceivedPayloadData(const uint8_t* payload_data,
>-                                        size_t payload_size,
>-                                        const WebRtcRTPHeader* rtp_header) = 0;
>-};
>-
> // Callback interface for packets recovered by FlexFEC or ULPFEC. In
> // the FlexFEC case, the implementation should be able to demultiplex
> // the recovered RTP packets based on SSRC.
>@@ -263,44 +247,20 @@ class RtcpBandwidthObserver {
> };
> 
> struct PacketFeedback {
>-  PacketFeedback(int64_t arrival_time_ms, uint16_t sequence_number)
>-      : PacketFeedback(-1,
>-                       arrival_time_ms,
>-                       kNoSendTime,
>-                       sequence_number,
>-                       0,
>-                       0,
>-                       0,
>-                       PacedPacketInfo()) {}
>+  PacketFeedback(int64_t arrival_time_ms, uint16_t sequence_number);
> 
>   PacketFeedback(int64_t arrival_time_ms,
>                  int64_t send_time_ms,
>                  uint16_t sequence_number,
>                  size_t payload_size,
>-                 const PacedPacketInfo& pacing_info)
>-      : PacketFeedback(-1,
>-                       arrival_time_ms,
>-                       send_time_ms,
>-                       sequence_number,
>-                       payload_size,
>-                       0,
>-                       0,
>-                       pacing_info) {}
>+                 const PacedPacketInfo& pacing_info);
> 
>   PacketFeedback(int64_t creation_time_ms,
>                  uint16_t sequence_number,
>                  size_t payload_size,
>                  uint16_t local_net_id,
>                  uint16_t remote_net_id,
>-                 const PacedPacketInfo& pacing_info)
>-      : PacketFeedback(creation_time_ms,
>-                       kNotReceived,
>-                       kNoSendTime,
>-                       sequence_number,
>-                       payload_size,
>-                       local_net_id,
>-                       remote_net_id,
>-                       pacing_info) {}
>+                 const PacedPacketInfo& pacing_info);
> 
>   PacketFeedback(int64_t creation_time_ms,
>                  int64_t arrival_time_ms,
>@@ -309,15 +269,10 @@ struct PacketFeedback {
>                  size_t payload_size,
>                  uint16_t local_net_id,
>                  uint16_t remote_net_id,
>-                 const PacedPacketInfo& pacing_info)
>-      : creation_time_ms(creation_time_ms),
>-        arrival_time_ms(arrival_time_ms),
>-        send_time_ms(send_time_ms),
>-        sequence_number(sequence_number),
>-        payload_size(payload_size),
>-        local_net_id(local_net_id),
>-        remote_net_id(remote_net_id),
>-        pacing_info(pacing_info) {}
>+                 const PacedPacketInfo& pacing_info);
>+  PacketFeedback(const PacketFeedback&);
>+  PacketFeedback& operator=(const PacketFeedback&);
>+  ~PacketFeedback();
> 
>   static constexpr int kNotAProbe = -1;
>   static constexpr int64_t kNotReceived = -1;
>@@ -328,12 +283,7 @@ struct PacketFeedback {
>   //       for book-keeping, and is of no interest outside that class.
>   // TODO(philipel): Remove |creation_time_ms| from PacketFeedback when cleaning
>   //                 up SendTimeHistory.
>-  bool operator==(const PacketFeedback& rhs) const {
>-    return arrival_time_ms == rhs.arrival_time_ms &&
>-           send_time_ms == rhs.send_time_ms &&
>-           sequence_number == rhs.sequence_number &&
>-           payload_size == rhs.payload_size && pacing_info == rhs.pacing_info;
>-  }
>+  bool operator==(const PacketFeedback& rhs) const;
> 
>   // Time corresponding to when this object was created.
>   int64_t creation_time_ms;
>@@ -352,6 +302,8 @@ struct PacketFeedback {
>   int64_t long_sequence_number;
>   // Size of the packet excluding RTP headers.
>   size_t payload_size;
>+  // Size of preceeding packets that are not part of feedback.
>+  size_t unacknowledged_data;
>   // The network route ids that this packet is associated with.
>   uint16_t local_net_id;
>   uint16_t remote_net_id;
>@@ -489,13 +441,8 @@ struct RtpPacketCounter {
>     packets -= other.packets;
>   }
> 
>-  void AddPacket(size_t packet_length, const RTPHeader& header) {
>-    ++packets;
>-    header_bytes += header.headerLength;
>-    padding_bytes += header.paddingLength;
>-    payload_bytes +=
>-        packet_length - (header.headerLength + header.paddingLength);
>-  }
>+  // Not inlined, since use of RtpPacket would result in circular includes.
>+  void AddPacket(const RtpPacket& packet);
> 
>   size_t TotalBytes() const {
>     return header_bytes + payload_bytes + padding_bytes;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.cc
>index d24c1b094c51f998edab65d6def9e37937a8bfe3..061f82765cc75baa3ca04cdd43c64f1992dd5eb4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.cc
>@@ -12,9 +12,6 @@
> 
> namespace webrtc {
> 
>-MockRtpData::MockRtpData() = default;
>-MockRtpData::~MockRtpData() = default;
>-
> MockRtpRtcp::MockRtpRtcp() = default;
> MockRtpRtcp::~MockRtpRtcp() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
>index 6d5bde4788f4a6ae48ff6a8ddc084f766344fade..3b9b943c4539504130d26d55bc4ecf34a9ccb5ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
>@@ -27,17 +27,6 @@
> 
> namespace webrtc {
> 
>-class MockRtpData : public RtpData {
>- public:
>-  MockRtpData();
>-  ~MockRtpData();
>-
>-  MOCK_METHOD3(OnReceivedPayloadData,
>-               int32_t(const uint8_t* payload_data,
>-                       size_t payload_size,
>-                       const WebRtcRTPHeader* rtp_header));
>-};
>-
> class MockRtpRtcp : public RtpRtcp {
>  public:
>   MockRtpRtcp();
>@@ -52,6 +41,7 @@ class MockRtpRtcp : public RtpRtcp {
>   MOCK_METHOD2(RegisterVideoSendPayload,
>                void(int payload_type, const char* payload_name));
>   MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type));
>+  MOCK_METHOD1(SetExtmapAllowMixed, void(bool extmap_allow_mixed));
>   MOCK_METHOD2(RegisterSendRtpHeaderExtension,
>                int32_t(RTPExtensionType type, uint8_t id));
>   MOCK_METHOD2(RegisterRtpHeaderExtension,
>@@ -83,6 +73,7 @@ class MockRtpRtcp : public RtpRtcp {
>   MOCK_CONST_METHOD0(Sending, bool());
>   MOCK_METHOD1(SetSendingMediaStatus, void(bool sending));
>   MOCK_CONST_METHOD0(SendingMedia, bool());
>+  MOCK_METHOD1(SetAsPartOfAllocation, void(bool));
>   MOCK_CONST_METHOD4(BitrateSent,
>                      void(uint32_t* total_rate,
>                           uint32_t* video_rate,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.cc
>index 853706c4f5dc140909ee22c9e26c02d47fac1b94..64dc443e295f5c46d8a70a0d71df3ffcf5c83736 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.cc
>@@ -25,9 +25,11 @@ ContributingSources::ContributingSources() = default;
> ContributingSources::~ContributingSources() = default;
> 
> void ContributingSources::Update(int64_t now_ms,
>-                                 rtc::ArrayView<const uint32_t> csrcs) {
>+                                 rtc::ArrayView<const uint32_t> csrcs,
>+                                 absl::optional<uint8_t> audio_level) {
>+  Entry entry = { now_ms, audio_level };
>   for (uint32_t csrc : csrcs) {
>-    last_seen_ms_[csrc] = now_ms;
>+    active_csrcs_[csrc] = entry;
>   }
>   if (!next_pruning_ms_) {
>     next_pruning_ms_ = now_ms + kPruningIntervalMs;
>@@ -43,9 +45,16 @@ void ContributingSources::Update(int64_t now_ms,
> // non-const.
> std::vector<RtpSource> ContributingSources::GetSources(int64_t now_ms) const {
>   std::vector<RtpSource> sources;
>-  for (auto& record : last_seen_ms_) {
>-    if (record.second >= now_ms - kHistoryMs) {
>-      sources.emplace_back(record.second, record.first, RtpSourceType::CSRC);
>+  for (auto& record : active_csrcs_) {
>+    if (record.second.last_seen_ms >= now_ms - kHistoryMs) {
>+      if (record.second.audio_level.has_value()) {
>+        sources.emplace_back(record.second.last_seen_ms, record.first,
>+                             RtpSourceType::CSRC,
>+                             *record.second.audio_level);
>+      } else {
>+        sources.emplace_back(record.second.last_seen_ms, record.first,
>+                             RtpSourceType::CSRC);
>+      }
>     }
>   }
> 
>@@ -54,15 +63,20 @@ std::vector<RtpSource> ContributingSources::GetSources(int64_t now_ms) const {
> 
> // Delete stale entries.
> void ContributingSources::DeleteOldEntries(int64_t now_ms) {
>-  for (auto it = last_seen_ms_.begin(); it != last_seen_ms_.end();) {
>-    if (it->second >= now_ms - kHistoryMs) {
>+  for (auto it = active_csrcs_.begin(); it != active_csrcs_.end();) {
>+    if (it->second.last_seen_ms >= now_ms - kHistoryMs) {
>       // Still relevant.
>       ++it;
>     } else {
>-      it = last_seen_ms_.erase(it);
>+      it = active_csrcs_.erase(it);
>     }
>   }
>   next_pruning_ms_ = now_ms + kPruningIntervalMs;
> }
> 
>+ContributingSources::Entry::Entry() = default;
>+ContributingSources::Entry::Entry(int64_t timestamp_ms,
>+                                  absl::optional<uint8_t> audio_level_arg)
>+    : last_seen_ms(timestamp_ms), audio_level(audio_level_arg) {}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.h
>index 1a4a572d8f54431ba6a523e40f5d33af01f8fb22..5e34539ce477e0b7bedaf80d099a744735114e63 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.h
>@@ -18,7 +18,8 @@
> 
> #include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/rtpreceiverinterface.h"
>+#include "api/rtpreceiverinterface.h"  // For RtpSource
>+#include "rtc_base/timeutils.h"        // For kNumMillisecsPerSec
> 
> namespace webrtc {
> 
>@@ -31,18 +32,25 @@ class ContributingSources {
>   ContributingSources();
>   ~ContributingSources();
> 
>-  // TODO(bugs.webrtc.org/3333): Needs to be extended with audio-level, to
>-  // support RFC6465.
>-  void Update(int64_t now_ms, rtc::ArrayView<const uint32_t> csrcs);
>+  void Update(int64_t now_ms, rtc::ArrayView<const uint32_t> csrcs,
>+              absl::optional<uint8_t> audio_level);
> 
>   // Returns contributing sources seen the last 10 s.
>   std::vector<RtpSource> GetSources(int64_t now_ms) const;
> 
>  private:
>+  struct Entry {
>+    Entry();
>+    Entry(int64_t timestamp_ms, absl::optional<uint8_t> audio_level);
>+
>+    int64_t last_seen_ms;
>+    absl::optional<uint8_t> audio_level;
>+  };
>+
>   void DeleteOldEntries(int64_t now_ms);
> 
>   // Indexed by csrc.
>-  std::map<uint32_t, int64_t> last_seen_ms_;
>+  std::map<uint32_t, Entry> active_csrcs_;
>   absl::optional<int64_t> next_pruning_ms_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources_unittest.cc
>index 8b22d26c94804cf4422c0db61d65873c18e4bcd2..5f1d8d3d112042c8942415679bd7dbb92e6fc1f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources_unittest.cc
>@@ -30,7 +30,7 @@ TEST(ContributingSourcesTest, RecordSources) {
>   ContributingSources csrcs;
>   constexpr uint32_t kCsrcs[] = {kCsrc1, kCsrc2};
>   constexpr int64_t kTime1 = 10;
>-  csrcs.Update(kTime1, kCsrcs);
>+  csrcs.Update(kTime1, kCsrcs, absl::nullopt);
>   EXPECT_THAT(
>       csrcs.GetSources(kTime1),
>       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>@@ -45,12 +45,12 @@ TEST(ContributingSourcesTest, UpdateSources) {
>   constexpr uint32_t kCsrcs2[] = {kCsrc2, kCsrc3};
>   constexpr int64_t kTime1 = 10;
>   constexpr int64_t kTime2 = kTime1 + 5 * rtc::kNumMillisecsPerSec;
>-  csrcs.Update(kTime1, kCsrcs1);
>+  csrcs.Update(kTime1, kCsrcs1, absl::nullopt);
>   EXPECT_THAT(
>       csrcs.GetSources(kTime1),
>       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>                            RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
>-  csrcs.Update(kTime2, kCsrcs2);
>+  csrcs.Update(kTime2, kCsrcs2, absl::nullopt);
>   EXPECT_THAT(
>       csrcs.GetSources(kTime2),
>       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>@@ -65,12 +65,12 @@ TEST(ContributingSourcesTest, ReturnRecentOnly) {
>   constexpr int64_t kTime1 = 10;
>   constexpr int64_t kTime2 = kTime1 + 5 * rtc::kNumMillisecsPerSec;
>   constexpr int64_t kTime3 = kTime1 + 12 * rtc::kNumMillisecsPerSec;
>-  csrcs.Update(kTime1, kCsrcs1);
>+  csrcs.Update(kTime1, kCsrcs1, absl::nullopt);
>   EXPECT_THAT(
>       csrcs.GetSources(kTime1),
>       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>                            RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
>-  csrcs.Update(kTime2, kCsrcs2);
>+  csrcs.Update(kTime2, kCsrcs2, absl::nullopt);
>   EXPECT_THAT(
>       csrcs.GetSources(kTime3),
>       UnorderedElementsAre(RtpSource(kTime2, kCsrc2, RtpSourceType::CSRC),
>@@ -84,18 +84,18 @@ TEST(ContributingSourcesTest, PurgeOldSources) {
>   constexpr int64_t kTime1 = 10;
>   constexpr int64_t kTime2 = kTime1 + 10 * rtc::kNumMillisecsPerSec;
>   constexpr int64_t kTime3 = kTime1 + 20 * rtc::kNumMillisecsPerSec;
>-  csrcs.Update(kTime1, kCsrcs1);
>+  csrcs.Update(kTime1, kCsrcs1, absl::nullopt);
>   EXPECT_THAT(
>       csrcs.GetSources(kTime2),
>       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>                            RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
>-  csrcs.Update(kTime2, kCsrcs2);
>+  csrcs.Update(kTime2, kCsrcs2, absl::nullopt);
>   EXPECT_THAT(
>       csrcs.GetSources(kTime2),
>       UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>                            RtpSource(kTime2, kCsrc2, RtpSourceType::CSRC),
>                            RtpSource(kTime2, kCsrc3, RtpSourceType::CSRC)));
>-  csrcs.Update(kTime3, kCsrcs2);
>+  csrcs.Update(kTime3, kCsrcs2, absl::nullopt);
>   EXPECT_THAT(
>       csrcs.GetSources(kTime3),
>       UnorderedElementsAre(RtpSource(kTime3, kCsrc2, RtpSourceType::CSRC),
>@@ -108,4 +108,22 @@ TEST(ContributingSourcesTest, PurgeOldSources) {
>                            RtpSource(kTime3, kCsrc3, RtpSourceType::CSRC)));
> }
> 
>+TEST(ContributingSourcesTest, AudioLevel) {
>+  ContributingSources csrcs;
>+  constexpr uint32_t kCsrcs[] = {kCsrc1, kCsrc2};
>+  constexpr int64_t kTime1 = 10;
>+  csrcs.Update(kTime1, kCsrcs, 47);
>+  EXPECT_THAT(
>+      csrcs.GetSources(kTime1),
>+      UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC, 47),
>+                           RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC, 47)));
>+
>+  constexpr uint32_t kCsrcsSubset[] = {kCsrc1};
>+  csrcs.Update(kTime1 + 1, kCsrcsSubset, absl::nullopt);
>+  EXPECT_THAT(
>+      csrcs.GetSources(kTime1 + 1),
>+      UnorderedElementsAre(RtpSource(kTime1 + 1, kCsrc1, RtpSourceType::CSRC),
>+                           RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC, 47)));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
>index 86ddb105f924e998f19dee296b0e19c0c5a184e1..10e674789a4ba0fcf6761b7fc33bb5a9d95a8f05 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
>@@ -10,6 +10,10 @@
> 
> #include "modules/rtp_rtcp/source/dtmf_queue.h"
> 
>+#include <stddef.h>
>+
>+#include "rtc_base/checks.h"
>+
> namespace {
> constexpr size_t kDtmfOutbandMax = 20;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/dtmf_queue.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
>index db70c9750800286fa7e381f229d2e5193e406961..e5955a1297c44383da19c55d413885ad3fd21ff3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/dtmf_queue.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
> #define MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
> 
>+#include <stdint.h>
> #include <list>
> 
> #include "rtc_base/criticalsection.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
>index d7666e1192a339cafb192516735309fac63b8add..b813340ea68d8555e4539687aec85f5fbf599553 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
>@@ -12,12 +12,11 @@
> 
> #include <string.h>
> 
>-#include <utility>
>-
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/scoped_ref_ptr.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h
>index 1d6ddda4b40985f4b612166a451129db3796c930..d305c4c288de14fdb840b7acca2ead83cf1c228b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h
>@@ -11,6 +11,9 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_FLEXFEC_HEADER_READER_WRITER_H_
> #define MODULES_RTP_RTCP_SOURCE_FLEXFEC_HEADER_READER_WRITER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc
>index c3ed4d53b2a43ff9e6dec909cfde4b165951d3a9..1a62bced68a4d2f22fb1a6e230f728c512b4f010 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc
>@@ -10,6 +10,9 @@
> 
> #include "modules/rtp_rtcp/include/flexfec_receiver.h"
> 
>+#include <string.h>
>+
>+#include "api/array_view.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/scoped_ref_ptr.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
>index 286f47c5fe808d26749f5abf0fafae542ea1e1bf..1204b2d978c44cff1dbb709f51bc2ad88006debf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
>@@ -10,11 +10,15 @@
> 
> #include "modules/rtp_rtcp/include/flexfec_sender.h"
> 
>+#include <string.h>
>+#include <list>
> #include <utility>
> 
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
>+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
>index b743110b2d6024ef0150aa793487fcd5ea1edcbb..a52fecad18e70f3c372c8ce9d088d168f60496aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
>@@ -11,11 +11,10 @@
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> 
> #include <string.h>
>-
> #include <algorithm>
>-#include <iterator>
> #include <utility>
> 
>+#include "modules/include/module_common_types_public.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/flexfec_header_reader_writer.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
>index 819f6bcec56a16ea7078b45fcbadd303f01fa364..adb7572ae670a964eef5c0fb1109afcdbd310d14 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
>@@ -11,16 +11,15 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
> #define MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
> 
>+#include <stddef.h>
> #include <stdint.h>
>-
> #include <list>
> #include <memory>
> #include <vector>
> 
>+#include "modules/include/module_fec_types.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
>index 7e5fd91e035644e01b6d193e24119ba38a680387..9b02026a77c505f38f8677c55e586bbcef8c8b9c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
> 
>+#include <string.h>
> #include <algorithm>
> 
> #include "modules/rtp_rtcp/source/fec_private_tables_bursty.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h
>index 2e8a202113af94867b1b1a5868d60be5f0aec30c..ed93f520e526fc788bc62eee3cd7e2e04c144bbc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h
>@@ -11,9 +11,11 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_
> #define MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_
> 
>-#include "modules/include/module_common_types.h"
>+#include <stddef.h>
>+#include <stdint.h>
> 
> #include "api/array_view.h"
>+#include "modules/include/module_fec_types.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc
>index 076348d2da3e0893e1b05490cf60da8279d784d6..36f0a63d596657be643391fdd4bde0dc8c0efdd7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/rtp_rtcp/source/packet_loss_stats.h"
> 
>+#include <cstdint>
>+#include <iterator>
> #include <vector>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc
>index d3a75dd13aa34a99f1dfc900c07f4423233806f5..dc33fad536ded3d067f752500209d129b3d28437 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.cc
>@@ -13,7 +13,6 @@
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h
>index 6e6e253c8c76e56d92d8ab5ae1c263f25ff83681..0e3bd39a7ef5ee1ae85dff256abbb7943268ee25 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h
>@@ -13,8 +13,10 @@
> 
> #include <stdint.h>
> 
>-#include "modules/include/module_common_types.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "modules/include/module_common_types_public.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
>index abe028d670237a8921a63d03e23e00bf7da709f9..bc742d1ea63ff7bd4794cceac3ddbd833d12c499 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
>@@ -11,10 +11,11 @@
> #include "modules/rtp_rtcp/source/receive_statistics_impl.h"
> 
> #include <math.h>
>-
> #include <cstdlib>
>+#include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
>@@ -33,13 +34,14 @@ StreamStatisticianImpl::StreamStatisticianImpl(
>     uint32_t ssrc,
>     Clock* clock,
>     bool enable_retransmit_detection,
>+    int max_reordering_threshold,
>     RtcpStatisticsCallback* rtcp_callback,
>     StreamDataCountersCallback* rtp_callback)
>     : ssrc_(ssrc),
>       clock_(clock),
>       incoming_bitrate_(kStatisticsProcessIntervalMs,
>                         RateStatistics::kBpsScale),
>-      max_reordering_threshold_(kDefaultMaxReorderingThreshold),
>+      max_reordering_threshold_(max_reordering_threshold),
>       enable_retransmit_detection_(enable_retransmit_detection),
>       jitter_q4_(0),
>       cumulative_loss_(0),
>@@ -48,7 +50,6 @@ StreamStatisticianImpl::StreamStatisticianImpl(
>       received_seq_first_(0),
>       received_seq_max_(0),
>       received_seq_wraps_(0),
>-      received_packet_overhead_(12),
>       last_report_inorder_packets_(0),
>       last_report_old_packets_(0),
>       last_report_seq_max_(0),
>@@ -57,79 +58,71 @@ StreamStatisticianImpl::StreamStatisticianImpl(
> 
> StreamStatisticianImpl::~StreamStatisticianImpl() = default;
> 
>-void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
>-                                            size_t packet_length) {
>-  StreamDataCounters counters;
>-  {
>-    rtc::CritScope cs(&stream_lock_);
>-
>-    bool retransmitted =
>-        enable_retransmit_detection_ && IsRetransmitOfOldPacket(header);
>-    counters = UpdateCounters(header, packet_length, retransmitted);
>-  }
>-  rtp_callback_->DataCountersUpdated(counters, ssrc_);
>+void StreamStatisticianImpl::OnRtpPacket(const RtpPacketReceived& packet) {
>+  StreamDataCounters counters = UpdateCounters(packet);
>+  if (rtp_callback_)
>+    rtp_callback_->DataCountersUpdated(counters, ssrc_);
> }
> 
> StreamDataCounters StreamStatisticianImpl::UpdateCounters(
>-    const RTPHeader& header,
>-    size_t packet_length,
>-    bool retransmitted) {
>-  bool in_order = InOrderPacketInternal(header.sequenceNumber);
>-  RTC_DCHECK_EQ(ssrc_, header.ssrc);
>-  incoming_bitrate_.Update(packet_length, clock_->TimeInMilliseconds());
>-  receive_counters_.transmitted.AddPacket(packet_length, header);
>-  if (!in_order && retransmitted) {
>-    receive_counters_.retransmitted.AddPacket(packet_length, header);
>+    const RtpPacketReceived& packet) {
>+  rtc::CritScope cs(&stream_lock_);
>+  RTC_DCHECK_EQ(ssrc_, packet.Ssrc());
>+  uint16_t sequence_number = packet.SequenceNumber();
>+  bool in_order =
>+      // First packet is always in order.
>+      last_receive_time_ms_ == 0 ||
>+      IsNewerSequenceNumber(sequence_number, received_seq_max_) ||
>+      // If we have a restart of the remote side this packet is still in order.
>+      !IsNewerSequenceNumber(sequence_number,
>+                             received_seq_max_ - max_reordering_threshold_);
>+  int64_t now_ms = clock_->TimeInMilliseconds();
>+
>+  incoming_bitrate_.Update(packet.size(), now_ms);
>+  receive_counters_.transmitted.AddPacket(packet);
>+  if (!in_order && enable_retransmit_detection_ &&
>+      IsRetransmitOfOldPacket(packet, now_ms)) {
>+    receive_counters_.retransmitted.AddPacket(packet);
>   }
> 
>   if (receive_counters_.transmitted.packets == 1) {
>-    received_seq_first_ = header.sequenceNumber;
>-    receive_counters_.first_packet_time_ms = clock_->TimeInMilliseconds();
>+    received_seq_first_ = packet.SequenceNumber();
>+    receive_counters_.first_packet_time_ms = now_ms;
>   }
> 
>   // Count only the new packets received. That is, if packets 1, 2, 3, 5, 4, 6
>   // are received, 4 will be ignored.
>   if (in_order) {
>-    // Current time in samples.
>-    NtpTime receive_time = clock_->CurrentNtpTime();
>-
>     // Wrong if we use RetransmitOfOldPacket.
>     if (receive_counters_.transmitted.packets > 1 &&
>-        received_seq_max_ > header.sequenceNumber) {
>+        received_seq_max_ > packet.SequenceNumber()) {
>       // Wrap around detected.
>       received_seq_wraps_++;
>     }
>     // New max.
>-    received_seq_max_ = header.sequenceNumber;
>+    received_seq_max_ = packet.SequenceNumber();
> 
>     // If new time stamp and more than one in-order packet received, calculate
>     // new jitter statistics.
>-    if (header.timestamp != last_received_timestamp_ &&
>+    if (packet.Timestamp() != last_received_timestamp_ &&
>         (receive_counters_.transmitted.packets -
>          receive_counters_.retransmitted.packets) > 1) {
>-      UpdateJitter(header, receive_time);
>+      UpdateJitter(packet, now_ms);
>     }
>-    last_received_timestamp_ = header.timestamp;
>-    last_receive_time_ntp_ = receive_time;
>-    last_receive_time_ms_ = clock_->TimeInMilliseconds();
>+    last_received_timestamp_ = packet.Timestamp();
>+    last_receive_time_ms_ = now_ms;
>   }
>-
>-  size_t packet_oh = header.headerLength + header.paddingLength;
>-
>-  // Our measured overhead. Filter from RFC 5104 4.2.1.2:
>-  // avg_OH (new) = 15/16*avg_OH (old) + 1/16*pckt_OH,
>-  received_packet_overhead_ = (15 * received_packet_overhead_ + packet_oh) >> 4;
>   return receive_counters_;
> }
> 
>-void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
>-                                          NtpTime receive_time) {
>-  uint32_t receive_time_rtp =
>-      NtpToRtp(receive_time, header.payload_type_frequency);
>-  uint32_t last_receive_time_rtp =
>-      NtpToRtp(last_receive_time_ntp_, header.payload_type_frequency);
>-  int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
>-                              (header.timestamp - last_received_timestamp_);
>+void StreamStatisticianImpl::UpdateJitter(const RtpPacketReceived& packet,
>+                                          int64_t receive_time_ms) {
>+  int64_t receive_diff_ms = receive_time_ms - last_receive_time_ms_;
>+  RTC_DCHECK_GE(receive_diff_ms, 0);
>+  uint32_t receive_diff_rtp = static_cast<uint32_t>(
>+      (receive_diff_ms * packet.payload_type_frequency()) / 1000);
>+  int32_t time_diff_samples =
>+      receive_diff_rtp - (packet.Timestamp() - last_received_timestamp_);
> 
>   time_diff_samples = std::abs(time_diff_samples);
> 
>@@ -143,15 +136,16 @@ void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
>   }
> }
> 
>-void StreamStatisticianImpl::FecPacketReceived(const RTPHeader& header,
>-                                               size_t packet_length) {
>+void StreamStatisticianImpl::FecPacketReceived(
>+    const RtpPacketReceived& packet) {
>   StreamDataCounters counters;
>   {
>     rtc::CritScope cs(&stream_lock_);
>-    receive_counters_.fec.AddPacket(packet_length, header);
>+    receive_counters_.fec.AddPacket(packet);
>     counters = receive_counters_;
>   }
>-  rtp_callback_->DataCountersUpdated(counters, ssrc_);
>+  if (rtp_callback_)
>+    rtp_callback_->DataCountersUpdated(counters, ssrc_);
> }
> 
> void StreamStatisticianImpl::SetMaxReorderingThreshold(
>@@ -188,7 +182,8 @@ bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics,
>     *statistics = CalculateRtcpStatistics();
>   }
> 
>-  rtcp_callback_->StatisticsUpdated(*statistics, ssrc_);
>+  if (rtcp_callback_)
>+    rtcp_callback_->StatisticsUpdated(*statistics, ssrc_);
>   return true;
> }
> 
>@@ -196,7 +191,7 @@ bool StreamStatisticianImpl::GetActiveStatisticsAndReset(
>     RtcpStatistics* statistics) {
>   {
>     rtc::CritScope cs(&stream_lock_);
>-    if (clock_->CurrentNtpInMilliseconds() - last_receive_time_ntp_.ToMs() >=
>+    if (clock_->TimeInMilliseconds() - last_receive_time_ms_ >=
>         kStatisticsTimeoutMs) {
>       // Not active.
>       return false;
>@@ -210,7 +205,8 @@ bool StreamStatisticianImpl::GetActiveStatisticsAndReset(
>     *statistics = CalculateRtcpStatistics();
>   }
> 
>-  rtcp_callback_->StatisticsUpdated(*statistics, ssrc_);
>+  if (rtcp_callback_)
>+    rtcp_callback_->StatisticsUpdated(*statistics, ssrc_);
>   return true;
> }
> 
>@@ -312,17 +308,15 @@ uint32_t StreamStatisticianImpl::BitrateReceived() const {
> }
> 
> bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
>-    const RTPHeader& header) const {
>-  if (InOrderPacketInternal(header.sequenceNumber)) {
>-    return false;
>-  }
>-  uint32_t frequency_khz = header.payload_type_frequency / 1000;
>-  assert(frequency_khz > 0);
>+    const RtpPacketReceived& packet,
>+    int64_t now_ms) const {
>+  uint32_t frequency_khz = packet.payload_type_frequency() / 1000;
>+  RTC_DCHECK_GT(frequency_khz, 0);
> 
>-  int64_t time_diff_ms = clock_->TimeInMilliseconds() - last_receive_time_ms_;
>+  int64_t time_diff_ms = now_ms - last_receive_time_ms_;
> 
>   // Diff in time stamp since last received in order.
>-  uint32_t timestamp_diff = header.timestamp - last_received_timestamp_;
>+  uint32_t timestamp_diff = packet.Timestamp() - last_received_timestamp_;
>   uint32_t rtp_time_stamp_diff_ms = timestamp_diff / frequency_khz;
> 
>   int64_t max_delay_ms = 0;
>@@ -341,30 +335,23 @@ bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
>   return time_diff_ms > rtp_time_stamp_diff_ms + max_delay_ms;
> }
> 
>-bool StreamStatisticianImpl::InOrderPacketInternal(
>-    uint16_t sequence_number) const {
>-  // First packet is always in order.
>-  if (last_receive_time_ms_ == 0)
>-    return true;
>-
>-  if (IsNewerSequenceNumber(sequence_number, received_seq_max_)) {
>-    return true;
>-  } else {
>-    // If we have a restart of the remote side this packet is still in order.
>-    return !IsNewerSequenceNumber(
>-        sequence_number, received_seq_max_ - max_reordering_threshold_);
>-  }
>-}
>-
>-ReceiveStatistics* ReceiveStatistics::Create(Clock* clock) {
>-  return new ReceiveStatisticsImpl(clock);
>+std::unique_ptr<ReceiveStatistics> ReceiveStatistics::Create(
>+    Clock* clock,
>+    RtcpStatisticsCallback* rtcp_callback,
>+    StreamDataCountersCallback* rtp_callback) {
>+  return absl::make_unique<ReceiveStatisticsImpl>(clock, rtcp_callback,
>+                                                  rtp_callback);
> }
> 
>-ReceiveStatisticsImpl::ReceiveStatisticsImpl(Clock* clock)
>+ReceiveStatisticsImpl::ReceiveStatisticsImpl(
>+    Clock* clock,
>+    RtcpStatisticsCallback* rtcp_callback,
>+    StreamDataCountersCallback* rtp_callback)
>     : clock_(clock),
>       last_returned_ssrc_(0),
>-      rtcp_stats_callback_(NULL),
>-      rtp_stats_callback_(NULL) {}
>+      max_reordering_threshold_(kDefaultMaxReorderingThreshold),
>+      rtcp_stats_callback_(rtcp_callback),
>+      rtp_stats_callback_(rtp_callback) {}
> 
> ReceiveStatisticsImpl::~ReceiveStatisticsImpl() {
>   while (!statisticians_.empty()) {
>@@ -374,31 +361,24 @@ ReceiveStatisticsImpl::~ReceiveStatisticsImpl() {
> }
> 
> void ReceiveStatisticsImpl::OnRtpPacket(const RtpPacketReceived& packet) {
>-  RTPHeader header;
>-  packet.GetHeader(&header);
>-  IncomingPacket(header, packet.size());
>-}
>-
>-void ReceiveStatisticsImpl::IncomingPacket(const RTPHeader& header,
>-                                           size_t packet_length) {
>   StreamStatisticianImpl* impl;
>   {
>     rtc::CritScope cs(&receive_statistics_lock_);
>-    auto it = statisticians_.find(header.ssrc);
>+    auto it = statisticians_.find(packet.Ssrc());
>     if (it != statisticians_.end()) {
>       impl = it->second;
>     } else {
>       impl = new StreamStatisticianImpl(
>-          header.ssrc, clock_, /* enable_retransmit_detection = */ false, this,
>-          this);
>-      statisticians_[header.ssrc] = impl;
>+          packet.Ssrc(), clock_, /* enable_retransmit_detection = */ false,
>+          max_reordering_threshold_, rtcp_stats_callback_, rtp_stats_callback_);
>+      statisticians_[packet.Ssrc()] = impl;
>     }
>   }
>   // StreamStatisticianImpl instance is created once and only destroyed when
>   // this whole ReceiveStatisticsImpl is destroyed. StreamStatisticianImpl has
>   // it's own locking so don't hold receive_statistics_lock_ (potential
>   // deadlock).
>-  impl->IncomingPacket(header, packet_length);
>+  impl->OnRtpPacket(packet);
> }
> 
> void ReceiveStatisticsImpl::FecPacketReceived(const RtpPacketReceived& packet) {
>@@ -411,9 +391,7 @@ void ReceiveStatisticsImpl::FecPacketReceived(const RtpPacketReceived& packet) {
>       return;
>     impl = it->second;
>   }
>-  RTPHeader header;
>-  packet.GetHeader(&header);
>-  impl->FecPacketReceived(header, packet.size());
>+  impl->FecPacketReceived(packet);
> }
> 
> StreamStatistician* ReceiveStatisticsImpl::GetStatistician(
>@@ -427,8 +405,13 @@ StreamStatistician* ReceiveStatisticsImpl::GetStatistician(
> 
> void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
>     int max_reordering_threshold) {
>-  rtc::CritScope cs(&receive_statistics_lock_);
>-  for (auto& statistician : statisticians_) {
>+  std::map<uint32_t, StreamStatisticianImpl*> statisticians;
>+  {
>+    rtc::CritScope cs(&receive_statistics_lock_);
>+    max_reordering_threshold_ = max_reordering_threshold;
>+    statisticians = statisticians_;
>+  }
>+  for (auto& statistician : statisticians) {
>     statistician.second->SetMaxReorderingThreshold(max_reordering_threshold);
>   }
> }
>@@ -440,7 +423,9 @@ void ReceiveStatisticsImpl::EnableRetransmitDetection(uint32_t ssrc,
>     rtc::CritScope cs(&receive_statistics_lock_);
>     StreamStatisticianImpl*& impl_ref = statisticians_[ssrc];
>     if (impl_ref == nullptr) {  // new element
>-      impl_ref = new StreamStatisticianImpl(ssrc, clock_, enable, this, this);
>+      impl_ref = new StreamStatisticianImpl(
>+          ssrc, clock_, enable, max_reordering_threshold_, rtcp_stats_callback_,
>+          rtp_stats_callback_);
>       return;
>     }
>     impl = impl_ref;
>@@ -448,43 +433,6 @@ void ReceiveStatisticsImpl::EnableRetransmitDetection(uint32_t ssrc,
>   impl->EnableRetransmitDetection(enable);
> }
> 
>-void ReceiveStatisticsImpl::RegisterRtcpStatisticsCallback(
>-    RtcpStatisticsCallback* callback) {
>-  rtc::CritScope cs(&receive_statistics_lock_);
>-  if (callback != NULL)
>-    assert(rtcp_stats_callback_ == NULL);
>-  rtcp_stats_callback_ = callback;
>-}
>-
>-void ReceiveStatisticsImpl::StatisticsUpdated(const RtcpStatistics& statistics,
>-                                              uint32_t ssrc) {
>-  rtc::CritScope cs(&receive_statistics_lock_);
>-  if (rtcp_stats_callback_)
>-    rtcp_stats_callback_->StatisticsUpdated(statistics, ssrc);
>-}
>-
>-void ReceiveStatisticsImpl::CNameChanged(const char* cname, uint32_t ssrc) {
>-  rtc::CritScope cs(&receive_statistics_lock_);
>-  if (rtcp_stats_callback_)
>-    rtcp_stats_callback_->CNameChanged(cname, ssrc);
>-}
>-
>-void ReceiveStatisticsImpl::RegisterRtpStatisticsCallback(
>-    StreamDataCountersCallback* callback) {
>-  rtc::CritScope cs(&receive_statistics_lock_);
>-  if (callback != NULL)
>-    assert(rtp_stats_callback_ == NULL);
>-  rtp_stats_callback_ = callback;
>-}
>-
>-void ReceiveStatisticsImpl::DataCountersUpdated(const StreamDataCounters& stats,
>-                                                uint32_t ssrc) {
>-  rtc::CritScope cs(&receive_statistics_lock_);
>-  if (rtp_stats_callback_) {
>-    rtp_stats_callback_->DataCountersUpdated(stats, ssrc);
>-  }
>-}
>-
> std::vector<rtcp::ReportBlock> ReceiveStatisticsImpl::RtcpReportBlocks(
>     size_t max_blocks) {
>   std::map<uint32_t, StreamStatisticianImpl*> statisticians;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
>index 56bfd2b4a405ce6d1786566bdd69a1b9c035010d..8153c4400d40bcf595fdb820db829a5aad72fd9c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
>@@ -19,15 +19,17 @@
> 
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/rate_statistics.h"
>-#include "system_wrappers/include/ntp_time.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
> 
>-class StreamStatisticianImpl : public StreamStatistician {
>+class StreamStatisticianImpl : public StreamStatistician,
>+                               public RtpPacketSinkInterface {
>  public:
>   StreamStatisticianImpl(uint32_t ssrc,
>                          Clock* clock,
>                          bool enable_retransmit_detection,
>+                         int max_reordering_threshold,
>                          RtcpStatisticsCallback* rtcp_callback,
>                          StreamDataCountersCallback* rtp_callback);
>   ~StreamStatisticianImpl() override;
>@@ -41,24 +43,22 @@ class StreamStatisticianImpl : public StreamStatistician {
>       StreamDataCounters* data_counters) const override;
>   uint32_t BitrateReceived() const override;
> 
>-  void IncomingPacket(const RTPHeader& rtp_header, size_t packet_length);
>-  void FecPacketReceived(const RTPHeader& header, size_t packet_length);
>+  // Implements RtpPacketSinkInterface
>+  void OnRtpPacket(const RtpPacketReceived& packet) override;
>+
>+  void FecPacketReceived(const RtpPacketReceived& packet);
>   void SetMaxReorderingThreshold(int max_reordering_threshold);
>   void EnableRetransmitDetection(bool enable);
> 
>  private:
>-  bool IsRetransmitOfOldPacket(const RTPHeader& header) const
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
>-  bool InOrderPacketInternal(uint16_t sequence_number) const
>+  bool IsRetransmitOfOldPacket(const RtpPacketReceived& packet,
>+                               int64_t now_ms) const
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
>   RtcpStatistics CalculateRtcpStatistics()
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
>-  void UpdateJitter(const RTPHeader& header, NtpTime receive_time)
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
>-  StreamDataCounters UpdateCounters(const RTPHeader& rtp_header,
>-                                    size_t packet_length,
>-                                    bool retransmitted)
>+  void UpdateJitter(const RtpPacketReceived& packet, int64_t receive_time_ms)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
>+  StreamDataCounters UpdateCounters(const RtpPacketReceived& packet);
> 
>   const uint32_t ssrc_;
>   Clock* const clock_;
>@@ -73,14 +73,12 @@ class StreamStatisticianImpl : public StreamStatistician {
>   uint32_t cumulative_loss_ RTC_GUARDED_BY(&stream_lock_);
> 
>   int64_t last_receive_time_ms_ RTC_GUARDED_BY(&stream_lock_);
>-  NtpTime last_receive_time_ntp_ RTC_GUARDED_BY(&stream_lock_);
>   uint32_t last_received_timestamp_ RTC_GUARDED_BY(&stream_lock_);
>   uint16_t received_seq_first_ RTC_GUARDED_BY(&stream_lock_);
>   uint16_t received_seq_max_ RTC_GUARDED_BY(&stream_lock_);
>   uint16_t received_seq_wraps_ RTC_GUARDED_BY(&stream_lock_);
> 
>   // Current counter values.
>-  size_t received_packet_overhead_ RTC_GUARDED_BY(&stream_lock_);
>   StreamDataCounters receive_counters_ RTC_GUARDED_BY(&stream_lock_);
> 
>   // Counter values when we sent the last report.
>@@ -94,47 +92,36 @@ class StreamStatisticianImpl : public StreamStatistician {
>   StreamDataCountersCallback* const rtp_callback_;
> };
> 
>-class ReceiveStatisticsImpl : public ReceiveStatistics,
>-                              public RtcpStatisticsCallback,
>-                              public StreamDataCountersCallback {
>+class ReceiveStatisticsImpl : public ReceiveStatistics {
>  public:
>-  explicit ReceiveStatisticsImpl(Clock* clock);
>+  ReceiveStatisticsImpl(Clock* clock,
>+                        RtcpStatisticsCallback* rtcp_callback,
>+                        StreamDataCountersCallback* rtp_callback);
> 
>   ~ReceiveStatisticsImpl() override;
> 
>-  // Implement ReceiveStatisticsProvider.
>+  // Implements ReceiveStatisticsProvider.
>   std::vector<rtcp::ReportBlock> RtcpReportBlocks(size_t max_blocks) override;
> 
>-  // Implement RtpPacketSinkInterface
>+  // Implements RtpPacketSinkInterface
>   void OnRtpPacket(const RtpPacketReceived& packet) override;
> 
>-  // Implement ReceiveStatistics.
>-  void IncomingPacket(const RTPHeader& header, size_t packet_length) override;
>+  // Implements ReceiveStatistics.
>   void FecPacketReceived(const RtpPacketReceived& packet) override;
>   StreamStatistician* GetStatistician(uint32_t ssrc) const override;
>   void SetMaxReorderingThreshold(int max_reordering_threshold) override;
>   void EnableRetransmitDetection(uint32_t ssrc, bool enable) override;
> 
>-  void RegisterRtcpStatisticsCallback(
>-      RtcpStatisticsCallback* callback) override;
>-
>-  void RegisterRtpStatisticsCallback(
>-      StreamDataCountersCallback* callback) override;
>-
>  private:
>-  void StatisticsUpdated(const RtcpStatistics& statistics,
>-                         uint32_t ssrc) override;
>-  void CNameChanged(const char* cname, uint32_t ssrc) override;
>-  void DataCountersUpdated(const StreamDataCounters& counters,
>-                           uint32_t ssrc) override;
>-
>   Clock* const clock_;
>   rtc::CriticalSection receive_statistics_lock_;
>   uint32_t last_returned_ssrc_;
>-  std::map<uint32_t, StreamStatisticianImpl*> statisticians_;
>+  int max_reordering_threshold_ RTC_GUARDED_BY(receive_statistics_lock_);
>+  std::map<uint32_t, StreamStatisticianImpl*> statisticians_
>+      RTC_GUARDED_BY(receive_statistics_lock_);
> 
>-  RtcpStatisticsCallback* rtcp_stats_callback_;
>-  StreamDataCountersCallback* rtp_stats_callback_;
>+  RtcpStatisticsCallback* const rtcp_stats_callback_;
>+  StreamDataCountersCallback* const rtp_stats_callback_;
> };
> }  // namespace webrtc
> #endif  // MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
>index 18448cb77ec6e5551b92d81d9dea2d55aabff8e7..25393631df9b89bce1d719791b3db06d848265d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
>@@ -48,10 +48,7 @@ RtpPacketReceived CreateRtpPacket(uint32_t ssrc,
>     packet.SetCsrcs(csrcs);
>   }
>   packet.SetPayloadSize(payload_size);
>-  if (padding_size > 0) {
>-    Random random(17);
>-    packet.SetPadding(padding_size, &random);
>-  }
>+  packet.SetPadding(padding_size);
>   return packet;
> }
> 
>@@ -74,7 +71,9 @@ void IncrementTimestamp(RtpPacketReceived* packet, uint32_t incr) {
> class ReceiveStatisticsTest : public ::testing::Test {
>  public:
>   ReceiveStatisticsTest()
>-      : clock_(0), receive_statistics_(ReceiveStatistics::Create(&clock_)) {
>+      : clock_(0),
>+        receive_statistics_(
>+            ReceiveStatistics::Create(&clock_, nullptr, nullptr)) {
>     packet1_ = CreateRtpPacket(kSsrc1, kPacketSize1);
>     packet2_ = CreateRtpPacket(kSsrc2, kPacketSize2);
>   }
>@@ -254,7 +253,7 @@ TEST_F(ReceiveStatisticsTest, RtcpCallbacks) {
>     RtcpStatistics stats_;
>   } callback;
> 
>-  receive_statistics_->RegisterRtcpStatisticsCallback(&callback);
>+  receive_statistics_ = ReceiveStatistics::Create(&clock_, &callback, nullptr);
>   receive_statistics_->EnableRetransmitDetection(kSsrc1, true);
> 
>   // Add some arbitrary data, with loss and jitter.
>@@ -294,33 +293,6 @@ TEST_F(ReceiveStatisticsTest, RtcpCallbacks) {
>   EXPECT_EQ(1, statistics.packets_lost);
>   EXPECT_EQ(5u, statistics.extended_highest_sequence_number);
>   EXPECT_EQ(177u, statistics.jitter);
>-
>-  receive_statistics_->RegisterRtcpStatisticsCallback(NULL);
>-
>-  // Add some more data.
>-  packet1_.SetSequenceNumber(1);
>-  clock_.AdvanceTimeMilliseconds(7);
>-  IncrementTimestamp(&packet1_, 3);
>-  receive_statistics_->OnRtpPacket(packet1_);
>-  IncrementSequenceNumber(&packet1_, 2);
>-  clock_.AdvanceTimeMilliseconds(9);
>-  IncrementTimestamp(&packet1_, 9);
>-  receive_statistics_->OnRtpPacket(packet1_);
>-  IncrementSequenceNumber(&packet1_, -1);
>-  clock_.AdvanceTimeMilliseconds(13);
>-  IncrementTimestamp(&packet1_, 47);
>-  receive_statistics_->OnRtpPacket(packet1_);
>-  IncrementSequenceNumber(&packet1_, 3);
>-  clock_.AdvanceTimeMilliseconds(11);
>-  IncrementTimestamp(&packet1_, 17);
>-  receive_statistics_->OnRtpPacket(packet1_);
>-  IncrementSequenceNumber(&packet1_);
>-
>-  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(&statistics,
>-                                                              true);
>-
>-  // Should not have been called after deregister.
>-  EXPECT_EQ(1u, callback.num_calls_);
> }
> 
> class RtpTestCallback : public StreamDataCountersCallback {
>@@ -361,7 +333,7 @@ class RtpTestCallback : public StreamDataCountersCallback {
> 
> TEST_F(ReceiveStatisticsTest, RtpCallbacks) {
>   RtpTestCallback callback;
>-  receive_statistics_->RegisterRtpStatisticsCallback(&callback);
>+  receive_statistics_ = ReceiveStatistics::Create(&clock_, nullptr, &callback);
>   receive_statistics_->EnableRetransmitDetection(kSsrc1, true);
> 
>   const size_t kHeaderLength = 20;
>@@ -420,19 +392,11 @@ TEST_F(ReceiveStatisticsTest, RtpCallbacks) {
>   expected.fec.header_bytes = kHeaderLength;
>   expected.fec.packets = 1;
>   callback.Matches(5, kSsrc1, expected);
>-
>-  receive_statistics_->RegisterRtpStatisticsCallback(NULL);
>-
>-  // New stats, but callback should not be called.
>-  IncrementSequenceNumber(&packet1);
>-  clock_.AdvanceTimeMilliseconds(5);
>-  receive_statistics_->OnRtpPacket(packet1);
>-  callback.Matches(5, kSsrc1, expected);
> }
> 
> TEST_F(ReceiveStatisticsTest, RtpCallbacksFecFirst) {
>   RtpTestCallback callback;
>-  receive_statistics_->RegisterRtpStatisticsCallback(&callback);
>+  receive_statistics_ = ReceiveStatistics::Create(&clock_, nullptr, &callback);
> 
>   const uint32_t kHeaderLength = 20;
>   RtpPacketReceived packet =
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
>index fc867a491a3749896620fcf87630beaf1f18208e..fd19b1383a44e3c8e9faeb6837b3dbd3f208e201 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
>@@ -10,8 +10,9 @@
> 
> #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
> 
>+#include <cstdint>
>+
> #include "rtc_base/logging.h"
>-#include "rtc_base/time/timestamp_extrapolator.h"
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
>index 5254cd57bb933b410894d12fac7715eaf91d2f0f..b301461427b697c95049619b0ba5b7da8c3e8882 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
>@@ -56,7 +56,8 @@ class RemoteNtpTimeEstimatorTest : public ::testing::Test {
>                               int64_t networking_delay_ms) {
>     uint32_t rtcp_timestamp = GetRemoteTimestamp();
>     int64_t ntp_error_fractions =
>-        ntp_error_ms * NtpTime::kFractionsPerSecond / 1000;
>+        ntp_error_ms * static_cast<int64_t>(NtpTime::kFractionsPerSecond) /
>+        1000;
>     NtpTime ntp(static_cast<uint64_t>(remote_clock_.CurrentNtpTime()) +
>                 ntp_error_fractions);
>     AdvanceTimeMilliseconds(kTestRtt / 2 + networking_delay_ms);
>@@ -110,7 +111,17 @@ TEST_F(RemoteNtpTimeEstimatorTest, Estimate) {
> }
> 
> TEST_F(RemoteNtpTimeEstimatorTest, AveragesErrorsOut) {
>-  // Remote peer sends first 5 RTCP SR without errors.
>+  // Remote peer sends first 10 RTCP SR without errors.
>+  AdvanceTimeMilliseconds(1000);
>+  SendRtcpSr();
>+  AdvanceTimeMilliseconds(1000);
>+  SendRtcpSr();
>+  AdvanceTimeMilliseconds(1000);
>+  SendRtcpSr();
>+  AdvanceTimeMilliseconds(1000);
>+  SendRtcpSr();
>+  AdvanceTimeMilliseconds(1000);
>+  SendRtcpSr();
>   AdvanceTimeMilliseconds(1000);
>   SendRtcpSr();
>   AdvanceTimeMilliseconds(1000);
>@@ -122,18 +133,17 @@ TEST_F(RemoteNtpTimeEstimatorTest, AveragesErrorsOut) {
>   AdvanceTimeMilliseconds(1000);
>   SendRtcpSr();
> 
>-  AdvanceTimeMilliseconds(15);
>+  AdvanceTimeMilliseconds(150);
>   uint32_t rtp_timestamp = GetRemoteTimestamp();
>   int64_t capture_ntp_time_ms = local_clock_.CurrentNtpInMilliseconds();
>-
>   // Local peer gets enough RTCP SR to calculate the capture time.
>   EXPECT_EQ(capture_ntp_time_ms, estimator_->Estimate(rtp_timestamp));
> 
>   // Remote sends corrupted RTCP SRs
>   AdvanceTimeMilliseconds(1000);
>-  SendRtcpSrInaccurately(10, 10);
>+  SendRtcpSrInaccurately(/*ntp_error_ms=*/2, /*networking_delay_ms=*/-1);
>   AdvanceTimeMilliseconds(1000);
>-  SendRtcpSrInaccurately(-20, 5);
>+  SendRtcpSrInaccurately(/*ntp_error_ms=*/-2, /*networking_delay_ms=*/1);
> 
>   // New RTP packet to estimate timestamp.
>   AdvanceTimeMilliseconds(150);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc
>index 24b708502ee6c2751aeeb0c1b41d43dc0316644d..1d652d0b5ba7bc7581be0bf920aa514a68e2be9b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc
>@@ -10,7 +10,7 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_nack_stats.h"
> 
>-#include "modules/include/module_common_types.h"
>+#include "modules/include/module_common_types_public.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
>index 11037cbd82d627376bcdaab497924dd53dba56d5..40e51e87ad33d6f0a00417ace80b2d0b0e1ebda5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
>@@ -11,6 +11,10 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
>+#include "api/array_view.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/function_view.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
>index 4e21bc93e4879900a798934476467c4213ee99e0..eadd4d9c352f8fff473fa36d14558b335da882be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
>@@ -10,6 +10,9 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/app.h"
> 
>+#include <string.h>
>+#include <cstdint>
>+
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
>index 19a97e0ed920a31a54ae25d37b6435de1b9065ff..a9602a80cfb5a878b8b5bcd369f045b809846126 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h
>@@ -11,6 +11,9 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_APP_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
> #include "rtc_base/buffer.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
>index 0e2eb9e41a3e1d2c27652c5e10580f61999973a4..23ac35f856b5389133dd0fb54d8f7e59713e3fb1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
> 
>+#include <string.h>
>+#include <cstdint>
> #include <utility>
> 
> #include "modules/rtp_rtcp/source/byte_io.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc
>index 27ed4ccef4f6321aa23e861fd3df2c3cc4d92654..5e7dadd1f482a11d38dea0ba7f14b1e4ad6c5975 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
> 
>+#include <cstdint>
> #include <utility>
> 
> #include "modules/rtp_rtcp/source/byte_io.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc
>index 5513f37dba0b1f374b7c13a23f9aa2c76af740ac..2b5f9cacd72821392bb88b2e391917c9b32f8c1a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
> 
>+#include <vector>
>+
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc
>index 6a4a0bdc0e43ba7c348eff75c09fa39f9821d56a..6fe7eade62b6478096b2dbbade23cb32835d91fa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.cc
>@@ -11,6 +11,7 @@
> #include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
> 
> #include <algorithm>
>+#include <cstdint>
> #include <utility>
> 
> #include "modules/rtp_rtcp/source/byte_io.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
>index ae66a172387ecc86e177f1357c166e12339daec6..46ee291285046085bf43363cfe5f965bc16287ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
>@@ -12,6 +12,9 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
>index 8f143da72c0901ef63c0aada2c289dec94057a82..7470d1d2e10abb72cdb7daf5304bf34412a8291a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
>@@ -11,6 +11,8 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RECEIVER_REPORT_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <vector>
> 
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc
>index 02406116f6e793e6b02b6466fa8939022165b9c9..3ed1fbdb803e6434a1f25585babbe4a59e6aecf1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
> 
>+#include <cstdint>
> #include <utility>
> 
> #include "modules/rtp_rtcp/source/byte_io.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
>index a470b1a217c556b530486e8a805f4b2bebd81c33..8eb4ce62adef8b86a8ad6d9fd5d7cd7566811b99 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
>@@ -12,6 +12,9 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "system_wrappers/include/ntp_time.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
>index a0407411ab38800c096bbb16a182fa68594d14df..21977736b619e487be8d9e9c0dafc29ed65766cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
>@@ -12,6 +12,9 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.cc
>index 337c8b04a8487092e46ce0a36828e43f442ced7b..0ef432903db55eff4721632455b083c9417ea825 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
> 
>+#include <string.h>
> #include <utility>
> 
> #include "modules/rtp_rtcp/source/byte_io.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
>index 4d38b3b6dbb3091d5d29f8cf01ea5d5c3d3b6e28..f57e5749c2ded05a9ca1bc2f8620c7ae231fccd1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
>@@ -10,7 +10,6 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
> 
>-#include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
>index d8f073b87f309df4649b569d3958b97d97554a97..9dc745e509c48a090e81f92e98308339a1f35a59 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
>@@ -10,7 +10,6 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
> 
>-#include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
>index 4703d3132ee872b3b8917dad60851c3b4893a354..28165594bec4c995e090aabecdcbabb6647786e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
>@@ -11,9 +11,10 @@
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
> 
> #include <algorithm>
>+#include <cstdint>
> #include <utility>
> 
>-#include "modules/include/module_common_types.h"
>+#include "modules/include/module_common_types_public.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
>index 496675497ad23e3b5cecc44118f9bc47c682268c..383f785dfea82062251d0a4fca71e6fab9a278a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
>@@ -131,6 +131,7 @@ RTCPReceiver::RTCPReceiver(
>     RtcpIntraFrameObserver* rtcp_intra_frame_observer,
>     TransportFeedbackObserver* transport_feedback_observer,
>     VideoBitrateAllocationObserver* bitrate_allocation_observer,
>+    int report_interval_ms,
>     ModuleRtpRtcp* owner)
>     : clock_(clock),
>       receiver_only_(receiver_only),
>@@ -139,6 +140,7 @@ RTCPReceiver::RTCPReceiver(
>       rtcp_intra_frame_observer_(rtcp_intra_frame_observer),
>       transport_feedback_observer_(transport_feedback_observer),
>       bitrate_allocation_observer_(bitrate_allocation_observer),
>+      report_interval_ms_(report_interval_ms),
>       main_ssrc_(0),
>       remote_ssrc_(0),
>       remote_sender_rtp_time_(0),
>@@ -280,7 +282,8 @@ RTCPReceiver::ConsumeReceivedXrReferenceTimeInfo() {
>   std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis;
>   last_xr_rtis.reserve(last_xr_rtis_size);
> 
>-  const uint32_t now_ntp = CompactNtp(clock_->CurrentNtpTime());
>+  const uint32_t now_ntp =
>+      CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds()));
> 
>   for (size_t i = 0; i < last_xr_rtis_size; ++i) {
>     RrtrInformation& rrtr = received_rrtrs_.front();
>@@ -427,7 +430,7 @@ void RTCPReceiver::HandleSenderReport(const CommonHeader& rtcp_block,
> 
>     remote_sender_ntp_time_ = sender_report.ntp();
>     remote_sender_rtp_time_ = sender_report.rtp_timestamp();
>-    last_received_sr_ntp_ = clock_->CurrentNtpTime();
>+    last_received_sr_ntp_ = TimeMicrosToNtp(clock_->TimeInMicroseconds());
>   } else {
>     // We will only store the send report from one source, but
>     // we will store all the receive blocks.
>@@ -503,10 +506,18 @@ void RTCPReceiver::HandleReportBlock(const ReportBlock& report_block,
>   // If no SR has been received yet, the field is set to zero.
>   // Receiver rtp_rtcp module is not expected to calculate rtt using
>   // Sender Reports even if it accidentally can.
>-  if (!receiver_only_ && send_time_ntp != 0) {
>+
>+  // TODO(nisse): Use this way to determine the RTT only when |receiver_only_|
>+  // is false. However, that currently breaks the tests of the
>+  // googCaptureStartNtpTimeMs stat for audio receive streams. To fix, either
>+  // delete all dependencies on RTT measurements for audio receive streams, or
>+  // ensure that audio receive streams that need RTT and stats that depend on it
>+  // are configured with an associated audio send stream.
>+  if (send_time_ntp != 0) {
>     uint32_t delay_ntp = report_block.delay_since_last_sr();
>     // Local NTP time.
>-    uint32_t receive_time_ntp = CompactNtp(clock_->CurrentNtpTime());
>+    uint32_t receive_time_ntp =
>+        CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds()));
> 
>     // RTT in 1/(2^16) seconds.
>     uint32_t rtt_ntp = receive_time_ntp - delay_ntp - send_time_ntp;
>@@ -552,12 +563,12 @@ RTCPReceiver::TmmbrInformation* RTCPReceiver::GetTmmbrInformation(
>   return &it->second;
> }
> 
>-bool RTCPReceiver::RtcpRrTimeout(int64_t rtcp_interval_ms) {
>+bool RTCPReceiver::RtcpRrTimeout() {
>   rtc::CritScope lock(&rtcp_receiver_lock_);
>   if (last_received_rb_ms_ == 0)
>     return false;
> 
>-  int64_t time_out_ms = kRrTimeoutIntervals * rtcp_interval_ms;
>+  int64_t time_out_ms = kRrTimeoutIntervals * report_interval_ms_;
>   if (clock_->TimeInMilliseconds() > last_received_rb_ms_ + time_out_ms) {
>     // Reset the timer to only trigger one log.
>     last_received_rb_ms_ = 0;
>@@ -566,12 +577,12 @@ bool RTCPReceiver::RtcpRrTimeout(int64_t rtcp_interval_ms) {
>   return false;
> }
> 
>-bool RTCPReceiver::RtcpRrSequenceNumberTimeout(int64_t rtcp_interval_ms) {
>+bool RTCPReceiver::RtcpRrSequenceNumberTimeout() {
>   rtc::CritScope lock(&rtcp_receiver_lock_);
>   if (last_increased_sequence_number_ms_ == 0)
>     return false;
> 
>-  int64_t time_out_ms = kRrTimeoutIntervals * rtcp_interval_ms;
>+  int64_t time_out_ms = kRrTimeoutIntervals * report_interval_ms_;
>   if (clock_->TimeInMilliseconds() >
>       last_increased_sequence_number_ms_ + time_out_ms) {
>     // Reset the timer to only trigger one log.
>@@ -720,7 +731,8 @@ void RTCPReceiver::HandleXr(const CommonHeader& rtcp_block,
> void RTCPReceiver::HandleXrReceiveReferenceTime(uint32_t sender_ssrc,
>                                                 const rtcp::Rrtr& rrtr) {
>   uint32_t received_remote_mid_ntp_time = CompactNtp(rrtr.ntp());
>-  uint32_t local_receive_mid_ntp_time = CompactNtp(clock_->CurrentNtpTime());
>+  uint32_t local_receive_mid_ntp_time =
>+      CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds()));
> 
>   auto it = received_rrtrs_ssrc_it_.find(sender_ssrc);
>   if (it != received_rrtrs_ssrc_it_.end()) {
>@@ -754,7 +766,7 @@ void RTCPReceiver::HandleXrDlrrReportBlock(const rtcp::ReceiveTimeInfo& rti) {
>     return;
> 
>   uint32_t delay_ntp = rti.delay_since_last_rr;
>-  uint32_t now_ntp = CompactNtp(clock_->CurrentNtpTime());
>+  uint32_t now_ntp = CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds()));
> 
>   uint32_t rtt_ntp = now_ntp - delay_ntp - send_time_ntp;
>   xr_rr_rtt_ms_ = CompactNtpRttToMs(rtt_ntp);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
>index a863caeff7b69e6c97b034eb7306013e18713b31..be4c70e3cea49186e009bd17f6d0cd4e1b448645 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
>@@ -17,6 +17,7 @@
> #include <string>
> #include <vector>
> 
>+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_nack_stats.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
>@@ -56,6 +57,7 @@ class RTCPReceiver {
>                RtcpIntraFrameObserver* rtcp_intra_frame_observer,
>                TransportFeedbackObserver* transport_feedback_observer,
>                VideoBitrateAllocationObserver* bitrate_allocation_observer,
>+               int report_interval_ms,
>                ModuleRtpRtcp* owner);
>   virtual ~RTCPReceiver();
> 
>@@ -94,13 +96,13 @@ class RTCPReceiver {
> 
>   // Returns true if we haven't received an RTCP RR for several RTCP
>   // intervals, but only triggers true once.
>-  bool RtcpRrTimeout(int64_t rtcp_interval_ms);
>+  bool RtcpRrTimeout();
> 
>   // Returns true if we haven't received an RTCP RR telling the receive side
>   // has not received RTP packets for too long, i.e. extended highest sequence
>   // number hasn't increased for several RTCP intervals. The function only
>   // returns true once until a new RR is received.
>-  bool RtcpRrSequenceNumberTimeout(int64_t rtcp_interval_ms);
>+  bool RtcpRrSequenceNumberTimeout();
> 
>   std::vector<rtcp::TmmbItem> TmmbrReceived();
>   // Return true if new bandwidth should be set.
>@@ -215,6 +217,7 @@ class RTCPReceiver {
>   RtcpIntraFrameObserver* const rtcp_intra_frame_observer_;
>   TransportFeedbackObserver* const transport_feedback_observer_;
>   VideoBitrateAllocationObserver* const bitrate_allocation_observer_;
>+  const int report_interval_ms_;
> 
>   rtc::CriticalSection rtcp_receiver_lock_;
>   uint32_t main_ssrc_ RTC_GUARDED_BY(rtcp_receiver_lock_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
>index c7708f0bddd97caa4bbd64df48360f68662ff04a..a576fdc5e9a576391985fe84379d8c143204db2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
>@@ -107,6 +107,8 @@ constexpr uint32_t kReceiverExtraSsrc = 0x1234567;
> constexpr uint32_t kNotToUsSsrc = 0x654321;
> constexpr uint32_t kUnknownSenderSsrc = 0x54321;
> 
>+constexpr int64_t kRtcpIntervalMs = 1000;
>+
> }  // namespace
> 
> class RtcpReceiverTest : public ::testing::Test {
>@@ -120,6 +122,7 @@ class RtcpReceiverTest : public ::testing::Test {
>                        &intra_frame_observer_,
>                        &transport_feedback_observer_,
>                        &bitrate_allocation_observer_,
>+                       kRtcpIntervalMs,
>                        &rtp_rtcp_impl_) {}
>   void SetUp() {
>     std::set<uint32_t> ssrcs = {kReceiverMainSsrc, kReceiverExtraSsrc};
>@@ -208,7 +211,8 @@ TEST_F(RtcpReceiverTest, InjectSrPacketCalculatesRTT) {
>   EXPECT_EQ(
>       -1, rtcp_receiver_.RTT(kSenderSsrc, &rtt_ms, nullptr, nullptr, nullptr));
> 
>-  uint32_t sent_ntp = CompactNtp(system_clock_.CurrentNtpTime());
>+  uint32_t sent_ntp =
>+      CompactNtp(TimeMicrosToNtp(system_clock_.TimeInMicroseconds()));
>   system_clock_.AdvanceTimeMilliseconds(kRttMs + kDelayMs);
> 
>   rtcp::SenderReport sr;
>@@ -238,7 +242,8 @@ TEST_F(RtcpReceiverTest, InjectSrPacketCalculatesNegativeRTTAsOne) {
>   EXPECT_EQ(
>       -1, rtcp_receiver_.RTT(kSenderSsrc, &rtt_ms, nullptr, nullptr, nullptr));
> 
>-  uint32_t sent_ntp = CompactNtp(system_clock_.CurrentNtpTime());
>+  uint32_t sent_ntp =
>+      CompactNtp(TimeMicrosToNtp(system_clock_.TimeInMicroseconds()));
>   system_clock_.AdvanceTimeMilliseconds(kRttMs + kDelayMs);
> 
>   rtcp::SenderReport sr;
>@@ -266,7 +271,8 @@ TEST_F(
>   const uint32_t kDelayNtp = 123000;
>   const int64_t kDelayMs = CompactNtpRttToMs(kDelayNtp);
> 
>-  uint32_t sent_ntp = CompactNtp(system_clock_.CurrentNtpTime());
>+  uint32_t sent_ntp =
>+      CompactNtp(TimeMicrosToNtp(system_clock_.TimeInMicroseconds()));
>   system_clock_.AdvanceTimeMilliseconds(kRttMs + kDelayMs);
> 
>   rtcp::SenderReport sr;
>@@ -737,7 +743,8 @@ TEST_F(RtcpReceiverTest, InjectExtendedReportsDlrrPacketWithSubBlock) {
> 
>   InjectRtcpPacket(xr);
> 
>-  uint32_t compact_ntp_now = CompactNtp(system_clock_.CurrentNtpTime());
>+  uint32_t compact_ntp_now =
>+      CompactNtp(TimeMicrosToNtp(system_clock_.TimeInMicroseconds()));
>   EXPECT_TRUE(rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms));
>   uint32_t rtt_ntp = compact_ntp_now - kDelay - kLastRR;
>   EXPECT_NEAR(CompactNtpRttToMs(rtt_ntp), rtt_ms, 1);
>@@ -756,7 +763,8 @@ TEST_F(RtcpReceiverTest, InjectExtendedReportsDlrrPacketWithMultipleSubBlocks) {
> 
>   InjectRtcpPacket(xr);
> 
>-  uint32_t compact_ntp_now = CompactNtp(system_clock_.CurrentNtpTime());
>+  uint32_t compact_ntp_now =
>+      CompactNtp(TimeMicrosToNtp(system_clock_.TimeInMicroseconds()));
>   int64_t rtt_ms = 0;
>   EXPECT_TRUE(rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms));
>   uint32_t rtt_ntp = compact_ntp_now - kDelay - kLastRR;
>@@ -818,7 +826,7 @@ TEST_F(RtcpReceiverTest, RttCalculatedAfterExtendedReportsDlrr) {
>   const uint32_t kDelayNtp = rand.Rand(0, 0x7fffffff);
>   const int64_t kDelayMs = CompactNtpRttToMs(kDelayNtp);
>   rtcp_receiver_.SetRtcpXrRrtrStatus(true);
>-  NtpTime now = system_clock_.CurrentNtpTime();
>+  NtpTime now = TimeMicrosToNtp(system_clock_.TimeInMicroseconds());
>   uint32_t sent_ntp = CompactNtp(now);
>   system_clock_.AdvanceTimeMilliseconds(kRttMs + kDelayMs);
> 
>@@ -838,7 +846,7 @@ TEST_F(RtcpReceiverTest, XrDlrrCalculatesNegativeRttAsOne) {
>   const int64_t kRttMs = rand.Rand(-3600 * 1000, -1);
>   const uint32_t kDelayNtp = rand.Rand(0, 0x7fffffff);
>   const int64_t kDelayMs = CompactNtpRttToMs(kDelayNtp);
>-  NtpTime now = system_clock_.CurrentNtpTime();
>+  NtpTime now = TimeMicrosToNtp(system_clock_.TimeInMicroseconds());
>   uint32_t sent_ntp = CompactNtp(now);
>   system_clock_.AdvanceTimeMilliseconds(kRttMs + kDelayMs);
>   rtcp_receiver_.SetRtcpXrRrtrStatus(true);
>@@ -936,13 +944,12 @@ TEST_F(RtcpReceiverTest, StoresLastReceivedRrtrPerSsrc) {
> }
> 
> TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
>-  const int64_t kRtcpIntervalMs = 1000;
>   const uint16_t kSequenceNumber = 1234;
>   system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs);
> 
>   // No RR received, shouldn't trigger a timeout.
>-  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout(kRtcpIntervalMs));
>-  EXPECT_FALSE(rtcp_receiver_.RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
>+  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout());
>+  EXPECT_FALSE(rtcp_receiver_.RtcpRrSequenceNumberTimeout());
> 
>   // Add a RR and advance the clock just enough to not trigger a timeout.
>   rtcp::ReportBlock rb1;
>@@ -957,8 +964,8 @@ TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
>   InjectRtcpPacket(rr1);
> 
>   system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs - 1);
>-  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout(kRtcpIntervalMs));
>-  EXPECT_FALSE(rtcp_receiver_.RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
>+  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout());
>+  EXPECT_FALSE(rtcp_receiver_.RtcpRrSequenceNumberTimeout());
> 
>   // Add a RR with the same extended max as the previous RR to trigger a
>   // sequence number timeout, but not a RR timeout.
>@@ -967,17 +974,17 @@ TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
>   InjectRtcpPacket(rr1);
> 
>   system_clock_.AdvanceTimeMilliseconds(2);
>-  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout(kRtcpIntervalMs));
>-  EXPECT_TRUE(rtcp_receiver_.RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
>+  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout());
>+  EXPECT_TRUE(rtcp_receiver_.RtcpRrSequenceNumberTimeout());
> 
>   // Advance clock enough to trigger an RR timeout too.
>   system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs);
>-  EXPECT_TRUE(rtcp_receiver_.RtcpRrTimeout(kRtcpIntervalMs));
>+  EXPECT_TRUE(rtcp_receiver_.RtcpRrTimeout());
> 
>   // We should only get one timeout even though we still haven't received a new
>   // RR.
>-  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout(kRtcpIntervalMs));
>-  EXPECT_FALSE(rtcp_receiver_.RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
>+  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout());
>+  EXPECT_FALSE(rtcp_receiver_.RtcpRrSequenceNumberTimeout());
> 
>   // Add a new RR with increase sequence number to reset timers.
>   rtcp::ReportBlock rb2;
>@@ -991,8 +998,8 @@ TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
>   EXPECT_CALL(bandwidth_observer_, OnReceivedRtcpReceiverReport(_, _, _));
>   InjectRtcpPacket(rr2);
> 
>-  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout(kRtcpIntervalMs));
>-  EXPECT_FALSE(rtcp_receiver_.RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
>+  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout());
>+  EXPECT_FALSE(rtcp_receiver_.RtcpRrSequenceNumberTimeout());
> 
>   // Verify we can get a timeout again once we've received new RR.
>   system_clock_.AdvanceTimeMilliseconds(2 * kRtcpIntervalMs);
>@@ -1001,11 +1008,11 @@ TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
>   InjectRtcpPacket(rr2);
> 
>   system_clock_.AdvanceTimeMilliseconds(kRtcpIntervalMs + 1);
>-  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout(kRtcpIntervalMs));
>-  EXPECT_TRUE(rtcp_receiver_.RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
>+  EXPECT_FALSE(rtcp_receiver_.RtcpRrTimeout());
>+  EXPECT_TRUE(rtcp_receiver_.RtcpRrSequenceNumberTimeout());
> 
>   system_clock_.AdvanceTimeMilliseconds(2 * kRtcpIntervalMs);
>-  EXPECT_TRUE(rtcp_receiver_.RtcpRrTimeout(kRtcpIntervalMs));
>+  EXPECT_TRUE(rtcp_receiver_.RtcpRrTimeout());
> }
> 
> TEST_F(RtcpReceiverTest, TmmbrReceivedWithNoIncomingPacket) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
>index bfddc422f0128be27fccd188690a18b2c0fd425c..2581487927bf84094ebfcf0de6deb5943292fa18 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
>@@ -10,7 +10,8 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_sender.h"
> 
>-#include <string.h>  // memcpy
>+#include <string.h>   // memcpy
>+#include <algorithm>  // std::min
> 
> #include <utility>
> 
>@@ -99,16 +100,16 @@ class RTCPSender::RtcpContext {
>   RtcpContext(const FeedbackState& feedback_state,
>               int32_t nack_size,
>               const uint16_t* nack_list,
>-              NtpTime now)
>+              int64_t now_us)
>       : feedback_state_(feedback_state),
>         nack_size_(nack_size),
>         nack_list_(nack_list),
>-        now_(now) {}
>+        now_us_(now_us) {}
> 
>   const FeedbackState& feedback_state_;
>   const int32_t nack_size_;
>   const uint16_t* nack_list_;
>-  const NtpTime now_;
>+  const int64_t now_us_;
> };
> 
> RTCPSender::RTCPSender(
>@@ -118,15 +119,14 @@ RTCPSender::RTCPSender(
>     RtcpPacketTypeCounterObserver* packet_type_counter_observer,
>     RtcEventLog* event_log,
>     Transport* outgoing_transport,
>-    RtcpIntervalConfig interval_config)
>+    int report_interval_ms)
>     : audio_(audio),
>       clock_(clock),
>       random_(clock_->TimeInMicroseconds()),
>       method_(RtcpMode::kOff),
>       event_log_(event_log),
>       transport_(outgoing_transport),
>-      interval_config_(interval_config),
>-      using_nack_(false),
>+      report_interval_ms_(report_interval_ms),
>       sending_(false),
>       next_time_to_send_rtcp_(0),
>       timestamp_offset_(0),
>@@ -151,7 +151,8 @@ RTCPSender::RTCPSender(
> 
>       xr_send_receiver_reference_time_enabled_(false),
>       packet_type_counter_observer_(packet_type_counter_observer),
>-      send_video_bitrate_allocation_(false) {
>+      send_video_bitrate_allocation_(false),
>+      last_payload_type_(-1) {
>   RTC_DCHECK(transport_ != nullptr);
> 
>   builders_[kRtcpSr] = &RTCPSender::BuildSR;
>@@ -180,9 +181,8 @@ void RTCPSender::SetRTCPStatus(RtcpMode new_method) {
> 
>   if (method_ == RtcpMode::kOff && new_method != RtcpMode::kOff) {
>     // When switching on, reschedule the next packet
>-    int64_t interval_ms = audio_ ? interval_config_.audio_interval_ms
>-                                 : interval_config_.video_interval_ms;
>-    next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + (interval_ms / 2);
>+    next_time_to_send_rtcp_ =
>+        clock_->TimeInMilliseconds() + (report_interval_ms_ / 2);
>   }
>   method_ = new_method;
> }
>@@ -254,8 +254,14 @@ void RTCPSender::SetTimestampOffset(uint32_t timestamp_offset) {
> }
> 
> void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
>-                                int64_t capture_time_ms) {
>+                                int64_t capture_time_ms,
>+                                int8_t payload_type) {
>   rtc::CritScope lock(&critical_section_rtcp_sender_);
>+  // For compatibility with clients who don't set payload type correctly on all
>+  // calls.
>+  if (payload_type != -1) {
>+    last_payload_type_ = payload_type;
>+  }
>   last_rtp_timestamp_ = rtp_timestamp;
>   if (capture_time_ms < 0) {
>     // We don't currently get a capture time from VoiceEngine.
>@@ -265,6 +271,11 @@ void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
>   }
> }
> 
>+void RTCPSender::SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz) {
>+  rtc::CritScope lock(&critical_section_rtcp_sender_);
>+  rtp_clock_rates_khz_[payload_type] = rtp_clock_rate_hz / 1000;
>+}
>+
> uint32_t RTCPSender::SSRC() const {
>   rtc::CritScope lock(&critical_section_rtcp_sender_);
>   return ssrc_;
>@@ -411,15 +422,21 @@ std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildSR(const RtcpContext& ctx) {
>   // the frame being captured at this moment. We are calculating that
>   // timestamp as the last frame's timestamp + the time since the last frame
>   // was captured.
>-  uint32_t rtp_rate =
>-      (audio_ ? kBogusRtpRateForAudioRtcp : kVideoPayloadTypeFrequency) / 1000;
>+  int rtp_rate = rtp_clock_rates_khz_[last_payload_type_];
>+  if (rtp_rate <= 0) {
>+    rtp_rate =
>+        (audio_ ? kBogusRtpRateForAudioRtcp : kVideoPayloadTypeFrequency) /
>+        1000;
>+  }
>+  // Round now_us_ to the closest millisecond, because Ntp time is rounded
>+  // when converted to milliseconds,
>   uint32_t rtp_timestamp =
>       timestamp_offset_ + last_rtp_timestamp_ +
>-      (clock_->TimeInMilliseconds() - last_frame_capture_time_ms_) * rtp_rate;
>+      ((ctx.now_us_ + 500) / 1000 - last_frame_capture_time_ms_) * rtp_rate;
> 
>   rtcp::SenderReport* report = new rtcp::SenderReport();
>   report->SetSenderSsrc(ssrc_);
>-  report->SetNtp(ctx.now_);
>+  report->SetNtp(TimeMicrosToNtp(ctx.now_us_));
>   report->SetRtpTimestamp(rtp_timestamp);
>   report->SetPacketCount(ctx.feedback_state_.packets_sent);
>   report->SetOctetCount(ctx.feedback_state_.media_bytes_sent);
>@@ -600,7 +617,7 @@ std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildExtendedReports(
> 
>   if (!sending_ && xr_send_receiver_reference_time_enabled_) {
>     rtcp::Rrtr rrtr;
>-    rrtr.SetNtp(ctx.now_);
>+    rrtr.SetNtp(TimeMicrosToNtp(ctx.now_us_));
>     xr->SetRrtr(rrtr);
>   }
> 
>@@ -675,7 +692,7 @@ int32_t RTCPSender::SendCompoundRTCP(
> 
>     // We need to send our NTP even if we haven't received any reports.
>     RtcpContext context(feedback_state, nack_size, nack_list,
>-                        clock_->CurrentNtpTime());
>+                        clock_->TimeInMicroseconds());
> 
>     PrepareReport(feedback_state);
> 
>@@ -748,28 +765,24 @@ void RTCPSender::PrepareReport(const FeedbackState& feedback_state) {
>     }
> 
>     // generate next time to send an RTCP report
>-    uint32_t minIntervalMs =
>-        rtc::dchecked_cast<uint32_t>(interval_config_.audio_interval_ms);
>-
>-    if (!audio_) {
>-      if (sending_) {
>-        // Calculate bandwidth for video; 360 / send bandwidth in kbit/s.
>-        uint32_t send_bitrate_kbit = feedback_state.send_bitrate / 1000;
>-        if (send_bitrate_kbit != 0)
>-          minIntervalMs = 360000 / send_bitrate_kbit;
>-      }
>-      if (minIntervalMs >
>-          rtc::dchecked_cast<uint32_t>(interval_config_.video_interval_ms)) {
>-        minIntervalMs =
>-            rtc::dchecked_cast<uint32_t>(interval_config_.video_interval_ms);
>+    int min_interval_ms = report_interval_ms_;
>+
>+    if (!audio_ && sending_) {
>+      // Calculate bandwidth for video; 360 / send bandwidth in kbit/s.
>+      int send_bitrate_kbit = feedback_state.send_bitrate / 1000;
>+      if (send_bitrate_kbit != 0) {
>+        min_interval_ms = 360000 / send_bitrate_kbit;
>+        min_interval_ms = std::min(min_interval_ms, report_interval_ms_);
>       }
>     }
> 
>     // The interval between RTCP packets is varied randomly over the
>     // range [1/2,3/2] times the calculated interval.
>-    uint32_t timeToNext =
>-        random_.Rand(minIntervalMs * 1 / 2, minIntervalMs * 3 / 2);
>-    next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + timeToNext;
>+    int time_to_next =
>+        random_.Rand(min_interval_ms * 1 / 2, min_interval_ms * 3 / 2);
>+
>+    RTC_DCHECK_GT(time_to_next, 0);
>+    next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + time_to_next;
> 
>     // RtcpSender expected to be used for sending either just sender reports
>     // or just receiver reports.
>@@ -791,7 +804,7 @@ std::vector<rtcp::ReportBlock> RTCPSender::CreateReportBlocks(
>   if (!result.empty() && ((feedback_state.last_rr_ntp_secs != 0) ||
>                           (feedback_state.last_rr_ntp_frac != 0))) {
>     // Get our NTP as late as possible to avoid a race.
>-    uint32_t now = CompactNtp(clock_->CurrentNtpTime());
>+    uint32_t now = CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds()));
> 
>     uint32_t receive_time = feedback_state.last_rr_ntp_secs & 0x0000FFFF;
>     receive_time <<= 16;
>@@ -949,12 +962,4 @@ bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
>   return packet.Build(max_packet_size, callback) && !send_failure;
> }
> 
>-int64_t RTCPSender::RtcpAudioReportInverval() const {
>-  return interval_config_.audio_interval_ms;
>-}
>-
>-int64_t RTCPSender::RtcpVideoReportInverval() const {
>-  return interval_config_.video_interval_ms;
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
>index c9220dd35c7a9dfa00ad2ac5d0971b5d335737f9..0845397da3bf6d224578f51b057accc813ac2495 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
>@@ -68,7 +68,7 @@ class RTCPSender {
>              RtcpPacketTypeCounterObserver* packet_type_counter_observer,
>              RtcEventLog* event_log,
>              Transport* outgoing_transport,
>-             RtcpIntervalConfig interval_config);
>+             int report_interval_ms);
>   virtual ~RTCPSender();
> 
>   RtcpMode Status() const;
>@@ -82,7 +82,13 @@ class RTCPSender {
> 
>   void SetTimestampOffset(uint32_t timestamp_offset);
> 
>-  void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms);
>+  // TODO(bugs.webrtc.org/6458): Remove default parameter value when all the
>+  // depending projects are updated to correctly set payload type.
>+  void SetLastRtpTime(uint32_t rtp_timestamp,
>+                      int64_t capture_time_ms,
>+                      int8_t payload_type = -1);
>+
>+  void SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz);
> 
>   uint32_t SSRC() const;
> 
>@@ -135,9 +141,6 @@ class RTCPSender {
>   void SetVideoBitrateAllocation(const VideoBitrateAllocation& bitrate);
>   bool SendFeedbackPacket(const rtcp::TransportFeedback& packet);
> 
>-  int64_t RtcpAudioReportInverval() const;
>-  int64_t RtcpVideoReportInverval() const;
>-
>  private:
>   class RtcpContext;
> 
>@@ -184,10 +187,9 @@ class RTCPSender {
>   RtcEventLog* const event_log_;
>   Transport* const transport_;
> 
>-  const RtcpIntervalConfig interval_config_;
>+  const int report_interval_ms_;
> 
>   rtc::CriticalSection critical_section_rtcp_sender_;
>-  bool using_nack_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
>   bool sending_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
> 
>   int64_t next_time_to_send_rtcp_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
>@@ -244,6 +246,11 @@ class RTCPSender {
>       RTC_GUARDED_BY(critical_section_rtcp_sender_);
>   bool send_video_bitrate_allocation_
>       RTC_GUARDED_BY(critical_section_rtcp_sender_);
>+
>+  std::map<int8_t, int> rtp_clock_rates_khz_
>+      RTC_GUARDED_BY(critical_section_rtcp_sender_);
>+  int8_t last_payload_type_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
>+
>   absl::optional<VideoBitrateAllocation> CheckAndUpdateLayerStructure(
>       const VideoBitrateAllocation& bitrate) const
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
>index bcffc81213d9f1249c833433883e655f359223be..e1a436b8452ff7291c94b8ff7180edb5f1c3b85c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
>@@ -11,11 +11,13 @@
> #include <memory>
> 
> #include "common_types.h"  // NOLINT(build/include)
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
> #include "modules/rtp_rtcp/source/rtcp_sender.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
>+#include "modules/rtp_rtcp/source/time_util.h"
> #include "rtc_base/rate_limiter.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>@@ -43,7 +45,7 @@ class RtcpPacketTypeCounterObserverImpl : public RtcpPacketTypeCounterObserver {
>   RtcpPacketTypeCounter counter_;
> };
> 
>-class TestTransport : public Transport, public RtpData {
>+class TestTransport : public Transport {
>  public:
>   TestTransport() {}
> 
>@@ -56,11 +58,6 @@ class TestTransport : public Transport, public RtpData {
>     parser_.Parse(data, len);
>     return true;
>   }
>-  int OnReceivedPayloadData(const uint8_t* payload_data,
>-                            size_t payload_size,
>-                            const WebRtcRTPHeader* rtp_header) override {
>-    return 0;
>-  }
>   test::RtcpPacketParser parser_;
> };
> 
>@@ -82,15 +79,17 @@ class RtcpSenderTest : public ::testing::Test {
>     configuration.clock = &clock_;
>     configuration.outgoing_transport = &test_transport_;
>     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
>+    configuration.rtcp_report_interval_ms = 1000;
> 
>     rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
>     rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
>                                       nullptr, nullptr, &test_transport_,
>-                                      configuration.rtcp_interval_config));
>+                                      configuration.rtcp_report_interval_ms));
>     rtcp_sender_->SetSSRC(kSenderSsrc);
>     rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
>     rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
>-    rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds());
>+    rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds(),
>+                                 /*paylpad_type=*/0);
>   }
> 
>   void InsertIncomingPacket(uint32_t remote_ssrc, uint16_t seq_num) {
>@@ -141,7 +140,7 @@ TEST_F(RtcpSenderTest, SendSr) {
>   rtcp_sender_->SetSendingStatus(feedback_state, true);
>   feedback_state.packets_sent = kPacketCount;
>   feedback_state.media_bytes_sent = kOctetCount;
>-  NtpTime ntp = clock_.CurrentNtpTime();
>+  NtpTime ntp = TimeMicrosToNtp(clock_.TimeInMicroseconds());
>   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpSr));
>   EXPECT_EQ(1, parser()->sender_report()->num_packets());
>   EXPECT_EQ(kSenderSsrc, parser()->sender_report()->sender_ssrc());
>@@ -153,10 +152,42 @@ TEST_F(RtcpSenderTest, SendSr) {
>   EXPECT_EQ(0U, parser()->sender_report()->report_blocks().size());
> }
> 
>+TEST_F(RtcpSenderTest, SendConsecutiveSrWithExactSlope) {
>+  const uint32_t kPacketCount = 0x12345;
>+  const uint32_t kOctetCount = 0x23456;
>+  const int kTimeBetweenSRsUs = 10043;  // Not exact value in milliseconds.
>+  const int kExtraPackets = 30;
>+  // Make sure clock is not exactly at some milliseconds point.
>+  clock_.AdvanceTimeMicroseconds(kTimeBetweenSRsUs);
>+  rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
>+  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
>+  rtcp_sender_->SetSendingStatus(feedback_state, true);
>+  feedback_state.packets_sent = kPacketCount;
>+  feedback_state.media_bytes_sent = kOctetCount;
>+
>+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpSr));
>+  EXPECT_EQ(1, parser()->sender_report()->num_packets());
>+  NtpTime ntp1 = parser()->sender_report()->ntp();
>+  uint32_t rtp1 = parser()->sender_report()->rtp_timestamp();
>+
>+  // Send more SRs to ensure slope is always exact for different offsets
>+  for (int packets = 1; packets <= kExtraPackets; ++packets) {
>+    clock_.AdvanceTimeMicroseconds(kTimeBetweenSRsUs);
>+    EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpSr));
>+    EXPECT_EQ(packets + 1, parser()->sender_report()->num_packets());
>+
>+    NtpTime ntp2 = parser()->sender_report()->ntp();
>+    uint32_t rtp2 = parser()->sender_report()->rtp_timestamp();
>+
>+    uint32_t ntp_diff_in_rtp_units =
>+        (ntp2.ToMs() - ntp1.ToMs()) * (kVideoPayloadTypeFrequency / 1000);
>+    EXPECT_EQ(rtp2 - rtp1, ntp_diff_in_rtp_units);
>+  }
>+}
>+
> TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
>   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
>-                                    nullptr, nullptr, &test_transport_,
>-                                    RtcpIntervalConfig{}));
>+                                    nullptr, nullptr, &test_transport_, 1000));
>   rtcp_sender_->SetSSRC(kSenderSsrc);
>   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
>   rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
>@@ -174,8 +205,7 @@ TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
> 
> TEST_F(RtcpSenderTest, DoNotSendCompundBeforeRtp) {
>   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
>-                                    nullptr, nullptr, &test_transport_,
>-                                    RtcpIntervalConfig{}));
>+                                    nullptr, nullptr, &test_transport_, 1000));
>   rtcp_sender_->SetSSRC(kSenderSsrc);
>   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
>   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
>@@ -447,7 +477,7 @@ TEST_F(RtcpSenderTest, SendXrWithRrtr) {
>   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
>   EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false));
>   rtcp_sender_->SendRtcpXrReceiverReferenceTime(true);
>-  NtpTime ntp = clock_.CurrentNtpTime();
>+  NtpTime ntp = TimeMicrosToNtp(clock_.TimeInMicroseconds());
>   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
>   EXPECT_EQ(1, parser()->xr()->num_packets());
>   EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
>@@ -476,7 +506,7 @@ TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
>   RtcpPacketTypeCounterObserverImpl observer;
>   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
>                                     &observer, nullptr, &test_transport_,
>-                                    RtcpIntervalConfig{}));
>+                                    1000));
>   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
>   rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
>   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
>@@ -598,12 +628,12 @@ TEST_F(RtcpSenderTest, ByeMustBeLast) {
> 
>   // Re-configure rtcp_sender_ with mock_transport_
>   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
>-                                    nullptr, nullptr, &mock_transport,
>-                                    RtcpIntervalConfig{}));
>+                                    nullptr, nullptr, &mock_transport, 1000));
>   rtcp_sender_->SetSSRC(kSenderSsrc);
>   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
>   rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
>-  rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds());
>+  rtcp_sender_->SetLastRtpTime(kRtpTimestamp, clock_.TimeInMilliseconds(),
>+                               /*paylpad_type=*/0);
> 
>   // Set up REMB info to be included with BYE.
>   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.cc
>index 77b10ca052d9864993fda1beae4ade0c71b22346..57d2142ffad4fff1e172955fc6f31d6ce89de9fd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.cc
>@@ -39,10 +39,10 @@ RtcpTransceiver::~RtcpTransceiver() {
>   RTC_DCHECK(!rtcp_transceiver_);
> }
> 
>-void RtcpTransceiver::Stop(std::unique_ptr<rtc::QueuedTask> on_destroyed) {
>+void RtcpTransceiver::Stop(std::function<void()> on_destroyed) {
>   RTC_DCHECK(rtcp_transceiver_);
>-  task_queue_->PostTaskAndReply(Destructor{std::move(rtcp_transceiver_)},
>-                                std::move(on_destroyed));
>+  task_queue_->PostTask(rtc::NewClosure(
>+      Destructor{std::move(rtcp_transceiver_)}, std::move(on_destroyed)));
>   RTC_DCHECK(!rtcp_transceiver_);
> }
> 
>@@ -59,13 +59,14 @@ void RtcpTransceiver::AddMediaReceiverRtcpObserver(
> void RtcpTransceiver::RemoveMediaReceiverRtcpObserver(
>     uint32_t remote_ssrc,
>     MediaReceiverRtcpObserver* observer,
>-    std::unique_ptr<rtc::QueuedTask> on_removed) {
>+    std::function<void()> on_removed) {
>   RTC_CHECK(rtcp_transceiver_);
>   RtcpTransceiverImpl* ptr = rtcp_transceiver_.get();
>   auto remove = [ptr, remote_ssrc, observer] {
>     ptr->RemoveMediaReceiverRtcpObserver(remote_ssrc, observer);
>   };
>-  task_queue_->PostTaskAndReply(std::move(remove), std::move(on_removed));
>+  task_queue_->PostTask(
>+      rtc::NewClosure(std::move(remove), std::move(on_removed)));
> }
> 
> void RtcpTransceiver::SetReadyToSend(bool ready) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.h
>index fc9488c4abd640c507f5fce439f37b8b3e55bbea..9c9675110fa60e395f51250341f1b0e0c90c98b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_H_
> 
>+#include <functional>
> #include <memory>
> #include <string>
> #include <vector>
>@@ -42,18 +43,17 @@ class RtcpTransceiver : public RtcpFeedbackSenderInterface {
>   // Note that interfaces provided in constructor or registered with AddObserver
>   // still might be used by the transceiver on the task queue
>   // until |on_destroyed| runs.
>-  void Stop(std::unique_ptr<rtc::QueuedTask> on_destroyed);
>+  void Stop(std::function<void()> on_destroyed);
> 
>   // Registers observer to be notified about incoming rtcp packets.
>   // Calls to observer will be done on the |config.task_queue|.
>   void AddMediaReceiverRtcpObserver(uint32_t remote_ssrc,
>                                     MediaReceiverRtcpObserver* observer);
>   // Deregisters the observer. Might return before observer is deregistered.
>-  // Posts |on_removed| task when observer is deregistered.
>-  void RemoveMediaReceiverRtcpObserver(
>-      uint32_t remote_ssrc,
>-      MediaReceiverRtcpObserver* observer,
>-      std::unique_ptr<rtc::QueuedTask> on_removed);
>+  // Runs |on_removed| when observer is deregistered.
>+  void RemoveMediaReceiverRtcpObserver(uint32_t remote_ssrc,
>+                                       MediaReceiverRtcpObserver* observer,
>+                                       std::function<void()> on_removed);
> 
>   // Enables/disables sending rtcp packets eventually.
>   // Packets may be sent after the SetReadyToSend(false) returns, but no new
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h
>index 0da18e2cc1b128a3fd3bf2703255a8bff0c8ffdf..01330d0bc7dfb0aff1649760fafb5e7f1f6ff2d8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h
>@@ -97,11 +97,6 @@ struct RtcpTransceiverConfig {
>   // Estimate RTT as non-sender as described in
>   // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5
>   bool non_sender_rtt_measurement = false;
>-  // Copies LastSR/DelaySinceLastSR for previous report block to avoid
>-  // triggering bug in older version of RtcpReceiver.
>-  // TODO(bugs.webrtc.org/8805): Change to false by default then remove when
>-  // all major webrtc clients updated with the fix in RtcpReceiver.
>-  bool avoid_zero_last_sr_in_last_report_block = true;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
>index 2c6c3acc391365ca58f66e3dc7e7569630274cc9..97c2ac018b31d0a56f222a0d08442e5f3cb7ad9e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
>@@ -421,12 +421,6 @@ std::vector<rtcp::ReportBlock> RtcpTransceiverImpl::CreateReportBlocks(
>     auto it = remote_senders_.find(report_block.source_ssrc());
>     if (it == remote_senders_.end() ||
>         !it->second.last_received_sender_report) {
>-      if (config_.avoid_zero_last_sr_in_last_report_block && last_sr != 0) {
>-        // Simulate behaviour of the RtcpSender to avoid hitting bug in
>-        // RtcpReceiver.
>-        report_block.SetLastSr(last_sr);
>-        report_block.SetDelayLastSr(last_delay);
>-      }
>       continue;
>     }
>     const SenderReportTimes& last_sender_report =
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
>index e5db0862cc476109bcc93b2abb4b1df6eee765dc..e86d67f5790c7f2ccea2db271ffde79c15ef9ce6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
>@@ -77,7 +77,6 @@ constexpr int kAlmostForeverMs = 1000;
> // Helper to wait for an rtcp packet produced on a different thread/task queue.
> class FakeRtcpTransport : public webrtc::Transport {
>  public:
>-  FakeRtcpTransport() : sent_rtcp_(false, false) {}
>   bool SendRtcp(const uint8_t* data, size_t size) override {
>     sent_rtcp_.Set();
>     return true;
>@@ -148,7 +147,7 @@ TEST(RtcpTransceiverImplTest, CanDestroyOnTaskQueue) {
>   // Wait for a periodic packet.
>   EXPECT_TRUE(transport.WaitPacket());
> 
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   queue.PostTask([rtcp_transceiver, &done] {
>     delete rtcp_transceiver;
>     done.Set();
>@@ -187,7 +186,7 @@ TEST(RtcpTransceiverImplTest, DelaysSendingFirstCompondPacket) {
>   EXPECT_GE(rtc::TimeMillis() - started_ms, config.initial_report_delay_ms);
> 
>   // Cleanup.
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   queue.PostTask([&] {
>     rtcp_transceiver.reset();
>     done.Set();
>@@ -220,7 +219,7 @@ TEST(RtcpTransceiverImplTest, PeriodicallySendsPackets) {
>             config.report_period_ms - 1);
> 
>   // Cleanup.
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   queue.PostTask([&] {
>     rtcp_transceiver.reset();
>     done.Set();
>@@ -242,7 +241,7 @@ TEST(RtcpTransceiverImplTest, SendCompoundPacketDelaysPeriodicSendPackets) {
>   // Wait for first packet.
>   EXPECT_TRUE(transport.WaitPacket());
>   // Send non periodic one after half period.
>-  rtc::Event non_periodic(false, false);
>+  rtc::Event non_periodic;
>   int64_t time_of_non_periodic_packet_ms = 0;
>   queue.PostDelayedTask(
>       [&] {
>@@ -265,7 +264,7 @@ TEST(RtcpTransceiverImplTest, SendCompoundPacketDelaysPeriodicSendPackets) {
>             config.report_period_ms - 1);
> 
>   // Cleanup.
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   queue.PostTask([&] {
>     rtcp_transceiver.reset();
>     done.Set();
>@@ -329,7 +328,7 @@ TEST(RtcpTransceiverImplTest, SendsPeriodicRtcpWhenNetworkStateIsUp) {
>   EXPECT_TRUE(transport.WaitPacket());
> 
>   // Cleanup.
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   queue.PostTask([&] {
>     rtcp_transceiver.reset();
>     done.Set();
>@@ -671,7 +670,6 @@ TEST(RtcpTransceiverImplTest,
> 
>   RtcpTransceiverConfig config;
>   config.schedule_periodic_compound_packets = false;
>-  config.avoid_zero_last_sr_in_last_report_block = false;
>   RtcpPacketParser rtcp_parser;
>   RtcpParserTransport transport(&rtcp_parser);
>   config.outgoing_transport = &transport;
>@@ -703,52 +701,6 @@ TEST(RtcpTransceiverImplTest,
>   EXPECT_EQ(report_blocks[1].last_sr(), 0u);
> }
> 
>-TEST(RtcpTransceiverImplTest, AvoidLastReportBlockToHaveZeroLastSrField) {
>-  const uint32_t kRemoteSsrc1 = 54321;
>-  const uint32_t kRemoteSsrc2 = 54323;
>-  MockReceiveStatisticsProvider receive_statistics;
>-  std::vector<ReportBlock> statistics_report_blocks(2);
>-  statistics_report_blocks[0].SetMediaSsrc(kRemoteSsrc1);
>-  statistics_report_blocks[1].SetMediaSsrc(kRemoteSsrc2);
>-  ON_CALL(receive_statistics, RtcpReportBlocks(_))
>-      .WillByDefault(Return(statistics_report_blocks));
>-
>-  RtcpTransceiverConfig config;
>-  config.schedule_periodic_compound_packets = false;
>-  config.avoid_zero_last_sr_in_last_report_block = true;
>-  RtcpPacketParser rtcp_parser;
>-  RtcpParserTransport transport(&rtcp_parser);
>-  config.outgoing_transport = &transport;
>-  config.receive_statistics = &receive_statistics;
>-  RtcpTransceiverImpl rtcp_transceiver(config);
>-
>-  const NtpTime kRemoteNtp(0x9876543211);
>-  // Receive SenderReport for RemoteSsrc1, but no report for RemoteSsrc2.
>-  SenderReport sr;
>-  sr.SetSenderSsrc(kRemoteSsrc1);
>-  sr.SetNtp(kRemoteNtp);
>-  auto raw_packet = sr.Build();
>-  rtcp_transceiver.ReceivePacket(raw_packet, /*now_us=*/0);
>-
>-  // Trigger sending ReceiverReport.
>-  rtcp_transceiver.SendCompoundPacket();
>-
>-  EXPECT_GT(rtcp_parser.receiver_report()->num_packets(), 0);
>-  const auto& report_blocks = rtcp_parser.receiver_report()->report_blocks();
>-  ASSERT_EQ(report_blocks.size(), 2u);
>-  // RtcpTransceiverImpl doesn't guarantee order of the report blocks
>-  // match result of ReceiveStatisticsProvider::RtcpReportBlocks callback,
>-  // but for simplicity of the test asume it is the same.
>-  ASSERT_EQ(report_blocks[0].source_ssrc(), kRemoteSsrc1);
>-  EXPECT_NE(report_blocks[0].last_sr(), 0u);
>-
>-  ASSERT_EQ(report_blocks[1].source_ssrc(), kRemoteSsrc2);
>-  // No Sender Report for kRemoteSsrc2, use same LastSR as for kRemoteSsrc1
>-  EXPECT_EQ(report_blocks[1].last_sr(), report_blocks[0].last_sr());
>-  EXPECT_EQ(report_blocks[1].delay_since_last_sr(),
>-            report_blocks[0].delay_since_last_sr());
>-}
>-
> TEST(RtcpTransceiverImplTest,
>      WhenSendsReceiverReportCalculatesDelaySinceLastSenderReport) {
>   const uint32_t kRemoteSsrc1 = 4321;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc
>index 2ea5bc9b5377d212c87e3190ba5073429536d70b..d71918db03b4f7a56c338c76b5d48ac0e12aabf2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc
>@@ -41,7 +41,7 @@ class MockMediaReceiverRtcpObserver : public webrtc::MediaReceiverRtcpObserver {
> constexpr int kTimeoutMs = 1000;
> 
> void WaitPostedTasks(rtc::TaskQueue* queue) {
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   queue->PostTask([&done] { done.Set(); });
>   ASSERT_TRUE(done.Wait(kTimeoutMs));
> }
>@@ -108,8 +108,8 @@ TEST(RtcpTransceiverTest, CanBeDestroyedWithoutBlocking) {
>   auto* rtcp_transceiver = new RtcpTransceiver(config);
>   rtcp_transceiver->SendCompoundPacket();
> 
>-  rtc::Event done(false, false);
>-  rtc::Event heavy_task(false, false);
>+  rtc::Event done;
>+  rtc::Event heavy_task;
>   queue.PostTask([&] {
>     EXPECT_TRUE(heavy_task.Wait(kTimeoutMs));
>     done.Set();
>@@ -128,7 +128,7 @@ TEST(RtcpTransceiverTest, MaySendPacketsAfterDestructor) {  // i.e. Be careful!
>   config.task_queue = &queue;
>   auto* rtcp_transceiver = new RtcpTransceiver(config);
> 
>-  rtc::Event heavy_task(false, false);
>+  rtc::Event heavy_task;
>   queue.PostTask([&] { EXPECT_TRUE(heavy_task.Wait(kTimeoutMs)); });
>   rtcp_transceiver->SendCompoundPacket();
>   delete rtcp_transceiver;
>@@ -158,7 +158,7 @@ TEST(RtcpTransceiverTest, DoesntPostToRtcpObserverAfterCallToRemove) {
>   config.outgoing_transport = &null_transport;
>   config.task_queue = &queue;
>   RtcpTransceiver rtcp_transceiver(config);
>-  rtc::Event observer_deleted(false, false);
>+  rtc::Event observer_deleted;
> 
>   auto observer = absl::make_unique<MockMediaReceiverRtcpObserver>();
>   EXPECT_CALL(*observer, OnSenderReport(kRemoteSsrc, _, 1));
>@@ -166,12 +166,11 @@ TEST(RtcpTransceiverTest, DoesntPostToRtcpObserverAfterCallToRemove) {
> 
>   rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, observer.get());
>   rtcp_transceiver.ReceivePacket(CreateSenderReport(kRemoteSsrc, 1));
>-  rtcp_transceiver.RemoveMediaReceiverRtcpObserver(
>-      kRemoteSsrc, observer.get(),
>-      /*on_removed=*/rtc::NewClosure([&] {
>-        observer.reset();
>-        observer_deleted.Set();
>-      }));
>+  rtcp_transceiver.RemoveMediaReceiverRtcpObserver(kRemoteSsrc, observer.get(),
>+                                                   /*on_removed=*/[&] {
>+                                                     observer.reset();
>+                                                     observer_deleted.Set();
>+                                                   });
>   rtcp_transceiver.ReceivePacket(CreateSenderReport(kRemoteSsrc, 2));
> 
>   EXPECT_TRUE(observer_deleted.Wait(kTimeoutMs));
>@@ -189,15 +188,14 @@ TEST(RtcpTransceiverTest, RemoveMediaReceiverRtcpObserverIsNonBlocking) {
>   auto observer = absl::make_unique<MockMediaReceiverRtcpObserver>();
>   rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, observer.get());
> 
>-  rtc::Event queue_blocker(false, false);
>-  rtc::Event observer_deleted(false, false);
>+  rtc::Event queue_blocker;
>+  rtc::Event observer_deleted;
>   queue.PostTask([&] { EXPECT_TRUE(queue_blocker.Wait(kTimeoutMs)); });
>-  rtcp_transceiver.RemoveMediaReceiverRtcpObserver(
>-      kRemoteSsrc, observer.get(),
>-      /*on_removed=*/rtc::NewClosure([&] {
>-        observer.reset();
>-        observer_deleted.Set();
>-      }));
>+  rtcp_transceiver.RemoveMediaReceiverRtcpObserver(kRemoteSsrc, observer.get(),
>+                                                   /*on_removed=*/[&] {
>+                                                     observer.reset();
>+                                                     observer_deleted.Set();
>+                                                   });
> 
>   EXPECT_THAT(observer, Not(IsNull()));
>   queue_blocker.Set();
>@@ -242,12 +240,12 @@ TEST(RtcpTransceiverTest, DoesntSendPacketsAfterStopCallback) {
>   config.schedule_periodic_compound_packets = true;
> 
>   auto rtcp_transceiver = absl::make_unique<RtcpTransceiver>(config);
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   rtcp_transceiver->SendCompoundPacket();
>-  rtcp_transceiver->Stop(rtc::NewClosure([&] {
>+  rtcp_transceiver->Stop([&] {
>     EXPECT_CALL(outgoing_transport, SendRtcp).Times(0);
>     done.Set();
>-  }));
>+  });
>   rtcp_transceiver = nullptr;
>   EXPECT_TRUE(done.Wait(kTimeoutMs));
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc
>index 13ec0afe9f7bd675ac9841be41cb998b9ef151b0..0010d90750f29efd001a8a5d2b81136befe9c396 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc
>@@ -10,13 +10,15 @@
> 
> #include "modules/rtp_rtcp/source/rtp_format.h"
> 
>-#include <utility>
>-
> #include "absl/memory/memory.h"
>+#include "absl/types/variant.h"
> #include "modules/rtp_rtcp/source/rtp_format_h264.h"
> #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
> #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
> #include "modules/rtp_rtcp/source/rtp_format_vp9.h"
>+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
>+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>@@ -63,6 +65,11 @@ std::vector<int> RtpPacketizer::SplitAboutEqually(
>   RTC_DCHECK_GE(limits.last_packet_reduction_len, 0);
> 
>   std::vector<int> result;
>+  if (limits.max_payload_len >=
>+      limits.single_packet_reduction_len + payload_len) {
>+    result.push_back(payload_len);
>+    return result;
>+  }
>   if (limits.max_payload_len - limits.first_packet_reduction_len < 1 ||
>       limits.max_payload_len - limits.last_packet_reduction_len < 1) {
>     // Capacity is not enough to put a single byte into one of the packets.
>@@ -77,6 +84,10 @@ std::vector<int> RtpPacketizer::SplitAboutEqually(
>   // Integer divisions with rounding up.
>   int num_packets_left =
>       (total_bytes + limits.max_payload_len - 1) / limits.max_payload_len;
>+  if (num_packets_left == 1) {
>+    // Single packet is a special case handled above.
>+    num_packets_left = 2;
>+  }
> 
>   if (payload_len < num_packets_left) {
>     // Edge case where limits force to have more packets than there are payload
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.h
>index 9fad4cf8f9761cacbf3c87ca4051d5b4be621e4d..71c7dc5e0ecd8d4fa80a079b191a78f423cfb421 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.h
>@@ -11,17 +11,17 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
> 
>+#include <stdint.h>
> #include <memory>
>-#include <string>
> #include <vector>
> 
> #include "api/array_view.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/include/module_common_types.h"
>-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "rtc_base/constructormagic.h"
>+#include "modules/rtp_rtcp/source/rtp_video_header.h"
> 
> namespace webrtc {
>+
> class RtpPacketToSend;
> 
> class RtpPacketizer {
>@@ -30,6 +30,8 @@ class RtpPacketizer {
>     int max_payload_len = 1200;
>     int first_packet_reduction_len = 0;
>     int last_packet_reduction_len = 0;
>+    // Reduction len for packet that is first & last at the same time.
>+    int single_packet_reduction_len = 0;
>   };
>   static std::unique_ptr<RtpPacketizer> Create(
>       VideoCodecType type,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
>index 9793ebae259131ddfec81e3176e2788fa248e07d..7a7bcdfbb89c874891fa2ce22022ec5c9e71af8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
>@@ -11,10 +11,16 @@
> #include "modules/rtp_rtcp/source/rtp_format_h264.h"
> 
> #include <string.h>
>+#include <cstddef>
>+#include <cstdint>
>+#include <iterator>
> #include <memory>
> #include <utility>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "absl/types/variant.h"
>+#include "common_types.h"  // NOLINT(build/include)
> #include "common_video/h264/h264_common.h"
> #include "common_video/h264/pps_parser.h"
> #include "common_video/h264/sps_parser.h"
>@@ -186,9 +192,11 @@ bool RtpPacketizerH264::GeneratePackets(
>       case H264PacketizationMode::NonInterleaved:
>         int fragment_len = input_fragments_[i].length;
>         int single_packet_capacity = limits_.max_payload_len;
>-        if (i == 0)
>+        if (input_fragments_.size() == 1)
>+          single_packet_capacity -= limits_.single_packet_reduction_len;
>+        else if (i == 0)
>           single_packet_capacity -= limits_.first_packet_reduction_len;
>-        if (i + 1 == input_fragments_.size())
>+        else if (i + 1 == input_fragments_.size())
>           single_packet_capacity -= limits_.last_packet_reduction_len;
> 
>         if (fragment_len > single_packet_capacity) {
>@@ -211,7 +219,10 @@ bool RtpPacketizerH264::PacketizeFuA(size_t fragment_index) {
>   PayloadSizeLimits limits = limits_;
>   // Leave room for the FU-A header.
>   limits.max_payload_len -= kFuAHeaderSize;
>-  // Ignore first/last packet reductions unless it is first/last fragment.
>+  // Ignore single/first/last packet reductions unless it is single/first/last
>+  // fragment.
>+  if (input_fragments_.size() != 1)
>+    limits.single_packet_reduction_len = 0;
>   if (fragment_index != 0)
>     limits.first_packet_reduction_len = 0;
>   if (fragment_index != input_fragments_.size() - 1)
>@@ -243,17 +254,31 @@ bool RtpPacketizerH264::PacketizeFuA(size_t fragment_index) {
> size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) {
>   // Aggregate fragments into one packet (STAP-A).
>   size_t payload_size_left = limits_.max_payload_len;
>-  if (fragment_index == 0)
>+  if (input_fragments_.size() == 1)
>+    payload_size_left -= limits_.single_packet_reduction_len;
>+  else if (fragment_index == 0)
>     payload_size_left -= limits_.first_packet_reduction_len;
>   int aggregated_fragments = 0;
>   size_t fragment_headers_length = 0;
>   const Fragment* fragment = &input_fragments_[fragment_index];
>   RTC_CHECK_GE(payload_size_left, fragment->length);
>   ++num_packets_left_;
>-  while (payload_size_left >= fragment->length + fragment_headers_length &&
>-         (fragment_index + 1 < input_fragments_.size() ||
>-          payload_size_left >= fragment->length + fragment_headers_length +
>-                                   limits_.last_packet_reduction_len)) {
>+
>+  auto payload_size_needed = [&] {
>+    size_t fragment_size = fragment->length + fragment_headers_length;
>+    if (input_fragments_.size() == 1) {
>+      // Single fragment, single packet, payload_size_left already adjusted
>+      // with limits_.single_packet_reduction_len.
>+      return fragment_size;
>+    }
>+    if (fragment_index == input_fragments_.size() - 1) {
>+      // Last fragment, so StrapA might be the last packet.
>+      return fragment_size + limits_.last_packet_reduction_len;
>+    }
>+    return fragment_size;
>+  };
>+
>+  while (payload_size_left >= payload_size_needed()) {
>     RTC_CHECK_GT(fragment->length, 0);
>     packets_.push(PacketUnit(*fragment, aggregated_fragments == 0, false, true,
>                              fragment->buffer[0]));
>@@ -282,9 +307,11 @@ size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) {
> bool RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) {
>   // Add a single NALU to the queue, no aggregation.
>   size_t payload_size_left = limits_.max_payload_len;
>-  if (fragment_index == 0)
>+  if (input_fragments_.size() == 1)
>+    payload_size_left -= limits_.single_packet_reduction_len;
>+  else if (fragment_index == 0)
>     payload_size_left -= limits_.first_packet_reduction_len;
>-  if (fragment_index + 1 == input_fragments_.size())
>+  else if (fragment_index + 1 == input_fragments_.size())
>     payload_size_left -= limits_.last_packet_reduction_len;
>   const Fragment* fragment = &input_fragments_[fragment_index];
>   if (payload_size_left < fragment->length) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
>index 73e40878a18bd9edd1d5016f0bf6b9303b0ceaa1..fbd4fd9509075d8fb852e33d4526bb32e338d394 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
>@@ -11,12 +11,17 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <deque>
> #include <memory>
> #include <queue>
>-#include <string>
> 
>+#include "api/array_view.h"
>+#include "modules/include/module_common_types.h"
> #include "modules/rtp_rtcp/source/rtp_format.h"
>+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
>index edf907aac859cf47ef10f06a9be0af86cc9d9ff8..aeab813b53e2651c4deb02094474654acfb54ba4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
>@@ -390,7 +390,7 @@ std::vector<int> TestFua(size_t frame_payload_size,
>                                NoFragmentation(frame));
>   std::vector<RtpPacketToSend> packets = FetchAllPackets(&packetizer);
> 
>-  RTC_CHECK_GE(packets.size(), 2);  // Single packet indicates it is not FuA.
>+  EXPECT_GE(packets.size(), 2u);  // Single packet indicates it is not FuA.
>   std::vector<uint16_t> fua_header;
>   std::vector<int> payload_sizes;
> 
>@@ -422,6 +422,7 @@ TEST(RtpPacketizerH264Test, FUAWithFirstPacketReduction) {
>   RtpPacketizer::PayloadSizeLimits limits;
>   limits.max_payload_len = 1200;
>   limits.first_packet_reduction_len = 4;
>+  limits.single_packet_reduction_len = 4;
>   EXPECT_THAT(TestFua(1198, limits), ElementsAre(597, 601));
> }
> 
>@@ -429,14 +430,14 @@ TEST(RtpPacketizerH264Test, FUAWithLastPacketReduction) {
>   RtpPacketizer::PayloadSizeLimits limits;
>   limits.max_payload_len = 1200;
>   limits.last_packet_reduction_len = 4;
>+  limits.single_packet_reduction_len = 4;
>   EXPECT_THAT(TestFua(1198, limits), ElementsAre(601, 597));
> }
> 
>-TEST(RtpPacketizerH264Test, FUAWithFirstAndLastPacketReduction) {
>+TEST(RtpPacketizerH264Test, FUAWithSinglePacketReduction) {
>   RtpPacketizer::PayloadSizeLimits limits;
>   limits.max_payload_len = 1199;
>-  limits.first_packet_reduction_len = 100;
>-  limits.last_packet_reduction_len = 100;
>+  limits.single_packet_reduction_len = 200;
>   EXPECT_THAT(TestFua(1000, limits), ElementsAre(500, 500));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_unittest.cc
>index a79c434b629f325b1ef6424a265be3630c79183c..ae1b5b054adfa3e4e2150eab41871eb50198f3bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_unittest.cc
>@@ -199,21 +199,32 @@ TEST(RtpPacketizerSplitAboutEqually, GivesNonZeroPayloadLengthEachPacket) {
> }
> 
> TEST(RtpPacketizerSplitAboutEqually,
>-     OnePacketWhenExtraSpaceIsEnoughForSumOfFirstAndLastPacketReductions) {
>+     IgnoresFirstAndLastPacketReductionWhenPayloadFitsIntoSinglePacket) {
>   RtpPacketizer::PayloadSizeLimits limits;
>   limits.max_payload_len = 30;
>-  limits.first_packet_reduction_len = 6;
>-  limits.last_packet_reduction_len = 4;
>+  limits.first_packet_reduction_len = 29;
>+  limits.last_packet_reduction_len = 29;
>+  limits.single_packet_reduction_len = 10;
> 
>   EXPECT_THAT(RtpPacketizer::SplitAboutEqually(20, limits), ElementsAre(20));
> }
> 
> TEST(RtpPacketizerSplitAboutEqually,
>-     TwoPacketsWhenExtraSpaceIsTooSmallForSumOfFirstAndLastPacketReductions) {
>+     OnePacketWhenExtraSpaceIsEnoughForSinglePacketReduction) {
>+  RtpPacketizer::PayloadSizeLimits limits;
>+  limits.max_payload_len = 30;
>+  limits.single_packet_reduction_len = 10;
>+
>+  EXPECT_THAT(RtpPacketizer::SplitAboutEqually(20, limits), ElementsAre(20));
>+}
>+
>+TEST(RtpPacketizerSplitAboutEqually,
>+     TwoPacketsWhenExtraSpaceIsTooSmallForSinglePacketReduction) {
>   RtpPacketizer::PayloadSizeLimits limits;
>   limits.max_payload_len = 29;
>-  limits.first_packet_reduction_len = 6;
>-  limits.last_packet_reduction_len = 4;
>+  limits.first_packet_reduction_len = 3;
>+  limits.last_packet_reduction_len = 1;
>+  limits.single_packet_reduction_len = 10;
> 
>   // First packet needs two more extra bytes compared to last one,
>   // so should have two less payload bytes.
>@@ -246,8 +257,7 @@ TEST(RtpPacketizerSplitAboutEqually, RejectsZeroLastPacketLen) {
> TEST(RtpPacketizerSplitAboutEqually, CantPutSinglePayloadByteInTwoPackets) {
>   RtpPacketizer::PayloadSizeLimits limits;
>   limits.max_payload_len = 10;
>-  limits.first_packet_reduction_len = 6;
>-  limits.last_packet_reduction_len = 4;
>+  limits.single_packet_reduction_len = 10;
> 
>   EXPECT_THAT(RtpPacketizer::SplitAboutEqually(1, limits), IsEmpty());
> }
>@@ -255,8 +265,7 @@ TEST(RtpPacketizerSplitAboutEqually, CantPutSinglePayloadByteInTwoPackets) {
> TEST(RtpPacketizerSplitAboutEqually, CanPutTwoPayloadBytesInTwoPackets) {
>   RtpPacketizer::PayloadSizeLimits limits;
>   limits.max_payload_len = 10;
>-  limits.first_packet_reduction_len = 6;
>-  limits.last_packet_reduction_len = 4;
>+  limits.single_packet_reduction_len = 10;
> 
>   EXPECT_THAT(RtpPacketizer::SplitAboutEqually(2, limits), ElementsAre(1, 1));
> }
>@@ -264,8 +273,7 @@ TEST(RtpPacketizerSplitAboutEqually, CanPutTwoPayloadBytesInTwoPackets) {
> TEST(RtpPacketizerSplitAboutEqually, CanPutSinglePayloadByteInOnePacket) {
>   RtpPacketizer::PayloadSizeLimits limits;
>   limits.max_payload_len = 11;
>-  limits.first_packet_reduction_len = 6;
>-  limits.last_packet_reduction_len = 4;
>+  limits.single_packet_reduction_len = 10;
> 
>   EXPECT_THAT(RtpPacketizer::SplitAboutEqually(1, limits), ElementsAre(1));
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
>index edd1e3c53051635435b8c0e11a85113cacb38425..92aada4688a85fa5cbcc758720a785691d509868 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
>@@ -8,11 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <string>
>+#include <assert.h>
>+#include <string.h>
> 
>-#include "modules/include/module_common_types.h"
>+#include "absl/types/optional.h"
> #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
>index 03509f5041df6f7cf6d48ae788b889565e32dcb0..3458d4942329a15c120dcff59e30ff7cb126db11 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
>@@ -10,6 +10,7 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
> 
>+#include <stdint.h>
> #include <vector>
> 
> #include "api/array_view.h"
>@@ -18,6 +19,10 @@
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>+
>+class RtpPacketToSend;
>+struct RTPVideoHeader;
>+
> namespace RtpFormatVideoGeneric {
> static const uint8_t kKeyFrameBit = 0x01;
> static const uint8_t kFirstPacketBit = 0x02;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
>index f40434e7e031f729f0d43122855c95bc923dc200..a1248cf040b173c5c07ad4b74fdddc1794258dad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
>@@ -10,13 +10,13 @@
> 
> #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
> 
>+#include <stdint.h>
> #include <string.h>  // memcpy
>-
>-#include <limits>
>-#include <utility>
> #include <vector>
> 
>+#include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>+#include "modules/video_coding/codecs/interface/common_constants.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
>index e4bc36e3f3f4ee25f636c9c1ee785ca7b847b636..444298fbbf809d555bbcd408f62d5c8e674b56c5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
>@@ -25,13 +25,15 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
> 
>-#include <string>
>+#include <stddef.h>
>+#include <cstdint>
> #include <vector>
> 
> #include "absl/container/inlined_vector.h"
> #include "api/array_view.h"
>-#include "modules/include/module_common_types.h"
> #include "modules/rtp_rtcp/source/rtp_format.h"
>+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
>index 4419d1a30c0b5f983b542428ab498e3c6ad80e17..9cd75143b8c2971af32aab540a48fc643d54ee74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
>@@ -12,9 +12,9 @@
> 
> #include <string.h>
> 
>-#include <cmath>
>-
>+#include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>+#include "modules/video_coding/codecs/interface/common_constants.h"
> #include "rtc_base/bitbuffer.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>@@ -451,6 +451,7 @@ RtpPacketizerVp9::RtpPacketizerVp9(rtc::ArrayView<const uint8_t> payload,
>       remaining_payload_(payload) {
>   limits.max_payload_len -= header_size_;
>   limits.first_packet_reduction_len += first_packet_extra_header_size_;
>+  limits.single_packet_reduction_len += first_packet_extra_header_size_;
> 
>   payload_sizes_ = SplitAboutEqually(payload.size(), limits);
>   current_packet_ = payload_sizes_.begin();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
>index 8a2a2a604c814d32255589343419efdb926e9ea9..c3b8f17a4b0981bac1481c964e5338474e4ae693 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
>@@ -21,12 +21,14 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <vector>
> 
> #include "api/array_view.h"
>-#include "modules/include/module_common_types.h"
> #include "modules/rtp_rtcp/source/rtp_format.h"
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
>index 1b931327bad1f8a791297420b635763836ce1d94..c27fb6e3f6f85fb91da9409e86326ed8c8333dde 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
> 
>+#include <cstdint>
>+
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>@@ -56,6 +58,16 @@ void RtpGenericFrameDescriptor::SetSpatialLayersBitmask(
>   spatial_layers_ = spatial_layers;
> }
> 
>+void RtpGenericFrameDescriptor::SetResolution(int width, int height) {
>+  RTC_DCHECK(FirstPacketInSubFrame());
>+  RTC_DCHECK_GE(width, 0);
>+  RTC_DCHECK_LE(width, 0xFFFF);
>+  RTC_DCHECK_GE(height, 0);
>+  RTC_DCHECK_LE(height, 0xFFFF);
>+  width_ = width;
>+  height_ = height;
>+}
>+
> uint16_t RtpGenericFrameDescriptor::FrameId() const {
>   RTC_DCHECK(FirstPacketInSubFrame());
>   return frame_id_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
>index 1bde4fa56dcc789f8f6cc3b6c7eba1f861e429e3..3a6c34d59ff2630fa1436de57f4fce6968a1049c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
>@@ -50,6 +50,10 @@ class RtpGenericFrameDescriptor {
>   uint8_t SpatialLayersBitmask() const;
>   void SetSpatialLayersBitmask(uint8_t spatial_layers);
> 
>+  int Width() const { return width_; }
>+  int Height() const { return height_; }
>+  void SetResolution(int width, int height);
>+
>   uint16_t FrameId() const;
>   void SetFrameId(uint16_t frame_id);
> 
>@@ -72,6 +76,9 @@ class RtpGenericFrameDescriptor {
>   uint8_t temporal_layer_ = 0;
>   size_t num_frame_deps_ = 0;
>   uint16_t frame_deps_id_diffs_[kMaxNumFrameDependencies];
>+  int width_ = 0;
>+  int height_ = 0;
>+
>   std::vector<uint8_t> byte_representation_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
>index c7b52d5ba680d14fd86562fa4a85bb10f090278c..7cd120d9f45c395706e7519350d9591f82c32293 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
>@@ -36,6 +36,14 @@ constexpr uint8_t kFlageXtendedOffset = 0x02;
> // B:   +      FID      +
> //      |               |
> //      +-+-+-+-+-+-+-+-+
>+//      |               |
>+//      +     Width     +
>+// B=1  |               |
>+// and  +-+-+-+-+-+-+-+-+
>+// D=0  |               |
>+//      +     Height    +
>+//      |               |
>+//      +-+-+-+-+-+-+-+-+
> // D:   |    FDIFF  |X|M|
> //      +---------------+
> // X:   |      ...      |
>@@ -75,6 +83,12 @@ bool RtpGenericFrameDescriptorExtension::Parse(
>   descriptor->ClearFrameDependencies();
>   size_t offset = 4;
>   bool has_more_dependencies = (data[0] & kFlagDependencies) != 0;
>+  if (!has_more_dependencies && data.size() >= offset + 4) {
>+    uint16_t width = (data[offset] << 8) | data[offset + 1];
>+    uint16_t height = (data[offset + 2] << 8) | data[offset + 3];
>+    descriptor->SetResolution(width, height);
>+    offset += 4;
>+  }
>   while (has_more_dependencies) {
>     if (data.size() == offset)
>       return false;
>@@ -91,7 +105,7 @@ bool RtpGenericFrameDescriptorExtension::Parse(
>     if (!descriptor->AddFrameDependencyDiff(fdiff))
>       return false;
>   }
>-  return data.size() == offset;
>+  return true;
> }
> 
> size_t RtpGenericFrameDescriptorExtension::ValueSize(
>@@ -103,6 +117,11 @@ size_t RtpGenericFrameDescriptorExtension::ValueSize(
>   for (uint16_t fdiff : descriptor.FrameDependenciesDiffs()) {
>     size += (fdiff >= (1 << 6)) ? 2 : 1;
>   }
>+  if (descriptor.FirstPacketInSubFrame() &&
>+      descriptor.FrameDependenciesDiffs().empty() && descriptor.Width() > 0 &&
>+      descriptor.Height() > 0) {
>+    size += 4;
>+  }
>   return size;
> }
> 
>@@ -129,6 +148,13 @@ bool RtpGenericFrameDescriptorExtension::Write(
>   data[3] = frame_id >> 8;
>   rtc::ArrayView<const uint16_t> fdiffs = descriptor.FrameDependenciesDiffs();
>   size_t offset = 4;
>+  if (descriptor.FirstPacketInSubFrame() && fdiffs.empty() &&
>+      descriptor.Width() > 0 && descriptor.Height() > 0) {
>+    data[offset++] = (descriptor.Width() >> 8);
>+    data[offset++] = (descriptor.Width() & 0xFF);
>+    data[offset++] = (descriptor.Height() >> 8);
>+    data[offset++] = (descriptor.Height() & 0xFF);
>+  }
>   for (size_t i = 0; i < fdiffs.size(); i++) {
>     bool extended = fdiffs[i] >= (1 << 6);
>     bool more = i < fdiffs.size() - 1;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
>index 2dd9dd49ebe12e02a9c05fe26c3bdb431e182ee0..0d673e060198d81dc0faf378cfe8e853bb280a27 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
>@@ -21,6 +21,7 @@ namespace webrtc {
> 
> class RtpGenericFrameDescriptorExtension {
>  public:
>+  using value_type = RtpGenericFrameDescriptor;
>   static constexpr RTPExtensionType kId = kRtpExtensionGenericFrameDescriptor;
>   static constexpr char kUri[] =
>       "http://www.webrtc.org/experiments/rtp-hdrext/"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension_unittest.cc
>index bb69e1455dd940caf57bb15280bc3195f5ce7473..7f8fa2f5a41e447fcd935c59c4e55046f8e724f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension_unittest.cc
>@@ -279,5 +279,32 @@ TEST(RtpGenericFrameDescriptorExtensionTest, WriteTwoFrameDependencies) {
>   EXPECT_THAT(buffer, ElementsAreArray(kRaw));
> }
> 
>+TEST(RtpGenericFrameDescriptorExtensionTest,
>+     ParseResolutionOnIndependentFrame) {
>+  constexpr int kWidth = 0x2468;
>+  constexpr int kHeight = 0x6543;
>+  constexpr uint8_t kRaw[] = {0x80, 0x01, 0x00, 0x00, 0x24, 0x68, 0x65, 0x43};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension::Parse(kRaw, &descriptor));
>+  EXPECT_EQ(descriptor.Width(), kWidth);
>+  EXPECT_EQ(descriptor.Height(), kHeight);
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest,
>+     WriteResolutionOnIndependentFrame) {
>+  constexpr int kWidth = 0x2468;
>+  constexpr int kHeight = 0x6543;
>+  constexpr uint8_t kRaw[] = {0x80, 0x01, 0x00, 0x00, 0x24, 0x68, 0x65, 0x43};
>+  RtpGenericFrameDescriptor descriptor;
>+  descriptor.SetFirstPacketInSubFrame(true);
>+  descriptor.SetResolution(kWidth, kHeight);
>+
>+  ASSERT_EQ(RtpGenericFrameDescriptorExtension::ValueSize(descriptor),
>+            sizeof(kRaw));
>+  uint8_t buffer[sizeof(kRaw)];
>+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension::Write(buffer, descriptor));
>+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
>+}
> }  // namespace
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
>index 168665afc4532deb4d9e8635a2d7e0ab320842ab..8e0a484d9774e606ead845cda46e4087266cc72f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
>@@ -43,6 +43,7 @@ constexpr ExtensionInfo kExtensions[] = {
>     CreateExtensionInfo<RepairedRtpStreamId>(),
>     CreateExtensionInfo<RtpMid>(),
>     CreateExtensionInfo<RtpGenericFrameDescriptorExtension>(),
>+    CreateExtensionInfo<ColorSpaceExtension>(),
> };
> 
> // Because of kRtpExtensionNone, NumberOfExtension is 1 bigger than the actual
>@@ -56,15 +57,17 @@ static_assert(arraysize(kExtensions) ==
> constexpr RTPExtensionType RtpHeaderExtensionMap::kInvalidType;
> constexpr int RtpHeaderExtensionMap::kInvalidId;
> 
>-RtpHeaderExtensionMap::RtpHeaderExtensionMap()
>-    : mixed_one_two_byte_header_supported_(false) {
>+RtpHeaderExtensionMap::RtpHeaderExtensionMap() : RtpHeaderExtensionMap(false) {}
>+
>+RtpHeaderExtensionMap::RtpHeaderExtensionMap(bool extmap_allow_mixed)
>+    : extmap_allow_mixed_(extmap_allow_mixed) {
>   for (auto& id : ids_)
>     id = kInvalidId;
> }
> 
> RtpHeaderExtensionMap::RtpHeaderExtensionMap(
>     rtc::ArrayView<const RtpExtension> extensions)
>-    : RtpHeaderExtensionMap() {
>+    : RtpHeaderExtensionMap(false) {
>   for (const RtpExtension& extension : extensions)
>     RegisterByUri(extension.id, extension.uri);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
>index fc4b7ceb9281af5c122e5b49bda50d54ccafe759..92694cd5a3a466a328d81c4377f04561b28b2080 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
>@@ -10,10 +10,14 @@
> 
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> 
>+#include <string.h>
>+#include <cmath>
>+
> #include "modules/rtp_rtcp/include/rtp_cvo.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
>+// TODO(bug:9855) Move kNoSpatialIdx from vp9_globals.h to common_constants
>+#include "modules/video_coding/codecs/interface/common_constants.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
> 
> namespace webrtc {
> // Absolute send time in RTP streams.
>@@ -430,6 +434,172 @@ bool FrameMarkingExtension::Write(rtc::ArrayView<uint8_t> data,
>   return true;
> }
> 
>+// Color space including HDR metadata as an optional field.
>+//
>+// RTP header extension to carry HDR metadata.
>+// Float values are upscaled by a static factor and transmitted as integers.
>+//
>+// Data layout with HDR metadata
>+//    0                   1                   2                   3
>+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+//   |       ID      |   length=30    |   Primaries   |    Transfer    |
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+//   |     Matrix    |      Range     |                 luminance_max  |
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+//   |               |                  luminance_min                  |
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+//   |              mastering_metadata.primary_r.x and .y              |
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+//   |              mastering_metadata.primary_g.x and .y              |
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+//   |              mastering_metadata.primary_b.x and .y              |
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+//   |                mastering_metadata.white.x and .y                |
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+//   |     max_content_light_level    | max_frame_average_light_level  |
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+//
>+// Data layout without HDR metadata
>+//    0                   1                   2                   3
>+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+//   |       ID      |    length=4    |   Primaries   |    Transfer    |
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+//   |     Matrix    |      Range     |
>+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
>+
>+constexpr RTPExtensionType ColorSpaceExtension::kId;
>+constexpr uint8_t ColorSpaceExtension::kValueSizeBytes;
>+constexpr const char ColorSpaceExtension::kUri[];
>+
>+bool ColorSpaceExtension::Parse(rtc::ArrayView<const uint8_t> data,
>+                                ColorSpace* color_space) {
>+  RTC_DCHECK(color_space);
>+  if (data.size() != kValueSizeBytes &&
>+      data.size() != kValueSizeBytesWithoutHdrMetadata)
>+    return false;
>+
>+  size_t offset = 0;
>+  // Read color space information.
>+  if (!color_space->set_primaries_from_uint8(data.data()[offset++]))
>+    return false;
>+  if (!color_space->set_transfer_from_uint8(data.data()[offset++]))
>+    return false;
>+  if (!color_space->set_matrix_from_uint8(data.data()[offset++]))
>+    return false;
>+  if (!color_space->set_range_from_uint8(data.data()[offset++]))
>+    return false;
>+
>+  // Read HDR metadata if it exists, otherwise clear it.
>+  if (data.size() == kValueSizeBytesWithoutHdrMetadata) {
>+    color_space->set_hdr_metadata(nullptr);
>+  } else {
>+    HdrMetadata hdr_metadata;
>+    offset += ParseLuminance(data.data() + offset,
>+                             &hdr_metadata.mastering_metadata.luminance_max,
>+                             kLuminanceMaxDenominator);
>+    offset += ParseLuminance(data.data() + offset,
>+                             &hdr_metadata.mastering_metadata.luminance_min,
>+                             kLuminanceMinDenominator);
>+    offset += ParseChromaticity(data.data() + offset,
>+                                &hdr_metadata.mastering_metadata.primary_r);
>+    offset += ParseChromaticity(data.data() + offset,
>+                                &hdr_metadata.mastering_metadata.primary_g);
>+    offset += ParseChromaticity(data.data() + offset,
>+                                &hdr_metadata.mastering_metadata.primary_b);
>+    offset += ParseChromaticity(data.data() + offset,
>+                                &hdr_metadata.mastering_metadata.white_point);
>+    hdr_metadata.max_content_light_level =
>+        ByteReader<uint16_t>::ReadBigEndian(data.data() + offset);
>+    offset += 2;
>+    hdr_metadata.max_frame_average_light_level =
>+        ByteReader<uint16_t>::ReadBigEndian(data.data() + offset);
>+    offset += 2;
>+    color_space->set_hdr_metadata(&hdr_metadata);
>+  }
>+  RTC_DCHECK_EQ(ValueSize(*color_space), offset);
>+  return true;
>+}
>+
>+bool ColorSpaceExtension::Write(rtc::ArrayView<uint8_t> data,
>+                                const ColorSpace& color_space) {
>+  RTC_DCHECK(data.size() >= ValueSize(color_space));
>+  size_t offset = 0;
>+  // Write color space information.
>+  data.data()[offset++] = static_cast<uint8_t>(color_space.primaries());
>+  data.data()[offset++] = static_cast<uint8_t>(color_space.transfer());
>+  data.data()[offset++] = static_cast<uint8_t>(color_space.matrix());
>+  data.data()[offset++] = static_cast<uint8_t>(color_space.range());
>+
>+  // Write HDR metadata if it exists.
>+  if (color_space.hdr_metadata()) {
>+    const HdrMetadata& hdr_metadata = *color_space.hdr_metadata();
>+    offset += WriteLuminance(data.data() + offset,
>+                             hdr_metadata.mastering_metadata.luminance_max,
>+                             kLuminanceMaxDenominator);
>+    offset += WriteLuminance(data.data() + offset,
>+                             hdr_metadata.mastering_metadata.luminance_min,
>+                             kLuminanceMinDenominator);
>+    offset += WriteChromaticity(data.data() + offset,
>+                                hdr_metadata.mastering_metadata.primary_r);
>+    offset += WriteChromaticity(data.data() + offset,
>+                                hdr_metadata.mastering_metadata.primary_g);
>+    offset += WriteChromaticity(data.data() + offset,
>+                                hdr_metadata.mastering_metadata.primary_b);
>+    offset += WriteChromaticity(data.data() + offset,
>+                                hdr_metadata.mastering_metadata.white_point);
>+
>+    ByteWriter<uint16_t>::WriteBigEndian(data.data() + offset,
>+                                         hdr_metadata.max_content_light_level);
>+    offset += 2;
>+    ByteWriter<uint16_t>::WriteBigEndian(
>+        data.data() + offset, hdr_metadata.max_frame_average_light_level);
>+    offset += 2;
>+  }
>+  RTC_DCHECK_EQ(ValueSize(color_space), offset);
>+  return true;
>+}
>+
>+size_t ColorSpaceExtension::ParseChromaticity(
>+    const uint8_t* data,
>+    HdrMasteringMetadata::Chromaticity* p) {
>+  uint16_t chromaticity_x_scaled = ByteReader<uint16_t>::ReadBigEndian(data);
>+  uint16_t chromaticity_y_scaled =
>+      ByteReader<uint16_t>::ReadBigEndian(data + 2);
>+  p->x = static_cast<float>(chromaticity_x_scaled) / kChromaticityDenominator;
>+  p->y = static_cast<float>(chromaticity_y_scaled) / kChromaticityDenominator;
>+  return 4;  // Return number of bytes read.
>+}
>+
>+size_t ColorSpaceExtension::ParseLuminance(const uint8_t* data,
>+                                           float* f,
>+                                           int denominator) {
>+  uint32_t luminance_scaled = ByteReader<uint32_t, 3>::ReadBigEndian(data);
>+  *f = static_cast<float>(luminance_scaled) / denominator;
>+  return 3;  // Return number of bytes read.
>+}
>+
>+size_t ColorSpaceExtension::WriteChromaticity(
>+    uint8_t* data,
>+    const HdrMasteringMetadata::Chromaticity& p) {
>+  RTC_DCHECK_GE(p.x, 0.0f);
>+  RTC_DCHECK_GE(p.y, 0.0f);
>+  ByteWriter<uint16_t>::WriteBigEndian(
>+      data, std::round(p.x * kChromaticityDenominator));
>+  ByteWriter<uint16_t>::WriteBigEndian(
>+      data + 2, std::round(p.y * kChromaticityDenominator));
>+  return 4;  // Return number of bytes written.
>+}
>+
>+size_t ColorSpaceExtension::WriteLuminance(uint8_t* data,
>+                                           float f,
>+                                           int denominator) {
>+  RTC_DCHECK_GE(f, 0.0f);
>+  ByteWriter<uint32_t, 3>::WriteBigEndian(data, std::round(f * denominator));
>+  return 3;  // Return number of bytes written.
>+}
>+
> bool BaseRtpStringExtension::Parse(rtc::ArrayView<const uint8_t> data,
>                                    StringRtpHeaderExtension* str) {
>   if (data.empty() || data[0] == 0)  // Valid string extension can't be empty.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
>index 84e9831479db3e94e995b6fb65fa403a3fcce1b0..42a6216c7b529892f5bd58d27cd940c895012dda 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
>@@ -10,19 +10,25 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
> 
>+#include <stddef.h>
> #include <stdint.h>
> #include <string>
> 
> #include "api/array_view.h"
>+#include "api/rtp_headers.h"
>+#include "api/video/color_space.h"
> #include "api/video/video_content_type.h"
>+#include "api/video/video_frame_marking.h"
> #include "api/video/video_rotation.h"
> #include "api/video/video_timing.h"
>+#include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> 
> namespace webrtc {
> 
> class AbsoluteSendTime {
>  public:
>+  using value_type = uint32_t;
>   static constexpr RTPExtensionType kId = kRtpExtensionAbsoluteSendTime;
>   static constexpr uint8_t kValueSizeBytes = 3;
>   static constexpr const char kUri[] =
>@@ -57,6 +63,7 @@ class AudioLevel {
> 
> class TransmissionOffset {
>  public:
>+  using value_type = int32_t;
>   static constexpr RTPExtensionType kId = kRtpExtensionTransmissionTimeOffset;
>   static constexpr uint8_t kValueSizeBytes = 3;
>   static constexpr const char kUri[] = "urn:ietf:params:rtp-hdrext:toffset";
>@@ -68,6 +75,7 @@ class TransmissionOffset {
> 
> class TransportSequenceNumber {
>  public:
>+  using value_type = uint16_t;
>   static constexpr RTPExtensionType kId = kRtpExtensionTransportSequenceNumber;
>   static constexpr uint8_t kValueSizeBytes = 2;
>   static constexpr const char kUri[] =
>@@ -80,6 +88,7 @@ class TransportSequenceNumber {
> 
> class VideoOrientation {
>  public:
>+  using value_type = VideoRotation;
>   static constexpr RTPExtensionType kId = kRtpExtensionVideoRotation;
>   static constexpr uint8_t kValueSizeBytes = 1;
>   static constexpr const char kUri[] = "urn:3gpp:video-orientation";
>@@ -94,6 +103,7 @@ class VideoOrientation {
> 
> class PlayoutDelayLimits {
>  public:
>+  using value_type = PlayoutDelay;
>   static constexpr RTPExtensionType kId = kRtpExtensionPlayoutDelay;
>   static constexpr uint8_t kValueSizeBytes = 3;
>   static constexpr const char kUri[] =
>@@ -117,6 +127,7 @@ class PlayoutDelayLimits {
> 
> class VideoContentTypeExtension {
>  public:
>+  using value_type = VideoContentType;
>   static constexpr RTPExtensionType kId = kRtpExtensionVideoContentType;
>   static constexpr uint8_t kValueSizeBytes = 1;
>   static constexpr const char kUri[] =
>@@ -133,6 +144,7 @@ class VideoContentTypeExtension {
> 
> class VideoTimingExtension {
>  public:
>+  using value_type = VideoSendTiming;
>   static constexpr RTPExtensionType kId = kRtpExtensionVideoTiming;
>   static constexpr uint8_t kValueSizeBytes = 13;
>   static constexpr const char kUri[] =
>@@ -155,6 +167,7 @@ class VideoTimingExtension {
> 
> class FrameMarkingExtension {
>  public:
>+  using value_type = FrameMarking;
>   static constexpr RTPExtensionType kId = kRtpExtensionFrameMarking;
>   static constexpr const char kUri[] =
>       "http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07";
>@@ -169,10 +182,41 @@ class FrameMarkingExtension {
>   static bool IsScalable(uint8_t temporal_id, uint8_t layer_id);
> };
> 
>+class ColorSpaceExtension {
>+ public:
>+  using value_type = ColorSpace;
>+  static constexpr RTPExtensionType kId = kRtpExtensionColorSpace;
>+  static constexpr uint8_t kValueSizeBytes = 30;
>+  static constexpr uint8_t kValueSizeBytesWithoutHdrMetadata = 4;
>+  // TODO(webrtc:8651): Change to a valid uri.
>+  static constexpr const char kUri[] = "rtp-colorspace-uri-placeholder";
>+
>+  static bool Parse(rtc::ArrayView<const uint8_t> data,
>+                    ColorSpace* color_space);
>+  static size_t ValueSize(const ColorSpace& color_space) {
>+    return color_space.hdr_metadata() ? kValueSizeBytes
>+                                      : kValueSizeBytesWithoutHdrMetadata;
>+  }
>+  static bool Write(rtc::ArrayView<uint8_t> data,
>+                    const ColorSpace& color_space);
>+
>+ private:
>+  static constexpr int kChromaticityDenominator = 10000;  // 0.0001 resolution.
>+  static constexpr int kLuminanceMaxDenominator = 100;    // 0.01 resolution.
>+  static constexpr int kLuminanceMinDenominator = 10000;  // 0.0001 resolution.
>+  static size_t ParseChromaticity(const uint8_t* data,
>+                                  HdrMasteringMetadata::Chromaticity* p);
>+  static size_t ParseLuminance(const uint8_t* data, float* f, int denominator);
>+  static size_t WriteChromaticity(uint8_t* data,
>+                                  const HdrMasteringMetadata::Chromaticity& p);
>+  static size_t WriteLuminance(uint8_t* data, float f, int denominator);
>+};
>+
> // Base extension class for RTP header extensions which are strings.
> // Subclasses must defined kId and kUri static constexpr members.
> class BaseRtpStringExtension {
>  public:
>+  using value_type = std::string;
>   // String RTP header extensions are limited to 16 bytes because it is the
>   // maximum length that can be encoded with one-byte header extensions.
>   static constexpr uint8_t kMaxValueSizeBytes = 16;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
>index df68f74dd9377222319d6fe0a30fc1aa8327ccbf..6481a403a0436a3d3e46f45cc77b2e2cb14a12b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
>@@ -9,9 +9,12 @@
>  */
> #include "modules/rtp_rtcp/include/rtp_header_parser.h"
> 
>+#include <string.h>
>+
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
> 
>@@ -25,8 +28,10 @@ class RtpHeaderParserImpl : public RtpHeaderParser {
>              RTPHeader* header) const override;
> 
>   bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id) override;
>+  bool RegisterRtpHeaderExtension(RtpExtension extension) override;
> 
>   bool DeregisterRtpHeaderExtension(RTPExtensionType type) override;
>+  bool DeregisterRtpHeaderExtension(RtpExtension extension) override;
> 
>  private:
>   rtc::CriticalSection critical_section_;
>@@ -63,6 +68,10 @@ bool RtpHeaderParserImpl::Parse(const uint8_t* packet,
>   }
>   return true;
> }
>+bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RtpExtension extension) {
>+  rtc::CritScope cs(&critical_section_);
>+  return rtp_header_extension_map_.RegisterByUri(extension.id, extension.uri);
>+}
> 
> bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
>                                                      uint8_t id) {
>@@ -70,6 +79,12 @@ bool RtpHeaderParserImpl::RegisterRtpHeaderExtension(RTPExtensionType type,
>   return rtp_header_extension_map_.RegisterByType(id, type);
> }
> 
>+bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RtpExtension extension) {
>+  rtc::CritScope cs(&critical_section_);
>+  return rtp_header_extension_map_.Deregister(
>+      rtp_header_extension_map_.GetType(extension.id));
>+}
>+
> bool RtpHeaderParserImpl::DeregisterRtpHeaderExtension(RTPExtensionType type) {
>   rtc::CritScope cs(&critical_section_);
>   return rtp_header_extension_map_.Deregister(type) == 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
>index 0b9ed80af56ed867a54c9580fcbcc6344b3ecdc8..9d4dce4dea27e4db04951ffc4f14d7f5b004fb41 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
>@@ -10,16 +10,15 @@
> 
> #include "modules/rtp_rtcp/source/rtp_packet.h"
> 
>+#include <cstdint>
> #include <cstring>
> #include <utility>
> 
> #include "api/rtpparameters.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/random.h"
> 
> namespace webrtc {
> namespace {
>@@ -212,8 +211,7 @@ rtc::ArrayView<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
>   const bool two_byte_header_required =
>       id > RtpExtension::kOneByteHeaderExtensionMaxId ||
>       length > RtpExtension::kOneByteHeaderExtensionMaxValueSize || length == 0;
>-  RTC_CHECK(!two_byte_header_required ||
>-            extensions_.IsMixedOneTwoByteHeaderSupported());
>+  RTC_CHECK(!two_byte_header_required || extensions_.ExtmapAllowMixed());
> 
>   uint16_t profile_id;
>   if (extensions_size_ > 0) {
>@@ -359,22 +357,20 @@ uint8_t* RtpPacket::SetPayloadSize(size_t size_bytes) {
>   return WriteAt(payload_offset_);
> }
> 
>-bool RtpPacket::SetPadding(uint8_t size_bytes, Random* random) {
>-  RTC_DCHECK(random);
>-  if (payload_offset_ + payload_size_ + size_bytes > capacity()) {
>-    RTC_LOG(LS_WARNING) << "Cannot set padding size " << size_bytes << ", only "
>+bool RtpPacket::SetPadding(size_t padding_bytes) {
>+  if (payload_offset_ + payload_size_ + padding_bytes > capacity()) {
>+    RTC_LOG(LS_WARNING) << "Cannot set padding size " << padding_bytes
>+                        << ", only "
>                         << (capacity() - payload_offset_ - payload_size_)
>                         << " bytes left in buffer.";
>     return false;
>   }
>-  padding_size_ = size_bytes;
>+  padding_size_ = rtc::dchecked_cast<uint8_t>(padding_bytes);
>   buffer_.SetSize(payload_offset_ + payload_size_ + padding_size_);
>   if (padding_size_ > 0) {
>     size_t padding_offset = payload_offset_ + payload_size_;
>     size_t padding_end = padding_offset + padding_size_;
>-    for (size_t offset = padding_offset; offset < padding_end - 1; ++offset) {
>-      WriteAt(offset, random->Rand<uint8_t>());
>-    }
>+    memset(WriteAt(padding_offset), 0, padding_size_ - 1);
>     WriteAt(padding_end - 1, padding_size_);
>     WriteAt(0, data()[0] | 0x20);  // Set padding bit.
>   } else {
>@@ -556,7 +552,7 @@ rtc::ArrayView<uint8_t> RtpPacket::AllocateExtension(ExtensionType type,
>                                                      size_t length) {
>   // TODO(webrtc:7990): Add support for empty extensions (length==0).
>   if (length == 0 || length > RtpExtension::kMaxValueSize ||
>-      (!extensions_.IsMixedOneTwoByteHeaderSupported() &&
>+      (!extensions_.ExtmapAllowMixed() &&
>        length > RtpExtension::kOneByteHeaderExtensionMaxValueSize)) {
>     return nullptr;
>   }
>@@ -566,7 +562,7 @@ rtc::ArrayView<uint8_t> RtpPacket::AllocateExtension(ExtensionType type,
>     // Extension not registered.
>     return nullptr;
>   }
>-  if (!extensions_.IsMixedOneTwoByteHeaderSupported() &&
>+  if (!extensions_.ExtmapAllowMixed() &&
>       id > RtpExtension::kOneByteHeaderExtensionMaxId) {
>     return nullptr;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.h
>index 15c3865b3594dc876e41806a4783f1746253dd74..76666b7f12d7637519f052339309123ce215e7b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.h
>@@ -12,10 +12,12 @@
> 
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/copyonwritebuffer.h"
>+#include "rtc_base/deprecation.h"
> 
> namespace webrtc {
> class Random;
>@@ -94,8 +96,11 @@ class RtpPacket {
>   template <typename Extension>
>   bool HasExtension() const;
> 
>-  template <typename Extension, typename... Values>
>-  bool GetExtension(Values...) const;
>+  template <typename Extension, typename FirstValue, typename... Values>
>+  bool GetExtension(FirstValue, Values...) const;
>+
>+  template <typename Extension>
>+  absl::optional<typename Extension::value_type> GetExtension() const;
> 
>   // Returns view of the raw extension or empty view on failure.
>   template <typename Extension>
>@@ -111,7 +116,12 @@ class RtpPacket {
>   uint8_t* SetPayloadSize(size_t size_bytes);
>   // Same as SetPayloadSize but doesn't guarantee to keep current payload.
>   uint8_t* AllocatePayload(size_t size_bytes);
>-  bool SetPadding(uint8_t size_bytes, Random* random);
>+  RTC_DEPRECATED
>+  bool SetPadding(uint8_t size_bytes, Random* random) {
>+    return SetPadding(size_bytes);
>+  }
>+
>+  bool SetPadding(size_t padding_size);
> 
>  private:
>   struct ExtensionInfo {
>@@ -177,12 +187,21 @@ bool RtpPacket::HasExtension() const {
>   return !FindExtension(Extension::kId).empty();
> }
> 
>-template <typename Extension, typename... Values>
>-bool RtpPacket::GetExtension(Values... values) const {
>+template <typename Extension, typename FirstValue, typename... Values>
>+bool RtpPacket::GetExtension(FirstValue first, Values... values) const {
>   auto raw = FindExtension(Extension::kId);
>   if (raw.empty())
>     return false;
>-  return Extension::Parse(raw, values...);
>+  return Extension::Parse(raw, first, values...);
>+}
>+
>+template <typename Extension>
>+absl::optional<typename Extension::value_type> RtpPacket::GetExtension() const {
>+  absl::optional<typename Extension::value_type> result;
>+  auto raw = FindExtension(Extension::kId);
>+  if (raw.empty() || !Extension::Parse(raw, &result.emplace()))
>+    result = absl::nullopt;
>+  return result;
> }
> 
> template <typename Extension>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
>index e1374feb57fcd9961215128e111f2337ec68fae0..8c1c8ebbbfae91cab6ea04c528a36d9c9974b9ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
>@@ -97,6 +97,15 @@ void RtpPacketHistory::PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
>   const uint16_t rtp_seq_no = packet->SequenceNumber();
>   StoredPacket& stored_packet = packet_history_[rtp_seq_no];
>   RTC_DCHECK(stored_packet.packet == nullptr);
>+  if (stored_packet.packet) {
>+    // It is an error if this happen. But it can happen if the sequence numbers
>+    // for some reason restart without that the history has been reset.
>+    auto size_iterator = packet_size_.find(stored_packet.packet->size());
>+    if (size_iterator != packet_size_.end() &&
>+        size_iterator->second == stored_packet.packet->SequenceNumber()) {
>+      packet_size_.erase(size_iterator);
>+    }
>+  }
>   stored_packet.packet = std::move(packet);
> 
>   if (stored_packet.packet->capture_time_ms() <= 0) {
>@@ -116,8 +125,7 @@ void RtpPacketHistory::PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
> }
> 
> std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPacketAndSetSendTime(
>-    uint16_t sequence_number,
>-    bool verify_rtt) {
>+    uint16_t sequence_number) {
>   rtc::CritScope cs(&lock_);
>   if (mode_ == StorageMode::kDisabled) {
>     return nullptr;
>@@ -130,7 +138,7 @@ std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPacketAndSetSendTime(
>   }
> 
>   StoredPacket& packet = rtp_it->second;
>-  if (verify_rtt && !VerifyRtt(rtp_it->second, now_ms)) {
>+  if (!VerifyRtt(rtp_it->second, now_ms)) {
>     return nullptr;
>   }
> 
>@@ -150,8 +158,7 @@ std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPacketAndSetSendTime(
> }
> 
> absl::optional<RtpPacketHistory::PacketState> RtpPacketHistory::GetPacketState(
>-    uint16_t sequence_number,
>-    bool verify_rtt) const {
>+    uint16_t sequence_number) const {
>   rtc::CritScope cs(&lock_);
>   if (mode_ == StorageMode::kDisabled) {
>     return absl::nullopt;
>@@ -162,7 +169,7 @@ absl::optional<RtpPacketHistory::PacketState> RtpPacketHistory::GetPacketState(
>     return absl::nullopt;
>   }
> 
>-  if (verify_rtt && !VerifyRtt(rtp_it->second, clock_->TimeInMilliseconds())) {
>+  if (!VerifyRtt(rtp_it->second, clock_->TimeInMilliseconds())) {
>     return absl::nullopt;
>   }
> 
>@@ -209,8 +216,19 @@ std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetBestFittingPacket(
>   const uint16_t seq_no = upper_bound_diff < lower_bound_diff
>                               ? size_iter_upper->second
>                               : size_iter_lower->second;
>-  RtpPacketToSend* best_packet =
>-      packet_history_.find(seq_no)->second.packet.get();
>+  auto history_it = packet_history_.find(seq_no);
>+  if (history_it == packet_history_.end()) {
>+    RTC_LOG(LS_ERROR) << "Can't find packet in history with seq_no" << seq_no;
>+    RTC_DCHECK(false);
>+    return nullptr;
>+  }
>+  if (!history_it->second.packet) {
>+    RTC_LOG(LS_ERROR) << "Packet pointer is null in history for seq_no"
>+                      << seq_no;
>+    RTC_DCHECK(false);
>+    return nullptr;
>+  }
>+  RtpPacketToSend* best_packet = history_it->second.packet.get();
>   return absl::make_unique<RtpPacketToSend>(*best_packet);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
>index 1646ba7c76f3afa5e9265e2630929a76ff3da072..095424e3d713123c9c3bb3b7096dbcdd5ea50760 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
>@@ -76,16 +76,13 @@ class RtpPacketHistory {
>                     absl::optional<int64_t> send_time_ms);
> 
>   // Gets stored RTP packet corresponding to the input |sequence number|.
>-  // Returns nullptr if packet is not found. If |verify_rtt| is true, doesn't
>-  // return packet that was (re)sent too recently.
>+  // Returns nullptr if packet is not found or was (re)sent too recently.
>   std::unique_ptr<RtpPacketToSend> GetPacketAndSetSendTime(
>-      uint16_t sequence_number,
>-      bool verify_rtt);
>+      uint16_t sequence_number);
> 
>   // Similar to GetPacketAndSetSendTime(), but only returns a snapshot of the
>   // current state for packet, and never updates internal state.
>-  absl::optional<PacketState> GetPacketState(uint16_t sequence_number,
>-                                             bool verify_rtt) const;
>+  absl::optional<PacketState> GetPacketState(uint16_t sequence_number) const;
> 
>   // Get the packet (if any) from the history, with size closest to
>   // |packet_size|. The exact size of the packet is not guaranteed.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
>index 85e9eb28cd7453a462d287a610d3d46e426c9020..140434cbbde79a6d1a25942ff801eaf1f6d9936f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
>@@ -63,11 +63,11 @@ TEST_F(RtpPacketHistoryTest, ClearsHistoryAfterSetStoreStatus) {
>   // Store a packet, but with send-time. It should then not be removed.
>   hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), kAllowRetransmission,
>                      absl::nullopt);
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
> 
>   // Changing store status, even to the current one, will clear the history.
>   hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
> }
> 
> TEST_F(RtpPacketHistoryTest, StartSeqResetAfterReset) {
>@@ -75,16 +75,16 @@ TEST_F(RtpPacketHistoryTest, StartSeqResetAfterReset) {
>   // Store a packet, but with send-time. It should then not be removed.
>   hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), kAllowRetransmission,
>                      absl::nullopt);
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
> 
>   // Changing store status, to clear the history.
>   hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
> 
>   // Add a new packet.
>   hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1), kAllowRetransmission,
>                      absl::nullopt);
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1));
> 
>   // Advance time past where packet expires.
>   fake_clock_.AdvanceTimeMilliseconds(
>@@ -94,9 +94,9 @@ TEST_F(RtpPacketHistoryTest, StartSeqResetAfterReset) {
>   // Add one more packet and verify no state left from packet before reset.
>   hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 2)),
>                      kAllowRetransmission, absl::nullopt);
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
>-  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 2), false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1));
>+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 2)));
> }
> 
> TEST_F(RtpPacketHistoryTest, NoStoreStatus) {
>@@ -104,21 +104,21 @@ TEST_F(RtpPacketHistoryTest, NoStoreStatus) {
>   std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
>   hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, absl::nullopt);
>   // Packet should not be stored.
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
> }
> 
> TEST_F(RtpPacketHistoryTest, GetRtpPacket_NotStored) {
>   hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>-  EXPECT_FALSE(hist_.GetPacketState(0, false));
>+  EXPECT_FALSE(hist_.GetPacketState(0));
> }
> 
> TEST_F(RtpPacketHistoryTest, PutRtpPacket) {
>   hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>   std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
> 
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
>   hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, absl::nullopt);
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
> }
> 
> TEST_F(RtpPacketHistoryTest, GetRtpPacket) {
>@@ -130,7 +130,7 @@ TEST_F(RtpPacketHistoryTest, GetRtpPacket) {
>   hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, absl::nullopt);
> 
>   std::unique_ptr<RtpPacketToSend> packet_out =
>-      hist_.GetPacketAndSetSendTime(kStartSeqNum, false);
>+      hist_.GetPacketAndSetSendTime(kStartSeqNum);
>   EXPECT_TRUE(packet_out);
>   EXPECT_EQ(buffer, packet_out->Buffer());
>   EXPECT_EQ(capture_time_ms, packet_out->capture_time_ms());
>@@ -146,7 +146,7 @@ TEST_F(RtpPacketHistoryTest, NoCaptureTime) {
>   hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, absl::nullopt);
> 
>   std::unique_ptr<RtpPacketToSend> packet_out =
>-      hist_.GetPacketAndSetSendTime(kStartSeqNum, false);
>+      hist_.GetPacketAndSetSendTime(kStartSeqNum);
>   EXPECT_TRUE(packet_out);
>   EXPECT_EQ(buffer, packet_out->Buffer());
>   EXPECT_EQ(capture_time_ms, packet_out->capture_time_ms());
>@@ -161,19 +161,21 @@ TEST_F(RtpPacketHistoryTest, DontRetransmit) {
> 
>   // Get the packet and verify data.
>   std::unique_ptr<RtpPacketToSend> packet_out;
>-  packet_out = hist_.GetPacketAndSetSendTime(kStartSeqNum, false);
>+  packet_out = hist_.GetPacketAndSetSendTime(kStartSeqNum);
>   ASSERT_TRUE(packet_out);
>   EXPECT_EQ(buffer.size(), packet_out->size());
>   EXPECT_EQ(capture_time_ms, packet_out->capture_time_ms());
> 
>   // Non-retransmittable packets are immediately removed, so getting in again
>   // should fail.
>-  EXPECT_FALSE(hist_.GetPacketAndSetSendTime(kStartSeqNum, false));
>+  EXPECT_FALSE(hist_.GetPacketAndSetSendTime(kStartSeqNum));
> }
> 
> TEST_F(RtpPacketHistoryTest, PacketStateIsCorrect) {
>   const uint32_t kSsrc = 92384762;
>+  const int64_t kRttMs = 100;
>   hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
>+  hist_.SetRtt(kRttMs);
>   std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
>   packet->SetSsrc(kSsrc);
>   packet->SetPayloadSize(1234);
>@@ -183,7 +185,7 @@ TEST_F(RtpPacketHistoryTest, PacketStateIsCorrect) {
>                      fake_clock_.TimeInMilliseconds());
> 
>   absl::optional<RtpPacketHistory::PacketState> state =
>-      hist_.GetPacketState(kStartSeqNum, false);
>+      hist_.GetPacketState(kStartSeqNum);
>   ASSERT_TRUE(state);
>   EXPECT_EQ(state->rtp_sequence_number, kStartSeqNum);
>   EXPECT_EQ(state->send_time_ms, fake_clock_.TimeInMilliseconds());
>@@ -193,9 +195,10 @@ TEST_F(RtpPacketHistoryTest, PacketStateIsCorrect) {
>   EXPECT_EQ(state->times_retransmitted, 0u);
> 
>   fake_clock_.AdvanceTimeMilliseconds(1);
>-  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum));
>+  fake_clock_.AdvanceTimeMilliseconds(kRttMs + 1);
> 
>-  state = hist_.GetPacketState(kStartSeqNum, false);
>+  state = hist_.GetPacketState(kStartSeqNum);
>   ASSERT_TRUE(state);
>   EXPECT_EQ(state->times_retransmitted, 1u);
> }
>@@ -211,7 +214,7 @@ TEST_F(RtpPacketHistoryTest, MinResendTimeWithPacer) {
>   hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, absl::nullopt);
> 
>   // First transmission: TimeToSendPacket() call from pacer.
>-  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum));
> 
>   // First retransmission - allow early retransmission.
>   fake_clock_.AdvanceTimeMilliseconds(1);
>@@ -221,26 +224,24 @@ TEST_F(RtpPacketHistoryTest, MinResendTimeWithPacer) {
>   //    packet is there and verify RTT constraints. Then we use the ssrc
>   //    and sequence number to enqueue the retransmission in the pacer
>   // 2) When the pacer determines that it is time to send the packet, it calls
>-  //    GetPacketAndSetSendTime(). This time we do not need to verify RTT as
>-  //    has that has already been done.
>+  //    GetPacketAndSetSendTime().
>   absl::optional<RtpPacketHistory::PacketState> packet_state =
>-      hist_.GetPacketState(kStartSeqNum, /*verify_rtt=*/true);
>+      hist_.GetPacketState(kStartSeqNum);
>   EXPECT_TRUE(packet_state);
>   EXPECT_EQ(len, packet_state->payload_size);
>   EXPECT_EQ(capture_time_ms, packet_state->capture_time_ms);
> 
>   // Retransmission was allowed, next send it from pacer.
>-  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum,
>-                                            /*verify_rtt=*/false));
>+  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum));
> 
>   // Second retransmission - advance time to just before retransmission OK.
>   fake_clock_.AdvanceTimeMilliseconds(kMinRetransmitIntervalMs - 1);
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, /*verify_rtt=*/true));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
> 
>   // Advance time to just after retransmission OK.
>   fake_clock_.AdvanceTimeMilliseconds(1);
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, /*verify_rtt=*/true));
>-  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
>+  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum));
> }
> 
> TEST_F(RtpPacketHistoryTest, MinResendTimeWithoutPacer) {
>@@ -256,18 +257,18 @@ TEST_F(RtpPacketHistoryTest, MinResendTimeWithoutPacer) {
> 
>   // First retransmission - allow early retransmission.
>   fake_clock_.AdvanceTimeMilliseconds(1);
>-  packet = hist_.GetPacketAndSetSendTime(kStartSeqNum, true);
>+  packet = hist_.GetPacketAndSetSendTime(kStartSeqNum);
>   EXPECT_TRUE(packet);
>   EXPECT_EQ(len, packet->size());
>   EXPECT_EQ(capture_time_ms, packet->capture_time_ms());
> 
>   // Second retransmission - advance time to just before retransmission OK.
>   fake_clock_.AdvanceTimeMilliseconds(kMinRetransmitIntervalMs - 1);
>-  EXPECT_FALSE(hist_.GetPacketAndSetSendTime(kStartSeqNum, true));
>+  EXPECT_FALSE(hist_.GetPacketAndSetSendTime(kStartSeqNum));
> 
>   // Advance time to just after retransmission OK.
>   fake_clock_.AdvanceTimeMilliseconds(1);
>-  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum, true));
>+  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum));
> }
> 
> TEST_F(RtpPacketHistoryTest, RemovesOldestSentPacketWhenAtMaxSize) {
>@@ -289,7 +290,7 @@ TEST_F(RtpPacketHistoryTest, RemovesOldestSentPacketWhenAtMaxSize) {
>   }
> 
>   // First packet should still be there.
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
> 
>   // History is full, oldest one should be overwritten.
>   std::unique_ptr<RtpPacketToSend> packet =
>@@ -298,8 +299,8 @@ TEST_F(RtpPacketHistoryTest, RemovesOldestSentPacketWhenAtMaxSize) {
>                      fake_clock_.TimeInMilliseconds());
> 
>   // Oldest packet should be gone, but packet after than one still present.
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1));
> }
> 
> TEST_F(RtpPacketHistoryTest, RemovesOldestPacketWhenAtMaxCapacity) {
>@@ -317,7 +318,7 @@ TEST_F(RtpPacketHistoryTest, RemovesOldestPacketWhenAtMaxCapacity) {
>   }
> 
>   // First packet should still be there.
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
> 
>   // History is full, oldest one should be overwritten.
>   std::unique_ptr<RtpPacketToSend> packet =
>@@ -326,8 +327,8 @@ TEST_F(RtpPacketHistoryTest, RemovesOldestPacketWhenAtMaxCapacity) {
>                      fake_clock_.TimeInMilliseconds());
> 
>   // Oldest packet should be gone, but packet after than one still present.
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1));
> }
> 
> TEST_F(RtpPacketHistoryTest, DontRemoveUnsentPackets) {
>@@ -343,25 +344,25 @@ TEST_F(RtpPacketHistoryTest, DontRemoveUnsentPackets) {
>   fake_clock_.AdvanceTimeMilliseconds(RtpPacketHistory::kMinPacketDurationMs);
> 
>   // First packet should still be there.
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
> 
>   // History is full, but old packets not sent, so allow expansion.
>   hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + kMaxNumPackets)),
>                      kAllowRetransmission, fake_clock_.TimeInMilliseconds());
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
> 
>   // Set all packet as sent and advance time past min packet duration time,
>   // otherwise packets till still be prevented from being removed.
>   for (size_t i = 0; i <= kMaxNumPackets; ++i) {
>-    EXPECT_TRUE(hist_.GetPacketAndSetSendTime(To16u(kStartSeqNum + i), false));
>+    EXPECT_TRUE(hist_.GetPacketAndSetSendTime(To16u(kStartSeqNum + i)));
>   }
>   fake_clock_.AdvanceTimeMilliseconds(RtpPacketHistory::kMinPacketDurationMs);
>   // Add a new packet, this means the two oldest ones will be culled.
>   hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + kMaxNumPackets + 1)),
>                      kAllowRetransmission, fake_clock_.TimeInMilliseconds());
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum + 1, false));
>-  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 2), false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum + 1));
>+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 2)));
> }
> 
> TEST_F(RtpPacketHistoryTest, DontRemoveTooRecentlyTransmittedPackets) {
>@@ -378,15 +379,15 @@ TEST_F(RtpPacketHistoryTest, DontRemoveTooRecentlyTransmittedPackets) {
>   hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1), kAllowRetransmission,
>                      fake_clock_.TimeInMilliseconds());
>   // First packet should still be there.
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
> 
>   // Advance time to where packet will be eligible for removal and try again.
>   fake_clock_.AdvanceTimeMilliseconds(1);
>   hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 2)),
>                      kAllowRetransmission, fake_clock_.TimeInMilliseconds());
>   // First packet should no be gone, but next one still there.
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1));
> }
> 
> TEST_F(RtpPacketHistoryTest, DontRemoveTooRecentlyTransmittedPacketsHighRtt) {
>@@ -407,15 +408,15 @@ TEST_F(RtpPacketHistoryTest, DontRemoveTooRecentlyTransmittedPacketsHighRtt) {
>   hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1), kAllowRetransmission,
>                      fake_clock_.TimeInMilliseconds());
>   // First packet should still be there.
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
> 
>   // Advance time to where packet will be eligible for removal and try again.
>   fake_clock_.AdvanceTimeMilliseconds(1);
>   hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 2)),
>                      kAllowRetransmission, fake_clock_.TimeInMilliseconds());
>   // First packet should no be gone, but next one still there.
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1));
> }
> 
> TEST_F(RtpPacketHistoryTest, RemovesOldWithCulling) {
>@@ -431,14 +432,14 @@ TEST_F(RtpPacketHistoryTest, RemovesOldWithCulling) {
>   fake_clock_.AdvanceTimeMilliseconds(kMaxPacketDurationMs - 1);
> 
>   // First packet should still be there.
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
> 
>   // Advance to where packet can be culled, even if buffer is not full.
>   fake_clock_.AdvanceTimeMilliseconds(1);
>   hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1), kAllowRetransmission,
>                      fake_clock_.TimeInMilliseconds());
> 
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
> }
> 
> TEST_F(RtpPacketHistoryTest, RemovesOldWithCullingHighRtt) {
>@@ -457,14 +458,14 @@ TEST_F(RtpPacketHistoryTest, RemovesOldWithCullingHighRtt) {
>   fake_clock_.AdvanceTimeMilliseconds(kMaxPacketDurationMs - 1);
> 
>   // First packet should still be there.
>-  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum));
> 
>   // Advance to where packet can be culled, even if buffer is not full.
>   fake_clock_.AdvanceTimeMilliseconds(1);
>   hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1), kAllowRetransmission,
>                      fake_clock_.TimeInMilliseconds());
> 
>-  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum));
> }
> 
> TEST_F(RtpPacketHistoryTest, GetBestFittingPacket) {
>@@ -503,7 +504,7 @@ TEST_F(RtpPacketHistoryTest,
>   ASSERT_THAT(packet, ::testing::NotNull());
> 
>   // Send the packet and advance time past where packet expires.
>-  ASSERT_THAT(hist_.GetPacketAndSetSendTime(kStartSeqNum, false),
>+  ASSERT_THAT(hist_.GetPacketAndSetSendTime(kStartSeqNum),
>               ::testing::NotNull());
>   fake_clock_.AdvanceTimeMilliseconds(
>       RtpPacketHistory::kPacketCullingDelayFactor *
>@@ -513,7 +514,7 @@ TEST_F(RtpPacketHistoryTest,
>   packet->SetPayloadSize(100);
>   hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
>                      fake_clock_.TimeInMilliseconds());
>-  ASSERT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  ASSERT_FALSE(hist_.GetPacketState(kStartSeqNum));
> 
>   auto best_packet = hist_.GetBestFittingPacket(target_packet_size + 2);
>   ASSERT_THAT(best_packet, ::testing::NotNull());
>@@ -573,7 +574,7 @@ TEST_F(RtpPacketHistoryTest,
>   hist_.PutRtpPacket(std::move(packet), kDontRetransmit,
>                      fake_clock_.TimeInMilliseconds());
>   EXPECT_THAT(hist_.GetBestFittingPacket(50), ::testing::IsNull());
>-  EXPECT_THAT(hist_.GetPacketAndSetSendTime(kStartSeqNum, false),
>+  EXPECT_THAT(hist_.GetPacketAndSetSendTime(kStartSeqNum),
>               ::testing::NotNull());
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
>index c8deb99d90e5586fa4c41525df4adf7f95524dd3..f80fad68e0d326d6effeb5595a240f24173a980c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> 
>+#include <stddef.h>
>+#include <cstdint>
> #include <vector>
> 
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
>@@ -67,6 +69,7 @@ void RtpPacketReceived::GetHeader(RTPHeader* header) const {
>   GetExtension<RepairedRtpStreamId>(&header->extension.repaired_stream_id);
>   GetExtension<RtpMid>(&header->extension.mid);
>   GetExtension<PlayoutDelayLimits>(&header->extension.playout_delay);
>+  header->extension.color_space = GetExtension<ColorSpaceExtension>();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h
>index 86900310e49e3198e9e49125d35b38a670858851..566b116a10a613c246093f37929bdfd3d6159f7e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h
>@@ -10,9 +10,11 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
> 
>+#include <stdint.h>
> #include <vector>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "api/array_view.h"
>+#include "api/rtp_headers.h"
> #include "modules/rtp_rtcp/source/rtp_packet.h"
> #include "system_wrappers/include/ntp_time.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
>index 0153bbe73ac229ed3400728f5c0be062b2d01d93..b55e74aaf01f807dd29758521a23e8de9e30fd99 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> 
>+#include <cstdint>
>+
> namespace webrtc {
> 
> RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
>index 1ed7dda478269772047eb670f3a87b43d29ef7ea..56b1024a059d7ac5bdaae1d3d0b22303f4ccdf2d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
>@@ -10,9 +10,12 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <vector>
> 
> #include "api/array_view.h"
>+#include "api/video/video_timing.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "modules/rtp_rtcp/source/rtp_packet.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
>index 125000d2dc94ed4be17862e04b24406bf3d2747c..b1c0e42525eae52677c0f80a0ccf64171459e4e7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
>@@ -19,6 +19,7 @@
> namespace webrtc {
> namespace {
> 
>+using ::testing::Each;
> using ::testing::ElementsAre;
> using ::testing::ElementsAreArray;
> using ::testing::IsEmpty;
>@@ -184,6 +185,52 @@ constexpr uint8_t kPacketWithLegacyTimingExtension[] = {
>     0x04, 0x00, 0x00, 0x00,
>     0x00, 0x00, 0x00, 0x00};
> // clang-format on
>+
>+HdrMetadata CreateTestHdrMetadata() {
>+  // Random but reasonable HDR metadata.
>+  HdrMetadata hdr_metadata;
>+  hdr_metadata.mastering_metadata.luminance_max = 2000.0;
>+  hdr_metadata.mastering_metadata.luminance_min = 2.0001;
>+  hdr_metadata.mastering_metadata.primary_r.x = 0.3003;
>+  hdr_metadata.mastering_metadata.primary_r.y = 0.4004;
>+  hdr_metadata.mastering_metadata.primary_g.x = 0.3201;
>+  hdr_metadata.mastering_metadata.primary_g.y = 0.4604;
>+  hdr_metadata.mastering_metadata.primary_b.x = 0.3409;
>+  hdr_metadata.mastering_metadata.primary_b.y = 0.4907;
>+  hdr_metadata.mastering_metadata.white_point.x = 0.4103;
>+  hdr_metadata.mastering_metadata.white_point.y = 0.4806;
>+  hdr_metadata.max_content_light_level = 2345;
>+  hdr_metadata.max_frame_average_light_level = 1789;
>+  return hdr_metadata;
>+}
>+
>+ColorSpace CreateTestColorSpace(bool with_hdr_metadata) {
>+  ColorSpace color_space(
>+      ColorSpace::PrimaryID::kBT709, ColorSpace::TransferID::kGAMMA22,
>+      ColorSpace::MatrixID::kSMPTE2085, ColorSpace::RangeID::kFull);
>+  if (with_hdr_metadata) {
>+    HdrMetadata hdr_metadata = CreateTestHdrMetadata();
>+    color_space.set_hdr_metadata(&hdr_metadata);
>+  }
>+  return color_space;
>+}
>+
>+void TestCreateAndParseColorSpaceExtension(bool with_hdr_metadata) {
>+  // Create packet with extension.
>+  RtpPacket::ExtensionManager extensions(/*extmap-allow-mixed=*/true);
>+  extensions.Register<ColorSpaceExtension>(1);
>+  RtpPacket packet(&extensions);
>+  const ColorSpace kColorSpace = CreateTestColorSpace(with_hdr_metadata);
>+  EXPECT_TRUE(packet.SetExtension<ColorSpaceExtension>(kColorSpace));
>+  packet.SetPayloadSize(42);
>+
>+  // Read packet with the extension.
>+  RtpPacketReceived parsed(&extensions);
>+  EXPECT_TRUE(parsed.Parse(packet.Buffer()));
>+  ColorSpace parsed_color_space;
>+  EXPECT_TRUE(parsed.GetExtension<ColorSpaceExtension>(&parsed_color_space));
>+  EXPECT_EQ(kColorSpace, parsed_color_space);
>+}
> }  // namespace
> 
> TEST(RtpPacketTest, CreateMinimum) {
>@@ -225,8 +272,7 @@ TEST(RtpPacketTest, CreateWith2Extensions) {
> }
> 
> TEST(RtpPacketTest, CreateWithTwoByteHeaderExtensionFirst) {
>-  RtpPacketToSend::ExtensionManager extensions;
>-  extensions.SetMixedOneTwoByteHeaderSupported(true);
>+  RtpPacketToSend::ExtensionManager extensions(true);
>   extensions.Register(kRtpExtensionTransmissionTimeOffset,
>                       kTransmissionOffsetExtensionId);
>   extensions.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
>@@ -247,8 +293,7 @@ TEST(RtpPacketTest, CreateWithTwoByteHeaderExtensionFirst) {
> 
> TEST(RtpPacketTest, CreateWithTwoByteHeaderExtensionLast) {
>   // This test will trigger RtpPacket::PromoteToTwoByteHeaderExtension().
>-  RtpPacketToSend::ExtensionManager extensions;
>-  extensions.SetMixedOneTwoByteHeaderSupported(true);
>+  RtpPacketToSend::ExtensionManager extensions(true);
>   extensions.Register(kRtpExtensionTransmissionTimeOffset,
>                       kTransmissionOffsetExtensionId);
>   extensions.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
>@@ -367,11 +412,10 @@ TEST(RtpPacketTest, CreatePurePadding) {
>   packet.SetSequenceNumber(kSeqNum);
>   packet.SetTimestamp(kTimestamp);
>   packet.SetSsrc(kSsrc);
>-  Random random(0x123456789);
> 
>   EXPECT_LT(packet.size(), packet.capacity());
>-  EXPECT_FALSE(packet.SetPadding(kPaddingSize + 1, &random));
>-  EXPECT_TRUE(packet.SetPadding(kPaddingSize, &random));
>+  EXPECT_FALSE(packet.SetPadding(kPaddingSize + 1));
>+  EXPECT_TRUE(packet.SetPadding(kPaddingSize));
>   EXPECT_EQ(packet.size(), packet.capacity());
> }
> 
>@@ -383,13 +427,48 @@ TEST(RtpPacketTest, CreateUnalignedPadding) {
>   packet.SetTimestamp(kTimestamp);
>   packet.SetSsrc(kSsrc);
>   packet.SetPayloadSize(kPayloadSize);
>-  Random r(0x123456789);
> 
>   EXPECT_LT(packet.size(), packet.capacity());
>-  EXPECT_TRUE(packet.SetPadding(kMaxPaddingSize, &r));
>+  EXPECT_TRUE(packet.SetPadding(kMaxPaddingSize));
>   EXPECT_EQ(packet.size(), packet.capacity());
> }
> 
>+TEST(RtpPacketTest, WritesPaddingSizeToLastByte) {
>+  const size_t kPaddingSize = 5;
>+  RtpPacket packet;
>+
>+  EXPECT_TRUE(packet.SetPadding(kPaddingSize));
>+  EXPECT_EQ(packet.data()[packet.size() - 1], kPaddingSize);
>+}
>+
>+TEST(RtpPacketTest, UsesZerosForPadding) {
>+  const size_t kPaddingSize = 5;
>+  RtpPacket packet;
>+
>+  EXPECT_TRUE(packet.SetPadding(kPaddingSize));
>+  EXPECT_THAT(rtc::MakeArrayView(packet.data() + 12, kPaddingSize - 1),
>+              Each(0));
>+}
>+
>+TEST(RtpPacketTest, CreateOneBytePadding) {
>+  size_t kPayloadSize = 123;
>+  RtpPacket packet(nullptr, 12 + kPayloadSize + 1);
>+  packet.SetPayloadSize(kPayloadSize);
>+
>+  EXPECT_TRUE(packet.SetPadding(1));
>+
>+  EXPECT_EQ(packet.size(), 12 + kPayloadSize + 1);
>+  EXPECT_EQ(packet.padding_size(), 1u);
>+}
>+
>+TEST(RtpPacketTest, FailsToAddPaddingWithoutCapacity) {
>+  size_t kPayloadSize = 123;
>+  RtpPacket packet(nullptr, 12 + kPayloadSize);
>+  packet.SetPayloadSize(kPayloadSize);
>+
>+  EXPECT_FALSE(packet.SetPadding(1));
>+}
>+
> TEST(RtpPacketTest, ParseMinimum) {
>   RtpPacketReceived packet;
>   EXPECT_TRUE(packet.Parse(kMinimumPacket, sizeof(kMinimumPacket)));
>@@ -433,6 +512,23 @@ TEST(RtpPacketTest, ParseWithExtension) {
>   EXPECT_EQ(0u, packet.padding_size());
> }
> 
>+TEST(RtpPacketTest, GetExtensionWithoutParametersReturnsOptionalValue) {
>+  RtpPacket::ExtensionManager extensions;
>+  extensions.Register<TransmissionOffset>(kTransmissionOffsetExtensionId);
>+  extensions.Register<RtpStreamId>(kRtpStreamIdExtensionId);
>+
>+  RtpPacketReceived packet(&extensions);
>+  EXPECT_TRUE(packet.Parse(kPacketWithTO, sizeof(kPacketWithTO)));
>+
>+  auto time_offset = packet.GetExtension<TransmissionOffset>();
>+  static_assert(
>+      std::is_same<decltype(time_offset),
>+                   absl::optional<TransmissionOffset::value_type>>::value,
>+      "");
>+  EXPECT_EQ(time_offset, kTimeOffset);
>+  EXPECT_FALSE(packet.GetExtension<RtpStreamId>().has_value());
>+}
>+
> TEST(RtpPacketTest, GetRawExtensionWhenPresent) {
>   constexpr uint8_t kRawPacket[] = {
>       // comment for clang-format to align kRawPacket nicer.
>@@ -751,4 +847,12 @@ TEST(RtpPacketTest, ParseLegacyTimingFrameExtension) {
>   EXPECT_EQ(receivied_timing.flags, 0);
> }
> 
>+TEST(RtpPacketTest, CreateAndParseColorSpaceExtension) {
>+  TestCreateAndParseColorSpaceExtension(/*with_hdr_metadata=*/true);
>+}
>+
>+TEST(RtpPacketTest, CreateAndParseColorSpaceExtensionWithoutHdrMetadata) {
>+  TestCreateAndParseColorSpaceExtension(/*with_hdr_metadata=*/false);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
>index 175a94be802161ef0779960c58adf30f0e46cbd1..0d0ca9618880373bd4ef10687439dbea27b659b0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
>@@ -11,12 +11,14 @@
> #include "modules/rtp_rtcp/source/rtp_rtcp_impl.h"
> 
> #include <string.h>
>-
> #include <algorithm>
>+#include <cstdint>
> #include <set>
> #include <string>
>+#include <utility>
> 
>-#include "api/rtpparameters.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
>+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
>@@ -31,6 +33,8 @@ const int64_t kRtpRtcpMaxIdleTimeProcessMs = 5;
> const int64_t kRtpRtcpRttProcessTimeMs = 1000;
> const int64_t kRtpRtcpBitrateProcessTimeMs = 10;
> const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
>+constexpr int32_t kDefaultVideoReportInterval = 1000;
>+constexpr int32_t kDefaultAudioReportInterval = 5000;
> }  // namespace
> 
> RtpRtcp::Configuration::Configuration() = default;
>@@ -62,7 +66,10 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
>                    configuration.rtcp_packet_type_counter_observer,
>                    configuration.event_log,
>                    configuration.outgoing_transport,
>-                   configuration.rtcp_interval_config),
>+                   configuration.rtcp_report_interval_ms > 0
>+                       ? configuration.rtcp_report_interval_ms
>+                       : (configuration.audio ? kDefaultAudioReportInterval
>+                                              : kDefaultVideoReportInterval)),
>       rtcp_receiver_(configuration.clock,
>                      configuration.receiver_only,
>                      configuration.rtcp_packet_type_counter_observer,
>@@ -70,6 +77,10 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
>                      configuration.intra_frame_callback,
>                      configuration.transport_feedback_callback,
>                      configuration.bitrate_allocation_observer,
>+                     configuration.rtcp_report_interval_ms > 0
>+                         ? configuration.rtcp_report_interval_ms
>+                         : (configuration.audio ? kDefaultAudioReportInterval
>+                                                : kDefaultVideoReportInterval),
>                      this),
>       clock_(configuration.clock),
>       audio_(configuration.audio),
>@@ -99,7 +110,9 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
>         configuration.send_packet_observer,
>         configuration.retransmission_rate_limiter,
>         configuration.overhead_observer,
>-        configuration.populate_network2_timestamp));
>+        configuration.populate_network2_timestamp,
>+        configuration.frame_encryptor, configuration.require_frame_encryption,
>+        configuration.extmap_allow_mixed));
>     // Make sure rtcp sender use same timestamp offset as rtp sender.
>     rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset());
> 
>@@ -175,10 +188,9 @@ void ModuleRtpRtcpImpl::Process() {
> 
>     // Verify receiver reports are delivered and the reported sequence number
>     // is increasing.
>-    int64_t rtcp_interval = RtcpReportInterval();
>-    if (rtcp_receiver_.RtcpRrTimeout(rtcp_interval)) {
>+    if (rtcp_receiver_.RtcpRrTimeout()) {
>       RTC_LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
>-    } else if (rtcp_receiver_.RtcpRrSequenceNumberTimeout(rtcp_interval)) {
>+    } else if (rtcp_receiver_.RtcpRrSequenceNumberTimeout()) {
>       RTC_LOG_F(LS_WARNING) << "Timeout: No increase in RTCP RR extended "
>                                "highest sequence number.";
>     }
>@@ -253,6 +265,7 @@ void ModuleRtpRtcpImpl::IncomingRtcpPacket(const uint8_t* rtcp_packet,
> }
> 
> int32_t ModuleRtpRtcpImpl::RegisterSendPayload(const CodecInst& voice_codec) {
>+  rtcp_sender_.SetRtpClockRate(voice_codec.pltype, voice_codec.plfreq);
>   return rtp_sender_->RegisterPayload(
>       voice_codec.plname, voice_codec.pltype, voice_codec.plfreq,
>       voice_codec.channels, (voice_codec.rate < 0) ? 0 : voice_codec.rate);
>@@ -260,8 +273,10 @@ int32_t ModuleRtpRtcpImpl::RegisterSendPayload(const CodecInst& voice_codec) {
> 
> void ModuleRtpRtcpImpl::RegisterVideoSendPayload(int payload_type,
>                                                  const char* payload_name) {
>-  RTC_CHECK_EQ(
>-      0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 0, 0));
>+  rtcp_sender_.SetRtpClockRate(payload_type, kVideoPayloadTypeFrequency);
>+  RTC_CHECK_EQ(0,
>+               rtp_sender_->RegisterPayload(payload_name, payload_type,
>+                                            kVideoPayloadTypeFrequency, 0, 0));
> }
> 
> int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) {
>@@ -392,6 +407,11 @@ bool ModuleRtpRtcpImpl::SendingMedia() const {
>   return rtp_sender_ ? rtp_sender_->SendingMedia() : false;
> }
> 
>+void ModuleRtpRtcpImpl::SetAsPartOfAllocation(bool part_of_allocation) {
>+  RTC_CHECK(rtp_sender_);
>+  rtp_sender_->SetAsPartOfAllocation(part_of_allocation);
>+}
>+
> bool ModuleRtpRtcpImpl::SendOutgoingData(
>     FrameType frame_type,
>     int8_t payload_type,
>@@ -402,7 +422,7 @@ bool ModuleRtpRtcpImpl::SendOutgoingData(
>     const RTPFragmentationHeader* fragmentation,
>     const RTPVideoHeader* rtp_video_header,
>     uint32_t* transport_frame_id_out) {
>-  rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
>+  rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms, payload_type);
>   // Make sure an RTCP report isn't queued behind a key frame.
>   if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
>     rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
>@@ -604,6 +624,10 @@ void ModuleRtpRtcpImpl::UnsetRemb() {
>   rtcp_sender_.UnsetRemb();
> }
> 
>+void ModuleRtpRtcpImpl::SetExtmapAllowMixed(bool extmap_allow_mixed) {
>+  rtp_sender_->SetExtmapAllowMixed(extmap_allow_mixed);
>+}
>+
> int32_t ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension(
>     const RTPExtensionType type,
>     const uint8_t id) {
>@@ -843,13 +867,6 @@ std::vector<rtcp::TmmbItem> ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner) {
>   return rtcp_receiver_.BoundingSet(tmmbr_owner);
> }
> 
>-int64_t ModuleRtpRtcpImpl::RtcpReportInterval() {
>-  if (audio_)
>-    return rtcp_sender_.RtcpAudioReportInverval();
>-  else
>-    return rtcp_sender_.RtcpVideoReportInverval();
>-}
>-
> void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) {
>   std::set<uint32_t> ssrcs;
>   ssrcs.insert(main_ssrc);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
>index 604414adf2ff4541185298a1550e342adeceed84..8e9751dcc1f0b237960e4fd5496ca739210302a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
>@@ -11,17 +11,24 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> #include <set>
> #include <string>
>-#include <utility>
> #include <vector>
> 
> #include "absl/types/optional.h"
>+#include "api/rtp_headers.h"
> #include "api/video/video_bitrate_allocation.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "modules/include/module_common_types.h"
>+#include "modules/include/module_fec_types.h"
>+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
>-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"  // RTCPPacketType
> #include "modules/rtp_rtcp/source/packet_loss_stats.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
> #include "modules/rtp_rtcp/source/rtcp_receiver.h"
> #include "modules/rtp_rtcp/source/rtcp_sender.h"
> #include "modules/rtp_rtcp/source/rtp_sender.h"
>@@ -30,6 +37,10 @@
> 
> namespace webrtc {
> 
>+class Clock;
>+struct PacedPacketInfo;
>+struct RTPVideoHeader;
>+
> class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
>  public:
>   explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
>@@ -59,6 +70,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
> 
>   int32_t DeRegisterSendPayload(int8_t payload_type) override;
> 
>+  void SetExtmapAllowMixed(bool extmap_allow_mixed) override;
>+
>   // Register RTP header extension.
>   int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
>                                          uint8_t id) override;
>@@ -115,6 +128,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
> 
>   bool SendingMedia() const override;
> 
>+  void SetAsPartOfAllocation(bool part_of_allocation) override;
>+
>   // Used by the codec module to deliver a video or audio frame for
>   // packetization.
>   bool SendOutgoingData(FrameType frame_type,
>@@ -311,7 +326,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
>  private:
>   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
>   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
>-  int64_t RtcpReportInterval();
>   void SetRtcpReceiverSsrcs(uint32_t main_ssrc);
> 
>   void set_rtt_ms(int64_t rtt_ms);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
>index 5160a64be73d892270ec3ce811ead803eab67d24..632a537e8029101b5069fd26913dc7445f90a47a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
>@@ -49,7 +49,7 @@ class RtcpRttStatsTestImpl : public RtcpRttStats {
>   int64_t rtt_ms_;
> };
> 
>-class SendTransport : public Transport, public RtpData {
>+class SendTransport : public Transport {
>  public:
>   SendTransport()
>       : receiver_(nullptr),
>@@ -90,11 +90,6 @@ class SendTransport : public Transport, public RtpData {
>     ++rtcp_packets_sent_;
>     return true;
>   }
>-  int32_t OnReceivedPayloadData(const uint8_t* payload_data,
>-                                size_t payload_size,
>-                                const WebRtcRTPHeader* rtp_header) override {
>-    return 0;
>-  }
>   void SetKeepalivePayloadType(uint8_t payload_type) {
>     keepalive_payload_type_ = payload_type;
>   }
>@@ -129,7 +124,7 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
>   std::unique_ptr<ModuleRtpRtcpImpl> impl_;
>   uint32_t remote_ssrc_;
>   RtpKeepAliveConfig keepalive_config_;
>-  RtcpIntervalConfig rtcp_interval_config_;
>+  int rtcp_report_interval_ms_ = 0;
> 
>   void SetRemoteSsrc(uint32_t ssrc) {
>     remote_ssrc_ = ssrc;
>@@ -164,8 +159,8 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
>     CreateModuleImpl();
>     transport_.SetKeepalivePayloadType(config.payload_type);
>   }
>-  void SetRtcpIntervalConfigAndReset(const RtcpIntervalConfig& config) {
>-    rtcp_interval_config_ = config;
>+  void SetRtcpReportIntervalAndReset(int rtcp_report_interval_ms) {
>+    rtcp_report_interval_ms_ = rtcp_report_interval_ms;
>     CreateModuleImpl();
>   }
> 
>@@ -179,7 +174,7 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
>     config.rtcp_packet_type_counter_observer = this;
>     config.rtt_stats = &rtt_stats_;
>     config.keepalive_config = keepalive_config_;
>-    config.rtcp_interval_config = rtcp_interval_config_;
>+    config.rtcp_report_interval_ms = rtcp_report_interval_ms_;
> 
>     impl_.reset(new ModuleRtpRtcpImpl(config));
>     impl_->SetRTCPStatus(RtcpMode::kCompound);
>@@ -648,11 +643,8 @@ TEST_F(RtpRtcpImplTest, SendsKeepaliveAfterTimout) {
> TEST_F(RtpRtcpImplTest, ConfigurableRtcpReportInterval) {
>   const int kVideoReportInterval = 3000;
> 
>-  RtcpIntervalConfig config;
>-  config.video_interval_ms = kVideoReportInterval;
>-
>   // Recreate sender impl with new configuration, and redo setup.
>-  sender_.SetRtcpIntervalConfigAndReset(config);
>+  sender_.SetRtcpReportIntervalAndReset(kVideoReportInterval);
>   SetUp();
> 
>   SendFrame(&sender_, kBaseLayerTid);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
>index 9f10c40e5754026955234852007197e175bd7369..ddf91f5f5a2d580d5f99ff7f92e777f40436e049 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
>@@ -16,6 +16,7 @@
> #include <utility>
> 
> #include "absl/memory/memory.h"
>+#include "absl/strings/match.h"
> #include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
>@@ -94,14 +95,6 @@ const char* FrameTypeToString(FrameType frame_type) {
>   }
>   return "";
> }
>-
>-void CountPacket(RtpPacketCounter* counter, const RtpPacketToSend& packet) {
>-  ++counter->packets;
>-  counter->header_bytes += packet.headers_size();
>-  counter->padding_bytes += packet.padding_size();
>-  counter->payload_bytes += packet.payload_size();
>-}
>-
> }  // namespace
> 
> RTPSender::RTPSender(
>@@ -119,24 +112,33 @@ RTPSender::RTPSender(
>     SendPacketObserver* send_packet_observer,
>     RateLimiter* retransmission_rate_limiter,
>     OverheadObserver* overhead_observer,
>-    bool populate_network2_timestamp)
>+    bool populate_network2_timestamp,
>+    FrameEncryptorInterface* frame_encryptor,
>+    bool require_frame_encryption,
>+    bool extmap_allow_mixed)
>     : clock_(clock),
>       // TODO(holmer): Remove this conversion?
>       clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
>       random_(clock_->TimeInMicroseconds()),
>       audio_configured_(audio),
>       audio_(audio ? new RTPSenderAudio(clock, this) : nullptr),
>-      video_(audio ? nullptr : new RTPSenderVideo(clock, this, flexfec_sender)),
>+      video_(audio ? nullptr
>+                   : new RTPSenderVideo(clock,
>+                                        this,
>+                                        flexfec_sender,
>+                                        frame_encryptor,
>+                                        require_frame_encryption)),
>       paced_sender_(paced_sender),
>       transport_sequence_number_allocator_(sequence_number_allocator),
>       transport_feedback_observer_(transport_feedback_observer),
>       last_capture_time_ms_sent_(0),
>       transport_(transport),
>-      sending_media_(true),                   // Default to sending media.
>+      sending_media_(true),  // Default to sending media.
>+      force_part_of_allocation_(false),
>       max_packet_size_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
>       last_payload_type_(-1),
>       payload_type_map_(),
>-      rtp_header_extension_map_(),
>+      rtp_header_extension_map_(extmap_allow_mixed),
>       packet_history_(clock),
>       flexfec_packet_history_(clock),
>       // Statistics
>@@ -167,9 +169,7 @@ RTPSender::RTPSender(
>       overhead_observer_(overhead_observer),
>       populate_network2_timestamp_(populate_network2_timestamp),
>       send_side_bwe_with_overhead_(
>-          webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
>-      unlimited_retransmission_experiment_(
>-          field_trial::IsEnabled("WebRTC-UnlimitedScreenshareRetransmission")) {
>+          webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
>   // This random initialization is not intended to be cryptographic strong.
>   timestamp_offset_ = random_.Rand<uint32_t>();
>   // Random start, 16 bits. Can't be 0.
>@@ -239,6 +239,11 @@ uint32_t RTPSender::NackOverheadRate() const {
>   return nack_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
> }
> 
>+void RTPSender::SetExtmapAllowMixed(bool extmap_allow_mixed) {
>+  rtc::CritScope lock(&send_critsect_);
>+  rtp_header_extension_map_.SetExtmapAllowMixed(extmap_allow_mixed);
>+}
>+
> int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type,
>                                               uint8_t id) {
>   rtc::CritScope lock(&send_critsect_);
>@@ -260,13 +265,12 @@ int32_t RTPSender::DeregisterRtpHeaderExtension(RTPExtensionType type) {
>   return rtp_header_extension_map_.Deregister(type);
> }
> 
>-int32_t RTPSender::RegisterPayload(
>-    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
>-    int8_t payload_number,
>-    uint32_t frequency,
>-    size_t channels,
>-    uint32_t rate) {
>-  RTC_DCHECK_LT(strlen(payload_name), RTP_PAYLOAD_NAME_SIZE);
>+int32_t RTPSender::RegisterPayload(absl::string_view payload_name,
>+                                   int8_t payload_number,
>+                                   uint32_t frequency,
>+                                   size_t channels,
>+                                   uint32_t rate) {
>+  RTC_DCHECK_LT(payload_name.size(), RTP_PAYLOAD_NAME_SIZE);
>   rtc::CritScope lock(&send_critsect_);
> 
>   std::map<int8_t, RtpUtility::Payload*>::iterator it =
>@@ -278,8 +282,7 @@ int32_t RTPSender::RegisterPayload(
>     RTC_DCHECK(payload);
> 
>     // Check if it's the same as we already have.
>-    if (RtpUtility::StringCompare(payload->name, payload_name,
>-                                  RTP_PAYLOAD_NAME_SIZE - 1)) {
>+    if (absl::EqualsIgnoreCase(payload->name, payload_name)) {
>       if (audio_configured_ && payload->typeSpecific.is_audio()) {
>         auto& p = payload->typeSpecific.audio_payload();
>         if (rtc::SafeEq(p.format.clockrate_hz, frequency) &&
>@@ -426,11 +429,6 @@ bool RTPSender::SendOutgoingData(FrameType frame_type,
>       *transport_frame_id_out = rtp_timestamp;
>     if (!sending_media_)
>       return true;
>-
>-    // Cache video content type.
>-    if (!audio_configured_ && rtp_header) {
>-      video_content_type_ = rtp_header->content_type;
>-    }
>   }
>   VideoCodecType video_type = kVideoCodecGeneric;
>   if (CheckPayloadType(payload_type, &video_type) != 0) {
>@@ -619,10 +617,16 @@ size_t RTPSender::SendPadData(size_t bytes,
>     PacketOptions options;
>     // Padding packets are never retransmissions.
>     options.is_retransmit = false;
>-    bool has_transport_seq_num =
>-        UpdateTransportSequenceNumber(&padding_packet, &options.packet_id);
>-    padding_packet.SetPadding(padding_bytes_in_packet, &random_);
>-
>+    bool has_transport_seq_num;
>+    {
>+      rtc::CritScope lock(&send_critsect_);
>+      has_transport_seq_num =
>+          UpdateTransportSequenceNumber(&padding_packet, &options.packet_id);
>+      options.included_in_allocation =
>+          has_transport_seq_num || force_part_of_allocation_;
>+      options.included_in_feedback = has_transport_seq_num;
>+    }
>+    padding_packet.SetPadding(padding_bytes_in_packet);
>     if (has_transport_seq_num) {
>       AddPacketToTransportFeedback(options.packet_id, padding_packet,
>                                    pacing_info);
>@@ -654,7 +658,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
>   // Try to find packet in RTP packet history. Also verify RTT here, so that we
>   // don't retransmit too often.
>   absl::optional<RtpPacketHistory::PacketState> stored_packet =
>-      packet_history_.GetPacketState(packet_id, true);
>+      packet_history_.GetPacketState(packet_id);
>   if (!stored_packet) {
>     // Packet not found.
>     return 0;
>@@ -664,20 +668,9 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
> 
>   // Skip retransmission rate check if not configured.
>   if (retransmission_rate_limiter_) {
>-    // Skip retransmission rate check if sending screenshare and the experiment
>-    // is on.
>-    bool skip_retransmission_rate_limit = false;
>-    if (unlimited_retransmission_experiment_) {
>-      rtc::CritScope lock(&send_critsect_);
>-      skip_retransmission_rate_limit =
>-          video_content_type_ &&
>-          videocontenttypehelpers::IsScreenshare(*video_content_type_);
>-    }
>-
>     // Check if we're overusing retransmission bitrate.
>     // TODO(sprang): Add histograms for nack success or failure reasons.
>-    if (!skip_retransmission_rate_limit &&
>-        !retransmission_rate_limiter_->TryUseRate(packet_size)) {
>+    if (!retransmission_rate_limiter_->TryUseRate(packet_size)) {
>       return -1;
>     }
>   }
>@@ -696,7 +689,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
>   }
> 
>   std::unique_ptr<RtpPacketToSend> packet =
>-      packet_history_.GetPacketAndSetSendTime(packet_id, true);
>+      packet_history_.GetPacketAndSetSendTime(packet_id);
>   if (!packet) {
>     // Packet could theoretically time out between the first check and this one.
>     return 0;
>@@ -774,17 +767,14 @@ bool RTPSender::TimeToSendPacket(uint32_t ssrc,
>     return true;
> 
>   std::unique_ptr<RtpPacketToSend> packet;
>-  // No need to verify RTT here, it has already been checked before putting the
>-  // packet into the pacer. But _do_ update the send time.
>   if (ssrc == SSRC()) {
>-    packet = packet_history_.GetPacketAndSetSendTime(sequence_number, false);
>+    packet = packet_history_.GetPacketAndSetSendTime(sequence_number);
>   } else if (ssrc == FlexfecSsrc()) {
>-    packet =
>-        flexfec_packet_history_.GetPacketAndSetSendTime(sequence_number, false);
>+    packet = flexfec_packet_history_.GetPacketAndSetSendTime(sequence_number);
>   }
> 
>   if (!packet) {
>-    // Packet cannot be found.
>+    // Packet cannot be found or was resend too recently.
>     return true;
>   }
> 
>@@ -838,7 +828,16 @@ bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
>   // E.g. RTPSender::TrySendRedundantPayloads calls PrepareAndSendPacket with
>   // send_over_rtx = true but is_retransmit = false.
>   options.is_retransmit = is_retransmit || send_over_rtx;
>-  if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id)) {
>+  bool has_transport_seq_num;
>+  {
>+    rtc::CritScope lock(&send_critsect_);
>+    has_transport_seq_num =
>+        UpdateTransportSequenceNumber(packet_to_send, &options.packet_id);
>+    options.included_in_allocation =
>+        has_transport_seq_num || force_part_of_allocation_;
>+    options.included_in_feedback = has_transport_seq_num;
>+  }
>+  if (has_transport_seq_num) {
>     AddPacketToTransportFeedback(options.packet_id, *packet_to_send,
>                                  pacing_info);
>   }
>@@ -876,13 +875,13 @@ void RTPSender::UpdateRtpStats(const RtpPacketToSend& packet,
>     counters->first_packet_time_ms = now_ms;
> 
>   if (IsFecPacket(packet))
>-    CountPacket(&counters->fec, packet);
>+    counters->fec.AddPacket(packet);
> 
>   if (is_retransmit) {
>-    CountPacket(&counters->retransmitted, packet);
>+    counters->retransmitted.AddPacket(packet);
>     nack_bitrate_sent_.Update(packet.size(), now_ms);
>   }
>-  CountPacket(&counters->transmitted, packet);
>+  counters->transmitted.AddPacket(packet);
> 
>   if (rtp_stats_callback_)
>     rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc());
>@@ -920,15 +919,6 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
>   RTC_DCHECK(packet);
>   int64_t now_ms = clock_->TimeInMilliseconds();
> 
>-  // |capture_time_ms| <= 0 is considered invalid.
>-  // TODO(holmer): This should be changed all over Video Engine so that negative
>-  // time is consider invalid, while 0 is considered a valid time.
>-  if (packet->capture_time_ms() > 0) {
>-    packet->SetExtension<TransmissionOffset>(
>-        kTimestampTicksPerMs * (now_ms - packet->capture_time_ms()));
>-  }
>-  packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::MsTo24Bits(now_ms));
>-
>   if (video_) {
>     BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
>                                     ActualSendBitrateKbit(), packet->Ssrc());
>@@ -971,7 +961,31 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
> 
>   PacketOptions options;
>   options.is_retransmit = false;
>-  if (UpdateTransportSequenceNumber(packet.get(), &options.packet_id)) {
>+
>+  // |capture_time_ms| <= 0 is considered invalid.
>+  // TODO(holmer): This should be changed all over Video Engine so that negative
>+  // time is consider invalid, while 0 is considered a valid time.
>+  if (packet->capture_time_ms() > 0) {
>+    packet->SetExtension<TransmissionOffset>(
>+        kTimestampTicksPerMs * (now_ms - packet->capture_time_ms()));
>+
>+    if (populate_network2_timestamp_ &&
>+        packet->HasExtension<VideoTimingExtension>()) {
>+      packet->set_network2_time_ms(now_ms);
>+    }
>+  }
>+  packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::MsTo24Bits(now_ms));
>+
>+  bool has_transport_seq_num;
>+  {
>+    rtc::CritScope lock(&send_critsect_);
>+    has_transport_seq_num =
>+        UpdateTransportSequenceNumber(packet.get(), &options.packet_id);
>+    options.included_in_allocation =
>+        has_transport_seq_num || force_part_of_allocation_;
>+    options.included_in_feedback = has_transport_seq_num;
>+  }
>+  if (has_transport_seq_num) {
>     AddPacketToTransportFeedback(options.packet_id, *packet.get(),
>                                  PacedPacketInfo());
>   }
>@@ -1141,8 +1155,15 @@ void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
> 
> std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
>   rtc::CritScope lock(&send_critsect_);
>-  std::unique_ptr<RtpPacketToSend> packet(
>-      new RtpPacketToSend(&rtp_header_extension_map_, max_packet_size_));
>+  // TODO(danilchap): Find better motivator and value for extra capacity.
>+  // RtpPacketizer might slightly miscalulate needed size,
>+  // SRTP may benefit from extra space in the buffer and do encryption in place
>+  // saving reallocation.
>+  // While sending slightly oversized packet increase chance of dropped packet,
>+  // it is better than crash on drop packet without trying to send it.
>+  static constexpr int kExtraCapacity = 16;
>+  auto packet = absl::make_unique<RtpPacketToSend>(
>+      &rtp_header_extension_map_, max_packet_size_ + kExtraCapacity);
>   RTC_DCHECK(ssrc_);
>   packet->SetSsrc(*ssrc_);
>   packet->SetCsrcs(csrcs_);
>@@ -1181,10 +1202,9 @@ bool RTPSender::AssignSequenceNumber(RtpPacketToSend* packet) {
> }
> 
> bool RTPSender::UpdateTransportSequenceNumber(RtpPacketToSend* packet,
>-                                              int* packet_id) const {
>+                                              int* packet_id) {
>   RTC_DCHECK(packet);
>   RTC_DCHECK(packet_id);
>-  rtc::CritScope lock(&send_critsect_);
>   if (!rtp_header_extension_map_.IsRegistered(TransportSequenceNumber::kId))
>     return false;
> 
>@@ -1209,6 +1229,11 @@ bool RTPSender::SendingMedia() const {
>   return sending_media_;
> }
> 
>+void RTPSender::SetAsPartOfAllocation(bool part_of_allocation) {
>+  rtc::CritScope lock(&send_critsect_);
>+  force_part_of_allocation_ = part_of_allocation;
>+}
>+
> void RTPSender::SetTimestampOffset(uint32_t timestamp) {
>   rtc::CritScope lock(&send_critsect_);
>   timestamp_offset_ = timestamp;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.h
>index 5a93d7a52021e14c508d754d306e0a2d884ebf1b..f9bbbdd9e22a23df01d22c19d87cc72b69c6dcd7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.h
>@@ -17,10 +17,10 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/strings/string_view.h"
> #include "absl/types/optional.h"
> #include "api/array_view.h"
> #include "api/call/transport.h"
>-#include "api/video/video_content_type.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/include/flexfec_sender.h"
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
>@@ -38,6 +38,7 @@
> 
> namespace webrtc {
> 
>+class FrameEncryptorInterface;
> class OverheadObserver;
> class RateLimiter;
> class RtcEventLog;
>@@ -63,7 +64,10 @@ class RTPSender {
>             SendPacketObserver* send_packet_observer,
>             RateLimiter* nack_rate_limiter,
>             OverheadObserver* overhead_observer,
>-            bool populate_network2_timestamp);
>+            bool populate_network2_timestamp,
>+            FrameEncryptorInterface* frame_encryptor,
>+            bool require_frame_encryption,
>+            bool extmap_allow_mixed);
> 
>   ~RTPSender();
> 
>@@ -75,7 +79,7 @@ class RTPSender {
>   uint32_t FecOverheadRate() const;
>   uint32_t NackOverheadRate() const;
> 
>-  int32_t RegisterPayload(const char* payload_name,
>+  int32_t RegisterPayload(absl::string_view payload_name,
>                           const int8_t payload_type,
>                           const uint32_t frequency,
>                           const size_t channels,
>@@ -86,6 +90,8 @@ class RTPSender {
>   void SetSendingMediaStatus(bool enabled);
>   bool SendingMedia() const;
> 
>+  void SetAsPartOfAllocation(bool part_of_allocation);
>+
>   void GetDataCounters(StreamDataCounters* rtp_stats,
>                        StreamDataCounters* rtx_stats) const;
> 
>@@ -114,6 +120,8 @@ class RTPSender {
>                         uint32_t* transport_frame_id_out,
>                         int64_t expected_retransmission_time_ms);
> 
>+  void SetExtmapAllowMixed(bool extmap_allow_mixed);
>+
>   // RTP header extension
>   int32_t RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
>   bool RegisterRtpHeaderExtension(const std::string& uri, int id);
>@@ -247,8 +255,8 @@ class RTPSender {
>                           int64_t capture_time_ms,
>                           uint32_t ssrc);
> 
>-  bool UpdateTransportSequenceNumber(RtpPacketToSend* packet,
>-                                     int* packet_id) const;
>+  bool UpdateTransportSequenceNumber(RtpPacketToSend* packet, int* packet_id)
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(send_critsect_);
> 
>   void UpdateRtpStats(const RtpPacketToSend& packet,
>                       bool is_rtx,
>@@ -277,7 +285,7 @@ class RTPSender {
> 
>   Transport* transport_;
>   bool sending_media_ RTC_GUARDED_BY(send_critsect_);
>-
>+  bool force_part_of_allocation_ RTC_GUARDED_BY(send_critsect_);
>   size_t max_packet_size_;
> 
>   int8_t last_payload_type_ RTC_GUARDED_BY(send_critsect_);
>@@ -343,11 +351,6 @@ class RTPSender {
> 
>   const bool send_side_bwe_with_overhead_;
> 
>-  const bool unlimited_retransmission_experiment_;
>-
>-  absl::optional<VideoContentType> video_content_type_
>-      RTC_GUARDED_BY(send_critsect_);
>-
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
>index bf86a39339be73f9078359c7755bded87741ec46..636ccccf0d9b7ff90a0dad4d3d529aab6fdb0fa6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
>@@ -11,16 +11,18 @@
> #include "modules/rtp_rtcp/source/rtp_sender_audio.h"
> 
> #include <string.h>
>-
> #include <memory>
> #include <utility>
> 
>+#include "absl/strings/match.h"
>+#include "api/audio_codecs/audio_format.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
>+#include "modules/rtp_rtcp/source/rtp_packet.h"
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
> 
> namespace webrtc {
>@@ -30,14 +32,13 @@ RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
> 
> RTPSenderAudio::~RTPSenderAudio() {}
> 
>-int32_t RTPSenderAudio::RegisterAudioPayload(
>-    const char payloadName[RTP_PAYLOAD_NAME_SIZE],
>-    const int8_t payload_type,
>-    const uint32_t frequency,
>-    const size_t channels,
>-    const uint32_t rate,
>-    RtpUtility::Payload** payload) {
>-  if (RtpUtility::StringCompare(payloadName, "cn", 2)) {
>+int32_t RTPSenderAudio::RegisterAudioPayload(absl::string_view payload_name,
>+                                             const int8_t payload_type,
>+                                             const uint32_t frequency,
>+                                             const size_t channels,
>+                                             const uint32_t rate,
>+                                             RtpUtility::Payload** payload) {
>+  if (absl::EqualsIgnoreCase(payload_name, "cn")) {
>     rtc::CritScope cs(&send_audio_critsect_);
>     //  we can have multiple CNG payload types
>     switch (frequency) {
>@@ -56,7 +57,7 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
>       default:
>         return -1;
>     }
>-  } else if (RtpUtility::StringCompare(payloadName, "telephone-event", 15)) {
>+  } else if (absl::EqualsIgnoreCase(payload_name, "telephone-event")) {
>     rtc::CritScope cs(&send_audio_critsect_);
>     // Don't add it to the list
>     // we dont want to allow send with a DTMF payloadtype
>@@ -65,9 +66,9 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
>     return 0;
>   }
>   *payload = new RtpUtility::Payload(
>-      payloadName,
>+      payload_name,
>       PayloadUnion(AudioPayload{
>-          SdpAudioFormat(payloadName, frequency, channels), rate}));
>+          SdpAudioFormat(payload_name, frequency, channels), rate}));
>   return 0;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
>index 63dfc2b99c393852783d6c80f797a4a50d3ca2cd..1dbe5b5817cc3d1674629711b106343445b100af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
>@@ -11,14 +11,19 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
>+#include "absl/strings/string_view.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/source/dtmf_queue.h"
>-#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
> #include "modules/rtp_rtcp/source/rtp_sender.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/onetimeevent.h"
>+#include "rtc_base/thread_annotations.h"
>+#include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
> 
>@@ -27,7 +32,7 @@ class RTPSenderAudio {
>   RTPSenderAudio(Clock* clock, RTPSender* rtp_sender);
>   ~RTPSenderAudio();
> 
>-  int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
>+  int32_t RegisterAudioPayload(absl::string_view payload_name,
>                                int8_t payload_type,
>                                uint32_t frequency,
>                                size_t channels,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
>index 86b92da2e00b9cd14cbd3606c3555b8bb41f7816..a687bcbb35b27adc73b67ecfa082044b822916b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
>@@ -184,7 +184,8 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
>         false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,
>         nullptr, &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
>         &mock_rtc_event_log_, &send_packet_observer_,
>-        &retransmission_rate_limiter_, nullptr, populate_network2));
>+        &retransmission_rate_limiter_, nullptr, populate_network2, nullptr,
>+        false, false));
>     rtp_sender_->SetSequenceNumber(kSeqNum);
>     rtp_sender_->SetTimestampOffset(0);
>     rtp_sender_->SetSSRC(kSsrc);
>@@ -276,7 +277,7 @@ class TestRtpSenderVideo : public RTPSenderVideo {
>   TestRtpSenderVideo(Clock* clock,
>                      RTPSender* rtp_sender,
>                      FlexfecSender* flexfec_sender)
>-      : RTPSenderVideo(clock, rtp_sender, flexfec_sender) {}
>+      : RTPSenderVideo(clock, rtp_sender, flexfec_sender, nullptr, false) {}
>   ~TestRtpSenderVideo() override {}
> 
>   StorageType GetStorageType(const RTPVideoHeader& header,
>@@ -382,7 +383,8 @@ TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) {
>   rtp_sender_.reset(new RTPSender(
>       kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
>       nullptr, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
>-      nullptr, &retransmission_rate_limiter_, nullptr, false));
>+      nullptr, &retransmission_rate_limiter_, nullptr, false, nullptr, false,
>+      false));
>   rtp_sender_->SetTimestampOffset(0);
>   rtp_sender_->SetSSRC(kSsrc);
> 
>@@ -428,7 +430,8 @@ TEST_P(RtpSenderTestWithoutPacer,
>   rtp_sender_.reset(new RTPSender(
>       false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
>       &feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
>-      nullptr, &retransmission_rate_limiter_, &mock_overhead_observer, false));
>+      nullptr, &retransmission_rate_limiter_, &mock_overhead_observer, false,
>+      nullptr, false, false));
>   rtp_sender_->SetSSRC(kSsrc);
>   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>                    kRtpExtensionTransportSequenceNumber,
>@@ -455,7 +458,8 @@ TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
>   rtp_sender_.reset(new RTPSender(
>       false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
>       &feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
>-      &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false));
>+      &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
>+      nullptr, false, false));
>   rtp_sender_->SetSSRC(kSsrc);
>   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>                    kRtpExtensionTransportSequenceNumber,
>@@ -479,13 +483,15 @@ TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
>   ASSERT_TRUE(packet.GetExtension<TransportSequenceNumber>(&transport_seq_no));
>   EXPECT_EQ(kTransportSequenceNumber, transport_seq_no);
>   EXPECT_EQ(transport_.last_options_.packet_id, transport_seq_no);
>+  EXPECT_TRUE(transport_.last_options_.included_in_allocation);
> }
> 
> TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
>   rtp_sender_.reset(new RTPSender(
>       false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
>       &feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
>-      &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false));
>+      &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
>+      nullptr, false, false));
>   rtp_sender_->SetSSRC(kSsrc);
> 
>   SendGenericPayload();
>@@ -493,16 +499,53 @@ TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
>   EXPECT_FALSE(transport_.last_options_.is_retransmit);
> }
> 
>-TEST_P(RtpSenderTestWithoutPacer, NoAllocationIfNotRegistered) {
>+TEST_P(RtpSenderTestWithoutPacer,
>+       SetsIncludedInFeedbackWhenTransportSequenceNumberExtensionIsRegistered) {
>+  SetUpRtpSender(false, false);
>+  rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
>+                                          kTransportSequenceNumberExtensionId);
>+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
>+      .WillOnce(testing::Return(kTransportSequenceNumber));
>+  EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
>+  SendGenericPayload();
>+  EXPECT_TRUE(transport_.last_options_.included_in_feedback);
>+}
>+
>+TEST_P(
>+    RtpSenderTestWithoutPacer,
>+    SetsIncludedInAllocationWhenTransportSequenceNumberExtensionIsRegistered) {
>+  SetUpRtpSender(false, false);
>+  rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
>+                                          kTransportSequenceNumberExtensionId);
>+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
>+      .WillOnce(testing::Return(kTransportSequenceNumber));
>+  EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
>   SendGenericPayload();
>+  EXPECT_TRUE(transport_.last_options_.included_in_allocation);
>+}
>+
>+TEST_P(RtpSenderTestWithoutPacer,
>+       SetsIncludedInAllocationWhenForcedAsPartOfAllocation) {
>+  SetUpRtpSender(false, false);
>+  rtp_sender_->SetAsPartOfAllocation(true);
>+  SendGenericPayload();
>+  EXPECT_FALSE(transport_.last_options_.included_in_feedback);
>+  EXPECT_TRUE(transport_.last_options_.included_in_allocation);
>+}
>+
>+TEST_P(RtpSenderTestWithoutPacer, DoesnSetIncludedInAllocationByDefault) {
>+  SetUpRtpSender(false, false);
>+  SendGenericPayload();
>+  EXPECT_FALSE(transport_.last_options_.included_in_feedback);
>+  EXPECT_FALSE(transport_.last_options_.included_in_allocation);
> }
> 
> TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
>   testing::StrictMock<MockSendSideDelayObserver> send_side_delay_observer_;
>-  rtp_sender_.reset(
>-      new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
>-                    nullptr, nullptr, nullptr, &send_side_delay_observer_,
>-                    &mock_rtc_event_log_, nullptr, nullptr, nullptr, false));
>+  rtp_sender_.reset(new RTPSender(
>+      false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
>+      nullptr, nullptr, &send_side_delay_observer_, &mock_rtc_event_log_,
>+      nullptr, nullptr, nullptr, false, nullptr, false, false));
>   rtp_sender_->SetSSRC(kSsrc);
> 
>   const uint8_t kPayloadType = 127;
>@@ -580,7 +623,7 @@ TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
>       false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
>       &seq_num_allocator_, &feedback_observer_, nullptr, nullptr, nullptr,
>       &mock_rtc_event_log_, &send_packet_observer_,
>-      &retransmission_rate_limiter_, nullptr, false));
>+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>   rtp_sender_->SetSSRC(kSsrc);
>   rtp_sender_->SetStorePacketsStatus(true, 10);
>@@ -647,8 +690,8 @@ TEST_P(RtpSenderTest, WritesPacerExitToTimingExtension) {
>   EXPECT_EQ(kStoredTimeInMs, video_timing.pacer_exit_delta_ms);
> }
> 
>-TEST_P(RtpSenderTest, WritesNetwork2ToTimingExtension) {
>-  SetUpRtpSender(true, true);
>+TEST_P(RtpSenderTest, WritesNetwork2ToTimingExtensionWithPacer) {
>+  SetUpRtpSender(/*pacer=*/true, /*populate_network2=*/true);
>   rtp_sender_->SetStorePacketsStatus(true, 10);
>   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>                    kRtpExtensionVideoTiming, kVideoTimingExtensionId));
>@@ -686,6 +729,31 @@ TEST_P(RtpSenderTest, WritesNetwork2ToTimingExtension) {
>   EXPECT_EQ(kPacerExitMs, video_timing.pacer_exit_delta_ms);
> }
> 
>+TEST_P(RtpSenderTest, WritesNetwork2ToTimingExtensionWithoutPacer) {
>+  SetUpRtpSender(/*pacer=*/false, /*populate_network2=*/true);
>+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>+                   kRtpExtensionVideoTiming, kVideoTimingExtensionId));
>+  auto packet = rtp_sender_->AllocatePacket();
>+  packet->SetMarker(true);
>+  packet->set_capture_time_ms(fake_clock_.TimeInMilliseconds());
>+  const VideoSendTiming kVideoTiming = {0u, 0u, 0u, 0u, 0u, 0u, true};
>+  packet->SetExtension<VideoTimingExtension>(kVideoTiming);
>+  EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
>+
>+  const int kPropagateTimeMs = 10;
>+  fake_clock_.AdvanceTimeMilliseconds(kPropagateTimeMs);
>+
>+  EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
>+                                         kAllowRetransmission,
>+                                         RtpPacketSender::kNormalPriority));
>+
>+  EXPECT_EQ(1, transport_.packets_sent());
>+  absl::optional<VideoSendTiming> video_timing =
>+      transport_.last_sent_packet().GetExtension<VideoTimingExtension>();
>+  ASSERT_TRUE(video_timing);
>+  EXPECT_EQ(kPropagateTimeMs, video_timing->network2_timestamp_delta_ms);
>+}
>+
> TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {
>   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
>                                                kSsrc, kSeqNum, _, _, _));
>@@ -940,7 +1008,7 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
>       false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
>       nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
>       nullptr, nullptr, &send_packet_observer_, &retransmission_rate_limiter_,
>-      nullptr, false));
>+      nullptr, false, nullptr, false, false));
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>   rtp_sender_->SetSSRC(kSsrc);
>   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>@@ -966,7 +1034,7 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
>   rtp_sender_.reset(new RTPSender(
>       false, &fake_clock_, &transport, &mock_paced_sender_, nullptr, nullptr,
>       nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
>-      &retransmission_rate_limiter_, nullptr, false));
>+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>   rtp_sender_->SetSSRC(kSsrc);
>   rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
>@@ -1090,7 +1158,7 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
>       false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
>       &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
>       &mock_rtc_event_log_, &send_packet_observer_,
>-      &retransmission_rate_limiter_, nullptr, false));
>+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
>   rtp_sender_->SetSSRC(kMediaSsrc);
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>   rtp_sender_->SetStorePacketsStatus(true, 10);
>@@ -1150,7 +1218,7 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
>       false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
>       &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
>       &mock_rtc_event_log_, &send_packet_observer_,
>-      &retransmission_rate_limiter_, nullptr, false));
>+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
>   rtp_sender_->SetSSRC(kMediaSsrc);
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>   rtp_sender_->SetStorePacketsStatus(true, 10);
>@@ -1245,11 +1313,11 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
>                                nullptr /* rtp_state */, &fake_clock_);
> 
>   // Reset |rtp_sender_| to use FlexFEC.
>-  rtp_sender_.reset(
>-      new RTPSender(false, &fake_clock_, &transport_, nullptr, &flexfec_sender,
>-                    &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
>-                    &mock_rtc_event_log_, &send_packet_observer_,
>-                    &retransmission_rate_limiter_, nullptr, false));
>+  rtp_sender_.reset(new RTPSender(
>+      false, &fake_clock_, &transport_, nullptr, &flexfec_sender,
>+      &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
>+      &mock_rtc_event_log_, &send_packet_observer_,
>+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
>   rtp_sender_->SetSSRC(kMediaSsrc);
>   rtp_sender_->SetSequenceNumber(kSeqNum);
> 
>@@ -1313,7 +1381,7 @@ TEST_P(RtpSenderTest, FecOverheadRate) {
>       false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
>       &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
>       &mock_rtc_event_log_, &send_packet_observer_,
>-      &retransmission_rate_limiter_, nullptr, false));
>+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
>   rtp_sender_->SetSSRC(kMediaSsrc);
>   rtp_sender_->SetSequenceNumber(kSeqNum);
> 
>@@ -1365,7 +1433,7 @@ TEST_P(RtpSenderTest, FrameCountCallbacks) {
>   rtp_sender_.reset(new RTPSender(
>       false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr, nullptr,
>       nullptr, nullptr, &callback, nullptr, nullptr, nullptr,
>-      &retransmission_rate_limiter_, nullptr, false));
>+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
>   rtp_sender_->SetSSRC(kSsrc);
>   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
>   const uint8_t payload_type = 127;
>@@ -1425,10 +1493,10 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
>     uint32_t total_bitrate_;
>     uint32_t retransmit_bitrate_;
>   } callback;
>-  rtp_sender_.reset(
>-      new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
>-                    nullptr, &callback, nullptr, nullptr, nullptr, nullptr,
>-                    &retransmission_rate_limiter_, nullptr, false));
>+  rtp_sender_.reset(new RTPSender(
>+      false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
>+      &callback, nullptr, nullptr, nullptr, nullptr,
>+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
>   rtp_sender_->SetSSRC(kSsrc);
> 
>   // Simulate kNumPackets sent with kPacketInterval ms intervals, with the
>@@ -1485,10 +1553,10 @@ class RtpSenderAudioTest : public RtpSenderTest {
> 
>   void SetUp() override {
>     payload_ = kAudioPayload;
>-    rtp_sender_.reset(
>-        new RTPSender(true, &fake_clock_, &transport_, nullptr, nullptr,
>-                      nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
>-                      nullptr, &retransmission_rate_limiter_, nullptr, false));
>+    rtp_sender_.reset(new RTPSender(
>+        true, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
>+        nullptr, nullptr, nullptr, nullptr, nullptr,
>+        &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
>     rtp_sender_->SetSSRC(kSsrc);
>     rtp_sender_->SetSequenceNumber(kSeqNum);
>   }
>@@ -2151,10 +2219,11 @@ TEST_P(RtpSenderVideoTest,
> 
> TEST_P(RtpSenderTest, OnOverheadChanged) {
>   MockOverheadObserver mock_overhead_observer;
>-  rtp_sender_.reset(new RTPSender(
>-      false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
>-      nullptr, nullptr, nullptr, nullptr, nullptr,
>-      &retransmission_rate_limiter_, &mock_overhead_observer, false));
>+  rtp_sender_.reset(
>+      new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
>+                    nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
>+                    &retransmission_rate_limiter_, &mock_overhead_observer,
>+                    false, nullptr, false, false));
>   rtp_sender_->SetSSRC(kSsrc);
> 
>   // RTP overhead is 12B.
>@@ -2172,10 +2241,11 @@ TEST_P(RtpSenderTest, OnOverheadChanged) {
> 
> TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
>   MockOverheadObserver mock_overhead_observer;
>-  rtp_sender_.reset(new RTPSender(
>-      false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
>-      nullptr, nullptr, nullptr, nullptr, nullptr,
>-      &retransmission_rate_limiter_, &mock_overhead_observer, false));
>+  rtp_sender_.reset(
>+      new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
>+                    nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
>+                    &retransmission_rate_limiter_, &mock_overhead_observer,
>+                    false, nullptr, false, false));
>   rtp_sender_->SetSSRC(kSsrc);
> 
>   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
>@@ -2185,10 +2255,10 @@ TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
> 
> TEST_P(RtpSenderTest, SendsKeepAlive) {
>   MockTransport transport;
>-  rtp_sender_.reset(
>-      new RTPSender(false, &fake_clock_, &transport, nullptr, nullptr, nullptr,
>-                    nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
>-                    nullptr, &retransmission_rate_limiter_, nullptr, false));
>+  rtp_sender_.reset(new RTPSender(
>+      false, &fake_clock_, &transport, nullptr, nullptr, nullptr, nullptr,
>+      nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
>+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>   rtp_sender_->SetTimestampOffset(0);
>   rtp_sender_->SetSSRC(kSsrc);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
>index e8f0ea5c8de07180acbf1930c652a28aa1db8635..cb0b665608455745ccd05cb7a5ca9e9b50b7122f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
>@@ -19,6 +19,8 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>+#include "absl/strings/match.h"
>+#include "api/crypto/frameencryptorinterface.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
>@@ -91,6 +93,11 @@ void AddRtpHeaderExtensions(const RTPVideoHeader& video_header,
>       generic_descriptor.SetSpatialLayersBitmask(spatial_bimask);
> 
>       generic_descriptor.SetTemporalLayer(video_header.generic->temporal_index);
>+
>+      if (frame_type == kVideoFrameKey) {
>+        generic_descriptor.SetResolution(video_header.width,
>+                                         video_header.height);
>+      }
>     }
>     packet->SetExtension<RtpGenericFrameDescriptorExtension>(
>         generic_descriptor);
>@@ -115,7 +122,9 @@ bool MinimizeDescriptor(const RTPVideoHeader& full, RTPVideoHeader* minimized) {
> 
> RTPSenderVideo::RTPSenderVideo(Clock* clock,
>                                RTPSender* rtp_sender,
>-                               FlexfecSender* flexfec_sender)
>+                               FlexfecSender* flexfec_sender,
>+                               FrameEncryptorInterface* frame_encryptor,
>+                               bool require_frame_encryption)
>     : rtp_sender_(rtp_sender),
>       clock_(clock),
>       video_type_(kVideoCodecGeneric),
>@@ -128,7 +137,9 @@ RTPSenderVideo::RTPSenderVideo(Clock* clock,
>       delta_fec_params_{0, 1, kFecMaskRandom},
>       key_fec_params_{0, 1, kFecMaskRandom},
>       fec_bitrate_(1000, RateStatistics::kBpsScale),
>-      video_bitrate_(1000, RateStatistics::kBpsScale) {}
>+      video_bitrate_(1000, RateStatistics::kBpsScale),
>+      frame_encryptor_(frame_encryptor),
>+      require_frame_encryption_(require_frame_encryption) {}
> 
> RTPSenderVideo::~RTPSenderVideo() {}
> 
>@@ -142,18 +153,18 @@ VideoCodecType RTPSenderVideo::VideoCodecType() const {
> 
> // Static.
> RtpUtility::Payload* RTPSenderVideo::CreateVideoPayload(
>-    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
>+    absl::string_view payload_name,
>     int8_t payload_type) {
>   enum VideoCodecType video_type = kVideoCodecGeneric;
>-  if (RtpUtility::StringCompare(payload_name, "VP8", 3)) {
>+  if (absl::EqualsIgnoreCase(payload_name, "VP8")) {
>     video_type = kVideoCodecVP8;
>-  } else if (RtpUtility::StringCompare(payload_name, "VP9", 3)) {
>+  } else if (absl::EqualsIgnoreCase(payload_name, "VP9")) {
>     video_type = kVideoCodecVP9;
>-  } else if (RtpUtility::StringCompare(payload_name, "H264", 4)) {
>+  } else if (absl::EqualsIgnoreCase(payload_name, "H264")) {
>     video_type = kVideoCodecH264;
>-  } else if (RtpUtility::StringCompare(payload_name, "I420", 4)) {
>+  } else if (absl::EqualsIgnoreCase(payload_name, "I420")) {
>     video_type = kVideoCodecGeneric;
>-  } else if (RtpUtility::StringCompare(payload_name, "stereo", 6)) {
>+  } else if (absl::EqualsIgnoreCase(payload_name, "stereo")) {
>     video_type = kVideoCodecGeneric;
>   } else {
>     video_type = kVideoCodecGeneric;
>@@ -387,20 +398,19 @@ bool RTPSenderVideo::SendVideo(enum VideoCodecType video_type,
>   int packet_capacity = rtp_sender_->MaxRtpPacketSize() - fec_packet_overhead -
>                         (rtp_sender_->RtxStatus() ? kRtxHeaderSize : 0);
> 
>-  auto create_packet = [&] {
>-    std::unique_ptr<RtpPacketToSend> rtp_packet = rtp_sender_->AllocatePacket();
>-    RTC_DCHECK_LE(packet_capacity, rtp_packet->capacity());
>-
>-    rtp_packet->SetPayloadType(payload_type);
>-    rtp_packet->SetTimestamp(rtp_timestamp);
>-    rtp_packet->set_capture_time_ms(capture_time_ms);
>-    return rtp_packet;
>-  };
>+  std::unique_ptr<RtpPacketToSend> single_packet =
>+      rtp_sender_->AllocatePacket();
>+  RTC_DCHECK_LE(packet_capacity, single_packet->capacity());
>+  single_packet->SetPayloadType(payload_type);
>+  single_packet->SetTimestamp(rtp_timestamp);
>+  single_packet->set_capture_time_ms(capture_time_ms);
> 
>-  auto first_packet = create_packet();
>-  auto middle_packet = absl::make_unique<RtpPacketToSend>(*first_packet);
>-  auto last_packet = absl::make_unique<RtpPacketToSend>(*first_packet);
>+  auto first_packet = absl::make_unique<RtpPacketToSend>(*single_packet);
>+  auto middle_packet = absl::make_unique<RtpPacketToSend>(*single_packet);
>+  auto last_packet = absl::make_unique<RtpPacketToSend>(*single_packet);
>   // Simplest way to estimate how much extensions would occupy is to set them.
>+  AddRtpHeaderExtensions(*video_header, frame_type, set_video_rotation,
>+                         /*first=*/true, /*last=*/true, single_packet.get());
>   AddRtpHeaderExtensions(*video_header, frame_type, set_video_rotation,
>                          /*first=*/true, /*last=*/false, first_packet.get());
>   AddRtpHeaderExtensions(*video_header, frame_type, set_video_rotation,
>@@ -408,12 +418,17 @@ bool RTPSenderVideo::SendVideo(enum VideoCodecType video_type,
>   AddRtpHeaderExtensions(*video_header, frame_type, set_video_rotation,
>                          /*first=*/false, /*last=*/true, last_packet.get());
> 
>+  RTC_DCHECK_GT(packet_capacity, single_packet->headers_size());
>   RTC_DCHECK_GT(packet_capacity, first_packet->headers_size());
>   RTC_DCHECK_GT(packet_capacity, middle_packet->headers_size());
>   RTC_DCHECK_GT(packet_capacity, last_packet->headers_size());
>   RtpPacketizer::PayloadSizeLimits limits;
>   limits.max_payload_len = packet_capacity - middle_packet->headers_size();
> 
>+  RTC_DCHECK_GE(single_packet->headers_size(), middle_packet->headers_size());
>+  limits.single_packet_reduction_len =
>+      single_packet->headers_size() - middle_packet->headers_size();
>+
>   RTC_DCHECK_GE(first_packet->headers_size(), middle_packet->headers_size());
>   limits.first_packet_reduction_len =
>       first_packet->headers_size() - middle_packet->headers_size();
>@@ -424,9 +439,42 @@ bool RTPSenderVideo::SendVideo(enum VideoCodecType video_type,
> 
>   RTPVideoHeader minimized_video_header;
>   const RTPVideoHeader* packetize_video_header = video_header;
>-  if (first_packet->HasExtension<RtpGenericFrameDescriptorExtension>() &&
>-      MinimizeDescriptor(*video_header, &minimized_video_header)) {
>-    packetize_video_header = &minimized_video_header;
>+  rtc::ArrayView<const uint8_t> generic_descriptor_raw =
>+      first_packet->GetRawExtension<RtpGenericFrameDescriptorExtension>();
>+  if (!generic_descriptor_raw.empty()) {
>+    if (MinimizeDescriptor(*video_header, &minimized_video_header)) {
>+      packetize_video_header = &minimized_video_header;
>+    }
>+  }
>+
>+  // TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
>+  rtc::Buffer encrypted_video_payload;
>+  if (frame_encryptor_ != nullptr) {
>+    if (generic_descriptor_raw.empty()) {
>+      return false;
>+    }
>+
>+    const size_t max_ciphertext_size =
>+        frame_encryptor_->GetMaxCiphertextByteSize(cricket::MEDIA_TYPE_VIDEO,
>+                                                   payload_size);
>+    encrypted_video_payload.SetSize(max_ciphertext_size);
>+
>+    size_t bytes_written = 0;
>+    if (frame_encryptor_->Encrypt(
>+            cricket::MEDIA_TYPE_VIDEO, first_packet->Ssrc(),
>+            /*additional_data=*/nullptr,
>+            rtc::MakeArrayView(payload_data, payload_size),
>+            encrypted_video_payload, &bytes_written) != 0) {
>+      return false;
>+    }
>+
>+    encrypted_video_payload.SetSize(bytes_written);
>+    payload_data = encrypted_video_payload.data();
>+    payload_size = encrypted_video_payload.size();
>+  } else if (require_frame_encryption_) {
>+    RTC_LOG(LS_WARNING)
>+        << "No FrameEncryptor is attached to this video sending stream but "
>+        << "one is required since require_frame_encryptor is set";
>   }
> 
>   std::unique_ptr<RtpPacketizer> packetizer = RtpPacketizer::Create(
>@@ -447,16 +495,9 @@ bool RTPSenderVideo::SendVideo(enum VideoCodecType video_type,
>     int expected_payload_capacity;
>     // Choose right packet template:
>     if (num_packets == 1) {
>-      // No prepared template, create a new packet.
>-      packet = create_packet();
>-      AddRtpHeaderExtensions(*video_header, frame_type, set_video_rotation,
>-                             /*first=*/true, /*last=*/true, packet.get());
>-      // TODO(bugs.webrtc.org/7990): Revisit this case when two byte header
>-      // extension are implemented because then single packet might need more
>-      // space for extensions than sum of first and last packet reductions.
>-      expected_payload_capacity = limits.max_payload_len -
>-                                  limits.first_packet_reduction_len -
>-                                  limits.last_packet_reduction_len;
>+      packet = std::move(single_packet);
>+      expected_payload_capacity =
>+          limits.max_payload_len - limits.single_packet_reduction_len;
>     } else if (i == 0) {
>       packet = std::move(first_packet);
>       expected_payload_capacity =
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
>index ce7be16149c3b37a44d74693117d4dc4ea639c9a..d3a898b40a306a7dbc4106029c99de61fea539e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
>@@ -14,6 +14,7 @@
> #include <map>
> #include <memory>
> 
>+#include "absl/strings/string_view.h"
> #include "absl/types/optional.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/include/flexfec_sender.h"
>@@ -29,6 +30,8 @@
> #include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
>+
>+class FrameEncryptorInterface;
> class RtpPacketizer;
> class RtpPacketToSend;
> 
>@@ -38,14 +41,15 @@ class RTPSenderVideo {
> 
>   RTPSenderVideo(Clock* clock,
>                  RTPSender* rtpSender,
>-                 FlexfecSender* flexfec_sender);
>+                 FlexfecSender* flexfec_sender,
>+                 FrameEncryptorInterface* frame_encryptor,
>+                 bool require_frame_encryption);
>   virtual ~RTPSenderVideo();
> 
>   virtual enum VideoCodecType VideoCodecType() const;
> 
>-  static RtpUtility::Payload* CreateVideoPayload(
>-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
>-      int8_t payload_type);
>+  static RtpUtility::Payload* CreateVideoPayload(absl::string_view payload_name,
>+                                                 int8_t payload_type);
> 
>   bool SendVideo(enum VideoCodecType video_type,
>                  FrameType frame_type,
>@@ -158,6 +162,13 @@ class RTPSenderVideo {
>       RTC_GUARDED_BY(stats_crit_);
> 
>   OneTimeEvent first_frame_sent_;
>+
>+  // E2EE Custom Video Frame Encryptor (optional)
>+  FrameEncryptorInterface* const frame_encryptor_ = nullptr;
>+  // If set to true will require all outgoing frames to pass through an
>+  // initialized frame_encryptor_ before being sent out of the network.
>+  // Otherwise these payloads will be dropped.
>+  bool require_frame_encryption_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
>index 228572f8d169c21cb0b76d6ba04539cee4647cf3..44c671f5067eaf37bfadf8f960afb8fc8c53cf81 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
>@@ -10,9 +10,19 @@
> 
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> 
>+#include <assert.h>
>+#include <stddef.h>
>+
>+#include "api/array_view.h"
>+#include "api/video/video_content_type.h"
>+#include "api/video/video_frame_marking.h"
>+#include "api/video/video_rotation.h"
>+#include "api/video/video_timing.h"
> #include "modules/rtp_rtcp/include/rtp_cvo.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
>+#include "modules/video_coding/codecs/interface/common_constants.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/stringutils.h"
> 
>@@ -33,10 +43,6 @@ enum {
>  * Misc utility routines
>  */
> 
>-bool StringCompare(const char* str1, const char* str2, const uint32_t length) {
>-  return _strnicmp(str1, str2, length) == 0;
>-}
>-
> size_t Word32Align(size_t size) {
>   uint32_t remainder = size % 4;
>   if (remainder != 0)
>@@ -501,6 +507,10 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
>           RTC_LOG(WARNING)
>               << "RtpGenericFrameDescriptor unsupported by rtp header parser.";
>           break;
>+        case kRtpExtensionColorSpace:
>+          RTC_LOG(WARNING)
>+              << "RtpExtensionColorSpace unsupported by rtp header parser.";
>+          break;
>         case kRtpExtensionNone:
>         case kRtpExtensionNumberOfExtensions: {
>           RTC_NOTREACHED() << "Invalid extension type: " << type;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.h
>index 762f964a2e2f31eaace15485960f3891e3db96bf..408517481bb495e5424eb66cf0d078e5d24cb803 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.h
>@@ -11,14 +11,14 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
> 
>-#include <cstring>
>-#include <map>
>+#include <stdint.h>
>+#include <algorithm>
> 
>-#include "modules/rtp_rtcp/include/receive_statistics.h"
>+#include "absl/strings/string_view.h"
>+#include "api/rtp_headers.h"
>+#include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
>-#include "rtc_base/deprecation.h"
> 
> namespace webrtc {
> 
>@@ -27,16 +27,15 @@ const uint8_t kRtpMarkerBitMask = 0x80;
> namespace RtpUtility {
> 
> struct Payload {
>-  Payload(const char* name, const PayloadUnion& pu) : typeSpecific(pu) {
>-    std::strncpy(this->name, name, sizeof(this->name) - 1);
>-    this->name[sizeof(this->name) - 1] = '\0';
>+  Payload(absl::string_view payload_name, const PayloadUnion& pu)
>+      : typeSpecific(pu) {
>+    size_t clipped_size = payload_name.copy(name, sizeof(name) - 1);
>+    name[clipped_size] = '\0';
>   }
>   char name[RTP_PAYLOAD_NAME_SIZE];
>   PayloadUnion typeSpecific;
> };
> 
>-bool StringCompare(const char* str1, const char* str2, const uint32_t length);
>-
> // Round up to the nearest size that is a multiple of 4.
> size_t Word32Align(size_t size);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.cc
>index a3ee8baa83e13afbd55f528901c2903bfcf8faf7..bb9413ddd5e28a89f39d77c3452ab1be3ba6165b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.cc
>@@ -12,7 +12,7 @@
> 
> namespace webrtc {
> 
>-RTPVideoHeader::RTPVideoHeader() : playout_delay(), video_timing() {}
>+RTPVideoHeader::RTPVideoHeader() : video_timing() {}
> RTPVideoHeader::RTPVideoHeader(const RTPVideoHeader& other) = default;
> RTPVideoHeader::~RTPVideoHeader() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.h
>index 288b7d03a0260e52f3de0e12597c4465ec66d0ce..1c75f539a1e86bb63ff1a717e0e1ab256da473b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.h
>@@ -10,8 +10,12 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
> 
>+#include <cstdint>
>+
> #include "absl/container/inlined_vector.h"
>+#include "absl/types/optional.h"
> #include "absl/types/variant.h"
>+#include "api/video/video_codec_type.h"
> #include "api/video/video_content_type.h"
> #include "api/video/video_frame_marking.h"
> #include "api/video/video_rotation.h"
>@@ -56,7 +60,7 @@ struct RTPVideoHeader {
>   uint8_t simulcastIdx = 0;
>   VideoCodecType codec = VideoCodecType::kVideoCodecGeneric;
> 
>-  PlayoutDelay playout_delay;
>+  PlayoutDelay playout_delay = {-1, -1};
>   VideoSendTiming video_timing;
>   FrameMarking frame_marking;
>   RTPVideoTypeHeader video_type_header;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.cc
>index 6ac280a61db7e93e757a8cf7a5d2ce800ca858d5..e65329d06b78052b204053b2fd9da457160de03a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.cc
>@@ -12,6 +12,7 @@
> 
> #include <algorithm>
> 
>+#include "rtc_base/checks.h"
> #include "rtc_base/timeutils.h"
> 
> namespace webrtc {
>@@ -22,20 +23,27 @@ inline int64_t DivideRoundToNearest(int64_t x, uint32_t y) {
>   return (x + y / 2) / y;
> }
> 
>-int64_t NtpOffsetUs() {
>+int64_t NtpOffsetMsCalledOnce() {
>   constexpr int64_t kNtpJan1970Sec = 2208988800;
>-  int64_t clock_time = rtc::TimeMicros();
>-  int64_t utc_time = rtc::TimeUTCMicros();
>-  return utc_time - clock_time + kNtpJan1970Sec * rtc::kNumMicrosecsPerSec;
>+  int64_t clock_time = rtc::TimeMillis();
>+  int64_t utc_time = rtc::TimeUTCMillis();
>+  return utc_time - clock_time + kNtpJan1970Sec * rtc::kNumMillisecsPerSec;
> }
> 
> }  // namespace
> 
>-NtpTime TimeMicrosToNtp(int64_t time_us) {
>+int64_t NtpOffsetMs() {
>   // Calculate the offset once.
>-  static int64_t ntp_offset_us = NtpOffsetUs();
>+  static int64_t ntp_offset_ms = NtpOffsetMsCalledOnce();
>+  return ntp_offset_ms;
>+}
> 
>-  int64_t time_ntp_us = time_us + ntp_offset_us;
>+NtpTime TimeMicrosToNtp(int64_t time_us) {
>+  // Since this doesn't return a wallclock time, but only NTP representation
>+  // of rtc::TimeMillis() clock, the exact offset doesn't matter.
>+  // To simplify conversions between NTP and RTP time, this offset is
>+  // limited to milliseconds in resolution.
>+  int64_t time_ntp_us = time_us + NtpOffsetMs() * 1000;
>   RTC_DCHECK_GE(time_ntp_us, 0);  // Time before year 1900 is unsupported.
> 
>   // TODO(danilchap): Convert both seconds and fraction together using int128
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.h
>index 672722c07f99ac0fb72773177600e27a509e0194..94b914310ccf20dcab6ca60448b29a9ae1f39243 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.h
>@@ -22,13 +22,14 @@ namespace webrtc {
> // difference of the passed values.
> // As a result TimeMicrosToNtp(rtc::TimeMicros()) doesn't guarantee to match
> // system time.
>+// However, TimeMicrosToNtp Guarantees that returned NtpTime will be offsetted
>+// from rtc::TimeMicros() by integral number of milliseconds.
>+// Use NtpOffsetMs() to get that offset value.
> NtpTime TimeMicrosToNtp(int64_t time_us);
> 
>-// Converts NTP timestamp to RTP timestamp.
>-inline uint32_t NtpToRtp(NtpTime ntp, uint32_t freq) {
>-  uint32_t tmp = (static_cast<uint64_t>(ntp.fractions()) * freq) >> 32;
>-  return ntp.seconds() * freq + tmp;
>-}
>+// Difference between Ntp time and local relative time returned by
>+// rtc::TimeMicros()
>+int64_t NtpOffsetMs();
> 
> // Helper function for compact ntp representation:
> // RFC 3550, Section 4. Time Format.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
>index 8aa4530211353c5ad584b4a9c2ff795a49cab0a6..315a4c21209eadbe2baf82363d38f49a14318ea8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/rtp_rtcp/source/tmmbr_help.h"
> 
>+#include <stddef.h>
> #include <algorithm>
> #include <limits>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
>index 91aeaf464101c2cfc0b36a832e90918d074a7e06..bf86f65222c7e092cc848b46e42c0cb4665d17c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
>@@ -11,7 +11,9 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
> #define MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
> 
>+#include <stdint.h>
> #include <vector>
>+
> #include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.cc
>index e5777edab8504b1740b164a6b19dff9ab84d60c5..56dae29450e950edd56a75f330e3b059859ca3be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.cc
>@@ -10,12 +10,15 @@
> 
> #include "modules/rtp_rtcp/source/ulpfec_generator.h"
> 
>+#include <string.h>
>+#include <cstdint>
> #include <memory>
> #include <utility>
> 
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
>+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.h
>index efc753f4c74132331ae6c5357ccccc2b9ece9634..74a1d80256ea9c9c3b96564cb30d7abaaf197891 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.h
>@@ -11,10 +11,13 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_ULPFEC_GENERATOR_H_
> #define MODULES_RTP_RTCP_SOURCE_ULPFEC_GENERATOR_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <list>
> #include <memory>
> #include <vector>
> 
>+#include "modules/include/module_fec_types.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
>index c54d3cdd8f90b1f34124c9fd489e4b183a0ab181..22af7e7736904bb24d9b83767cbb45bbc973c95f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
>@@ -10,11 +10,12 @@
> 
> #include "modules/rtp_rtcp/source/ulpfec_header_reader_writer.h"
> 
>-#include <utility>
>+#include <string.h>
> 
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/scoped_ref_ptr.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h
>index fc83afdf0cc025a92966ba355f6de6a6cb744bc1..a8bb737dbb58da05eb2dea68e1e5f26857c20ff8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h
>@@ -11,6 +11,9 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_ULPFEC_HEADER_READER_WRITER_H_
> #define MODULES_RTP_RTCP_SOURCE_ULPFEC_HEADER_READER_WRITER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
>+
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
>index eb09c95813413d57595ebd94ab899684ab838dd4..7da6b88a007cf5ddefdca8c218ebb1975f86f66d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
>@@ -10,12 +10,13 @@
> 
> #include "modules/rtp_rtcp/source/ulpfec_receiver_impl.h"
> 
>+#include <string.h>
> #include <memory>
> #include <utility>
> 
> #include "modules/rtp_rtcp/source/byte_io.h"
>-#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/scoped_ref_ptr.h"
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.h
>index 96367dc7aa821edcc856e7eda61ee3c35439faf8..0943266161802c7842dd246084a6f9c507887291 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.h
>@@ -11,9 +11,12 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_ULPFEC_RECEIVER_IMPL_H_
> #define MODULES_RTP_RTCP_SOURCE_ULPFEC_RECEIVER_IMPL_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> #include <vector>
> 
>+#include "api/rtp_headers.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/include/ulpfec_receiver.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/BUILD.gn
>index 32da10950f293e626a92f97f68f0af55f7adc028..1d780b0e4a770fd1eff3565ef11e6449572f68da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/BUILD.gn
>@@ -26,7 +26,6 @@ rtc_static_library("utility") {
> 
>   deps = [
>     "..:module_api",
>-    "../..:webrtc_common",
>     "../../common_audio",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.cc
>index 97272da2bf10a76e32baf105e34ab7ce3f518e90..20f26d927a39463898572ba9f5f877126a54d796 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.cc
>@@ -10,6 +10,8 @@
> 
> #include "modules/utility/source/process_thread_impl.h"
> 
>+#include <string>
>+
> #include "modules/include/module.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/task_queue.h"
>@@ -42,9 +44,7 @@ std::unique_ptr<ProcessThread> ProcessThread::Create(const char* thread_name) {
> }
> 
> ProcessThreadImpl::ProcessThreadImpl(const char* thread_name)
>-    : wake_up_(EventWrapper::Create()),
>-      stop_(false),
>-      thread_name_(thread_name) {}
>+    : stop_(false), thread_name_(thread_name) {}
> 
> ProcessThreadImpl::~ProcessThreadImpl() {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>@@ -83,7 +83,7 @@ void ProcessThreadImpl::Stop() {
>     stop_ = true;
>   }
> 
>-  wake_up_->Set();
>+  wake_up_.Set();
> 
>   thread_->Stop();
>   stop_ = false;
>@@ -102,7 +102,7 @@ void ProcessThreadImpl::WakeUp(Module* module) {
>         m.next_callback = kCallProcessImmediately;
>     }
>   }
>-  wake_up_->Set();
>+  wake_up_.Set();
> }
> 
> void ProcessThreadImpl::PostTask(std::unique_ptr<rtc::QueuedTask> task) {
>@@ -111,7 +111,7 @@ void ProcessThreadImpl::PostTask(std::unique_ptr<rtc::QueuedTask> task) {
>     rtc::CritScope lock(&lock_);
>     queue_.push(task.release());
>   }
>-  wake_up_->Set();
>+  wake_up_.Set();
> }
> 
> void ProcessThreadImpl::RegisterModule(Module* module,
>@@ -145,7 +145,7 @@ void ProcessThreadImpl::RegisterModule(Module* module,
>   // Wake the thread calling ProcessThreadImpl::Process() to update the
>   // waiting time. The waiting time for the just registered module may be
>   // shorter than all other registered modules.
>-  wake_up_->Set();
>+  wake_up_.Set();
> }
> 
> void ProcessThreadImpl::DeRegisterModule(Module* module) {
>@@ -215,7 +215,7 @@ bool ProcessThreadImpl::Process() {
> 
>   int64_t time_to_wait = next_checkpoint - rtc::TimeMillis();
>   if (time_to_wait > 0)
>-    wake_up_->Wait(static_cast<unsigned long>(time_to_wait));
>+    wake_up_.Wait(static_cast<int>(time_to_wait));
> 
>   return true;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.h
>index c1f6ed44475d6d7485704247eb6bd0dba2581ff1..fff21d04619c029641a690ccb47dd63f27f7eff2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.h
>@@ -11,16 +11,19 @@
> #ifndef MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_
> #define MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_
> 
>+#include <stdint.h>
> #include <list>
> #include <memory>
> #include <queue>
> 
>+#include "modules/include/module.h"
> #include "modules/utility/include/process_thread.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/location.h"
> #include "rtc_base/platform_thread.h"
>+#include "rtc_base/task_queue.h"
> #include "rtc_base/thread_checker.h"
>-#include "system_wrappers/include/event_wrapper.h"
> 
> namespace webrtc {
> 
>@@ -72,7 +75,7 @@ class ProcessThreadImpl : public ProcessThread {
>   rtc::CriticalSection lock_;  // Used to guard modules_, tasks_ and stop_.
> 
>   rtc::ThreadChecker thread_checker_;
>-  const std::unique_ptr<EventWrapper> wake_up_;
>+  rtc::Event wake_up_;
>   // TODO(pbos): Remove unique_ptr and stop recreating the thread.
>   std::unique_ptr<rtc::PlatformThread> thread_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl_unittest.cc
>index d5926b26c3e2ca4b07722663cac6bb0c46e15c70..19157963a3bbbf9f93d78dc33ba055b4299d8ed7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl_unittest.cc
>@@ -42,14 +42,14 @@ class MockModule : public Module {
> 
> class RaiseEventTask : public rtc::QueuedTask {
>  public:
>-  RaiseEventTask(EventWrapper* event) : event_(event) {}
>+  RaiseEventTask(rtc::Event* event) : event_(event) {}
>   bool Run() override {
>     event_->Set();
>     return true;
>   }
> 
>  private:
>-  EventWrapper* event_;
>+  rtc::Event* event_;
> };
> 
> ACTION_P(SetEvent, event) {
>@@ -83,19 +83,19 @@ TEST(ProcessThreadImpl, ProcessCall) {
>   ProcessThreadImpl thread("ProcessThread");
>   thread.Start();
> 
>-  std::unique_ptr<EventWrapper> event(EventWrapper::Create());
>+  rtc::Event event;
> 
>   MockModule module;
>   EXPECT_CALL(module, TimeUntilNextProcess())
>       .WillOnce(Return(0))
>       .WillRepeatedly(Return(1));
>   EXPECT_CALL(module, Process())
>-      .WillOnce(DoAll(SetEvent(event.get()), Return()))
>+      .WillOnce(DoAll(SetEvent(&event), Return()))
>       .WillRepeatedly(Return());
>   EXPECT_CALL(module, ProcessThreadAttached(&thread)).Times(1);
> 
>   thread.RegisterModule(&module, RTC_FROM_HERE);
>-  EXPECT_EQ(kEventSignaled, event->Wait(kEventWaitTimeout));
>+  EXPECT_TRUE(event.Wait(kEventWaitTimeout));
> 
>   EXPECT_CALL(module, ProcessThreadAttached(nullptr)).Times(1);
>   thread.Stop();
>@@ -105,21 +105,21 @@ TEST(ProcessThreadImpl, ProcessCall) {
> // call to Start().
> TEST(ProcessThreadImpl, ProcessCall2) {
>   ProcessThreadImpl thread("ProcessThread");
>-  std::unique_ptr<EventWrapper> event(EventWrapper::Create());
>+  rtc::Event event;
> 
>   MockModule module;
>   EXPECT_CALL(module, TimeUntilNextProcess())
>       .WillOnce(Return(0))
>       .WillRepeatedly(Return(1));
>   EXPECT_CALL(module, Process())
>-      .WillOnce(DoAll(SetEvent(event.get()), Return()))
>+      .WillOnce(DoAll(SetEvent(&event), Return()))
>       .WillRepeatedly(Return());
> 
>   thread.RegisterModule(&module, RTC_FROM_HERE);
> 
>   EXPECT_CALL(module, ProcessThreadAttached(&thread)).Times(1);
>   thread.Start();
>-  EXPECT_EQ(kEventSignaled, event->Wait(kEventWaitTimeout));
>+  EXPECT_TRUE(event.Wait(kEventWaitTimeout));
> 
>   EXPECT_CALL(module, ProcessThreadAttached(nullptr)).Times(1);
>   thread.Stop();
>@@ -129,7 +129,7 @@ TEST(ProcessThreadImpl, ProcessCall2) {
> // After unregistration, we should not receive any further callbacks.
> TEST(ProcessThreadImpl, Deregister) {
>   ProcessThreadImpl thread("ProcessThread");
>-  std::unique_ptr<EventWrapper> event(EventWrapper::Create());
>+  rtc::Event event;
> 
>   int process_count = 0;
>   MockModule module;
>@@ -137,8 +137,7 @@ TEST(ProcessThreadImpl, Deregister) {
>       .WillOnce(Return(0))
>       .WillRepeatedly(Return(1));
>   EXPECT_CALL(module, Process())
>-      .WillOnce(
>-          DoAll(SetEvent(event.get()), Increment(&process_count), Return()))
>+      .WillOnce(DoAll(SetEvent(&event), Increment(&process_count), Return()))
>       .WillRepeatedly(DoAll(Increment(&process_count), Return()));
> 
>   thread.RegisterModule(&module, RTC_FROM_HERE);
>@@ -146,7 +145,7 @@ TEST(ProcessThreadImpl, Deregister) {
>   EXPECT_CALL(module, ProcessThreadAttached(&thread)).Times(1);
>   thread.Start();
> 
>-  EXPECT_EQ(kEventSignaled, event->Wait(kEventWaitTimeout));
>+  EXPECT_TRUE(event.Wait(kEventWaitTimeout));
> 
>   EXPECT_CALL(module, ProcessThreadAttached(nullptr)).Times(1);
>   thread.DeRegisterModule(&module);
>@@ -155,7 +154,7 @@ TEST(ProcessThreadImpl, Deregister) {
>   int count_after_deregister = process_count;
> 
>   // We shouldn't get any more callbacks.
>-  EXPECT_EQ(kEventTimeout, event->Wait(20));
>+  EXPECT_FALSE(event.Wait(20));
>   EXPECT_EQ(count_after_deregister, process_count);
>   thread.Stop();
> }
>@@ -167,7 +166,7 @@ void ProcessCallAfterAFewMs(int64_t milliseconds) {
>   ProcessThreadImpl thread("ProcessThread");
>   thread.Start();
> 
>-  std::unique_ptr<EventWrapper> event(EventWrapper::Create());
>+  rtc::Event event;
> 
>   MockModule module;
>   int64_t start_time = 0;
>@@ -176,8 +175,7 @@ void ProcessCallAfterAFewMs(int64_t milliseconds) {
>       .WillOnce(DoAll(SetTimestamp(&start_time), Return(milliseconds)))
>       .WillRepeatedly(Return(milliseconds));
>   EXPECT_CALL(module, Process())
>-      .WillOnce(
>-          DoAll(SetTimestamp(&called_time), SetEvent(event.get()), Return()))
>+      .WillOnce(DoAll(SetTimestamp(&called_time), SetEvent(&event), Return()))
>       .WillRepeatedly(Return());
> 
>   EXPECT_CALL(module, ProcessThreadAttached(&thread)).Times(1);
>@@ -185,7 +183,7 @@ void ProcessCallAfterAFewMs(int64_t milliseconds) {
> 
>   // Add a buffer of 50ms due to slowness of some trybots
>   // (e.g. win_drmemory_light)
>-  EXPECT_EQ(kEventSignaled, event->Wait(milliseconds + 50));
>+  EXPECT_TRUE(event.Wait(milliseconds + 50));
> 
>   EXPECT_CALL(module, ProcessThreadAttached(nullptr)).Times(1);
>   thread.Stop();
>@@ -230,7 +228,7 @@ TEST(ProcessThreadImpl, DISABLED_Process50Times) {
>   ProcessThreadImpl thread("ProcessThread");
>   thread.Start();
> 
>-  std::unique_ptr<EventWrapper> event(EventWrapper::Create());
>+  rtc::Event event;
> 
>   MockModule module;
>   int callback_count = 0;
>@@ -242,7 +240,7 @@ TEST(ProcessThreadImpl, DISABLED_Process50Times) {
>   EXPECT_CALL(module, ProcessThreadAttached(&thread)).Times(1);
>   thread.RegisterModule(&module, RTC_FROM_HERE);
> 
>-  EXPECT_EQ(kEventTimeout, event->Wait(1000));
>+  EXPECT_TRUE(event.Wait(1000));
> 
>   EXPECT_CALL(module, ProcessThreadAttached(nullptr)).Times(1);
>   thread.Stop();
>@@ -261,8 +259,8 @@ TEST(ProcessThreadImpl, WakeUp) {
>   ProcessThreadImpl thread("ProcessThread");
>   thread.Start();
> 
>-  std::unique_ptr<EventWrapper> started(EventWrapper::Create());
>-  std::unique_ptr<EventWrapper> called(EventWrapper::Create());
>+  rtc::Event started;
>+  rtc::Event called;
> 
>   MockModule module;
>   int64_t start_time;
>@@ -276,20 +274,19 @@ TEST(ProcessThreadImpl, WakeUp) {
>   // The second time TimeUntilNextProcess is then called, is after Process
>   // has been called and we don't expect any more calls.
>   EXPECT_CALL(module, TimeUntilNextProcess())
>-      .WillOnce(DoAll(SetTimestamp(&start_time), SetEvent(started.get()),
>-                      Return(1000)))
>+      .WillOnce(
>+          DoAll(SetTimestamp(&start_time), SetEvent(&started), Return(1000)))
>       .WillOnce(Return(1000));
>   EXPECT_CALL(module, Process())
>-      .WillOnce(
>-          DoAll(SetTimestamp(&called_time), SetEvent(called.get()), Return()))
>+      .WillOnce(DoAll(SetTimestamp(&called_time), SetEvent(&called), Return()))
>       .WillRepeatedly(Return());
> 
>   EXPECT_CALL(module, ProcessThreadAttached(&thread)).Times(1);
>   thread.RegisterModule(&module, RTC_FROM_HERE);
> 
>-  EXPECT_EQ(kEventSignaled, started->Wait(kEventWaitTimeout));
>+  EXPECT_TRUE(started.Wait(kEventWaitTimeout));
>   thread.WakeUp(&module);
>-  EXPECT_EQ(kEventSignaled, called->Wait(kEventWaitTimeout));
>+  EXPECT_TRUE(called.Wait(kEventWaitTimeout));
> 
>   EXPECT_CALL(module, ProcessThreadAttached(nullptr)).Times(1);
>   thread.Stop();
>@@ -304,11 +301,11 @@ TEST(ProcessThreadImpl, WakeUp) {
> // thread.
> TEST(ProcessThreadImpl, PostTask) {
>   ProcessThreadImpl thread("ProcessThread");
>-  std::unique_ptr<EventWrapper> task_ran(EventWrapper::Create());
>-  std::unique_ptr<RaiseEventTask> task(new RaiseEventTask(task_ran.get()));
>+  rtc::Event task_ran;
>+  std::unique_ptr<RaiseEventTask> task(new RaiseEventTask(&task_ran));
>   thread.Start();
>   thread.PostTask(std::move(task));
>-  EXPECT_EQ(kEventSignaled, task_ran->Wait(kEventWaitTimeout));
>+  EXPECT_TRUE(task_ran.Wait(kEventWaitTimeout));
>   thread.Stop();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/BUILD.gn
>index ef6f335e4939d576ce6181e89094aebffc10c4ab..44db2eb477aa28a53fc6b1091baee0cf764bc798 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/BUILD.gn
>@@ -38,6 +38,7 @@ rtc_static_library("video_capture_module") {
>     "../../rtc_base:stringutils",
>     "../../rtc_base/synchronization:rw_lock_wrapper",
>     "../../system_wrappers",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/libyuv",
>   ]
> }
>@@ -190,6 +191,7 @@ if (!build_with_chromium) {
>         "../../common_video:common_video",
>         "../../rtc_base:rtc_base_approved",
>         "../../system_wrappers:system_wrappers",
>+        "../../test:test_support",
>         "../../test:video_test_common",
>         "../utility",
>         "//testing/gtest",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/device_info_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/device_info_impl.cc
>index 2f0a39f2964fb37c703f9864f14ad77a0eb361fa..cdcb5ddc4a82744cc298de512d8204e27a8026b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/device_info_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/device_info_impl.cc
>@@ -11,6 +11,7 @@
> #include <assert.h>
> #include <stdlib.h>
> 
>+#include "absl/strings/match.h"
> #include "modules/video_capture/device_info_impl.h"
> #include "modules/video_capture/video_capture_config.h"
> #include "rtc_base/logging.h"
>@@ -42,14 +43,12 @@ int32_t DeviceInfoImpl::NumberOfCapabilities(const char* deviceUniqueIdUTF8) {
> 
>   _apiLock.AcquireLockShared();
> 
>-  if (_lastUsedDeviceNameLength == strlen((char*)deviceUniqueIdUTF8)) {
>-    // Is it the same device that is asked for again.
>-    if (_strnicmp((char*)_lastUsedDeviceName, (char*)deviceUniqueIdUTF8,
>-                  _lastUsedDeviceNameLength) == 0) {
>-      // yes
>-      _apiLock.ReleaseLockShared();
>-      return static_cast<int32_t>(_captureCapabilities.size());
>-    }
>+  // Is it the same device that is asked for again.
>+  if (absl::EqualsIgnoreCase(
>+          deviceUniqueIdUTF8,
>+          absl::string_view(_lastUsedDeviceName, _lastUsedDeviceNameLength))) {
>+    _apiLock.ReleaseLockShared();
>+    return static_cast<int32_t>(_captureCapabilities.size());
>   }
>   // Need to get exclusive rights to create the new capability map.
>   _apiLock.ReleaseLockShared();
>@@ -66,9 +65,9 @@ int32_t DeviceInfoImpl::GetCapability(const char* deviceUniqueIdUTF8,
> 
>   ReadLockScoped cs(_apiLock);
> 
>-  if ((_lastUsedDeviceNameLength != strlen((char*)deviceUniqueIdUTF8)) ||
>-      (_strnicmp((char*)_lastUsedDeviceName, (char*)deviceUniqueIdUTF8,
>-                 _lastUsedDeviceNameLength) != 0)) {
>+  if (!absl::EqualsIgnoreCase(
>+          deviceUniqueIdUTF8,
>+          absl::string_view(_lastUsedDeviceName, _lastUsedDeviceNameLength))) {
>     _apiLock.ReleaseLockShared();
>     _apiLock.AcquireLockExclusive();
>     if (-1 == CreateCapabilityMap(deviceUniqueIdUTF8)) {
>@@ -100,9 +99,9 @@ int32_t DeviceInfoImpl::GetBestMatchedCapability(
>     return -1;
> 
>   ReadLockScoped cs(_apiLock);
>-  if ((_lastUsedDeviceNameLength != strlen((char*)deviceUniqueIdUTF8)) ||
>-      (_strnicmp((char*)_lastUsedDeviceName, (char*)deviceUniqueIdUTF8,
>-                 _lastUsedDeviceNameLength) != 0)) {
>+  if (!absl::EqualsIgnoreCase(
>+          deviceUniqueIdUTF8,
>+          absl::string_view(_lastUsedDeviceName, _lastUsedDeviceNameLength))) {
>     _apiLock.ReleaseLockShared();
>     _apiLock.AcquireLockExclusive();
>     if (-1 == CreateCapabilityMap(deviceUniqueIdUTF8)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_impl.cc
>index 53afed00b644e895fd2da5e083b328770713044a..235fd3438a216469e78297acade568132574aac7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_impl.cc
>@@ -143,7 +143,7 @@ int32_t VideoCaptureImpl::IncomingFrame(uint8_t* videoFrame,
>   int stride_y = width;
>   int stride_uv = (width + 1) / 2;
>   int target_width = width;
>-  int target_height = height;
>+  int target_height = abs(height);
> 
>   // SetApplyRotation doesn't take any lock. Make a local copy here.
>   bool apply_rotation = apply_rotation_;
>@@ -163,7 +163,7 @@ int32_t VideoCaptureImpl::IncomingFrame(uint8_t* videoFrame,
> 
>   // TODO(nisse): Use a pool?
>   rtc::scoped_refptr<I420Buffer> buffer = I420Buffer::Create(
>-      target_width, abs(target_height), stride_y, stride_uv, stride_uv);
>+      target_width, target_height, stride_y, stride_uv, stride_uv);
> 
>   libyuv::RotationMode rotation_mode = libyuv::kRotate0;
>   if (apply_rotation) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.cc
>index 778fd8761ac19b3961ac25747a5c743ac581157c..ea377a3b05cb6d5d5a454fbef6307e5bcb31b784 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.cc
>@@ -95,6 +95,10 @@ int32_t VideoCaptureDS::Init(const char* deviceUniqueIdUTF8) {
>   }
> 
>   _outputCapturePin = GetOutputPin(_captureFilter, PIN_CATEGORY_CAPTURE);
>+  if (!_outputCapturePin) {
>+    RTC_LOG(LS_INFO) << "Failed to get output capture pin";
>+    return -1;
>+  }
> 
>   // Create the sink filte used for receiving Captured frames.
>   _sinkFilter = new CaptureSinkFilter(SINK_FILTER_NAME, NULL, &hr, *this);
>@@ -109,7 +113,12 @@ int32_t VideoCaptureDS::Init(const char* deviceUniqueIdUTF8) {
>     RTC_LOG(LS_INFO) << "Failed to add the send filter to the graph.";
>     return -1;
>   }
>+
>   _inputSendPin = GetInputPin(_sinkFilter);
>+  if (!_inputSendPin) {
>+    RTC_LOG(LS_INFO) << "Failed to get input send pin";
>+    return -1;
>+  }
> 
>   // Temporary connect here.
>   // This is done so that no one else can use the capture device.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/BUILD.gn
>index e759a4a7e8dd5e635dd7f2cef40e55848d6bacb8..ac359bc64f827712ef25406d7d2cb06f5a9da475 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/BUILD.gn
>@@ -17,8 +17,8 @@ rtc_static_library("encoded_frame") {
>   deps = [
>     ":video_codec_interface",
>     "../../:webrtc_common",
>+    "../../api/video:encoded_image",
>     "../../api/video:video_frame_i420",
>-    "../../common_video:common_video",
>     "../../modules:module_api",
>     "../../modules:module_api_public",
>     "../../modules/video_coding:video_coding_utility",
>@@ -57,6 +57,7 @@ rtc_static_library("nack_module") {
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:rtc_numerics",
>     "../../system_wrappers",
>+    "../../system_wrappers:field_trial",
>     "../utility:utility",
>   ]
> }
>@@ -158,6 +159,7 @@ rtc_static_library("video_coding") {
>     "..:module_api_public",
>     "../..:webrtc_common",
>     "../../api:fec_controller_api",
>+    "../../api/video:builtin_video_bitrate_allocator_factory",
>     "../../api/video:encoded_frame",
>     "../../api/video:video_bitrate_allocator",
>     "../../api/video:video_frame",
>@@ -165,12 +167,14 @@ rtc_static_library("video_coding") {
>     "../../api/video_codecs:video_codecs_api",
>     "../../common_video",
>     "../../rtc_base:checks",
>+    "../../rtc_base:deprecation",
>     "../../rtc_base:rtc_base",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:rtc_numerics",
>     "../../rtc_base:rtc_task_queue",
>     "../../rtc_base:sequenced_task_checker",
>     "../../rtc_base/experiments:alr_experiment",
>+    "../../rtc_base/experiments:jitter_upper_bound_experiment",
>     "../../rtc_base/experiments:rtt_mult_experiment",
>     "../../rtc_base/system:fallthrough",
>     "../../rtc_base/third_party/base64",
>@@ -238,8 +242,6 @@ rtc_source_set("video_coding_utility") {
>     "utility/framerate_controller.h",
>     "utility/ivf_file_writer.cc",
>     "utility/ivf_file_writer.h",
>-    "utility/moving_average.cc",
>-    "utility/moving_average.h",
>     "utility/quality_scaler.cc",
>     "utility/quality_scaler.h",
>     "utility/simulcast_rate_allocator.cc",
>@@ -261,6 +263,7 @@ rtc_source_set("video_coding_utility") {
>     ":video_codec_interface",
>     "..:module_api",
>     "../..:webrtc_common",
>+    "../../api/video:encoded_image",
>     "../../api/video:video_bitrate_allocator",
>     "../../api/video_codecs:video_codecs_api",
>     "../../common_video",
>@@ -302,8 +305,10 @@ rtc_static_library("webrtc_h264") {
>     "../../media:rtc_media_base",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base",
>+    "../../rtc_base/system:rtc_export",
>     "../../system_wrappers:metrics",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/libyuv",
>   ]
> 
>@@ -376,6 +381,7 @@ rtc_static_library("webrtc_multiplex") {
>     ":video_coding_utility",
>     "..:module_api",
>     "../..:webrtc_common",
>+    "../../api/video:encoded_image",
>     "../../api/video:video_frame",
>     "../../api/video:video_frame_i420",
>     "../../api/video_codecs:video_codecs_api",
>@@ -391,22 +397,13 @@ rtc_static_library("webrtc_vp8") {
>   visibility = [ "*" ]
>   poisonous = [ "software_video_codecs" ]
>   sources = [
>-    "codecs/vp8/default_temporal_layers.cc",
>-    "codecs/vp8/default_temporal_layers.h",
>-    "codecs/vp8/include/temporal_layers_checker.h",
>     "codecs/vp8/include/vp8.h",
>-    "codecs/vp8/include/vp8_temporal_layers.h",
>     "codecs/vp8/libvpx_interface.cc",
>     "codecs/vp8/libvpx_interface.h",
>     "codecs/vp8/libvpx_vp8_decoder.cc",
>     "codecs/vp8/libvpx_vp8_decoder.h",
>     "codecs/vp8/libvpx_vp8_encoder.cc",
>     "codecs/vp8/libvpx_vp8_encoder.h",
>-    "codecs/vp8/screenshare_layers.cc",
>-    "codecs/vp8/screenshare_layers.h",
>-    "codecs/vp8/temporal_layers.h",
>-    "codecs/vp8/temporal_layers_checker.cc",
>-    "codecs/vp8/vp8_temporal_layers.cc",
>   ]
> 
>   if (!build_with_chromium && is_clang) {
>@@ -418,14 +415,18 @@ rtc_static_library("webrtc_vp8") {
>     ":codec_globals_headers",
>     ":video_codec_interface",
>     ":video_coding_utility",
>+    ":webrtc_vp8_temporal_layers",
>     "..:module_api",
>     "../..:webrtc_common",
>+    "../../api/video:encoded_image",
>     "../../api/video:video_frame",
>+    "../../api/video_codecs:create_vp8_temporal_layers",
>     "../../api/video_codecs:video_codecs_api",
>     "../../common_video",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:rtc_numerics",
>+    "../../rtc_base/experiments:cpu_speed_experiment",
>     "../../system_wrappers",
>     "../../system_wrappers:field_trial",
>     "../../system_wrappers:metrics",
>@@ -438,6 +439,41 @@ rtc_static_library("webrtc_vp8") {
>   }
> }
> 
>+rtc_static_library("webrtc_vp8_temporal_layers") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "codecs/vp8/default_temporal_layers.cc",
>+    "codecs/vp8/default_temporal_layers.h",
>+    "codecs/vp8/include/temporal_layers_checker.h",
>+    "codecs/vp8/screenshare_layers.cc",
>+    "codecs/vp8/screenshare_layers.h",
>+    "codecs/vp8/temporal_layers.h",
>+    "codecs/vp8/temporal_layers_checker.cc",
>+  ]
>+
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+
>+  deps = [
>+    ":codec_globals_headers",
>+    ":video_codec_interface",
>+    ":video_coding_utility",
>+    "..:module_api",
>+    "../..:webrtc_common",
>+    "../../api/video_codecs:video_codecs_api",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:rtc_numerics",
>+    "../../system_wrappers",
>+    "../../system_wrappers:field_trial",
>+    "../../system_wrappers:metrics",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
> # This target includes VP9 files that may be used for any VP9 codec, internal SW or external HW.
> rtc_static_library("webrtc_vp9_helpers") {
>   sources = [
>@@ -465,19 +501,14 @@ rtc_static_library("webrtc_vp9_helpers") {
> rtc_static_library("webrtc_vp9") {
>   visibility = [ "*" ]
>   poisonous = [ "software_video_codecs" ]
>-  if (rtc_libvpx_build_vp9) {
>-    sources = [
>-      "codecs/vp9/include/vp9.h",
>-      "codecs/vp9/vp9_frame_buffer_pool.cc",
>-      "codecs/vp9/vp9_frame_buffer_pool.h",
>-      "codecs/vp9/vp9_impl.cc",
>-      "codecs/vp9/vp9_impl.h",
>-    ]
>-  } else {
>-    sources = [
>-      "codecs/vp9/vp9_noop.cc",
>-    ]
>-  }
>+  sources = [
>+    "codecs/vp9/include/vp9.h",
>+    "codecs/vp9/vp9.cc",
>+    "codecs/vp9/vp9_frame_buffer_pool.cc",
>+    "codecs/vp9/vp9_frame_buffer_pool.h",
>+    "codecs/vp9/vp9_impl.cc",
>+    "codecs/vp9/vp9_impl.h",
>+  ]
> 
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -574,7 +605,10 @@ if (rtc_include_tests) {
>       ":video_coding",
>       ":video_coding_utility",
>       "../../:webrtc_common",
>+      "../../api:mock_video_decoder",
>+      "../../api:mock_video_encoder",
>       "../../api:simulcast_test_fixture_api",
>+      "../../api/video:encoded_image",
>       "../../api/video:video_frame",
>       "../../api/video:video_frame_i420",
>       "../../api/video_codecs:video_codecs_api",
>@@ -606,6 +640,7 @@ if (rtc_include_tests) {
>       ":webrtc_vp9_helpers",
>       "../..:webrtc_common",
>       "../../api:videocodec_test_fixture_api",
>+      "../../api/video:builtin_video_bitrate_allocator_factory",
>       "../../api/video:video_bitrate_allocator",
>       "../../api/video:video_frame",
>       "../../api/video:video_frame_i420",
>@@ -674,6 +709,7 @@ if (rtc_include_tests) {
>       ":webrtc_vp9_helpers",
>       "../..:webrtc_common",
>       "../../api:videocodec_test_fixture_api",
>+      "../../api/test/video:function_video_factory",
>       "../../api/video_codecs:video_codecs_api",
>       "../../call:video_stream_api",
>       "../../common_video",
>@@ -719,6 +755,7 @@ if (rtc_include_tests) {
>     }
> 
>     deps = [
>+      ":mock_headers",
>       ":video_codecs_test_framework",
>       ":video_coding_utility",
>       ":videocodec_test_impl",
>@@ -730,7 +767,10 @@ if (rtc_include_tests) {
>       "../..:webrtc_common",
>       "../../api:create_videocodec_test_fixture_api",
>       "../../api:mock_video_codec_factory",
>+      "../../api:mock_video_decoder",
>+      "../../api:mock_video_encoder",
>       "../../api:videocodec_test_fixture_api",
>+      "../../api/test/video:function_video_factory",
>       "../../api/video:video_frame",
>       "../../api/video:video_frame_i420",
>       "../../api/video_codecs:rtc_software_fallback_wrappers",
>@@ -739,11 +779,11 @@ if (rtc_include_tests) {
>       "../../media:rtc_h264_profile_id",
>       "../../media:rtc_internal_video_codecs",
>       "../../media:rtc_media_base",
>+      "../../media:rtc_simulcast_encoder_adapter",
>       "../../media:rtc_vp9_profile",
>       "../../rtc_base:rtc_base",
>       "../../test:field_trial",
>       "../../test:fileutils",
>-      "../../test:test_common",
>       "../../test:test_support",
>       "../../test:video_test_common",
>       "../rtp_rtcp:rtp_rtcp_format",
>@@ -774,6 +814,10 @@ if (rtc_include_tests) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>     }
>+
>+    if (rtc_build_libvpx) {
>+      deps += [ rtc_libvpx_dir ]
>+    }
>   }
> 
>   rtc_source_set("video_coding_unittests") {
>@@ -804,13 +848,11 @@ if (rtc_include_tests) {
>       "session_info_unittest.cc",
>       "test/stream_generator.cc",
>       "test/stream_generator.h",
>-      "test/test_util.h",
>       "timing_unittest.cc",
>       "utility/default_video_bitrate_allocator_unittest.cc",
>       "utility/frame_dropper_unittest.cc",
>       "utility/framerate_controller_unittest.cc",
>       "utility/ivf_file_writer_unittest.cc",
>-      "utility/moving_average_unittest.cc",
>       "utility/quality_scaler_unittest.cc",
>       "utility/simulcast_rate_allocator_unittest.cc",
>       "video_codec_initializer_unittest.cc",
>@@ -839,16 +881,23 @@ if (rtc_include_tests) {
>       ":videocodec_test_impl",
>       ":webrtc_h264",
>       ":webrtc_vp8",
>+      ":webrtc_vp8_temporal_layers",
>       ":webrtc_vp9",
>       ":webrtc_vp9_helpers",
>       "..:module_api",
>       "../..:webrtc_common",
>       "../../api:create_simulcast_test_fixture_api",
>+      "../../api:mock_video_decoder",
>+      "../../api:mock_video_encoder",
>       "../../api:simulcast_test_fixture_api",
>       "../../api:videocodec_test_fixture_api",
>+      "../../api/test/video:function_video_factory",
>+      "../../api/video:builtin_video_bitrate_allocator_factory",
>       "../../api/video:video_bitrate_allocator",
>+      "../../api/video:video_bitrate_allocator_factory",
>       "../../api/video:video_frame",
>       "../../api/video:video_frame_i420",
>+      "../../api/video_codecs:create_vp8_temporal_layers",
>       "../../api/video_codecs:video_codecs_api",
>       "../../common_video:common_video",
>       "../../media:rtc_media_base",
>@@ -859,6 +908,7 @@ if (rtc_include_tests) {
>       "../../rtc_base:rtc_numerics",
>       "../../rtc_base:rtc_task_queue",
>       "../../rtc_base:rtc_task_queue_for_test",
>+      "../../rtc_base/experiments:jitter_upper_bound_experiment",
>       "../../system_wrappers",
>       "../../system_wrappers:field_trial",
>       "../../system_wrappers:metrics",
>@@ -870,6 +920,7 @@ if (rtc_include_tests) {
>       "../../test:video_test_support",
>       "../rtp_rtcp:rtp_rtcp_format",
>       "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>     if (rtc_build_libvpx) {
>       deps += [ rtc_libvpx_dir ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
>index 73d45837af77d699f6a796c1dcf51df3216eadeb..e213223ca4f811d71e99b5fc8bf1398c89cc7726 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
>@@ -323,13 +323,11 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
>   // without copying the underlying buffer.
>   if (av_frame_->width != i420_buffer->width() ||
>       av_frame_->height != i420_buffer->height()) {
>-    rtc::scoped_refptr<VideoFrameBuffer> cropped_buf(
>-        new rtc::RefCountedObject<WrappedI420Buffer>(
>-            av_frame_->width, av_frame_->height,
>-            i420_buffer->DataY(), i420_buffer->StrideY(),
>-            i420_buffer->DataU(), i420_buffer->StrideU(),
>-            i420_buffer->DataV(), i420_buffer->StrideV(),
>-            rtc::KeepRefUntilDone(i420_buffer)));
>+    rtc::scoped_refptr<VideoFrameBuffer> cropped_buf = WrapI420Buffer(
>+        av_frame_->width, av_frame_->height, i420_buffer->DataY(),
>+        i420_buffer->StrideY(), i420_buffer->DataU(), i420_buffer->StrideU(),
>+        i420_buffer->DataV(), i420_buffer->StrideV(),
>+        rtc::KeepRefUntilDone(i420_buffer));
>     VideoFrame cropped_frame =
>         VideoFrame::Builder()
>             .set_video_frame_buffer(cropped_buf)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
>index 4effcdbc524fea997727f2be4ef686efd9421e41..f191f2c64032aa97453c0c8199df4245fb6db8c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
>@@ -19,6 +19,7 @@
> #include "third_party/openh264/src/codec/api/svc/codec_def.h"
> #include "third_party/openh264/src/codec/api/svc/codec_ver.h"
> 
>+#include "absl/strings/match.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "modules/video_coding/utility/simulcast_rate_allocator.h"
> #include "modules/video_coding/utility/simulcast_utility.h"
>@@ -167,7 +168,7 @@ H264EncoderImpl::H264EncoderImpl(const cricket::VideoCodec& codec)
>       encoded_image_callback_(nullptr),
>       has_reported_init_(false),
>       has_reported_error_(false) {
>-  RTC_CHECK(cricket::CodecNamesEq(codec.name, cricket::kH264CodecName));
>+  RTC_CHECK(absl::EqualsIgnoreCase(codec.name, cricket::kH264CodecName));
>   std::string packetization_mode_string;
>   if (codec.GetParam(cricket::kH264FmtpPacketizationMode,
>                      &packetization_mode_string) &&
>@@ -310,7 +311,7 @@ int32_t H264EncoderImpl::InitEncode(const VideoCodec* inst,
>   }
> 
>   SimulcastRateAllocator init_allocator(codec_);
>-  BitrateAllocation allocation = init_allocator.GetAllocation(
>+  VideoBitrateAllocation allocation = init_allocator.GetAllocation(
>       codec_.startBitrate * 1000, codec_.maxFramerate);
>   return SetRateAllocation(allocation, codec_.maxFramerate);
> }
>@@ -339,7 +340,7 @@ int32_t H264EncoderImpl::RegisterEncodeCompleteCallback(
> }
> 
> int32_t H264EncoderImpl::SetRateAllocation(
>-    const BitrateAllocation& bitrate,
>+    const VideoBitrateAllocation& bitrate,
>     uint32_t new_framerate) {
>   if (encoders_.empty())
>     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>@@ -534,10 +535,6 @@ int32_t H264EncoderImpl::Encode(const VideoFrame& input_frame,
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-const char* H264EncoderImpl::ImplementationName() const {
>-  return "OpenH264";
>-}
>-
> // Initialization parameters.
> // There are two ways to initialize. There is SEncParamBase (cleared with
> // memset(&p, 0, sizeof(SEncParamBase)) used in Initialize, and SEncParamExt
>@@ -624,14 +621,13 @@ void H264EncoderImpl::ReportError() {
>   has_reported_error_ = true;
> }
> 
>-int32_t H264EncoderImpl::SetChannelParameters(uint32_t packet_loss,
>-                                              int64_t rtt) {
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
>-VideoEncoder::ScalingSettings H264EncoderImpl::GetScalingSettings() const {
>-  return VideoEncoder::ScalingSettings(kLowH264QpThreshold,
>-                                       kHighH264QpThreshold);
>+VideoEncoder::EncoderInfo H264EncoderImpl::GetEncoderInfo() const {
>+  EncoderInfo info;
>+  info.supports_native_handle = false;
>+  info.implementation_name = "OpenH264";
>+  info.scaling_settings =
>+      VideoEncoder::ScalingSettings(kLowH264QpThreshold, kHighH264QpThreshold);
>+  return info;
> }
> 
> void H264EncoderImpl::LayerConfig::SetStreamState(bool send_stream) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h
>index 0d259660f699b6b55ab3b5631bc76274d6d3a138..da32563c9b2de2f7ef2f09484d0f88d1141fcf28 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h
>@@ -70,12 +70,7 @@ class H264EncoderImpl : public H264Encoder {
>                  const CodecSpecificInfo* codec_specific_info,
>                  const std::vector<FrameType>* frame_types) override;
> 
>-  const char* ImplementationName() const override;
>-
>-  VideoEncoder::ScalingSettings GetScalingSettings() const override;
>-
>-  // Unsupported / Do nothing.
>-  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>+  EncoderInfo GetEncoderInfo() const override;
> 
>   // Exposed for testing.
>   H264PacketizationMode PacketizationModeForTesting() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc
>index 6c6fae8c1a0d143aaa12a2731e41a7ec881249f0..3b720b38430b470973ee4454fbdc29cf7b7e7e61 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc
>@@ -13,9 +13,9 @@
> #include "absl/memory/memory.h"
> #include "api/test/create_simulcast_test_fixture.h"
> #include "api/test/simulcast_test_fixture.h"
>+#include "api/test/video/function_video_decoder_factory.h"
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
>-#include "test/function_video_decoder_factory.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/include/h264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/include/h264.h
>index e23818b756b86f3033eb6b953a3688ba3c806a32..f5cebcfe62122c5ef727f3770503b63f7fd4f1ff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/include/h264.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/include/h264.h
>@@ -17,6 +17,7 @@
> 
> #include "media/base/codec.h"
> #include "modules/video_coding/include/video_codec_interface.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>@@ -26,13 +27,13 @@ struct SdpVideoFormat;
> // |rtc_use_h264| build flag is true (if false, this function does nothing).
> // This function should only be called before or during WebRTC initialization
> // and is not thread-safe.
>-void DisableRtcUseH264();
>+RTC_EXPORT void DisableRtcUseH264();
> 
> // Returns a vector with all supported internal H264 profiles that we can
> // negotiate in SDP, in order of preference.
> std::vector<SdpVideoFormat> SupportedH264Codecs();
> 
>-class H264Encoder : public VideoEncoder {
>+class RTC_EXPORT H264Encoder : public VideoEncoder {
>  public:
>   static std::unique_ptr<H264Encoder> Create(const cricket::VideoCodec& codec);
>   // If H.264 is supported (any implementation).
>@@ -41,7 +42,7 @@ class H264Encoder : public VideoEncoder {
>   ~H264Encoder() override {}
> };
> 
>-class H264Decoder : public VideoDecoder {
>+class RTC_EXPORT H264Decoder : public VideoDecoder {
>  public:
>   static std::unique_ptr<H264Decoder> Create();
>   static bool IsSupported();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
>index 14bb6bcc47e63f152f8d1a3af9035ae5e38f75a4..82c14b7fd529c06e46aca010aaf9f6f2c954aa6b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
>@@ -56,7 +56,7 @@ TEST_F(TestH264Impl, MAYBE_EncodeDecode) {
>   ASSERT_TRUE(decoded_frame);
>   EXPECT_GT(I420PSNR(input_frame, decoded_frame.get()), 36);
> 
>-  const ColorSpace color_space = decoded_frame->color_space().value();
>+  const ColorSpace color_space = *decoded_frame->color_space();
>   EXPECT_EQ(ColorSpace::PrimaryID::kInvalid, color_space.primaries());
>   EXPECT_EQ(ColorSpace::TransferID::kInvalid, color_space.transfer());
>   EXPECT_EQ(ColorSpace::MatrixID::kInvalid, color_space.matrix());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/include/i420.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/include/i420.h
>index 580bc16c2f8a9c0345e77f9530c964ff05e4125a..967286f42cb43e0fc36b78aab2a6edc0682ee1be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/include/i420.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/include/i420.h
>@@ -65,10 +65,6 @@ class I420Encoder : public VideoEncoder {
>   // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
>   int Release() override;
> 
>-  int SetChannelParameters(uint32_t /*packetLoss*/, int64_t /*rtt*/) override {
>-    return WEBRTC_VIDEO_CODEC_OK;
>-  }
>-
>  private:
>   static uint8_t* InsertHeader(uint8_t* buffer,
>                                uint16_t width,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h
>index 531d07870e76de1ea47de70a9efbc9dee9c14663..235a36030b078cfaea2bf3108d4295b5eb360096 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h
>@@ -46,11 +46,10 @@ class MultiplexEncoderAdapter : public VideoEncoder {
>              const CodecSpecificInfo* codec_specific_info,
>              const std::vector<FrameType>* frame_types) override;
>   int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
>-  int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>   int SetRateAllocation(const VideoBitrateAllocation& bitrate,
>                         uint32_t new_framerate) override;
>   int Release() override;
>-  const char* ImplementationName() const override;
>+  EncoderInfo GetEncoderInfo() const override;
> 
>   EncodedImageCallback::Result OnEncodedImage(
>       AlphaCodecStream stream_idx,
>@@ -81,6 +80,8 @@ class MultiplexEncoderAdapter : public VideoEncoder {
> 
>   const bool supports_augmented_data_;
>   int augmenting_data_size_ = 0;
>+
>+  EncoderInfo encoder_info_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
>index cc588d0347a58552702135b368526b4a1df73a96..a27bc8d2588345f5048a2c4d79354b41f178fb98 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
>@@ -10,9 +10,9 @@
> 
> #include "modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h"
> 
>+#include "api/video/encoded_image.h"
> #include "api/video/i420_buffer.h"
> #include "api/video/video_frame_buffer.h"
>-#include "common_video/include/video_frame.h"
> #include "common_video/include/video_frame_buffer.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.h
>index ea37f0c4d9ff465160ce8fa5357aedf75593d654..0d2f9fa26ea35d5a8c4b154354796ee057c302d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.h
>@@ -14,8 +14,8 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/video/encoded_image.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/video_frame.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoder_adapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoder_adapter.cc
>index c3a75062b54ebe58914939ad2714c8b2d6d12db1..4dabca77fd0059a51b383231601d03593500e1ba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoder_adapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoder_adapter.cc
>@@ -12,7 +12,7 @@
> 
> #include <cstring>
> 
>-#include "common_video/include/video_frame.h"
>+#include "api/video/encoded_image.h"
> #include "common_video/include/video_frame_buffer.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "modules/include/module_common_types.h"
>@@ -53,6 +53,7 @@ MultiplexEncoderAdapter::MultiplexEncoderAdapter(
>     : factory_(factory),
>       associated_format_(associated_format),
>       encoded_complete_callback_(nullptr),
>+      key_frame_interval_(0),
>       supports_augmented_data_(supports_augmented_data) {}
> 
> MultiplexEncoderAdapter::~MultiplexEncoderAdapter() {
>@@ -92,6 +93,11 @@ int MultiplexEncoderAdapter::InitEncode(const VideoCodec* inst,
>       break;
>   }
> 
>+  encoder_info_ = EncoderInfo();
>+  encoder_info_.implementation_name = "MultiplexEncoderAdapter (";
>+  // This needs to be false so that we can do the split in Encode().
>+  encoder_info_.supports_native_handle = false;
>+
>   for (size_t i = 0; i < kAlphaCodecStreams; ++i) {
>     std::unique_ptr<VideoEncoder> encoder =
>         factory_->CreateVideoEncoder(associated_format_);
>@@ -104,8 +110,17 @@ int MultiplexEncoderAdapter::InitEncode(const VideoCodec* inst,
>     adapter_callbacks_.emplace_back(new AdapterEncodedImageCallback(
>         this, static_cast<AlphaCodecStream>(i)));
>     encoder->RegisterEncodeCompleteCallback(adapter_callbacks_.back().get());
>+
>+    const EncoderInfo& encoder_impl_info = encoder->GetEncoderInfo();
>+    encoder_info_.implementation_name += encoder_impl_info.implementation_name;
>+    if (i != kAlphaCodecStreams - 1) {
>+      encoder_info_.implementation_name += ", ";
>+    }
>+
>     encoders_.emplace_back(std::move(encoder));
>   }
>+  encoder_info_.implementation_name += ")";
>+
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>@@ -187,16 +202,6 @@ int MultiplexEncoderAdapter::RegisterEncodeCompleteCallback(
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-int MultiplexEncoderAdapter::SetChannelParameters(uint32_t packet_loss,
>-                                                  int64_t rtt) {
>-  for (auto& encoder : encoders_) {
>-    const int rv = encoder->SetChannelParameters(packet_loss, rtt);
>-    if (rv)
>-      return rv;
>-  }
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
> int MultiplexEncoderAdapter::SetRateAllocation(
>     const VideoBitrateAllocation& bitrate,
>     uint32_t framerate) {
>@@ -238,8 +243,8 @@ int MultiplexEncoderAdapter::Release() {
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-const char* MultiplexEncoderAdapter::ImplementationName() const {
>-  return "MultiplexEncoderAdapter";
>+VideoEncoder::EncoderInfo MultiplexEncoderAdapter::GetEncoderInfo() const {
>+  return encoder_info_;
> }
> 
> EncodedImageCallback::Result MultiplexEncoderAdapter::OnEncodedImage(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_unittest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_unittest.h
>index 48860c35a6f34971592e3d3ec9b4c707eddb7e51..9a4d60cec2678726613a1a72c186a705e715c54d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_unittest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_unittest.h
>@@ -32,11 +32,7 @@ class VideoCodecUnitTest : public ::testing::Test {
>   VideoCodecUnitTest()
>       : encode_complete_callback_(this),
>         decode_complete_callback_(this),
>-        encoded_frame_event_(false /* manual reset */,
>-                             false /* initially signaled */),
>         wait_for_encoded_frames_threshold_(1),
>-        decoded_frame_event_(false /* manual reset */,
>-                             false /* initially signaled */),
>         last_input_frame_timestamp_(0) {}
> 
>  protected:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
>index 4dc268a0431f095efd9443c73accd88969986e52..847e84785bcb4326d47336131fe6b4fcc076749a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
>@@ -130,9 +130,7 @@ bool RunEncodeInRealTime(const VideoCodecTestFixtureImpl::Config& config) {
> 
> std::string FilenameWithParams(
>     const VideoCodecTestFixtureImpl::Config& config) {
>-  std::string implementation_type = config.hw_encoder ? "hw" : "sw";
>   return config.filename + "_" + config.CodecName() + "_" +
>-         implementation_type + "_" +
>          std::to_string(config.codec_settings.startBitrate);
> }
> 
>@@ -446,7 +444,7 @@ void VideoCodecTestFixtureImpl::ProcessAllFrames(
>   }
> 
>   // Wait until we know that the last frame has been sent for encode.
>-  rtc::Event sync_event(false, false);
>+  rtc::Event sync_event;
>   task_queue->PostTask([&sync_event] { sync_event.Set(); });
>   sync_event.Wait(rtc::Event::kForever);
> 
>@@ -673,7 +671,7 @@ void VideoCodecTestFixtureImpl::PrintSettings(
>   std::string encoder_name;
>   std::string decoder_name;
>   task_queue->SendTask([this, &encoder_name, &decoder_name] {
>-    encoder_name = encoder_->ImplementationName();
>+    encoder_name = encoder_->GetEncoderInfo().implementation_name;
>     decoder_name = decoders_.at(0)->ImplementationName();
>   });
>   printf("enc_impl_name: %s\n", encoder_name.c_str());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_libvpx.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_libvpx.cc
>index 9586ce9c792382133421aed816cbaf5edee8d048..f69fde6884010fc35fc396702e243bb276ab4302 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_libvpx.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_libvpx.cc
>@@ -12,6 +12,7 @@
> 
> #include "absl/memory/memory.h"
> #include "api/test/create_videocodec_test_fixture.h"
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "api/video_codecs/sdp_video_format.h"
> #include "media/base/mediaconstants.h"
> #include "media/engine/internaldecoderfactory.h"
>@@ -19,7 +20,6 @@
> #include "media/engine/simulcast_encoder_adapter.h"
> #include "modules/video_coding/utility/vp8_header_parser.h"
> #include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
> 
>@@ -63,8 +63,6 @@ VideoCodecTestFixture::Config CreateConfig() {
>   config.filepath = ResourcePath(config.filename, "yuv");
>   config.num_frames = kNumFramesLong;
>   config.use_single_core = true;
>-  config.hw_encoder = false;
>-  config.hw_decoder = false;
>   return config;
> }
> 
>@@ -92,7 +90,7 @@ void PrintRdPerf(std::map<size_t, std::vector<VideoStatistics>> rd_stats) {
> }
> }  // namespace
> 
>-#if !defined(RTC_DISABLE_VP9)
>+#if defined(RTC_ENABLE_VP9)
> TEST(VideoCodecTestLibvpx, HighBitrateVP9) {
>   auto config = CreateConfig();
>   config.SetCodecSettings(cricket::kVp9CodecName, 1, 1, 1, false, true, false,
>@@ -105,7 +103,7 @@ TEST(VideoCodecTestLibvpx, HighBitrateVP9) {
>   std::vector<RateProfile> rate_profiles = {{500, 30, kNumFramesShort}};
> 
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {5, 1, 0, 0.11, 0.3, 0.1, 0, 1}};
>+      {5, 1, 0, 1, 0.3, 0.1, 0, 1}};
> 
>   std::vector<QualityThresholds> quality_thresholds = {{37, 36, 0.94, 0.92}};
> 
>@@ -126,9 +124,9 @@ TEST(VideoCodecTestLibvpx, ChangeBitrateVP9) {
>       {500, 30, kNumFramesLong}};
> 
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {5, 1, 0, 0.15, 0.5, 0.1, 0, 1},
>-      {15, 2, 0, 0.2, 0.5, 0.1, 0, 0},
>-      {10, 1, 0, 0.3, 0.5, 0.1, 0, 0}};
>+      {5, 1, 0, 1, 0.5, 0.1, 0, 1},
>+      {15, 2, 0, 1, 0.5, 0.1, 0, 0},
>+      {10, 1, 0, 1, 0.5, 0.1, 0, 0}};
> 
>   std::vector<QualityThresholds> quality_thresholds = {
>       {34, 33, 0.90, 0.88}, {38, 35, 0.95, 0.91}, {35, 34, 0.93, 0.90}};
>@@ -151,9 +149,9 @@ TEST(VideoCodecTestLibvpx, ChangeFramerateVP9) {
> 
>   // Framerate mismatch should be lower for lower framerate.
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {10, 2, 40, 0.4, 0.5, 0.2, 0, 1},
>-      {8, 2, 5, 0.2, 0.5, 0.2, 0, 0},
>-      {5, 2, 0, 0.21, 0.5, 0.3, 0, 0}};
>+      {10, 2, 40, 1, 0.5, 0.2, 0, 1},
>+      {8, 2, 5, 1, 0.5, 0.2, 0, 0},
>+      {5, 2, 0, 1, 0.5, 0.3, 0, 0}};
> 
>   // Quality should be higher for lower framerates for the same content.
>   std::vector<QualityThresholds> quality_thresholds = {
>@@ -174,7 +172,7 @@ TEST(VideoCodecTestLibvpx, DenoiserOnVP9) {
>   std::vector<RateProfile> rate_profiles = {{500, 30, kNumFramesShort}};
> 
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {5, 1, 0, 0.11, 0.3, 0.1, 0, 1}};
>+      {5, 1, 0, 1, 0.3, 0.1, 0, 1}};
> 
>   std::vector<QualityThresholds> quality_thresholds = {{37.5, 36, 0.94, 0.93}};
> 
>@@ -192,7 +190,7 @@ TEST(VideoCodecTestLibvpx, VeryLowBitrateVP9) {
>   std::vector<RateProfile> rate_profiles = {{50, 30, kNumFramesLong}};
> 
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {15, 3, 75, 1.0, 0.5, 0.4, 1, 1}};
>+      {15, 3, 75, 1, 0.5, 0.4, 1, 1}};
> 
>   std::vector<QualityThresholds> quality_thresholds = {{28, 25, 0.80, 0.65}};
> 
>@@ -202,7 +200,7 @@ TEST(VideoCodecTestLibvpx, VeryLowBitrateVP9) {
> // TODO(marpan): Add temporal layer test for VP9, once changes are in
> // vp9 wrapper for this.
> 
>-#endif  // !defined(RTC_DISABLE_VP9)
>+#endif  // defined(RTC_ENABLE_VP9)
> 
> TEST(VideoCodecTestLibvpx, HighBitrateVP8) {
>   auto config = CreateConfig();
>@@ -216,7 +214,7 @@ TEST(VideoCodecTestLibvpx, HighBitrateVP8) {
>   std::vector<RateProfile> rate_profiles = {{500, 30, kNumFramesShort}};
> 
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {5, 1, 0, 0.1, 0.2, 0.1, 0, 1}};
>+      {5, 1, 0, 1, 0.2, 0.1, 0, 1}};
> 
> #if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
>   std::vector<QualityThresholds> quality_thresholds = {{35, 33, 0.91, 0.89}};
>@@ -255,9 +253,9 @@ TEST(VideoCodecTestLibvpx, MAYBE_ChangeBitrateVP8) {
>       {500, 30, kNumFramesLong}};
> 
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {5, 1, 0, 0.1, 0.2, 0.1, 0, 1},
>-      {15.5, 1, 0, 0.1, 0.2, 0.1, 0, 0},
>-      {15, 1, 0, 0.3, 0.2, 0.1, 0, 0}};
>+      {5, 1, 0, 1, 0.2, 0.1, 0, 1},
>+      {15.5, 1, 0, 1, 0.2, 0.1, 0, 0},
>+      {15, 1, 0, 1, 0.2, 0.1, 0, 0}};
> 
> #if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
>   std::vector<QualityThresholds> quality_thresholds = {
>@@ -290,14 +288,14 @@ TEST(VideoCodecTestLibvpx, MAYBE_ChangeFramerateVP8) {
> 
> #if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {10, 2, 60, 0.5, 0.3, 0.3, 0, 1},
>-      {10, 2, 30, 0.3, 0.3, 0.3, 0, 0},
>-      {10, 2, 10, 0.2, 0.3, 0.2, 0, 0}};
>+      {10, 2, 60, 1, 0.3, 0.3, 0, 1},
>+      {10, 2, 30, 1, 0.3, 0.3, 0, 0},
>+      {10, 2, 10, 1, 0.3, 0.2, 0, 0}};
> #else
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {10, 2, 20, 0.4, 0.3, 0.1, 0, 1},
>-      {5, 2, 5, 0.3, 0.3, 0.1, 0, 0},
>-      {4, 2, 1, 0.2, 0.3, 0.2, 0, 0}};
>+      {10, 2, 20, 1, 0.3, 0.1, 0, 1},
>+      {5, 2, 5, 1, 0.3, 0.1, 0, 0},
>+      {4, 2, 1, 1, 0.3, 0.2, 0, 0}};
> #endif
> 
> #if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
>@@ -328,10 +326,10 @@ TEST(VideoCodecTestLibvpx, MAYBE_TemporalLayersVP8) {
> 
> #if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {10, 1, 2, 0.3, 0.2, 0.1, 0, 1}, {12, 2, 3, 0.1, 0.2, 0.1, 0, 1}};
>+      {10, 1, 2, 1, 0.2, 0.1, 0, 1}, {12, 2, 3, 1, 0.2, 0.1, 0, 1}};
> #else
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {5, 1, 0, 0.1, 0.2, 0.1, 0, 1}, {10, 2, 0, 0.1, 0.2, 0.1, 0, 1}};
>+      {5, 1, 0, 1, 0.2, 0.1, 0, 1}, {10, 2, 0, 1, 0.2, 0.1, 0, 1}};
> #endif
> // Min SSIM drops because of high motion scene with complex backgound (trees).
> #if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
>@@ -364,7 +362,7 @@ TEST(VideoCodecTestLibvpx, MAYBE_MultiresVP8) {
>   std::vector<RateProfile> rate_profiles = {{1500, 30, config.num_frames}};
> 
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {5, 1, 5, 0.2, 0.3, 0.1, 0, 1}};
>+      {5, 1, 5, 1, 0.3, 0.1, 0, 1}};
>   std::vector<QualityThresholds> quality_thresholds = {{34, 32, 0.90, 0.88}};
> 
>   fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>@@ -401,7 +399,7 @@ TEST(VideoCodecTestLibvpx, MAYBE_SimulcastVP8) {
>   std::vector<RateProfile> rate_profiles = {{1500, 30, config.num_frames}};
> 
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {20, 5, 90, 0.8, 0.5, 0.3, 0, 1}};
>+      {20, 5, 90, 1, 0.5, 0.3, 0, 1}};
>   std::vector<QualityThresholds> quality_thresholds = {{34, 32, 0.90, 0.88}};
> 
>   fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>@@ -426,7 +424,7 @@ TEST(VideoCodecTestLibvpx, MAYBE_SvcVP9) {
>   std::vector<RateProfile> rate_profiles = {{1500, 30, config.num_frames}};
> 
>   std::vector<RateControlThresholds> rc_thresholds = {
>-      {5, 1, 5, 0.2, 0.3, 0.1, 0, 1}};
>+      {5, 1, 5, 1, 0.3, 0.1, 0, 1}};
>   std::vector<QualityThresholds> quality_thresholds = {{36, 34, 0.93, 0.90}};
> 
>   fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_mediacodec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_mediacodec.cc
>index 99b1aed8c0a580536798025a8c658f57b36b7245..edfc211c226d549cc0c5934f723f35ce07926e49 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_mediacodec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_mediacodec.cc
>@@ -33,8 +33,6 @@ VideoCodecTestFixture::Config CreateConfig() {
>   config.filename = "foreman_cif";
>   config.filepath = ResourcePath(config.filename, "yuv");
>   config.num_frames = kForemanNumFrames;
>-  config.hw_encoder = true;
>-  config.hw_decoder = true;
>   // In order to not overwhelm the OpenMAX buffers in the Android MediaCodec.
>   config.encode_in_real_time = true;
>   return config;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_openh264.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_openh264.cc
>index e5cf71ba90f9042ee693c116d93baba13edde0f3..bd4bb52b85191dc560af3a8d09c4e39f82c3734e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_openh264.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_openh264.cc
>@@ -32,8 +32,6 @@ VideoCodecTestFixture::Config CreateConfig() {
>   config.num_frames = kNumFrames;
>   // Only allow encoder/decoder to use single core, for predictability.
>   config.use_single_core = true;
>-  config.hw_encoder = false;
>-  config.hw_decoder = false;
>   return config;
> }
> }  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_parameterized.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_parameterized.cc
>index 0d49b80d95af30c5b2914013c4a4d56c549fd913..efc80add2da5c1d23923917169f301242662e2fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_parameterized.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_parameterized.cc
>@@ -20,7 +20,6 @@ namespace {
> // Loop variables.
> const size_t kBitrates[] = {500};
> const VideoCodecType kVideoCodecType[] = {kVideoCodecVP8};
>-const bool kHwCodec[] = {false};
> 
> // Codec settings.
> const int kNumSpatialLayers = 1;
>@@ -39,12 +38,11 @@ const int kNumFrames = 30;
> class VideoCodecTestParameterized
>     : public ::testing::Test,
>       public ::testing::WithParamInterface<
>-          ::testing::tuple<size_t, VideoCodecType, bool>> {
>+          ::testing::tuple<size_t, VideoCodecType>> {
>  protected:
>   VideoCodecTestParameterized()
>       : bitrate_(::testing::get<0>(GetParam())),
>-        codec_type_(::testing::get<1>(GetParam())),
>-        hw_codec_(::testing::get<2>(GetParam())) {}
>+        codec_type_(::testing::get<1>(GetParam())) {}
>   ~VideoCodecTestParameterized() override = default;
> 
>   void RunTest(size_t width,
>@@ -56,8 +54,6 @@ class VideoCodecTestParameterized
>     config.filepath = ResourcePath(filename, "yuv");
>     config.use_single_core = kUseSingleCore;
>     config.measure_cpu = kMeasureCpu;
>-    config.hw_encoder = hw_codec_;
>-    config.hw_decoder = hw_codec_;
>     config.num_frames = kNumFrames;
> 
>     const size_t num_simulcast_streams =
>@@ -80,14 +76,13 @@ class VideoCodecTestParameterized
>   std::unique_ptr<VideoCodecTestFixture> fixture_;
>   const size_t bitrate_;
>   const VideoCodecType codec_type_;
>-  const bool hw_codec_;
> };
> 
>-INSTANTIATE_TEST_CASE_P(CodecSettings,
>-                        VideoCodecTestParameterized,
>-                        ::testing::Combine(::testing::ValuesIn(kBitrates),
>-                                           ::testing::ValuesIn(kVideoCodecType),
>-                                           ::testing::ValuesIn(kHwCodec)));
>+INSTANTIATE_TEST_CASE_P(
>+    CodecSettings,
>+    VideoCodecTestParameterized,
>+    ::testing::Combine(::testing::ValuesIn(kBitrates),
>+                       ::testing::ValuesIn(kVideoCodecType)));
> 
> TEST_P(VideoCodecTestParameterized, Foreman_352x288_30) {
>   RunTest(352, 288, 30, "foreman_cif");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_videotoolbox.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_videotoolbox.cc
>index f31ff548a81255bec445973ec750c3670328f057..6dfb993ce881c7eeed46c8164ef1db8865b4d6aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_videotoolbox.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_videotoolbox.cc
>@@ -28,8 +28,6 @@ VideoCodecTestFixture::Config CreateConfig() {
>   config.filename = "foreman_cif";
>   config.filepath = ResourcePath(config.filename, "yuv");
>   config.num_frames = kForemanNumFrames;
>-  config.hw_encoder = true;
>-  config.hw_decoder = true;
>   return config;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
>index 5e980cdcf1acc8155f841e4f7e047793ba206d74..69582660f91188bab809dae344149f1a6fc9f5c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
>@@ -14,6 +14,7 @@
> #include <limits>
> #include <utility>
> 
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "api/video/i420_buffer.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "common_video/h264/h264_common.h"
>@@ -173,8 +174,9 @@ VideoProcessor::VideoProcessor(webrtc::VideoEncoder* encoder,
>       stats_(stats),
>       encoder_(encoder),
>       decoders_(decoders),
>-      bitrate_allocator_(VideoCodecInitializer::CreateBitrateAllocator(
>-          config_.codec_settings)),
>+      bitrate_allocator_(
>+          CreateBuiltinVideoBitrateAllocatorFactory()
>+              ->CreateVideoBitrateAllocator(config_.codec_settings)),
>       framerate_fps_(0),
>       encode_callback_(this),
>       input_frame_reader_(input_frame_reader),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
>index 589a3d898a8ee488a4016eb623c9ead43854a578..3d41ad998febfc62063787ca0bd35fde18f7f36e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
>@@ -11,13 +11,14 @@
> #include <memory>
> 
> #include "absl/memory/memory.h"
>+#include "api/test/mock_video_decoder.h"
>+#include "api/test/mock_video_encoder.h"
> #include "api/test/videocodec_test_fixture.h"
> #include "api/video/i420_buffer.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "media/base/mediaconstants.h"
> #include "modules/video_coding/codecs/test/videocodec_test_stats_impl.h"
> #include "modules/video_coding/codecs/test/videoprocessor.h"
>-#include "modules/video_coding/include/mock/mock_video_codec_interface.h"
> #include "modules/video_coding/include/video_coding.h"
> #include "rtc_base/event.h"
> #include "rtc_base/task_queue_for_test.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
>index d53f46994d3608ef39e3e06e61edafaac3a9b875..c5e163fcf720b8b28ec98275321774735a907734 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
>@@ -7,8 +7,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
>-
> #include <stdlib.h>
> #include <string.h>
> 
>@@ -20,6 +18,7 @@
> #include <vector>
> 
> #include "modules/include/module_common_types.h"
>+#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>@@ -27,27 +26,30 @@
> 
> namespace webrtc {
> 
>-TemporalLayers::FrameConfig::FrameConfig()
>+Vp8TemporalLayers::FrameConfig::FrameConfig()
>     : FrameConfig(kNone, kNone, kNone, false) {}
> 
>-TemporalLayers::FrameConfig::FrameConfig(TemporalLayers::BufferFlags last,
>-                                         TemporalLayers::BufferFlags golden,
>-                                         TemporalLayers::BufferFlags arf)
>+Vp8TemporalLayers::FrameConfig::FrameConfig(
>+    Vp8TemporalLayers::BufferFlags last,
>+    Vp8TemporalLayers::BufferFlags golden,
>+    Vp8TemporalLayers::BufferFlags arf)
>     : FrameConfig(last, golden, arf, false) {}
> 
>-TemporalLayers::FrameConfig::FrameConfig(TemporalLayers::BufferFlags last,
>-                                         TemporalLayers::BufferFlags golden,
>-                                         TemporalLayers::BufferFlags arf,
>-                                         FreezeEntropy)
>+Vp8TemporalLayers::FrameConfig::FrameConfig(
>+    Vp8TemporalLayers::BufferFlags last,
>+    Vp8TemporalLayers::BufferFlags golden,
>+    Vp8TemporalLayers::BufferFlags arf,
>+    FreezeEntropy)
>     : FrameConfig(last, golden, arf, true) {}
> 
>-TemporalLayers::FrameConfig::FrameConfig(TemporalLayers::BufferFlags last,
>-                                         TemporalLayers::BufferFlags golden,
>-                                         TemporalLayers::BufferFlags arf,
>-                                         bool freeze_entropy)
>-    : drop_frame(last == TemporalLayers::kNone &&
>-                 golden == TemporalLayers::kNone &&
>-                 arf == TemporalLayers::kNone),
>+Vp8TemporalLayers::FrameConfig::FrameConfig(
>+    Vp8TemporalLayers::BufferFlags last,
>+    Vp8TemporalLayers::BufferFlags golden,
>+    Vp8TemporalLayers::BufferFlags arf,
>+    bool freeze_entropy)
>+    : drop_frame(last == Vp8TemporalLayers::kNone &&
>+                 golden == Vp8TemporalLayers::kNone &&
>+                 arf == Vp8TemporalLayers::kNone),
>       last_buffer_flags(last),
>       golden_buffer_flags(golden),
>       arf_buffer_flags(arf),
>@@ -106,15 +108,15 @@ std::vector<unsigned int> GetTemporalIds(size_t num_layers) {
>   return {0};
> }
> 
>-uint8_t GetUpdatedBuffers(const TemporalLayers::FrameConfig& config) {
>+uint8_t GetUpdatedBuffers(const Vp8TemporalLayers::FrameConfig& config) {
>   uint8_t flags = 0;
>-  if (config.last_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>+  if (config.last_buffer_flags & Vp8TemporalLayers::BufferFlags::kUpdate) {
>     flags |= static_cast<uint8_t>(Vp8BufferReference::kLast);
>   }
>-  if (config.golden_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>+  if (config.golden_buffer_flags & Vp8TemporalLayers::BufferFlags::kUpdate) {
>     flags |= static_cast<uint8_t>(Vp8BufferReference::kGolden);
>   }
>-  if (config.arf_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>+  if (config.arf_buffer_flags & Vp8TemporalLayers::BufferFlags::kUpdate) {
>     flags |= static_cast<uint8_t>(Vp8BufferReference::kAltref);
>   }
>   return flags;
>@@ -122,10 +124,10 @@ uint8_t GetUpdatedBuffers(const TemporalLayers::FrameConfig& config) {
> 
> // Find the set of buffers that are never updated by the given pattern.
> std::set<Vp8BufferReference> FindKfBuffers(
>-    const std::vector<TemporalLayers::FrameConfig>& frame_configs) {
>+    const std::vector<Vp8TemporalLayers::FrameConfig>& frame_configs) {
>   std::set<Vp8BufferReference> kf_buffers(kAllBuffers.begin(),
>                                           kAllBuffers.end());
>-  for (TemporalLayers::FrameConfig config : frame_configs) {
>+  for (Vp8TemporalLayers::FrameConfig config : frame_configs) {
>     // Get bit-masked set of update buffers for this frame config.
>     uint8_t updated_buffers = GetUpdatedBuffers(config);
>     for (Vp8BufferReference buffer : kAllBuffers) {
>@@ -138,7 +140,7 @@ std::set<Vp8BufferReference> FindKfBuffers(
> }
> }  // namespace
> 
>-std::vector<TemporalLayers::FrameConfig>
>+std::vector<Vp8TemporalLayers::FrameConfig>
> DefaultTemporalLayers::GetTemporalPattern(size_t num_layers) {
>   // For indexing in the patterns described below (which temporal layers they
>   // belong to), see the diagram above.
>@@ -262,7 +264,7 @@ DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers)
>       kf_buffers_(FindKfBuffers(temporal_pattern_)),
>       pattern_idx_(kUninitializedPatternIndex),
>       checker_(TemporalLayersChecker::CreateTemporalLayersChecker(
>-          TemporalLayersType::kFixedPattern,
>+          Vp8TemporalLayersType::kFixedPattern,
>           number_of_temporal_layers)) {
>   RTC_CHECK_GE(kMaxTemporalStreams, number_of_temporal_layers);
>   RTC_CHECK_GE(number_of_temporal_layers, 0);
>@@ -278,6 +280,8 @@ DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers)
>   }
> }
> 
>+DefaultTemporalLayers::~DefaultTemporalLayers() = default;
>+
> bool DefaultTemporalLayers::SupportsEncoderFrameDropping() const {
>   // This class allows the encoder drop frames as it sees fit.
>   return true;
>@@ -346,13 +350,13 @@ bool DefaultTemporalLayers::IsSyncFrame(const FrameConfig& config) const {
>   return true;
> }
> 
>-TemporalLayers::FrameConfig DefaultTemporalLayers::UpdateLayerConfig(
>+Vp8TemporalLayers::FrameConfig DefaultTemporalLayers::UpdateLayerConfig(
>     uint32_t timestamp) {
>   RTC_DCHECK_GT(num_layers_, 0);
>   RTC_DCHECK_LT(0, temporal_pattern_.size());
> 
>   pattern_idx_ = (pattern_idx_ + 1) % temporal_pattern_.size();
>-  TemporalLayers::FrameConfig tl_config = temporal_pattern_[pattern_idx_];
>+  Vp8TemporalLayers::FrameConfig tl_config = temporal_pattern_[pattern_idx_];
>   tl_config.encoder_layer_id = tl_config.packetizer_temporal_idx =
>       temporal_ids_[pattern_idx_ % temporal_ids_.size()];
> 
>@@ -560,9 +564,11 @@ DefaultTemporalLayersChecker::DefaultTemporalLayersChecker(
>   }
> }
> 
>+DefaultTemporalLayersChecker::~DefaultTemporalLayersChecker() = default;
>+
> bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>     bool frame_is_keyframe,
>-    const TemporalLayers::FrameConfig& frame_config) {
>+    const Vp8TemporalLayers::FrameConfig& frame_config) {
>   if (!TemporalLayersChecker::CheckTemporalConfig(frame_is_keyframe,
>                                                   frame_config)) {
>     return false;
>@@ -613,7 +619,7 @@ bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>   std::vector<int> dependencies;
> 
>   if (frame_config.last_buffer_flags &
>-      TemporalLayers::BufferFlags::kReference) {
>+      Vp8TemporalLayers::BufferFlags::kReference) {
>     uint8_t referenced_layer = temporal_ids_[last_.pattern_idx];
>     if (referenced_layer > 0) {
>       need_sync = false;
>@@ -628,7 +634,8 @@ bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>     return false;
>   }
> 
>-  if (frame_config.arf_buffer_flags & TemporalLayers::BufferFlags::kReference) {
>+  if (frame_config.arf_buffer_flags &
>+      Vp8TemporalLayers::BufferFlags::kReference) {
>     uint8_t referenced_layer = temporal_ids_[arf_.pattern_idx];
>     if (referenced_layer > 0) {
>       need_sync = false;
>@@ -644,7 +651,7 @@ bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>   }
> 
>   if (frame_config.golden_buffer_flags &
>-      TemporalLayers::BufferFlags::kReference) {
>+      Vp8TemporalLayers::BufferFlags::kReference) {
>     uint8_t referenced_layer = temporal_ids_[golden_.pattern_idx];
>     if (referenced_layer > 0) {
>       need_sync = false;
>@@ -680,17 +687,19 @@ bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>     }
>   }
> 
>-  if (frame_config.last_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>+  if (frame_config.last_buffer_flags &
>+      Vp8TemporalLayers::BufferFlags::kUpdate) {
>     last_.is_updated_this_cycle = true;
>     last_.pattern_idx = pattern_idx_;
>     last_.is_keyframe = false;
>   }
>-  if (frame_config.arf_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>+  if (frame_config.arf_buffer_flags & Vp8TemporalLayers::BufferFlags::kUpdate) {
>     arf_.is_updated_this_cycle = true;
>     arf_.pattern_idx = pattern_idx_;
>     arf_.is_keyframe = false;
>   }
>-  if (frame_config.golden_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>+  if (frame_config.golden_buffer_flags &
>+      Vp8TemporalLayers::BufferFlags::kUpdate) {
>     golden_.is_updated_this_cycle = true;
>     golden_.pattern_idx = pattern_idx_;
>     golden_.is_keyframe = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
>index bda43cd6d93b414bbe7a4baaa5bd2661fc749d6b..fff62a557d36e0de9ffdabfb0ccd008153202385 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
>@@ -18,23 +18,23 @@
> #include <set>
> #include <vector>
> 
>-#include "modules/video_coding/codecs/vp8/include/temporal_layers_checker.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
>-
> #include "absl/types/optional.h"
> 
>+#include "api/video_codecs/vp8_temporal_layers.h"
>+#include "modules/video_coding/codecs/vp8/include/temporal_layers_checker.h"
>+
> namespace webrtc {
> 
>-class DefaultTemporalLayers : public TemporalLayers {
>+class DefaultTemporalLayers : public Vp8TemporalLayers {
>  public:
>   explicit DefaultTemporalLayers(int number_of_temporal_layers);
>-  virtual ~DefaultTemporalLayers() {}
>+  ~DefaultTemporalLayers() override;
> 
>   bool SupportsEncoderFrameDropping() const override;
> 
>   // Returns the recommended VP8 encode flags needed. May refresh the decoder
>   // and/or update the reference buffers.
>-  TemporalLayers::FrameConfig UpdateLayerConfig(uint32_t timestamp) override;
>+  Vp8TemporalLayers::FrameConfig UpdateLayerConfig(uint32_t timestamp) override;
> 
>   // New target bitrate, per temporal layer.
>   void OnRatesUpdated(const std::vector<uint32_t>& bitrates_bps,
>@@ -50,7 +50,7 @@ class DefaultTemporalLayers : public TemporalLayers {
> 
>  private:
>   static constexpr size_t kKeyframeBuffer = std::numeric_limits<size_t>::max();
>-  static std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(
>+  static std::vector<Vp8TemporalLayers::FrameConfig> GetTemporalPattern(
>       size_t num_layers);
>   bool IsSyncFrame(const FrameConfig& config) const;
>   void ValidateReferences(BufferFlags* flags, Vp8BufferReference ref) const;
>@@ -58,7 +58,7 @@ class DefaultTemporalLayers : public TemporalLayers {
> 
>   const size_t num_layers_;
>   const std::vector<unsigned int> temporal_ids_;
>-  const std::vector<TemporalLayers::FrameConfig> temporal_pattern_;
>+  const std::vector<Vp8TemporalLayers::FrameConfig> temporal_pattern_;
>   // Set of buffers that are never updated except by keyframes.
>   const std::set<Vp8BufferReference> kf_buffers_;
> 
>@@ -95,9 +95,11 @@ class DefaultTemporalLayers : public TemporalLayers {
> class DefaultTemporalLayersChecker : public TemporalLayersChecker {
>  public:
>   explicit DefaultTemporalLayersChecker(int number_of_temporal_layers);
>+  ~DefaultTemporalLayersChecker() override;
>+
>   bool CheckTemporalConfig(
>       bool frame_is_keyframe,
>-      const TemporalLayers::FrameConfig& frame_config) override;
>+      const Vp8TemporalLayers::FrameConfig& frame_config) override;
> 
>  private:
>   struct BufferState {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
>index ad31a6ef4a0aecbb332b10156fe9d79988fd0f1c..5ee652055cea517782b3ccedcf09943daf71c7da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
>@@ -74,7 +74,7 @@ constexpr int kDefaultBytesPerFrame =
> constexpr int kDefaultQp = 2;
> }  // namespace
> 
>-using BufferFlags = TemporalLayers::BufferFlags;
>+using BufferFlags = Vp8TemporalLayers::BufferFlags;
> 
> TEST(TemporalLayersTest, 2Layers) {
>   constexpr int kNumLayers = 2;
>@@ -114,7 +114,7 @@ TEST(TemporalLayersTest, 2Layers) {
> 
>   uint32_t timestamp = 0;
>   for (int i = 0; i < 16; ++i) {
>-    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+    Vp8TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>     EXPECT_EQ(expected_flags[i], LibvpxVp8Encoder::EncodeFlags(tl_config)) << i;
>     tl.OnEncodeDone(timestamp, kDefaultBytesPerFrame, i == 0, kDefaultQp,
>                     &vp8_info);
>@@ -166,7 +166,7 @@ TEST(TemporalLayersTest, 3Layers) {
> 
>   unsigned int timestamp = 0;
>   for (int i = 0; i < 16; ++i) {
>-    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+    Vp8TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>     EXPECT_EQ(expected_flags[i], LibvpxVp8Encoder::EncodeFlags(tl_config)) << i;
>     tl.OnEncodeDone(timestamp, kDefaultBytesPerFrame, i == 0, kDefaultQp,
>                     &vp8_info);
>@@ -207,7 +207,7 @@ TEST(TemporalLayersTest, Alternative3Layers) {
> 
>   unsigned int timestamp = 0;
>   for (int i = 0; i < 8; ++i) {
>-    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+    Vp8TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>     EXPECT_EQ(expected_flags[i], LibvpxVp8Encoder::EncodeFlags(tl_config)) << i;
>     tl.OnEncodeDone(timestamp, kDefaultBytesPerFrame, i == 0, kDefaultQp,
>                     &vp8_info);
>@@ -238,7 +238,7 @@ TEST(TemporalLayersTest, SearchOrder) {
> 
>   // Start with a key-frame. tl_config flags can be ignored.
>   uint32_t timestamp = 0;
>-  TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+  Vp8TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>   tl.OnEncodeDone(timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
>                   &vp8_info);
> 
>@@ -282,7 +282,7 @@ TEST(TemporalLayersTest, SearchOrderWithDrop) {
> 
>   // Start with a key-frame. tl_config flags can be ignored.
>   uint32_t timestamp = 0;
>-  TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+  Vp8TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>   tl.OnEncodeDone(timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
>                   &vp8_info);
> 
>@@ -343,7 +343,7 @@ TEST(TemporalLayersTest, 4Layers) {
> 
>   uint32_t timestamp = 0;
>   for (int i = 0; i < 16; ++i) {
>-    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+    Vp8TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>     EXPECT_EQ(expected_flags[i], LibvpxVp8Encoder::EncodeFlags(tl_config)) << i;
>     tl.OnEncodeDone(timestamp, kDefaultBytesPerFrame, i == 0, kDefaultQp,
>                     &vp8_info);
>@@ -373,7 +373,7 @@ TEST(TemporalLayersTest, DoesNotReferenceDroppedFrames) {
> 
>   // Start with a keyframe.
>   uint32_t timestamp = 0;
>-  TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+  Vp8TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>   tl.OnEncodeDone(timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
>                   &vp8_info);
> 
>@@ -460,7 +460,7 @@ TEST(TemporalLayersTest, DoesNotReferenceUnlessGuaranteedToExist) {
> 
>   // Start with a keyframe.
>   uint32_t timestamp = 0;
>-  TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+  Vp8TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>   tl.OnEncodeDone(timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
>                   &vp8_info);
> 
>@@ -530,7 +530,7 @@ TEST(TemporalLayersTest, DoesNotReferenceUnlessGuaranteedToExistLongDelay) {
> 
>   // Start with a keyframe.
>   uint32_t timestamp = 0;
>-  TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+  Vp8TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>   tl.OnEncodeDone(timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
>                   &vp8_info);
> 
>@@ -610,7 +610,8 @@ TEST(TemporalLayersTest, KeyFrame) {
>     for (int j = 1; j <= i; ++j) {
>       // Since last frame was always a keyframe and thus index 0 in the pattern,
>       // this loop starts at index 1.
>-      TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+      Vp8TemporalLayers::FrameConfig tl_config =
>+          tl.UpdateLayerConfig(timestamp);
>       EXPECT_EQ(expected_flags[j], LibvpxVp8Encoder::EncodeFlags(tl_config))
>           << j;
>       tl.OnEncodeDone(timestamp, kDefaultBytesPerFrame, false, kDefaultQp,
>@@ -622,7 +623,7 @@ TEST(TemporalLayersTest, KeyFrame) {
>       timestamp += 3000;
>     }
> 
>-    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+    Vp8TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>     tl.OnEncodeDone(timestamp, kDefaultBytesPerFrame, true, kDefaultQp,
>                     &vp8_info);
>     EXPECT_TRUE(vp8_info.layerSync) << "Key frame should be marked layer sync.";
>@@ -652,9 +653,9 @@ class TemporalLayersReferenceTest : public ::testing::TestWithParam<int> {
>     bool sync;
>   };
> 
>-  bool UpdateSyncRefState(const TemporalLayers::BufferFlags& flags,
>+  bool UpdateSyncRefState(const Vp8TemporalLayers::BufferFlags& flags,
>                           BufferState* buffer_state) {
>-    if (flags & TemporalLayers::kReference) {
>+    if (flags & Vp8TemporalLayers::kReference) {
>       if (buffer_state->temporal_idx == -1)
>         return true;  // References key-frame.
>       if (buffer_state->temporal_idx == 0) {
>@@ -668,10 +669,10 @@ class TemporalLayersReferenceTest : public ::testing::TestWithParam<int> {
>     return true;  // No reference, does not affect sync frame status.
>   }
> 
>-  void ValidateReference(const TemporalLayers::BufferFlags& flags,
>+  void ValidateReference(const Vp8TemporalLayers::BufferFlags& flags,
>                          const BufferState& buffer_state,
>                          int temporal_layer) {
>-    if (flags & TemporalLayers::kReference) {
>+    if (flags & Vp8TemporalLayers::kReference) {
>       if (temporal_layer > 0 && buffer_state.timestamp > 0) {
>         // Check that high layer reference does not go past last sync frame.
>         EXPECT_GE(buffer_state.timestamp, last_sync_timestamp_);
>@@ -709,9 +710,9 @@ TEST_P(TemporalLayersReferenceTest, ValidFrameConfigs) {
>   // (any). If a given buffer is never updated, it is legal to reference it
>   // even for sync frames. In order to be general, don't assume TL0 always
>   // updates |last|.
>-  std::vector<TemporalLayers::FrameConfig> tl_configs(kMaxPatternLength);
>+  std::vector<Vp8TemporalLayers::FrameConfig> tl_configs(kMaxPatternLength);
>   for (int i = 0; i < kMaxPatternLength; ++i) {
>-    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp_);
>+    Vp8TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp_);
>     tl.OnEncodeDone(timestamp_, kDefaultBytesPerFrame, i == 0, kDefaultQp,
>                     &vp8_specifics);
>     ++timestamp_;
>@@ -752,11 +753,11 @@ TEST_P(TemporalLayersReferenceTest, ValidFrameConfigs) {
> 
>     // Update the current layer state.
>     BufferState state = {temporal_idx, timestamp_, is_sync_frame};
>-    if (tl_config.last_buffer_flags & TemporalLayers::kUpdate)
>+    if (tl_config.last_buffer_flags & Vp8TemporalLayers::kUpdate)
>       last_state = state;
>-    if (tl_config.golden_buffer_flags & TemporalLayers::kUpdate)
>+    if (tl_config.golden_buffer_flags & Vp8TemporalLayers::kUpdate)
>       golden_state = state;
>-    if (tl_config.arf_buffer_flags & TemporalLayers::kUpdate)
>+    if (tl_config.arf_buffer_flags & Vp8TemporalLayers::kUpdate)
>       altref_state = state;
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/temporal_layers_checker.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/temporal_layers_checker.h
>index 9878ac9f3a1c6bdcd15bcfe79f94b14f02810090..ae14f682361c3f4549fbbb7661ea60fa09826eb0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/temporal_layers_checker.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/temporal_layers_checker.h
>@@ -15,7 +15,7 @@
> 
> #include <memory>
> 
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> 
> namespace webrtc {
> 
>@@ -29,10 +29,10 @@ class TemporalLayersChecker {
> 
>   virtual bool CheckTemporalConfig(
>       bool frame_is_keyframe,
>-      const TemporalLayers::FrameConfig& frame_config);
>+      const Vp8TemporalLayers::FrameConfig& frame_config);
> 
>   static std::unique_ptr<TemporalLayersChecker> CreateTemporalLayersChecker(
>-      TemporalLayersType type,
>+      Vp8TemporalLayersType type,
>       int num_temporal_layers);
> 
>  private:
>@@ -46,7 +46,7 @@ class TemporalLayersChecker {
>                                  bool* need_sync,
>                                  bool frame_is_keyframe,
>                                  uint8_t temporal_layer,
>-                                 webrtc::TemporalLayers::BufferFlags flags,
>+                                 webrtc::Vp8TemporalLayers::BufferFlags flags,
>                                  uint32_t sequence_number,
>                                  uint32_t* lowest_sequence_referenced);
>   BufferState last_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h
>deleted file mode 100644
>index b5dbc4ec2bfc2b6013ba4aa222a07de0e4fc38f9..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h
>+++ /dev/null
>@@ -1,198 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_TEMPORAL_LAYERS_H_
>-#define MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_TEMPORAL_LAYERS_H_
>-
>-#include <memory>
>-#include <vector>
>-
>-namespace webrtc {
>-
>-// Some notes on the prerequisites of the TemporalLayers interface.
>-// * Implementations of TemporalLayers may not contain internal synchronization
>-//   so caller must make sure doing so thread safe.
>-// * The encoder is assumed to encode all frames in order, and callbacks to
>-//   PopulateCodecSpecific() / FrameEncoded() must happen in the same order.
>-//
>-// This means that in the case of pipelining encoders, it is OK to have a chain
>-// of calls such as this:
>-// - UpdateLayerConfig(timestampA)
>-// - UpdateLayerConfig(timestampB)
>-// - PopulateCodecSpecific(timestampA, ...)
>-// - UpdateLayerConfig(timestampC)
>-// - OnEncodeDone(timestampA, 1234, ...)
>-// - UpdateLayerConfig(timestampC)
>-// - OnEncodeDone(timestampB, 0, ...)
>-// - OnEncodeDone(timestampC, 1234, ...)
>-// Note that UpdateLayerConfig() for a new frame can happen before
>-// FrameEncoded() for a previous one, but calls themselves must be both
>-// synchronized (e.g. run on a task queue) and in order (per type).
>-
>-enum class TemporalLayersType { kFixedPattern, kBitrateDynamic };
>-
>-struct CodecSpecificInfoVP8;
>-enum class Vp8BufferReference : uint8_t {
>-  kNone = 0,
>-  kLast = 1,
>-  kGolden = 2,
>-  kAltref = 4
>-};
>-
>-struct Vp8EncoderConfig {
>-  static constexpr size_t kMaxPeriodicity = 16;
>-  static constexpr size_t kMaxLayers = 5;
>-
>-  // Number of active temporal layers. Set to 0 if not used.
>-  uint32_t ts_number_layers;
>-  // Arrays of length |ts_number_layers|, indicating (cumulative) target bitrate
>-  // and rate decimator (e.g. 4 if every 4th frame is in the given layer) for
>-  // each active temporal layer, starting with temporal id 0.
>-  uint32_t ts_target_bitrate[kMaxLayers];
>-  uint32_t ts_rate_decimator[kMaxLayers];
>-
>-  // The periodicity of the temporal pattern. Set to 0 if not used.
>-  uint32_t ts_periodicity;
>-  // Array of length |ts_periodicity| indicating the sequence of temporal id's
>-  // to assign to incoming frames.
>-  uint32_t ts_layer_id[kMaxPeriodicity];
>-
>-  // Target bitrate, in bps.
>-  uint32_t rc_target_bitrate;
>-
>-  // Clamp QP to min/max. Use 0 to disable clamping.
>-  uint32_t rc_min_quantizer;
>-  uint32_t rc_max_quantizer;
>-};
>-
>-// This interface defines a way of getting the encoder settings needed to
>-// realize a temporal layer structure of predefined size.
>-class TemporalLayers {
>- public:
>-  enum BufferFlags : int {
>-    kNone = 0,
>-    kReference = 1,
>-    kUpdate = 2,
>-    kReferenceAndUpdate = kReference | kUpdate,
>-  };
>-  enum FreezeEntropy { kFreezeEntropy };
>-
>-  struct FrameConfig {
>-    FrameConfig();
>-
>-    FrameConfig(BufferFlags last, BufferFlags golden, BufferFlags arf);
>-    FrameConfig(BufferFlags last,
>-                BufferFlags golden,
>-                BufferFlags arf,
>-                FreezeEntropy);
>-
>-    bool drop_frame;
>-    BufferFlags last_buffer_flags;
>-    BufferFlags golden_buffer_flags;
>-    BufferFlags arf_buffer_flags;
>-
>-    // The encoder layer ID is used to utilize the correct bitrate allocator
>-    // inside the encoder. It does not control references nor determine which
>-    // "actual" temporal layer this is. The packetizer temporal index determines
>-    // which layer the encoded frame should be packetized into.
>-    // Normally these are the same, but current temporal-layer strategies for
>-    // screenshare use one bitrate allocator for all layers, but attempt to
>-    // packetize / utilize references to split a stream into multiple layers,
>-    // with different quantizer settings, to hit target bitrate.
>-    // TODO(pbos): Screenshare layers are being reconsidered at the time of
>-    // writing, we might be able to remove this distinction, and have a temporal
>-    // layer imply both (the normal case).
>-    int encoder_layer_id;
>-    int packetizer_temporal_idx;
>-
>-    bool layer_sync;
>-
>-    bool freeze_entropy;
>-
>-    // Indicates in which order the encoder should search the reference buffers
>-    // when doing motion prediction. Set to kNone to use unspecified order. Any
>-    // buffer indicated here must not have the corresponding no_ref bit set.
>-    // If all three buffers can be reference, the one not listed here should be
>-    // searched last.
>-    Vp8BufferReference first_reference;
>-    Vp8BufferReference second_reference;
>-
>-    bool operator==(const FrameConfig& o) const;
>-    bool operator!=(const FrameConfig& o) const { return !(*this == o); }
>-
>-   private:
>-    FrameConfig(BufferFlags last,
>-                BufferFlags golden,
>-                BufferFlags arf,
>-                bool freeze_entropy);
>-  };
>-
>-  // Factory for TemporalLayer strategy. Default behavior is a fixed pattern
>-  // of temporal layers. See default_temporal_layers.cc
>-  static std::unique_ptr<TemporalLayers> CreateTemporalLayers(
>-      TemporalLayersType type,
>-      int num_temporal_layers);
>-
>-  virtual ~TemporalLayers() = default;
>-
>-  // If this method returns true, the encoder is free to drop frames for
>-  // instance in an effort to uphold encoding bitrate.
>-  // If this return false, the encoder must not drop any frames unless:
>-  //  1. Requested to do so via FrameConfig.drop_frame
>-  //  2. The frame to be encoded is requested to be a keyframe
>-  //  3. The encoded detected a large overshoot and decided to drop and then
>-  //     re-encode the image at a low bitrate. In this case the encoder should
>-  //     call OnEncodeDone() once with size = 0 to indicate drop, and then call
>-  //     OnEncodeDone() again when the frame has actually been encoded.
>-  virtual bool SupportsEncoderFrameDropping() const = 0;
>-
>-  // New target bitrate, per temporal layer.
>-  virtual void OnRatesUpdated(const std::vector<uint32_t>& bitrates_bps,
>-                              int framerate_fps) = 0;
>-
>-  // Called by the encoder before encoding a frame. |cfg| contains the current
>-  // configuration. If the TemporalLayers instance wishes any part of that
>-  // to be changed before the encode step, |cfg| should be changed and then
>-  // return true. If false is returned, the encoder will proceed without
>-  // updating the configuration.
>-  virtual bool UpdateConfiguration(Vp8EncoderConfig* cfg) = 0;
>-
>-  // Returns the recommended VP8 encode flags needed, and moves the temporal
>-  // pattern to the next frame.
>-  // The timestamp may be used as both a time and a unique identifier, and so
>-  // the caller must make sure no two frames use the same timestamp.
>-  // The timestamp uses a 90kHz RTP clock.
>-  // After calling this method, first call the actual encoder with the provided
>-  // frame configuration, and then OnEncodeDone() below.
>-  virtual FrameConfig UpdateLayerConfig(uint32_t rtp_timestamp) = 0;
>-
>-  // Called after the encode step is done. |rtp_timestamp| must match the
>-  // parameter use in the UpdateLayerConfig() call.
>-  // |is_keyframe| must be true iff the encoder decided to encode this frame as
>-  // a keyframe.
>-  // If the encoder decided to drop this frame, |size_bytes| must be set to 0,
>-  // otherwise it should indicate the size in bytes of the encoded frame.
>-  // If |size_bytes| > 0, and |vp8_info| is not null, the TemporalLayers
>-  // instance my update |vp8_info| with codec specific data such as temporal id.
>-  // Some fields of this struct may have already been populated by the encoder,
>-  // check before overwriting.
>-  // If |size_bytes| > 0, |qp| should indicate the frame-level QP this frame was
>-  // encoded at. If the encoder does not support extracting this, |qp| should be
>-  // set to 0.
>-  virtual void OnEncodeDone(uint32_t rtp_timestamp,
>-                            size_t size_bytes,
>-                            bool is_keyframe,
>-                            int qp,
>-                            CodecSpecificInfoVP8* vp8_info) = 0;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_TEMPORAL_LAYERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
>index 20684d1d39185e62e10fc713e3d9953588682913..48838e93e7f369b6ca0dc672f5545809bcc257ec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
>@@ -12,7 +12,6 @@
> #include <string>
> 
> #include "absl/memory/memory.h"
>-#include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/exp_filter.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h
>index 3a6cf92405cc7edc8b5a304a454d0cd63205cb48..cff9e8998841b9c3848469455dff36bbf88f78f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h
>@@ -13,10 +13,10 @@
> 
> #include <memory>
> 
>+#include "api/video/encoded_image.h"
> #include "api/video_codecs/video_decoder.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "common_video/include/i420_buffer_pool.h"
>-#include "common_video/include/video_frame.h"
> #include "modules/include/module_common_types.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "modules/video_coding/include/video_codec_interface.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
>index 41dad40d9e40bf3f9c6fa0ea3bd6463de960a17b..be5242c1b9168b06f9adb1d6604e4a9c016f8c53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
>@@ -14,6 +14,8 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>+#include "api/video_codecs/create_vp8_temporal_layers.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h"
> #include "modules/video_coding/utility/simulcast_rate_allocator.h"
>@@ -27,7 +29,8 @@
> 
> namespace webrtc {
> namespace {
>-const char kVp8GfBoostFieldTrial[] = "WebRTC-VP8-GfBoost";
>+const char kVp8TrustedRateControllerFieldTrial[] =
>+    "WebRTC-LibvpxVp8TrustedRateController";
> 
> // QP is obtained from VP8-bitstream for HW, so the QP corresponds to the
> // bitstream range of [0, 127] and not the user-level range of [0,63].
>@@ -59,20 +62,6 @@ static int GCD(int a, int b) {
>   return b;
> }
> 
>-bool GetGfBoostPercentageFromFieldTrialGroup(int* boost_percentage) {
>-  std::string group = webrtc::field_trial::FindFullName(kVp8GfBoostFieldTrial);
>-  if (group.empty())
>-    return false;
>-
>-  if (sscanf(group.c_str(), "Enabled-%d", boost_percentage) != 1)
>-    return false;
>-
>-  if (*boost_percentage < 0 || *boost_percentage > 100)
>-    return false;
>-
>-  return true;
>-}
>-
> static_assert(Vp8EncoderConfig::kMaxPeriodicity == VPX_TS_MAX_PERIODICITY,
>               "Vp8EncoderConfig::kMaxPeriodicity must be kept in sync with the "
>               "constant in libvpx.");
>@@ -113,7 +102,7 @@ static void FillInEncoderConfig(vpx_codec_enc_cfg* vpx_config,
>   vpx_config->rc_max_quantizer = config.rc_max_quantizer;
> }
> 
>-bool UpdateVpxConfiguration(TemporalLayers* temporal_layers,
>+bool UpdateVpxConfiguration(Vp8TemporalLayers* temporal_layers,
>                             vpx_codec_enc_cfg_t* cfg) {
>   Vp8EncoderConfig config = GetEncoderConfig(cfg);
>   const bool res = temporal_layers->UpdateConfiguration(&config);
>@@ -129,22 +118,22 @@ std::unique_ptr<VideoEncoder> VP8Encoder::Create() {
> }
> 
> vpx_enc_frame_flags_t LibvpxVp8Encoder::EncodeFlags(
>-    const TemporalLayers::FrameConfig& references) {
>+    const Vp8TemporalLayers::FrameConfig& references) {
>   RTC_DCHECK(!references.drop_frame);
> 
>   vpx_enc_frame_flags_t flags = 0;
> 
>-  if ((references.last_buffer_flags & TemporalLayers::kReference) == 0)
>+  if ((references.last_buffer_flags & Vp8TemporalLayers::kReference) == 0)
>     flags |= VP8_EFLAG_NO_REF_LAST;
>-  if ((references.last_buffer_flags & TemporalLayers::kUpdate) == 0)
>+  if ((references.last_buffer_flags & Vp8TemporalLayers::kUpdate) == 0)
>     flags |= VP8_EFLAG_NO_UPD_LAST;
>-  if ((references.golden_buffer_flags & TemporalLayers::kReference) == 0)
>+  if ((references.golden_buffer_flags & Vp8TemporalLayers::kReference) == 0)
>     flags |= VP8_EFLAG_NO_REF_GF;
>-  if ((references.golden_buffer_flags & TemporalLayers::kUpdate) == 0)
>+  if ((references.golden_buffer_flags & Vp8TemporalLayers::kUpdate) == 0)
>     flags |= VP8_EFLAG_NO_UPD_GF;
>-  if ((references.arf_buffer_flags & TemporalLayers::kReference) == 0)
>+  if ((references.arf_buffer_flags & Vp8TemporalLayers::kReference) == 0)
>     flags |= VP8_EFLAG_NO_REF_ARF;
>-  if ((references.arf_buffer_flags & TemporalLayers::kUpdate) == 0)
>+  if ((references.arf_buffer_flags & Vp8TemporalLayers::kUpdate) == 0)
>     flags |= VP8_EFLAG_NO_UPD_ARF;
>   if (references.freeze_entropy)
>     flags |= VP8_EFLAG_NO_UPD_ENTROPY;
>@@ -157,7 +146,9 @@ LibvpxVp8Encoder::LibvpxVp8Encoder()
> 
> LibvpxVp8Encoder::LibvpxVp8Encoder(std::unique_ptr<LibvpxInterface> interface)
>     : libvpx_(std::move(interface)),
>-      use_gf_boost_(webrtc::field_trial::IsEnabled(kVp8GfBoostFieldTrial)),
>+      experimental_cpu_speed_config_arm_(CpuSpeedExperiment::GetConfigs()),
>+      trusted_rate_controller_(
>+          field_trial::IsEnabled(kVp8TrustedRateControllerFieldTrial)),
>       encoded_complete_callback_(nullptr),
>       inited_(false),
>       timestamp_(0),
>@@ -167,7 +158,6 @@ LibvpxVp8Encoder::LibvpxVp8Encoder(std::unique_ptr<LibvpxInterface> interface)
>       rc_max_intra_target_(0),
>       key_frame_request_(kMaxSimulcastStreams, false) {
>   temporal_layers_.reserve(kMaxSimulcastStreams);
>-  temporal_layers_checkers_.reserve(kMaxSimulcastStreams);
>   raw_images_.reserve(kMaxSimulcastStreams);
>   encoded_images_.reserve(kMaxSimulcastStreams);
>   send_stream_.reserve(kMaxSimulcastStreams);
>@@ -206,7 +196,6 @@ int LibvpxVp8Encoder::Release() {
>     raw_images_.pop_back();
>   }
>   temporal_layers_.clear();
>-  temporal_layers_checkers_.clear();
>   inited_ = false;
>   return ret_val;
> }
>@@ -278,10 +267,6 @@ int LibvpxVp8Encoder::SetRateAllocation(const VideoBitrateAllocation& bitrate,
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-const char* LibvpxVp8Encoder::ImplementationName() const {
>-  return "libvpx";
>-}
>-
> void LibvpxVp8Encoder::SetStreamState(bool send_stream, int stream_idx) {
>   if (send_stream && !send_stream_[stream_idx]) {
>     // Need a key frame if we have not sent this stream before.
>@@ -294,21 +279,18 @@ void LibvpxVp8Encoder::SetupTemporalLayers(const VideoCodec& codec) {
>   RTC_DCHECK(temporal_layers_.empty());
>   int num_streams = SimulcastUtility::NumberOfSimulcastStreams(codec);
>   for (int i = 0; i < num_streams; ++i) {
>-    TemporalLayersType type;
>+    Vp8TemporalLayersType type;
>     int num_temporal_layers =
>         SimulcastUtility::NumberOfTemporalLayers(codec, i);
>     if (SimulcastUtility::IsConferenceModeScreenshare(codec) && i == 0) {
>-      type = TemporalLayersType::kBitrateDynamic;
>+      type = Vp8TemporalLayersType::kBitrateDynamic;
>       // Legacy screenshare layers supports max 2 layers.
>       num_temporal_layers = std::max<int>(2, num_temporal_layers);
>     } else {
>-      type = TemporalLayersType::kFixedPattern;
>+      type = Vp8TemporalLayersType::kFixedPattern;
>     }
>     temporal_layers_.emplace_back(
>-        TemporalLayers::CreateTemporalLayers(type, num_temporal_layers));
>-    temporal_layers_checkers_.emplace_back(
>-        TemporalLayersChecker::CreateTemporalLayersChecker(
>-            type, num_temporal_layers));
>+        CreateVp8TemporalLayers(type, num_temporal_layers));
>   }
> }
> 
>@@ -453,10 +435,10 @@ int LibvpxVp8Encoder::InitEncode(const VideoCodec* inst,
>   }
>   cpu_speed_default_ = cpu_speed_[0];
>   // Set encoding complexity (cpu_speed) based on resolution and/or platform.
>-  cpu_speed_[0] = SetCpuSpeed(inst->width, inst->height);
>+  cpu_speed_[0] = GetCpuSpeed(inst->width, inst->height);
>   for (int i = 1; i < number_of_streams; ++i) {
>     cpu_speed_[i] =
>-        SetCpuSpeed(inst->simulcastStream[number_of_streams - 1 - i].width,
>+        GetCpuSpeed(inst->simulcastStream[number_of_streams - 1 - i].width,
>                     inst->simulcastStream[number_of_streams - 1 - i].height);
>   }
>   configurations_[0].g_w = inst->width;
>@@ -528,7 +510,7 @@ int LibvpxVp8Encoder::InitEncode(const VideoCodec* inst,
>   return InitAndSetControlSettings();
> }
> 
>-int LibvpxVp8Encoder::SetCpuSpeed(int width, int height) {
>+int LibvpxVp8Encoder::GetCpuSpeed(int width, int height) {
> #if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
>     defined(WEBRTC_ANDROID)
>   // On mobile platform, use a lower speed setting for lower resolutions for
>@@ -537,6 +519,11 @@ int LibvpxVp8Encoder::SetCpuSpeed(int width, int height) {
>   if (number_of_cores_ <= 3)
>     return -12;
> 
>+  if (experimental_cpu_speed_config_arm_) {
>+    return CpuSpeedExperiment::GetValue(width * height,
>+                                        *experimental_cpu_speed_config_arm_);
>+  }
>+
>   if (width * height <= 352 * 288)
>     return -8;
>   else if (width * height <= 640 * 480)
>@@ -644,14 +631,6 @@ int LibvpxVp8Encoder::InitAndSetControlSettings() {
>     libvpx_->codec_control(
>         &(encoders_[i]), VP8E_SET_SCREEN_CONTENT_MODE,
>         codec_.mode == VideoCodecMode::kScreensharing ? 2u : 0u);
>-    // Apply boost on golden frames (has only effect when resilience is off).
>-    if (use_gf_boost_ && configurations_[0].g_error_resilient == 0) {
>-      int gf_boost_percent;
>-      if (GetGfBoostPercentageFromFieldTrialGroup(&gf_boost_percent)) {
>-        libvpx_->codec_control(&(encoders_[i]), VP8E_SET_GF_CBR_BOOST_PCT,
>-                               gf_boost_percent);
>-      }
>-    }
>   }
>   inited_ = true;
>   return WEBRTC_VIDEO_CODEC_OK;
>@@ -750,7 +729,7 @@ int LibvpxVp8Encoder::Encode(const VideoFrame& frame,
>     }
>   }
>   vpx_enc_frame_flags_t flags[kMaxSimulcastStreams];
>-  TemporalLayers::FrameConfig tl_configs[kMaxSimulcastStreams];
>+  Vp8TemporalLayers::FrameConfig tl_configs[kMaxSimulcastStreams];
>   for (size_t i = 0; i < encoders_.size(); ++i) {
>     tl_configs[i] = temporal_layers_[i]->UpdateLayerConfig(frame.timestamp());
>     if (tl_configs[i].drop_frame) {
>@@ -831,10 +810,11 @@ int LibvpxVp8Encoder::Encode(const VideoFrame& frame,
>     }
>     if (error)
>       return WEBRTC_VIDEO_CODEC_ERROR;
>-    timestamp_ += duration;
>     // Examines frame timestamps only.
>     error = GetEncodedPartitions(frame);
>   }
>+  // TODO(sprang): Shouldn't we use the frame timestamp instead?
>+  timestamp_ += duration;
>   return error;
> }
> 
>@@ -845,7 +825,6 @@ void LibvpxVp8Encoder::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>                                              uint32_t timestamp) {
>   assert(codec_specific != NULL);
>   codec_specific->codecType = kVideoCodecVP8;
>-  codec_specific->codec_name = ImplementationName();
>   CodecSpecificInfoVP8* vp8Info = &(codec_specific->codecSpecific.VP8);
>   vp8Info->keyIdx = kNoKeyIdx;  // TODO(hlundin) populate this
>   vp8Info->nonReference = (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE) != 0;
>@@ -863,14 +842,9 @@ int LibvpxVp8Encoder::GetEncodedPartitions(const VideoFrame& input_image) {
>   for (size_t encoder_idx = 0; encoder_idx < encoders_.size();
>        ++encoder_idx, --stream_idx) {
>     vpx_codec_iter_t iter = NULL;
>-    int part_idx = 0;
>     encoded_images_[encoder_idx]._length = 0;
>     encoded_images_[encoder_idx]._frameType = kVideoFrameDelta;
>-    RTPFragmentationHeader frag_info;
>-    // kTokenPartitions is number of bits used.
>-    frag_info.VerifyAndAllocateFragmentationHeader((1 << kTokenPartitions) + 1);
>     CodecSpecificInfo codec_specific;
>-    bool is_keyframe = false;
>     const vpx_codec_cx_pkt_t* pkt = NULL;
>     while ((pkt = libvpx_->codec_get_cx_data(&encoders_[encoder_idx], &iter)) !=
>            NULL) {
>@@ -887,13 +861,8 @@ int LibvpxVp8Encoder::GetEncodedPartitions(const VideoFrame& input_image) {
>           }
>           memcpy(&encoded_images_[encoder_idx]._buffer[length],
>                  pkt->data.frame.buf, pkt->data.frame.sz);
>-          frag_info.fragmentationOffset[part_idx] = length;
>-          frag_info.fragmentationLength[part_idx] = pkt->data.frame.sz;
>-          frag_info.fragmentationPlType[part_idx] = 0;  // not known here
>-          frag_info.fragmentationTimeDiff[part_idx] = 0;
>           encoded_images_[encoder_idx]._length += pkt->data.frame.sz;
>           assert(length <= encoded_images_[encoder_idx]._size);
>-          ++part_idx;
>           break;
>         }
>         default:
>@@ -904,7 +873,6 @@ int LibvpxVp8Encoder::GetEncodedPartitions(const VideoFrame& input_image) {
>         // check if encoded frame is a key frame
>         if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
>           encoded_images_[encoder_idx]._frameType = kVideoFrameKey;
>-          is_keyframe = true;
>         }
>         encoded_images_[encoder_idx].SetSpatialIndex(stream_idx);
>         PopulateCodecSpecific(&codec_specific, *pkt, stream_idx, encoder_idx,
>@@ -935,7 +903,7 @@ int LibvpxVp8Encoder::GetEncodedPartitions(const VideoFrame& input_image) {
>                                &qp_128);
>         encoded_images_[encoder_idx].qp_ = qp_128;
>         encoded_complete_callback_->OnEncodedImage(encoded_images_[encoder_idx],
>-                                                   &codec_specific, &frag_info);
>+                                                   &codec_specific, nullptr);
>       } else if (!temporal_layers_[stream_idx]
>                       ->SupportsEncoderFrameDropping()) {
>         result = WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT;
>@@ -950,17 +918,21 @@ int LibvpxVp8Encoder::GetEncodedPartitions(const VideoFrame& input_image) {
>   return result;
> }
> 
>-VideoEncoder::ScalingSettings LibvpxVp8Encoder::GetScalingSettings() const {
>+VideoEncoder::EncoderInfo LibvpxVp8Encoder::GetEncoderInfo() const {
>+  EncoderInfo info;
>+  info.supports_native_handle = false;
>+  info.implementation_name = "libvpx";
>+  info.has_trusted_rate_controller = trusted_rate_controller_;
>+
>   const bool enable_scaling = encoders_.size() == 1 &&
>                               configurations_[0].rc_dropframe_thresh > 0 &&
>                               codec_.VP8().automaticResizeOn;
>-  return enable_scaling ? VideoEncoder::ScalingSettings(kLowVp8QpThreshold,
>-                                                        kHighVp8QpThreshold)
>-                        : VideoEncoder::ScalingSettings::kOff;
>-}
>+  info.scaling_settings = enable_scaling
>+                              ? VideoEncoder::ScalingSettings(
>+                                    kLowVp8QpThreshold, kHighVp8QpThreshold)
>+                              : VideoEncoder::ScalingSettings::kOff;
> 
>-int LibvpxVp8Encoder::SetChannelParameters(uint32_t packetLoss, int64_t rtt) {
>-  return WEBRTC_VIDEO_CODEC_OK;
>+  return info;
> }
> 
> int LibvpxVp8Encoder::RegisterEncodeCompleteCallback(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h
>index 3c20672df2cd4e6041a0fb5389a8425e425b6a31..df2dbcee59e663ac27d3a7b11e2d11ad4f499e92 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h
>@@ -14,15 +14,15 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/video/encoded_image.h"
> #include "api/video/video_frame.h"
> #include "api/video_codecs/video_encoder.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/video_frame.h"
>-#include "modules/video_coding/codecs/vp8/include/temporal_layers_checker.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
> #include "modules/video_coding/codecs/vp8/libvpx_interface.h"
> #include "modules/video_coding/include/video_codec_interface.h"
>+#include "rtc_base/experiments/cpu_speed_experiment.h"
> 
> #include "vpx/vp8cx.h"
> #include "vpx/vpx_encoder.h"
>@@ -47,23 +47,19 @@ class LibvpxVp8Encoder : public VideoEncoder {
> 
>   int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
> 
>-  int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>-
>   int SetRateAllocation(const VideoBitrateAllocation& bitrate,
>                         uint32_t new_framerate) override;
> 
>-  ScalingSettings GetScalingSettings() const override;
>-
>-  const char* ImplementationName() const override;
>+  EncoderInfo GetEncoderInfo() const override;
> 
>   static vpx_enc_frame_flags_t EncodeFlags(
>-      const TemporalLayers::FrameConfig& references);
>+      const Vp8TemporalLayers::FrameConfig& references);
> 
>  private:
>   void SetupTemporalLayers(const VideoCodec& codec);
> 
>-  // Set the cpu_speed setting for encoder based on resolution and/or platform.
>-  int SetCpuSpeed(int width, int height);
>+  // Get the cpu_speed setting for encoder based on resolution and/or platform.
>+  int GetCpuSpeed(int width, int height);
> 
>   // Determine number of encoder threads to use.
>   int NumberOfThreads(int width, int height, int number_of_cores);
>@@ -87,7 +83,10 @@ class LibvpxVp8Encoder : public VideoEncoder {
>   uint32_t FrameDropThreshold(size_t spatial_idx) const;
> 
>   const std::unique_ptr<LibvpxInterface> libvpx_;
>-  const bool use_gf_boost_;
>+
>+  const absl::optional<std::vector<CpuSpeedExperiment::Config>>
>+      experimental_cpu_speed_config_arm_;
>+  const bool trusted_rate_controller_;
> 
>   EncodedImageCallback* encoded_complete_callback_;
>   VideoCodec codec_;
>@@ -97,8 +96,7 @@ class LibvpxVp8Encoder : public VideoEncoder {
>   int cpu_speed_default_;
>   int number_of_cores_;
>   uint32_t rc_max_intra_target_;
>-  std::vector<std::unique_ptr<TemporalLayers>> temporal_layers_;
>-  std::vector<std::unique_ptr<TemporalLayersChecker>> temporal_layers_checkers_;
>+  std::vector<std::unique_ptr<Vp8TemporalLayers>> temporal_layers_;
>   std::vector<bool> key_frame_request_;
>   std::vector<bool> send_stream_;
>   std::vector<int> cpu_speed_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_simulcast_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_simulcast_test.cc
>index 5f974d643bb0ee32cd6f76dbe00ae3b19f155d1e..d493e2aceb648771fa95ed5e23cc208e30ab2c1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_simulcast_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_simulcast_test.cc
>@@ -13,9 +13,9 @@
> #include "absl/memory/memory.h"
> #include "api/test/create_simulcast_test_fixture.h"
> #include "api/test/simulcast_test_fixture.h"
>+#include "api/test/video/function_video_decoder_factory.h"
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>-#include "test/function_video_decoder_factory.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
>index 76c4382a21f8dfeece7ee26cc0ff78d31e2fed46..ede21e554d72648a99005e8a77d9cbda9f6a725e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
>@@ -53,7 +53,7 @@ ScreenshareLayers::ScreenshareLayers(int num_temporal_layers, Clock* clock)
>       encode_framerate_(1000.0f, 1000.0f),  // 1 second window, second scale.
>       bitrate_updated_(false),
>       checker_(TemporalLayersChecker::CreateTemporalLayersChecker(
>-          TemporalLayersType::kBitrateDynamic,
>+          Vp8TemporalLayersType::kBitrateDynamic,
>           num_temporal_layers)) {
>   RTC_CHECK_GT(number_of_temporal_layers_, 0);
>   RTC_CHECK_LE(number_of_temporal_layers_, kMaxNumTemporalLayers);
>@@ -68,7 +68,7 @@ bool ScreenshareLayers::SupportsEncoderFrameDropping() const {
>   return false;
> }
> 
>-TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>+Vp8TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>     uint32_t timestamp) {
>   auto it = pending_frame_configs_.find(timestamp);
>   if (it != pending_frame_configs_.end()) {
>@@ -79,7 +79,7 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>   if (number_of_temporal_layers_ <= 1) {
>     // No flags needed for 1 layer screenshare.
>     // TODO(pbos): Consider updating only last, and not all buffers.
>-    TemporalLayers::FrameConfig tl_config(
>+    Vp8TemporalLayers::FrameConfig tl_config(
>         kReferenceAndUpdate, kReferenceAndUpdate, kReferenceAndUpdate);
>     pending_frame_configs_[timestamp] = tl_config;
>     return tl_config;
>@@ -100,7 +100,7 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>     // averaging window, or if frame interval is below 90% of desired value,
>     // drop frame.
>     if (encode_framerate_.Rate(now_ms).value_or(0) > *target_framerate_)
>-      return TemporalLayers::FrameConfig(kNone, kNone, kNone);
>+      return Vp8TemporalLayers::FrameConfig(kNone, kNone, kNone);
> 
>     // Primarily check if frame interval is too short using frame timestamps,
>     // as if they are correct they won't be affected by queuing in webrtc.
>@@ -108,7 +108,7 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>         kOneSecond90Khz / *target_framerate_;
>     if (last_timestamp_ != -1 && ts_diff > 0) {
>       if (ts_diff < 85 * expected_frame_interval_90khz / 100) {
>-        return TemporalLayers::FrameConfig(kNone, kNone, kNone);
>+        return Vp8TemporalLayers::FrameConfig(kNone, kNone, kNone);
>       }
>     } else {
>       // Timestamps looks off, use realtime clock here instead.
>@@ -116,7 +116,7 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>       if (last_frame_time_ms_ != -1 &&
>           now_ms - last_frame_time_ms_ <
>               (85 * expected_frame_interval_ms) / 100) {
>-        return TemporalLayers::FrameConfig(kNone, kNone, kNone);
>+        return Vp8TemporalLayers::FrameConfig(kNone, kNone, kNone);
>       }
>     }
>   }
>@@ -182,30 +182,30 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>       RTC_NOTREACHED();
>   }
> 
>-  TemporalLayers::FrameConfig tl_config;
>+  Vp8TemporalLayers::FrameConfig tl_config;
>   // TODO(pbos): Consider referencing but not updating the 'alt' buffer for all
>   // layers.
>   switch (layer_state) {
>     case TemporalLayerState::kDrop:
>-      tl_config = TemporalLayers::FrameConfig(kNone, kNone, kNone);
>+      tl_config = Vp8TemporalLayers::FrameConfig(kNone, kNone, kNone);
>       break;
>     case TemporalLayerState::kTl0:
>       // TL0 only references and updates 'last'.
>       tl_config =
>-          TemporalLayers::FrameConfig(kReferenceAndUpdate, kNone, kNone);
>+          Vp8TemporalLayers::FrameConfig(kReferenceAndUpdate, kNone, kNone);
>       tl_config.packetizer_temporal_idx = 0;
>       break;
>     case TemporalLayerState::kTl1:
>       // TL1 references both 'last' and 'golden' but only updates 'golden'.
>-      tl_config =
>-          TemporalLayers::FrameConfig(kReference, kReferenceAndUpdate, kNone);
>+      tl_config = Vp8TemporalLayers::FrameConfig(kReference,
>+                                                 kReferenceAndUpdate, kNone);
>       tl_config.packetizer_temporal_idx = 1;
>       break;
>     case TemporalLayerState::kTl1Sync:
>       // Predict from only TL0 to allow participants to switch to the high
>       // bitrate stream. Updates 'golden' so that TL1 can continue to refer to
>       // and update 'golden' from this point on.
>-      tl_config = TemporalLayers::FrameConfig(kReference, kUpdate, kNone);
>+      tl_config = Vp8TemporalLayers::FrameConfig(kReference, kUpdate, kNone);
>       tl_config.packetizer_temporal_idx = 1;
>       break;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
>index 6193be360010df66b7b4f281f2a4c6b4908aa238..0359b2c33fc590cc0fcc1ceca4425e5576b57d43 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
>@@ -13,8 +13,8 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "modules/video_coding/codecs/vp8/include/temporal_layers_checker.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
> #include "modules/video_coding/utility/frame_dropper.h"
> #include "rtc_base/rate_statistics.h"
> #include "rtc_base/timeutils.h"
>@@ -24,7 +24,7 @@ namespace webrtc {
> struct CodecSpecificInfoVP8;
> class Clock;
> 
>-class ScreenshareLayers : public TemporalLayers {
>+class ScreenshareLayers : public Vp8TemporalLayers {
>  public:
>   static const double kMaxTL0FpsReduction;
>   static const double kAcceptableTargetOvershoot;
>@@ -38,7 +38,7 @@ class ScreenshareLayers : public TemporalLayers {
> 
>   // Returns the recommended VP8 encode flags needed. May refresh the decoder
>   // and/or update the reference buffers.
>-  TemporalLayers::FrameConfig UpdateLayerConfig(
>+  Vp8TemporalLayers::FrameConfig UpdateLayerConfig(
>       uint32_t rtp_timestamp) override;
> 
>   // New target bitrate, per temporal layer.
>@@ -74,7 +74,7 @@ class ScreenshareLayers : public TemporalLayers {
>   int max_qp_;
>   uint32_t max_debt_bytes_;
> 
>-  std::map<uint32_t, TemporalLayers::FrameConfig> pending_frame_configs_;
>+  std::map<uint32_t, Vp8TemporalLayers::FrameConfig> pending_frame_configs_;
> 
>   // Configured max framerate.
>   absl::optional<uint32_t> target_framerate_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
>index 4ae7ed5e7e425b6ae5c708db404dbcc0c53f4cf4..474e4f825b597ee10e4d964415fa7bc5f679c9fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
>@@ -87,7 +87,7 @@ class ScreenshareLayerTest : public ::testing::Test {
>     return flags;
>   }
> 
>-  TemporalLayers::FrameConfig UpdateLayerConfig(uint32_t timestamp) {
>+  Vp8TemporalLayers::FrameConfig UpdateLayerConfig(uint32_t timestamp) {
>     int64_t timestamp_ms = timestamp / 90;
>     clock_.AdvanceTimeMilliseconds(timestamp_ms - clock_.TimeInMilliseconds());
>     return layers_->UpdateLayerConfig(timestamp);
>@@ -167,7 +167,7 @@ class ScreenshareLayerTest : public ::testing::Test {
>   std::unique_ptr<ScreenshareLayers> layers_;
> 
>   uint32_t timestamp_;
>-  TemporalLayers::FrameConfig tl_config_;
>+  Vp8TemporalLayers::FrameConfig tl_config_;
>   Vp8EncoderConfig cfg_;
>   bool config_updated_;
>   CodecSpecificInfoVP8 vp8_info_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
>index 788d69a27bb3174e622af0450809ce3851c75880..9576fb27beaa6317c11945e13da08b43707078e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
>@@ -12,6 +12,6 @@
> #define MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
> 
> // TODO(webrtc:9012) Remove this file when downstream projects have updated.
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> 
> #endif  // MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc
>index c0a736a4edda9fd94c01c07a2f564afe19e070ad..4dba7a366a9d928fbe408c0c42faa7854e3e5235 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers_checker.cc
>@@ -17,13 +17,13 @@
> namespace webrtc {
> 
> std::unique_ptr<TemporalLayersChecker>
>-TemporalLayersChecker::CreateTemporalLayersChecker(TemporalLayersType type,
>+TemporalLayersChecker::CreateTemporalLayersChecker(Vp8TemporalLayersType type,
>                                                    int num_temporal_layers) {
>   switch (type) {
>-    case TemporalLayersType::kFixedPattern:
>+    case Vp8TemporalLayersType::kFixedPattern:
>       return absl::make_unique<DefaultTemporalLayersChecker>(
>           num_temporal_layers);
>-    case TemporalLayersType::kBitrateDynamic:
>+    case Vp8TemporalLayersType::kBitrateDynamic:
>       // Conference mode temporal layering for screen content in base stream.
>       return absl::make_unique<TemporalLayersChecker>(num_temporal_layers);
>   }
>@@ -40,10 +40,10 @@ bool TemporalLayersChecker::CheckAndUpdateBufferState(
>     bool* need_sync,
>     bool frame_is_keyframe,
>     uint8_t temporal_layer,
>-    webrtc::TemporalLayers::BufferFlags flags,
>+    webrtc::Vp8TemporalLayers::BufferFlags flags,
>     uint32_t sequence_number,
>     uint32_t* lowest_sequence_referenced) {
>-  if (flags & TemporalLayers::BufferFlags::kReference) {
>+  if (flags & Vp8TemporalLayers::BufferFlags::kReference) {
>     if (state->temporal_layer > 0 && !state->is_keyframe) {
>       *need_sync = false;
>     }
>@@ -57,7 +57,7 @@ bool TemporalLayersChecker::CheckAndUpdateBufferState(
>       return false;
>     }
>   }
>-  if ((flags & TemporalLayers::BufferFlags::kUpdate)) {
>+  if ((flags & Vp8TemporalLayers::BufferFlags::kUpdate)) {
>     state->temporal_layer = temporal_layer;
>     state->sequence_number = sequence_number;
>     state->is_keyframe = frame_is_keyframe;
>@@ -69,7 +69,7 @@ bool TemporalLayersChecker::CheckAndUpdateBufferState(
> 
> bool TemporalLayersChecker::CheckTemporalConfig(
>     bool frame_is_keyframe,
>-    const TemporalLayers::FrameConfig& frame_config) {
>+    const Vp8TemporalLayers::FrameConfig& frame_config) {
>   if (frame_config.drop_frame ||
>       frame_config.packetizer_temporal_idx == kNoTemporalIdx) {
>     return true;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/mock_libvpx_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/mock_libvpx_interface.h
>index 5804e814ec9148858ddc83dc7e776b527c57eef2..dcff1e6a18846adbe3de419026e1773e798c6ac7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/mock_libvpx_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/mock_libvpx_interface.h
>@@ -32,7 +32,7 @@ class MockLibvpxVp8Interface : public LibvpxInterface {
>                                   unsigned int,
>                                   unsigned int,
>                                   unsigned char*));
>-  MOCK_CONST_METHOD1(img_free, vpx_codec_err_t(const vpx_codec_enc_cfg_t*));
>+  MOCK_CONST_METHOD1(img_free, void(vpx_image_t* img));
>   MOCK_CONST_METHOD2(codec_enc_config_set,
>                      vpx_codec_err_t(vpx_codec_ctx_t*,
>                                      const vpx_codec_enc_cfg_t*));
>@@ -84,7 +84,7 @@ class MockLibvpxVp8Interface : public LibvpxInterface {
>                                      uint64_t,
>                                      vpx_enc_frame_flags_t,
>                                      uint64_t));
>-  MOCK_CONST_METHOD3(codec_get_cx_data,
>+  MOCK_CONST_METHOD2(codec_get_cx_data,
>                      const vpx_codec_cx_pkt_t*(vpx_codec_ctx_t*,
>                                                vpx_codec_iter_t*));
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
>index 45a090df3736de910663445cabd769a92b19c109..5eed5d380cf6a6f012ce1f80d2d90d7550a49a84 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
>@@ -12,16 +12,25 @@
> 
> #include <memory>
> 
>+#include "api/test/mock_video_decoder.h"
>+#include "api/test/mock_video_encoder.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "modules/video_coding/codecs/test/video_codec_unittest.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
>+#include "modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h"
>+#include "modules/video_coding/codecs/vp8/test/mock_libvpx_interface.h"
> #include "modules/video_coding/utility/vp8_header_parser.h"
> #include "rtc_base/timeutils.h"
> #include "test/video_codec_settings.h"
> 
> namespace webrtc {
> 
>+using testing::Invoke;
>+using testing::NiceMock;
>+using testing::Return;
>+using testing::_;
>+
> namespace {
> constexpr uint32_t kInitialTimestampRtp = 123;
> constexpr int64_t kTestNtpTimeMs = 456;
>@@ -68,7 +77,7 @@ class TestVp8Impl : public VideoCodecUnitTest {
>               encoder_->Encode(input_frame, nullptr, &frame_types));
>     ASSERT_TRUE(WaitForEncodedFrame(encoded_frame, codec_specific_info));
>     VerifyQpParser(*encoded_frame);
>-    EXPECT_STREQ("libvpx", codec_specific_info->codec_name);
>+    EXPECT_EQ("libvpx", encoder_->GetEncoderInfo().implementation_name);
>     EXPECT_EQ(kVideoCodecVP8, codec_specific_info->codecType);
>     EXPECT_EQ(0, encoded_frame->SpatialIndex());
>   }
>@@ -325,7 +334,8 @@ TEST_F(TestVp8Impl, ScalingDisabledIfAutomaticResizeOff) {
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>             encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
> 
>-  VideoEncoder::ScalingSettings settings = encoder_->GetScalingSettings();
>+  VideoEncoder::ScalingSettings settings =
>+      encoder_->GetEncoderInfo().scaling_settings;
>   EXPECT_FALSE(settings.thresholds.has_value());
> }
> 
>@@ -335,7 +345,8 @@ TEST_F(TestVp8Impl, ScalingEnabledIfAutomaticResizeOn) {
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>             encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
> 
>-  VideoEncoder::ScalingSettings settings = encoder_->GetScalingSettings();
>+  VideoEncoder::ScalingSettings settings =
>+      encoder_->GetEncoderInfo().scaling_settings;
>   EXPECT_TRUE(settings.thresholds.has_value());
>   EXPECT_EQ(kDefaultMinPixelsPerFrame, settings.min_pixels_per_frame);
> }
>@@ -380,4 +391,41 @@ TEST_F(TestVp8Impl, DontDropKeyframes) {
>   EncodeAndExpectFrameWith(*NextInputFrame(), 0, true);
> }
> 
>+TEST_F(TestVp8Impl, KeepsTimestampOnReencode) {
>+  auto* const vpx = new NiceMock<MockLibvpxVp8Interface>();
>+  LibvpxVp8Encoder encoder((std::unique_ptr<LibvpxInterface>(vpx)));
>+
>+  // Settings needed to trigger ScreenshareLayers usage, which is required for
>+  // overshoot-drop-reencode logic.
>+  codec_settings_.targetBitrate = 200;
>+  codec_settings_.maxBitrate = 1000;
>+  codec_settings_.mode = VideoCodecMode::kScreensharing;
>+  codec_settings_.VP8()->numberOfTemporalLayers = 2;
>+
>+  EXPECT_CALL(*vpx, img_wrap(_, _, _, _, _, _))
>+      .WillOnce(Invoke([](vpx_image_t* img, vpx_img_fmt_t fmt, unsigned int d_w,
>+                          unsigned int d_h, unsigned int stride_align,
>+                          unsigned char* img_data) {
>+        img->fmt = fmt;
>+        img->d_w = d_w;
>+        img->d_h = d_h;
>+        img->img_data = img_data;
>+        return img;
>+      }));
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder.InitEncode(&codec_settings_, 1, 1000));
>+  MockEncodedImageCallback callback;
>+  encoder.RegisterEncodeCompleteCallback(&callback);
>+
>+  // Simulate overshoot drop, re-encode: encode function will be called twice
>+  // with the same parameters. codec_get_cx_data() will by default return no
>+  // image data and be interpreted as drop.
>+  EXPECT_CALL(*vpx, codec_encode(_, _, /* pts = */ 0, _, _, _))
>+      .Times(2)
>+      .WillRepeatedly(Return(vpx_codec_err_t::VPX_CODEC_OK));
>+
>+  auto delta_frame = std::vector<FrameType>{kVideoFrameDelta};
>+  encoder.Encode(*NextInputFrame(), nullptr, &delta_frame);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_temporal_layers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_temporal_layers.cc
>deleted file mode 100644
>index 49584414a911370d749686b430ca1e5098efcd6a..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_temporal_layers.cc
>+++ /dev/null
>@@ -1,42 +0,0 @@
>-/* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
>-
>-#include "absl/memory/memory.h"
>-#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
>-#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>-#include "system_wrappers/include/clock.h"
>-
>-namespace webrtc {
>-
>-bool TemporalLayers::FrameConfig::operator==(const FrameConfig& o) const {
>-  return drop_frame == o.drop_frame &&
>-         last_buffer_flags == o.last_buffer_flags &&
>-         golden_buffer_flags == o.golden_buffer_flags &&
>-         arf_buffer_flags == o.arf_buffer_flags && layer_sync == o.layer_sync &&
>-         freeze_entropy == o.freeze_entropy &&
>-         encoder_layer_id == o.encoder_layer_id &&
>-         packetizer_temporal_idx == o.packetizer_temporal_idx;
>-}
>-
>-std::unique_ptr<TemporalLayers> TemporalLayers::CreateTemporalLayers(
>-    TemporalLayersType type,
>-    int num_temporal_layers) {
>-  switch (type) {
>-    case TemporalLayersType::kFixedPattern:
>-      return absl::make_unique<DefaultTemporalLayers>(num_temporal_layers);
>-    case TemporalLayersType::kBitrateDynamic:
>-      // Conference mode temporal layering for screen content in base stream.
>-      return absl::make_unique<ScreenshareLayers>(num_temporal_layers,
>-                                                  Clock::GetRealTimeClock());
>-  }
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config.cc
>index 6807698246e8ff3355e9ac2ac0452fb91aea6ab0..3e922801483928cc13ea31c7dea13a54bdeef3b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config.cc
>@@ -22,9 +22,9 @@ namespace webrtc {
> namespace {
> const size_t kMinVp9SvcBitrateKbps = 30;
> 
>-const size_t kMaxNumLayersForScreenSharing = 2;
>-const float kMaxScreenSharingLayerFramerateFps[] = {5.0, 5.0};
>-const size_t kMaxScreenSharingLayerBitrateKbps[] = {200, 500};
>+const size_t kMaxNumLayersForScreenSharing = 3;
>+const float kMaxScreenSharingLayerFramerateFps[] = {5.0, 5.0, 30.0};
>+const size_t kMaxScreenSharingLayerBitrateKbps[] = {200, 500, 1250};
> }  // namespace
> 
> std::vector<SpatialLayer> ConfigureSvcScreenSharing(size_t input_width,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config_unittest.cc
>index 257c5df2f037ba0f502dcf5a4f7ef0c2e781dc5e..b997767465661e8b1844b45750fd176ec8d7e11b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config_unittest.cc
>@@ -49,12 +49,13 @@ TEST(SvcConfig, ScreenSharing) {
>   std::vector<SpatialLayer> spatial_layers =
>       GetSvcConfig(1920, 1080, 30, 3, 3, true);
> 
>-  EXPECT_EQ(spatial_layers.size(), 2UL);
>+  EXPECT_EQ(spatial_layers.size(), 3UL);
> 
>-  for (const SpatialLayer& layer : spatial_layers) {
>+  for (size_t i = 0; i < 3; ++i) {
>+    const SpatialLayer& layer = spatial_layers[i];
>     EXPECT_EQ(layer.width, 1920);
>     EXPECT_EQ(layer.height, 1080);
>-    EXPECT_EQ(layer.maxFramerate, 5);
>+    EXPECT_EQ(layer.maxFramerate, (i < 2) ? 5 : 30);
>     EXPECT_EQ(layer.numberOfTemporalLayers, 1);
>     EXPECT_LE(layer.minBitrate, layer.maxBitrate);
>     EXPECT_LE(layer.minBitrate, layer.targetBitrate);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator_unittest.cc
>index 048bf7d69471111e77461aa1cb8592817452cb69..eec2b9d4199ea9f77e3c99b6cb2d47849bd9f57c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator_unittest.cc
>@@ -149,7 +149,7 @@ TEST(SvcRateAllocatorTest, MinBitrateToGetQualityLayer) {
> 
>   const SpatialLayer* layers = codec.spatialLayers;
> 
>-  EXPECT_LE(codec.VP9()->numberOfSpatialLayers, 2U);
>+  EXPECT_LE(codec.VP9()->numberOfSpatialLayers, 3U);
> 
>   VideoBitrateAllocation allocation =
>       allocator.GetAllocation(layers[0].minBitrate * 1000, 30);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
>index 4733ad979a4995a1acb3b6fd00bb9ccdfa4a22ca..85fa278a0d376eaf5512ec0758b8cb7dd721eaf2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
>@@ -104,6 +104,33 @@ class TestVp9Impl : public VideoCodecUnitTest {
>       codec_settings_.spatialLayers[i] = layers[i];
>     }
>   }
>+
>+  HdrMetadata CreateTestHdrMetadata() const {
>+    // Random but reasonable HDR metadata.
>+    HdrMetadata hdr_metadata;
>+    hdr_metadata.mastering_metadata.luminance_max = 2000.0;
>+    hdr_metadata.mastering_metadata.luminance_min = 2.0001;
>+    hdr_metadata.mastering_metadata.primary_r.x = 0.30;
>+    hdr_metadata.mastering_metadata.primary_r.y = 0.40;
>+    hdr_metadata.mastering_metadata.primary_g.x = 0.32;
>+    hdr_metadata.mastering_metadata.primary_g.y = 0.46;
>+    hdr_metadata.mastering_metadata.primary_b.x = 0.34;
>+    hdr_metadata.mastering_metadata.primary_b.y = 0.49;
>+    hdr_metadata.mastering_metadata.white_point.x = 0.41;
>+    hdr_metadata.mastering_metadata.white_point.y = 0.48;
>+    hdr_metadata.max_content_light_level = 2345;
>+    hdr_metadata.max_frame_average_light_level = 1789;
>+    return hdr_metadata;
>+  }
>+
>+  ColorSpace CreateTestColorSpace() const {
>+    HdrMetadata hdr_metadata = CreateTestHdrMetadata();
>+    ColorSpace color_space(ColorSpace::PrimaryID::kBT709,
>+                           ColorSpace::TransferID::kGAMMA22,
>+                           ColorSpace::MatrixID::kSMPTE2085,
>+                           ColorSpace::RangeID::kFull, &hdr_metadata);
>+    return color_space;
>+  }
> };
> 
> // Disabled on ios as flake, see https://crbug.com/webrtc/7057
>@@ -128,7 +155,7 @@ TEST_F(TestVp9Impl, EncodeDecode) {
>   ASSERT_TRUE(decoded_frame);
>   EXPECT_GT(I420PSNR(input_frame, decoded_frame.get()), 36);
> 
>-  const ColorSpace color_space = decoded_frame->color_space().value();
>+  const ColorSpace color_space = *decoded_frame->color_space();
>   EXPECT_EQ(ColorSpace::PrimaryID::kInvalid, color_space.primaries());
>   EXPECT_EQ(ColorSpace::TransferID::kInvalid, color_space.transfer());
>   EXPECT_EQ(ColorSpace::MatrixID::kInvalid, color_space.matrix());
>@@ -157,6 +184,60 @@ TEST_F(TestVp9Impl, EncodedRotationEqualsInputRotation) {
>   EXPECT_EQ(kVideoRotation_90, encoded_frame.rotation_);
> }
> 
>+TEST_F(TestVp9Impl, EncodedColorSpaceEqualsInputColorSpace) {
>+  // Video frame without explicit color space information.
>+  VideoFrame* input_frame = NextInputFrame();
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->Encode(*input_frame, nullptr, nullptr));
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>+  EXPECT_FALSE(encoded_frame.ColorSpace());
>+
>+  // Video frame with explicit color space information.
>+  ColorSpace color_space = CreateTestColorSpace();
>+  VideoFrame input_frame_w_hdr =
>+      VideoFrame::Builder()
>+          .set_video_frame_buffer(input_frame->video_frame_buffer())
>+          .set_color_space(&color_space)
>+          .build();
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->Encode(input_frame_w_hdr, nullptr, nullptr));
>+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>+  ASSERT_TRUE(encoded_frame.ColorSpace());
>+  EXPECT_EQ(*encoded_frame.ColorSpace(), color_space);
>+}
>+
>+TEST_F(TestVp9Impl, DecodedHdrMetadataEqualsEncodedHdrMetadata) {
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>+
>+  // Encoded frame without explicit color space information.
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            decoder_->Decode(encoded_frame, false, nullptr, 0));
>+  std::unique_ptr<VideoFrame> decoded_frame;
>+  absl::optional<uint8_t> decoded_qp;
>+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>+  ASSERT_TRUE(decoded_frame);
>+  // Color space present from encoded bitstream.
>+  ASSERT_TRUE(decoded_frame->color_space());
>+  // No HDR metadata present.
>+  EXPECT_FALSE(decoded_frame->color_space()->hdr_metadata());
>+
>+  // Encoded frame with explicit color space information.
>+  ColorSpace color_space = CreateTestColorSpace();
>+  encoded_frame.SetColorSpace(&color_space);
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            decoder_->Decode(encoded_frame, false, nullptr, 0));
>+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>+  ASSERT_TRUE(decoded_frame);
>+  ASSERT_TRUE(decoded_frame->color_space());
>+  EXPECT_EQ(color_space, *decoded_frame->color_space());
>+}
>+
> TEST_F(TestVp9Impl, DecodedQpEqualsEncodedQp) {
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>             encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>@@ -393,7 +474,7 @@ TEST_F(TestVp9Impl, InterLayerPred) {
>   ConfigureSvc(num_spatial_layers);
>   codec_settings_.VP9()->frameDroppingOn = false;
> 
>-  BitrateAllocation bitrate_allocation;
>+  VideoBitrateAllocation bitrate_allocation;
>   for (size_t i = 0; i < num_spatial_layers; ++i) {
>     bitrate_allocation.SetBitrate(
>         i, 0, codec_settings_.spatialLayers[i].targetBitrate * 1000);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..22a08881208a129eb52e35a60d69f96b432f32bd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9.cc
>@@ -0,0 +1,75 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/vp9/include/vp9.h"
>+
>+#include "absl/memory/memory.h"
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "modules/video_coding/codecs/vp9/vp9_impl.h"
>+#include "rtc_base/checks.h"
>+#include "vpx/vp8cx.h"
>+#include "vpx/vp8dx.h"
>+#include "vpx/vpx_codec.h"
>+
>+namespace webrtc {
>+
>+std::vector<SdpVideoFormat> SupportedVP9Codecs() {
>+#ifdef RTC_ENABLE_VP9
>+  // Profile 2 might not be available on some platforms until
>+  // https://bugs.chromium.org/p/webm/issues/detail?id=1544 is solved.
>+  static bool vpx_supports_high_bit_depth =
>+      (vpx_codec_get_caps(vpx_codec_vp9_cx()) & VPX_CODEC_CAP_HIGHBITDEPTH) !=
>+          0 &&
>+      (vpx_codec_get_caps(vpx_codec_vp9_dx()) & VPX_CODEC_CAP_HIGHBITDEPTH) !=
>+          0;
>+
>+  std::vector<SdpVideoFormat> supported_formats{SdpVideoFormat(
>+      cricket::kVp9CodecName,
>+      {{kVP9FmtpProfileId, VP9ProfileToString(VP9Profile::kProfile0)}})};
>+  if (vpx_supports_high_bit_depth) {
>+    supported_formats.push_back(SdpVideoFormat(
>+        cricket::kVp9CodecName,
>+        {{kVP9FmtpProfileId, VP9ProfileToString(VP9Profile::kProfile2)}}));
>+  }
>+  return supported_formats;
>+#else
>+  return std::vector<SdpVideoFormat>();
>+#endif
>+}
>+
>+std::unique_ptr<VP9Encoder> VP9Encoder::Create() {
>+#ifdef RTC_ENABLE_VP9
>+  return absl::make_unique<VP9EncoderImpl>(cricket::VideoCodec());
>+#else
>+  RTC_NOTREACHED();
>+  return nullptr;
>+#endif
>+}
>+
>+std::unique_ptr<VP9Encoder> VP9Encoder::Create(
>+    const cricket::VideoCodec& codec) {
>+#ifdef RTC_ENABLE_VP9
>+  return absl::make_unique<VP9EncoderImpl>(codec);
>+#else
>+  RTC_NOTREACHED();
>+  return nullptr;
>+#endif
>+}
>+
>+std::unique_ptr<VP9Decoder> VP9Decoder::Create() {
>+#ifdef RTC_ENABLE_VP9
>+  return absl::make_unique<VP9DecoderImpl>();
>+#else
>+  RTC_NOTREACHED();
>+  return nullptr;
>+#endif
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
>index 69101cb506f350431990081f3cbf66f50085ebf7..ecbda4cb3bd0f24d36969746d8d58968154a0b95 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
>@@ -9,6 +9,8 @@
>  *
>  */
> 
>+#ifdef RTC_ENABLE_VP9
>+
> #include "modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h"
> 
> #include "vpx/vpx_codec.h"
>@@ -138,3 +140,5 @@ int32_t Vp9FrameBufferPool::VpxReleaseFrameBuffer(void* user_priv,
> }
> 
> }  // namespace webrtc
>+
>+#endif  // RTC_ENABLE_VP9
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h
>index 9b035eb94ade737ded3b35bb67a27dcf98893f6c..02ede2432955fdea572b1c59ecca8f513f9c3a4c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h
>@@ -12,6 +12,8 @@
> #ifndef MODULES_VIDEO_CODING_CODECS_VP9_VP9_FRAME_BUFFER_POOL_H_
> #define MODULES_VIDEO_CODING_CODECS_VP9_VP9_FRAME_BUFFER_POOL_H_
> 
>+#ifdef RTC_ENABLE_VP9
>+
> #include <vector>
> 
> #include "rtc_base/buffer.h"
>@@ -120,4 +122,6 @@ class Vp9FrameBufferPool {
> 
> }  // namespace webrtc
> 
>+#endif  // RTC_ENABLE_VP9
>+
> #endif  // MODULES_VIDEO_CODING_CODECS_VP9_VP9_FRAME_BUFFER_POOL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
>index 6b982a8a132389d2aea51dfeaac92bedf9c2af7f..61542c508211496297b36cecde4794b46284e605 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
>@@ -9,6 +9,8 @@
>  *
>  */
> 
>+#ifdef RTC_ENABLE_VP9
>+
> #include "modules/video_coding/codecs/vp9/vp9_impl.h"
> 
> #include <algorithm>
>@@ -37,11 +39,16 @@
> namespace webrtc {
> 
> namespace {
>+const char kVp9TrustedRateControllerFieldTrial[] =
>+    "WebRTC-LibvpxVp9TrustedRateController";
>+
> // Maps from gof_idx to encoder internal reference frame buffer index. These
> // maps work for 1,2 and 3 temporal layers with GOF length of 1,2 and 4 frames.
> uint8_t kRefBufIdx[4] = {0, 0, 0, 1};
> uint8_t kUpdBufIdx[4] = {0, 0, 1, 0};
> 
>+int kMaxNumTiles4kVideo = 8;
>+
> // Only positive speeds, range for real-time coding currently is: 5 - 8.
> // Lower means slower/better quality, higher means fastest/lower quality.
> int GetCpuSpeed(int width, int height) {
>@@ -119,23 +126,6 @@ ColorSpace ExtractVP9ColorSpace(vpx_color_space_t space_t,
> }
> }  // namespace
> 
>-std::vector<SdpVideoFormat> SupportedVP9Codecs() {
>-  // TODO(emircan): Add Profile 2 support after fixing browser_tests.
>-  std::vector<SdpVideoFormat> supported_formats{SdpVideoFormat(
>-      cricket::kVp9CodecName,
>-      {{kVP9FmtpProfileId, VP9ProfileToString(VP9Profile::kProfile0)}})};
>-  return supported_formats;
>-}
>-
>-std::unique_ptr<VP9Encoder> VP9Encoder::Create() {
>-  return absl::make_unique<VP9EncoderImpl>(cricket::VideoCodec());
>-}
>-
>-std::unique_ptr<VP9Encoder> VP9Encoder::Create(
>-    const cricket::VideoCodec& codec) {
>-  return absl::make_unique<VP9EncoderImpl>(codec);
>-}
>-
> void VP9EncoderImpl::EncoderOutputCodedPacketCallback(vpx_codec_cx_pkt* pkt,
>                                                       void* user_data) {
>   VP9EncoderImpl* enc = static_cast<VP9EncoderImpl*>(user_data);
>@@ -160,12 +150,16 @@ VP9EncoderImpl::VP9EncoderImpl(const cricket::VideoCodec& codec)
>       num_temporal_layers_(0),
>       num_spatial_layers_(0),
>       num_active_spatial_layers_(0),
>-      layer_deactivation_requires_key_frame_(webrtc::field_trial::IsEnabled(
>-          "WebRTC-Vp9IssueKeyFrameOnLayerDeactivation")),
>+      layer_deactivation_requires_key_frame_(
>+          field_trial::IsEnabled("WebRTC-Vp9IssueKeyFrameOnLayerDeactivation")),
>       is_svc_(false),
>       inter_layer_pred_(InterLayerPredMode::kOn),
>       external_ref_control_(
>-          webrtc::field_trial::IsEnabled("WebRTC-Vp9ExternalRefCtrl")),
>+          field_trial::IsEnabled("WebRTC-Vp9ExternalRefCtrl")),
>+      trusted_rate_controller_(
>+          field_trial::IsEnabled(kVp9TrustedRateControllerFieldTrial)),
>+      full_superframe_drop_(true),
>+      first_frame_in_picture_(true),
>       is_flexible_mode_(false) {
>   memset(&codec_, 0, sizeof(codec_));
>   memset(&svc_params_, 0, sizeof(vpx_svc_extra_cfg_t));
>@@ -627,7 +621,8 @@ int VP9EncoderImpl::InitAndSetControlSettings(const VideoCodec* inst) {
>     // quality flickering and is not compatible with RTP non-flexible mode.
>     vpx_svc_frame_drop_t svc_drop_frame;
>     memset(&svc_drop_frame, 0, sizeof(svc_drop_frame));
>-    svc_drop_frame.framedrop_mode = FULL_SUPERFRAME_DROP;
>+    svc_drop_frame.framedrop_mode =
>+        full_superframe_drop_ ? FULL_SUPERFRAME_DROP : CONSTRAINED_LAYER_DROP;
>     svc_drop_frame.max_consec_drop = std::numeric_limits<int>::max();
>     for (size_t i = 0; i < num_spatial_layers_; ++i) {
>       svc_drop_frame.framedrop_thresh[i] = config_->rc_dropframe_thresh;
>@@ -701,30 +696,33 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
>     }
>   }
> 
>-  if (VideoCodecMode::kScreensharing == codec_.mode && !force_key_frame_) {
>-    // Skip encoding spatial layer frames if their target frame rate is lower
>-    // than actual input frame rate.
>+  size_t first_active_spatial_layer_id = 0;
>+  if (VideoCodecMode::kScreensharing == codec_.mode) {
>     vpx_svc_layer_id_t layer_id = {0};
>-    const size_t gof_idx = (pics_since_key_ + 1) % gof_.num_frames_in_gof;
>-    layer_id.temporal_layer_id = gof_.temporal_idx[gof_idx];
>+    if (!force_key_frame_) {
>+      // Skip encoding spatial layer frames if their target frame rate is lower
>+      // than actual input frame rate.
>+      const size_t gof_idx = (pics_since_key_ + 1) % gof_.num_frames_in_gof;
>+      layer_id.temporal_layer_id = gof_.temporal_idx[gof_idx];
> 
>-    const uint32_t frame_timestamp_ms =
>-        1000 * input_image.timestamp() / kVideoPayloadTypeFrequency;
>+      const uint32_t frame_timestamp_ms =
>+          1000 * input_image.timestamp() / kVideoPayloadTypeFrequency;
> 
>-    for (uint8_t sl_idx = 0; sl_idx < num_active_spatial_layers_; ++sl_idx) {
>-      if (framerate_controller_[sl_idx].DropFrame(frame_timestamp_ms)) {
>-        ++layer_id.spatial_layer_id;
>-      } else {
>-        break;
>+      for (uint8_t sl_idx = 0; sl_idx < num_active_spatial_layers_; ++sl_idx) {
>+        if (framerate_controller_[sl_idx].DropFrame(frame_timestamp_ms)) {
>+          ++layer_id.spatial_layer_id;
>+        } else {
>+          break;
>+        }
>       }
>-    }
> 
>-    RTC_DCHECK_LE(layer_id.spatial_layer_id, num_active_spatial_layers_);
>-    if (layer_id.spatial_layer_id >= num_active_spatial_layers_) {
>-      // Drop entire picture.
>-      return WEBRTC_VIDEO_CODEC_OK;
>+      RTC_DCHECK_LE(layer_id.spatial_layer_id, num_active_spatial_layers_);
>+      if (layer_id.spatial_layer_id >= num_active_spatial_layers_) {
>+        // Drop entire picture.
>+        return WEBRTC_VIDEO_CODEC_OK;
>+      }
>     }
>-
>+    first_active_spatial_layer_id = layer_id.spatial_layer_id;
>     vpx_codec_control(encoder_, VP9E_SET_SVC_LAYER_ID, &layer_id);
>   }
> 
>@@ -785,10 +783,21 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
>   }
> 
>   if (external_ref_control_) {
>-    vpx_svc_ref_frame_config_t ref_config = SetReferences(force_key_frame_);
>+    vpx_svc_ref_frame_config_t ref_config =
>+        SetReferences(force_key_frame_, first_active_spatial_layer_id);
>+
>+    if (VideoCodecMode::kScreensharing == codec_.mode) {
>+      for (uint8_t sl_idx = 0; sl_idx < num_active_spatial_layers_; ++sl_idx) {
>+        ref_config.duration[sl_idx] = static_cast<int64_t>(
>+            90000 / framerate_controller_[sl_idx].GetTargetRate());
>+      }
>+    }
>+
>     vpx_codec_control(encoder_, VP9E_SET_SVC_REF_FRAME_CONFIG, &ref_config);
>   }
> 
>+  first_frame_in_picture_ = true;
>+
>   // TODO(ssilkin): Frame duration should be specified per spatial layer
>   // since their frame rate can be different. For now calculate frame duration
>   // based on target frame rate of the highest spatial layer, which frame rate
>@@ -814,8 +823,10 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
>   }
>   timestamp_ += duration;
> 
>-  const bool end_of_picture = true;
>-  DeliverBufferedFrame(end_of_picture);
>+  if (!full_superframe_drop_) {
>+    const bool end_of_picture = true;
>+    DeliverBufferedFrame(end_of_picture);
>+  }
> 
>   return WEBRTC_VIDEO_CODEC_OK;
> }
>@@ -823,14 +834,12 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
> void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>                                            absl::optional<int>* spatial_idx,
>                                            const vpx_codec_cx_pkt& pkt,
>-                                           uint32_t timestamp,
>-                                           bool first_frame_in_picture) {
>+                                           uint32_t timestamp) {
>   RTC_CHECK(codec_specific != nullptr);
>   codec_specific->codecType = kVideoCodecVP9;
>-  codec_specific->codec_name = ImplementationName();
>   CodecSpecificInfoVP9* vp9_info = &(codec_specific->codecSpecific.VP9);
> 
>-  vp9_info->first_frame_in_picture = first_frame_in_picture;
>+  vp9_info->first_frame_in_picture = first_frame_in_picture_;
>   vp9_info->flexible_mode = is_flexible_mode_;
>   vp9_info->ss_data_available =
>       (pkt.data.frame.flags & VPX_FRAME_IS_KEY) ? true : false;
>@@ -861,7 +870,7 @@ void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
> 
>   if (pkt.data.frame.flags & VPX_FRAME_IS_KEY) {
>     pics_since_key_ = 0;
>-  } else if (first_frame_in_picture) {
>+  } else if (first_frame_in_picture_) {
>     ++pics_since_key_;
>   }
> 
>@@ -877,7 +886,7 @@ void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>   // if low layer frame is lost) then receiver won't be able to decode next high
>   // layer frame which uses ILP.
>   vp9_info->inter_layer_predicted =
>-      first_frame_in_picture ? false : is_inter_layer_pred_allowed;
>+      first_frame_in_picture_ ? false : is_inter_layer_pred_allowed;
> 
>   // Mark all low spatial layer frames as references (not just frames of
>   // active low spatial layers) if inter-layer prediction is enabled since
>@@ -921,6 +930,8 @@ void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>       vp9_info->gof.CopyGofInfoVP9(gof_);
>     }
>   }
>+
>+  first_frame_in_picture_ = false;
> }
> 
> void VP9EncoderImpl::FillReferenceIndices(const vpx_codec_cx_pkt& pkt,
>@@ -978,6 +989,8 @@ void VP9EncoderImpl::FillReferenceIndices(const vpx_codec_cx_pkt& pkt,
> 
>   size_t max_ref_temporal_layer_id = 0;
> 
>+  std::vector<size_t> ref_pid_list;
>+
>   vp9_info->num_ref_pics = 0;
>   for (const RefFrameBuffer& ref_buf : ref_buf_list) {
>     RTC_DCHECK_LE(ref_buf.pic_num, pic_num);
>@@ -990,6 +1003,16 @@ void VP9EncoderImpl::FillReferenceIndices(const vpx_codec_cx_pkt& pkt,
>       }
>       RTC_DCHECK_LE(ref_buf.temporal_layer_id, layer_id.temporal_layer_id);
> 
>+      // Encoder may reference several spatial layers on the same previous
>+      // frame in case if some spatial layers are skipped on the current frame.
>+      // We shouldn't put duplicate references as it may break some old
>+      // clients and isn't RTP compatible.
>+      if (std::find(ref_pid_list.begin(), ref_pid_list.end(),
>+                    ref_buf.pic_num) != ref_pid_list.end()) {
>+        continue;
>+      }
>+      ref_pid_list.push_back(ref_buf.pic_num);
>+
>       const size_t p_diff = pic_num - ref_buf.pic_num;
>       RTC_DCHECK_LE(p_diff, 127UL);
> 
>@@ -1054,7 +1077,9 @@ void VP9EncoderImpl::UpdateReferenceBuffers(const vpx_codec_cx_pkt& pkt,
>   }
> }
> 
>-vpx_svc_ref_frame_config_t VP9EncoderImpl::SetReferences(bool is_key_pic) {
>+vpx_svc_ref_frame_config_t VP9EncoderImpl::SetReferences(
>+    bool is_key_pic,
>+    size_t first_active_spatial_layer_id) {
>   // kRefBufIdx, kUpdBufIdx need to be updated to support longer GOFs.
>   RTC_DCHECK_LE(gof_.num_frames_in_gof, 4);
> 
>@@ -1106,13 +1131,14 @@ vpx_svc_ref_frame_config_t VP9EncoderImpl::SetReferences(bool is_key_pic) {
>       }
>     }
> 
>-    if (is_inter_layer_pred_allowed && sl_idx > 0) {
>+    if (is_inter_layer_pred_allowed && sl_idx > first_active_spatial_layer_id) {
>       // Set up spatial reference.
>       RTC_DCHECK(last_updated_buf_idx);
>       ref_config.gld_fb_idx[sl_idx] = *last_updated_buf_idx;
>       ref_config.reference_golden[sl_idx] = 1;
>     } else {
>-      RTC_DCHECK(ref_config.reference_last[sl_idx] != 0 || sl_idx == 0 ||
>+      RTC_DCHECK(ref_config.reference_last[sl_idx] != 0 ||
>+                 sl_idx == first_active_spatial_layer_id ||
>                  inter_layer_pred_ == InterLayerPredMode::kOff);
>     }
> 
>@@ -1147,12 +1173,11 @@ int VP9EncoderImpl::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
>   vpx_svc_layer_id_t layer_id = {0};
>   vpx_codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
> 
>-  const bool first_frame_in_picture = encoded_image_._length == 0;
>-  // Ensure we don't buffer layers of previous picture (superframe).
>-  RTC_DCHECK(first_frame_in_picture || layer_id.spatial_layer_id > 0);
>-
>-  const bool end_of_picture = false;
>-  DeliverBufferedFrame(end_of_picture);
>+  if (!full_superframe_drop_) {
>+    // Deliver buffered low spatial layer frame.
>+    const bool end_of_picture = false;
>+    DeliverBufferedFrame(end_of_picture);
>+  }
> 
>   if (pkt->data.frame.sz > encoded_image_._size) {
>     delete[] encoded_image_._buffer;
>@@ -1178,7 +1203,7 @@ int VP9EncoderImpl::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
>   memset(&codec_specific_, 0, sizeof(codec_specific_));
>   absl::optional<int> spatial_index;
>   PopulateCodecSpecific(&codec_specific_, &spatial_index, *pkt,
>-                        input_image_->timestamp(), first_frame_in_picture);
>+                        input_image_->timestamp());
>   encoded_image_.SetSpatialIndex(spatial_index);
> 
>   if (is_flexible_mode_) {
>@@ -1200,6 +1225,13 @@ int VP9EncoderImpl::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
>   int qp = -1;
>   vpx_codec_control(encoder_, VP8E_GET_LAST_QUANTIZER, &qp);
>   encoded_image_.qp_ = qp;
>+  encoded_image_.SetColorSpace(input_image_->color_space());
>+
>+  if (full_superframe_drop_) {
>+    const bool end_of_picture = encoded_image_.SpatialIndex().value_or(0) + 1 ==
>+                                num_active_spatial_layers_;
>+    DeliverBufferedFrame(end_of_picture);
>+  }
> 
>   return WEBRTC_VIDEO_CODEC_OK;
> }
>@@ -1230,22 +1262,19 @@ void VP9EncoderImpl::DeliverBufferedFrame(bool end_of_picture) {
>   }
> }
> 
>-int VP9EncoderImpl::SetChannelParameters(uint32_t packet_loss, int64_t rtt) {
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
> int VP9EncoderImpl::RegisterEncodeCompleteCallback(
>     EncodedImageCallback* callback) {
>   encoded_complete_callback_ = callback;
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-const char* VP9EncoderImpl::ImplementationName() const {
>-  return "libvpx";
>-}
>-
>-std::unique_ptr<VP9Decoder> VP9Decoder::Create() {
>-  return absl::make_unique<VP9DecoderImpl>();
>+VideoEncoder::EncoderInfo VP9EncoderImpl::GetEncoderInfo() const {
>+  EncoderInfo info;
>+  info.supports_native_handle = false;
>+  info.implementation_name = "libvpx";
>+  info.scaling_settings = VideoEncoder::ScalingSettings::kOff;
>+  info.has_trusted_rate_controller = trusted_rate_controller_;
>+  return info;
> }
> 
> VP9DecoderImpl::VP9DecoderImpl()
>@@ -1272,13 +1301,18 @@ int VP9DecoderImpl::InitDecode(const VideoCodec* inst, int number_of_cores) {
>   if (ret_val < 0) {
>     return ret_val;
>   }
>+
>   if (decoder_ == nullptr) {
>     decoder_ = new vpx_codec_ctx_t;
>   }
>   vpx_codec_dec_cfg_t cfg;
>-  // Setting number of threads to a constant value (1)
>-  cfg.threads = 1;
>-  cfg.h = cfg.w = 0;  // set after decode
>+  memset(&cfg, 0, sizeof(cfg));
>+
>+  // We want to use multithreading when decoding high resolution videos. But,
>+  // since we don't know resolution of input stream at this stage, we always
>+  // enable it.
>+  cfg.threads = std::min(number_of_cores, kMaxNumTiles4kVideo);
>+
>   vpx_codec_flags_t flags = 0;
>   if (vpx_codec_dec_init(decoder_, vpx_codec_vp9_dx(), &cfg, flags)) {
>     return WEBRTC_VIDEO_CODEC_MEMORY;
>@@ -1336,8 +1370,8 @@ int VP9DecoderImpl::Decode(const EncodedImage& input_image,
>   vpx_codec_err_t vpx_ret =
>       vpx_codec_control(decoder_, VPXD_GET_LAST_QUANTIZER, &qp);
>   RTC_DCHECK_EQ(vpx_ret, VPX_CODEC_OK);
>-  int ret =
>-      ReturnFrame(img, input_image.Timestamp(), input_image.ntp_time_ms_, qp);
>+  int ret = ReturnFrame(img, input_image.Timestamp(), input_image.ntp_time_ms_,
>+                        qp, input_image.ColorSpace());
>   if (ret != 0) {
>     return ret;
>   }
>@@ -1347,7 +1381,8 @@ int VP9DecoderImpl::Decode(const EncodedImage& input_image,
> int VP9DecoderImpl::ReturnFrame(const vpx_image_t* img,
>                                 uint32_t timestamp,
>                                 int64_t ntp_time_ms,
>-                                int qp) {
>+                                int qp,
>+                                const ColorSpace* explicit_color_space) {
>   if (img == nullptr) {
>     // Decoder OK and nullptr image => No show frame.
>     return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>@@ -1389,15 +1424,21 @@ int VP9DecoderImpl::ReturnFrame(const vpx_image_t* img,
>       return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>   }
> 
>-  VideoFrame decoded_image = VideoFrame::Builder()
>-                                 .set_video_frame_buffer(img_wrapped_buffer)
>-                                 .set_timestamp_ms(0)
>-                                 .set_timestamp_rtp(timestamp)
>-                                 .set_ntp_time_ms(ntp_time_ms)
>-                                 .set_rotation(webrtc::kVideoRotation_0)
>-                                 .set_color_space(ExtractVP9ColorSpace(
>-                                     img->cs, img->range, img->bit_depth))
>-                                 .build();
>+  auto builder = VideoFrame::Builder()
>+                     .set_video_frame_buffer(img_wrapped_buffer)
>+                     .set_timestamp_ms(0)
>+                     .set_timestamp_rtp(timestamp)
>+                     .set_ntp_time_ms(ntp_time_ms)
>+                     .set_rotation(webrtc::kVideoRotation_0);
>+  if (explicit_color_space) {
>+    builder.set_color_space(explicit_color_space);
>+  } else {
>+    builder.set_color_space(
>+        ExtractVP9ColorSpace(img->cs, img->range, img->bit_depth));
>+  }
>+
>+  VideoFrame decoded_image = builder.build();
>+
>   decode_complete_callback_->Decoded(decoded_image, absl::nullopt, qp);
>   return WEBRTC_VIDEO_CODEC_OK;
> }
>@@ -1435,3 +1476,5 @@ const char* VP9DecoderImpl::ImplementationName() const {
> }
> 
> }  // namespace webrtc
>+
>+#endif  // RTC_ENABLE_VP9
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
>index d8ce1ff2ab5efb23773cf7f202808bf314fb9e72..3bfab9ad5fd589badd523b419f818d1f2e8207b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
>@@ -12,6 +12,8 @@
> #ifndef MODULES_VIDEO_CODING_CODECS_VP9_VP9_IMPL_H_
> #define MODULES_VIDEO_CODING_CODECS_VP9_VP9_IMPL_H_
> 
>+#ifdef RTC_ENABLE_VP9
>+
> #include <map>
> #include <memory>
> #include <vector>
>@@ -46,12 +48,10 @@ class VP9EncoderImpl : public VP9Encoder {
> 
>   int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
> 
>-  int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>-
>   int SetRateAllocation(const VideoBitrateAllocation& bitrate_allocation,
>                         uint32_t frame_rate) override;
> 
>-  const char* ImplementationName() const override;
>+  EncoderInfo GetEncoderInfo() const override;
> 
>  private:
>   // Determine number of encoder threads to use.
>@@ -63,15 +63,16 @@ class VP9EncoderImpl : public VP9Encoder {
>   void PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>                              absl::optional<int>* spatial_idx,
>                              const vpx_codec_cx_pkt& pkt,
>-                             uint32_t timestamp,
>-                             bool first_frame_in_picture);
>+                             uint32_t timestamp);
>   void FillReferenceIndices(const vpx_codec_cx_pkt& pkt,
>                             const size_t pic_num,
>                             const bool inter_layer_predicted,
>                             CodecSpecificInfoVP9* vp9_info);
>   void UpdateReferenceBuffers(const vpx_codec_cx_pkt& pkt,
>                               const size_t pic_num);
>-  vpx_svc_ref_frame_config_t SetReferences(bool is_key_pic);
>+  vpx_svc_ref_frame_config_t SetReferences(
>+      bool is_key_pic,
>+      size_t first_active_spatial_layer_id);
> 
>   bool ExplicitlyConfiguredSpatialLayers() const;
>   bool SetSvcRates(const VideoBitrateAllocation& bitrate_allocation);
>@@ -119,6 +120,9 @@ class VP9EncoderImpl : public VP9Encoder {
>   bool is_svc_;
>   InterLayerPredMode inter_layer_pred_;
>   bool external_ref_control_;
>+  const bool trusted_rate_controller_;
>+  const bool full_superframe_drop_;
>+  bool first_frame_in_picture_;
> 
>   std::vector<FramerateController> framerate_controller_;
> 
>@@ -168,7 +172,8 @@ class VP9DecoderImpl : public VP9Decoder {
>   int ReturnFrame(const vpx_image_t* img,
>                   uint32_t timestamp,
>                   int64_t ntp_time_ms,
>-                  int qp);
>+                  int qp,
>+                  const ColorSpace* explicit_color_space);
> 
>   // Memory pool used to share buffers between libvpx and webrtc.
>   Vp9FrameBufferPool frame_buffer_pool_;
>@@ -179,4 +184,6 @@ class VP9DecoderImpl : public VP9Decoder {
> };
> }  // namespace webrtc
> 
>+#endif  // RTC_ENABLE_VP9
>+
> #endif  // MODULES_VIDEO_CODING_CODECS_VP9_VP9_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.h
>index a08eb0769c77f9f4aeb33fc9e1f15c2daec148a5..c7efd400727b864afdee490d629b1ea1747e99c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.h
>@@ -13,8 +13,8 @@
> 
> #include <vector>
> 
>+#include "api/video/encoded_image.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/video_frame.h"
> #include "modules/include/module_common_types.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/include/video_coding_defines.h"
>@@ -60,11 +60,21 @@ class VCMEncodedFrame : protected EncodedImage {
>    *   Get pointer to frame buffer
>    */
>   const uint8_t* Buffer() const { return _buffer; }
>+  /**
>+   *   Get pointer to frame buffer that can be mutated.
>+   */
>+  uint8_t* MutableBuffer() { return _buffer; }
>   /**
>    *   Get frame length
>    */
>   size_t Length() const { return _length; }
>-
>+  /**
>+   *   Set frame length
>+   */
>+  void SetLength(size_t length) {
>+    RTC_DCHECK(length <= _size);
>+    _length = length;
>+  }
>   /**
>    *   Frame RTP timestamp (90kHz)
>    */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.cc
>index 5fe38e6297dd1379b0e3483bcfda82e45ccbaf03..841ed742dac509e5772a02674044c3fbef474a8f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.cc
>@@ -8,10 +8,17 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/video_coding/fec_controller_default.h"
>+#include <algorithm>
>+
>+#include "modules/video_coding/fec_controller_default.h"  // NOLINT
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> using rtc::CritScope;
>+
>+const float kProtectionOverheadRateThreshold = 0.5;
>+
> FecControllerDefault::FecControllerDefault(
>     Clock* clock,
>     VCMProtectionCallback* protection_callback)
>@@ -19,13 +26,15 @@ FecControllerDefault::FecControllerDefault(
>       protection_callback_(protection_callback),
>       loss_prot_logic_(new media_optimization::VCMLossProtectionLogic(
>           clock_->TimeInMilliseconds())),
>-      max_payload_size_(1460) {}
>+      max_payload_size_(1460),
>+      overhead_threshold_(GetProtectionOverheadRateThreshold()) {}
> 
> FecControllerDefault::FecControllerDefault(Clock* clock)
>     : clock_(clock),
>       loss_prot_logic_(new media_optimization::VCMLossProtectionLogic(
>           clock_->TimeInMilliseconds())),
>-      max_payload_size_(1460) {}
>+      max_payload_size_(1460),
>+      overhead_threshold_(GetProtectionOverheadRateThreshold()) {}
> 
> FecControllerDefault::~FecControllerDefault(void) {
>   loss_prot_logic_->Release();
>@@ -46,6 +55,25 @@ void FecControllerDefault::SetEncodingData(size_t width,
>   max_payload_size_ = max_payload_size;
> }
> 
>+float FecControllerDefault::GetProtectionOverheadRateThreshold() {
>+  float overhead_threshold =
>+      strtof(webrtc::field_trial::FindFullName(
>+                 "WebRTC-ProtectionOverheadRateThreshold")
>+                 .c_str(),
>+             nullptr);
>+  if (overhead_threshold > 0 && overhead_threshold <= 1) {
>+    RTC_LOG(LS_INFO) << "ProtectionOverheadRateThreshold is set to "
>+                     << overhead_threshold;
>+    return overhead_threshold;
>+  } else if (overhead_threshold < 0 || overhead_threshold > 1) {
>+    RTC_LOG(WARNING) << "ProtectionOverheadRateThreshold field trial is set to "
>+                        "an invalid value, expecting a value between (0, 1].";
>+  }
>+  // WebRTC-ProtectionOverheadRateThreshold field trial string is not found, use
>+  // the default value.
>+  return kProtectionOverheadRateThreshold;
>+}
>+
> uint32_t FecControllerDefault::UpdateFecRates(
>     uint32_t estimated_bitrate_bps,
>     int actual_framerate_fps,
>@@ -125,9 +153,9 @@ uint32_t FecControllerDefault::UpdateFecRates(
>         static_cast<float>(sent_nack_rate_bps + sent_fec_rate_bps) /
>         sent_total_rate_bps;
>   }
>-  // Cap the overhead estimate to 50%.
>-  if (protection_overhead_rate > 0.5)
>-    protection_overhead_rate = 0.5;
>+  // Cap the overhead estimate to a threshold, default is 50%.
>+  protection_overhead_rate =
>+      std::min(protection_overhead_rate, overhead_threshold_);
>   // Source coding rate: total rate - protection overhead.
>   return estimated_bitrate_bps * (1.0 - protection_overhead_rate);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.h
>index a95eced245a2208a02420d12bacd56891f617900..9d79a61d28ebbaeed7f0f31a51c7f7680ae44ae5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.h
>@@ -42,6 +42,7 @@ class FecControllerDefault : public FecController {
>   void UpdateWithEncodedData(const size_t encoded_image_length,
>                              const FrameType encoded_image_frametype) override;
>   bool UseLossVectorMask() override;
>+  float GetProtectionOverheadRateThreshold();
> 
>  private:
>   enum { kBitrateAverageWinMs = 1000 };
>@@ -52,6 +53,7 @@ class FecControllerDefault : public FecController {
>       RTC_GUARDED_BY(crit_sect_);
>   size_t max_payload_size_ RTC_GUARDED_BY(crit_sect_);
>   RTC_DISALLOW_COPY_AND_ASSIGN(FecControllerDefault);
>+  const float overhead_threshold_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.cc
>index 377611e5e271437506ad9847ea6840c9e1d6d606..04fba84caf55b447c057da93d06e654016ec1ec3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.cc
>@@ -46,11 +46,9 @@ FrameBuffer::FrameBuffer(Clock* clock,
>                          VCMTiming* timing,
>                          VCMReceiveStatisticsCallback* stats_callback)
>     : clock_(clock),
>-      new_continuous_frame_event_(false, false),
>       jitter_estimator_(jitter_estimator),
>       timing_(timing),
>       inter_frame_delay_(clock_->TimeInMilliseconds()),
>-      last_decoded_frame_timestamp_(0),
>       last_decoded_frame_it_(frames_.end()),
>       last_continuous_frame_it_(frames_.end()),
>       num_frames_history_(0),
>@@ -115,10 +113,19 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
>         if (keyframe_required && !frame->is_keyframe())
>           continue;
> 
>+        // TODO(https://bugs.webrtc.org/9974): consider removing this check
>+        // as it may make a stream undecodable after a very long delay between
>+        // frames.
>+        if (last_decoded_frame_timestamp_ &&
>+            AheadOf(*last_decoded_frame_timestamp_, frame->Timestamp())) {
>+          continue;
>+        }
>+
>         next_frame_it_ = frame_it;
>-        if (frame->RenderTime() == -1)
>+        if (frame->RenderTime() == -1) {
>           frame->SetRenderTime(
>               timing_->RenderTimeMs(frame->Timestamp(), now_ms));
>+        }
>         wait_ms = timing_->MaxWaitingTime(frame->RenderTime(), now_ms);
> 
>         // This will cause the frame buffer to prefer high framerate rather
>@@ -175,33 +182,6 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
>       UpdateTimingFrameInfo();
>       PropagateDecodability(next_frame_it_->second);
> 
>-      // Sanity check for RTP timestamp monotonicity.
>-      if (last_decoded_frame_it_ != frames_.end()) {
>-        const VideoLayerFrameId& last_decoded_frame_key =
>-            last_decoded_frame_it_->first;
>-        const VideoLayerFrameId& frame_key = next_frame_it_->first;
>-
>-        const bool frame_is_higher_spatial_layer_of_last_decoded_frame =
>-            last_decoded_frame_timestamp_ == frame->Timestamp() &&
>-            last_decoded_frame_key.picture_id == frame_key.picture_id &&
>-            last_decoded_frame_key.spatial_layer < frame_key.spatial_layer;
>-
>-        if (AheadOrAt(last_decoded_frame_timestamp_, frame->Timestamp()) &&
>-            !frame_is_higher_spatial_layer_of_last_decoded_frame) {
>-          // TODO(brandtr): Consider clearing the entire buffer when we hit
>-          // these conditions.
>-          RTC_LOG(LS_WARNING)
>-              << "Frame with (timestamp:picture_id:spatial_id) ("
>-              << frame->Timestamp() << ":" << frame->id.picture_id << ":"
>-              << static_cast<int>(frame->id.spatial_layer) << ")"
>-              << " sent to decoder after frame with"
>-              << " (timestamp:picture_id:spatial_id) ("
>-              << last_decoded_frame_timestamp_ << ":"
>-              << last_decoded_frame_key.picture_id << ":"
>-              << static_cast<int>(last_decoded_frame_key.spatial_layer) << ").";
>-        }
>-      }
>-
>       AdvanceLastDecodedFrame(next_frame_it_);
>       last_decoded_frame_timestamp_ = frame->Timestamp();
>       *frame_out = std::move(frame);
>@@ -267,6 +247,11 @@ void FrameBuffer::Stop() {
>   new_continuous_frame_event_.Set();
> }
> 
>+void FrameBuffer::Clear() {
>+  rtc::CritScope lock(&crit_);
>+  ClearFramesAndHistory();
>+}
>+
> void FrameBuffer::UpdateRtt(int64_t rtt_ms) {
>   rtc::CritScope lock(&crit_);
>   jitter_estimator_->UpdateRtt(rtt_ms);
>@@ -348,7 +333,7 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
> 
>   if (last_decoded_frame_it_ != frames_.end() &&
>       id <= last_decoded_frame_it_->first) {
>-    if (AheadOf(frame->Timestamp(), last_decoded_frame_timestamp_) &&
>+    if (AheadOf(frame->Timestamp(), *last_decoded_frame_timestamp_) &&
>         frame->is_keyframe()) {
>       // If this frame has a newer timestamp but an earlier picture id then we
>       // assume there has been a jump in the picture id due to some encoder
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.h
>index ae84fcf9e6c293810d44432ef3a009db41eef74e..dc5e5a2e372fdc10ac80ac6ee71f769c8d86c983 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.h
>@@ -79,6 +79,9 @@ class FrameBuffer {
>   // Updates the RTT for jitter buffer estimation.
>   void UpdateRtt(int64_t rtt_ms);
> 
>+  // Clears the FrameBuffer, removing all the buffered frames.
>+  void Clear();
>+
>  private:
>   struct FrameInfo {
>     FrameInfo();
>@@ -161,7 +164,7 @@ class FrameBuffer {
>   VCMJitterEstimator* const jitter_estimator_ RTC_GUARDED_BY(crit_);
>   VCMTiming* const timing_ RTC_GUARDED_BY(crit_);
>   VCMInterFrameDelay inter_frame_delay_ RTC_GUARDED_BY(crit_);
>-  uint32_t last_decoded_frame_timestamp_ RTC_GUARDED_BY(crit_);
>+  absl::optional<uint32_t> last_decoded_frame_timestamp_ RTC_GUARDED_BY(crit_);
>   FrameMap::iterator last_decoded_frame_it_ RTC_GUARDED_BY(crit_);
>   FrameMap::iterator last_continuous_frame_it_ RTC_GUARDED_BY(crit_);
>   FrameMap::iterator next_frame_it_ RTC_GUARDED_BY(crit_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2_unittest.cc
>index 357ba86540551f992aed561c5ff9138532de02d3..e10f78527957e77c075a0f0827b37d6d7bf4caf5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2_unittest.cc
>@@ -137,9 +137,7 @@ class TestFrameBuffer2 : public ::testing::Test {
>                                 &stats_callback_)),
>         rand_(0x34678213),
>         tear_down_(false),
>-        extract_thread_(&ExtractLoop, this, "Extract Thread"),
>-        trigger_extract_event_(false, false),
>-        crit_acquired_event_(false, false) {}
>+        extract_thread_(&ExtractLoop, this, "Extract Thread") {}
> 
>   void SetUp() override { extract_thread_.Start(); }
> 
>@@ -600,5 +598,21 @@ TEST_F(TestFrameBuffer2, DontUpdateOnUndecodableFrame) {
>   ExtractFrame(0, true);
> }
> 
>+TEST_F(TestFrameBuffer2, DontDecodeOlderTimestamp) {
>+  InsertFrame(2, 0, 1, false);
>+  InsertFrame(1, 0, 2, false);  // Older picture id but newer timestamp.
>+  ExtractFrame(0);
>+  ExtractFrame(0);
>+  CheckFrame(0, 1, 0);
>+  CheckNoFrame(1);
>+
>+  InsertFrame(3, 0, 4, false);
>+  InsertFrame(4, 0, 3, false);  // Newer picture id but older timestamp.
>+  ExtractFrame(0);
>+  ExtractFrame(0);
>+  CheckFrame(2, 3, 0);
>+  CheckNoFrame(3);
>+}
>+
> }  // namespace video_coding
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.cc
>index b1d7c28abd24733738f6087f8c9393aa05fee0b3..d5cc0a8ba068dcfdfef5eaf24d78911156ebb9d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.cc
>@@ -101,26 +101,14 @@ int32_t VCMGenericEncoder::Encode(const VideoFrame& frame,
> 
> void VCMGenericEncoder::SetEncoderParameters(const EncoderParameters& params) {
>   RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
>-  bool channel_parameters_have_changed;
>   bool rates_have_changed;
>   {
>     rtc::CritScope lock(&params_lock_);
>-    channel_parameters_have_changed =
>-        params.loss_rate != encoder_params_.loss_rate ||
>-        params.rtt != encoder_params_.rtt;
>     rates_have_changed =
>         params.target_bitrate != encoder_params_.target_bitrate ||
>         params.input_frame_rate != encoder_params_.input_frame_rate;
>     encoder_params_ = params;
>   }
>-  if (channel_parameters_have_changed) {
>-    int res = encoder_->SetChannelParameters(params.loss_rate, params.rtt);
>-    if (res != 0) {
>-      RTC_LOG(LS_WARNING) << "Error set encoder parameters (loss = "
>-                          << params.loss_rate << ", rtt = " << params.rtt
>-                          << "): " << res;
>-    }
>-  }
>   if (rates_have_changed) {
>     int res = encoder_->SetRateAllocation(params.target_bitrate,
>                                           params.input_frame_rate);
>@@ -164,9 +152,9 @@ bool VCMGenericEncoder::InternalSource() const {
>   return internal_source_;
> }
> 
>-bool VCMGenericEncoder::SupportsNativeHandle() const {
>+VideoEncoder::EncoderInfo VCMGenericEncoder::GetEncoderInfo() const {
>   RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
>-  return encoder_->SupportsNativeHandle();
>+  return encoder_->GetEncoderInfo();
> }
> 
> VCMEncodedFrameCallback::VCMEncodedFrameCallback(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.h
>index 151e93e49759796095aedfeef9a40d13b410ad9c..2f841b34dd815d2995e48b4ac96813b0f0874dd8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.h
>@@ -146,7 +146,7 @@ class VCMGenericEncoder {
> 
>   int32_t RequestFrame(const std::vector<FrameType>& frame_types);
>   bool InternalSource() const;
>-  bool SupportsNativeHandle() const;
>+  VideoEncoder::EncoderInfo GetEncoderInfo() const;
> 
>  private:
>   rtc::RaceChecker race_checker_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h
>index c29efeead68d4630c3917b610ba35182ab14493c..7d003591ea3d95cc7ea744103da39d4267b710d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h
>@@ -19,6 +19,8 @@
> 
> namespace webrtc {
> 
>+// NOTE: Deprecated file, include api/mock_video_(encoder|decoder).h instead.
>+
> class MockEncodedImageCallback : public EncodedImageCallback {
>  public:
>   MOCK_METHOD3(OnEncodedImage,
>@@ -42,7 +44,6 @@ class MockVideoEncoder : public VideoEncoder {
>                int32_t(EncodedImageCallback* callback));
>   MOCK_METHOD0(Release, int32_t());
>   MOCK_METHOD0(Reset, int32_t());
>-  MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
>   MOCK_METHOD2(SetRates, int32_t(uint32_t newBitRate, uint32_t frameRate));
>   MOCK_METHOD2(SetRateAllocation,
>                int32_t(const VideoBitrateAllocation& newBitRate,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_initializer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_initializer.h
>index ee70810f62ed73d358491b17216ca3210713b7b9..e979f9c86776e2b9edbe746bd6d95823dbc923d8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_initializer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_initializer.h
>@@ -30,15 +30,9 @@ class VideoCodecInitializer {
>   // type used. For instance, VP8 will create an allocator than can handle
>   // simulcast and temporal layering.
>   // GetBitrateAllocator is called implicitly from here, no need to call again.
>-  static bool SetupCodec(
>-      const VideoEncoderConfig& config,
>-      const std::vector<VideoStream>& streams,
>-      VideoCodec* codec,
>-      std::unique_ptr<VideoBitrateAllocator>* bitrate_allocator);
>-
>-  // Create a bitrate allocator for the specified codec.
>-  static std::unique_ptr<VideoBitrateAllocator> CreateBitrateAllocator(
>-      const VideoCodec& codec);
>+  static bool SetupCodec(const VideoEncoderConfig& config,
>+                         const std::vector<VideoStream>& streams,
>+                         VideoCodec* codec);
> 
>  private:
>   static VideoCodec VideoEncoderConfigToVideoCodec(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_interface.h
>index d2badf9201b21ee4aa326386d201c49541a19b4e..38583bfa3b410c6285b32cf2b9654b7e076f2f8d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_interface.h
>@@ -79,6 +79,7 @@ struct CodecSpecificInfo {
>     memset(&codecSpecific, 0, sizeof(codecSpecific));
>   }
>   VideoCodecType codecType;
>+  // |codec_name| is deprecated, use name provided by VideoEncoder instead.
>   const char* codec_name;
>   CodecSpecificInfoUnion codecSpecific;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding.h
>index 8ef046aa23193f7f30f2b1ecded11ff73c834d65..0889b231dea347cce047c0547dbea7ce80534120 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding.h
>@@ -27,6 +27,7 @@
> #include "modules/include/module.h"
> #include "modules/include/module_common_types.h"
> #include "modules/video_coding/include/video_coding_defines.h"
>+#include "rtc_base/deprecation.h"
> #include "system_wrappers/include/event_wrapper.h"
> 
> namespace webrtc {
>@@ -37,20 +38,6 @@ class VideoDecoder;
> class VideoEncoder;
> struct CodecSpecificInfo;
> 
>-class EventFactory {
>- public:
>-  virtual ~EventFactory() {}
>-
>-  virtual EventWrapper* CreateEvent() = 0;
>-};
>-
>-class EventFactoryImpl : public EventFactory {
>- public:
>-  ~EventFactoryImpl() override {}
>-
>-  EventWrapper* CreateEvent() override;
>-};
>-
> // Used to indicate which decode with errors mode should be used.
> enum VCMDecodeErrorMode {
>   kNoErrors,         // Never decode with errors. Video will freeze
>@@ -69,7 +56,7 @@ class VideoCodingModule : public Module {
>   enum SenderNackMode { kNackNone, kNackAll, kNackSelective };
> 
>   // DEPRECATED.
>-  static VideoCodingModule* Create(Clock* clock, EventFactory* event_factory);
>+  static VideoCodingModule* Create(Clock* clock);
> 
>   /*
>    *   Sender
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding_defines.h
>index 7fb6f58364773f8fa818396e1cb7f684f8edb06f..7bbc98fd4831a8e28a440cee9ad27f6826960c15 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding_defines.h
>@@ -15,8 +15,6 @@
> #include <vector>
> 
> #include "api/video/video_frame.h"
>-// For EncodedImage
>-#include "common_video/include/video_frame.h"
> #include "modules/include/module_common_types.h"
> 
> namespace webrtc {
>@@ -43,7 +41,7 @@ enum {
>   // |kDefaultTimingFramesDelayMs|, or if the frame is at least
>   // |kDefaultOutliserFrameSizePercent| in size of average frame.
>   kDefaultTimingFramesDelayMs = 200,
>-  kDefaultOutlierFrameSizePercent = 250,
>+  kDefaultOutlierFrameSizePercent = 500,
>   // Maximum number of frames for what we store encode start timing information.
>   kMaxEncodeStartTimeListSize = 50,
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.h
>index f649adc8b3ab359304ad1c5ccbce3b246f884800..482627604553cd321222f2bda84d6ce5d9759044 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.h
>@@ -35,7 +35,6 @@ enum VCMNackMode { kNack, kNoNack };
> 
> // forward declarations
> class Clock;
>-class EventFactory;
> class EventWrapper;
> class VCMFrameBuffer;
> class VCMPacket;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_unittest.cc
>index 6ae65d62e986c728cad98d47d4eacbbb1ecb6d0c..3ed18a938650b7be0dcdc6539ecebd261a0bee5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_unittest.cc
>@@ -20,7 +20,6 @@
> #include "modules/video_coding/media_opt_util.h"
> #include "modules/video_coding/packet.h"
> #include "modules/video_coding/test/stream_generator.h"
>-#include "modules/video_coding/test/test_util.h"
> #include "rtc_base/location.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/field_trial.h"
>@@ -220,9 +219,7 @@ class TestBasicJitterBuffer : public ::testing::TestWithParam<std::string>,
>   void SetUp() override {
>     clock_.reset(new SimulatedClock(0));
>     jitter_buffer_.reset(new VCMJitterBuffer(
>-        clock_.get(),
>-        std::unique_ptr<EventWrapper>(event_factory_.CreateEvent()), this,
>-        this));
>+        clock_.get(), absl::WrapUnique(EventWrapper::Create()), this, this));
>     jitter_buffer_->Start();
>     seq_num_ = 1234;
>     timestamp_ = 0;
>@@ -313,7 +310,6 @@ class TestBasicJitterBuffer : public ::testing::TestWithParam<std::string>,
>   uint8_t data_[1500];
>   std::unique_ptr<VCMPacket> packet_;
>   std::unique_ptr<SimulatedClock> clock_;
>-  NullEventFactory event_factory_;
>   std::unique_ptr<VCMJitterBuffer> jitter_buffer_;
> };
> 
>@@ -339,9 +335,7 @@ class TestRunningJitterBuffer : public ::testing::TestWithParam<std::string>,
>     max_nack_list_size_ = 150;
>     oldest_packet_to_nack_ = 250;
>     jitter_buffer_ = new VCMJitterBuffer(
>-        clock_.get(),
>-        std::unique_ptr<EventWrapper>(event_factory_.CreateEvent()), this,
>-        this);
>+        clock_.get(), absl::WrapUnique(EventWrapper::Create()), this, this);
>     stream_generator_ = new StreamGenerator(0, clock_->TimeInMilliseconds());
>     jitter_buffer_->Start();
>     jitter_buffer_->SetNackSettings(max_nack_list_size_, oldest_packet_to_nack_,
>@@ -433,7 +427,6 @@ class TestRunningJitterBuffer : public ::testing::TestWithParam<std::string>,
>   VCMJitterBuffer* jitter_buffer_;
>   StreamGenerator* stream_generator_;
>   std::unique_ptr<SimulatedClock> clock_;
>-  NullEventFactory event_factory_;
>   size_t max_nack_list_size_;
>   int oldest_packet_to_nack_;
>   uint8_t data_buffer_[kDataBufferSize];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.cc
>index 5e754f10885edc6655e2e7152a447bed4038597b..0d0fac7291e8fddd41dc6b355a935f6496d164c1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.cc
>@@ -14,18 +14,24 @@
> #include <math.h>
> #include <stdlib.h>
> #include <string.h>
>+#include <algorithm>
> #include <string>
> 
>+#include "absl/types/optional.h"
> #include "modules/video_coding/internal_defines.h"
> #include "modules/video_coding/rtt_filter.h"
>+#include "rtc_base/experiments/jitter_upper_bound_experiment.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
>-
>-enum { kStartupDelaySamples = 30 };
>-enum { kFsAccuStartupSamples = 5 };
>-enum { kMaxFramerateEstimate = 200 };
>+namespace {
>+static constexpr uint32_t kStartupDelaySamples = 30;
>+static constexpr int64_t kFsAccuStartupSamples = 5;
>+static constexpr double kMaxFramerateEstimate = 200.0;
>+static constexpr int64_t kNackCountTimeoutMs = 60000;
>+static constexpr double kDefaultMaxTimestampDeviationInSigmas = 3.5;
>+}  // namespace
> 
> VCMJitterEstimator::VCMJitterEstimator(const Clock* clock,
>                                        int32_t vcmId,
>@@ -45,7 +51,9 @@ VCMJitterEstimator::VCMJitterEstimator(const Clock* clock,
>       _rttFilter(),
>       fps_counter_(30),  // TODO(sprang): Use an estimator with limit based on
>                          // time, rather than number of samples.
>-      low_rate_experiment_(kInit),
>+      time_deviation_upper_bound_(
>+          JitterUpperBoundExperiment::GetUpperBoundSigmas().value_or(
>+              kDefaultMaxTimestampDeviationInSigmas)),
>       clock_(clock) {
>   Reset();
> }
>@@ -75,6 +83,7 @@ VCMJitterEstimator& VCMJitterEstimator::operator=(
>     _latestNackTimestamp = rhs._latestNackTimestamp;
>     _nackCount = rhs._nackCount;
>     _rttFilter = rhs._rttFilter;
>+    clock_ = rhs.clock_;
>   }
>   return *this;
> }
>@@ -102,6 +111,7 @@ void VCMJitterEstimator::Reset() {
>   _filterJitterEstimate = 0.0;
>   _latestNackTimestamp = 0;
>   _nackCount = 0;
>+  _latestNackTimestamp = 0;
>   _fsSum = 0;
>   _fsCount = 0;
>   _startupCount = 0;
>@@ -155,6 +165,12 @@ void VCMJitterEstimator::UpdateEstimate(int64_t frameDelayMS,
>   }
>   _prevFrameSize = frameSizeBytes;
> 
>+  // Cap frameDelayMS based on the current time deviation noise.
>+  int64_t max_time_deviation_ms =
>+      static_cast<int64_t>(time_deviation_upper_bound_ * sqrt(_varNoise) + 0.5);
>+  frameDelayMS = std::max(std::min(frameDelayMS, max_time_deviation_ms),
>+                          -max_time_deviation_ms);
>+
>   // Only update the Kalman filter if the sample is not considered
>   // an extreme outlier. Even if it is an extreme outlier from a
>   // delay point of view, if the frame size also is large the
>@@ -193,15 +209,10 @@ void VCMJitterEstimator::UpdateEstimate(int64_t frameDelayMS,
> 
> // Updates the nack/packet ratio
> void VCMJitterEstimator::FrameNacked() {
>-  // Wait until _nackLimit retransmissions has been received,
>-  // then always add ~1 RTT delay.
>-  // TODO(holmer): Should we ever remove the additional delay if the
>-  // the packet losses seem to have stopped? We could for instance scale
>-  // the number of RTTs to add with the amount of retransmissions in a given
>-  // time interval, or similar.
>   if (_nackCount < _nackLimit) {
>     _nackCount++;
>   }
>+  _latestNackTimestamp = clock_->TimeInMicroseconds();
> }
> 
> // Updates Kalman estimate of the channel
>@@ -309,22 +320,20 @@ void VCMJitterEstimator::EstimateRandomJitter(double d_dT,
>   if (_alphaCount > _alphaCountMax)
>     _alphaCount = _alphaCountMax;
> 
>-  if (LowRateExperimentEnabled()) {
>-    // In order to avoid a low frame rate stream to react slower to changes,
>-    // scale the alpha weight relative a 30 fps stream.
>-    double fps = GetFrameRate();
>-    if (fps > 0.0) {
>-      double rate_scale = 30.0 / fps;
>-      // At startup, there can be a lot of noise in the fps estimate.
>-      // Interpolate rate_scale linearly, from 1.0 at sample #1, to 30.0 / fps
>-      // at sample #kStartupDelaySamples.
>-      if (_alphaCount < kStartupDelaySamples) {
>-        rate_scale =
>-            (_alphaCount * rate_scale + (kStartupDelaySamples - _alphaCount)) /
>-            kStartupDelaySamples;
>-      }
>-      alpha = pow(alpha, rate_scale);
>+  // In order to avoid a low frame rate stream to react slower to changes,
>+  // scale the alpha weight relative a 30 fps stream.
>+  double fps = GetFrameRate();
>+  if (fps > 0.0) {
>+    double rate_scale = 30.0 / fps;
>+    // At startup, there can be a lot of noise in the fps estimate.
>+    // Interpolate rate_scale linearly, from 1.0 at sample #1, to 30.0 / fps
>+    // at sample #kStartupDelaySamples.
>+    if (_alphaCount < kStartupDelaySamples) {
>+      rate_scale =
>+          (_alphaCount * rate_scale + (kStartupDelaySamples - _alphaCount)) /
>+          kStartupDelaySamples;
>     }
>+    alpha = pow(alpha, rate_scale);
>   }
> 
>   double avgNoise = alpha * _avgNoise + (1 - alpha) * d_dT;
>@@ -386,46 +395,35 @@ void VCMJitterEstimator::UpdateMaxFrameSize(uint32_t frameSizeBytes) {
> // otherwise tries to calculate an estimate.
> int VCMJitterEstimator::GetJitterEstimate(double rttMultiplier) {
>   double jitterMS = CalculateEstimate() + OPERATING_SYSTEM_JITTER;
>+  uint64_t now = clock_->TimeInMicroseconds();
>+
>+  if (now - _latestNackTimestamp > kNackCountTimeoutMs * 1000)
>+    _nackCount = 0;
>+
>   if (_filterJitterEstimate > jitterMS)
>     jitterMS = _filterJitterEstimate;
>   if (_nackCount >= _nackLimit)
>     jitterMS += _rttFilter.RttMs() * rttMultiplier;
> 
>-  if (LowRateExperimentEnabled()) {
>-    static const double kJitterScaleLowThreshold = 5.0;
>-    static const double kJitterScaleHighThreshold = 10.0;
>-    double fps = GetFrameRate();
>-    // Ignore jitter for very low fps streams.
>-    if (fps < kJitterScaleLowThreshold) {
>-      if (fps == 0.0) {
>-        return jitterMS;
>-      }
>-      return 0;
>-    }
>-
>-    // Semi-low frame rate; scale by factor linearly interpolated from 0.0 at
>-    // kJitterScaleLowThreshold to 1.0 at kJitterScaleHighThreshold.
>-    if (fps < kJitterScaleHighThreshold) {
>-      jitterMS =
>-          (1.0 / (kJitterScaleHighThreshold - kJitterScaleLowThreshold)) *
>-          (fps - kJitterScaleLowThreshold) * jitterMS;
>+  static const double kJitterScaleLowThreshold = 5.0;
>+  static const double kJitterScaleHighThreshold = 10.0;
>+  double fps = GetFrameRate();
>+  // Ignore jitter for very low fps streams.
>+  if (fps < kJitterScaleLowThreshold) {
>+    if (fps == 0.0) {
>+      return rtc::checked_cast<int>(std::max(0.0, jitterMS) + 0.5);
>     }
>+    return 0;
>   }
> 
>-  return static_cast<uint32_t>(jitterMS + 0.5);
>-}
>-
>-bool VCMJitterEstimator::LowRateExperimentEnabled() {
>-  if (low_rate_experiment_ == kInit) {
>-    std::string group =
>-        webrtc::field_trial::FindFullName("WebRTC-ReducedJitterDelay");
>-    if (group == "Disabled") {
>-      low_rate_experiment_ = kDisabled;
>-    } else {
>-      low_rate_experiment_ = kEnabled;
>-    }
>+  // Semi-low frame rate; scale by factor linearly interpolated from 0.0 at
>+  // kJitterScaleLowThreshold to 1.0 at kJitterScaleHighThreshold.
>+  if (fps < kJitterScaleHighThreshold) {
>+    jitterMS = (1.0 / (kJitterScaleHighThreshold - kJitterScaleLowThreshold)) *
>+               (fps - kJitterScaleLowThreshold) * jitterMS;
>   }
>-  return low_rate_experiment_ == kEnabled ? true : false;
>+
>+  return rtc::checked_cast<int>(std::max(0.0, jitterMS) + 0.5);
> }
> 
> double VCMJitterEstimator::GetFrameRate() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.h
>index aeba3eded33cd157c54ce75ca31d1fcf43386c0d..56b532851d939ce8936606dc62cfccce4b71871e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.h
>@@ -72,8 +72,6 @@ class VCMJitterEstimator {
>   double _theta[2];  // Estimated line parameters (slope, offset)
>   double _varNoise;  // Variance of the time-deviation from the line
> 
>-  virtual bool LowRateExperimentEnabled();
>-
>  private:
>   // Updates the Kalman filter for the line describing
>   // the frame size dependent jitter.
>@@ -159,8 +157,7 @@ class VCMJitterEstimator {
>   VCMRttFilter _rttFilter;
> 
>   rtc::RollingAccumulator<uint64_t> fps_counter_;
>-  enum ExperimentFlag { kInit, kEnabled, kDisabled };
>-  ExperimentFlag low_rate_experiment_;
>+  const double time_deviation_upper_bound_;
>   const Clock* clock_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator_tests.cc
>index f00e1ef6da8b64565972c8cdacd7493ed2422fb7..45262ef83602e7c15e2a9cf3b9e8066f35a92764 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator_tests.cc
>@@ -9,38 +9,30 @@
> 
> #include "modules/video_coding/jitter_estimator.h"
> 
>+#include "rtc_base/experiments/jitter_upper_bound_experiment.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/histogram_percentile_counter.h"
>+#include "rtc_base/timeutils.h"
> #include "system_wrappers/include/clock.h"
>+#include "test/field_trial.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> 
>-class TestEstimator : public VCMJitterEstimator {
>- public:
>-  explicit TestEstimator(bool exp_enabled)
>-      : VCMJitterEstimator(&fake_clock_, 0, 0),
>-        fake_clock_(0),
>-        exp_enabled_(exp_enabled) {}
>+class TestVCMJitterEstimator : public ::testing::Test {
>+ protected:
>+  TestVCMJitterEstimator() : fake_clock_(0) {}
> 
>-  virtual bool LowRateExperimentEnabled() { return exp_enabled_; }
>+  virtual void SetUp() {
>+    estimator_ = absl::make_unique<VCMJitterEstimator>(&fake_clock_, 0, 0);
>+  }
> 
>   void AdvanceClock(int64_t microseconds) {
>     fake_clock_.AdvanceTimeMicroseconds(microseconds);
>   }
> 
>- private:
>   SimulatedClock fake_clock_;
>-  const bool exp_enabled_;
>-};
>-
>-class TestVCMJitterEstimator : public ::testing::Test {
>- protected:
>-  TestVCMJitterEstimator()
>-      : regular_estimator_(false), low_rate_estimator_(true) {}
>-
>-  virtual void SetUp() { regular_estimator_.Reset(); }
>-
>-  TestEstimator regular_estimator_;
>-  TestEstimator low_rate_estimator_;
>+  std::unique_ptr<VCMJitterEstimator> estimator_;
> };
> 
> // Generates some simple test data in the form of a sawtooth wave.
>@@ -64,98 +56,56 @@ class ValueGenerator {
> // 5 fps, disable jitter delay altogether.
> TEST_F(TestVCMJitterEstimator, TestLowRate) {
>   ValueGenerator gen(10);
>-  uint64_t time_delta = 1000000 / 5;
>+  uint64_t time_delta_us = rtc::kNumMicrosecsPerSec / 5;
>   for (int i = 0; i < 60; ++i) {
>-    regular_estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
>-    regular_estimator_.AdvanceClock(time_delta);
>-    low_rate_estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
>-    low_rate_estimator_.AdvanceClock(time_delta);
>-    EXPECT_GT(regular_estimator_.GetJitterEstimate(0), 0);
>+    estimator_->UpdateEstimate(gen.Delay(), gen.FrameSize());
>+    AdvanceClock(time_delta_us);
>     if (i > 2)
>-      EXPECT_EQ(low_rate_estimator_.GetJitterEstimate(0), 0);
>-    gen.Advance();
>-  }
>-}
>-
>-// 8 fps, steady state estimate should be in interpolated interval between 0
>-// and value of previous method.
>-TEST_F(TestVCMJitterEstimator, TestMidRate) {
>-  ValueGenerator gen(10);
>-  uint64_t time_delta = 1000000 / 8;
>-  for (int i = 0; i < 60; ++i) {
>-    regular_estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
>-    regular_estimator_.AdvanceClock(time_delta);
>-    low_rate_estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
>-    low_rate_estimator_.AdvanceClock(time_delta);
>-    EXPECT_GT(regular_estimator_.GetJitterEstimate(0), 0);
>-    EXPECT_GT(low_rate_estimator_.GetJitterEstimate(0), 0);
>-    EXPECT_GE(regular_estimator_.GetJitterEstimate(0),
>-              low_rate_estimator_.GetJitterEstimate(0));
>+      EXPECT_EQ(estimator_->GetJitterEstimate(0), 0);
>     gen.Advance();
>   }
> }
> 
>-// 30 fps, steady state estimate should be same as previous method.
>-TEST_F(TestVCMJitterEstimator, TestHighRate) {
>-  ValueGenerator gen(10);
>-  uint64_t time_delta = 1000000 / 30;
>-  for (int i = 0; i < 60; ++i) {
>-    regular_estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
>-    regular_estimator_.AdvanceClock(time_delta);
>-    low_rate_estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
>-    low_rate_estimator_.AdvanceClock(time_delta);
>-    EXPECT_EQ(regular_estimator_.GetJitterEstimate(0),
>-              low_rate_estimator_.GetJitterEstimate(0));
>-    gen.Advance();
>-  }
>-}
>-
>-// 10 fps, high jitter then low jitter. Low rate estimator should converge
>-// faster to low noise estimate.
>-TEST_F(TestVCMJitterEstimator, TestConvergence) {
>-  // Reach a steady state with high noise.
>-  ValueGenerator gen(50);
>-  uint64_t time_delta = 1000000 / 10;
>-  for (int i = 0; i < 100; ++i) {
>-    regular_estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
>-    regular_estimator_.AdvanceClock(time_delta * 2);
>-    low_rate_estimator_.UpdateEstimate(gen.Delay(), gen.FrameSize());
>-    low_rate_estimator_.AdvanceClock(time_delta * 2);
>-    gen.Advance();
>-  }
>-
>-  int threshold = regular_estimator_.GetJitterEstimate(0) / 2;
>-
>-  // New generator with zero noise.
>-  ValueGenerator low_gen(0);
>-  int regular_iterations = 0;
>-  int low_rate_iterations = 0;
>-  for (int i = 0; i < 500; ++i) {
>-    if (regular_iterations == 0) {
>-      regular_estimator_.UpdateEstimate(low_gen.Delay(), low_gen.FrameSize());
>-      regular_estimator_.AdvanceClock(time_delta);
>-      if (regular_estimator_.GetJitterEstimate(0) < threshold) {
>-        regular_iterations = i;
>-      }
>+TEST_F(TestVCMJitterEstimator, TestUpperBound) {
>+  struct TestContext {
>+    TestContext() : upper_bound(0.0), percentiles(1000) {}
>+    double upper_bound;
>+    rtc::HistogramPercentileCounter percentiles;
>+  };
>+  std::vector<TestContext> test_cases(2);
>+
>+  test_cases[0].upper_bound = 100.0;  // First use essentially no cap.
>+  test_cases[1].upper_bound = 3.5;    // Second, reasonably small cap.
>+
>+  for (TestContext& context : test_cases) {
>+    // Set up field trial and reset jitter estimator.
>+    char string_buf[64];
>+    rtc::SimpleStringBuilder ssb(string_buf);
>+    ssb << JitterUpperBoundExperiment::kJitterUpperBoundExperimentName
>+        << "/Enabled-" << context.upper_bound << "/";
>+    test::ScopedFieldTrials field_trials(ssb.str());
>+    SetUp();
>+
>+    ValueGenerator gen(50);
>+    uint64_t time_delta_us = rtc::kNumMicrosecsPerSec / 30;
>+    for (int i = 0; i < 100; ++i) {
>+      estimator_->UpdateEstimate(gen.Delay(), gen.FrameSize());
>+      AdvanceClock(time_delta_us);
>+      context.percentiles.Add(
>+          static_cast<uint32_t>(estimator_->GetJitterEstimate(0)));
>+      gen.Advance();
>     }
>-
>-    if (low_rate_iterations == 0) {
>-      low_rate_estimator_.UpdateEstimate(low_gen.Delay(), low_gen.FrameSize());
>-      low_rate_estimator_.AdvanceClock(time_delta);
>-      if (low_rate_estimator_.GetJitterEstimate(0) < threshold) {
>-        low_rate_iterations = i;
>-      }
>-    }
>-
>-    if (regular_iterations != 0 && low_rate_iterations != 0) {
>-      break;
>-    }
>-
>-    gen.Advance();
>   }
> 
>-  EXPECT_NE(regular_iterations, 0);
>-  EXPECT_NE(low_rate_iterations, 0);
>-  EXPECT_LE(low_rate_iterations, regular_iterations);
>+  // Median should be similar after three seconds. Allow 5% error margin.
>+  uint32_t median_unbound = *test_cases[0].percentiles.GetPercentile(0.5);
>+  uint32_t median_bounded = *test_cases[1].percentiles.GetPercentile(0.5);
>+  EXPECT_NEAR(median_unbound, median_bounded, (median_unbound * 5) / 100);
>+
>+  // Max should be lower for the bounded case.
>+  uint32_t max_unbound = *test_cases[0].percentiles.GetPercentile(1.0);
>+  uint32_t max_bounded = *test_cases[1].percentiles.GetPercentile(1.0);
>+  EXPECT_GT(max_unbound, static_cast<uint32_t>(max_bounded * 1.25));
> }
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.cc
>index 7fbcf34b41a91373bd7861ae0f5dfe3e8e1f7fd1..0c69b57749349ffafce1d0877e5fd50704a71071 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.cc
>@@ -16,6 +16,7 @@
> #include "modules/utility/include/process_thread.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
>@@ -28,14 +29,29 @@ const int kProcessFrequency = 50;
> const int kProcessIntervalMs = 1000 / kProcessFrequency;
> const int kMaxReorderedPackets = 128;
> const int kNumReorderingBuckets = 10;
>+const int kDefaultSendNackDelayMs = 0;
>+
>+int64_t GetSendNackDelay() {
>+  int64_t delay_ms = strtol(
>+      webrtc::field_trial::FindFullName("WebRTC-SendNackDelayMs").c_str(),
>+      nullptr, 10);
>+  if (delay_ms > 0 && delay_ms <= 20) {
>+    RTC_LOG(LS_INFO) << "SendNackDelay is set to " << delay_ms;
>+    return delay_ms;
>+  }
>+  return kDefaultSendNackDelayMs;
>+}
> }  // namespace
> 
> NackModule::NackInfo::NackInfo()
>     : seq_num(0), send_at_seq_num(0), sent_at_time(-1), retries(0) {}
> 
>-NackModule::NackInfo::NackInfo(uint16_t seq_num, uint16_t send_at_seq_num)
>+NackModule::NackInfo::NackInfo(uint16_t seq_num,
>+                               uint16_t send_at_seq_num,
>+                               int64_t created_at_time)
>     : seq_num(seq_num),
>       send_at_seq_num(send_at_seq_num),
>+      created_at_time(created_at_time),
>       sent_at_time(-1),
>       retries(0) {}
> 
>@@ -49,13 +65,20 @@ NackModule::NackModule(Clock* clock,
>       initialized_(false),
>       rtt_ms_(kDefaultRttMs),
>       newest_seq_num_(0),
>-      next_process_time_ms_(-1) {
>+      next_process_time_ms_(-1),
>+      send_nack_delay_ms_(GetSendNackDelay()) {
>   RTC_DCHECK(clock_);
>   RTC_DCHECK(nack_sender_);
>   RTC_DCHECK(keyframe_request_sender_);
> }
> 
> int NackModule::OnReceivedPacket(uint16_t seq_num, bool is_keyframe) {
>+  return OnReceivedPacket(seq_num, is_keyframe, false);
>+}
>+
>+int NackModule::OnReceivedPacket(uint16_t seq_num,
>+                                 bool is_keyframe,
>+                                 bool is_recovered) {
>   rtc::CritScope lock(&crit_);
>   // TODO(philipel): When the packet includes information whether it is
>   //                 retransmitted or not, use that value instead. For
>@@ -88,8 +111,6 @@ int NackModule::OnReceivedPacket(uint16_t seq_num, bool is_keyframe) {
>       UpdateReorderingStatistics(seq_num);
>     return nacks_sent_for_packet;
>   }
>-  AddPacketsToNack(newest_seq_num_ + 1, seq_num);
>-  newest_seq_num_ = seq_num;
> 
>   // Keep track of new keyframes.
>   if (is_keyframe)
>@@ -100,6 +121,21 @@ int NackModule::OnReceivedPacket(uint16_t seq_num, bool is_keyframe) {
>   if (it != keyframe_list_.begin())
>     keyframe_list_.erase(keyframe_list_.begin(), it);
> 
>+  if (is_recovered) {
>+    recovered_list_.insert(seq_num);
>+
>+    // Remove old ones so we don't accumulate recovered packets.
>+    auto it = recovered_list_.lower_bound(seq_num - kMaxPacketAge);
>+    if (it != recovered_list_.begin())
>+      recovered_list_.erase(recovered_list_.begin(), it);
>+
>+    // Do not send nack for packets recovered by FEC or RTX.
>+    return 0;
>+  }
>+
>+  AddPacketsToNack(newest_seq_num_ + 1, seq_num);
>+  newest_seq_num_ = seq_num;
>+
>   // Are there any nacks that are waiting for this seq_num.
>   std::vector<uint16_t> nack_batch = GetNackBatch(kSeqNumOnly);
>   if (!nack_batch.empty())
>@@ -113,6 +149,8 @@ void NackModule::ClearUpTo(uint16_t seq_num) {
>   nack_list_.erase(nack_list_.begin(), nack_list_.lower_bound(seq_num));
>   keyframe_list_.erase(keyframe_list_.begin(),
>                        keyframe_list_.lower_bound(seq_num));
>+  recovered_list_.erase(recovered_list_.begin(),
>+                        recovered_list_.lower_bound(seq_num));
> }
> 
> void NackModule::UpdateRtt(int64_t rtt_ms) {
>@@ -124,6 +162,7 @@ void NackModule::Clear() {
>   rtc::CritScope lock(&crit_);
>   nack_list_.clear();
>   keyframe_list_.clear();
>+  recovered_list_.clear();
> }
> 
> int64_t NackModule::TimeUntilNextProcess() {
>@@ -200,7 +239,11 @@ void NackModule::AddPacketsToNack(uint16_t seq_num_start,
>   }
> 
>   for (uint16_t seq_num = seq_num_start; seq_num != seq_num_end; ++seq_num) {
>-    NackInfo nack_info(seq_num, seq_num + WaitNumberOfPackets(0.5));
>+    // Do not send nack for packets that are already recovered by FEC or RTX
>+    if (recovered_list_.find(seq_num) != recovered_list_.end())
>+      continue;
>+    NackInfo nack_info(seq_num, seq_num + WaitNumberOfPackets(0.5),
>+                       clock_->TimeInMilliseconds());
>     RTC_DCHECK(nack_list_.find(seq_num) == nack_list_.end());
>     nack_list_[seq_num] = nack_info;
>   }
>@@ -213,22 +256,14 @@ std::vector<uint16_t> NackModule::GetNackBatch(NackFilterOptions options) {
>   std::vector<uint16_t> nack_batch;
>   auto it = nack_list_.begin();
>   while (it != nack_list_.end()) {
>-    if (consider_seq_num && it->second.sent_at_time == -1 &&
>-        AheadOrAt(newest_seq_num_, it->second.send_at_seq_num)) {
>-      nack_batch.emplace_back(it->second.seq_num);
>-      ++it->second.retries;
>-      it->second.sent_at_time = now_ms;
>-      if (it->second.retries >= kMaxNackRetries) {
>-        RTC_LOG(LS_WARNING) << "Sequence number " << it->second.seq_num
>-                            << " removed from NACK list due to max retries.";
>-        it = nack_list_.erase(it);
>-      } else {
>-        ++it;
>-      }
>-      continue;
>-    }
>-
>-    if (consider_timestamp && it->second.sent_at_time + rtt_ms_ <= now_ms) {
>+    bool delay_timed_out =
>+        now_ms - it->second.created_at_time >= send_nack_delay_ms_;
>+    bool nack_on_rtt_passed = now_ms - it->second.sent_at_time >= rtt_ms_;
>+    bool nack_on_seq_num_passed =
>+        it->second.sent_at_time == -1 &&
>+        AheadOrAt(newest_seq_num_, it->second.send_at_seq_num);
>+    if (delay_timed_out && ((consider_seq_num && nack_on_seq_num_passed) ||
>+                            (consider_timestamp && nack_on_rtt_passed))) {
>       nack_batch.emplace_back(it->second.seq_num);
>       ++it->second.retries;
>       it->second.sent_at_time = now_ms;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.h
>index 44340e9d4ca476454ca7d0b3b5a687f57dd743e9..f86dd16db28ac30e0f4a136c04c5b53a9a9532bc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.h
>@@ -32,6 +32,8 @@ class NackModule : public Module {
>              KeyFrameRequestSender* keyframe_request_sender);
> 
>   int OnReceivedPacket(uint16_t seq_num, bool is_keyframe);
>+  int OnReceivedPacket(uint16_t seq_num, bool is_keyframe, bool is_recovered);
>+
>   void ClearUpTo(uint16_t seq_num);
>   void UpdateRtt(int64_t rtt_ms);
>   void Clear();
>@@ -50,10 +52,13 @@ class NackModule : public Module {
>   // we have tried to nack this packet.
>   struct NackInfo {
>     NackInfo();
>-    NackInfo(uint16_t seq_num, uint16_t send_at_seq_num);
>+    NackInfo(uint16_t seq_num,
>+             uint16_t send_at_seq_num,
>+             int64_t created_at_time);
> 
>     uint16_t seq_num;
>     uint16_t send_at_seq_num;
>+    int64_t created_at_time;
>     int64_t sent_at_time;
>     int retries;
>   };
>@@ -87,6 +92,8 @@ class NackModule : public Module {
>       RTC_GUARDED_BY(crit_);
>   std::set<uint16_t, DescendingSeqNumComp<uint16_t>> keyframe_list_
>       RTC_GUARDED_BY(crit_);
>+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> recovered_list_
>+      RTC_GUARDED_BY(crit_);
>   video_coding::Histogram reordering_histogram_ RTC_GUARDED_BY(crit_);
>   bool initialized_ RTC_GUARDED_BY(crit_);
>   int64_t rtt_ms_ RTC_GUARDED_BY(crit_);
>@@ -94,6 +101,9 @@ class NackModule : public Module {
> 
>   // Only touched on the process thread.
>   int64_t next_process_time_ms_;
>+
>+  // Adds a delay before send nack on packet received.
>+  const int64_t send_nack_delay_ms_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module_unittest.cc
>index 8a5f46bf7fba56872f259363287c03d9d3871608..8efb8ac1549f9d4b4cb798e45a9ba366597ed113 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module_unittest.cc
>@@ -14,6 +14,7 @@
> #include "modules/video_coding/include/video_coding_defines.h"
> #include "modules/video_coding/nack_module.h"
> #include "system_wrappers/include/clock.h"
>+#include "test/field_trial.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -40,38 +41,38 @@ class TestNackModule : public ::testing::Test,
> };
> 
> TEST_F(TestNackModule, NackOnePacket) {
>-  nack_module_.OnReceivedPacket(1, false);
>-  nack_module_.OnReceivedPacket(3, false);
>+  nack_module_.OnReceivedPacket(1, false, false);
>+  nack_module_.OnReceivedPacket(3, false, false);
>   EXPECT_EQ(1u, sent_nacks_.size());
>   EXPECT_EQ(2, sent_nacks_[0]);
> }
> 
> TEST_F(TestNackModule, WrappingSeqNum) {
>-  nack_module_.OnReceivedPacket(0xfffe, false);
>-  nack_module_.OnReceivedPacket(1, false);
>+  nack_module_.OnReceivedPacket(0xfffe, false, false);
>+  nack_module_.OnReceivedPacket(1, false, false);
>   EXPECT_EQ(2u, sent_nacks_.size());
>   EXPECT_EQ(0xffff, sent_nacks_[0]);
>   EXPECT_EQ(0, sent_nacks_[1]);
> }
> 
> TEST_F(TestNackModule, WrappingSeqNumClearToKeyframe) {
>-  nack_module_.OnReceivedPacket(0xfffe, false);
>-  nack_module_.OnReceivedPacket(1, false);
>+  nack_module_.OnReceivedPacket(0xfffe, false, false);
>+  nack_module_.OnReceivedPacket(1, false, false);
>   EXPECT_EQ(2u, sent_nacks_.size());
>   EXPECT_EQ(0xffff, sent_nacks_[0]);
>   EXPECT_EQ(0, sent_nacks_[1]);
> 
>   sent_nacks_.clear();
>-  nack_module_.OnReceivedPacket(2, true);
>+  nack_module_.OnReceivedPacket(2, true, false);
>   EXPECT_EQ(0u, sent_nacks_.size());
> 
>-  nack_module_.OnReceivedPacket(501, true);
>+  nack_module_.OnReceivedPacket(501, true, false);
>   EXPECT_EQ(498u, sent_nacks_.size());
>   for (int seq_num = 3; seq_num < 501; ++seq_num)
>     EXPECT_EQ(seq_num, sent_nacks_[seq_num - 3]);
> 
>   sent_nacks_.clear();
>-  nack_module_.OnReceivedPacket(1001, false);
>+  nack_module_.OnReceivedPacket(1001, false, false);
>   EXPECT_EQ(499u, sent_nacks_.size());
>   for (int seq_num = 502; seq_num < 1001; ++seq_num)
>     EXPECT_EQ(seq_num, sent_nacks_[seq_num - 502]);
>@@ -91,7 +92,7 @@ TEST_F(TestNackModule, WrappingSeqNumClearToKeyframe) {
>   // It will then clear all nacks up to the next keyframe (seq num 2),
>   // thus removing 0xffff and 0 from the nack list.
>   sent_nacks_.clear();
>-  nack_module_.OnReceivedPacket(1004, false);
>+  nack_module_.OnReceivedPacket(1004, false, false);
>   EXPECT_EQ(2u, sent_nacks_.size());
>   EXPECT_EQ(1002, sent_nacks_[0]);
>   EXPECT_EQ(1003, sent_nacks_[1]);
>@@ -107,7 +108,7 @@ TEST_F(TestNackModule, WrappingSeqNumClearToKeyframe) {
> 
>   // Adding packet 1007 will cause the nack module to overflow again, thus
>   // clearing everything up to 501 which is the next keyframe.
>-  nack_module_.OnReceivedPacket(1007, false);
>+  nack_module_.OnReceivedPacket(1007, false, false);
>   sent_nacks_.clear();
>   clock_->AdvanceTimeMilliseconds(100);
>   nack_module_.Process();
>@@ -146,8 +147,8 @@ TEST_F(TestNackModule, DontBurstOnTimeSkip) {
> }
> 
> TEST_F(TestNackModule, ResendNack) {
>-  nack_module_.OnReceivedPacket(1, false);
>-  nack_module_.OnReceivedPacket(3, false);
>+  nack_module_.OnReceivedPacket(1, false, false);
>+  nack_module_.OnReceivedPacket(3, false, false);
>   EXPECT_EQ(1u, sent_nacks_.size());
>   EXPECT_EQ(2, sent_nacks_[0]);
> 
>@@ -169,15 +170,15 @@ TEST_F(TestNackModule, ResendNack) {
>   nack_module_.Process();
>   EXPECT_EQ(4u, sent_nacks_.size());
> 
>-  nack_module_.OnReceivedPacket(2, false);
>+  nack_module_.OnReceivedPacket(2, false, false);
>   clock_->AdvanceTimeMilliseconds(50);
>   nack_module_.Process();
>   EXPECT_EQ(4u, sent_nacks_.size());
> }
> 
> TEST_F(TestNackModule, ResendPacketMaxRetries) {
>-  nack_module_.OnReceivedPacket(1, false);
>-  nack_module_.OnReceivedPacket(3, false);
>+  nack_module_.OnReceivedPacket(1, false, false);
>+  nack_module_.OnReceivedPacket(3, false, false);
>   EXPECT_EQ(1u, sent_nacks_.size());
>   EXPECT_EQ(2, sent_nacks_[0]);
> 
>@@ -193,35 +194,35 @@ TEST_F(TestNackModule, ResendPacketMaxRetries) {
> }
> 
> TEST_F(TestNackModule, TooLargeNackList) {
>-  nack_module_.OnReceivedPacket(0, false);
>-  nack_module_.OnReceivedPacket(1001, false);
>+  nack_module_.OnReceivedPacket(0, false, false);
>+  nack_module_.OnReceivedPacket(1001, false, false);
>   EXPECT_EQ(1000u, sent_nacks_.size());
>   EXPECT_EQ(0, keyframes_requested_);
>-  nack_module_.OnReceivedPacket(1003, false);
>+  nack_module_.OnReceivedPacket(1003, false, false);
>   EXPECT_EQ(1000u, sent_nacks_.size());
>   EXPECT_EQ(1, keyframes_requested_);
>-  nack_module_.OnReceivedPacket(1004, false);
>+  nack_module_.OnReceivedPacket(1004, false, false);
>   EXPECT_EQ(1000u, sent_nacks_.size());
>   EXPECT_EQ(1, keyframes_requested_);
> }
> 
> TEST_F(TestNackModule, TooLargeNackListWithKeyFrame) {
>-  nack_module_.OnReceivedPacket(0, false);
>-  nack_module_.OnReceivedPacket(1, true);
>-  nack_module_.OnReceivedPacket(1001, false);
>+  nack_module_.OnReceivedPacket(0, false, false);
>+  nack_module_.OnReceivedPacket(1, true, false);
>+  nack_module_.OnReceivedPacket(1001, false, false);
>   EXPECT_EQ(999u, sent_nacks_.size());
>   EXPECT_EQ(0, keyframes_requested_);
>-  nack_module_.OnReceivedPacket(1003, false);
>+  nack_module_.OnReceivedPacket(1003, false, false);
>   EXPECT_EQ(1000u, sent_nacks_.size());
>   EXPECT_EQ(0, keyframes_requested_);
>-  nack_module_.OnReceivedPacket(1005, false);
>+  nack_module_.OnReceivedPacket(1005, false, false);
>   EXPECT_EQ(1000u, sent_nacks_.size());
>   EXPECT_EQ(1, keyframes_requested_);
> }
> 
> TEST_F(TestNackModule, ClearUpTo) {
>-  nack_module_.OnReceivedPacket(0, false);
>-  nack_module_.OnReceivedPacket(100, false);
>+  nack_module_.OnReceivedPacket(0, false, false);
>+  nack_module_.OnReceivedPacket(100, false, false);
>   EXPECT_EQ(99u, sent_nacks_.size());
> 
>   sent_nacks_.clear();
>@@ -233,8 +234,8 @@ TEST_F(TestNackModule, ClearUpTo) {
> }
> 
> TEST_F(TestNackModule, ClearUpToWrap) {
>-  nack_module_.OnReceivedPacket(0xfff0, false);
>-  nack_module_.OnReceivedPacket(0xf, false);
>+  nack_module_.OnReceivedPacket(0xfff0, false, false);
>+  nack_module_.OnReceivedPacket(0xf, false, false);
>   EXPECT_EQ(30u, sent_nacks_.size());
> 
>   sent_nacks_.clear();
>@@ -246,20 +247,20 @@ TEST_F(TestNackModule, ClearUpToWrap) {
> }
> 
> TEST_F(TestNackModule, PacketNackCount) {
>-  EXPECT_EQ(0, nack_module_.OnReceivedPacket(0, false));
>-  EXPECT_EQ(0, nack_module_.OnReceivedPacket(2, false));
>-  EXPECT_EQ(1, nack_module_.OnReceivedPacket(1, false));
>+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(0, false, false));
>+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(2, false, false));
>+  EXPECT_EQ(1, nack_module_.OnReceivedPacket(1, false, false));
> 
>   sent_nacks_.clear();
>   nack_module_.UpdateRtt(100);
>-  EXPECT_EQ(0, nack_module_.OnReceivedPacket(5, false));
>+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(5, false, false));
>   clock_->AdvanceTimeMilliseconds(100);
>   nack_module_.Process();
>   clock_->AdvanceTimeMilliseconds(100);
>   nack_module_.Process();
>-  EXPECT_EQ(3, nack_module_.OnReceivedPacket(3, false));
>-  EXPECT_EQ(3, nack_module_.OnReceivedPacket(4, false));
>-  EXPECT_EQ(0, nack_module_.OnReceivedPacket(4, false));
>+  EXPECT_EQ(3, nack_module_.OnReceivedPacket(3, false, false));
>+  EXPECT_EQ(3, nack_module_.OnReceivedPacket(4, false, false));
>+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(4, false, false));
> }
> 
> TEST_F(TestNackModule, NackListFullAndNoOverlapWithKeyframes) {
>@@ -267,14 +268,63 @@ TEST_F(TestNackModule, NackListFullAndNoOverlapWithKeyframes) {
>   const unsigned int kFirstGap = kMaxNackPackets - 20;
>   const unsigned int kSecondGap = 200;
>   uint16_t seq_num = 0;
>-  nack_module_.OnReceivedPacket(seq_num++, true);
>+  nack_module_.OnReceivedPacket(seq_num++, true, false);
>   seq_num += kFirstGap;
>-  nack_module_.OnReceivedPacket(seq_num++, true);
>+  nack_module_.OnReceivedPacket(seq_num++, true, false);
>   EXPECT_EQ(kFirstGap, sent_nacks_.size());
>   sent_nacks_.clear();
>   seq_num += kSecondGap;
>-  nack_module_.OnReceivedPacket(seq_num, true);
>+  nack_module_.OnReceivedPacket(seq_num, true, false);
>   EXPECT_EQ(kSecondGap, sent_nacks_.size());
> }
> 
>+TEST_F(TestNackModule, HandleFecRecoveredPacket) {
>+  nack_module_.OnReceivedPacket(1, false, false);
>+  nack_module_.OnReceivedPacket(4, false, true);
>+  EXPECT_EQ(0u, sent_nacks_.size());
>+  nack_module_.OnReceivedPacket(5, false, false);
>+  EXPECT_EQ(2u, sent_nacks_.size());
>+}
>+
>+TEST_F(TestNackModule, SendNackWithoutDelay) {
>+  nack_module_.OnReceivedPacket(0, false, false);
>+  nack_module_.OnReceivedPacket(100, false, false);
>+  EXPECT_EQ(99u, sent_nacks_.size());
>+}
>+
>+class TestNackModuleWithFieldTrial : public ::testing::Test,
>+                                     public NackSender,
>+                                     public KeyFrameRequestSender {
>+ protected:
>+  TestNackModuleWithFieldTrial()
>+      : nack_delay_field_trial_("WebRTC-SendNackDelayMs/10/"),
>+        clock_(new SimulatedClock(0)),
>+        nack_module_(clock_.get(), this, this),
>+        keyframes_requested_(0) {}
>+
>+  void SendNack(const std::vector<uint16_t>& sequence_numbers) override {
>+    sent_nacks_.insert(sent_nacks_.end(), sequence_numbers.begin(),
>+                       sequence_numbers.end());
>+  }
>+
>+  void RequestKeyFrame() override { ++keyframes_requested_; }
>+
>+  test::ScopedFieldTrials nack_delay_field_trial_;
>+  std::unique_ptr<SimulatedClock> clock_;
>+  NackModule nack_module_;
>+  std::vector<uint16_t> sent_nacks_;
>+  int keyframes_requested_;
>+};
>+
>+TEST_F(TestNackModuleWithFieldTrial, SendNackWithDelay) {
>+  nack_module_.OnReceivedPacket(0, false, false);
>+  nack_module_.OnReceivedPacket(100, false, false);
>+  EXPECT_EQ(0u, sent_nacks_.size());
>+  clock_->AdvanceTimeMilliseconds(10);
>+  nack_module_.OnReceivedPacket(106, false, false);
>+  EXPECT_EQ(99u, sent_nacks_.size());
>+  clock_->AdvanceTimeMilliseconds(10);
>+  nack_module_.OnReceivedPacket(109, false, false);
>+  EXPECT_EQ(104u, sent_nacks_.size());
>+}
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.cc
>index 7f22b73a5a621a289e860a0bd4f9d926354360a5..5bf27257b07a31a6b9d0afc7c5cec422027ada6a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.cc
>@@ -28,32 +28,25 @@ namespace webrtc {
> 
> enum { kMaxReceiverDelayMs = 10000 };
> 
>-VCMReceiver::VCMReceiver(VCMTiming* timing,
>-                         Clock* clock,
>-                         EventFactory* event_factory)
>+VCMReceiver::VCMReceiver(VCMTiming* timing, Clock* clock)
>     : VCMReceiver::VCMReceiver(timing,
>                                clock,
>-                               event_factory,
>+                               absl::WrapUnique(EventWrapper::Create()),
>+                               absl::WrapUnique(EventWrapper::Create()),
>                                nullptr,  // NackSender
>                                nullptr)  // KeyframeRequestSender
> {}
> 
> VCMReceiver::VCMReceiver(VCMTiming* timing,
>                          Clock* clock,
>-                         EventFactory* event_factory,
>                          NackSender* nack_sender,
>                          KeyFrameRequestSender* keyframe_request_sender)
>-    : VCMReceiver(
>-          timing,
>-          clock,
>-          std::unique_ptr<EventWrapper>(event_factory
>-                                            ? event_factory->CreateEvent()
>-                                            : EventWrapper::Create()),
>-          std::unique_ptr<EventWrapper>(event_factory
>-                                            ? event_factory->CreateEvent()
>-                                            : EventWrapper::Create()),
>-          nack_sender,
>-          keyframe_request_sender) {}
>+    : VCMReceiver(timing,
>+                  clock,
>+                  absl::WrapUnique(EventWrapper::Create()),
>+                  absl::WrapUnique(EventWrapper::Create()),
>+                  nack_sender,
>+                  keyframe_request_sender) {}
> 
> VCMReceiver::VCMReceiver(VCMTiming* timing,
>                          Clock* clock,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.h
>index 4f6590d274822f392dc2adeaaca1063fd36b9c41..503cae7c8deedd67e9c4ddb0c21a59660ade26d1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.h
>@@ -30,17 +30,16 @@ class VCMReceiver {
>  public:
>   // Constructor for current interface, will be removed when the
>   // new jitter buffer is in place.
>-  VCMReceiver(VCMTiming* timing, Clock* clock, EventFactory* event_factory);
>+  VCMReceiver(VCMTiming* timing, Clock* clock);
> 
>   // Create method for the new jitter buffer.
>   VCMReceiver(VCMTiming* timing,
>               Clock* clock,
>-              EventFactory* event_factory,
>               NackSender* nack_sender,
>               KeyFrameRequestSender* keyframe_request_sender);
> 
>-  // Using this constructor, you can specify a different event factory for the
>-  // jitter buffer. Useful for unit tests when you want to simulate incoming
>+  // Using this constructor, you can specify a different event implemetation for
>+  // the jitter buffer. Useful for unit tests when you want to simulate incoming
>   // packets, in which case the jitter buffer's wait event is different from
>   // that of VCMReceiver itself.
>   //
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver_unittest.cc
>index 1ff866298e2ccdf4616631fb911dd8ac8049fefd..ba35f693582d481c3cbb1204cd3fd407913a4eba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver_unittest.cc
>@@ -18,7 +18,6 @@
> #include "modules/video_coding/packet.h"
> #include "modules/video_coding/receiver.h"
> #include "modules/video_coding/test/stream_generator.h"
>-#include "modules/video_coding/test/test_util.h"
> #include "modules/video_coding/timing.h"
> #include "rtc_base/checks.h"
> #include "system_wrappers/include/clock.h"
>@@ -31,7 +30,7 @@ class TestVCMReceiver : public ::testing::Test {
>   TestVCMReceiver()
>       : clock_(new SimulatedClock(0)),
>         timing_(clock_.get()),
>-        receiver_(&timing_, clock_.get(), &event_factory_) {
>+        receiver_(&timing_, clock_.get()) {
>     stream_generator_.reset(
>         new StreamGenerator(0, clock_->TimeInMilliseconds()));
>   }
>@@ -81,7 +80,6 @@ class TestVCMReceiver : public ::testing::Test {
> 
>   std::unique_ptr<SimulatedClock> clock_;
>   VCMTiming timing_;
>-  NullEventFactory event_factory_;
>   VCMReceiver receiver_;
>   std::unique_ptr<StreamGenerator> stream_generator_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/test/test_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/test/test_util.h
>deleted file mode 100644
>index a38fc58b1284bc781b1c6f5cd642f7f66dc3ae48..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/test/test_util.h
>+++ /dev/null
>@@ -1,34 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
>-#define MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
>-
>-#include "system_wrappers/include/event_wrapper.h"
>-
>-class NullEventFactory : public webrtc::EventFactory {
>- public:
>-  virtual ~NullEventFactory() {}
>-
>-  webrtc::EventWrapper* CreateEvent() override { return new NullEvent; }
>-
>- private:
>-  // Private class to avoid more dependencies on it in tests.
>-  class NullEvent : public webrtc::EventWrapper {
>-   public:
>-    ~NullEvent() override {}
>-    bool Set() override { return true; }
>-    webrtc::EventTypeWrapper Wait(unsigned long max_time) override {  // NOLINT
>-      return webrtc::kEventTimeout;
>-    }
>-  };
>-};
>-
>-#endif  // MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/ivf_file_writer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/ivf_file_writer.h
>index 7d917ed35ee9b10f987ecfc94a1cea8803aafd32..4775beddac85a51b69298c5b13552e74a5395de3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/ivf_file_writer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/ivf_file_writer.h
>@@ -14,7 +14,7 @@
> #include <memory>
> #include <string>
> 
>-#include "common_video/include/video_frame.h"
>+#include "api/video/encoded_image.h"
> #include "modules/include/module_common_types.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/file.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.cc
>deleted file mode 100644
>index eb23e3d92f3e6063eacdedfd8802a9ffa83d1c72..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.cc
>+++ /dev/null
>@@ -1,47 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/utility/moving_average.h"
>-
>-#include <algorithm>
>-
>-namespace webrtc {
>-
>-MovingAverage::MovingAverage(size_t s) : sum_history_(s + 1, 0) {}
>-MovingAverage::~MovingAverage() = default;
>-
>-void MovingAverage::AddSample(int sample) {
>-  count_++;
>-  sum_ += sample;
>-  sum_history_[count_ % sum_history_.size()] = sum_;
>-}
>-
>-absl::optional<int> MovingAverage::GetAverage() const {
>-  return GetAverage(size());
>-}
>-
>-absl::optional<int> MovingAverage::GetAverage(size_t num_samples) const {
>-  if (num_samples > size() || num_samples == 0)
>-    return absl::nullopt;
>-  int sum = sum_ - sum_history_[(count_ - num_samples) % sum_history_.size()];
>-  return sum / static_cast<int>(num_samples);
>-}
>-
>-void MovingAverage::Reset() {
>-  count_ = 0;
>-  sum_ = 0;
>-  std::fill(sum_history_.begin(), sum_history_.end(), 0);
>-}
>-
>-size_t MovingAverage::size() const {
>-  return std::min(count_, sum_history_.size() - 1);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.h
>deleted file mode 100644
>index 77ce633b18fcc16f6f001d717d01d16dd5adb796..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.h
>+++ /dev/null
>@@ -1,36 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_UTILITY_MOVING_AVERAGE_H_
>-#define MODULES_VIDEO_CODING_UTILITY_MOVING_AVERAGE_H_
>-
>-#include <vector>
>-
>-#include "absl/types/optional.h"
>-
>-namespace webrtc {
>-class MovingAverage {
>- public:
>-  explicit MovingAverage(size_t s);
>-  ~MovingAverage();
>-  void AddSample(int sample);
>-  absl::optional<int> GetAverage() const;
>-  absl::optional<int> GetAverage(size_t num_samples) const;
>-  void Reset();
>-  size_t size() const;
>-
>- private:
>-  size_t count_ = 0;
>-  int sum_ = 0;
>-  std::vector<int> sum_history_;
>-};
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_UTILITY_MOVING_AVERAGE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average_unittest.cc
>deleted file mode 100644
>index 72fc775a82732dcd175be761eb620d85dd60d6a9..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average_unittest.cc
>+++ /dev/null
>@@ -1,62 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/utility/moving_average.h"
>-
>-#include "test/gtest.h"
>-
>-TEST(MovingAverageTest, EmptyAverage) {
>-  webrtc::MovingAverage moving_average(1);
>-  EXPECT_EQ(0u, moving_average.size());
>-  EXPECT_FALSE(moving_average.GetAverage(0));
>-}
>-
>-// Test single value.
>-TEST(MovingAverageTest, OneElement) {
>-  webrtc::MovingAverage moving_average(1);
>-  moving_average.AddSample(3);
>-  EXPECT_EQ(1u, moving_average.size());
>-  EXPECT_EQ(3, *moving_average.GetAverage());
>-  EXPECT_EQ(3, *moving_average.GetAverage(1));
>-  EXPECT_FALSE(moving_average.GetAverage(2));
>-}
>-
>-TEST(MovingAverageTest, GetAverage) {
>-  webrtc::MovingAverage moving_average(1024);
>-  moving_average.AddSample(1);
>-  moving_average.AddSample(1);
>-  moving_average.AddSample(3);
>-  moving_average.AddSample(3);
>-  EXPECT_EQ(*moving_average.GetAverage(4), 2);
>-  EXPECT_EQ(*moving_average.GetAverage(2), 3);
>-  EXPECT_FALSE(moving_average.GetAverage(0));
>-}
>-
>-TEST(MovingAverageTest, Reset) {
>-  webrtc::MovingAverage moving_average(5);
>-  moving_average.AddSample(1);
>-  EXPECT_EQ(1, *moving_average.GetAverage(1));
>-  moving_average.Reset();
>-  EXPECT_FALSE(moving_average.GetAverage(1));
>-  EXPECT_FALSE(moving_average.GetAverage(6));
>-}
>-
>-TEST(MovingAverageTest, ManySamples) {
>-  webrtc::MovingAverage moving_average(10);
>-  for (int i = 1; i < 11; i++) {
>-    moving_average.AddSample(i);
>-  }
>-  EXPECT_EQ(*moving_average.GetAverage(), 5);
>-  moving_average.Reset();
>-  for (int i = 1; i < 2001; i++) {
>-    moving_average.AddSample(i);
>-  }
>-  EXPECT_EQ(*moving_average.GetAverage(), 1995);
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.cc
>index ea19e418c9d4e921fce0c97d6d6a93431f6ce725..32ae166a9c677a21cc3946f2a26572f52d5db35b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.cc
>@@ -173,8 +173,8 @@ void QualityScaler::CheckQp() {
>   // If we have not observed at least this many frames we can't make a good
>   // scaling decision.
>   const size_t frames = config_.use_all_drop_reasons
>-                            ? framedrop_percent_all_.size()
>-                            : framedrop_percent_media_opt_.size();
>+                            ? framedrop_percent_all_.Size()
>+                            : framedrop_percent_media_opt_.Size();
>   if (frames < kMinFramesNeededToScale) {
>     observed_enough_frames_ = false;
>     return;
>@@ -183,8 +183,9 @@ void QualityScaler::CheckQp() {
> 
>   // Check if we should scale down due to high frame drop.
>   const absl::optional<int> drop_rate =
>-      config_.use_all_drop_reasons ? framedrop_percent_all_.GetAverage()
>-                                   : framedrop_percent_media_opt_.GetAverage();
>+      config_.use_all_drop_reasons
>+          ? framedrop_percent_all_.GetAverageRoundedDown()
>+          : framedrop_percent_media_opt_.GetAverageRoundedDown();
>   if (drop_rate && *drop_rate >= kFramedropPercentThreshold) {
>     RTC_LOG(LS_INFO) << "Reporting high QP, framedrop percent " << *drop_rate;
>     ReportQpHigh();
>@@ -192,11 +193,12 @@ void QualityScaler::CheckQp() {
>   }
> 
>   // Check if we should scale up or down based on QP.
>-  const absl::optional<int> avg_qp_high = qp_smoother_high_
>-                                              ? qp_smoother_high_->GetAvg()
>-                                              : average_qp_.GetAverage();
>+  const absl::optional<int> avg_qp_high =
>+      qp_smoother_high_ ? qp_smoother_high_->GetAvg()
>+                        : average_qp_.GetAverageRoundedDown();
>   const absl::optional<int> avg_qp_low =
>-      qp_smoother_low_ ? qp_smoother_low_->GetAvg() : average_qp_.GetAverage();
>+      qp_smoother_low_ ? qp_smoother_low_->GetAvg()
>+                       : average_qp_.GetAverageRoundedDown();
>   if (avg_qp_high && avg_qp_low) {
>     RTC_LOG(LS_INFO) << "Checking average QP " << *avg_qp_high << " ("
>                      << *avg_qp_low << ").";
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.h
>index 3091ebf36c42e9d96728e007e716642b7a77239e..272c02e53a80381513493d12739546932fb714e0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.h
>@@ -17,8 +17,8 @@
> #include "absl/types/optional.h"
> #include "api/video_codecs/video_encoder.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "modules/video_coding/utility/moving_average.h"
> #include "rtc_base/experiments/quality_scaling_experiment.h"
>+#include "rtc_base/numerics/moving_average.h"
> #include "rtc_base/sequenced_task_checker.h"
> 
> namespace webrtc {
>@@ -79,9 +79,10 @@ class QualityScaler {
>   const VideoEncoder::QpThresholds thresholds_;
>   const int64_t sampling_period_ms_;
>   bool fast_rampup_ RTC_GUARDED_BY(&task_checker_);
>-  MovingAverage average_qp_ RTC_GUARDED_BY(&task_checker_);
>-  MovingAverage framedrop_percent_media_opt_ RTC_GUARDED_BY(&task_checker_);
>-  MovingAverage framedrop_percent_all_ RTC_GUARDED_BY(&task_checker_);
>+  rtc::MovingAverage average_qp_ RTC_GUARDED_BY(&task_checker_);
>+  rtc::MovingAverage framedrop_percent_media_opt_
>+      RTC_GUARDED_BY(&task_checker_);
>+  rtc::MovingAverage framedrop_percent_all_ RTC_GUARDED_BY(&task_checker_);
> 
>   // Used by QualityScalingExperiment.
>   const bool experiment_enabled_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
>index b17062df5c158ca5afdbba22a2a79dcee366a17a..ed7937992979b30b5bfb68466f24027f2300cc8f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
>@@ -28,19 +28,18 @@ static const int kMinFramesNeededToScale = 60;  // From quality_scaler.cc.
> static const size_t kDefaultTimeoutMs = 150;
> }  // namespace
> 
>-#define DO_SYNC(q, block)           \
>-  do {                              \
>-    rtc::Event event(false, false); \
>-    q->PostTask([this, &event] {    \
>-      block;                        \
>-      event.Set();                  \
>-    });                             \
>-    RTC_CHECK(event.Wait(1000));    \
>+#define DO_SYNC(q, block)        \
>+  do {                           \
>+    rtc::Event event;            \
>+    q->PostTask([this, &event] { \
>+      block;                     \
>+      event.Set();               \
>+    });                          \
>+    RTC_CHECK(event.Wait(1000)); \
>   } while (0)
> 
> class MockAdaptationObserver : public AdaptationObserverInterface {
>  public:
>-  MockAdaptationObserver() : event(false, false) {}
>   virtual ~MockAdaptationObserver() {}
> 
>   void AdaptUp(AdaptReason r) override {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc
>index 7f54932a813ddb5141c8d3b14f8d136276ad30e6..bbe3f19fb7320d9e641e5286fe35817249f9f5a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc
>@@ -15,8 +15,7 @@
> #include <utility>
> #include <vector>
> 
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
>-
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "test/field_trial.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>@@ -34,9 +33,9 @@ constexpr uint32_t kLegacyScreenshareMaxBitrateKbps = 1000;
> constexpr uint32_t kSimulcastScreenshareMinBitrateKbps = 600;
> constexpr uint32_t kSimulcastScreenshareMaxBitrateKbps = 1250;
> 
>-class MockTemporalLayers : public TemporalLayers {
>+class MockTemporalLayers : public Vp8TemporalLayers {
>  public:
>-  MOCK_METHOD1(UpdateLayerConfig, TemporalLayers::FrameConfig(uint32_t));
>+  MOCK_METHOD1(UpdateLayerConfig, Vp8TemporalLayers::FrameConfig(uint32_t));
>   MOCK_METHOD2(OnRatesUpdated, void(const std::vector<uint32_t>&, int));
>   MOCK_METHOD1(UpdateConfiguration, bool(Vp8EncoderConfig*));
>   MOCK_METHOD5(OnEncodeDone,
>@@ -44,7 +43,7 @@ class MockTemporalLayers : public TemporalLayers {
>   MOCK_METHOD3(FrameEncoded, void(uint32_t, size_t, int));
>   MOCK_CONST_METHOD0(Tl0PicIdx, uint8_t());
>   MOCK_CONST_METHOD1(GetTemporalLayerId,
>-                     int(const TemporalLayers::FrameConfig&));
>+                     int(const Vp8TemporalLayers::FrameConfig&));
> };
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.cc
>index 4af526c66319cd772dce81c5a0c99bf0753a440c..992773ecc609eefd898610786f2522d06dbc2f4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.cc
>@@ -15,9 +15,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/video/encoded_image.h"
> #include "api/video_codecs/sdp_video_format.h"
>-#include "common_video/include/video_frame.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/include/video_coding_defines.h"
> #include "rtc_base/checks.h"
> #include "test/gtest.h"
>@@ -819,5 +820,67 @@ void SimulcastTestFixtureImpl::TestStrideEncodeDecode() {
>   EXPECT_EQ(2, decoder_callback.DecodedFrames());
> }
> 
>+void SimulcastTestFixtureImpl::TestDecodeWidthHeightSet() {
>+  MockEncodedImageCallback encoder_callback;
>+  MockDecodedImageCallback decoder_callback;
>+
>+  EncodedImage encoded_frame[3];
>+  SetRates(kMaxBitrates[2], 30);  // To get all three streams.
>+  encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
>+  decoder_->RegisterDecodeCompleteCallback(&decoder_callback);
>+
>+  EXPECT_CALL(encoder_callback, OnEncodedImage(_, _, _))
>+      .Times(3)
>+      .WillRepeatedly(
>+          testing::Invoke([&](const EncodedImage& encoded_image,
>+                              const CodecSpecificInfo* codec_specific_info,
>+                              const RTPFragmentationHeader* fragmentation) {
>+            EXPECT_EQ(encoded_image._frameType, kVideoFrameKey);
>+
>+            size_t index = encoded_image.SpatialIndex().value_or(0);
>+            encoded_frame[index]._buffer = new uint8_t[encoded_image._size];
>+            encoded_frame[index]._size = encoded_image._size;
>+            encoded_frame[index]._length = encoded_image._length;
>+            encoded_frame[index]._frameType = encoded_image._frameType;
>+            encoded_frame[index]._completeFrame = encoded_image._completeFrame;
>+            memcpy(encoded_frame[index]._buffer, encoded_image._buffer,
>+                   encoded_image._length);
>+            return EncodedImageCallback::Result(
>+                EncodedImageCallback::Result::OK, 0);
>+          }));
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+
>+  EXPECT_CALL(decoder_callback, Decoded(_, _, _))
>+      .WillOnce(testing::Invoke([](VideoFrame& decodedImage,
>+                                   absl::optional<int32_t> decode_time_ms,
>+                                   absl::optional<uint8_t> qp) {
>+        EXPECT_EQ(decodedImage.width(), kDefaultWidth / 4);
>+        EXPECT_EQ(decodedImage.height(), kDefaultHeight / 4);
>+      }));
>+  EXPECT_EQ(0, decoder_->Decode(encoded_frame[0], false, NULL, 0));
>+
>+  EXPECT_CALL(decoder_callback, Decoded(_, _, _))
>+      .WillOnce(testing::Invoke([](VideoFrame& decodedImage,
>+                                   absl::optional<int32_t> decode_time_ms,
>+                                   absl::optional<uint8_t> qp) {
>+        EXPECT_EQ(decodedImage.width(), kDefaultWidth / 2);
>+        EXPECT_EQ(decodedImage.height(), kDefaultHeight / 2);
>+      }));
>+  EXPECT_EQ(0, decoder_->Decode(encoded_frame[1], false, NULL, 0));
>+
>+  EXPECT_CALL(decoder_callback, Decoded(_, _, _))
>+      .WillOnce(testing::Invoke([](VideoFrame& decodedImage,
>+                                   absl::optional<int32_t> decode_time_ms,
>+                                   absl::optional<uint8_t> qp) {
>+        EXPECT_EQ(decodedImage.width(), kDefaultWidth);
>+        EXPECT_EQ(decodedImage.height(), kDefaultHeight);
>+      }));
>+  EXPECT_EQ(0, decoder_->Decode(encoded_frame[2], false, NULL, 0));
>+
>+  for (int i = 0; i < 3; ++i) {
>+    delete [] encoded_frame[i]._buffer;
>+  }
>+}
>+
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.h
>index 6634a69ad1ca543402f7e3fef09411b7ce26e800..2f834bdfb55fc2439b2888fa29a21e26d9898a8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.h
>@@ -14,6 +14,8 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/test/mock_video_decoder.h"
>+#include "api/test/mock_video_encoder.h"
> #include "api/test/simulcast_test_fixture.h"
> #include "api/video/i420_buffer.h"
> #include "api/video/video_frame.h"
>@@ -21,7 +23,6 @@
> #include "api/video_codecs/video_encoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/video_coding/utility/simulcast_rate_allocator.h"
>-#include "modules/video_coding/include/mock/mock_video_codec_interface.h"
> 
> namespace webrtc {
> namespace test {
>@@ -50,6 +51,7 @@ class SimulcastTestFixtureImpl final : public SimulcastTestFixture {
>   void TestSpatioTemporalLayers333PatternEncoder() override;
>   void TestSpatioTemporalLayers321PatternEncoder() override;
>   void TestStrideEncodeDecode() override;
>+  void TestDecodeWidthHeightSet() override;
> 
>   static void DefaultSettings(VideoCodec* settings,
>                               const int* temporal_layer_profile,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc
>index e7125ce2782313f8fbc2606b96dbe46ab3ec4e5c..86a5ab2a4820994fd7402874e9f34f2bbe2ef381 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc
>@@ -24,15 +24,13 @@
> 
> namespace webrtc {
> 
>-bool VideoCodecInitializer::SetupCodec(
>-    const VideoEncoderConfig& config,
>-    const std::vector<VideoStream>& streams,
>-    VideoCodec* codec,
>-    std::unique_ptr<VideoBitrateAllocator>* bitrate_allocator) {
>+bool VideoCodecInitializer::SetupCodec(const VideoEncoderConfig& config,
>+                                       const std::vector<VideoStream>& streams,
>+                                       VideoCodec* codec) {
>   if (config.codec_type == kVideoCodecMultiplex) {
>     VideoEncoderConfig associated_config = config.Copy();
>     associated_config.codec_type = kVideoCodecVP9;
>-    if (!SetupCodec(associated_config, streams, codec, bitrate_allocator)) {
>+    if (!SetupCodec(associated_config, streams, codec)) {
>       RTC_LOG(LS_ERROR) << "Failed to create stereo encoder configuration.";
>       return false;
>     }
>@@ -41,31 +39,9 @@ bool VideoCodecInitializer::SetupCodec(
>   }
> 
>   *codec = VideoEncoderConfigToVideoCodec(config, streams);
>-  *bitrate_allocator = CreateBitrateAllocator(*codec);
>-
>   return true;
> }
> 
>-std::unique_ptr<VideoBitrateAllocator>
>-VideoCodecInitializer::CreateBitrateAllocator(const VideoCodec& codec) {
>-  std::unique_ptr<VideoBitrateAllocator> rate_allocator;
>-
>-  switch (codec.codecType) {
>-    case kVideoCodecVP8:
>-      RTC_FALLTHROUGH();
>-    case kVideoCodecH264:
>-      rate_allocator.reset(new SimulcastRateAllocator(codec));
>-      break;
>-    case kVideoCodecVP9:
>-      rate_allocator.reset(new SvcRateAllocator(codec));
>-      break;
>-    default:
>-      rate_allocator.reset(new DefaultVideoBitrateAllocator(codec));
>-  }
>-
>-  return rate_allocator;
>-}
>-
> // TODO(sprang): Split this up and separate the codec specific parts.
> VideoCodec VideoCodecInitializer::VideoEncoderConfigToVideoCodec(
>     const VideoEncoderConfig& config,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer_unittest.cc
>index fb601f0d4299ec952c2f4657606bbd618f5bde23..ceff1ebaf7427bf928fe7e6adbe03c530a187ce3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer_unittest.cc
>@@ -9,11 +9,13 @@
>  */
> 
> #include "modules/video_coding/include/video_codec_initializer.h"
>-#include "api/video/video_bitrate_allocator.h"
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
>+#include "api/video_codecs/create_vp8_temporal_layers.h"
> #include "api/video_codecs/video_encoder.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
> #include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/refcountedobject.h"
> #include "test/gtest.h"
> 
>@@ -74,21 +76,22 @@ class VideoCodecInitializerTest : public ::testing::Test {
> 
>   bool InitializeCodec() {
>     codec_out_ = VideoCodec();
>-    bitrate_allocator_out_.reset();
>     temporal_layers_.clear();
>-    if (!VideoCodecInitializer::SetupCodec(config_, streams_, &codec_out_,
>-                                           &bitrate_allocator_out_)) {
>+    if (!VideoCodecInitializer::SetupCodec(config_, streams_, &codec_out_)) {
>       return false;
>     }
>+    bitrate_allocator_ = CreateBuiltinVideoBitrateAllocatorFactory()
>+                             ->CreateVideoBitrateAllocator(codec_out_);
>+    RTC_CHECK(bitrate_allocator_);
>     if (codec_out_.codecType == VideoCodecType::kVideoCodecMultiplex)
>       return true;
> 
>     // Make sure temporal layers instances have been created.
>     if (codec_out_.codecType == VideoCodecType::kVideoCodecVP8) {
>       for (int i = 0; i < codec_out_.numberOfSimulcastStreams; ++i) {
>-        temporal_layers_.emplace_back(TemporalLayers::CreateTemporalLayers(
>-            TemporalLayersType::kFixedPattern,
>-            codec_out_.VP8()->numberOfTemporalLayers));
>+        temporal_layers_.emplace_back(
>+            CreateVp8TemporalLayers(Vp8TemporalLayersType::kFixedPattern,
>+                                    codec_out_.VP8()->numberOfTemporalLayers));
>       }
>     }
>     return true;
>@@ -125,8 +128,8 @@ class VideoCodecInitializerTest : public ::testing::Test {
> 
>   // Output.
>   VideoCodec codec_out_;
>-  std::unique_ptr<VideoBitrateAllocator> bitrate_allocator_out_;
>-  std::vector<std::unique_ptr<TemporalLayers>> temporal_layers_;
>+  std::unique_ptr<VideoBitrateAllocator> bitrate_allocator_;
>+  std::vector<std::unique_ptr<Vp8TemporalLayers>> temporal_layers_;
> };
> 
> TEST_F(VideoCodecInitializerTest, SingleStreamVp8Screenshare) {
>@@ -134,9 +137,8 @@ TEST_F(VideoCodecInitializerTest, SingleStreamVp8Screenshare) {
>   streams_.push_back(DefaultStream());
>   EXPECT_TRUE(InitializeCodec());
> 
>-  VideoBitrateAllocation bitrate_allocation =
>-      bitrate_allocator_out_->GetAllocation(kDefaultTargetBitrateBps,
>-                                            kDefaultFrameRate);
>+  VideoBitrateAllocation bitrate_allocation = bitrate_allocator_->GetAllocation(
>+      kDefaultTargetBitrateBps, kDefaultFrameRate);
>   EXPECT_EQ(1u, codec_out_.numberOfSimulcastStreams);
>   EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
>   EXPECT_EQ(kDefaultTargetBitrateBps, bitrate_allocation.get_sum_bps());
>@@ -149,9 +151,8 @@ TEST_F(VideoCodecInitializerTest, SingleStreamVp8ScreenshareInactive) {
>   streams_.push_back(inactive_stream);
>   EXPECT_TRUE(InitializeCodec());
> 
>-  VideoBitrateAllocation bitrate_allocation =
>-      bitrate_allocator_out_->GetAllocation(kDefaultTargetBitrateBps,
>-                                            kDefaultFrameRate);
>+  VideoBitrateAllocation bitrate_allocation = bitrate_allocator_->GetAllocation(
>+      kDefaultTargetBitrateBps, kDefaultFrameRate);
>   EXPECT_EQ(1u, codec_out_.numberOfSimulcastStreams);
>   EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
>   EXPECT_EQ(0U, bitrate_allocation.get_sum_bps());
>@@ -164,9 +165,8 @@ TEST_F(VideoCodecInitializerTest, TemporalLayeredVp8Screenshare) {
> 
>   EXPECT_EQ(1u, codec_out_.numberOfSimulcastStreams);
>   EXPECT_EQ(2u, codec_out_.VP8()->numberOfTemporalLayers);
>-  VideoBitrateAllocation bitrate_allocation =
>-      bitrate_allocator_out_->GetAllocation(kScreenshareCodecTargetBitrateBps,
>-                                            kScreenshareDefaultFramerate);
>+  VideoBitrateAllocation bitrate_allocation = bitrate_allocator_->GetAllocation(
>+      kScreenshareCodecTargetBitrateBps, kScreenshareDefaultFramerate);
>   EXPECT_EQ(kScreenshareCodecTargetBitrateBps,
>             bitrate_allocation.get_sum_bps());
>   EXPECT_EQ(kScreenshareTl0BitrateBps, bitrate_allocation.GetBitrate(0, 0));
>@@ -184,9 +184,8 @@ TEST_F(VideoCodecInitializerTest, SimulcastVp8Screenshare) {
>   EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
>   const uint32_t max_bitrate_bps =
>       streams_[0].target_bitrate_bps + streams_[1].max_bitrate_bps;
>-  VideoBitrateAllocation bitrate_allocation =
>-      bitrate_allocator_out_->GetAllocation(max_bitrate_bps,
>-                                            kScreenshareDefaultFramerate);
>+  VideoBitrateAllocation bitrate_allocation = bitrate_allocator_->GetAllocation(
>+      max_bitrate_bps, kScreenshareDefaultFramerate);
>   EXPECT_EQ(max_bitrate_bps, bitrate_allocation.get_sum_bps());
>   EXPECT_EQ(static_cast<uint32_t>(streams_[0].target_bitrate_bps),
>             bitrate_allocation.GetSpatialLayerSum(0));
>@@ -209,9 +208,8 @@ TEST_F(VideoCodecInitializerTest, SimulcastVp8ScreenshareInactive) {
>   EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
>   const uint32_t target_bitrate =
>       streams_[0].target_bitrate_bps + streams_[1].target_bitrate_bps;
>-  VideoBitrateAllocation bitrate_allocation =
>-      bitrate_allocator_out_->GetAllocation(target_bitrate,
>-                                            kScreenshareDefaultFramerate);
>+  VideoBitrateAllocation bitrate_allocation = bitrate_allocator_->GetAllocation(
>+      target_bitrate, kScreenshareDefaultFramerate);
>   EXPECT_EQ(static_cast<uint32_t>(streams_[0].max_bitrate_bps),
>             bitrate_allocation.get_sum_bps());
>   EXPECT_EQ(static_cast<uint32_t>(streams_[0].max_bitrate_bps),
>@@ -234,7 +232,7 @@ TEST_F(VideoCodecInitializerTest, HighFpsSimulcastVp8Screenshare) {
>   const uint32_t max_bitrate_bps =
>       streams_[0].target_bitrate_bps + streams_[1].max_bitrate_bps;
>   VideoBitrateAllocation bitrate_allocation =
>-      bitrate_allocator_out_->GetAllocation(max_bitrate_bps, kDefaultFrameRate);
>+      bitrate_allocator_->GetAllocation(max_bitrate_bps, kDefaultFrameRate);
>   EXPECT_EQ(max_bitrate_bps, bitrate_allocation.get_sum_bps());
>   EXPECT_EQ(static_cast<uint32_t>(streams_[0].target_bitrate_bps),
>             bitrate_allocation.GetSpatialLayerSum(0));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.cc
>index a061c18fd70e222ecccb742fa075df31516c2509..58397ad386d3221e03fb72b6dbfe0716578c83b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.cc
>@@ -13,6 +13,7 @@
> #include <algorithm>
> #include <utility>
> 
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "api/video/video_bitrate_allocator.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>@@ -27,10 +28,6 @@
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
>-EventWrapper* EventFactoryImpl::CreateEvent() {
>-  return EventWrapper::Create();
>-}
>-
> namespace vcm {
> 
> int64_t VCMProcessTimer::Period() const {
>@@ -81,17 +78,13 @@ class EncodedImageCallbackWrapper : public EncodedImageCallback {
> class VideoCodingModuleImpl : public VideoCodingModule {
>  public:
>   VideoCodingModuleImpl(Clock* clock,
>-                        EventFactory* event_factory,
>                         NackSender* nack_sender,
>                         KeyFrameRequestSender* keyframe_request_sender)
>       : VideoCodingModule(),
>         sender_(clock, &post_encode_callback_),
>+        rate_allocator_factory_(CreateBuiltinVideoBitrateAllocatorFactory()),
>         timing_(new VCMTiming(clock)),
>-        receiver_(clock,
>-                  event_factory,
>-                  timing_.get(),
>-                  nack_sender,
>-                  keyframe_request_sender) {}
>+        receiver_(clock, timing_.get(), nack_sender, keyframe_request_sender) {}
> 
>   virtual ~VideoCodingModuleImpl() {}
> 
>@@ -114,7 +107,8 @@ class VideoCodingModuleImpl : public VideoCodingModule {
>       // asynchronously keep the instance alive until destruction or until a
>       // new send codec is registered.
>       VideoCodec codec = *sendCodec;
>-      rate_allocator_ = VideoCodecInitializer::CreateBitrateAllocator(codec);
>+      rate_allocator_ =
>+          rate_allocator_factory_->CreateVideoBitrateAllocator(codec);
>       return sender_.RegisterSendCodec(&codec, numberOfCores, maxPayloadSize);
>     }
>     return sender_.RegisterSendCodec(sendCodec, numberOfCores, maxPayloadSize);
>@@ -142,7 +136,7 @@ class VideoCodingModuleImpl : public VideoCodingModule {
> 
>   int32_t AddVideoFrame(const VideoFrame& videoFrame,
>                         const CodecSpecificInfo* codecSpecificInfo) override {
>-    return sender_.AddVideoFrame(videoFrame, codecSpecificInfo);
>+    return sender_.AddVideoFrame(videoFrame, codecSpecificInfo, absl::nullopt);
>   }
> 
>   int32_t IntraFrameRequest(size_t stream_index) override {
>@@ -213,19 +207,18 @@ class VideoCodingModuleImpl : public VideoCodingModule {
>   rtc::ThreadChecker construction_thread_;
>   EncodedImageCallbackWrapper post_encode_callback_;
>   vcm::VideoSender sender_;
>+  const std::unique_ptr<VideoBitrateAllocatorFactory> rate_allocator_factory_;
>   std::unique_ptr<VideoBitrateAllocator> rate_allocator_;
>-  std::unique_ptr<VCMTiming> timing_;
>+  const std::unique_ptr<VCMTiming> timing_;
>   vcm::VideoReceiver receiver_;
> };
> }  // namespace
> 
> // DEPRECATED.  Create method for current interface, will be removed when the
> // new jitter buffer is in place.
>-VideoCodingModule* VideoCodingModule::Create(Clock* clock,
>-                                             EventFactory* event_factory) {
>+VideoCodingModule* VideoCodingModule::Create(Clock* clock) {
>   RTC_DCHECK(clock);
>-  RTC_DCHECK(event_factory);
>-  return new VideoCodingModuleImpl(clock, event_factory, nullptr, nullptr);
>+  return new VideoCodingModuleImpl(clock, nullptr, nullptr);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.h
>index 3c09f96e269f9fc0aa3e4d0c148fc2d22ff30665..e27a2fe76011ee5d2fa64d352bec87a55ce28fba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.h
>@@ -17,7 +17,7 @@
> #include <string>
> #include <vector>
> 
>-#include "common_video/include/frame_callback.h"
>+#include "absl/types/optional.h"
> #include "modules/video_coding/decoder_database.h"
> #include "modules/video_coding/encoder_database.h"
> #include "modules/video_coding/frame_buffer.h"
>@@ -94,7 +94,8 @@ class VideoSender {
>       VideoBitrateAllocationObserver* bitrate_updated_callback);
> 
>   int32_t AddVideoFrame(const VideoFrame& videoFrame,
>-                        const CodecSpecificInfo* codecSpecificInfo);
>+                        const CodecSpecificInfo* codecSpecificInfo,
>+                        absl::optional<VideoEncoder::EncoderInfo> encoder_info);
> 
>   int32_t IntraFrameRequest(size_t stream_index);
>   int32_t EnableFrameDropper(bool enable);
>@@ -113,7 +114,18 @@ class VideoSender {
>   VCMEncodedFrameCallback _encodedFrameCallback RTC_GUARDED_BY(encoder_crit_);
>   EncodedImageCallback* const post_encode_callback_;
>   VCMEncoderDataBase _codecDataBase RTC_GUARDED_BY(encoder_crit_);
>-  bool frame_dropper_enabled_ RTC_GUARDED_BY(encoder_crit_);
>+
>+  // |frame_dropper_requested_| specifies if the user of this class has
>+  // requested frame dropping to be enabled, via EnableFrameDropper().
>+  // Depending on video encoder configuration, this setting may be overridden
>+  // and the frame dropper be force disabled. If so,
>+  // |force_disable_frame_dropper_| will be set to true.
>+  // If frame dropper is requested, and is not force disabled, frame dropping
>+  // might still be disabled if VideoEncoder::GetEncoderInfo() indicates that
>+  // the encoder has a trusted rate controller. This is determined on a
>+  // per-frame basis, as the encoder behavior might dynamically change.
>+  bool frame_dropper_requested_ RTC_GUARDED_BY(encoder_crit_);
>+  bool force_disable_frame_dropper_ RTC_GUARDED_BY(encoder_crit_);
> 
>   // Must be accessed on the construction thread of VideoSender.
>   VideoCodec current_codec_;
>@@ -128,7 +140,6 @@ class VideoSender {
> class VideoReceiver : public Module {
>  public:
>   VideoReceiver(Clock* clock,
>-                EventFactory* event_factory,
>                 VCMTiming* timing,
>                 NackSender* nack_sender = nullptr,
>                 KeyFrameRequestSender* keyframe_request_sender = nullptr);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_packet_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_packet_buffer_unittest.cc
>index 425542c949c0365222e4aca04b5f1c6713b8a68d..13e64eebab41f09d73a8c51a8b439a247f4f8a99 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_packet_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_packet_buffer_unittest.cc
>@@ -120,14 +120,7 @@ TEST_F(TestPacketBuffer, InsertDuplicatePacket) {
>   EXPECT_TRUE(Insert(seq_num, kKeyFrame, kFirst, kLast));
> }
> 
>-#if defined(WEBRTC_ANDROID)
>-// Fails on android after clang update
>-// TODO(crbug.com/887464): Reenable this
>-#define MAYBE_SeqNumWrapOneFrame DISABLED_SeqNumWrapOneFrame
>-#else
>-#define MAYBE_SeqNumWrapOneFrame SeqNumWrapOneFrame
>-#endif
>-TEST_F(TestPacketBuffer, MAYBE_SeqNumWrapOneFrame) {
>+TEST_F(TestPacketBuffer, SeqNumWrapOneFrame) {
>   EXPECT_TRUE(Insert(0xFFFF, kKeyFrame, kFirst, kNotLast));
>   EXPECT_TRUE(Insert(0x0, kKeyFrame, kNotFirst, kLast));
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver.cc
>index d94ba5e69f9ef70c0458dcb2a2e526e8ae3388b1..8b4ee785eef7987c249e04251b32f49063329712 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver.cc
>@@ -26,7 +26,6 @@ namespace webrtc {
> namespace vcm {
> 
> VideoReceiver::VideoReceiver(Clock* clock,
>-                             EventFactory* event_factory,
>                              VCMTiming* timing,
>                              NackSender* nack_sender,
>                              KeyFrameRequestSender* keyframe_request_sender)
>@@ -34,7 +33,6 @@ VideoReceiver::VideoReceiver(Clock* clock,
>       _timing(timing),
>       _receiver(_timing,
>                 clock_,
>-                event_factory,
>                 nack_sender,
>                 keyframe_request_sender),
>       _decodedFrameCallback(_timing, clock_),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver_unittest.cc
>index b7a46f92ea88e7ee2d5bd1514e35d01b14922bf3..f99dac49200a3bc955a8875bec7b6dafdf6da074 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver_unittest.cc
>@@ -11,10 +11,9 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/test/mock_video_decoder.h"
> #include "modules/video_coding/include/mock/mock_vcm_callbacks.h"
>-#include "modules/video_coding/include/mock/mock_video_codec_interface.h"
> #include "modules/video_coding/include/video_coding.h"
>-#include "modules/video_coding/test/test_util.h"
> #include "modules/video_coding/timing.h"
> #include "modules/video_coding/video_coding_impl.h"
> #include "system_wrappers/include/clock.h"
>@@ -37,7 +36,7 @@ class TestVideoReceiver : public ::testing::Test {
> 
>   virtual void SetUp() {
>     timing_.reset(new VCMTiming(&clock_));
>-    receiver_.reset(new VideoReceiver(&clock_, &event_factory_, timing_.get()));
>+    receiver_.reset(new VideoReceiver(&clock_, timing_.get()));
>     receiver_->RegisterExternalDecoder(&decoder_, kUnusedPayloadType);
>     const size_t kMaxNackListSize = 250;
>     const int kMaxPacketAgeToNack = 450;
>@@ -81,7 +80,6 @@ class TestVideoReceiver : public ::testing::Test {
>   }
> 
>   SimulatedClock clock_;
>-  NullEventFactory event_factory_;
>   VideoCodec settings_;
>   NiceMock<MockVideoDecoder> decoder_;
>   NiceMock<MockPacketRequestCallback> packet_request_callback_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender.cc
>index 4c1e24361204dc059b8e37adaffa401b6e5363ef..4bd52530c9f5020a7fbc2786c8cf459a0ab32a77 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender.cc
>@@ -40,7 +40,8 @@ VideoSender::VideoSender(Clock* clock,
>       _encodedFrameCallback(post_encode_callback, &_mediaOpt),
>       post_encode_callback_(post_encode_callback),
>       _codecDataBase(&_encodedFrameCallback),
>-      frame_dropper_enabled_(true),
>+      frame_dropper_requested_(true),
>+      force_disable_frame_dropper_(false),
>       current_codec_(),
>       encoder_params_({VideoBitrateAllocation(), 0, 0, 0}),
>       encoder_has_internal_source_(false),
>@@ -96,15 +97,12 @@ int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec,
>     numLayers = 1;
>   }
> 
>-  // If we have screensharing and we have layers, we disable frame dropper.
>-  const bool disable_frame_dropper =
>+  // Force-disable frame dropper if either:
>+  //  * We have screensharing with layers.
>+  //  * "WebRTC-FrameDropper" field trial is "Disabled".
>+  force_disable_frame_dropper_ =
>       field_trial::IsDisabled(kFrameDropperFieldTrial) ||
>       (numLayers > 1 && sendCodec->mode == VideoCodecMode::kScreensharing);
>-  if (disable_frame_dropper) {
>-    _mediaOpt.EnableFrameDropper(false);
>-  } else if (frame_dropper_enabled_) {
>-    _mediaOpt.EnableFrameDropper(true);
>-  }
> 
>   {
>     rtc::CritScope cs(&params_crit_);
>@@ -247,8 +245,10 @@ void VideoSender::SetEncoderParameters(EncoderParameters params,
> }
> 
> // Add one raw video frame to the encoder, blocking.
>-int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
>-                                   const CodecSpecificInfo* codecSpecificInfo) {
>+int32_t VideoSender::AddVideoFrame(
>+    const VideoFrame& videoFrame,
>+    const CodecSpecificInfo* codecSpecificInfo,
>+    absl::optional<VideoEncoder::EncoderInfo> encoder_info) {
>   EncoderParameters encoder_params;
>   std::vector<FrameType> next_frame_types;
>   bool encoder_has_internal_source = false;
>@@ -262,6 +262,17 @@ int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
>   if (_encoder == nullptr)
>     return VCM_UNINITIALIZED;
>   SetEncoderParameters(encoder_params, encoder_has_internal_source);
>+  if (!encoder_info) {
>+    encoder_info = _encoder->GetEncoderInfo();
>+  }
>+
>+  // Frame dropping is enabled iff frame dropping has been requested, and
>+  // frame dropping is not force-disabled, and rate controller is not trusted.
>+  const bool frame_dropping_enabled =
>+      frame_dropper_requested_ && !force_disable_frame_dropper_ &&
>+      !encoder_info->has_trusted_rate_controller;
>+  _mediaOpt.EnableFrameDropper(frame_dropping_enabled);
>+
>   if (_mediaOpt.DropFrame()) {
>     RTC_LOG(LS_VERBOSE) << "Drop Frame "
>                         << "target bitrate "
>@@ -287,7 +298,7 @@ int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
>   const bool is_buffer_type_supported =
>       buffer_type == VideoFrameBuffer::Type::kI420 ||
>       (buffer_type == VideoFrameBuffer::Type::kNative &&
>-       _encoder->SupportsNativeHandle());
>+       encoder_info->supports_native_handle);
>   if (!is_buffer_type_supported) {
>     // This module only supports software encoding.
>     // TODO(pbos): Offload conversion from the encoder thread.
>@@ -355,7 +366,7 @@ int32_t VideoSender::IntraFrameRequest(size_t stream_index) {
> 
> int32_t VideoSender::EnableFrameDropper(bool enable) {
>   rtc::CritScope lock(&encoder_crit_);
>-  frame_dropper_enabled_ = enable;
>+  frame_dropper_requested_ = enable;
>   _mediaOpt.EnableFrameDropper(enable);
>   return VCM_OK;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender_unittest.cc
>index 3cd9caf65c306798f888f9dc5ca7c701ad32d9b8..c2ff0f6207c129a9dc838bb7e855fb1ff3aaa805 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender_unittest.cc
>@@ -11,11 +11,11 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/test/mock_video_encoder.h"
> #include "api/video/i420_buffer.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
> #include "modules/video_coding/include/mock/mock_vcm_callbacks.h"
>-#include "modules/video_coding/include/mock/mock_video_codec_interface.h"
> #include "modules/video_coding/include/video_coding.h"
> #include "modules/video_coding/utility/default_video_bitrate_allocator.h"
> #include "modules/video_coding/utility/simulcast_rate_allocator.h"
>@@ -186,7 +186,10 @@ class TestVideoSender : public ::testing::Test {
> 
>   void AddFrame() {
>     assert(generator_.get());
>-    sender_->AddVideoFrame(*generator_->NextFrame(), NULL);
>+    sender_->AddVideoFrame(*generator_->NextFrame(), nullptr,
>+                           encoder_ ? absl::optional<VideoEncoder::EncoderInfo>(
>+                                          encoder_->GetEncoderInfo())
>+                                    : absl::nullopt);
>   }
> 
>   SimulatedClock clock_;
>@@ -369,9 +372,6 @@ TEST_F(TestVideoSenderWithMockEncoder,
>   // Expect initial call to SetChannelParameters. Rates are initialized through
>   // InitEncode and expects no additional call before the framerate (or bitrate)
>   // updates.
>-  EXPECT_CALL(encoder_, SetChannelParameters(kLossRate, kRtt))
>-      .Times(1)
>-      .WillOnce(Return(0));
>   sender_->SetChannelParameters(settings_.startBitrate * 1000, kLossRate, kRtt,
>                                 rate_allocator_.get(), nullptr);
>   while (clock_.TimeInMilliseconds() < start_time + kRateStatsWindowMs) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/BUILD.gn
>deleted file mode 100644
>index b65165a037be5fcedfb11ec0f8dbf50ac6285d55..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/BUILD.gn
>+++ /dev/null
>@@ -1,108 +0,0 @@
>-# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-import("../webrtc.gni")
>-if (is_android) {
>-  import("//build/config/android/config.gni")
>-  import("//build/config/android/rules.gni")
>-}
>-
>-rtc_static_library("ortc") {
>-  defines = []
>-  sources = [
>-    "ortcfactory.cc",
>-    "ortcfactory.h",
>-    "ortcrtpreceiveradapter.cc",
>-    "ortcrtpreceiveradapter.h",
>-    "ortcrtpsenderadapter.cc",
>-    "ortcrtpsenderadapter.h",
>-    "rtptransportadapter.cc",
>-    "rtptransportadapter.h",
>-    "rtptransportcontrolleradapter.cc",
>-    "rtptransportcontrolleradapter.h",
>-  ]
>-
>-  # TODO(deadbeef): Create a separate target for the common things ORTC and
>-  # PeerConnection code shares, so that ortc can depend on that instead of
>-  # libjingle_peerconnection.
>-  deps = [
>-    "../api:libjingle_peerconnection_api",
>-    "../api:ortc_api",
>-    "../api/video_codecs:builtin_video_decoder_factory",
>-    "../api/video_codecs:builtin_video_encoder_factory",
>-    "../call:call_interfaces",
>-    "../call:rtp_sender",
>-    "../logging:rtc_event_log_api",
>-    "../logging:rtc_event_log_impl_base",
>-    "../media:rtc_audio_video",
>-    "../media:rtc_media",
>-    "../media:rtc_media_base",
>-    "../modules/audio_processing:audio_processing",
>-    "../p2p:rtc_p2p",
>-    "../pc:libjingle_peerconnection",
>-    "../pc:peerconnection",
>-    "../pc:rtc_pc",
>-    "../pc:rtc_pc_base",
>-    "../rtc_base:checks",
>-    "../rtc_base:rtc_base",
>-    "../rtc_base:rtc_base_approved",
>-    "../rtc_base/third_party/sigslot",
>-    "//third_party/abseil-cpp/absl/memory",
>-    "//third_party/abseil-cpp/absl/types:optional",
>-  ]
>-
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-}
>-
>-if (rtc_include_tests) {
>-  rtc_test("ortc_unittests") {
>-    testonly = true
>-
>-    sources = [
>-      "ortcfactory_integrationtest.cc",
>-      "ortcfactory_unittest.cc",
>-      "ortcrtpreceiver_unittest.cc",
>-      "ortcrtpsender_unittest.cc",
>-      "rtptransport_unittest.cc",
>-      "rtptransportcontroller_unittest.cc",
>-      "srtptransport_unittest.cc",
>-      "testrtpparameters.cc",
>-      "testrtpparameters.h",
>-    ]
>-
>-    deps = [
>-      ":ortc",
>-      "../api:libjingle_peerconnection_api",
>-      "../api:ortc_api",
>-      "../api/audio_codecs:builtin_audio_decoder_factory",
>-      "../api/audio_codecs:builtin_audio_encoder_factory",
>-      "../media:rtc_media_tests_utils",
>-      "../p2p:p2p_test_utils",
>-      "../p2p:rtc_p2p",
>-      "../pc:pc_test_utils",
>-      "../pc:peerconnection",
>-      "../rtc_base:rtc_base",
>-      "../rtc_base:rtc_base_approved",
>-      "../rtc_base:rtc_base_tests_main",
>-      "../rtc_base:rtc_base_tests_utils",
>-      "../rtc_base/system:arch",
>-    ]
>-
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-
>-    if (is_android) {
>-      deps += [ "//testing/android/native_test:native_test_support" ]
>-    }
>-  }
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/DEPS
>deleted file mode 100644
>index bebf030b8290e6290c9326999d1e5eadc17fe6fd..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/DEPS
>+++ /dev/null
>@@ -1,17 +0,0 @@
>-include_rules = [
>-  "+api",
>-  "+call",
>-  "+logging/rtc_event_log",
>-  "+media",
>-  "+modules/audio_coding",
>-  "+modules/audio_processing",
>-  "+p2p",
>-  "+pc",
>-
>-  "+modules/rtp_rtcp",
>-  "+system_wrappers",
>-
>-  "+modules/audio_device",
>-  "+modules/video_coding",
>-  "+modules/video_render",
>-]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/OWNERS
>deleted file mode 100644
>index 4c08fff50b8f2ea7393d9ab9d3e75d7e4b6cd301..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/OWNERS
>+++ /dev/null
>@@ -1,7 +0,0 @@
>-pthatcher@webrtc.org
>-steveanton@webrtc.org
>-
>-# These are for the common case of adding or renaming files. If you're doing
>-# structural changes, please get a review from a reviewer in this file.
>-per-file *.gn=*
>-per-file *.gni=*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc
>deleted file mode 100644
>index cb63d26174ba6786409544fc154e2b1355ef2f59..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc
>+++ /dev/null
>@@ -1,563 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "ortc/ortcfactory.h"
>-
>-#include <utility>  // For std::move.
>-#include <vector>
>-
>-#include "absl/memory/memory.h"
>-#include "api/mediastreamtrackproxy.h"
>-#include "api/proxy.h"
>-#include "api/rtcerror.h"
>-#include "api/video_codecs/builtin_video_decoder_factory.h"
>-#include "api/video_codecs/builtin_video_encoder_factory.h"
>-#include "api/videosourceproxy.h"
>-#include "logging/rtc_event_log/rtc_event_log.h"
>-#include "media/base/mediaconstants.h"
>-#include "media/base/rtpdataengine.h"
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "ortc/ortcrtpreceiveradapter.h"
>-#include "ortc/ortcrtpsenderadapter.h"
>-#include "ortc/rtptransportadapter.h"
>-#include "ortc/rtptransportcontrolleradapter.h"
>-#include "p2p/base/basicpacketsocketfactory.h"
>-#include "p2p/base/udptransport.h"
>-#include "pc/audiotrack.h"
>-#include "pc/channelmanager.h"
>-#include "pc/localaudiosource.h"
>-#include "pc/rtpparametersconversion.h"
>-#include "pc/videotrack.h"
>-#include "rtc_base/asyncpacketsocket.h"
>-#include "rtc_base/bind.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/helpers.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/strings/string_builder.h"
>-
>-namespace {
>-
>-const int kDefaultRtcpCnameLength = 16;
>-
>-// Asserts that all of the built-in capabilities can be converted to
>-// RtpCapabilities. If they can't, something's wrong (for example, maybe a new
>-// feedback mechanism is supported, but an enum value wasn't added to
>-// rtpparameters.h).
>-template <typename C>
>-webrtc::RtpCapabilities ToRtpCapabilitiesWithAsserts(
>-    const std::vector<C>& cricket_codecs,
>-    const cricket::RtpHeaderExtensions& cricket_extensions) {
>-  webrtc::RtpCapabilities capabilities =
>-      webrtc::ToRtpCapabilities(cricket_codecs, cricket_extensions);
>-  RTC_DCHECK_EQ(capabilities.codecs.size(), cricket_codecs.size());
>-  for (size_t i = 0; i < capabilities.codecs.size(); ++i) {
>-    RTC_DCHECK_EQ(capabilities.codecs[i].rtcp_feedback.size(),
>-                  cricket_codecs[i].feedback_params.params().size());
>-  }
>-  RTC_DCHECK_EQ(capabilities.header_extensions.size(),
>-                cricket_extensions.size());
>-  return capabilities;
>-}
>-
>-}  // namespace
>-
>-namespace webrtc {
>-
>-// Note that this proxy class uses the network thread as the "worker" thread.
>-BEGIN_OWNED_PROXY_MAP(OrtcFactory)
>-PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-PROXY_METHOD0(RTCErrorOr<std::unique_ptr<RtpTransportControllerInterface>>,
>-              CreateRtpTransportController)
>-PROXY_METHOD4(RTCErrorOr<std::unique_ptr<RtpTransportInterface>>,
>-              CreateRtpTransport,
>-              const RtpTransportParameters&,
>-              PacketTransportInterface*,
>-              PacketTransportInterface*,
>-              RtpTransportControllerInterface*)
>-
>-PROXY_METHOD4(RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>,
>-              CreateSrtpTransport,
>-              const RtpTransportParameters&,
>-              PacketTransportInterface*,
>-              PacketTransportInterface*,
>-              RtpTransportControllerInterface*)
>-
>-PROXY_CONSTMETHOD1(RtpCapabilities,
>-                   GetRtpSenderCapabilities,
>-                   cricket::MediaType)
>-PROXY_METHOD2(RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>>,
>-              CreateRtpSender,
>-              rtc::scoped_refptr<MediaStreamTrackInterface>,
>-              RtpTransportInterface*)
>-PROXY_METHOD2(RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>>,
>-              CreateRtpSender,
>-              cricket::MediaType,
>-              RtpTransportInterface*)
>-PROXY_CONSTMETHOD1(RtpCapabilities,
>-                   GetRtpReceiverCapabilities,
>-                   cricket::MediaType)
>-PROXY_METHOD2(RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>>,
>-              CreateRtpReceiver,
>-              cricket::MediaType,
>-              RtpTransportInterface*)
>-PROXY_WORKER_METHOD3(RTCErrorOr<std::unique_ptr<UdpTransportInterface>>,
>-                     CreateUdpTransport,
>-                     int,
>-                     uint16_t,
>-                     uint16_t)
>-PROXY_METHOD1(rtc::scoped_refptr<AudioSourceInterface>,
>-              CreateAudioSource,
>-              const cricket::AudioOptions&)
>-PROXY_METHOD2(rtc::scoped_refptr<VideoTrackInterface>,
>-              CreateVideoTrack,
>-              const std::string&,
>-              VideoTrackSourceInterface*)
>-PROXY_METHOD2(rtc::scoped_refptr<AudioTrackInterface>,
>-              CreateAudioTrack,
>-              const std::string&,
>-              AudioSourceInterface*)
>-END_PROXY_MAP()
>-
>-// static
>-RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> OrtcFactory::Create(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* signaling_thread,
>-    rtc::NetworkManager* network_manager,
>-    rtc::PacketSocketFactory* socket_factory,
>-    AudioDeviceModule* adm,
>-    std::unique_ptr<cricket::MediaEngineInterface> media_engine,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory) {
>-  // Hop to signaling thread if needed.
>-  if (signaling_thread && !signaling_thread->IsCurrent()) {
>-    return signaling_thread
>-        ->Invoke<RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>>>(
>-            RTC_FROM_HERE,
>-            rtc::Bind(&OrtcFactory::Create_s, network_thread, signaling_thread,
>-                      network_manager, socket_factory, adm,
>-                      media_engine.release(), audio_encoder_factory,
>-                      audio_decoder_factory));
>-  }
>-  return Create_s(network_thread, signaling_thread, network_manager,
>-                  socket_factory, adm, media_engine.release(),
>-                  audio_encoder_factory, audio_decoder_factory);
>-}
>-
>-RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> OrtcFactoryInterface::Create(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* signaling_thread,
>-    rtc::NetworkManager* network_manager,
>-    rtc::PacketSocketFactory* socket_factory,
>-    AudioDeviceModule* adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory) {
>-  return OrtcFactory::Create(network_thread, signaling_thread, network_manager,
>-                             socket_factory, adm, nullptr,
>-                             audio_encoder_factory, audio_decoder_factory);
>-}
>-
>-OrtcFactory::OrtcFactory(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* signaling_thread,
>-    rtc::NetworkManager* network_manager,
>-    rtc::PacketSocketFactory* socket_factory,
>-    AudioDeviceModule* adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory)
>-    : network_thread_(network_thread),
>-      signaling_thread_(signaling_thread),
>-      network_manager_(network_manager),
>-      socket_factory_(socket_factory),
>-      adm_(adm),
>-      null_event_log_(RtcEventLog::CreateNull()),
>-      audio_encoder_factory_(audio_encoder_factory),
>-      audio_decoder_factory_(audio_decoder_factory) {
>-  if (!rtc::CreateRandomString(kDefaultRtcpCnameLength, &default_cname_)) {
>-    RTC_LOG(LS_ERROR) << "Failed to generate CNAME?";
>-    RTC_NOTREACHED();
>-  }
>-  if (!network_thread_) {
>-    owned_network_thread_ = rtc::Thread::CreateWithSocketServer();
>-    owned_network_thread_->Start();
>-    network_thread_ = owned_network_thread_.get();
>-  }
>-
>-  // The worker thread is created internally because it's an implementation
>-  // detail, and consumers of the API don't need to really know about it.
>-  worker_thread_ = rtc::Thread::Create();
>-  worker_thread_->SetName("ORTC-worker", this);
>-  worker_thread_->Start();
>-
>-  if (signaling_thread_) {
>-    RTC_DCHECK_RUN_ON(signaling_thread_);
>-  } else {
>-    signaling_thread_ = rtc::Thread::Current();
>-    if (!signaling_thread_) {
>-      // If this thread isn't already wrapped by an rtc::Thread, create a
>-      // wrapper and own it in this class.
>-      signaling_thread_ = rtc::ThreadManager::Instance()->WrapCurrentThread();
>-      wraps_signaling_thread_ = true;
>-    }
>-  }
>-
>-  if (signaling_thread_->name().empty()) {
>-    signaling_thread_->SetName("ORTC-signaling", this);
>-  }
>-
>-  if (!network_manager_) {
>-    owned_network_manager_.reset(new rtc::BasicNetworkManager());
>-    network_manager_ = owned_network_manager_.get();
>-  }
>-  if (!socket_factory_) {
>-    owned_socket_factory_.reset(
>-        new rtc::BasicPacketSocketFactory(network_thread_));
>-    socket_factory_ = owned_socket_factory_.get();
>-  }
>-}
>-
>-OrtcFactory::~OrtcFactory() {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  if (wraps_signaling_thread_) {
>-    rtc::ThreadManager::Instance()->UnwrapCurrentThread();
>-  }
>-}
>-
>-RTCErrorOr<std::unique_ptr<RtpTransportControllerInterface>>
>-OrtcFactory::CreateRtpTransportController() {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  return RtpTransportControllerAdapter::CreateProxied(
>-      cricket::MediaConfig(), channel_manager_.get(), null_event_log_.get(),
>-      signaling_thread_, worker_thread_.get(), network_thread_);
>-}
>-
>-RTCErrorOr<std::unique_ptr<RtpTransportInterface>>
>-OrtcFactory::CreateRtpTransport(
>-    const RtpTransportParameters& parameters,
>-    PacketTransportInterface* rtp,
>-    PacketTransportInterface* rtcp,
>-    RtpTransportControllerInterface* transport_controller) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  RtpTransportParameters copied_parameters = parameters;
>-  if (copied_parameters.rtcp.cname.empty()) {
>-    copied_parameters.rtcp.cname = default_cname_;
>-  }
>-  if (transport_controller) {
>-    return transport_controller->GetInternal()->CreateProxiedRtpTransport(
>-        copied_parameters, rtp, rtcp);
>-  } else {
>-    // If |transport_controller| is null, create one automatically, which the
>-    // returned RtpTransport will own.
>-    auto controller_result = CreateRtpTransportController();
>-    if (!controller_result.ok()) {
>-      return controller_result.MoveError();
>-    }
>-    auto controller = controller_result.MoveValue();
>-    auto transport_result =
>-        controller->GetInternal()->CreateProxiedRtpTransport(copied_parameters,
>-                                                             rtp, rtcp);
>-    // If RtpTransport was successfully created, transfer ownership of
>-    // |rtp_transport_controller|. Otherwise it will go out of scope and be
>-    // deleted automatically.
>-    if (transport_result.ok()) {
>-      transport_result.value()
>-          ->GetInternal()
>-          ->TakeOwnershipOfRtpTransportController(std::move(controller));
>-    }
>-    return transport_result;
>-  }
>-}
>-
>-RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
>-OrtcFactory::CreateSrtpTransport(
>-    const RtpTransportParameters& parameters,
>-    PacketTransportInterface* rtp,
>-    PacketTransportInterface* rtcp,
>-    RtpTransportControllerInterface* transport_controller) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  RtpTransportParameters copied_parameters = parameters;
>-  if (copied_parameters.rtcp.cname.empty()) {
>-    copied_parameters.rtcp.cname = default_cname_;
>-  }
>-  if (transport_controller) {
>-    return transport_controller->GetInternal()->CreateProxiedSrtpTransport(
>-        copied_parameters, rtp, rtcp);
>-  } else {
>-    // If |transport_controller| is null, create one automatically, which the
>-    // returned SrtpTransport will own.
>-    auto controller_result = CreateRtpTransportController();
>-    if (!controller_result.ok()) {
>-      return controller_result.MoveError();
>-    }
>-    auto controller = controller_result.MoveValue();
>-    auto transport_result =
>-        controller->GetInternal()->CreateProxiedSrtpTransport(copied_parameters,
>-                                                              rtp, rtcp);
>-    // If SrtpTransport was successfully created, transfer ownership of
>-    // |rtp_transport_controller|. Otherwise it will go out of scope and be
>-    // deleted automatically.
>-    if (transport_result.ok()) {
>-      transport_result.value()
>-          ->GetInternal()
>-          ->TakeOwnershipOfRtpTransportController(std::move(controller));
>-    }
>-    return transport_result;
>-  }
>-}
>-
>-RtpCapabilities OrtcFactory::GetRtpSenderCapabilities(
>-    cricket::MediaType kind) const {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  switch (kind) {
>-    case cricket::MEDIA_TYPE_AUDIO: {
>-      cricket::AudioCodecs cricket_codecs;
>-      cricket::RtpHeaderExtensions cricket_extensions;
>-      channel_manager_->GetSupportedAudioSendCodecs(&cricket_codecs);
>-      channel_manager_->GetSupportedAudioRtpHeaderExtensions(
>-          &cricket_extensions);
>-      return ToRtpCapabilitiesWithAsserts(cricket_codecs, cricket_extensions);
>-    }
>-    case cricket::MEDIA_TYPE_VIDEO: {
>-      cricket::VideoCodecs cricket_codecs;
>-      cricket::RtpHeaderExtensions cricket_extensions;
>-      channel_manager_->GetSupportedVideoCodecs(&cricket_codecs);
>-      channel_manager_->GetSupportedVideoRtpHeaderExtensions(
>-          &cricket_extensions);
>-      return ToRtpCapabilitiesWithAsserts(cricket_codecs, cricket_extensions);
>-    }
>-    case cricket::MEDIA_TYPE_DATA:
>-      return RtpCapabilities();
>-  }
>-  // Not reached; avoids compile warning.
>-  FATAL();
>-}
>-
>-RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>>
>-OrtcFactory::CreateRtpSender(
>-    rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-    RtpTransportInterface* transport) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  if (!track) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Cannot pass null track into CreateRtpSender.");
>-  }
>-  auto result =
>-      CreateRtpSender(cricket::MediaTypeFromString(track->kind()), transport);
>-  if (!result.ok()) {
>-    return result;
>-  }
>-  auto err = result.value()->SetTrack(track);
>-  if (!err.ok()) {
>-    return std::move(err);
>-  }
>-  return result;
>-}
>-
>-RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>>
>-OrtcFactory::CreateRtpSender(cricket::MediaType kind,
>-                             RtpTransportInterface* transport) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  if (kind == cricket::MEDIA_TYPE_DATA) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Cannot create data RtpSender.");
>-  }
>-  if (!transport) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Cannot pass null transport into CreateRtpSender.");
>-  }
>-  return transport->GetInternal()
>-      ->rtp_transport_controller()
>-      ->CreateProxiedRtpSender(kind, transport);
>-}
>-
>-RtpCapabilities OrtcFactory::GetRtpReceiverCapabilities(
>-    cricket::MediaType kind) const {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  switch (kind) {
>-    case cricket::MEDIA_TYPE_AUDIO: {
>-      cricket::AudioCodecs cricket_codecs;
>-      cricket::RtpHeaderExtensions cricket_extensions;
>-      channel_manager_->GetSupportedAudioReceiveCodecs(&cricket_codecs);
>-      channel_manager_->GetSupportedAudioRtpHeaderExtensions(
>-          &cricket_extensions);
>-      return ToRtpCapabilitiesWithAsserts(cricket_codecs, cricket_extensions);
>-    }
>-    case cricket::MEDIA_TYPE_VIDEO: {
>-      cricket::VideoCodecs cricket_codecs;
>-      cricket::RtpHeaderExtensions cricket_extensions;
>-      channel_manager_->GetSupportedVideoCodecs(&cricket_codecs);
>-      channel_manager_->GetSupportedVideoRtpHeaderExtensions(
>-          &cricket_extensions);
>-      return ToRtpCapabilitiesWithAsserts(cricket_codecs, cricket_extensions);
>-    }
>-    case cricket::MEDIA_TYPE_DATA:
>-      return RtpCapabilities();
>-  }
>-  // Not reached; avoids compile warning.
>-  FATAL();
>-}
>-
>-RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>>
>-OrtcFactory::CreateRtpReceiver(cricket::MediaType kind,
>-                               RtpTransportInterface* transport) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  if (kind == cricket::MEDIA_TYPE_DATA) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Cannot create data RtpReceiver.");
>-  }
>-  if (!transport) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Cannot pass null transport into CreateRtpReceiver.");
>-  }
>-  return transport->GetInternal()
>-      ->rtp_transport_controller()
>-      ->CreateProxiedRtpReceiver(kind, transport);
>-}
>-
>-// UdpTransport expects all methods to be called on one thread, which needs to
>-// be the network thread, since that's where its socket can safely be used. So
>-// return a proxy to the created UdpTransport.
>-BEGIN_OWNED_PROXY_MAP(UdpTransport)
>-PROXY_WORKER_THREAD_DESTRUCTOR()
>-PROXY_WORKER_CONSTMETHOD0(rtc::SocketAddress, GetLocalAddress)
>-PROXY_WORKER_METHOD1(bool, SetRemoteAddress, const rtc::SocketAddress&)
>-PROXY_WORKER_CONSTMETHOD0(rtc::SocketAddress, GetRemoteAddress)
>-protected:
>-rtc::PacketTransportInternal* GetInternal() override {
>-  return internal();
>-}
>-END_PROXY_MAP()
>-
>-RTCErrorOr<std::unique_ptr<UdpTransportInterface>>
>-OrtcFactory::CreateUdpTransport(int family,
>-                                uint16_t min_port,
>-                                uint16_t max_port) {
>-  RTC_DCHECK_RUN_ON(network_thread_);
>-  if (family != AF_INET && family != AF_INET6) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Address family must be AF_INET or AF_INET6.");
>-  }
>-  if (min_port > max_port) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
>-                         "Port range invalid; minimum port must be less than "
>-                         "or equal to max port.");
>-  }
>-  std::unique_ptr<rtc::AsyncPacketSocket> socket(
>-      socket_factory_->CreateUdpSocket(
>-          rtc::SocketAddress(rtc::GetAnyIP(family), 0), min_port, max_port));
>-  if (!socket) {
>-    // Only log at warning level, because this method may be called with
>-    // specific port ranges to determine if a port is available, expecting the
>-    // possibility of an error.
>-    LOG_AND_RETURN_ERROR_EX(RTCErrorType::RESOURCE_EXHAUSTED,
>-                            "Local socket allocation failure.", LS_WARNING);
>-  }
>-  RTC_LOG(LS_INFO) << "Created UDP socket with address "
>-                   << socket->GetLocalAddress().ToSensitiveString() << ".";
>-  // Make a unique debug name (for logging/diagnostics only).
>-  rtc::StringBuilder oss;
>-  static int udp_id = 0;
>-  oss << "udp" << udp_id++;
>-  return UdpTransportProxyWithInternal<cricket::UdpTransport>::Create(
>-      signaling_thread_, network_thread_,
>-      std::unique_ptr<cricket::UdpTransport>(
>-          new cricket::UdpTransport(oss.str(), std::move(socket))));
>-}
>-
>-rtc::scoped_refptr<AudioSourceInterface> OrtcFactory::CreateAudioSource(
>-    const cricket::AudioOptions& options) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  return rtc::scoped_refptr<LocalAudioSource>(
>-      LocalAudioSource::Create(&options));
>-}
>-
>-rtc::scoped_refptr<VideoTrackInterface> OrtcFactory::CreateVideoTrack(
>-    const std::string& id,
>-    VideoTrackSourceInterface* source) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  rtc::scoped_refptr<VideoTrackInterface> track(
>-      VideoTrack::Create(id, source, worker_thread_.get()));
>-  return VideoTrackProxy::Create(signaling_thread_, worker_thread_.get(),
>-                                 track);
>-}
>-
>-rtc::scoped_refptr<AudioTrackInterface> OrtcFactory::CreateAudioTrack(
>-    const std::string& id,
>-    AudioSourceInterface* source) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  rtc::scoped_refptr<AudioTrackInterface> track(AudioTrack::Create(id, source));
>-  return AudioTrackProxy::Create(signaling_thread_, track);
>-}
>-
>-// static
>-RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> OrtcFactory::Create_s(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* signaling_thread,
>-    rtc::NetworkManager* network_manager,
>-    rtc::PacketSocketFactory* socket_factory,
>-    AudioDeviceModule* adm,
>-    cricket::MediaEngineInterface* media_engine,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory) {
>-  // Add the unique_ptr wrapper back.
>-  std::unique_ptr<cricket::MediaEngineInterface> owned_media_engine(
>-      media_engine);
>-  std::unique_ptr<OrtcFactory> new_factory(new OrtcFactory(
>-      network_thread, signaling_thread, network_manager, socket_factory, adm,
>-      audio_encoder_factory, audio_decoder_factory));
>-  RTCError err = new_factory->Initialize(std::move(owned_media_engine));
>-  if (!err.ok()) {
>-    return std::move(err);
>-  }
>-  // Return a proxy so that any calls on the returned object (including
>-  // destructor) happen on the signaling thread.
>-  rtc::Thread* signaling = new_factory->signaling_thread();
>-  rtc::Thread* network = new_factory->network_thread();
>-  return OrtcFactoryProxy::Create(signaling, network, std::move(new_factory));
>-}
>-
>-RTCError OrtcFactory::Initialize(
>-    std::unique_ptr<cricket::MediaEngineInterface> media_engine) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  // TODO(deadbeef): Get rid of requirement to hop to worker thread here.
>-  if (!media_engine) {
>-    media_engine =
>-        worker_thread_->Invoke<std::unique_ptr<cricket::MediaEngineInterface>>(
>-            RTC_FROM_HERE, rtc::Bind(&OrtcFactory::CreateMediaEngine_w, this));
>-  }
>-
>-  channel_manager_.reset(new cricket::ChannelManager(
>-      std::move(media_engine), absl::make_unique<cricket::RtpDataEngine>(),
>-      worker_thread_.get(), network_thread_));
>-  channel_manager_->SetVideoRtxEnabled(true);
>-  if (!channel_manager_->Init()) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>-                         "Failed to initialize ChannelManager.");
>-  }
>-  return RTCError::OK();
>-}
>-
>-std::unique_ptr<cricket::MediaEngineInterface>
>-OrtcFactory::CreateMediaEngine_w() {
>-  RTC_DCHECK_RUN_ON(worker_thread_.get());
>-  // The null arguments are optional factories that could be passed into the
>-  // OrtcFactory, but aren't yet.
>-  //
>-  // Note that |adm_| may be null, in which case the platform-specific default
>-  // AudioDeviceModule will be used.
>-  return std::unique_ptr<cricket::MediaEngineInterface>(
>-      cricket::WebRtcMediaEngineFactory::Create(
>-          rtc::scoped_refptr<webrtc::AudioDeviceModule>(adm_),
>-          audio_encoder_factory_, audio_decoder_factory_,
>-          webrtc::CreateBuiltinVideoEncoderFactory(),
>-          webrtc::CreateBuiltinVideoDecoderFactory(), nullptr,
>-          webrtc::AudioProcessingBuilder().Create()));
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.h
>deleted file mode 100644
>index d72f6121dbbd92a38e69af5ca703c81a4358232e..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.h
>+++ /dev/null
>@@ -1,156 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef ORTC_ORTCFACTORY_H_
>-#define ORTC_ORTCFACTORY_H_
>-
>-#include <memory>
>-#include <string>
>-
>-#include "api/ortc/ortcfactoryinterface.h"
>-#include "media/base/mediaengine.h"
>-#include "media/engine/webrtcmediaengine.h"
>-#include "pc/channelmanager.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/scoped_ref_ptr.h"
>-
>-namespace webrtc {
>-
>-// Implementation of OrtcFactoryInterface.
>-//
>-// See ortcfactoryinterface.h for documentation.
>-class OrtcFactory : public OrtcFactoryInterface {
>- public:
>-  ~OrtcFactory() override;
>-
>-  // Internal-only Create method that allows passing in a fake media engine,
>-  // for testing.
>-  static RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> Create(
>-      rtc::Thread* network_thread,
>-      rtc::Thread* signaling_thread,
>-      rtc::NetworkManager* network_manager,
>-      rtc::PacketSocketFactory* socket_factory,
>-      AudioDeviceModule* adm,
>-      std::unique_ptr<cricket::MediaEngineInterface> media_engine,
>-      rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-      rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory);
>-
>-  RTCErrorOr<std::unique_ptr<RtpTransportControllerInterface>>
>-  CreateRtpTransportController() override;
>-
>-  RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateRtpTransport(
>-      const RtpTransportParameters& parameters,
>-      PacketTransportInterface* rtp,
>-      PacketTransportInterface* rtcp,
>-      RtpTransportControllerInterface* transport_controller) override;
>-
>-  RTCErrorOr<std::unique_ptr<SrtpTransportInterface>> CreateSrtpTransport(
>-      const RtpTransportParameters& parameters,
>-      PacketTransportInterface* rtp,
>-      PacketTransportInterface* rtcp,
>-      RtpTransportControllerInterface* transport_controller) override;
>-
>-  RtpCapabilities GetRtpSenderCapabilities(
>-      cricket::MediaType kind) const override;
>-
>-  RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateRtpSender(
>-      rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-      RtpTransportInterface* transport) override;
>-
>-  RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateRtpSender(
>-      cricket::MediaType kind,
>-      RtpTransportInterface* transport) override;
>-
>-  RtpCapabilities GetRtpReceiverCapabilities(
>-      cricket::MediaType kind) const override;
>-
>-  RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>> CreateRtpReceiver(
>-      cricket::MediaType kind,
>-      RtpTransportInterface* transport) override;
>-
>-  RTCErrorOr<std::unique_ptr<UdpTransportInterface>>
>-  CreateUdpTransport(int family, uint16_t min_port, uint16_t max_port) override;
>-
>-  rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
>-      const cricket::AudioOptions& options) override;
>-
>-  rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
>-      const std::string& id,
>-      VideoTrackSourceInterface* source) override;
>-
>-  rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
>-      const std::string& id,
>-      AudioSourceInterface* source) override;
>-
>-  rtc::Thread* network_thread() { return network_thread_; }
>-  rtc::Thread* worker_thread() { return worker_thread_.get(); }
>-  rtc::Thread* signaling_thread() { return signaling_thread_; }
>-
>- private:
>-  // Should only be called by OrtcFactoryInterface::Create.
>-  OrtcFactory(rtc::Thread* network_thread,
>-              rtc::Thread* signaling_thread,
>-              rtc::NetworkManager* network_manager,
>-              rtc::PacketSocketFactory* socket_factory,
>-              AudioDeviceModule* adm,
>-              rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-              rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory);
>-
>-  RTCErrorOr<std::unique_ptr<RtpTransportControllerInterface>>
>-  CreateRtpTransportController(const RtpTransportParameters& parameters);
>-
>-  // Thread::Invoke doesn't support move-only arguments, so we need to remove
>-  // the unique_ptr wrapper from media_engine. TODO(deadbeef): Fix this.
>-  static RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> Create_s(
>-      rtc::Thread* network_thread,
>-      rtc::Thread* signaling_thread,
>-      rtc::NetworkManager* network_manager,
>-      rtc::PacketSocketFactory* socket_factory,
>-      AudioDeviceModule* adm,
>-      cricket::MediaEngineInterface* media_engine,
>-      rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-      rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory);
>-
>-  // Performs initialization that can fail. Called by factory method after
>-  // construction, and if it fails, no object is returned.
>-  RTCError Initialize(
>-      std::unique_ptr<cricket::MediaEngineInterface> media_engine);
>-  std::unique_ptr<cricket::MediaEngineInterface> CreateMediaEngine_w();
>-
>-  // Threads and networking objects.
>-  rtc::Thread* network_thread_;
>-  rtc::Thread* signaling_thread_;
>-  rtc::NetworkManager* network_manager_;
>-  rtc::PacketSocketFactory* socket_factory_;
>-  AudioDeviceModule* adm_;
>-  // If we created/own the objects above, these will be non-null and thus will
>-  // be released automatically upon destruction.
>-  std::unique_ptr<rtc::Thread> owned_network_thread_;
>-  bool wraps_signaling_thread_ = false;
>-  std::unique_ptr<rtc::NetworkManager> owned_network_manager_;
>-  std::unique_ptr<rtc::PacketSocketFactory> owned_socket_factory_;
>-  // We always own the worker thread.
>-  std::unique_ptr<rtc::Thread> worker_thread_;
>-  // Media-releated objects.
>-  std::unique_ptr<RtcEventLog> null_event_log_;
>-  rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
>-  rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
>-  std::unique_ptr<cricket::ChannelManager> channel_manager_;
>-  // Default CNAME to use for RtpTransports if none is passed in.
>-  std::string default_cname_;
>-
>-  friend class OrtcFactoryInterface;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(OrtcFactory);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // ORTC_ORTCFACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory_integrationtest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory_integrationtest.cc
>deleted file mode 100644
>index 62f29520ed8d76888a71890a7715967a19118f15..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory_integrationtest.cc
>+++ /dev/null
>@@ -1,724 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-#include <utility>  // For std::pair, std::move.
>-
>-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>-#include "api/ortc/ortcfactoryinterface.h"
>-#include "ortc/testrtpparameters.h"
>-#include "p2p/base/udptransport.h"
>-#include "pc/test/fakeaudiocapturemodule.h"
>-#include "pc/test/fakeperiodicvideotracksource.h"
>-#include "pc/test/fakevideotrackrenderer.h"
>-#include "pc/videotracksource.h"
>-#include "rtc_base/criticalsection.h"
>-#include "rtc_base/fakenetwork.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/system/arch.h"
>-#include "rtc_base/timeutils.h"
>-#include "rtc_base/virtualsocketserver.h"
>-
>-namespace {
>-
>-const int kDefaultTimeout = 10000;    // 10 seconds.
>-const int kReceivingDuration = 1000;  // 1 second.
>-
>-// Default number of audio/video frames to wait for before considering a test a
>-// success.
>-const int kDefaultNumFrames = 3;
>-const rtc::IPAddress kIPv4LocalHostAddress =
>-    rtc::IPAddress(0x7F000001);  // 127.0.0.1
>-
>-static const char kTestKeyParams1[] =
>-    "inline:WVNfX19zZW1jdGwgKskgewkyMjA7fQp9CnVubGVz";
>-static const char kTestKeyParams2[] =
>-    "inline:PS1uQCVeeCFCanVmcjkpaywjNWhcYD0mXXtxaVBR";
>-static const char kTestKeyParams3[] =
>-    "inline:WVNfX19zZW1jdGwgKskgewkyMjA7fQp9CnVubGVa";
>-static const char kTestKeyParams4[] =
>-    "inline:WVNfX19zZW1jdGwgKskgewkyMjA7fQp9CnVubGVb";
>-static const cricket::CryptoParams kTestCryptoParams1(1,
>-                                                      "AES_CM_128_HMAC_SHA1_80",
>-                                                      kTestKeyParams1,
>-                                                      "");
>-static const cricket::CryptoParams kTestCryptoParams2(1,
>-                                                      "AES_CM_128_HMAC_SHA1_80",
>-                                                      kTestKeyParams2,
>-                                                      "");
>-static const cricket::CryptoParams kTestCryptoParams3(1,
>-                                                      "AES_CM_128_HMAC_SHA1_80",
>-                                                      kTestKeyParams3,
>-                                                      "");
>-static const cricket::CryptoParams kTestCryptoParams4(1,
>-                                                      "AES_CM_128_HMAC_SHA1_80",
>-                                                      kTestKeyParams4,
>-                                                      "");
>-}  // namespace
>-
>-namespace webrtc {
>-
>-// Used to test that things work end-to-end when using the default
>-// implementations of threads/etc. provided by OrtcFactory, with the exception
>-// of using a virtual network.
>-//
>-// By default, the virtual network manager doesn't enumerate any networks, but
>-// sockets can still be created in this state.
>-class OrtcFactoryIntegrationTest : public testing::Test {
>- public:
>-  OrtcFactoryIntegrationTest()
>-      : network_thread_(&virtual_socket_server_),
>-        fake_audio_capture_module1_(FakeAudioCaptureModule::Create()),
>-        fake_audio_capture_module2_(FakeAudioCaptureModule::Create()) {
>-    // Sockets are bound to the ANY address, so this is needed to tell the
>-    // virtual network which address to use in this case.
>-    virtual_socket_server_.SetDefaultRoute(kIPv4LocalHostAddress);
>-    network_thread_.SetName("TestNetworkThread", this);
>-    network_thread_.Start();
>-    // Need to create after network thread is started.
>-    ortc_factory1_ =
>-        OrtcFactoryInterface::Create(
>-            &network_thread_, nullptr, &fake_network_manager_, nullptr,
>-            fake_audio_capture_module1_, CreateBuiltinAudioEncoderFactory(),
>-            CreateBuiltinAudioDecoderFactory())
>-            .MoveValue();
>-    ortc_factory2_ =
>-        OrtcFactoryInterface::Create(
>-            &network_thread_, nullptr, &fake_network_manager_, nullptr,
>-            fake_audio_capture_module2_, CreateBuiltinAudioEncoderFactory(),
>-            CreateBuiltinAudioDecoderFactory())
>-            .MoveValue();
>-  }
>-
>- protected:
>-  typedef std::pair<std::unique_ptr<UdpTransportInterface>,
>-                    std::unique_ptr<UdpTransportInterface>>
>-      UdpTransportPair;
>-  typedef std::pair<std::unique_ptr<RtpTransportInterface>,
>-                    std::unique_ptr<RtpTransportInterface>>
>-      RtpTransportPair;
>-  typedef std::pair<std::unique_ptr<SrtpTransportInterface>,
>-                    std::unique_ptr<SrtpTransportInterface>>
>-      SrtpTransportPair;
>-  typedef std::pair<std::unique_ptr<RtpTransportControllerInterface>,
>-                    std::unique_ptr<RtpTransportControllerInterface>>
>-      RtpTransportControllerPair;
>-
>-  // Helper function that creates one UDP transport each for |ortc_factory1_|
>-  // and |ortc_factory2_|, and connects them.
>-  UdpTransportPair CreateAndConnectUdpTransportPair() {
>-    auto transport1 = ortc_factory1_->CreateUdpTransport(AF_INET).MoveValue();
>-    auto transport2 = ortc_factory2_->CreateUdpTransport(AF_INET).MoveValue();
>-    transport1->SetRemoteAddress(
>-        rtc::SocketAddress(virtual_socket_server_.GetDefaultRoute(AF_INET),
>-                           transport2->GetLocalAddress().port()));
>-    transport2->SetRemoteAddress(
>-        rtc::SocketAddress(virtual_socket_server_.GetDefaultRoute(AF_INET),
>-                           transport1->GetLocalAddress().port()));
>-    return {std::move(transport1), std::move(transport2)};
>-  }
>-
>-  // Creates one transport controller each for |ortc_factory1_| and
>-  // |ortc_factory2_|.
>-  RtpTransportControllerPair CreateRtpTransportControllerPair() {
>-    return {ortc_factory1_->CreateRtpTransportController().MoveValue(),
>-            ortc_factory2_->CreateRtpTransportController().MoveValue()};
>-  }
>-
>-  // Helper function that creates a pair of RtpTransports between
>-  // |ortc_factory1_| and |ortc_factory2_|. Expected to be called with the
>-  // result of CreateAndConnectUdpTransportPair. |rtcp_udp_transports| can be
>-  // empty if RTCP muxing is used. |transport_controllers| can be empty if
>-  // these transports are being created using a default transport controller.
>-  RtpTransportPair CreateRtpTransportPair(
>-      const RtpTransportParameters& parameters,
>-      const UdpTransportPair& rtp_udp_transports,
>-      const UdpTransportPair& rtcp_udp_transports,
>-      const RtpTransportControllerPair& transport_controllers) {
>-    auto transport_result1 = ortc_factory1_->CreateRtpTransport(
>-        parameters, rtp_udp_transports.first.get(),
>-        rtcp_udp_transports.first.get(), transport_controllers.first.get());
>-    auto transport_result2 = ortc_factory2_->CreateRtpTransport(
>-        parameters, rtp_udp_transports.second.get(),
>-        rtcp_udp_transports.second.get(), transport_controllers.second.get());
>-    return {transport_result1.MoveValue(), transport_result2.MoveValue()};
>-  }
>-
>-  SrtpTransportPair CreateSrtpTransportPair(
>-      const RtpTransportParameters& parameters,
>-      const UdpTransportPair& rtp_udp_transports,
>-      const UdpTransportPair& rtcp_udp_transports,
>-      const RtpTransportControllerPair& transport_controllers) {
>-    auto transport_result1 = ortc_factory1_->CreateSrtpTransport(
>-        parameters, rtp_udp_transports.first.get(),
>-        rtcp_udp_transports.first.get(), transport_controllers.first.get());
>-    auto transport_result2 = ortc_factory2_->CreateSrtpTransport(
>-        parameters, rtp_udp_transports.second.get(),
>-        rtcp_udp_transports.second.get(), transport_controllers.second.get());
>-    return {transport_result1.MoveValue(), transport_result2.MoveValue()};
>-  }
>-
>-  // For convenience when |rtcp_udp_transports| and |transport_controllers|
>-  // aren't needed.
>-  RtpTransportPair CreateRtpTransportPair(
>-      const RtpTransportParameters& parameters,
>-      const UdpTransportPair& rtp_udp_transports) {
>-    return CreateRtpTransportPair(parameters, rtp_udp_transports,
>-                                  UdpTransportPair(),
>-                                  RtpTransportControllerPair());
>-  }
>-
>-  SrtpTransportPair CreateSrtpTransportPairAndSetKeys(
>-      const RtpTransportParameters& parameters,
>-      const UdpTransportPair& rtp_udp_transports) {
>-    SrtpTransportPair srtp_transports = CreateSrtpTransportPair(
>-        parameters, rtp_udp_transports, UdpTransportPair(),
>-        RtpTransportControllerPair());
>-    EXPECT_TRUE(srtp_transports.first->SetSrtpSendKey(kTestCryptoParams1).ok());
>-    EXPECT_TRUE(
>-        srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams2).ok());
>-    EXPECT_TRUE(
>-        srtp_transports.second->SetSrtpSendKey(kTestCryptoParams2).ok());
>-    EXPECT_TRUE(
>-        srtp_transports.second->SetSrtpReceiveKey(kTestCryptoParams1).ok());
>-    return srtp_transports;
>-  }
>-
>-  SrtpTransportPair CreateSrtpTransportPairAndSetMismatchingKeys(
>-      const RtpTransportParameters& parameters,
>-      const UdpTransportPair& rtp_udp_transports) {
>-    SrtpTransportPair srtp_transports = CreateSrtpTransportPair(
>-        parameters, rtp_udp_transports, UdpTransportPair(),
>-        RtpTransportControllerPair());
>-    EXPECT_TRUE(srtp_transports.first->SetSrtpSendKey(kTestCryptoParams1).ok());
>-    EXPECT_TRUE(
>-        srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams2).ok());
>-    EXPECT_TRUE(
>-        srtp_transports.second->SetSrtpSendKey(kTestCryptoParams1).ok());
>-    EXPECT_TRUE(
>-        srtp_transports.second->SetSrtpReceiveKey(kTestCryptoParams2).ok());
>-    return srtp_transports;
>-  }
>-
>-  // Ends up using fake audio capture module, which was passed into OrtcFactory
>-  // on creation.
>-  rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
>-      const std::string& id,
>-      OrtcFactoryInterface* ortc_factory) {
>-    // Disable echo cancellation to make test more efficient.
>-    cricket::AudioOptions options;
>-    options.echo_cancellation.emplace(true);
>-    rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
>-        ortc_factory->CreateAudioSource(options);
>-    return ortc_factory->CreateAudioTrack(id, source);
>-  }
>-
>-  // Stores created video source in |fake_video_track_sources_|.
>-  rtc::scoped_refptr<webrtc::VideoTrackInterface>
>-  CreateLocalVideoTrackAndFakeSource(const std::string& id,
>-                                     OrtcFactoryInterface* ortc_factory) {
>-    FakePeriodicVideoSource::Config config;
>-    config.timestamp_offset_ms = rtc::TimeMillis();
>-    fake_video_track_sources_.emplace_back(
>-        new rtc::RefCountedObject<FakePeriodicVideoTrackSource>(
>-            config, false /* remote */));
>-    return rtc::scoped_refptr<VideoTrackInterface>(
>-        ortc_factory->CreateVideoTrack(id, fake_video_track_sources_.back()));
>-  }
>-
>-  // Helper function used to test two way RTP senders and receivers with basic
>-  // configurations.
>-  // If |expect_success| is true, waits for kDefaultTimeout for
>-  // kDefaultNumFrames frames to be received by all RtpReceivers.
>-  // If |expect_success| is false, simply waits for |kReceivingDuration|, and
>-  // stores the number of received frames in |received_audio_frame1_| etc.
>-  void BasicTwoWayRtpSendersAndReceiversTest(RtpTransportPair srtp_transports,
>-                                             bool expect_success) {
>-    received_audio_frames1_ = 0;
>-    received_audio_frames2_ = 0;
>-    rendered_video_frames1_ = 0;
>-    rendered_video_frames2_ = 0;
>-    // Create all the senders and receivers (four per endpoint).
>-    auto audio_sender_result1 = ortc_factory1_->CreateRtpSender(
>-        cricket::MEDIA_TYPE_AUDIO, srtp_transports.first.get());
>-    auto video_sender_result1 = ortc_factory1_->CreateRtpSender(
>-        cricket::MEDIA_TYPE_VIDEO, srtp_transports.first.get());
>-    auto audio_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
>-        cricket::MEDIA_TYPE_AUDIO, srtp_transports.first.get());
>-    auto video_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
>-        cricket::MEDIA_TYPE_VIDEO, srtp_transports.first.get());
>-    ASSERT_TRUE(audio_sender_result1.ok());
>-    ASSERT_TRUE(video_sender_result1.ok());
>-    ASSERT_TRUE(audio_receiver_result1.ok());
>-    ASSERT_TRUE(video_receiver_result1.ok());
>-    auto audio_sender1 = audio_sender_result1.MoveValue();
>-    auto video_sender1 = video_sender_result1.MoveValue();
>-    auto audio_receiver1 = audio_receiver_result1.MoveValue();
>-    auto video_receiver1 = video_receiver_result1.MoveValue();
>-
>-    auto audio_sender_result2 = ortc_factory2_->CreateRtpSender(
>-        cricket::MEDIA_TYPE_AUDIO, srtp_transports.second.get());
>-    auto video_sender_result2 = ortc_factory2_->CreateRtpSender(
>-        cricket::MEDIA_TYPE_VIDEO, srtp_transports.second.get());
>-    auto audio_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
>-        cricket::MEDIA_TYPE_AUDIO, srtp_transports.second.get());
>-    auto video_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
>-        cricket::MEDIA_TYPE_VIDEO, srtp_transports.second.get());
>-    ASSERT_TRUE(audio_sender_result2.ok());
>-    ASSERT_TRUE(video_sender_result2.ok());
>-    ASSERT_TRUE(audio_receiver_result2.ok());
>-    ASSERT_TRUE(video_receiver_result2.ok());
>-    auto audio_sender2 = audio_sender_result2.MoveValue();
>-    auto video_sender2 = video_sender_result2.MoveValue();
>-    auto audio_receiver2 = audio_receiver_result2.MoveValue();
>-    auto video_receiver2 = video_receiver_result2.MoveValue();
>-
>-    // Add fake tracks.
>-    RTCError error = audio_sender1->SetTrack(
>-        CreateLocalAudioTrack("audio", ortc_factory1_.get()));
>-    EXPECT_TRUE(error.ok());
>-    error = video_sender1->SetTrack(
>-        CreateLocalVideoTrackAndFakeSource("video", ortc_factory1_.get()));
>-    EXPECT_TRUE(error.ok());
>-    error = audio_sender2->SetTrack(
>-        CreateLocalAudioTrack("audio", ortc_factory2_.get()));
>-    EXPECT_TRUE(error.ok());
>-    error = video_sender2->SetTrack(
>-        CreateLocalVideoTrackAndFakeSource("video", ortc_factory2_.get()));
>-    EXPECT_TRUE(error.ok());
>-
>-    // "sent_X_parameters1" are the parameters that endpoint 1 sends with and
>-    // endpoint 2 receives with.
>-    RtpParameters sent_opus_parameters1 =
>-        MakeMinimalOpusParametersWithSsrc(0xdeadbeef);
>-    RtpParameters sent_vp8_parameters1 =
>-        MakeMinimalVp8ParametersWithSsrc(0xbaadfeed);
>-    RtpParameters sent_opus_parameters2 =
>-        MakeMinimalOpusParametersWithSsrc(0x13333337);
>-    RtpParameters sent_vp8_parameters2 =
>-        MakeMinimalVp8ParametersWithSsrc(0x12345678);
>-
>-    // Configure the senders' and receivers' parameters.
>-    EXPECT_TRUE(audio_receiver1->Receive(sent_opus_parameters2).ok());
>-    EXPECT_TRUE(video_receiver1->Receive(sent_vp8_parameters2).ok());
>-    EXPECT_TRUE(audio_receiver2->Receive(sent_opus_parameters1).ok());
>-    EXPECT_TRUE(video_receiver2->Receive(sent_vp8_parameters1).ok());
>-    EXPECT_TRUE(audio_sender1->Send(sent_opus_parameters1).ok());
>-    EXPECT_TRUE(video_sender1->Send(sent_vp8_parameters1).ok());
>-    EXPECT_TRUE(audio_sender2->Send(sent_opus_parameters2).ok());
>-    EXPECT_TRUE(video_sender2->Send(sent_vp8_parameters2).ok());
>-
>-    FakeVideoTrackRenderer fake_video_renderer1(
>-        static_cast<VideoTrackInterface*>(video_receiver1->GetTrack().get()));
>-    FakeVideoTrackRenderer fake_video_renderer2(
>-        static_cast<VideoTrackInterface*>(video_receiver2->GetTrack().get()));
>-
>-    if (expect_success) {
>-      EXPECT_TRUE_WAIT(
>-          fake_audio_capture_module1_->frames_received() > kDefaultNumFrames &&
>-              fake_video_renderer1.num_rendered_frames() > kDefaultNumFrames &&
>-              fake_audio_capture_module2_->frames_received() >
>-                  kDefaultNumFrames &&
>-              fake_video_renderer2.num_rendered_frames() > kDefaultNumFrames,
>-          kDefaultTimeout)
>-          << "Audio capture module 1 received "
>-          << fake_audio_capture_module1_->frames_received()
>-          << " frames, Video renderer 1 rendered "
>-          << fake_video_renderer1.num_rendered_frames()
>-          << " frames, Audio capture module 2 received "
>-          << fake_audio_capture_module2_->frames_received()
>-          << " frames, Video renderer 2 rendered "
>-          << fake_video_renderer2.num_rendered_frames() << " frames.";
>-    } else {
>-      WAIT(false, kReceivingDuration);
>-      rendered_video_frames1_ = fake_video_renderer1.num_rendered_frames();
>-      rendered_video_frames2_ = fake_video_renderer2.num_rendered_frames();
>-      received_audio_frames1_ = fake_audio_capture_module1_->frames_received();
>-      received_audio_frames2_ = fake_audio_capture_module2_->frames_received();
>-    }
>-  }
>-
>-  rtc::VirtualSocketServer virtual_socket_server_;
>-  rtc::Thread network_thread_;
>-  rtc::FakeNetworkManager fake_network_manager_;
>-  rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module1_;
>-  rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module2_;
>-  std::unique_ptr<OrtcFactoryInterface> ortc_factory1_;
>-  std::unique_ptr<OrtcFactoryInterface> ortc_factory2_;
>-  std::vector<rtc::scoped_refptr<VideoTrackSource>> fake_video_track_sources_;
>-  int received_audio_frames1_ = 0;
>-  int received_audio_frames2_ = 0;
>-  int rendered_video_frames1_ = 0;
>-  int rendered_video_frames2_ = 0;
>-};
>-
>-// Disable for TSan v2, see
>-// https://bugs.chromium.org/p/webrtc/issues/detail?id=7366 for details.
>-#if !defined(THREAD_SANITIZER)
>-
>-// Very basic end-to-end test with a single pair of audio RTP sender and
>-// receiver.
>-//
>-// Uses muxed RTCP, and minimal parameters with a hard-coded config that's
>-// known to work.
>-TEST_F(OrtcFactoryIntegrationTest, BasicOneWayAudioRtpSenderAndReceiver) {
>-  auto udp_transports = CreateAndConnectUdpTransportPair();
>-  auto rtp_transports =
>-      CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
>-
>-  auto sender_result = ortc_factory1_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transports.first.get());
>-  auto receiver_result = ortc_factory2_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transports.second.get());
>-  ASSERT_TRUE(sender_result.ok());
>-  ASSERT_TRUE(receiver_result.ok());
>-  auto sender = sender_result.MoveValue();
>-  auto receiver = receiver_result.MoveValue();
>-
>-  RTCError error =
>-      sender->SetTrack(CreateLocalAudioTrack("audio", ortc_factory1_.get()));
>-  EXPECT_TRUE(error.ok());
>-
>-  RtpParameters opus_parameters = MakeMinimalOpusParameters();
>-  EXPECT_TRUE(receiver->Receive(opus_parameters).ok());
>-  EXPECT_TRUE(sender->Send(opus_parameters).ok());
>-  // Sender and receiver are connected and configured; audio frames should be
>-  // able to flow at this point.
>-  EXPECT_TRUE_WAIT(
>-      fake_audio_capture_module2_->frames_received() > kDefaultNumFrames,
>-      kDefaultTimeout);
>-}
>-
>-// Very basic end-to-end test with a single pair of video RTP sender and
>-// receiver.
>-//
>-// Uses muxed RTCP, and minimal parameters with a hard-coded config that's
>-// known to work.
>-TEST_F(OrtcFactoryIntegrationTest, BasicOneWayVideoRtpSenderAndReceiver) {
>-  auto udp_transports = CreateAndConnectUdpTransportPair();
>-  auto rtp_transports =
>-      CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
>-
>-  auto sender_result = ortc_factory1_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get());
>-  auto receiver_result = ortc_factory2_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get());
>-  ASSERT_TRUE(sender_result.ok());
>-  ASSERT_TRUE(receiver_result.ok());
>-  auto sender = sender_result.MoveValue();
>-  auto receiver = receiver_result.MoveValue();
>-
>-  RTCError error = sender->SetTrack(
>-      CreateLocalVideoTrackAndFakeSource("video", ortc_factory1_.get()));
>-  EXPECT_TRUE(error.ok());
>-
>-  RtpParameters vp8_parameters = MakeMinimalVp8Parameters();
>-  EXPECT_TRUE(receiver->Receive(vp8_parameters).ok());
>-  EXPECT_TRUE(sender->Send(vp8_parameters).ok());
>-  FakeVideoTrackRenderer fake_renderer(
>-      static_cast<VideoTrackInterface*>(receiver->GetTrack().get()));
>-  // Sender and receiver are connected and configured; video frames should be
>-  // able to flow at this point.
>-  EXPECT_TRUE_WAIT(fake_renderer.num_rendered_frames() > kDefaultNumFrames,
>-                   kDefaultTimeout);
>-}
>-
>-// Test that if the track is changed while sending, the sender seamlessly
>-// transitions to sending it and frames are received end-to-end.
>-//
>-// Only doing this for video, since given that audio is sourced from a single
>-// fake audio capture module, the audio track is just a dummy object.
>-// TODO(deadbeef): Change this when possible.
>-TEST_F(OrtcFactoryIntegrationTest, SetTrackWhileSending) {
>-  auto udp_transports = CreateAndConnectUdpTransportPair();
>-  auto rtp_transports =
>-      CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
>-
>-  auto sender_result = ortc_factory1_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get());
>-  auto receiver_result = ortc_factory2_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get());
>-  ASSERT_TRUE(sender_result.ok());
>-  ASSERT_TRUE(receiver_result.ok());
>-  auto sender = sender_result.MoveValue();
>-  auto receiver = receiver_result.MoveValue();
>-
>-  RTCError error = sender->SetTrack(
>-      CreateLocalVideoTrackAndFakeSource("video_1", ortc_factory1_.get()));
>-  EXPECT_TRUE(error.ok());
>-  RtpParameters vp8_parameters = MakeMinimalVp8Parameters();
>-  EXPECT_TRUE(receiver->Receive(vp8_parameters).ok());
>-  EXPECT_TRUE(sender->Send(vp8_parameters).ok());
>-  FakeVideoTrackRenderer fake_renderer(
>-      static_cast<VideoTrackInterface*>(receiver->GetTrack().get()));
>-  // Expect for some initial number of frames to be received.
>-  EXPECT_TRUE_WAIT(fake_renderer.num_rendered_frames() > kDefaultNumFrames,
>-                   kDefaultTimeout);
>-  // Destroy old source, set a new track, and verify new frames are received
>-  // from the new track. The VideoTrackSource is reference counted and may live
>-  // a little longer, so tell it that its source is going away now.
>-  fake_video_track_sources_[0] = nullptr;
>-  int prev_num_frames = fake_renderer.num_rendered_frames();
>-  error = sender->SetTrack(
>-      CreateLocalVideoTrackAndFakeSource("video_2", ortc_factory1_.get()));
>-  EXPECT_TRUE(error.ok());
>-  EXPECT_TRUE_WAIT(
>-      fake_renderer.num_rendered_frames() > kDefaultNumFrames + prev_num_frames,
>-      kDefaultTimeout);
>-}
>-
>-// TODO(webrtc:7915, webrtc:9184): Tests below are disabled for iOS 64 on debug
>-// builds because of flakiness.
>-#if !(defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_64_BITS) && !defined(NDEBUG))
>-#define MAYBE_BasicTwoWayAudioVideoRtpSendersAndReceivers \
>-  BasicTwoWayAudioVideoRtpSendersAndReceivers
>-#define MAYBE_BasicTwoWayAudioVideoSrtpSendersAndReceivers \
>-  BasicTwoWayAudioVideoSrtpSendersAndReceivers
>-#define MAYBE_SrtpSendersAndReceiversWithMismatchingKeys \
>-  SrtpSendersAndReceiversWithMismatchingKeys
>-#define MAYBE_OneSideSrtpSenderAndReceiver OneSideSrtpSenderAndReceiver
>-#define MAYBE_FullTwoWayAudioVideoSrtpSendersAndReceivers \
>-  FullTwoWayAudioVideoSrtpSendersAndReceivers
>-#else
>-#define MAYBE_BasicTwoWayAudioVideoRtpSendersAndReceivers \
>-  DISABLED_BasicTwoWayAudioVideoRtpSendersAndReceivers
>-#define MAYBE_BasicTwoWayAudioVideoSrtpSendersAndReceivers \
>-  DISABLED_BasicTwoWayAudioVideoSrtpSendersAndReceivers
>-#define MAYBE_SrtpSendersAndReceiversWithMismatchingKeys \
>-  DISABLED_SrtpSendersAndReceiversWithMismatchingKeys
>-#define MAYBE_OneSideSrtpSenderAndReceiver DISABLED_OneSideSrtpSenderAndReceiver
>-#define MAYBE_FullTwoWayAudioVideoSrtpSendersAndReceivers \
>-  DISABLED_FullTwoWayAudioVideoSrtpSendersAndReceivers
>-#endif
>-
>-// End-to-end test with two pairs of RTP senders and receivers, for audio and
>-// video.
>-//
>-// Uses muxed RTCP, and minimal parameters with hard-coded configs that are
>-// known to work.
>-TEST_F(OrtcFactoryIntegrationTest,
>-       MAYBE_BasicTwoWayAudioVideoRtpSendersAndReceivers) {
>-  auto udp_transports = CreateAndConnectUdpTransportPair();
>-  auto rtp_transports =
>-      CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
>-  bool expect_success = true;
>-  BasicTwoWayRtpSendersAndReceiversTest(std::move(rtp_transports),
>-                                        expect_success);
>-}
>-
>-TEST_F(OrtcFactoryIntegrationTest,
>-       MAYBE_BasicTwoWayAudioVideoSrtpSendersAndReceivers) {
>-  auto udp_transports = CreateAndConnectUdpTransportPair();
>-  auto srtp_transports = CreateSrtpTransportPairAndSetKeys(
>-      MakeRtcpMuxParameters(), udp_transports);
>-  bool expect_success = true;
>-  BasicTwoWayRtpSendersAndReceiversTest(std::move(srtp_transports),
>-                                        expect_success);
>-}
>-
>-// Tests that the packets cannot be decoded if the keys are mismatched.
>-// TODO(webrtc:9184): Disabled because this test is flaky.
>-TEST_F(OrtcFactoryIntegrationTest,
>-       MAYBE_SrtpSendersAndReceiversWithMismatchingKeys) {
>-  auto udp_transports = CreateAndConnectUdpTransportPair();
>-  auto srtp_transports = CreateSrtpTransportPairAndSetMismatchingKeys(
>-      MakeRtcpMuxParameters(), udp_transports);
>-  bool expect_success = false;
>-  BasicTwoWayRtpSendersAndReceiversTest(std::move(srtp_transports),
>-                                        expect_success);
>-  // No frames are expected to be decoded.
>-  EXPECT_TRUE(received_audio_frames1_ == 0 && received_audio_frames2_ == 0 &&
>-              rendered_video_frames1_ == 0 && rendered_video_frames2_ == 0);
>-}
>-
>-// Tests that the frames cannot be decoded if only one side uses SRTP.
>-TEST_F(OrtcFactoryIntegrationTest, MAYBE_OneSideSrtpSenderAndReceiver) {
>-  auto rtcp_parameters = MakeRtcpMuxParameters();
>-  auto udp_transports = CreateAndConnectUdpTransportPair();
>-  auto rtcp_udp_transports = UdpTransportPair();
>-  auto transport_controllers = RtpTransportControllerPair();
>-  auto transport_result1 = ortc_factory1_->CreateRtpTransport(
>-      rtcp_parameters, udp_transports.first.get(),
>-      rtcp_udp_transports.first.get(), transport_controllers.first.get());
>-  auto transport_result2 = ortc_factory2_->CreateSrtpTransport(
>-      rtcp_parameters, udp_transports.second.get(),
>-      rtcp_udp_transports.second.get(), transport_controllers.second.get());
>-
>-  auto rtp_transport = transport_result1.MoveValue();
>-  auto srtp_transport = transport_result2.MoveValue();
>-  EXPECT_TRUE(srtp_transport->SetSrtpSendKey(kTestCryptoParams1).ok());
>-  EXPECT_TRUE(srtp_transport->SetSrtpReceiveKey(kTestCryptoParams2).ok());
>-  bool expect_success = false;
>-  BasicTwoWayRtpSendersAndReceiversTest(
>-      {std::move(rtp_transport), std::move(srtp_transport)}, expect_success);
>-
>-  // The SRTP side is not expected to decode any audio or video frames.
>-  // The RTP side is not expected to decode any video frames while it is
>-  // possible that the encrypted audio frames can be accidentally decoded which
>-  // is why received_audio_frames1_ is not validated.
>-  EXPECT_TRUE(received_audio_frames2_ == 0 && rendered_video_frames1_ == 0 &&
>-              rendered_video_frames2_ == 0);
>-}
>-
>-// End-to-end test with two pairs of RTP senders and receivers, for audio and
>-// video. Unlike the test above, this attempts to make the parameters as
>-// complex as possible. The senders and receivers use the SRTP transport with
>-// different keys.
>-//
>-// Uses non-muxed RTCP, with separate audio/video transports, and a full set of
>-// parameters, as would normally be used in a PeerConnection.
>-//
>-// TODO(deadbeef): Update this test as more audio/video features become
>-// supported.
>-TEST_F(OrtcFactoryIntegrationTest,
>-       MAYBE_FullTwoWayAudioVideoSrtpSendersAndReceivers) {
>-  // We want four pairs of UDP transports for this test, for audio/video and
>-  // RTP/RTCP.
>-  auto audio_rtp_udp_transports = CreateAndConnectUdpTransportPair();
>-  auto audio_rtcp_udp_transports = CreateAndConnectUdpTransportPair();
>-  auto video_rtp_udp_transports = CreateAndConnectUdpTransportPair();
>-  auto video_rtcp_udp_transports = CreateAndConnectUdpTransportPair();
>-
>-  // Since we have multiple RTP transports on each side, we need an RTP
>-  // transport controller.
>-  auto transport_controllers = CreateRtpTransportControllerPair();
>-
>-  RtpTransportParameters audio_rtp_transport_parameters;
>-  audio_rtp_transport_parameters.rtcp.mux = false;
>-  auto audio_srtp_transports = CreateSrtpTransportPair(
>-      audio_rtp_transport_parameters, audio_rtp_udp_transports,
>-      audio_rtcp_udp_transports, transport_controllers);
>-
>-  RtpTransportParameters video_rtp_transport_parameters;
>-  video_rtp_transport_parameters.rtcp.mux = false;
>-  video_rtp_transport_parameters.rtcp.reduced_size = true;
>-  auto video_srtp_transports = CreateSrtpTransportPair(
>-      video_rtp_transport_parameters, video_rtp_udp_transports,
>-      video_rtcp_udp_transports, transport_controllers);
>-
>-  // Set keys for SRTP transports.
>-  audio_srtp_transports.first->SetSrtpSendKey(kTestCryptoParams1);
>-  audio_srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams2);
>-  video_srtp_transports.first->SetSrtpSendKey(kTestCryptoParams3);
>-  video_srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams4);
>-
>-  audio_srtp_transports.second->SetSrtpSendKey(kTestCryptoParams2);
>-  audio_srtp_transports.second->SetSrtpReceiveKey(kTestCryptoParams1);
>-  video_srtp_transports.second->SetSrtpSendKey(kTestCryptoParams4);
>-  video_srtp_transports.second->SetSrtpReceiveKey(kTestCryptoParams3);
>-
>-  // Create all the senders and receivers (four per endpoint).
>-  auto audio_sender_result1 = ortc_factory1_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, audio_srtp_transports.first.get());
>-  auto video_sender_result1 = ortc_factory1_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, video_srtp_transports.first.get());
>-  auto audio_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, audio_srtp_transports.first.get());
>-  auto video_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, video_srtp_transports.first.get());
>-  ASSERT_TRUE(audio_sender_result1.ok());
>-  ASSERT_TRUE(video_sender_result1.ok());
>-  ASSERT_TRUE(audio_receiver_result1.ok());
>-  ASSERT_TRUE(video_receiver_result1.ok());
>-  auto audio_sender1 = audio_sender_result1.MoveValue();
>-  auto video_sender1 = video_sender_result1.MoveValue();
>-  auto audio_receiver1 = audio_receiver_result1.MoveValue();
>-  auto video_receiver1 = video_receiver_result1.MoveValue();
>-
>-  auto audio_sender_result2 = ortc_factory2_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, audio_srtp_transports.second.get());
>-  auto video_sender_result2 = ortc_factory2_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, video_srtp_transports.second.get());
>-  auto audio_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, audio_srtp_transports.second.get());
>-  auto video_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, video_srtp_transports.second.get());
>-  ASSERT_TRUE(audio_sender_result2.ok());
>-  ASSERT_TRUE(video_sender_result2.ok());
>-  ASSERT_TRUE(audio_receiver_result2.ok());
>-  ASSERT_TRUE(video_receiver_result2.ok());
>-  auto audio_sender2 = audio_sender_result2.MoveValue();
>-  auto video_sender2 = video_sender_result2.MoveValue();
>-  auto audio_receiver2 = audio_receiver_result2.MoveValue();
>-  auto video_receiver2 = video_receiver_result2.MoveValue();
>-
>-  RTCError error = audio_sender1->SetTrack(
>-      CreateLocalAudioTrack("audio", ortc_factory1_.get()));
>-  EXPECT_TRUE(error.ok());
>-  error = video_sender1->SetTrack(
>-      CreateLocalVideoTrackAndFakeSource("video", ortc_factory1_.get()));
>-  EXPECT_TRUE(error.ok());
>-  error = audio_sender2->SetTrack(
>-      CreateLocalAudioTrack("audio", ortc_factory2_.get()));
>-  EXPECT_TRUE(error.ok());
>-  error = video_sender2->SetTrack(
>-      CreateLocalVideoTrackAndFakeSource("video", ortc_factory2_.get()));
>-  EXPECT_TRUE(error.ok());
>-
>-  // Use different codecs in different directions for extra challenge.
>-  RtpParameters opus_send_parameters = MakeFullOpusParameters();
>-  RtpParameters isac_send_parameters = MakeFullIsacParameters();
>-  RtpParameters vp8_send_parameters = MakeFullVp8Parameters();
>-  RtpParameters vp9_send_parameters = MakeFullVp9Parameters();
>-
>-  // Remove "payload_type" from receive parameters. Receiver will need to
>-  // discern the payload type from packets received.
>-  RtpParameters opus_receive_parameters = opus_send_parameters;
>-  RtpParameters isac_receive_parameters = isac_send_parameters;
>-  RtpParameters vp8_receive_parameters = vp8_send_parameters;
>-  RtpParameters vp9_receive_parameters = vp9_send_parameters;
>-  opus_receive_parameters.encodings[0].codec_payload_type.reset();
>-  isac_receive_parameters.encodings[0].codec_payload_type.reset();
>-  vp8_receive_parameters.encodings[0].codec_payload_type.reset();
>-  vp9_receive_parameters.encodings[0].codec_payload_type.reset();
>-
>-  // Configure the senders' and receivers' parameters.
>-  //
>-  // Note: Intentionally, the top codec in the receive parameters does not
>-  // match the codec sent by the other side. If "Receive" is called with a list
>-  // of codecs, the receiver should be prepared to receive any of them, not
>-  // just the one on top.
>-  EXPECT_TRUE(audio_receiver1->Receive(opus_receive_parameters).ok());
>-  EXPECT_TRUE(video_receiver1->Receive(vp8_receive_parameters).ok());
>-  EXPECT_TRUE(audio_receiver2->Receive(isac_receive_parameters).ok());
>-  EXPECT_TRUE(video_receiver2->Receive(vp9_receive_parameters).ok());
>-  EXPECT_TRUE(audio_sender1->Send(opus_send_parameters).ok());
>-  EXPECT_TRUE(video_sender1->Send(vp8_send_parameters).ok());
>-  EXPECT_TRUE(audio_sender2->Send(isac_send_parameters).ok());
>-  EXPECT_TRUE(video_sender2->Send(vp9_send_parameters).ok());
>-
>-  FakeVideoTrackRenderer fake_video_renderer1(
>-      static_cast<VideoTrackInterface*>(video_receiver1->GetTrack().get()));
>-  FakeVideoTrackRenderer fake_video_renderer2(
>-      static_cast<VideoTrackInterface*>(video_receiver2->GetTrack().get()));
>-
>-  // Senders and receivers are connected and configured; audio and video frames
>-  // should be able to flow at this point.
>-  EXPECT_TRUE_WAIT(
>-      fake_audio_capture_module1_->frames_received() > kDefaultNumFrames &&
>-          fake_video_renderer1.num_rendered_frames() > kDefaultNumFrames &&
>-          fake_audio_capture_module2_->frames_received() > kDefaultNumFrames &&
>-          fake_video_renderer2.num_rendered_frames() > kDefaultNumFrames,
>-      kDefaultTimeout);
>-}
>-
>-// TODO(deadbeef): End-to-end test for multiple senders/receivers of the same
>-// media type, once that's supported. Currently, it is not because the
>-// BaseChannel model relies on there being a single VoiceChannel and
>-// VideoChannel, and these only support a single set of codecs/etc. per
>-// send/receive direction.
>-
>-// TODO(deadbeef): End-to-end test for simulcast, once that's supported by this
>-// API.
>-
>-#endif  // if !defined(THREAD_SANITIZER)
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory_unittest.cc
>deleted file mode 100644
>index 40afbd328b6dfe176a68e11b2323c84e33593855..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory_unittest.cc
>+++ /dev/null
>@@ -1,250 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-
>-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>-#include "media/base/fakemediaengine.h"
>-#include "ortc/ortcfactory.h"
>-#include "ortc/testrtpparameters.h"
>-#include "p2p/base/fakepackettransport.h"
>-#include "rtc_base/fakenetwork.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/virtualsocketserver.h"
>-
>-namespace webrtc {
>-
>-// This test uses a virtual network and fake media engine, in order to test the
>-// OrtcFactory at only an API level. Any end-to-end test should go in
>-// ortcfactory_integrationtest.cc instead.
>-class OrtcFactoryTest : public testing::Test {
>- public:
>-  OrtcFactoryTest()
>-      : thread_(&virtual_socket_server_),
>-        fake_packet_transport_("fake transport") {
>-    ortc_factory_ =
>-        OrtcFactory::Create(&thread_, nullptr, &fake_network_manager_, nullptr,
>-                            nullptr,
>-                            std::unique_ptr<cricket::MediaEngineInterface>(
>-                                new cricket::FakeMediaEngine()),
>-                            CreateBuiltinAudioEncoderFactory(),
>-                            CreateBuiltinAudioDecoderFactory())
>-            .MoveValue();
>-  }
>-
>- protected:
>-  // Uses a single pre-made FakePacketTransport, so shouldn't be called twice in
>-  // the same test.
>-  std::unique_ptr<RtpTransportInterface>
>-  CreateRtpTransportWithFakePacketTransport() {
>-    return ortc_factory_
>-        ->CreateRtpTransport(MakeRtcpMuxParameters(), &fake_packet_transport_,
>-                             nullptr, nullptr)
>-        .MoveValue();
>-  }
>-
>-  rtc::VirtualSocketServer virtual_socket_server_;
>-  rtc::AutoSocketServerThread thread_;
>-  rtc::FakeNetworkManager fake_network_manager_;
>-  rtc::FakePacketTransport fake_packet_transport_;
>-  std::unique_ptr<OrtcFactoryInterface> ortc_factory_;
>-};
>-
>-TEST_F(OrtcFactoryTest, CanCreateMultipleRtpTransportControllers) {
>-  auto controller_result1 = ortc_factory_->CreateRtpTransportController();
>-  EXPECT_TRUE(controller_result1.ok());
>-  auto controller_result2 = ortc_factory_->CreateRtpTransportController();
>-  EXPECT_TRUE(controller_result1.ok());
>-}
>-
>-// Simple test for the successful cases of CreateRtpTransport.
>-TEST_F(OrtcFactoryTest, CreateRtpTransportWithAndWithoutMux) {
>-  rtc::FakePacketTransport rtp("rtp");
>-  rtc::FakePacketTransport rtcp("rtcp");
>-  // With muxed RTCP.
>-  RtpTransportParameters parameters = MakeRtcpMuxParameters();
>-  auto result =
>-      ortc_factory_->CreateRtpTransport(parameters, &rtp, nullptr, nullptr);
>-  EXPECT_TRUE(result.ok());
>-  result.MoveValue().reset();
>-  // With non-muxed RTCP.
>-  parameters.rtcp.mux = false;
>-  result = ortc_factory_->CreateRtpTransport(parameters, &rtp, &rtcp, nullptr);
>-  EXPECT_TRUE(result.ok());
>-}
>-
>-// Simple test for the successful cases of CreateSrtpTransport.
>-TEST_F(OrtcFactoryTest, CreateSrtpTransport) {
>-  rtc::FakePacketTransport rtp("rtp");
>-  rtc::FakePacketTransport rtcp("rtcp");
>-  // With muxed RTCP.
>-  RtpTransportParameters parameters = MakeRtcpMuxParameters();
>-  auto result =
>-      ortc_factory_->CreateSrtpTransport(parameters, &rtp, nullptr, nullptr);
>-  EXPECT_TRUE(result.ok());
>-  result.MoveValue().reset();
>-  // With non-muxed RTCP.
>-  parameters.rtcp.mux = false;
>-  result = ortc_factory_->CreateSrtpTransport(parameters, &rtp, &rtcp, nullptr);
>-  EXPECT_TRUE(result.ok());
>-}
>-
>-// If no CNAME is provided, one should be generated and returned by
>-// GetRtpParameters.
>-TEST_F(OrtcFactoryTest, CreateRtpTransportGeneratesCname) {
>-  rtc::FakePacketTransport rtp("rtp");
>-  auto result = ortc_factory_->CreateRtpTransport(MakeRtcpMuxParameters(), &rtp,
>-                                                  nullptr, nullptr);
>-  ASSERT_TRUE(result.ok());
>-  EXPECT_FALSE(result.value()->GetParameters().rtcp.cname.empty());
>-}
>-
>-// Extension of the above test; multiple transports created by the same factory
>-// should use the same generated CNAME.
>-TEST_F(OrtcFactoryTest, MultipleRtpTransportsUseSameGeneratedCname) {
>-  rtc::FakePacketTransport packet_transport1("1");
>-  rtc::FakePacketTransport packet_transport2("2");
>-  RtpTransportParameters parameters = MakeRtcpMuxParameters();
>-  // Sanity check.
>-  ASSERT_TRUE(parameters.rtcp.cname.empty());
>-  auto result = ortc_factory_->CreateRtpTransport(
>-      parameters, &packet_transport1, nullptr, nullptr);
>-  ASSERT_TRUE(result.ok());
>-  auto rtp_transport1 = result.MoveValue();
>-  result = ortc_factory_->CreateRtpTransport(parameters, &packet_transport2,
>-                                             nullptr, nullptr);
>-  ASSERT_TRUE(result.ok());
>-  auto rtp_transport2 = result.MoveValue();
>-  RtcpParameters params1 = rtp_transport1->GetParameters().rtcp;
>-  RtcpParameters params2 = rtp_transport2->GetParameters().rtcp;
>-  EXPECT_FALSE(params1.cname.empty());
>-  EXPECT_EQ(params1.cname, params2.cname);
>-}
>-
>-TEST_F(OrtcFactoryTest, CreateRtpTransportWithNoPacketTransport) {
>-  auto result = ortc_factory_->CreateRtpTransport(MakeRtcpMuxParameters(),
>-                                                  nullptr, nullptr, nullptr);
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
>-}
>-
>-// If the |mux| member of the RtcpParameters is false, both an RTP and RTCP
>-// packet transport are needed.
>-TEST_F(OrtcFactoryTest, CreateRtpTransportWithMissingRtcpTransport) {
>-  rtc::FakePacketTransport rtp("rtp");
>-  RtpTransportParameters parameters;
>-  parameters.rtcp.mux = false;
>-  auto result =
>-      ortc_factory_->CreateRtpTransport(parameters, &rtp, nullptr, nullptr);
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
>-}
>-
>-// If the |mux| member of the RtcpParameters is true, only an RTP packet
>-// transport is necessary. So, passing in an RTCP transport is most likely
>-// an accident, and thus should be treated as an error.
>-TEST_F(OrtcFactoryTest, CreateRtpTransportWithExtraneousRtcpTransport) {
>-  rtc::FakePacketTransport rtp("rtp");
>-  rtc::FakePacketTransport rtcp("rtcp");
>-  auto result = ortc_factory_->CreateRtpTransport(MakeRtcpMuxParameters(), &rtp,
>-                                                  &rtcp, nullptr);
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
>-}
>-
>-// Basic test that CreateUdpTransport works with AF_INET and AF_INET6.
>-TEST_F(OrtcFactoryTest, CreateUdpTransport) {
>-  auto result = ortc_factory_->CreateUdpTransport(AF_INET);
>-  EXPECT_TRUE(result.ok());
>-  result = ortc_factory_->CreateUdpTransport(AF_INET6);
>-  EXPECT_TRUE(result.ok());
>-}
>-
>-// Test CreateUdpPort with the |min_port| and |max_port| arguments.
>-TEST_F(OrtcFactoryTest, CreateUdpTransportWithPortRange) {
>-  auto socket_result1 = ortc_factory_->CreateUdpTransport(AF_INET, 2000, 2002);
>-  ASSERT_TRUE(socket_result1.ok());
>-  EXPECT_EQ(2000, socket_result1.value()->GetLocalAddress().port());
>-  auto socket_result2 = ortc_factory_->CreateUdpTransport(AF_INET, 2000, 2002);
>-  ASSERT_TRUE(socket_result2.ok());
>-  EXPECT_EQ(2001, socket_result2.value()->GetLocalAddress().port());
>-  auto socket_result3 = ortc_factory_->CreateUdpTransport(AF_INET, 2000, 2002);
>-  ASSERT_TRUE(socket_result3.ok());
>-  EXPECT_EQ(2002, socket_result3.value()->GetLocalAddress().port());
>-
>-  // All sockets in the range have been exhausted, so the next call should
>-  // fail.
>-  auto failed_result = ortc_factory_->CreateUdpTransport(AF_INET, 2000, 2002);
>-  EXPECT_EQ(RTCErrorType::RESOURCE_EXHAUSTED, failed_result.error().type());
>-
>-  // If one socket is destroyed, that port should be freed up again.
>-  socket_result2.MoveValue().reset();
>-  auto socket_result4 = ortc_factory_->CreateUdpTransport(AF_INET, 2000, 2002);
>-  ASSERT_TRUE(socket_result4.ok());
>-  EXPECT_EQ(2001, socket_result4.value()->GetLocalAddress().port());
>-}
>-
>-// Basic test that CreateUdpTransport works with AF_INET and AF_INET6.
>-TEST_F(OrtcFactoryTest, CreateUdpTransportWithInvalidAddressFamily) {
>-  auto result = ortc_factory_->CreateUdpTransport(12345);
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
>-}
>-
>-TEST_F(OrtcFactoryTest, CreateUdpTransportWithInvalidPortRange) {
>-  auto result = ortc_factory_->CreateUdpTransport(AF_INET, 3000, 2000);
>-  EXPECT_EQ(RTCErrorType::INVALID_RANGE, result.error().type());
>-}
>-
>-// Just sanity check that each "GetCapabilities" method returns some codecs.
>-TEST_F(OrtcFactoryTest, GetSenderAndReceiverCapabilities) {
>-  RtpCapabilities audio_send_caps =
>-      ortc_factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO);
>-  EXPECT_GT(audio_send_caps.codecs.size(), 0u);
>-  RtpCapabilities video_send_caps =
>-      ortc_factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
>-  EXPECT_GT(video_send_caps.codecs.size(), 0u);
>-  RtpCapabilities audio_receive_caps =
>-      ortc_factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_AUDIO);
>-  EXPECT_GT(audio_receive_caps.codecs.size(), 0u);
>-  RtpCapabilities video_receive_caps =
>-      ortc_factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
>-  EXPECT_GT(video_receive_caps.codecs.size(), 0u);
>-}
>-
>-// Calling CreateRtpSender with a null track should fail, since that makes it
>-// impossible to know whether to create an audio or video sender. The
>-// application should be using the method that takes a cricket::MediaType
>-// instead.
>-TEST_F(OrtcFactoryTest, CreateSenderWithNullTrack) {
>-  auto rtp_transport = CreateRtpTransportWithFakePacketTransport();
>-  auto result = ortc_factory_->CreateRtpSender(nullptr, rtp_transport.get());
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
>-}
>-
>-// Calling CreateRtpSender or CreateRtpReceiver with MEDIA_TYPE_DATA should
>-// fail.
>-TEST_F(OrtcFactoryTest, CreateSenderOrReceieverWithInvalidKind) {
>-  auto rtp_transport = CreateRtpTransportWithFakePacketTransport();
>-  auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_DATA,
>-                                                      rtp_transport.get());
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, sender_result.error().type());
>-  auto receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_DATA, rtp_transport.get());
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, receiver_result.error().type());
>-}
>-
>-TEST_F(OrtcFactoryTest, CreateSendersOrReceieversWithNullTransport) {
>-  auto sender_result =
>-      ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO, nullptr);
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, sender_result.error().type());
>-  auto receiver_result =
>-      ortc_factory_->CreateRtpReceiver(cricket::MEDIA_TYPE_AUDIO, nullptr);
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, receiver_result.error().type());
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiver_unittest.cc
>deleted file mode 100644
>index f0e92387145eac64a9f6d808f24efafa9fea35df..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiver_unittest.cc
>+++ /dev/null
>@@ -1,550 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-
>-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>-#include "media/base/fakemediaengine.h"
>-#include "ortc/ortcfactory.h"
>-#include "ortc/testrtpparameters.h"
>-#include "p2p/base/fakepackettransport.h"
>-#include "pc/test/fakevideotracksource.h"
>-#include "rtc_base/gunit.h"
>-
>-namespace webrtc {
>-
>-// This test uses an individual RtpReceiver using only the public interface,
>-// and verifies that it behaves as designed at an API level. Also tests that
>-// parameters are applied to the audio/video engines as expected. Network and
>-// media interfaces are faked to isolate what's being tested.
>-//
>-// This test shouldn't result any any actual media being sent. That sort of
>-// test should go in ortcfactory_integrationtest.cc.
>-class OrtcRtpReceiverTest : public testing::Test {
>- public:
>-  OrtcRtpReceiverTest() : fake_packet_transport_("fake") {
>-    fake_media_engine_ = new cricket::FakeMediaEngine();
>-    // Note: This doesn't need to use fake network classes, since we already
>-    // use FakePacketTransport.
>-    auto ortc_factory_result = OrtcFactory::Create(
>-        nullptr, nullptr, nullptr, nullptr, nullptr,
>-        std::unique_ptr<cricket::MediaEngineInterface>(fake_media_engine_),
>-        CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory());
>-    ortc_factory_ = ortc_factory_result.MoveValue();
>-    RtpTransportParameters parameters;
>-    parameters.rtcp.mux = true;
>-    auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
>-        parameters, &fake_packet_transport_, nullptr, nullptr);
>-    rtp_transport_ = rtp_transport_result.MoveValue();
>-  }
>-
>- protected:
>-  // Owned by |ortc_factory_|.
>-  cricket::FakeMediaEngine* fake_media_engine_;
>-  rtc::FakePacketTransport fake_packet_transport_;
>-  std::unique_ptr<OrtcFactoryInterface> ortc_factory_;
>-  std::unique_ptr<RtpTransportInterface> rtp_transport_;
>-};
>-
>-// See ortcrtpreceiverinterface.h for the current expectations of what GetTrack
>-// will return after calls to Receive.
>-// TODO(deadbeef): Replace this test when the non-standard behavior is fixed
>-// and GetTrack starts returning the same track for the lifetime of the
>-// receiver.
>-TEST_F(OrtcRtpReceiverTest, GetTrack) {
>-  auto receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  ASSERT_TRUE(receiver_result.ok());
>-  auto receiver = receiver_result.MoveValue();
>-
>-  // Track initially expected to be null.
>-  EXPECT_EQ(nullptr, receiver_result.value().get());
>-
>-  EXPECT_TRUE(receiver->Receive(MakeMinimalVp8ParametersWithNoSsrc()).ok());
>-  auto initial_track = receiver->GetTrack();
>-  EXPECT_NE(nullptr, initial_track);
>-
>-  // Codec changing but SSRC (or lack thereof) isn't; shouldn't create new track
>-  EXPECT_TRUE(receiver->Receive(MakeMinimalVp9ParametersWithNoSsrc()).ok());
>-  EXPECT_EQ(initial_track, receiver->GetTrack());
>-
>-  // Explicitly set SSRC and expect a different track.
>-  EXPECT_TRUE(
>-      receiver->Receive(MakeMinimalVp9ParametersWithSsrc(0xdeadbeef)).ok());
>-  auto next_track = receiver->GetTrack();
>-  EXPECT_NE(next_track, initial_track);
>-
>-  // Deactivating the encoding shouldn't change the track.
>-  RtpParameters inactive_encoding =
>-      MakeMinimalVp9ParametersWithSsrc(0xdeadbeef);
>-  inactive_encoding.encodings[0].active = false;
>-  EXPECT_TRUE(receiver->Receive(inactive_encoding).ok());
>-  EXPECT_EQ(next_track, receiver->GetTrack());
>-
>-  // Removing all encodings *is* expected to clear the track.
>-  RtpParameters no_encodings = MakeMinimalVp9ParametersWithSsrc(0xdeadbeef);
>-  no_encodings.encodings.clear();
>-  EXPECT_TRUE(receiver->Receive(no_encodings).ok());
>-  EXPECT_EQ(nullptr, receiver->GetTrack());
>-}
>-
>-// Currently SetTransport isn't supported. When it is, replace this test with a
>-// test/tests for it.
>-TEST_F(OrtcRtpReceiverTest, SetTransportFails) {
>-  rtc::FakePacketTransport fake_packet_transport("another_transport");
>-  RtpTransportParameters parameters;
>-  parameters.rtcp.mux = true;
>-  auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
>-      parameters, &fake_packet_transport, nullptr, nullptr);
>-  auto rtp_transport = rtp_transport_result.MoveValue();
>-
>-  auto receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto receiver = receiver_result.MoveValue();
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            receiver->SetTransport(rtp_transport.get()).type());
>-}
>-
>-TEST_F(OrtcRtpReceiverTest, GetTransport) {
>-  auto result = ortc_factory_->CreateRtpReceiver(cricket::MEDIA_TYPE_AUDIO,
>-                                                 rtp_transport_.get());
>-  EXPECT_EQ(rtp_transport_.get(), result.value()->GetTransport());
>-}
>-
>-// Test that "Receive" causes the expected parameters to be applied to the media
>-// engine level, for an audio receiver.
>-TEST_F(OrtcRtpReceiverTest, ReceiveAppliesAudioParametersToMediaEngine) {
>-  auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_receiver = audio_receiver_result.MoveValue();
>-
>-  // First, create parameters with all the bells and whistles.
>-  RtpParameters parameters;
>-
>-  RtpCodecParameters opus_codec;
>-  opus_codec.name = "opus";
>-  opus_codec.kind = cricket::MEDIA_TYPE_AUDIO;
>-  opus_codec.payload_type = 120;
>-  opus_codec.clock_rate.emplace(48000);
>-  opus_codec.num_channels.emplace(2);
>-  opus_codec.parameters["minptime"] = "10";
>-  opus_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::TRANSPORT_CC);
>-  parameters.codecs.push_back(std::move(opus_codec));
>-
>-  // Add two codecs, expecting the first to be used.
>-  // TODO(deadbeef): Once "codec_payload_type" is supported, use it to select a
>-  // codec that's not at the top of the list.
>-  RtpCodecParameters isac_codec;
>-  isac_codec.name = "ISAC";
>-  isac_codec.kind = cricket::MEDIA_TYPE_AUDIO;
>-  isac_codec.payload_type = 110;
>-  isac_codec.clock_rate.emplace(16000);
>-  parameters.codecs.push_back(std::move(isac_codec));
>-
>-  RtpEncodingParameters encoding;
>-  encoding.ssrc.emplace(0xdeadbeef);
>-  parameters.encodings.push_back(std::move(encoding));
>-
>-  parameters.header_extensions.emplace_back(
>-      "urn:ietf:params:rtp-hdrext:ssrc-audio-level", 3);
>-
>-  EXPECT_TRUE(audio_receiver->Receive(parameters).ok());
>-
>-  // Now verify that the parameters were applied to the fake media engine layer
>-  // that exists below BaseChannel.
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  EXPECT_TRUE(fake_voice_channel->playout());
>-
>-  // Verify codec parameters.
>-  ASSERT_GT(fake_voice_channel->recv_codecs().size(), 0u);
>-  const cricket::AudioCodec& top_codec = fake_voice_channel->recv_codecs()[0];
>-  EXPECT_EQ("opus", top_codec.name);
>-  EXPECT_EQ(120, top_codec.id);
>-  EXPECT_EQ(48000, top_codec.clockrate);
>-  EXPECT_EQ(2u, top_codec.channels);
>-  ASSERT_NE(top_codec.params.end(), top_codec.params.find("minptime"));
>-  EXPECT_EQ("10", top_codec.params.at("minptime"));
>-
>-  // Verify encoding parameters.
>-  ASSERT_EQ(1u, fake_voice_channel->recv_streams().size());
>-  const cricket::StreamParams& recv_stream =
>-      fake_voice_channel->recv_streams()[0];
>-  EXPECT_EQ(1u, recv_stream.ssrcs.size());
>-  EXPECT_EQ(0xdeadbeef, recv_stream.first_ssrc());
>-
>-  // Verify header extensions.
>-  ASSERT_EQ(1u, fake_voice_channel->recv_extensions().size());
>-  const RtpExtension& extension = fake_voice_channel->recv_extensions()[0];
>-  EXPECT_EQ("urn:ietf:params:rtp-hdrext:ssrc-audio-level", extension.uri);
>-  EXPECT_EQ(3, extension.id);
>-}
>-
>-// Test that "Receive" causes the expected parameters to be applied to the media
>-// engine level, for a video receiver.
>-TEST_F(OrtcRtpReceiverTest, ReceiveAppliesVideoParametersToMediaEngine) {
>-  auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_receiver = video_receiver_result.MoveValue();
>-
>-  // First, create parameters with all the bells and whistles.
>-  RtpParameters parameters;
>-
>-  RtpCodecParameters vp8_codec;
>-  vp8_codec.name = "VP8";
>-  vp8_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp8_codec.payload_type = 99;
>-  // Try a couple types of feedback params. "Generic NACK" is a bit of a
>-  // special case, so test it here.
>-  vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::CCM,
>-                                       RtcpFeedbackMessageType::FIR);
>-  vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::NACK,
>-                                       RtcpFeedbackMessageType::GENERIC_NACK);
>-  parameters.codecs.push_back(std::move(vp8_codec));
>-
>-  RtpCodecParameters vp8_rtx_codec;
>-  vp8_rtx_codec.name = "rtx";
>-  vp8_rtx_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp8_rtx_codec.payload_type = 100;
>-  vp8_rtx_codec.parameters["apt"] = "99";
>-  parameters.codecs.push_back(std::move(vp8_rtx_codec));
>-
>-  // Add two codecs, expecting the first to be used.
>-  // TODO(deadbeef): Once "codec_payload_type" is supported, use it to select a
>-  // codec that's not at the top of the list.
>-  RtpCodecParameters vp9_codec;
>-  vp9_codec.name = "VP9";
>-  vp9_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp9_codec.payload_type = 102;
>-  parameters.codecs.push_back(std::move(vp9_codec));
>-
>-  RtpCodecParameters vp9_rtx_codec;
>-  vp9_rtx_codec.name = "rtx";
>-  vp9_rtx_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp9_rtx_codec.payload_type = 103;
>-  vp9_rtx_codec.parameters["apt"] = "102";
>-  parameters.codecs.push_back(std::move(vp9_rtx_codec));
>-
>-  RtpEncodingParameters encoding;
>-  encoding.ssrc.emplace(0xdeadbeef);
>-  encoding.rtx.emplace(0xbaadfeed);
>-  parameters.encodings.push_back(std::move(encoding));
>-
>-  parameters.header_extensions.emplace_back("urn:3gpp:video-orientation", 4);
>-  parameters.header_extensions.emplace_back(
>-      "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", 6);
>-
>-  EXPECT_TRUE(video_receiver->Receive(parameters).ok());
>-
>-  // Now verify that the parameters were applied to the fake media engine layer
>-  // that exists below BaseChannel.
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-
>-  // Verify codec parameters.
>-  ASSERT_GE(fake_video_channel->recv_codecs().size(), 2u);
>-  const cricket::VideoCodec& top_codec = fake_video_channel->recv_codecs()[0];
>-  EXPECT_EQ("VP8", top_codec.name);
>-  EXPECT_EQ(99, top_codec.id);
>-  EXPECT_TRUE(top_codec.feedback_params.Has({"ccm", "fir"}));
>-  EXPECT_TRUE(top_codec.feedback_params.Has(cricket::FeedbackParam("nack")));
>-
>-  const cricket::VideoCodec& rtx_codec = fake_video_channel->recv_codecs()[1];
>-  EXPECT_EQ("rtx", rtx_codec.name);
>-  EXPECT_EQ(100, rtx_codec.id);
>-  ASSERT_NE(rtx_codec.params.end(), rtx_codec.params.find("apt"));
>-  EXPECT_EQ("99", rtx_codec.params.at("apt"));
>-
>-  // Verify encoding parameters.
>-  ASSERT_EQ(1u, fake_video_channel->recv_streams().size());
>-  const cricket::StreamParams& recv_stream =
>-      fake_video_channel->recv_streams()[0];
>-  EXPECT_EQ(2u, recv_stream.ssrcs.size());
>-  EXPECT_EQ(0xdeadbeef, recv_stream.first_ssrc());
>-  uint32_t rtx_ssrc = 0u;
>-  EXPECT_TRUE(recv_stream.GetFidSsrc(recv_stream.first_ssrc(), &rtx_ssrc));
>-  EXPECT_EQ(0xbaadfeed, rtx_ssrc);
>-
>-  // Verify header extensions.
>-  ASSERT_EQ(2u, fake_video_channel->recv_extensions().size());
>-  const RtpExtension& extension1 = fake_video_channel->recv_extensions()[0];
>-  EXPECT_EQ("urn:3gpp:video-orientation", extension1.uri);
>-  EXPECT_EQ(4, extension1.id);
>-  const RtpExtension& extension2 = fake_video_channel->recv_extensions()[1];
>-  EXPECT_EQ("http://www.webrtc.org/experiments/rtp-hdrext/playout-delay",
>-            extension2.uri);
>-  EXPECT_EQ(6, extension2.id);
>-}
>-
>-// Test changing both the receive codec and SSRC at the same time, and verify
>-// that the new parameters are applied to the media engine level.
>-TEST_F(OrtcRtpReceiverTest, CallingReceiveTwiceChangesParameters) {
>-  auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_receiver = audio_receiver_result.MoveValue();
>-  RTCError error =
>-      audio_receiver->Receive(MakeMinimalOpusParametersWithSsrc(0x11111111));
>-  EXPECT_TRUE(error.ok());
>-  error =
>-      audio_receiver->Receive(MakeMinimalIsacParametersWithSsrc(0x22222222));
>-  EXPECT_TRUE(error.ok());
>-
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  ASSERT_GT(fake_voice_channel->recv_codecs().size(), 0u);
>-  EXPECT_EQ("ISAC", fake_voice_channel->recv_codecs()[0].name);
>-  ASSERT_EQ(1u, fake_voice_channel->recv_streams().size());
>-  EXPECT_EQ(0x22222222u, fake_voice_channel->recv_streams()[0].first_ssrc());
>-
>-  auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_receiver = video_receiver_result.MoveValue();
>-  error = video_receiver->Receive(MakeMinimalVp8ParametersWithSsrc(0x33333333));
>-  EXPECT_TRUE(error.ok());
>-  error = video_receiver->Receive(MakeMinimalVp9ParametersWithSsrc(0x44444444));
>-  EXPECT_TRUE(error.ok());
>-
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-  ASSERT_GT(fake_video_channel->recv_codecs().size(), 0u);
>-  EXPECT_EQ("VP9", fake_video_channel->recv_codecs()[0].name);
>-  ASSERT_EQ(1u, fake_video_channel->recv_streams().size());
>-  EXPECT_EQ(0x44444444u, fake_video_channel->recv_streams()[0].first_ssrc());
>-}
>-
>-// Ensure that if the |active| flag of RtpEncodingParameters is set to false,
>-// playout stops at the media engine level. Note that this is only applicable
>-// to audio (at least currently).
>-TEST_F(OrtcRtpReceiverTest, DeactivatingEncodingStopsPlayout) {
>-  auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_receiver = audio_receiver_result.MoveValue();
>-  RtpParameters parameters = MakeMinimalOpusParameters();
>-  EXPECT_TRUE(audio_receiver->Receive(parameters).ok());
>-
>-  // Expect "playout" flag to initially be true.
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  EXPECT_TRUE(fake_voice_channel->playout());
>-
>-  // Deactivate encoding and expect it to change to false.
>-  parameters.encodings[0].active = false;
>-  EXPECT_TRUE(audio_receiver->Receive(parameters).ok());
>-  EXPECT_FALSE(fake_voice_channel->playout());
>-}
>-
>-// Ensure that calling Receive with an empty list of encodings causes receive
>-// streams at the media engine level to be cleared.
>-TEST_F(OrtcRtpReceiverTest,
>-       CallingReceiveWithEmptyEncodingsClearsReceiveStreams) {
>-  auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_receiver = audio_receiver_result.MoveValue();
>-  RtpParameters parameters = MakeMinimalOpusParameters();
>-  EXPECT_TRUE(audio_receiver->Receive(parameters).ok());
>-  parameters.encodings.clear();
>-  EXPECT_TRUE(audio_receiver->Receive(parameters).ok());
>-
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  EXPECT_TRUE(fake_voice_channel->recv_streams().empty());
>-
>-  auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_receiver = video_receiver_result.MoveValue();
>-  parameters = MakeMinimalVp8Parameters();
>-  EXPECT_TRUE(video_receiver->Receive(parameters).ok());
>-  parameters.encodings.clear();
>-  EXPECT_TRUE(video_receiver->Receive(parameters).ok());
>-
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-  EXPECT_TRUE(fake_video_channel->recv_streams().empty());
>-}
>-
>-// These errors should be covered by rtpparametersconversion_unittest.cc, but
>-// we should at least test that those errors are propogated from calls to
>-// Receive, with a few examples.
>-TEST_F(OrtcRtpReceiverTest, ReceiveReturnsErrorOnInvalidParameters) {
>-  auto result = ortc_factory_->CreateRtpReceiver(cricket::MEDIA_TYPE_AUDIO,
>-                                                 rtp_transport_.get());
>-  auto receiver = result.MoveValue();
>-  // CCM feedback missing message type.
>-  RtpParameters invalid_feedback = MakeMinimalOpusParameters();
>-  invalid_feedback.codecs[0].rtcp_feedback.emplace_back(RtcpFeedbackType::CCM);
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            receiver->Receive(invalid_feedback).type());
>-  // Payload type greater than 127.
>-  RtpParameters invalid_pt = MakeMinimalOpusParameters();
>-  invalid_pt.codecs[0].payload_type = 128;
>-  EXPECT_EQ(RTCErrorType::INVALID_RANGE, receiver->Receive(invalid_pt).type());
>-  // Duplicate header extension IDs.
>-  RtpParameters duplicate_ids = MakeMinimalOpusParameters();
>-  duplicate_ids.header_extensions.emplace_back("foo", 5);
>-  duplicate_ids.header_extensions.emplace_back("bar", 5);
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            receiver->Receive(duplicate_ids).type());
>-}
>-
>-// Two receivers using the same transport shouldn't be able to use the same
>-// payload type to refer to different codecs, same header extension IDs to
>-// refer to different extensions, or same SSRC.
>-TEST_F(OrtcRtpReceiverTest, ReceiveReturnsErrorOnIdConflicts) {
>-  auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto audio_receiver = audio_receiver_result.MoveValue();
>-  auto video_receiver = video_receiver_result.MoveValue();
>-
>-  // First test payload type conflict.
>-  RtpParameters audio_parameters = MakeMinimalOpusParameters();
>-  RtpParameters video_parameters = MakeMinimalVp8Parameters();
>-  audio_parameters.codecs[0].payload_type = 100;
>-  video_parameters.codecs[0].payload_type = 100;
>-  EXPECT_TRUE(audio_receiver->Receive(audio_parameters).ok());
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            video_receiver->Receive(video_parameters).type());
>-
>-  // Test header extension ID conflict.
>-  video_parameters.codecs[0].payload_type = 110;
>-  audio_parameters.header_extensions.emplace_back("foo", 4);
>-  video_parameters.header_extensions.emplace_back("bar", 4);
>-  EXPECT_TRUE(audio_receiver->Receive(audio_parameters).ok());
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            video_receiver->Receive(video_parameters).type());
>-
>-  // Test SSRC conflict. Have an RTX SSRC that conflicts with a primary SSRC
>-  // for extra challenge.
>-  video_parameters.header_extensions[0].uri = "foo";
>-  audio_parameters.encodings[0].ssrc.emplace(0xabbaabba);
>-  audio_parameters.encodings[0].rtx.emplace(0xdeadbeef);
>-  video_parameters.encodings[0].ssrc.emplace(0xdeadbeef);
>-  EXPECT_TRUE(audio_receiver->Receive(audio_parameters).ok());
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            video_receiver->Receive(video_parameters).type());
>-
>-  // Sanity check that parameters can be set if the conflicts are all resolved.
>-  video_parameters.encodings[0].ssrc.emplace(0xbaadf00d);
>-  EXPECT_TRUE(video_receiver->Receive(video_parameters).ok());
>-}
>-
>-// Ensure that deleting a receiver causes receive streams at the media engine
>-// level to be cleared.
>-TEST_F(OrtcRtpReceiverTest, DeletingReceiverClearsReceiveStreams) {
>-  auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_receiver = audio_receiver_result.MoveValue();
>-  EXPECT_TRUE(audio_receiver->Receive(MakeMinimalOpusParameters()).ok());
>-
>-  // Also create an audio sender, to prevent the voice channel from being
>-  // completely deleted.
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  EXPECT_TRUE(audio_sender->Send(MakeMinimalOpusParameters()).ok());
>-
>-  audio_receiver.reset(nullptr);
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  EXPECT_TRUE(fake_voice_channel->recv_streams().empty());
>-
>-  auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_receiver = video_receiver_result.MoveValue();
>-  EXPECT_TRUE(video_receiver->Receive(MakeMinimalVp8Parameters()).ok());
>-
>-  // Also create an video sender, to prevent the video channel from being
>-  // completely deleted.
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-  EXPECT_TRUE(video_sender->Send(MakeMinimalVp8Parameters()).ok());
>-
>-  video_receiver.reset(nullptr);
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-  EXPECT_TRUE(fake_video_channel->recv_streams().empty());
>-}
>-
>-// If Receive hasn't been called, GetParameters should return empty parameters.
>-TEST_F(OrtcRtpReceiverTest, GetDefaultParameters) {
>-  auto result = ortc_factory_->CreateRtpReceiver(cricket::MEDIA_TYPE_AUDIO,
>-                                                 rtp_transport_.get());
>-  EXPECT_EQ(RtpParameters(), result.value()->GetParameters());
>-  result = ortc_factory_->CreateRtpReceiver(cricket::MEDIA_TYPE_VIDEO,
>-                                            rtp_transport_.get());
>-  EXPECT_EQ(RtpParameters(), result.value()->GetParameters());
>-}
>-
>-// Test that GetParameters returns the last parameters passed into Receive,
>-// along with the implementation-default values filled in where they were left
>-// unset.
>-TEST_F(OrtcRtpReceiverTest,
>-       GetParametersReturnsLastSetParametersWithDefaultsFilled) {
>-  auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_receiver = audio_receiver_result.MoveValue();
>-
>-  RtpParameters opus_parameters = MakeMinimalOpusParameters();
>-  EXPECT_TRUE(audio_receiver->Receive(opus_parameters).ok());
>-  EXPECT_EQ(opus_parameters, audio_receiver->GetParameters());
>-
>-  RtpParameters isac_parameters = MakeMinimalIsacParameters();
>-  // Sanity check that num_channels actually is left unset.
>-  ASSERT_FALSE(isac_parameters.codecs[0].num_channels);
>-  EXPECT_TRUE(audio_receiver->Receive(isac_parameters).ok());
>-  // Should be filled with a default "num channels" of 1.
>-  // TODO(deadbeef): This should actually default to 2 for some codecs. Update
>-  // this test once that's implemented.
>-  isac_parameters.codecs[0].num_channels.emplace(1);
>-  EXPECT_EQ(isac_parameters, audio_receiver->GetParameters());
>-
>-  auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_receiver = video_receiver_result.MoveValue();
>-
>-  RtpParameters vp8_parameters = MakeMinimalVp8Parameters();
>-  // Sanity check that clock_rate actually is left unset.
>-  EXPECT_TRUE(video_receiver->Receive(vp8_parameters).ok());
>-  // Should be filled with a default clock rate of 90000.
>-  vp8_parameters.codecs[0].clock_rate.emplace(90000);
>-  EXPECT_EQ(vp8_parameters, video_receiver->GetParameters());
>-
>-  RtpParameters vp9_parameters = MakeMinimalVp9Parameters();
>-  // Sanity check that clock_rate actually is left unset.
>-  EXPECT_TRUE(video_receiver->Receive(vp9_parameters).ok());
>-  // Should be filled with a default clock rate of 90000.
>-  vp9_parameters.codecs[0].clock_rate.emplace(90000);
>-  EXPECT_EQ(vp9_parameters, video_receiver->GetParameters());
>-}
>-
>-TEST_F(OrtcRtpReceiverTest, GetKind) {
>-  auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto audio_receiver = audio_receiver_result.MoveValue();
>-  auto video_receiver = video_receiver_result.MoveValue();
>-  EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, audio_receiver->GetKind());
>-  EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, video_receiver->GetKind());
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.cc
>deleted file mode 100644
>index 689412217acc8b5d0fc84ea69b344ca73b2fad06..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.cc
>+++ /dev/null
>@@ -1,181 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "ortc/ortcrtpreceiveradapter.h"
>-
>-#include <string>
>-#include <utility>
>-#include <vector>
>-
>-#include "media/base/mediaconstants.h"
>-#include "ortc/rtptransportadapter.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/helpers.h"  // For "CreateRandomX".
>-
>-namespace {
>-
>-void FillAudioReceiverParameters(webrtc::RtpParameters* parameters) {
>-  for (webrtc::RtpCodecParameters& codec : parameters->codecs) {
>-    if (!codec.num_channels) {
>-      codec.num_channels = 1;
>-    }
>-  }
>-}
>-
>-void FillVideoReceiverParameters(webrtc::RtpParameters* parameters) {
>-  for (webrtc::RtpCodecParameters& codec : parameters->codecs) {
>-    if (!codec.clock_rate) {
>-      codec.clock_rate = cricket::kVideoCodecClockrate;
>-    }
>-  }
>-}
>-
>-}  // namespace
>-
>-namespace webrtc {
>-
>-BEGIN_OWNED_PROXY_MAP(OrtcRtpReceiver)
>-PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, GetTrack)
>-PROXY_METHOD1(RTCError, SetTransport, RtpTransportInterface*)
>-PROXY_CONSTMETHOD0(RtpTransportInterface*, GetTransport)
>-PROXY_METHOD1(RTCError, Receive, const RtpParameters&)
>-PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
>-PROXY_CONSTMETHOD0(cricket::MediaType, GetKind)
>-END_PROXY_MAP()
>-
>-// static
>-std::unique_ptr<OrtcRtpReceiverInterface> OrtcRtpReceiverAdapter::CreateProxy(
>-    std::unique_ptr<OrtcRtpReceiverAdapter> wrapped_receiver) {
>-  RTC_DCHECK(wrapped_receiver);
>-  rtc::Thread* signaling =
>-      wrapped_receiver->rtp_transport_controller_->signaling_thread();
>-  rtc::Thread* worker =
>-      wrapped_receiver->rtp_transport_controller_->worker_thread();
>-  return OrtcRtpReceiverProxy::Create(signaling, worker,
>-                                      std::move(wrapped_receiver));
>-}
>-
>-OrtcRtpReceiverAdapter::~OrtcRtpReceiverAdapter() {
>-  internal_receiver_ = nullptr;
>-  SignalDestroyed();
>-}
>-
>-rtc::scoped_refptr<MediaStreamTrackInterface> OrtcRtpReceiverAdapter::GetTrack()
>-    const {
>-  return internal_receiver_ ? internal_receiver_->track() : nullptr;
>-}
>-
>-RTCError OrtcRtpReceiverAdapter::SetTransport(
>-    RtpTransportInterface* transport) {
>-  LOG_AND_RETURN_ERROR(
>-      RTCErrorType::UNSUPPORTED_OPERATION,
>-      "Changing the transport of an RtpReceiver is not yet supported.");
>-}
>-
>-RtpTransportInterface* OrtcRtpReceiverAdapter::GetTransport() const {
>-  return transport_;
>-}
>-
>-RTCError OrtcRtpReceiverAdapter::Receive(const RtpParameters& parameters) {
>-  RtpParameters filled_parameters = parameters;
>-  RTCError err;
>-  switch (kind_) {
>-    case cricket::MEDIA_TYPE_AUDIO:
>-      FillAudioReceiverParameters(&filled_parameters);
>-      err = rtp_transport_controller_->ValidateAndApplyAudioReceiverParameters(
>-          filled_parameters);
>-      if (!err.ok()) {
>-        return err;
>-      }
>-      break;
>-    case cricket::MEDIA_TYPE_VIDEO:
>-      FillVideoReceiverParameters(&filled_parameters);
>-      err = rtp_transport_controller_->ValidateAndApplyVideoReceiverParameters(
>-          filled_parameters);
>-      if (!err.ok()) {
>-        return err;
>-      }
>-      break;
>-    case cricket::MEDIA_TYPE_DATA:
>-      RTC_NOTREACHED();
>-      return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
>-  }
>-  last_applied_parameters_ = filled_parameters;
>-
>-  // Now that parameters were applied, can create (or recreate) the internal
>-  // receiver.
>-  //
>-  // This is analogous to a PeerConnection creating a receiver after
>-  // SetRemoteDescription is successful.
>-  MaybeRecreateInternalReceiver();
>-  return RTCError::OK();
>-}
>-
>-RtpParameters OrtcRtpReceiverAdapter::GetParameters() const {
>-  return last_applied_parameters_;
>-}
>-
>-cricket::MediaType OrtcRtpReceiverAdapter::GetKind() const {
>-  return kind_;
>-}
>-
>-OrtcRtpReceiverAdapter::OrtcRtpReceiverAdapter(
>-    cricket::MediaType kind,
>-    RtpTransportInterface* transport,
>-    RtpTransportControllerAdapter* rtp_transport_controller)
>-    : kind_(kind),
>-      transport_(transport),
>-      rtp_transport_controller_(rtp_transport_controller) {}
>-
>-void OrtcRtpReceiverAdapter::MaybeRecreateInternalReceiver() {
>-  if (last_applied_parameters_.encodings.empty()) {
>-    internal_receiver_ = nullptr;
>-    return;
>-  }
>-  // An SSRC of 0 is valid; this is used to identify "the default SSRC" (which
>-  // is the first one seen by the underlying media engine).
>-  uint32_t ssrc = 0;
>-  if (last_applied_parameters_.encodings[0].ssrc) {
>-    ssrc = *last_applied_parameters_.encodings[0].ssrc;
>-  }
>-  if (internal_receiver_ && ssrc == internal_receiver_->ssrc()) {
>-    // SSRC not changing; nothing to do.
>-    return;
>-  }
>-  internal_receiver_ = nullptr;
>-  switch (kind_) {
>-    case cricket::MEDIA_TYPE_AUDIO: {
>-      auto* audio_receiver = new AudioRtpReceiver(
>-          rtp_transport_controller_->worker_thread(), rtc::CreateRandomUuid(),
>-          std::vector<std::string>({}));
>-      auto* voice_channel = rtp_transport_controller_->voice_channel();
>-      RTC_DCHECK(voice_channel);
>-      audio_receiver->SetVoiceMediaChannel(voice_channel->media_channel());
>-      internal_receiver_ = audio_receiver;
>-      break;
>-    }
>-    case cricket::MEDIA_TYPE_VIDEO: {
>-      auto* video_receiver = new VideoRtpReceiver(
>-          rtp_transport_controller_->worker_thread(), rtc::CreateRandomUuid(),
>-          std::vector<std::string>({}));
>-      auto* video_channel = rtp_transport_controller_->video_channel();
>-      RTC_DCHECK(video_channel);
>-      video_receiver->SetVideoMediaChannel(video_channel->media_channel());
>-      internal_receiver_ = video_receiver;
>-      break;
>-    }
>-    case cricket::MEDIA_TYPE_DATA:
>-      RTC_NOTREACHED();
>-  }
>-  internal_receiver_->SetupMediaChannel(ssrc);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.h
>deleted file mode 100644
>index 4647b043145adae4b02f87275133c7ebea59d3ce..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.h
>+++ /dev/null
>@@ -1,79 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef ORTC_ORTCRTPRECEIVERADAPTER_H_
>-#define ORTC_ORTCRTPRECEIVERADAPTER_H_
>-
>-#include <memory>
>-
>-#include "api/ortc/ortcrtpreceiverinterface.h"
>-#include "api/rtcerror.h"
>-#include "api/rtpparameters.h"
>-#include "ortc/rtptransportcontrolleradapter.h"
>-#include "pc/rtpreceiver.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/third_party/sigslot/sigslot.h"
>-#include "rtc_base/thread.h"
>-
>-namespace webrtc {
>-
>-// Implementation of OrtcRtpReceiverInterface that works with
>-// RtpTransportAdapter, and wraps a VideoRtpReceiver/AudioRtpReceiver that's
>-// normally used with the PeerConnection.
>-//
>-// TODO(deadbeef): When BaseChannel is split apart into separate
>-// "RtpReceiver"/"RtpTransceiver"/"RtpReceiver"/"RtpReceiver" objects, this
>-// adapter object can be removed.
>-class OrtcRtpReceiverAdapter : public OrtcRtpReceiverInterface {
>- public:
>-  // Wraps |wrapped_receiver| in a proxy that will safely call methods on the
>-  // correct thread.
>-  static std::unique_ptr<OrtcRtpReceiverInterface> CreateProxy(
>-      std::unique_ptr<OrtcRtpReceiverAdapter> wrapped_receiver);
>-
>-  // Should only be called by RtpTransportControllerAdapter.
>-  OrtcRtpReceiverAdapter(
>-      cricket::MediaType kind,
>-      RtpTransportInterface* transport,
>-      RtpTransportControllerAdapter* rtp_transport_controller);
>-  ~OrtcRtpReceiverAdapter() override;
>-
>-  // OrtcRtpReceiverInterface implementation.
>-  rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const override;
>-
>-  RTCError SetTransport(RtpTransportInterface* transport) override;
>-  RtpTransportInterface* GetTransport() const override;
>-
>-  RTCError Receive(const RtpParameters& parameters) override;
>-  RtpParameters GetParameters() const override;
>-
>-  cricket::MediaType GetKind() const override;
>-
>-  // Used so that the RtpTransportControllerAdapter knows when it can
>-  // deallocate resources allocated for this object.
>-  sigslot::signal0<> SignalDestroyed;
>-
>- private:
>-  void MaybeRecreateInternalReceiver();
>-
>-  cricket::MediaType kind_;
>-  RtpTransportInterface* transport_;
>-  RtpTransportControllerAdapter* rtp_transport_controller_;
>-  // Scoped refptr due to ref-counted interface, but we should be the only
>-  // reference holder.
>-  rtc::scoped_refptr<RtpReceiverInternal> internal_receiver_;
>-  RtpParameters last_applied_parameters_;
>-
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(OrtcRtpReceiverAdapter);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // ORTC_ORTCRTPRECEIVERADAPTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsender_unittest.cc
>deleted file mode 100644
>index c8bd8f93934538b51ac8f062fe0907e40031052d..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsender_unittest.cc
>+++ /dev/null
>@@ -1,670 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-
>-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>-#include "media/base/fakemediaengine.h"
>-#include "ortc/ortcfactory.h"
>-#include "ortc/testrtpparameters.h"
>-#include "p2p/base/fakepackettransport.h"
>-#include "pc/test/fakevideotracksource.h"
>-#include "rtc_base/gunit.h"
>-
>-namespace webrtc {
>-
>-// This test uses an individual RtpSender using only the public interface, and
>-// verifies that its behaves as designed at an API level. Also tests that
>-// parameters are applied to the audio/video engines as expected. Network and
>-// media interfaces are faked to isolate what's being tested.
>-//
>-// This test shouldn't result any any actual media being sent. That sort of
>-// test should go in ortcfactory_integrationtest.cc.
>-class OrtcRtpSenderTest : public testing::Test {
>- public:
>-  OrtcRtpSenderTest() : fake_packet_transport_("fake") {
>-    // Need to set the fake packet transport to writable, in order to test that
>-    // the "send" flag is applied to the media engine based on the encoding
>-    // |active| flag.
>-    fake_packet_transport_.SetWritable(true);
>-    fake_media_engine_ = new cricket::FakeMediaEngine();
>-    // Note: This doesn't need to use fake network classes, since we already
>-    // use FakePacketTransport.
>-    auto ortc_factory_result = OrtcFactory::Create(
>-        nullptr, nullptr, nullptr, nullptr, nullptr,
>-        std::unique_ptr<cricket::MediaEngineInterface>(fake_media_engine_),
>-        CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory());
>-    ortc_factory_ = ortc_factory_result.MoveValue();
>-    RtpTransportParameters parameters;
>-    parameters.rtcp.mux = true;
>-    auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
>-        parameters, &fake_packet_transport_, nullptr, nullptr);
>-    rtp_transport_ = rtp_transport_result.MoveValue();
>-  }
>-
>- protected:
>-  rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
>-      const std::string& id) {
>-    return ortc_factory_->CreateAudioTrack(id, nullptr);
>-  }
>-
>-  rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
>-      const std::string& id) {
>-    return rtc::scoped_refptr<webrtc::VideoTrackInterface>(
>-        ortc_factory_->CreateVideoTrack(id, FakeVideoTrackSource::Create()));
>-  }
>-
>-  // Owned by |ortc_factory_|.
>-  cricket::FakeMediaEngine* fake_media_engine_;
>-  rtc::FakePacketTransport fake_packet_transport_;
>-  std::unique_ptr<OrtcFactoryInterface> ortc_factory_;
>-  std::unique_ptr<RtpTransportInterface> rtp_transport_;
>-};
>-
>-TEST_F(OrtcRtpSenderTest, GetAndSetTrack) {
>-  // Test GetTrack with a sender constructed with a track.
>-  auto audio_track = CreateAudioTrack("audio");
>-  auto audio_sender_result =
>-      ortc_factory_->CreateRtpSender(audio_track, rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  EXPECT_EQ(audio_track, audio_sender->GetTrack());
>-
>-  // Test GetTrack after SetTrack.
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-  auto video_track = CreateVideoTrack("video1");
>-  EXPECT_TRUE(video_sender->SetTrack(video_track).ok());
>-  EXPECT_EQ(video_track, video_sender->GetTrack());
>-  video_track = CreateVideoTrack("video2");
>-  EXPECT_TRUE(video_sender->SetTrack(video_track).ok());
>-  EXPECT_EQ(video_track, video_sender->GetTrack());
>-}
>-
>-// Test that track can be set when previously unset, even after Send has been
>-// called.
>-TEST_F(OrtcRtpSenderTest, SetTrackWhileSending) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  EXPECT_TRUE(audio_sender->Send(MakeMinimalOpusParameters()).ok());
>-  EXPECT_TRUE(audio_sender->SetTrack(CreateAudioTrack("audio")).ok());
>-
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-  EXPECT_TRUE(video_sender->Send(MakeMinimalVp8Parameters()).ok());
>-  EXPECT_TRUE(video_sender->SetTrack(CreateVideoTrack("video")).ok());
>-}
>-
>-// Test that track can be changed mid-sending. Differs from the above test in
>-// that the track is set and being changed, rather than unset and being set for
>-// the first time.
>-TEST_F(OrtcRtpSenderTest, ChangeTrackWhileSending) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      CreateAudioTrack("audio1"), rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  EXPECT_TRUE(audio_sender->Send(MakeMinimalOpusParameters()).ok());
>-  EXPECT_TRUE(audio_sender->SetTrack(CreateAudioTrack("audio2")).ok());
>-
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      CreateVideoTrack("video1"), rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-  EXPECT_TRUE(video_sender->Send(MakeMinimalVp8Parameters()).ok());
>-  EXPECT_TRUE(video_sender->SetTrack(CreateVideoTrack("video2")).ok());
>-}
>-
>-// Test that track can be set to null while sending.
>-TEST_F(OrtcRtpSenderTest, UnsetTrackWhileSending) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      CreateAudioTrack("audio"), rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  EXPECT_TRUE(audio_sender->Send(MakeMinimalOpusParameters()).ok());
>-  EXPECT_TRUE(audio_sender->SetTrack(nullptr).ok());
>-
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      CreateVideoTrack("video"), rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-  EXPECT_TRUE(video_sender->Send(MakeMinimalVp8Parameters()).ok());
>-  EXPECT_TRUE(video_sender->SetTrack(nullptr).ok());
>-}
>-
>-// Shouldn't be able to set an audio track on a video sender or vice versa.
>-TEST_F(OrtcRtpSenderTest, SetTrackOfWrongKindFails) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            audio_sender->SetTrack(CreateVideoTrack("video")).type());
>-
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            video_sender->SetTrack(CreateAudioTrack("audio")).type());
>-}
>-
>-// Currently SetTransport isn't supported. When it is, replace this test with a
>-// test/tests for it.
>-TEST_F(OrtcRtpSenderTest, SetTransportFails) {
>-  rtc::FakePacketTransport fake_packet_transport("another_transport");
>-  RtpTransportParameters parameters;
>-  parameters.rtcp.mux = true;
>-  auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
>-      parameters, &fake_packet_transport, nullptr, nullptr);
>-  auto rtp_transport = rtp_transport_result.MoveValue();
>-
>-  auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO,
>-                                                      rtp_transport_.get());
>-  auto sender = sender_result.MoveValue();
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            sender->SetTransport(rtp_transport.get()).type());
>-}
>-
>-TEST_F(OrtcRtpSenderTest, GetTransport) {
>-  auto result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO,
>-                                               rtp_transport_.get());
>-  EXPECT_EQ(rtp_transport_.get(), result.value()->GetTransport());
>-}
>-
>-// Test that "Send" causes the expected parameters to be applied to the media
>-// engine level, for an audio sender.
>-TEST_F(OrtcRtpSenderTest, SendAppliesAudioParametersToMediaEngine) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-
>-  // First, create parameters with all the bells and whistles.
>-  RtpParameters parameters;
>-
>-  RtpCodecParameters opus_codec;
>-  opus_codec.name = "opus";
>-  opus_codec.kind = cricket::MEDIA_TYPE_AUDIO;
>-  opus_codec.payload_type = 120;
>-  opus_codec.clock_rate.emplace(48000);
>-  opus_codec.num_channels.emplace(2);
>-  opus_codec.parameters["minptime"] = "10";
>-  opus_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::TRANSPORT_CC);
>-  parameters.codecs.push_back(std::move(opus_codec));
>-
>-  // Add two codecs, expecting the first to be used.
>-  // TODO(deadbeef): Once "codec_payload_type" is supported, use it to select a
>-  // codec that's not at the top of the list.
>-  RtpCodecParameters isac_codec;
>-  isac_codec.name = "ISAC";
>-  isac_codec.kind = cricket::MEDIA_TYPE_AUDIO;
>-  isac_codec.payload_type = 110;
>-  isac_codec.clock_rate.emplace(16000);
>-  parameters.codecs.push_back(std::move(isac_codec));
>-
>-  RtpEncodingParameters encoding;
>-  encoding.ssrc.emplace(0xdeadbeef);
>-  encoding.max_bitrate_bps.emplace(20000);
>-  parameters.encodings.push_back(std::move(encoding));
>-
>-  parameters.header_extensions.emplace_back(
>-      "urn:ietf:params:rtp-hdrext:ssrc-audio-level", 3);
>-
>-  EXPECT_TRUE(audio_sender->Send(parameters).ok());
>-
>-  // Now verify that the parameters were applied to the fake media engine layer
>-  // that exists below BaseChannel.
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  EXPECT_TRUE(fake_voice_channel->sending());
>-
>-  // Verify codec parameters.
>-  ASSERT_GT(fake_voice_channel->send_codecs().size(), 0u);
>-  const cricket::AudioCodec& top_codec = fake_voice_channel->send_codecs()[0];
>-  EXPECT_EQ("opus", top_codec.name);
>-  EXPECT_EQ(120, top_codec.id);
>-  EXPECT_EQ(48000, top_codec.clockrate);
>-  EXPECT_EQ(2u, top_codec.channels);
>-  ASSERT_NE(top_codec.params.end(), top_codec.params.find("minptime"));
>-  EXPECT_EQ("10", top_codec.params.at("minptime"));
>-
>-  // Verify encoding parameters.
>-  EXPECT_EQ(20000, fake_voice_channel->max_bps());
>-  ASSERT_EQ(1u, fake_voice_channel->send_streams().size());
>-  const cricket::StreamParams& send_stream =
>-      fake_voice_channel->send_streams()[0];
>-  EXPECT_EQ(1u, send_stream.ssrcs.size());
>-  EXPECT_EQ(0xdeadbeef, send_stream.first_ssrc());
>-
>-  // Verify header extensions.
>-  ASSERT_EQ(1u, fake_voice_channel->send_extensions().size());
>-  const RtpExtension& extension = fake_voice_channel->send_extensions()[0];
>-  EXPECT_EQ("urn:ietf:params:rtp-hdrext:ssrc-audio-level", extension.uri);
>-  EXPECT_EQ(3, extension.id);
>-}
>-
>-// Test that "Send" causes the expected parameters to be applied to the media
>-// engine level, for a video sender.
>-TEST_F(OrtcRtpSenderTest, SendAppliesVideoParametersToMediaEngine) {
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-
>-  // First, create parameters with all the bells and whistles.
>-  RtpParameters parameters;
>-
>-  RtpCodecParameters vp8_codec;
>-  vp8_codec.name = "VP8";
>-  vp8_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp8_codec.payload_type = 99;
>-  // Try a couple types of feedback params. "Generic NACK" is a bit of a
>-  // special case, so test it here.
>-  vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::CCM,
>-                                       RtcpFeedbackMessageType::FIR);
>-  vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::NACK,
>-                                       RtcpFeedbackMessageType::GENERIC_NACK);
>-  parameters.codecs.push_back(std::move(vp8_codec));
>-
>-  RtpCodecParameters vp8_rtx_codec;
>-  vp8_rtx_codec.name = "rtx";
>-  vp8_rtx_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp8_rtx_codec.payload_type = 100;
>-  vp8_rtx_codec.parameters["apt"] = "99";
>-  parameters.codecs.push_back(std::move(vp8_rtx_codec));
>-
>-  // Add two codecs, expecting the first to be used.
>-  // TODO(deadbeef): Once "codec_payload_type" is supported, use it to select a
>-  // codec that's not at the top of the list.
>-  RtpCodecParameters vp9_codec;
>-  vp9_codec.name = "VP9";
>-  vp9_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp9_codec.payload_type = 102;
>-  parameters.codecs.push_back(std::move(vp9_codec));
>-
>-  RtpCodecParameters vp9_rtx_codec;
>-  vp9_rtx_codec.name = "rtx";
>-  vp9_rtx_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp9_rtx_codec.payload_type = 103;
>-  vp9_rtx_codec.parameters["apt"] = "102";
>-  parameters.codecs.push_back(std::move(vp9_rtx_codec));
>-
>-  RtpEncodingParameters encoding;
>-  encoding.ssrc.emplace(0xdeadbeef);
>-  encoding.rtx.emplace(0xbaadfeed);
>-  encoding.max_bitrate_bps.emplace(99999);
>-  parameters.encodings.push_back(std::move(encoding));
>-
>-  parameters.header_extensions.emplace_back("urn:3gpp:video-orientation", 4);
>-  parameters.header_extensions.emplace_back(
>-      "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", 6);
>-
>-  EXPECT_TRUE(video_sender->Send(parameters).ok());
>-
>-  // Now verify that the parameters were applied to the fake media engine layer
>-  // that exists below BaseChannel.
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-  EXPECT_TRUE(fake_video_channel->sending());
>-
>-  // Verify codec parameters.
>-  ASSERT_GE(fake_video_channel->send_codecs().size(), 2u);
>-  const cricket::VideoCodec& top_codec = fake_video_channel->send_codecs()[0];
>-  EXPECT_EQ("VP8", top_codec.name);
>-  EXPECT_EQ(99, top_codec.id);
>-  EXPECT_TRUE(top_codec.feedback_params.Has({"ccm", "fir"}));
>-  EXPECT_TRUE(top_codec.feedback_params.Has(cricket::FeedbackParam("nack")));
>-
>-  const cricket::VideoCodec& rtx_codec = fake_video_channel->send_codecs()[1];
>-  EXPECT_EQ("rtx", rtx_codec.name);
>-  EXPECT_EQ(100, rtx_codec.id);
>-  ASSERT_NE(rtx_codec.params.end(), rtx_codec.params.find("apt"));
>-  EXPECT_EQ("99", rtx_codec.params.at("apt"));
>-
>-  // Verify encoding parameters.
>-  EXPECT_EQ(99999, fake_video_channel->max_bps());
>-  ASSERT_EQ(1u, fake_video_channel->send_streams().size());
>-  const cricket::StreamParams& send_stream =
>-      fake_video_channel->send_streams()[0];
>-  EXPECT_EQ(2u, send_stream.ssrcs.size());
>-  EXPECT_EQ(0xdeadbeef, send_stream.first_ssrc());
>-  uint32_t rtx_ssrc = 0u;
>-  EXPECT_TRUE(send_stream.GetFidSsrc(send_stream.first_ssrc(), &rtx_ssrc));
>-  EXPECT_EQ(0xbaadfeed, rtx_ssrc);
>-
>-  // Verify header extensions.
>-  ASSERT_EQ(2u, fake_video_channel->send_extensions().size());
>-  const RtpExtension& extension1 = fake_video_channel->send_extensions()[0];
>-  EXPECT_EQ("urn:3gpp:video-orientation", extension1.uri);
>-  EXPECT_EQ(4, extension1.id);
>-  const RtpExtension& extension2 = fake_video_channel->send_extensions()[1];
>-  EXPECT_EQ("http://www.webrtc.org/experiments/rtp-hdrext/playout-delay",
>-            extension2.uri);
>-  EXPECT_EQ(6, extension2.id);
>-}
>-
>-// Ensure that when primary or RTX SSRCs are left unset, they're generated
>-// automatically.
>-TEST_F(OrtcRtpSenderTest, SendGeneratesSsrcsWhenEmpty) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  RtpParameters parameters = MakeMinimalOpusParametersWithNoSsrc();
>-  // Default RTX parameters, with no SSRC.
>-  parameters.encodings[0].rtx.emplace();
>-  EXPECT_TRUE(audio_sender->Send(parameters).ok());
>-
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  ASSERT_EQ(1u, fake_voice_channel->send_streams().size());
>-  const cricket::StreamParams& audio_send_stream =
>-      fake_voice_channel->send_streams()[0];
>-  EXPECT_NE(0u, audio_send_stream.first_ssrc());
>-  uint32_t rtx_ssrc = 0u;
>-  EXPECT_TRUE(
>-      audio_send_stream.GetFidSsrc(audio_send_stream.first_ssrc(), &rtx_ssrc));
>-  EXPECT_NE(0u, rtx_ssrc);
>-  EXPECT_NE(audio_send_stream.first_ssrc(), rtx_ssrc);
>-
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-  parameters = MakeMinimalVp8ParametersWithNoSsrc();
>-  // Default RTX parameters, with no SSRC.
>-  parameters.encodings[0].rtx.emplace();
>-  EXPECT_TRUE(video_sender->Send(parameters).ok());
>-
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-  ASSERT_EQ(1u, fake_video_channel->send_streams().size());
>-  const cricket::StreamParams& video_send_stream =
>-      fake_video_channel->send_streams()[0];
>-  EXPECT_NE(0u, video_send_stream.first_ssrc());
>-  rtx_ssrc = 0u;
>-  EXPECT_TRUE(
>-      video_send_stream.GetFidSsrc(video_send_stream.first_ssrc(), &rtx_ssrc));
>-  EXPECT_NE(0u, rtx_ssrc);
>-  EXPECT_NE(video_send_stream.first_ssrc(), rtx_ssrc);
>-  EXPECT_NE(video_send_stream.first_ssrc(), audio_send_stream.first_ssrc());
>-}
>-
>-// Test changing both the send codec and SSRC at the same time, and verify that
>-// the new parameters are applied to the media engine level.
>-TEST_F(OrtcRtpSenderTest, CallingSendTwiceChangesParameters) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  EXPECT_TRUE(
>-      audio_sender->Send(MakeMinimalOpusParametersWithSsrc(0x11111111)).ok());
>-  EXPECT_TRUE(
>-      audio_sender->Send(MakeMinimalIsacParametersWithSsrc(0x22222222)).ok());
>-
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  ASSERT_GT(fake_voice_channel->send_codecs().size(), 0u);
>-  EXPECT_EQ("ISAC", fake_voice_channel->send_codecs()[0].name);
>-  ASSERT_EQ(1u, fake_voice_channel->send_streams().size());
>-  EXPECT_EQ(0x22222222u, fake_voice_channel->send_streams()[0].first_ssrc());
>-
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-  EXPECT_TRUE(
>-      video_sender->Send(MakeMinimalVp8ParametersWithSsrc(0x33333333)).ok());
>-  EXPECT_TRUE(
>-      video_sender->Send(MakeMinimalVp9ParametersWithSsrc(0x44444444)).ok());
>-
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-  ASSERT_GT(fake_video_channel->send_codecs().size(), 0u);
>-  EXPECT_EQ("VP9", fake_video_channel->send_codecs()[0].name);
>-  ASSERT_EQ(1u, fake_video_channel->send_streams().size());
>-  EXPECT_EQ(0x44444444u, fake_video_channel->send_streams()[0].first_ssrc());
>-}
>-
>-// Ensure that if the |active| flag of RtpEncodingParameters is set to false,
>-// sending stops at the media engine level.
>-TEST_F(OrtcRtpSenderTest, DeactivatingEncodingStopsSending) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  RtpParameters parameters = MakeMinimalOpusParameters();
>-  EXPECT_TRUE(audio_sender->Send(parameters).ok());
>-
>-  // Expect "sending" flag to initially be true.
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  EXPECT_TRUE(fake_voice_channel->sending());
>-
>-  // Deactivate encoding and expect it to change to false.
>-  parameters.encodings[0].active = false;
>-  EXPECT_TRUE(audio_sender->Send(parameters).ok());
>-  EXPECT_FALSE(fake_voice_channel->sending());
>-
>-  // Try the same thing for video now.
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-  parameters = MakeMinimalVp8Parameters();
>-  EXPECT_TRUE(video_sender->Send(parameters).ok());
>-
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-  EXPECT_TRUE(fake_video_channel->sending());
>-
>-  parameters.encodings[0].active = false;
>-  EXPECT_TRUE(video_sender->Send(parameters).ok());
>-  EXPECT_FALSE(fake_video_channel->sending());
>-}
>-
>-// Ensure that calling Send with an empty list of encodings causes send streams
>-// at the media engine level to be cleared.
>-TEST_F(OrtcRtpSenderTest, CallingSendWithEmptyEncodingsClearsSendStreams) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  RtpParameters parameters = MakeMinimalOpusParameters();
>-  EXPECT_TRUE(audio_sender->Send(parameters).ok());
>-  parameters.encodings.clear();
>-  EXPECT_TRUE(audio_sender->Send(parameters).ok());
>-
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  EXPECT_TRUE(fake_voice_channel->send_streams().empty());
>-
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-  parameters = MakeMinimalVp8Parameters();
>-  EXPECT_TRUE(video_sender->Send(parameters).ok());
>-  parameters.encodings.clear();
>-  EXPECT_TRUE(video_sender->Send(parameters).ok());
>-
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-  EXPECT_TRUE(fake_video_channel->send_streams().empty());
>-}
>-
>-// These errors should be covered by rtpparametersconversion_unittest.cc, but
>-// we should at least test that those errors are propogated from calls to Send,
>-// with a few examples.
>-TEST_F(OrtcRtpSenderTest, SendReturnsErrorOnInvalidParameters) {
>-  auto result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO,
>-                                               rtp_transport_.get());
>-  auto sender = result.MoveValue();
>-  // NACK feedback missing message type.
>-  RtpParameters invalid_feedback = MakeMinimalVp8Parameters();
>-  invalid_feedback.codecs[0].rtcp_feedback.emplace_back(RtcpFeedbackType::NACK);
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            sender->Send(invalid_feedback).type());
>-  // Negative payload type.
>-  RtpParameters invalid_pt = MakeMinimalVp8Parameters();
>-  invalid_pt.codecs[0].payload_type = -1;
>-  EXPECT_EQ(RTCErrorType::INVALID_RANGE, sender->Send(invalid_pt).type());
>-  // Duplicate codec payload types.
>-  RtpParameters duplicate_payload_types = MakeMinimalVp8Parameters();
>-  duplicate_payload_types.codecs.push_back(duplicate_payload_types.codecs[0]);
>-  duplicate_payload_types.codecs.back().name = "VP9";
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            sender->Send(duplicate_payload_types).type());
>-}
>-
>-// Two senders using the same transport shouldn't be able to use the same
>-// payload type to refer to different codecs, same header extension IDs to
>-// refer to different extensions, or same SSRC.
>-TEST_F(OrtcRtpSenderTest, SendReturnsErrorOnIdConflicts) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  auto video_sender = video_sender_result.MoveValue();
>-
>-  // First test payload type conflict.
>-  RtpParameters audio_parameters = MakeMinimalOpusParameters();
>-  RtpParameters video_parameters = MakeMinimalVp8Parameters();
>-  audio_parameters.codecs[0].payload_type = 100;
>-  video_parameters.codecs[0].payload_type = 100;
>-  EXPECT_TRUE(audio_sender->Send(audio_parameters).ok());
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            video_sender->Send(video_parameters).type());
>-
>-  // Test header extension ID conflict.
>-  video_parameters.codecs[0].payload_type = 110;
>-  audio_parameters.header_extensions.emplace_back("foo", 4);
>-  video_parameters.header_extensions.emplace_back("bar", 4);
>-  EXPECT_TRUE(audio_sender->Send(audio_parameters).ok());
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            video_sender->Send(video_parameters).type());
>-
>-  // Test SSRC conflict. Have an RTX SSRC that conflicts with a primary SSRC
>-  // for extra challenge.
>-  video_parameters.header_extensions[0].uri = "foo";
>-  audio_parameters.encodings[0].ssrc.emplace(0xdeadbeef);
>-  video_parameters.encodings[0].ssrc.emplace(0xabbaabba);
>-  video_parameters.encodings[0].rtx.emplace(0xdeadbeef);
>-  EXPECT_TRUE(audio_sender->Send(audio_parameters).ok());
>-  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER,
>-            video_sender->Send(video_parameters).type());
>-
>-  // Sanity check that parameters can be set if the conflicts are all resolved.
>-  video_parameters.encodings[0].rtx->ssrc.emplace(0xbaadf00d);
>-  EXPECT_TRUE(video_sender->Send(video_parameters).ok());
>-}
>-
>-// Ensure that deleting a sender causes send streams at the media engine level
>-// to be cleared.
>-TEST_F(OrtcRtpSenderTest, DeletingSenderClearsSendStreams) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  EXPECT_TRUE(audio_sender->Send(MakeMinimalOpusParameters()).ok());
>-
>-  // Also create an audio receiver, to prevent the voice channel from being
>-  // completely deleted.
>-  auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto audio_receiver = audio_receiver_result.MoveValue();
>-  EXPECT_TRUE(audio_receiver->Receive(MakeMinimalOpusParameters()).ok());
>-
>-  audio_sender.reset(nullptr);
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  EXPECT_TRUE(fake_voice_channel->send_streams().empty());
>-
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-  EXPECT_TRUE(video_sender->Send(MakeMinimalVp8Parameters()).ok());
>-
>-  // Also create an video receiver, to prevent the video channel from being
>-  // completely deleted.
>-  auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport_.get());
>-  auto video_receiver = video_receiver_result.MoveValue();
>-  EXPECT_TRUE(video_receiver->Receive(MakeMinimalVp8Parameters()).ok());
>-
>-  video_sender.reset(nullptr);
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-  EXPECT_TRUE(fake_video_channel->send_streams().empty());
>-}
>-
>-// If Send hasn't been called, GetParameters should return empty parameters.
>-TEST_F(OrtcRtpSenderTest, GetDefaultParameters) {
>-  auto result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO,
>-                                               rtp_transport_.get());
>-  EXPECT_EQ(RtpParameters(), result.value()->GetParameters());
>-  result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO,
>-                                          rtp_transport_.get());
>-  EXPECT_EQ(RtpParameters(), result.value()->GetParameters());
>-}
>-
>-// Test that GetParameters returns the last parameters passed into Send, along
>-// with the implementation-default values filled in where they were left unset.
>-TEST_F(OrtcRtpSenderTest,
>-       GetParametersReturnsLastSetParametersWithDefaultsFilled) {
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      CreateAudioTrack("audio"), rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-
>-  RtpParameters opus_parameters = MakeMinimalOpusParameters();
>-  EXPECT_TRUE(audio_sender->Send(opus_parameters).ok());
>-  EXPECT_EQ(opus_parameters, audio_sender->GetParameters());
>-
>-  RtpParameters isac_parameters = MakeMinimalIsacParameters();
>-  // Sanity check that num_channels actually is left unset.
>-  ASSERT_FALSE(isac_parameters.codecs[0].num_channels);
>-  EXPECT_TRUE(audio_sender->Send(isac_parameters).ok());
>-  // Should be filled with a default "num channels" of 1.
>-  // TODO(deadbeef): This should actually default to 2 for some codecs. Update
>-  // this test once that's implemented.
>-  isac_parameters.codecs[0].num_channels.emplace(1);
>-  EXPECT_EQ(isac_parameters, audio_sender->GetParameters());
>-
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      CreateVideoTrack("video"), rtp_transport_.get());
>-  auto video_sender = video_sender_result.MoveValue();
>-
>-  RtpParameters vp8_parameters = MakeMinimalVp8Parameters();
>-  // Sanity check that clock_rate actually is left unset.
>-  EXPECT_TRUE(video_sender->Send(vp8_parameters).ok());
>-  // Should be filled with a default clock rate of 90000.
>-  vp8_parameters.codecs[0].clock_rate.emplace(90000);
>-  EXPECT_EQ(vp8_parameters, video_sender->GetParameters());
>-
>-  RtpParameters vp9_parameters = MakeMinimalVp9Parameters();
>-  // Sanity check that clock_rate actually is left unset.
>-  EXPECT_TRUE(video_sender->Send(vp9_parameters).ok());
>-  // Should be filled with a default clock rate of 90000.
>-  vp9_parameters.codecs[0].clock_rate.emplace(90000);
>-  EXPECT_EQ(vp9_parameters, video_sender->GetParameters());
>-}
>-
>-TEST_F(OrtcRtpSenderTest, GetKind) {
>-  // Construct one sender from the "kind" enum and another from a track.
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport_.get());
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      CreateVideoTrack("video"), rtp_transport_.get());
>-  auto audio_sender = audio_sender_result.MoveValue();
>-  auto video_sender = video_sender_result.MoveValue();
>-  EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, audio_sender->GetKind());
>-  EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, video_sender->GetKind());
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.cc
>deleted file mode 100644
>index f2541cb8f85f3a64ae1a0f4594a42755de51cbd4..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.cc
>+++ /dev/null
>@@ -1,188 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "ortc/ortcrtpsenderadapter.h"
>-
>-#include <utility>
>-
>-#include "media/base/mediaconstants.h"
>-#include "ortc/rtptransportadapter.h"
>-#include "rtc_base/checks.h"
>-
>-namespace {
>-
>-void FillAudioSenderParameters(webrtc::RtpParameters* parameters) {
>-  for (webrtc::RtpCodecParameters& codec : parameters->codecs) {
>-    if (!codec.num_channels) {
>-      codec.num_channels = 1;
>-    }
>-  }
>-}
>-
>-void FillVideoSenderParameters(webrtc::RtpParameters* parameters) {
>-  for (webrtc::RtpCodecParameters& codec : parameters->codecs) {
>-    if (!codec.clock_rate) {
>-      codec.clock_rate = cricket::kVideoCodecClockrate;
>-    }
>-  }
>-}
>-
>-}  // namespace
>-
>-namespace webrtc {
>-
>-BEGIN_OWNED_PROXY_MAP(OrtcRtpSender)
>-PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-PROXY_METHOD1(RTCError, SetTrack, MediaStreamTrackInterface*)
>-PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, GetTrack)
>-PROXY_METHOD1(RTCError, SetTransport, RtpTransportInterface*)
>-PROXY_CONSTMETHOD0(RtpTransportInterface*, GetTransport)
>-PROXY_METHOD1(RTCError, Send, const RtpParameters&)
>-PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
>-PROXY_CONSTMETHOD0(cricket::MediaType, GetKind)
>-END_PROXY_MAP()
>-
>-// static
>-std::unique_ptr<OrtcRtpSenderInterface> OrtcRtpSenderAdapter::CreateProxy(
>-    std::unique_ptr<OrtcRtpSenderAdapter> wrapped_sender) {
>-  RTC_DCHECK(wrapped_sender);
>-  rtc::Thread* signaling =
>-      wrapped_sender->rtp_transport_controller_->signaling_thread();
>-  rtc::Thread* worker =
>-      wrapped_sender->rtp_transport_controller_->worker_thread();
>-  return OrtcRtpSenderProxy::Create(signaling, worker,
>-                                    std::move(wrapped_sender));
>-}
>-
>-OrtcRtpSenderAdapter::~OrtcRtpSenderAdapter() {
>-  internal_sender_ = nullptr;
>-  SignalDestroyed();
>-}
>-
>-RTCError OrtcRtpSenderAdapter::SetTrack(MediaStreamTrackInterface* track) {
>-  if (track && cricket::MediaTypeFromString(track->kind()) != kind_) {
>-    LOG_AND_RETURN_ERROR(
>-        RTCErrorType::INVALID_PARAMETER,
>-        "Track kind (audio/video) doesn't match the kind of this sender.");
>-  }
>-  if (internal_sender_ && !internal_sender_->SetTrack(track)) {
>-    // Since we checked the track type above, this should never happen...
>-    RTC_NOTREACHED();
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>-                         "Failed to set track on RtpSender.");
>-  }
>-  track_ = track;
>-  return RTCError::OK();
>-}
>-
>-rtc::scoped_refptr<MediaStreamTrackInterface> OrtcRtpSenderAdapter::GetTrack()
>-    const {
>-  return track_;
>-}
>-
>-RTCError OrtcRtpSenderAdapter::SetTransport(RtpTransportInterface* transport) {
>-  LOG_AND_RETURN_ERROR(
>-      RTCErrorType::UNSUPPORTED_OPERATION,
>-      "Changing the transport of an RtpSender is not yet supported.");
>-}
>-
>-RtpTransportInterface* OrtcRtpSenderAdapter::GetTransport() const {
>-  return transport_;
>-}
>-
>-RTCError OrtcRtpSenderAdapter::Send(const RtpParameters& parameters) {
>-  RtpParameters filled_parameters = parameters;
>-  RTCError err;
>-  uint32_t ssrc = 0;
>-  switch (kind_) {
>-    case cricket::MEDIA_TYPE_AUDIO:
>-      FillAudioSenderParameters(&filled_parameters);
>-      err = rtp_transport_controller_->ValidateAndApplyAudioSenderParameters(
>-          filled_parameters, &ssrc);
>-      if (!err.ok()) {
>-        return err;
>-      }
>-      break;
>-    case cricket::MEDIA_TYPE_VIDEO:
>-      FillVideoSenderParameters(&filled_parameters);
>-      err = rtp_transport_controller_->ValidateAndApplyVideoSenderParameters(
>-          filled_parameters, &ssrc);
>-      if (!err.ok()) {
>-        return err;
>-      }
>-      break;
>-    case cricket::MEDIA_TYPE_DATA:
>-      RTC_NOTREACHED();
>-      return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
>-  }
>-  last_applied_parameters_ = filled_parameters;
>-
>-  // Now that parameters were applied, can call SetSsrc on the internal sender.
>-  // This is analogous to a PeerConnection calling SetSsrc after
>-  // SetLocalDescription is successful.
>-  //
>-  // If there were no encodings, this SSRC may be 0, which is valid.
>-  if (!internal_sender_) {
>-    CreateInternalSender();
>-  }
>-  internal_sender_->SetSsrc(ssrc);
>-
>-  return RTCError::OK();
>-}
>-
>-RtpParameters OrtcRtpSenderAdapter::GetParameters() const {
>-  return last_applied_parameters_;
>-}
>-
>-cricket::MediaType OrtcRtpSenderAdapter::GetKind() const {
>-  return kind_;
>-}
>-
>-OrtcRtpSenderAdapter::OrtcRtpSenderAdapter(
>-    cricket::MediaType kind,
>-    RtpTransportInterface* transport,
>-    RtpTransportControllerAdapter* rtp_transport_controller)
>-    : kind_(kind),
>-      transport_(transport),
>-      rtp_transport_controller_(rtp_transport_controller) {}
>-
>-void OrtcRtpSenderAdapter::CreateInternalSender() {
>-  switch (kind_) {
>-    case cricket::MEDIA_TYPE_AUDIO: {
>-      auto* audio_sender = new AudioRtpSender(
>-          rtp_transport_controller_->worker_thread(), /*id=*/"", nullptr);
>-      auto* voice_channel = rtp_transport_controller_->voice_channel();
>-      RTC_DCHECK(voice_channel);
>-      audio_sender->SetVoiceMediaChannel(voice_channel->media_channel());
>-      internal_sender_ = audio_sender;
>-      break;
>-    }
>-    case cricket::MEDIA_TYPE_VIDEO: {
>-      auto* video_sender = new VideoRtpSender(
>-          rtp_transport_controller_->worker_thread(), /*id=*/"");
>-      auto* video_channel = rtp_transport_controller_->video_channel();
>-      RTC_DCHECK(video_channel);
>-      video_sender->SetVideoMediaChannel(video_channel->media_channel());
>-      internal_sender_ = video_sender;
>-      break;
>-    }
>-    case cricket::MEDIA_TYPE_DATA:
>-      RTC_NOTREACHED();
>-  }
>-  if (track_) {
>-    if (!internal_sender_->SetTrack(track_)) {
>-      // Since we checked the track type when it was set, this should never
>-      // happen...
>-      RTC_NOTREACHED();
>-    }
>-  }
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.h
>deleted file mode 100644
>index b078da985c9d18a06886220d249f8a9094fc5abb..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.h
>+++ /dev/null
>@@ -1,79 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef ORTC_ORTCRTPSENDERADAPTER_H_
>-#define ORTC_ORTCRTPSENDERADAPTER_H_
>-
>-#include <memory>
>-
>-#include "api/ortc/ortcrtpsenderinterface.h"
>-#include "api/rtcerror.h"
>-#include "api/rtpparameters.h"
>-#include "ortc/rtptransportcontrolleradapter.h"
>-#include "pc/rtpsender.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/third_party/sigslot/sigslot.h"
>-
>-namespace webrtc {
>-
>-// Implementation of OrtcRtpSenderInterface that works with RtpTransportAdapter,
>-// and wraps a VideoRtpSender/AudioRtpSender that's normally used with the
>-// PeerConnection.
>-//
>-// TODO(deadbeef): When BaseChannel is split apart into separate
>-// "RtpSender"/"RtpTransceiver"/"RtpSender"/"RtpReceiver" objects, this adapter
>-// object can be removed.
>-class OrtcRtpSenderAdapter : public OrtcRtpSenderInterface {
>- public:
>-  // Wraps |wrapped_sender| in a proxy that will safely call methods on the
>-  // correct thread.
>-  static std::unique_ptr<OrtcRtpSenderInterface> CreateProxy(
>-      std::unique_ptr<OrtcRtpSenderAdapter> wrapped_sender);
>-
>-  // Should only be called by RtpTransportControllerAdapter.
>-  OrtcRtpSenderAdapter(cricket::MediaType kind,
>-                       RtpTransportInterface* transport,
>-                       RtpTransportControllerAdapter* rtp_transport_controller);
>-  ~OrtcRtpSenderAdapter() override;
>-
>-  // OrtcRtpSenderInterface implementation.
>-  RTCError SetTrack(MediaStreamTrackInterface* track) override;
>-  rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const override;
>-
>-  RTCError SetTransport(RtpTransportInterface* transport) override;
>-  RtpTransportInterface* GetTransport() const override;
>-
>-  RTCError Send(const RtpParameters& parameters) override;
>-  RtpParameters GetParameters() const override;
>-
>-  cricket::MediaType GetKind() const override;
>-
>-  // Used so that the RtpTransportControllerAdapter knows when it can
>-  // deallocate resources allocated for this object.
>-  sigslot::signal0<> SignalDestroyed;
>-
>- private:
>-  void CreateInternalSender();
>-
>-  cricket::MediaType kind_;
>-  RtpTransportInterface* transport_;
>-  RtpTransportControllerAdapter* rtp_transport_controller_;
>-  // Scoped refptr due to ref-counted interface, but we should be the only
>-  // reference holder.
>-  rtc::scoped_refptr<RtpSenderInternal> internal_sender_;
>-  rtc::scoped_refptr<MediaStreamTrackInterface> track_;
>-  RtpParameters last_applied_parameters_;
>-
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(OrtcRtpSenderAdapter);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // ORTC_ORTCRTPSENDERADAPTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransport_unittest.cc
>deleted file mode 100644
>index 92ad43fab3edee70405ba5521d915dfb59309b8a..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransport_unittest.cc
>+++ /dev/null
>@@ -1,286 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-
>-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>-#include "media/base/fakemediaengine.h"
>-#include "ortc/ortcfactory.h"
>-#include "ortc/testrtpparameters.h"
>-#include "p2p/base/fakepackettransport.h"
>-#include "rtc_base/gunit.h"
>-
>-namespace webrtc {
>-
>-// This test uses fake packet transports and a fake media engine, in order to
>-// test the RtpTransport at only an API level. Any end-to-end test should go in
>-// ortcfactory_integrationtest.cc instead.
>-class RtpTransportTest : public testing::Test {
>- public:
>-  RtpTransportTest() {
>-    fake_media_engine_ = new cricket::FakeMediaEngine();
>-    // Note: This doesn't need to use fake network classes, since it uses
>-    // FakePacketTransports.
>-    auto result = OrtcFactory::Create(
>-        nullptr, nullptr, nullptr, nullptr, nullptr,
>-        std::unique_ptr<cricket::MediaEngineInterface>(fake_media_engine_),
>-        CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory());
>-    ortc_factory_ = result.MoveValue();
>-  }
>-
>- protected:
>-  // Owned by |ortc_factory_|.
>-  cricket::FakeMediaEngine* fake_media_engine_;
>-  std::unique_ptr<OrtcFactoryInterface> ortc_factory_;
>-};
>-
>-// Test GetRtpPacketTransport and GetRtcpPacketTransport, with and without RTCP
>-// muxing.
>-TEST_F(RtpTransportTest, GetPacketTransports) {
>-  rtc::FakePacketTransport rtp("rtp");
>-  rtc::FakePacketTransport rtcp("rtcp");
>-  // With muxed RTCP.
>-  RtpTransportParameters parameters;
>-  parameters.rtcp.mux = true;
>-  auto result =
>-      ortc_factory_->CreateRtpTransport(parameters, &rtp, nullptr, nullptr);
>-  ASSERT_TRUE(result.ok());
>-  EXPECT_EQ(&rtp, result.value()->GetRtpPacketTransport());
>-  EXPECT_EQ(nullptr, result.value()->GetRtcpPacketTransport());
>-  result.MoveValue().reset();
>-  // With non-muxed RTCP.
>-  parameters.rtcp.mux = false;
>-  result = ortc_factory_->CreateRtpTransport(parameters, &rtp, &rtcp, nullptr);
>-  ASSERT_TRUE(result.ok());
>-  EXPECT_EQ(&rtp, result.value()->GetRtpPacketTransport());
>-  EXPECT_EQ(&rtcp, result.value()->GetRtcpPacketTransport());
>-}
>-
>-// If an RtpTransport starts out un-muxed and then starts muxing, the RTCP
>-// packet transport should be forgotten and GetRtcpPacketTransport should
>-// return null.
>-TEST_F(RtpTransportTest, EnablingRtcpMuxingUnsetsRtcpTransport) {
>-  rtc::FakePacketTransport rtp("rtp");
>-  rtc::FakePacketTransport rtcp("rtcp");
>-
>-  // Create non-muxed.
>-  RtpTransportParameters parameters;
>-  parameters.rtcp.mux = false;
>-  auto result =
>-      ortc_factory_->CreateRtpTransport(parameters, &rtp, &rtcp, nullptr);
>-  ASSERT_TRUE(result.ok());
>-  auto rtp_transport = result.MoveValue();
>-
>-  // Enable muxing.
>-  parameters.rtcp.mux = true;
>-  EXPECT_TRUE(rtp_transport->SetParameters(parameters).ok());
>-  EXPECT_EQ(nullptr, rtp_transport->GetRtcpPacketTransport());
>-}
>-
>-TEST_F(RtpTransportTest, GetAndSetRtcpParameters) {
>-  rtc::FakePacketTransport rtp("rtp");
>-  rtc::FakePacketTransport rtcp("rtcp");
>-  // Start with non-muxed RTCP.
>-  RtpTransportParameters parameters;
>-  parameters.rtcp.mux = false;
>-  parameters.rtcp.cname = "teST";
>-  parameters.rtcp.reduced_size = false;
>-  auto result =
>-      ortc_factory_->CreateRtpTransport(parameters, &rtp, &rtcp, nullptr);
>-  ASSERT_TRUE(result.ok());
>-  auto transport = result.MoveValue();
>-  EXPECT_EQ(parameters, transport->GetParameters());
>-
>-  // Changing the CNAME is currently unsupported.
>-  parameters.rtcp.cname = "different";
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            transport->SetParameters(parameters).type());
>-  parameters.rtcp.cname = "teST";
>-
>-  // Enable RTCP muxing and reduced-size RTCP.
>-  parameters.rtcp.mux = true;
>-  parameters.rtcp.reduced_size = true;
>-  EXPECT_TRUE(transport->SetParameters(parameters).ok());
>-  EXPECT_EQ(parameters, transport->GetParameters());
>-
>-  // Empty CNAME should result in the existing CNAME being used.
>-  parameters.rtcp.cname.clear();
>-  EXPECT_TRUE(transport->SetParameters(parameters).ok());
>-  EXPECT_EQ("teST", transport->GetParameters().rtcp.cname);
>-
>-  // Disabling RTCP muxing after enabling shouldn't be allowed, since enabling
>-  // muxing should have made the RTP transport forget about the RTCP packet
>-  // transport initially passed into it.
>-  parameters.rtcp.mux = false;
>-  EXPECT_EQ(RTCErrorType::INVALID_STATE,
>-            transport->SetParameters(parameters).type());
>-}
>-
>-// When Send or Receive is called on a sender or receiver, the RTCP parameters
>-// from the RtpTransport underneath the sender should be applied to the created
>-// media stream. The only relevant parameters (currently) are |cname| and
>-// |reduced_size|.
>-TEST_F(RtpTransportTest, SendAndReceiveApplyRtcpParametersToMediaEngine) {
>-  // First, create video transport with reduced-size RTCP.
>-  rtc::FakePacketTransport fake_packet_transport1("1");
>-  RtpTransportParameters parameters;
>-  parameters.rtcp.mux = true;
>-  parameters.rtcp.reduced_size = true;
>-  parameters.rtcp.cname = "foo";
>-  auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
>-      parameters, &fake_packet_transport1, nullptr, nullptr);
>-  auto video_transport = rtp_transport_result.MoveValue();
>-
>-  // Create video sender and call Send, expecting parameters to be applied.
>-  auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO,
>-                                                      video_transport.get());
>-  auto video_sender = sender_result.MoveValue();
>-  EXPECT_TRUE(video_sender->Send(MakeMinimalVp8Parameters()).ok());
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-  EXPECT_TRUE(fake_video_channel->send_rtcp_parameters().reduced_size);
>-  ASSERT_EQ(1u, fake_video_channel->send_streams().size());
>-  const cricket::StreamParams& video_send_stream =
>-      fake_video_channel->send_streams()[0];
>-  EXPECT_EQ("foo", video_send_stream.cname);
>-
>-  // Create video receiver and call Receive, expecting parameters to be applied
>-  // (minus |cname|, since that's the sent cname, not received).
>-  auto receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, video_transport.get());
>-  auto video_receiver = receiver_result.MoveValue();
>-  EXPECT_TRUE(
>-      video_receiver->Receive(MakeMinimalVp8ParametersWithSsrc(0xdeadbeef))
>-          .ok());
>-  EXPECT_TRUE(fake_video_channel->recv_rtcp_parameters().reduced_size);
>-
>-  // Create audio transport with non-reduced size RTCP.
>-  rtc::FakePacketTransport fake_packet_transport2("2");
>-  parameters.rtcp.reduced_size = false;
>-  parameters.rtcp.cname = "bar";
>-  rtp_transport_result = ortc_factory_->CreateRtpTransport(
>-      parameters, &fake_packet_transport2, nullptr, nullptr);
>-  auto audio_transport = rtp_transport_result.MoveValue();
>-
>-  // Create audio sender and call Send, expecting parameters to be applied.
>-  sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO,
>-                                                 audio_transport.get());
>-  auto audio_sender = sender_result.MoveValue();
>-  EXPECT_TRUE(audio_sender->Send(MakeMinimalIsacParameters()).ok());
>-
>-  cricket::FakeVoiceMediaChannel* fake_voice_channel =
>-      fake_media_engine_->GetVoiceChannel(0);
>-  ASSERT_NE(nullptr, fake_voice_channel);
>-  EXPECT_FALSE(fake_voice_channel->send_rtcp_parameters().reduced_size);
>-  ASSERT_EQ(1u, fake_voice_channel->send_streams().size());
>-  const cricket::StreamParams& audio_send_stream =
>-      fake_voice_channel->send_streams()[0];
>-  EXPECT_EQ("bar", audio_send_stream.cname);
>-
>-  // Create audio receiver and call Receive, expecting parameters to be applied
>-  // (minus |cname|, since that's the sent cname, not received).
>-  receiver_result = ortc_factory_->CreateRtpReceiver(cricket::MEDIA_TYPE_AUDIO,
>-                                                     audio_transport.get());
>-  auto audio_receiver = receiver_result.MoveValue();
>-  EXPECT_TRUE(
>-      audio_receiver->Receive(MakeMinimalOpusParametersWithSsrc(0xbaadf00d))
>-          .ok());
>-  EXPECT_FALSE(fake_voice_channel->recv_rtcp_parameters().reduced_size);
>-}
>-
>-// When SetParameters is called, the modified parameters should be applied
>-// to the media engine.
>-// TODO(deadbeef): Once the implementation supports changing the CNAME,
>-// test that here.
>-TEST_F(RtpTransportTest, SetRtcpParametersAppliesParametersToMediaEngine) {
>-  rtc::FakePacketTransport fake_packet_transport("fake");
>-  RtpTransportParameters parameters;
>-  parameters.rtcp.mux = true;
>-  parameters.rtcp.reduced_size = false;
>-  auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
>-      parameters, &fake_packet_transport, nullptr, nullptr);
>-  auto rtp_transport = rtp_transport_result.MoveValue();
>-
>-  // Create video sender and call Send, applying an initial set of parameters.
>-  auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO,
>-                                                      rtp_transport.get());
>-  auto sender = sender_result.MoveValue();
>-  EXPECT_TRUE(sender->Send(MakeMinimalVp8Parameters()).ok());
>-
>-  // Modify parameters and expect them to be changed at the media engine level.
>-  parameters.rtcp.reduced_size = true;
>-  EXPECT_TRUE(rtp_transport->SetParameters(parameters).ok());
>-
>-  cricket::FakeVideoMediaChannel* fake_video_channel =
>-      fake_media_engine_->GetVideoChannel(0);
>-  ASSERT_NE(nullptr, fake_video_channel);
>-  EXPECT_TRUE(fake_video_channel->send_rtcp_parameters().reduced_size);
>-}
>-
>-// SetParameters should set keepalive for all RTP transports.
>-// It is impossible to modify keepalive parameters if any streams are created.
>-// Note: This is an implementation detail for current way of configuring the
>-// keep-alive. It may change in the future.
>-TEST_F(RtpTransportTest, CantChangeKeepAliveAfterCreatedSendStreams) {
>-  rtc::FakePacketTransport fake_packet_transport("fake");
>-  RtpTransportParameters parameters;
>-  parameters.keepalive.timeout_interval_ms = 100;
>-  auto rtp_transport_result = ortc_factory_->CreateRtpTransport(
>-      parameters, &fake_packet_transport, nullptr, nullptr);
>-  ASSERT_TRUE(rtp_transport_result.ok());
>-  std::unique_ptr<RtpTransportInterface> rtp_transport =
>-      rtp_transport_result.MoveValue();
>-
>-  // Updating keepalive parameters is ok, since no rtp sender created.
>-  parameters.keepalive.timeout_interval_ms = 200;
>-  EXPECT_TRUE(rtp_transport->SetParameters(parameters).ok());
>-
>-  // Create video sender. Note: |sender_result| scope must extend past the
>-  // SetParameters() call below.
>-  auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_VIDEO,
>-                                                      rtp_transport.get());
>-  EXPECT_TRUE(sender_result.ok());
>-
>-  // Modify parameters second time after video send stream created.
>-  parameters.keepalive.timeout_interval_ms = 10;
>-  EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION,
>-            rtp_transport->SetParameters(parameters).type());
>-}
>-
>-// Note: This is an implementation detail for current way of configuring the
>-// keep-alive. It may change in the future.
>-TEST_F(RtpTransportTest, KeepAliveMustBeSameAcrossTransportController) {
>-  rtc::FakePacketTransport fake_packet_transport("fake");
>-  RtpTransportParameters parameters;
>-  parameters.keepalive.timeout_interval_ms = 100;
>-
>-  // Manually create a controller, that can be shared by multiple transports.
>-  auto controller_result = ortc_factory_->CreateRtpTransportController();
>-  ASSERT_TRUE(controller_result.ok());
>-  std::unique_ptr<RtpTransportControllerInterface> controller =
>-      controller_result.MoveValue();
>-
>-  // Create a first transport.
>-  auto first_transport_result = ortc_factory_->CreateRtpTransport(
>-      parameters, &fake_packet_transport, nullptr, controller.get());
>-  ASSERT_TRUE(first_transport_result.ok());
>-
>-  // Update the parameters, and create another transport for the same
>-  // controller.
>-  parameters.keepalive.timeout_interval_ms = 10;
>-  auto seconds_transport_result = ortc_factory_->CreateRtpTransport(
>-      parameters, &fake_packet_transport, nullptr, controller.get());
>-  EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION,
>-            seconds_transport_result.error().type());
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.cc
>deleted file mode 100644
>index 84c2ef09e66eddb9ac784c371611de723fea7ae0..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.cc
>+++ /dev/null
>@@ -1,230 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "ortc/rtptransportadapter.h"
>-
>-#include <algorithm>  // For std::find.
>-#include <set>
>-#include <utility>  // For std::move.
>-
>-#include "absl/memory/memory.h"
>-#include "api/proxy.h"
>-#include "rtc_base/logging.h"
>-
>-namespace webrtc {
>-
>-BEGIN_OWNED_PROXY_MAP(RtpTransport)
>-PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-PROXY_CONSTMETHOD0(PacketTransportInterface*, GetRtpPacketTransport)
>-PROXY_CONSTMETHOD0(PacketTransportInterface*, GetRtcpPacketTransport)
>-PROXY_METHOD1(RTCError, SetParameters, const RtpTransportParameters&)
>-PROXY_CONSTMETHOD0(RtpTransportParameters, GetParameters)
>-protected:
>-RtpTransportAdapter* GetInternal() override {
>-  return internal();
>-}
>-END_PROXY_MAP()
>-
>-BEGIN_OWNED_PROXY_MAP(SrtpTransport)
>-PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-PROXY_CONSTMETHOD0(PacketTransportInterface*, GetRtpPacketTransport)
>-PROXY_CONSTMETHOD0(PacketTransportInterface*, GetRtcpPacketTransport)
>-PROXY_METHOD1(RTCError, SetParameters, const RtpTransportParameters&)
>-PROXY_CONSTMETHOD0(RtpTransportParameters, GetParameters)
>-PROXY_METHOD1(RTCError, SetSrtpSendKey, const cricket::CryptoParams&)
>-PROXY_METHOD1(RTCError, SetSrtpReceiveKey, const cricket::CryptoParams&)
>-protected:
>-RtpTransportAdapter* GetInternal() override {
>-  return internal();
>-}
>-END_PROXY_MAP()
>-
>-// static
>-RTCErrorOr<std::unique_ptr<RtpTransportInterface>>
>-RtpTransportAdapter::CreateProxied(
>-    const RtpTransportParameters& parameters,
>-    PacketTransportInterface* rtp,
>-    PacketTransportInterface* rtcp,
>-    RtpTransportControllerAdapter* rtp_transport_controller) {
>-  if (!rtp) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Must provide an RTP packet transport.");
>-  }
>-  if (!parameters.rtcp.mux && !rtcp) {
>-    LOG_AND_RETURN_ERROR(
>-        RTCErrorType::INVALID_PARAMETER,
>-        "Must provide an RTCP packet transport when RTCP muxing is not used.");
>-  }
>-  if (parameters.rtcp.mux && rtcp) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Creating an RtpTransport with RTCP muxing enabled, "
>-                         "with a separate RTCP packet transport?");
>-  }
>-  if (!rtp_transport_controller) {
>-    // Since OrtcFactory::CreateRtpTransport creates an RtpTransportController
>-    // automatically when one isn't passed in, this should never be reached.
>-    RTC_NOTREACHED();
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Must provide an RTP transport controller.");
>-  }
>-  std::unique_ptr<RtpTransportAdapter> transport_adapter(
>-      new RtpTransportAdapter(parameters.rtcp, rtp, rtcp,
>-                              rtp_transport_controller,
>-                              false /*is_srtp_transport*/));
>-  RTCError params_result = transport_adapter->SetParameters(parameters);
>-  if (!params_result.ok()) {
>-    return std::move(params_result);
>-  }
>-
>-  return RtpTransportProxyWithInternal<RtpTransportAdapter>::Create(
>-      rtp_transport_controller->signaling_thread(),
>-      rtp_transport_controller->worker_thread(), std::move(transport_adapter));
>-}
>-
>-RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
>-RtpTransportAdapter::CreateSrtpProxied(
>-    const RtpTransportParameters& parameters,
>-    PacketTransportInterface* rtp,
>-    PacketTransportInterface* rtcp,
>-    RtpTransportControllerAdapter* rtp_transport_controller) {
>-  if (!rtp) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Must provide an RTP packet transport.");
>-  }
>-  if (!parameters.rtcp.mux && !rtcp) {
>-    LOG_AND_RETURN_ERROR(
>-        RTCErrorType::INVALID_PARAMETER,
>-        "Must provide an RTCP packet transport when RTCP muxing is not used.");
>-  }
>-  if (parameters.rtcp.mux && rtcp) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Creating an RtpTransport with RTCP muxing enabled, "
>-                         "with a separate RTCP packet transport?");
>-  }
>-  if (!rtp_transport_controller) {
>-    // Since OrtcFactory::CreateRtpTransport creates an RtpTransportController
>-    // automatically when one isn't passed in, this should never be reached.
>-    RTC_NOTREACHED();
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Must provide an RTP transport controller.");
>-  }
>-
>-  std::unique_ptr<RtpTransportAdapter> transport_adapter;
>-  transport_adapter.reset(new RtpTransportAdapter(parameters.rtcp, rtp, rtcp,
>-                                                  rtp_transport_controller,
>-                                                  true /*is_srtp_transport*/));
>-  RTCError params_result = transport_adapter->SetParameters(parameters);
>-  if (!params_result.ok()) {
>-    return std::move(params_result);
>-  }
>-
>-  return SrtpTransportProxyWithInternal<RtpTransportAdapter>::Create(
>-      rtp_transport_controller->signaling_thread(),
>-      rtp_transport_controller->worker_thread(), std::move(transport_adapter));
>-}
>-
>-void RtpTransportAdapter::TakeOwnershipOfRtpTransportController(
>-    std::unique_ptr<RtpTransportControllerInterface> controller) {
>-  RTC_DCHECK_EQ(rtp_transport_controller_, controller->GetInternal());
>-  RTC_DCHECK(owned_rtp_transport_controller_.get() == nullptr);
>-  owned_rtp_transport_controller_ = std::move(controller);
>-}
>-
>-RtpTransportAdapter::RtpTransportAdapter(
>-    const RtcpParameters& rtcp_params,
>-    PacketTransportInterface* rtp,
>-    PacketTransportInterface* rtcp,
>-    RtpTransportControllerAdapter* rtp_transport_controller,
>-    bool is_srtp_transport)
>-    : rtp_packet_transport_(rtp),
>-      rtcp_packet_transport_(rtcp),
>-      rtp_transport_controller_(rtp_transport_controller),
>-      network_thread_(rtp_transport_controller_->network_thread()) {
>-  parameters_.rtcp = rtcp_params;
>-  // CNAME should have been filled by OrtcFactory if empty.
>-  RTC_DCHECK(!parameters_.rtcp.cname.empty());
>-  RTC_DCHECK(rtp_transport_controller);
>-
>-  if (is_srtp_transport) {
>-    srtp_transport_ = absl::make_unique<SrtpTransport>(rtcp == nullptr);
>-    transport_ = srtp_transport_.get();
>-  } else {
>-    unencrypted_rtp_transport_ =
>-        absl::make_unique<RtpTransport>(rtcp == nullptr);
>-    transport_ = unencrypted_rtp_transport_.get();
>-  }
>-  RTC_DCHECK(transport_);
>-
>-  network_thread_->Invoke<void>(RTC_FROM_HERE, [=] {
>-    SetRtpPacketTransport(rtp->GetInternal());
>-    if (rtcp) {
>-      SetRtcpPacketTransport(rtcp->GetInternal());
>-    }
>-  });
>-
>-  transport_->SignalReadyToSend.connect(this,
>-                                        &RtpTransportAdapter::OnReadyToSend);
>-  transport_->SignalRtcpPacketReceived.connect(
>-      this, &RtpTransportAdapter::OnRtcpPacketReceived);
>-  transport_->SignalWritableState.connect(
>-      this, &RtpTransportAdapter::OnWritableState);
>-}
>-
>-RtpTransportAdapter::~RtpTransportAdapter() {
>-  SignalDestroyed(this);
>-}
>-
>-RTCError RtpTransportAdapter::SetParameters(
>-    const RtpTransportParameters& parameters) {
>-  if (!parameters.rtcp.mux && parameters_.rtcp.mux) {
>-    LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::INVALID_STATE,
>-                         "Can't disable RTCP muxing after enabling.");
>-  }
>-  if (!parameters.rtcp.cname.empty() &&
>-      parameters.rtcp.cname != parameters_.rtcp.cname) {
>-    LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
>-                         "Changing the RTCP CNAME is currently unsupported.");
>-  }
>-  // If the CNAME is empty, use the existing one.
>-  RtpTransportParameters copy = parameters;
>-  if (copy.rtcp.cname.empty()) {
>-    copy.rtcp.cname = parameters_.rtcp.cname;
>-  }
>-  RTCError err =
>-      rtp_transport_controller_->SetRtpTransportParameters(copy, this);
>-  if (!err.ok()) {
>-    return err;
>-  }
>-  parameters_ = copy;
>-  if (parameters_.rtcp.mux) {
>-    rtcp_packet_transport_ = nullptr;
>-  }
>-  return RTCError::OK();
>-}
>-
>-RTCError RtpTransportAdapter::SetSrtpSendKey(
>-    const cricket::CryptoParams& params) {
>-  if (!network_thread_->IsCurrent()) {
>-    return network_thread_->Invoke<RTCError>(
>-        RTC_FROM_HERE, [&] { return SetSrtpSendKey(params); });
>-  }
>-  return transport_->SetSrtpSendKey(params);
>-}
>-
>-RTCError RtpTransportAdapter::SetSrtpReceiveKey(
>-    const cricket::CryptoParams& params) {
>-  if (!network_thread_->IsCurrent()) {
>-    return network_thread_->Invoke<RTCError>(
>-        RTC_FROM_HERE, [&] { return SetSrtpReceiveKey(params); });
>-  }
>-  return transport_->SetSrtpReceiveKey(params);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.h
>deleted file mode 100644
>index 81914f6b8d1b475f75b0485595a18a7f1641a544..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.h
>+++ /dev/null
>@@ -1,121 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef ORTC_RTPTRANSPORTADAPTER_H_
>-#define ORTC_RTPTRANSPORTADAPTER_H_
>-
>-#include <memory>
>-#include <vector>
>-
>-#include "api/rtcerror.h"
>-#include "media/base/streamparams.h"
>-#include "ortc/rtptransportcontrolleradapter.h"
>-#include "pc/channel.h"
>-#include "pc/rtptransportinternaladapter.h"
>-#include "pc/srtptransport.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/third_party/sigslot/sigslot.h"
>-
>-namespace webrtc {
>-// This class is a wrapper over an RtpTransport or an SrtpTransport. The base
>-// class RtpTransportInternalAdapter keeps a raw pointer, |transport_|, of the
>-// transport object and implements both the public SrtpTransportInterface and
>-// RtpTransport internal interface by calling the |transport_| underneath.
>-//
>-// This adapter can be used as an unencrypted RTP transport or an SrtpTransport
>-// with RtpSenderAdapter, RtpReceiverAdapter, and RtpTransportControllerAdapter.
>-//
>-// TODO(deadbeef): When BaseChannel is split apart into separate
>-// "RtpTransport"/"RtpTransceiver"/"RtpSender"/"RtpReceiver" objects, this
>-// adapter object can be removed.
>-class RtpTransportAdapter : public RtpTransportInternalAdapter {
>- public:
>-  // |rtp| can't be null. |rtcp| can if RTCP muxing is used immediately (meaning
>-  // |rtcp_parameters.mux| is also true).
>-  static RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateProxied(
>-      const RtpTransportParameters& rtcp_parameters,
>-      PacketTransportInterface* rtp,
>-      PacketTransportInterface* rtcp,
>-      RtpTransportControllerAdapter* rtp_transport_controller);
>-
>-  static RTCErrorOr<std::unique_ptr<SrtpTransportInterface>> CreateSrtpProxied(
>-      const RtpTransportParameters& rtcp_parameters,
>-      PacketTransportInterface* rtp,
>-      PacketTransportInterface* rtcp,
>-      RtpTransportControllerAdapter* rtp_transport_controller);
>-
>-  ~RtpTransportAdapter() override;
>-
>-  // RtpTransportInterface implementation.
>-  PacketTransportInterface* GetRtpPacketTransport() const override {
>-    return rtp_packet_transport_;
>-  }
>-  PacketTransportInterface* GetRtcpPacketTransport() const override {
>-    return rtcp_packet_transport_;
>-  }
>-  RTCError SetParameters(const RtpTransportParameters& parameters) override;
>-  RtpTransportParameters GetParameters() const override { return parameters_; }
>-
>-  // SRTP specific implementation.
>-  RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override;
>-  RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) override;
>-
>-  // Methods used internally by OrtcFactory.
>-  RtpTransportControllerAdapter* rtp_transport_controller() {
>-    return rtp_transport_controller_;
>-  }
>-  void TakeOwnershipOfRtpTransportController(
>-      std::unique_ptr<RtpTransportControllerInterface> controller);
>-
>-  // Used by RtpTransportControllerAdapter to tell when it should stop
>-  // returning this transport from GetTransports().
>-  sigslot::signal1<RtpTransportAdapter*> SignalDestroyed;
>-
>-  bool IsSrtpActive() const override { return transport_->IsSrtpActive(); }
>-
>- protected:
>-  RtpTransportAdapter* GetInternal() override { return this; }
>-
>- private:
>-  RtpTransportAdapter(const RtcpParameters& rtcp_params,
>-                      PacketTransportInterface* rtp,
>-                      PacketTransportInterface* rtcp,
>-                      RtpTransportControllerAdapter* rtp_transport_controller,
>-                      bool is_srtp_transport);
>-
>-  void OnReadyToSend(bool ready) { SignalReadyToSend(ready); }
>-
>-  void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                            const rtc::PacketTime& time) {
>-    SignalRtcpPacketReceived(packet, time);
>-  }
>-
>-  void OnWritableState(bool writable) { SignalWritableState(writable); }
>-
>-  PacketTransportInterface* rtp_packet_transport_ = nullptr;
>-  PacketTransportInterface* rtcp_packet_transport_ = nullptr;
>-  RtpTransportControllerAdapter* const rtp_transport_controller_ = nullptr;
>-  // Non-null if this class owns the transport controller.
>-  std::unique_ptr<RtpTransportControllerInterface>
>-      owned_rtp_transport_controller_;
>-  RtpTransportParameters parameters_;
>-
>-  // Only one of them is non-null;
>-  std::unique_ptr<RtpTransport> unencrypted_rtp_transport_;
>-  std::unique_ptr<SrtpTransport> srtp_transport_;
>-
>-  rtc::Thread* network_thread_ = nullptr;
>-
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpTransportAdapter);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // ORTC_RTPTRANSPORTADAPTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontroller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontroller_unittest.cc
>deleted file mode 100644
>index 49548417ee7cf67f4297df77a87c6b0f77f397c3..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontroller_unittest.cc
>+++ /dev/null
>@@ -1,199 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-
>-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>-#include "media/base/fakemediaengine.h"
>-#include "ortc/ortcfactory.h"
>-#include "ortc/testrtpparameters.h"
>-#include "p2p/base/fakepackettransport.h"
>-#include "rtc_base/gunit.h"
>-
>-namespace webrtc {
>-
>-// This test uses fake packet transports and a fake media engine, in order to
>-// test the RtpTransportController at only an API level. Any end-to-end test
>-// should go in ortcfactory_integrationtest.cc instead.
>-//
>-// Currently, this test mainly focuses on the limitations of the "adapter"
>-// RtpTransportController implementation. Only one of each type of
>-// sender/receiver can be created, and the sender/receiver of the same media
>-// type must use the same transport.
>-class RtpTransportControllerTest : public testing::Test {
>- public:
>-  RtpTransportControllerTest() {
>-    // Note: This doesn't need to use fake network classes, since it uses
>-    // FakePacketTransports.
>-    auto result = OrtcFactory::Create(
>-        nullptr, nullptr, nullptr, nullptr, nullptr,
>-        std::unique_ptr<cricket::MediaEngineInterface>(
>-            new cricket::FakeMediaEngine()),
>-        CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory());
>-    ortc_factory_ = result.MoveValue();
>-    rtp_transport_controller_ =
>-        ortc_factory_->CreateRtpTransportController().MoveValue();
>-  }
>-
>- protected:
>-  std::unique_ptr<OrtcFactoryInterface> ortc_factory_;
>-  std::unique_ptr<RtpTransportControllerInterface> rtp_transport_controller_;
>-};
>-
>-TEST_F(RtpTransportControllerTest, GetTransports) {
>-  rtc::FakePacketTransport packet_transport1("one");
>-  rtc::FakePacketTransport packet_transport2("two");
>-
>-  auto rtp_transport_result1 = ortc_factory_->CreateRtpTransport(
>-      MakeRtcpMuxParameters(), &packet_transport1, nullptr,
>-      rtp_transport_controller_.get());
>-  ASSERT_TRUE(rtp_transport_result1.ok());
>-
>-  auto rtp_transport_result2 = ortc_factory_->CreateRtpTransport(
>-      MakeRtcpMuxParameters(), &packet_transport2, nullptr,
>-      rtp_transport_controller_.get());
>-  ASSERT_TRUE(rtp_transport_result2.ok());
>-
>-  auto returned_transports = rtp_transport_controller_->GetTransports();
>-  ASSERT_EQ(2u, returned_transports.size());
>-  EXPECT_EQ(rtp_transport_result1.value().get(), returned_transports[0]);
>-  EXPECT_EQ(rtp_transport_result2.value().get(), returned_transports[1]);
>-
>-  // If a transport is deleted, it shouldn't be returned any more.
>-  rtp_transport_result1.MoveValue().reset();
>-  returned_transports = rtp_transport_controller_->GetTransports();
>-  ASSERT_EQ(1u, returned_transports.size());
>-  EXPECT_EQ(rtp_transport_result2.value().get(), returned_transports[0]);
>-}
>-
>-// Create RtpSenders and RtpReceivers on top of RtpTransports controlled by the
>-// same RtpTransportController. Currently only one each of audio/video is
>-// supported.
>-TEST_F(RtpTransportControllerTest, AttachMultipleSendersAndReceivers) {
>-  rtc::FakePacketTransport audio_packet_transport("audio");
>-  rtc::FakePacketTransport video_packet_transport("video");
>-
>-  auto audio_rtp_transport_result = ortc_factory_->CreateRtpTransport(
>-      MakeRtcpMuxParameters(), &audio_packet_transport, nullptr,
>-      rtp_transport_controller_.get());
>-  ASSERT_TRUE(audio_rtp_transport_result.ok());
>-  auto audio_rtp_transport = audio_rtp_transport_result.MoveValue();
>-
>-  auto video_rtp_transport_result = ortc_factory_->CreateRtpTransport(
>-      MakeRtcpMuxParameters(), &video_packet_transport, nullptr,
>-      rtp_transport_controller_.get());
>-  ASSERT_TRUE(video_rtp_transport_result.ok());
>-  auto video_rtp_transport = video_rtp_transport_result.MoveValue();
>-
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
>-  EXPECT_TRUE(audio_sender_result.ok());
>-  auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
>-  EXPECT_TRUE(audio_receiver_result.ok());
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
>-  EXPECT_TRUE(video_sender_result.ok());
>-  auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
>-  EXPECT_TRUE(video_receiver_result.ok());
>-
>-  // Now that we have one each of audio/video senders/receivers, trying to
>-  // create more on top of the same controller is expected to fail.
>-  // TODO(deadbeef): Update this test once multiple senders/receivers on top of
>-  // the same controller is supported.
>-  auto failed_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            failed_sender_result.error().type());
>-  auto failed_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            failed_receiver_result.error().type());
>-  failed_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            failed_sender_result.error().type());
>-  failed_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            failed_receiver_result.error().type());
>-
>-  // If we destroy the existing sender/receiver using a transport controller,
>-  // we should be able to make a new one, despite the above limitation.
>-  audio_sender_result.MoveValue().reset();
>-  audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
>-  EXPECT_TRUE(audio_sender_result.ok());
>-  audio_receiver_result.MoveValue().reset();
>-  audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, audio_rtp_transport.get());
>-  EXPECT_TRUE(audio_receiver_result.ok());
>-  video_sender_result.MoveValue().reset();
>-  video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
>-  EXPECT_TRUE(video_sender_result.ok());
>-  video_receiver_result.MoveValue().reset();
>-  video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, video_rtp_transport.get());
>-  EXPECT_TRUE(video_receiver_result.ok());
>-}
>-
>-// Given the current limitations of the BaseChannel-based implementation, it's
>-// not possible for an audio sender and receiver to use different RtpTransports.
>-// TODO(deadbeef): Once this is supported, update/replace this test.
>-TEST_F(RtpTransportControllerTest,
>-       SenderAndReceiverUsingDifferentTransportsUnsupported) {
>-  rtc::FakePacketTransport packet_transport1("one");
>-  rtc::FakePacketTransport packet_transport2("two");
>-
>-  auto rtp_transport_result1 = ortc_factory_->CreateRtpTransport(
>-      MakeRtcpMuxParameters(), &packet_transport1, nullptr,
>-      rtp_transport_controller_.get());
>-  ASSERT_TRUE(rtp_transport_result1.ok());
>-  auto rtp_transport1 = rtp_transport_result1.MoveValue();
>-
>-  auto rtp_transport_result2 = ortc_factory_->CreateRtpTransport(
>-      MakeRtcpMuxParameters(), &packet_transport2, nullptr,
>-      rtp_transport_controller_.get());
>-  ASSERT_TRUE(rtp_transport_result2.ok());
>-  auto rtp_transport2 = rtp_transport_result2.MoveValue();
>-
>-  // Create an audio sender on transport 1, then try to create a receiver on 2.
>-  auto audio_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport1.get());
>-  EXPECT_TRUE(audio_sender_result.ok());
>-  auto audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport2.get());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            audio_receiver_result.error().type());
>-  // Delete the sender; now we should be ok to create the receiver on 2.
>-  audio_sender_result.MoveValue().reset();
>-  audio_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, rtp_transport2.get());
>-  EXPECT_TRUE(audio_receiver_result.ok());
>-
>-  // Do the same thing for video, reversing 1 and 2 (for variety).
>-  auto video_sender_result = ortc_factory_->CreateRtpSender(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport2.get());
>-  EXPECT_TRUE(video_sender_result.ok());
>-
>-  auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport1.get());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            video_receiver_result.error().type());
>-  video_sender_result.MoveValue().reset();
>-  video_receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, rtp_transport1.get());
>-  EXPECT_TRUE(video_receiver_result.ok());
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.cc
>deleted file mode 100644
>index dbb03ea321f546737ef31fd31eab8648d5d00886..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.cc
>+++ /dev/null
>@@ -1,971 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "ortc/rtptransportcontrolleradapter.h"
>-
>-#include <algorithm>  // For "remove", "find".
>-#include <set>
>-#include <unordered_map>
>-#include <utility>  // For std::move.
>-
>-#include "absl/memory/memory.h"
>-#include "api/proxy.h"
>-#include "media/base/mediaconstants.h"
>-#include "ortc/ortcrtpreceiveradapter.h"
>-#include "ortc/ortcrtpsenderadapter.h"
>-#include "ortc/rtptransportadapter.h"
>-#include "pc/rtpmediautils.h"
>-#include "pc/rtpparametersconversion.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/strings/string_builder.h"
>-
>-namespace webrtc {
>-
>-// Note: It's assumed that each individual list doesn't have conflicts, since
>-// they should have been detected already by rtpparametersconversion.cc. This
>-// only needs to detect conflicts *between* A and B.
>-template <typename C1, typename C2>
>-static RTCError CheckForIdConflicts(
>-    const std::vector<C1>& codecs_a,
>-    const cricket::RtpHeaderExtensions& extensions_a,
>-    const cricket::StreamParamsVec& streams_a,
>-    const std::vector<C2>& codecs_b,
>-    const cricket::RtpHeaderExtensions& extensions_b,
>-    const cricket::StreamParamsVec& streams_b) {
>-  rtc::StringBuilder oss;
>-  // Since it's assumed that C1 and C2 are different types, codecs_a and
>-  // codecs_b should never contain the same payload type, and thus we can just
>-  // use a set.
>-  std::set<int> seen_payload_types;
>-  for (const C1& codec : codecs_a) {
>-    seen_payload_types.insert(codec.id);
>-  }
>-  for (const C2& codec : codecs_b) {
>-    if (!seen_payload_types.insert(codec.id).second) {
>-      oss << "Same payload type used for audio and video codecs: " << codec.id;
>-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, oss.str());
>-    }
>-  }
>-  // Audio and video *may* use the same header extensions, so use a map.
>-  std::unordered_map<int, std::string> seen_extensions;
>-  for (const webrtc::RtpExtension& extension : extensions_a) {
>-    seen_extensions[extension.id] = extension.uri;
>-  }
>-  for (const webrtc::RtpExtension& extension : extensions_b) {
>-    if (seen_extensions.find(extension.id) != seen_extensions.end() &&
>-        seen_extensions.at(extension.id) != extension.uri) {
>-      oss << "Same ID used for different RTP header extensions: "
>-          << extension.id;
>-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, oss.str());
>-    }
>-  }
>-  std::set<uint32_t> seen_ssrcs;
>-  for (const cricket::StreamParams& stream : streams_a) {
>-    seen_ssrcs.insert(stream.ssrcs.begin(), stream.ssrcs.end());
>-  }
>-  for (const cricket::StreamParams& stream : streams_b) {
>-    for (uint32_t ssrc : stream.ssrcs) {
>-      if (!seen_ssrcs.insert(ssrc).second) {
>-        oss << "Same SSRC used for audio and video senders: " << ssrc;
>-        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, oss.str());
>-      }
>-    }
>-  }
>-  return RTCError::OK();
>-}
>-
>-BEGIN_OWNED_PROXY_MAP(RtpTransportController)
>-PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-PROXY_CONSTMETHOD0(std::vector<RtpTransportInterface*>, GetTransports)
>-protected:
>-RtpTransportControllerAdapter* GetInternal() override {
>-  return internal();
>-}
>-END_PROXY_MAP()
>-
>-// static
>-std::unique_ptr<RtpTransportControllerInterface>
>-RtpTransportControllerAdapter::CreateProxied(
>-    const cricket::MediaConfig& config,
>-    cricket::ChannelManager* channel_manager,
>-    webrtc::RtcEventLog* event_log,
>-    rtc::Thread* signaling_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* network_thread) {
>-  std::unique_ptr<RtpTransportControllerAdapter> wrapped(
>-      new RtpTransportControllerAdapter(config, channel_manager, event_log,
>-                                        signaling_thread, worker_thread,
>-                                        network_thread));
>-  return RtpTransportControllerProxyWithInternal<
>-      RtpTransportControllerAdapter>::Create(signaling_thread, worker_thread,
>-                                             std::move(wrapped));
>-}
>-
>-RtpTransportControllerAdapter::~RtpTransportControllerAdapter() {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  if (!transport_proxies_.empty()) {
>-    RTC_LOG(LS_ERROR)
>-        << "Destroying RtpTransportControllerAdapter while RtpTransports "
>-           "are still using it; this is unsafe.";
>-  }
>-  if (voice_channel_) {
>-    // This would mean audio RTP senders/receivers that are using us haven't
>-    // been destroyed. This isn't safe (see error log above).
>-    DestroyVoiceChannel();
>-  }
>-  if (voice_channel_) {
>-    // This would mean video RTP senders/receivers that are using us haven't
>-    // been destroyed. This isn't safe (see error log above).
>-    DestroyVideoChannel();
>-  }
>-  // Call must be destroyed on the worker thread.
>-  worker_thread_->Invoke<void>(
>-      RTC_FROM_HERE, rtc::Bind(&RtpTransportControllerAdapter::Close_w, this));
>-}
>-
>-RTCErrorOr<std::unique_ptr<RtpTransportInterface>>
>-RtpTransportControllerAdapter::CreateProxiedRtpTransport(
>-    const RtpTransportParameters& parameters,
>-    PacketTransportInterface* rtp,
>-    PacketTransportInterface* rtcp) {
>-  if (!transport_proxies_.empty() && (parameters.keepalive != keepalive_)) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
>-                         "Cannot create RtpTransport with different keep-alive "
>-                         "from the RtpTransports already associated with this "
>-                         "transport controller.");
>-  }
>-  auto result = RtpTransportAdapter::CreateProxied(parameters, rtp, rtcp, this);
>-  if (result.ok()) {
>-    transport_proxies_.push_back(result.value().get());
>-    transport_proxies_.back()->GetInternal()->SignalDestroyed.connect(
>-        this, &RtpTransportControllerAdapter::OnRtpTransportDestroyed);
>-  }
>-  return result;
>-}
>-
>-RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
>-RtpTransportControllerAdapter::CreateProxiedSrtpTransport(
>-    const RtpTransportParameters& parameters,
>-    PacketTransportInterface* rtp,
>-    PacketTransportInterface* rtcp) {
>-  auto result =
>-      RtpTransportAdapter::CreateSrtpProxied(parameters, rtp, rtcp, this);
>-  if (result.ok()) {
>-    transport_proxies_.push_back(result.value().get());
>-    transport_proxies_.back()->GetInternal()->SignalDestroyed.connect(
>-        this, &RtpTransportControllerAdapter::OnRtpTransportDestroyed);
>-  }
>-  return result;
>-}
>-
>-RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>>
>-RtpTransportControllerAdapter::CreateProxiedRtpSender(
>-    cricket::MediaType kind,
>-    RtpTransportInterface* transport_proxy) {
>-  RTC_DCHECK(transport_proxy);
>-  RTC_DCHECK(std::find(transport_proxies_.begin(), transport_proxies_.end(),
>-                       transport_proxy) != transport_proxies_.end());
>-  std::unique_ptr<OrtcRtpSenderAdapter> new_sender(
>-      new OrtcRtpSenderAdapter(kind, transport_proxy, this));
>-  RTCError err;
>-  switch (kind) {
>-    case cricket::MEDIA_TYPE_AUDIO:
>-      err = AttachAudioSender(new_sender.get(), transport_proxy->GetInternal());
>-      break;
>-    case cricket::MEDIA_TYPE_VIDEO:
>-      err = AttachVideoSender(new_sender.get(), transport_proxy->GetInternal());
>-      break;
>-    case cricket::MEDIA_TYPE_DATA:
>-      RTC_NOTREACHED();
>-  }
>-  if (!err.ok()) {
>-    return std::move(err);
>-  }
>-
>-  return OrtcRtpSenderAdapter::CreateProxy(std::move(new_sender));
>-}
>-
>-RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>>
>-RtpTransportControllerAdapter::CreateProxiedRtpReceiver(
>-    cricket::MediaType kind,
>-    RtpTransportInterface* transport_proxy) {
>-  RTC_DCHECK(transport_proxy);
>-  RTC_DCHECK(std::find(transport_proxies_.begin(), transport_proxies_.end(),
>-                       transport_proxy) != transport_proxies_.end());
>-  std::unique_ptr<OrtcRtpReceiverAdapter> new_receiver(
>-      new OrtcRtpReceiverAdapter(kind, transport_proxy, this));
>-  RTCError err;
>-  switch (kind) {
>-    case cricket::MEDIA_TYPE_AUDIO:
>-      err = AttachAudioReceiver(new_receiver.get(),
>-                                transport_proxy->GetInternal());
>-      break;
>-    case cricket::MEDIA_TYPE_VIDEO:
>-      err = AttachVideoReceiver(new_receiver.get(),
>-                                transport_proxy->GetInternal());
>-      break;
>-    case cricket::MEDIA_TYPE_DATA:
>-      RTC_NOTREACHED();
>-  }
>-  if (!err.ok()) {
>-    return std::move(err);
>-  }
>-
>-  return OrtcRtpReceiverAdapter::CreateProxy(std::move(new_receiver));
>-}
>-
>-std::vector<RtpTransportInterface*>
>-RtpTransportControllerAdapter::GetTransports() const {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  return transport_proxies_;
>-}
>-
>-RTCError RtpTransportControllerAdapter::SetRtpTransportParameters(
>-    const RtpTransportParameters& parameters,
>-    RtpTransportInterface* inner_transport) {
>-  if ((video_channel_ != nullptr || voice_channel_ != nullptr) &&
>-      (parameters.keepalive != keepalive_)) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
>-                         "Cannot change keep-alive settings after creating "
>-                         "media streams or additional transports for the same "
>-                         "transport controller.");
>-  }
>-  // Call must be configured on the worker thread.
>-  worker_thread_->Invoke<void>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&RtpTransportControllerAdapter::SetRtpTransportParameters_w,
>-                this, parameters));
>-
>-  do {
>-    if (inner_transport == inner_audio_transport_) {
>-      CopyRtcpParametersToDescriptions(parameters.rtcp,
>-                                       &local_audio_description_,
>-                                       &remote_audio_description_);
>-      if (!voice_channel_->SetLocalContent(&local_audio_description_,
>-                                           SdpType::kOffer, nullptr)) {
>-        break;
>-      }
>-      if (!voice_channel_->SetRemoteContent(&remote_audio_description_,
>-                                            SdpType::kAnswer, nullptr)) {
>-        break;
>-      }
>-    } else if (inner_transport == inner_video_transport_) {
>-      CopyRtcpParametersToDescriptions(parameters.rtcp,
>-                                       &local_video_description_,
>-                                       &remote_video_description_);
>-      if (!video_channel_->SetLocalContent(&local_video_description_,
>-                                           SdpType::kOffer, nullptr)) {
>-        break;
>-      }
>-      if (!video_channel_->SetRemoteContent(&remote_video_description_,
>-                                            SdpType::kAnswer, nullptr)) {
>-        break;
>-      }
>-    }
>-    return RTCError::OK();
>-  } while (false);
>-  LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>-                       "Failed to apply new RTCP parameters.");
>-}
>-
>-void RtpTransportControllerAdapter::SetRtpTransportParameters_w(
>-    const RtpTransportParameters& parameters) {
>-  call_send_rtp_transport_controller_->SetKeepAliveConfig(parameters.keepalive);
>-}
>-
>-RTCError RtpTransportControllerAdapter::ValidateAndApplyAudioSenderParameters(
>-    const RtpParameters& parameters,
>-    uint32_t* primary_ssrc) {
>-  RTC_DCHECK(voice_channel_);
>-  RTC_DCHECK(have_audio_sender_);
>-
>-  auto codecs_result = ToCricketCodecs<cricket::AudioCodec>(parameters.codecs);
>-  if (!codecs_result.ok()) {
>-    return codecs_result.MoveError();
>-  }
>-
>-  auto extensions_result =
>-      ToCricketRtpHeaderExtensions(parameters.header_extensions);
>-  if (!extensions_result.ok()) {
>-    return extensions_result.MoveError();
>-  }
>-
>-  auto stream_params_result = MakeSendStreamParamsVec(
>-      parameters.encodings, inner_audio_transport_->GetParameters().rtcp.cname,
>-      local_audio_description_);
>-  if (!stream_params_result.ok()) {
>-    return stream_params_result.MoveError();
>-  }
>-
>-  // Check that audio/video sender aren't using the same IDs to refer to
>-  // different things, if they share the same transport.
>-  if (inner_audio_transport_ == inner_video_transport_) {
>-    RTCError err = CheckForIdConflicts(
>-        codecs_result.value(), extensions_result.value(),
>-        stream_params_result.value(), remote_video_description_.codecs(),
>-        remote_video_description_.rtp_header_extensions(),
>-        local_video_description_.streams());
>-    if (!err.ok()) {
>-      return err;
>-    }
>-  }
>-
>-  bool local_send = false;
>-  int bandwidth = cricket::kAutoBandwidth;
>-  if (parameters.encodings.size() == 1u) {
>-    if (parameters.encodings[0].max_bitrate_bps) {
>-      bandwidth = *parameters.encodings[0].max_bitrate_bps;
>-    }
>-    local_send = parameters.encodings[0].active;
>-  }
>-  const bool local_recv =
>-      RtpTransceiverDirectionHasRecv(local_audio_description_.direction());
>-  const auto local_direction =
>-      RtpTransceiverDirectionFromSendRecv(local_send, local_recv);
>-  if (primary_ssrc && !stream_params_result.value().empty()) {
>-    *primary_ssrc = stream_params_result.value()[0].first_ssrc();
>-  }
>-
>-  // Validation is done, so we can attempt applying the descriptions. Sent
>-  // codecs and header extensions go in remote description, streams go in
>-  // local.
>-  //
>-  // If there are no codecs or encodings, just leave the previous set of
>-  // codecs. The media engine doesn't like an empty set of codecs.
>-  if (local_audio_description_.streams().empty() &&
>-      remote_audio_description_.codecs().empty()) {
>-  } else {
>-    remote_audio_description_.set_codecs(codecs_result.MoveValue());
>-  }
>-  remote_audio_description_.set_rtp_header_extensions(
>-      extensions_result.MoveValue());
>-  remote_audio_description_.set_bandwidth(bandwidth);
>-  local_audio_description_.mutable_streams() = stream_params_result.MoveValue();
>-  // Direction set based on encoding "active" flag.
>-  local_audio_description_.set_direction(local_direction);
>-  remote_audio_description_.set_direction(
>-      RtpTransceiverDirectionReversed(local_direction));
>-
>-  // Set remote content first, to ensure the stream is created with the correct
>-  // codec.
>-  if (!voice_channel_->SetRemoteContent(&remote_audio_description_,
>-                                        SdpType::kOffer, nullptr)) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>-                         "Failed to apply remote parameters to media channel.");
>-  }
>-  if (!voice_channel_->SetLocalContent(&local_audio_description_,
>-                                       SdpType::kAnswer, nullptr)) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>-                         "Failed to apply local parameters to media channel.");
>-  }
>-  return RTCError::OK();
>-}
>-
>-RTCError RtpTransportControllerAdapter::ValidateAndApplyVideoSenderParameters(
>-    const RtpParameters& parameters,
>-    uint32_t* primary_ssrc) {
>-  RTC_DCHECK(video_channel_);
>-  RTC_DCHECK(have_video_sender_);
>-
>-  auto codecs_result = ToCricketCodecs<cricket::VideoCodec>(parameters.codecs);
>-  if (!codecs_result.ok()) {
>-    return codecs_result.MoveError();
>-  }
>-
>-  auto extensions_result =
>-      ToCricketRtpHeaderExtensions(parameters.header_extensions);
>-  if (!extensions_result.ok()) {
>-    return extensions_result.MoveError();
>-  }
>-
>-  auto stream_params_result = MakeSendStreamParamsVec(
>-      parameters.encodings, inner_video_transport_->GetParameters().rtcp.cname,
>-      local_video_description_);
>-  if (!stream_params_result.ok()) {
>-    return stream_params_result.MoveError();
>-  }
>-
>-  // Check that audio/video sender aren't using the same IDs to refer to
>-  // different things, if they share the same transport.
>-  if (inner_audio_transport_ == inner_video_transport_) {
>-    RTCError err = CheckForIdConflicts(
>-        codecs_result.value(), extensions_result.value(),
>-        stream_params_result.value(), remote_audio_description_.codecs(),
>-        remote_audio_description_.rtp_header_extensions(),
>-        local_audio_description_.streams());
>-    if (!err.ok()) {
>-      return err;
>-    }
>-  }
>-
>-  bool local_send = false;
>-  int bandwidth = cricket::kAutoBandwidth;
>-  if (parameters.encodings.size() == 1u) {
>-    if (parameters.encodings[0].max_bitrate_bps) {
>-      bandwidth = *parameters.encodings[0].max_bitrate_bps;
>-    }
>-    local_send = parameters.encodings[0].active;
>-  }
>-  const bool local_recv =
>-      RtpTransceiverDirectionHasRecv(local_audio_description_.direction());
>-  const auto local_direction =
>-      RtpTransceiverDirectionFromSendRecv(local_send, local_recv);
>-  if (primary_ssrc && !stream_params_result.value().empty()) {
>-    *primary_ssrc = stream_params_result.value()[0].first_ssrc();
>-  }
>-
>-  // Validation is done, so we can attempt applying the descriptions. Sent
>-  // codecs and header extensions go in remote description, streams go in
>-  // local.
>-  //
>-  // If there are no codecs or encodings, just leave the previous set of
>-  // codecs. The media engine doesn't like an empty set of codecs.
>-  if (local_video_description_.streams().empty() &&
>-      remote_video_description_.codecs().empty()) {
>-  } else {
>-    remote_video_description_.set_codecs(codecs_result.MoveValue());
>-  }
>-  remote_video_description_.set_rtp_header_extensions(
>-      extensions_result.MoveValue());
>-  remote_video_description_.set_bandwidth(bandwidth);
>-  local_video_description_.mutable_streams() = stream_params_result.MoveValue();
>-  // Direction set based on encoding "active" flag.
>-  local_video_description_.set_direction(local_direction);
>-  remote_video_description_.set_direction(
>-      RtpTransceiverDirectionReversed(local_direction));
>-
>-  // Set remote content first, to ensure the stream is created with the correct
>-  // codec.
>-  if (!video_channel_->SetRemoteContent(&remote_video_description_,
>-                                        SdpType::kOffer, nullptr)) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>-                         "Failed to apply remote parameters to media channel.");
>-  }
>-  if (!video_channel_->SetLocalContent(&local_video_description_,
>-                                       SdpType::kAnswer, nullptr)) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>-                         "Failed to apply local parameters to media channel.");
>-  }
>-  return RTCError::OK();
>-}
>-
>-RTCError RtpTransportControllerAdapter::ValidateAndApplyAudioReceiverParameters(
>-    const RtpParameters& parameters) {
>-  RTC_DCHECK(voice_channel_);
>-  RTC_DCHECK(have_audio_receiver_);
>-
>-  auto codecs_result = ToCricketCodecs<cricket::AudioCodec>(parameters.codecs);
>-  if (!codecs_result.ok()) {
>-    return codecs_result.MoveError();
>-  }
>-
>-  auto extensions_result =
>-      ToCricketRtpHeaderExtensions(parameters.header_extensions);
>-  if (!extensions_result.ok()) {
>-    return extensions_result.MoveError();
>-  }
>-
>-  auto stream_params_result = ToCricketStreamParamsVec(parameters.encodings);
>-  if (!stream_params_result.ok()) {
>-    return stream_params_result.MoveError();
>-  }
>-
>-  // Check that audio/video receive aren't using the same IDs to refer to
>-  // different things, if they share the same transport.
>-  if (inner_audio_transport_ == inner_video_transport_) {
>-    RTCError err = CheckForIdConflicts(
>-        codecs_result.value(), extensions_result.value(),
>-        stream_params_result.value(), local_video_description_.codecs(),
>-        local_video_description_.rtp_header_extensions(),
>-        remote_video_description_.streams());
>-    if (!err.ok()) {
>-      return err;
>-    }
>-  }
>-
>-  const bool local_send =
>-      RtpTransceiverDirectionHasSend(local_audio_description_.direction());
>-  const bool local_recv =
>-      !parameters.encodings.empty() && parameters.encodings[0].active;
>-  const auto local_direction =
>-      RtpTransceiverDirectionFromSendRecv(local_send, local_recv);
>-
>-  // Validation is done, so we can attempt applying the descriptions. Received
>-  // codecs and header extensions go in local description, streams go in
>-  // remote.
>-  //
>-  // If there are no codecs or encodings, just leave the previous set of
>-  // codecs. The media engine doesn't like an empty set of codecs.
>-  if (remote_audio_description_.streams().empty() &&
>-      local_audio_description_.codecs().empty()) {
>-  } else {
>-    local_audio_description_.set_codecs(codecs_result.MoveValue());
>-  }
>-  local_audio_description_.set_rtp_header_extensions(
>-      extensions_result.MoveValue());
>-  remote_audio_description_.mutable_streams() =
>-      stream_params_result.MoveValue();
>-  // Direction set based on encoding "active" flag.
>-  local_audio_description_.set_direction(local_direction);
>-  remote_audio_description_.set_direction(
>-      RtpTransceiverDirectionReversed(local_direction));
>-
>-  if (!voice_channel_->SetLocalContent(&local_audio_description_,
>-                                       SdpType::kOffer, nullptr)) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>-                         "Failed to apply local parameters to media channel.");
>-  }
>-  if (!voice_channel_->SetRemoteContent(&remote_audio_description_,
>-                                        SdpType::kAnswer, nullptr)) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>-                         "Failed to apply remote parameters to media channel.");
>-  }
>-  return RTCError::OK();
>-}
>-
>-RTCError RtpTransportControllerAdapter::ValidateAndApplyVideoReceiverParameters(
>-    const RtpParameters& parameters) {
>-  RTC_DCHECK(video_channel_);
>-  RTC_DCHECK(have_video_receiver_);
>-
>-  auto codecs_result = ToCricketCodecs<cricket::VideoCodec>(parameters.codecs);
>-  if (!codecs_result.ok()) {
>-    return codecs_result.MoveError();
>-  }
>-
>-  auto extensions_result =
>-      ToCricketRtpHeaderExtensions(parameters.header_extensions);
>-  if (!extensions_result.ok()) {
>-    return extensions_result.MoveError();
>-  }
>-
>-  int bandwidth = cricket::kAutoBandwidth;
>-  auto stream_params_result = ToCricketStreamParamsVec(parameters.encodings);
>-  if (!stream_params_result.ok()) {
>-    return stream_params_result.MoveError();
>-  }
>-
>-  // Check that audio/video receiver aren't using the same IDs to refer to
>-  // different things, if they share the same transport.
>-  if (inner_audio_transport_ == inner_video_transport_) {
>-    RTCError err = CheckForIdConflicts(
>-        codecs_result.value(), extensions_result.value(),
>-        stream_params_result.value(), local_audio_description_.codecs(),
>-        local_audio_description_.rtp_header_extensions(),
>-        remote_audio_description_.streams());
>-    if (!err.ok()) {
>-      return err;
>-    }
>-  }
>-
>-  const bool local_send =
>-      RtpTransceiverDirectionHasSend(local_video_description_.direction());
>-  const bool local_recv =
>-      !parameters.encodings.empty() && parameters.encodings[0].active;
>-  const auto local_direction =
>-      RtpTransceiverDirectionFromSendRecv(local_send, local_recv);
>-
>-  // Validation is done, so we can attempt applying the descriptions. Received
>-  // codecs and header extensions go in local description, streams go in
>-  // remote.
>-  //
>-  // If there are no codecs or encodings, just leave the previous set of
>-  // codecs. The media engine doesn't like an empty set of codecs.
>-  if (remote_video_description_.streams().empty() &&
>-      local_video_description_.codecs().empty()) {
>-  } else {
>-    local_video_description_.set_codecs(codecs_result.MoveValue());
>-  }
>-  local_video_description_.set_rtp_header_extensions(
>-      extensions_result.MoveValue());
>-  local_video_description_.set_bandwidth(bandwidth);
>-  remote_video_description_.mutable_streams() =
>-      stream_params_result.MoveValue();
>-  // Direction set based on encoding "active" flag.
>-  local_video_description_.set_direction(local_direction);
>-  remote_video_description_.set_direction(
>-      RtpTransceiverDirectionReversed(local_direction));
>-
>-  if (!video_channel_->SetLocalContent(&local_video_description_,
>-                                       SdpType::kOffer, nullptr)) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>-                         "Failed to apply local parameters to media channel.");
>-  }
>-  if (!video_channel_->SetRemoteContent(&remote_video_description_,
>-                                        SdpType::kAnswer, nullptr)) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>-                         "Failed to apply remote parameters to media channel.");
>-  }
>-  return RTCError::OK();
>-}
>-
>-RtpTransportControllerAdapter::RtpTransportControllerAdapter(
>-    const cricket::MediaConfig& config,
>-    cricket::ChannelManager* channel_manager,
>-    webrtc::RtcEventLog* event_log,
>-    rtc::Thread* signaling_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* network_thread)
>-    : signaling_thread_(signaling_thread),
>-      worker_thread_(worker_thread),
>-      network_thread_(network_thread),
>-      media_config_(config),
>-      channel_manager_(channel_manager),
>-      event_log_(event_log),
>-      call_send_rtp_transport_controller_(nullptr) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  RTC_DCHECK(channel_manager_);
>-  // Add "dummy" codecs to the descriptions, because the media engines
>-  // currently reject empty lists of codecs. Note that these codecs will never
>-  // actually be used, because when parameters are set, the dummy codecs will
>-  // be replaced by actual codecs before any send/receive streams are created.
>-  const cricket::AudioCodec dummy_audio(0, cricket::kPcmuCodecName, 8000, 0, 1);
>-  const cricket::VideoCodec dummy_video(96, cricket::kVp8CodecName);
>-  local_audio_description_.AddCodec(dummy_audio);
>-  remote_audio_description_.AddCodec(dummy_audio);
>-  local_video_description_.AddCodec(dummy_video);
>-  remote_video_description_.AddCodec(dummy_video);
>-
>-  worker_thread_->Invoke<void>(
>-      RTC_FROM_HERE, rtc::Bind(&RtpTransportControllerAdapter::Init_w, this));
>-}
>-
>-// TODO(nisse): Duplicates corresponding method in PeerConnection (used
>-// to be in MediaController).
>-void RtpTransportControllerAdapter::Init_w() {
>-  RTC_DCHECK(worker_thread_->IsCurrent());
>-  RTC_DCHECK(!call_);
>-
>-  const int kMinBandwidthBps = 30000;
>-  const int kStartBandwidthBps = 300000;
>-  const int kMaxBandwidthBps = 2000000;
>-
>-  webrtc::Call::Config call_config(event_log_);
>-  call_config.audio_state = channel_manager_->media_engine()->GetAudioState();
>-  call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
>-  call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
>-  call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
>-  std::unique_ptr<RtpTransportControllerSend> controller_send =
>-      absl::make_unique<RtpTransportControllerSend>(
>-          Clock::GetRealTimeClock(), event_log_,
>-          call_config.network_controller_factory, call_config.bitrate_config);
>-  call_send_rtp_transport_controller_ = controller_send.get();
>-  call_.reset(webrtc::Call::Create(call_config, std::move(controller_send)));
>-}
>-
>-void RtpTransportControllerAdapter::Close_w() {
>-  call_.reset();
>-  call_send_rtp_transport_controller_ = nullptr;
>-}
>-
>-RTCError RtpTransportControllerAdapter::AttachAudioSender(
>-    OrtcRtpSenderAdapter* sender,
>-    RtpTransportInterface* inner_transport) {
>-  if (have_audio_sender_) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
>-                         "Using two audio RtpSenders with the same "
>-                         "RtpTransportControllerAdapter is not currently "
>-                         "supported.");
>-  }
>-  if (inner_audio_transport_ && inner_audio_transport_ != inner_transport) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
>-                         "Using different transports for the audio "
>-                         "RtpSender and RtpReceiver is not currently "
>-                         "supported.");
>-  }
>-
>-  // If setting new transport, extract its RTCP parameters and create voice
>-  // channel.
>-  if (!inner_audio_transport_) {
>-    CopyRtcpParametersToDescriptions(inner_transport->GetParameters().rtcp,
>-                                     &local_audio_description_,
>-                                     &remote_audio_description_);
>-    inner_audio_transport_ = inner_transport;
>-    CreateVoiceChannel();
>-  }
>-  have_audio_sender_ = true;
>-  sender->SignalDestroyed.connect(
>-      this, &RtpTransportControllerAdapter::OnAudioSenderDestroyed);
>-  return RTCError::OK();
>-}
>-
>-RTCError RtpTransportControllerAdapter::AttachVideoSender(
>-    OrtcRtpSenderAdapter* sender,
>-    RtpTransportInterface* inner_transport) {
>-  if (have_video_sender_) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
>-                         "Using two video RtpSenders with the same "
>-                         "RtpTransportControllerAdapter is not currently "
>-                         "supported.");
>-  }
>-  if (inner_video_transport_ && inner_video_transport_ != inner_transport) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
>-                         "Using different transports for the video "
>-                         "RtpSender and RtpReceiver is not currently "
>-                         "supported.");
>-  }
>-
>-  // If setting new transport, extract its RTCP parameters and create video
>-  // channel.
>-  if (!inner_video_transport_) {
>-    CopyRtcpParametersToDescriptions(inner_transport->GetParameters().rtcp,
>-                                     &local_video_description_,
>-                                     &remote_video_description_);
>-    inner_video_transport_ = inner_transport;
>-    CreateVideoChannel();
>-  }
>-  have_video_sender_ = true;
>-  sender->SignalDestroyed.connect(
>-      this, &RtpTransportControllerAdapter::OnVideoSenderDestroyed);
>-  return RTCError::OK();
>-}
>-
>-RTCError RtpTransportControllerAdapter::AttachAudioReceiver(
>-    OrtcRtpReceiverAdapter* receiver,
>-    RtpTransportInterface* inner_transport) {
>-  if (have_audio_receiver_) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
>-                         "Using two audio RtpReceivers with the same "
>-                         "RtpTransportControllerAdapter is not currently "
>-                         "supported.");
>-  }
>-  if (inner_audio_transport_ && inner_audio_transport_ != inner_transport) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
>-                         "Using different transports for the audio "
>-                         "RtpReceiver and RtpReceiver is not currently "
>-                         "supported.");
>-  }
>-
>-  // If setting new transport, extract its RTCP parameters and create voice
>-  // channel.
>-  if (!inner_audio_transport_) {
>-    CopyRtcpParametersToDescriptions(inner_transport->GetParameters().rtcp,
>-                                     &local_audio_description_,
>-                                     &remote_audio_description_);
>-    inner_audio_transport_ = inner_transport;
>-    CreateVoiceChannel();
>-  }
>-  have_audio_receiver_ = true;
>-  receiver->SignalDestroyed.connect(
>-      this, &RtpTransportControllerAdapter::OnAudioReceiverDestroyed);
>-  return RTCError::OK();
>-}
>-
>-RTCError RtpTransportControllerAdapter::AttachVideoReceiver(
>-    OrtcRtpReceiverAdapter* receiver,
>-    RtpTransportInterface* inner_transport) {
>-  if (have_video_receiver_) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
>-                         "Using two video RtpReceivers with the same "
>-                         "RtpTransportControllerAdapter is not currently "
>-                         "supported.");
>-  }
>-  if (inner_video_transport_ && inner_video_transport_ != inner_transport) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
>-                         "Using different transports for the video "
>-                         "RtpReceiver and RtpReceiver is not currently "
>-                         "supported.");
>-  }
>-  // If setting new transport, extract its RTCP parameters and create video
>-  // channel.
>-  if (!inner_video_transport_) {
>-    CopyRtcpParametersToDescriptions(inner_transport->GetParameters().rtcp,
>-                                     &local_video_description_,
>-                                     &remote_video_description_);
>-    inner_video_transport_ = inner_transport;
>-    CreateVideoChannel();
>-  }
>-  have_video_receiver_ = true;
>-  receiver->SignalDestroyed.connect(
>-      this, &RtpTransportControllerAdapter::OnVideoReceiverDestroyed);
>-  return RTCError::OK();
>-}
>-
>-void RtpTransportControllerAdapter::OnRtpTransportDestroyed(
>-    RtpTransportAdapter* transport) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  auto it = std::find_if(transport_proxies_.begin(), transport_proxies_.end(),
>-                         [transport](RtpTransportInterface* proxy) {
>-                           return proxy->GetInternal() == transport;
>-                         });
>-  if (it == transport_proxies_.end()) {
>-    RTC_NOTREACHED();
>-    return;
>-  }
>-  transport_proxies_.erase(it);
>-}
>-
>-void RtpTransportControllerAdapter::OnAudioSenderDestroyed() {
>-  if (!have_audio_sender_) {
>-    RTC_NOTREACHED();
>-    return;
>-  }
>-  // Empty parameters should result in sending being stopped.
>-  RTCError err =
>-      ValidateAndApplyAudioSenderParameters(RtpParameters(), nullptr);
>-  RTC_DCHECK(err.ok());
>-  have_audio_sender_ = false;
>-  if (!have_audio_receiver_) {
>-    DestroyVoiceChannel();
>-  }
>-}
>-
>-void RtpTransportControllerAdapter::OnVideoSenderDestroyed() {
>-  if (!have_video_sender_) {
>-    RTC_NOTREACHED();
>-    return;
>-  }
>-  // Empty parameters should result in sending being stopped.
>-  RTCError err =
>-      ValidateAndApplyVideoSenderParameters(RtpParameters(), nullptr);
>-  RTC_DCHECK(err.ok());
>-  have_video_sender_ = false;
>-  if (!have_video_receiver_) {
>-    DestroyVideoChannel();
>-  }
>-}
>-
>-void RtpTransportControllerAdapter::OnAudioReceiverDestroyed() {
>-  if (!have_audio_receiver_) {
>-    RTC_NOTREACHED();
>-    return;
>-  }
>-  // Empty parameters should result in receiving being stopped.
>-  RTCError err = ValidateAndApplyAudioReceiverParameters(RtpParameters());
>-  RTC_DCHECK(err.ok());
>-  have_audio_receiver_ = false;
>-  if (!have_audio_sender_) {
>-    DestroyVoiceChannel();
>-  }
>-}
>-
>-void RtpTransportControllerAdapter::OnVideoReceiverDestroyed() {
>-  if (!have_video_receiver_) {
>-    RTC_NOTREACHED();
>-    return;
>-  }
>-  // Empty parameters should result in receiving being stopped.
>-  RTCError err = ValidateAndApplyVideoReceiverParameters(RtpParameters());
>-  RTC_DCHECK(err.ok());
>-  have_video_receiver_ = false;
>-  if (!have_video_sender_) {
>-    DestroyVideoChannel();
>-  }
>-}
>-
>-void RtpTransportControllerAdapter::CreateVoiceChannel() {
>-  voice_channel_ = channel_manager_->CreateVoiceChannel(
>-      call_.get(), media_config_, inner_audio_transport_->GetInternal(),
>-      signaling_thread_, "audio", false, rtc::CryptoOptions(),
>-      cricket::AudioOptions());
>-  RTC_DCHECK(voice_channel_);
>-  voice_channel_->Enable(true);
>-}
>-
>-void RtpTransportControllerAdapter::CreateVideoChannel() {
>-  video_channel_ = channel_manager_->CreateVideoChannel(
>-      call_.get(), media_config_, inner_video_transport_->GetInternal(),
>-      signaling_thread_, "video", false, rtc::CryptoOptions(),
>-      cricket::VideoOptions());
>-  RTC_DCHECK(video_channel_);
>-  video_channel_->Enable(true);
>-}
>-
>-void RtpTransportControllerAdapter::DestroyVoiceChannel() {
>-  RTC_DCHECK(voice_channel_);
>-  channel_manager_->DestroyVoiceChannel(voice_channel_);
>-  voice_channel_ = nullptr;
>-  inner_audio_transport_ = nullptr;
>-}
>-
>-void RtpTransportControllerAdapter::DestroyVideoChannel() {
>-  RTC_DCHECK(video_channel_);
>-  channel_manager_->DestroyVideoChannel(video_channel_);
>-  video_channel_ = nullptr;
>-  inner_video_transport_ = nullptr;
>-}
>-
>-void RtpTransportControllerAdapter::CopyRtcpParametersToDescriptions(
>-    const RtcpParameters& params,
>-    cricket::MediaContentDescription* local,
>-    cricket::MediaContentDescription* remote) {
>-  local->set_rtcp_mux(params.mux);
>-  remote->set_rtcp_mux(params.mux);
>-  local->set_rtcp_reduced_size(params.reduced_size);
>-  remote->set_rtcp_reduced_size(params.reduced_size);
>-  for (cricket::StreamParams& stream_params : local->mutable_streams()) {
>-    stream_params.cname = params.cname;
>-  }
>-}
>-
>-uint32_t RtpTransportControllerAdapter::GenerateUnusedSsrc(
>-    std::set<uint32_t>* new_ssrcs) const {
>-  uint32_t ssrc;
>-  do {
>-    ssrc = rtc::CreateRandomNonZeroId();
>-  } while (
>-      cricket::GetStreamBySsrc(local_audio_description_.streams(), ssrc) ||
>-      cricket::GetStreamBySsrc(remote_audio_description_.streams(), ssrc) ||
>-      cricket::GetStreamBySsrc(local_video_description_.streams(), ssrc) ||
>-      cricket::GetStreamBySsrc(remote_video_description_.streams(), ssrc) ||
>-      !new_ssrcs->insert(ssrc).second);
>-  return ssrc;
>-}
>-
>-RTCErrorOr<cricket::StreamParamsVec>
>-RtpTransportControllerAdapter::MakeSendStreamParamsVec(
>-    std::vector<RtpEncodingParameters> encodings,
>-    const std::string& cname,
>-    const cricket::MediaContentDescription& description) const {
>-  if (encodings.size() > 1u) {
>-    LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::UNSUPPORTED_PARAMETER,
>-                         "ORTC API implementation doesn't currently "
>-                         "support simulcast or layered encodings.");
>-  } else if (encodings.empty()) {
>-    return cricket::StreamParamsVec();
>-  }
>-  RtpEncodingParameters& encoding = encodings[0];
>-  std::set<uint32_t> new_ssrcs;
>-  if (encoding.ssrc) {
>-    new_ssrcs.insert(*encoding.ssrc);
>-  }
>-  if (encoding.rtx && encoding.rtx->ssrc) {
>-    new_ssrcs.insert(*encoding.rtx->ssrc);
>-  }
>-  // May need to fill missing SSRCs with generated ones.
>-  if (!encoding.ssrc) {
>-    if (!description.streams().empty()) {
>-      encoding.ssrc.emplace(description.streams()[0].first_ssrc());
>-    } else {
>-      encoding.ssrc.emplace(GenerateUnusedSsrc(&new_ssrcs));
>-    }
>-  }
>-  if (encoding.rtx && !encoding.rtx->ssrc) {
>-    uint32_t existing_rtx_ssrc;
>-    if (!description.streams().empty() &&
>-        description.streams()[0].GetFidSsrc(
>-            description.streams()[0].first_ssrc(), &existing_rtx_ssrc)) {
>-      encoding.rtx->ssrc.emplace(existing_rtx_ssrc);
>-    } else {
>-      encoding.rtx->ssrc.emplace(GenerateUnusedSsrc(&new_ssrcs));
>-    }
>-  }
>-
>-  auto result = ToCricketStreamParamsVec(encodings);
>-  if (!result.ok()) {
>-    return result.MoveError();
>-  }
>-  // If conversion was successful, there should be one StreamParams.
>-  RTC_DCHECK_EQ(1u, result.value().size());
>-  result.value()[0].cname = cname;
>-  return result;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.h
>deleted file mode 100644
>index 14c578aa46f5caabb770d2340de7d44c59cab5bb..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.h
>+++ /dev/null
>@@ -1,221 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef ORTC_RTPTRANSPORTCONTROLLERADAPTER_H_
>-#define ORTC_RTPTRANSPORTCONTROLLERADAPTER_H_
>-
>-#include <memory>
>-#include <set>
>-#include <string>
>-#include <vector>
>-
>-#include "api/ortc/ortcrtpreceiverinterface.h"
>-#include "api/ortc/ortcrtpsenderinterface.h"
>-#include "api/ortc/rtptransportcontrollerinterface.h"
>-#include "api/ortc/srtptransportinterface.h"
>-#include "call/call.h"
>-#include "call/rtp_transport_controller_send.h"
>-#include "logging/rtc_event_log/rtc_event_log.h"
>-#include "media/base/mediachannel.h"  // For MediaConfig.
>-#include "pc/channelmanager.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/third_party/sigslot/sigslot.h"
>-#include "rtc_base/thread.h"
>-
>-namespace webrtc {
>-
>-class RtpTransportAdapter;
>-class OrtcRtpSenderAdapter;
>-class OrtcRtpReceiverAdapter;
>-
>-// Implementation of RtpTransportControllerInterface. Wraps a Call,
>-// a VoiceChannel and VideoChannel, and maintains a list of dependent RTP
>-// transports.
>-//
>-// When used along with an RtpSenderAdapter or RtpReceiverAdapter, the
>-// sender/receiver passes its parameters along to this class, which turns them
>-// into cricket:: media descriptions (the interface used by BaseChannel).
>-//
>-// Due to the fact that BaseChannel has different subclasses for audio/video,
>-// the actual BaseChannel object is not created until an RtpSender/RtpReceiver
>-// needs them.
>-//
>-// All methods should be called on the signaling thread.
>-//
>-// TODO(deadbeef): When BaseChannel is split apart into separate
>-// "RtpSender"/"RtpTransceiver"/"RtpSender"/"RtpReceiver" objects, this adapter
>-// object can be replaced by a "real" one.
>-class RtpTransportControllerAdapter : public RtpTransportControllerInterface,
>-                                      public sigslot::has_slots<> {
>- public:
>-  // Creates a proxy that will call "public interface" methods on the correct
>-  // thread.
>-  //
>-  // Doesn't take ownership of any objects passed in.
>-  //
>-  // |channel_manager| must not be null.
>-  static std::unique_ptr<RtpTransportControllerInterface> CreateProxied(
>-      const cricket::MediaConfig& config,
>-      cricket::ChannelManager* channel_manager,
>-      webrtc::RtcEventLog* event_log,
>-      rtc::Thread* signaling_thread,
>-      rtc::Thread* worker_thread,
>-      rtc::Thread* network_thread);
>-
>-  ~RtpTransportControllerAdapter() override;
>-
>-  // RtpTransportControllerInterface implementation.
>-  std::vector<RtpTransportInterface*> GetTransports() const override;
>-
>-  // These methods are used by OrtcFactory to create RtpTransports, RtpSenders
>-  // and RtpReceivers using this controller. Called "CreateProxied" because
>-  // these methods return proxies that will safely call methods on the correct
>-  // thread.
>-  RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateProxiedRtpTransport(
>-      const RtpTransportParameters& rtcp_parameters,
>-      PacketTransportInterface* rtp,
>-      PacketTransportInterface* rtcp);
>-
>-  RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
>-  CreateProxiedSrtpTransport(const RtpTransportParameters& rtcp_parameters,
>-                             PacketTransportInterface* rtp,
>-                             PacketTransportInterface* rtcp);
>-
>-  // |transport_proxy| needs to be a proxy to a transport because the
>-  // application may call GetTransport() on the returned sender or receiver,
>-  // and expects it to return a thread-safe transport proxy.
>-  RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateProxiedRtpSender(
>-      cricket::MediaType kind,
>-      RtpTransportInterface* transport_proxy);
>-  RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>>
>-  CreateProxiedRtpReceiver(cricket::MediaType kind,
>-                           RtpTransportInterface* transport_proxy);
>-
>-  // Methods used internally by other "adapter" classes.
>-  rtc::Thread* signaling_thread() const { return signaling_thread_; }
>-  rtc::Thread* worker_thread() const { return worker_thread_; }
>-  rtc::Thread* network_thread() const { return network_thread_; }
>-
>-  // |parameters.keepalive| will be set for ALL RTP transports in the call.
>-  RTCError SetRtpTransportParameters(const RtpTransportParameters& parameters,
>-                                     RtpTransportInterface* inner_transport);
>-  void SetRtpTransportParameters_w(const RtpTransportParameters& parameters);
>-
>-  cricket::VoiceChannel* voice_channel() { return voice_channel_; }
>-  cricket::VideoChannel* video_channel() { return video_channel_; }
>-
>-  // |primary_ssrc| out parameter is filled with either
>-  // |parameters.encodings[0].ssrc|, or a generated SSRC if that's left unset.
>-  RTCError ValidateAndApplyAudioSenderParameters(
>-      const RtpParameters& parameters,
>-      uint32_t* primary_ssrc);
>-  RTCError ValidateAndApplyVideoSenderParameters(
>-      const RtpParameters& parameters,
>-      uint32_t* primary_ssrc);
>-  RTCError ValidateAndApplyAudioReceiverParameters(
>-      const RtpParameters& parameters);
>-  RTCError ValidateAndApplyVideoReceiverParameters(
>-      const RtpParameters& parameters);
>-
>- protected:
>-  RtpTransportControllerAdapter* GetInternal() override { return this; }
>-
>- private:
>-  // Only expected to be called by RtpTransportControllerAdapter::CreateProxied.
>-  RtpTransportControllerAdapter(const cricket::MediaConfig& config,
>-                                cricket::ChannelManager* channel_manager,
>-                                webrtc::RtcEventLog* event_log,
>-                                rtc::Thread* signaling_thread,
>-                                rtc::Thread* worker_thread,
>-                                rtc::Thread* network_thread);
>-  void Init_w();
>-  void Close_w();
>-
>-  // These return an error if another of the same type of object is already
>-  // attached, or if |transport_proxy| can't be used with the sender/receiver
>-  // due to the limitation that the sender/receiver of the same media type must
>-  // use the same transport.
>-  RTCError AttachAudioSender(OrtcRtpSenderAdapter* sender,
>-                             RtpTransportInterface* inner_transport);
>-  RTCError AttachVideoSender(OrtcRtpSenderAdapter* sender,
>-                             RtpTransportInterface* inner_transport);
>-  RTCError AttachAudioReceiver(OrtcRtpReceiverAdapter* receiver,
>-                               RtpTransportInterface* inner_transport);
>-  RTCError AttachVideoReceiver(OrtcRtpReceiverAdapter* receiver,
>-                               RtpTransportInterface* inner_transport);
>-
>-  void OnRtpTransportDestroyed(RtpTransportAdapter* transport);
>-
>-  void OnAudioSenderDestroyed();
>-  void OnVideoSenderDestroyed();
>-  void OnAudioReceiverDestroyed();
>-  void OnVideoReceiverDestroyed();
>-
>-  void CreateVoiceChannel();
>-  void CreateVideoChannel();
>-  void DestroyVoiceChannel();
>-  void DestroyVideoChannel();
>-
>-  void CopyRtcpParametersToDescriptions(
>-      const RtcpParameters& params,
>-      cricket::MediaContentDescription* local,
>-      cricket::MediaContentDescription* remote);
>-
>-  // Helper function to generate an SSRC that doesn't match one in any of the
>-  // "content description" structs, or in |new_ssrcs| (which is needed since
>-  // multiple SSRCs may be generated in one go).
>-  uint32_t GenerateUnusedSsrc(std::set<uint32_t>* new_ssrcs) const;
>-
>-  // |description| is the matching description where existing SSRCs can be
>-  // found.
>-  //
>-  // This is a member function because it may need to generate SSRCs that don't
>-  // match existing ones, which is more than ToStreamParamsVec does.
>-  RTCErrorOr<cricket::StreamParamsVec> MakeSendStreamParamsVec(
>-      std::vector<RtpEncodingParameters> encodings,
>-      const std::string& cname,
>-      const cricket::MediaContentDescription& description) const;
>-
>-  rtc::Thread* signaling_thread_;
>-  rtc::Thread* worker_thread_;
>-  rtc::Thread* network_thread_;
>-  // |transport_proxies_| and |inner_audio_transport_|/|inner_audio_transport_|
>-  // are somewhat redundant, but the latter are only set when
>-  // RtpSenders/RtpReceivers are attached to the transport.
>-  std::vector<RtpTransportInterface*> transport_proxies_;
>-  RtpTransportInterface* inner_audio_transport_ = nullptr;
>-  RtpTransportInterface* inner_video_transport_ = nullptr;
>-  const cricket::MediaConfig media_config_;
>-  RtpKeepAliveConfig keepalive_;
>-  cricket::ChannelManager* channel_manager_;
>-  webrtc::RtcEventLog* event_log_;
>-  std::unique_ptr<Call> call_;
>-  webrtc::RtpTransportControllerSend* call_send_rtp_transport_controller_;
>-
>-  // BaseChannel takes content descriptions as input, so we store them here
>-  // such that they can be updated when a new RtpSenderAdapter/
>-  // RtpReceiverAdapter attaches itself.
>-  cricket::AudioContentDescription local_audio_description_;
>-  cricket::AudioContentDescription remote_audio_description_;
>-  cricket::VideoContentDescription local_video_description_;
>-  cricket::VideoContentDescription remote_video_description_;
>-  cricket::VoiceChannel* voice_channel_ = nullptr;
>-  cricket::VideoChannel* video_channel_ = nullptr;
>-  bool have_audio_sender_ = false;
>-  bool have_video_sender_ = false;
>-  bool have_audio_receiver_ = false;
>-  bool have_video_receiver_ = false;
>-
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpTransportControllerAdapter);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // ORTC_RTPTRANSPORTCONTROLLERADAPTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/srtptransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/srtptransport_unittest.cc
>deleted file mode 100644
>index 1506aa70414e29ae27b0fb9ad81d2d498fe212ac..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/srtptransport_unittest.cc
>+++ /dev/null
>@@ -1,125 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-
>-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>-#include "media/base/fakemediaengine.h"
>-#include "ortc/ortcfactory.h"
>-#include "ortc/testrtpparameters.h"
>-#include "p2p/base/fakepackettransport.h"
>-#include "rtc_base/gunit.h"
>-
>-namespace webrtc {
>-
>-static const char kTestSha1KeyParams1[] =
>-    "inline:WVNfX19zZW1jdGwgKCkgewkyMjA7fQp9CnVubGVz";
>-static const char kTestSha1KeyParams2[] =
>-    "inline:PS1uQCVeeCFCanVmcjkpPywjNWhcYD0mXXtxaVBR";
>-static const char kTestGcmKeyParams3[] =
>-    "inline:e166KFlKzJsGW0d5apX+rrI05vxbrvMJEzFI14aTDCa63IRTlLK4iH66uOI=";
>-
>-static const cricket::CryptoParams kTestSha1CryptoParams1(
>-    1,
>-    "AES_CM_128_HMAC_SHA1_80",
>-    kTestSha1KeyParams1,
>-    "");
>-static const cricket::CryptoParams kTestSha1CryptoParams2(
>-    1,
>-    "AES_CM_128_HMAC_SHA1_80",
>-    kTestSha1KeyParams2,
>-    "");
>-static const cricket::CryptoParams kTestGcmCryptoParams3(1,
>-                                                         "AEAD_AES_256_GCM",
>-                                                         kTestGcmKeyParams3,
>-                                                         "");
>-
>-// This test uses fake packet transports and a fake media engine, in order to
>-// test the SrtpTransport at only an API level. Any end-to-end test should go in
>-// ortcfactory_integrationtest.cc instead.
>-class SrtpTransportTest : public testing::Test {
>- public:
>-  SrtpTransportTest() {
>-    fake_media_engine_ = new cricket::FakeMediaEngine();
>-    // Note: This doesn't need to use fake network classes, since it uses
>-    // FakePacketTransports.
>-    auto result = OrtcFactory::Create(
>-        nullptr, nullptr, nullptr, nullptr, nullptr,
>-        std::unique_ptr<cricket::MediaEngineInterface>(fake_media_engine_),
>-        CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory());
>-    ortc_factory_ = result.MoveValue();
>-    rtp_transport_controller_ =
>-        ortc_factory_->CreateRtpTransportController().MoveValue();
>-
>-    fake_packet_transport_.reset(new rtc::FakePacketTransport("fake"));
>-    auto srtp_transport_result = ortc_factory_->CreateSrtpTransport(
>-        rtp_transport_parameters_, fake_packet_transport_.get(), nullptr,
>-        rtp_transport_controller_.get());
>-    srtp_transport_ = srtp_transport_result.MoveValue();
>-  }
>-
>- protected:
>-  // Owned by |ortc_factory_|.
>-  cricket::FakeMediaEngine* fake_media_engine_;
>-  std::unique_ptr<OrtcFactoryInterface> ortc_factory_;
>-  std::unique_ptr<RtpTransportControllerInterface> rtp_transport_controller_;
>-  std::unique_ptr<SrtpTransportInterface> srtp_transport_;
>-  RtpTransportParameters rtp_transport_parameters_;
>-  std::unique_ptr<rtc::FakePacketTransport> fake_packet_transport_;
>-};
>-
>-// Tests that setting the SRTP send/receive key succeeds.
>-TEST_F(SrtpTransportTest, SetSrtpSendAndReceiveKey) {
>-  EXPECT_TRUE(srtp_transport_->SetSrtpSendKey(kTestSha1CryptoParams1).ok());
>-  EXPECT_TRUE(srtp_transport_->SetSrtpReceiveKey(kTestSha1CryptoParams2).ok());
>-  auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO,
>-                                                      srtp_transport_.get());
>-  EXPECT_TRUE(sender_result.ok());
>-  auto receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, srtp_transport_.get());
>-  EXPECT_TRUE(receiver_result.ok());
>-}
>-
>-// Tests that setting the SRTP send/receive key twice is not supported.
>-TEST_F(SrtpTransportTest, SetSrtpSendAndReceiveKeyTwice) {
>-  EXPECT_TRUE(srtp_transport_->SetSrtpSendKey(kTestSha1CryptoParams1).ok());
>-  EXPECT_TRUE(srtp_transport_->SetSrtpReceiveKey(kTestSha1CryptoParams2).ok());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            srtp_transport_->SetSrtpSendKey(kTestSha1CryptoParams2).type());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            srtp_transport_->SetSrtpReceiveKey(kTestSha1CryptoParams1).type());
>-  // Ensure that the senders and receivers can be created despite the previous
>-  // errors.
>-  auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO,
>-                                                      srtp_transport_.get());
>-  EXPECT_TRUE(sender_result.ok());
>-  auto receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, srtp_transport_.get());
>-  EXPECT_TRUE(receiver_result.ok());
>-}
>-
>-// Test that the SRTP send key and receive key must have the same cipher suite.
>-TEST_F(SrtpTransportTest, SetSrtpSendAndReceiveKeyDifferentCipherSuite) {
>-  EXPECT_TRUE(srtp_transport_->SetSrtpSendKey(kTestSha1CryptoParams1).ok());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>-            srtp_transport_->SetSrtpReceiveKey(kTestGcmCryptoParams3).type());
>-  EXPECT_TRUE(srtp_transport_->SetSrtpReceiveKey(kTestSha1CryptoParams2).ok());
>-  // Ensure that the senders and receivers can be created despite the previous
>-  // error.
>-  auto sender_result = ortc_factory_->CreateRtpSender(cricket::MEDIA_TYPE_AUDIO,
>-                                                      srtp_transport_.get());
>-  EXPECT_TRUE(sender_result.ok());
>-  auto receiver_result = ortc_factory_->CreateRtpReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, srtp_transport_.get());
>-  EXPECT_TRUE(receiver_result.ok());
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/testrtpparameters.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/testrtpparameters.cc
>deleted file mode 100644
>index 82d96590fb9655ea7885305724893116f67af29f..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/testrtpparameters.cc
>+++ /dev/null
>@@ -1,315 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "ortc/testrtpparameters.h"
>-
>-#include <algorithm>
>-#include <utility>
>-
>-namespace webrtc {
>-
>-RtpParameters MakeMinimalOpusParameters() {
>-  RtpParameters parameters;
>-  RtpCodecParameters opus_codec;
>-  opus_codec.name = "opus";
>-  opus_codec.kind = cricket::MEDIA_TYPE_AUDIO;
>-  opus_codec.payload_type = 111;
>-  opus_codec.clock_rate.emplace(48000);
>-  opus_codec.num_channels.emplace(2);
>-  parameters.codecs.push_back(std::move(opus_codec));
>-  RtpEncodingParameters encoding;
>-  encoding.codec_payload_type.emplace(111);
>-  parameters.encodings.push_back(std::move(encoding));
>-  return parameters;
>-}
>-
>-RtpParameters MakeMinimalIsacParameters() {
>-  RtpParameters parameters;
>-  RtpCodecParameters isac_codec;
>-  isac_codec.name = "ISAC";
>-  isac_codec.kind = cricket::MEDIA_TYPE_AUDIO;
>-  isac_codec.payload_type = 103;
>-  isac_codec.clock_rate.emplace(16000);
>-  parameters.codecs.push_back(std::move(isac_codec));
>-  RtpEncodingParameters encoding;
>-  encoding.codec_payload_type.emplace(111);
>-  parameters.encodings.push_back(std::move(encoding));
>-  return parameters;
>-}
>-
>-RtpParameters MakeMinimalOpusParametersWithSsrc(uint32_t ssrc) {
>-  RtpParameters parameters = MakeMinimalOpusParameters();
>-  parameters.encodings[0].ssrc.emplace(ssrc);
>-  return parameters;
>-}
>-
>-RtpParameters MakeMinimalIsacParametersWithSsrc(uint32_t ssrc) {
>-  RtpParameters parameters = MakeMinimalIsacParameters();
>-  parameters.encodings[0].ssrc.emplace(ssrc);
>-  return parameters;
>-}
>-
>-RtpParameters MakeMinimalVideoParameters(const char* codec_name) {
>-  RtpParameters parameters;
>-  RtpCodecParameters vp8_codec;
>-  vp8_codec.name = codec_name;
>-  vp8_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp8_codec.payload_type = 96;
>-  parameters.codecs.push_back(std::move(vp8_codec));
>-  RtpEncodingParameters encoding;
>-  encoding.codec_payload_type.emplace(96);
>-  parameters.encodings.push_back(std::move(encoding));
>-  return parameters;
>-}
>-
>-RtpParameters MakeMinimalVp8Parameters() {
>-  return MakeMinimalVideoParameters("VP8");
>-}
>-
>-RtpParameters MakeMinimalVp9Parameters() {
>-  return MakeMinimalVideoParameters("VP9");
>-}
>-
>-RtpParameters MakeMinimalVp8ParametersWithSsrc(uint32_t ssrc) {
>-  RtpParameters parameters = MakeMinimalVp8Parameters();
>-  parameters.encodings[0].ssrc.emplace(ssrc);
>-  return parameters;
>-}
>-
>-RtpParameters MakeMinimalVp9ParametersWithSsrc(uint32_t ssrc) {
>-  RtpParameters parameters = MakeMinimalVp9Parameters();
>-  parameters.encodings[0].ssrc.emplace(ssrc);
>-  return parameters;
>-}
>-
>-// Note: Currently, these "WithNoSsrc" methods are identical to the normal
>-// "MakeMinimal" methods, but with the added guarantee that they will never be
>-// changed to include an SSRC.
>-
>-RtpParameters MakeMinimalOpusParametersWithNoSsrc() {
>-  RtpParameters parameters = MakeMinimalOpusParameters();
>-  RTC_DCHECK(!parameters.encodings[0].ssrc);
>-  return parameters;
>-}
>-
>-RtpParameters MakeMinimalIsacParametersWithNoSsrc() {
>-  RtpParameters parameters = MakeMinimalIsacParameters();
>-  RTC_DCHECK(!parameters.encodings[0].ssrc);
>-  return parameters;
>-}
>-
>-RtpParameters MakeMinimalVp8ParametersWithNoSsrc() {
>-  RtpParameters parameters = MakeMinimalVp8Parameters();
>-  RTC_DCHECK(!parameters.encodings[0].ssrc);
>-  return parameters;
>-}
>-
>-RtpParameters MakeMinimalVp9ParametersWithNoSsrc() {
>-  RtpParameters parameters = MakeMinimalVp9Parameters();
>-  RTC_DCHECK(!parameters.encodings[0].ssrc);
>-  return parameters;
>-}
>-
>-// Make audio parameters with all the available properties configured and
>-// features used, and with multiple codecs offered. Obtained by taking a
>-// snapshot of a default PeerConnection offer (and adding other things, like
>-// bitrate limit).
>-//
>-// See "MakeFullOpusParameters"/"MakeFullIsacParameters" below.
>-RtpParameters MakeFullAudioParameters(int preferred_payload_type) {
>-  RtpParameters parameters;
>-
>-  RtpCodecParameters opus_codec;
>-  opus_codec.name = "opus";
>-  opus_codec.kind = cricket::MEDIA_TYPE_AUDIO;
>-  opus_codec.payload_type = 111;
>-  opus_codec.clock_rate.emplace(48000);
>-  opus_codec.num_channels.emplace(2);
>-  opus_codec.parameters["minptime"] = "10";
>-  opus_codec.parameters["useinbandfec"] = "1";
>-  opus_codec.parameters["usedtx"] = "1";
>-  opus_codec.parameters["stereo"] = "1";
>-  opus_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::TRANSPORT_CC);
>-  parameters.codecs.push_back(std::move(opus_codec));
>-
>-  RtpCodecParameters isac_codec;
>-  isac_codec.name = "ISAC";
>-  isac_codec.kind = cricket::MEDIA_TYPE_AUDIO;
>-  isac_codec.payload_type = 103;
>-  isac_codec.clock_rate.emplace(16000);
>-  parameters.codecs.push_back(std::move(isac_codec));
>-
>-  RtpCodecParameters cn_codec;
>-  cn_codec.name = "CN";
>-  cn_codec.kind = cricket::MEDIA_TYPE_AUDIO;
>-  cn_codec.payload_type = 106;
>-  cn_codec.clock_rate.emplace(32000);
>-  parameters.codecs.push_back(std::move(cn_codec));
>-
>-  RtpCodecParameters dtmf_codec;
>-  dtmf_codec.name = "telephone-event";
>-  dtmf_codec.kind = cricket::MEDIA_TYPE_AUDIO;
>-  dtmf_codec.payload_type = 126;
>-  dtmf_codec.clock_rate.emplace(8000);
>-  parameters.codecs.push_back(std::move(dtmf_codec));
>-
>-  // "codec_payload_type" isn't implemented, so we need to reorder codecs to
>-  // cause one to be used.
>-  // TODO(deadbeef): Remove this when it becomes unnecessary.
>-  auto it = std::find_if(parameters.codecs.begin(), parameters.codecs.end(),
>-                         [preferred_payload_type](const RtpCodecParameters& p) {
>-                           return p.payload_type == preferred_payload_type;
>-                         });
>-  RtpCodecParameters preferred = *it;
>-  parameters.codecs.erase(it);
>-  parameters.codecs.insert(parameters.codecs.begin(), preferred);
>-
>-  // Intentionally leave out SSRC so one's chosen automatically.
>-  RtpEncodingParameters encoding;
>-  encoding.codec_payload_type.emplace(preferred_payload_type);
>-  encoding.dtx.emplace(DtxStatus::ENABLED);
>-  // 20 kbps.
>-  encoding.max_bitrate_bps.emplace(20000);
>-  parameters.encodings.push_back(std::move(encoding));
>-
>-  parameters.header_extensions.emplace_back(
>-      "urn:ietf:params:rtp-hdrext:ssrc-audio-level", 1);
>-  return parameters;
>-}
>-
>-RtpParameters MakeFullOpusParameters() {
>-  return MakeFullAudioParameters(111);
>-}
>-
>-RtpParameters MakeFullIsacParameters() {
>-  return MakeFullAudioParameters(103);
>-}
>-
>-// Make video parameters with all the available properties configured and
>-// features used, and with multiple codecs offered. Obtained by taking a
>-// snapshot of a default PeerConnection offer (and adding other things, like
>-// bitrate limit).
>-//
>-// See "MakeFullVp8Parameters"/"MakeFullVp9Parameters" below.
>-RtpParameters MakeFullVideoParameters(int preferred_payload_type) {
>-  RtpParameters parameters;
>-
>-  RtpCodecParameters vp8_codec;
>-  vp8_codec.name = "VP8";
>-  vp8_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp8_codec.payload_type = 100;
>-  vp8_codec.clock_rate.emplace(90000);
>-  vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::CCM,
>-                                       RtcpFeedbackMessageType::FIR);
>-  vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::NACK,
>-                                       RtcpFeedbackMessageType::GENERIC_NACK);
>-  vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::NACK,
>-                                       RtcpFeedbackMessageType::PLI);
>-  vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::REMB);
>-  vp8_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::TRANSPORT_CC);
>-  parameters.codecs.push_back(std::move(vp8_codec));
>-
>-  RtpCodecParameters vp8_rtx_codec;
>-  vp8_rtx_codec.name = "rtx";
>-  vp8_rtx_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp8_rtx_codec.payload_type = 96;
>-  vp8_rtx_codec.clock_rate.emplace(90000);
>-  vp8_rtx_codec.parameters["apt"] = "100";
>-  parameters.codecs.push_back(std::move(vp8_rtx_codec));
>-
>-  RtpCodecParameters vp9_codec;
>-  vp9_codec.name = "VP9";
>-  vp9_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp9_codec.payload_type = 101;
>-  vp9_codec.clock_rate.emplace(90000);
>-  vp9_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::CCM,
>-                                       RtcpFeedbackMessageType::FIR);
>-  vp9_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::NACK,
>-                                       RtcpFeedbackMessageType::GENERIC_NACK);
>-  vp9_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::NACK,
>-                                       RtcpFeedbackMessageType::PLI);
>-  vp9_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::REMB);
>-  vp9_codec.rtcp_feedback.emplace_back(RtcpFeedbackType::TRANSPORT_CC);
>-  parameters.codecs.push_back(std::move(vp9_codec));
>-
>-  RtpCodecParameters vp9_rtx_codec;
>-  vp9_rtx_codec.name = "rtx";
>-  vp9_rtx_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  vp9_rtx_codec.payload_type = 97;
>-  vp9_rtx_codec.clock_rate.emplace(90000);
>-  vp9_rtx_codec.parameters["apt"] = "101";
>-  parameters.codecs.push_back(std::move(vp9_rtx_codec));
>-
>-  RtpCodecParameters red_codec;
>-  red_codec.name = "red";
>-  red_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  red_codec.payload_type = 116;
>-  red_codec.clock_rate.emplace(90000);
>-  parameters.codecs.push_back(std::move(red_codec));
>-
>-  RtpCodecParameters red_rtx_codec;
>-  red_rtx_codec.name = "rtx";
>-  red_rtx_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  red_rtx_codec.payload_type = 98;
>-  red_rtx_codec.clock_rate.emplace(90000);
>-  red_rtx_codec.parameters["apt"] = "116";
>-  parameters.codecs.push_back(std::move(red_rtx_codec));
>-
>-  RtpCodecParameters ulpfec_codec;
>-  ulpfec_codec.name = "ulpfec";
>-  ulpfec_codec.kind = cricket::MEDIA_TYPE_VIDEO;
>-  ulpfec_codec.payload_type = 117;
>-  ulpfec_codec.clock_rate.emplace(90000);
>-  parameters.codecs.push_back(std::move(ulpfec_codec));
>-
>-  // "codec_payload_type" isn't implemented, so we need to reorder codecs to
>-  // cause one to be used.
>-  // TODO(deadbeef): Remove this when it becomes unnecessary.
>-  auto it = std::find_if(parameters.codecs.begin(), parameters.codecs.end(),
>-                         [preferred_payload_type](const RtpCodecParameters& p) {
>-                           return p.payload_type == preferred_payload_type;
>-                         });
>-  RtpCodecParameters preferred = *it;
>-  parameters.codecs.erase(it);
>-  parameters.codecs.insert(parameters.codecs.begin(), preferred);
>-
>-  // Intentionally leave out SSRC so one's chosen automatically.
>-  RtpEncodingParameters encoding;
>-  encoding.codec_payload_type.emplace(preferred_payload_type);
>-  encoding.fec.emplace(FecMechanism::RED_AND_ULPFEC);
>-  // Will create default RtxParameters, with unset SSRC.
>-  encoding.rtx.emplace();
>-  // 100 kbps.
>-  encoding.max_bitrate_bps.emplace(100000);
>-  parameters.encodings.push_back(std::move(encoding));
>-
>-  parameters.header_extensions.emplace_back(
>-      "urn:ietf:params:rtp-hdrext:toffset", 2);
>-  parameters.header_extensions.emplace_back(
>-      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", 3);
>-  parameters.header_extensions.emplace_back("urn:3gpp:video-orientation", 4);
>-  parameters.header_extensions.emplace_back(
>-      "http://www.ietf.org/id/"
>-      "draft-holmer-rmcat-transport-wide-cc-extensions-01",
>-      5);
>-  parameters.header_extensions.emplace_back(
>-      "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", 6);
>-  return parameters;
>-}
>-
>-RtpParameters MakeFullVp8Parameters() {
>-  return MakeFullVideoParameters(100);
>-}
>-
>-RtpParameters MakeFullVp9Parameters() {
>-  return MakeFullVideoParameters(101);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/testrtpparameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/testrtpparameters.h
>deleted file mode 100644
>index 47e51e140837d4b9c496c56e3c49895197970586..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/testrtpparameters.h
>+++ /dev/null
>@@ -1,72 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef ORTC_TESTRTPPARAMETERS_H_
>-#define ORTC_TESTRTPPARAMETERS_H_
>-
>-#include "api/ortc/rtptransportinterface.h"
>-#include "api/rtpparameters.h"
>-
>-namespace webrtc {
>-
>-// Helper methods to create RtpParameters to use for sending/receiving.
>-//
>-// "MakeMinimal" methods contain the minimal necessary information for an
>-// RtpSender or RtpReceiver to function. The "MakeFull" methods are the
>-// opposite, and include all features that would normally be offered by a
>-// PeerConnection, and in some cases additional ones.
>-//
>-// These methods are intended to be used for end-to-end testing (such as in
>-// ortcfactory_integrationtest.cc), or unit testing that doesn't care about the
>-// specific contents of the parameters. Tests should NOT assume that these
>-// methods will not change; tests that are testing that a specific value in the
>-// parameters is applied properly should construct the parameters in the test
>-// itself.
>-
>-inline RtpTransportParameters MakeRtcpMuxParameters() {
>-  RtpTransportParameters parameters;
>-  parameters.rtcp.mux = true;
>-  return parameters;
>-}
>-
>-RtpParameters MakeMinimalOpusParameters();
>-RtpParameters MakeMinimalIsacParameters();
>-RtpParameters MakeMinimalOpusParametersWithSsrc(uint32_t ssrc);
>-RtpParameters MakeMinimalIsacParametersWithSsrc(uint32_t ssrc);
>-
>-RtpParameters MakeMinimalVp8Parameters();
>-RtpParameters MakeMinimalVp9Parameters();
>-RtpParameters MakeMinimalVp8ParametersWithSsrc(uint32_t ssrc);
>-RtpParameters MakeMinimalVp9ParametersWithSsrc(uint32_t ssrc);
>-
>-// Will create an encoding with no SSRC (meaning "match first SSRC seen" for a
>-// receiver, or "pick one automatically" for a sender).
>-RtpParameters MakeMinimalOpusParametersWithNoSsrc();
>-RtpParameters MakeMinimalIsacParametersWithNoSsrc();
>-RtpParameters MakeMinimalVp8ParametersWithNoSsrc();
>-RtpParameters MakeMinimalVp9ParametersWithNoSsrc();
>-
>-// Make audio parameters with all the available properties configured and
>-// features used, and with multiple codecs offered. Obtained by taking a
>-// snapshot of a default PeerConnection offer (and adding other things, like
>-// bitrate limit).
>-RtpParameters MakeFullOpusParameters();
>-RtpParameters MakeFullIsacParameters();
>-
>-// Make video parameters with all the available properties configured and
>-// features used, and with multiple codecs offered. Obtained by taking a
>-// snapshot of a default PeerConnection offer (and adding other things, like
>-// bitrate limit).
>-RtpParameters MakeFullVp8Parameters();
>-RtpParameters MakeFullVp9Parameters();
>-
>-}  // namespace webrtc
>-
>-#endif  // ORTC_TESTRTPPARAMETERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/BUILD.gn
>index 2d351bb0fa2cff038923bf6127e7bf37e5fbb1a7..29e633c5d227f924350c94b2b5e351a9718afe9b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/BUILD.gn
>@@ -29,6 +29,8 @@ rtc_static_library("rtc_p2p") {
>     "base/dtlstransport.h",
>     "base/dtlstransportinternal.cc",
>     "base/dtlstransportinternal.h",
>+    "base/icecredentialsiterator.cc",
>+    "base/icecredentialsiterator.h",
>     "base/icetransportinternal.cc",
>     "base/icetransportinternal.h",
>     "base/mdns_message.cc",
>@@ -74,8 +76,6 @@ rtc_static_library("rtc_p2p") {
>     "base/turnport.cc",
>     "base/turnport.h",
>     "base/udpport.h",
>-    "base/udptransport.cc",
>-    "base/udptransport.h",
>     "client/basicportallocator.cc",
>     "client/basicportallocator.h",
>     "client/relayportfactoryinterface.h",
>@@ -86,17 +86,20 @@ rtc_static_library("rtc_p2p") {
>   deps = [
>     "../api:libjingle_peerconnection_api",
>     "../api:ortc_api",
>+    "../api/transport:enums",
>     "../logging:ice_log",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base",
>     "../rtc_base:safe_minmax",
>     "../rtc_base:stringutils",
>     "../rtc_base:weak_ptr",
>+    "../rtc_base/system:rtc_export",
>     "../rtc_base/third_party/base64",
>     "../rtc_base/third_party/sigslot",
>     "../system_wrappers:field_trial",
>     "../system_wrappers:metrics",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>@@ -137,6 +140,7 @@ if (rtc_include_tests) {
>       ":rtc_p2p",
>       "../api:libjingle_peerconnection_api",
>       "../api:ortc_api",
>+      "../rtc_base:gunit_helpers",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>@@ -153,6 +157,7 @@ if (rtc_include_tests) {
>       "base/asyncstuntcpsocket_unittest.cc",
>       "base/basicasyncresolverfactory_unittest.cc",
>       "base/dtlstransport_unittest.cc",
>+      "base/icecredentialsiterator_unittest.cc",
>       "base/mdns_message_unittest.cc",
>       "base/p2ptransportchannel_unittest.cc",
>       "base/packetlossestimator_unittest.cc",
>@@ -170,7 +175,6 @@ if (rtc_include_tests) {
>       "base/transportdescriptionfactory_unittest.cc",
>       "base/turnport_unittest.cc",
>       "base/turnserver_unittest.cc",
>-      "base/udptransport_unittest.cc",
>       "client/basicportallocator_unittest.cc",
>     ]
>     deps = [
>@@ -178,10 +182,12 @@ if (rtc_include_tests) {
>       ":rtc_p2p",
>       "../api:ortc_api",
>       "../rtc_base:checks",
>+      "../rtc_base:gunit_helpers",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:stringutils",
>+      "../rtc_base:testclient",
>       "../system_wrappers:metrics",
>       "../test:test_support",
>       "//testing/gtest",
>@@ -206,6 +212,7 @@ rtc_static_library("libstunprober") {
>     "..:webrtc_common",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base",
>+    "../rtc_base/system:rtc_export",
>   ]
> }
> 
>@@ -221,6 +228,7 @@ if (rtc_include_tests) {
>       ":p2p_test_utils",
>       ":rtc_p2p",
>       "../rtc_base:checks",
>+      "../rtc_base:gunit_helpers",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_tests_utils",
>       "//testing/gtest",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/OWNERS
>index 18f6044b163d85f92f01d74f9068422ea21d9027..84a19ed596cced35d3dcb7031ee8d9c059cafb72 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/OWNERS
>@@ -6,6 +6,9 @@ mflodman@webrtc.org
> perkj@webrtc.org
> pthatcher@webrtc.org
> qingsi@webrtc.org
>+jeroendb@webrtc.org
>+emadomara@webrtc.org
>+steveanton@webrtc.org
> sergeyu@chromium.org
> tommi@webrtc.org
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.cc
>index 8bb796fd1d59cb59f8904751ee2b1bc0be482a26..d0185c3eb290ee872b9330c8b2a0751762209432 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.cc
>@@ -112,7 +112,7 @@ void AsyncStunTCPSocket::ProcessInput(char* data, size_t* len) {
>     }
> 
>     SignalReadPacket(this, data, expected_pkt_len, remote_addr,
>-                     rtc::CreatePacketTime(0));
>+                     rtc::TimeMicros());
> 
>     *len -= actual_length;
>     if (*len > 0) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc
>index de9541941f418da14581e14788b9cb28e87a8a11..b5fa14edc14bf8f436d6a62b335e6e1b1d553430 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc
>@@ -81,7 +81,7 @@ class AsyncStunTCPSocketTest : public testing::Test,
>                     const char* data,
>                     size_t len,
>                     const rtc::SocketAddress& remote_addr,
>-                    const rtc::PacketTime& packet_time) {
>+                    const int64_t& /* packet_time_us */) {
>     recv_packets_.push_back(std::string(data, len));
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.cc
>index 314f63ce74f865f1af346928b5d4e117d6bfaf34..97c6b13fb7a7b0d1b3a80769cfb3a25299fd03a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.cc
>@@ -20,6 +20,7 @@
> #include "rtc_base/dscp.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/messagequeue.h"
>+#include "rtc_base/rtccertificate.h"
> #include "rtc_base/sslstreamadapter.h"
> #include "rtc_base/stream.h"
> #include "rtc_base/thread.h"
>@@ -113,37 +114,23 @@ void StreamInterfaceChannel::Close() {
>   state_ = rtc::SS_CLOSED;
> }
> 
>-DtlsTransport::DtlsTransport(IceTransportInternal* ice_transport,
>-                             const rtc::CryptoOptions& crypto_options)
>-    : transport_name_(ice_transport->transport_name()),
>-      component_(ice_transport->component()),
>-      ice_transport_(ice_transport),
>-      downward_(NULL),
>-      srtp_ciphers_(GetSupportedDtlsSrtpCryptoSuites(crypto_options)),
>-      ssl_max_version_(rtc::SSL_PROTOCOL_DTLS_12),
>-      crypto_options_(crypto_options) {
>-  RTC_DCHECK(ice_transport_);
>-  ConnectToIceTransport();
>-}
>-
> DtlsTransport::DtlsTransport(
>     std::unique_ptr<IceTransportInternal> ice_transport,
>-    const rtc::CryptoOptions& crypto_options)
>+    const webrtc::CryptoOptions& crypto_options)
>     : transport_name_(ice_transport->transport_name()),
>       component_(ice_transport->component()),
>-      ice_transport_(ice_transport.get()),
>-      owned_ice_transport_(std::move(ice_transport)),
>+      ice_transport_(std::move(ice_transport)),
>       downward_(NULL),
>-      srtp_ciphers_(GetSupportedDtlsSrtpCryptoSuites(crypto_options)),
>+      srtp_ciphers_(crypto_options.GetSupportedDtlsSrtpCryptoSuites()),
>       ssl_max_version_(rtc::SSL_PROTOCOL_DTLS_12),
>       crypto_options_(crypto_options) {
>-  RTC_DCHECK(owned_ice_transport_);
>+  RTC_DCHECK(ice_transport_);
>   ConnectToIceTransport();
> }
> 
> DtlsTransport::~DtlsTransport() = default;
> 
>-const rtc::CryptoOptions& DtlsTransport::crypto_options() const {
>+const webrtc::CryptoOptions& DtlsTransport::crypto_options() const {
>   return crypto_options_;
> }
> 
>@@ -335,7 +322,8 @@ bool DtlsTransport::ExportKeyingMaterial(const std::string& label,
> 
> bool DtlsTransport::SetupDtls() {
>   RTC_DCHECK(dtls_role_);
>-  StreamInterfaceChannel* downward = new StreamInterfaceChannel(ice_transport_);
>+  StreamInterfaceChannel* downward =
>+      new StreamInterfaceChannel(ice_transport_.get());
> 
>   dtls_.reset(rtc::SSLStreamAdapter::Create(downward));
>   if (!dtls_) {
>@@ -431,7 +419,7 @@ int DtlsTransport::SendPacket(const char* data,
> }
> 
> IceTransportInternal* DtlsTransport::ice_transport() {
>-  return ice_transport_;
>+  return ice_transport_.get();
> }
> 
> bool DtlsTransport::IsDtlsConnected() {
>@@ -488,7 +476,7 @@ void DtlsTransport::ConnectToIceTransport() {
> //       impl again
> void DtlsTransport::OnWritableState(rtc::PacketTransportInternal* transport) {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>-  RTC_DCHECK(transport == ice_transport_);
>+  RTC_DCHECK(transport == ice_transport_.get());
>   RTC_LOG(LS_VERBOSE) << ToString()
>                       << ": ice_transport writable state changed to "
>                       << ice_transport_->writable();
>@@ -520,7 +508,7 @@ void DtlsTransport::OnWritableState(rtc::PacketTransportInternal* transport) {
> 
> void DtlsTransport::OnReceivingState(rtc::PacketTransportInternal* transport) {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>-  RTC_DCHECK(transport == ice_transport_);
>+  RTC_DCHECK(transport == ice_transport_.get());
>   RTC_LOG(LS_VERBOSE) << ToString()
>                       << ": ice_transport "
>                          "receiving state changed to "
>@@ -534,15 +522,15 @@ void DtlsTransport::OnReceivingState(rtc::PacketTransportInternal* transport) {
> void DtlsTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
>                                  const char* data,
>                                  size_t size,
>-                                 const rtc::PacketTime& packet_time,
>+                                 const int64_t& packet_time_us,
>                                  int flags) {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>-  RTC_DCHECK(transport == ice_transport_);
>+  RTC_DCHECK(transport == ice_transport_.get());
>   RTC_DCHECK(flags == 0);
> 
>   if (!dtls_active_) {
>     // Not doing DTLS.
>-    SignalReadPacket(this, data, size, packet_time, 0);
>+    SignalReadPacket(this, data, size, packet_time_us, 0);
>     return;
>   }
> 
>@@ -605,7 +593,7 @@ void DtlsTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
>         RTC_DCHECK(!srtp_ciphers_.empty());
> 
>         // Signal this upwards as a bypass packet.
>-        SignalReadPacket(this, data, size, packet_time, PF_SRTP_BYPASS);
>+        SignalReadPacket(this, data, size, packet_time_us, PF_SRTP_BYPASS);
>       }
>       break;
>     case DTLS_TRANSPORT_FAILED:
>@@ -651,7 +639,7 @@ void DtlsTransport::OnDtlsEvent(rtc::StreamInterface* dtls, int sig, int err) {
>     do {
>       ret = dtls_->Read(buf, sizeof(buf), &read, &read_error);
>       if (ret == rtc::SR_SUCCESS) {
>-        SignalReadPacket(this, buf, read, rtc::CreatePacketTime(0), 0);
>+        SignalReadPacket(this, buf, read, rtc::TimeMicros(), 0);
>       } else if (ret == rtc::SR_EOS) {
>         // Remote peer shut down the association with no error.
>         RTC_LOG(LS_INFO) << ToString() << ": DTLS transport closed";
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.h
>index 456f375a1070807e7a7b278001a759cabfb1c10a..ce95803aecb213de73e34844ab4bcd33c44042cb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.h
>@@ -15,6 +15,7 @@
> #include <string>
> #include <vector>
> 
>+#include "api/crypto/cryptooptions.h"
> #include "p2p/base/dtlstransportinternal.h"
> #include "p2p/base/icetransportinternal.h"
> #include "rtc_base/buffer.h"
>@@ -94,15 +95,12 @@ class DtlsTransport : public DtlsTransportInternal {
>   //
>   // |crypto_options| are the options used for the DTLS handshake. This affects
>   // whether GCM crypto suites are negotiated.
>-  // TODO(zhihuang): Remove this once we switch to JsepTransportController.
>-  explicit DtlsTransport(IceTransportInternal* ice_transport,
>-                         const rtc::CryptoOptions& crypto_options);
>   explicit DtlsTransport(std::unique_ptr<IceTransportInternal> ice_transport,
>-                         const rtc::CryptoOptions& crypto_options);
>+                         const webrtc::CryptoOptions& crypto_options);
> 
>   ~DtlsTransport() override;
> 
>-  const rtc::CryptoOptions& crypto_options() const override;
>+  const webrtc::CryptoOptions& crypto_options() const override;
>   DtlsTransportState dtls_state() const override;
>   const std::string& transport_name() const override;
>   int component() const override;
>@@ -196,7 +194,7 @@ class DtlsTransport : public DtlsTransportInternal {
>   void OnReadPacket(rtc::PacketTransportInternal* transport,
>                     const char* data,
>                     size_t size,
>-                    const rtc::PacketTime& packet_time,
>+                    const int64_t& packet_time_us,
>                     int flags);
>   void OnSentPacket(rtc::PacketTransportInternal* transport,
>                     const rtc::SentPacket& sent_packet);
>@@ -220,9 +218,8 @@ class DtlsTransport : public DtlsTransportInternal {
>   std::string transport_name_;
>   int component_;
>   DtlsTransportState dtls_state_ = DTLS_TRANSPORT_NEW;
>-  // Underlying ice_transport, not owned by this class.
>-  IceTransportInternal* const ice_transport_;
>-  std::unique_ptr<IceTransportInternal> owned_ice_transport_;
>+  // Underlying ice_transport, owned by this class.
>+  std::unique_ptr<IceTransportInternal> ice_transport_;
>   std::unique_ptr<rtc::SSLStreamAdapter> dtls_;  // The DTLS stream
>   StreamInterfaceChannel*
>       downward_;  // Wrapper for ice_transport_, owned by dtls_.
>@@ -231,7 +228,7 @@ class DtlsTransport : public DtlsTransportInternal {
>   rtc::scoped_refptr<rtc::RTCCertificate> local_certificate_;
>   absl::optional<rtc::SSLRole> dtls_role_;
>   rtc::SSLProtocolVersion ssl_max_version_;
>-  rtc::CryptoOptions crypto_options_;
>+  webrtc::CryptoOptions crypto_options_;
>   rtc::Buffer remote_fingerprint_value_;
>   std::string remote_fingerprint_algorithm_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport_unittest.cc
>index 3cf423e8513cdf9f784a73f679dc51f51a6cfd92..93f3da337d303084f07bf90aaf3f2cb403aabf22 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport_unittest.cc
>@@ -11,6 +11,7 @@
> #include <algorithm>
> #include <memory>
> #include <set>
>+#include <utility>
> 
> #include "p2p/base/dtlstransport.h"
> #include "p2p/base/fakeicetransport.h"
>@@ -19,6 +20,7 @@
> #include "rtc_base/dscp.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/helpers.h"
>+#include "rtc_base/rtccertificate.h"
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/sslidentity.h"
> #include "rtc_base/sslstreamadapter.h"
>@@ -47,8 +49,8 @@ void SetRemoteFingerprintFromCert(
>     DtlsTransport* transport,
>     const rtc::scoped_refptr<rtc::RTCCertificate>& cert,
>     bool modify_digest = false) {
>-  rtc::SSLFingerprint* fingerprint =
>-      rtc::SSLFingerprint::CreateFromCertificate(cert);
>+  std::unique_ptr<rtc::SSLFingerprint> fingerprint =
>+      rtc::SSLFingerprint::CreateFromCertificate(*cert);
>   if (modify_digest) {
>     ++fingerprint->digest[0];
>   }
>@@ -57,7 +59,6 @@ void SetRemoteFingerprintFromCert(
>       fingerprint->algorithm,
>       reinterpret_cast<const uint8_t*>(fingerprint->digest.data()),
>       fingerprint->digest.size()));
>-  delete fingerprint;
> }
> 
> class DtlsTestClient : public sigslot::has_slots<> {
>@@ -76,18 +77,18 @@ class DtlsTestClient : public sigslot::has_slots<> {
>   }
>   // Set up fake ICE transport and real DTLS transport under test.
>   void SetupTransports(IceRole role, int async_delay_ms = 0) {
>-    fake_ice_transport_.reset(new FakeIceTransport("fake", 0));
>-    fake_ice_transport_->SetAsync(true);
>-    fake_ice_transport_->SetAsyncDelay(async_delay_ms);
>-    fake_ice_transport_->SetIceRole(role);
>-    fake_ice_transport_->SetIceTiebreaker((role == ICEROLE_CONTROLLING) ? 1
>-                                                                        : 2);
>+    std::unique_ptr<FakeIceTransport> fake_ice_transport;
>+    fake_ice_transport.reset(new FakeIceTransport("fake", 0));
>+    fake_ice_transport->SetAsync(true);
>+    fake_ice_transport->SetAsyncDelay(async_delay_ms);
>+    fake_ice_transport->SetIceRole(role);
>+    fake_ice_transport->SetIceTiebreaker((role == ICEROLE_CONTROLLING) ? 1 : 2);
>     // Hook the raw packets so that we can verify they are encrypted.
>-    fake_ice_transport_->SignalReadPacket.connect(
>+    fake_ice_transport->SignalReadPacket.connect(
>         this, &DtlsTestClient::OnFakeIceTransportReadPacket);
> 
>-    dtls_transport_.reset(
>-        new DtlsTransport(fake_ice_transport_.get(), rtc::CryptoOptions()));
>+    dtls_transport_ = absl::make_unique<DtlsTransport>(
>+        std::move(fake_ice_transport), webrtc::CryptoOptions());
>     dtls_transport_->SetSslMaxProtocolVersion(ssl_max_version_);
>     // Note: Certificate may be null here if testing passthrough.
>     dtls_transport_->SetLocalCertificate(certificate_);
>@@ -99,13 +100,16 @@ class DtlsTestClient : public sigslot::has_slots<> {
>         this, &DtlsTestClient::OnTransportSentPacket);
>   }
> 
>-  FakeIceTransport* fake_ice_transport() { return fake_ice_transport_.get(); }
>+  FakeIceTransport* fake_ice_transport() {
>+    return static_cast<FakeIceTransport*>(dtls_transport_->ice_transport());
>+  }
> 
>   DtlsTransport* dtls_transport() { return dtls_transport_.get(); }
> 
>   // Simulate fake ICE transports connecting.
>   bool Connect(DtlsTestClient* peer, bool asymmetric) {
>-    fake_ice_transport_->SetDestination(peer->fake_ice_transport(), asymmetric);
>+    fake_ice_transport()->SetDestination(peer->fake_ice_transport(),
>+                                         asymmetric);
>     return true;
>   }
> 
>@@ -232,7 +236,7 @@ class DtlsTestClient : public sigslot::has_slots<> {
>   void OnTransportReadPacket(rtc::PacketTransportInternal* transport,
>                              const char* data,
>                              size_t size,
>-                             const rtc::PacketTime& packet_time,
>+                             const int64_t& /* packet_time_us */,
>                              int flags) {
>     uint32_t packet_num = 0;
>     ASSERT_TRUE(VerifyPacket(data, size, &packet_num));
>@@ -254,7 +258,7 @@ class DtlsTestClient : public sigslot::has_slots<> {
>   void OnFakeIceTransportReadPacket(rtc::PacketTransportInternal* transport,
>                                     const char* data,
>                                     size_t size,
>-                                    const rtc::PacketTime& time,
>+                                    const int64_t& /* packet_time_us */,
>                                     int flags) {
>     // Flags shouldn't be set on the underlying Transport packets.
>     ASSERT_EQ(0, flags);
>@@ -513,8 +517,8 @@ TEST_F(DtlsTransportTest, TestCertificatesBeforeConnect) {
>   // remote certificate, because connection has not yet occurred.
>   auto certificate1 = client1_.dtls_transport()->GetLocalCertificate();
>   auto certificate2 = client2_.dtls_transport()->GetLocalCertificate();
>-  ASSERT_NE(certificate1->ssl_certificate().ToPEMString(),
>-            certificate2->ssl_certificate().ToPEMString());
>+  ASSERT_NE(certificate1->GetSSLCertificate().ToPEMString(),
>+            certificate2->GetSSLCertificate().ToPEMString());
>   ASSERT_FALSE(client1_.dtls_transport()->GetRemoteSSLCertChain());
>   ASSERT_FALSE(client2_.dtls_transport()->GetRemoteSSLCertChain());
> }
>@@ -527,8 +531,8 @@ TEST_F(DtlsTransportTest, TestCertificatesAfterConnect) {
>   // After connection, each side has a distinct local certificate.
>   auto certificate1 = client1_.dtls_transport()->GetLocalCertificate();
>   auto certificate2 = client2_.dtls_transport()->GetLocalCertificate();
>-  ASSERT_NE(certificate1->ssl_certificate().ToPEMString(),
>-            certificate2->ssl_certificate().ToPEMString());
>+  ASSERT_NE(certificate1->GetSSLCertificate().ToPEMString(),
>+            certificate2->GetSSLCertificate().ToPEMString());
> 
>   // Each side's remote certificate is the other side's local certificate.
>   std::unique_ptr<rtc::SSLCertChain> remote_cert1 =
>@@ -536,13 +540,13 @@ TEST_F(DtlsTransportTest, TestCertificatesAfterConnect) {
>   ASSERT_TRUE(remote_cert1);
>   ASSERT_EQ(1u, remote_cert1->GetSize());
>   ASSERT_EQ(remote_cert1->Get(0).ToPEMString(),
>-            certificate2->ssl_certificate().ToPEMString());
>+            certificate2->GetSSLCertificate().ToPEMString());
>   std::unique_ptr<rtc::SSLCertChain> remote_cert2 =
>       client2_.dtls_transport()->GetRemoteSSLCertChain();
>   ASSERT_TRUE(remote_cert2);
>   ASSERT_EQ(1u, remote_cert2->GetSize());
>   ASSERT_EQ(remote_cert2->Get(0).ToPEMString(),
>-            certificate1->ssl_certificate().ToPEMString());
>+            certificate1->GetSSLCertificate().ToPEMString());
> }
> 
> // Test that packets are retransmitted according to the expected schedule.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransportinternal.h
>index 58d36fa08dbc83f6606f4ce98e8446f25adbfaad..a7137449f4cf159844338d25aac4130d85299464 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransportinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransportinternal.h
>@@ -15,10 +15,10 @@
> #include <string>
> #include <vector>
> 
>+#include "api/crypto/cryptooptions.h"
> #include "p2p/base/icetransportinternal.h"
> #include "p2p/base/packettransportinternal.h"
> #include "rtc_base/sslstreamadapter.h"
>-#include "rtc_base/stringencode.h"
> 
> namespace cricket {
> 
>@@ -51,7 +51,7 @@ class DtlsTransportInternal : public rtc::PacketTransportInternal {
>  public:
>   ~DtlsTransportInternal() override;
> 
>-  virtual const rtc::CryptoOptions& crypto_options() const = 0;
>+  virtual const webrtc::CryptoOptions& crypto_options() const = 0;
> 
>   virtual DtlsTransportState dtls_state() const = 0;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakedtlstransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakedtlstransport.h
>index cb944d3db5b39da760d4542245792df52bdb67ad..daef5c77fc998a41190ba88e4bf4209a5f5fc3f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakedtlstransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakedtlstransport.h
>@@ -17,9 +17,11 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>+#include "api/crypto/cryptooptions.h"
> #include "p2p/base/dtlstransportinternal.h"
> #include "p2p/base/fakeicetransport.h"
> #include "rtc_base/fakesslidentity.h"
>+#include "rtc_base/rtccertificate.h"
> 
> namespace cricket {
> 
>@@ -32,7 +34,7 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>       : ice_transport_(ice_transport),
>         transport_name_(ice_transport->transport_name()),
>         component_(ice_transport->component()),
>-        dtls_fingerprint_("", nullptr, 0) {
>+        dtls_fingerprint_("", nullptr) {
>     RTC_DCHECK(ice_transport_);
>     ice_transport_->SignalReadPacket.connect(
>         this, &FakeDtlsTransport::OnIceTransportReadPacket);
>@@ -44,7 +46,7 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>       : owned_ice_transport_(std::move(ice)),
>         transport_name_(owned_ice_transport_->transport_name()),
>         component_(owned_ice_transport_->component()),
>-        dtls_fingerprint_("", nullptr, 0) {
>+        dtls_fingerprint_("", rtc::ArrayView<const uint8_t>()) {
>     ice_transport_ = owned_ice_transport_.get();
>     ice_transport_->SignalReadPacket.connect(
>         this, &FakeDtlsTransport::OnIceTransportReadPacket);
>@@ -82,6 +84,7 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>     ice_transport_->SetReceiving(receiving);
>     set_receiving(receiving);
>   }
>+  void SetDtlsState(DtlsTransportState state) { dtls_state_ = state; }
> 
>   // Simulates the two DTLS transports connecting to each other.
>   // If |asymmetric| is true this method only affects this FakeDtlsTransport.
>@@ -132,7 +135,8 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>   bool SetRemoteFingerprint(const std::string& alg,
>                             const uint8_t* digest,
>                             size_t digest_len) override {
>-    dtls_fingerprint_ = rtc::SSLFingerprint(alg, digest, digest_len);
>+    dtls_fingerprint_ =
>+        rtc::SSLFingerprint(alg, rtc::MakeArrayView(digest, digest_len));
>     return true;
>   }
>   bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) override {
>@@ -149,10 +153,10 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>     *role = *dtls_role_;
>     return true;
>   }
>-  const rtc::CryptoOptions& crypto_options() const override {
>+  const webrtc::CryptoOptions& crypto_options() const override {
>     return crypto_options_;
>   }
>-  void SetCryptoOptions(const rtc::CryptoOptions& crypto_options) {
>+  void SetCryptoOptions(const webrtc::CryptoOptions& crypto_options) {
>     crypto_options_ = crypto_options;
>   }
>   bool SetLocalCertificate(
>@@ -178,8 +182,10 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>     return local_cert_;
>   }
>   std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain() const override {
>-    return remote_cert_ ? absl::make_unique<rtc::SSLCertChain>(remote_cert_)
>-                        : nullptr;
>+    if (!remote_cert_) {
>+      return nullptr;
>+    }
>+    return absl::make_unique<rtc::SSLCertChain>(remote_cert_->Clone());
>   }
>   bool ExportKeyingMaterial(const std::string& label,
>                             const uint8_t* context,
>@@ -232,9 +238,9 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>   void OnIceTransportReadPacket(PacketTransportInternal* ice_,
>                                 const char* data,
>                                 size_t len,
>-                                const rtc::PacketTime& time,
>+                                const int64_t& packet_time_us,
>                                 int flags) {
>-    SignalReadPacket(this, data, len, time, flags);
>+    SignalReadPacket(this, data, len, packet_time_us, flags);
>   }
> 
>   void set_receiving(bool receiving) {
>@@ -272,7 +278,7 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>   rtc::SSLFingerprint dtls_fingerprint_;
>   absl::optional<rtc::SSLRole> dtls_role_;
>   int crypto_suite_ = rtc::SRTP_AES128_CM_SHA1_80;
>-  rtc::CryptoOptions crypto_options_;
>+  webrtc::CryptoOptions crypto_options_;
> 
>   DtlsTransportState dtls_state_ = DTLS_TRANSPORT_NEW;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeicetransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeicetransport.h
>index 48157451ea69a1b1522d418654e007bb9ae435d0..88afc382ecbb851915328b36936e740cacee7868 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeicetransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeicetransport.h
>@@ -23,8 +23,13 @@ namespace cricket {
> 
> class FakeIceTransport : public IceTransportInternal {
>  public:
>-  explicit FakeIceTransport(const std::string& name, int component)
>-      : name_(name), component_(component) {}
>+  explicit FakeIceTransport(const std::string& name,
>+                            int component,
>+                            rtc::Thread* network_thread = nullptr)
>+      : name_(name),
>+        component_(component),
>+        network_thread_(network_thread ? network_thread
>+                                       : rtc::Thread::Current()) {}
>   ~FakeIceTransport() override {
>     if (dest_ && dest_->dest_ == this) {
>       dest_->dest_ = nullptr;
>@@ -114,6 +119,19 @@ class FakeIceTransport : public IceTransportInternal {
>     return IceTransportState::STATE_CONNECTING;
>   }
> 
>+  webrtc::IceTransportState GetIceTransportState() const override {
>+    if (connection_count_ == 0) {
>+      return had_connection_ ? webrtc::IceTransportState::kFailed
>+                             : webrtc::IceTransportState::kNew;
>+    }
>+
>+    if (connection_count_ == 1) {
>+      return webrtc::IceTransportState::kCompleted;
>+    }
>+
>+    return webrtc::IceTransportState::kConnected;
>+  }
>+
>   void SetIceRole(IceRole role) override { role_ = role; }
>   IceRole GetIceRole() const override { return role_; }
>   void SetIceTiebreaker(uint64_t tiebreaker) override {
>@@ -226,6 +244,8 @@ class FakeIceTransport : public IceTransportInternal {
>   }
>   void SetNetworkRoute(absl::optional<rtc::NetworkRoute> network_route) {
>     network_route_ = network_route;
>+    network_thread_->Invoke<void>(
>+        RTC_FROM_HERE, [this] { SignalNetworkRouteChanged(network_route_); });
>   }
> 
>  private:
>@@ -253,7 +273,7 @@ class FakeIceTransport : public IceTransportInternal {
>     if (dest_) {
>       last_sent_packet_ = packet;
>       dest_->SignalReadPacket(dest_, packet.data<char>(), packet.size(),
>-                              rtc::CreatePacketTime(0), 0);
>+                              rtc::TimeMicros(), 0);
>     }
>   }
> 
>@@ -282,6 +302,7 @@ class FakeIceTransport : public IceTransportInternal {
>   absl::optional<rtc::NetworkRoute> network_route_;
>   std::map<rtc::Socket::Option, int> socket_options_;
>   rtc::CopyOnWriteBuffer last_sent_packet_;
>+  rtc::Thread* const network_thread_;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakepackettransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakepackettransport.h
>index e57bc179a7991714c1b4b4323b57f5472d19db7f..52b39215c7524ac345bc539dca075c21827f818e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakepackettransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakepackettransport.h
>@@ -119,7 +119,7 @@ class FakePacketTransport : public PacketTransportInternal {
>     last_sent_packet_ = packet;
>     if (dest_) {
>       dest_->SignalReadPacket(dest_, packet.data<char>(), packet.size(),
>-                              CreatePacketTime(0), 0);
>+                              TimeMicros(), 0);
>     }
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icecredentialsiterator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icecredentialsiterator.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..7d29653440d7cb2da80c6d5acf10e339c123ed0d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icecredentialsiterator.cc
>@@ -0,0 +1,36 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "p2p/base/icecredentialsiterator.h"
>+#include "rtc_base/helpers.h"
>+
>+namespace cricket {
>+
>+IceCredentialsIterator::IceCredentialsIterator(
>+    const std::vector<IceParameters>& pooled_credentials)
>+    : pooled_ice_credentials_(pooled_credentials) {}
>+
>+IceCredentialsIterator::~IceCredentialsIterator() = default;
>+
>+IceParameters IceCredentialsIterator::CreateRandomIceCredentials() {
>+  return IceParameters(rtc::CreateRandomString(ICE_UFRAG_LENGTH),
>+                       rtc::CreateRandomString(ICE_PWD_LENGTH), false);
>+}
>+
>+IceParameters IceCredentialsIterator::GetIceCredentials() {
>+  if (pooled_ice_credentials_.empty()) {
>+    return CreateRandomIceCredentials();
>+  }
>+  IceParameters credentials = pooled_ice_credentials_.back();
>+  pooled_ice_credentials_.pop_back();
>+  return credentials;
>+}
>+
>+}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icecredentialsiterator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icecredentialsiterator.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..33e1d6460ae906b356897c4edb8ac6b9e904ec3a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icecredentialsiterator.h
>@@ -0,0 +1,37 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef P2P_BASE_ICECREDENTIALSITERATOR_H_
>+#define P2P_BASE_ICECREDENTIALSITERATOR_H_
>+
>+#include <vector>
>+
>+#include "p2p/base/transportdescription.h"
>+
>+namespace cricket {
>+
>+class IceCredentialsIterator {
>+ public:
>+  explicit IceCredentialsIterator(const std::vector<IceParameters>&);
>+  virtual ~IceCredentialsIterator();
>+
>+  // Get next pooled ice credentials.
>+  // Returns a new random credential if the pool is empty.
>+  IceParameters GetIceCredentials();
>+
>+  static IceParameters CreateRandomIceCredentials();
>+
>+ private:
>+  std::vector<IceParameters> pooled_ice_credentials_;
>+};
>+
>+}  // namespace cricket
>+
>+#endif  // P2P_BASE_ICECREDENTIALSITERATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icecredentialsiterator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icecredentialsiterator_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..00facfbb88d1efc9ce639b89d7016ba53d4d49ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icecredentialsiterator_unittest.cc
>@@ -0,0 +1,49 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "p2p/base/icecredentialsiterator.h"
>+#include "rtc_base/gunit.h"
>+
>+using cricket::IceParameters;
>+using cricket::IceCredentialsIterator;
>+
>+TEST(IceCredentialsIteratorTest, GetEmpty) {
>+  std::vector<IceParameters> empty;
>+  IceCredentialsIterator iterator(empty);
>+  // Verify that we can get credentials even if input is empty.
>+  IceParameters credentials1 = iterator.GetIceCredentials();
>+}
>+
>+TEST(IceCredentialsIteratorTest, GetOne) {
>+  std::vector<IceParameters> one = {
>+      IceCredentialsIterator::CreateRandomIceCredentials()};
>+  IceCredentialsIterator iterator(one);
>+  EXPECT_EQ(iterator.GetIceCredentials(), one[0]);
>+  auto random = iterator.GetIceCredentials();
>+  EXPECT_NE(random, one[0]);
>+  EXPECT_NE(random, iterator.GetIceCredentials());
>+}
>+
>+TEST(IceCredentialsIteratorTest, GetTwo) {
>+  std::vector<IceParameters> two = {
>+      IceCredentialsIterator::CreateRandomIceCredentials(),
>+      IceCredentialsIterator::CreateRandomIceCredentials()};
>+  IceCredentialsIterator iterator(two);
>+  EXPECT_EQ(iterator.GetIceCredentials(), two[1]);
>+  EXPECT_EQ(iterator.GetIceCredentials(), two[0]);
>+  auto random = iterator.GetIceCredentials();
>+  EXPECT_NE(random, two[0]);
>+  EXPECT_NE(random, two[1]);
>+  EXPECT_NE(random, iterator.GetIceCredentials());
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>index 7483c3babc2e21ecc2d397628db6441fa073a42b..099cea70e8f890837d9b420f089bb86c691b820f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>@@ -15,11 +15,12 @@
> #include <vector>
> 
> #include "api/candidate.h"
>+#include "api/transport/enums.h"
> #include "p2p/base/candidatepairinterface.h"
> #include "p2p/base/packettransportinternal.h"
> #include "p2p/base/port.h"
> #include "p2p/base/transportdescription.h"
>-#include "rtc_base/stringencode.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace cricket {
> 
>@@ -181,14 +182,17 @@ enum IceProtocolType {
> 
> // IceTransportInternal is an internal abstract class that does ICE.
> // Once the public interface is supported,
>-// (https://www.w3.org/TR/webrtc/#rtcicetransport-interface)
>+// (https://www.w3.org/TR/webrtc/#rtcicetransport)
> // the IceTransportInterface will be split from this class.
>-class IceTransportInternal : public rtc::PacketTransportInternal {
>+class RTC_EXPORT IceTransportInternal : public rtc::PacketTransportInternal {
>  public:
>   IceTransportInternal();
>   ~IceTransportInternal() override;
> 
>+  // TODO(bugs.webrtc.org/9308): Remove GetState once all uses have been
>+  // migrated to GetIceTransportState.
>   virtual IceTransportState GetState() const = 0;
>+  virtual webrtc::IceTransportState GetIceTransportState() const = 0;
> 
>   virtual int component() const = 0;
> 
>@@ -258,8 +262,13 @@ class IceTransportInternal : public rtc::PacketTransportInternal {
>   sigslot::signal1<IceTransportInternal*> SignalRoleConflict;
> 
>   // Emitted whenever the transport state changed.
>+  // TODO(bugs.webrtc.org/9308): Remove once all uses have migrated to the new
>+  // IceTransportState.
>   sigslot::signal1<IceTransportInternal*> SignalStateChanged;
> 
>+  // Emitted whenever the new standards-compliant transport state changed.
>+  sigslot::signal1<IceTransportInternal*> SignalIceTransportStateChanged;
>+
>   // Invoked when the transport is being destroyed.
>   sigslot::signal1<IceTransportInternal*> SignalDestroyed;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.cc
>index 61af6f39f29c898e3c2a0db21ac5097c7381c60b..f14a0d117e35ea39c221ee1fff0cf7823d994f81 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.cc
>@@ -19,13 +19,13 @@ namespace {
> // RFC 1035, Section 4.1.1.
> //
> // QR bit.
>-constexpr uint16_t kMDnsFlagMaskQueryOrResponse = 0x8000;
>+constexpr uint16_t kMdnsFlagMaskQueryOrResponse = 0x8000;
> // AA bit.
>-constexpr uint16_t kMDnsFlagMaskAuthoritative = 0x0400;
>+constexpr uint16_t kMdnsFlagMaskAuthoritative = 0x0400;
> // RFC 1035, Section 4.1.2, QCLASS and RFC 6762, Section 18.12, repurposing of
> // top bit of QCLASS as the unicast response bit.
>-constexpr uint16_t kMDnsQClassMaskUnicastResponse = 0x8000;
>-constexpr size_t kMDnsHeaderSizeBytes = 12;
>+constexpr uint16_t kMdnsQClassMaskUnicastResponse = 0x8000;
>+constexpr size_t kMdnsHeaderSizeBytes = 12;
> 
> bool ReadDomainName(MessageBufferReader* buf, std::string* name) {
>   size_t name_start_pos = buf->CurrentOffset();
>@@ -64,7 +64,7 @@ bool ReadDomainName(MessageBufferReader* buf, std::string* name) {
>     // A legitimate pointer only refers to a prior occurrence of the same name,
>     // and we should only move strictly backward to a prior name field after the
>     // header.
>-    if (pos_jump_to >= name_start_pos || pos_jump_to < kMDnsHeaderSizeBytes) {
>+    if (pos_jump_to >= name_start_pos || pos_jump_to < kMdnsHeaderSizeBytes) {
>       return false;
>     }
>     MessageBufferReader new_buf(buf->MessageData(), buf->MessageLength());
>@@ -88,27 +88,27 @@ void WriteDomainName(rtc::ByteBufferWriter* buf, const std::string& name) {
> 
> }  // namespace
> 
>-void MDnsHeader::SetQueryOrResponse(bool is_query) {
>+void MdnsHeader::SetQueryOrResponse(bool is_query) {
>   if (is_query) {
>-    flags &= ~kMDnsFlagMaskQueryOrResponse;
>+    flags &= ~kMdnsFlagMaskQueryOrResponse;
>   } else {
>-    flags |= kMDnsFlagMaskQueryOrResponse;
>+    flags |= kMdnsFlagMaskQueryOrResponse;
>   }
> }
> 
>-void MDnsHeader::SetAuthoritative(bool is_authoritative) {
>+void MdnsHeader::SetAuthoritative(bool is_authoritative) {
>   if (is_authoritative) {
>-    flags |= kMDnsFlagMaskAuthoritative;
>+    flags |= kMdnsFlagMaskAuthoritative;
>   } else {
>-    flags &= ~kMDnsFlagMaskAuthoritative;
>+    flags &= ~kMdnsFlagMaskAuthoritative;
>   }
> }
> 
>-bool MDnsHeader::IsAuthoritative() const {
>-  return flags & kMDnsFlagMaskAuthoritative;
>+bool MdnsHeader::IsAuthoritative() const {
>+  return flags & kMdnsFlagMaskAuthoritative;
> }
> 
>-bool MDnsHeader::Read(MessageBufferReader* buf) {
>+bool MdnsHeader::Read(MessageBufferReader* buf) {
>   if (!buf->ReadUInt16(&id) || !buf->ReadUInt16(&flags) ||
>       !buf->ReadUInt16(&qdcount) || !buf->ReadUInt16(&ancount) ||
>       !buf->ReadUInt16(&nscount) || !buf->ReadUInt16(&arcount)) {
>@@ -118,7 +118,7 @@ bool MDnsHeader::Read(MessageBufferReader* buf) {
>   return true;
> }
> 
>-void MDnsHeader::Write(rtc::ByteBufferWriter* buf) const {
>+void MdnsHeader::Write(rtc::ByteBufferWriter* buf) const {
>   buf->WriteUInt16(id);
>   buf->WriteUInt16(flags);
>   buf->WriteUInt16(qdcount);
>@@ -127,15 +127,15 @@ void MDnsHeader::Write(rtc::ByteBufferWriter* buf) const {
>   buf->WriteUInt16(arcount);
> }
> 
>-bool MDnsHeader::IsQuery() const {
>-  return !(flags & kMDnsFlagMaskQueryOrResponse);
>+bool MdnsHeader::IsQuery() const {
>+  return !(flags & kMdnsFlagMaskQueryOrResponse);
> }
> 
>-MDnsSectionEntry::MDnsSectionEntry() = default;
>-MDnsSectionEntry::~MDnsSectionEntry() = default;
>-MDnsSectionEntry::MDnsSectionEntry(const MDnsSectionEntry& other) = default;
>+MdnsSectionEntry::MdnsSectionEntry() = default;
>+MdnsSectionEntry::~MdnsSectionEntry() = default;
>+MdnsSectionEntry::MdnsSectionEntry(const MdnsSectionEntry& other) = default;
> 
>-void MDnsSectionEntry::SetType(SectionEntryType type) {
>+void MdnsSectionEntry::SetType(SectionEntryType type) {
>   switch (type) {
>     case SectionEntryType::kA:
>       type_ = 1;
>@@ -148,7 +148,7 @@ void MDnsSectionEntry::SetType(SectionEntryType type) {
>   }
> }
> 
>-SectionEntryType MDnsSectionEntry::GetType() const {
>+SectionEntryType MdnsSectionEntry::GetType() const {
>   switch (type_) {
>     case 1:
>       return SectionEntryType::kA;
>@@ -159,7 +159,7 @@ SectionEntryType MDnsSectionEntry::GetType() const {
>   }
> }
> 
>-void MDnsSectionEntry::SetClass(SectionEntryClass cls) {
>+void MdnsSectionEntry::SetClass(SectionEntryClass cls) {
>   switch (cls) {
>     case SectionEntryClass::kIN:
>       class_ = 1;
>@@ -169,7 +169,7 @@ void MDnsSectionEntry::SetClass(SectionEntryClass cls) {
>   }
> }
> 
>-SectionEntryClass MDnsSectionEntry::GetClass() const {
>+SectionEntryClass MdnsSectionEntry::GetClass() const {
>   switch (class_) {
>     case 1:
>       return SectionEntryClass::kIN;
>@@ -178,11 +178,11 @@ SectionEntryClass MDnsSectionEntry::GetClass() const {
>   }
> }
> 
>-MDnsQuestion::MDnsQuestion() = default;
>-MDnsQuestion::MDnsQuestion(const MDnsQuestion& other) = default;
>-MDnsQuestion::~MDnsQuestion() = default;
>+MdnsQuestion::MdnsQuestion() = default;
>+MdnsQuestion::MdnsQuestion(const MdnsQuestion& other) = default;
>+MdnsQuestion::~MdnsQuestion() = default;
> 
>-bool MDnsQuestion::Read(MessageBufferReader* buf) {
>+bool MdnsQuestion::Read(MessageBufferReader* buf) {
>   if (!ReadDomainName(buf, &name_)) {
>     RTC_LOG(LS_ERROR) << "Invalid name.";
>     return false;
>@@ -194,31 +194,31 @@ bool MDnsQuestion::Read(MessageBufferReader* buf) {
>   return true;
> }
> 
>-bool MDnsQuestion::Write(rtc::ByteBufferWriter* buf) const {
>+bool MdnsQuestion::Write(rtc::ByteBufferWriter* buf) const {
>   WriteDomainName(buf, name_);
>   buf->WriteUInt16(type_);
>   buf->WriteUInt16(class_);
>   return true;
> }
> 
>-void MDnsQuestion::SetUnicastResponse(bool should_unicast) {
>+void MdnsQuestion::SetUnicastResponse(bool should_unicast) {
>   if (should_unicast) {
>-    class_ |= kMDnsQClassMaskUnicastResponse;
>+    class_ |= kMdnsQClassMaskUnicastResponse;
>   } else {
>-    class_ &= ~kMDnsQClassMaskUnicastResponse;
>+    class_ &= ~kMdnsQClassMaskUnicastResponse;
>   }
> }
> 
>-bool MDnsQuestion::ShouldUnicastResponse() const {
>-  return class_ & kMDnsQClassMaskUnicastResponse;
>+bool MdnsQuestion::ShouldUnicastResponse() const {
>+  return class_ & kMdnsQClassMaskUnicastResponse;
> }
> 
>-MDnsResourceRecord::MDnsResourceRecord() = default;
>-MDnsResourceRecord::MDnsResourceRecord(const MDnsResourceRecord& other) =
>+MdnsResourceRecord::MdnsResourceRecord() = default;
>+MdnsResourceRecord::MdnsResourceRecord(const MdnsResourceRecord& other) =
>     default;
>-MDnsResourceRecord::~MDnsResourceRecord() = default;
>+MdnsResourceRecord::~MdnsResourceRecord() = default;
> 
>-bool MDnsResourceRecord::Read(MessageBufferReader* buf) {
>+bool MdnsResourceRecord::Read(MessageBufferReader* buf) {
>   if (!ReadDomainName(buf, &name_)) {
>     return false;
>   }
>@@ -239,17 +239,17 @@ bool MDnsResourceRecord::Read(MessageBufferReader* buf) {
>   }
>   return false;
> }
>-bool MDnsResourceRecord::ReadARData(MessageBufferReader* buf) {
>+bool MdnsResourceRecord::ReadARData(MessageBufferReader* buf) {
>   // A RDATA contains a 32-bit IPv4 address.
>   return buf->ReadString(&rdata_, 4);
> }
> 
>-bool MDnsResourceRecord::ReadQuadARData(MessageBufferReader* buf) {
>+bool MdnsResourceRecord::ReadQuadARData(MessageBufferReader* buf) {
>   // AAAA RDATA contains a 128-bit IPv6 address.
>   return buf->ReadString(&rdata_, 16);
> }
> 
>-bool MDnsResourceRecord::Write(rtc::ByteBufferWriter* buf) const {
>+bool MdnsResourceRecord::Write(rtc::ByteBufferWriter* buf) const {
>   WriteDomainName(buf, name_);
>   buf->WriteUInt16(type_);
>   buf->WriteUInt16(class_);
>@@ -270,15 +270,15 @@ bool MDnsResourceRecord::Write(rtc::ByteBufferWriter* buf) const {
>   return true;
> }
> 
>-void MDnsResourceRecord::WriteARData(rtc::ByteBufferWriter* buf) const {
>+void MdnsResourceRecord::WriteARData(rtc::ByteBufferWriter* buf) const {
>   buf->WriteString(rdata_);
> }
> 
>-void MDnsResourceRecord::WriteQuadARData(rtc::ByteBufferWriter* buf) const {
>+void MdnsResourceRecord::WriteQuadARData(rtc::ByteBufferWriter* buf) const {
>   buf->WriteString(rdata_);
> }
> 
>-bool MDnsResourceRecord::SetIPAddressInRecordData(
>+bool MdnsResourceRecord::SetIPAddressInRecordData(
>     const rtc::IPAddress& address) {
>   int af = address.family();
>   if (af != AF_INET && af != AF_INET6) {
>@@ -293,7 +293,7 @@ bool MDnsResourceRecord::SetIPAddressInRecordData(
>   return true;
> }
> 
>-bool MDnsResourceRecord::GetIPAddressFromRecordData(
>+bool MdnsResourceRecord::GetIPAddressFromRecordData(
>     rtc::IPAddress* address) const {
>   if (GetType() != SectionEntryType::kA &&
>       GetType() != SectionEntryType::kAAAA) {
>@@ -310,16 +310,16 @@ bool MDnsResourceRecord::GetIPAddressFromRecordData(
>   return rtc::IPFromString(std::string(out), address);
> }
> 
>-MDnsMessage::MDnsMessage() = default;
>-MDnsMessage::~MDnsMessage() = default;
>+MdnsMessage::MdnsMessage() = default;
>+MdnsMessage::~MdnsMessage() = default;
> 
>-bool MDnsMessage::Read(MessageBufferReader* buf) {
>+bool MdnsMessage::Read(MessageBufferReader* buf) {
>   RTC_DCHECK_EQ(0u, buf->CurrentOffset());
>   if (!header_.Read(buf)) {
>     return false;
>   }
> 
>-  auto read_question = [&buf](std::vector<MDnsQuestion>* section,
>+  auto read_question = [&buf](std::vector<MdnsQuestion>* section,
>                               uint16_t count) {
>     section->resize(count);
>     for (auto& question : (*section)) {
>@@ -329,7 +329,7 @@ bool MDnsMessage::Read(MessageBufferReader* buf) {
>     }
>     return true;
>   };
>-  auto read_rr = [&buf](std::vector<MDnsResourceRecord>* section,
>+  auto read_rr = [&buf](std::vector<MdnsResourceRecord>* section,
>                         uint16_t count) {
>     section->resize(count);
>     for (auto& rr : (*section)) {
>@@ -349,10 +349,10 @@ bool MDnsMessage::Read(MessageBufferReader* buf) {
>   return true;
> }
> 
>-bool MDnsMessage::Write(rtc::ByteBufferWriter* buf) const {
>+bool MdnsMessage::Write(rtc::ByteBufferWriter* buf) const {
>   header_.Write(buf);
> 
>-  auto write_rr = [&buf](const std::vector<MDnsResourceRecord>& section) {
>+  auto write_rr = [&buf](const std::vector<MdnsResourceRecord>& section) {
>     for (auto rr : section) {
>       if (!rr.Write(buf)) {
>         return false;
>@@ -374,7 +374,7 @@ bool MDnsMessage::Write(rtc::ByteBufferWriter* buf) const {
>   return true;
> }
> 
>-bool MDnsMessage::ShouldUnicastResponse() const {
>+bool MdnsMessage::ShouldUnicastResponse() const {
>   bool should_unicast = false;
>   for (const auto& question : question_section_) {
>     should_unicast |= question.ShouldUnicastResponse();
>@@ -382,12 +382,12 @@ bool MDnsMessage::ShouldUnicastResponse() const {
>   return should_unicast;
> }
> 
>-void MDnsMessage::AddQuestion(const MDnsQuestion& question) {
>+void MdnsMessage::AddQuestion(const MdnsQuestion& question) {
>   question_section_.push_back(question);
>   header_.qdcount = question_section_.size();
> }
> 
>-void MDnsMessage::AddAnswerRecord(const MDnsResourceRecord& answer) {
>+void MdnsMessage::AddAnswerRecord(const MdnsResourceRecord& answer) {
>   answer_section_.push_back(answer);
>   header_.ancount = answer_section_.size();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.h
>index f6dec20a956d53020748dfc9cb2b489fd4098207..7aa1b21039a530f11f8c8f8551c9b50cad3ddc52 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message.h
>@@ -15,7 +15,7 @@
> // 6762 and RFC 1025 (DNS messages). Note that it is recommended by RFC 6762 to
> // use the name compression scheme defined in RFC 1035 whenever possible. We
> // currently only implement the capability of reading compressed names in mDNS
>-// messages in MDnsMessage::Read(); however, the MDnsMessage::Write() does not
>+// messages in MdnsMessage::Read(); however, the MdnsMessage::Write() does not
> // support name compression yet.
> //
> // Fuzzer tests (test/fuzzers/mdns_parser_fuzzer.cc) MUST always be performed
>@@ -50,7 +50,7 @@ enum class SectionEntryClass {
> };
> 
> // RFC 1035, Section 4.1.1.
>-class MDnsHeader final {
>+class MdnsHeader final {
>  public:
>   bool Read(MessageBufferReader* buf);
>   void Write(rtc::ByteBufferWriter* buf) const;
>@@ -74,11 +74,11 @@ class MDnsHeader final {
> 
> // Entries in each section after the header share a common structure. Note that
> // this is not a concept defined in RFC 1035.
>-class MDnsSectionEntry {
>+class MdnsSectionEntry {
>  public:
>-  MDnsSectionEntry();
>-  MDnsSectionEntry(const MDnsSectionEntry& other);
>-  virtual ~MDnsSectionEntry();
>+  MdnsSectionEntry();
>+  MdnsSectionEntry(const MdnsSectionEntry& other);
>+  virtual ~MdnsSectionEntry();
>   virtual bool Read(MessageBufferReader* buf) = 0;
>   virtual bool Write(rtc::ByteBufferWriter* buf) const = 0;
> 
>@@ -99,11 +99,11 @@ class MDnsSectionEntry {
> };
> 
> // RFC 1035, Section 4.1.2.
>-class MDnsQuestion final : public MDnsSectionEntry {
>+class MdnsQuestion final : public MdnsSectionEntry {
>  public:
>-  MDnsQuestion();
>-  MDnsQuestion(const MDnsQuestion& other);
>-  ~MDnsQuestion() override;
>+  MdnsQuestion();
>+  MdnsQuestion(const MdnsQuestion& other);
>+  ~MdnsQuestion() override;
> 
>   bool Read(MessageBufferReader* buf) override;
>   bool Write(rtc::ByteBufferWriter* buf) const override;
>@@ -113,11 +113,11 @@ class MDnsQuestion final : public MDnsSectionEntry {
> };
> 
> // RFC 1035, Section 4.1.3.
>-class MDnsResourceRecord final : public MDnsSectionEntry {
>+class MdnsResourceRecord final : public MdnsSectionEntry {
>  public:
>-  MDnsResourceRecord();
>-  MDnsResourceRecord(const MDnsResourceRecord& other);
>-  ~MDnsResourceRecord() override;
>+  MdnsResourceRecord();
>+  MdnsResourceRecord(const MdnsResourceRecord& other);
>+  ~MdnsResourceRecord() override;
> 
>   bool Read(MessageBufferReader* buf) override;
>   bool Write(rtc::ByteBufferWriter* buf) const override;
>@@ -145,17 +145,17 @@ class MDnsResourceRecord final : public MDnsSectionEntry {
>   std::string rdata_;
> };
> 
>-class MDnsMessage final {
>+class MdnsMessage final {
>  public:
>   // RFC 1035, Section 4.1.
>   enum class Section { kQuestion, kAnswer, kAuthority, kAdditional };
> 
>-  MDnsMessage();
>-  ~MDnsMessage();
>+  MdnsMessage();
>+  ~MdnsMessage();
>   // Reads the mDNS message in |buf| and populates the corresponding fields in
>-  // MDnsMessage.
>+  // MdnsMessage.
>   bool Read(MessageBufferReader* buf);
>-  // Write an mDNS message to |buf| based on the fields in MDnsMessage.
>+  // Write an mDNS message to |buf| based on the fields in MdnsMessage.
>   //
>   // TODO(qingsi): Implement name compression when writing mDNS messages.
>   bool Write(rtc::ByteBufferWriter* buf) const;
>@@ -177,29 +177,29 @@ class MDnsMessage final {
>   // preferred. False otherwise.
>   bool ShouldUnicastResponse() const;
> 
>-  void AddQuestion(const MDnsQuestion& question);
>+  void AddQuestion(const MdnsQuestion& question);
>   // TODO(qingsi): Implement AddXRecord for name server and additional records.
>-  void AddAnswerRecord(const MDnsResourceRecord& answer);
>+  void AddAnswerRecord(const MdnsResourceRecord& answer);
> 
>-  const std::vector<MDnsQuestion>& question_section() const {
>+  const std::vector<MdnsQuestion>& question_section() const {
>     return question_section_;
>   }
>-  const std::vector<MDnsResourceRecord>& answer_section() const {
>+  const std::vector<MdnsResourceRecord>& answer_section() const {
>     return answer_section_;
>   }
>-  const std::vector<MDnsResourceRecord>& authority_section() const {
>+  const std::vector<MdnsResourceRecord>& authority_section() const {
>     return authority_section_;
>   }
>-  const std::vector<MDnsResourceRecord>& additional_section() const {
>+  const std::vector<MdnsResourceRecord>& additional_section() const {
>     return additional_section_;
>   }
> 
>  private:
>-  MDnsHeader header_;
>-  std::vector<MDnsQuestion> question_section_;
>-  std::vector<MDnsResourceRecord> answer_section_;
>-  std::vector<MDnsResourceRecord> authority_section_;
>-  std::vector<MDnsResourceRecord> additional_section_;
>+  MdnsHeader header_;
>+  std::vector<MdnsQuestion> question_section_;
>+  std::vector<MdnsResourceRecord> answer_section_;
>+  std::vector<MdnsResourceRecord> authority_section_;
>+  std::vector<MdnsResourceRecord> additional_section_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message_unittest.cc
>index fb4a3b1e7a2615609d1a5a43d2d3abef00448f72..7f816f740b8a40d471fa1b1ecb372485f4ffb682 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mdns_message_unittest.cc
>@@ -19,9 +19,9 @@
> #include "rtc_base/socketaddress.h"
> #include "test/gmock.h"
> 
>-#define ReadMDnsMessage(X, Y) ReadMDnsMessageTestCase(X, Y, sizeof(Y))
>-#define WriteMDnsMessageAndCompare(X, Y) \
>-  WriteMDnsMessageAndCompareWithTestCast(X, Y, sizeof(Y))
>+#define ReadMdnsMessage(X, Y) ReadMdnsMessageTestCase(X, Y, sizeof(Y))
>+#define WriteMdnsMessageAndCompare(X, Y) \
>+  WriteMdnsMessageAndCompareWithTestCast(X, Y, sizeof(Y))
> 
> using ::testing::ElementsAre;
> using ::testing::Pair;
>@@ -255,14 +255,14 @@ const uint8_t kCorruptedAnswerWithNameCompression2[] = {
>     0xc0, 0xA8, 0x00, 0x01,  // 192.168.0.1
> };
> 
>-bool ReadMDnsMessageTestCase(MDnsMessage* msg,
>+bool ReadMdnsMessageTestCase(MdnsMessage* msg,
>                              const uint8_t* testcase,
>                              size_t size) {
>   MessageBufferReader buf(reinterpret_cast<const char*>(testcase), size);
>   return msg->Read(&buf);
> }
> 
>-void WriteMDnsMessageAndCompareWithTestCast(MDnsMessage* msg,
>+void WriteMdnsMessageAndCompareWithTestCast(MdnsMessage* msg,
>                                             const uint8_t* testcase,
>                                             size_t size) {
>   rtc::ByteBufferWriter out;
>@@ -276,7 +276,7 @@ void WriteMDnsMessageAndCompareWithTestCast(MDnsMessage* msg,
>   EXPECT_EQ(testcase_bytes, bytes);
> }
> 
>-bool GetQueriedNames(MDnsMessage* msg, std::set<std::string>* names) {
>+bool GetQueriedNames(MdnsMessage* msg, std::set<std::string>* names) {
>   if (!msg->IsQuery() || msg->question_section().empty()) {
>     return false;
>   }
>@@ -286,7 +286,7 @@ bool GetQueriedNames(MDnsMessage* msg, std::set<std::string>* names) {
>   return true;
> }
> 
>-bool GetResolution(MDnsMessage* msg,
>+bool GetResolution(MdnsMessage* msg,
>                    std::map<std::string, rtc::IPAddress>* names) {
>   if (msg->IsQuery() || msg->answer_section().empty()) {
>     return false;
>@@ -303,10 +303,10 @@ bool GetResolution(MDnsMessage* msg,
> 
> }  // namespace
> 
>-TEST(MDnsMessageTest, ReadSingleQuestionForIPv4Address) {
>-  MDnsMessage msg;
>+TEST(MdnsMessageTest, ReadSingleQuestionForIPv4Address) {
>+  MdnsMessage msg;
>   ASSERT_TRUE(
>-      ReadMDnsMessage(&msg, kSingleQuestionForIPv4AddrWithUnicastResponse));
>+      ReadMdnsMessage(&msg, kSingleQuestionForIPv4AddrWithUnicastResponse));
>   EXPECT_TRUE(msg.IsQuery());
>   EXPECT_EQ(0x1234, msg.GetId());
>   ASSERT_EQ(1u, msg.question_section().size());
>@@ -323,9 +323,9 @@ TEST(MDnsMessageTest, ReadSingleQuestionForIPv4Address) {
>   EXPECT_THAT(queried_names, ElementsAre("webrtc.org."));
> }
> 
>-TEST(MDnsMessageTest, ReadTwoQuestionsForIPv4AndIPv6Addr) {
>-  MDnsMessage msg;
>-  ASSERT_TRUE(ReadMDnsMessage(
>+TEST(MdnsMessageTest, ReadTwoQuestionsForIPv4AndIPv6Addr) {
>+  MdnsMessage msg;
>+  ASSERT_TRUE(ReadMdnsMessage(
>       &msg, kTwoQuestionsForIPv4AndIPv6AddrWithMulticastResponse));
>   EXPECT_TRUE(msg.IsQuery());
>   EXPECT_EQ(0x1234, msg.GetId());
>@@ -345,9 +345,9 @@ TEST(MDnsMessageTest, ReadTwoQuestionsForIPv4AndIPv6Addr) {
>               UnorderedElementsAre("webrtc4.org.", "webrtc6.org."));
> }
> 
>-TEST(MDnsMessageTest, ReadTwoQuestionsForIPv4AndIPv6AddrWithNameCompression) {
>-  MDnsMessage msg;
>-  ASSERT_TRUE(ReadMDnsMessage(
>+TEST(MdnsMessageTest, ReadTwoQuestionsForIPv4AndIPv6AddrWithNameCompression) {
>+  MdnsMessage msg;
>+  ASSERT_TRUE(ReadMdnsMessage(
>       &msg,
>       kTwoQuestionsForIPv4AndIPv6AddrWithMulticastResponseAndNameCompression));
> 
>@@ -363,10 +363,10 @@ TEST(MDnsMessageTest, ReadTwoQuestionsForIPv4AndIPv6AddrWithNameCompression) {
>               UnorderedElementsAre("www.webrtc.org.", "mdns.webrtc.org."));
> }
> 
>-TEST(MDnsMessageTest, ReadThreeQuestionsWithTwoPointersToTheSameNameSuffix) {
>-  MDnsMessage msg;
>+TEST(MdnsMessageTest, ReadThreeQuestionsWithTwoPointersToTheSameNameSuffix) {
>+  MdnsMessage msg;
>   ASSERT_TRUE(
>-      ReadMDnsMessage(&msg, kThreeQuestionsWithTwoPointersToTheSameNameSuffix));
>+      ReadMdnsMessage(&msg, kThreeQuestionsWithTwoPointersToTheSameNameSuffix));
> 
>   ASSERT_EQ(3u, msg.question_section().size());
>   const auto& question1 = msg.question_section()[0];
>@@ -383,10 +383,10 @@ TEST(MDnsMessageTest, ReadThreeQuestionsWithTwoPointersToTheSameNameSuffix) {
>                                    "webrtc.org."));
> }
> 
>-TEST(MDnsMessageTest,
>+TEST(MdnsMessageTest,
>      ReadThreeQuestionsWithPointerToNameSuffixContainingAnotherPointer) {
>-  MDnsMessage msg;
>-  ASSERT_TRUE(ReadMDnsMessage(
>+  MdnsMessage msg;
>+  ASSERT_TRUE(ReadMdnsMessage(
>       &msg, kThreeQuestionsWithPointerToNameSuffixContainingAnotherPointer));
> 
>   ASSERT_EQ(3u, msg.question_section().size());
>@@ -404,16 +404,16 @@ TEST(MDnsMessageTest,
>                                    "www.mdns.webrtc.org."));
> }
> 
>-TEST(MDnsMessageTest,
>+TEST(MdnsMessageTest,
>      ReadQuestionWithCorruptedPointerInNameCompressionShouldFail) {
>-  MDnsMessage msg;
>-  EXPECT_FALSE(ReadMDnsMessage(&msg, kCorruptedQuestionWithNameCompression1));
>-  EXPECT_FALSE(ReadMDnsMessage(&msg, kCorruptedQuestionWithNameCompression2));
>+  MdnsMessage msg;
>+  EXPECT_FALSE(ReadMdnsMessage(&msg, kCorruptedQuestionWithNameCompression1));
>+  EXPECT_FALSE(ReadMdnsMessage(&msg, kCorruptedQuestionWithNameCompression2));
> }
> 
>-TEST(MDnsMessageTest, ReadSingleAnswerForIPv4Addr) {
>-  MDnsMessage msg;
>-  ASSERT_TRUE(ReadMDnsMessage(&msg, kSingleAuthoritativeAnswerWithIPv4Addr));
>+TEST(MdnsMessageTest, ReadSingleAnswerForIPv4Addr) {
>+  MdnsMessage msg;
>+  ASSERT_TRUE(ReadMdnsMessage(&msg, kSingleAuthoritativeAnswerWithIPv4Addr));
>   EXPECT_FALSE(msg.IsQuery());
>   EXPECT_TRUE(msg.IsAuthoritative());
>   EXPECT_EQ(0x1234, msg.GetId());
>@@ -432,10 +432,10 @@ TEST(MDnsMessageTest, ReadSingleAnswerForIPv4Addr) {
>   EXPECT_THAT(resolution, ElementsAre(Pair("webrtc.org.", expected_addr)));
> }
> 
>-TEST(MDnsMessageTest, ReadTwoAnswersForIPv4AndIPv6Addr) {
>-  MDnsMessage msg;
>+TEST(MdnsMessageTest, ReadTwoAnswersForIPv4AndIPv6Addr) {
>+  MdnsMessage msg;
>   ASSERT_TRUE(
>-      ReadMDnsMessage(&msg, kTwoAuthoritativeAnswersWithIPv4AndIPv6Addr));
>+      ReadMdnsMessage(&msg, kTwoAuthoritativeAnswersWithIPv4AndIPv6Addr));
>   EXPECT_FALSE(msg.IsQuery());
>   EXPECT_TRUE(msg.IsAuthoritative());
>   EXPECT_EQ(0x1234, msg.GetId());
>@@ -462,9 +462,9 @@ TEST(MDnsMessageTest, ReadTwoAnswersForIPv4AndIPv6Addr) {
>                                    Pair("webrtc6.org.", expected_addr_ipv6)));
> }
> 
>-TEST(MDnsMessageTest, ReadTwoAnswersForIPv4AndIPv6AddrWithNameCompression) {
>-  MDnsMessage msg;
>-  ASSERT_TRUE(ReadMDnsMessage(
>+TEST(MdnsMessageTest, ReadTwoAnswersForIPv4AndIPv6AddrWithNameCompression) {
>+  MdnsMessage msg;
>+  ASSERT_TRUE(ReadMdnsMessage(
>       &msg, kTwoAuthoritativeAnswersWithIPv4AndIPv6AddrWithNameCompression));
> 
>   std::map<std::string, rtc::IPAddress> resolution;
>@@ -478,57 +478,57 @@ TEST(MDnsMessageTest, ReadTwoAnswersForIPv4AndIPv6AddrWithNameCompression) {
>                                    Pair("webrtc.org.", expected_addr_ipv6)));
> }
> 
>-TEST(MDnsMessageTest,
>+TEST(MdnsMessageTest,
>      ReadAnswerWithCorruptedPointerInNameCompressionShouldFail) {
>-  MDnsMessage msg;
>-  EXPECT_FALSE(ReadMDnsMessage(&msg, kCorruptedAnswerWithNameCompression1));
>-  EXPECT_FALSE(ReadMDnsMessage(&msg, kCorruptedAnswerWithNameCompression2));
>+  MdnsMessage msg;
>+  EXPECT_FALSE(ReadMdnsMessage(&msg, kCorruptedAnswerWithNameCompression1));
>+  EXPECT_FALSE(ReadMdnsMessage(&msg, kCorruptedAnswerWithNameCompression2));
> }
> 
>-TEST(MDnsMessageTest, WriteSingleQuestionForIPv4Addr) {
>-  MDnsMessage msg;
>+TEST(MdnsMessageTest, WriteSingleQuestionForIPv4Addr) {
>+  MdnsMessage msg;
>   msg.SetId(0x1234);
>   msg.SetQueryOrResponse(true);
> 
>-  MDnsQuestion question;
>+  MdnsQuestion question;
>   question.SetName("webrtc.org.");
>   question.SetType(SectionEntryType::kA);
>   question.SetClass(SectionEntryClass::kIN);
>   question.SetUnicastResponse(true);
>   msg.AddQuestion(question);
> 
>-  WriteMDnsMessageAndCompare(&msg,
>+  WriteMdnsMessageAndCompare(&msg,
>                              kSingleQuestionForIPv4AddrWithUnicastResponse);
> }
> 
>-TEST(MDnsMessageTest, WriteTwoQuestionsForIPv4AndIPv6Addr) {
>-  MDnsMessage msg;
>+TEST(MdnsMessageTest, WriteTwoQuestionsForIPv4AndIPv6Addr) {
>+  MdnsMessage msg;
>   msg.SetId(0x1234);
>   msg.SetQueryOrResponse(true);
> 
>-  MDnsQuestion question1;
>+  MdnsQuestion question1;
>   question1.SetName("webrtc4.org.");
>   question1.SetType(SectionEntryType::kA);
>   question1.SetClass(SectionEntryClass::kIN);
>   msg.AddQuestion(question1);
> 
>-  MDnsQuestion question2;
>+  MdnsQuestion question2;
>   question2.SetName("webrtc6.org.");
>   question2.SetType(SectionEntryType::kAAAA);
>   question2.SetClass(SectionEntryClass::kIN);
>   msg.AddQuestion(question2);
> 
>-  WriteMDnsMessageAndCompare(
>+  WriteMdnsMessageAndCompare(
>       &msg, kTwoQuestionsForIPv4AndIPv6AddrWithMulticastResponse);
> }
> 
>-TEST(MDnsMessageTest, WriteSingleAnswerToIPv4Addr) {
>-  MDnsMessage msg;
>+TEST(MdnsMessageTest, WriteSingleAnswerToIPv4Addr) {
>+  MdnsMessage msg;
>   msg.SetId(0x1234);
>   msg.SetQueryOrResponse(false);
>   msg.SetAuthoritative(true);
> 
>-  MDnsResourceRecord answer;
>+  MdnsResourceRecord answer;
>   answer.SetName("webrtc.org.");
>   answer.SetType(SectionEntryType::kA);
>   answer.SetClass(SectionEntryClass::kIN);
>@@ -537,16 +537,16 @@ TEST(MDnsMessageTest, WriteSingleAnswerToIPv4Addr) {
>   answer.SetTtlSeconds(120);
>   msg.AddAnswerRecord(answer);
> 
>-  WriteMDnsMessageAndCompare(&msg, kSingleAuthoritativeAnswerWithIPv4Addr);
>+  WriteMdnsMessageAndCompare(&msg, kSingleAuthoritativeAnswerWithIPv4Addr);
> }
> 
>-TEST(MDnsMessageTest, WriteTwoAnswersToIPv4AndIPv6Addr) {
>-  MDnsMessage msg;
>+TEST(MdnsMessageTest, WriteTwoAnswersToIPv4AndIPv6Addr) {
>+  MdnsMessage msg;
>   msg.SetId(0x1234);
>   msg.SetQueryOrResponse(false);
>   msg.SetAuthoritative(true);
> 
>-  MDnsResourceRecord answer1;
>+  MdnsResourceRecord answer1;
>   answer1.SetName("webrtc4.org.");
>   answer1.SetType(SectionEntryType::kA);
>   answer1.SetClass(SectionEntryClass::kIN);
>@@ -555,7 +555,7 @@ TEST(MDnsMessageTest, WriteTwoAnswersToIPv4AndIPv6Addr) {
>   answer1.SetTtlSeconds(60);
>   msg.AddAnswerRecord(answer1);
> 
>-  MDnsResourceRecord answer2;
>+  MdnsResourceRecord answer2;
>   answer2.SetName("webrtc6.org.");
>   answer2.SetType(SectionEntryType::kAAAA);
>   answer2.SetClass(SectionEntryClass::kIN);
>@@ -564,7 +564,7 @@ TEST(MDnsMessageTest, WriteTwoAnswersToIPv4AndIPv6Addr) {
>   answer2.SetTtlSeconds(120);
>   msg.AddAnswerRecord(answer2);
> 
>-  WriteMDnsMessageAndCompare(&msg, kTwoAuthoritativeAnswersWithIPv4AndIPv6Addr);
>+  WriteMdnsMessageAndCompare(&msg, kTwoAuthoritativeAnswersWithIPv4AndIPv6Addr);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mockicetransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mockicetransport.h
>index d1863c52935209eee442a483c4265018f6eb2c70..b18ce3d5ade1da9400ccad202bd8adda8caa5fe4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mockicetransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mockicetransport.h
>@@ -47,6 +47,10 @@ class MockIceTransport : public IceTransportInternal {
>   IceTransportState GetState() const override {
>     return IceTransportState::STATE_INIT;
>   }
>+  webrtc::IceTransportState GetIceTransportState() const override {
>+    return webrtc::IceTransportState::kNew;
>+  }
>+
>   const std::string& transport_name() const override { return transport_name_; }
>   int component() const override { return 0; }
>   void SetIceRole(IceRole role) override {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.h
>index b2e92eff1352ac8c605ca0148498194a6a711bef..2948be0bb837d4c04dcbf2081f98b02d0d47d94b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.h
>@@ -13,6 +13,8 @@
> 
> #include <string>
> 
>+#include "rtc_base/system/rtc_export.h"
>+
> namespace cricket {
> 
> // CN_ == "content name".  When we initiate a session, we choose the
>@@ -29,7 +31,7 @@ extern const char CN_OTHER[];
> extern const char GROUP_TYPE_BUNDLE[];
> 
> extern const int ICE_UFRAG_LENGTH;
>-extern const int ICE_PWD_LENGTH;
>+RTC_EXPORT extern const int ICE_PWD_LENGTH;
> extern const size_t ICE_UFRAG_MIN_LENGTH;
> extern const size_t ICE_PWD_MIN_LENGTH;
> extern const size_t ICE_UFRAG_MAX_LENGTH;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.cc
>index b0abe24a0580a12a0fbec025e63af120dec67fb0..f61291c5ddc00826eb4247364ac4f5a1ba251203 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.cc
>@@ -323,6 +323,10 @@ IceTransportState P2PTransportChannel::GetState() const {
>   return state_;
> }
> 
>+webrtc::IceTransportState P2PTransportChannel::GetIceTransportState() const {
>+  return standardized_state_;
>+}
>+
> const std::string& P2PTransportChannel::transport_name() const {
>   return transport_name_;
> }
>@@ -387,6 +391,41 @@ IceTransportState P2PTransportChannel::ComputeState() const {
>   return IceTransportState::STATE_COMPLETED;
> }
> 
>+// Compute the current RTCIceTransportState as described in
>+// https://www.w3.org/TR/webrtc/#dom-rtcicetransportstate
>+// TODO(bugs.webrtc.org/9308): Return IceTransportState::kDisconnected when it
>+// makes sense.
>+// TODO(bugs.webrtc.org/9218): Avoid prematurely signalling kFailed once we have
>+// implemented end-of-candidates signalling.
>+webrtc::IceTransportState P2PTransportChannel::ComputeIceTransportState()
>+    const {
>+  bool has_connection = false;
>+  for (Connection* connection : connections_) {
>+    if (connection->active()) {
>+      has_connection = true;
>+      break;
>+    }
>+  }
>+
>+  switch (gathering_state_) {
>+    case kIceGatheringComplete:
>+      if (has_connection)
>+        return webrtc::IceTransportState::kCompleted;
>+      else
>+        return webrtc::IceTransportState::kFailed;
>+    case kIceGatheringNew:
>+      return webrtc::IceTransportState::kNew;
>+    case kIceGatheringGathering:
>+      if (has_connection)
>+        return webrtc::IceTransportState::kConnected;
>+      else
>+        return webrtc::IceTransportState::kChecking;
>+    default:
>+      RTC_NOTREACHED();
>+      return webrtc::IceTransportState::kFailed;
>+  }
>+}
>+
> void P2PTransportChannel::SetIceParameters(const IceParameters& ice_params) {
>   RTC_DCHECK(network_thread_ == rtc::Thread::Current());
>   RTC_LOG(LS_INFO) << "Set ICE ufrag: " << ice_params.ufrag
>@@ -1827,6 +1866,9 @@ void P2PTransportChannel::SwitchSelectedConnection(Connection* conn) {
> // example, we call this at the end of SortConnectionsAndUpdateState.
> void P2PTransportChannel::UpdateState() {
>   IceTransportState state = ComputeState();
>+  webrtc::IceTransportState current_standardized_state =
>+      ComputeIceTransportState();
>+
>   if (state_ != state) {
>     RTC_LOG(LS_INFO) << ToString()
>                      << ": Transport channel state changed from "
>@@ -1868,6 +1910,10 @@ void P2PTransportChannel::UpdateState() {
>     SignalStateChanged(this);
>   }
> 
>+  if (standardized_state_ != current_standardized_state) {
>+    standardized_state_ = current_standardized_state;
>+    SignalIceTransportStateChanged(this);
>+  }
>   // If our selected connection is "presumed writable" (TURN-TURN with no
>   // CreatePermission required), act like we're already writable to the upper
>   // layers, so they can start media quicker.
>@@ -2319,7 +2365,7 @@ bool P2PTransportChannel::PrunePort(PortInterface* port) {
> void P2PTransportChannel::OnReadPacket(Connection* connection,
>                                        const char* data,
>                                        size_t len,
>-                                       const rtc::PacketTime& packet_time) {
>+                                       int64_t packet_time_us) {
>   RTC_DCHECK(network_thread_ == rtc::Thread::Current());
> 
>   // Do not deliver, if packet doesn't belong to the correct transport channel.
>@@ -2327,7 +2373,7 @@ void P2PTransportChannel::OnReadPacket(Connection* connection,
>     return;
> 
>   // Let the client know of an incoming packet
>-  SignalReadPacket(this, data, len, packet_time, 0);
>+  SignalReadPacket(this, data, len, packet_time_us, 0);
> 
>   // May need to switch the sending connection based on the receiving media path
>   // if this is the controlled side.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.h
>index fbed344cc33e6ec731c9f08600bddb5a8db01fc9..abedd0bdd3489375e52f9436122fcd619ca75369 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.h
>@@ -42,6 +42,7 @@
> #include "rtc_base/asyncpacketsocket.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/strings/string_builder.h"
>+#include "rtc_base/system/rtc_export.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace webrtc {
>@@ -75,7 +76,7 @@ class RemoteCandidate : public Candidate {
> 
> // P2PTransportChannel manages the candidates and connection process to keep
> // two P2P clients connected to each other.
>-class P2PTransportChannel : public IceTransportInternal {
>+class RTC_EXPORT P2PTransportChannel : public IceTransportInternal {
>  public:
>   // For testing only.
>   // TODO(zstein): Remove once AsyncResolverFactory is required.
>@@ -91,6 +92,8 @@ class P2PTransportChannel : public IceTransportInternal {
> 
>   // From TransportChannelImpl:
>   IceTransportState GetState() const override;
>+  webrtc::IceTransportState GetIceTransportState() const override;
>+
>   const std::string& transport_name() const override;
>   int component() const override;
>   bool writable() const override;
>@@ -243,7 +246,13 @@ class P2PTransportChannel : public IceTransportInternal {
>   void UpdateState();
>   void HandleAllTimedOut();
>   void MaybeStopPortAllocatorSessions();
>+
>+  // ComputeIceTransportState computes the RTCIceTransportState as described in
>+  // https://w3c.github.io/webrtc-pc/#dom-rtcicetransportstate. ComputeState
>+  // computes the value we currently export as RTCIceTransportState.
>+  // TODO(bugs.webrtc.org/9308): Remove ComputeState once it's no longer used.
>   IceTransportState ComputeState() const;
>+  webrtc::IceTransportState ComputeIceTransportState() const;
> 
>   Connection* GetBestConnectionOnNetwork(rtc::Network* network) const;
>   bool CreateConnections(const Candidate& remote_candidate,
>@@ -295,7 +304,7 @@ class P2PTransportChannel : public IceTransportInternal {
>   void OnReadPacket(Connection* connection,
>                     const char* data,
>                     size_t len,
>-                    const rtc::PacketTime& packet_time);
>+                    int64_t packet_time_us);
>   void OnSentPacket(const rtc::SentPacket& sent_packet);
>   void OnReadyToSend(Connection* connection);
>   void OnConnectionDestroyed(Connection* connection);
>@@ -407,7 +416,11 @@ class P2PTransportChannel : public IceTransportInternal {
>   std::unique_ptr<webrtc::BasicRegatheringController> regathering_controller_;
>   int64_t last_ping_sent_ms_ = 0;
>   int weak_ping_interval_ = WEAK_PING_INTERVAL;
>+  // TODO(jonasolsson): Remove state_ and rename standardized_state_ once state_
>+  // is no longer used to compute the ICE connection state.
>   IceTransportState state_ = IceTransportState::STATE_INIT;
>+  webrtc::IceTransportState standardized_state_ =
>+      webrtc::IceTransportState::kNew;
>   IceConfig config_;
>   int last_sent_packet_id_ = -1;  // -1 indicates no packet was sent before.
>   bool started_pinging_ = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel_unittest.cc
>index 76b918ca1e4bce92763bdee4571137a73eb5cf5f..2ab3d88afbc597ad7e520c7e8f1bf9ffc0c48cf6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel_unittest.cc
>@@ -699,7 +699,6 @@ class P2PTransportChannelTestBase : public testing::Test,
>     EXPECT_NE(info.packet_type, rtc::PacketType::kUnknown);
>     EXPECT_NE(info.protocol, rtc::PacketInfoProtocolType::kUnknown);
>     EXPECT_TRUE(info.network_id.has_value());
>-    EXPECT_FALSE(info.local_socket_address.IsNil());
>   }
> 
>   void OnReadyToSend(rtc::PacketTransportInternal* transport) {
>@@ -811,7 +810,7 @@ class P2PTransportChannelTestBase : public testing::Test,
>   void OnReadPacket(rtc::PacketTransportInternal* transport,
>                     const char* data,
>                     size_t len,
>-                    const rtc::PacketTime& packet_time,
>+                    const int64_t& /* packet_time_us */,
>                     int flags) {
>     std::list<std::string>& packets = GetPacketList(transport);
>     packets.push_front(std::string(data, len));
>@@ -3204,7 +3203,7 @@ class P2PTransportChannelPingTest : public testing::Test,
>     msg.AddFingerprint();
>     rtc::ByteBufferWriter buf;
>     msg.Write(&buf);
>-    conn->OnReadPacket(buf.Data(), buf.Length(), rtc::CreatePacketTime(0));
>+    conn->OnReadPacket(buf.Data(), buf.Length(), rtc::TimeMicros());
>   }
> 
>   void OnReadyToSend(rtc::PacketTransportInternal* transport) {
>@@ -3613,7 +3612,7 @@ TEST_F(P2PTransportChannelPingTest, TestReceivingStateChange) {
> 
>   clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
>   conn1->ReceivedPing();
>-  conn1->OnReadPacket("ABC", 3, rtc::CreatePacketTime(0));
>+  conn1->OnReadPacket("ABC", 3, rtc::TimeMicros());
>   EXPECT_TRUE_SIMULATED_WAIT(ch.receiving(), kShortTimeout, clock);
>   EXPECT_TRUE_SIMULATED_WAIT(!ch.receiving(), kShortTimeout, clock);
> }
>@@ -3804,7 +3803,7 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBasedOnMediaReceived) {
>   Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2);
>   ASSERT_TRUE(conn2 != nullptr);
>   conn2->ReceivedPingResponse(LOW_RTT, "id");  // Become writable and receiving.
>-  conn2->OnReadPacket("ABC", 3, rtc::CreatePacketTime(0));
>+  conn2->OnReadPacket("ABC", 3, rtc::TimeMicros());
>   EXPECT_EQ(conn2, ch.selected_connection());
>   conn2->ReceivedPingResponse(LOW_RTT, "id");  // Become writable.
> 
>@@ -3832,7 +3831,7 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBasedOnMediaReceived) {
>   // selected connection was nominated by the controlling side.
>   conn2->ReceivedPing();
>   conn2->ReceivedPingResponse(LOW_RTT, "id");
>-  conn2->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
>+  conn2->OnReadPacket("XYZ", 3, rtc::TimeMicros());
>   EXPECT_EQ_WAIT(conn3, ch.selected_connection(), kDefaultTimeout);
> }
> 
>@@ -3861,12 +3860,12 @@ TEST_F(P2PTransportChannelPingTest,
>   // Advance the clock by 1ms so that the last data receiving timestamp of
>   // conn2 is larger.
>   SIMULATED_WAIT(false, 1, clock);
>-  conn2->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
>+  conn2->OnReadPacket("XYZ", 3, rtc::TimeMicros());
>   EXPECT_EQ(1, reset_selected_candidate_pair_switches());
>   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
> 
>   // conn1 also receives data; it becomes selected due to priority again.
>-  conn1->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
>+  conn1->OnReadPacket("XYZ", 3, rtc::TimeMicros());
>   EXPECT_EQ(1, reset_selected_candidate_pair_switches());
>   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
> 
>@@ -3875,7 +3874,7 @@ TEST_F(P2PTransportChannelPingTest,
>   SIMULATED_WAIT(false, 1, clock);
>   // Need to become writable again because it was pruned.
>   conn2->ReceivedPingResponse(LOW_RTT, "id");
>-  conn2->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
>+  conn2->OnReadPacket("XYZ", 3, rtc::TimeMicros());
>   EXPECT_EQ(1, reset_selected_candidate_pair_switches());
>   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn2));
> 
>@@ -3905,7 +3904,7 @@ TEST_F(P2PTransportChannelPingTest,
>   // conn1 received data; it is the selected connection.
>   // Advance the clock to have a non-zero last-data-receiving time.
>   SIMULATED_WAIT(false, 1, clock);
>-  conn1->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
>+  conn1->OnReadPacket("XYZ", 3, rtc::TimeMicros());
>   EXPECT_EQ(1, reset_selected_candidate_pair_switches());
>   EXPECT_TRUE(CandidatePairMatchesNetworkRoute(conn1));
> 
>@@ -4126,7 +4125,7 @@ TEST_F(P2PTransportChannelPingTest, TestDontPruneHighPriorityConnections) {
>   // conn2.
>   NominateConnection(conn1);
>   SIMULATED_WAIT(false, 1, clock);
>-  conn1->OnReadPacket("XYZ", 3, rtc::CreatePacketTime(0));
>+  conn1->OnReadPacket("XYZ", 3, rtc::TimeMicros());
>   SIMULATED_WAIT(conn2->pruned(), 100, clock);
>   EXPECT_FALSE(conn2->pruned());
> }
>@@ -4137,6 +4136,7 @@ TEST_F(P2PTransportChannelPingTest, TestGetState) {
>   clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
>   FakePortAllocator pa(rtc::Thread::Current(), nullptr);
>   P2PTransportChannel ch("test channel", 1, &pa);
>+  EXPECT_EQ(webrtc::IceTransportState::kNew, ch.GetIceTransportState());
>   PrepareChannel(&ch);
>   ch.MaybeStartGathering();
>   EXPECT_EQ(IceTransportState::STATE_INIT, ch.GetState());
>@@ -4144,6 +4144,8 @@ TEST_F(P2PTransportChannelPingTest, TestGetState) {
>   ch.AddRemoteCandidate(CreateUdpCandidate(LOCAL_PORT_TYPE, "2.2.2.2", 2, 1));
>   Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1, &clock);
>   Connection* conn2 = WaitForConnectionTo(&ch, "2.2.2.2", 2, &clock);
>+  // Gathering complete with candidates.
>+  EXPECT_EQ(webrtc::IceTransportState::kCompleted, ch.GetIceTransportState());
>   ASSERT_TRUE(conn1 != nullptr);
>   ASSERT_TRUE(conn2 != nullptr);
>   // Now there are two connections, so the transport channel is connecting.
>@@ -4156,6 +4158,7 @@ TEST_F(P2PTransportChannelPingTest, TestGetState) {
>   // Need to wait until the channel state is updated.
>   EXPECT_EQ_SIMULATED_WAIT(IceTransportState::STATE_FAILED, ch.GetState(),
>                            kShortTimeout, clock);
>+  EXPECT_EQ(webrtc::IceTransportState::kFailed, ch.GetIceTransportState());
> }
> 
> // Test that when a low-priority connection is pruned, it is not deleted
>@@ -4590,7 +4593,7 @@ TEST(P2PTransportChannelResolverTest, HostnameCandidateIsResolved) {
> // prflx candidate is updated to a host candidate after the name resolution is
> // done.
> TEST_F(P2PTransportChannelTest,
>-       PeerReflexiveCandidateBeforeSignalingWithMDnsName) {
>+       PeerReflexiveCandidateBeforeSignalingWithMdnsName) {
>   rtc::MockAsyncResolver mock_async_resolver;
>   webrtc::MockAsyncResolverFactory mock_async_resolver_factory;
>   EXPECT_CALL(mock_async_resolver_factory, Create())
>@@ -4601,7 +4604,7 @@ TEST_F(P2PTransportChannelTest,
>   ConfigureEndpoints(OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
>   // ICE parameter will be set up when creating the channels.
>   set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
>-  GetEndpoint(0)->network_manager_.CreateMDnsResponder();
>+  GetEndpoint(0)->network_manager_.CreateMdnsResponder();
>   GetEndpoint(1)->async_resolver_factory_ = &mock_async_resolver_factory;
>   CreateChannels();
>   // Pause sending candidates from both endpoints until we find out what port
>@@ -4656,7 +4659,7 @@ TEST_F(P2PTransportChannelTest,
> // a host candidate if the hostname candidate turns out to have the same IP
> // address after the resolution completes.
> TEST_F(P2PTransportChannelTest,
>-       PeerReflexiveCandidateDuringResolvingHostCandidateWithMDnsName) {
>+       PeerReflexiveCandidateDuringResolvingHostCandidateWithMdnsName) {
>   NiceMock<rtc::MockAsyncResolver> mock_async_resolver;
>   webrtc::MockAsyncResolverFactory mock_async_resolver_factory;
>   EXPECT_CALL(mock_async_resolver_factory, Create())
>@@ -4667,7 +4670,7 @@ TEST_F(P2PTransportChannelTest,
>   ConfigureEndpoints(OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
>   // ICE parameter will be set up when creating the channels.
>   set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
>-  GetEndpoint(0)->network_manager_.CreateMDnsResponder();
>+  GetEndpoint(0)->network_manager_.CreateMdnsResponder();
>   GetEndpoint(1)->async_resolver_factory_ = &mock_async_resolver_factory;
>   CreateChannels();
>   // Pause sending candidates from both endpoints until we find out what port
>@@ -4723,7 +4726,7 @@ TEST_F(P2PTransportChannelTest,
> // Test that if we only gather and signal a host candidate, the IP address of
> // which is obfuscated by an mDNS name, and if the peer can complete the name
> // resolution with the correct IP address, we can have a p2p connection.
>-TEST_F(P2PTransportChannelTest, CanConnectWithHostCandidateWithMDnsName) {
>+TEST_F(P2PTransportChannelTest, CanConnectWithHostCandidateWithMdnsName) {
>   NiceMock<rtc::MockAsyncResolver> mock_async_resolver;
>   webrtc::MockAsyncResolverFactory mock_async_resolver_factory;
>   EXPECT_CALL(mock_async_resolver_factory, Create())
>@@ -4734,7 +4737,7 @@ TEST_F(P2PTransportChannelTest, CanConnectWithHostCandidateWithMDnsName) {
>   ConfigureEndpoints(OPEN, OPEN, kOnlyLocalPorts, kOnlyLocalPorts);
>   // ICE parameter will be set up when creating the channels.
>   set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
>-  GetEndpoint(0)->network_manager_.CreateMDnsResponder();
>+  GetEndpoint(0)->network_manager_.CreateMdnsResponder();
>   GetEndpoint(1)->async_resolver_factory_ = &mock_async_resolver_factory;
>   CreateChannels();
>   // Pause sending candidates from both endpoints until we find out what port
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetsocketfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetsocketfactory.h
>index 4667bb1fd70028b9e06d4f655756306b62363ed6..c903df06c8a98dd45653cd4e03fa4f050c1be575 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetsocketfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetsocketfactory.h
>@@ -17,6 +17,7 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/proxyinfo.h"
> #include "rtc_base/sslcertificate.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace rtc {
> 
>@@ -36,7 +37,7 @@ struct PacketSocketTcpOptions {
> class AsyncPacketSocket;
> class AsyncResolverInterface;
> 
>-class PacketSocketFactory {
>+class RTC_EXPORT PacketSocketFactory {
>  public:
>   enum Options {
>     OPT_STUN = 0x04,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.h
>index ca7f0036114df836f1bccc68e0c5855987b3371b..7d50f9c0eea1d977bb805a2fb6478deddb591264 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.h
>@@ -21,15 +21,16 @@
> #include "rtc_base/asyncpacketsocket.h"
> #include "rtc_base/networkroute.h"
> #include "rtc_base/socket.h"
>+#include "rtc_base/system/rtc_export.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> struct PacketOptions;
>-struct PacketTime;
> struct SentPacket;
> 
>-class PacketTransportInternal : public virtual webrtc::PacketTransportInterface,
>-                                public sigslot::has_slots<> {
>+class RTC_EXPORT PacketTransportInternal
>+    : public virtual webrtc::PacketTransportInterface,
>+      public sigslot::has_slots<> {
>  public:
>   virtual const std::string& transport_name() const = 0;
> 
>@@ -85,7 +86,9 @@ class PacketTransportInternal : public virtual webrtc::PacketTransportInterface,
>   sigslot::signal5<PacketTransportInternal*,
>                    const char*,
>                    size_t,
>-                   const rtc::PacketTime&,
>+                   // TODO(bugs.webrtc.org/9584): Change to passing the int64_t
>+                   // timestamp by value.
>+                   const int64_t&,
>                    int>
>       SignalReadPacket;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc
>index ac7521e66727f6e9434807f650c1ae3bd2f63179..5b8e02d28ff894d3d376efab3b07f0f3bee40cbf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc
>@@ -17,6 +17,7 @@
> #include <vector>
> 
> #include "absl/memory/memory.h"
>+#include "absl/strings/match.h"
> #include "p2p/base/portallocator.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/crc32.h"
>@@ -27,7 +28,6 @@
> #include "rtc_base/network.h"
> #include "rtc_base/numerics/safe_minmax.h"
> #include "rtc_base/stringencode.h"
>-#include "rtc_base/stringutils.h"
> #include "rtc_base/third_party/base64/base64.h"
> 
> namespace {
>@@ -183,7 +183,7 @@ const char* ProtoToString(ProtocolType proto) {
> 
> bool StringToProto(const char* value, ProtocolType* proto) {
>   for (size_t i = 0; i <= PROTO_LAST; ++i) {
>-    if (_stricmp(PROTO_NAMES[i], value) == 0) {
>+    if (absl::EqualsIgnoreCase(PROTO_NAMES[i], value)) {
>       *proto = static_cast<ProtocolType>(i);
>       return true;
>     }
>@@ -399,9 +399,9 @@ void Port::AddAddress(const rtc::SocketAddress& address,
>                       const std::string& type,
>                       uint32_t type_preference,
>                       uint32_t relay_preference,
>-                      bool final) {
>+                      bool is_final) {
>   AddAddress(address, base_address, related_address, protocol, relay_protocol,
>-             tcptype, type, type_preference, relay_preference, "", final);
>+             tcptype, type, type_preference, relay_preference, "", is_final);
> }
> 
> void Port::AddAddress(const rtc::SocketAddress& address,
>@@ -430,44 +430,60 @@ void Port::AddAddress(const rtc::SocketAddress& address,
>   c.set_network_name(network_->name());
>   c.set_network_type(network_->type());
>   c.set_url(url);
>+  c.set_related_address(related_address);
>+
>+  bool pending = MaybeObfuscateAddress(&c, type, is_final);
>+
>+  if (!pending) {
>+    FinishAddingAddress(c, is_final);
>+  }
>+}
>+
>+bool Port::MaybeObfuscateAddress(Candidate* c,
>+                                 const std::string& type,
>+                                 bool is_final) {
>   // TODO(bugs.webrtc.org/9723): Use a config to control the feature of IP
>   // handling with mDNS.
>-  if (network_->GetMDnsResponder() != nullptr) {
>-    // Obfuscate the IP address of a host candidates by an mDNS hostname.
>-    if (type == LOCAL_PORT_TYPE) {
>-      auto weak_ptr = weak_factory_.GetWeakPtr();
>-      auto callback = [weak_ptr, c, is_final](const rtc::IPAddress& addr,
>-                                              const std::string& name) mutable {
>-        RTC_DCHECK(c.address().ipaddr() == addr);
>-        rtc::SocketAddress hostname_address(name, c.address().port());
>-        // In Port and Connection, we need the IP address information to
>-        // correctly handle the update of candidate type to prflx. The removal
>-        // of IP address when signaling this candidate will take place in
>-        // BasicPortAllocatorSession::OnCandidateReady, via SanitizeCandidate.
>-        hostname_address.SetResolvedIP(addr);
>-        c.set_address(hostname_address);
>-        RTC_DCHECK(c.related_address() == rtc::SocketAddress());
>-        if (weak_ptr != nullptr) {
>-          weak_ptr->FinishAddingAddress(c, is_final);
>-        }
>-      };
>-      network_->GetMDnsResponder()->CreateNameForAddress(c.address().ipaddr(),
>-                                                         callback);
>-      return;
>-    }
>-    // For other types of candidates, the related address should be set to
>-    // 0.0.0.0 or ::0.
>-    c.set_related_address(rtc::SocketAddress());
>-  } else {
>-    c.set_related_address(related_address);
>+  if (network_->GetMdnsResponder() == nullptr) {
>+    return false;
>+  }
>+  if (type != LOCAL_PORT_TYPE) {
>+    return false;
>   }
>-  FinishAddingAddress(c, is_final);
>+
>+  auto copy = *c;
>+  auto weak_ptr = weak_factory_.GetWeakPtr();
>+  auto callback = [weak_ptr, copy, is_final](const rtc::IPAddress& addr,
>+                                             const std::string& name) mutable {
>+    RTC_DCHECK(copy.address().ipaddr() == addr);
>+    rtc::SocketAddress hostname_address(name, copy.address().port());
>+    // In Port and Connection, we need the IP address information to
>+    // correctly handle the update of candidate type to prflx. The removal
>+    // of IP address when signaling this candidate will take place in
>+    // BasicPortAllocatorSession::OnCandidateReady, via SanitizeCandidate.
>+    hostname_address.SetResolvedIP(addr);
>+    copy.set_address(hostname_address);
>+    copy.set_related_address(rtc::SocketAddress());
>+    if (weak_ptr != nullptr) {
>+      weak_ptr->set_mdns_name_registration_status(
>+          MdnsNameRegistrationStatus::kCompleted);
>+      weak_ptr->FinishAddingAddress(copy, is_final);
>+    }
>+  };
>+  set_mdns_name_registration_status(MdnsNameRegistrationStatus::kInProgress);
>+  network_->GetMdnsResponder()->CreateNameForAddress(copy.address().ipaddr(),
>+                                                     callback);
>+  return true;
> }
> 
> void Port::FinishAddingAddress(const Candidate& c, bool is_final) {
>   candidates_.push_back(c);
>   SignalCandidateReady(this, c);
> 
>+  PostAddAddress(is_final);
>+}
>+
>+void Port::PostAddAddress(bool is_final) {
>   if (is_final) {
>     SignalPortComplete(this);
>   }
>@@ -773,7 +789,7 @@ bool Port::HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
>                                 const char* data,
>                                 size_t size,
>                                 const rtc::SocketAddress& remote_addr,
>-                                const rtc::PacketTime& packet_time) {
>+                                int64_t packet_time_us) {
>   RTC_NOTREACHED();
>   return false;
> }
>@@ -1236,7 +1252,7 @@ void Connection::OnSendStunPacket(const void* data,
> 
> void Connection::OnReadPacket(const char* data,
>                               size_t size,
>-                              const rtc::PacketTime& packet_time) {
>+                              int64_t packet_time_us) {
>   std::unique_ptr<IceMessage> msg;
>   std::string remote_ufrag;
>   const rtc::SocketAddress& addr(remote_candidate_.address());
>@@ -1246,7 +1262,7 @@ void Connection::OnReadPacket(const char* data,
>     last_data_received_ = rtc::TimeMillis();
>     UpdateReceiving(last_data_received_);
>     recv_rate_tracker_.AddSamples(size);
>-    SignalReadPacket(this, data, size, packet_time);
>+    SignalReadPacket(this, data, size, packet_time_us);
> 
>     // If timed out sending writability checks, start up again
>     if (!pruned_ && (write_state_ == STATE_WRITE_TIMEOUT)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>index 85711e41f56f93892e43d733b9a5cb3048cda967..9a8f92a96e45fd9a39e8cf8db4b53c5356e640bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>@@ -37,6 +37,7 @@
> #include "rtc_base/proxyinfo.h"
> #include "rtc_base/ratetracker.h"
> #include "rtc_base/socketaddress.h"
>+#include "rtc_base/system/rtc_export.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/weak_ptr.h"
>@@ -49,7 +50,7 @@ class ConnectionRequest;
> extern const char LOCAL_PORT_TYPE[];
> extern const char STUN_PORT_TYPE[];
> extern const char PRFLX_PORT_TYPE[];
>-extern const char RELAY_PORT_TYPE[];
>+RTC_EXPORT extern const char RELAY_PORT_TYPE[];
> 
> // RFC 6544, TCP candidate encoding rules.
> extern const int DISCARD_PORT;
>@@ -83,6 +84,18 @@ enum class IceCandidatePairState {
>   // frozen because we have not implemented ICE freezing logic.
> };
> 
>+enum class MdnsNameRegistrationStatus {
>+  // IP concealment with mDNS is not enabled or the name registration process is
>+  // not started yet.
>+  kNotStarted,
>+  // A request to create and register an mDNS name for a local IP address of a
>+  // host candidate is sent to the mDNS responder.
>+  kInProgress,
>+  // The name registration is complete and the created name is returned by the
>+  // mDNS responder.
>+  kCompleted,
>+};
>+
> // Stats that we can return about the port of a STUN candidate.
> class StunStats {
>  public:
>@@ -305,7 +318,7 @@ class Port : public PortInterface,
>                                     const char* data,
>                                     size_t size,
>                                     const rtc::SocketAddress& remote_addr,
>-                                    const rtc::PacketTime& packet_time);
>+                                    int64_t packet_time_us);
> 
>   // Shall the port handle packet from this |remote_addr|.
>   // This method is overridden by TurnPort.
>@@ -392,7 +405,7 @@ class Port : public PortInterface,
>                   const std::string& type,
>                   uint32_t type_preference,
>                   uint32_t relay_preference,
>-                  bool final);
>+                  bool is_final);
> 
>   void AddAddress(const rtc::SocketAddress& address,
>                   const rtc::SocketAddress& base_address,
>@@ -408,6 +421,8 @@ class Port : public PortInterface,
> 
>   void FinishAddingAddress(const Candidate& c, bool is_final);
> 
>+  virtual void PostAddAddress(bool is_final);
>+
>   // Adds the given connection to the map keyed by the remote candidate address.
>   // If an existing connection has the same address, the existing one will be
>   // replaced and destroyed.
>@@ -443,6 +458,13 @@ class Port : public PortInterface,
> 
>   void CopyPortInformationToPacketInfo(rtc::PacketInfo* info) const;
> 
>+  MdnsNameRegistrationStatus mdns_name_registration_status() const {
>+    return mdns_name_registration_status_;
>+  }
>+  void set_mdns_name_registration_status(MdnsNameRegistrationStatus status) {
>+    mdns_name_registration_status_ = status;
>+  }
>+
>  private:
>   void Construct();
>   // Called when one of our connections deletes itself.
>@@ -487,9 +509,15 @@ class Port : public PortInterface,
>   int16_t network_cost_;
>   State state_ = State::INIT;
>   int64_t last_time_all_connections_removed_ = 0;
>+  MdnsNameRegistrationStatus mdns_name_registration_status_ =
>+      MdnsNameRegistrationStatus::kNotStarted;
> 
>   rtc::WeakPtrFactory<Port> weak_factory_;
> 
>+  bool MaybeObfuscateAddress(Candidate* c,
>+                             const std::string& type,
>+                             bool is_final);
>+
>   friend class Connection;
> };
> 
>@@ -578,15 +606,12 @@ class Connection : public CandidatePairInterface,
>   // Error if Send() returns < 0
>   virtual int GetError() = 0;
> 
>-  sigslot::signal4<Connection*, const char*, size_t, const rtc::PacketTime&>
>-      SignalReadPacket;
>+  sigslot::signal4<Connection*, const char*, size_t, int64_t> SignalReadPacket;
> 
>   sigslot::signal1<Connection*> SignalReadyToSend;
> 
>   // Called when a packet is received on this connection.
>-  void OnReadPacket(const char* data,
>-                    size_t size,
>-                    const rtc::PacketTime& packet_time);
>+  void OnReadPacket(const char* data, size_t size, int64_t packet_time_us);
> 
>   // Called when the socket is currently able to send.
>   void OnReadyToSend();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port_unittest.cc
>index 50280301d4ce18ffbf7b1eb57b0b8fd3ad55b712..71929423cb620048e0e32183362b66216d994270 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port_unittest.cc
>@@ -91,18 +91,18 @@ SocketAddress GetAddress(Port* port) {
>   return GetCandidate(port).address();
> }
> 
>-IceMessage* CopyStunMessage(const IceMessage* src) {
>-  IceMessage* dst = new IceMessage();
>+std::unique_ptr<IceMessage> CopyStunMessage(const IceMessage& src) {
>+  auto dst = absl::make_unique<IceMessage>();
>   ByteBufferWriter buf;
>-  src->Write(&buf);
>+  src.Write(&buf);
>   ByteBufferReader read_buf(buf);
>   dst->Read(&read_buf);
>   return dst;
> }
> 
>-bool WriteStunMessage(const StunMessage* msg, ByteBufferWriter* buf) {
>+bool WriteStunMessage(const StunMessage& msg, ByteBufferWriter* buf) {
>   buf->Resize(0);  // clear out any existing buffer contents
>-  return msg->Write(buf);
>+  return msg.Write(buf);
> }
> 
> }  // namespace
>@@ -189,17 +189,15 @@ class TestPort : public Port {
>                      const rtc::PacketOptions& options,
>                      bool payload) {
>     if (!payload) {
>-      IceMessage* msg = new IceMessage;
>-      auto* buf =
>-          new rtc::BufferT<uint8_t>(static_cast<const char*>(data), size);
>+      auto msg = absl::make_unique<IceMessage>();
>+      auto buf = absl::make_unique<rtc::BufferT<uint8_t>>(
>+          static_cast<const char*>(data), size);
>       ByteBufferReader read_buf(*buf);
>       if (!msg->Read(&read_buf)) {
>-        delete msg;
>-        delete buf;
>         return -1;
>       }
>-      last_stun_buf_.reset(buf);
>-      last_stun_msg_.reset(msg);
>+      last_stun_buf_ = std::move(buf);
>+      last_stun_msg_ = std::move(msg);
>     }
>     return static_cast<int>(size);
>   }
>@@ -234,24 +232,18 @@ static void SendPingAndReceiveResponse(Connection* lconn,
>   ASSERT_TRUE_WAIT(lport->last_stun_msg(), kDefaultTimeout);
>   ASSERT_TRUE(lport->last_stun_buf());
>   rconn->OnReadPacket(lport->last_stun_buf()->data<char>(),
>-                      lport->last_stun_buf()->size(), rtc::PacketTime());
>+                      lport->last_stun_buf()->size(), /* packet_time_us */ -1);
>   clock->AdvanceTime(webrtc::TimeDelta::ms(ms));
>   ASSERT_TRUE_WAIT(rport->last_stun_msg(), kDefaultTimeout);
>   ASSERT_TRUE(rport->last_stun_buf());
>   lconn->OnReadPacket(rport->last_stun_buf()->data<char>(),
>-                      rport->last_stun_buf()->size(), rtc::PacketTime());
>+                      rport->last_stun_buf()->size(), /* packet_time_us */ -1);
> }
> 
> class TestChannel : public sigslot::has_slots<> {
>  public:
>   // Takes ownership of |p1| (but not |p2|).
>-  explicit TestChannel(Port* p1)
>-      : ice_mode_(ICEMODE_FULL),
>-        port_(p1),
>-        complete_count_(0),
>-        conn_(NULL),
>-        remote_request_(),
>-        nominated_(false) {
>+  explicit TestChannel(std::unique_ptr<Port> p1) : port_(std::move(p1)) {
>     port_->SignalPortComplete.connect(this, &TestChannel::OnPortComplete);
>     port_->SignalUnknownAddress.connect(this, &TestChannel::OnUnknownAddress);
>     port_->SignalDestroyed.connect(this, &TestChannel::OnSrcPortDestroyed);
>@@ -327,7 +319,7 @@ class TestChannel : public sigslot::has_slots<> {
>     EXPECT_TRUE(mi_attr != NULL);
>     EXPECT_TRUE(fingerprint_attr != NULL);
>     remote_address_ = addr;
>-    remote_request_.reset(CopyStunMessage(msg));
>+    remote_request_ = CopyStunMessage(*msg);
>     remote_frag_ = rf;
>   }
> 
>@@ -362,15 +354,15 @@ class TestChannel : public sigslot::has_slots<> {
>     connection_ready_to_send_ = true;
>   }
> 
>-  IceMode ice_mode_;
>+  IceMode ice_mode_ = ICEMODE_FULL;
>   std::unique_ptr<Port> port_;
> 
>-  int complete_count_;
>-  Connection* conn_;
>+  int complete_count_ = 0;
>+  Connection* conn_ = nullptr;
>   SocketAddress remote_address_;
>   std::unique_ptr<StunMessage> remote_request_;
>   std::string remote_frag_;
>-  bool nominated_;
>+  bool nominated_ = false;
>   bool connection_ready_to_send_ = false;
> };
> 
>@@ -400,82 +392,85 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
> 
>  protected:
>   void TestLocalToLocal() {
>-    Port* port1 = CreateUdpPort(kLocalAddr1);
>+    auto port1 = CreateUdpPort(kLocalAddr1);
>     port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-    Port* port2 = CreateUdpPort(kLocalAddr2);
>+    auto port2 = CreateUdpPort(kLocalAddr2);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-    TestConnectivity("udp", port1, "udp", port2, true, true, true, true);
>+    TestConnectivity("udp", std::move(port1), "udp", std::move(port2), true,
>+                     true, true, true);
>   }
>   void TestLocalToStun(NATType ntype) {
>-    Port* port1 = CreateUdpPort(kLocalAddr1);
>+    auto port1 = CreateUdpPort(kLocalAddr1);
>     port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-    nat_server2_.reset(CreateNatServer(kNatAddr2, ntype));
>-    Port* port2 = CreateStunPort(kLocalAddr2, &nat_socket_factory2_);
>+    nat_server2_ = CreateNatServer(kNatAddr2, ntype);
>+    auto port2 = CreateStunPort(kLocalAddr2, &nat_socket_factory2_);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-    TestConnectivity("udp", port1, StunName(ntype), port2,
>+    TestConnectivity("udp", std::move(port1), StunName(ntype), std::move(port2),
>                      ntype == NAT_OPEN_CONE, true, ntype != NAT_SYMMETRIC,
>                      true);
>   }
>   void TestLocalToRelay(RelayType rtype, ProtocolType proto) {
>-    Port* port1 = CreateUdpPort(kLocalAddr1);
>+    auto port1 = CreateUdpPort(kLocalAddr1);
>     port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-    Port* port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_UDP);
>+    auto port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_UDP);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-    TestConnectivity("udp", port1, RelayName(rtype, proto), port2,
>-                     rtype == RELAY_GTURN, true, true, true);
>+    TestConnectivity("udp", std::move(port1), RelayName(rtype, proto),
>+                     std::move(port2), rtype == RELAY_GTURN, true, true, true);
>   }
>   void TestStunToLocal(NATType ntype) {
>-    nat_server1_.reset(CreateNatServer(kNatAddr1, ntype));
>-    Port* port1 = CreateStunPort(kLocalAddr1, &nat_socket_factory1_);
>+    nat_server1_ = CreateNatServer(kNatAddr1, ntype);
>+    auto port1 = CreateStunPort(kLocalAddr1, &nat_socket_factory1_);
>     port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-    Port* port2 = CreateUdpPort(kLocalAddr2);
>+    auto port2 = CreateUdpPort(kLocalAddr2);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-    TestConnectivity(StunName(ntype), port1, "udp", port2, true,
>-                     ntype != NAT_SYMMETRIC, true, true);
>+    TestConnectivity(StunName(ntype), std::move(port1), "udp", std::move(port2),
>+                     true, ntype != NAT_SYMMETRIC, true, true);
>   }
>   void TestStunToStun(NATType ntype1, NATType ntype2) {
>-    nat_server1_.reset(CreateNatServer(kNatAddr1, ntype1));
>-    Port* port1 = CreateStunPort(kLocalAddr1, &nat_socket_factory1_);
>+    nat_server1_ = CreateNatServer(kNatAddr1, ntype1);
>+    auto port1 = CreateStunPort(kLocalAddr1, &nat_socket_factory1_);
>     port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-    nat_server2_.reset(CreateNatServer(kNatAddr2, ntype2));
>-    Port* port2 = CreateStunPort(kLocalAddr2, &nat_socket_factory2_);
>+    nat_server2_ = CreateNatServer(kNatAddr2, ntype2);
>+    auto port2 = CreateStunPort(kLocalAddr2, &nat_socket_factory2_);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-    TestConnectivity(StunName(ntype1), port1, StunName(ntype2), port2,
>-                     ntype2 == NAT_OPEN_CONE, ntype1 != NAT_SYMMETRIC,
>-                     ntype2 != NAT_SYMMETRIC,
>+    TestConnectivity(StunName(ntype1), std::move(port1), StunName(ntype2),
>+                     std::move(port2), ntype2 == NAT_OPEN_CONE,
>+                     ntype1 != NAT_SYMMETRIC, ntype2 != NAT_SYMMETRIC,
>                      ntype1 + ntype2 < (NAT_PORT_RESTRICTED + NAT_SYMMETRIC));
>   }
>   void TestStunToRelay(NATType ntype, RelayType rtype, ProtocolType proto) {
>-    nat_server1_.reset(CreateNatServer(kNatAddr1, ntype));
>-    Port* port1 = CreateStunPort(kLocalAddr1, &nat_socket_factory1_);
>+    nat_server1_ = CreateNatServer(kNatAddr1, ntype);
>+    auto port1 = CreateStunPort(kLocalAddr1, &nat_socket_factory1_);
>     port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-    Port* port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_UDP);
>+    auto port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_UDP);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-    TestConnectivity(StunName(ntype), port1, RelayName(rtype, proto), port2,
>-                     rtype == RELAY_GTURN, ntype != NAT_SYMMETRIC, true, true);
>+    TestConnectivity(StunName(ntype), std::move(port1), RelayName(rtype, proto),
>+                     std::move(port2), rtype == RELAY_GTURN,
>+                     ntype != NAT_SYMMETRIC, true, true);
>   }
>   void TestTcpToTcp() {
>-    Port* port1 = CreateTcpPort(kLocalAddr1);
>+    auto port1 = CreateTcpPort(kLocalAddr1);
>     port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-    Port* port2 = CreateTcpPort(kLocalAddr2);
>+    auto port2 = CreateTcpPort(kLocalAddr2);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-    TestConnectivity("tcp", port1, "tcp", port2, true, false, true, true);
>+    TestConnectivity("tcp", std::move(port1), "tcp", std::move(port2), true,
>+                     false, true, true);
>   }
>   void TestTcpToRelay(RelayType rtype, ProtocolType proto) {
>-    Port* port1 = CreateTcpPort(kLocalAddr1);
>+    auto port1 = CreateTcpPort(kLocalAddr1);
>     port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-    Port* port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_TCP);
>+    auto port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_TCP);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-    TestConnectivity("tcp", port1, RelayName(rtype, proto), port2,
>-                     rtype == RELAY_GTURN, false, true, true);
>+    TestConnectivity("tcp", std::move(port1), RelayName(rtype, proto),
>+                     std::move(port2), rtype == RELAY_GTURN, false, true, true);
>   }
>   void TestSslTcpToRelay(RelayType rtype, ProtocolType proto) {
>-    Port* port1 = CreateTcpPort(kLocalAddr1);
>+    auto port1 = CreateTcpPort(kLocalAddr1);
>     port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-    Port* port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_SSLTCP);
>+    auto port2 = CreateRelayPort(kLocalAddr2, rtype, proto, PROTO_SSLTCP);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-    TestConnectivity("ssltcp", port1, RelayName(rtype, proto), port2,
>-                     rtype == RELAY_GTURN, false, true, true);
>+    TestConnectivity("ssltcp", std::move(port1), RelayName(rtype, proto),
>+                     std::move(port2), rtype == RELAY_GTURN, false, true, true);
>   }
> 
>   rtc::Network* MakeNetwork(const SocketAddress& addr) {
>@@ -485,71 +480,71 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>   }
> 
>   // helpers for above functions
>-  UDPPort* CreateUdpPort(const SocketAddress& addr) {
>+  std::unique_ptr<UDPPort> CreateUdpPort(const SocketAddress& addr) {
>     return CreateUdpPort(addr, &socket_factory_);
>   }
>-  UDPPort* CreateUdpPort(const SocketAddress& addr,
>-                         PacketSocketFactory* socket_factory) {
>+  std::unique_ptr<UDPPort> CreateUdpPort(const SocketAddress& addr,
>+                                         PacketSocketFactory* socket_factory) {
>     return UDPPort::Create(&main_, socket_factory, MakeNetwork(addr), 0, 0,
>                            username_, password_, std::string(), true,
>                            absl::nullopt);
>   }
>-  TCPPort* CreateTcpPort(const SocketAddress& addr) {
>+  std::unique_ptr<TCPPort> CreateTcpPort(const SocketAddress& addr) {
>     return CreateTcpPort(addr, &socket_factory_);
>   }
>-  TCPPort* CreateTcpPort(const SocketAddress& addr,
>-                         PacketSocketFactory* socket_factory) {
>+  std::unique_ptr<TCPPort> CreateTcpPort(const SocketAddress& addr,
>+                                         PacketSocketFactory* socket_factory) {
>     return TCPPort::Create(&main_, socket_factory, MakeNetwork(addr), 0, 0,
>                            username_, password_, true);
>   }
>-  StunPort* CreateStunPort(const SocketAddress& addr,
>-                           rtc::PacketSocketFactory* factory) {
>+  std::unique_ptr<StunPort> CreateStunPort(const SocketAddress& addr,
>+                                           rtc::PacketSocketFactory* factory) {
>     ServerAddresses stun_servers;
>     stun_servers.insert(kStunAddr);
>     return StunPort::Create(&main_, factory, MakeNetwork(addr), 0, 0, username_,
>                             password_, stun_servers, std::string(),
>                             absl::nullopt);
>   }
>-  Port* CreateRelayPort(const SocketAddress& addr,
>-                        RelayType rtype,
>-                        ProtocolType int_proto,
>-                        ProtocolType ext_proto) {
>+  std::unique_ptr<Port> CreateRelayPort(const SocketAddress& addr,
>+                                        RelayType rtype,
>+                                        ProtocolType int_proto,
>+                                        ProtocolType ext_proto) {
>     if (rtype == RELAY_TURN) {
>       return CreateTurnPort(addr, &socket_factory_, int_proto, ext_proto);
>     } else {
>       return CreateGturnPort(addr, int_proto, ext_proto);
>     }
>   }
>-  TurnPort* CreateTurnPort(const SocketAddress& addr,
>-                           PacketSocketFactory* socket_factory,
>-                           ProtocolType int_proto,
>-                           ProtocolType ext_proto) {
>+  std::unique_ptr<TurnPort> CreateTurnPort(const SocketAddress& addr,
>+                                           PacketSocketFactory* socket_factory,
>+                                           ProtocolType int_proto,
>+                                           ProtocolType ext_proto) {
>     SocketAddress server_addr =
>         int_proto == PROTO_TCP ? kTurnTcpIntAddr : kTurnUdpIntAddr;
>     return CreateTurnPort(addr, socket_factory, int_proto, ext_proto,
>                           server_addr);
>   }
>-  TurnPort* CreateTurnPort(const SocketAddress& addr,
>-                           PacketSocketFactory* socket_factory,
>-                           ProtocolType int_proto,
>-                           ProtocolType ext_proto,
>-                           const rtc::SocketAddress& server_addr) {
>-    return TurnPort::Create(
>-        &main_, socket_factory, MakeNetwork(addr), 0, 0, username_, password_,
>-        ProtocolAddress(server_addr, int_proto), kRelayCredentials, 0,
>-        std::string(), std::vector<std::string>(), std::vector<std::string>(),
>-        nullptr, nullptr);
>-  }
>-  RelayPort* CreateGturnPort(const SocketAddress& addr,
>-                             ProtocolType int_proto,
>-                             ProtocolType ext_proto) {
>-    RelayPort* port = CreateGturnPort(addr);
>+  std::unique_ptr<TurnPort> CreateTurnPort(
>+      const SocketAddress& addr,
>+      PacketSocketFactory* socket_factory,
>+      ProtocolType int_proto,
>+      ProtocolType ext_proto,
>+      const rtc::SocketAddress& server_addr) {
>+    return TurnPort::Create(&main_, socket_factory, MakeNetwork(addr), 0, 0,
>+                            username_, password_,
>+                            ProtocolAddress(server_addr, int_proto),
>+                            kRelayCredentials, 0, "", {}, {}, nullptr, nullptr);
>+  }
>+  std::unique_ptr<RelayPort> CreateGturnPort(const SocketAddress& addr,
>+                                             ProtocolType int_proto,
>+                                             ProtocolType ext_proto) {
>+    std::unique_ptr<RelayPort> port = CreateGturnPort(addr);
>     SocketAddress addrs[] = {kRelayUdpIntAddr, kRelayTcpIntAddr,
>                              kRelaySslTcpIntAddr};
>     port->AddServerAddress(ProtocolAddress(addrs[int_proto], int_proto));
>     return port;
>   }
>-  RelayPort* CreateGturnPort(const SocketAddress& addr) {
>+  std::unique_ptr<RelayPort> CreateGturnPort(const SocketAddress& addr) {
>     // TODO(pthatcher):  Remove GTURN.
>     // Generate a username with length of 16 for Gturn only.
>     std::string username = rtc::CreateRandomString(kGturnUserNameLength);
>@@ -558,9 +553,10 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>     // TODO(?): Add an external address for ext_proto, so that the
>     // other side can connect to this port using a non-UDP protocol.
>   }
>-  rtc::NATServer* CreateNatServer(const SocketAddress& addr,
>-                                  rtc::NATType type) {
>-    return new rtc::NATServer(type, ss_.get(), addr, addr, ss_.get(), addr);
>+  std::unique_ptr<rtc::NATServer> CreateNatServer(const SocketAddress& addr,
>+                                                  rtc::NATType type) {
>+    return absl::make_unique<rtc::NATServer>(type, ss_.get(), addr, addr,
>+                                             ss_.get(), addr);
>   }
>   static const char* StunName(NATType type) {
>     switch (type) {
>@@ -612,9 +608,9 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
> 
>   // This does all the work and then deletes |port1| and |port2|.
>   void TestConnectivity(const char* name1,
>-                        Port* port1,
>+                        std::unique_ptr<Port> port1,
>                         const char* name2,
>-                        Port* port2,
>+                        std::unique_ptr<Port> port2,
>                         bool accept,
>                         bool same_addr1,
>                         bool same_addr2,
>@@ -680,17 +676,17 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
> 
>   void TestTcpReconnect(bool ping_after_disconnected,
>                         bool send_after_disconnected) {
>-    Port* port1 = CreateTcpPort(kLocalAddr1);
>+    auto port1 = CreateTcpPort(kLocalAddr1);
>     port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-    Port* port2 = CreateTcpPort(kLocalAddr2);
>+    auto port2 = CreateTcpPort(kLocalAddr2);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
> 
>     port1->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
>     port2->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
> 
>     // Set up channels and ensure both ports will be deleted.
>-    TestChannel ch1(port1);
>-    TestChannel ch2(port2);
>+    TestChannel ch1(std::move(port1));
>+    TestChannel ch2(std::move(port2));
>     EXPECT_EQ(0, ch1.complete_count());
>     EXPECT_EQ(0, ch2.complete_count());
> 
>@@ -700,7 +696,7 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>     ASSERT_EQ_WAIT(1, ch2.complete_count(), kDefaultTimeout);
> 
>     // Initial connecting the channel, create connection on channel1.
>-    ch1.CreateConnection(GetCandidate(port2));
>+    ch1.CreateConnection(GetCandidate(ch2.port()));
>     ConnectStartedChannels(&ch1, &ch2);
> 
>     // Shorten the timeout period.
>@@ -753,43 +749,45 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>     EXPECT_TRUE_WAIT(ch2.conn() == NULL, kDefaultTimeout);
>   }
> 
>-  IceMessage* CreateStunMessage(int type) {
>-    IceMessage* msg = new IceMessage();
>+  std::unique_ptr<IceMessage> CreateStunMessage(int type) {
>+    auto msg = absl::make_unique<IceMessage>();
>     msg->SetType(type);
>     msg->SetTransactionID("TESTTESTTEST");
>     return msg;
>   }
>-  IceMessage* CreateStunMessageWithUsername(int type,
>-                                            const std::string& username) {
>-    IceMessage* msg = CreateStunMessage(type);
>+  std::unique_ptr<IceMessage> CreateStunMessageWithUsername(
>+      int type,
>+      const std::string& username) {
>+    std::unique_ptr<IceMessage> msg = CreateStunMessage(type);
>     msg->AddAttribute(absl::make_unique<StunByteStringAttribute>(
>         STUN_ATTR_USERNAME, username));
>     return msg;
>   }
>-  TestPort* CreateTestPort(const rtc::SocketAddress& addr,
>-                           const std::string& username,
>-                           const std::string& password) {
>-    TestPort* port = new TestPort(&main_, "test", &socket_factory_,
>-                                  MakeNetwork(addr), 0, 0, username, password);
>+  std::unique_ptr<TestPort> CreateTestPort(const rtc::SocketAddress& addr,
>+                                           const std::string& username,
>+                                           const std::string& password) {
>+    auto port = absl::make_unique<TestPort>(&main_, "test", &socket_factory_,
>+                                            MakeNetwork(addr), 0, 0, username,
>+                                            password);
>     port->SignalRoleConflict.connect(this, &PortTest::OnRoleConflict);
>     return port;
>   }
>-  TestPort* CreateTestPort(const rtc::SocketAddress& addr,
>-                           const std::string& username,
>-                           const std::string& password,
>-                           cricket::IceRole role,
>-                           int tiebreaker) {
>-    TestPort* port = CreateTestPort(addr, username, password);
>+  std::unique_ptr<TestPort> CreateTestPort(const rtc::SocketAddress& addr,
>+                                           const std::string& username,
>+                                           const std::string& password,
>+                                           cricket::IceRole role,
>+                                           int tiebreaker) {
>+    auto port = CreateTestPort(addr, username, password);
>     port->SetIceRole(role);
>     port->SetIceTiebreaker(tiebreaker);
>     return port;
>   }
>   // Overload to create a test port given an rtc::Network directly.
>-  TestPort* CreateTestPort(rtc::Network* network,
>-                           const std::string& username,
>-                           const std::string& password) {
>-    TestPort* port = new TestPort(&main_, "test", &socket_factory_, network, 0,
>-                                  0, username, password);
>+  std::unique_ptr<TestPort> CreateTestPort(rtc::Network* network,
>+                                           const std::string& username,
>+                                           const std::string& password) {
>+    auto port = absl::make_unique<TestPort>(&main_, "test", &socket_factory_,
>+                                            network, 0, 0, username, password);
>     port->SignalRoleConflict.connect(this, &PortTest::OnRoleConflict);
>     return port;
>   }
>@@ -834,9 +832,9 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
> };
> 
> void PortTest::TestConnectivity(const char* name1,
>-                                Port* port1,
>+                                std::unique_ptr<Port> port1,
>                                 const char* name2,
>-                                Port* port2,
>+                                std::unique_ptr<Port> port2,
>                                 bool accept,
>                                 bool same_addr1,
>                                 bool same_addr2,
>@@ -847,8 +845,8 @@ void PortTest::TestConnectivity(const char* name1,
>   port2->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
> 
>   // Set up channels and ensure both ports will be deleted.
>-  TestChannel ch1(port1);
>-  TestChannel ch2(port2);
>+  TestChannel ch1(std::move(port1));
>+  TestChannel ch2(std::move(port2));
>   EXPECT_EQ(0, ch1.complete_count());
>   EXPECT_EQ(0, ch2.complete_count());
> 
>@@ -859,7 +857,7 @@ void PortTest::TestConnectivity(const char* name1,
>   ASSERT_EQ_SIMULATED_WAIT(1, ch2.complete_count(), kDefaultTimeout, clock);
> 
>   // Send a ping from src to dst. This may or may not make it.
>-  ch1.CreateConnection(GetCandidate(port2));
>+  ch1.CreateConnection(GetCandidate(ch2.port()));
>   ASSERT_TRUE(ch1.conn() != NULL);
>   EXPECT_TRUE_SIMULATED_WAIT(ch1.conn()->connected(), kDefaultTimeout,
>                              clock);  // for TCP connect
>@@ -870,16 +868,16 @@ void PortTest::TestConnectivity(const char* name1,
>     // We are able to send a ping from src to dst. This is the case when
>     // sending to UDP ports and cone NATs.
>     EXPECT_TRUE(ch1.remote_address().IsNil());
>-    EXPECT_EQ(ch2.remote_fragment(), port1->username_fragment());
>+    EXPECT_EQ(ch2.remote_fragment(), ch1.port()->username_fragment());
> 
>     // Ensure the ping came from the same address used for src.
>     // This is the case unless the source NAT was symmetric.
>     if (same_addr1)
>-      EXPECT_EQ(ch2.remote_address(), GetAddress(port1));
>+      EXPECT_EQ(ch2.remote_address(), GetAddress(ch1.port()));
>     EXPECT_TRUE(same_addr2);
> 
>     // Send a ping from dst to src.
>-    ch2.AcceptConnection(GetCandidate(port1));
>+    ch2.AcceptConnection(GetCandidate(ch1.port()));
>     ASSERT_TRUE(ch2.conn() != NULL);
>     ch2.Ping();
>     EXPECT_EQ_SIMULATED_WAIT(Connection::STATE_WRITABLE,
>@@ -891,7 +889,7 @@ void PortTest::TestConnectivity(const char* name1,
>     EXPECT_TRUE(ch2.remote_address().IsNil());
> 
>     // Send a ping from dst to src. Again, this may or may not make it.
>-    ch2.CreateConnection(GetCandidate(port1));
>+    ch2.CreateConnection(GetCandidate(ch1.port()));
>     ASSERT_TRUE(ch2.conn() != NULL);
>     ch2.Ping();
>     SIMULATED_WAIT(ch2.conn()->write_state() == Connection::STATE_WRITABLE,
>@@ -925,7 +923,7 @@ void PortTest::TestConnectivity(const char* name1,
>       EXPECT_TRUE(ch1.remote_address().IsNil());
> 
>       // Pick up the actual address and establish the connection.
>-      ch2.AcceptConnection(GetCandidate(port1));
>+      ch2.AcceptConnection(GetCandidate(ch1.port()));
>       ASSERT_TRUE(ch2.conn() != NULL);
>       ch2.Ping();
>       EXPECT_EQ_SIMULATED_WAIT(Connection::STATE_WRITABLE,
>@@ -938,7 +936,7 @@ void PortTest::TestConnectivity(const char* name1,
>       EXPECT_FALSE(ch1.conn()->receiving());
> 
>       // Update our address and complete the connection.
>-      ch1.AcceptConnection(GetCandidate(port2));
>+      ch1.AcceptConnection(GetCandidate(ch2.port()));
>       ch1.Ping();
>       EXPECT_EQ_SIMULATED_WAIT(Connection::STATE_WRITABLE,
>                                ch1.conn()->write_state(), kDefaultTimeout,
>@@ -1261,12 +1259,12 @@ TEST_F(PortTest, TestTcpReconnectTimeout) {
> // Test when TcpConnection never connects, the OnClose() will be called to
> // destroy the connection.
> TEST_F(PortTest, TestTcpNeverConnect) {
>-  Port* port1 = CreateTcpPort(kLocalAddr1);
>+  auto port1 = CreateTcpPort(kLocalAddr1);
>   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   port1->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
> 
>   // Set up a channel and ensure the port will be deleted.
>-  TestChannel ch1(port1);
>+  TestChannel ch1(std::move(port1));
>   EXPECT_EQ(0, ch1.complete_count());
> 
>   ch1.Start();
>@@ -1277,7 +1275,7 @@ TEST_F(PortTest, TestTcpNeverConnect) {
>   // Bind but not listen.
>   EXPECT_EQ(0, server->Bind(kLocalAddr2));
> 
>-  Candidate c = GetCandidate(port1);
>+  Candidate c = GetCandidate(ch1.port());
>   c.set_address(server->GetLocalAddress());
> 
>   ch1.CreateConnection(c);
>@@ -1312,10 +1310,8 @@ TEST_F(PortTest, TestSslTcpToSslTcpRelay) {
> // ii) it has not received anything for DEAD_CONNECTION_RECEIVE_TIMEOUT
> //     milliseconds since last receiving.
> TEST_F(PortTest, TestConnectionDead) {
>-  UDPPort* port1 = CreateUdpPort(kLocalAddr1);
>-  UDPPort* port2 = CreateUdpPort(kLocalAddr2);
>-  TestChannel ch1(port1);
>-  TestChannel ch2(port2);
>+  TestChannel ch1(CreateUdpPort(kLocalAddr1));
>+  TestChannel ch2(CreateUdpPort(kLocalAddr2));
>   // Acquire address.
>   ch1.Start();
>   ch2.Start();
>@@ -1324,7 +1320,7 @@ TEST_F(PortTest, TestConnectionDead) {
> 
>   // Test case that the connection has never received anything.
>   int64_t before_created = rtc::TimeMillis();
>-  ch1.CreateConnection(GetCandidate(port2));
>+  ch1.CreateConnection(GetCandidate(ch2.port()));
>   int64_t after_created = rtc::TimeMillis();
>   Connection* conn = ch1.conn();
>   ASSERT_NE(conn, nullptr);
>@@ -1343,7 +1339,7 @@ TEST_F(PortTest, TestConnectionDead) {
> 
>   // Test case that the connection has received something.
>   // Create a connection again and receive a ping.
>-  ch1.CreateConnection(GetCandidate(port2));
>+  ch1.CreateConnection(GetCandidate(ch2.port()));
>   conn = ch1.conn();
>   ASSERT_NE(conn, nullptr);
>   int64_t before_last_receiving = rtc::TimeMillis();
>@@ -1362,14 +1358,15 @@ TEST_F(PortTest, TestConnectionDead) {
> // verifies Message Integrity attribute in STUN messages and username in STUN
> // binding request will have colon (":") between remote and local username.
> TEST_F(PortTest, TestLocalToLocalStandard) {
>-  UDPPort* port1 = CreateUdpPort(kLocalAddr1);
>+  auto port1 = CreateUdpPort(kLocalAddr1);
>   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   port1->SetIceTiebreaker(kTiebreaker1);
>-  UDPPort* port2 = CreateUdpPort(kLocalAddr2);
>+  auto port2 = CreateUdpPort(kLocalAddr2);
>   port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>   port2->SetIceTiebreaker(kTiebreaker2);
>   // Same parameters as TestLocalToLocal above.
>-  TestConnectivity("udp", port1, "udp", port2, true, true, true, true);
>+  TestConnectivity("udp", std::move(port1), "udp", std::move(port2), true, true,
>+                   true, true);
> }
> 
> // This test is trying to validate a successful and failure scenario in a
>@@ -1377,8 +1374,7 @@ TEST_F(PortTest, TestLocalToLocalStandard) {
> // should remain equal to the request generated by the port and role of port
> // must be in controlling.
> TEST_F(PortTest, TestLoopbackCall) {
>-  std::unique_ptr<TestPort> lport(
>-      CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
>+  auto lport = CreateTestPort(kLocalAddr1, "lfrag", "lpass");
>   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   lport->SetIceTiebreaker(kTiebreaker1);
>   lport->PrepareAddress();
>@@ -1391,7 +1387,7 @@ TEST_F(PortTest, TestLoopbackCall) {
>   IceMessage* msg = lport->last_stun_msg();
>   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
>   conn->OnReadPacket(lport->last_stun_buf()->data<char>(),
>-                     lport->last_stun_buf()->size(), rtc::PacketTime());
>+                     lport->last_stun_buf()->size(), /* packet_time_us */ -1);
>   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
>   msg = lport->last_stun_msg();
>   EXPECT_EQ(STUN_BINDING_RESPONSE, msg->type());
>@@ -1422,9 +1418,9 @@ TEST_F(PortTest, TestLoopbackCall) {
>   modified_req->AddFingerprint();
> 
>   lport->Reset();
>-  std::unique_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
>-  WriteStunMessage(modified_req.get(), buf.get());
>-  conn1->OnReadPacket(buf->Data(), buf->Length(), rtc::PacketTime());
>+  auto buf = absl::make_unique<ByteBufferWriter>();
>+  WriteStunMessage(*modified_req, buf.get());
>+  conn1->OnReadPacket(buf->Data(), buf->Length(), /* packet_time_us */ -1);
>   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
>   msg = lport->last_stun_msg();
>   EXPECT_EQ(STUN_BINDING_ERROR_RESPONSE, msg->type());
>@@ -1436,12 +1432,10 @@ TEST_F(PortTest, TestLoopbackCall) {
> // value of tiebreaker, when it receives ping request from |rport| it will
> // send role conflict signal.
> TEST_F(PortTest, TestIceRoleConflict) {
>-  std::unique_ptr<TestPort> lport(
>-      CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
>+  auto lport = CreateTestPort(kLocalAddr1, "lfrag", "lpass");
>   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   lport->SetIceTiebreaker(kTiebreaker1);
>-  std::unique_ptr<TestPort> rport(
>-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
>+  auto rport = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
>   rport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   rport->SetIceTiebreaker(kTiebreaker2);
> 
>@@ -1460,7 +1454,7 @@ TEST_F(PortTest, TestIceRoleConflict) {
>   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
>   // Send rport binding request to lport.
>   lconn->OnReadPacket(rport->last_stun_buf()->data<char>(),
>-                      rport->last_stun_buf()->size(), rtc::PacketTime());
>+                      rport->last_stun_buf()->size(), /* packet_time_us */ -1);
> 
>   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
>   EXPECT_EQ(STUN_BINDING_RESPONSE, lport->last_stun_msg()->type());
>@@ -1468,13 +1462,12 @@ TEST_F(PortTest, TestIceRoleConflict) {
> }
> 
> TEST_F(PortTest, TestTcpNoDelay) {
>-  TCPPort* port1 = CreateTcpPort(kLocalAddr1);
>+  auto port1 = CreateTcpPort(kLocalAddr1);
>   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   int option_value = -1;
>   int success = port1->GetOption(rtc::Socket::OPT_NODELAY, &option_value);
>   ASSERT_EQ(0, success);  // GetOption() should complete successfully w/ 0
>   ASSERT_EQ(1, option_value);
>-  delete port1;
> }
> 
> TEST_F(PortTest, TestDelayedBindingUdp) {
>@@ -1482,7 +1475,7 @@ TEST_F(PortTest, TestDelayedBindingUdp) {
>   FakePacketSocketFactory socket_factory;
> 
>   socket_factory.set_next_udp_socket(socket);
>-  std::unique_ptr<UDPPort> port(CreateUdpPort(kLocalAddr1, &socket_factory));
>+  auto port = CreateUdpPort(kLocalAddr1, &socket_factory);
> 
>   socket->set_state(AsyncPacketSocket::STATE_BINDING);
>   port->PrepareAddress();
>@@ -1498,7 +1491,7 @@ TEST_F(PortTest, TestDelayedBindingTcp) {
>   FakePacketSocketFactory socket_factory;
> 
>   socket_factory.set_next_server_tcp_socket(socket);
>-  std::unique_ptr<TCPPort> port(CreateTcpPort(kLocalAddr1, &socket_factory));
>+  auto port = CreateTcpPort(kLocalAddr1, &socket_factory);
> 
>   socket->set_state(AsyncPacketSocket::STATE_BINDING);
>   port->PrepareAddress();
>@@ -1519,10 +1512,10 @@ void PortTest::TestCrossFamilyPorts(int type) {
>     FakeAsyncPacketSocket* socket = new FakeAsyncPacketSocket();
>     if (type == SOCK_DGRAM) {
>       factory.set_next_udp_socket(socket);
>-      ports[i].reset(CreateUdpPort(addresses[i], &factory));
>+      ports[i] = CreateUdpPort(addresses[i], &factory);
>     } else if (type == SOCK_STREAM) {
>       factory.set_next_server_tcp_socket(socket);
>-      ports[i].reset(CreateTcpPort(addresses[i], &factory));
>+      ports[i] = CreateTcpPort(addresses[i], &factory);
>     }
>     socket->set_state(AsyncPacketSocket::STATE_BINDING);
>     socket->SignalAddressReady(socket, addresses[i]);
>@@ -1586,7 +1579,7 @@ TEST_F(PortTest, TestUdpV6CrossTypePorts) {
>   for (int i = 0; i < 4; i++) {
>     FakeAsyncPacketSocket* socket = new FakeAsyncPacketSocket();
>     factory.set_next_udp_socket(socket);
>-    ports[i].reset(CreateUdpPort(addresses[i], &factory));
>+    ports[i] = CreateUdpPort(addresses[i], &factory);
>     socket->set_state(AsyncPacketSocket::STATE_BINDING);
>     socket->SignalAddressReady(socket, addresses[i]);
>     ports[i]->PrepareAddress();
>@@ -1611,28 +1604,27 @@ TEST_F(PortTest, TestUdpV6CrossTypePorts) {
> // get through DefaultDscpValue.
> TEST_F(PortTest, TestDefaultDscpValue) {
>   int dscp;
>-  std::unique_ptr<UDPPort> udpport(CreateUdpPort(kLocalAddr1));
>+  auto udpport = CreateUdpPort(kLocalAddr1);
>   EXPECT_EQ(0, udpport->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_CS6));
>   EXPECT_EQ(0, udpport->GetOption(rtc::Socket::OPT_DSCP, &dscp));
>-  std::unique_ptr<TCPPort> tcpport(CreateTcpPort(kLocalAddr1));
>+  auto tcpport = CreateTcpPort(kLocalAddr1);
>   EXPECT_EQ(0, tcpport->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_AF31));
>   EXPECT_EQ(0, tcpport->GetOption(rtc::Socket::OPT_DSCP, &dscp));
>   EXPECT_EQ(rtc::DSCP_AF31, dscp);
>-  std::unique_ptr<StunPort> stunport(
>-      CreateStunPort(kLocalAddr1, nat_socket_factory1()));
>+  auto stunport = CreateStunPort(kLocalAddr1, nat_socket_factory1());
>   EXPECT_EQ(0, stunport->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_AF41));
>   EXPECT_EQ(0, stunport->GetOption(rtc::Socket::OPT_DSCP, &dscp));
>   EXPECT_EQ(rtc::DSCP_AF41, dscp);
>-  std::unique_ptr<TurnPort> turnport1(
>-      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
>+  auto turnport1 =
>+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP);
>   // Socket is created in PrepareAddress.
>   turnport1->PrepareAddress();
>   EXPECT_EQ(0, turnport1->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_CS7));
>   EXPECT_EQ(0, turnport1->GetOption(rtc::Socket::OPT_DSCP, &dscp));
>   EXPECT_EQ(rtc::DSCP_CS7, dscp);
>   // This will verify correct value returned without the socket.
>-  std::unique_ptr<TurnPort> turnport2(
>-      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
>+  auto turnport2 =
>+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP);
>   EXPECT_EQ(0, turnport2->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_CS6));
>   EXPECT_EQ(0, turnport2->GetOption(rtc::Socket::OPT_DSCP, &dscp));
>   EXPECT_EQ(rtc::DSCP_CS6, dscp);
>@@ -1640,10 +1632,8 @@ TEST_F(PortTest, TestDefaultDscpValue) {
> 
> // Test sending STUN messages.
> TEST_F(PortTest, TestSendStunMessage) {
>-  std::unique_ptr<TestPort> lport(
>-      CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
>-  std::unique_ptr<TestPort> rport(
>-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
>+  auto lport = CreateTestPort(kLocalAddr1, "lfrag", "lpass");
>+  auto rport = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
>   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   lport->SetIceTiebreaker(kTiebreaker1);
>   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
>@@ -1689,17 +1679,17 @@ TEST_F(PortTest, TestSendStunMessage) {
>   ASSERT_TRUE(msg->GetUInt32(STUN_ATTR_RETRANSMIT_COUNT) == NULL);
> 
>   // Save a copy of the BINDING-REQUEST for use below.
>-  std::unique_ptr<IceMessage> request(CopyStunMessage(msg));
>+  std::unique_ptr<IceMessage> request = CopyStunMessage(*msg);
> 
>   // Receive the BINDING-REQUEST and respond with BINDING-RESPONSE.
>   rconn->OnReadPacket(lport->last_stun_buf()->data<char>(),
>-                      lport->last_stun_buf()->size(), rtc::PacketTime());
>+                      lport->last_stun_buf()->size(), /* packet_time_us */ -1);
>   msg = rport->last_stun_msg();
>   ASSERT_TRUE(msg != NULL);
>   EXPECT_EQ(STUN_BINDING_RESPONSE, msg->type());
>   // Received a BINDING-RESPONSE.
>   lconn->OnReadPacket(rport->last_stun_buf()->data<char>(),
>-                      rport->last_stun_buf()->size(), rtc::PacketTime());
>+                      rport->last_stun_buf()->size(), /* packet_time_us */ -1);
>   // Verify the STUN Stats.
>   EXPECT_EQ(1U, lconn->stats().sent_ping_requests_total);
>   EXPECT_EQ(1U, lconn->stats().sent_ping_requests_before_first_response);
>@@ -1777,13 +1767,13 @@ TEST_F(PortTest, TestSendStunMessage) {
>   EXPECT_EQ(2U, retransmit_attr->value());
> 
>   // Respond with a BINDING-RESPONSE.
>-  request.reset(CopyStunMessage(msg));
>+  request = CopyStunMessage(*msg);
>   lconn->OnReadPacket(rport->last_stun_buf()->data<char>(),
>-                      rport->last_stun_buf()->size(), rtc::PacketTime());
>+                      rport->last_stun_buf()->size(), /* packet_time_us */ -1);
>   msg = lport->last_stun_msg();
>   // Receive the BINDING-RESPONSE.
>   rconn->OnReadPacket(lport->last_stun_buf()->data<char>(),
>-                      lport->last_stun_buf()->size(), rtc::PacketTime());
>+                      lport->last_stun_buf()->size(), /* packet_time_us */ -1);
> 
>   // Verify the Stun ping stats.
>   EXPECT_EQ(3U, rconn->stats().sent_ping_requests_total);
>@@ -1804,10 +1794,8 @@ TEST_F(PortTest, TestSendStunMessage) {
> }
> 
> TEST_F(PortTest, TestNomination) {
>-  std::unique_ptr<TestPort> lport(
>-      CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
>-  std::unique_ptr<TestPort> rport(
>-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
>+  auto lport = CreateTestPort(kLocalAddr1, "lfrag", "lpass");
>+  auto rport = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
>   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   lport->SetIceTiebreaker(kTiebreaker1);
>   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
>@@ -1837,7 +1825,7 @@ TEST_F(PortTest, TestNomination) {
>   ASSERT_TRUE_WAIT(lport->last_stun_msg(), kDefaultTimeout);
>   ASSERT_TRUE(lport->last_stun_buf());
>   rconn->OnReadPacket(lport->last_stun_buf()->data<char>(),
>-                      lport->last_stun_buf()->size(), rtc::PacketTime());
>+                      lport->last_stun_buf()->size(), /* packet_time_us */ -1);
>   EXPECT_EQ(nomination, rconn->remote_nomination());
>   EXPECT_FALSE(lconn->nominated());
>   EXPECT_TRUE(rconn->nominated());
>@@ -1849,7 +1837,7 @@ TEST_F(PortTest, TestNomination) {
>   ASSERT_TRUE_WAIT(rport->last_stun_msg(), kDefaultTimeout);
>   ASSERT_TRUE(rport->last_stun_buf());
>   lconn->OnReadPacket(rport->last_stun_buf()->data<char>(),
>-                      rport->last_stun_buf()->size(), rtc::PacketTime());
>+                      rport->last_stun_buf()->size(), /* packet_time_us */ -1);
>   EXPECT_EQ(nomination, lconn->acked_nomination());
>   EXPECT_TRUE(lconn->nominated());
>   EXPECT_TRUE(rconn->nominated());
>@@ -1860,10 +1848,8 @@ TEST_F(PortTest, TestNomination) {
> TEST_F(PortTest, TestRoundTripTime) {
>   rtc::ScopedFakeClock clock;
> 
>-  std::unique_ptr<TestPort> lport(
>-      CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
>-  std::unique_ptr<TestPort> rport(
>-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
>+  auto lport = CreateTestPort(kLocalAddr1, "lfrag", "lpass");
>+  auto rport = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
>   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   lport->SetIceTiebreaker(kTiebreaker1);
>   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
>@@ -1901,10 +1887,8 @@ TEST_F(PortTest, TestRoundTripTime) {
> }
> 
> TEST_F(PortTest, TestUseCandidateAttribute) {
>-  std::unique_ptr<TestPort> lport(
>-      CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
>-  std::unique_ptr<TestPort> rport(
>-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
>+  auto lport = CreateTestPort(kLocalAddr1, "lfrag", "lpass");
>+  auto rport = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
>   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   lport->SetIceTiebreaker(kTiebreaker1);
>   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
>@@ -1932,10 +1916,8 @@ TEST_F(PortTest, TestUseCandidateAttribute) {
> // the remote network costs are updated with the stun binding requests.
> TEST_F(PortTest, TestNetworkCostChange) {
>   rtc::Network* test_network = MakeNetwork(kLocalAddr1);
>-  std::unique_ptr<TestPort> lport(
>-      CreateTestPort(test_network, "lfrag", "lpass"));
>-  std::unique_ptr<TestPort> rport(
>-      CreateTestPort(test_network, "rfrag", "rpass"));
>+  auto lport = CreateTestPort(test_network, "lfrag", "lpass");
>+  auto rport = CreateTestPort(test_network, "rfrag", "rpass");
>   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   lport->SetIceTiebreaker(kTiebreaker1);
>   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
>@@ -1982,7 +1964,7 @@ TEST_F(PortTest, TestNetworkCostChange) {
>   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
>   // Pass the binding request to rport.
>   rconn->OnReadPacket(lport->last_stun_buf()->data<char>(),
>-                      lport->last_stun_buf()->size(), rtc::PacketTime());
>+                      lport->last_stun_buf()->size(), /* packet_time_us */ -1);
>   // Wait until rport sends the response and then check the remote network cost.
>   ASSERT_TRUE_WAIT(rport->last_stun_msg() != NULL, kDefaultTimeout);
>   EXPECT_EQ(rtc::kNetworkCostHigh, rconn->remote_candidate().network_cost());
>@@ -1990,10 +1972,8 @@ TEST_F(PortTest, TestNetworkCostChange) {
> 
> TEST_F(PortTest, TestNetworkInfoAttribute) {
>   rtc::Network* test_network = MakeNetwork(kLocalAddr1);
>-  std::unique_ptr<TestPort> lport(
>-      CreateTestPort(test_network, "lfrag", "lpass"));
>-  std::unique_ptr<TestPort> rport(
>-      CreateTestPort(test_network, "rfrag", "rpass"));
>+  auto lport = CreateTestPort(test_network, "lfrag", "lpass");
>+  auto rport = CreateTestPort(test_network, "rfrag", "rpass");
>   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   lport->SetIceTiebreaker(kTiebreaker1);
>   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
>@@ -2037,44 +2017,43 @@ TEST_F(PortTest, TestNetworkInfoAttribute) {
> // Test handling STUN messages.
> TEST_F(PortTest, TestHandleStunMessage) {
>   // Our port will act as the "remote" port.
>-  std::unique_ptr<TestPort> port(CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
>+  auto port = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
> 
>   std::unique_ptr<IceMessage> in_msg, out_msg;
>-  std::unique_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
>+  auto buf = absl::make_unique<ByteBufferWriter>();
>   rtc::SocketAddress addr(kLocalAddr1);
>   std::string username;
> 
>   // BINDING-REQUEST from local to remote with valid ICE username,
>   // MESSAGE-INTEGRITY, and FINGERPRINT.
>-  in_msg.reset(
>-      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag"));
>+  in_msg = CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag");
>   in_msg->AddMessageIntegrity("rpass");
>   in_msg->AddFingerprint();
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                    &username));
>   EXPECT_TRUE(out_msg.get() != NULL);
>   EXPECT_EQ("lfrag", username);
> 
>   // BINDING-RESPONSE without username, with MESSAGE-INTEGRITY and FINGERPRINT.
>-  in_msg.reset(CreateStunMessage(STUN_BINDING_RESPONSE));
>+  in_msg = CreateStunMessage(STUN_BINDING_RESPONSE);
>   in_msg->AddAttribute(absl::make_unique<StunXorAddressAttribute>(
>       STUN_ATTR_XOR_MAPPED_ADDRESS, kLocalAddr2));
>   in_msg->AddMessageIntegrity("rpass");
>   in_msg->AddFingerprint();
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                    &username));
>   EXPECT_TRUE(out_msg.get() != NULL);
>   EXPECT_EQ("", username);
> 
>   // BINDING-ERROR-RESPONSE without username, with error, M-I, and FINGERPRINT.
>-  in_msg.reset(CreateStunMessage(STUN_BINDING_ERROR_RESPONSE));
>+  in_msg = CreateStunMessage(STUN_BINDING_ERROR_RESPONSE);
>   in_msg->AddAttribute(absl::make_unique<StunErrorCodeAttribute>(
>       STUN_ATTR_ERROR_CODE, STUN_ERROR_SERVER_ERROR,
>       STUN_ERROR_REASON_SERVER_ERROR));
>   in_msg->AddFingerprint();
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                    &username));
>   EXPECT_TRUE(out_msg.get() != NULL);
>@@ -2087,18 +2066,18 @@ TEST_F(PortTest, TestHandleStunMessage) {
> 
> // Tests handling of ICE binding requests with missing or incorrect usernames.
> TEST_F(PortTest, TestHandleStunMessageBadUsername) {
>-  std::unique_ptr<TestPort> port(CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
>+  auto port = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
> 
>   std::unique_ptr<IceMessage> in_msg, out_msg;
>-  std::unique_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
>+  auto buf = absl::make_unique<ByteBufferWriter>();
>   rtc::SocketAddress addr(kLocalAddr1);
>   std::string username;
> 
>   // BINDING-REQUEST with no username.
>-  in_msg.reset(CreateStunMessage(STUN_BINDING_REQUEST));
>+  in_msg = CreateStunMessage(STUN_BINDING_REQUEST);
>   in_msg->AddMessageIntegrity("rpass");
>   in_msg->AddFingerprint();
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                    &username));
>   EXPECT_TRUE(out_msg.get() == NULL);
>@@ -2106,10 +2085,10 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) {
>   EXPECT_EQ(STUN_ERROR_BAD_REQUEST, port->last_stun_error_code());
> 
>   // BINDING-REQUEST with empty username.
>-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST, ""));
>+  in_msg = CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "");
>   in_msg->AddMessageIntegrity("rpass");
>   in_msg->AddFingerprint();
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                    &username));
>   EXPECT_TRUE(out_msg.get() == NULL);
>@@ -2117,10 +2096,10 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) {
>   EXPECT_EQ(STUN_ERROR_UNAUTHORIZED, port->last_stun_error_code());
> 
>   // BINDING-REQUEST with too-short username.
>-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfra"));
>+  in_msg = CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfra");
>   in_msg->AddMessageIntegrity("rpass");
>   in_msg->AddFingerprint();
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                    &username));
>   EXPECT_TRUE(out_msg.get() == NULL);
>@@ -2128,11 +2107,10 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) {
>   EXPECT_EQ(STUN_ERROR_UNAUTHORIZED, port->last_stun_error_code());
> 
>   // BINDING-REQUEST with reversed username.
>-  in_msg.reset(
>-      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "lfrag:rfrag"));
>+  in_msg = CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "lfrag:rfrag");
>   in_msg->AddMessageIntegrity("rpass");
>   in_msg->AddFingerprint();
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                    &username));
>   EXPECT_TRUE(out_msg.get() == NULL);
>@@ -2140,11 +2118,10 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) {
>   EXPECT_EQ(STUN_ERROR_UNAUTHORIZED, port->last_stun_error_code());
> 
>   // BINDING-REQUEST with garbage username.
>-  in_msg.reset(
>-      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "abcd:efgh"));
>+  in_msg = CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "abcd:efgh");
>   in_msg->AddMessageIntegrity("rpass");
>   in_msg->AddFingerprint();
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                    &username));
>   EXPECT_TRUE(out_msg.get() == NULL);
>@@ -2155,19 +2132,18 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) {
> // Test handling STUN messages with missing or malformed M-I.
> TEST_F(PortTest, TestHandleStunMessageBadMessageIntegrity) {
>   // Our port will act as the "remote" port.
>-  std::unique_ptr<TestPort> port(CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
>+  auto port = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
> 
>   std::unique_ptr<IceMessage> in_msg, out_msg;
>-  std::unique_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
>+  auto buf = absl::make_unique<ByteBufferWriter>();
>   rtc::SocketAddress addr(kLocalAddr1);
>   std::string username;
> 
>   // BINDING-REQUEST from local to remote with valid ICE username and
>   // FINGERPRINT, but no MESSAGE-INTEGRITY.
>-  in_msg.reset(
>-      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag"));
>+  in_msg = CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag");
>   in_msg->AddFingerprint();
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                    &username));
>   EXPECT_TRUE(out_msg.get() == NULL);
>@@ -2176,11 +2152,10 @@ TEST_F(PortTest, TestHandleStunMessageBadMessageIntegrity) {
> 
>   // BINDING-REQUEST from local to remote with valid ICE username and
>   // FINGERPRINT, but invalid MESSAGE-INTEGRITY.
>-  in_msg.reset(
>-      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag"));
>+  in_msg = CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag");
>   in_msg->AddMessageIntegrity("invalid");
>   in_msg->AddFingerprint();
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                    &username));
>   EXPECT_TRUE(out_msg.get() == NULL);
>@@ -2195,19 +2170,18 @@ TEST_F(PortTest, TestHandleStunMessageBadMessageIntegrity) {
> // Test handling STUN messages with missing or malformed FINGERPRINT.
> TEST_F(PortTest, TestHandleStunMessageBadFingerprint) {
>   // Our port will act as the "remote" port.
>-  std::unique_ptr<TestPort> port(CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
>+  auto port = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
> 
>   std::unique_ptr<IceMessage> in_msg, out_msg;
>-  std::unique_ptr<ByteBufferWriter> buf(new ByteBufferWriter());
>+  auto buf = absl::make_unique<ByteBufferWriter>();
>   rtc::SocketAddress addr(kLocalAddr1);
>   std::string username;
> 
>   // BINDING-REQUEST from local to remote with valid ICE username and
>   // MESSAGE-INTEGRITY, but no FINGERPRINT; GetStunMessage should fail.
>-  in_msg.reset(
>-      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag"));
>+  in_msg = CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag");
>   in_msg->AddMessageIntegrity("rpass");
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_FALSE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                     &username));
>   EXPECT_EQ(0, port->last_stun_error_code());
>@@ -2215,17 +2189,17 @@ TEST_F(PortTest, TestHandleStunMessageBadFingerprint) {
>   // Now, add a fingerprint, but munge the message so it's not valid.
>   in_msg->AddFingerprint();
>   in_msg->SetTransactionID("TESTTESTBADD");
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_FALSE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                     &username));
>   EXPECT_EQ(0, port->last_stun_error_code());
> 
>   // Valid BINDING-RESPONSE, except no FINGERPRINT.
>-  in_msg.reset(CreateStunMessage(STUN_BINDING_RESPONSE));
>+  in_msg = CreateStunMessage(STUN_BINDING_RESPONSE);
>   in_msg->AddAttribute(absl::make_unique<StunXorAddressAttribute>(
>       STUN_ATTR_XOR_MAPPED_ADDRESS, kLocalAddr2));
>   in_msg->AddMessageIntegrity("rpass");
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_FALSE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                     &username));
>   EXPECT_EQ(0, port->last_stun_error_code());
>@@ -2233,18 +2207,18 @@ TEST_F(PortTest, TestHandleStunMessageBadFingerprint) {
>   // Now, add a fingerprint, but munge the message so it's not valid.
>   in_msg->AddFingerprint();
>   in_msg->SetTransactionID("TESTTESTBADD");
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_FALSE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                     &username));
>   EXPECT_EQ(0, port->last_stun_error_code());
> 
>   // Valid BINDING-ERROR-RESPONSE, except no FINGERPRINT.
>-  in_msg.reset(CreateStunMessage(STUN_BINDING_ERROR_RESPONSE));
>+  in_msg = CreateStunMessage(STUN_BINDING_ERROR_RESPONSE);
>   in_msg->AddAttribute(absl::make_unique<StunErrorCodeAttribute>(
>       STUN_ATTR_ERROR_CODE, STUN_ERROR_SERVER_ERROR,
>       STUN_ERROR_REASON_SERVER_ERROR));
>   in_msg->AddMessageIntegrity("rpass");
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_FALSE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                     &username));
>   EXPECT_EQ(0, port->last_stun_error_code());
>@@ -2252,7 +2226,7 @@ TEST_F(PortTest, TestHandleStunMessageBadFingerprint) {
>   // Now, add a fingerprint, but munge the message so it's not valid.
>   in_msg->AddFingerprint();
>   in_msg->SetTransactionID("TESTTESTBADD");
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_FALSE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                     &username));
>   EXPECT_EQ(0, port->last_stun_error_code());
>@@ -2261,8 +2235,7 @@ TEST_F(PortTest, TestHandleStunMessageBadFingerprint) {
> // Test handling of STUN binding indication messages . STUN binding
> // indications are allowed only to the connection which is in read mode.
> TEST_F(PortTest, TestHandleStunBindingIndication) {
>-  std::unique_ptr<TestPort> lport(
>-      CreateTestPort(kLocalAddr2, "lfrag", "lpass"));
>+  auto lport = CreateTestPort(kLocalAddr2, "lfrag", "lpass");
>   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   lport->SetIceTiebreaker(kTiebreaker1);
> 
>@@ -2272,9 +2245,9 @@ TEST_F(PortTest, TestHandleStunBindingIndication) {
>   rtc::SocketAddress addr(kLocalAddr1);
>   std::string username;
> 
>-  in_msg.reset(CreateStunMessage(STUN_BINDING_INDICATION));
>+  in_msg = CreateStunMessage(STUN_BINDING_INDICATION);
>   in_msg->AddFingerprint();
>-  WriteStunMessage(in_msg.get(), buf.get());
>+  WriteStunMessage(*in_msg, buf.get());
>   EXPECT_TRUE(lport->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>                                     &username));
>   EXPECT_TRUE(out_msg.get() != NULL);
>@@ -2283,8 +2256,7 @@ TEST_F(PortTest, TestHandleStunBindingIndication) {
> 
>   // Verify connection can handle STUN indication and updates
>   // last_ping_received.
>-  std::unique_ptr<TestPort> rport(
>-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
>+  auto rport = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
>   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
>   rport->SetIceTiebreaker(kTiebreaker2);
> 
>@@ -2304,7 +2276,7 @@ TEST_F(PortTest, TestHandleStunBindingIndication) {
>   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
>   // Send rport binding request to lport.
>   lconn->OnReadPacket(rport->last_stun_buf()->data<char>(),
>-                      rport->last_stun_buf()->size(), rtc::PacketTime());
>+                      rport->last_stun_buf()->size(), /* packet_time_us */ -1);
>   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
>   EXPECT_EQ(STUN_BINDING_RESPONSE, lport->last_stun_msg()->type());
>   int64_t last_ping_received1 = lconn->last_ping_received();
>@@ -2312,13 +2284,13 @@ TEST_F(PortTest, TestHandleStunBindingIndication) {
>   // Adding a delay of 100ms.
>   rtc::Thread::Current()->ProcessMessages(100);
>   // Pinging lconn using stun indication message.
>-  lconn->OnReadPacket(buf->Data(), buf->Length(), rtc::PacketTime());
>+  lconn->OnReadPacket(buf->Data(), buf->Length(), /* packet_time_us */ -1);
>   int64_t last_ping_received2 = lconn->last_ping_received();
>   EXPECT_GT(last_ping_received2, last_ping_received1);
> }
> 
> TEST_F(PortTest, TestComputeCandidatePriority) {
>-  std::unique_ptr<TestPort> port(CreateTestPort(kLocalAddr1, "name", "pass"));
>+  auto port = CreateTestPort(kLocalAddr1, "name", "pass");
>   port->set_type_preference(90);
>   port->set_component(177);
>   port->AddCandidateAddress(SocketAddress("192.168.1.4", 1234));
>@@ -2355,8 +2327,7 @@ TEST_F(PortTest, TestComputeCandidatePriority) {
> // In the case of shared socket, one port may be shared by local and stun.
> // Test that candidates with different types will have different foundation.
> TEST_F(PortTest, TestFoundation) {
>-  std::unique_ptr<TestPort> testport(
>-      CreateTestPort(kLocalAddr1, "name", "pass"));
>+  auto testport = CreateTestPort(kLocalAddr1, "name", "pass");
>   testport->AddCandidateAddress(kLocalAddr1, kLocalAddr1, LOCAL_PORT_TYPE,
>                                 cricket::ICE_TYPE_PREFERENCE_HOST, false);
>   testport->AddCandidateAddress(kLocalAddr2, kLocalAddr1, STUN_PORT_TYPE,
>@@ -2369,20 +2340,19 @@ TEST_F(PortTest, TestFoundation) {
> TEST_F(PortTest, TestCandidateFoundation) {
>   std::unique_ptr<rtc::NATServer> nat_server(
>       CreateNatServer(kNatAddr1, NAT_OPEN_CONE));
>-  std::unique_ptr<UDPPort> udpport1(CreateUdpPort(kLocalAddr1));
>+  auto udpport1 = CreateUdpPort(kLocalAddr1);
>   udpport1->PrepareAddress();
>-  std::unique_ptr<UDPPort> udpport2(CreateUdpPort(kLocalAddr1));
>+  auto udpport2 = CreateUdpPort(kLocalAddr1);
>   udpport2->PrepareAddress();
>   EXPECT_EQ(udpport1->Candidates()[0].foundation(),
>             udpport2->Candidates()[0].foundation());
>-  std::unique_ptr<TCPPort> tcpport1(CreateTcpPort(kLocalAddr1));
>+  auto tcpport1 = CreateTcpPort(kLocalAddr1);
>   tcpport1->PrepareAddress();
>-  std::unique_ptr<TCPPort> tcpport2(CreateTcpPort(kLocalAddr1));
>+  auto tcpport2 = CreateTcpPort(kLocalAddr1);
>   tcpport2->PrepareAddress();
>   EXPECT_EQ(tcpport1->Candidates()[0].foundation(),
>             tcpport2->Candidates()[0].foundation());
>-  std::unique_ptr<Port> stunport(
>-      CreateStunPort(kLocalAddr1, nat_socket_factory1()));
>+  auto stunport = CreateStunPort(kLocalAddr1, nat_socket_factory1());
>   stunport->PrepareAddress();
>   ASSERT_EQ_WAIT(1U, stunport->Candidates().size(), kDefaultTimeout);
>   EXPECT_NE(tcpport1->Candidates()[0].foundation(),
>@@ -2394,7 +2364,7 @@ TEST_F(PortTest, TestCandidateFoundation) {
>   EXPECT_NE(udpport2->Candidates()[0].foundation(),
>             stunport->Candidates()[0].foundation());
>   // Verify GTURN candidate foundation.
>-  std::unique_ptr<RelayPort> relayport(CreateGturnPort(kLocalAddr1));
>+  auto relayport = CreateGturnPort(kLocalAddr1);
>   relayport->AddServerAddress(
>       cricket::ProtocolAddress(kRelayUdpIntAddr, cricket::PROTO_UDP));
>   relayport->PrepareAddress();
>@@ -2404,8 +2374,8 @@ TEST_F(PortTest, TestCandidateFoundation) {
>   EXPECT_NE(udpport2->Candidates()[0].foundation(),
>             relayport->Candidates()[0].foundation());
>   // Verifying TURN candidate foundation.
>-  std::unique_ptr<Port> turnport1(
>-      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
>+  auto turnport1 =
>+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP);
>   turnport1->PrepareAddress();
>   ASSERT_EQ_WAIT(1U, turnport1->Candidates().size(), kDefaultTimeout);
>   EXPECT_NE(udpport1->Candidates()[0].foundation(),
>@@ -2414,8 +2384,8 @@ TEST_F(PortTest, TestCandidateFoundation) {
>             turnport1->Candidates()[0].foundation());
>   EXPECT_NE(stunport->Candidates()[0].foundation(),
>             turnport1->Candidates()[0].foundation());
>-  std::unique_ptr<Port> turnport2(
>-      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
>+  auto turnport2 =
>+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP);
>   turnport2->PrepareAddress();
>   ASSERT_EQ_WAIT(1U, turnport2->Candidates().size(), kDefaultTimeout);
>   EXPECT_EQ(turnport1->Candidates()[0].foundation(),
>@@ -2426,9 +2396,8 @@ TEST_F(PortTest, TestCandidateFoundation) {
>   SocketAddress kTurnUdpExtAddr2("99.99.98.5", 0);
>   TestTurnServer turn_server2(rtc::Thread::Current(), kTurnUdpIntAddr2,
>                               kTurnUdpExtAddr2);
>-  std::unique_ptr<Port> turnport3(
>-      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP,
>-                     kTurnUdpIntAddr2));
>+  auto turnport3 = CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP,
>+                                  PROTO_UDP, kTurnUdpIntAddr2);
>   turnport3->PrepareAddress();
>   ASSERT_EQ_WAIT(1U, turnport3->Candidates().size(), kDefaultTimeout);
>   EXPECT_NE(turnport3->Candidates()[0].foundation(),
>@@ -2438,8 +2407,8 @@ TEST_F(PortTest, TestCandidateFoundation) {
>   // different foundations if their relay protocols are different.
>   TestTurnServer turn_server3(rtc::Thread::Current(), kTurnTcpIntAddr,
>                               kTurnUdpExtAddr, PROTO_TCP);
>-  std::unique_ptr<Port> turnport4(
>-      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_TCP, PROTO_UDP));
>+  auto turnport4 =
>+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_TCP, PROTO_UDP);
>   turnport4->PrepareAddress();
>   ASSERT_EQ_WAIT(1U, turnport4->Candidates().size(), kDefaultTimeout);
>   EXPECT_NE(turnport2->Candidates()[0].foundation(),
>@@ -2449,17 +2418,15 @@ TEST_F(PortTest, TestCandidateFoundation) {
> // This test verifies the related addresses of different types of
> // ICE candiates.
> TEST_F(PortTest, TestCandidateRelatedAddress) {
>-  std::unique_ptr<rtc::NATServer> nat_server(
>-      CreateNatServer(kNatAddr1, NAT_OPEN_CONE));
>-  std::unique_ptr<UDPPort> udpport(CreateUdpPort(kLocalAddr1));
>+  auto nat_server = CreateNatServer(kNatAddr1, NAT_OPEN_CONE);
>+  auto udpport = CreateUdpPort(kLocalAddr1);
>   udpport->PrepareAddress();
>   // For UDPPort, related address will be empty.
>   EXPECT_TRUE(udpport->Candidates()[0].related_address().IsNil());
>   // Testing related address for stun candidates.
>   // For stun candidate related address must be equal to the base
>   // socket address.
>-  std::unique_ptr<StunPort> stunport(
>-      CreateStunPort(kLocalAddr1, nat_socket_factory1()));
>+  auto stunport = CreateStunPort(kLocalAddr1, nat_socket_factory1());
>   stunport->PrepareAddress();
>   ASSERT_EQ_WAIT(1U, stunport->Candidates().size(), kDefaultTimeout);
>   // Check STUN candidate address.
>@@ -2470,7 +2437,7 @@ TEST_F(PortTest, TestCandidateRelatedAddress) {
>   // Verifying the related address for the GTURN candidates.
>   // NOTE: In case of GTURN related address will be equal to the mapped
>   // address, but address(mapped) will not be XOR.
>-  std::unique_ptr<RelayPort> relayport(CreateGturnPort(kLocalAddr1));
>+  auto relayport = CreateGturnPort(kLocalAddr1);
>   relayport->AddServerAddress(
>       cricket::ProtocolAddress(kRelayUdpIntAddr, cricket::PROTO_UDP));
>   relayport->PrepareAddress();
>@@ -2479,8 +2446,8 @@ TEST_F(PortTest, TestCandidateRelatedAddress) {
>   EXPECT_EQ(rtc::SocketAddress(), relayport->Candidates()[0].related_address());
>   // Verifying the related address for TURN candidate.
>   // For TURN related address must be equal to the mapped address.
>-  std::unique_ptr<Port> turnport(
>-      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
>+  auto turnport =
>+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP);
>   turnport->PrepareAddress();
>   ASSERT_EQ_WAIT(1U, turnport->Candidates().size(), kDefaultTimeout);
>   EXPECT_EQ(kTurnUdpExtAddr.ipaddr(),
>@@ -2500,11 +2467,9 @@ TEST_F(PortTest, TestCandidatePriority) {
> 
> // Test the Connection priority is calculated correctly.
> TEST_F(PortTest, TestConnectionPriority) {
>-  std::unique_ptr<TestPort> lport(
>-      CreateTestPort(kLocalAddr1, "lfrag", "lpass"));
>+  auto lport = CreateTestPort(kLocalAddr1, "lfrag", "lpass");
>   lport->set_type_preference(cricket::ICE_TYPE_PREFERENCE_HOST);
>-  std::unique_ptr<TestPort> rport(
>-      CreateTestPort(kLocalAddr2, "rfrag", "rpass"));
>+  auto rport = CreateTestPort(kLocalAddr2, "rfrag", "rpass");
>   rport->set_type_preference(cricket::ICE_TYPE_PREFERENCE_RELAY_UDP);
>   lport->set_component(123);
>   lport->AddCandidateAddress(SocketAddress("192.168.1.4", 1234));
>@@ -2545,14 +2510,14 @@ TEST_F(PortTest, TestConnectionPriority) {
> // estimate given by |MINIMUM_RTT| = 100.
> TEST_F(PortTest, TestWritableState) {
>   rtc::ScopedFakeClock clock;
>-  UDPPort* port1 = CreateUdpPort(kLocalAddr1);
>+  auto port1 = CreateUdpPort(kLocalAddr1);
>   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-  UDPPort* port2 = CreateUdpPort(kLocalAddr2);
>+  auto port2 = CreateUdpPort(kLocalAddr2);
>   port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
> 
>   // Set up channels.
>-  TestChannel ch1(port1);
>-  TestChannel ch2(port2);
>+  TestChannel ch1(std::move(port1));
>+  TestChannel ch2(std::move(port2));
> 
>   // Acquire addresses.
>   ch1.Start();
>@@ -2561,7 +2526,7 @@ TEST_F(PortTest, TestWritableState) {
>   ASSERT_EQ_SIMULATED_WAIT(1, ch2.complete_count(), kDefaultTimeout, clock);
> 
>   // Send a ping from src to dst.
>-  ch1.CreateConnection(GetCandidate(port2));
>+  ch1.CreateConnection(GetCandidate(ch2.port()));
>   ASSERT_TRUE(ch1.conn() != NULL);
>   EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state());
>   // for TCP connect
>@@ -2577,7 +2542,7 @@ TEST_F(PortTest, TestWritableState) {
> 
>   // Accept the connection to return the binding response, transition to
>   // writable, and allow data to be sent.
>-  ch2.AcceptConnection(GetCandidate(port1));
>+  ch2.AcceptConnection(GetCandidate(ch1.port()));
>   EXPECT_EQ_SIMULATED_WAIT(Connection::STATE_WRITABLE,
>                            ch1.conn()->write_state(), kDefaultTimeout, clock);
>   EXPECT_EQ(data_size, ch1.conn()->Send(data, data_size, options));
>@@ -2622,14 +2587,14 @@ TEST_F(PortTest, TestWritableState) {
> // |CONNECTION_WRITE_CONNECT_FAILURES|.
> TEST_F(PortTest, TestWritableStateWithConfiguredThreshold) {
>   rtc::ScopedFakeClock clock;
>-  UDPPort* port1 = CreateUdpPort(kLocalAddr1);
>+  auto port1 = CreateUdpPort(kLocalAddr1);
>   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-  UDPPort* port2 = CreateUdpPort(kLocalAddr2);
>+  auto port2 = CreateUdpPort(kLocalAddr2);
>   port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
> 
>   // Set up channels.
>-  TestChannel ch1(port1);
>-  TestChannel ch2(port2);
>+  TestChannel ch1(std::move(port1));
>+  TestChannel ch2(std::move(port2));
> 
>   // Acquire addresses.
>   ch1.Start();
>@@ -2638,14 +2603,14 @@ TEST_F(PortTest, TestWritableStateWithConfiguredThreshold) {
>   ASSERT_EQ_SIMULATED_WAIT(1, ch2.complete_count(), kDefaultTimeout, clock);
> 
>   // Send a ping from src to dst.
>-  ch1.CreateConnection(GetCandidate(port2));
>+  ch1.CreateConnection(GetCandidate(ch2.port()));
>   ASSERT_TRUE(ch1.conn() != NULL);
>   ch1.Ping();
>   SIMULATED_WAIT(!ch2.remote_address().IsNil(), kShortTimeout, clock);
> 
>   // Accept the connection to return the binding response, transition to
>   // writable, and allow data to be sent.
>-  ch2.AcceptConnection(GetCandidate(port1));
>+  ch2.AcceptConnection(GetCandidate(ch1.port()));
>   EXPECT_EQ_SIMULATED_WAIT(Connection::STATE_WRITABLE,
>                            ch1.conn()->write_state(), kDefaultTimeout, clock);
> 
>@@ -2676,20 +2641,20 @@ TEST_F(PortTest, TestWritableStateWithConfiguredThreshold) {
> }
> 
> TEST_F(PortTest, TestTimeoutForNeverWritable) {
>-  UDPPort* port1 = CreateUdpPort(kLocalAddr1);
>+  auto port1 = CreateUdpPort(kLocalAddr1);
>   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-  UDPPort* port2 = CreateUdpPort(kLocalAddr2);
>+  auto port2 = CreateUdpPort(kLocalAddr2);
>   port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
> 
>   // Set up channels.
>-  TestChannel ch1(port1);
>-  TestChannel ch2(port2);
>+  TestChannel ch1(std::move(port1));
>+  TestChannel ch2(std::move(port2));
> 
>   // Acquire addresses.
>   ch1.Start();
>   ch2.Start();
> 
>-  ch1.CreateConnection(GetCandidate(port2));
>+  ch1.CreateConnection(GetCandidate(ch2.port()));
>   ASSERT_TRUE(ch1.conn() != NULL);
>   EXPECT_EQ(Connection::STATE_WRITE_INIT, ch1.conn()->write_state());
> 
>@@ -2706,15 +2671,15 @@ TEST_F(PortTest, TestTimeoutForNeverWritable) {
> // In this test |ch1| behaves like FULL mode client and we have created
> // port which responds to the ping message just like LITE client.
> TEST_F(PortTest, TestIceLiteConnectivity) {
>-  TestPort* ice_full_port =
>+  auto ice_full_port =
>       CreateTestPort(kLocalAddr1, "lfrag", "lpass",
>                      cricket::ICEROLE_CONTROLLING, kTiebreaker1);
>+  auto* ice_full_port_ptr = ice_full_port.get();
> 
>-  std::unique_ptr<TestPort> ice_lite_port(
>-      CreateTestPort(kLocalAddr2, "rfrag", "rpass", cricket::ICEROLE_CONTROLLED,
>-                     kTiebreaker2));
>+  auto ice_lite_port = CreateTestPort(
>+      kLocalAddr2, "rfrag", "rpass", cricket::ICEROLE_CONTROLLED, kTiebreaker2);
>   // Setup TestChannel. This behaves like FULL mode client.
>-  TestChannel ch1(ice_full_port);
>+  TestChannel ch1(std::move(ice_full_port));
>   ch1.SetIceMode(ICEMODE_FULL);
> 
>   // Start gathering candidates.
>@@ -2734,24 +2699,24 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
> 
>   // Verify stun ping is without USE_CANDIDATE_ATTR. Getting message directly
>   // from port.
>-  ASSERT_TRUE_WAIT(ice_full_port->last_stun_msg() != NULL, kDefaultTimeout);
>-  IceMessage* msg = ice_full_port->last_stun_msg();
>+  ASSERT_TRUE_WAIT(ice_full_port_ptr->last_stun_msg() != NULL, kDefaultTimeout);
>+  IceMessage* msg = ice_full_port_ptr->last_stun_msg();
>   EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) == NULL);
> 
>   // Respond with a BINDING-RESPONSE from litemode client.
>   // NOTE: Ideally we should't create connection at this stage from lite
>   // port, as it should be done only after receiving ping with USE_CANDIDATE.
>   // But we need a connection to send a response message.
>-  ice_lite_port->CreateConnection(ice_full_port->Candidates()[0],
>+  ice_lite_port->CreateConnection(ice_full_port_ptr->Candidates()[0],
>                                   cricket::Port::ORIGIN_MESSAGE);
>-  std::unique_ptr<IceMessage> request(CopyStunMessage(msg));
>-  ice_lite_port->SendBindingResponse(request.get(),
>-                                     ice_full_port->Candidates()[0].address());
>+  std::unique_ptr<IceMessage> request = CopyStunMessage(*msg);
>+  ice_lite_port->SendBindingResponse(
>+      request.get(), ice_full_port_ptr->Candidates()[0].address());
> 
>   // Feeding the respone message from litemode to the full mode connection.
>   ch1.conn()->OnReadPacket(ice_lite_port->last_stun_buf()->data<char>(),
>                            ice_lite_port->last_stun_buf()->size(),
>-                           rtc::PacketTime());
>+                           /* packet_time_us */ -1);
>   // Verifying full mode connection becomes writable from the response.
>   EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, ch1.conn()->write_state(),
>                  kDefaultTimeout);
>@@ -2759,11 +2724,11 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
> 
>   // Clear existing stun messsages. Otherwise we will process old stun
>   // message right after we send ping.
>-  ice_full_port->Reset();
>+  ice_full_port_ptr->Reset();
>   // Send ping. This must have USE_CANDIDATE_ATTR.
>   ch1.Ping();
>-  ASSERT_TRUE_WAIT(ice_full_port->last_stun_msg() != NULL, kDefaultTimeout);
>-  msg = ice_full_port->last_stun_msg();
>+  ASSERT_TRUE_WAIT(ice_full_port_ptr->last_stun_msg() != NULL, kDefaultTimeout);
>+  msg = ice_full_port_ptr->last_stun_msg();
>   EXPECT_TRUE(msg->GetByteString(STUN_ATTR_USE_CANDIDATE) != NULL);
>   ch1.Stop();
> }
>@@ -2774,21 +2739,21 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
> TEST_F(PortTest, TestPortTimeoutIfNotKeptAlive) {
>   rtc::ScopedFakeClock clock;
>   int timeout_delay = 100;
>-  UDPPort* port1 = CreateUdpPort(kLocalAddr1);
>-  ConnectToSignalDestroyed(port1);
>+  auto port1 = CreateUdpPort(kLocalAddr1);
>+  ConnectToSignalDestroyed(port1.get());
>   port1->set_timeout_delay(timeout_delay);  // milliseconds
>   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   port1->SetIceTiebreaker(kTiebreaker1);
> 
>-  UDPPort* port2 = CreateUdpPort(kLocalAddr2);
>-  ConnectToSignalDestroyed(port2);
>+  auto port2 = CreateUdpPort(kLocalAddr2);
>+  ConnectToSignalDestroyed(port2.get());
>   port2->set_timeout_delay(timeout_delay);  // milliseconds
>   port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>   port2->SetIceTiebreaker(kTiebreaker2);
> 
>   // Set up channels and ensure both ports will be deleted.
>-  TestChannel ch1(port1);
>-  TestChannel ch2(port2);
>+  TestChannel ch1(std::move(port1));
>+  TestChannel ch2(std::move(port2));
> 
>   // Simulate a connection that succeeds, and then is destroyed.
>   StartConnectAndStopChannels(&ch1, &ch2);
>@@ -2803,22 +2768,22 @@ TEST_F(PortTest, TestPortTimeoutIfNotKeptAlive) {
> TEST_F(PortTest, TestPortTimeoutAfterNewConnectionCreatedAndDestroyed) {
>   rtc::ScopedFakeClock clock;
>   int timeout_delay = 100;
>-  UDPPort* port1 = CreateUdpPort(kLocalAddr1);
>-  ConnectToSignalDestroyed(port1);
>+  auto port1 = CreateUdpPort(kLocalAddr1);
>+  ConnectToSignalDestroyed(port1.get());
>   port1->set_timeout_delay(timeout_delay);  // milliseconds
>   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   port1->SetIceTiebreaker(kTiebreaker1);
> 
>-  UDPPort* port2 = CreateUdpPort(kLocalAddr2);
>-  ConnectToSignalDestroyed(port2);
>+  auto port2 = CreateUdpPort(kLocalAddr2);
>+  ConnectToSignalDestroyed(port2.get());
>   port2->set_timeout_delay(timeout_delay);  // milliseconds
> 
>   port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>   port2->SetIceTiebreaker(kTiebreaker2);
> 
>   // Set up channels and ensure both ports will be deleted.
>-  TestChannel ch1(port1);
>-  TestChannel ch2(port2);
>+  TestChannel ch1(std::move(port1));
>+  TestChannel ch2(std::move(port2));
> 
>   // Simulate a connection that succeeds, and then is destroyed.
>   StartConnectAndStopChannels(&ch1, &ch2);
>@@ -2844,14 +2809,14 @@ TEST_F(PortTest, TestPortTimeoutAfterNewConnectionCreatedAndDestroyed) {
> TEST_F(PortTest, TestPortNotTimeoutUntilPruned) {
>   rtc::ScopedFakeClock clock;
>   int timeout_delay = 100;
>-  UDPPort* port1 = CreateUdpPort(kLocalAddr1);
>-  ConnectToSignalDestroyed(port1);
>+  auto port1 = CreateUdpPort(kLocalAddr1);
>+  ConnectToSignalDestroyed(port1.get());
>   port1->set_timeout_delay(timeout_delay);  // milliseconds
>   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   port1->SetIceTiebreaker(kTiebreaker1);
> 
>-  UDPPort* port2 = CreateUdpPort(kLocalAddr2);
>-  ConnectToSignalDestroyed(port2);
>+  auto port2 = CreateUdpPort(kLocalAddr2);
>+  ConnectToSignalDestroyed(port2.get());
>   port2->set_timeout_delay(timeout_delay);  // milliseconds
>   port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>   port2->SetIceTiebreaker(kTiebreaker2);
>@@ -2862,40 +2827,39 @@ TEST_F(PortTest, TestPortNotTimeoutUntilPruned) {
>   port2->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
> 
>   // Set up channels and keep the port alive.
>-  TestChannel ch1(port1);
>-  TestChannel ch2(port2);
>+  TestChannel ch1(std::move(port1));
>+  TestChannel ch2(std::move(port2));
>   // Simulate a connection that succeeds, and then is destroyed. But ports
>   // are kept alive. Ports won't be destroyed.
>   StartConnectAndStopChannels(&ch1, &ch2);
>-  port1->KeepAliveUntilPruned();
>-  port2->KeepAliveUntilPruned();
>+  ch1.port()->KeepAliveUntilPruned();
>+  ch2.port()->KeepAliveUntilPruned();
>   SIMULATED_WAIT(ports_destroyed() > 0, 150, clock);
>   EXPECT_EQ(0, ports_destroyed());
> 
>   // If they are pruned now, they will be destroyed right away.
>-  port1->Prune();
>-  port2->Prune();
>+  ch1.port()->Prune();
>+  ch2.port()->Prune();
>   // The ports on both sides should be destroyed after timeout.
>   EXPECT_TRUE_SIMULATED_WAIT(ports_destroyed() == 2, 1, clock);
> }
> 
> TEST_F(PortTest, TestSupportsProtocol) {
>-  std::unique_ptr<Port> udp_port(CreateUdpPort(kLocalAddr1));
>+  auto udp_port = CreateUdpPort(kLocalAddr1);
>   EXPECT_TRUE(udp_port->SupportsProtocol(UDP_PROTOCOL_NAME));
>   EXPECT_FALSE(udp_port->SupportsProtocol(TCP_PROTOCOL_NAME));
> 
>-  std::unique_ptr<Port> stun_port(
>-      CreateStunPort(kLocalAddr1, nat_socket_factory1()));
>+  auto stun_port = CreateStunPort(kLocalAddr1, nat_socket_factory1());
>   EXPECT_TRUE(stun_port->SupportsProtocol(UDP_PROTOCOL_NAME));
>   EXPECT_FALSE(stun_port->SupportsProtocol(TCP_PROTOCOL_NAME));
> 
>-  std::unique_ptr<Port> tcp_port(CreateTcpPort(kLocalAddr1));
>+  auto tcp_port = CreateTcpPort(kLocalAddr1);
>   EXPECT_TRUE(tcp_port->SupportsProtocol(TCP_PROTOCOL_NAME));
>   EXPECT_TRUE(tcp_port->SupportsProtocol(SSLTCP_PROTOCOL_NAME));
>   EXPECT_FALSE(tcp_port->SupportsProtocol(UDP_PROTOCOL_NAME));
> 
>-  std::unique_ptr<Port> turn_port(
>-      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
>+  auto turn_port =
>+      CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP);
>   EXPECT_TRUE(turn_port->SupportsProtocol(UDP_PROTOCOL_NAME));
>   EXPECT_FALSE(turn_port->SupportsProtocol(TCP_PROTOCOL_NAME));
> }
>@@ -2903,8 +2867,7 @@ TEST_F(PortTest, TestSupportsProtocol) {
> // Test that SetIceParameters updates the component, ufrag and password
> // on both the port itself and its candidates.
> TEST_F(PortTest, TestSetIceParameters) {
>-  std::unique_ptr<TestPort> port(
>-      CreateTestPort(kLocalAddr1, "ufrag1", "password1"));
>+  auto port = CreateTestPort(kLocalAddr1, "ufrag1", "password1");
>   port->PrepareAddress();
>   EXPECT_EQ(1UL, port->Candidates().size());
>   port->SetIceParameters(1, "ufrag2", "password2");
>@@ -2918,8 +2881,7 @@ TEST_F(PortTest, TestSetIceParameters) {
> }
> 
> TEST_F(PortTest, TestAddConnectionWithSameAddress) {
>-  std::unique_ptr<TestPort> port(
>-      CreateTestPort(kLocalAddr1, "ufrag1", "password1"));
>+  auto port = CreateTestPort(kLocalAddr1, "ufrag1", "password1");
>   port->PrepareAddress();
>   EXPECT_EQ(1u, port->Candidates().size());
>   rtc::SocketAddress address("1.1.1.1", 5000);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.cc
>index 5470b5b82e655e3643bee0ed9f42eb9eb706578a..d3b3a56edd926c7c0a0314bc53c86a472974c22b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.cc
>@@ -10,8 +10,10 @@
> 
> #include "p2p/base/portallocator.h"
> 
>+#include <iterator>
> #include <utility>
> 
>+#include "p2p/base/icecredentialsiterator.h"
> #include "rtc_base/checks.h"
> 
> namespace cricket {
>@@ -121,6 +123,10 @@ PortAllocator::~PortAllocator() {
>   CheckRunOnValidThreadIfInitialized();
> }
> 
>+void PortAllocator::set_restrict_ice_credentials_change(bool value) {
>+  restrict_ice_credentials_change_ = value;
>+}
>+
> bool PortAllocator::SetConfiguration(
>     const ServerAddresses& stun_servers,
>     const std::vector<RelayServerConfig>& turn_servers,
>@@ -166,8 +172,8 @@ bool PortAllocator::SetConfiguration(
>   // If |candidate_pool_size_| is less than the number of pooled sessions, get
>   // rid of the extras.
>   while (candidate_pool_size_ < static_cast<int>(pooled_sessions_.size())) {
>-    pooled_sessions_.front().reset(nullptr);
>-    pooled_sessions_.pop_front();
>+    pooled_sessions_.back().reset(nullptr);
>+    pooled_sessions_.pop_back();
>   }
> 
>   // |stun_candidate_keepalive_interval_| will be used in STUN port allocation
>@@ -183,7 +189,11 @@ bool PortAllocator::SetConfiguration(
>   // If |candidate_pool_size_| is greater than the number of pooled sessions,
>   // create new sessions.
>   while (static_cast<int>(pooled_sessions_.size()) < candidate_pool_size_) {
>-    PortAllocatorSession* pooled_session = CreateSessionInternal("", 0, "", "");
>+    IceParameters iceCredentials =
>+        IceCredentialsIterator::CreateRandomIceCredentials();
>+    PortAllocatorSession* pooled_session =
>+        CreateSessionInternal("", 0, iceCredentials.ufrag, iceCredentials.pwd);
>+    pooled_session->set_pooled(true);
>     pooled_session->StartGettingPorts();
>     pooled_sessions_.push_back(
>         std::unique_ptr<PortAllocatorSession>(pooled_session));
>@@ -214,22 +224,50 @@ std::unique_ptr<PortAllocatorSession> PortAllocator::TakePooledSession(
>   if (pooled_sessions_.empty()) {
>     return nullptr;
>   }
>-  std::unique_ptr<PortAllocatorSession> ret =
>-      std::move(pooled_sessions_.front());
>+
>+  IceParameters credentials(ice_ufrag, ice_pwd, false);
>+  // If restrict_ice_credentials_change_ is TRUE, then call FindPooledSession
>+  // with ice credentials. Otherwise call it with nullptr which means
>+  // "find any" pooled session.
>+  auto cit = FindPooledSession(restrict_ice_credentials_change_ ? &credentials
>+                                                                : nullptr);
>+  if (cit == pooled_sessions_.end()) {
>+    return nullptr;
>+  }
>+
>+  auto it =
>+      pooled_sessions_.begin() + std::distance(pooled_sessions_.cbegin(), cit);
>+  std::unique_ptr<PortAllocatorSession> ret = std::move(*it);
>   ret->SetIceParameters(content_name, component, ice_ufrag, ice_pwd);
>-  // According to JSEP, a pooled session should filter candidates only after
>-  // it's taken out of the pool.
>+  ret->set_pooled(false);
>+  // According to JSEP, a pooled session should filter candidates only
>+  // after it's taken out of the pool.
>   ret->SetCandidateFilter(candidate_filter());
>-  pooled_sessions_.pop_front();
>+  pooled_sessions_.erase(it);
>   return ret;
> }
> 
>-const PortAllocatorSession* PortAllocator::GetPooledSession() const {
>+const PortAllocatorSession* PortAllocator::GetPooledSession(
>+    const IceParameters* ice_credentials) const {
>   CheckRunOnValidThreadAndInitialized();
>-  if (pooled_sessions_.empty()) {
>+  auto it = FindPooledSession(ice_credentials);
>+  if (it == pooled_sessions_.end()) {
>     return nullptr;
>+  } else {
>+    return it->get();
>+  }
>+}
>+
>+std::vector<std::unique_ptr<PortAllocatorSession>>::const_iterator
>+PortAllocator::FindPooledSession(const IceParameters* ice_credentials) const {
>+  for (auto it = pooled_sessions_.begin(); it != pooled_sessions_.end(); ++it) {
>+    if (ice_credentials == nullptr ||
>+        ((*it)->ice_ufrag() == ice_credentials->ufrag &&
>+         (*it)->ice_pwd() == ice_credentials->pwd)) {
>+      return it;
>+    }
>   }
>-  return pooled_sessions_.front().get();
>+  return pooled_sessions_.end();
> }
> 
> void PortAllocator::FreezeCandidatePool() {
>@@ -250,4 +288,14 @@ void PortAllocator::GetCandidateStatsFromPooledSessions(
>   }
> }
> 
>+std::vector<IceParameters> PortAllocator::GetPooledIceCredentials() {
>+  CheckRunOnValidThreadAndInitialized();
>+  std::vector<IceParameters> list;
>+  for (const auto& session : pooled_sessions_) {
>+    list.push_back(
>+        IceParameters(session->ice_ufrag(), session->ice_pwd(), false));
>+  }
>+  return list;
>+}
>+
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.h
>index 8bd709642c8aec3846581be8d6c6b06e78bce020..7026f2b2b63204f72aabd723369d5c3f19d2497e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.h
>@@ -21,6 +21,7 @@
> #include "rtc_base/helpers.h"
> #include "rtc_base/proxyinfo.h"
> #include "rtc_base/sslcertificate.h"
>+#include "rtc_base/system/rtc_export.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/thread_checker.h"
>@@ -146,7 +147,7 @@ struct RelayCredentials {
> 
> typedef std::vector<ProtocolAddress> PortList;
> // TODO(deadbeef): Rename to TurnServerConfig.
>-struct RelayServerConfig {
>+struct RTC_EXPORT RelayServerConfig {
>   explicit RelayServerConfig(RelayType type);
>   RelayServerConfig(const rtc::SocketAddress& address,
>                     const std::string& username,
>@@ -183,7 +184,7 @@ struct RelayServerConfig {
>   rtc::SSLCertificateVerifier* tls_cert_verifier = nullptr;
> };
> 
>-class PortAllocatorSession : public sigslot::has_slots<> {
>+class RTC_EXPORT PortAllocatorSession : public sigslot::has_slots<> {
>  public:
>   // Content name passed in mostly for logging and debugging.
>   PortAllocatorSession(const std::string& content_name,
>@@ -201,7 +202,7 @@ class PortAllocatorSession : public sigslot::has_slots<> {
>   int component() const { return component_; }
>   const std::string& ice_ufrag() const { return ice_ufrag_; }
>   const std::string& ice_pwd() const { return ice_pwd_; }
>-  bool pooled() const { return ice_ufrag_.empty(); }
>+  bool pooled() const { return pooled_; }
> 
>   // Setting this filter should affect not only candidates gathered in the
>   // future, but candidates already gathered and ports already "ready",
>@@ -309,6 +310,8 @@ class PortAllocatorSession : public sigslot::has_slots<> {
>     UpdateIceParametersInternal();
>   }
> 
>+  void set_pooled(bool value) { pooled_ = value; }
>+
>   uint32_t flags_;
>   uint32_t generation_;
>   std::string content_name_;
>@@ -316,6 +319,8 @@ class PortAllocatorSession : public sigslot::has_slots<> {
>   std::string ice_ufrag_;
>   std::string ice_pwd_;
> 
>+  bool pooled_ = false;
>+
>   // SetIceParameters is an implementation detail which only PortAllocator
>   // should be able to call.
>   friend class PortAllocator;
>@@ -326,7 +331,7 @@ class PortAllocatorSession : public sigslot::has_slots<> {
> //
> // This allows a PortAllocator subclass to be constructed and configured on one
> // thread, and passed into an object that uses it on a different thread.
>-class PortAllocator : public sigslot::has_slots<> {
>+class RTC_EXPORT PortAllocator : public sigslot::has_slots<> {
>  public:
>   PortAllocator();
>   ~PortAllocator() override;
>@@ -335,6 +340,11 @@ class PortAllocator : public sigslot::has_slots<> {
>   // constructing and configuring the PortAllocator subclasses.
>   virtual void Initialize();
> 
>+  // Set to true if some Ports need to know the ICE credentials when they are
>+  // created. This will ensure that the PortAllocator will only match pooled
>+  // allocator sessions to the ICE transport with the same credentials.
>+  virtual void set_restrict_ice_credentials_change(bool value);
>+
>   // Set STUN and TURN servers to be used in future sessions, and set
>   // candidate pool size, as described in JSEP.
>   //
>@@ -392,6 +402,8 @@ class PortAllocator : public sigslot::has_slots<> {
>   //
>   // Caller takes ownership of the returned session.
>   //
>+  // If restrict_ice_credentials_change is TRUE, then it will only
>+  //   return a pooled session with matching ice credentials.
>   // If no pooled sessions are available, returns null.
>   std::unique_ptr<PortAllocatorSession> TakePooledSession(
>       const std::string& content_name,
>@@ -399,8 +411,10 @@ class PortAllocator : public sigslot::has_slots<> {
>       const std::string& ice_ufrag,
>       const std::string& ice_pwd);
> 
>-  // Returns the next session that would be returned by TakePooledSession.
>-  const PortAllocatorSession* GetPooledSession() const;
>+  // Returns the next session that would be returned by TakePooledSession
>+  // optionally restricting it to sessions with specified ice credentials.
>+  const PortAllocatorSession* GetPooledSession(
>+      const IceParameters* ice_credentials = nullptr) const;
> 
>   // After FreezeCandidatePool is called, changing the candidate pool size will
>   // no longer be allowed, and changing ICE servers will not cause pooled
>@@ -548,6 +562,9 @@ class PortAllocator : public sigslot::has_slots<> {
>   virtual void GetCandidateStatsFromPooledSessions(
>       CandidateStatsList* candidate_stats_list);
> 
>+  // Return IceParameters of the pooled sessions.
>+  std::vector<IceParameters> GetPooledIceCredentials();
>+
>  protected:
>   virtual PortAllocatorSession* CreateSessionInternal(
>       const std::string& content_name,
>@@ -555,7 +572,7 @@ class PortAllocator : public sigslot::has_slots<> {
>       const std::string& ice_ufrag,
>       const std::string& ice_pwd) = 0;
> 
>-  const std::deque<std::unique_ptr<PortAllocatorSession>>& pooled_sessions() {
>+  const std::vector<std::unique_ptr<PortAllocatorSession>>& pooled_sessions() {
>     return pooled_sessions_;
>   }
> 
>@@ -586,7 +603,7 @@ class PortAllocator : public sigslot::has_slots<> {
>   ServerAddresses stun_servers_;
>   std::vector<RelayServerConfig> turn_servers_;
>   int candidate_pool_size_ = 0;  // Last value passed into SetConfiguration.
>-  std::deque<std::unique_ptr<PortAllocatorSession>> pooled_sessions_;
>+  std::vector<std::unique_ptr<PortAllocatorSession>> pooled_sessions_;
>   bool candidate_pool_frozen_ = false;
>   bool prune_turn_ports_ = false;
> 
>@@ -596,6 +613,15 @@ class PortAllocator : public sigslot::has_slots<> {
>   webrtc::TurnCustomizer* turn_customizer_ = nullptr;
> 
>   absl::optional<int> stun_candidate_keepalive_interval_;
>+
>+  // If true, TakePooledSession() will only return sessions that has same ice
>+  // credentials as requested.
>+  bool restrict_ice_credentials_change_ = false;
>+
>+  // Returns iterator to pooled session with specified ice_credentials or first
>+  // if ice_credentials is nullptr.
>+  std::vector<std::unique_ptr<PortAllocatorSession>>::const_iterator
>+  FindPooledSession(const IceParameters* ice_credentials = nullptr) const;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator_unittest.cc
>index 3887a90db1f29ba6f08123bb13e2d8cbf5327f17..8b317f4d5a953c847332809b6e66c2031ba63600 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator_unittest.cc
>@@ -71,8 +71,7 @@ class PortAllocatorTest : public testing::Test, public sigslot::has_slots<> {
> 
>   int GetAllPooledSessionsReturnCount() {
>     int count = 0;
>-    while (GetPooledSession()) {
>-      TakePooledSession();
>+    while (TakePooledSession() != nullptr) {
>       ++count;
>     }
>     return count;
>@@ -275,3 +274,29 @@ TEST_F(PortAllocatorTest, DiscardCandidatePool) {
>   allocator_->DiscardCandidatePool();
>   EXPECT_EQ(0, GetAllPooledSessionsReturnCount());
> }
>+
>+TEST_F(PortAllocatorTest, RestrictIceCredentialsChange) {
>+  SetConfigurationWithPoolSize(1);
>+  EXPECT_EQ(1, GetAllPooledSessionsReturnCount());
>+  allocator_->DiscardCandidatePool();
>+
>+  // Only return pooled sessions with the ice credentials that
>+  // match those requested in TakePooledSession().
>+  allocator_->set_restrict_ice_credentials_change(true);
>+  SetConfigurationWithPoolSize(1);
>+  EXPECT_EQ(0, GetAllPooledSessionsReturnCount());
>+  allocator_->DiscardCandidatePool();
>+
>+  SetConfigurationWithPoolSize(1);
>+  auto credentials = allocator_->GetPooledIceCredentials();
>+  ASSERT_EQ(1u, credentials.size());
>+  EXPECT_EQ(nullptr,
>+            allocator_->TakePooledSession(kContentName, 0, kIceUfrag, kIcePwd));
>+  EXPECT_NE(nullptr,
>+            allocator_->TakePooledSession(kContentName, 0, credentials[0].ufrag,
>+                                          credentials[0].pwd));
>+  EXPECT_EQ(nullptr,
>+            allocator_->TakePooledSession(kContentName, 0, credentials[0].ufrag,
>+                                          credentials[0].pwd));
>+  allocator_->DiscardCandidatePool();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.cc
>index 161122b6d5c73842b77521fc11e286c6d857c0a9..c74298584aee1af1bb2382d98cfb003cdf3741c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.cc
>@@ -24,7 +24,6 @@
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_minmax.h"
> #include "rtc_base/socket.h"
>-#include "rtc_base/stringutils.h"
> #include "rtc_base/timeutils.h"
> 
> // The following logging is for detailed (packet-level) analysis only.
>@@ -186,8 +185,8 @@ void ReportStats() {
>   char buffer[256];
>   size_t len = 0;
>   for (int i = 0; i < S_NUM_STATS; ++i) {
>-    len += rtc::sprintfn(buffer, arraysize(buffer), "%s%s:%d",
>-                               (i == 0) ? "" : ",", STAT_NAMES[i], g_stats[i]);
>+    len += snprintf(buffer, arraysize(buffer), "%s%s:%d",
>+                          (i == 0) ? "" : ",", STAT_NAMES[i], g_stats[i]);
>     g_stats[i] = 0;
>   }
>   RTC_LOG(LS_INFO) << "Stats[" << buffer << "]";
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.h
>index c363ef64b74af9b1223ff7ca533179ae82c1acb4..5b26aeb21a5bce435a19857ca7d051fadd91370a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.h
>@@ -14,6 +14,7 @@
> #include <list>
> 
> #include "rtc_base/stream.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace cricket {
> 
>@@ -45,7 +46,7 @@ class IPseudoTcpNotify {
> // PseudoTcp
> //////////////////////////////////////////////////////////////////////
> 
>-class PseudoTcp {
>+class RTC_EXPORT PseudoTcp {
>  public:
>   static uint32_t Now();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp_unittest.cc
>index dd86f921d52faa15692ca23a2498e0ca76fe1ff2..2fc9900cbcca5b1fbf4960bc15ea1b8490a82a80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp_unittest.cc
>@@ -15,8 +15,8 @@
> #include "p2p/base/pseudotcp.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/helpers.h"
>+#include "rtc_base/memory_stream.h"
> #include "rtc_base/messagehandler.h"
>-#include "rtc_base/stream.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/timeutils.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.cc
>index fb830ce66a943cd249dda40ad28e9dce4baa38f0..b1b924b80274c394242fce5da359d4632d6398b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.cc
>@@ -139,7 +139,7 @@ class RelayEntry : public rtc::MessageHandler, public sigslot::has_slots<> {
>                     const char* data,
>                     size_t size,
>                     const rtc::SocketAddress& remote_addr,
>-                    const rtc::PacketTime& packet_time);
>+                    const int64_t& packet_time_us);
> 
>   void OnSentPacket(rtc::AsyncPacketSocket* socket,
>                     const rtc::SentPacket& sent_packet);
>@@ -397,9 +397,9 @@ void RelayPort::OnReadPacket(const char* data,
>                              size_t size,
>                              const rtc::SocketAddress& remote_addr,
>                              ProtocolType proto,
>-                             const rtc::PacketTime& packet_time) {
>+                             int64_t packet_time_us) {
>   if (Connection* conn = GetConnection(remote_addr)) {
>-    conn->OnReadPacket(data, size, packet_time);
>+    conn->OnReadPacket(data, size, packet_time_us);
>   } else {
>     Port::OnReadPacket(data, size, remote_addr, proto);
>   }
>@@ -686,7 +686,7 @@ void RelayEntry::OnReadPacket(rtc::AsyncPacketSocket* socket,
>                               const char* data,
>                               size_t size,
>                               const rtc::SocketAddress& remote_addr,
>-                              const rtc::PacketTime& packet_time) {
>+                              const int64_t& packet_time_us) {
>   // RTC_DCHECK(remote_addr == port_->server_addr());
>   // TODO(?): are we worried about this?
> 
>@@ -700,7 +700,7 @@ void RelayEntry::OnReadPacket(rtc::AsyncPacketSocket* socket,
>   // by the server,  The actual remote address is the one we recorded.
>   if (!port_->HasMagicCookie(data, size)) {
>     if (locked_) {
>-      port_->OnReadPacket(data, size, ext_addr_, PROTO_UDP, packet_time);
>+      port_->OnReadPacket(data, size, ext_addr_, PROTO_UDP, packet_time_us);
>     } else {
>       RTC_LOG(WARNING) << "Dropping packet: entry not locked";
>     }
>@@ -753,7 +753,7 @@ void RelayEntry::OnReadPacket(rtc::AsyncPacketSocket* socket,
> 
>   // Process the actual data and remote address in the normal manner.
>   port_->OnReadPacket(data_attr->bytes(), data_attr->length(), remote_addr2,
>-                      PROTO_UDP, packet_time);
>+                      PROTO_UDP, packet_time_us);
> }
> 
> void RelayEntry::OnSentPacket(rtc::AsyncPacketSocket* socket,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.h
>index 48a939e0cfd10e8f7b2fdff7adcdc3013cd54dbb..5989f76e81f5f664fc1d8c63788381379189388f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.h
>@@ -12,10 +12,12 @@
> #define P2P_BASE_RELAYPORT_H_
> 
> #include <deque>
>+#include <memory>
> #include <string>
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/port.h"
> #include "p2p/base/stunrequest.h"
> 
>@@ -35,15 +37,16 @@ class RelayPort : public Port {
>   typedef std::pair<rtc::Socket::Option, int> OptionValue;
> 
>   // RelayPort doesn't yet do anything fancy in the ctor.
>-  static RelayPort* Create(rtc::Thread* thread,
>-                           rtc::PacketSocketFactory* factory,
>-                           rtc::Network* network,
>-                           uint16_t min_port,
>-                           uint16_t max_port,
>-                           const std::string& username,
>-                           const std::string& password) {
>-    return new RelayPort(thread, factory, network, min_port, max_port, username,
>-                         password);
>+  static std::unique_ptr<RelayPort> Create(rtc::Thread* thread,
>+                                           rtc::PacketSocketFactory* factory,
>+                                           rtc::Network* network,
>+                                           uint16_t min_port,
>+                                           uint16_t max_port,
>+                                           const std::string& username,
>+                                           const std::string& password) {
>+    // Using `new` to access a non-public constructor.
>+    return absl::WrapUnique(new RelayPort(thread, factory, network, min_port,
>+                                          max_port, username, password));
>   }
>   ~RelayPort() override;
> 
>@@ -92,7 +95,7 @@ class RelayPort : public Port {
>                     size_t size,
>                     const rtc::SocketAddress& remote_addr,
>                     ProtocolType proto,
>-                    const rtc::PacketTime& packet_time);
>+                    int64_t packet_time_us);
> 
>   // The OnSentPacket callback is left empty here since they are handled by
>   // RelayEntry.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport_unittest.cc
>index e6e922fdaf77fea7d0443474a7392e0d06badfff..1ca69f4ef96ce20bf98eac37f0263e60e144e1e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport_unittest.cc
>@@ -62,10 +62,10 @@ class RelayPortTest : public testing::Test, public sigslot::has_slots<> {
>   }
> 
>   void OnReadPacket(rtc::AsyncPacketSocket* socket,
>-                    const char* data,
>-                    size_t size,
>-                    const rtc::SocketAddress& remote_addr,
>-                    const rtc::PacketTime& packet_time) {
>+                    const char* /* data */,
>+                    size_t /* size */,
>+                    const rtc::SocketAddress& /* remote_addr */,
>+                    const int64_t& /* packet_time_us */) {
>     received_packet_count_[socket]++;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.cc
>index 8e17a79db8d444d56c33dfffc748ce6bea43bbde..ad6b487282e148c0927457c7de19013edf5b84dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.cc
>@@ -115,7 +115,7 @@ void RelayServer::AddInternalSocket(rtc::AsyncPacketSocket* socket) {
> }
> 
> void RelayServer::RemoveInternalSocket(rtc::AsyncPacketSocket* socket) {
>-  SocketList::iterator iter =
>+  auto iter =
>       std::find(internal_sockets_.begin(), internal_sockets_.end(), socket);
>   RTC_DCHECK(iter != internal_sockets_.end());
>   internal_sockets_.erase(iter);
>@@ -132,7 +132,7 @@ void RelayServer::AddExternalSocket(rtc::AsyncPacketSocket* socket) {
> }
> 
> void RelayServer::RemoveExternalSocket(rtc::AsyncPacketSocket* socket) {
>-  SocketList::iterator iter =
>+  auto iter =
>       std::find(external_sockets_.begin(), external_sockets_.end(), socket);
>   RTC_DCHECK(iter != external_sockets_.end());
>   external_sockets_.erase(iter);
>@@ -148,7 +148,7 @@ void RelayServer::AddInternalServerSocket(rtc::AsyncSocket* socket,
> }
> 
> void RelayServer::RemoveInternalServerSocket(rtc::AsyncSocket* socket) {
>-  ServerSocketMap::iterator iter = server_sockets_.find(socket);
>+  auto iter = server_sockets_.find(socket);
>   RTC_DCHECK(iter != server_sockets_.end());
>   server_sockets_.erase(iter);
>   socket->SignalReadEvent.disconnect(this);
>@@ -160,10 +160,9 @@ int RelayServer::GetConnectionCount() const {
> 
> rtc::SocketAddressPair RelayServer::GetConnection(int connection) const {
>   int i = 0;
>-  for (ConnectionMap::const_iterator it = connections_.begin();
>-       it != connections_.end(); ++it) {
>+  for (const auto& entry : connections_) {
>     if (i == connection) {
>-      return it->second->addr_pair();
>+      return entry.second->addr_pair();
>     }
>     ++i;
>   }
>@@ -171,9 +170,8 @@ rtc::SocketAddressPair RelayServer::GetConnection(int connection) const {
> }
> 
> bool RelayServer::HasConnection(const rtc::SocketAddress& address) const {
>-  for (ConnectionMap::const_iterator it = connections_.begin();
>-       it != connections_.end(); ++it) {
>-    if (it->second->addr_pair().destination() == address) {
>+  for (const auto& entry : connections_) {
>+    if (entry.second->addr_pair().destination() == address) {
>       return true;
>     }
>   }
>@@ -189,14 +187,14 @@ void RelayServer::OnInternalPacket(rtc::AsyncPacketSocket* socket,
>                                    const char* bytes,
>                                    size_t size,
>                                    const rtc::SocketAddress& remote_addr,
>-                                   const rtc::PacketTime& packet_time) {
>+                                   const int64_t& /* packet_time_us */) {
>   // Get the address of the connection we just received on.
>   rtc::SocketAddressPair ap(remote_addr, socket->GetLocalAddress());
>   RTC_DCHECK(!ap.destination().IsNil());
> 
>   // If this did not come from an existing connection, it should be a STUN
>   // allocate request.
>-  ConnectionMap::iterator piter = connections_.find(ap);
>+  auto piter = connections_.find(ap);
>   if (piter == connections_.end()) {
>     HandleStunAllocate(bytes, size, ap, socket);
>     return;
>@@ -234,13 +232,13 @@ void RelayServer::OnExternalPacket(rtc::AsyncPacketSocket* socket,
>                                    const char* bytes,
>                                    size_t size,
>                                    const rtc::SocketAddress& remote_addr,
>-                                   const rtc::PacketTime& packet_time) {
>+                                   const int64_t& /* packet_time_us */) {
>   // Get the address of the connection we just received on.
>   rtc::SocketAddressPair ap(remote_addr, socket->GetLocalAddress());
>   RTC_DCHECK(!ap.destination().IsNil());
> 
>   // If this connection already exists, then forward the traffic.
>-  ConnectionMap::iterator piter = connections_.find(ap);
>+  auto piter = connections_.find(ap);
>   if (piter != connections_.end()) {
>     // TODO(?): Check the HMAC.
>     RelayServerConnection* ext_conn = piter->second;
>@@ -276,7 +274,7 @@ void RelayServer::OnExternalPacket(rtc::AsyncPacketSocket* socket,
>   // TODO(?): Check the HMAC.
> 
>   // The binding should already be present.
>-  BindingMap::iterator biter = bindings_.find(username);
>+  auto biter = bindings_.find(username);
>   if (biter == bindings_.end()) {
>     RTC_LOG(LS_WARNING) << "Dropping packet: no binding with username";
>     return;
>@@ -351,7 +349,7 @@ void RelayServer::HandleStunAllocate(const char* bytes,
> 
>   RelayServerBinding* binding;
> 
>-  BindingMap::iterator biter = bindings_.find(username);
>+  auto biter = bindings_.find(username);
>   if (biter != bindings_.end()) {
>     binding = biter->second;
>   } else {
>@@ -514,13 +512,13 @@ void RelayServer::AddConnection(RelayServerConnection* conn) {
> }
> 
> void RelayServer::RemoveConnection(RelayServerConnection* conn) {
>-  ConnectionMap::iterator iter = connections_.find(conn->addr_pair());
>+  auto iter = connections_.find(conn->addr_pair());
>   RTC_DCHECK(iter != connections_.end());
>   connections_.erase(iter);
> }
> 
> void RelayServer::RemoveBinding(RelayServerBinding* binding) {
>-  BindingMap::iterator iter = bindings_.find(binding->username());
>+  auto iter = bindings_.find(binding->username());
>   RTC_DCHECK(iter != bindings_.end());
>   bindings_.erase(iter);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.h
>index c7661204f8199099064bbcbb13ef470ed3ae25b3..5ba5e065a86812058223efca5188c62c98a79e2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.h
>@@ -68,33 +68,27 @@ class RelayServer : public rtc::MessageHandler, public sigslot::has_slots<> {
>   bool HasConnection(const rtc::SocketAddress& address) const;
> 
>  private:
>-  typedef std::vector<rtc::AsyncPacketSocket*> SocketList;
>-  typedef std::map<rtc::AsyncSocket*, cricket::ProtocolType> ServerSocketMap;
>-  typedef std::map<std::string, RelayServerBinding*> BindingMap;
>-  typedef std::map<rtc::SocketAddressPair, RelayServerConnection*>
>-      ConnectionMap;
>-
>   rtc::Thread* thread_;
>   webrtc::Random random_;
>   bool log_bindings_;
>-  SocketList internal_sockets_;
>-  SocketList external_sockets_;
>-  SocketList removed_sockets_;
>-  ServerSocketMap server_sockets_;
>-  BindingMap bindings_;
>-  ConnectionMap connections_;
>+  std::vector<rtc::AsyncPacketSocket*> internal_sockets_;
>+  std::vector<rtc::AsyncPacketSocket*> external_sockets_;
>+  std::vector<rtc::AsyncPacketSocket*> removed_sockets_;
>+  std::map<rtc::AsyncSocket*, cricket::ProtocolType> server_sockets_;
>+  std::map<std::string, RelayServerBinding*> bindings_;
>+  std::map<rtc::SocketAddressPair, RelayServerConnection*> connections_;
> 
>   // Called when a packet is received by the server on one of its sockets.
>   void OnInternalPacket(rtc::AsyncPacketSocket* socket,
>                         const char* bytes,
>                         size_t size,
>                         const rtc::SocketAddress& remote_addr,
>-                        const rtc::PacketTime& packet_time);
>+                        const int64_t& packet_time_us);
>   void OnExternalPacket(rtc::AsyncPacketSocket* socket,
>                         const char* bytes,
>                         size_t size,
>                         const rtc::SocketAddress& remote_addr,
>-                        const rtc::PacketTime& packet_time);
>+                        const int64_t& packet_time_us);
> 
>   void OnReadEvent(rtc::AsyncSocket* socket);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.cc
>index cc9fb4e4c090c33b57c999badf0696812b77b942..fa6b6f832e08575a98a320f815f260b050aacce9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.cc
>@@ -22,7 +22,6 @@
> #include "rtc_base/crc32.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/messagedigest.h"
>-#include "rtc_base/stringencode.h"
> 
> using rtc::ByteBufferReader;
> using rtc::ByteBufferWriter;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.cc
>index c60bf92eb4920c3defdc2c51d60029f2ccef3e2f..727312329ee4d09967be717dd8b640c998700f76 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.cc
>@@ -319,9 +319,9 @@ bool UDPPort::HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
>                                    const char* data,
>                                    size_t size,
>                                    const rtc::SocketAddress& remote_addr,
>-                                   const rtc::PacketTime& packet_time) {
>+                                   int64_t packet_time_us) {
>   // All packets given to UDP port will be consumed.
>-  OnReadPacket(socket, data, size, remote_addr, packet_time);
>+  OnReadPacket(socket, data, size, remote_addr, packet_time_us);
>   return true;
> }
> 
>@@ -358,11 +358,15 @@ void UDPPort::OnLocalAddressReady(rtc::AsyncPacketSocket* socket,
>   MaybePrepareStunCandidate();
> }
> 
>+void UDPPort::PostAddAddress(bool is_final) {
>+  MaybeSetPortCompleteOrError();
>+}
>+
> void UDPPort::OnReadPacket(rtc::AsyncPacketSocket* socket,
>                            const char* data,
>                            size_t size,
>                            const rtc::SocketAddress& remote_addr,
>-                           const rtc::PacketTime& packet_time) {
>+                           const int64_t& packet_time_us) {
>   RTC_DCHECK(socket == socket_);
>   RTC_DCHECK(!remote_addr.IsUnresolvedIP());
> 
>@@ -376,7 +380,7 @@ void UDPPort::OnReadPacket(rtc::AsyncPacketSocket* socket,
>   }
> 
>   if (Connection* conn = GetConnection(remote_addr)) {
>-    conn->OnReadPacket(data, size, packet_time);
>+    conn->OnReadPacket(data, size, packet_time_us);
>   } else {
>     Port::OnReadPacket(data, size, remote_addr, PROTO_UDP);
>   }
>@@ -517,8 +521,14 @@ void UDPPort::OnStunBindingOrResolveRequestFailed(
> }
> 
> void UDPPort::MaybeSetPortCompleteOrError() {
>-  if (ready_)
>+  if (mdns_name_registration_status() ==
>+      MdnsNameRegistrationStatus::kInProgress) {
>     return;
>+  }
>+
>+  if (ready_) {
>+    return;
>+  }
> 
>   // Do not set port ready if we are still waiting for bind responses.
>   const size_t servers_done_bind_request =
>@@ -563,22 +573,24 @@ bool UDPPort::HasCandidateWithAddress(const rtc::SocketAddress& addr) const {
>   return false;
> }
> 
>-StunPort* StunPort::Create(rtc::Thread* thread,
>-                           rtc::PacketSocketFactory* factory,
>-                           rtc::Network* network,
>-                           uint16_t min_port,
>-                           uint16_t max_port,
>-                           const std::string& username,
>-                           const std::string& password,
>-                           const ServerAddresses& servers,
>-                           const std::string& origin,
>-                           absl::optional<int> stun_keepalive_interval) {
>-  StunPort* port = new StunPort(thread, factory, network, min_port, max_port,
>-                                username, password, servers, origin);
>+std::unique_ptr<StunPort> StunPort::Create(
>+    rtc::Thread* thread,
>+    rtc::PacketSocketFactory* factory,
>+    rtc::Network* network,
>+    uint16_t min_port,
>+    uint16_t max_port,
>+    const std::string& username,
>+    const std::string& password,
>+    const ServerAddresses& servers,
>+    const std::string& origin,
>+    absl::optional<int> stun_keepalive_interval) {
>+  // Using `new` to access a non-public constructor.
>+  auto port = absl::WrapUnique(new StunPort(thread, factory, network, min_port,
>+                                            max_port, username, password,
>+                                            servers, origin));
>   port->set_stun_keepalive_delay(stun_keepalive_interval);
>   if (!port->Init()) {
>-    delete port;
>-    port = NULL;
>+    return nullptr;
>   }
>   return port;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.h
>index 5e397835dbeb6518669fd387427ceaf974e4fe90..ca43cbc8c483d0749193c6b1e44894160d4cff15 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.h
>@@ -15,6 +15,7 @@
> #include <memory>
> #include <string>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/port.h"
> #include "p2p/base/stunrequest.h"
> #include "rtc_base/asyncpacketsocket.h"
>@@ -35,42 +36,45 @@ static const int HIGH_COST_PORT_KEEPALIVE_LIFETIME = 2 * 60 * 1000;
> // Communicates using the address on the outside of a NAT.
> class UDPPort : public Port {
>  public:
>-  static UDPPort* Create(rtc::Thread* thread,
>-                         rtc::PacketSocketFactory* factory,
>-                         rtc::Network* network,
>-                         rtc::AsyncPacketSocket* socket,
>-                         const std::string& username,
>-                         const std::string& password,
>-                         const std::string& origin,
>-                         bool emit_local_for_anyaddress,
>-                         absl::optional<int> stun_keepalive_interval) {
>-    UDPPort* port = new UDPPort(thread, factory, network, socket, username,
>-                                password, origin, emit_local_for_anyaddress);
>+  static std::unique_ptr<UDPPort> Create(
>+      rtc::Thread* thread,
>+      rtc::PacketSocketFactory* factory,
>+      rtc::Network* network,
>+      rtc::AsyncPacketSocket* socket,
>+      const std::string& username,
>+      const std::string& password,
>+      const std::string& origin,
>+      bool emit_local_for_anyaddress,
>+      absl::optional<int> stun_keepalive_interval) {
>+    // Using `new` to access a non-public constructor.
>+    auto port = absl::WrapUnique(new UDPPort(thread, factory, network, socket,
>+                                             username, password, origin,
>+                                             emit_local_for_anyaddress));
>     port->set_stun_keepalive_delay(stun_keepalive_interval);
>     if (!port->Init()) {
>-      delete port;
>-      port = NULL;
>+      return nullptr;
>     }
>     return port;
>   }
> 
>-  static UDPPort* Create(rtc::Thread* thread,
>-                         rtc::PacketSocketFactory* factory,
>-                         rtc::Network* network,
>-                         uint16_t min_port,
>-                         uint16_t max_port,
>-                         const std::string& username,
>-                         const std::string& password,
>-                         const std::string& origin,
>-                         bool emit_local_for_anyaddress,
>-                         absl::optional<int> stun_keepalive_interval) {
>-    UDPPort* port =
>+  static std::unique_ptr<UDPPort> Create(
>+      rtc::Thread* thread,
>+      rtc::PacketSocketFactory* factory,
>+      rtc::Network* network,
>+      uint16_t min_port,
>+      uint16_t max_port,
>+      const std::string& username,
>+      const std::string& password,
>+      const std::string& origin,
>+      bool emit_local_for_anyaddress,
>+      absl::optional<int> stun_keepalive_interval) {
>+    // Using `new` to access a non-public constructor.
>+    auto port = absl::WrapUnique(
>         new UDPPort(thread, factory, network, min_port, max_port, username,
>-                    password, origin, emit_local_for_anyaddress);
>+                    password, origin, emit_local_for_anyaddress));
>     port->set_stun_keepalive_delay(stun_keepalive_interval);
>     if (!port->Init()) {
>-      delete port;
>-      port = NULL;
>+      return nullptr;
>     }
>     return port;
>   }
>@@ -100,7 +104,7 @@ class UDPPort : public Port {
>                             const char* data,
>                             size_t size,
>                             const rtc::SocketAddress& remote_addr,
>-                            const rtc::PacketTime& packet_time) override;
>+                            int64_t packet_time_us) override;
> 
>   bool SupportsProtocol(const std::string& protocol) const override;
>   ProtocolType GetProtocol() const override;
>@@ -156,10 +160,14 @@ class UDPPort : public Port {
> 
>   void OnLocalAddressReady(rtc::AsyncPacketSocket* socket,
>                            const rtc::SocketAddress& address);
>+
>+  void PostAddAddress(bool is_final) override;
>+
>   void OnReadPacket(rtc::AsyncPacketSocket* socket,
>-                    const char* data, size_t size,
>+                    const char* data,
>+                    size_t size,
>                     const rtc::SocketAddress& remote_addr,
>-                    const rtc::PacketTime& packet_time);
>+                    const int64_t& packet_time_us);
> 
>   void OnSentPacket(rtc::AsyncPacketSocket* socket,
>                     const rtc::SentPacket& sent_packet) override;
>@@ -262,16 +270,17 @@ class UDPPort : public Port {
> 
> class StunPort : public UDPPort {
>  public:
>-  static StunPort* Create(rtc::Thread* thread,
>-                          rtc::PacketSocketFactory* factory,
>-                          rtc::Network* network,
>-                          uint16_t min_port,
>-                          uint16_t max_port,
>-                          const std::string& username,
>-                          const std::string& password,
>-                          const ServerAddresses& servers,
>-                          const std::string& origin,
>-                          absl::optional<int> stun_keepalive_interval);
>+  static std::unique_ptr<StunPort> Create(
>+      rtc::Thread* thread,
>+      rtc::PacketSocketFactory* factory,
>+      rtc::Network* network,
>+      uint16_t min_port,
>+      uint16_t max_port,
>+      const std::string& username,
>+      const std::string& password,
>+      const ServerAddresses& servers,
>+      const std::string& origin,
>+      absl::optional<int> stun_keepalive_interval);
> 
>   void PrepareAddress() override;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport_unittest.cc
>index 29754e861c938a41a7ea8fbc867cafc3f0f7b1d2..01518a9468364d10ec341dc307e83b5a282a01cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport_unittest.cc
>@@ -75,10 +75,10 @@ class StunPortTestBase : public testing::Test, public sigslot::has_slots<> {
>   }
> 
>   void CreateStunPort(const ServerAddresses& stun_servers) {
>-    stun_port_.reset(cricket::StunPort::Create(
>+    stun_port_ = cricket::StunPort::Create(
>         rtc::Thread::Current(), &socket_factory_, &network_, 0, 0,
>         rtc::CreateRandomString(16), rtc::CreateRandomString(22), stun_servers,
>-        std::string(), absl::nullopt));
>+        std::string(), absl::nullopt);
>     stun_port_->set_stun_keepalive_delay(stun_keepalive_delay_);
>     // If |stun_keepalive_lifetime_| is negative, let the stun port
>     // choose its lifetime from the network type.
>@@ -100,10 +100,10 @@ class StunPortTestBase : public testing::Test, public sigslot::has_slots<> {
>     }
>     ASSERT_TRUE(socket_ != NULL);
>     socket_->SignalReadPacket.connect(this, &StunPortTestBase::OnReadPacket);
>-    stun_port_.reset(cricket::UDPPort::Create(
>+    stun_port_ = cricket::UDPPort::Create(
>         rtc::Thread::Current(), &socket_factory_, &network_, socket_.get(),
>         rtc::CreateRandomString(16), rtc::CreateRandomString(22), std::string(),
>-        false, absl::nullopt));
>+        false, absl::nullopt);
>     ASSERT_TRUE(stun_port_ != NULL);
>     ServerAddresses stun_servers;
>     stun_servers.insert(server_addr);
>@@ -119,15 +119,15 @@ class StunPortTestBase : public testing::Test, public sigslot::has_slots<> {
>                     const char* data,
>                     size_t size,
>                     const rtc::SocketAddress& remote_addr,
>-                    const rtc::PacketTime& packet_time) {
>+                    const int64_t& /* packet_time_us */) {
>     stun_port_->HandleIncomingPacket(socket, data, size, remote_addr,
>-                                     rtc::PacketTime());
>+                                     /* packet_time_us */ -1);
>   }
> 
>   void SendData(const char* data, size_t len) {
>     stun_port_->HandleIncomingPacket(socket_.get(), data, len,
>                                      rtc::SocketAddress("22.22.22.22", 0),
>-                                     rtc::PacketTime());
>+                                     /* packet_time_us */ -1);
>   }
> 
>  protected:
>@@ -216,7 +216,7 @@ TEST_F(StunPortTest, TestPrepareAddressFail) {
> 
> // Test that we can get an address from a STUN server specified by a hostname.
> // Crashes on Linux, see webrtc:7416
>-#if defined(WEBRTC_LINUX)
>+#if defined(WEBRTC_LINUX) || defined(WEBRTC_WIN)
> #define MAYBE_TestPrepareAddressHostname DISABLED_TestPrepareAddressHostname
> #else
> #define MAYBE_TestPrepareAddressHostname TestPrepareAddressHostname
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc
>index 56410bea089d9b269e2ffe216f7cc9ad4477724c..edba4d68e54bc20a0ef3fd466a84e5c9d3632fce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc
>@@ -18,7 +18,7 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/stringencode.h"
>+#include "rtc_base/timeutils.h"  // For TimeMillis
> 
> namespace cricket {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.cc
>index 00eacf17bc9e19a6aebe7efc2cee615da4e6a44d..95f0cebe0a70d88cf039c855aff2544182156bac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.cc
>@@ -29,7 +29,7 @@ void StunServer::OnPacket(rtc::AsyncPacketSocket* socket,
>                           const char* buf,
>                           size_t size,
>                           const rtc::SocketAddress& remote_addr,
>-                          const rtc::PacketTime& packet_time) {
>+                          const int64_t& /* packet_time_us */) {
>   // Parse the STUN message; eat any messages that fail to parse.
>   rtc::ByteBufferReader bbuf(buf, size);
>   StunMessage msg;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.h
>index 9016d06c20aca8d91860969b1b6a1a559bdc88fa..0061d382bc1492524064a23f720f8c2bd3b72645 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.h
>@@ -33,7 +33,7 @@ class StunServer : public sigslot::has_slots<> {
>                 const char* buf,
>                 size_t size,
>                 const rtc::SocketAddress& remote_addr,
>-                const rtc::PacketTime& packet_time);
>+                const int64_t& packet_time_us);
> 
>   // Handlers for the different types of STUN/TURN requests:
>   virtual void OnBindingRequest(StunMessage* msg,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>index c66c7307375e03fb8974b90a0ef2bf92084b7f83..0d7aea9d60a483f15c2511c1a0a05f8c792dbd80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>@@ -307,7 +307,7 @@ void TCPPort::OnReadPacket(rtc::AsyncPacketSocket* socket,
>                            const char* data,
>                            size_t size,
>                            const rtc::SocketAddress& remote_addr,
>-                           const rtc::PacketTime& packet_time) {
>+                           const int64_t& packet_time_us) {
>   Port::OnReadPacket(data, size, remote_addr, PROTO_TCP);
> }
> 
>@@ -545,9 +545,9 @@ void TCPConnection::OnReadPacket(rtc::AsyncPacketSocket* socket,
>                                  const char* data,
>                                  size_t size,
>                                  const rtc::SocketAddress& remote_addr,
>-                                 const rtc::PacketTime& packet_time) {
>+                                 const int64_t& packet_time_us) {
>   RTC_DCHECK(socket == socket_.get());
>-  Connection::OnReadPacket(data, size, packet_time);
>+  Connection::OnReadPacket(data, size, packet_time_us);
> }
> 
> void TCPConnection::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.h
>index fa5c958c21c263bbb4b1ea019b8a170bfa7ad697..4d06a6507c353421eacce9132a9e31dd1628c091 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.h
>@@ -15,6 +15,7 @@
> #include <memory>
> #include <string>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/port.h"
> #include "rtc_base/asyncpacketsocket.h"
> 
>@@ -30,16 +31,18 @@ class TCPConnection;
> // call this TCPPort::OnReadPacket (3 arg) to dispatch to a connection.
> class TCPPort : public Port {
>  public:
>-  static TCPPort* Create(rtc::Thread* thread,
>-                         rtc::PacketSocketFactory* factory,
>-                         rtc::Network* network,
>-                         uint16_t min_port,
>-                         uint16_t max_port,
>-                         const std::string& username,
>-                         const std::string& password,
>-                         bool allow_listen) {
>-    return new TCPPort(thread, factory, network, min_port, max_port, username,
>-                       password, allow_listen);
>+  static std::unique_ptr<TCPPort> Create(rtc::Thread* thread,
>+                                         rtc::PacketSocketFactory* factory,
>+                                         rtc::Network* network,
>+                                         uint16_t min_port,
>+                                         uint16_t max_port,
>+                                         const std::string& username,
>+                                         const std::string& password,
>+                                         bool allow_listen) {
>+    // Using `new` to access a non-public constructor.
>+    return absl::WrapUnique(new TCPPort(thread, factory, network, min_port,
>+                                        max_port, username, password,
>+                                        allow_listen));
>   }
>   ~TCPPort() override;
> 
>@@ -91,7 +94,7 @@ class TCPPort : public Port {
>                     const char* data,
>                     size_t size,
>                     const rtc::SocketAddress& remote_addr,
>-                    const rtc::PacketTime& packet_time);
>+                    const int64_t& packet_time_us);
> 
>   void OnSentPacket(rtc::AsyncPacketSocket* socket,
>                     const rtc::SentPacket& sent_packet) override;
>@@ -159,7 +162,7 @@ class TCPConnection : public Connection {
>                     const char* data,
>                     size_t size,
>                     const rtc::SocketAddress& remote_addr,
>-                    const rtc::PacketTime& packet_time);
>+                    const int64_t& packet_time_us);
>   void OnReadyToSend(rtc::AsyncPacketSocket* socket);
> 
>   std::unique_ptr<rtc::AsyncPacketSocket> socket_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport_unittest.cc
>index ef302f9026032d04320bd1cc08efe700598054ee..34385a7f9bee694084cdb5b6364ddee61cd7301c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport_unittest.cc
>@@ -14,6 +14,7 @@
> #include "p2p/base/basicpacketsocketfactory.h"
> #include "p2p/base/tcpport.h"
> #include "rtc_base/gunit.h"
>+#include "rtc_base/helpers.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/virtualsocketserver.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.cc
>index ad29a1199e7d531b087d0c0a26cce8e91db49913..377a4c3eee327854166e22e0a7dae4196c95aa2f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.cc
>@@ -10,9 +10,9 @@
> 
> #include "p2p/base/transportdescription.h"
> 
>+#include "absl/strings/match.h"
> #include "p2p/base/p2pconstants.h"
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/stringutils.h"
> 
> namespace cricket {
> 
>@@ -22,7 +22,7 @@ bool StringToConnectionRole(const std::string& role_str, ConnectionRole* role) {
>       CONNECTIONROLE_ACTPASS_STR, CONNECTIONROLE_HOLDCONN_STR};
> 
>   for (size_t i = 0; i < arraysize(roles); ++i) {
>-    if (_stricmp(roles[i], role_str.c_str()) == 0) {
>+    if (absl::EqualsIgnoreCase(roles[i], role_str)) {
>       *role = static_cast<ConnectionRole>(CONNECTIONROLE_ACTIVE + i);
>       return true;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.h
>index 3bffdf971a0e936f20081163c9893fcfbddc4ddc..2ab973278faf72aa6034d883483d0e857730ffeb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.h
>@@ -67,11 +67,13 @@ struct IceParameters {
>                 bool ice_renomination)
>       : ufrag(ice_ufrag), pwd(ice_pwd), renomination(ice_renomination) {}
> 
>-  bool operator==(const IceParameters& other) {
>+  bool operator==(const IceParameters& other) const {
>     return ufrag == other.ufrag && pwd == other.pwd &&
>            renomination == other.renomination;
>   }
>-  bool operator!=(const IceParameters& other) { return !(*this == other); }
>+  bool operator!=(const IceParameters& other) const {
>+    return !(*this == other);
>+  }
> };
> 
> extern const char CONNECTIONROLE_ACTIVE_STR[];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.cc
>index 618726e841164c096ae05089c54d15ec1745d046..689cd4f7a2d0bdc7f051f3a44ca750a96afe60e6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.cc
>@@ -27,13 +27,15 @@ TransportDescriptionFactory::~TransportDescriptionFactory() = default;
> 
> TransportDescription* TransportDescriptionFactory::CreateOffer(
>     const TransportOptions& options,
>-    const TransportDescription* current_description) const {
>+    const TransportDescription* current_description,
>+    IceCredentialsIterator* ice_credentials) const {
>   std::unique_ptr<TransportDescription> desc(new TransportDescription());
> 
>   // Generate the ICE credentials if we don't already have them.
>   if (!current_description || options.ice_restart) {
>-    desc->ice_ufrag = rtc::CreateRandomString(ICE_UFRAG_LENGTH);
>-    desc->ice_pwd = rtc::CreateRandomString(ICE_PWD_LENGTH);
>+    IceParameters credentials = ice_credentials->GetIceCredentials();
>+    desc->ice_ufrag = credentials.ufrag;
>+    desc->ice_pwd = credentials.pwd;
>   } else {
>     desc->ice_ufrag = current_description->ice_ufrag;
>     desc->ice_pwd = current_description->ice_pwd;
>@@ -59,7 +61,8 @@ TransportDescription* TransportDescriptionFactory::CreateAnswer(
>     const TransportDescription* offer,
>     const TransportOptions& options,
>     bool require_transport_attributes,
>-    const TransportDescription* current_description) const {
>+    const TransportDescription* current_description,
>+    IceCredentialsIterator* ice_credentials) const {
>   // TODO(juberti): Figure out why we get NULL offers, and fix this upstream.
>   if (!offer) {
>     RTC_LOG(LS_WARNING) << "Failed to create TransportDescription answer "
>@@ -71,8 +74,9 @@ TransportDescription* TransportDescriptionFactory::CreateAnswer(
>   // Generate the ICE credentials if we don't already have them or ice is
>   // being restarted.
>   if (!current_description || options.ice_restart) {
>-    desc->ice_ufrag = rtc::CreateRandomString(ICE_UFRAG_LENGTH);
>-    desc->ice_pwd = rtc::CreateRandomString(ICE_PWD_LENGTH);
>+    IceParameters credentials = ice_credentials->GetIceCredentials();
>+    desc->ice_ufrag = credentials.ufrag;
>+    desc->ice_pwd = credentials.pwd;
>   } else {
>     desc->ice_ufrag = current_description->ice_ufrag;
>     desc->ice_pwd = current_description->ice_pwd;
>@@ -116,8 +120,8 @@ bool TransportDescriptionFactory::SetSecurityInfo(TransportDescription* desc,
>   // This digest algorithm is used to produce the a=fingerprint lines in SDP.
>   // RFC 4572 Section 5 requires that those lines use the same hash function as
>   // the certificate's signature, which is what CreateFromCertificate does.
>-  desc->identity_fingerprint.reset(
>-      rtc::SSLFingerprint::CreateFromCertificate(certificate_));
>+  desc->identity_fingerprint =
>+      rtc::SSLFingerprint::CreateFromCertificate(*certificate_);
>   if (!desc->identity_fingerprint) {
>     return false;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.h
>index 937c5fa1fea4a21faaccbf5e5195e32bf2d10b3e..dc1476a80f5071b9068c014c27fd134cd4dd972e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.h
>@@ -11,6 +11,7 @@
> #ifndef P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
> #define P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
> 
>+#include "p2p/base/icecredentialsiterator.h"
> #include "p2p/base/transportdescription.h"
> #include "rtc_base/rtccertificate.h"
> 
>@@ -54,7 +55,8 @@ class TransportDescriptionFactory {
>   // Creates a transport description suitable for use in an offer.
>   TransportDescription* CreateOffer(
>       const TransportOptions& options,
>-      const TransportDescription* current_description) const;
>+      const TransportDescription* current_description,
>+      IceCredentialsIterator* ice_credentials) const;
>   // Create a transport description that is a response to an offer.
>   //
>   // If |require_transport_attributes| is true, then TRANSPORT category
>@@ -66,7 +68,8 @@ class TransportDescriptionFactory {
>       const TransportDescription* offer,
>       const TransportOptions& options,
>       bool require_transport_attributes,
>-      const TransportDescription* current_description) const;
>+      const TransportDescription* current_description,
>+      IceCredentialsIterator* ice_credentials) const;
> 
>  private:
>   bool SetSecurityInfo(TransportDescription* description,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
>index a7c34b56a04d35c6df90b08dae943cf879351ded..c46630ace2192f1586a76ce3ea6370e96c16cbc7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
>@@ -26,7 +26,8 @@ using cricket::TransportOptions;
> class TransportDescriptionFactoryTest : public testing::Test {
>  public:
>   TransportDescriptionFactoryTest()
>-      : cert1_(rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>+      : ice_credentials_({}),
>+        cert1_(rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>             new rtc::FakeSSLIdentity("User1")))),
>         cert2_(rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>             new rtc::FakeSSLIdentity("User2")))) {}
>@@ -64,21 +65,22 @@ class TransportDescriptionFactoryTest : public testing::Test {
>     SetDtls(dtls);
>     cricket::TransportOptions options;
>     // The initial offer / answer exchange.
>-    std::unique_ptr<TransportDescription> offer(f1_.CreateOffer(options, NULL));
>+    std::unique_ptr<TransportDescription> offer(
>+        f1_.CreateOffer(options, NULL, &ice_credentials_));
>     std::unique_ptr<TransportDescription> answer(
>-        f2_.CreateAnswer(offer.get(), options, true, NULL));
>+        f2_.CreateAnswer(offer.get(), options, true, NULL, &ice_credentials_));
> 
>     // Create an updated offer where we restart ice.
>     options.ice_restart = true;
>     std::unique_ptr<TransportDescription> restart_offer(
>-        f1_.CreateOffer(options, offer.get()));
>+        f1_.CreateOffer(options, offer.get(), &ice_credentials_));
> 
>     VerifyUfragAndPasswordChanged(dtls, offer.get(), restart_offer.get());
> 
>     // Create a new answer. The transport ufrag and password is changed since
>     // |options.ice_restart == true|
>-    std::unique_ptr<TransportDescription> restart_answer(
>-        f2_.CreateAnswer(restart_offer.get(), options, true, answer.get()));
>+    std::unique_ptr<TransportDescription> restart_answer(f2_.CreateAnswer(
>+        restart_offer.get(), options, true, answer.get(), &ice_credentials_));
>     ASSERT_TRUE(restart_answer.get() != NULL);
> 
>     VerifyUfragAndPasswordChanged(dtls, answer.get(), restart_answer.get());
>@@ -108,19 +110,20 @@ class TransportDescriptionFactoryTest : public testing::Test {
>     cricket::TransportOptions options;
>     // The initial offer / answer exchange.
>     std::unique_ptr<TransportDescription> offer(
>-        f1_.CreateOffer(options, nullptr));
>-    std::unique_ptr<TransportDescription> answer(
>-        f2_.CreateAnswer(offer.get(), options, true, nullptr));
>+        f1_.CreateOffer(options, nullptr, &ice_credentials_));
>+    std::unique_ptr<TransportDescription> answer(f2_.CreateAnswer(
>+        offer.get(), options, true, nullptr, &ice_credentials_));
>     VerifyRenomination(offer.get(), false);
>     VerifyRenomination(answer.get(), false);
> 
>     options.enable_ice_renomination = true;
>     std::unique_ptr<TransportDescription> renomination_offer(
>-        f1_.CreateOffer(options, offer.get()));
>+        f1_.CreateOffer(options, offer.get(), &ice_credentials_));
>     VerifyRenomination(renomination_offer.get(), true);
> 
>-    std::unique_ptr<TransportDescription> renomination_answer(f2_.CreateAnswer(
>-        renomination_offer.get(), options, true, answer.get()));
>+    std::unique_ptr<TransportDescription> renomination_answer(
>+        f2_.CreateAnswer(renomination_offer.get(), options, true, answer.get(),
>+                         &ice_credentials_));
>     VerifyRenomination(renomination_answer.get(), true);
>   }
> 
>@@ -145,6 +148,7 @@ class TransportDescriptionFactoryTest : public testing::Test {
>     }
>   }
> 
>+  cricket::IceCredentialsIterator ice_credentials_;
>   TransportDescriptionFactory f1_;
>   TransportDescriptionFactory f2_;
> 
>@@ -154,7 +158,7 @@ class TransportDescriptionFactoryTest : public testing::Test {
> 
> TEST_F(TransportDescriptionFactoryTest, TestOfferDefault) {
>   std::unique_ptr<TransportDescription> desc(
>-      f1_.CreateOffer(TransportOptions(), NULL));
>+      f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_));
>   CheckDesc(desc.get(), "", "", "", "");
> }
> 
>@@ -163,13 +167,13 @@ TEST_F(TransportDescriptionFactoryTest, TestOfferDtls) {
>   f1_.set_certificate(cert1_);
>   std::string digest_alg;
>   ASSERT_TRUE(
>-      cert1_->ssl_certificate().GetSignatureDigestAlgorithm(&digest_alg));
>+      cert1_->GetSSLCertificate().GetSignatureDigestAlgorithm(&digest_alg));
>   std::unique_ptr<TransportDescription> desc(
>-      f1_.CreateOffer(TransportOptions(), NULL));
>+      f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_));
>   CheckDesc(desc.get(), "", "", "", digest_alg);
>   // Ensure it also works with SEC_REQUIRED.
>   f1_.set_secure(cricket::SEC_REQUIRED);
>-  desc.reset(f1_.CreateOffer(TransportOptions(), NULL));
>+  desc.reset(f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_));
>   CheckDesc(desc.get(), "", "", "", digest_alg);
> }
> 
>@@ -177,7 +181,7 @@ TEST_F(TransportDescriptionFactoryTest, TestOfferDtls) {
> TEST_F(TransportDescriptionFactoryTest, TestOfferDtlsWithNoIdentity) {
>   f1_.set_secure(cricket::SEC_ENABLED);
>   std::unique_ptr<TransportDescription> desc(
>-      f1_.CreateOffer(TransportOptions(), NULL));
>+      f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_));
>   ASSERT_TRUE(desc.get() == NULL);
> }
> 
>@@ -188,36 +192,38 @@ TEST_F(TransportDescriptionFactoryTest, TestOfferDtlsReofferDtls) {
>   f1_.set_certificate(cert1_);
>   std::string digest_alg;
>   ASSERT_TRUE(
>-      cert1_->ssl_certificate().GetSignatureDigestAlgorithm(&digest_alg));
>+      cert1_->GetSSLCertificate().GetSignatureDigestAlgorithm(&digest_alg));
>   std::unique_ptr<TransportDescription> old_desc(
>-      f1_.CreateOffer(TransportOptions(), NULL));
>+      f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_));
>   ASSERT_TRUE(old_desc.get() != NULL);
>   std::unique_ptr<TransportDescription> desc(
>-      f1_.CreateOffer(TransportOptions(), old_desc.get()));
>+      f1_.CreateOffer(TransportOptions(), old_desc.get(), &ice_credentials_));
>   CheckDesc(desc.get(), "", old_desc->ice_ufrag, old_desc->ice_pwd, digest_alg);
> }
> 
> TEST_F(TransportDescriptionFactoryTest, TestAnswerDefault) {
>   std::unique_ptr<TransportDescription> offer(
>-      f1_.CreateOffer(TransportOptions(), NULL));
>+      f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_));
>   ASSERT_TRUE(offer.get() != NULL);
>-  std::unique_ptr<TransportDescription> desc(
>-      f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL));
>+  std::unique_ptr<TransportDescription> desc(f2_.CreateAnswer(
>+      offer.get(), TransportOptions(), true, NULL, &ice_credentials_));
>   CheckDesc(desc.get(), "", "", "", "");
>-  desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL));
>+  desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL,
>+                              &ice_credentials_));
>   CheckDesc(desc.get(), "", "", "", "");
> }
> 
> // Test that we can update an answer properly; ICE credentials shouldn't change.
> TEST_F(TransportDescriptionFactoryTest, TestReanswer) {
>   std::unique_ptr<TransportDescription> offer(
>-      f1_.CreateOffer(TransportOptions(), NULL));
>+      f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_));
>   ASSERT_TRUE(offer.get() != NULL);
>-  std::unique_ptr<TransportDescription> old_desc(
>-      f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL));
>+  std::unique_ptr<TransportDescription> old_desc(f2_.CreateAnswer(
>+      offer.get(), TransportOptions(), true, NULL, &ice_credentials_));
>   ASSERT_TRUE(old_desc.get() != NULL);
>   std::unique_ptr<TransportDescription> desc(
>-      f2_.CreateAnswer(offer.get(), TransportOptions(), true, old_desc.get()));
>+      f2_.CreateAnswer(offer.get(), TransportOptions(), true, old_desc.get(),
>+                       &ice_credentials_));
>   ASSERT_TRUE(desc.get() != NULL);
>   CheckDesc(desc.get(), "", old_desc->ice_ufrag, old_desc->ice_pwd, "");
> }
>@@ -227,10 +233,10 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerDtlsToNoDtls) {
>   f1_.set_secure(cricket::SEC_ENABLED);
>   f1_.set_certificate(cert1_);
>   std::unique_ptr<TransportDescription> offer(
>-      f1_.CreateOffer(TransportOptions(), NULL));
>+      f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_));
>   ASSERT_TRUE(offer.get() != NULL);
>-  std::unique_ptr<TransportDescription> desc(
>-      f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL));
>+  std::unique_ptr<TransportDescription> desc(f2_.CreateAnswer(
>+      offer.get(), TransportOptions(), true, NULL, &ice_credentials_));
>   CheckDesc(desc.get(), "", "", "", "");
> }
> 
>@@ -240,13 +246,14 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerNoDtlsToDtls) {
>   f2_.set_secure(cricket::SEC_ENABLED);
>   f2_.set_certificate(cert2_);
>   std::unique_ptr<TransportDescription> offer(
>-      f1_.CreateOffer(TransportOptions(), NULL));
>+      f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_));
>   ASSERT_TRUE(offer.get() != NULL);
>-  std::unique_ptr<TransportDescription> desc(
>-      f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL));
>+  std::unique_ptr<TransportDescription> desc(f2_.CreateAnswer(
>+      offer.get(), TransportOptions(), true, NULL, &ice_credentials_));
>   CheckDesc(desc.get(), "", "", "", "");
>   f2_.set_secure(cricket::SEC_REQUIRED);
>-  desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL));
>+  desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL,
>+                              &ice_credentials_));
>   ASSERT_TRUE(desc.get() == NULL);
> }
> 
>@@ -262,16 +269,17 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerDtlsToDtls) {
>   // answer must contain fingerprint lines with cert2_'s digest algorithm.
>   std::string digest_alg2;
>   ASSERT_TRUE(
>-      cert2_->ssl_certificate().GetSignatureDigestAlgorithm(&digest_alg2));
>+      cert2_->GetSSLCertificate().GetSignatureDigestAlgorithm(&digest_alg2));
> 
>   std::unique_ptr<TransportDescription> offer(
>-      f1_.CreateOffer(TransportOptions(), NULL));
>+      f1_.CreateOffer(TransportOptions(), NULL, &ice_credentials_));
>   ASSERT_TRUE(offer.get() != NULL);
>-  std::unique_ptr<TransportDescription> desc(
>-      f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL));
>+  std::unique_ptr<TransportDescription> desc(f2_.CreateAnswer(
>+      offer.get(), TransportOptions(), true, NULL, &ice_credentials_));
>   CheckDesc(desc.get(), "", "", "", digest_alg2);
>   f2_.set_secure(cricket::SEC_REQUIRED);
>-  desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL));
>+  desc.reset(f2_.CreateAnswer(offer.get(), TransportOptions(), true, NULL,
>+                              &ice_credentials_));
>   CheckDesc(desc.get(), "", "", "", digest_alg2);
> }
> 
>@@ -304,9 +312,36 @@ TEST_F(TransportDescriptionFactoryTest, TestIceRenominationWithDtls) {
> TEST_F(TransportDescriptionFactoryTest, AddsTrickleIceOption) {
>   cricket::TransportOptions options;
>   std::unique_ptr<TransportDescription> offer(
>-      f1_.CreateOffer(options, nullptr));
>+      f1_.CreateOffer(options, nullptr, &ice_credentials_));
>   EXPECT_TRUE(offer->HasOption("trickle"));
>   std::unique_ptr<TransportDescription> answer(
>-      f2_.CreateAnswer(offer.get(), options, true, nullptr));
>+      f2_.CreateAnswer(offer.get(), options, true, nullptr, &ice_credentials_));
>   EXPECT_TRUE(answer->HasOption("trickle"));
> }
>+
>+// Test CreateOffer with IceCredentialsIterator.
>+TEST_F(TransportDescriptionFactoryTest, CreateOfferIceCredentialsIterator) {
>+  std::vector<cricket::IceParameters> credentials = {
>+      cricket::IceParameters("kalle", "anka", false)};
>+  cricket::IceCredentialsIterator credentialsIterator(credentials);
>+  cricket::TransportOptions options;
>+  std::unique_ptr<TransportDescription> offer(
>+      f1_.CreateOffer(options, nullptr, &credentialsIterator));
>+  EXPECT_EQ(offer->GetIceParameters().ufrag, credentials[0].ufrag);
>+  EXPECT_EQ(offer->GetIceParameters().pwd, credentials[0].pwd);
>+}
>+
>+// Test CreateAnswer with IceCredentialsIterator.
>+TEST_F(TransportDescriptionFactoryTest, CreateAnswerIceCredentialsIterator) {
>+  cricket::TransportOptions options;
>+  std::unique_ptr<TransportDescription> offer(
>+      f1_.CreateOffer(options, nullptr, &ice_credentials_));
>+
>+  std::vector<cricket::IceParameters> credentials = {
>+      cricket::IceParameters("kalle", "anka", false)};
>+  cricket::IceCredentialsIterator credentialsIterator(credentials);
>+  std::unique_ptr<TransportDescription> answer(f1_.CreateAnswer(
>+      offer.get(), options, false, nullptr, &credentialsIterator));
>+  EXPECT_EQ(answer->GetIceParameters().ufrag, credentials[0].ufrag);
>+  EXPECT_EQ(answer->GetIceParameters().pwd, credentials[0].pwd);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportfactoryinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportfactoryinterface.h
>index ce32ee8483edeb7ee2f084578323024aa783a67e..9805db005cc02155fbd22df6cedea03b0240994e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportfactoryinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportfactoryinterface.h
>@@ -34,7 +34,7 @@ class TransportFactoryInterface {
> 
>   virtual std::unique_ptr<DtlsTransportInternal> CreateDtlsTransport(
>       std::unique_ptr<IceTransportInternal> ice,
>-      const rtc::CryptoOptions& crypto_options) = 0;
>+      const webrtc::CryptoOptions& crypto_options) = 0;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.cc
>index 97541f60c02dd9b742d7def22c3509fbbb84b211..eb0adfa3ce4e1db990c303972a6c0e85587093a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.cc
>@@ -24,7 +24,6 @@
> #include "rtc_base/logging.h"
> #include "rtc_base/nethelpers.h"
> #include "rtc_base/socketaddress.h"
>-#include "rtc_base/stringencode.h"
> #include "rtc_base/strings/string_builder.h"
> 
> namespace cricket {
>@@ -623,7 +622,7 @@ bool TurnPort::HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
>                                     const char* data,
>                                     size_t size,
>                                     const rtc::SocketAddress& remote_addr,
>-                                    const rtc::PacketTime& packet_time) {
>+                                    int64_t packet_time_us) {
>   if (socket != socket_) {
>     // The packet was received on a shared socket after we've allocated a new
>     // socket for this TURN port.
>@@ -660,12 +659,12 @@ bool TurnPort::HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
>   // a response to a previous request.
>   uint16_t msg_type = rtc::GetBE16(data);
>   if (IsTurnChannelData(msg_type)) {
>-    HandleChannelData(msg_type, data, size, packet_time);
>+    HandleChannelData(msg_type, data, size, packet_time_us);
>     return true;
>   }
> 
>   if (msg_type == TURN_DATA_INDICATION) {
>-    HandleDataIndication(data, size, packet_time);
>+    HandleDataIndication(data, size, packet_time_us);
>     return true;
>   }
> 
>@@ -696,8 +695,8 @@ void TurnPort::OnReadPacket(rtc::AsyncPacketSocket* socket,
>                             const char* data,
>                             size_t size,
>                             const rtc::SocketAddress& remote_addr,
>-                            const rtc::PacketTime& packet_time) {
>-  HandleIncomingPacket(socket, data, size, remote_addr, packet_time);
>+                            const int64_t& packet_time_us) {
>+  HandleIncomingPacket(socket, data, size, remote_addr, packet_time_us);
> }
> 
> void TurnPort::OnSentPacket(rtc::AsyncPacketSocket* socket,
>@@ -933,7 +932,7 @@ void TurnPort::OnAllocateRequestTimeout() {
> 
> void TurnPort::HandleDataIndication(const char* data,
>                                     size_t size,
>-                                    const rtc::PacketTime& packet_time) {
>+                                    int64_t packet_time_us) {
>   // Read in the message, and process according to RFC5766, Section 10.4.
>   rtc::ByteBufferReader buf(data, size);
>   TurnMessage msg;
>@@ -972,13 +971,13 @@ void TurnPort::HandleDataIndication(const char* data,
>   }
> 
>   DispatchPacket(data_attr->bytes(), data_attr->length(), ext_addr, PROTO_UDP,
>-                 packet_time);
>+                 packet_time_us);
> }
> 
> void TurnPort::HandleChannelData(int channel_id,
>                                  const char* data,
>                                  size_t size,
>-                                 const rtc::PacketTime& packet_time) {
>+                                 int64_t packet_time_us) {
>   // Read the message, and process according to RFC5766, Section 11.6.
>   //    0                   1                   2                   3
>   //    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>@@ -1014,16 +1013,16 @@ void TurnPort::HandleChannelData(int channel_id,
>   }
> 
>   DispatchPacket(data + TURN_CHANNEL_HEADER_SIZE, len, entry->address(),
>-                 PROTO_UDP, packet_time);
>+                 PROTO_UDP, packet_time_us);
> }
> 
> void TurnPort::DispatchPacket(const char* data,
>                               size_t size,
>                               const rtc::SocketAddress& remote_addr,
>                               ProtocolType proto,
>-                              const rtc::PacketTime& packet_time) {
>+                              int64_t packet_time_us) {
>   if (Connection* conn = GetConnection(remote_addr)) {
>-    conn->OnReadPacket(data, size, packet_time);
>+    conn->OnReadPacket(data, size, packet_time_us);
>   } else {
>     Port::OnReadPacket(data, size, remote_addr, proto);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.h
>index 11b58d4f6c99faa9428b825bac67ae863cd1d3aa..5db1956a9b1afa7ca647c26eb320e5f4b36a950b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.h
>@@ -14,10 +14,12 @@
> #include <stdio.h>
> #include <list>
> #include <map>
>+#include <memory>
> #include <set>
> #include <string>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/port.h"
> #include "p2p/client/basicportallocator.h"
> #include "rtc_base/asyncinvoker.h"
>@@ -50,25 +52,44 @@ class TurnPort : public Port {
>                          // packets.
>   };
>   // Create a TURN port using the shared UDP socket, |socket|.
>-  static TurnPort* Create(rtc::Thread* thread,
>-                          rtc::PacketSocketFactory* factory,
>-                          rtc::Network* network,
>-                          rtc::AsyncPacketSocket* socket,
>-                          const std::string& username,  // ice username.
>-                          const std::string& password,  // ice password.
>-                          const ProtocolAddress& server_address,
>-                          const RelayCredentials& credentials,
>-                          int server_priority,
>-                          const std::string& origin,
>-                          webrtc::TurnCustomizer* customizer) {
>-    return new TurnPort(thread, factory, network, socket, username, password,
>-                        server_address, credentials, server_priority, origin,
>-                        customizer);
>+  static std::unique_ptr<TurnPort> Create(
>+      rtc::Thread* thread,
>+      rtc::PacketSocketFactory* factory,
>+      rtc::Network* network,
>+      rtc::AsyncPacketSocket* socket,
>+      const std::string& username,  // ice username.
>+      const std::string& password,  // ice password.
>+      const ProtocolAddress& server_address,
>+      const RelayCredentials& credentials,
>+      int server_priority,
>+      const std::string& origin,
>+      webrtc::TurnCustomizer* customizer) {
>+    // Using `new` to access a non-public constructor.
>+    return absl::WrapUnique(new TurnPort(
>+        thread, factory, network, socket, username, password, server_address,
>+        credentials, server_priority, origin, customizer));
>+  }
>+  // TODO(steveanton): Remove once downstream clients have moved to |Create|.
>+  static std::unique_ptr<TurnPort> CreateUnique(
>+      rtc::Thread* thread,
>+      rtc::PacketSocketFactory* factory,
>+      rtc::Network* network,
>+      rtc::AsyncPacketSocket* socket,
>+      const std::string& username,  // ice username.
>+      const std::string& password,  // ice password.
>+      const ProtocolAddress& server_address,
>+      const RelayCredentials& credentials,
>+      int server_priority,
>+      const std::string& origin,
>+      webrtc::TurnCustomizer* customizer) {
>+    return Create(thread, factory, network, socket, username, password,
>+                  server_address, credentials, server_priority, origin,
>+                  customizer);
>   }
> 
>   // Create a TURN port that will use a new socket, bound to |network| and
>   // using a port in the range between |min_port| and |max_port|.
>-  static TurnPort* Create(
>+  static std::unique_ptr<TurnPort> Create(
>       rtc::Thread* thread,
>       rtc::PacketSocketFactory* factory,
>       rtc::Network* network,
>@@ -84,10 +105,34 @@ class TurnPort : public Port {
>       const std::vector<std::string>& tls_elliptic_curves,
>       webrtc::TurnCustomizer* customizer,
>       rtc::SSLCertificateVerifier* tls_cert_verifier = nullptr) {
>-    return new TurnPort(thread, factory, network, min_port, max_port, username,
>-                        password, server_address, credentials, server_priority,
>-                        origin, tls_alpn_protocols, tls_elliptic_curves,
>-                        customizer, tls_cert_verifier);
>+    // Using `new` to access a non-public constructor.
>+    return absl::WrapUnique(
>+        new TurnPort(thread, factory, network, min_port, max_port, username,
>+                     password, server_address, credentials, server_priority,
>+                     origin, tls_alpn_protocols, tls_elliptic_curves,
>+                     customizer, tls_cert_verifier));
>+  }
>+  // TODO(steveanton): Remove once downstream clients have moved to |Create|.
>+  static std::unique_ptr<TurnPort> CreateUnique(
>+      rtc::Thread* thread,
>+      rtc::PacketSocketFactory* factory,
>+      rtc::Network* network,
>+      uint16_t min_port,
>+      uint16_t max_port,
>+      const std::string& username,  // ice username.
>+      const std::string& password,  // ice password.
>+      const ProtocolAddress& server_address,
>+      const RelayCredentials& credentials,
>+      int server_priority,
>+      const std::string& origin,
>+      const std::vector<std::string>& tls_alpn_protocols,
>+      const std::vector<std::string>& tls_elliptic_curves,
>+      webrtc::TurnCustomizer* customizer,
>+      rtc::SSLCertificateVerifier* tls_cert_verifier = nullptr) {
>+    return Create(thread, factory, network, min_port, max_port, username,
>+                  password, server_address, credentials, server_priority,
>+                  origin, tls_alpn_protocols, tls_elliptic_curves, customizer,
>+                  tls_cert_verifier);
>   }
> 
>   ~TurnPort() override;
>@@ -130,13 +175,14 @@ class TurnPort : public Port {
>                             const char* data,
>                             size_t size,
>                             const rtc::SocketAddress& remote_addr,
>-                            const rtc::PacketTime& packet_time) override;
>+                            int64_t packet_time_us) override;
>   bool CanHandleIncomingPacketsFrom(
>       const rtc::SocketAddress& addr) const override;
>   virtual void OnReadPacket(rtc::AsyncPacketSocket* socket,
>-                            const char* data, size_t size,
>+                            const char* data,
>+                            size_t size,
>                             const rtc::SocketAddress& remote_addr,
>-                            const rtc::PacketTime& packet_time);
>+                            const int64_t& packet_time_us);
> 
>   void OnSentPacket(rtc::AsyncPacketSocket* socket,
>                     const rtc::SentPacket& sent_packet) override;
>@@ -268,13 +314,18 @@ class TurnPort : public Port {
>   void OnAllocateError();
>   void OnAllocateRequestTimeout();
> 
>-  void HandleDataIndication(const char* data, size_t size,
>-                            const rtc::PacketTime& packet_time);
>-  void HandleChannelData(int channel_id, const char* data, size_t size,
>-                         const rtc::PacketTime& packet_time);
>-  void DispatchPacket(const char* data, size_t size,
>-      const rtc::SocketAddress& remote_addr,
>-      ProtocolType proto, const rtc::PacketTime& packet_time);
>+  void HandleDataIndication(const char* data,
>+                            size_t size,
>+                            int64_t packet_time_us);
>+  void HandleChannelData(int channel_id,
>+                         const char* data,
>+                         size_t size,
>+                         int64_t packet_time_us);
>+  void DispatchPacket(const char* data,
>+                      size_t size,
>+                      const rtc::SocketAddress& remote_addr,
>+                      ProtocolType proto,
>+                      int64_t packet_time_us);
> 
>   bool ScheduleRefresh(uint32_t lifetime);
>   void SendRequest(StunRequest* request, int delay);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport_unittest.cc
>index bac35e891cd49dcde0ca36941cccb6bc89e2b041..617e7798adf53fae544e00c19c444611cc9dc347 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport_unittest.cc
>@@ -194,23 +194,23 @@ class TurnPortTest : public testing::Test,
>   void OnTurnReadPacket(Connection* conn,
>                         const char* data,
>                         size_t size,
>-                        const rtc::PacketTime& packet_time) {
>+                        int64_t packet_time_us) {
>     turn_packets_.push_back(rtc::Buffer(data, size));
>   }
>   void OnUdpPortComplete(Port* port) { udp_ready_ = true; }
>   void OnUdpReadPacket(Connection* conn,
>                        const char* data,
>                        size_t size,
>-                       const rtc::PacketTime& packet_time) {
>+                       int64_t packet_time_us) {
>     udp_packets_.push_back(rtc::Buffer(data, size));
>   }
>   void OnSocketReadPacket(rtc::AsyncPacketSocket* socket,
>                           const char* data,
>                           size_t size,
>                           const rtc::SocketAddress& remote_addr,
>-                          const rtc::PacketTime& packet_time) {
>+                          const int64_t& packet_time_us) {
>     turn_port_->HandleIncomingPacket(socket, data, size, remote_addr,
>-                                     packet_time);
>+                                     packet_time_us);
>   }
>   void OnTurnPortClosed(TurnPort* port) { turn_port_closed_ = true; }
>   void OnTurnPortDestroyed(PortInterface* port) { turn_port_destroyed_ = true; }
>@@ -270,10 +270,9 @@ class TurnPortTest : public testing::Test,
>                                    const ProtocolAddress& server_address,
>                                    const std::string& origin) {
>     RelayCredentials credentials(username, password);
>-    turn_port_.reset(TurnPort::Create(
>+    turn_port_ = TurnPort::Create(
>         &main_, &socket_factory_, network, 0, 0, kIceUfrag1, kIcePwd1,
>-        server_address, credentials, 0, origin, std::vector<std::string>(),
>-        std::vector<std::string>(), turn_customizer_.get()));
>+        server_address, credentials, 0, origin, {}, {}, turn_customizer_.get());
>     // This TURN port will be the controlling.
>     turn_port_->SetIceRole(ICEROLE_CONTROLLING);
>     ConnectSignals();
>@@ -301,10 +300,10 @@ class TurnPortTest : public testing::Test,
>     }
> 
>     RelayCredentials credentials(username, password);
>-    turn_port_.reset(TurnPort::Create(&main_, &socket_factory_,
>-                                      MakeNetwork(kLocalAddr1), socket_.get(),
>-                                      kIceUfrag1, kIcePwd1, server_address,
>-                                      credentials, 0, std::string(), nullptr));
>+    turn_port_ =
>+        TurnPort::Create(&main_, &socket_factory_, MakeNetwork(kLocalAddr1),
>+                         socket_.get(), kIceUfrag1, kIcePwd1, server_address,
>+                         credentials, 0, std::string(), nullptr);
>     // This TURN port will be the controlling.
>     turn_port_->SetIceRole(ICEROLE_CONTROLLING);
>     ConnectSignals();
>@@ -329,9 +328,9 @@ class TurnPortTest : public testing::Test,
>   void CreateUdpPort() { CreateUdpPort(kLocalAddr2); }
> 
>   void CreateUdpPort(const SocketAddress& address) {
>-    udp_port_.reset(UDPPort::Create(
>-        &main_, &socket_factory_, MakeNetwork(address), 0, 0, kIceUfrag2,
>-        kIcePwd2, std::string(), false, absl::nullopt));
>+    udp_port_ = UDPPort::Create(&main_, &socket_factory_, MakeNetwork(address),
>+                                0, 0, kIceUfrag2, kIcePwd2, std::string(),
>+                                false, absl::nullopt);
>     // UDP port will be controlled.
>     udp_port_->SetIceRole(ICEROLE_CONTROLLED);
>     udp_port_->SignalPortComplete.connect(this,
>@@ -1032,8 +1031,7 @@ TEST_F(TurnPortTest, TestTurnAllocateMismatch) {
>   std::string test_packet = "Test packet";
>   EXPECT_FALSE(turn_port_->HandleIncomingPacket(
>       socket_.get(), test_packet.data(), test_packet.size(),
>-      rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0),
>-      rtc::CreatePacketTime(0)));
>+      rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0), rtc::TimeMicros()));
> }
> 
> // Tests that a shared-socket-TurnPort creates its own socket after
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.cc
>index 022113e7e746a0c59076b8b062cb310992a94e30..a8bb41c87a6acad2e284aa783101537f7774ffd0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.cc
>@@ -24,7 +24,6 @@
> #include "rtc_base/logging.h"
> #include "rtc_base/messagedigest.h"
> #include "rtc_base/socketadapters.h"
>-#include "rtc_base/stringencode.h"
> #include "rtc_base/strings/string_builder.h"
> #include "rtc_base/thread.h"
> 
>@@ -199,9 +198,10 @@ void TurnServer::OnInternalSocketClose(rtc::AsyncPacketSocket* socket,
> }
> 
> void TurnServer::OnInternalPacket(rtc::AsyncPacketSocket* socket,
>-                                  const char* data, size_t size,
>+                                  const char* data,
>+                                  size_t size,
>                                   const rtc::SocketAddress& addr,
>-                                  const rtc::PacketTime& packet_time) {
>+                                  const int64_t& /* packet_time_us */) {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   // Fail if the packet is too small to even contain a channel header.
>   if (size < TURN_CHANNEL_HEADER_SIZE) {
>@@ -842,9 +842,10 @@ void TurnServerAllocation::HandleChannelData(const char* data, size_t size) {
> 
> void TurnServerAllocation::OnExternalPacket(
>     rtc::AsyncPacketSocket* socket,
>-    const char* data, size_t size,
>+    const char* data,
>+    size_t size,
>     const rtc::SocketAddress& addr,
>-    const rtc::PacketTime& packet_time) {
>+    const int64_t& /* packet_time_us */) {
>   RTC_DCHECK(external_socket_.get() == socket);
>   Channel* channel = FindChannel(addr);
>   if (channel) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.h
>index e7f886cf8fb553de3494cbe72813a9d88d3f984b..af51251bbcd0cf03984f031bb272897e343e439c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.h
>@@ -105,9 +105,10 @@ class TurnServerAllocation : public rtc::MessageHandler,
>   void HandleChannelBindRequest(const TurnMessage* msg);
> 
>   void OnExternalPacket(rtc::AsyncPacketSocket* socket,
>-                        const char* data, size_t size,
>+                        const char* data,
>+                        size_t size,
>                         const rtc::SocketAddress& addr,
>-                        const rtc::PacketTime& packet_time);
>+                        const int64_t& packet_time_us);
> 
>   static int ComputeLifetime(const TurnMessage* msg);
>   bool HasPermission(const rtc::IPAddress& addr);
>@@ -256,9 +257,11 @@ class TurnServer : public sigslot::has_slots<> {
> 
>  private:
>   std::string GenerateNonce(int64_t now) const;
>-  void OnInternalPacket(rtc::AsyncPacketSocket* socket, const char* data,
>-                        size_t size, const rtc::SocketAddress& address,
>-                        const rtc::PacketTime& packet_time);
>+  void OnInternalPacket(rtc::AsyncPacketSocket* socket,
>+                        const char* data,
>+                        size_t size,
>+                        const rtc::SocketAddress& address,
>+                        const int64_t& packet_time_us);
> 
>   void OnNewInternalConnection(rtc::AsyncSocket* socket);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.cc
>deleted file mode 100644
>index c21e5b8769ee6dc66f5a3b724b37f8304e11026d..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.cc
>+++ /dev/null
>@@ -1,133 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "p2p/base/udptransport.h"
>-
>-#include <string>
>-#include <utility>  // For std::move.
>-
>-#include "rtc_base/asyncpacketsocket.h"
>-#include "rtc_base/asyncudpsocket.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/nethelper.h"
>-#include "rtc_base/socketaddress.h"
>-#include "rtc_base/thread.h"
>-#include "rtc_base/thread_checker.h"
>-
>-namespace cricket {
>-
>-UdpTransport::UdpTransport(const std::string& transport_name,
>-                           std::unique_ptr<rtc::AsyncPacketSocket> socket)
>-    : transport_name_(transport_name), socket_(std::move(socket)) {
>-  RTC_DCHECK(socket_);
>-  socket_->SignalReadPacket.connect(this, &UdpTransport::OnSocketReadPacket);
>-  socket_->SignalSentPacket.connect(this, &UdpTransport::OnSocketSentPacket);
>-}
>-
>-UdpTransport::~UdpTransport() {
>-  RTC_DCHECK_RUN_ON(&network_thread_checker_);
>-}
>-
>-rtc::SocketAddress UdpTransport::GetLocalAddress() const {
>-  RTC_DCHECK_RUN_ON(&network_thread_checker_);
>-  return socket_->GetLocalAddress();
>-}
>-
>-bool UdpTransport::SetRemoteAddress(const rtc::SocketAddress& addr) {
>-  RTC_DCHECK_RUN_ON(&network_thread_checker_);
>-  if (!addr.IsComplete()) {
>-    RTC_LOG(LS_WARNING) << "Remote address not complete.";
>-    return false;
>-  }
>-  // TODO(johan): check for ipv4, other settings.
>-  bool prev_destination_nil = remote_address_.IsNil();
>-  remote_address_ = addr;
>-  // Going from "didn't have destination" to "have destination" or vice versa.
>-  if (prev_destination_nil != remote_address_.IsNil()) {
>-    SignalWritableState(this);
>-    if (prev_destination_nil) {
>-      SignalReadyToSend(this);
>-    }
>-  }
>-  return true;
>-}
>-
>-rtc::SocketAddress UdpTransport::GetRemoteAddress() const {
>-  RTC_DCHECK_RUN_ON(&network_thread_checker_);
>-  return remote_address_;
>-}
>-
>-const std::string& UdpTransport::transport_name() const {
>-  return transport_name_;
>-}
>-
>-bool UdpTransport::receiving() const {
>-  // TODO(johan): Implement method and signal.
>-  return true;
>-}
>-
>-bool UdpTransport::writable() const {
>-  RTC_DCHECK_RUN_ON(&network_thread_checker_);
>-  return !remote_address_.IsNil();
>-}
>-
>-int UdpTransport::SendPacket(const char* data,
>-                             size_t len,
>-                             const rtc::PacketOptions& options,
>-                             int flags) {
>-  // No thread_checker in high frequency network function.
>-  if (remote_address_.IsNil()) {
>-    RTC_LOG(LS_WARNING) << "Remote address not set.";
>-    send_error_ = ENOTCONN;
>-    return -1;
>-  }
>-  int result =
>-      socket_->SendTo((const void*)data, len, remote_address_, options);
>-  if (result <= 0) {
>-    RTC_LOG(LS_VERBOSE) << "SendPacket() " << result;
>-  }
>-  return result;
>-}
>-
>-absl::optional<rtc::NetworkRoute> UdpTransport::network_route() const {
>-  rtc::NetworkRoute network_route;
>-  network_route.packet_overhead =
>-      /*kUdpOverhead=*/8 + GetIpOverhead(GetLocalAddress().family());
>-  return absl::optional<rtc::NetworkRoute>(network_route);
>-}
>-
>-int UdpTransport::SetOption(rtc::Socket::Option opt, int value) {
>-  return 0;
>-}
>-
>-int UdpTransport::GetError() {
>-  return send_error_;
>-}
>-
>-rtc::PacketTransportInternal* UdpTransport::GetInternal() {
>-  return this;
>-}
>-
>-void UdpTransport::OnSocketReadPacket(rtc::AsyncPacketSocket* socket,
>-                                      const char* data,
>-                                      size_t len,
>-                                      const rtc::SocketAddress& remote_addr,
>-                                      const rtc::PacketTime& packet_time) {
>-  // No thread_checker in high frequency network function.
>-  SignalReadPacket(this, data, len, packet_time, 0);
>-}
>-
>-void UdpTransport::OnSocketSentPacket(rtc::AsyncPacketSocket* socket,
>-                                      const rtc::SentPacket& packet) {
>-  RTC_DCHECK_EQ(socket_.get(), socket);
>-  SignalSentPacket(this, packet);
>-}
>-
>-}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.h
>deleted file mode 100644
>index 06a795f2e1f9a5041146f8a9aadbc6d699026e4c..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.h
>+++ /dev/null
>@@ -1,89 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef P2P_BASE_UDPTRANSPORT_H_
>-#define P2P_BASE_UDPTRANSPORT_H_
>-
>-#include <memory>
>-#include <string>
>-
>-#include "absl/types/optional.h"
>-#include "api/ortc/udptransportinterface.h"
>-#include "p2p/base/packettransportinternal.h"
>-#include "rtc_base/asyncpacketsocket.h"  // For PacketOptions.
>-#include "rtc_base/thread_checker.h"
>-
>-namespace rtc {
>-class AsyncPacketSocket;
>-struct PacketTime;
>-struct SentPacket;
>-class SocketAddress;
>-}  // namespace rtc
>-
>-namespace cricket {
>-
>-// Implementation of UdpTransportInterface.
>-// Used by OrtcFactory.
>-class UdpTransport : public rtc::PacketTransportInternal,
>-                     public webrtc::UdpTransportInterface {
>- public:
>-  // |transport_name| is only used for identification/logging.
>-  // |socket| must be non-null.
>-  UdpTransport(const std::string& transport_name,
>-               std::unique_ptr<rtc::AsyncPacketSocket> socket);
>-  ~UdpTransport() override;
>-
>-  // Overrides of UdpTransportInterface, used by the API consumer.
>-  rtc::SocketAddress GetLocalAddress() const override;
>-  bool SetRemoteAddress(const rtc::SocketAddress& addr) override;
>-  rtc::SocketAddress GetRemoteAddress() const override;
>-
>-  // Overrides of PacketTransportInternal, used by webrtc internally.
>-  const std::string& transport_name() const override;
>-
>-  bool receiving() const override;
>-
>-  bool writable() const override;
>-
>-  int SendPacket(const char* data,
>-                 size_t len,
>-                 const rtc::PacketOptions& options,
>-                 int flags) override;
>-
>-  int SetOption(rtc::Socket::Option opt, int value) override;
>-
>-  int GetError() override;
>-
>-  absl::optional<rtc::NetworkRoute> network_route() const override;
>-
>- protected:
>-  PacketTransportInternal* GetInternal() override;
>-
>- private:
>-  void OnSocketReadPacket(rtc::AsyncPacketSocket* socket,
>-                          const char* data,
>-                          size_t len,
>-                          const rtc::SocketAddress& remote_addr,
>-                          const rtc::PacketTime& packet_time);
>-  void OnSocketSentPacket(rtc::AsyncPacketSocket* socket,
>-                          const rtc::SentPacket& packet);
>-  bool IsLocalConsistent();
>-
>-  std::string transport_name_;
>-  int send_error_ = 0;
>-  std::unique_ptr<rtc::AsyncPacketSocket> socket_;
>-  // If not set, will be an "nil" address ("IsNil" returns true).
>-  rtc::SocketAddress remote_address_;
>-  rtc::ThreadChecker network_thread_checker_;
>-};
>-
>-}  // namespace cricket
>-
>-#endif  // P2P_BASE_UDPTRANSPORT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport_unittest.cc
>deleted file mode 100644
>index ead366595f416b82ca22e1bdad890ba7857dfbdd..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport_unittest.cc
>+++ /dev/null
>@@ -1,189 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <algorithm>
>-#include <list>
>-#include <memory>
>-#include <utility>
>-#include <vector>
>-
>-#include "p2p/base/basicpacketsocketfactory.h"
>-#include "p2p/base/packettransportinternal.h"
>-#include "p2p/base/udptransport.h"
>-#include "rtc_base/asyncpacketsocket.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/ipaddress.h"
>-#include "rtc_base/socketaddress.h"
>-#include "rtc_base/socketserver.h"
>-#include "rtc_base/thread.h"
>-#include "rtc_base/virtualsocketserver.h"
>-
>-namespace cricket {
>-
>-constexpr int kTimeoutMs = 10000;
>-static const rtc::IPAddress kIPv4LocalHostAddress =
>-    rtc::IPAddress(0x7F000001);  // 127.0.0.1
>-
>-class UdpTransportTest : public testing::Test, public sigslot::has_slots<> {
>- public:
>-  UdpTransportTest()
>-      : virtual_socket_server_(new rtc::VirtualSocketServer()),
>-        network_thread_(virtual_socket_server_.get()),
>-        ep1_("Name1",
>-             std::unique_ptr<rtc::AsyncPacketSocket>(
>-                 socket_factory_.CreateUdpSocket(
>-                     rtc::SocketAddress(rtc::GetAnyIP(AF_INET), 0),
>-                     0,
>-                     0))),
>-        ep2_("Name2",
>-             std::unique_ptr<rtc::AsyncPacketSocket>(
>-                 socket_factory_.CreateUdpSocket(
>-                     rtc::SocketAddress(rtc::GetAnyIP(AF_INET), 0),
>-                     0,
>-                     0))) {
>-    // Setup IP Address for outgoing packets from sockets bound to IPV4
>-    // INADDR_ANY ("0.0.0.0."), as used above when creating the virtual
>-    // sockets. The virtual socket server sends these packets only if the
>-    // default address is explicit set. With a physical socket, the actual
>-    // network stack / operating system would set the IP address for outgoing
>-    // packets.
>-    virtual_socket_server_->SetDefaultRoute(kIPv4LocalHostAddress);
>-  }
>-
>-  struct Endpoint : public sigslot::has_slots<> {
>-    explicit Endpoint(std::string tch_name,
>-                      std::unique_ptr<rtc::AsyncPacketSocket> socket) {
>-      ch_.reset(new UdpTransport(std::move(tch_name), std::move(socket)));
>-      ch_->SignalReadPacket.connect(this, &Endpoint::OnReadPacket);
>-      ch_->SignalSentPacket.connect(this, &Endpoint::OnSentPacket);
>-      ch_->SignalReadyToSend.connect(this, &Endpoint::OnReadyToSend);
>-      ch_->SignalWritableState.connect(this, &Endpoint::OnWritableState);
>-    }
>-
>-    bool CheckData(const char* data, int len) {
>-      bool ret = false;
>-      if (!ch_packets_.empty()) {
>-        std::string packet = ch_packets_.front();
>-        ret = (packet == std::string(data, len));
>-        ch_packets_.pop_front();
>-      }
>-      return ret;
>-    }
>-
>-    void OnWritableState(rtc::PacketTransportInternal* transport) {
>-      num_sig_writable_++;
>-    }
>-
>-    void OnReadyToSend(rtc::PacketTransportInternal* transport) {
>-      num_sig_ready_to_send_++;
>-    }
>-
>-    void OnReadPacket(rtc::PacketTransportInternal* transport,
>-                      const char* data,
>-                      size_t len,
>-                      const rtc::PacketTime& packet_time,
>-                      int flags) {
>-      num_received_packets_++;
>-      RTC_LOG(LS_VERBOSE) << "OnReadPacket (unittest)";
>-      ch_packets_.push_front(std::string(data, len));
>-    }
>-
>-    void OnSentPacket(rtc::PacketTransportInternal* transport,
>-                      const rtc::SentPacket&) {
>-      num_sig_sent_packets_++;
>-    }
>-
>-    int SendData(const char* data, size_t len) {
>-      rtc::PacketOptions options;
>-      return ch_->SendPacket(data, len, options, 0);
>-    }
>-
>-    void GetLocalPort(uint16_t* local_port) {
>-      *local_port = ch_->GetLocalAddress().port();
>-    }
>-
>-    std::list<std::string> ch_packets_;
>-    std::unique_ptr<UdpTransport> ch_;
>-    uint32_t num_received_packets_ = 0;   // Increases on SignalReadPacket.
>-    uint32_t num_sig_sent_packets_ = 0;   // Increases on SignalSentPacket.
>-    uint32_t num_sig_writable_ = 0;       // Increases on SignalWritable.
>-    uint32_t num_sig_ready_to_send_ = 0;  // Increases on SignalReadyToSend.
>-  };
>-
>-  std::unique_ptr<rtc::VirtualSocketServer> virtual_socket_server_;
>-  rtc::AutoSocketServerThread network_thread_;
>-  // Uses current thread's socket server, which will be set by ss_scope_.
>-  rtc::BasicPacketSocketFactory socket_factory_;
>-
>-  Endpoint ep1_;
>-  Endpoint ep2_;
>-
>-  void TestSendRecv() {
>-    for (uint32_t i = 0; i < 5; ++i) {
>-      static const char* data = "ABCDEFGHIJKLMNOPQRSTUVWXYZ1234567890";
>-      int len = static_cast<int>(strlen(data));
>-      // local_channel <==> remote_channel
>-      EXPECT_EQ_WAIT(len, ep1_.SendData(data, len), kTimeoutMs);
>-      EXPECT_TRUE_WAIT(ep2_.CheckData(data, len), kTimeoutMs);
>-      EXPECT_EQ_WAIT(i + 1u, ep2_.num_received_packets_, kTimeoutMs);
>-      EXPECT_EQ_WAIT(len, ep2_.SendData(data, len), kTimeoutMs);
>-      EXPECT_TRUE_WAIT(ep1_.CheckData(data, len), kTimeoutMs);
>-      EXPECT_EQ_WAIT(i + 1u, ep1_.num_received_packets_, kTimeoutMs);
>-    }
>-  }
>-};
>-
>-TEST_F(UdpTransportTest, AddressGetters) {
>-  // Initially, remote address should be nil but local address shouldn't be.
>-  EXPECT_FALSE(ep1_.ch_->GetLocalAddress().IsNil());
>-  EXPECT_TRUE(ep1_.ch_->GetRemoteAddress().IsNil());
>-  rtc::SocketAddress destination("127.0.0.1", 1337);
>-  ASSERT_TRUE(ep1_.ch_->SetRemoteAddress(destination));
>-  EXPECT_EQ(destination, ep1_.ch_->GetRemoteAddress());
>-}
>-
>-// Setting an invalid address should fail and have no effect.
>-TEST_F(UdpTransportTest, SettingIncompleteRemoteAddressFails) {
>-  EXPECT_FALSE(ep1_.ch_->SetRemoteAddress(rtc::SocketAddress("127.0.0.1", 0)));
>-  EXPECT_TRUE(ep1_.ch_->GetRemoteAddress().IsNil());
>-}
>-
>-TEST_F(UdpTransportTest, SendRecvBasic) {
>-  uint16_t port;
>-  ep2_.GetLocalPort(&port);
>-  rtc::SocketAddress addr2 = rtc::SocketAddress("127.0.0.1", port);
>-  EXPECT_TRUE(ep1_.ch_->SetRemoteAddress(addr2));
>-  ep1_.GetLocalPort(&port);
>-  rtc::SocketAddress addr1 = rtc::SocketAddress("127.0.0.1", port);
>-  EXPECT_TRUE(ep2_.ch_->SetRemoteAddress(addr1));
>-  TestSendRecv();
>-}
>-
>-// Test the signals and state methods used internally by causing a UdpTransport
>-// to send a packet to itself.
>-TEST_F(UdpTransportTest, StatusAndSignals) {
>-  EXPECT_EQ(0u, ep1_.num_sig_writable_);
>-  EXPECT_EQ(0u, ep1_.num_sig_ready_to_send_);
>-  // Loopback
>-  EXPECT_TRUE(!ep1_.ch_->writable());
>-  rtc::SocketAddress addr = ep1_.ch_->GetLocalAddress();
>-  // Keep port, but explicitly set IP.
>-  addr.SetIP("127.0.0.1");
>-  ep1_.ch_->SetRemoteAddress(addr);
>-  EXPECT_TRUE(ep1_.ch_->writable());
>-  EXPECT_EQ(1u, ep1_.num_sig_writable_);
>-  EXPECT_EQ(1u, ep1_.num_sig_ready_to_send_);
>-  const char data[] = "abc";
>-  ep1_.SendData(data, sizeof(data));
>-  EXPECT_EQ_WAIT(1u, ep1_.ch_packets_.size(), kTimeoutMs);
>-  EXPECT_EQ_WAIT(1u, ep1_.num_sig_sent_packets_, kTimeoutMs);
>-}
>-
>-}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.cc
>index c1b1de58ddb60ff278ae53a1c863c994003b2466..0c2fef3112605796b683c5edd5801dfe7478c677 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.cc
>@@ -518,6 +518,10 @@ void BasicPortAllocatorSession::GetCandidatesFromPort(
>   }
> }
> 
>+bool BasicPortAllocatorSession::MdnsObfuscationEnabled() const {
>+  return allocator_->network_manager()->GetMdnsResponder() != nullptr;
>+}
>+
> Candidate BasicPortAllocatorSession::SanitizeCandidate(
>     const Candidate& c) const {
>   RTC_DCHECK_RUN_ON(network_thread_);
>@@ -534,7 +538,7 @@ Candidate BasicPortAllocatorSession::SanitizeCandidate(
>   bool filter_stun_related_address =
>       ((flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION) &&
>        (flags() & PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE)) ||
>-      !(candidate_filter_ & CF_HOST);
>+      !(candidate_filter_ & CF_HOST) || MdnsObfuscationEnabled();
>   // If the candidate filter doesn't allow reflexive addresses, empty TURN raddr
>   // to avoid reflexive address leakage.
>   bool filter_turn_related_address = !(candidate_filter_ & CF_REFLEXIVE);
>@@ -1362,7 +1366,7 @@ void AllocationSequence::CreateUDPPorts() {
> 
>   // TODO(mallinath) - Remove UDPPort creating socket after shared socket
>   // is enabled completely.
>-  UDPPort* port = NULL;
>+  std::unique_ptr<UDPPort> port;
>   bool emit_local_candidate_for_anyaddress =
>       !IsFlagSet(PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE);
>   if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET) && udp_socket_) {
>@@ -1384,7 +1388,7 @@ void AllocationSequence::CreateUDPPorts() {
>     // If shared socket is enabled, STUN candidate will be allocated by the
>     // UDPPort.
>     if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET)) {
>-      udp_port_ = port;
>+      udp_port_ = port.get();
>       port->SignalDestroyed.connect(this, &AllocationSequence::OnPortDestroyed);
> 
>       // If STUN is not disabled, setting stun server address to port.
>@@ -1398,7 +1402,7 @@ void AllocationSequence::CreateUDPPorts() {
>       }
>     }
> 
>-    session_->AddAllocatedPort(port, this, true);
>+    session_->AddAllocatedPort(port.release(), this, true);
>   }
> }
> 
>@@ -1408,13 +1412,13 @@ void AllocationSequence::CreateTCPPorts() {
>     return;
>   }
> 
>-  Port* port = TCPPort::Create(
>+  std::unique_ptr<Port> port = TCPPort::Create(
>       session_->network_thread(), session_->socket_factory(), network_,
>       session_->allocator()->min_port(), session_->allocator()->max_port(),
>       session_->username(), session_->password(),
>       session_->allocator()->allow_tcp_listen());
>   if (port) {
>-    session_->AddAllocatedPort(port, this, true);
>+    session_->AddAllocatedPort(port.release(), this, true);
>     // Since TCPPort is not created using shared socket, |port| will not be
>     // added to the dequeue.
>   }
>@@ -1436,14 +1440,14 @@ void AllocationSequence::CreateStunPorts() {
>     return;
>   }
> 
>-  StunPort* port = StunPort::Create(
>+  std::unique_ptr<StunPort> port = StunPort::Create(
>       session_->network_thread(), session_->socket_factory(), network_,
>       session_->allocator()->min_port(), session_->allocator()->max_port(),
>       session_->username(), session_->password(), config_->StunServers(),
>       session_->allocator()->origin(),
>       session_->allocator()->stun_candidate_keepalive_interval());
>   if (port) {
>-    session_->AddAllocatedPort(port, this, true);
>+    session_->AddAllocatedPort(port.release(), this, true);
>     // Since StunPort is not created using shared socket, |port| will not be
>     // added to the dequeue.
>   }
>@@ -1479,11 +1483,12 @@ void AllocationSequence::CreateRelayPorts() {
> 
> void AllocationSequence::CreateGturnPort(const RelayServerConfig& config) {
>   // TODO(mallinath) - Rename RelayPort to GTurnPort.
>-  RelayPort* port = RelayPort::Create(
>+  std::unique_ptr<RelayPort> port = RelayPort::Create(
>       session_->network_thread(), session_->socket_factory(), network_,
>       session_->allocator()->min_port(), session_->allocator()->max_port(),
>       config_->username, config_->password);
>   if (port) {
>+    RelayPort* port_ptr = port.release();
>     // Since RelayPort is not created using shared socket, |port| will not be
>     // added to the dequeue.
>     // Note: We must add the allocated port before we add addresses because
>@@ -1491,17 +1496,17 @@ void AllocationSequence::CreateGturnPort(const RelayServerConfig& config) {
>     //       settings.  However, we also can't prepare the address (normally
>     //       done by AddAllocatedPort) until we have these addresses.  So we
>     //       wait to do that until below.
>-    session_->AddAllocatedPort(port, this, false);
>+    session_->AddAllocatedPort(port_ptr, this, false);
> 
>     // Add the addresses of this protocol.
>     PortList::const_iterator relay_port;
>     for (relay_port = config.ports.begin(); relay_port != config.ports.end();
>          ++relay_port) {
>-      port->AddServerAddress(*relay_port);
>-      port->AddExternalAddress(*relay_port);
>+      port_ptr->AddServerAddress(*relay_port);
>+      port_ptr->AddExternalAddress(*relay_port);
>     }
>     // Start fetching an address for this port.
>-    port->PrepareAddress();
>+    port_ptr->PrepareAddress();
>   }
> }
> 
>@@ -1579,7 +1584,7 @@ void AllocationSequence::OnReadPacket(rtc::AsyncPacketSocket* socket,
>                                       const char* data,
>                                       size_t size,
>                                       const rtc::SocketAddress& remote_addr,
>-                                      const rtc::PacketTime& packet_time) {
>+                                      const int64_t& packet_time_us) {
>   RTC_DCHECK(socket == udp_socket_.get());
> 
>   bool turn_port_found = false;
>@@ -1593,7 +1598,7 @@ void AllocationSequence::OnReadPacket(rtc::AsyncPacketSocket* socket,
>   for (auto* port : relay_ports_) {
>     if (port->CanHandleIncomingPacketsFrom(remote_addr)) {
>       if (port->HandleIncomingPacket(socket, data, size, remote_addr,
>-                                     packet_time)) {
>+                                     packet_time_us)) {
>         return;
>       }
>       turn_port_found = true;
>@@ -1609,7 +1614,7 @@ void AllocationSequence::OnReadPacket(rtc::AsyncPacketSocket* socket,
>         stun_servers.find(remote_addr) != stun_servers.end()) {
>       RTC_DCHECK(udp_port_->SharedSocket());
>       udp_port_->HandleIncomingPacket(socket, data, size, remote_addr,
>-                                      packet_time);
>+                                      packet_time_us);
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.h
>index 8b951cae9e62cbf7eec1872b4b99fcc3289d458f..672f3ddb7c263b1a55fa31422b9f103a5be7b601 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.h
>@@ -22,11 +22,12 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/messagequeue.h"
> #include "rtc_base/network.h"
>+#include "rtc_base/system/rtc_export.h"
> #include "rtc_base/thread.h"
> 
> namespace cricket {
> 
>-class BasicPortAllocator : public PortAllocator {
>+class RTC_EXPORT BasicPortAllocator : public PortAllocator {
>  public:
>   // note: The (optional) relay_port_factory is owned by caller
>   // and must have a life time that exceeds that of BasicPortAllocator.
>@@ -110,8 +111,8 @@ enum class SessionState {
>               // process will be started.
> };
> 
>-class BasicPortAllocatorSession : public PortAllocatorSession,
>-                                  public rtc::MessageHandler {
>+class RTC_EXPORT BasicPortAllocatorSession : public PortAllocatorSession,
>+                                             public rtc::MessageHandler {
>  public:
>   BasicPortAllocatorSession(BasicPortAllocator* allocator,
>                             const std::string& content_name,
>@@ -235,6 +236,10 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
> 
>   bool CheckCandidateFilter(const Candidate& c) const;
>   bool CandidatePairable(const Candidate& c, const Port* port) const;
>+
>+  // Returns true if there is an mDNS responder attached to the network manager
>+  bool MdnsObfuscationEnabled() const;
>+
>   // Clears 1) the address if the candidate is supposedly a hostname candidate;
>   // 2) the related address according to the flags and candidate filter in order
>   // to avoid leaking any information.
>@@ -274,7 +279,7 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
> 
> // Records configuration information useful in creating ports.
> // TODO(deadbeef): Rename "relay" to "turn_server" in this struct.
>-struct PortConfiguration : public rtc::MessageData {
>+struct RTC_EXPORT PortConfiguration : public rtc::MessageData {
>   // TODO(jiayl): remove |stun_address| when Chrome is updated.
>   rtc::SocketAddress stun_address;
>   ServerAddresses stun_servers;
>@@ -383,7 +388,7 @@ class AllocationSequence : public rtc::MessageHandler,
>                     const char* data,
>                     size_t size,
>                     const rtc::SocketAddress& remote_addr,
>-                    const rtc::PacketTime& packet_time);
>+                    const int64_t& packet_time_us);
> 
>   void OnPortDestroyed(PortInterface* port);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator_unittest.cc
>index a46e53ba9da6620e7b465d21ad2a59471e98821c..8943ebc73dba2db68a2fbcdbe40735786f4eaf0f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator_unittest.cc
>@@ -38,7 +38,6 @@
> 
> using rtc::IPAddress;
> using rtc::SocketAddress;
>-using rtc::Thread;
> 
> #define MAYBE_SKIP_IPV4                        \
>   if (!rtc::HasIPv4Enabled()) {                \
>@@ -147,15 +146,15 @@ class BasicPortAllocatorTestBase : public testing::Test,
>         // must be called.
>         nat_factory_(vss_.get(), kNatUdpAddr, kNatTcpAddr),
>         nat_socket_factory_(new rtc::BasicPacketSocketFactory(&nat_factory_)),
>-        stun_server_(TestStunServer::Create(Thread::Current(), kStunAddr)),
>-        relay_server_(Thread::Current(),
>+        stun_server_(TestStunServer::Create(rtc::Thread::Current(), kStunAddr)),
>+        relay_server_(rtc::Thread::Current(),
>                       kRelayUdpIntAddr,
>                       kRelayUdpExtAddr,
>                       kRelayTcpIntAddr,
>                       kRelayTcpExtAddr,
>                       kRelaySslTcpIntAddr,
>                       kRelaySslTcpExtAddr),
>-        turn_server_(Thread::Current(), kTurnUdpIntAddr, kTurnUdpExtAddr),
>+        turn_server_(rtc::Thread::Current(), kTurnUdpIntAddr, kTurnUdpExtAddr),
>         candidate_allocation_done_(false) {
>     ServerAddresses stun_servers;
>     stun_servers.insert(kStunAddr);
>@@ -2247,8 +2246,8 @@ TEST_F(BasicPortAllocatorTest, IceRegatheringMetricsLoggedWhenNetworkChanges) {
> }
> 
> // Test that when an mDNS responder is present, the local address of a host
>-// candidate is masked by an mDNS hostname and the related address of any other
>-// type of candidates is set to 0.0.0.0 or ::0.
>+// candidate is concealed by an mDNS hostname and the related address of a srflx
>+// candidate is set to 0.0.0.0 or ::0.
> TEST_F(BasicPortAllocatorTest, HostCandidateAddressIsReplacedByHostname) {
>   // Default config uses GTURN and no NAT, so replace that with the
>   // desired setup (NAT, STUN server, TURN server, UDP/TCP).
>@@ -2258,7 +2257,7 @@ TEST_F(BasicPortAllocatorTest, HostCandidateAddressIsReplacedByHostname) {
>   AddTurnServers(kTurnUdpIntIPv6Addr, kTurnTcpIntIPv6Addr);
> 
>   ASSERT_EQ(&network_manager_, allocator().network_manager());
>-  network_manager_.CreateMDnsResponder();
>+  network_manager_.CreateMdnsResponder();
>   AddInterface(kClientAddr);
>   ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>@@ -2270,23 +2269,29 @@ TEST_F(BasicPortAllocatorTest, HostCandidateAddressIsReplacedByHostname) {
>   int num_srflx_candidates = 0;
>   int num_relay_candidates = 0;
>   for (const auto& candidate : candidates_) {
>+    const auto& raddr = candidate.related_address();
>+
>     if (candidate.type() == LOCAL_PORT_TYPE) {
>-      EXPECT_TRUE(candidate.address().IsUnresolvedIP());
>+      EXPECT_FALSE(candidate.address().hostname().empty());
>+      EXPECT_TRUE(raddr.IsNil());
>       if (candidate.protocol() == UDP_PROTOCOL_NAME) {
>         ++num_host_udp_candidates;
>       } else {
>         ++num_host_tcp_candidates;
>       }
>+    } else if (candidate.type() == STUN_PORT_TYPE) {
>+      // For a srflx candidate, the related address should be set to 0.0.0.0 or
>+      // ::0
>+      EXPECT_TRUE(IPIsAny(raddr.ipaddr()));
>+      EXPECT_EQ(raddr.port(), 0);
>+      ++num_srflx_candidates;
>+    } else if (candidate.type() == RELAY_PORT_TYPE) {
>+      EXPECT_EQ(kNatUdpAddr.ipaddr(), raddr.ipaddr());
>+      EXPECT_EQ(kNatUdpAddr.family(), raddr.family());
>+      ++num_relay_candidates;
>     } else {
>-      EXPECT_NE(PRFLX_PORT_TYPE, candidate.type());
>-      // The related address should be set to 0.0.0.0 or ::0 for srflx and
>-      // relay candidates.
>-      EXPECT_EQ(rtc::SocketAddress(), candidate.related_address());
>-      if (candidate.type() == STUN_PORT_TYPE) {
>-        ++num_srflx_candidates;
>-      } else {
>-        ++num_relay_candidates;
>-      }
>+      // prflx candidates are not expected
>+      FAIL();
>     }
>   }
>   EXPECT_EQ(1, num_host_udp_candidates);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.cc
>index 6404134c4a19d8c5e1a0ee734a6b766e0fb4a778..c0c172076bc1eb7d22f75931a25a90504572c729 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.cc
>@@ -11,6 +11,7 @@
> #include "p2p/client/turnportfactory.h"
> 
> #include <memory>
>+#include <utility>
> 
> #include "p2p/base/turnport.h"
> 
>@@ -21,26 +22,26 @@ TurnPortFactory::~TurnPortFactory() {}
> std::unique_ptr<Port> TurnPortFactory::Create(
>     const CreateRelayPortArgs& args,
>     rtc::AsyncPacketSocket* udp_socket) {
>-  TurnPort* port = TurnPort::Create(
>+  auto port = TurnPort::CreateUnique(
>       args.network_thread, args.socket_factory, args.network, udp_socket,
>       args.username, args.password, *args.server_address,
>       args.config->credentials, args.config->priority, args.origin,
>       args.turn_customizer);
>   port->SetTlsCertPolicy(args.config->tls_cert_policy);
>-  return std::unique_ptr<Port>(port);
>+  return std::move(port);
> }
> 
> std::unique_ptr<Port> TurnPortFactory::Create(const CreateRelayPortArgs& args,
>                                               int min_port,
>                                               int max_port) {
>-  TurnPort* port = TurnPort::Create(
>+  auto port = TurnPort::CreateUnique(
>       args.network_thread, args.socket_factory, args.network, min_port,
>       max_port, args.username, args.password, *args.server_address,
>       args.config->credentials, args.config->priority, args.origin,
>       args.config->tls_alpn_protocols, args.config->tls_elliptic_curves,
>       args.turn_customizer, args.config->tls_cert_verifier);
>   port->SetTlsCertPolicy(args.config->tls_cert_policy);
>-  return std::unique_ptr<Port>(port);
>+  return std::move(port);
> }
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.cc
>index 51506b670d7518af371ad0acc0026ae2a3657716..9778416a1b0091205d27bddc67dfa5875c2a5c25 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.cc
>@@ -77,7 +77,7 @@ class StunProber::Requester : public sigslot::has_slots<> {
>                               const char* buf,
>                               size_t size,
>                               const rtc::SocketAddress& addr,
>-                              const rtc::PacketTime& time);
>+                              const int64_t& packet_time_us);
> 
>   const std::vector<Request*>& requests() { return requests_; }
> 
>@@ -204,7 +204,7 @@ void StunProber::Requester::OnStunResponseReceived(
>     const char* buf,
>     size_t size,
>     const rtc::SocketAddress& addr,
>-    const rtc::PacketTime& time) {
>+    const int64_t& /* packet_time_us */) {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   RTC_DCHECK(socket_);
>   Request* request = GetRequestByAddress(addr.ipaddr());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.h
>index 9f1d42f32ce37b1bb2e5638772a64a8b61a13a71..a4feecd8132a9f35b49cb978c33b64550ee66cea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.h
>@@ -22,6 +22,7 @@
> #include "rtc_base/ipaddress.h"
> #include "rtc_base/network.h"
> #include "rtc_base/socketaddress.h"
>+#include "rtc_base/system/rtc_export.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/thread_checker.h"
> 
>@@ -49,7 +50,7 @@ enum NatType {
>   NATTYPE_NON_SYMMETRIC  // Behind a non-symmetric NAT.
> };
> 
>-class StunProber : public sigslot::has_slots<> {
>+class RTC_EXPORT StunProber : public sigslot::has_slots<> {
>  public:
>   enum Status {       // Used in UMA_HISTOGRAM_ENUMERATION.
>     SUCCESS,          // Successfully received bytes from the server.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/BUILD.gn
>index 4e6e55810ad9c20b849de756c973747a7e638a36..65ab33a50b497d2dc2036c3650628b8d748bf8a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/BUILD.gn
>@@ -31,6 +31,7 @@ rtc_static_library("rtc_pc_base") {
>   sources = [
>     "channel.cc",
>     "channel.h",
>+    "channelinterface.h",
>     "channelmanager.cc",
>     "channelmanager.h",
>     "dtlssrtptransport.cc",
>@@ -87,6 +88,7 @@ rtc_static_library("rtc_pc_base") {
>     "../rtc_base/third_party/sigslot",
>     "../system_wrappers:metrics",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>@@ -203,6 +205,7 @@ rtc_static_library("peerconnection") {
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:stringutils",
>     "../rtc_base/experiments:congestion_controller_experiment",
>+    "../rtc_base/system:rtc_export",
>     "../rtc_base/third_party/base64",
>     "../rtc_base/third_party/sigslot",
>     "../stats",
>@@ -210,62 +213,11 @@ rtc_static_library("peerconnection") {
>     "../system_wrappers:field_trial",
>     "../system_wrappers:metrics",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>-rtc_static_library("builtin_video_bitrate_allocator_factory") {
>-  sources = [
>-    "builtin_video_bitrate_allocator_factory.cc",
>-    "builtin_video_bitrate_allocator_factory.h",
>-  ]
>-
>-  deps = [
>-    "../api/video:video_bitrate_allocator_factory",
>-    "../media:rtc_media_base",
>-    "../modules/video_coding:video_coding_utility",
>-    "../modules/video_coding:webrtc_vp9_helpers",
>-    "../rtc_base:ptr_util",
>-    "../rtc_base/system:fallthrough",
>-    "//third_party/abseil-cpp/absl/memory",
>-  ]
>-}
>-
>-# This target implements CreatePeerConnectionFactory methods that will create a
>-# PeerConnection will full functionality (audio, video and data). Applications
>-# that wish to reduce their binary size by ommitting functionality they don't
>-# need should use CreateModularCreatePeerConnectionFactory instead, using the
>-# "peerconnection" build target and other targets specific to their
>-# requrements. See comment in peerconnectionfactoryinterface.h.
>-rtc_static_library("create_pc_factory") {
>-  sources = [
>-    "createpeerconnectionfactory.cc",
>-  ]
>-
>-  deps = [
>-    "../api:callfactory_api",
>-    "../api:libjingle_peerconnection_api",
>-    "../api/audio:audio_mixer_api",
>-    "../api/audio_codecs:audio_codecs_api",
>-    "../api/video_codecs:video_codecs_api",
>-    "../call",
>-    "../call:call_interfaces",
>-    "../logging:rtc_event_log_api",
>-    "../logging:rtc_event_log_impl_base",
>-    "../media:rtc_audio_video",
>-    "../media:rtc_media_base",
>-    "../modules/audio_device:audio_device",
>-    "../modules/audio_processing:audio_processing",
>-    "../rtc_base:rtc_base",
>-    "../rtc_base:rtc_base_approved",
>-  ]
>-
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-}
>-
> rtc_source_set("libjingle_peerconnection") {
>   visibility = [ "*" ]
>   allow_poison = [
>@@ -273,8 +225,8 @@ rtc_source_set("libjingle_peerconnection") {
>     "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>   ]
>   deps = [
>-    ":create_pc_factory",
>     ":peerconnection",
>+    "../api:create_peerconnection_factory",
>     "../api:libjingle_peerconnection_api",
>   ]
> }
>@@ -293,6 +245,7 @@ if (rtc_include_tests) {
>       "rtcpmuxfilter_unittest.cc",
>       "rtptransport_unittest.cc",
>       "rtptransporttestutil.h",
>+      "sessiondescription_unittest.cc",
>       "srtpfilter_unittest.cc",
>       "srtpsession_unittest.cc",
>       "srtptestutil.h",
>@@ -316,6 +269,7 @@ if (rtc_include_tests) {
>       ":rtc_pc",
>       ":rtc_pc_base",
>       "../api:array_view",
>+      "../api:fake_media_transport",
>       "../api:libjingle_peerconnection_api",
>       "../call:rtp_interfaces",
>       "../logging:rtc_event_log_api",
>@@ -325,6 +279,7 @@ if (rtc_include_tests) {
>       "../p2p:p2p_test_utils",
>       "../p2p:rtc_p2p",
>       "../rtc_base:checks",
>+      "../rtc_base:gunit_helpers",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_main",
>@@ -353,6 +308,7 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       ":pc_test_utils",
>+      "../api:create_peerconnection_factory",
>       "../api:libjingle_peerconnection_api",
>       "../api:rtc_stats_api",
>       "../api/audio_codecs:builtin_audio_decoder_factory",
>@@ -363,6 +319,7 @@ if (rtc_include_tests) {
>       "../p2p:p2p_test_utils",
>       "../p2p:rtc_p2p",
>       "../pc:peerconnection",
>+      "../rtc_base:gunit_helpers",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>@@ -391,6 +348,7 @@ if (rtc_include_tests) {
>       "test/fakevideotrackrenderer.h",
>       "test/fakevideotracksource.h",
>       "test/framegeneratorcapturervideotracksource.h",
>+      "test/mock_channelinterface.h",
>       "test/mock_datachannel.h",
>       "test/mock_rtpreceiverinternal.h",
>       "test/mock_rtpsenderinternal.h",
>@@ -406,6 +364,7 @@ if (rtc_include_tests) {
>       ":peerconnection",
>       ":rtc_pc_base",
>       "..:webrtc_common",
>+      "../api:create_peerconnection_factory",
>       "../api:libjingle_peerconnection_api",
>       "../api:libjingle_peerconnection_test_api",
>       "../api:rtc_stats_api",
>@@ -419,12 +378,13 @@ if (rtc_include_tests) {
>       "../media:rtc_media_base",
>       "../media:rtc_media_tests_utils",
>       "../modules/audio_device:audio_device",
>+      "../modules/audio_processing:api",
>       "../modules/audio_processing:audio_processing",
>       "../p2p:p2p_test_utils",
>       "../rtc_base:checks",
>+      "../rtc_base:gunit_helpers",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>-      "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:rtc_task_queue",
>       "../rtc_base/third_party/sigslot",
>       "../test:test_support",
>@@ -470,6 +430,7 @@ if (rtc_include_tests) {
>       "rtpmediautils_unittest.cc",
>       "rtpparametersconversion_unittest.cc",
>       "rtpsenderreceiver_unittest.cc",
>+      "rtptransceiver_unittest.cc",
>       "sctputils_unittest.cc",
>       "statscollector_unittest.cc",
>       "test/fakeaudiocapturemodule_unittest.cc",
>@@ -492,17 +453,23 @@ if (rtc_include_tests) {
>     deps = [
>       ":peerconnection",
>       ":rtc_pc_base",
>-      "../api:fake_frame_crypto",
>+      "../api:create_peerconnection_factory",
>+      "../api:fake_frame_decryptor",
>+      "../api:fake_frame_encryptor",
>       "../api:libjingle_peerconnection_api",
>+      "../api:loopback_media_transport",
>       "../api:mock_rtp",
>       "../api/units:time_delta",
>       "../logging:fake_rtc_event_log",
>       "../rtc_base:checks",
>+      "../rtc_base:gunit_helpers",
>+      "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:stringutils",
>       "../rtc_base/third_party/base64",
>       "../system_wrappers:metrics",
>       "../test:fileutils",
>       "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/strings",
>     ]
>     if (is_android) {
>       deps += [ ":android_black_magic" ]
>@@ -513,6 +480,7 @@ if (rtc_include_tests) {
>       ":pc_test_utils",
>       "..:webrtc_common",
>       "../api:callfactory_api",
>+      "../api:fake_media_transport",
>       "../api:libjingle_peerconnection_test_api",
>       "../api:rtc_stats_api",
>       "../api/audio_codecs:audio_codecs_api",
>@@ -531,6 +499,7 @@ if (rtc_include_tests) {
>       "../media:rtc_data",  # TODO(phoglund): AFAIK only used for one sctp constant.
>       "../media:rtc_media_base",
>       "../media:rtc_media_tests_utils",
>+      "../modules/audio_processing:api",
>       "../modules/audio_processing:audio_processing",
>       "../modules/utility:utility",
>       "../p2p:p2p_test_utils",
>@@ -539,7 +508,6 @@ if (rtc_include_tests) {
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_main",
>-      "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:rtc_task_queue",
>       "../rtc_base:safe_conversions",
>       "../test:audio_codec_mocks",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/builtin_video_bitrate_allocator_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/builtin_video_bitrate_allocator_factory.cc
>deleted file mode 100644
>index 46d7daea6fa98af7f00a9bfff415f999f8b79e34..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/builtin_video_bitrate_allocator_factory.cc
>+++ /dev/null
>@@ -1,56 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "pc/builtin_video_bitrate_allocator_factory.h"
>-
>-#include "absl/memory/memory.h"
>-#include "media/base/codec.h"
>-#include "modules/video_coding/codecs/vp9/svc_rate_allocator.h"
>-#include "modules/video_coding/utility/default_video_bitrate_allocator.h"
>-#include "modules/video_coding/utility/simulcast_rate_allocator.h"
>-#include "rtc_base/system/fallthrough.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-
>-class BuiltinVideoBitrateAllocatorFactory
>-    : public VideoBitrateAllocatorFactory {
>- public:
>-  BuiltinVideoBitrateAllocatorFactory() = default;
>-  ~BuiltinVideoBitrateAllocatorFactory() override = default;
>-
>-  std::unique_ptr<VideoBitrateAllocator> CreateVideoBitrateAllocator(
>-      const VideoCodec& codec) override {
>-    std::unique_ptr<VideoBitrateAllocator> rate_allocator;
>-    switch (codec.codecType) {
>-      case kVideoCodecVP8:
>-        RTC_FALLTHROUGH();
>-      case kVideoCodecH264:
>-        rate_allocator.reset(new SimulcastRateAllocator(codec));
>-        break;
>-      case kVideoCodecVP9:
>-        rate_allocator.reset(new SvcRateAllocator(codec));
>-        break;
>-      default:
>-        rate_allocator.reset(new DefaultVideoBitrateAllocator(codec));
>-    }
>-    return rate_allocator;
>-  }
>-};
>-
>-}  // namespace
>-
>-std::unique_ptr<VideoBitrateAllocatorFactory>
>-CreateBuiltinVideoBitrateAllocatorFactory() {
>-  return absl::make_unique<BuiltinVideoBitrateAllocatorFactory>();
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/builtin_video_bitrate_allocator_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/builtin_video_bitrate_allocator_factory.h
>deleted file mode 100644
>index 60f2afcbaba127814d635bae42f003ad3020fe47..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/builtin_video_bitrate_allocator_factory.h
>+++ /dev/null
>@@ -1,25 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef PC_BUILTIN_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>-#define PC_BUILTIN_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>-
>-#include <memory>
>-
>-#include "api/video/video_bitrate_allocator_factory.h"
>-
>-namespace webrtc {
>-
>-std::unique_ptr<VideoBitrateAllocatorFactory>
>-CreateBuiltinVideoBitrateAllocatorFactory();
>-
>-}  // namespace webrtc
>-
>-#endif  // PC_BUILTIN_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.cc
>deleted file mode 100644
>index 201d2bf95cab547162092d9197e7910981ceefd6..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.cc
>+++ /dev/null
>@@ -1,47 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "pc/bundlefilter.h"
>-
>-#include "media/base/rtputils.h"
>-#include "rtc_base/logging.h"
>-
>-namespace cricket {
>-
>-BundleFilter::BundleFilter() {}
>-
>-BundleFilter::~BundleFilter() {}
>-
>-bool BundleFilter::DemuxPacket(const uint8_t* data, size_t len) {
>-  // For RTP packets, we check whether the payload type can be found.
>-  if (!IsRtpPacket(data, len)) {
>-    return false;
>-  }
>-
>-  int payload_type = 0;
>-  if (!GetRtpPayloadType(data, len, &payload_type)) {
>-    return false;
>-  }
>-  return FindPayloadType(payload_type);
>-}
>-
>-void BundleFilter::AddPayloadType(int payload_type) {
>-  payload_types_.insert(payload_type);
>-}
>-
>-bool BundleFilter::FindPayloadType(int pl_type) const {
>-  return payload_types_.find(pl_type) != payload_types_.end();
>-}
>-
>-void BundleFilter::ClearAllPayloadTypes() {
>-  payload_types_.clear();
>-}
>-
>-}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.h
>deleted file mode 100644
>index de477f457f2948bb4341573eabdce2d1ff899c50..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.h
>+++ /dev/null
>@@ -1,54 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef PC_BUNDLEFILTER_H_
>-#define PC_BUNDLEFILTER_H_
>-
>-#include <stdint.h>
>-
>-#include <set>
>-#include <vector>
>-
>-#include "media/base/streamparams.h"
>-//#include "rtc_base/basictypes.h"
>-
>-namespace cricket {
>-
>-// In case of single RTP session and single transport channel, all session
>-// (or media) channels share a common transport channel. Hence they all get
>-// SignalReadPacket when packet received on transport channel. This requires
>-// cricket::BaseChannel to know all the valid sources, else media channel
>-// will decode invalid packets.
>-//
>-// This class determines whether a packet is destined for cricket::BaseChannel.
>-// This is only to be used for RTP packets as RTCP packets are not filtered.
>-// For RTP packets, this is decided based on the payload type.
>-class BundleFilter {
>- public:
>-  BundleFilter();
>-  ~BundleFilter();
>-
>-  // Determines if a RTP packet belongs to valid cricket::BaseChannel.
>-  bool DemuxPacket(const uint8_t* data, size_t len);
>-
>-  // Adds the supported payload type.
>-  void AddPayloadType(int payload_type);
>-
>-  // Public for unittests.
>-  bool FindPayloadType(int pl_type) const;
>-  void ClearAllPayloadTypes();
>-
>- private:
>-  std::set<int> payload_types_;
>-};
>-
>-}  // namespace cricket
>-
>-#endif  // PC_BUNDLEFILTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter_unittest.cc
>deleted file mode 100644
>index e7cadbfbac613c6429a8be277e21f375cfc4bd71..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter_unittest.cc
>+++ /dev/null
>@@ -1,73 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "pc/bundlefilter.h"
>-#include "rtc_base/gunit.h"
>-
>-using cricket::StreamParams;
>-
>-static const int kPayloadType1 = 0x11;
>-static const int kPayloadType2 = 0x22;
>-static const int kPayloadType3 = 0x33;
>-
>-// SSRC = 0x1111, Payload type = 0x11
>-static const unsigned char kRtpPacketPt1Ssrc1[] = {
>-    0x80, kPayloadType1, 0x00, 0x01, 0x00, 0x00,
>-    0x00, 0x00,          0x00, 0x00, 0x11, 0x11,
>-};
>-
>-// SSRC = 0x2222, Payload type = 0x22
>-static const unsigned char kRtpPacketPt2Ssrc2[] = {
>-    0x80, 0x80 + kPayloadType2,
>-    0x00, 0x01,
>-    0x00, 0x00,
>-    0x00, 0x00,
>-    0x00, 0x00,
>-    0x22, 0x22,
>-};
>-
>-// SSRC = 0x2222, Payload type = 0x33
>-static const unsigned char kRtpPacketPt3Ssrc2[] = {
>-    0x80, kPayloadType3, 0x00, 0x01, 0x00, 0x00,
>-    0x00, 0x00,          0x00, 0x00, 0x22, 0x22,
>-};
>-
>-// An SCTP packet.
>-static const unsigned char kSctpPacket[] = {
>-    0x00, 0x01, 0x00, 0x01, 0xff, 0xff, 0xff, 0xff, 0x00, 0x00,
>-    0x00, 0x00, 0x03, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00,
>-};
>-
>-TEST(BundleFilterTest, RtpPacketTest) {
>-  cricket::BundleFilter bundle_filter;
>-  bundle_filter.AddPayloadType(kPayloadType1);
>-  EXPECT_TRUE(bundle_filter.DemuxPacket(kRtpPacketPt1Ssrc1,
>-                                        sizeof(kRtpPacketPt1Ssrc1)));
>-  bundle_filter.AddPayloadType(kPayloadType2);
>-  EXPECT_TRUE(bundle_filter.DemuxPacket(kRtpPacketPt2Ssrc2,
>-                                        sizeof(kRtpPacketPt2Ssrc2)));
>-
>-  // Payload type 0x33 is not added.
>-  EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt3Ssrc2,
>-                                         sizeof(kRtpPacketPt3Ssrc2)));
>-  // Size is too small.
>-  EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt1Ssrc1, 11));
>-
>-  bundle_filter.ClearAllPayloadTypes();
>-  EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt1Ssrc1,
>-                                         sizeof(kRtpPacketPt1Ssrc1)));
>-  EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt2Ssrc2,
>-                                         sizeof(kRtpPacketPt2Ssrc2)));
>-}
>-
>-TEST(BundleFilterTest, InvalidRtpPacket) {
>-  cricket::BundleFilter bundle_filter;
>-  EXPECT_FALSE(bundle_filter.DemuxPacket(kSctpPacket, sizeof(kSctpPacket)));
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.cc
>index b63cea7cecf3021c362a7a72d7846c7e2132acfa..67d698487ce4f89bcd2f9236b7cef024f6a9d4b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.cc
>@@ -93,6 +93,7 @@ void RtpSendParametersFromMediaDescription(
>     RtpSendParameters<Codec>* send_params) {
>   RtpParametersFromMediaDescription(desc, extensions, send_params);
>   send_params->max_bandwidth_bps = desc->bandwidth();
>+  send_params->extmap_allow_mixed = desc->extmap_allow_mixed();
> }
> 
> BaseChannel::BaseChannel(rtc::Thread* worker_thread,
>@@ -101,7 +102,7 @@ BaseChannel::BaseChannel(rtc::Thread* worker_thread,
>                          std::unique_ptr<MediaChannel> media_channel,
>                          const std::string& content_name,
>                          bool srtp_required,
>-                         rtc::CryptoOptions crypto_options)
>+                         webrtc::CryptoOptions crypto_options)
>     : worker_thread_(worker_thread),
>       network_thread_(network_thread),
>       signaling_thread_(signaling_thread),
>@@ -117,6 +118,11 @@ BaseChannel::BaseChannel(rtc::Thread* worker_thread,
> BaseChannel::~BaseChannel() {
>   TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel");
>   RTC_DCHECK_RUN_ON(worker_thread_);
>+
>+  if (media_transport_) {
>+    media_transport_->SetNetworkChangeCallback(nullptr);
>+  }
>+
>   // Eats any outstanding messages or packets.
>   worker_thread_->Clear(&invoker_);
>   worker_thread_->Clear(this);
>@@ -136,8 +142,15 @@ bool BaseChannel::ConnectToRtpTransport() {
>       this, &BaseChannel::OnTransportReadyToSend);
>   rtp_transport_->SignalRtcpPacketReceived.connect(
>       this, &BaseChannel::OnRtcpPacketReceived);
>-  rtp_transport_->SignalNetworkRouteChanged.connect(
>-      this, &BaseChannel::OnNetworkRouteChanged);
>+
>+  // If media transport is used, it's responsible for providing network
>+  // route changed callbacks.
>+  if (!media_transport_) {
>+    rtp_transport_->SignalNetworkRouteChanged.connect(
>+        this, &BaseChannel::OnNetworkRouteChanged);
>+  }
>+  // TODO(bugs.webrtc.org/9719): Media transport should also be used to provide
>+  // 'writable' state here.
>   rtp_transport_->SignalWritableState.connect(this,
>                                               &BaseChannel::OnWritableState);
>   rtp_transport_->SignalSentPacket.connect(this,
>@@ -155,19 +168,29 @@ void BaseChannel::DisconnectFromRtpTransport() {
>   rtp_transport_->SignalSentPacket.disconnect(this);
> }
> 
>-void BaseChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
>+void BaseChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport,
>+                         webrtc::MediaTransportInterface* media_transport) {
>   RTC_DCHECK_RUN_ON(worker_thread_);
>+  media_transport_ = media_transport;
>+
>   network_thread_->Invoke<void>(
>       RTC_FROM_HERE, [this, rtp_transport] { SetRtpTransport(rtp_transport); });
> 
>   // Both RTP and RTCP channels should be set, we can call SetInterface on
>   // the media channel and it can set network options.
>-  media_channel_->SetInterface(this);
>+  media_channel_->SetInterface(this, media_transport);
>+
>+  RTC_LOG(LS_INFO) << "BaseChannel::Init_w, media_transport="
>+                   << (media_transport_ != nullptr);
>+  if (media_transport_) {
>+    media_transport_->SetNetworkChangeCallback(this);
>+  }
> }
> 
> void BaseChannel::Deinit() {
>   RTC_DCHECK(worker_thread_->IsCurrent());
>-  media_channel_->SetInterface(NULL);
>+  media_channel_->SetInterface(/*iface=*/nullptr,
>+                               /*media_transport=*/nullptr);
>   // Packets arrive on the network thread, processing packets calls virtual
>   // functions, so need to stop this process in Deinit that is called in
>   // derived classes destructor.
>@@ -350,6 +373,8 @@ void BaseChannel::OnWritableState(bool writable) {
> 
> void BaseChannel::OnNetworkRouteChanged(
>     absl::optional<rtc::NetworkRoute> network_route) {
>+  RTC_LOG(LS_INFO) << "Network route was changed.";
>+
>   RTC_DCHECK(network_thread_->IsCurrent());
>   rtc::NetworkRoute new_route;
>   if (network_route) {
>@@ -439,13 +464,12 @@ void BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet) {
>   // RtpPacketReceived.arrival_time_ms = (PacketTime + 500) / 1000;
>   // Note: The |not_before| field is always 0 here. This field is not currently
>   // used, so it should be fine.
>-  int64_t timestamp = -1;
>+  int64_t timestamp_us = -1;
>   if (parsed_packet.arrival_time_ms() > 0) {
>-    timestamp = parsed_packet.arrival_time_ms() * 1000;
>+    timestamp_us = parsed_packet.arrival_time_ms() * 1000;
>   }
>-  rtc::PacketTime packet_time(timestamp, /*not_before=*/0);
> 
>-  OnPacketReceived(/*rtcp=*/false, parsed_packet.Buffer(), packet_time);
>+  OnPacketReceived(/*rtcp=*/false, parsed_packet.Buffer(), timestamp_us);
> }
> 
> void BaseChannel::UpdateRtpHeaderExtensionMap(
>@@ -472,13 +496,13 @@ bool BaseChannel::RegisterRtpDemuxerSink() {
> }
> 
> void BaseChannel::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                       const rtc::PacketTime& packet_time) {
>-  OnPacketReceived(/*rtcp=*/true, *packet, packet_time);
>+                                       int64_t packet_time_us) {
>+  OnPacketReceived(/*rtcp=*/true, *packet, packet_time_us);
> }
> 
> void BaseChannel::OnPacketReceived(bool rtcp,
>                                    const rtc::CopyOnWriteBuffer& packet,
>-                                   const rtc::PacketTime& packet_time) {
>+                                   int64_t packet_time_us) {
>   if (!has_received_packet_ && !rtcp) {
>     has_received_packet_ = true;
>     signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FIRSTPACKETRECEIVED);
>@@ -504,21 +528,21 @@ void BaseChannel::OnPacketReceived(bool rtcp,
> 
>   invoker_.AsyncInvoke<void>(
>       RTC_FROM_HERE, worker_thread_,
>-      Bind(&BaseChannel::ProcessPacket, this, rtcp, packet, packet_time));
>+      Bind(&BaseChannel::ProcessPacket, this, rtcp, packet, packet_time_us));
> }
> 
> void BaseChannel::ProcessPacket(bool rtcp,
>                                 const rtc::CopyOnWriteBuffer& packet,
>-                                const rtc::PacketTime& packet_time) {
>+                                int64_t packet_time_us) {
>   RTC_DCHECK(worker_thread_->IsCurrent());
> 
>   // Need to copy variable because OnRtcpReceived/OnPacketReceived
>   // requires non-const pointer to buffer. This doesn't memcpy the actual data.
>   rtc::CopyOnWriteBuffer data(packet);
>   if (rtcp) {
>-    media_channel_->OnRtcpReceived(&data, packet_time);
>+    media_channel_->OnRtcpReceived(&data, packet_time_us);
>   } else {
>-    media_channel_->OnPacketReceived(&data, packet_time);
>+    media_channel_->OnPacketReceived(&data, packet_time_us);
>   }
> }
> 
>@@ -673,7 +697,7 @@ bool BaseChannel::UpdateRemoteStreams_w(
> RtpHeaderExtensions BaseChannel::GetFilteredRtpHeaderExtensions(
>     const RtpHeaderExtensions& extensions) {
>   RTC_DCHECK(rtp_transport_);
>-  if (crypto_options_.enable_encrypted_rtp_header_extensions) {
>+  if (crypto_options_.srtp.enable_encrypted_rtp_header_extensions) {
>     RtpHeaderExtensions filtered;
>     auto pred = [](const webrtc::RtpExtension& extension) {
>       return !extension.encrypt;
>@@ -700,7 +724,7 @@ void BaseChannel::OnMessage(rtc::Message* pmsg) {
>       break;
>     }
>     case MSG_FIRSTPACKETRECEIVED: {
>-      SignalFirstPacketReceived(this);
>+      SignalFirstPacketReceived_(this);
>       break;
>     }
>   }
>@@ -742,7 +766,7 @@ VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
>                            std::unique_ptr<VoiceMediaChannel> media_channel,
>                            const std::string& content_name,
>                            bool srtp_required,
>-                           rtc::CryptoOptions crypto_options)
>+                           webrtc::CryptoOptions crypto_options)
>     : BaseChannel(worker_thread,
>                   network_thread,
>                   signaling_thread,
>@@ -765,6 +789,11 @@ void BaseChannel::UpdateMediaSendRecvState() {
>       Bind(&BaseChannel::UpdateMediaSendRecvState_w, this));
> }
> 
>+void BaseChannel::OnNetworkRouteChanged(
>+    const rtc::NetworkRoute& network_route) {
>+  OnNetworkRouteChanged(absl::make_optional(network_route));
>+}
>+
> void VoiceChannel::UpdateMediaSendRecvState_w() {
>   // Render incoming data if we're the active call, and we have the local
>   // content. We receive data on the default channel and multiplexed streams.
>@@ -797,6 +826,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
>   RtpHeaderExtensions rtp_header_extensions =
>       GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
>   UpdateRtpHeaderExtensionMap(rtp_header_extensions);
>+  media_channel()->SetExtmapAllowMixed(audio->extmap_allow_mixed());
> 
>   AudioRecvParameters recv_params = last_recv_params_;
>   RtpParametersFromMediaDescription(audio, rtp_header_extensions, &recv_params);
>@@ -881,7 +911,7 @@ VideoChannel::VideoChannel(rtc::Thread* worker_thread,
>                            std::unique_ptr<VideoMediaChannel> media_channel,
>                            const std::string& content_name,
>                            bool srtp_required,
>-                           rtc::CryptoOptions crypto_options)
>+                           webrtc::CryptoOptions crypto_options)
>     : BaseChannel(worker_thread,
>                   network_thread,
>                   signaling_thread,
>@@ -932,6 +962,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
>   RtpHeaderExtensions rtp_header_extensions =
>       GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
>   UpdateRtpHeaderExtensionMap(rtp_header_extensions);
>+  media_channel()->SetExtmapAllowMixed(video->extmap_allow_mixed());
> 
>   VideoRecvParameters recv_params = last_recv_params_;
>   RtpParametersFromMediaDescription(video, rtp_header_extensions, &recv_params);
>@@ -1019,7 +1050,7 @@ RtpDataChannel::RtpDataChannel(rtc::Thread* worker_thread,
>                                std::unique_ptr<DataMediaChannel> media_channel,
>                                const std::string& content_name,
>                                bool srtp_required,
>-                               rtc::CryptoOptions crypto_options)
>+                               webrtc::CryptoOptions crypto_options)
>     : BaseChannel(worker_thread,
>                   network_thread,
>                   signaling_thread,
>@@ -1036,7 +1067,7 @@ RtpDataChannel::~RtpDataChannel() {
> }
> 
> void RtpDataChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
>-  BaseChannel::Init_w(rtp_transport);
>+  BaseChannel::Init_w(rtp_transport, /*media_transport=*/nullptr);
>   media_channel()->SignalDataReceived.connect(this,
>                                               &RtpDataChannel::OnDataReceived);
>   media_channel()->SignalReadyToSend.connect(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.h
>index ba584690ba6ad07557bc196a160330e1d844f507..6264bb39041ae7d3d4a1a5e7270f4348c234a27e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.h
>@@ -29,6 +29,7 @@
> #include "media/base/streamparams.h"
> #include "p2p/base/dtlstransportinternal.h"
> #include "p2p/base/packettransportinternal.h"
>+#include "pc/channelinterface.h"
> #include "pc/dtlssrtptransport.h"
> #include "pc/mediasession.h"
> #include "pc/rtptransport.h"
>@@ -42,12 +43,12 @@
> 
> namespace webrtc {
> class AudioSinkInterface;
>+class MediaTransportInterface;
> }  // namespace webrtc
> 
> namespace cricket {
> 
> struct CryptoParams;
>-class MediaContentDescription;
> 
> // BaseChannel contains logic common to voice and video, including enable,
> // marshaling calls to a worker and network threads, and connection and media
>@@ -67,10 +68,12 @@ class MediaContentDescription;
> // vtable, and the media channel's thread using BaseChannel as the
> // NetworkInterface.
> 
>-class BaseChannel : public rtc::MessageHandler,
>+class BaseChannel : public ChannelInterface,
>+                    public rtc::MessageHandler,
>                     public sigslot::has_slots<>,
>                     public MediaChannel::NetworkInterface,
>-                    public webrtc::RtpPacketSinkInterface {
>+                    public webrtc::RtpPacketSinkInterface,
>+                    public webrtc::MediaTransportNetworkChangeCallback {
>  public:
>   // If |srtp_required| is true, the channel will not send or receive any
>   // RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP).
>@@ -82,9 +85,10 @@ class BaseChannel : public rtc::MessageHandler,
>               std::unique_ptr<MediaChannel> media_channel,
>               const std::string& content_name,
>               bool srtp_required,
>-              rtc::CryptoOptions crypto_options);
>+              webrtc::CryptoOptions crypto_options);
>   virtual ~BaseChannel();
>-  void Init_w(webrtc::RtpTransportInternal* rtp_transport);
>+  void Init_w(webrtc::RtpTransportInternal* rtp_transport,
>+              webrtc::MediaTransportInterface* media_transport);
> 
>   // Deinit may be called multiple times and is simply ignored if it's already
>   // done.
>@@ -92,10 +96,10 @@ class BaseChannel : public rtc::MessageHandler,
> 
>   rtc::Thread* worker_thread() const { return worker_thread_; }
>   rtc::Thread* network_thread() const { return network_thread_; }
>-  const std::string& content_name() const { return content_name_; }
>+  const std::string& content_name() const override { return content_name_; }
>   // TODO(deadbeef): This is redundant; remove this.
>-  const std::string& transport_name() const { return transport_name_; }
>-  bool enabled() const { return enabled_; }
>+  const std::string& transport_name() const override { return transport_name_; }
>+  bool enabled() const override { return enabled_; }
> 
>   // This function returns true if using SRTP (DTLS-based keying or SDES).
>   bool srtp_active() const {
>@@ -108,17 +112,17 @@ class BaseChannel : public rtc::MessageHandler,
>   // encryption, an SrtpTransport for SDES or a DtlsSrtpTransport for DTLS-SRTP.
>   // This can be called from any thread and it hops to the network thread
>   // internally. It would replace the |SetTransports| and its variants.
>-  bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport);
>+  bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) override;
> 
>   // Channel control
>   bool SetLocalContent(const MediaContentDescription* content,
>                        webrtc::SdpType type,
>-                       std::string* error_desc);
>+                       std::string* error_desc) override;
>   bool SetRemoteContent(const MediaContentDescription* content,
>                         webrtc::SdpType type,
>-                        std::string* error_desc);
>+                        std::string* error_desc) override;
> 
>-  bool Enable(bool enable);
>+  bool Enable(bool enable) override;
> 
>   // TODO(zhihuang): These methods are used for testing and can be removed.
>   bool AddRecvStream(const StreamParams& sp);
>@@ -138,7 +142,9 @@ class BaseChannel : public rtc::MessageHandler,
>   void SignalDtlsSrtpSetupFailure_s(bool rtcp);
> 
>   // Used for latency measurements.
>-  sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;
>+  sigslot::signal1<ChannelInterface*>& SignalFirstPacketReceived() override {
>+    return SignalFirstPacketReceived_;
>+  }
> 
>   // Forward SignalSentPacket to worker thread.
>   sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
>@@ -162,6 +168,11 @@ class BaseChannel : public rtc::MessageHandler,
>     return nullptr;
>   }
> 
>+  // Returns media transport, can be null if media transport is not available.
>+  webrtc::MediaTransportInterface* media_transport() {
>+    return media_transport_;
>+  }
>+
>   // From RtpTransport - public for testing only
>   void OnTransportReadyToSend(bool ready);
> 
>@@ -169,8 +180,6 @@ class BaseChannel : public rtc::MessageHandler,
>   int SetOption(SocketType type, rtc::Socket::Option o, int val) override;
>   int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
> 
>-  virtual cricket::MediaType media_type() = 0;
>-
>   // RtpPacketSinkInterface overrides.
>   void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
> 
>@@ -180,9 +189,9 @@ class BaseChannel : public rtc::MessageHandler,
>     transport_name_ = transport_name;
>   }
> 
>- protected:
>-  virtual MediaChannel* media_channel() const { return media_channel_.get(); }
>+  MediaChannel* media_channel() const override { return media_channel_.get(); }
> 
>+ protected:
>   bool was_ever_writable() const { return was_ever_writable_; }
>   void set_local_content_direction(webrtc::RtpTransceiverDirection direction) {
>     local_content_direction_ = direction;
>@@ -225,14 +234,14 @@ class BaseChannel : public rtc::MessageHandler,
>                   const rtc::PacketOptions& options);
> 
>   void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                            const rtc::PacketTime& packet_time);
>+                            int64_t packet_time_us);
> 
>   void OnPacketReceived(bool rtcp,
>                         const rtc::CopyOnWriteBuffer& packet,
>-                        const rtc::PacketTime& packet_time);
>+                        int64_t packet_time_us);
>   void ProcessPacket(bool rtcp,
>                      const rtc::CopyOnWriteBuffer& packet,
>-                     const rtc::PacketTime& packet_time);
>+                     int64_t packet_time_us);
> 
>   void EnableMedia_w();
>   void DisableMedia_w();
>@@ -295,10 +304,14 @@ class BaseChannel : public rtc::MessageHandler,
>   void SignalSentPacket_n(const rtc::SentPacket& sent_packet);
>   void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
>   bool IsReadyToSendMedia_n() const;
>+
>+  // MediaTransportNetworkChangeCallback override.
>+  void OnNetworkRouteChanged(const rtc::NetworkRoute& network_route) override;
>   rtc::Thread* const worker_thread_;
>   rtc::Thread* const network_thread_;
>   rtc::Thread* const signaling_thread_;
>   rtc::AsyncInvoker invoker_;
>+  sigslot::signal1<ChannelInterface*> SignalFirstPacketReceived_;
> 
>   const std::string content_name_;
> 
>@@ -307,13 +320,18 @@ class BaseChannel : public rtc::MessageHandler,
> 
>   webrtc::RtpTransportInternal* rtp_transport_ = nullptr;
> 
>+  // Optional media transport (experimental).
>+  // If provided, audio and video will be sent through media_transport instead
>+  // of RTP/RTCP. Currently media_transport can co-exist with rtp_transport.
>+  webrtc::MediaTransportInterface* media_transport_ = nullptr;
>+
>   std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
>   std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
>   bool writable_ = false;
>   bool was_ever_writable_ = false;
>   bool has_received_packet_ = false;
>   const bool srtp_required_ = true;
>-  rtc::CryptoOptions crypto_options_;
>+  webrtc::CryptoOptions crypto_options_;
> 
>   // MediaChannel related members that should be accessed from the worker
>   // thread.
>@@ -343,7 +361,7 @@ class VoiceChannel : public BaseChannel {
>                std::unique_ptr<VoiceMediaChannel> channel,
>                const std::string& content_name,
>                bool srtp_required,
>-               rtc::CryptoOptions crypto_options);
>+               webrtc::CryptoOptions crypto_options);
>   ~VoiceChannel();
> 
>   // downcasts a MediaChannel
>@@ -351,10 +369,9 @@ class VoiceChannel : public BaseChannel {
>     return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
>   }
> 
>-  webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const;
>-  webrtc::RTCError SetRtpSendParameters_w(uint32_t ssrc,
>-                                          webrtc::RtpParameters parameters);
>-  cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_AUDIO; }
>+  cricket::MediaType media_type() const override {
>+    return cricket::MEDIA_TYPE_AUDIO;
>+  }
> 
>  private:
>   // overrides from BaseChannel
>@@ -383,7 +400,7 @@ class VideoChannel : public BaseChannel {
>                std::unique_ptr<VideoMediaChannel> media_channel,
>                const std::string& content_name,
>                bool srtp_required,
>-               rtc::CryptoOptions crypto_options);
>+               webrtc::CryptoOptions crypto_options);
>   ~VideoChannel();
> 
>   // downcasts a MediaChannel
>@@ -393,7 +410,9 @@ class VideoChannel : public BaseChannel {
> 
>   void FillBitrateInfo(BandwidthEstimationInfo* bwe_info);
> 
>-  cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_VIDEO; }
>+  cricket::MediaType media_type() const override {
>+    return cricket::MEDIA_TYPE_VIDEO;
>+  }
> 
>  private:
>   // overrides from BaseChannel
>@@ -422,7 +441,7 @@ class RtpDataChannel : public BaseChannel {
>                  std::unique_ptr<DataMediaChannel> channel,
>                  const std::string& content_name,
>                  bool srtp_required,
>-                 rtc::CryptoOptions crypto_options);
>+                 webrtc::CryptoOptions crypto_options);
>   ~RtpDataChannel();
>   // TODO(zhihuang): Remove this once the RtpTransport can be shared between
>   // BaseChannels.
>@@ -445,7 +464,9 @@ class RtpDataChannel : public BaseChannel {
>   // That occurs when the channel is enabled, the transport is writable,
>   // both local and remote descriptions are set, and the channel is unblocked.
>   sigslot::signal1<bool> SignalReadyToSendData;
>-  cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_DATA; }
>+  cricket::MediaType media_type() const override {
>+    return cricket::MEDIA_TYPE_DATA;
>+  }
> 
>  protected:
>   // downcasts a MediaChannel.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel_unittest.cc
>index 58cb17b8bbf5dd1f7cbe400fc5c5d5009c6d72b7..5d4d4c7f9e004e56d5ecb8af207680b9da926337 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel_unittest.cc
>@@ -15,7 +15,6 @@
> #include "media/base/fakemediaengine.h"
> #include "media/base/fakertp.h"
> #include "media/base/mediachannel.h"
>-#include "media/base/testutils.h"
> #include "p2p/base/fakecandidatepair.h"
> #include "p2p/base/fakedtlstransport.h"
> #include "p2p/base/fakepackettransport.h"
>@@ -250,8 +249,8 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     rtc::Thread* signaling_thread = rtc::Thread::Current();
>     auto channel = absl::make_unique<typename T::Channel>(
>         worker_thread, network_thread, signaling_thread, engine, std::move(ch),
>-        cricket::CN_AUDIO, (flags & DTLS) != 0, rtc::CryptoOptions());
>-    channel->Init_w(rtp_transport);
>+        cricket::CN_AUDIO, (flags & DTLS) != 0, webrtc::CryptoOptions());
>+    channel->Init_w(rtp_transport, /*media_transport=*/nullptr);
>     return channel;
>   }
> 
>@@ -577,6 +576,37 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>         CodecMatches(content.codecs()[0], media_channel1_->codecs()[0]));
>   }
> 
>+  // Test that SetLocalContent and SetRemoteContent properly configure
>+  // extmap-allow-mixed.
>+  void TestSetContentsExtmapAllowMixedCaller(bool offer, bool answer) {
>+    // For a caller, SetLocalContent() is called first with an offer and next
>+    // SetRemoteContent() is called with the answer.
>+    CreateChannels(0, 0);
>+    typename T::Content content;
>+    CreateContent(0, kPcmuCodec, kH264Codec, &content);
>+    auto offer_enum = offer ? (T::Content::kSession) : (T::Content::kNo);
>+    auto answer_enum = answer ? (T::Content::kSession) : (T::Content::kNo);
>+    content.set_extmap_allow_mixed_enum(offer_enum);
>+    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, NULL));
>+    content.set_extmap_allow_mixed_enum(answer_enum);
>+    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, NULL));
>+    EXPECT_EQ(answer, media_channel1_->ExtmapAllowMixed());
>+  }
>+  void TestSetContentsExtmapAllowMixedCallee(bool offer, bool answer) {
>+    // For a callee, SetRemoteContent() is called first with an offer and next
>+    // SetLocalContent() is called with the answer.
>+    CreateChannels(0, 0);
>+    typename T::Content content;
>+    CreateContent(0, kPcmuCodec, kH264Codec, &content);
>+    auto offer_enum = offer ? (T::Content::kSession) : (T::Content::kNo);
>+    auto answer_enum = answer ? (T::Content::kSession) : (T::Content::kNo);
>+    content.set_extmap_allow_mixed_enum(offer_enum);
>+    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kOffer, NULL));
>+    content.set_extmap_allow_mixed_enum(answer_enum);
>+    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kAnswer, NULL));
>+    EXPECT_EQ(answer, media_channel1_->ExtmapAllowMixed());
>+  }
>+
>   // Test that SetLocalContent and SetRemoteContent properly deals
>   // with an empty offer.
>   void TestSetContentsNullOffer() {
>@@ -1545,8 +1575,8 @@ std::unique_ptr<cricket::VideoChannel> ChannelTest<VideoTraits>::CreateChannel(
>   rtc::Thread* signaling_thread = rtc::Thread::Current();
>   auto channel = absl::make_unique<cricket::VideoChannel>(
>       worker_thread, network_thread, signaling_thread, std::move(ch),
>-      cricket::CN_VIDEO, (flags & DTLS) != 0, rtc::CryptoOptions());
>-  channel->Init_w(rtp_transport);
>+      cricket::CN_VIDEO, (flags & DTLS) != 0, webrtc::CryptoOptions());
>+  channel->Init_w(rtp_transport, /*media_transport=*/nullptr);
>   return channel;
> }
> 
>@@ -1615,6 +1645,24 @@ TEST_F(VoiceChannelSingleThreadTest, TestSetContents) {
>   Base::TestSetContents();
> }
> 
>+TEST_F(VoiceChannelSingleThreadTest, TestSetContentsExtmapAllowMixedAsCaller) {
>+  Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/true);
>+}
>+
>+TEST_F(VoiceChannelSingleThreadTest,
>+       TestSetContentsExtmapAllowMixedNotSupportedAsCaller) {
>+  Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/false);
>+}
>+
>+TEST_F(VoiceChannelSingleThreadTest, TestSetContentsExtmapAllowMixedAsCallee) {
>+  Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/true);
>+}
>+
>+TEST_F(VoiceChannelSingleThreadTest,
>+       TestSetContentsExtmapAllowMixedNotSupportedAsCallee) {
>+  Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/false);
>+}
>+
> TEST_F(VoiceChannelSingleThreadTest, TestSetContentsNullOffer) {
>   Base::TestSetContentsNullOffer();
> }
>@@ -1750,6 +1798,24 @@ TEST_F(VoiceChannelDoubleThreadTest, TestSetContents) {
>   Base::TestSetContents();
> }
> 
>+TEST_F(VoiceChannelDoubleThreadTest, TestSetContentsExtmapAllowMixedAsCaller) {
>+  Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/true);
>+}
>+
>+TEST_F(VoiceChannelDoubleThreadTest,
>+       TestSetContentsExtmapAllowMixedNotSupportedAsCaller) {
>+  Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/false);
>+}
>+
>+TEST_F(VoiceChannelDoubleThreadTest, TestSetContentsExtmapAllowMixedAsCallee) {
>+  Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/true);
>+}
>+
>+TEST_F(VoiceChannelDoubleThreadTest,
>+       TestSetContentsExtmapAllowMixedNotSupportedAsCallee) {
>+  Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/false);
>+}
>+
> TEST_F(VoiceChannelDoubleThreadTest, TestSetContentsNullOffer) {
>   Base::TestSetContentsNullOffer();
> }
>@@ -1883,6 +1949,24 @@ TEST_F(VideoChannelSingleThreadTest, TestSetContents) {
>   Base::TestSetContents();
> }
> 
>+TEST_F(VideoChannelSingleThreadTest, TestSetContentsExtmapAllowMixedAsCaller) {
>+  Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/true);
>+}
>+
>+TEST_F(VideoChannelSingleThreadTest,
>+       TestSetContentsExtmapAllowMixedNotSupportedAsCaller) {
>+  Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/false);
>+}
>+
>+TEST_F(VideoChannelSingleThreadTest, TestSetContentsExtmapAllowMixedAsCallee) {
>+  Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/true);
>+}
>+
>+TEST_F(VideoChannelSingleThreadTest,
>+       TestSetContentsExtmapAllowMixedNotSupportedAsCallee) {
>+  Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/false);
>+}
>+
> TEST_F(VideoChannelSingleThreadTest, TestSetContentsNullOffer) {
>   Base::TestSetContentsNullOffer();
> }
>@@ -2016,6 +2100,24 @@ TEST_F(VideoChannelDoubleThreadTest, TestSetContents) {
>   Base::TestSetContents();
> }
> 
>+TEST_F(VideoChannelDoubleThreadTest, TestSetContentsExtmapAllowMixedAsCaller) {
>+  Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/true);
>+}
>+
>+TEST_F(VideoChannelDoubleThreadTest,
>+       TestSetContentsExtmapAllowMixedNotSupportedAsCaller) {
>+  Base::TestSetContentsExtmapAllowMixedCaller(/*offer=*/true, /*answer=*/false);
>+}
>+
>+TEST_F(VideoChannelDoubleThreadTest, TestSetContentsExtmapAllowMixedAsCallee) {
>+  Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/true);
>+}
>+
>+TEST_F(VideoChannelDoubleThreadTest,
>+       TestSetContentsExtmapAllowMixedNotSupportedAsCallee) {
>+  Base::TestSetContentsExtmapAllowMixedCallee(/*offer=*/true, /*answer=*/false);
>+}
>+
> TEST_F(VideoChannelDoubleThreadTest, TestSetContentsNullOffer) {
>   Base::TestSetContentsNullOffer();
> }
>@@ -2164,7 +2266,7 @@ std::unique_ptr<cricket::RtpDataChannel> ChannelTest<DataTraits>::CreateChannel(
>   rtc::Thread* signaling_thread = rtc::Thread::Current();
>   auto channel = absl::make_unique<cricket::RtpDataChannel>(
>       worker_thread, network_thread, signaling_thread, std::move(ch),
>-      cricket::CN_DATA, (flags & DTLS) != 0, rtc::CryptoOptions());
>+      cricket::CN_DATA, (flags & DTLS) != 0, webrtc::CryptoOptions());
>   channel->Init_w(rtp_transport);
>   return channel;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelinterface.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..8e4109a04542f535656bda412e92e652237161a9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelinterface.h
>@@ -0,0 +1,68 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_CHANNELINTERFACE_H_
>+#define PC_CHANNELINTERFACE_H_
>+
>+#include <string>
>+
>+#include "api/jsep.h"
>+#include "api/mediatypes.h"
>+#include "media/base/mediachannel.h"
>+#include "pc/rtptransportinternal.h"
>+
>+namespace cricket {
>+
>+class MediaContentDescription;
>+
>+// ChannelInterface contains methods common to voice, video and data channels.
>+// As more methods are added to BaseChannel, they should be included in the
>+// interface as well.
>+class ChannelInterface {
>+ public:
>+  virtual cricket::MediaType media_type() const = 0;
>+
>+  virtual MediaChannel* media_channel() const = 0;
>+
>+  // TODO(deadbeef): This is redundant; remove this.
>+  virtual const std::string& transport_name() const = 0;
>+
>+  virtual const std::string& content_name() const = 0;
>+
>+  virtual bool enabled() const = 0;
>+
>+  // Enables or disables this channel
>+  virtual bool Enable(bool enable) = 0;
>+
>+  // Used for latency measurements.
>+  virtual sigslot::signal1<ChannelInterface*>& SignalFirstPacketReceived() = 0;
>+
>+  // Channel control
>+  virtual bool SetLocalContent(const MediaContentDescription* content,
>+                               webrtc::SdpType type,
>+                               std::string* error_desc) = 0;
>+  virtual bool SetRemoteContent(const MediaContentDescription* content,
>+                                webrtc::SdpType type,
>+                                std::string* error_desc) = 0;
>+
>+  // Set an RTP level transport.
>+  // Some examples:
>+  //   * An RtpTransport without encryption.
>+  //   * An SrtpTransport for SDES.
>+  //   * A DtlsSrtpTransport for DTLS-SRTP.
>+  virtual bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) = 0;
>+
>+ protected:
>+  virtual ~ChannelInterface() = default;
>+};
>+
>+}  // namespace cricket
>+
>+#endif  // PC_CHANNELINTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.cc
>index dab622ad70e42f079fb430d2e901fca9dd49923c..eda5a2dd57922cbd7ae4fe84982ca7ef08dfec91 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.cc
>@@ -14,10 +14,10 @@
> #include <utility>
> 
> #include "absl/memory/memory.h"
>+#include "absl/strings/match.h"
> #include "media/base/rtpdataengine.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/stringutils.h"
> #include "rtc_base/trace_event.h"
> 
> namespace cricket {
>@@ -66,7 +66,7 @@ void ChannelManager::GetSupportedAudioSendCodecs(
>   if (!media_engine_) {
>     return;
>   }
>-  *codecs = media_engine_->audio_send_codecs();
>+  *codecs = media_engine_->voice().send_codecs();
> }
> 
> void ChannelManager::GetSupportedAudioReceiveCodecs(
>@@ -74,7 +74,7 @@ void ChannelManager::GetSupportedAudioReceiveCodecs(
>   if (!media_engine_) {
>     return;
>   }
>-  *codecs = media_engine_->audio_recv_codecs();
>+  *codecs = media_engine_->voice().recv_codecs();
> }
> 
> void ChannelManager::GetSupportedAudioRtpHeaderExtensions(
>@@ -82,7 +82,7 @@ void ChannelManager::GetSupportedAudioRtpHeaderExtensions(
>   if (!media_engine_) {
>     return;
>   }
>-  *ext = media_engine_->GetAudioCapabilities().header_extensions;
>+  *ext = media_engine_->voice().GetCapabilities().header_extensions;
> }
> 
> void ChannelManager::GetSupportedVideoCodecs(
>@@ -92,10 +92,10 @@ void ChannelManager::GetSupportedVideoCodecs(
>   }
>   codecs->clear();
> 
>-  std::vector<VideoCodec> video_codecs = media_engine_->video_codecs();
>+  std::vector<VideoCodec> video_codecs = media_engine_->video().codecs();
>   for (const auto& video_codec : video_codecs) {
>     if (!enable_rtx_ &&
>-        _stricmp(kRtxCodecName, video_codec.name.c_str()) == 0) {
>+        absl::EqualsIgnoreCase(kRtxCodecName, video_codec.name)) {
>       continue;
>     }
>     codecs->push_back(video_codec);
>@@ -107,7 +107,7 @@ void ChannelManager::GetSupportedVideoRtpHeaderExtensions(
>   if (!media_engine_) {
>     return;
>   }
>-  *ext = media_engine_->GetVideoCapabilities().header_extensions;
>+  *ext = media_engine_->video().GetCapabilities().header_extensions;
> }
> 
> void ChannelManager::GetSupportedDataCodecs(
>@@ -156,16 +156,17 @@ VoiceChannel* ChannelManager::CreateVoiceChannel(
>     webrtc::Call* call,
>     const cricket::MediaConfig& media_config,
>     webrtc::RtpTransportInternal* rtp_transport,
>+    webrtc::MediaTransportInterface* media_transport,
>     rtc::Thread* signaling_thread,
>     const std::string& content_name,
>     bool srtp_required,
>-    const rtc::CryptoOptions& crypto_options,
>+    const webrtc::CryptoOptions& crypto_options,
>     const AudioOptions& options) {
>   if (!worker_thread_->IsCurrent()) {
>     return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
>       return CreateVoiceChannel(call, media_config, rtp_transport,
>-                                signaling_thread, content_name, srtp_required,
>-                                crypto_options, options);
>+                                media_transport, signaling_thread, content_name,
>+                                srtp_required, crypto_options, options);
>     });
>   }
> 
>@@ -176,8 +177,8 @@ VoiceChannel* ChannelManager::CreateVoiceChannel(
>     return nullptr;
>   }
> 
>-  VoiceMediaChannel* media_channel =
>-      media_engine_->CreateChannel(call, media_config, options);
>+  VoiceMediaChannel* media_channel = media_engine_->voice().CreateMediaChannel(
>+      call, media_config, options, crypto_options);
>   if (!media_channel) {
>     return nullptr;
>   }
>@@ -187,7 +188,7 @@ VoiceChannel* ChannelManager::CreateVoiceChannel(
>       absl::WrapUnique(media_channel), content_name, srtp_required,
>       crypto_options);
> 
>-  voice_channel->Init_w(rtp_transport);
>+  voice_channel->Init_w(rtp_transport, media_transport);
> 
>   VoiceChannel* voice_channel_ptr = voice_channel.get();
>   voice_channels_.push_back(std::move(voice_channel));
>@@ -226,7 +227,7 @@ VideoChannel* ChannelManager::CreateVideoChannel(
>     rtc::Thread* signaling_thread,
>     const std::string& content_name,
>     bool srtp_required,
>-    const rtc::CryptoOptions& crypto_options,
>+    const webrtc::CryptoOptions& crypto_options,
>     const VideoOptions& options) {
>   if (!worker_thread_->IsCurrent()) {
>     return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
>@@ -243,8 +244,8 @@ VideoChannel* ChannelManager::CreateVideoChannel(
>     return nullptr;
>   }
> 
>-  VideoMediaChannel* media_channel =
>-      media_engine_->CreateVideoChannel(call, media_config, options);
>+  VideoMediaChannel* media_channel = media_engine_->video().CreateMediaChannel(
>+      call, media_config, options, crypto_options);
>   if (!media_channel) {
>     return nullptr;
>   }
>@@ -253,7 +254,9 @@ VideoChannel* ChannelManager::CreateVideoChannel(
>       worker_thread_, network_thread_, signaling_thread,
>       absl::WrapUnique(media_channel), content_name, srtp_required,
>       crypto_options);
>-  video_channel->Init_w(rtp_transport);
>+
>+  // TODO(sukhanov): Add media_transport support for video channel.
>+  video_channel->Init_w(rtp_transport, /*media_transport=*/nullptr);
> 
>   VideoChannel* video_channel_ptr = video_channel.get();
>   video_channels_.push_back(std::move(video_channel));
>@@ -291,7 +294,7 @@ RtpDataChannel* ChannelManager::CreateRtpDataChannel(
>     rtc::Thread* signaling_thread,
>     const std::string& content_name,
>     bool srtp_required,
>-    const rtc::CryptoOptions& crypto_options) {
>+    const webrtc::CryptoOptions& crypto_options) {
>   if (!worker_thread_->IsCurrent()) {
>     return worker_thread_->Invoke<RtpDataChannel*>(RTC_FROM_HERE, [&] {
>       return CreateRtpDataChannel(media_config, rtp_transport, signaling_thread,
>@@ -346,13 +349,13 @@ void ChannelManager::DestroyRtpDataChannel(RtpDataChannel* data_channel) {
> bool ChannelManager::StartAecDump(rtc::PlatformFile file,
>                                   int64_t max_size_bytes) {
>   return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
>-    return media_engine_->StartAecDump(file, max_size_bytes);
>+    return media_engine_->voice().StartAecDump(file, max_size_bytes);
>   });
> }
> 
> void ChannelManager::StopAecDump() {
>   worker_thread_->Invoke<void>(RTC_FROM_HERE,
>-                               [&] { media_engine_->StopAecDump(); });
>+                               [&] { media_engine_->voice().StopAecDump(); });
> }
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.h
>index c6b601ea8f6cb0e0017b0bf0aca703de0249658a..5cafd8cfa72f56377fc581573148a3a95b480c77 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.h
>@@ -80,14 +80,16 @@ class ChannelManager final {
>   // call the appropriate Destroy*Channel method when done.
> 
>   // Creates a voice channel, to be associated with the specified session.
>-  VoiceChannel* CreateVoiceChannel(webrtc::Call* call,
>-                                   const cricket::MediaConfig& media_config,
>-                                   webrtc::RtpTransportInternal* rtp_transport,
>-                                   rtc::Thread* signaling_thread,
>-                                   const std::string& content_name,
>-                                   bool srtp_required,
>-                                   const rtc::CryptoOptions& crypto_options,
>-                                   const AudioOptions& options);
>+  VoiceChannel* CreateVoiceChannel(
>+      webrtc::Call* call,
>+      const cricket::MediaConfig& media_config,
>+      webrtc::RtpTransportInternal* rtp_transport,
>+      webrtc::MediaTransportInterface* media_transport,
>+      rtc::Thread* signaling_thread,
>+      const std::string& content_name,
>+      bool srtp_required,
>+      const webrtc::CryptoOptions& crypto_options,
>+      const AudioOptions& options);
>   // Destroys a voice channel created by CreateVoiceChannel.
>   void DestroyVoiceChannel(VoiceChannel* voice_channel);
> 
>@@ -100,7 +102,7 @@ class ChannelManager final {
>                                    rtc::Thread* signaling_thread,
>                                    const std::string& content_name,
>                                    bool srtp_required,
>-                                   const rtc::CryptoOptions& crypto_options,
>+                                   const webrtc::CryptoOptions& crypto_options,
>                                    const VideoOptions& options);
>   // Destroys a video channel created by CreateVideoChannel.
>   void DestroyVideoChannel(VideoChannel* video_channel);
>@@ -111,7 +113,7 @@ class ChannelManager final {
>       rtc::Thread* signaling_thread,
>       const std::string& content_name,
>       bool srtp_required,
>-      const rtc::CryptoOptions& crypto_options);
>+      const webrtc::CryptoOptions& crypto_options);
>   // Destroys a data channel created by CreateRtpDataChannel.
>   void DestroyRtpDataChannel(RtpDataChannel* data_channel);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager_unittest.cc
>index 47c95306ee70f8b261b80ac04ece7d388402712d..6e9cab6cda81de6c60ebe7ee87fc1ab77bb95ac2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager_unittest.cc
>@@ -11,6 +11,7 @@
> #include <memory>
> #include <utility>
> 
>+#include "api/test/fake_media_transport.h"
> #include "media/base/fakemediaengine.h"
> #include "media/base/testutils.h"
> #include "media/engine/fakewebrtccall.h"
>@@ -61,20 +62,32 @@ class ChannelManagerTest : public testing::Test {
>     return dtls_srtp_transport;
>   }
> 
>-  void TestCreateDestroyChannels(webrtc::RtpTransportInternal* rtp_transport) {
>+  std::unique_ptr<webrtc::MediaTransportInterface> CreateMediaTransport(
>+      rtc::PacketTransportInternal* packet_transport) {
>+    auto media_transport_result =
>+        fake_media_transport_factory_.CreateMediaTransport(packet_transport,
>+                                                           network_.get(),
>+                                                           /*is_caller=*/true);
>+    RTC_CHECK(media_transport_result.ok());
>+    return media_transport_result.MoveValue();
>+  }
>+
>+  void TestCreateDestroyChannels(
>+      webrtc::RtpTransportInternal* rtp_transport,
>+      webrtc::MediaTransportInterface* media_transport) {
>     cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
>-        &fake_call_, cricket::MediaConfig(), rtp_transport,
>+        &fake_call_, cricket::MediaConfig(), rtp_transport, media_transport,
>         rtc::Thread::Current(), cricket::CN_AUDIO, kDefaultSrtpRequired,
>-        rtc::CryptoOptions(), AudioOptions());
>+        webrtc::CryptoOptions(), AudioOptions());
>     EXPECT_TRUE(voice_channel != nullptr);
>     cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
>         &fake_call_, cricket::MediaConfig(), rtp_transport,
>         rtc::Thread::Current(), cricket::CN_VIDEO, kDefaultSrtpRequired,
>-        rtc::CryptoOptions(), VideoOptions());
>+        webrtc::CryptoOptions(), VideoOptions());
>     EXPECT_TRUE(video_channel != nullptr);
>     cricket::RtpDataChannel* rtp_data_channel = cm_->CreateRtpDataChannel(
>         cricket::MediaConfig(), rtp_transport, rtc::Thread::Current(),
>-        cricket::CN_DATA, kDefaultSrtpRequired, rtc::CryptoOptions());
>+        cricket::CN_DATA, kDefaultSrtpRequired, webrtc::CryptoOptions());
>     EXPECT_TRUE(rtp_data_channel != nullptr);
>     cm_->DestroyVideoChannel(video_channel);
>     cm_->DestroyVoiceChannel(voice_channel);
>@@ -90,6 +103,7 @@ class ChannelManagerTest : public testing::Test {
>   cricket::FakeDataEngine* fdme_;
>   std::unique_ptr<cricket::ChannelManager> cm_;
>   cricket::FakeCall fake_call_;
>+  webrtc::FakeMediaTransportFactory fake_media_transport_factory_;
> };
> 
> // Test that we startup/shutdown properly.
>@@ -154,7 +168,15 @@ TEST_F(ChannelManagerTest, SetVideoRtxEnabled) {
> TEST_F(ChannelManagerTest, CreateDestroyChannels) {
>   EXPECT_TRUE(cm_->Init());
>   auto rtp_transport = CreateDtlsSrtpTransport();
>-  TestCreateDestroyChannels(rtp_transport.get());
>+  TestCreateDestroyChannels(rtp_transport.get(), /*media_transport=*/nullptr);
>+}
>+
>+TEST_F(ChannelManagerTest, CreateDestroyChannelsWithMediaTransport) {
>+  EXPECT_TRUE(cm_->Init());
>+  auto rtp_transport = CreateDtlsSrtpTransport();
>+  auto media_transport =
>+      CreateMediaTransport(rtp_transport->rtcp_packet_transport());
>+  TestCreateDestroyChannels(rtp_transport.get(), media_transport.get());
> }
> 
> TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
>@@ -164,7 +186,7 @@ TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
>   EXPECT_TRUE(cm_->set_network_thread(network_.get()));
>   EXPECT_TRUE(cm_->Init());
>   auto rtp_transport = CreateDtlsSrtpTransport();
>-  TestCreateDestroyChannels(rtp_transport.get());
>+  TestCreateDestroyChannels(rtp_transport.get(), /*media_transport=*/nullptr);
> }
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/createpeerconnectionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/createpeerconnectionfactory.cc
>deleted file mode 100644
>index bbc4ebbe730f99ae38027f23812918188b5a019c..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/createpeerconnectionfactory.cc
>+++ /dev/null
>@@ -1,172 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/call/callfactoryinterface.h"
>-#include "api/peerconnectioninterface.h"
>-#include "api/video_codecs/video_decoder_factory.h"
>-#include "api/video_codecs/video_encoder_factory.h"
>-#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
>-#include "media/engine/webrtcmediaengine.h"
>-#include "modules/audio_device/include/audio_device.h"
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "rtc_base/bind.h"
>-#include "rtc_base/scoped_ref_ptr.h"
>-#include "rtc_base/thread.h"
>-
>-namespace webrtc {
>-
>-#if defined(USE_BUILTIN_SW_CODECS)
>-rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory) {
>-  return CreatePeerConnectionFactoryWithAudioMixer(
>-      nullptr /*network_thread*/, nullptr /*worker_thread*/,
>-      nullptr /*signaling_thread*/, nullptr /*default_adm*/,
>-      audio_encoder_factory, audio_decoder_factory,
>-      nullptr /*video_encoder_factory*/, nullptr /*video_decoder_factory*/,
>-      nullptr /*audio_mixer*/);
>-}
>-
>-// Note: all the other CreatePeerConnectionFactory variants just end up calling
>-// this, ultimately.
>-rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* signaling_thread,
>-    AudioDeviceModule* default_adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>-    rtc::scoped_refptr<AudioMixer> audio_mixer,
>-    rtc::scoped_refptr<AudioProcessing> audio_processing) {
>-  rtc::scoped_refptr<AudioProcessing> audio_processing_use = audio_processing;
>-  if (!audio_processing_use) {
>-    audio_processing_use = AudioProcessingBuilder().Create();
>-  }
>-
>-  std::unique_ptr<cricket::MediaEngineInterface> media_engine(
>-      cricket::WebRtcMediaEngineFactory::Create(
>-          default_adm, audio_encoder_factory, audio_decoder_factory,
>-          video_encoder_factory, video_decoder_factory, audio_mixer,
>-          audio_processing_use));
>-
>-  std::unique_ptr<CallFactoryInterface> call_factory = CreateCallFactory();
>-
>-  std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory =
>-      CreateRtcEventLogFactory();
>-
>-  return CreateModularPeerConnectionFactory(
>-      network_thread, worker_thread, signaling_thread, std::move(media_engine),
>-      std::move(call_factory), std::move(event_log_factory));
>-}
>-
>-rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* signaling_thread,
>-    AudioDeviceModule* default_adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>-    rtc::scoped_refptr<AudioMixer> audio_mixer,
>-    rtc::scoped_refptr<AudioProcessing> audio_processing,
>-    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
>-    std::unique_ptr<NetworkControllerFactoryInterface>
>-        network_controller_factory) {
>-  rtc::scoped_refptr<AudioProcessing> audio_processing_use = audio_processing;
>-  if (!audio_processing_use) {
>-    audio_processing_use = AudioProcessingBuilder().Create();
>-  }
>-
>-  std::unique_ptr<cricket::MediaEngineInterface> media_engine(
>-      cricket::WebRtcMediaEngineFactory::Create(
>-          default_adm, audio_encoder_factory, audio_decoder_factory,
>-          video_encoder_factory, video_decoder_factory, audio_mixer,
>-          audio_processing_use));
>-
>-  std::unique_ptr<CallFactoryInterface> call_factory = CreateCallFactory();
>-
>-  std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory =
>-      CreateRtcEventLogFactory();
>-
>-  return CreateModularPeerConnectionFactory(
>-      network_thread, worker_thread, signaling_thread, std::move(media_engine),
>-      std::move(call_factory), std::move(event_log_factory),
>-      std::move(fec_controller_factory), std::move(network_controller_factory));
>-}
>-#endif
>-
>-rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* signaling_thread,
>-    rtc::scoped_refptr<AudioDeviceModule> default_adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>-    std::unique_ptr<VideoEncoderFactory> video_encoder_factory,
>-    std::unique_ptr<VideoDecoderFactory> video_decoder_factory,
>-    rtc::scoped_refptr<AudioMixer> audio_mixer,
>-    rtc::scoped_refptr<AudioProcessing> audio_processing) {
>-  if (!audio_processing)
>-    audio_processing = AudioProcessingBuilder().Create();
>-
>-  std::unique_ptr<cricket::MediaEngineInterface> media_engine =
>-      cricket::WebRtcMediaEngineFactory::Create(
>-          default_adm, audio_encoder_factory, audio_decoder_factory,
>-          std::move(video_encoder_factory), std::move(video_decoder_factory),
>-          audio_mixer, audio_processing);
>-
>-  std::unique_ptr<CallFactoryInterface> call_factory = CreateCallFactory();
>-
>-  std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory =
>-      CreateRtcEventLogFactory();
>-
>-  return CreateModularPeerConnectionFactory(
>-      network_thread, worker_thread, signaling_thread, std::move(media_engine),
>-      std::move(call_factory), std::move(event_log_factory));
>-}
>-
>-#if defined(USE_BUILTIN_SW_CODECS)
>-rtc::scoped_refptr<PeerConnectionFactoryInterface>
>-CreatePeerConnectionFactoryWithAudioMixer(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* signaling_thread,
>-    AudioDeviceModule* default_adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>-    rtc::scoped_refptr<AudioMixer> audio_mixer) {
>-  return CreatePeerConnectionFactory(
>-      network_thread, worker_thread, signaling_thread, default_adm,
>-      audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
>-      video_decoder_factory, audio_mixer, nullptr);
>-}
>-
>-rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>-    rtc::Thread* network_thread,
>-    rtc::Thread* worker_thread,
>-    rtc::Thread* signaling_thread,
>-    AudioDeviceModule* default_adm,
>-    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>-    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory) {
>-  return CreatePeerConnectionFactoryWithAudioMixer(
>-      network_thread, worker_thread, signaling_thread, default_adm,
>-      audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
>-      video_decoder_factory, nullptr);
>-}
>-#endif
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.cc
>index f819d26695760515d08fe194ba0131520370408d..e989586607d77bf3a41f9283d524d3ad3f73b0e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.cc
>@@ -118,6 +118,10 @@ rtc::scoped_refptr<DataChannel> DataChannel::Create(
>   return channel;
> }
> 
>+bool DataChannel::IsSctpLike(cricket::DataChannelType type) {
>+  return type == cricket::DCT_SCTP || type == cricket::DCT_MEDIA_TRANSPORT;
>+}
>+
> DataChannel::DataChannel(DataChannelProviderInterface* provider,
>                          cricket::DataChannelType dct,
>                          const std::string& label)
>@@ -147,7 +151,7 @@ bool DataChannel::Init(const InternalDataChannelInit& config) {
>       return false;
>     }
>     handshake_state_ = kHandshakeReady;
>-  } else if (data_channel_type_ == cricket::DCT_SCTP) {
>+  } else if (IsSctpLike(data_channel_type_)) {
>     if (config.id < -1 || config.maxRetransmits < -1 ||
>         config.maxRetransmitTime < -1) {
>       RTC_LOG(LS_ERROR) << "Failed to initialize the SCTP data channel due to "
>@@ -241,7 +245,7 @@ bool DataChannel::Send(const DataBuffer& buffer) {
>   if (!queued_send_data_.Empty()) {
>     // Only SCTP DataChannel queues the outgoing data when the transport is
>     // blocked.
>-    RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
>+    RTC_DCHECK(IsSctpLike(data_channel_type_));
>     if (!QueueSendDataMessage(buffer)) {
>       RTC_LOG(LS_ERROR) << "Closing the DataChannel due to a failure to queue "
>                            "additional data.";
>@@ -273,7 +277,7 @@ void DataChannel::SetReceiveSsrc(uint32_t receive_ssrc) {
> void DataChannel::SetSctpSid(int sid) {
>   RTC_DCHECK_LT(config_.id, 0);
>   RTC_DCHECK_GE(sid, 0);
>-  RTC_DCHECK_EQ(data_channel_type_, cricket::DCT_SCTP);
>+  RTC_DCHECK(IsSctpLike(data_channel_type_));
>   if (config_.id == sid) {
>     return;
>   }
>@@ -283,7 +287,7 @@ void DataChannel::SetSctpSid(int sid) {
> }
> 
> void DataChannel::OnClosingProcedureStartedRemotely(int sid) {
>-  if (data_channel_type_ == cricket::DCT_SCTP && sid == config_.id &&
>+  if (IsSctpLike(data_channel_type_) && sid == config_.id &&
>       state_ != kClosing && state_ != kClosed) {
>     // Don't bother sending queued data since the side that initiated the
>     // closure wouldn't receive it anyway. See crbug.com/559394 for a lengthy
>@@ -299,7 +303,7 @@ void DataChannel::OnClosingProcedureStartedRemotely(int sid) {
> }
> 
> void DataChannel::OnClosingProcedureComplete(int sid) {
>-  if (data_channel_type_ == cricket::DCT_SCTP && sid == config_.id) {
>+  if (IsSctpLike(data_channel_type_) && sid == config_.id) {
>     // If the closing procedure is complete, we should have finished sending
>     // all pending data and transitioned to kClosing already.
>     RTC_DCHECK_EQ(state_, kClosing);
>@@ -310,7 +314,7 @@ void DataChannel::OnClosingProcedureComplete(int sid) {
> }
> 
> void DataChannel::OnTransportChannelCreated() {
>-  RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
>+  RTC_DCHECK(IsSctpLike(data_channel_type_));
>   if (!connected_to_provider_) {
>     connected_to_provider_ = provider_->ConnectDataChannel(this);
>   }
>@@ -348,12 +352,12 @@ void DataChannel::OnDataReceived(const cricket::ReceiveDataParams& params,
>   if (data_channel_type_ == cricket::DCT_RTP && params.ssrc != receive_ssrc_) {
>     return;
>   }
>-  if (data_channel_type_ == cricket::DCT_SCTP && params.sid != config_.id) {
>+  if (IsSctpLike(data_channel_type_) && params.sid != config_.id) {
>     return;
>   }
> 
>   if (params.type == cricket::DMT_CONTROL) {
>-    RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
>+    RTC_DCHECK(IsSctpLike(data_channel_type_));
>     if (handshake_state_ != kHandshakeWaitingForAck) {
>       // Ignore it if we are not expecting an ACK message.
>       RTC_LOG(LS_WARNING)
>@@ -570,7 +574,7 @@ bool DataChannel::SendDataMessage(const DataBuffer& buffer,
>                                   bool queue_if_blocked) {
>   cricket::SendDataParams send_params;
> 
>-  if (data_channel_type_ == cricket::DCT_SCTP) {
>+  if (IsSctpLike(data_channel_type_)) {
>     send_params.ordered = config_.ordered;
>     // Send as ordered if it is still going through OPEN/ACK signaling.
>     if (handshake_state_ != kHandshakeReady && !config_.ordered) {
>@@ -597,7 +601,7 @@ bool DataChannel::SendDataMessage(const DataBuffer& buffer,
>     return true;
>   }
> 
>-  if (data_channel_type_ != cricket::DCT_SCTP) {
>+  if (!IsSctpLike(data_channel_type_)) {
>     return false;
>   }
> 
>@@ -649,7 +653,7 @@ void DataChannel::QueueControlMessage(const rtc::CopyOnWriteBuffer& buffer) {
> bool DataChannel::SendControlMessage(const rtc::CopyOnWriteBuffer& buffer) {
>   bool is_open_message = handshake_state_ == kHandshakeShouldSendOpen;
> 
>-  RTC_DCHECK_EQ(data_channel_type_, cricket::DCT_SCTP);
>+  RTC_DCHECK(IsSctpLike(data_channel_type_));
>   RTC_DCHECK(writable_);
>   RTC_DCHECK_GE(config_.id, 0);
>   RTC_DCHECK(!is_open_message || !config_.negotiated);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.h
>index cbb0c8b2cf0aba568ca7dbff4b3d1b93a5cb95b1..22ea354c211632db86e7942348bd5c328664f7f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.h
>@@ -122,6 +122,8 @@ class DataChannel : public DataChannelInterface, public sigslot::has_slots<> {
>       const std::string& label,
>       const InternalDataChannelInit& config);
> 
>+  static bool IsSctpLike(cricket::DataChannelType type);
>+
>   virtual void RegisterObserver(DataChannelObserver* observer);
>   virtual void UnregisterObserver();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.cc
>index e31b2f5872ca96b573bc8124a05cd1d38020cd84..2835d34a4677654972b54b7f6033a1f420396333 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.cc
>@@ -301,6 +301,8 @@ void DtlsSrtpTransport::OnDtlsState(cricket::DtlsTransportInternal* transport,
>   RTC_DCHECK(transport == rtp_dtls_transport_ ||
>              transport == rtcp_dtls_transport_);
> 
>+  SignalDtlsStateChange();
>+
>   if (state != cricket::DTLS_TRANSPORT_CONNECTED) {
>     ResetParams();
>     return;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.h
>index 498f02e917622b73004946061158fe0986f7eb71..cac560e79ace85af3672835c4dabf95c0ccaf3dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.h
>@@ -42,7 +42,8 @@ class DtlsSrtpTransport : public SrtpTransport {
>   void UpdateRecvEncryptedHeaderExtensionIds(
>       const std::vector<int>& recv_extension_ids);
> 
>-  sigslot::signal2<DtlsSrtpTransport*, bool> SignalDtlsSrtpSetupFailure;
>+  sigslot::signal<DtlsSrtpTransport*, bool> SignalDtlsSrtpSetupFailure;
>+  sigslot::signal<> SignalDtlsStateChange;
> 
>   RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override {
>     return RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepicecandidate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepicecandidate.cc
>index 277f1bab4fff10d384f7d13a74fae3a8d48a78be..2133d9dadd258184b49776897cc339ffa2c6b87a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepicecandidate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepicecandidate.cc
>@@ -14,7 +14,6 @@
> 
> #include "absl/memory/memory.h"
> #include "pc/webrtcsdp.h"
>-#include "rtc_base/stringencode.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription.cc
>index 64a7a2bd591b0a691ed21c91a6f8ba73954e4d32..13d074380820053428f23eda1985c182a42d0969 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription.cc
>@@ -17,7 +17,6 @@
> #include "pc/mediasession.h"
> #include "pc/webrtcsdp.h"
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/stringencode.h"
> 
> using cricket::SessionDescription;
> 
>@@ -93,37 +92,9 @@ void UpdateConnectionAddress(
> 
> }  // namespace
> 
>-const char SessionDescriptionInterface::kOffer[] = "offer";
>-const char SessionDescriptionInterface::kPrAnswer[] = "pranswer";
>-const char SessionDescriptionInterface::kAnswer[] = "answer";
>-
> const int JsepSessionDescription::kDefaultVideoCodecId = 100;
> const char JsepSessionDescription::kDefaultVideoCodecName[] = "VP8";
> 
>-const char* SdpTypeToString(SdpType type) {
>-  switch (type) {
>-    case SdpType::kOffer:
>-      return SessionDescriptionInterface::kOffer;
>-    case SdpType::kPrAnswer:
>-      return SessionDescriptionInterface::kPrAnswer;
>-    case SdpType::kAnswer:
>-      return SessionDescriptionInterface::kAnswer;
>-  }
>-  return "";
>-}
>-
>-absl::optional<SdpType> SdpTypeFromString(const std::string& type_str) {
>-  if (type_str == SessionDescriptionInterface::kOffer) {
>-    return SdpType::kOffer;
>-  } else if (type_str == SessionDescriptionInterface::kPrAnswer) {
>-    return SdpType::kPrAnswer;
>-  } else if (type_str == SessionDescriptionInterface::kAnswer) {
>-    return SdpType::kAnswer;
>-  } else {
>-    return absl::nullopt;
>-  }
>-}
>-
> // TODO(steveanton): Remove this default implementation once Chromium has been
> // updated.
> SdpType SessionDescriptionInterface::GetType() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.cc
>index c3dfd32464ded2579a194c01d7ec1da8102f6dd5..88d20a45b17927f41d3625b14b11cf05ec4a1cb0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.cc
>@@ -13,7 +13,6 @@
> #include <memory>
> #include <utility>  // for std::pair
> 
>-#include "absl/memory/memory.h"
> #include "api/candidate.h"
> #include "p2p/base/p2pconstants.h"
> #include "p2p/base/p2ptransportchannel.h"
>@@ -94,11 +93,13 @@ JsepTransport::JsepTransport(
>     std::unique_ptr<webrtc::SrtpTransport> sdes_transport,
>     std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport,
>     std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport,
>-    std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport)
>+    std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport,
>+    std::unique_ptr<webrtc::MediaTransportInterface> media_transport)
>     : mid_(mid),
>       local_certificate_(local_certificate),
>       rtp_dtls_transport_(std::move(rtp_dtls_transport)),
>-      rtcp_dtls_transport_(std::move(rtcp_dtls_transport)) {
>+      rtcp_dtls_transport_(std::move(rtcp_dtls_transport)),
>+      media_transport_(std::move(media_transport)) {
>   RTC_DCHECK(rtp_dtls_transport_);
>   if (unencrypted_rtp_transport) {
>     RTC_DCHECK(!sdes_transport);
>@@ -114,9 +115,17 @@ JsepTransport::JsepTransport(
>     RTC_DCHECK(!sdes_transport);
>     dtls_srtp_transport_ = std::move(dtls_srtp_transport);
>   }
>+
>+  if (media_transport_) {
>+    media_transport_->SetMediaTransportStateCallback(this);
>+  }
> }
> 
>-JsepTransport::~JsepTransport() {}
>+JsepTransport::~JsepTransport() {
>+  if (media_transport_) {
>+    media_transport_->SetMediaTransportStateCallback(nullptr);
>+  }
>+}
> 
> webrtc::RTCError JsepTransport::SetLocalJsepTransportDescription(
>     const JsepTransportDescription& jsep_description,
>@@ -315,8 +324,9 @@ webrtc::RTCError JsepTransport::VerifyCertificateFingerprint(
>     return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>                             "Fingerprint provided but no identity available.");
>   }
>-  std::unique_ptr<rtc::SSLFingerprint> fp_tmp(rtc::SSLFingerprint::Create(
>-      fingerprint->algorithm, certificate->identity()));
>+  std::unique_ptr<rtc::SSLFingerprint> fp_tmp =
>+      rtc::SSLFingerprint::CreateUnique(fingerprint->algorithm,
>+                                        *certificate->identity());
>   RTC_DCHECK(fp_tmp.get() != NULL);
>   if (*fp_tmp == *fingerprint) {
>     return webrtc::RTCError::OK();
>@@ -505,7 +515,8 @@ webrtc::RTCError JsepTransport::NegotiateAndSetDtlsParameters(
>         "Local fingerprint supplied when caller didn't offer DTLS.");
>   } else {
>     // We are not doing DTLS
>-    remote_fingerprint = absl::make_unique<rtc::SSLFingerprint>("", nullptr, 0);
>+    remote_fingerprint = absl::make_unique<rtc::SSLFingerprint>(
>+        "", rtc::ArrayView<const uint8_t>());
>   }
>   // Now that we have negotiated everything, push it downward.
>   // Note that we cache the result so that if we have race conditions
>@@ -633,4 +644,12 @@ bool JsepTransport::GetTransportStats(DtlsTransportInternal* dtls_transport,
>   return true;
> }
> 
>+void JsepTransport::OnStateChanged(webrtc::MediaTransportState state) {
>+  // TODO(bugs.webrtc.org/9719) This method currently fires on the network
>+  // thread, but media transport does not make such guarantees. We need to make
>+  // sure this callback is guaranteed to be executed on the network thread.
>+  media_transport_state_ = state;
>+  SignalMediaTransportStateChanged();
>+}
>+
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.h
>index 8e898533ef97c1ec6d2bee478ffa96d20501fed0..952f2ccb8f74848f6465871070dea35b08c16200 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.h
>@@ -19,6 +19,7 @@
> #include "absl/types/optional.h"
> #include "api/candidate.h"
> #include "api/jsep.h"
>+#include "api/media_transport_interface.h"
> #include "p2p/base/dtlstransport.h"
> #include "p2p/base/p2pconstants.h"
> #include "p2p/base/transportinfo.h"
>@@ -70,11 +71,16 @@ struct JsepTransportDescription {
> //
> // On Threading: JsepTransport performs work solely on the network thread, and
> // so its methods should only be called on the network thread.
>-class JsepTransport : public sigslot::has_slots<> {
>+class JsepTransport : public sigslot::has_slots<>,
>+                      public webrtc::MediaTransportStateCallback {
>  public:
>   // |mid| is just used for log statements in order to identify the Transport.
>   // Note that |local_certificate| is allowed to be null since a remote
>   // description may be set before a local certificate is generated.
>+  //
>+  // |media_trasport| is optional (experimental). If available it will be used
>+  // to send / receive encoded audio and video frames instead of RTP.
>+  // Currently |media_transport| can co-exist with RTP / RTCP transports.
>   JsepTransport(
>       const std::string& mid,
>       const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate,
>@@ -82,7 +88,8 @@ class JsepTransport : public sigslot::has_slots<> {
>       std::unique_ptr<webrtc::SrtpTransport> sdes_transport,
>       std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport,
>       std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport,
>-      std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport);
>+      std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport,
>+      std::unique_ptr<webrtc::MediaTransportInterface> media_transport);
> 
>   ~JsepTransport() override;
> 
>@@ -158,11 +165,26 @@ class JsepTransport : public sigslot::has_slots<> {
>     return rtcp_dtls_transport_.get();
>   }
> 
>+  // Returns media transport, if available.
>+  // Note that media transport is owned by jseptransport and the pointer
>+  // to media transport will becomes invalid after destruction of jseptransport.
>+  webrtc::MediaTransportInterface* media_transport() const {
>+    return media_transport_.get();
>+  }
>+
>+  // Returns the latest media transport state.
>+  webrtc::MediaTransportState media_transport_state() const {
>+    return media_transport_state_;
>+  }
>+
>   // This is signaled when RTCP-mux becomes active and
>   // |rtcp_dtls_transport_| is destroyed. The JsepTransportController will
>   // handle the signal and update the aggregate transport states.
>   sigslot::signal<> SignalRtcpMuxActive;
> 
>+  // This is signaled for changes in |media_transport_| state.
>+  sigslot::signal<> SignalMediaTransportStateChanged;
>+
>   // TODO(deadbeef): The methods below are only public for testing. Should make
>   // them utility functions or objects so they can be tested independently from
>   // this class.
>@@ -218,6 +240,9 @@ class JsepTransport : public sigslot::has_slots<> {
>   bool GetTransportStats(DtlsTransportInternal* dtls_transport,
>                          TransportStats* stats);
> 
>+  // Invoked whenever the state of the media transport changes.
>+  void OnStateChanged(webrtc::MediaTransportState state) override;
>+
>   const std::string mid_;
>   // needs-ice-restart bit as described in JSEP.
>   bool needs_ice_restart_ = false;
>@@ -241,6 +266,14 @@ class JsepTransport : public sigslot::has_slots<> {
>   absl::optional<std::vector<int>> send_extension_ids_;
>   absl::optional<std::vector<int>> recv_extension_ids_;
> 
>+  // Optional media transport (experimental).
>+  std::unique_ptr<webrtc::MediaTransportInterface> media_transport_;
>+
>+  // If |media_transport_| is provided, this variable represents the state of
>+  // media transport.
>+  webrtc::MediaTransportState media_transport_state_ =
>+      webrtc::MediaTransportState::kPending;
>+
>   RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransport);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport_unittest.cc
>index 56a742c338bc7fe03ad3c21878e1ca40f091cc75..e518301ce2c862fc75f6491aff4ad03fbc169e6b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport_unittest.cc
>@@ -98,11 +98,16 @@ class JsepTransport2Test : public testing::Test, public sigslot::has_slots<> {
>         RTC_NOTREACHED();
>     }
> 
>+    // TODO(sukhanov): Currently there is no media_transport specific
>+    // logic in jseptransport, so jseptransport unittests are created with
>+    // media_transport = nullptr. In the future we will probably add
>+    // more logic that require unit tests. Note that creation of media_transport
>+    // is covered in jseptransportcontroller_unittest.
>     auto jsep_transport = absl::make_unique<JsepTransport>(
>         kTransportName, /*local_certificate=*/nullptr,
>         std::move(unencrypted_rtp_transport), std::move(sdes_transport),
>         std::move(dtls_srtp_transport), std::move(rtp_dtls_transport),
>-        std::move(rtcp_dtls_transport));
>+        std::move(rtcp_dtls_transport), /*media_transport=*/nullptr);
> 
>     signal_rtcp_mux_active_received_ = false;
>     jsep_transport->SignalRtcpMuxActive.connect(
>@@ -121,7 +126,7 @@ class JsepTransport2Test : public testing::Test, public sigslot::has_slots<> {
> 
>     std::unique_ptr<rtc::SSLFingerprint> fingerprint;
>     if (cert) {
>-      fingerprint.reset(rtc::SSLFingerprint::CreateFromCertificate(cert));
>+      fingerprint = rtc::SSLFingerprint::CreateFromCertificate(*cert);
>     }
>     jsep_description.transport_desc =
>         TransportDescription(std::vector<std::string>(), ufrag, pwd,
>@@ -378,11 +383,12 @@ TEST_P(JsepTransport2WithRtcpMux, VerifyCertificateFingerprint) {
>     ASSERT_NE(nullptr, certificate);
> 
>     std::string digest_algorithm;
>-    ASSERT_TRUE(certificate->ssl_certificate().GetSignatureDigestAlgorithm(
>+    ASSERT_TRUE(certificate->GetSSLCertificate().GetSignatureDigestAlgorithm(
>         &digest_algorithm));
>     ASSERT_FALSE(digest_algorithm.empty());
>-    std::unique_ptr<rtc::SSLFingerprint> good_fingerprint(
>-        rtc::SSLFingerprint::Create(digest_algorithm, certificate->identity()));
>+    std::unique_ptr<rtc::SSLFingerprint> good_fingerprint =
>+        rtc::SSLFingerprint::CreateUnique(digest_algorithm,
>+                                          *certificate->identity());
>     ASSERT_NE(nullptr, good_fingerprint);
> 
>     EXPECT_TRUE(jsep_transport_
>@@ -1045,7 +1051,7 @@ class JsepTransport2HeaderExtensionTest
>   void OnReadPacket1(rtc::PacketTransportInternal* transport,
>                      const char* data,
>                      size_t size,
>-                     const rtc::PacketTime& time,
>+                     const int64_t& /* packet_time_us */,
>                      int flags) {
>     RTC_LOG(LS_INFO) << "JsepTransport 1 Received a packet.";
>     CompareHeaderExtensions(
>@@ -1058,7 +1064,7 @@ class JsepTransport2HeaderExtensionTest
>   void OnReadPacket2(rtc::PacketTransportInternal* transport,
>                      const char* data,
>                      size_t size,
>-                     const rtc::PacketTime& time,
>+                     const int64_t& /* packet_time_us */,
>                      int flags) {
>     RTC_LOG(LS_INFO) << "JsepTransport 2 Received a packet.";
>     CompareHeaderExtensions(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc
>index 4c69a6a07e29588ce9bfa4dfa7be34855a2ff012..78ecaf31deac4e41e8db488a0df6ca81504d2cb7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc
>@@ -14,10 +14,11 @@
> #include <memory>
> #include <utility>
> 
>-#include "absl/memory/memory.h"
> #include "p2p/base/port.h"
>+#include "pc/srtpfilter.h"
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/key_derivation.h"
> #include "rtc_base/thread.h"
> 
> using webrtc::SdpType;
>@@ -133,6 +134,24 @@ RtpTransportInternal* JsepTransportController::GetRtpTransport(
>   return jsep_transport->rtp_transport();
> }
> 
>+MediaTransportInterface* JsepTransportController::GetMediaTransport(
>+    const std::string& mid) const {
>+  auto jsep_transport = GetJsepTransportForMid(mid);
>+  if (!jsep_transport) {
>+    return nullptr;
>+  }
>+  return jsep_transport->media_transport();
>+}
>+
>+MediaTransportState JsepTransportController::GetMediaTransportState(
>+    const std::string& mid) const {
>+  auto jsep_transport = GetJsepTransportForMid(mid);
>+  if (!jsep_transport) {
>+    return MediaTransportState::kPending;
>+  }
>+  return jsep_transport->media_transport_state();
>+}
>+
> cricket::DtlsTransportInternal* JsepTransportController::GetDtlsTransport(
>     const std::string& mid) const {
>   auto jsep_transport = GetJsepTransportForMid(mid);
>@@ -370,6 +389,15 @@ void JsepTransportController::SetActiveResetSrtpParams(
>   }
> }
> 
>+void JsepTransportController::SetMediaTransportFactory(
>+    MediaTransportFactory* media_transport_factory) {
>+  RTC_DCHECK(media_transport_factory == config_.media_transport_factory ||
>+             jsep_transports_by_name_.empty())
>+      << "You can only call SetMediaTransportFactory before "
>+         "JsepTransportController created its first transport.";
>+  config_.media_transport_factory = media_transport_factory;
>+}
>+
> std::unique_ptr<cricket::DtlsTransportInternal>
> JsepTransportController::CreateDtlsTransport(const std::string& transport_name,
>                                              bool rtcp) {
>@@ -471,6 +499,8 @@ JsepTransportController::CreateDtlsSrtpTransport(
>                                          rtcp_dtls_transport);
>   dtls_srtp_transport->SetActiveResetSrtpParams(
>       config_.active_reset_srtp_params);
>+  dtls_srtp_transport->SignalDtlsStateChange.connect(
>+      this, &JsepTransportController::UpdateAggregateStates_n);
>   return dtls_srtp_transport;
> }
> 
>@@ -523,7 +553,7 @@ RTCError JsepTransportController::ApplyDescription_n(
>         (IsBundled(content_info.name) && content_info.name != *bundled_mid())) {
>       continue;
>     }
>-    error = MaybeCreateJsepTransport(content_info);
>+    error = MaybeCreateJsepTransport(local, content_info);
>     if (!error.ok()) {
>       return error;
>     }
>@@ -732,6 +762,10 @@ bool JsepTransportController::HandleBundledContent(
>   // BaseChannel/SctpTransport change the RtpTransport/DtlsTransport first,
>   // then destroy the cricket::JsepTransport.
>   if (SetTransportForMid(content_info.name, jsep_transport)) {
>+    // TODO(bugs.webrtc.org/9719) For media transport this is far from ideal,
>+    // because it means that we first create media transport and start
>+    // connecting it, and then we destroy it. We will need to address it before
>+    // video path is enabled.
>     MaybeDestroyJsepTransport(content_info.name);
>     return true;
>   }
>@@ -749,12 +783,12 @@ bool JsepTransportController::SetTransportForMid(
>   mid_to_transport_[mid] = jsep_transport;
>   return config_.transport_observer->OnTransportChanged(
>       mid, jsep_transport->rtp_transport(),
>-      jsep_transport->rtp_dtls_transport());
>+      jsep_transport->rtp_dtls_transport(), jsep_transport->media_transport());
> }
> 
> void JsepTransportController::RemoveTransportForMid(const std::string& mid) {
>-  bool ret =
>-      config_.transport_observer->OnTransportChanged(mid, nullptr, nullptr);
>+  bool ret = config_.transport_observer->OnTransportChanged(mid, nullptr,
>+                                                            nullptr, nullptr);
>   // Calling OnTransportChanged with nullptr should always succeed, since it is
>   // only expected to fail when adding media to a transport (not removing).
>   RTC_DCHECK(ret);
>@@ -806,7 +840,7 @@ std::vector<int> JsepTransportController::GetEncryptedHeaderExtensionIds(
>       static_cast<const cricket::MediaContentDescription*>(
>           content_info.description);
> 
>-  if (!config_.crypto_options.enable_encrypted_rtp_header_extensions) {
>+  if (!config_.crypto_options.srtp.enable_encrypted_rtp_header_extensions) {
>     return std::vector<int>();
>   }
> 
>@@ -888,7 +922,94 @@ cricket::JsepTransport* JsepTransportController::GetJsepTransportByName(
>   return (it == jsep_transports_by_name_.end()) ? nullptr : it->second.get();
> }
> 
>+std::unique_ptr<webrtc::MediaTransportInterface>
>+JsepTransportController::MaybeCreateMediaTransport(
>+    const cricket::ContentInfo& content_info,
>+    bool local,
>+    cricket::IceTransportInternal* ice_transport) {
>+  absl::optional<cricket::CryptoParams> selected_crypto_for_media_transport;
>+  if (content_info.media_description() &&
>+      !content_info.media_description()->cryptos().empty()) {
>+    // Order of cryptos is deterministic (rfc4568, 5.1.1), so we just select the
>+    // first one (in fact the first one should be the most preferred one.) We
>+    // ignore the HMAC size, as media transport crypto settings currently don't
>+    // expose HMAC size, nor crypto protocol for that matter.
>+    selected_crypto_for_media_transport =
>+        content_info.media_description()->cryptos()[0];
>+  }
>+
>+  if (config_.media_transport_factory != nullptr) {
>+    if (!selected_crypto_for_media_transport.has_value()) {
>+      RTC_LOG(LS_WARNING) << "a=cryto line was not found in the offer. Most "
>+                             "likely you did not enable SDES. "
>+                             "Make sure to pass config.enable_dtls_srtp=false "
>+                             "to RTCConfiguration. "
>+                             "Cannot continue with media transport. Falling "
>+                             "back to RTP. is_local="
>+                          << local;
>+
>+      // Remove media_transport_factory from config, because we don't want to
>+      // use it on the subsequent call (for the other side of the offer).
>+      config_.media_transport_factory = nullptr;
>+    } else {
>+      // Note that we ignore here lifetime and length.
>+      // In fact we take those bits (inline, lifetime and length) and keep it as
>+      // part of key derivation.
>+      //
>+      // Technically, we are also not following rfc4568, which requires us to
>+      // send and answer with the key that we chose. In practice, for media
>+      // transport, the current approach should be sufficient (we take the key
>+      // that sender offered, and caller assumes we will use it. We are not
>+      // signaling back that we indeed used it.)
>+      std::unique_ptr<rtc::KeyDerivation> key_derivation =
>+          rtc::KeyDerivation::Create(rtc::KeyDerivationAlgorithm::HKDF_SHA256);
>+      const std::string label = "MediaTransportLabel";
>+      constexpr int kDerivedKeyByteSize = 32;
>+
>+      int key_len, salt_len;
>+      if (!rtc::GetSrtpKeyAndSaltLengths(
>+              rtc::SrtpCryptoSuiteFromName(
>+                  selected_crypto_for_media_transport.value().cipher_suite),
>+              &key_len, &salt_len)) {
>+        RTC_CHECK(false) << "Cannot set up secure media transport";
>+      }
>+      rtc::ZeroOnFreeBuffer<uint8_t> raw_key(key_len + salt_len);
>+
>+      cricket::SrtpFilter::ParseKeyParams(
>+          selected_crypto_for_media_transport.value().key_params,
>+          raw_key.data(), raw_key.size());
>+      absl::optional<rtc::ZeroOnFreeBuffer<uint8_t>> key =
>+          key_derivation->DeriveKey(
>+              raw_key,
>+              /*salt=*/nullptr,
>+              rtc::ArrayView<const uint8_t>(
>+                  reinterpret_cast<const uint8_t*>(label.data()), label.size()),
>+              kDerivedKeyByteSize);
>+
>+      // We want to crash the app if we don't have a key, and not silently fall
>+      // back to the unsecure communication.
>+      RTC_CHECK(key.has_value());
>+      MediaTransportSettings settings;
>+      settings.is_caller = local;
>+      settings.pre_shared_key =
>+          std::string(reinterpret_cast<const char*>(key.value().data()),
>+                      key.value().size());
>+      auto media_transport_result =
>+          config_.media_transport_factory->CreateMediaTransport(
>+              ice_transport, network_thread_, settings);
>+
>+      // TODO(sukhanov): Proper error handling.
>+      RTC_CHECK(media_transport_result.ok());
>+
>+      return media_transport_result.MoveValue();
>+    }
>+  }
>+
>+  return nullptr;
>+}
>+
> RTCError JsepTransportController::MaybeCreateJsepTransport(
>+    bool local,
>     const cricket::ContentInfo& content_info) {
>   RTC_DCHECK(network_thread_->IsCurrent());
>   cricket::JsepTransport* transport = GetJsepTransportByName(content_info.name);
>@@ -906,17 +1027,24 @@ RTCError JsepTransportController::MaybeCreateJsepTransport(
> 
>   std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport =
>       CreateDtlsTransport(content_info.name, /*rtcp =*/false);
>+
>   std::unique_ptr<cricket::DtlsTransportInternal> rtcp_dtls_transport;
>+  std::unique_ptr<RtpTransport> unencrypted_rtp_transport;
>+  std::unique_ptr<SrtpTransport> sdes_transport;
>+  std::unique_ptr<DtlsSrtpTransport> dtls_srtp_transport;
>+  std::unique_ptr<MediaTransportInterface> media_transport;
>+
>   if (config_.rtcp_mux_policy !=
>           PeerConnectionInterface::kRtcpMuxPolicyRequire &&
>       content_info.type == cricket::MediaProtocolType::kRtp) {
>     rtcp_dtls_transport =
>         CreateDtlsTransport(content_info.name, /*rtcp =*/true);
>   }
>+  media_transport = MaybeCreateMediaTransport(
>+      content_info, local, rtp_dtls_transport->ice_transport());
> 
>-  std::unique_ptr<RtpTransport> unencrypted_rtp_transport;
>-  std::unique_ptr<SrtpTransport> sdes_transport;
>-  std::unique_ptr<DtlsSrtpTransport> dtls_srtp_transport;
>+  // TODO(sukhanov): Do not create RTP/RTCP transports if media transport is
>+  // used.
>   if (config_.disable_encryption) {
>     unencrypted_rtp_transport = CreateUnencryptedRtpTransport(
>         content_info.name, rtp_dtls_transport.get(), rtcp_dtls_transport.get());
>@@ -932,9 +1060,12 @@ RTCError JsepTransportController::MaybeCreateJsepTransport(
>       absl::make_unique<cricket::JsepTransport>(
>           content_info.name, certificate_, std::move(unencrypted_rtp_transport),
>           std::move(sdes_transport), std::move(dtls_srtp_transport),
>-          std::move(rtp_dtls_transport), std::move(rtcp_dtls_transport));
>+          std::move(rtp_dtls_transport), std::move(rtcp_dtls_transport),
>+          std::move(media_transport));
>   jsep_transport->SignalRtcpMuxActive.connect(
>       this, &JsepTransportController::UpdateAggregateStates_n);
>+  jsep_transport->SignalMediaTransportStateChanged.connect(
>+      this, &JsepTransportController::OnMediaTransportStateChanged_n);
>   SetTransportForMid(content_info.name, jsep_transport.get());
> 
>   jsep_transports_by_name_[content_info.name] = std::move(jsep_transport);
>@@ -956,12 +1087,19 @@ void JsepTransportController::MaybeDestroyJsepTransport(
>       return;
>     }
>   }
>+
>   jsep_transports_by_name_.erase(mid);
>   UpdateAggregateStates_n();
> }
> 
> void JsepTransportController::DestroyAllJsepTransports_n() {
>   RTC_DCHECK(network_thread_->IsCurrent());
>+
>+  for (const auto& jsep_transport : jsep_transports_by_name_) {
>+    config_.transport_observer->OnTransportChanged(jsep_transport.first,
>+                                                   nullptr, nullptr, nullptr);
>+  }
>+
>   jsep_transports_by_name_.clear();
> }
> 
>@@ -1116,18 +1254,35 @@ void JsepTransportController::OnTransportStateChanged_n(
>   UpdateAggregateStates_n();
> }
> 
>+void JsepTransportController::OnMediaTransportStateChanged_n() {
>+  SignalMediaTransportStateChanged();
>+  UpdateAggregateStates_n();
>+}
>+
> void JsepTransportController::UpdateAggregateStates_n() {
>   RTC_DCHECK(network_thread_->IsCurrent());
> 
>   auto dtls_transports = GetDtlsTransports();
>   cricket::IceConnectionState new_connection_state =
>       cricket::kIceConnectionConnecting;
>+  PeerConnectionInterface::IceConnectionState new_ice_connection_state =
>+      PeerConnectionInterface::IceConnectionState::kIceConnectionNew;
>+  PeerConnectionInterface::PeerConnectionState new_combined_state =
>+      PeerConnectionInterface::PeerConnectionState::kNew;
>   cricket::IceGatheringState new_gathering_state = cricket::kIceGatheringNew;
>   bool any_failed = false;
>+
>+  // TODO(http://bugs.webrtc.org/9719) If(when) media_transport disables
>+  // dtls_transports entirely, the below line will have to be changed to account
>+  // for the fact that dtls transports might be absent.
>   bool all_connected = !dtls_transports.empty();
>   bool all_completed = !dtls_transports.empty();
>   bool any_gathering = false;
>   bool all_done_gathering = !dtls_transports.empty();
>+
>+  std::map<IceTransportState, int> ice_state_counts;
>+  std::map<cricket::DtlsTransportState, int> dtls_state_counts;
>+
>   for (const auto& dtls : dtls_transports) {
>     any_failed = any_failed || dtls->ice_transport()->GetState() ==
>                                    cricket::IceTransportState::STATE_FAILED;
>@@ -1144,7 +1299,35 @@ void JsepTransportController::UpdateAggregateStates_n() {
>     all_done_gathering =
>         all_done_gathering && dtls->ice_transport()->gathering_state() ==
>                                   cricket::kIceGatheringComplete;
>+
>+    dtls_state_counts[dtls->dtls_state()]++;
>+    ice_state_counts[dtls->ice_transport()->GetIceTransportState()]++;
>   }
>+
>+  for (auto it = jsep_transports_by_name_.begin();
>+       it != jsep_transports_by_name_.end(); ++it) {
>+    auto jsep_transport = it->second.get();
>+    if (!jsep_transport->media_transport()) {
>+      continue;
>+    }
>+
>+    // There is no 'kIceConnectionDisconnected', so we only need to handle
>+    // connected and completed.
>+    // We treat kClosed as failed, because if it happens before shutting down
>+    // media transports it means that there was a failure.
>+    // MediaTransportInterface allows to flip back and forth between kWritable
>+    // and kPending, but there does not exist an implementation that does that,
>+    // and the contract of jsep transport controller doesn't quite expect that.
>+    // When this happens, we would go from connected to connecting state, but
>+    // this may change in future.
>+    any_failed |= jsep_transport->media_transport_state() ==
>+                  webrtc::MediaTransportState::kClosed;
>+    all_completed &= jsep_transport->media_transport_state() ==
>+                     webrtc::MediaTransportState::kWritable;
>+    all_connected &= jsep_transport->media_transport_state() ==
>+                     webrtc::MediaTransportState::kWritable;
>+  }
>+
>   if (any_failed) {
>     new_connection_state = cricket::kIceConnectionFailed;
>   } else if (all_completed) {
>@@ -1160,6 +1343,127 @@ void JsepTransportController::UpdateAggregateStates_n() {
>                                });
>   }
> 
>+  // Compute the current RTCIceConnectionState as described in
>+  // https://www.w3.org/TR/webrtc/#dom-rtciceconnectionstate.
>+  // The PeerConnection is responsible for handling the "closed" state.
>+  int total_ice_checking = ice_state_counts[IceTransportState::kChecking];
>+  int total_ice_connected = ice_state_counts[IceTransportState::kConnected];
>+  int total_ice_completed = ice_state_counts[IceTransportState::kCompleted];
>+  int total_ice_failed = ice_state_counts[IceTransportState::kFailed];
>+  int total_ice_disconnected =
>+      ice_state_counts[IceTransportState::kDisconnected];
>+  int total_ice_closed = ice_state_counts[IceTransportState::kClosed];
>+  int total_ice_new = ice_state_counts[IceTransportState::kNew];
>+  int total_ice = dtls_transports.size();
>+
>+  if (total_ice_failed > 0) {
>+    // Any of the RTCIceTransports are in the "failed" state.
>+    new_ice_connection_state = PeerConnectionInterface::kIceConnectionFailed;
>+  } else if (total_ice_disconnected > 0) {
>+    // Any of the RTCIceTransports are in the "disconnected" state and none of
>+    // them are in the "failed" state.
>+    new_ice_connection_state =
>+        PeerConnectionInterface::kIceConnectionDisconnected;
>+  } else if (total_ice_checking > 0) {
>+    // Any of the RTCIceTransports are in the "checking" state and none of them
>+    // are in the "disconnected" or "failed" state.
>+    new_ice_connection_state = PeerConnectionInterface::kIceConnectionChecking;
>+  } else if (total_ice_completed + total_ice_closed == total_ice &&
>+             total_ice_completed > 0) {
>+    // All RTCIceTransports are in the "completed" or "closed" state and at
>+    // least one of them is in the "completed" state.
>+    new_ice_connection_state = PeerConnectionInterface::kIceConnectionCompleted;
>+  } else if (total_ice_connected + total_ice_completed + total_ice_closed ==
>+                 total_ice &&
>+             total_ice_connected > 0) {
>+    // All RTCIceTransports are in the "connected", "completed" or "closed"
>+    // state and at least one of them is in the "connected" state.
>+    new_ice_connection_state = PeerConnectionInterface::kIceConnectionConnected;
>+  } else if ((total_ice_new > 0 &&
>+              total_ice_checking + total_ice_disconnected + total_ice_failed ==
>+                  0) ||
>+             total_ice == total_ice_closed) {
>+    // Any of the RTCIceTransports are in the "new" state and none of them are
>+    // in the "checking", "disconnected" or "failed" state, or all
>+    // RTCIceTransports are in the "closed" state, or there are no transports.
>+    new_ice_connection_state = PeerConnectionInterface::kIceConnectionNew;
>+  } else {
>+    RTC_NOTREACHED();
>+  }
>+
>+  if (standardized_ice_connection_state_ != new_ice_connection_state) {
>+    standardized_ice_connection_state_ = new_ice_connection_state;
>+    invoker_.AsyncInvoke<void>(
>+        RTC_FROM_HERE, signaling_thread_, [this, new_ice_connection_state] {
>+          SignalStandardizedIceConnectionState(new_ice_connection_state);
>+        });
>+  }
>+
>+  // Compute the current RTCPeerConnectionState as described in
>+  // https://www.w3.org/TR/webrtc/#dom-rtcpeerconnectionstate.
>+  // The PeerConnection is responsible for handling the "closed" state.
>+  // Note that "connecting" is only a valid state for DTLS transports while
>+  // "checking", "completed" and "disconnected" are only valid for ICE
>+  // transports.
>+  int total_connected = total_ice_connected +
>+                        dtls_state_counts[cricket::DTLS_TRANSPORT_CONNECTED];
>+  int total_dtls_connecting =
>+      dtls_state_counts[cricket::DTLS_TRANSPORT_CONNECTING];
>+  int total_failed =
>+      total_ice_failed + dtls_state_counts[cricket::DTLS_TRANSPORT_FAILED];
>+  int total_closed =
>+      total_ice_closed + dtls_state_counts[cricket::DTLS_TRANSPORT_CLOSED];
>+  int total_new =
>+      total_ice_new + dtls_state_counts[cricket::DTLS_TRANSPORT_NEW];
>+  int total_transports = total_ice * 2;
>+
>+  if (total_failed > 0) {
>+    // Any of the RTCIceTransports or RTCDtlsTransports are in a "failed" state.
>+    new_combined_state = PeerConnectionInterface::PeerConnectionState::kFailed;
>+  } else if (total_ice_disconnected > 0 &&
>+             total_dtls_connecting + total_ice_checking == 0) {
>+    // Any of the RTCIceTransports or RTCDtlsTransports are in the
>+    // "disconnected" state and none of them are in the "failed" or "connecting"
>+    // or "checking" state.
>+    new_combined_state =
>+        PeerConnectionInterface::PeerConnectionState::kDisconnected;
>+  } else if (total_dtls_connecting + total_ice_checking > 0) {
>+    // Any of the RTCIceTransports or RTCDtlsTransports are in the "connecting"
>+    // or "checking" state and none of them is in the "failed" state.
>+    new_combined_state =
>+        PeerConnectionInterface::PeerConnectionState::kConnecting;
>+  } else if (total_connected + total_ice_completed + total_closed ==
>+                 total_transports &&
>+             total_connected + total_ice_completed > 0) {
>+    // All RTCIceTransports and RTCDtlsTransports are in the "connected",
>+    // "completed" or "closed" state and at least one of them is in the
>+    // "connected" or "completed" state.
>+    new_combined_state =
>+        PeerConnectionInterface::PeerConnectionState::kConnected;
>+  } else if ((total_new > 0 && total_dtls_connecting + total_ice_checking +
>+                                       total_failed + total_ice_disconnected ==
>+                                   0) ||
>+             total_transports == total_closed) {
>+    // Any of the RTCIceTransports or RTCDtlsTransports are in the "new" state
>+    // and none of the transports are in the "connecting", "checking", "failed"
>+    // or "disconnected" state, or all transports are in the "closed" state, or
>+    // there are no transports.
>+    //
>+    // Note that if none of the other conditions hold this is guaranteed to be
>+    // true.
>+    new_combined_state = PeerConnectionInterface::PeerConnectionState::kNew;
>+  } else {
>+    RTC_NOTREACHED();
>+  }
>+
>+  if (combined_connection_state_ != new_combined_state) {
>+    combined_connection_state_ = new_combined_state;
>+    invoker_.AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread_,
>+                               [this, new_combined_state] {
>+                                 SignalConnectionState(new_combined_state);
>+                               });
>+  }
>+
>   if (all_done_gathering) {
>     new_gathering_state = cricket::kIceGatheringComplete;
>   } else if (any_gathering) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.h
>index d9340cf357d0e1345db635893e2fb27a34451480..3ed7f5f433b7a6250b345ed96086fe2f383d30c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.h
>@@ -18,6 +18,8 @@
> #include <vector>
> 
> #include "api/candidate.h"
>+#include "api/crypto/cryptooptions.h"
>+#include "api/media_transport_interface.h"
> #include "api/peerconnectioninterface.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "media/sctp/sctptransportinternal.h"
>@@ -32,7 +34,6 @@
> #include "rtc_base/asyncinvoker.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/refcountedobject.h"
>-#include "rtc_base/sslstreamadapter.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
>@@ -56,7 +57,8 @@ class JsepTransportController : public sigslot::has_slots<> {
>     virtual bool OnTransportChanged(
>         const std::string& mid,
>         RtpTransportInternal* rtp_transport,
>-        cricket::DtlsTransportInternal* dtls_transport) = 0;
>+        cricket::DtlsTransportInternal* dtls_transport,
>+        MediaTransportInterface* media_transport) = 0;
>   };
> 
>   struct Config {
>@@ -67,7 +69,7 @@ class JsepTransportController : public sigslot::has_slots<> {
>     rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
>     // |crypto_options| is used to determine if created DTLS transports
>     // negotiate GCM crypto suites or not.
>-    rtc::CryptoOptions crypto_options;
>+    webrtc::CryptoOptions crypto_options;
>     PeerConnectionInterface::BundlePolicy bundle_policy =
>         PeerConnectionInterface::kBundlePolicyBalanced;
>     PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy =
>@@ -79,6 +81,13 @@ class JsepTransportController : public sigslot::has_slots<> {
>     Observer* transport_observer = nullptr;
>     bool active_reset_srtp_params = false;
>     RtcEventLog* event_log = nullptr;
>+
>+    // Optional media transport factory (experimental). If provided it will be
>+    // used to create media_transport and will be used to send / receive
>+    // audio and video frames instead of RTP. Note that currently
>+    // media_transport co-exists with RTP / RTCP transports and uses the same
>+    // underlying ICE transport.
>+    MediaTransportFactory* media_transport_factory = nullptr;
>   };
> 
>   // The ICE related events are signaled on the |signaling_thread|.
>@@ -108,6 +117,9 @@ class JsepTransportController : public sigslot::has_slots<> {
>   cricket::DtlsTransportInternal* GetRtcpDtlsTransport(
>       const std::string& mid) const;
> 
>+  MediaTransportInterface* GetMediaTransport(const std::string& mid) const;
>+  MediaTransportState GetMediaTransportState(const std::string& mid) const;
>+
>   /*********************
>    * ICE-related methods
>    ********************/
>@@ -157,6 +169,12 @@ class JsepTransportController : public sigslot::has_slots<> {
> 
>   void SetActiveResetSrtpParams(bool active_reset_srtp_params);
> 
>+  // Allows to overwrite the settings from config. You may set or reset the
>+  // media transport factory on the jsep transport controller, as long as you
>+  // did not call 'GetMediaTransport' or 'MaybeCreateJsepTransport'. Once Jsep
>+  // transport is created, you can't change this setting.
>+  void SetMediaTransportFactory(MediaTransportFactory* media_transport_factory);
>+
>   // All of these signals are fired on the signaling thread.
> 
>   // If any transport failed => failed,
>@@ -165,6 +183,11 @@ class JsepTransportController : public sigslot::has_slots<> {
>   // Else => connecting
>   sigslot::signal1<cricket::IceConnectionState> SignalIceConnectionState;
> 
>+  sigslot::signal1<PeerConnectionInterface::PeerConnectionState>
>+      SignalConnectionState;
>+  sigslot::signal1<PeerConnectionInterface::IceConnectionState>
>+      SignalStandardizedIceConnectionState;
>+
>   // If all transports done gathering => complete,
>   // Else if any are gathering => gathering,
>   // Else => new
>@@ -179,6 +202,8 @@ class JsepTransportController : public sigslot::has_slots<> {
> 
>   sigslot::signal1<rtc::SSLHandshakeError> SignalDtlsHandshakeError;
> 
>+  sigslot::signal<> SignalMediaTransportStateChanged;
>+
>  private:
>   RTCError ApplyDescription_n(bool local,
>                               SdpType type,
>@@ -241,7 +266,20 @@ class JsepTransportController : public sigslot::has_slots<> {
>   cricket::JsepTransport* GetJsepTransportByName(
>       const std::string& transport_name);
> 
>-  RTCError MaybeCreateJsepTransport(const cricket::ContentInfo& content_info);
>+  // Creates jsep transport. Noop if transport is already created.
>+  // Transport is created either during SetLocalDescription (|local| == true) or
>+  // during SetRemoteDescription (|local| == false). Passing |local| helps to
>+  // differentiate initiator (caller) from answerer (callee).
>+  RTCError MaybeCreateJsepTransport(bool local,
>+                                    const cricket::ContentInfo& content_info);
>+
>+  // Creates media transport if config wants to use it, and pre-shared key is
>+  // provided in content info. It modifies the config to disable media transport
>+  // if pre-shared key is not provided.
>+  std::unique_ptr<webrtc::MediaTransportInterface> MaybeCreateMediaTransport(
>+      const cricket::ContentInfo& content_info,
>+      bool local,
>+      cricket::IceTransportInternal* ice_transport);
>   void MaybeDestroyJsepTransport(const std::string& mid);
>   void DestroyAllJsepTransports_n();
> 
>@@ -285,6 +323,7 @@ class JsepTransportController : public sigslot::has_slots<> {
>                                       const cricket::Candidates& candidates);
>   void OnTransportRoleConflict_n(cricket::IceTransportInternal* transport);
>   void OnTransportStateChanged_n(cricket::IceTransportInternal* transport);
>+  void OnMediaTransportStateChanged_n();
> 
>   void UpdateAggregateStates_n();
> 
>@@ -301,9 +340,16 @@ class JsepTransportController : public sigslot::has_slots<> {
>   // (BaseChannel/SctpTransport) and the JsepTransport underneath.
>   std::map<std::string, cricket::JsepTransport*> mid_to_transport_;
> 
>-  // Aggregate state for Transports.
>+  // Aggregate states for Transports.
>+  // standardized_ice_connection_state_ is intended to replace
>+  // ice_connection_state, see bugs.webrtc.org/9308
>   cricket::IceConnectionState ice_connection_state_ =
>       cricket::kIceConnectionConnecting;
>+  PeerConnectionInterface::IceConnectionState
>+      standardized_ice_connection_state_ =
>+          PeerConnectionInterface::kIceConnectionNew;
>+  PeerConnectionInterface::PeerConnectionState combined_connection_state_ =
>+      PeerConnectionInterface::PeerConnectionState::kNew;
>   cricket::IceGatheringState ice_gathering_state_ = cricket::kIceGatheringNew;
> 
>   Config config_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller_unittest.cc
>index 6f8693bea7a9b57150258d0486c0a7f24eefb3f7..129d22a4fcc84730e1f99b7a854e967ca2ef9a0f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller_unittest.cc
>@@ -11,7 +11,8 @@
> #include <map>
> #include <memory>
> 
>-#include "absl/memory/memory.h"
>+#include "api/media_transport_interface.h"
>+#include "api/test/fake_media_transport.h"
> #include "p2p/base/fakedtlstransport.h"
> #include "p2p/base/fakeicetransport.h"
> #include "p2p/base/transportfactoryinterface.h"
>@@ -41,6 +42,20 @@ static const char kDataMid1[] = "data1";
> 
> namespace webrtc {
> 
>+namespace {
>+
>+// Media transport factory requires crypto settings to be present in order to
>+// create media transport.
>+void AddCryptoSettings(cricket::SessionDescription* description) {
>+  for (auto& content : description->contents()) {
>+    content.media_description()->AddCrypto(cricket::CryptoParams(
>+        /*t=*/0, std::string(rtc::CS_AES_CM_128_HMAC_SHA1_80),
>+        "inline:YUJDZGVmZ2hpSktMbW9QUXJzVHVWd3l6MTIzNDU2", ""));
>+  }
>+}
>+
>+}  // namespace
>+
> class FakeTransportFactory : public cricket::TransportFactoryInterface {
>  public:
>   std::unique_ptr<cricket::IceTransportInternal> CreateIceTransport(
>@@ -52,7 +67,7 @@ class FakeTransportFactory : public cricket::TransportFactoryInterface {
> 
>   std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport(
>       std::unique_ptr<cricket::IceTransportInternal> ice,
>-      const rtc::CryptoOptions& crypto_options) override {
>+      const webrtc::CryptoOptions& crypto_options) override {
>     std::unique_ptr<cricket::FakeIceTransport> fake_ice(
>         static_cast<cricket::FakeIceTransport*>(ice.release()));
>     return absl::make_unique<FakeDtlsTransport>(std::move(fake_ice));
>@@ -84,6 +99,10 @@ class JsepTransportControllerTest : public JsepTransportController::Observer,
>   void ConnectTransportControllerSignals() {
>     transport_controller_->SignalIceConnectionState.connect(
>         this, &JsepTransportControllerTest::OnConnectionState);
>+    transport_controller_->SignalStandardizedIceConnectionState.connect(
>+        this, &JsepTransportControllerTest::OnStandardizedIceConnectionState);
>+    transport_controller_->SignalConnectionState.connect(
>+        this, &JsepTransportControllerTest::OnCombinedConnectionState);
>     transport_controller_->SignalIceGatheringState.connect(
>         this, &JsepTransportControllerTest::OnGatheringState);
>     transport_controller_->SignalIceCandidatesGathered.connect(
>@@ -170,7 +189,7 @@ class JsepTransportControllerTest : public JsepTransportController::Observer,
>                         rtc::scoped_refptr<rtc::RTCCertificate> cert) {
>     std::unique_ptr<rtc::SSLFingerprint> fingerprint;
>     if (cert) {
>-      fingerprint.reset(rtc::SSLFingerprint::CreateFromCertificate(cert));
>+      fingerprint = rtc::SSLFingerprint::CreateFromCertificate(*cert);
>     }
> 
>     cricket::TransportDescription transport_desc(std::vector<std::string>(),
>@@ -243,6 +262,24 @@ class JsepTransportControllerTest : public JsepTransportController::Observer,
>     ++connection_state_signal_count_;
>   }
> 
>+  void OnStandardizedIceConnectionState(
>+      PeerConnectionInterface::IceConnectionState state) {
>+    if (!signaling_thread_->IsCurrent()) {
>+      signaled_on_non_signaling_thread_ = true;
>+    }
>+    ice_connection_state_ = state;
>+    ++ice_connection_state_signal_count_;
>+  }
>+
>+  void OnCombinedConnectionState(
>+      PeerConnectionInterface::PeerConnectionState state) {
>+    if (!signaling_thread_->IsCurrent()) {
>+      signaled_on_non_signaling_thread_ = true;
>+    }
>+    combined_connection_state_ = state;
>+    ++combined_connection_state_signal_count_;
>+  }
>+
>   void OnGatheringState(cricket::IceGatheringState state) {
>     if (!signaling_thread_->IsCurrent()) {
>       signaled_on_non_signaling_thread_ = true;
>@@ -262,31 +299,37 @@ class JsepTransportControllerTest : public JsepTransportController::Observer,
>   }
> 
>   // JsepTransportController::Observer overrides.
>-  bool OnTransportChanged(
>-      const std::string& mid,
>-      RtpTransportInternal* rtp_transport,
>-      cricket::DtlsTransportInternal* dtls_transport) override {
>+  bool OnTransportChanged(const std::string& mid,
>+                          RtpTransportInternal* rtp_transport,
>+                          cricket::DtlsTransportInternal* dtls_transport,
>+                          MediaTransportInterface* media_transport) override {
>     changed_rtp_transport_by_mid_[mid] = rtp_transport;
>     changed_dtls_transport_by_mid_[mid] = dtls_transport;
>+    changed_media_transport_by_mid_[mid] = media_transport;
>     return true;
>   }
> 
>   // Information received from signals from transport controller.
>   cricket::IceConnectionState connection_state_ =
>       cricket::kIceConnectionConnecting;
>+  PeerConnectionInterface::IceConnectionState ice_connection_state_ =
>+      PeerConnectionInterface::kIceConnectionNew;
>+  PeerConnectionInterface::PeerConnectionState combined_connection_state_ =
>+      PeerConnectionInterface::PeerConnectionState::kNew;
>   bool receiving_ = false;
>   cricket::IceGatheringState gathering_state_ = cricket::kIceGatheringNew;
>   // transport_name => candidates
>   std::map<std::string, Candidates> candidates_;
>   // Counts of each signal emitted.
>   int connection_state_signal_count_ = 0;
>+  int ice_connection_state_signal_count_ = 0;
>+  int combined_connection_state_signal_count_ = 0;
>   int receiving_signal_count_ = 0;
>   int gathering_state_signal_count_ = 0;
>   int candidates_signal_count_ = 0;
> 
>   // |network_thread_| should be destroyed after |transport_controller_|
>   std::unique_ptr<rtc::Thread> network_thread_;
>-  std::unique_ptr<JsepTransportController> transport_controller_;
>   std::unique_ptr<FakeTransportFactory> fake_transport_factory_;
>   rtc::Thread* const signaling_thread_ = nullptr;
>   bool signaled_on_non_signaling_thread_ = false;
>@@ -295,6 +338,12 @@ class JsepTransportControllerTest : public JsepTransportController::Observer,
>   std::map<std::string, RtpTransportInternal*> changed_rtp_transport_by_mid_;
>   std::map<std::string, cricket::DtlsTransportInternal*>
>       changed_dtls_transport_by_mid_;
>+  std::map<std::string, MediaTransportInterface*>
>+      changed_media_transport_by_mid_;
>+
>+  // Transport controller needs to be destroyed first, because it may issue
>+  // callbacks that modify the changed_*_by_mid in the destructor.
>+  std::unique_ptr<JsepTransportController> transport_controller_;
> };
> 
> TEST_F(JsepTransportControllerTest, GetRtpTransport) {
>@@ -341,6 +390,118 @@ TEST_F(JsepTransportControllerTest, GetDtlsTransportWithRtcpMux) {
>   EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kAudioMid1));
>   EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kVideoMid1));
>   EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kVideoMid1));
>+  EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kAudioMid1));
>+}
>+
>+TEST_F(JsepTransportControllerTest, GetMediaTransportInCaller) {
>+  FakeMediaTransportFactory fake_media_transport_factory;
>+  JsepTransportController::Config config;
>+
>+  config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
>+  config.media_transport_factory = &fake_media_transport_factory;
>+  CreateJsepTransportController(config);
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  AddCryptoSettings(description.get());
>+
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
>+      transport_controller_->GetMediaTransport(kAudioMid1));
>+
>+  ASSERT_NE(nullptr, media_transport);
>+
>+  // After SetLocalDescription, media transport should be created as caller.
>+  EXPECT_TRUE(media_transport->is_caller());
>+  EXPECT_TRUE(media_transport->pre_shared_key().has_value());
>+
>+  // Return nullptr for non-existing mids.
>+  EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kVideoMid2));
>+}
>+
>+TEST_F(JsepTransportControllerTest, GetMediaTransportInCallee) {
>+  FakeMediaTransportFactory fake_media_transport_factory;
>+  JsepTransportController::Config config;
>+
>+  config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
>+  config.media_transport_factory = &fake_media_transport_factory;
>+  CreateJsepTransportController(config);
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  AddCryptoSettings(description.get());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
>+      transport_controller_->GetMediaTransport(kAudioMid1));
>+
>+  ASSERT_NE(nullptr, media_transport);
>+
>+  // After SetRemoteDescription, media transport should be created as callee.
>+  EXPECT_FALSE(media_transport->is_caller());
>+  EXPECT_TRUE(media_transport->pre_shared_key().has_value());
>+
>+  // Return nullptr for non-existing mids.
>+  EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kVideoMid2));
>+}
>+
>+TEST_F(JsepTransportControllerTest, GetMediaTransportIsNotSetIfNoSdes) {
>+  FakeMediaTransportFactory fake_media_transport_factory;
>+  JsepTransportController::Config config;
>+
>+  config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
>+  config.media_transport_factory = &fake_media_transport_factory;
>+  CreateJsepTransportController(config);
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kAudioMid1));
>+
>+  // Even if we set local description with crypto now (after the remote offer
>+  // was set), media transport won't be provided.
>+  auto description2 = CreateSessionDescriptionWithoutBundle();
>+  AddCryptoSettings(description2.get());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kAnswer, description2.get())
>+                  .ok());
>+
>+  EXPECT_EQ(nullptr, transport_controller_->GetMediaTransport(kAudioMid1));
>+}
>+
>+TEST_F(JsepTransportControllerTest,
>+       AfterSettingAnswerTheSameMediaTransportIsReturned) {
>+  FakeMediaTransportFactory fake_media_transport_factory;
>+  JsepTransportController::Config config;
>+
>+  config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
>+  config.media_transport_factory = &fake_media_transport_factory;
>+  CreateJsepTransportController(config);
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  AddCryptoSettings(description.get());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
>+      transport_controller_->GetMediaTransport(kAudioMid1));
>+  EXPECT_NE(nullptr, media_transport);
>+  EXPECT_TRUE(media_transport->pre_shared_key().has_value());
>+
>+  // Even if we set local description with crypto now (after the remote offer
>+  // was set), media transport won't be provided.
>+  auto description2 = CreateSessionDescriptionWithoutBundle();
>+  AddCryptoSettings(description2.get());
>+
>+  RTCError result = transport_controller_->SetLocalDescription(
>+      SdpType::kAnswer, description2.get());
>+  EXPECT_TRUE(result.ok()) << result.message();
>+
>+  // Media transport did not change.
>+  EXPECT_EQ(media_transport,
>+            transport_controller_->GetMediaTransport(kAudioMid1));
> }
> 
> TEST_F(JsepTransportControllerTest, SetIceConfig) {
>@@ -568,9 +729,16 @@ TEST_F(JsepTransportControllerTest, SignalConnectionStateFailed) {
>   fake_ice->SetConnectionCount(0);
>   EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
>   EXPECT_EQ(1, connection_state_signal_count_);
>+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
>+                 ice_connection_state_, kTimeout);
>+  EXPECT_EQ(1, ice_connection_state_signal_count_);
>+  EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kFailed,
>+                 combined_connection_state_, kTimeout);
>+  EXPECT_EQ(1, combined_connection_state_signal_count_);
> }
> 
>-TEST_F(JsepTransportControllerTest, SignalConnectionStateConnected) {
>+TEST_F(JsepTransportControllerTest,
>+       SignalConnectionStateConnectedNoMediaTransport) {
>   CreateJsepTransportController(JsepTransportController::Config());
>   auto description = CreateSessionDescriptionWithoutBundle();
>   EXPECT_TRUE(transport_controller_
>@@ -595,13 +763,114 @@ TEST_F(JsepTransportControllerTest, SignalConnectionStateConnected) {
> 
>   EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
>   EXPECT_EQ(1, connection_state_signal_count_);
>-
>+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
>+                 ice_connection_state_, kTimeout);
>+  EXPECT_EQ(1, ice_connection_state_signal_count_);
>+  EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kFailed,
>+                 combined_connection_state_, kTimeout);
>+  EXPECT_EQ(1, combined_connection_state_signal_count_);
>+
>+  fake_audio_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
>+  fake_video_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
>   // Set the connection count to be 2 and the cricket::FakeIceTransport will set
>   // the transport state to be STATE_CONNECTING.
>   fake_video_dtls->fake_ice_transport()->SetConnectionCount(2);
>   fake_video_dtls->SetWritable(true);
>   EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
>   EXPECT_EQ(2, connection_state_signal_count_);
>+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
>+                 ice_connection_state_, kTimeout);
>+  EXPECT_EQ(2, ice_connection_state_signal_count_);
>+  EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kConnected,
>+                 combined_connection_state_, kTimeout);
>+  EXPECT_EQ(2, combined_connection_state_signal_count_);
>+}
>+
>+TEST_F(JsepTransportControllerTest,
>+       SignalConnectionStateConnectedWithMediaTransport) {
>+  FakeMediaTransportFactory fake_media_transport_factory;
>+  JsepTransportController::Config config;
>+  config.media_transport_factory = &fake_media_transport_factory;
>+  CreateJsepTransportController(config);
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  AddCryptoSettings(description.get());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kVideoMid1));
>+  fake_audio_dtls->SetWritable(true);
>+  fake_video_dtls->SetWritable(true);
>+  // Decreasing connection count from 2 to 1 triggers connection state event.
>+  fake_audio_dtls->fake_ice_transport()->SetConnectionCount(2);
>+  fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  fake_video_dtls->fake_ice_transport()->SetConnectionCount(2);
>+  fake_video_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  fake_audio_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
>+  fake_video_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
>+
>+  // Still not connected, because we are waiting for media transport.
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnecting, connection_state_,
>+                 kTimeout);
>+
>+  FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
>+      transport_controller_->GetMediaTransport(kAudioMid1));
>+
>+  media_transport->SetState(webrtc::MediaTransportState::kWritable);
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnecting, connection_state_,
>+                 kTimeout);
>+
>+  // Still waiting for the second media transport.
>+  media_transport = static_cast<FakeMediaTransport*>(
>+      transport_controller_->GetMediaTransport(kVideoMid1));
>+  media_transport->SetState(webrtc::MediaTransportState::kWritable);
>+
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
>+}
>+
>+TEST_F(JsepTransportControllerTest,
>+       SignalConnectionStateFailedWhenMediaTransportClosed) {
>+  FakeMediaTransportFactory fake_media_transport_factory;
>+  JsepTransportController::Config config;
>+  config.media_transport_factory = &fake_media_transport_factory;
>+  CreateJsepTransportController(config);
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  AddCryptoSettings(description.get());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kVideoMid1));
>+  fake_audio_dtls->SetWritable(true);
>+  fake_video_dtls->SetWritable(true);
>+  // Decreasing connection count from 2 to 1 triggers connection state event.
>+  fake_audio_dtls->fake_ice_transport()->SetConnectionCount(2);
>+  fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  fake_video_dtls->fake_ice_transport()->SetConnectionCount(2);
>+  fake_video_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  fake_audio_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
>+  fake_video_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
>+
>+  FakeMediaTransport* media_transport = static_cast<FakeMediaTransport*>(
>+      transport_controller_->GetMediaTransport(kAudioMid1));
>+
>+  media_transport->SetState(webrtc::MediaTransportState::kWritable);
>+
>+  media_transport = static_cast<FakeMediaTransport*>(
>+      transport_controller_->GetMediaTransport(kVideoMid1));
>+
>+  media_transport->SetState(webrtc::MediaTransportState::kWritable);
>+
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
>+
>+  media_transport->SetState(webrtc::MediaTransportState::kClosed);
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
> }
> 
> TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) {
>@@ -629,13 +898,27 @@ TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) {
> 
>   EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
>   EXPECT_EQ(1, connection_state_signal_count_);
>-
>+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionFailed,
>+                 ice_connection_state_, kTimeout);
>+  EXPECT_EQ(1, ice_connection_state_signal_count_);
>+  EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kFailed,
>+                 combined_connection_state_, kTimeout);
>+  EXPECT_EQ(1, combined_connection_state_signal_count_);
>+
>+  fake_audio_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
>+  fake_video_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
>   // Set the connection count to be 1 and the cricket::FakeIceTransport will set
>   // the transport state to be STATE_COMPLETED.
>   fake_video_dtls->fake_ice_transport()->SetConnectionCount(1);
>   fake_video_dtls->SetWritable(true);
>   EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
>   EXPECT_EQ(2, connection_state_signal_count_);
>+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
>+                 ice_connection_state_, kTimeout);
>+  EXPECT_EQ(2, ice_connection_state_signal_count_);
>+  EXPECT_EQ_WAIT(PeerConnectionInterface::PeerConnectionState::kConnected,
>+                 combined_connection_state_, kTimeout);
>+  EXPECT_EQ(2, combined_connection_state_signal_count_);
> }
> 
> TEST_F(JsepTransportControllerTest, SignalIceGatheringStateGathering) {
>@@ -709,6 +992,7 @@ TEST_F(JsepTransportControllerTest,
>   fake_audio_dtls->SetWritable(true);
>   fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
>   fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  fake_audio_dtls->SetDtlsState(cricket::DTLS_TRANSPORT_CONNECTED);
>   EXPECT_EQ(1, gathering_state_signal_count_);
> 
>   // Set the remote description and enable the bundle.
>@@ -721,6 +1005,10 @@ TEST_F(JsepTransportControllerTest,
>       transport_controller_->GetDtlsTransport(kVideoMid1));
>   EXPECT_EQ(fake_audio_dtls, fake_video_dtls);
>   EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
>+  EXPECT_EQ(PeerConnectionInterface::kIceConnectionCompleted,
>+            ice_connection_state_);
>+  EXPECT_EQ(PeerConnectionInterface::PeerConnectionState::kConnected,
>+            combined_connection_state_);
>   EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
>   EXPECT_EQ(2, gathering_state_signal_count_);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.h
>index c5f65304d0b3ceb5b770406554424d34cc3c7f49..c48f5407d4c1e534f5e6cb36cb97ca2f54f6342b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.h
>@@ -29,7 +29,7 @@ class LocalAudioSource : public Notifier<AudioSourceInterface> {
>   SourceState state() const override { return kLive; }
>   bool remote() const override { return false; }
> 
>-  virtual const cricket::AudioOptions& options() const { return options_; }
>+  const cricket::AudioOptions options() const override { return options_; }
> 
>   void AddSink(AudioTrackSinkInterface* sink) override {}
>   void RemoveSink(AudioTrackSinkInterface* sink) override {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc
>index 9ae46be27c5bf85a03a2c82de7ae7f6a74a115b4..7239af86c399e7bdbe486d8b06bdaca69a612c0b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc
>@@ -18,9 +18,9 @@
> #include <unordered_map>
> #include <utility>
> 
>+#include "absl/strings/match.h"
> #include "absl/types/optional.h"
> #include "api/cryptoparams.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "media/base/h264_profile_level_id.h"
> #include "media/base/mediaconstants.h"
> #include "p2p/base/p2pconstants.h"
>@@ -39,10 +39,10 @@ using webrtc::RtpTransceiverDirection;
> 
> const char kInline[] = "inline:";
> 
>-void GetSupportedSdesCryptoSuiteNames(void (*func)(const rtc::CryptoOptions&,
>-                                                   std::vector<int>*),
>-                                      const rtc::CryptoOptions& crypto_options,
>-                                      std::vector<std::string>* names) {
>+void GetSupportedSdesCryptoSuiteNames(
>+    void (*func)(const webrtc::CryptoOptions&, std::vector<int>*),
>+    const webrtc::CryptoOptions& crypto_options,
>+    std::vector<std::string>* names) {
>   std::vector<int> crypto_suites;
>   func(crypto_options, &crypto_suites);
>   for (const auto crypto : crypto_suites) {
>@@ -195,28 +195,30 @@ bool FindMatchingCrypto(const CryptoParamsVec& cryptos,
> 
> // For audio, HMAC 32 (if enabled) is prefered over HMAC 80 because of the
> // low overhead.
>-void GetSupportedAudioSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
>-                                       std::vector<int>* crypto_suites) {
>-  if (crypto_options.enable_gcm_crypto_suites) {
>+void GetSupportedAudioSdesCryptoSuites(
>+    const webrtc::CryptoOptions& crypto_options,
>+    std::vector<int>* crypto_suites) {
>+  if (crypto_options.srtp.enable_gcm_crypto_suites) {
>     crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
>     crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
>   }
>-  if (crypto_options.enable_aes128_sha1_32_crypto_cipher) {
>+  if (crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher) {
>     crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_32);
>   }
>   crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
> }
> 
> void GetSupportedAudioSdesCryptoSuiteNames(
>-    const rtc::CryptoOptions& crypto_options,
>+    const webrtc::CryptoOptions& crypto_options,
>     std::vector<std::string>* crypto_suite_names) {
>   GetSupportedSdesCryptoSuiteNames(GetSupportedAudioSdesCryptoSuites,
>                                    crypto_options, crypto_suite_names);
> }
> 
>-void GetSupportedVideoSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
>-                                       std::vector<int>* crypto_suites) {
>-  if (crypto_options.enable_gcm_crypto_suites) {
>+void GetSupportedVideoSdesCryptoSuites(
>+    const webrtc::CryptoOptions& crypto_options,
>+    std::vector<int>* crypto_suites) {
>+  if (crypto_options.srtp.enable_gcm_crypto_suites) {
>     crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
>     crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
>   }
>@@ -224,15 +226,16 @@ void GetSupportedVideoSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
> }
> 
> void GetSupportedVideoSdesCryptoSuiteNames(
>-    const rtc::CryptoOptions& crypto_options,
>+    const webrtc::CryptoOptions& crypto_options,
>     std::vector<std::string>* crypto_suite_names) {
>   GetSupportedSdesCryptoSuiteNames(GetSupportedVideoSdesCryptoSuites,
>                                    crypto_options, crypto_suite_names);
> }
> 
>-void GetSupportedDataSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
>-                                      std::vector<int>* crypto_suites) {
>-  if (crypto_options.enable_gcm_crypto_suites) {
>+void GetSupportedDataSdesCryptoSuites(
>+    const webrtc::CryptoOptions& crypto_options,
>+    std::vector<int>* crypto_suites) {
>+  if (crypto_options.srtp.enable_gcm_crypto_suites) {
>     crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
>     crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
>   }
>@@ -240,7 +243,7 @@ void GetSupportedDataSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
> }
> 
> void GetSupportedDataSdesCryptoSuiteNames(
>-    const rtc::CryptoOptions& crypto_options,
>+    const webrtc::CryptoOptions& crypto_options,
>     std::vector<std::string>* crypto_suite_names) {
>   GetSupportedSdesCryptoSuiteNames(GetSupportedDataSdesCryptoSuites,
>                                    crypto_options, crypto_suite_names);
>@@ -252,17 +255,17 @@ void GetSupportedDataSdesCryptoSuiteNames(
> // Pick the crypto in the list that is supported.
> static bool SelectCrypto(const MediaContentDescription* offer,
>                          bool bundle,
>-                         const rtc::CryptoOptions& crypto_options,
>+                         const webrtc::CryptoOptions& crypto_options,
>                          CryptoParams* crypto_out) {
>   bool audio = offer->type() == MEDIA_TYPE_AUDIO;
>   const CryptoParamsVec& cryptos = offer->cryptos();
> 
>   for (const CryptoParams& crypto : cryptos) {
>-    if ((crypto_options.enable_gcm_crypto_suites &&
>+    if ((crypto_options.srtp.enable_gcm_crypto_suites &&
>          rtc::IsGcmCryptoSuiteName(crypto.cipher_suite)) ||
>         rtc::CS_AES_CM_128_HMAC_SHA1_80 == crypto.cipher_suite ||
>         (rtc::CS_AES_CM_128_HMAC_SHA1_32 == crypto.cipher_suite && audio &&
>-         !bundle && crypto_options.enable_aes128_sha1_32_crypto_cipher)) {
>+         !bundle && crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher)) {
>       return CreateCryptoParams(crypto.tag, crypto.cipher_suite, crypto_out);
>     }
>   }
>@@ -309,7 +312,8 @@ void FilterDataCodecs(std::vector<DataCodec>* codecs, bool sctp) {
>       sctp ? kGoogleRtpDataCodecName : kGoogleSctpDataCodecName;
>   codecs->erase(std::remove_if(codecs->begin(), codecs->end(),
>                                [&codec_name](const DataCodec& codec) {
>-                                 return CodecNamesEq(codec.name, codec_name);
>+                                 return absl::EqualsIgnoreCase(codec.name,
>+                                                               codec_name);
>                                }),
>                 codecs->end());
> }
>@@ -650,7 +654,7 @@ static bool ContainsRtxCodec(const std::vector<C>& codecs) {
> 
> template <class C>
> static bool IsRtxCodec(const C& codec) {
>-  return STR_CASE_CMP(codec.name.c_str(), kRtxCodecName) == 0;
>+  return absl::EqualsIgnoreCase(codec.name, kRtxCodecName);
> }
> 
> template <class C>
>@@ -665,7 +669,7 @@ static bool ContainsFlexfecCodec(const std::vector<C>& codecs) {
> 
> template <class C>
> static bool IsFlexfecCodec(const C& codec) {
>-  return STR_CASE_CMP(codec.name.c_str(), kFlexfecCodecName) == 0;
>+  return absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName);
> }
> 
> // Create a media content to be offered for the given |sender_options|,
>@@ -743,7 +747,7 @@ static void NegotiateCodecs(const std::vector<C>& local_codecs,
>         RTC_DCHECK(apt_it != theirs.params.end());
>         negotiated.SetParam(kCodecParamAssociatedPayloadType, apt_it->second);
>       }
>-      if (CodecNamesEq(ours.name.c_str(), kH264CodecName)) {
>+      if (absl::EqualsIgnoreCase(ours.name, kH264CodecName)) {
>         webrtc::H264::GenerateProfileLevelIdForAnswer(
>             ours.params, theirs.params, &negotiated.params);
>       }
>@@ -1038,9 +1042,8 @@ static void NegotiateRtpHeaderExtensions(
> static void StripCNCodecs(AudioCodecs* audio_codecs) {
>   audio_codecs->erase(std::remove_if(audio_codecs->begin(), audio_codecs->end(),
>                                      [](const AudioCodec& codec) {
>-                                       return STR_CASE_CMP(
>-                                                  codec.name.c_str(),
>-                                                  kComfortNoiseCodecName) == 0;
>+                                       return absl::EqualsIgnoreCase(
>+                                           codec.name, kComfortNoiseCodecName);
>                                      }),
>                       audio_codecs->end());
> }
>@@ -1069,6 +1072,8 @@ static bool CreateMediaContentAnswer(
>   NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs);
>   answer->AddCodecs(negotiated_codecs);
>   answer->set_protocol(offer->protocol());
>+
>+  answer->set_extmap_allow_mixed_enum(offer->extmap_allow_mixed_enum());
>   RtpHeaderExtensions negotiated_rtp_extensions;
>   NegotiateRtpHeaderExtensions(
>       local_rtp_extenstions, offer->rtp_header_extensions(),
>@@ -1262,6 +1267,8 @@ SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
>     const SessionDescription* current_description) const {
>   std::unique_ptr<SessionDescription> offer(new SessionDescription());
> 
>+  IceCredentialsIterator ice_credentials(
>+      session_options.pooled_ice_credentials);
>   StreamParamsVec current_streams;
>   GetCurrentStreamParams(current_description, &current_streams);
> 
>@@ -1305,18 +1312,18 @@ SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
>     }
>     switch (media_description_options.type) {
>       case MEDIA_TYPE_AUDIO:
>-        if (!AddAudioContentForOffer(media_description_options, session_options,
>-                                     current_content, current_description,
>-                                     audio_rtp_extensions, offer_audio_codecs,
>-                                     &current_streams, offer.get())) {
>+        if (!AddAudioContentForOffer(
>+                media_description_options, session_options, current_content,
>+                current_description, audio_rtp_extensions, offer_audio_codecs,
>+                &current_streams, offer.get(), &ice_credentials)) {
>           return nullptr;
>         }
>         break;
>       case MEDIA_TYPE_VIDEO:
>-        if (!AddVideoContentForOffer(media_description_options, session_options,
>-                                     current_content, current_description,
>-                                     video_rtp_extensions, offer_video_codecs,
>-                                     &current_streams, offer.get())) {
>+        if (!AddVideoContentForOffer(
>+                media_description_options, session_options, current_content,
>+                current_description, video_rtp_extensions, offer_video_codecs,
>+                &current_streams, offer.get(), &ice_credentials)) {
>           return nullptr;
>         }
>         break;
>@@ -1324,7 +1331,7 @@ SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
>         if (!AddDataContentForOffer(media_description_options, session_options,
>                                     current_content, current_description,
>                                     offer_data_codecs, &current_streams,
>-                                    offer.get())) {
>+                                    offer.get(), &ice_credentials)) {
>           return nullptr;
>         }
>         break;
>@@ -1371,6 +1378,8 @@ SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
>     offer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute);
>   }
> 
>+  offer->set_extmap_allow_mixed(session_options.offer_extmap_allow_mixed);
>+
>   return offer.release();
> }
> 
>@@ -1381,6 +1390,10 @@ SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
>   if (!offer) {
>     return nullptr;
>   }
>+
>+  IceCredentialsIterator ice_credentials(
>+      session_options.pooled_ice_credentials);
>+
>   // The answer contains the intersection of the codecs in the offer with the
>   // codecs we support. As indicated by XEP-0167, we retain the same payload ids
>   // from the offer in the answer.
>@@ -1396,6 +1409,8 @@ SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
>   // Transport info shared by the bundle group.
>   std::unique_ptr<TransportInfo> bundle_transport;
> 
>+  answer->set_extmap_allow_mixed(offer->extmap_allow_mixed());
>+
>   // Get list of all possible codecs that respects existing payload type
>   // mappings and uses a single payload type space.
>   //
>@@ -1440,7 +1455,7 @@ SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
>                 media_description_options, session_options, offer_content,
>                 offer, current_content, current_description,
>                 bundle_transport.get(), answer_audio_codecs, &current_streams,
>-                answer.get())) {
>+                answer.get(), &ice_credentials)) {
>           return nullptr;
>         }
>         break;
>@@ -1449,16 +1464,16 @@ SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
>                 media_description_options, session_options, offer_content,
>                 offer, current_content, current_description,
>                 bundle_transport.get(), answer_video_codecs, &current_streams,
>-                answer.get())) {
>+                answer.get(), &ice_credentials)) {
>           return nullptr;
>         }
>         break;
>       case MEDIA_TYPE_DATA:
>-        if (!AddDataContentForAnswer(media_description_options, session_options,
>-                                     offer_content, offer, current_content,
>-                                     current_description,
>-                                     bundle_transport.get(), answer_data_codecs,
>-                                     &current_streams, answer.get())) {
>+        if (!AddDataContentForAnswer(
>+                media_description_options, session_options, offer_content,
>+                offer, current_content, current_description,
>+                bundle_transport.get(), answer_data_codecs, &current_streams,
>+                answer.get(), &ice_credentials)) {
>           return nullptr;
>         }
>         break;
>@@ -1765,13 +1780,15 @@ bool MediaSessionDescriptionFactory::AddTransportOffer(
>     const std::string& content_name,
>     const TransportOptions& transport_options,
>     const SessionDescription* current_desc,
>-    SessionDescription* offer_desc) const {
>+    SessionDescription* offer_desc,
>+    IceCredentialsIterator* ice_credentials) const {
>   if (!transport_desc_factory_)
>     return false;
>   const TransportDescription* current_tdesc =
>       GetTransportDescription(content_name, current_desc);
>   std::unique_ptr<TransportDescription> new_tdesc(
>-      transport_desc_factory_->CreateOffer(transport_options, current_tdesc));
>+      transport_desc_factory_->CreateOffer(transport_options, current_tdesc,
>+                                           ice_credentials));
>   bool ret =
>       (new_tdesc.get() != NULL &&
>        offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc)));
>@@ -1787,7 +1804,8 @@ TransportDescription* MediaSessionDescriptionFactory::CreateTransportAnswer(
>     const SessionDescription* offer_desc,
>     const TransportOptions& transport_options,
>     const SessionDescription* current_desc,
>-    bool require_transport_attributes) const {
>+    bool require_transport_attributes,
>+    IceCredentialsIterator* ice_credentials) const {
>   if (!transport_desc_factory_)
>     return NULL;
>   const TransportDescription* offer_tdesc =
>@@ -1796,7 +1814,7 @@ TransportDescription* MediaSessionDescriptionFactory::CreateTransportAnswer(
>       GetTransportDescription(content_name, current_desc);
>   return transport_desc_factory_->CreateAnswer(offer_tdesc, transport_options,
>                                                require_transport_attributes,
>-                                               current_tdesc);
>+                                               current_tdesc, ice_credentials);
> }
> 
> bool MediaSessionDescriptionFactory::AddTransportAnswer(
>@@ -1832,7 +1850,8 @@ bool MediaSessionDescriptionFactory::AddAudioContentForOffer(
>     const RtpHeaderExtensions& audio_rtp_extensions,
>     const AudioCodecs& audio_codecs,
>     StreamParamsVec* current_streams,
>-    SessionDescription* desc) const {
>+    SessionDescription* desc,
>+    IceCredentialsIterator* ice_credentials) const {
>   // Filter audio_codecs (which includes all codecs, with correctly remapped
>   // payload types) based on transceiver direction.
>   const AudioCodecs& supported_audio_codecs =
>@@ -1888,7 +1907,7 @@ bool MediaSessionDescriptionFactory::AddAudioContentForOffer(
>                    media_description_options.stopped, audio.release());
>   if (!AddTransportOffer(media_description_options.mid,
>                          media_description_options.transport_options,
>-                         current_description, desc)) {
>+                         current_description, desc, ice_credentials)) {
>     return false;
>   }
> 
>@@ -1903,7 +1922,8 @@ bool MediaSessionDescriptionFactory::AddVideoContentForOffer(
>     const RtpHeaderExtensions& video_rtp_extensions,
>     const VideoCodecs& video_codecs,
>     StreamParamsVec* current_streams,
>-    SessionDescription* desc) const {
>+    SessionDescription* desc,
>+    IceCredentialsIterator* ice_credentials) const {
>   cricket::SecurePolicy sdes_policy =
>       IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
>                                                          : secure();
>@@ -1957,7 +1977,7 @@ bool MediaSessionDescriptionFactory::AddVideoContentForOffer(
>                    media_description_options.stopped, video.release());
>   if (!AddTransportOffer(media_description_options.mid,
>                          media_description_options.transport_options,
>-                         current_description, desc)) {
>+                         current_description, desc, ice_credentials)) {
>     return false;
>   }
>   return true;
>@@ -1970,7 +1990,8 @@ bool MediaSessionDescriptionFactory::AddDataContentForOffer(
>     const SessionDescription* current_description,
>     const DataCodecs& data_codecs,
>     StreamParamsVec* current_streams,
>-    SessionDescription* desc) const {
>+    SessionDescription* desc,
>+    IceCredentialsIterator* ice_credentials) const {
>   bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
> 
>   std::unique_ptr<DataContentDescription> data(new DataContentDescription());
>@@ -2024,7 +2045,7 @@ bool MediaSessionDescriptionFactory::AddDataContentForOffer(
>   }
>   if (!AddTransportOffer(media_description_options.mid,
>                          media_description_options.transport_options,
>-                         current_description, desc)) {
>+                         current_description, desc, ice_credentials)) {
>     return false;
>   }
>   return true;
>@@ -2052,15 +2073,16 @@ bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
>     const TransportInfo* bundle_transport,
>     const AudioCodecs& audio_codecs,
>     StreamParamsVec* current_streams,
>-    SessionDescription* answer) const {
>+    SessionDescription* answer,
>+    IceCredentialsIterator* ice_credentials) const {
>   RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_AUDIO));
>   const AudioContentDescription* offer_audio_description =
>       offer_content->media_description()->as_audio();
> 
>-  std::unique_ptr<TransportDescription> audio_transport(
>-      CreateTransportAnswer(media_description_options.mid, offer_description,
>-                            media_description_options.transport_options,
>-                            current_description, bundle_transport != nullptr));
>+  std::unique_ptr<TransportDescription> audio_transport(CreateTransportAnswer(
>+      media_description_options.mid, offer_description,
>+      media_description_options.transport_options, current_description,
>+      bundle_transport != nullptr, ice_credentials));
>   if (!audio_transport) {
>     return false;
>   }
>@@ -2146,15 +2168,16 @@ bool MediaSessionDescriptionFactory::AddVideoContentForAnswer(
>     const TransportInfo* bundle_transport,
>     const VideoCodecs& video_codecs,
>     StreamParamsVec* current_streams,
>-    SessionDescription* answer) const {
>+    SessionDescription* answer,
>+    IceCredentialsIterator* ice_credentials) const {
>   RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_VIDEO));
>   const VideoContentDescription* offer_video_description =
>       offer_content->media_description()->as_video();
> 
>-  std::unique_ptr<TransportDescription> video_transport(
>-      CreateTransportAnswer(media_description_options.mid, offer_description,
>-                            media_description_options.transport_options,
>-                            current_description, bundle_transport != nullptr));
>+  std::unique_ptr<TransportDescription> video_transport(CreateTransportAnswer(
>+      media_description_options.mid, offer_description,
>+      media_description_options.transport_options, current_description,
>+      bundle_transport != nullptr, ice_credentials));
>   if (!video_transport) {
>     return false;
>   }
>@@ -2232,11 +2255,12 @@ bool MediaSessionDescriptionFactory::AddDataContentForAnswer(
>     const TransportInfo* bundle_transport,
>     const DataCodecs& data_codecs,
>     StreamParamsVec* current_streams,
>-    SessionDescription* answer) const {
>-  std::unique_ptr<TransportDescription> data_transport(
>-      CreateTransportAnswer(media_description_options.mid, offer_description,
>-                            media_description_options.transport_options,
>-                            current_description, bundle_transport != nullptr));
>+    SessionDescription* answer,
>+    IceCredentialsIterator* ice_credentials) const {
>+  std::unique_ptr<TransportDescription> data_transport(CreateTransportAnswer(
>+      media_description_options.mid, offer_description,
>+      media_description_options.transport_options, current_description,
>+      bundle_transport != nullptr, ice_credentials));
>   if (!data_transport) {
>     return false;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.h
>index b40974039b820a965f57d57f35c689021c445430..5904605f10a4da25fbd7cdc95cfb90f949914630 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.h
>@@ -21,6 +21,7 @@
> #include "api/mediatypes.h"
> #include "media/base/mediaconstants.h"
> #include "media/base/mediaengine.h"  // For DataChannelType
>+#include "p2p/base/icecredentialsiterator.h"
> #include "p2p/base/transportdescriptionfactory.h"
> #include "pc/jseptransport.h"
> #include "pc/sessiondescription.h"
>@@ -93,11 +94,13 @@ struct MediaSessionOptions {
>   bool rtcp_mux_enabled = true;
>   bool bundle_enabled = false;
>   bool is_unified_plan = false;
>+  bool offer_extmap_allow_mixed = false;
>   std::string rtcp_cname = kDefaultRtcpCname;
>-  rtc::CryptoOptions crypto_options;
>+  webrtc::CryptoOptions crypto_options;
>   // List of media description options in the same order that the media
>   // descriptions will be generated.
>   std::vector<MediaDescriptionOptions> media_description_options;
>+  std::vector<IceParameters> pooled_ice_credentials;
> };
> 
> // Creates media session descriptions according to the supplied codecs and
>@@ -186,14 +189,16 @@ class MediaSessionDescriptionFactory {
>   bool AddTransportOffer(const std::string& content_name,
>                          const TransportOptions& transport_options,
>                          const SessionDescription* current_desc,
>-                         SessionDescription* offer) const;
>+                         SessionDescription* offer,
>+                         IceCredentialsIterator* ice_credentials) const;
> 
>   TransportDescription* CreateTransportAnswer(
>       const std::string& content_name,
>       const SessionDescription* offer_desc,
>       const TransportOptions& transport_options,
>       const SessionDescription* current_desc,
>-      bool require_transport_attributes) const;
>+      bool require_transport_attributes,
>+      IceCredentialsIterator* ice_credentials) const;
> 
>   bool AddTransportAnswer(const std::string& content_name,
>                           const TransportDescription& transport_desc,
>@@ -211,7 +216,8 @@ class MediaSessionDescriptionFactory {
>       const RtpHeaderExtensions& audio_rtp_extensions,
>       const AudioCodecs& audio_codecs,
>       StreamParamsVec* current_streams,
>-      SessionDescription* desc) const;
>+      SessionDescription* desc,
>+      IceCredentialsIterator* ice_credentials) const;
> 
>   bool AddVideoContentForOffer(
>       const MediaDescriptionOptions& media_description_options,
>@@ -221,7 +227,8 @@ class MediaSessionDescriptionFactory {
>       const RtpHeaderExtensions& video_rtp_extensions,
>       const VideoCodecs& video_codecs,
>       StreamParamsVec* current_streams,
>-      SessionDescription* desc) const;
>+      SessionDescription* desc,
>+      IceCredentialsIterator* ice_credentials) const;
> 
>   bool AddDataContentForOffer(
>       const MediaDescriptionOptions& media_description_options,
>@@ -230,7 +237,8 @@ class MediaSessionDescriptionFactory {
>       const SessionDescription* current_description,
>       const DataCodecs& data_codecs,
>       StreamParamsVec* current_streams,
>-      SessionDescription* desc) const;
>+      SessionDescription* desc,
>+      IceCredentialsIterator* ice_credentials) const;
> 
>   bool AddAudioContentForAnswer(
>       const MediaDescriptionOptions& media_description_options,
>@@ -242,7 +250,8 @@ class MediaSessionDescriptionFactory {
>       const TransportInfo* bundle_transport,
>       const AudioCodecs& audio_codecs,
>       StreamParamsVec* current_streams,
>-      SessionDescription* answer) const;
>+      SessionDescription* answer,
>+      IceCredentialsIterator* ice_credentials) const;
> 
>   bool AddVideoContentForAnswer(
>       const MediaDescriptionOptions& media_description_options,
>@@ -254,7 +263,8 @@ class MediaSessionDescriptionFactory {
>       const TransportInfo* bundle_transport,
>       const VideoCodecs& video_codecs,
>       StreamParamsVec* current_streams,
>-      SessionDescription* answer) const;
>+      SessionDescription* answer,
>+      IceCredentialsIterator* ice_credentials) const;
> 
>   bool AddDataContentForAnswer(
>       const MediaDescriptionOptions& media_description_options,
>@@ -266,7 +276,8 @@ class MediaSessionDescriptionFactory {
>       const TransportInfo* bundle_transport,
>       const DataCodecs& data_codecs,
>       StreamParamsVec* current_streams,
>-      SessionDescription* answer) const;
>+      SessionDescription* answer,
>+      IceCredentialsIterator* ice_credentials) const;
> 
>   void ComputeAudioCodecsIntersectionAndUnion();
> 
>@@ -327,20 +338,23 @@ DataContentDescription* GetFirstDataContentDescription(
>     SessionDescription* sdesc);
> 
> // Helper functions to return crypto suites used for SDES.
>-void GetSupportedAudioSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
>-                                       std::vector<int>* crypto_suites);
>-void GetSupportedVideoSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
>-                                       std::vector<int>* crypto_suites);
>-void GetSupportedDataSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
>-                                      std::vector<int>* crypto_suites);
>+void GetSupportedAudioSdesCryptoSuites(
>+    const webrtc::CryptoOptions& crypto_options,
>+    std::vector<int>* crypto_suites);
>+void GetSupportedVideoSdesCryptoSuites(
>+    const webrtc::CryptoOptions& crypto_options,
>+    std::vector<int>* crypto_suites);
>+void GetSupportedDataSdesCryptoSuites(
>+    const webrtc::CryptoOptions& crypto_options,
>+    std::vector<int>* crypto_suites);
> void GetSupportedAudioSdesCryptoSuiteNames(
>-    const rtc::CryptoOptions& crypto_options,
>+    const webrtc::CryptoOptions& crypto_options,
>     std::vector<std::string>* crypto_suite_names);
> void GetSupportedVideoSdesCryptoSuiteNames(
>-    const rtc::CryptoOptions& crypto_options,
>+    const webrtc::CryptoOptions& crypto_options,
>     std::vector<std::string>* crypto_suite_names);
> void GetSupportedDataSdesCryptoSuiteNames(
>-    const rtc::CryptoOptions& crypto_options,
>+    const webrtc::CryptoOptions& crypto_options,
>     std::vector<std::string>* crypto_suite_names);
> 
> // Returns true if the given media section protocol indicates use of RTP.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc
>index db325ed2ef167ebf9b3ade832cb4ccdbf7f73c16..076ad12a5b34980c00c31209749a1eb536fc34d4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc
>@@ -26,10 +26,11 @@
> #include "rtc_base/messagedigest.h"
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/strings/string_builder.h"
>+#include "test/gmock.h"
> 
> #define ASSERT_CRYPTO(cd, s, cs)      \
>   ASSERT_EQ(s, cd->cryptos().size()); \
>-  ASSERT_EQ(std::string(cs), cd->cryptos()[0].cipher_suite)
>+  ASSERT_EQ(cs, cd->cryptos()[0].cipher_suite)
> 
> typedef std::vector<cricket::Candidate> Candidates;
> 
>@@ -71,6 +72,7 @@ using rtc::CS_AES_CM_128_HMAC_SHA1_32;
> using rtc::CS_AES_CM_128_HMAC_SHA1_80;
> using rtc::CS_AEAD_AES_128_GCM;
> using rtc::CS_AEAD_AES_256_GCM;
>+using testing::ElementsAreArray;
> using webrtc::RtpExtension;
> using webrtc::RtpTransceiverDirection;
> 
>@@ -545,7 +547,7 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>     EXPECT_TRUE(CompareCryptoParams(audio_media_desc->cryptos(),
>                                     video_media_desc->cryptos()));
>     EXPECT_EQ(1u, audio_media_desc->cryptos().size());
>-    EXPECT_EQ(std::string(kDefaultSrtpCryptoSuite),
>+    EXPECT_EQ(kDefaultSrtpCryptoSuite,
>               audio_media_desc->cryptos()[0].cipher_suite);
> 
>     // Verify the selected crypto is one from the reference audio
>@@ -609,11 +611,11 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>   void TestVideoGcmCipher(bool gcm_offer, bool gcm_answer) {
>     MediaSessionOptions offer_opts;
>     AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &offer_opts);
>-    offer_opts.crypto_options.enable_gcm_crypto_suites = gcm_offer;
>+    offer_opts.crypto_options.srtp.enable_gcm_crypto_suites = gcm_offer;
> 
>     MediaSessionOptions answer_opts;
>     AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &answer_opts);
>-    answer_opts.crypto_options.enable_gcm_crypto_suites = gcm_answer;
>+    answer_opts.crypto_options.srtp.enable_gcm_crypto_suites = gcm_answer;
> 
>     f1_.set_secure(SEC_ENABLED);
>     f2_.set_secure(SEC_ENABLED);
>@@ -631,7 +633,7 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>     const AudioContentDescription* acd = ac->media_description()->as_audio();
>     const VideoContentDescription* vcd = vc->media_description()->as_video();
>     EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
>-    EXPECT_EQ(MAKE_VECTOR(kAudioCodecsAnswer), acd->codecs());
>+    EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
>     EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
>     EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>     EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>@@ -641,7 +643,7 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>       ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>     }
>     EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
>-    EXPECT_EQ(MAKE_VECTOR(kVideoCodecsAnswer), vcd->codecs());
>+    EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecsAnswer));
>     EXPECT_EQ(0U, vcd->first_ssrc());  // no sender is attached
>     EXPECT_TRUE(vcd->rtcp_mux());      // negotiated rtcp-mux
>     if (gcm_offer && gcm_answer) {
>@@ -649,7 +651,7 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>     } else {
>       ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
>     }
>-    EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), vcd->protocol());
>+    EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
>   }
> 
>  protected:
>@@ -677,7 +679,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioOffer) {
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // default bandwidth (auto)
>   EXPECT_TRUE(acd->rtcp_mux());                 // rtcp-mux defaults on
>   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
> }
> 
> // Create a typical video offer, and ensure it matches what we expect.
>@@ -701,14 +703,14 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoOffer) {
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // default bandwidth (auto)
>   EXPECT_TRUE(acd->rtcp_mux());                 // rtcp-mux defaults on
>   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
>   EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
>   EXPECT_EQ(f1_.video_codecs(), vcd->codecs());
>   EXPECT_EQ(0U, vcd->first_ssrc());             // no sender is attached
>   EXPECT_EQ(kAutoBandwidth, vcd->bandwidth());  // default bandwidth (auto)
>   EXPECT_TRUE(vcd->rtcp_mux());                 // rtcp-mux defaults on
>   ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), vcd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
> }
> 
> // Test creating an offer with bundle where the Codecs have the same dynamic
>@@ -779,11 +781,11 @@ TEST_F(MediaSessionDescriptionFactoryTest,
>   EXPECT_TRUE(NULL != dcd);
> 
>   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
>   ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), vcd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
>   ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), dcd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
> }
> 
> // Create a RTP data offer, and ensure it matches what we expect.
>@@ -808,7 +810,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateRtpDataOffer) {
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // default bandwidth (auto)
>   EXPECT_TRUE(acd->rtcp_mux());                 // rtcp-mux defaults on
>   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
>   EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
>   EXPECT_EQ(f1_.data_codecs(), dcd->codecs());
>   EXPECT_EQ(0U, dcd->first_ssrc());  // no sender is attached.
>@@ -816,7 +818,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateRtpDataOffer) {
>             dcd->bandwidth());   // default bandwidth (auto)
>   EXPECT_TRUE(dcd->rtcp_mux());  // rtcp-mux defaults on
>   ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), dcd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
> }
> 
> // Create an SCTP data offer with bundle without error.
>@@ -939,12 +941,12 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswer) {
>   EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
>   const AudioContentDescription* acd = ac->media_description()->as_audio();
>   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioCodecsAnswer), acd->codecs());
>+  EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
>   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
> }
> 
> // Create a typical audio answer with GCM ciphers enabled, and ensure it
>@@ -953,7 +955,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswerGcm) {
>   f1_.set_secure(SEC_ENABLED);
>   f2_.set_secure(SEC_ENABLED);
>   MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
>-  opts.crypto_options.enable_gcm_crypto_suites = true;
>+  opts.crypto_options.srtp.enable_gcm_crypto_suites = true;
>   std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
>   ASSERT_TRUE(offer.get() != NULL);
>   std::unique_ptr<SessionDescription> answer(
>@@ -965,12 +967,12 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswerGcm) {
>   EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
>   const AudioContentDescription* acd = ac->media_description()->as_audio();
>   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioCodecsAnswer), acd->codecs());
>+  EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
>   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuiteGcm);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
> }
> 
> // Create a typical video answer, and ensure it matches what we expect.
>@@ -992,17 +994,17 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswer) {
>   const AudioContentDescription* acd = ac->media_description()->as_audio();
>   const VideoContentDescription* vcd = vc->media_description()->as_video();
>   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioCodecsAnswer), acd->codecs());
>+  EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoCodecsAnswer), vcd->codecs());
>+  EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecsAnswer));
>   EXPECT_EQ(0U, vcd->first_ssrc());  // no sender is attached
>   EXPECT_TRUE(vcd->rtcp_mux());      // negotiated rtcp-mux
>   ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), vcd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
> }
> 
> // Create a typical video answer with GCM ciphers enabled, and ensure it
>@@ -1041,23 +1043,23 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswer) {
>   const AudioContentDescription* acd = ac->media_description()->as_audio();
>   const DataContentDescription* dcd = dc->media_description()->as_data();
>   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioCodecsAnswer), acd->codecs());
>+  EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
>-  EXPECT_EQ(MAKE_VECTOR(kDataCodecsAnswer), dcd->codecs());
>+  EXPECT_THAT(dcd->codecs(), ElementsAreArray(kDataCodecsAnswer));
>   EXPECT_EQ(0U, dcd->first_ssrc());  // no sender is attached
>   EXPECT_TRUE(dcd->rtcp_mux());      // negotiated rtcp-mux
>   ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), dcd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
> }
> 
> TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswerGcm) {
>   MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
>   AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly, &opts);
>-  opts.crypto_options.enable_gcm_crypto_suites = true;
>+  opts.crypto_options.srtp.enable_gcm_crypto_suites = true;
>   f1_.set_secure(SEC_ENABLED);
>   f2_.set_secure(SEC_ENABLED);
>   std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
>@@ -1073,17 +1075,17 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswerGcm) {
>   const AudioContentDescription* acd = ac->media_description()->as_audio();
>   const DataContentDescription* dcd = dc->media_description()->as_data();
>   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioCodecsAnswer), acd->codecs());
>+  EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuiteGcm);
>   EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
>-  EXPECT_EQ(MAKE_VECTOR(kDataCodecsAnswer), dcd->codecs());
>+  EXPECT_THAT(dcd->codecs(), ElementsAreArray(kDataCodecsAnswer));
>   EXPECT_EQ(0U, dcd->first_ssrc());  // no sender is attached
>   EXPECT_TRUE(dcd->rtcp_mux());      // negotiated rtcp-mux
>   ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuiteGcm);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), dcd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, dcd->protocol());
> }
> 
> // The use_sctpmap flag should be set in a DataContentDescription by default.
>@@ -1264,7 +1266,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, AudioOfferAnswerWithCryptoDisabled) {
>   const AudioContentDescription* offer_acd =
>       GetFirstAudioContentDescription(offer.get());
>   ASSERT_TRUE(offer_acd != NULL);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolAvpf), offer_acd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolAvpf, offer_acd->protocol());
> 
>   std::unique_ptr<SessionDescription> answer(
>       f2_.CreateAnswer(offer.get(), opts, NULL));
>@@ -1276,7 +1278,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, AudioOfferAnswerWithCryptoDisabled) {
>   const AudioContentDescription* answer_acd =
>       GetFirstAudioContentDescription(answer.get());
>   ASSERT_TRUE(answer_acd != NULL);
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolAvpf), answer_acd->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolAvpf, answer_acd->protocol());
> }
> 
> // Create a video offer and answer and ensure the RTP header extensions
>@@ -1581,6 +1583,67 @@ TEST_F(MediaSessionDescriptionFactoryTest,
>   EXPECT_TRUE(dc->rejected);
> }
> 
>+TEST_F(MediaSessionDescriptionFactoryTest,
>+       CreateAnswerSupportsMixedOneAndTwoByteHeaderExtensions) {
>+  MediaSessionOptions opts;
>+  std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
>+  // Offer without request of mixed one- and two-byte header extensions.
>+  offer->set_extmap_allow_mixed(false);
>+  ASSERT_TRUE(offer.get() != NULL);
>+  std::unique_ptr<SessionDescription> answer_no_support(
>+      f2_.CreateAnswer(offer.get(), opts, NULL));
>+  EXPECT_FALSE(answer_no_support->extmap_allow_mixed());
>+
>+  // Offer with request of mixed one- and two-byte header extensions.
>+  offer->set_extmap_allow_mixed(true);
>+  ASSERT_TRUE(offer.get() != NULL);
>+  std::unique_ptr<SessionDescription> answer_support(
>+      f2_.CreateAnswer(offer.get(), opts, NULL));
>+  EXPECT_TRUE(answer_support->extmap_allow_mixed());
>+}
>+
>+TEST_F(MediaSessionDescriptionFactoryTest,
>+       CreateAnswerSupportsMixedOneAndTwoByteHeaderExtensionsOnMediaLevel) {
>+  MediaSessionOptions opts;
>+  AddAudioVideoSections(RtpTransceiverDirection::kSendRecv, &opts);
>+  std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
>+  MediaContentDescription* video_offer =
>+      offer->GetContentDescriptionByName("video");
>+  ASSERT_TRUE(video_offer);
>+  MediaContentDescription* audio_offer =
>+      offer->GetContentDescriptionByName("audio");
>+  ASSERT_TRUE(audio_offer);
>+
>+  // Explicit disable of mixed one-two byte header support in offer.
>+  video_offer->set_extmap_allow_mixed_enum(MediaContentDescription::kNo);
>+  audio_offer->set_extmap_allow_mixed_enum(MediaContentDescription::kNo);
>+
>+  ASSERT_TRUE(offer.get() != NULL);
>+  std::unique_ptr<SessionDescription> answer_no_support(
>+      f2_.CreateAnswer(offer.get(), opts, NULL));
>+  MediaContentDescription* video_answer =
>+      answer_no_support->GetContentDescriptionByName("video");
>+  MediaContentDescription* audio_answer =
>+      answer_no_support->GetContentDescriptionByName("audio");
>+  EXPECT_EQ(MediaContentDescription::kNo,
>+            video_answer->extmap_allow_mixed_enum());
>+  EXPECT_EQ(MediaContentDescription::kNo,
>+            audio_answer->extmap_allow_mixed_enum());
>+
>+  // Enable mixed one-two byte header support in offer.
>+  video_offer->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
>+  audio_offer->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
>+  ASSERT_TRUE(offer.get() != NULL);
>+  std::unique_ptr<SessionDescription> answer_support(
>+      f2_.CreateAnswer(offer.get(), opts, NULL));
>+  video_answer = answer_support->GetContentDescriptionByName("video");
>+  audio_answer = answer_support->GetContentDescriptionByName("audio");
>+  EXPECT_EQ(MediaContentDescription::kMedia,
>+            video_answer->extmap_allow_mixed_enum());
>+  EXPECT_EQ(MediaContentDescription::kMedia,
>+            audio_answer->extmap_allow_mixed_enum());
>+}
>+
> // Create an audio and video offer with:
> // - one video track
> // - two audio tracks
>@@ -1814,7 +1877,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
>   ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
> 
>   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioCodecsAnswer), acd->codecs());
>+  EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
> 
>   const StreamParamsVec& audio_streams = acd->streams();
>   ASSERT_EQ(2U, audio_streams.size());
>@@ -1830,7 +1893,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
>   EXPECT_TRUE(acd->rtcp_mux());                 // rtcp-mux defaults on
> 
>   EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoCodecsAnswer), vcd->codecs());
>+  EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecsAnswer));
> 
>   const StreamParamsVec& video_streams = vcd->streams();
>   ASSERT_EQ(1U, video_streams.size());
>@@ -1840,7 +1903,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
>   EXPECT_TRUE(vcd->rtcp_mux());                 // rtcp-mux defaults on
> 
>   EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
>-  EXPECT_EQ(MAKE_VECTOR(kDataCodecsAnswer), dcd->codecs());
>+  EXPECT_THAT(dcd->codecs(), ElementsAreArray(kDataCodecsAnswer));
> 
>   const StreamParamsVec& data_streams = dcd->streams();
>   ASSERT_EQ(2U, data_streams.size());
>@@ -1923,11 +1986,11 @@ TEST_F(MediaSessionDescriptionFactoryTest,
> 
>   const AudioContentDescription* acd =
>       GetFirstAudioContentDescription(answer.get());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioCodecsAnswer), acd->codecs());
>+  EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
> 
>   const VideoContentDescription* vcd =
>       GetFirstVideoContentDescription(answer.get());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoCodecsAnswer), vcd->codecs());
>+  EXPECT_THAT(vcd->codecs(), ElementsAreArray(kVideoCodecsAnswer));
> 
>   std::unique_ptr<SessionDescription> updated_offer(
>       f2_.CreateOffer(opts, answer.get()));
>@@ -1950,11 +2013,11 @@ TEST_F(MediaSessionDescriptionFactoryTest,
> 
>   const AudioContentDescription* updated_acd =
>       GetFirstAudioContentDescription(updated_offer.get());
>-  EXPECT_EQ(MAKE_VECTOR(kUpdatedAudioCodecOffer), updated_acd->codecs());
>+  EXPECT_THAT(updated_acd->codecs(), ElementsAreArray(kUpdatedAudioCodecOffer));
> 
>   const VideoContentDescription* updated_vcd =
>       GetFirstVideoContentDescription(updated_offer.get());
>-  EXPECT_EQ(MAKE_VECTOR(kUpdatedVideoCodecOffer), updated_vcd->codecs());
>+  EXPECT_THAT(updated_vcd->codecs(), ElementsAreArray(kUpdatedVideoCodecOffer));
> }
> 
> // Create an updated offer after creating an answer to the original offer and
>@@ -2079,7 +2142,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
> 
>   const AudioContentDescription* acd =
>       GetFirstAudioContentDescription(answer.get());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioCodecsAnswer), acd->codecs());
>+  EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
> 
>   // Now - let |f2_| add video with RTX and let the payload type the RTX codec
>   // reference  be the same as an audio codec that was negotiated in the
>@@ -2101,13 +2164,13 @@ TEST_F(MediaSessionDescriptionFactoryTest,
> 
>   const AudioContentDescription* updated_acd =
>       GetFirstAudioContentDescription(answer.get());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioCodecsAnswer), updated_acd->codecs());
>+  EXPECT_THAT(updated_acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
> 
>   const VideoContentDescription* updated_vcd =
>       GetFirstVideoContentDescription(updated_answer.get());
> 
>   ASSERT_EQ("H264", updated_vcd->codecs()[0].name);
>-  ASSERT_EQ(std::string(cricket::kRtxCodecName), updated_vcd->codecs()[1].name);
>+  ASSERT_EQ(cricket::kRtxCodecName, updated_vcd->codecs()[1].name);
>   int new_h264_pl_type = updated_vcd->codecs()[0].id;
>   EXPECT_NE(used_pl_type, new_h264_pl_type);
>   VideoCodec rtx = updated_vcd->codecs()[1];
>@@ -2776,8 +2839,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestOfferDtlsSavpfCreateAnswer) {
> 
>   const AudioContentDescription* answer_audio_desc =
>       answer_content->media_description()->as_audio();
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolDtlsSavpf),
>-            answer_audio_desc->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolDtlsSavpf, answer_audio_desc->protocol());
> }
> 
> // Test that we include both SDES and DTLS in the offer, but only include SDES
>@@ -2842,10 +2904,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoDtls) {
>   ASSERT_TRUE(video_media_desc != NULL);
>   EXPECT_TRUE(audio_media_desc->cryptos().empty());
>   EXPECT_TRUE(video_media_desc->cryptos().empty());
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
>-            audio_media_desc->protocol());
>-  EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf),
>-            video_media_desc->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, audio_media_desc->protocol());
>+  EXPECT_EQ(cricket::kMediaProtocolSavpf, video_media_desc->protocol());
> 
>   audio_trans_desc = answer->GetTransportDescriptionByName("audio");
>   ASSERT_TRUE(audio_trans_desc != NULL);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>index 6d67329242d75c1ca8256d07a4514c72d9f40287..a6b47c16deba381e57abfbf1230118165fb158a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>@@ -22,6 +22,7 @@
> #include "api/jsepsessiondescription.h"
> #include "api/mediastreamproxy.h"
> #include "api/mediastreamtrackproxy.h"
>+#include "api/umametrics.h"
> #include "call/call.h"
> #include "logging/rtc_event_log/icelogger.h"
> #include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
>@@ -630,6 +631,35 @@ absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
>   return rtc_configuration_parameter;
> }
> 
>+cricket::DataMessageType ToCricketDataMessageType(DataMessageType type) {
>+  switch (type) {
>+    case DataMessageType::kText:
>+      return cricket::DMT_TEXT;
>+    case DataMessageType::kBinary:
>+      return cricket::DMT_BINARY;
>+    case DataMessageType::kControl:
>+      return cricket::DMT_CONTROL;
>+    default:
>+      return cricket::DMT_NONE;
>+  }
>+  return cricket::DMT_NONE;
>+}
>+
>+DataMessageType ToWebrtcDataMessageType(cricket::DataMessageType type) {
>+  switch (type) {
>+    case cricket::DMT_TEXT:
>+      return DataMessageType::kText;
>+    case cricket::DMT_BINARY:
>+      return DataMessageType::kBinary;
>+    case cricket::DMT_CONTROL:
>+      return DataMessageType::kControl;
>+    case cricket::DMT_NONE:
>+    default:
>+      RTC_NOTREACHED();
>+  }
>+  return DataMessageType::kControl;
>+}
>+
> }  // namespace
> 
> // Upon completion, posts a task to execute the callback of the
>@@ -684,6 +714,7 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
>     CandidateNetworkPolicy candidate_network_policy;
>     int audio_jitter_buffer_max_packets;
>     bool audio_jitter_buffer_fast_accelerate;
>+    int audio_jitter_buffer_min_delay_ms;
>     int ice_connection_receiving_timeout;
>     int ice_backup_candidate_pair_ping_interval;
>     ContinualGatheringPolicy continual_gathering_policy;
>@@ -704,6 +735,10 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
>     SdpSemantics sdp_semantics;
>     absl::optional<rtc::AdapterType> network_preference;
>     bool active_reset_srtp_params;
>+    bool use_media_transport;
>+    bool use_media_transport_for_data_channels;
>+    absl::optional<CryptoOptions> crypto_options;
>+    bool offer_extmap_allow_mixed;
>   };
>   static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
>                 "Did you add something to RTCConfiguration and forget to "
>@@ -716,6 +751,8 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
>          audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
>          audio_jitter_buffer_fast_accelerate ==
>              o.audio_jitter_buffer_fast_accelerate &&
>+         audio_jitter_buffer_min_delay_ms ==
>+             o.audio_jitter_buffer_min_delay_ms &&
>          ice_connection_receiving_timeout ==
>              o.ice_connection_receiving_timeout &&
>          ice_backup_candidate_pair_ping_interval ==
>@@ -751,7 +788,12 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
>          turn_customizer == o.turn_customizer &&
>          sdp_semantics == o.sdp_semantics &&
>          network_preference == o.network_preference &&
>-         active_reset_srtp_params == o.active_reset_srtp_params;
>+         active_reset_srtp_params == o.active_reset_srtp_params &&
>+         use_media_transport == o.use_media_transport &&
>+         use_media_transport_for_data_channels ==
>+             o.use_media_transport_for_data_channels &&
>+         crypto_options == o.crypto_options &&
>+         offer_extmap_allow_mixed == o.offer_extmap_allow_mixed;
> }
> 
> bool PeerConnectionInterface::RTCConfiguration::operator!=(
>@@ -820,6 +862,7 @@ PeerConnection::~PeerConnection() {
>   webrtc_session_desc_factory_.reset();
>   sctp_invoker_.reset();
>   sctp_factory_.reset();
>+  media_transport_invoker_.reset();
>   transport_controller_.reset();
> 
>   // port_allocator_ lives on the network thread and should be destroyed there.
>@@ -929,18 +972,40 @@ bool PeerConnection::Initialize(
>   config.disable_encryption = options.disable_encryption;
>   config.bundle_policy = configuration.bundle_policy;
>   config.rtcp_mux_policy = configuration.rtcp_mux_policy;
>-  config.crypto_options = options.crypto_options;
>+  // TODO(bugs.webrtc.org/9891) - Remove options.crypto_options then remove this
>+  // stub.
>+  config.crypto_options = configuration.crypto_options.has_value()
>+                              ? *configuration.crypto_options
>+                              : options.crypto_options;
>   config.transport_observer = this;
>   config.event_log = event_log_.get();
> #if defined(ENABLE_EXTERNAL_AUTH)
>   config.enable_external_auth = true;
> #endif
>   config.active_reset_srtp_params = configuration.active_reset_srtp_params;
>+
>+  if (configuration.use_media_transport ||
>+      configuration.use_media_transport_for_data_channels) {
>+    if (!factory_->media_transport_factory()) {
>+      RTC_DCHECK(false)
>+          << "PeerConnecton is initialized with use_media_transport = true or "
>+          << "use_media_transport_for_data_channels = true "
>+          << "but media transport factory is not set in PeerConnectionFactory";
>+      return false;
>+    }
>+
>+    config.media_transport_factory = factory_->media_transport_factory();
>+  }
>+
>   transport_controller_.reset(new JsepTransportController(
>       signaling_thread(), network_thread(), port_allocator_.get(),
>       async_resolver_factory_.get(), config));
>   transport_controller_->SignalIceConnectionState.connect(
>       this, &PeerConnection::OnTransportControllerConnectionState);
>+  transport_controller_->SignalStandardizedIceConnectionState.connect(
>+      this, &PeerConnection::SetStandardizedIceConnectionState);
>+  transport_controller_->SignalConnectionState.connect(
>+      this, &PeerConnection::SetConnectionState);
>   transport_controller_->SignalIceGatheringState.connect(
>       this, &PeerConnection::OnTransportControllerGatheringState);
>   transport_controller_->SignalIceCandidatesGathered.connect(
>@@ -979,10 +1044,18 @@ bool PeerConnection::Initialize(
>     }
>   }
> 
>-  // Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
>-  // It takes precendence over the disable_sctp_data_channels
>-  // PeerConnectionFactoryInterface::Options.
>-  if (configuration.enable_rtp_data_channel) {
>+  if (configuration.use_media_transport_for_data_channels) {
>+    if (configuration.enable_rtp_data_channel) {
>+      RTC_LOG(LS_ERROR) << "enable_rtp_data_channel and "
>+                           "use_media_transport_for_data_channels are "
>+                           "incompatible and cannot both be set to true";
>+      return false;
>+    }
>+    data_channel_type_ = cricket::DCT_MEDIA_TRANSPORT;
>+  } else if (configuration.enable_rtp_data_channel) {
>+    // Enable creation of RTP data channels if the kEnableRtpDataChannels is
>+    // set. It takes precendence over the disable_sctp_data_channels
>+    // PeerConnectionFactoryInterface::Options.
>     data_channel_type_ = cricket::DCT_RTP;
>   } else {
>     // DTLS has to be enabled to use SCTP.
>@@ -1002,6 +1075,9 @@ bool PeerConnection::Initialize(
>   audio_options_.audio_jitter_buffer_fast_accelerate =
>       configuration.audio_jitter_buffer_fast_accelerate;
> 
>+  audio_options_.audio_jitter_buffer_min_delay_ms =
>+      configuration.audio_jitter_buffer_min_delay_ms;
>+
>   // Whether the certificate generator/certificate is null or not determines
>   // what PeerConnectionDescriptionFactory will do, so make sure that we give it
>   // the right instructions by clearing the variables if needed.
>@@ -1024,7 +1100,7 @@ bool PeerConnection::Initialize(
>   }
> 
>   webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
>-      options.crypto_options.enable_encrypted_rtp_header_extensions);
>+      GetCryptoOptions().srtp.enable_encrypted_rtp_header_extensions);
> 
>   // Add default audio/video transceivers for Plan B SDP.
>   if (!IsUnifiedPlan()) {
>@@ -1183,7 +1259,7 @@ PeerConnection::AddTrackPlanB(
>   auto new_sender =
>       CreateSender(media_type, track->id(), track, adjusted_stream_ids, {});
>   if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
>-    new_sender->internal()->SetVoiceMediaChannel(voice_media_channel());
>+    new_sender->internal()->SetMediaChannel(voice_media_channel());
>     GetAudioTransceiver()->internal()->AddSender(new_sender);
>     const RtpSenderInfo* sender_info =
>         FindSenderInfo(local_audio_sender_infos_,
>@@ -1193,7 +1269,7 @@ PeerConnection::AddTrackPlanB(
>     }
>   } else {
>     RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
>-    new_sender->internal()->SetVideoMediaChannel(video_media_channel());
>+    new_sender->internal()->SetMediaChannel(video_media_channel());
>     GetVideoTransceiver()->internal()->AddSender(new_sender);
>     const RtpSenderInfo* sender_info =
>         FindSenderInfo(local_video_sender_infos_,
>@@ -1523,14 +1599,14 @@ rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
>   if (kind == MediaStreamTrackInterface::kAudioKind) {
>     auto* audio_sender = new AudioRtpSender(
>         worker_thread(), rtc::CreateRandomUuid(), stats_.get());
>-    audio_sender->SetVoiceMediaChannel(voice_media_channel());
>+    audio_sender->SetMediaChannel(voice_media_channel());
>     new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
>         signaling_thread(), audio_sender);
>     GetAudioTransceiver()->internal()->AddSender(new_sender);
>   } else if (kind == MediaStreamTrackInterface::kVideoKind) {
>     auto* video_sender =
>         new VideoRtpSender(worker_thread(), rtc::CreateRandomUuid());
>-    video_sender->SetVideoMediaChannel(video_media_channel());
>+    video_sender->SetMediaChannel(video_media_channel());
>     new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
>         signaling_thread(), video_sender);
>     GetVideoTransceiver()->internal()->AddSender(new_sender);
>@@ -1647,7 +1723,7 @@ void PeerConnection::GetStats(
>         break;
>     }
>   }
>-  // If there is no |internal_sender|Â then |selector| is either null or does not
>+  // If there is no |internal_sender| then |selector| is either null or does not
>   // belong to the PeerConnection (in Plan B, senders can be removed from the
>   // PeerConnection). This means that "all the stats objects representing the
>   // selector" is an empty set. Invoking GetStatsReport() with a null selector
>@@ -1675,7 +1751,7 @@ void PeerConnection::GetStats(
>         break;
>     }
>   }
>-  // If there is no |internal_receiver|Â then |selector| is either null or does
>+  // If there is no |internal_receiver| then |selector| is either null or does
>   // not belong to the PeerConnection (in Plan B, receivers can be removed from
>   // the PeerConnection). This means that "all the stats objects representing
>   // the selector" is an empty set. Invoking GetStatsReport() with a null
>@@ -1692,6 +1768,16 @@ PeerConnection::ice_connection_state() {
>   return ice_connection_state_;
> }
> 
>+PeerConnectionInterface::IceConnectionState
>+PeerConnection::standardized_ice_connection_state() {
>+  return standardized_ice_connection_state_;
>+}
>+
>+PeerConnectionInterface::PeerConnectionState
>+PeerConnection::peer_connection_state() {
>+  return connection_state_;
>+}
>+
> PeerConnectionInterface::IceGatheringState
> PeerConnection::ice_gathering_state() {
>   return ice_gathering_state_;
>@@ -1995,6 +2081,16 @@ RTCError PeerConnection::ApplyLocalDescription(
>   // |local_description()|.
>   RTC_DCHECK(local_description());
> 
>+  if (!is_caller_) {
>+    if (remote_description()) {
>+      // Remote description was applied first, so this PC is the callee.
>+      is_caller_ = false;
>+    } else {
>+      // Local description is applied first, so this PC is the caller.
>+      is_caller_ = true;
>+    }
>+  }
>+
>   RTCError error = PushdownTransportDescription(cricket::CS_LOCAL, type);
>   if (!error.ok()) {
>     return error;
>@@ -2077,7 +2173,7 @@ RTCError PeerConnection::ApplyLocalDescription(
>   // If setting the description decided our SSL role, allocate any necessary
>   // SCTP sids.
>   rtc::SSLRole role;
>-  if (data_channel_type() == cricket::DCT_SCTP && GetSctpSslRole(&role)) {
>+  if (DataChannel::IsSctpLike(data_channel_type_) && GetSctpSslRole(&role)) {
>     AllocateSctpSids(role);
>   }
> 
>@@ -2352,7 +2448,7 @@ RTCError PeerConnection::ApplyRemoteDescription(
>   // If setting the description decided our SSL role, allocate any necessary
>   // SCTP sids.
>   rtc::SSLRole role;
>-  if (data_channel_type() == cricket::DCT_SCTP && GetSctpSslRole(&role)) {
>+  if (DataChannel::IsSctpLike(data_channel_type_) && GetSctpSslRole(&role)) {
>     AllocateSctpSids(role);
>   }
> 
>@@ -2646,11 +2742,11 @@ RTCError PeerConnection::UpdateTransceiverChannel(
>     const cricket::ContentGroup* bundle_group) {
>   RTC_DCHECK(IsUnifiedPlan());
>   RTC_DCHECK(transceiver);
>-  cricket::BaseChannel* channel = transceiver->internal()->channel();
>+  cricket::ChannelInterface* channel = transceiver->internal()->channel();
>   if (content.rejected) {
>     if (channel) {
>       transceiver->internal()->SetChannel(nullptr);
>-      DestroyBaseChannel(channel);
>+      DestroyChannelInterface(channel);
>     }
>   } else {
>     if (!channel) {
>@@ -2683,7 +2779,7 @@ RTCError PeerConnection::UpdateDataChannel(
>   if (content.rejected) {
>     DestroyDataChannel();
>   } else {
>-    if (!rtp_data_channel_ && !sctp_transport_) {
>+    if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) {
>       if (!CreateDataChannel(content.name)) {
>         LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>                              "Failed to create data channel.");
>@@ -2870,6 +2966,49 @@ bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
>     return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
>   }
> 
>+  if (local_description() &&
>+      configuration.use_media_transport != configuration_.use_media_transport) {
>+    RTC_LOG(LS_ERROR) << "Can't change media_transport after calling "
>+                         "SetLocalDescription.";
>+    return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
>+  }
>+
>+  if (remote_description() &&
>+      configuration.use_media_transport != configuration_.use_media_transport) {
>+    RTC_LOG(LS_ERROR) << "Can't change media_transport after calling "
>+                         "SetRemoteDescription.";
>+    return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
>+  }
>+
>+  if (local_description() &&
>+      configuration.use_media_transport_for_data_channels !=
>+          configuration_.use_media_transport_for_data_channels) {
>+    RTC_LOG(LS_ERROR) << "Can't change media_transport_for_data_channels "
>+                         "after calling SetLocalDescription.";
>+    return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
>+  }
>+
>+  if (remote_description() &&
>+      configuration.use_media_transport_for_data_channels !=
>+          configuration_.use_media_transport_for_data_channels) {
>+    RTC_LOG(LS_ERROR) << "Can't change media_transport_for_data_channels "
>+                         "after calling SetRemoteDescription.";
>+    return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
>+  }
>+
>+  if (local_description() &&
>+      configuration.crypto_options != configuration_.crypto_options) {
>+    RTC_LOG(LS_ERROR) << "Can't change crypto_options after calling "
>+                         "SetLocalDescription.";
>+    return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
>+  }
>+
>+  if (configuration.use_media_transport_for_data_channels ||
>+      configuration.use_media_transport) {
>+    RTC_CHECK(configuration.bundle_policy == kBundlePolicyMaxBundle)
>+        << "Media transport requires MaxBundle policy.";
>+  }
>+
>   // The simplest (and most future-compatible) way to tell if the config was
>   // modified in an invalid way is to copy each property we do support
>   // modifying, then use operator==. There are far more properties we don't
>@@ -2894,6 +3033,9 @@ bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
>   modified_config.network_preference = configuration.network_preference;
>   modified_config.active_reset_srtp_params =
>       configuration.active_reset_srtp_params;
>+  modified_config.use_media_transport = configuration.use_media_transport;
>+  modified_config.use_media_transport_for_data_channels =
>+      configuration.use_media_transport_for_data_channels;
>   if (configuration != modified_config) {
>     RTC_LOG(LS_ERROR) << "Modifying the configuration in an unsupported way.";
>     return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
>@@ -2951,6 +3093,11 @@ bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
>   }
> 
>   transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
>+  transport_controller_->SetMediaTransportFactory(
>+      modified_config.use_media_transport ||
>+              modified_config.use_media_transport_for_data_channels
>+          ? factory_->media_transport_factory()
>+          : nullptr);
> 
>   if (configuration_.active_reset_srtp_params !=
>       modified_config.active_reset_srtp_params) {
>@@ -3119,7 +3266,7 @@ void PeerConnection::SetAudioPlayout(bool playout) {
>     return;
>   }
>   auto audio_state =
>-      factory_->channel_manager()->media_engine()->GetAudioState();
>+      factory_->channel_manager()->media_engine()->voice().GetAudioState();
>   audio_state->SetPlayout(playout);
> }
> 
>@@ -3131,7 +3278,7 @@ void PeerConnection::SetAudioRecording(bool recording) {
>     return;
>   }
>   auto audio_state =
>-      factory_->channel_manager()->media_engine()->GetAudioState();
>+      factory_->channel_manager()->media_engine()->voice().GetAudioState();
>   audio_state->SetRecording(recording);
> }
> 
>@@ -3141,7 +3288,7 @@ PeerConnection::GetRemoteAudioSSLCertificate() {
>   if (!chain || !chain->GetSize()) {
>     return nullptr;
>   }
>-  return chain->Get(0).GetUniqueReference();
>+  return chain->Get(0).Clone();
> }
> 
> std::unique_ptr<rtc::SSLCertChain>
>@@ -3170,9 +3317,13 @@ bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
>   const size_t max_size = (max_size_bytes < 0)
>                               ? RtcEventLog::kUnlimitedOutput
>                               : rtc::saturated_cast<size_t>(max_size_bytes);
>+  int64_t output_period_ms = webrtc::RtcEventLog::kImmediateOutput;
>+  if (field_trial::IsEnabled("WebRTC-RtcEventLogNewFormat")) {
>+    output_period_ms = 5000;
>+  }
>   return StartRtcEventLog(
>       absl::make_unique<RtcEventLogOutputFile>(file, max_size),
>-      webrtc::RtcEventLog::kImmediateOutput);
>+      output_period_ms);
> }
> 
> bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
>@@ -3347,7 +3498,7 @@ void PeerConnection::CreateAudioReceiver(
>   // the constructor taking stream IDs instead.
>   auto* audio_receiver = new AudioRtpReceiver(
>       worker_thread(), remote_sender_info.sender_id, streams);
>-  audio_receiver->SetVoiceMediaChannel(voice_media_channel());
>+  audio_receiver->SetMediaChannel(voice_media_channel());
>   audio_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
>   auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
>       signaling_thread(), audio_receiver);
>@@ -3365,7 +3516,7 @@ void PeerConnection::CreateVideoReceiver(
>   // the constructor taking stream IDs instead.
>   auto* video_receiver = new VideoRtpReceiver(
>       worker_thread(), remote_sender_info.sender_id, streams);
>-  video_receiver->SetVideoMediaChannel(video_media_channel());
>+  video_receiver->SetMediaChannel(video_media_channel());
>   video_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
>   auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
>       signaling_thread(), video_receiver);
>@@ -3408,7 +3559,7 @@ void PeerConnection::AddAudioTrack(AudioTrackInterface* track,
>   // Normal case; we've never seen this track before.
>   auto new_sender = CreateSender(cricket::MEDIA_TYPE_AUDIO, track->id(), track,
>                                  {stream->id()}, {});
>-  new_sender->internal()->SetVoiceMediaChannel(voice_media_channel());
>+  new_sender->internal()->SetMediaChannel(voice_media_channel());
>   GetAudioTransceiver()->internal()->AddSender(new_sender);
>   // If the sender has already been configured in SDP, we call SetSsrc,
>   // which will connect the sender to the underlying transport. This can
>@@ -3453,7 +3604,7 @@ void PeerConnection::AddVideoTrack(VideoTrackInterface* track,
>   // Normal case; we've never seen this track before.
>   auto new_sender = CreateSender(cricket::MEDIA_TYPE_VIDEO, track->id(), track,
>                                  {stream->id()}, {});
>-  new_sender->internal()->SetVideoMediaChannel(video_media_channel());
>+  new_sender->internal()->SetMediaChannel(video_media_channel());
>   GetVideoTransceiver()->internal()->AddSender(new_sender);
>   const RtpSenderInfo* sender_info =
>       FindSenderInfo(local_video_sender_infos_, stream->id(), track->id());
>@@ -3495,6 +3646,29 @@ void PeerConnection::SetIceConnectionState(IceConnectionState new_state) {
>   Observer()->OnIceConnectionChange(ice_connection_state_);
> }
> 
>+void PeerConnection::SetStandardizedIceConnectionState(
>+    PeerConnectionInterface::IceConnectionState new_state) {
>+  RTC_DCHECK(signaling_thread()->IsCurrent());
>+  if (standardized_ice_connection_state_ == new_state)
>+    return;
>+  if (IsClosed())
>+    return;
>+  standardized_ice_connection_state_ = new_state;
>+  // TODO(jonasolsson): Pass this value on to OnIceConnectionChange instead of
>+  // the old one once disconnects are handled properly.
>+}
>+
>+void PeerConnection::SetConnectionState(
>+    PeerConnectionInterface::PeerConnectionState new_state) {
>+  RTC_DCHECK(signaling_thread()->IsCurrent());
>+  if (connection_state_ == new_state)
>+    return;
>+  if (IsClosed())
>+    return;
>+  connection_state_ = new_state;
>+  Observer()->OnConnectionChange(new_state);
>+}
>+
> void PeerConnection::OnIceGatheringChange(
>     PeerConnectionInterface::IceGatheringState new_state) {
>   RTC_DCHECK(signaling_thread()->IsCurrent());
>@@ -3542,6 +3716,10 @@ void PeerConnection::ChangeSignalingState(
>   if (signaling_state == kClosed) {
>     ice_connection_state_ = kIceConnectionClosed;
>     Observer()->OnIceConnectionChange(ice_connection_state_);
>+    standardized_ice_connection_state_ =
>+        PeerConnectionInterface::IceConnectionState::kIceConnectionClosed;
>+    connection_state_ = PeerConnectionInterface::PeerConnectionState::kClosed;
>+    Observer()->OnConnectionChange(connection_state_);
>     if (ice_gathering_state_ != kIceGatheringComplete) {
>       ice_gathering_state_ = kIceGatheringComplete;
>       Observer()->OnIceGatheringChange(ice_gathering_state_);
>@@ -3641,8 +3819,15 @@ void PeerConnection::GetOptionsForOffer(
>   }
> 
>   session_options->rtcp_cname = rtcp_cname_;
>-  session_options->crypto_options = factory_->options().crypto_options;
>+  session_options->crypto_options = GetCryptoOptions();
>   session_options->is_unified_plan = IsUnifiedPlan();
>+  session_options->pooled_ice_credentials =
>+      network_thread()->Invoke<std::vector<cricket::IceParameters>>(
>+          RTC_FROM_HERE,
>+          rtc::Bind(&cricket::PortAllocator::GetPooledIceCredentials,
>+                    port_allocator_.get()));
>+  session_options->offer_extmap_allow_mixed =
>+      configuration_.offer_extmap_allow_mixed;
> }
> 
> void PeerConnection::GetOptionsForPlanBOffer(
>@@ -3901,8 +4086,13 @@ void PeerConnection::GetOptionsForAnswer(
>   }
> 
>   session_options->rtcp_cname = rtcp_cname_;
>-  session_options->crypto_options = factory_->options().crypto_options;
>+  session_options->crypto_options = GetCryptoOptions();
>   session_options->is_unified_plan = IsUnifiedPlan();
>+  session_options->pooled_ice_credentials =
>+      network_thread()->Invoke<std::vector<cricket::IceParameters>>(
>+          RTC_FROM_HERE,
>+          rtc::Bind(&cricket::PortAllocator::GetPooledIceCredentials,
>+                    port_allocator_.get()));
> }
> 
> void PeerConnection::GetOptionsForPlanBAnswer(
>@@ -4069,6 +4259,8 @@ absl::optional<std::string> PeerConnection::GetDataMid() const {
>       return rtp_data_channel_->content_name();
>     case cricket::DCT_SCTP:
>       return sctp_mid_;
>+    case cricket::DCT_MEDIA_TRANSPORT:
>+      return media_transport_data_mid_;
>     default:
>       return absl::nullopt;
>   }
>@@ -4431,7 +4623,7 @@ rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
>   }
>   InternalDataChannelInit new_config =
>       config ? (*config) : InternalDataChannelInit();
>-  if (data_channel_type() == cricket::DCT_SCTP) {
>+  if (DataChannel::IsSctpLike(data_channel_type_)) {
>     if (new_config.id < 0) {
>       rtc::SSLRole role;
>       if ((GetSctpSslRole(&role)) &&
>@@ -4462,7 +4654,7 @@ rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
>     }
>     rtp_data_channels_[channel->label()] = channel;
>   } else {
>-    RTC_DCHECK(channel->data_channel_type() == cricket::DCT_SCTP);
>+    RTC_DCHECK(DataChannel::IsSctpLike(data_channel_type_));
>     sctp_data_channels_.push_back(channel);
>     channel->SignalClosed.connect(this,
>                                   &PeerConnection::OnSctpDataChannelClosed);
>@@ -4542,6 +4734,27 @@ void PeerConnection::OnDataChannelOpenMessage(
>   NoteUsageEvent(UsageEvent::DATA_ADDED);
> }
> 
>+bool PeerConnection::HandleOpenMessage_s(
>+    const cricket::ReceiveDataParams& params,
>+    const rtc::CopyOnWriteBuffer& buffer) {
>+  if (params.type == cricket::DMT_CONTROL && IsOpenMessage(buffer)) {
>+    // Received OPEN message; parse and signal that a new data channel should
>+    // be created.
>+    std::string label;
>+    InternalDataChannelInit config;
>+    config.id = params.ssrc;
>+    if (!ParseDataChannelOpenMessage(buffer, &label, &config)) {
>+      RTC_LOG(LS_WARNING) << "Failed to parse the OPEN message for ssrc "
>+                          << params.ssrc;
>+      return true;
>+    }
>+    config.open_handshake_role = InternalDataChannelInit::kAcker;
>+    OnDataChannelOpenMessage(label, config);
>+    return true;
>+  }
>+  return false;
>+}
>+
> rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
> PeerConnection::GetAudioTransceiver() const {
>   // This method only works with Plan B SDP, where there is a single
>@@ -4769,10 +4982,10 @@ void PeerConnection::StopRtcEventLog_w() {
>   }
> }
> 
>-cricket::BaseChannel* PeerConnection::GetChannel(
>+cricket::ChannelInterface* PeerConnection::GetChannel(
>     const std::string& content_name) {
>   for (auto transceiver : transceivers_) {
>-    cricket::BaseChannel* channel = transceiver->internal()->channel();
>+    cricket::ChannelInterface* channel = transceiver->internal()->channel();
>     if (channel && channel->content_name() == content_name) {
>       return channel;
>     }
>@@ -4785,19 +4998,25 @@ cricket::BaseChannel* PeerConnection::GetChannel(
> }
> 
> bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
>+  RTC_DCHECK_RUN_ON(signaling_thread());
>   if (!local_description() || !remote_description()) {
>     RTC_LOG(LS_INFO)
>         << "Local and Remote descriptions must be applied to get the "
>            "SSL Role of the SCTP transport.";
>     return false;
>   }
>-  if (!sctp_transport_) {
>+  if (!sctp_transport_ && !media_transport_) {
>     RTC_LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
>                         "SSL Role of the SCTP transport.";
>     return false;
>   }
> 
>-  auto dtls_role = transport_controller_->GetDtlsRole(*sctp_mid_);
>+  absl::optional<rtc::SSLRole> dtls_role;
>+  if (sctp_mid_) {
>+    dtls_role = transport_controller_->GetDtlsRole(*sctp_mid_);
>+  } else if (is_caller_) {
>+    dtls_role = *is_caller_ ? rtc::SSL_SERVER : rtc::SSL_CLIENT;
>+  }
>   if (dtls_role) {
>     *role = *dtls_role;
>     return true;
>@@ -4883,7 +5102,7 @@ RTCError PeerConnection::PushdownMediaDescription(
>   for (auto transceiver : transceivers_) {
>     const ContentInfo* content_info =
>         FindMediaSectionForTransceiver(transceiver, sdesc);
>-    cricket::BaseChannel* channel = transceiver->internal()->channel();
>+    cricket::ChannelInterface* channel = transceiver->internal()->channel();
>     if (!channel || !content_info || content_info->rejected) {
>       continue;
>     }
>@@ -5043,11 +5262,22 @@ bool PeerConnection::GetRemoteTrackIdBySsrc(uint32_t ssrc,
> bool PeerConnection::SendData(const cricket::SendDataParams& params,
>                               const rtc::CopyOnWriteBuffer& payload,
>                               cricket::SendDataResult* result) {
>-  if (!rtp_data_channel_ && !sctp_transport_) {
>-    RTC_LOG(LS_ERROR) << "SendData called when rtp_data_channel_ "
>-                         "and sctp_transport_ are NULL.";
>+  if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) {
>+    RTC_LOG(LS_ERROR) << "SendData called when rtp_data_channel_, "
>+                         "sctp_transport_, and media_transport_ are NULL.";
>     return false;
>   }
>+  if (media_transport_) {
>+    SendDataParams send_params;
>+    send_params.type = ToWebrtcDataMessageType(params.type);
>+    send_params.ordered = params.ordered;
>+    if (params.max_rtx_count >= 0) {
>+      send_params.max_rtx_count = params.max_rtx_count;
>+    } else if (params.max_rtx_ms >= 0) {
>+      send_params.max_rtx_ms = params.max_rtx_ms;
>+    }
>+    return media_transport_->SendData(params.sid, send_params, payload).ok();
>+  }
>   return rtp_data_channel_
>              ? rtp_data_channel_->SendData(params, payload, result)
>              : network_thread()->Invoke<bool>(
>@@ -5057,13 +5287,23 @@ bool PeerConnection::SendData(const cricket::SendDataParams& params,
> }
> 
> bool PeerConnection::ConnectDataChannel(DataChannel* webrtc_data_channel) {
>-  if (!rtp_data_channel_ && !sctp_transport_) {
>+  RTC_DCHECK_RUN_ON(signaling_thread());
>+  if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) {
>     // Don't log an error here, because DataChannels are expected to call
>     // ConnectDataChannel in this state. It's the only way to initially tell
>     // whether or not the underlying transport is ready.
>     return false;
>   }
>-  if (rtp_data_channel_) {
>+  if (media_transport_) {
>+    SignalMediaTransportWritable_s.connect(webrtc_data_channel,
>+                                           &DataChannel::OnChannelReady);
>+    SignalMediaTransportReceivedData_s.connect(webrtc_data_channel,
>+                                               &DataChannel::OnDataReceived);
>+    SignalMediaTransportChannelClosing_s.connect(
>+        webrtc_data_channel, &DataChannel::OnClosingProcedureStartedRemotely);
>+    SignalMediaTransportChannelClosed_s.connect(
>+        webrtc_data_channel, &DataChannel::OnClosingProcedureComplete);
>+  } else if (rtp_data_channel_) {
>     rtp_data_channel_->SignalReadyToSendData.connect(
>         webrtc_data_channel, &DataChannel::OnChannelReady);
>     rtp_data_channel_->SignalDataReceived.connect(webrtc_data_channel,
>@@ -5082,13 +5322,19 @@ bool PeerConnection::ConnectDataChannel(DataChannel* webrtc_data_channel) {
> }
> 
> void PeerConnection::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
>-  if (!rtp_data_channel_ && !sctp_transport_) {
>+  RTC_DCHECK_RUN_ON(signaling_thread());
>+  if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) {
>     RTC_LOG(LS_ERROR)
>         << "DisconnectDataChannel called when rtp_data_channel_ and "
>            "sctp_transport_ are NULL.";
>     return;
>   }
>-  if (rtp_data_channel_) {
>+  if (media_transport_) {
>+    SignalMediaTransportWritable_s.disconnect(webrtc_data_channel);
>+    SignalMediaTransportReceivedData_s.disconnect(webrtc_data_channel);
>+    SignalMediaTransportChannelClosing_s.disconnect(webrtc_data_channel);
>+    SignalMediaTransportChannelClosed_s.disconnect(webrtc_data_channel);
>+  } else if (rtp_data_channel_) {
>     rtp_data_channel_->SignalReadyToSendData.disconnect(webrtc_data_channel);
>     rtp_data_channel_->SignalDataReceived.disconnect(webrtc_data_channel);
>   } else {
>@@ -5100,6 +5346,10 @@ void PeerConnection::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
> }
> 
> void PeerConnection::AddSctpDataStream(int sid) {
>+  if (media_transport_) {
>+    // No-op.  Media transport does not need to add streams.
>+    return;
>+  }
>   if (!sctp_transport_) {
>     RTC_LOG(LS_ERROR)
>         << "AddSctpDataStream called when sctp_transport_ is NULL.";
>@@ -5111,6 +5361,10 @@ void PeerConnection::AddSctpDataStream(int sid) {
> }
> 
> void PeerConnection::RemoveSctpDataStream(int sid) {
>+  if (media_transport_) {
>+    media_transport_->CloseChannel(sid);
>+    return;
>+  }
>   if (!sctp_transport_) {
>     RTC_LOG(LS_ERROR) << "RemoveSctpDataStream called when sctp_transport_ is "
>                          "NULL.";
>@@ -5122,10 +5376,43 @@ void PeerConnection::RemoveSctpDataStream(int sid) {
> }
> 
> bool PeerConnection::ReadyToSendData() const {
>+  RTC_DCHECK_RUN_ON(signaling_thread());
>   return (rtp_data_channel_ && rtp_data_channel_->ready_to_send_data()) ||
>+         (media_transport_ && media_transport_ready_to_send_data_) ||
>          sctp_ready_to_send_data_;
> }
> 
>+void PeerConnection::OnDataReceived(int channel_id,
>+                                    DataMessageType type,
>+                                    const rtc::CopyOnWriteBuffer& buffer) {
>+  cricket::ReceiveDataParams params;
>+  params.sid = channel_id;
>+  params.type = ToCricketDataMessageType(type);
>+  media_transport_invoker_->AsyncInvoke<void>(
>+      RTC_FROM_HERE, signaling_thread(), [this, params, buffer] {
>+        RTC_DCHECK_RUN_ON(signaling_thread());
>+        if (!HandleOpenMessage_s(params, buffer)) {
>+          SignalMediaTransportReceivedData_s(params, buffer);
>+        }
>+      });
>+}
>+
>+void PeerConnection::OnChannelClosing(int channel_id) {
>+  media_transport_invoker_->AsyncInvoke<void>(
>+      RTC_FROM_HERE, signaling_thread(), [this, channel_id] {
>+        RTC_DCHECK_RUN_ON(signaling_thread());
>+        SignalMediaTransportChannelClosing_s(channel_id);
>+      });
>+}
>+
>+void PeerConnection::OnChannelClosed(int channel_id) {
>+  media_transport_invoker_->AsyncInvoke<void>(
>+      RTC_FROM_HERE, signaling_thread(), [this, channel_id] {
>+        RTC_DCHECK_RUN_ON(signaling_thread());
>+        SignalMediaTransportChannelClosed_s(channel_id);
>+      });
>+}
>+
> absl::optional<std::string> PeerConnection::sctp_transport_name() const {
>   if (sctp_mid_ && transport_controller_) {
>     auto dtls_transport = transport_controller_->GetDtlsTransport(*sctp_mid_);
>@@ -5150,7 +5437,7 @@ std::map<std::string, std::string> PeerConnection::GetTransportNamesByMid()
>     const {
>   std::map<std::string, std::string> transport_names_by_mid;
>   for (auto transceiver : transceivers_) {
>-    cricket::BaseChannel* channel = transceiver->internal()->channel();
>+    cricket::ChannelInterface* channel = transceiver->internal()->channel();
>     if (channel) {
>       transport_names_by_mid[channel->content_name()] =
>           channel->transport_name();
>@@ -5327,7 +5614,7 @@ void PeerConnection::OnTransportControllerDtlsHandshakeError(
> 
> void PeerConnection::EnableSending() {
>   for (auto transceiver : transceivers_) {
>-    cricket::BaseChannel* channel = transceiver->internal()->channel();
>+    cricket::ChannelInterface* channel = transceiver->internal()->channel();
>     if (channel && !channel->enabled()) {
>       channel->Enable(true);
>     }
>@@ -5486,7 +5773,7 @@ RTCError PeerConnection::CreateChannels(const SessionDescription& desc) {
> 
>   const cricket::ContentInfo* data = cricket::GetFirstDataContent(&desc);
>   if (data_channel_type_ != cricket::DCT_NONE && data && !data->rejected &&
>-      !rtp_data_channel_ && !sctp_transport_) {
>+      !rtp_data_channel_ && !sctp_transport_ && !media_transport_) {
>     if (!CreateDataChannel(data->name)) {
>       LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>                            "Failed to create data channel.");
>@@ -5499,13 +5786,16 @@ RTCError PeerConnection::CreateChannels(const SessionDescription& desc) {
> // TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
> cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
>     const std::string& mid) {
>-  RtpTransportInternal* rtp_transport =
>-      transport_controller_->GetRtpTransport(mid);
>-  RTC_DCHECK(rtp_transport);
>+  RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
>+  MediaTransportInterface* media_transport = nullptr;
>+  if (configuration_.use_media_transport) {
>+    media_transport = GetMediaTransport(mid);
>+  }
>+
>   cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
>-      call_.get(), configuration_.media_config, rtp_transport,
>-      signaling_thread(), mid, SrtpRequired(),
>-      factory_->options().crypto_options, audio_options_);
>+      call_.get(), configuration_.media_config, rtp_transport, media_transport,
>+      signaling_thread(), mid, SrtpRequired(), GetCryptoOptions(),
>+      audio_options_);
>   if (!voice_channel) {
>     return nullptr;
>   }
>@@ -5521,13 +5811,13 @@ cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
> // TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
> cricket::VideoChannel* PeerConnection::CreateVideoChannel(
>     const std::string& mid) {
>-  RtpTransportInternal* rtp_transport =
>-      transport_controller_->GetRtpTransport(mid);
>-  RTC_DCHECK(rtp_transport);
>+  RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
>+
>+  // TODO(sukhanov): Propagate media_transport to video channel.
>   cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
>       call_.get(), configuration_.media_config, rtp_transport,
>-      signaling_thread(), mid, SrtpRequired(),
>-      factory_->options().crypto_options, video_options_);
>+      signaling_thread(), mid, SrtpRequired(), GetCryptoOptions(),
>+      video_options_);
>   if (!video_channel) {
>     return nullptr;
>   }
>@@ -5541,37 +5831,49 @@ cricket::VideoChannel* PeerConnection::CreateVideoChannel(
> }
> 
> bool PeerConnection::CreateDataChannel(const std::string& mid) {
>-  bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
>-  if (sctp) {
>-    if (!sctp_factory_) {
>-      RTC_LOG(LS_ERROR)
>-          << "Trying to create SCTP transport, but didn't compile with "
>-             "SCTP support (HAVE_SCTP)";
>-      return false;
>-    }
>-    if (!network_thread()->Invoke<bool>(
>-            RTC_FROM_HERE,
>-            rtc::Bind(&PeerConnection::CreateSctpTransport_n, this, mid))) {
>-      return false;
>-    }
>-    for (const auto& channel : sctp_data_channels_) {
>-      channel->OnTransportChannelCreated();
>-    }
>-  } else {
>-    RtpTransportInternal* rtp_transport =
>-        transport_controller_->GetRtpTransport(mid);
>-    RTC_DCHECK(rtp_transport);
>-    rtp_data_channel_ = channel_manager()->CreateRtpDataChannel(
>-        configuration_.media_config, rtp_transport, signaling_thread(), mid,
>-        SrtpRequired(), factory_->options().crypto_options);
>-    if (!rtp_data_channel_) {
>+  switch (data_channel_type_) {
>+    case cricket::DCT_MEDIA_TRANSPORT:
>+      if (network_thread()->Invoke<bool>(
>+              RTC_FROM_HERE,
>+              rtc::Bind(&PeerConnection::SetupMediaTransportForDataChannels_n,
>+                        this, mid))) {
>+        for (const auto& channel : sctp_data_channels_) {
>+          channel->OnTransportChannelCreated();
>+        }
>+        return true;
>+      }
>       return false;
>-    }
>-    rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect(
>-        this, &PeerConnection::OnDtlsSrtpSetupFailure);
>-    rtp_data_channel_->SignalSentPacket.connect(
>-        this, &PeerConnection::OnSentPacket_w);
>-    rtp_data_channel_->SetRtpTransport(rtp_transport);
>+    case cricket::DCT_SCTP:
>+      if (!sctp_factory_) {
>+        RTC_LOG(LS_ERROR)
>+            << "Trying to create SCTP transport, but didn't compile with "
>+               "SCTP support (HAVE_SCTP)";
>+        return false;
>+      }
>+      if (!network_thread()->Invoke<bool>(
>+              RTC_FROM_HERE,
>+              rtc::Bind(&PeerConnection::CreateSctpTransport_n, this, mid))) {
>+        return false;
>+      }
>+      for (const auto& channel : sctp_data_channels_) {
>+        channel->OnTransportChannelCreated();
>+      }
>+      return true;
>+    case cricket::DCT_RTP:
>+    default:
>+      RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
>+      rtp_data_channel_ = channel_manager()->CreateRtpDataChannel(
>+          configuration_.media_config, rtp_transport, signaling_thread(), mid,
>+          SrtpRequired(), GetCryptoOptions());
>+      if (!rtp_data_channel_) {
>+        return false;
>+      }
>+      rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect(
>+          this, &PeerConnection::OnDtlsSrtpSetupFailure);
>+      rtp_data_channel_->SignalSentPacket.connect(
>+          this, &PeerConnection::OnSentPacket_w);
>+      rtp_data_channel_->SetRtpTransport(rtp_transport);
>+      return true;
>   }
> 
>   return true;
>@@ -5661,22 +5963,8 @@ void PeerConnection::OnSctpTransportDataReceived_n(
> void PeerConnection::OnSctpTransportDataReceived_s(
>     const cricket::ReceiveDataParams& params,
>     const rtc::CopyOnWriteBuffer& payload) {
>-  RTC_DCHECK(signaling_thread()->IsCurrent());
>-  if (params.type == cricket::DMT_CONTROL && IsOpenMessage(payload)) {
>-    // Received OPEN message; parse and signal that a new data channel should
>-    // be created.
>-    std::string label;
>-    InternalDataChannelInit config;
>-    config.id = params.ssrc;
>-    if (!ParseDataChannelOpenMessage(payload, &label, &config)) {
>-      RTC_LOG(LS_WARNING) << "Failed to parse the OPEN message for sid "
>-                          << params.ssrc;
>-      return;
>-    }
>-    config.open_handshake_role = InternalDataChannelInit::kAcker;
>-    OnDataChannelOpenMessage(label, config);
>-  } else {
>-    // Otherwise just forward the signal.
>+  RTC_DCHECK_RUN_ON(signaling_thread());
>+  if (!HandleOpenMessage_s(params, payload)) {
>     SignalSctpDataReceived(params, payload);
>   }
> }
>@@ -5699,6 +5987,49 @@ void PeerConnection::OnSctpClosingProcedureComplete_n(int sid) {
>                 &SignalSctpClosingProcedureComplete, sid));
> }
> 
>+bool PeerConnection::SetupMediaTransportForDataChannels_n(
>+    const std::string& mid) {
>+  media_transport_ = transport_controller_->GetMediaTransport(mid);
>+  if (!media_transport_) {
>+    RTC_LOG(LS_ERROR) << "Media transport is not available for data channels";
>+    return false;
>+  }
>+
>+  media_transport_invoker_ = absl::make_unique<rtc::AsyncInvoker>();
>+  media_transport_->SetDataSink(this);
>+  media_transport_data_mid_ = mid;
>+  transport_controller_->SignalMediaTransportStateChanged.connect(
>+      this, &PeerConnection::OnMediaTransportStateChanged_n);
>+  // Check the initial state right away, in case transport is already writable.
>+  OnMediaTransportStateChanged_n();
>+  return true;
>+}
>+
>+void PeerConnection::TeardownMediaTransportForDataChannels_n() {
>+  if (!media_transport_) {
>+    return;
>+  }
>+  transport_controller_->SignalMediaTransportStateChanged.disconnect(this);
>+  media_transport_data_mid_.reset();
>+  media_transport_->SetDataSink(nullptr);
>+  media_transport_invoker_ = nullptr;
>+  media_transport_ = nullptr;
>+}
>+
>+void PeerConnection::OnMediaTransportStateChanged_n() {
>+  if (!media_transport_data_mid_ ||
>+      transport_controller_->GetMediaTransportState(
>+          *media_transport_data_mid_) != MediaTransportState::kWritable) {
>+    return;
>+  }
>+  media_transport_invoker_->AsyncInvoke<void>(
>+      RTC_FROM_HERE, signaling_thread(), [this] {
>+        RTC_DCHECK_RUN_ON(signaling_thread());
>+        media_transport_ready_to_send_data_ = true;
>+        SignalMediaTransportWritable_s(media_transport_ready_to_send_data_);
>+      });
>+}
>+
> // Returns false if bundle is enabled and rtcp_mux is disabled.
> bool PeerConnection::ValidateBundleSettings(const SessionDescription* desc) {
>   bool bundle_enabled = desc->HasGroup(cricket::GROUP_TYPE_BUNDLE);
>@@ -6169,7 +6500,7 @@ void PeerConnection::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
> 
> const std::string PeerConnection::GetTransportName(
>     const std::string& content_name) {
>-  cricket::BaseChannel* channel = GetChannel(content_name);
>+  cricket::ChannelInterface* channel = GetChannel(content_name);
>   if (channel) {
>     return channel->transport_name();
>   }
>@@ -6188,17 +6519,17 @@ void PeerConnection::DestroyTransceiverChannel(
>         transceiver) {
>   RTC_DCHECK(transceiver);
> 
>-  cricket::BaseChannel* channel = transceiver->internal()->channel();
>+  cricket::ChannelInterface* channel = transceiver->internal()->channel();
>   if (channel) {
>     transceiver->internal()->SetChannel(nullptr);
>-    DestroyBaseChannel(channel);
>+    DestroyChannelInterface(channel);
>   }
> }
> 
> void PeerConnection::DestroyDataChannel() {
>   if (rtp_data_channel_) {
>     OnDataChannelDestroyed();
>-    DestroyBaseChannel(rtp_data_channel_);
>+    DestroyChannelInterface(rtp_data_channel_);
>     rtp_data_channel_ = nullptr;
>   }
> 
>@@ -6213,9 +6544,18 @@ void PeerConnection::DestroyDataChannel() {
>     network_thread()->Invoke<void>(RTC_FROM_HERE,
>                                    [this] { DestroySctpTransport_n(); });
>   }
>+
>+  if (media_transport_) {
>+    OnDataChannelDestroyed();
>+    network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
>+      RTC_DCHECK_RUN_ON(network_thread());
>+      TeardownMediaTransportForDataChannels_n();
>+    });
>+  }
> }
> 
>-void PeerConnection::DestroyBaseChannel(cricket::BaseChannel* channel) {
>+void PeerConnection::DestroyChannelInterface(
>+    cricket::ChannelInterface* channel) {
>   RTC_DCHECK(channel);
>   switch (channel->media_type()) {
>     case cricket::MEDIA_TYPE_AUDIO:
>@@ -6239,7 +6579,8 @@ void PeerConnection::DestroyBaseChannel(cricket::BaseChannel* channel) {
> bool PeerConnection::OnTransportChanged(
>     const std::string& mid,
>     RtpTransportInternal* rtp_transport,
>-    cricket::DtlsTransportInternal* dtls_transport) {
>+    cricket::DtlsTransportInternal* dtls_transport,
>+    MediaTransportInterface* media_transport) {
>   bool ret = true;
>   auto base_channel = GetChannel(mid);
>   if (base_channel) {
>@@ -6248,6 +6589,9 @@ bool PeerConnection::OnTransportChanged(
>   if (sctp_transport_ && mid == sctp_mid_) {
>     sctp_transport_->SetDtlsTransport(dtls_transport);
>   }
>+
>+  call_->MediaTransportChange(media_transport);
>+
>   return ret;
> }
> 
>@@ -6261,6 +6605,14 @@ PeerConnectionObserver* PeerConnection::Observer() const {
>   return observer_;
> }
> 
>+CryptoOptions PeerConnection::GetCryptoOptions() {
>+  // TODO(bugs.webrtc.org/9891) - Remove PeerConnectionFactory::CryptoOptions
>+  // after it has been removed.
>+  return configuration_.crypto_options.has_value()
>+             ? *configuration_.crypto_options
>+             : factory_->options().crypto_options;
>+}
>+
> void PeerConnection::ClearStatsCache() {
>   if (stats_collector_) {
>     stats_collector_->ClearCachedStatsReport();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h
>index 37c9d73bd46c881c91c5abdda27a97a8aadac677..7e97afab7c605395db0c9b404463682e515d376b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h
>@@ -17,6 +17,7 @@
> #include <string>
> #include <vector>
> 
>+#include "api/media_transport_interface.h"
> #include "api/peerconnectioninterface.h"
> #include "api/turncustomizer.h"
> #include "pc/iceserverparsing.h"
>@@ -51,6 +52,7 @@ class RtcEventLog;
> // - Generating stats.
> class PeerConnection : public PeerConnectionInternal,
>                        public DataChannelProviderInterface,
>+                       public DataChannelSink,
>                        public JsepTransportController::Observer,
>                        public rtc::MessageHandler,
>                        public sigslot::has_slots<> {
>@@ -145,6 +147,8 @@ class PeerConnection : public PeerConnectionInternal,
>   SignalingState signaling_state() override;
> 
>   IceConnectionState ice_connection_state() override;
>+  IceConnectionState standardized_ice_connection_state();
>+  PeerConnectionState peer_connection_state() override;
>   IceGatheringState ice_gathering_state() override;
> 
>   const SessionDescriptionInterface* local_description() const override;
>@@ -372,6 +376,11 @@ class PeerConnection : public PeerConnectionInternal,
>           receiver);
> 
>   void SetIceConnectionState(IceConnectionState new_state);
>+  void SetStandardizedIceConnectionState(
>+      PeerConnectionInterface::IceConnectionState new_state);
>+  void SetConnectionState(
>+      PeerConnectionInterface::PeerConnectionState new_state);
>+
>   // Called any time the IceGatheringState changes
>   void OnIceGatheringChange(IceGatheringState new_state);
>   // New ICE candidate has been gathered.
>@@ -624,6 +633,11 @@ class PeerConnection : public PeerConnectionInternal,
>   // Called when a valid data channel OPEN message is received.
>   void OnDataChannelOpenMessage(const std::string& label,
>                                 const InternalDataChannelInit& config);
>+  // Parses and handles open messages.  Returns true if the message is an open
>+  // message, false otherwise.
>+  bool HandleOpenMessage_s(const cricket::ReceiveDataParams& params,
>+                           const rtc::CopyOnWriteBuffer& buffer)
>+      RTC_RUN_ON(signaling_thread());
> 
>   // Returns true if the PeerConnection is configured to use Unified Plan
>   // semantics for creating offers/answers and setting local/remote
>@@ -705,7 +719,7 @@ class PeerConnection : public PeerConnectionInternal,
>   SessionError session_error() const { return session_error_; }
>   const std::string& session_error_desc() const { return session_error_desc_; }
> 
>-  cricket::BaseChannel* GetChannel(const std::string& content_name);
>+  cricket::ChannelInterface* GetChannel(const std::string& content_name);
> 
>   // Get current SSL role used by SCTP's underlying transport.
>   bool GetSctpSslRole(rtc::SSLRole* role);
>@@ -725,6 +739,13 @@ class PeerConnection : public PeerConnectionInternal,
> 
>   cricket::DataChannelType data_channel_type() const;
> 
>+  // Implements DataChannelSink.
>+  void OnDataReceived(int channel_id,
>+                      DataMessageType type,
>+                      const rtc::CopyOnWriteBuffer& buffer) override;
>+  void OnChannelClosing(int channel_id) override;
>+  void OnChannelClosed(int channel_id) override;
>+
>   // Called when an RTCCertificate is generated or retrieved by
>   // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
>   void OnCertificateReady(
>@@ -822,6 +843,11 @@ class PeerConnection : public PeerConnectionInternal,
>   void OnSctpClosingProcedureStartedRemotely_n(int sid);
>   void OnSctpClosingProcedureComplete_n(int sid);
> 
>+  bool SetupMediaTransportForDataChannels_n(const std::string& mid)
>+      RTC_RUN_ON(network_thread());
>+  void OnMediaTransportStateChanged_n() RTC_RUN_ON(network_thread());
>+  void TeardownMediaTransportForDataChannels_n() RTC_RUN_ON(network_thread());
>+
>   bool ValidateBundleSettings(const cricket::SessionDescription* desc);
>   bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
>   // Below methods are helper methods which verifies SDP.
>@@ -896,9 +922,9 @@ class PeerConnection : public PeerConnectionInternal,
>   // Destroys the RTP data channel and/or the SCTP data channel and clears it.
>   void DestroyDataChannel();
> 
>-  // Destroys the given BaseChannel. The channel cannot be accessed after this
>-  // method is called.
>-  void DestroyBaseChannel(cricket::BaseChannel* channel);
>+  // Destroys the given ChannelInterface.
>+  // The channel cannot be accessed after this method is called.
>+  void DestroyChannelInterface(cricket::ChannelInterface* channel);
> 
>   // JsepTransportController::Observer override.
>   //
>@@ -906,14 +932,41 @@ class PeerConnection : public PeerConnectionInternal,
>   // from a session description, and the mapping from m= sections to transports
>   // changed (as a result of BUNDLE negotiation, or m= sections being
>   // rejected).
>-  bool OnTransportChanged(
>-      const std::string& mid,
>-      RtpTransportInternal* rtp_transport,
>-      cricket::DtlsTransportInternal* dtls_transport) override;
>+  bool OnTransportChanged(const std::string& mid,
>+                          RtpTransportInternal* rtp_transport,
>+                          cricket::DtlsTransportInternal* dtls_transport,
>+                          MediaTransportInterface* media_transport) override;
> 
>   // Returns the observer. Will crash on CHECK if the observer is removed.
>   PeerConnectionObserver* Observer() const;
> 
>+  // Returns the CryptoOptions for this PeerConnection. This will always
>+  // return the RTCConfiguration.crypto_options if set and will only default
>+  // back to the PeerConnectionFactory settings if nothing was set.
>+  CryptoOptions GetCryptoOptions();
>+
>+  // Returns rtp transport, result can not be nullptr.
>+  RtpTransportInternal* GetRtpTransport(const std::string& mid) {
>+    auto rtp_transport = transport_controller_->GetRtpTransport(mid);
>+    RTC_DCHECK(rtp_transport);
>+    return rtp_transport;
>+  }
>+
>+  // Returns media transport, if PeerConnection was created with configuration
>+  // to use media transport. Otherwise returns nullptr.
>+  MediaTransportInterface* GetMediaTransport(const std::string& mid) {
>+    auto media_transport = transport_controller_->GetMediaTransport(mid);
>+    RTC_DCHECK((configuration_.use_media_transport ||
>+                configuration_.use_media_transport_for_data_channels) ==
>+               (media_transport != nullptr))
>+        << "configuration_.use_media_transport="
>+        << configuration_.use_media_transport
>+        << ", configuration_.use_media_transport_for_data_channels="
>+        << configuration_.use_media_transport_for_data_channels
>+        << ", (media_transport != nullptr)=" << (media_transport != nullptr);
>+    return media_transport;
>+  }
>+
>   sigslot::signal1<DataChannel*> SignalDataChannelCreated_;
> 
>   // Storing the factory as a scoped reference pointer ensures that the memory
>@@ -930,6 +983,11 @@ class PeerConnection : public PeerConnectionInternal,
> 
>   SignalingState signaling_state_ = kStable;
>   IceConnectionState ice_connection_state_ = kIceConnectionNew;
>+  PeerConnectionInterface::IceConnectionState
>+      standardized_ice_connection_state_ = kIceConnectionNew;
>+  PeerConnectionInterface::PeerConnectionState connection_state_ =
>+      PeerConnectionState::kNew;
>+
>   IceGatheringState ice_gathering_state_ = kIceGatheringNew;
>   PeerConnectionInterface::RTCConfiguration configuration_;
> 
>@@ -1010,6 +1068,33 @@ class PeerConnection : public PeerConnectionInternal,
>   sigslot::signal1<int> SignalSctpClosingProcedureStartedRemotely;
>   sigslot::signal1<int> SignalSctpClosingProcedureComplete;
> 
>+  // Whether this peer is the caller. Set when the local description is applied.
>+  absl::optional<bool> is_caller_ RTC_GUARDED_BY(signaling_thread());
>+
>+  // Content name (MID) for media transport data channels in SDP.
>+  absl::optional<std::string> media_transport_data_mid_;
>+
>+  // Media transport used for data channels.  Thread-safe.
>+  MediaTransportInterface* media_transport_ = nullptr;
>+
>+  // Cached value of whether the media transport is ready to send.
>+  bool media_transport_ready_to_send_data_ RTC_GUARDED_BY(signaling_thread()) =
>+      false;
>+
>+  // Used to invoke media transport signals on the signaling thread.
>+  std::unique_ptr<rtc::AsyncInvoker> media_transport_invoker_;
>+
>+  // Identical to the signals for SCTP, but from media transport:
>+  sigslot::signal1<bool> SignalMediaTransportWritable_s
>+      RTC_GUARDED_BY(signaling_thread());
>+  sigslot::signal2<const cricket::ReceiveDataParams&,
>+                   const rtc::CopyOnWriteBuffer&>
>+      SignalMediaTransportReceivedData_s RTC_GUARDED_BY(signaling_thread());
>+  sigslot::signal1<int> SignalMediaTransportChannelClosing_s
>+      RTC_GUARDED_BY(signaling_thread());
>+  sigslot::signal1<int> SignalMediaTransportChannelClosed_s
>+      RTC_GUARDED_BY(signaling_thread());
>+
>   std::unique_ptr<SessionDescriptionInterface> current_local_description_;
>   std::unique_ptr<SessionDescriptionInterface> pending_local_description_;
>   std::unique_ptr<SessionDescriptionInterface> current_remote_description_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_bundle_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_bundle_unittest.cc
>index 907669b5ad30c4866aeaac83b3c626b4076f2c42..6c372b4b8982b8289921c75c07a2a69bfa2e6df4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_bundle_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_bundle_unittest.cc
>@@ -10,6 +10,7 @@
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/create_peerconnection_factory.h"
> #include "api/peerconnectionproxy.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_crypto_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_crypto_unittest.cc
>index 1ccfe5560f09509f5143c60c4f87f484a4b5b033..6f7e23b94f5ad350534a3ed71432ad5ed1e5cee1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_crypto_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_crypto_unittest.cc
>@@ -10,6 +10,7 @@
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/create_peerconnection_factory.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "p2p/base/fakeportallocator.h"
>@@ -76,6 +77,7 @@ class PeerConnectionCryptoBaseTest : public ::testing::Test {
>       return nullptr;
>     }
> 
>+    observer->SetPeerConnectionInterface(pc.get());
>     return absl::make_unique<PeerConnectionWrapper>(pc_factory_, pc,
>                                                     std::move(observer));
>   }
>@@ -278,13 +280,35 @@ TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenEncryptionDisabled) {
>                              answer->description()));
> }
> 
>+// CryptoOptions has been promoted to RTCConfiguration. As such if it is ever
>+// set in the configuration it should overrite the settings set in the factory.
>+TEST_P(PeerConnectionCryptoTest, RTCConfigurationCryptoOptionOverridesFactory) {
>+  PeerConnectionFactoryInterface::Options options;
>+  options.crypto_options.srtp.enable_gcm_crypto_suites = true;
>+  pc_factory_->SetOptions(options);
>+
>+  RTCConfiguration config;
>+  config.enable_dtls_srtp.emplace(false);
>+  CryptoOptions crypto_options;
>+  crypto_options.srtp.enable_gcm_crypto_suites = false;
>+  config.crypto_options = crypto_options;
>+  auto caller = CreatePeerConnectionWithAudioVideo(config);
>+
>+  auto offer = caller->CreateOffer();
>+  ASSERT_TRUE(offer);
>+
>+  ASSERT_FALSE(offer->description()->contents().empty());
>+  // This should exist if GCM is enabled see CorrectCryptoInOfferWithSdesAndGcm
>+  EXPECT_FALSE(SdpContentsAll(HaveSdesGcmCryptos(3), offer->description()));
>+}
>+
> // When DTLS is disabled and GCM cipher suites are enabled, the SDP offer/answer
> // should have the correct ciphers in the SDES crypto options.
> // With GCM cipher suites enabled, there will be 3 cryptos in the offer and 1
> // in the answer.
> TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWithSdesAndGcm) {
>   PeerConnectionFactoryInterface::Options options;
>-  options.crypto_options.enable_gcm_crypto_suites = true;
>+  options.crypto_options.srtp.enable_gcm_crypto_suites = true;
>   pc_factory_->SetOptions(options);
> 
>   RTCConfiguration config;
>@@ -297,9 +321,10 @@ TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWithSdesAndGcm) {
>   ASSERT_FALSE(offer->description()->contents().empty());
>   EXPECT_TRUE(SdpContentsAll(HaveSdesGcmCryptos(3), offer->description()));
> }
>+
> TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWithSdesAndGcm) {
>   PeerConnectionFactoryInterface::Options options;
>-  options.crypto_options.enable_gcm_crypto_suites = true;
>+  options.crypto_options.srtp.enable_gcm_crypto_suites = true;
>   pc_factory_->SetOptions(options);
> 
>   RTCConfiguration config;
>@@ -317,7 +342,7 @@ TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWithSdesAndGcm) {
> 
> TEST_P(PeerConnectionCryptoTest, CanSetSdesGcmRemoteOfferAndLocalAnswer) {
>   PeerConnectionFactoryInterface::Options options;
>-  options.crypto_options.enable_gcm_crypto_suites = true;
>+  options.crypto_options.srtp.enable_gcm_crypto_suites = true;
>   pc_factory_->SetOptions(options);
> 
>   RTCConfiguration config;
>@@ -703,8 +728,8 @@ TEST_P(PeerConnectionCryptoTest, SessionErrorIfFingerprintInvalid) {
>       invalid_answer->description()->GetTransportInfoByName(
>           audio_content->name);
>   ASSERT_TRUE(audio_transport_info);
>-  audio_transport_info->description.identity_fingerprint.reset(
>-      rtc::SSLFingerprint::CreateFromCertificate(other_certificate));
>+  audio_transport_info->description.identity_fingerprint =
>+      rtc::SSLFingerprint::CreateFromCertificate(*other_certificate);
> 
>   // Set the invalid answer and expect a fingerprint error.
>   std::string error;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_datachannel_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_datachannel_unittest.cc
>index 033a609f98adab2391119ce96dfc715eab6651e1..cfb5dde22af3cbe0487d46e9cf9c932351c04879 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_datachannel_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_datachannel_unittest.cc
>@@ -11,6 +11,7 @@
> #include <tuple>
> 
> #include "api/peerconnectionproxy.h"
>+#include "api/test/fake_media_transport.h"
> #include "media/base/fakemediaengine.h"
> #include "pc/mediasession.h"
> #include "pc/peerconnection.h"
>@@ -31,17 +32,39 @@ using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
> using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
> using ::testing::Values;
> 
>+namespace {
>+
>+PeerConnectionFactoryDependencies CreatePeerConnectionFactoryDependencies(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    std::unique_ptr<cricket::MediaEngineInterface> media_engine,
>+    std::unique_ptr<CallFactoryInterface> call_factory,
>+    std::unique_ptr<MediaTransportFactory> media_transport_factory) {
>+  PeerConnectionFactoryDependencies deps;
>+  deps.network_thread = network_thread;
>+  deps.worker_thread = worker_thread;
>+  deps.signaling_thread = signaling_thread;
>+  deps.media_engine = std::move(media_engine);
>+  deps.call_factory = std::move(call_factory);
>+  deps.media_transport_factory = std::move(media_transport_factory);
>+  return deps;
>+}
>+
>+}  // namespace
>+
> class PeerConnectionFactoryForDataChannelTest
>     : public rtc::RefCountedObject<PeerConnectionFactory> {
>  public:
>   PeerConnectionFactoryForDataChannelTest()
>       : rtc::RefCountedObject<PeerConnectionFactory>(
>-            rtc::Thread::Current(),
>-            rtc::Thread::Current(),
>-            rtc::Thread::Current(),
>-            absl::make_unique<cricket::FakeMediaEngine>(),
>-            CreateCallFactory(),
>-            nullptr) {}
>+            CreatePeerConnectionFactoryDependencies(
>+                rtc::Thread::Current(),
>+                rtc::Thread::Current(),
>+                rtc::Thread::Current(),
>+                absl::make_unique<cricket::FakeMediaEngine>(),
>+                CreateCallFactory(),
>+                absl::make_unique<FakeMediaTransportFactory>())) {}
> 
>   std::unique_ptr<cricket::SctpTransportInternalFactory>
>   CreateSctpTransportInternalFactory() {
>@@ -123,6 +146,7 @@ class PeerConnectionDataChannelBaseTest : public ::testing::Test {
>       return nullptr;
>     }
> 
>+    observer->SetPeerConnectionInterface(pc.get());
>     auto wrapper = absl::make_unique<PeerConnectionWrapperForDataChannelTest>(
>         pc_factory, pc, std::move(observer));
>     RTC_DCHECK(pc_factory->last_fake_sctp_transport_factory_);
>@@ -323,6 +347,52 @@ TEST_P(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) {
>   EXPECT_EQ(kNewRecvPort, callee_transport->local_port());
> }
> 
>+TEST_P(PeerConnectionDataChannelTest,
>+       NoSctpTransportCreatedIfMediaTransportDataChannelsEnabled) {
>+  RTCConfiguration config;
>+  config.use_media_transport_for_data_channels = true;
>+  config.enable_dtls_srtp = false;  // SDES is required to use media transport.
>+  auto caller = CreatePeerConnectionWithDataChannel(config);
>+
>+  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
>+  EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport());
>+}
>+
>+TEST_P(PeerConnectionDataChannelTest,
>+       MediaTransportDataChannelCreatedEvenIfSctpAvailable) {
>+  RTCConfiguration config;
>+  config.use_media_transport_for_data_channels = true;
>+  config.enable_dtls_srtp = false;  // SDES is required to use media transport.
>+  PeerConnectionFactoryInterface::Options options;
>+  options.disable_sctp_data_channels = false;
>+  auto caller = CreatePeerConnectionWithDataChannel(config, options);
>+
>+  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
>+  EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport());
>+}
>+
>+TEST_P(PeerConnectionDataChannelTest,
>+       CannotEnableBothMediaTransportAndRtpDataChannels) {
>+  RTCConfiguration config;
>+  config.enable_rtp_data_channel = true;
>+  config.use_media_transport_for_data_channels = true;
>+  config.enable_dtls_srtp = false;  // SDES is required to use media transport.
>+  EXPECT_EQ(CreatePeerConnection(config), nullptr);
>+}
>+
>+TEST_P(PeerConnectionDataChannelTest,
>+       MediaTransportDataChannelFailsWithoutSdes) {
>+  RTCConfiguration config;
>+  config.use_media_transport_for_data_channels = true;
>+  config.enable_dtls_srtp = true;  // Disables SDES for data sections.
>+  auto caller = CreatePeerConnectionWithDataChannel(config);
>+
>+  std::string error;
>+  ASSERT_FALSE(caller->SetLocalDescription(caller->CreateOffer(), &error));
>+  EXPECT_EQ(error,
>+            "Failed to set local offer sdp: Failed to create data channel.");
>+}
>+
> INSTANTIATE_TEST_CASE_P(PeerConnectionDataChannelTest,
>                         PeerConnectionDataChannelTest,
>                         Values(SdpSemantics::kPlanB,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_histogram_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_histogram_unittest.cc
>index b5c07f49ec411b0680ca13bdd750ba86e044795e..497e33a688c65db49107768a2c9b2c9977c096da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_histogram_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_histogram_unittest.cc
>@@ -251,6 +251,7 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test {
>       return nullptr;
>     }
> 
>+    observer->SetPeerConnectionInterface(pc.get());
>     auto wrapper =
>         absl::make_unique<PeerConnectionWrapperForUsageHistogramTest>(
>             pc_factory, pc, std::move(observer));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_ice_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_ice_unittest.cc
>index 965956c42a3241b1180e5db8735814ede36f5375..928549f3800f2564fe9b674c2d4c6f805ef0aa92 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_ice_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_ice_unittest.cc
>@@ -21,7 +21,9 @@
> #include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/create_peerconnection_factory.h"
> #include "api/peerconnectionproxy.h"
>+#include "api/umametrics.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "pc/test/fakeaudiocapturemodule.h"
>@@ -77,6 +79,9 @@ class PeerConnectionWrapperForIceTest : public PeerConnectionWrapper {
> 
>   void set_network(rtc::FakeNetworkManager* network) { network_ = network; }
> 
>+  // The port allocator used by this PC.
>+  cricket::PortAllocator* port_allocator_;
>+
>  private:
>   rtc::FakeNetworkManager* network_;
> };
>@@ -114,15 +119,18 @@ class PeerConnectionIceBaseTest : public ::testing::Test {
>     RTCConfiguration modified_config = config;
>     modified_config.sdp_semantics = sdp_semantics_;
>     auto observer = absl::make_unique<MockPeerConnectionObserver>();
>+    auto port_allocator_copy = port_allocator.get();
>     auto pc = pc_factory_->CreatePeerConnection(
>         modified_config, std::move(port_allocator), nullptr, observer.get());
>     if (!pc) {
>       return nullptr;
>     }
> 
>+    observer->SetPeerConnectionInterface(pc.get());
>     auto wrapper = absl::make_unique<PeerConnectionWrapperForIceTest>(
>         pc_factory_, pc, std::move(observer));
>     wrapper->set_network(fake_network);
>+    wrapper->port_allocator_ = port_allocator_copy;
>     return wrapper;
>   }
> 
>@@ -205,7 +213,12 @@ class PeerConnectionIceBaseTest : public ::testing::Test {
>     PeerConnection* pc = static_cast<PeerConnection*>(pc_proxy->internal());
>     for (auto transceiver : pc->GetTransceiversInternal()) {
>       if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>-        cricket::BaseChannel* channel = transceiver->internal()->channel();
>+        // TODO(amithi): This test seems to be using a method that should not
>+        // be public |rtp_packet_transport|. Because the test is not mocking
>+        // the channels or transceiver, workaround will be to |static_cast|
>+        // the channel until the method is rewritten.
>+        cricket::BaseChannel* channel = static_cast<cricket::BaseChannel*>(
>+            transceiver->internal()->channel());
>         if (channel) {
>           auto dtls_transport = static_cast<cricket::DtlsTransportInternal*>(
>               channel->rtp_packet_transport());
>@@ -1007,4 +1020,41 @@ TEST_F(PeerConnectionIceConfigTest, SetStunCandidateKeepaliveInterval) {
>   EXPECT_EQ(actual_stun_keepalive_interval.value_or(-1), 321);
> }
> 
>+TEST_P(PeerConnectionIceTest, IceCredentialsCreateOffer) {
>+  RTCConfiguration config;
>+  config.ice_candidate_pool_size = 1;
>+  auto pc = CreatePeerConnectionWithAudioVideo(config);
>+  ASSERT_NE(pc->port_allocator_, nullptr);
>+  auto offer = pc->CreateOffer();
>+  auto credentials = pc->port_allocator_->GetPooledIceCredentials();
>+  ASSERT_EQ(1u, credentials.size());
>+
>+  auto* desc = offer->description();
>+  for (const auto& content : desc->contents()) {
>+    auto* transport_info = desc->GetTransportInfoByName(content.name);
>+    EXPECT_EQ(transport_info->description.ice_ufrag, credentials[0].ufrag);
>+    EXPECT_EQ(transport_info->description.ice_pwd, credentials[0].pwd);
>+  }
>+}
>+
>+TEST_P(PeerConnectionIceTest, IceCredentialsCreateAnswer) {
>+  RTCConfiguration config;
>+  config.ice_candidate_pool_size = 1;
>+  auto pc = CreatePeerConnectionWithAudioVideo(config);
>+  ASSERT_NE(pc->port_allocator_, nullptr);
>+  auto offer = pc->CreateOffer();
>+  ASSERT_TRUE(pc->SetRemoteDescription(std::move(offer)));
>+  auto answer = pc->CreateAnswer();
>+
>+  auto credentials = pc->port_allocator_->GetPooledIceCredentials();
>+  ASSERT_EQ(1u, credentials.size());
>+
>+  auto* desc = answer->description();
>+  for (const auto& content : desc->contents()) {
>+    auto* transport_info = desc->GetTransportInfoByName(content.name);
>+    EXPECT_EQ(transport_info->description.ice_ufrag, credentials[0].ufrag);
>+    EXPECT_EQ(transport_info->description.ice_pwd, credentials[0].pwd);
>+  }
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_integrationtest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_integrationtest.cc
>index 28e0110355037ed4d7ec916326ae68f171ce13d3..a7f7aad0f54b31caed3ea7fd45f63f47c94a9ee0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_integrationtest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_integrationtest.cc
>@@ -28,6 +28,8 @@
> #include "api/peerconnectioninterface.h"
> #include "api/peerconnectionproxy.h"
> #include "api/rtpreceiverinterface.h"
>+#include "api/test/loopback_media_transport.h"
>+#include "api/umametrics.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "api/video_codecs/sdp_video_format.h"
>@@ -250,7 +252,8 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>     webrtc::PeerConnectionDependencies dependencies(nullptr);
>     dependencies.cert_generator = std::move(cert_generator);
>     if (!client->Init(nullptr, nullptr, std::move(dependencies), network_thread,
>-                      worker_thread, nullptr)) {
>+                      worker_thread, nullptr,
>+                      /*media_transport_factory=*/nullptr)) {
>       delete client;
>       return nullptr;
>     }
>@@ -317,6 +320,12 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>     ice_connection_state_history_.clear();
>   }
> 
>+  // Every PeerConnection state in order that has been seen by the observer.
>+  std::vector<PeerConnectionInterface::PeerConnectionState>
>+  peer_connection_state_history() const {
>+    return peer_connection_state_history_;
>+  }
>+
>   // Every ICE gathering state in order that has been seen by the observer.
>   std::vector<PeerConnectionInterface::IceGatheringState>
>   ice_gathering_state_history() const {
>@@ -581,12 +590,14 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>   explicit PeerConnectionWrapper(const std::string& debug_name)
>       : debug_name_(debug_name) {}
> 
>-  bool Init(const PeerConnectionFactory::Options* options,
>-            const PeerConnectionInterface::RTCConfiguration* config,
>-            webrtc::PeerConnectionDependencies dependencies,
>-            rtc::Thread* network_thread,
>-            rtc::Thread* worker_thread,
>-            std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory) {
>+  bool Init(
>+      const PeerConnectionFactory::Options* options,
>+      const PeerConnectionInterface::RTCConfiguration* config,
>+      webrtc::PeerConnectionDependencies dependencies,
>+      rtc::Thread* network_thread,
>+      rtc::Thread* worker_thread,
>+      std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
>+      std::unique_ptr<webrtc::MediaTransportFactory> media_transport_factory) {
>     // There's an error in this test code if Init ends up being called twice.
>     RTC_DCHECK(!peer_connection_);
>     RTC_DCHECK(!peer_connection_factory_);
>@@ -624,6 +635,10 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>       pc_factory_dependencies.event_log_factory =
>           webrtc::CreateRtcEventLogFactory();
>     }
>+    if (media_transport_factory) {
>+      pc_factory_dependencies.media_transport_factory =
>+          std::move(media_transport_factory);
>+    }
>     peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(
>         std::move(pc_factory_dependencies));
> 
>@@ -913,6 +928,11 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>     EXPECT_EQ(pc()->ice_connection_state(), new_state);
>     ice_connection_state_history_.push_back(new_state);
>   }
>+  void OnConnectionChange(
>+      webrtc::PeerConnectionInterface::PeerConnectionState new_state) override {
>+    peer_connection_state_history_.push_back(new_state);
>+  }
>+
>   void OnIceGatheringChange(
>       webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
>     EXPECT_EQ(pc()->ice_gathering_state(), new_state);
>@@ -1009,6 +1029,8 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
> 
>   std::vector<PeerConnectionInterface::IceConnectionState>
>       ice_connection_state_history_;
>+  std::vector<PeerConnectionInterface::PeerConnectionState>
>+      peer_connection_state_history_;
>   std::vector<PeerConnectionInterface::IceGatheringState>
>       ice_gathering_state_history_;
> 
>@@ -1142,7 +1164,8 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
>         ss_(new rtc::VirtualSocketServer()),
>         fss_(new rtc::FirewallSocketServer(ss_.get())),
>         network_thread_(new rtc::Thread(fss_.get())),
>-        worker_thread_(rtc::Thread::Create()) {
>+        worker_thread_(rtc::Thread::Create()),
>+        loopback_media_transports_(network_thread_.get()) {
>     network_thread_->SetName("PCNetworkThread", this);
>     worker_thread_->SetName("PCWorkerThread", this);
>     RTC_CHECK(network_thread_->Start());
>@@ -1198,7 +1221,8 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
>       const PeerConnectionFactory::Options* options,
>       const RTCConfiguration* config,
>       webrtc::PeerConnectionDependencies dependencies,
>-      std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory) {
>+      std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
>+      std::unique_ptr<webrtc::MediaTransportFactory> media_transport_factory) {
>     RTCConfiguration modified_config;
>     if (config) {
>       modified_config = *config;
>@@ -1213,7 +1237,8 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
> 
>     if (!client->Init(options, &modified_config, std::move(dependencies),
>                       network_thread_.get(), worker_thread_.get(),
>-                      std::move(event_log_factory))) {
>+                      std::move(event_log_factory),
>+                      std::move(media_transport_factory))) {
>       return nullptr;
>     }
>     return client;
>@@ -1229,7 +1254,8 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
>         new webrtc::FakeRtcEventLogFactory(rtc::Thread::Current()));
>     return CreatePeerConnectionWrapper(debug_name, options, config,
>                                        std::move(dependencies),
>-                                       std::move(event_log_factory));
>+                                       std::move(event_log_factory),
>+                                       /*media_transport_factory=*/nullptr);
>   }
> 
>   bool CreatePeerConnectionWrappers() {
>@@ -1250,11 +1276,11 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
>     sdp_semantics_ = caller_semantics;
>     caller_ = CreatePeerConnectionWrapper(
>         "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
>-        nullptr);
>+        nullptr, /*media_transport_factory=*/nullptr);
>     sdp_semantics_ = callee_semantics;
>     callee_ = CreatePeerConnectionWrapper(
>         "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
>-        nullptr);
>+        nullptr, /*media_transport_factory=*/nullptr);
>     sdp_semantics_ = original_semantics;
>     return caller_ && callee_;
>   }
>@@ -1264,10 +1290,28 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
>       const PeerConnectionInterface::RTCConfiguration& callee_config) {
>     caller_ = CreatePeerConnectionWrapper(
>         "Caller", nullptr, &caller_config,
>-        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr,
>+        /*media_transport_factory=*/nullptr);
>     callee_ = CreatePeerConnectionWrapper(
>         "Callee", nullptr, &callee_config,
>-        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr,
>+        /*media_transport_factory=*/nullptr);
>+    return caller_ && callee_;
>+  }
>+
>+  bool CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
>+      const PeerConnectionInterface::RTCConfiguration& caller_config,
>+      const PeerConnectionInterface::RTCConfiguration& callee_config,
>+      std::unique_ptr<webrtc::MediaTransportFactory> caller_factory,
>+      std::unique_ptr<webrtc::MediaTransportFactory> callee_factory) {
>+    caller_ =
>+        CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
>+                                    webrtc::PeerConnectionDependencies(nullptr),
>+                                    nullptr, std::move(caller_factory));
>+    callee_ =
>+        CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
>+                                    webrtc::PeerConnectionDependencies(nullptr),
>+                                    nullptr, std::move(callee_factory));
>     return caller_ && callee_;
>   }
> 
>@@ -1278,10 +1322,12 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
>       webrtc::PeerConnectionDependencies callee_dependencies) {
>     caller_ =
>         CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
>-                                    std::move(caller_dependencies), nullptr);
>+                                    std::move(caller_dependencies), nullptr,
>+                                    /*media_transport_factory=*/nullptr);
>     callee_ =
>         CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
>-                                    std::move(callee_dependencies), nullptr);
>+                                    std::move(callee_dependencies), nullptr,
>+                                    /*media_transport_factory=*/nullptr);
>     return caller_ && callee_;
>   }
> 
>@@ -1290,10 +1336,12 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
>       const PeerConnectionFactory::Options& callee_options) {
>     caller_ = CreatePeerConnectionWrapper(
>         "Caller", &caller_options, nullptr,
>-        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr,
>+        /*media_transport_factory=*/nullptr);
>     callee_ = CreatePeerConnectionWrapper(
>         "Callee", &callee_options, nullptr,
>-        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr,
>+        /*media_transport_factory=*/nullptr);
>     return caller_ && callee_;
>   }
> 
>@@ -1317,7 +1365,8 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
>     webrtc::PeerConnectionDependencies dependencies(nullptr);
>     dependencies.cert_generator = std::move(cert_generator);
>     return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr,
>-                                       std::move(dependencies), nullptr);
>+                                       std::move(dependencies), nullptr,
>+                                       /*media_transport_factory=*/nullptr);
>   }
> 
>   cricket::TestTurnServer* CreateTurnServer(
>@@ -1405,6 +1454,10 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
> 
>   rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); }
> 
>+  webrtc::MediaTransportPair* loopback_media_transports() {
>+    return &loopback_media_transports_;
>+  }
>+
>   PeerConnectionWrapper* caller() { return caller_.get(); }
> 
>   // Set the |caller_| to the |wrapper| passed in and return the
>@@ -1557,9 +1610,11 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
>                                          bool remote_gcm_enabled,
>                                          int expected_cipher_suite) {
>     PeerConnectionFactory::Options caller_options;
>-    caller_options.crypto_options.enable_gcm_crypto_suites = local_gcm_enabled;
>+    caller_options.crypto_options.srtp.enable_gcm_crypto_suites =
>+        local_gcm_enabled;
>     PeerConnectionFactory::Options callee_options;
>-    callee_options.crypto_options.enable_gcm_crypto_suites = remote_gcm_enabled;
>+    callee_options.crypto_options.srtp.enable_gcm_crypto_suites =
>+        remote_gcm_enabled;
>     TestNegotiatedCipherSuite(caller_options, callee_options,
>                               expected_cipher_suite);
>   }
>@@ -1581,6 +1636,7 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
>   // on the network thread.
>   std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_;
>   std::vector<std::unique_ptr<cricket::TestTurnCustomizer>> turn_customizers_;
>+  webrtc::MediaTransportPair loopback_media_transports_;
>   std::unique_ptr<PeerConnectionWrapper> caller_;
>   std::unique_ptr<PeerConnectionWrapper> callee_;
> };
>@@ -1805,26 +1861,26 @@ TEST_P(PeerConnectionIntegrationTest,
> 
>   auto caller_remote_cert = GetRemoteAudioSSLCertificate(caller());
>   ASSERT_TRUE(caller_remote_cert);
>-  EXPECT_EQ(callee_cert->ssl_certificate().ToPEMString(),
>+  EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
>             caller_remote_cert->ToPEMString());
> 
>   auto callee_remote_cert = GetRemoteAudioSSLCertificate(callee());
>   ASSERT_TRUE(callee_remote_cert);
>-  EXPECT_EQ(caller_cert->ssl_certificate().ToPEMString(),
>+  EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
>             callee_remote_cert->ToPEMString());
> 
>   auto caller_remote_cert_chain = GetRemoteAudioSSLCertChain(caller());
>   ASSERT_TRUE(caller_remote_cert_chain);
>   ASSERT_EQ(1U, caller_remote_cert_chain->GetSize());
>   auto remote_cert = &caller_remote_cert_chain->Get(0);
>-  EXPECT_EQ(callee_cert->ssl_certificate().ToPEMString(),
>+  EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(),
>             remote_cert->ToPEMString());
> 
>   auto callee_remote_cert_chain = GetRemoteAudioSSLCertChain(callee());
>   ASSERT_TRUE(callee_remote_cert_chain);
>   ASSERT_EQ(1U, callee_remote_cert_chain->GetSize());
>   remote_cert = &callee_remote_cert_chain->Get(0);
>-  EXPECT_EQ(caller_cert->ssl_certificate().ToPEMString(),
>+  EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(),
>             remote_cert->ToPEMString());
> }
> 
>@@ -2842,9 +2898,10 @@ TEST_P(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) {
> TEST_P(PeerConnectionIntegrationTest,
>        Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported) {
>   PeerConnectionFactory::Options caller_options;
>-  caller_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
>+  caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
>   PeerConnectionFactory::Options callee_options;
>-  callee_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = false;
>+  callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
>+      false;
>   int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
>   TestNegotiatedCipherSuite(caller_options, callee_options,
>                             expected_cipher_suite);
>@@ -2853,9 +2910,10 @@ TEST_P(PeerConnectionIntegrationTest,
> TEST_P(PeerConnectionIntegrationTest,
>        Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported) {
>   PeerConnectionFactory::Options caller_options;
>-  caller_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = false;
>+  caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
>+      false;
>   PeerConnectionFactory::Options callee_options;
>-  callee_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
>+  callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
>   int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
>   TestNegotiatedCipherSuite(caller_options, callee_options,
>                             expected_cipher_suite);
>@@ -2863,9 +2921,9 @@ TEST_P(PeerConnectionIntegrationTest,
> 
> TEST_P(PeerConnectionIntegrationTest, Aes128Sha1_32_CipherUsedWhenSupported) {
>   PeerConnectionFactory::Options caller_options;
>-  caller_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
>+  caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
>   PeerConnectionFactory::Options callee_options;
>-  callee_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
>+  callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
>   int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_32;
>   TestNegotiatedCipherSuite(caller_options, callee_options,
>                             expected_cipher_suite);
>@@ -2915,7 +2973,7 @@ TEST_P(PeerConnectionIntegrationTest,
> // works with it.
> TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
>   PeerConnectionFactory::Options gcm_options;
>-  gcm_options.crypto_options.enable_gcm_crypto_suites = true;
>+  gcm_options.crypto_options.srtp.enable_gcm_crypto_suites = true;
>   ASSERT_TRUE(
>       CreatePeerConnectionWrappersWithOptions(gcm_options, gcm_options));
>   ConnectFakeSignaling();
>@@ -3329,6 +3387,146 @@ TEST_P(PeerConnectionIntegrationTest,
> 
> #endif  // HAVE_SCTP
> 
>+// This test sets up a call between two parties with a media transport data
>+// channel.
>+TEST_P(PeerConnectionIntegrationTest, MediaTransportDataChannelEndToEnd) {
>+  PeerConnectionInterface::RTCConfiguration rtc_config;
>+  rtc_config.use_media_transport_for_data_channels = true;
>+  rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
>+      rtc_config, rtc_config, loopback_media_transports()->first_factory(),
>+      loopback_media_transports()->second_factory()));
>+  ConnectFakeSignaling();
>+
>+  // Expect that data channel created on caller side will show up for callee as
>+  // well.
>+  caller()->CreateDataChannel();
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>+  // Ensure that the media transport is ready.
>+  loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
>+  loopback_media_transports()->FlushAsyncInvokes();
>+
>+  // Caller data channel should already exist (it created one). Callee data
>+  // channel may not exist yet, since negotiation happens in-band, not in SDP.
>+  ASSERT_NE(nullptr, caller()->data_channel());
>+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
>+
>+  // Ensure data can be sent in both directions.
>+  std::string data = "hello world";
>+  caller()->data_channel()->Send(DataBuffer(data));
>+  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
>+                 kDefaultTimeout);
>+  callee()->data_channel()->Send(DataBuffer(data));
>+  EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
>+                 kDefaultTimeout);
>+}
>+
>+// Ensure that when the callee closes a media transport data channel, the
>+// closing procedure results in the data channel being closed for the caller
>+// as well.
>+TEST_P(PeerConnectionIntegrationTest, MediaTransportDataChannelCalleeCloses) {
>+  PeerConnectionInterface::RTCConfiguration rtc_config;
>+  rtc_config.use_media_transport_for_data_channels = true;
>+  rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
>+      rtc_config, rtc_config, loopback_media_transports()->first_factory(),
>+      loopback_media_transports()->second_factory()));
>+  ConnectFakeSignaling();
>+
>+  // Create a data channel on the caller and signal it to the callee.
>+  caller()->CreateDataChannel();
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>+  // Ensure that the media transport is ready.
>+  loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
>+  loopback_media_transports()->FlushAsyncInvokes();
>+
>+  // Data channels exist and open on both ends of the connection.
>+  ASSERT_NE(nullptr, caller()->data_channel());
>+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
>+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
>+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
>+
>+  // Close the data channel on the callee side, and wait for it to reach the
>+  // "closed" state on both sides.
>+  callee()->data_channel()->Close();
>+  EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
>+}
>+
>+TEST_P(PeerConnectionIntegrationTest,
>+       MediaTransportDataChannelConfigSentToOtherSide) {
>+  PeerConnectionInterface::RTCConfiguration rtc_config;
>+  rtc_config.use_media_transport_for_data_channels = true;
>+  rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
>+      rtc_config, rtc_config, loopback_media_transports()->first_factory(),
>+      loopback_media_transports()->second_factory()));
>+  ConnectFakeSignaling();
>+
>+  // Create a data channel with a non-default configuration and signal it to the
>+  // callee.
>+  webrtc::DataChannelInit init;
>+  init.id = 53;
>+  init.maxRetransmits = 52;
>+  caller()->CreateDataChannel("data-channel", &init);
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>+  // Ensure that the media transport is ready.
>+  loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
>+  loopback_media_transports()->FlushAsyncInvokes();
>+
>+  // Ensure that the data channel exists on the callee with the correct
>+  // configuration.
>+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
>+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
>+  EXPECT_EQ(init.id, callee()->data_channel()->id());
>+  EXPECT_EQ("data-channel", callee()->data_channel()->label());
>+  EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits());
>+  EXPECT_FALSE(callee()->data_channel()->negotiated());
>+}
>+
>+TEST_P(PeerConnectionIntegrationTest, MediaTransportBidirectionalAudio) {
>+  PeerConnectionInterface::RTCConfiguration rtc_config;
>+  rtc_config.use_media_transport = true;
>+  rtc_config.enable_dtls_srtp = false;  // SDES is required for media transport.
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
>+      rtc_config, rtc_config, loopback_media_transports()->first_factory(),
>+      loopback_media_transports()->second_factory()));
>+  ConnectFakeSignaling();
>+
>+  caller()->AddAudioTrack();
>+  callee()->AddAudioTrack();
>+  // Start offer/answer exchange and wait for it to complete.
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>+  // Ensure that the media transport is ready.
>+  loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
>+  loopback_media_transports()->FlushAsyncInvokes();
>+
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudio();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+
>+  webrtc::MediaTransportPair::Stats first_stats =
>+      loopback_media_transports()->FirstStats();
>+  webrtc::MediaTransportPair::Stats second_stats =
>+      loopback_media_transports()->SecondStats();
>+
>+  EXPECT_GT(first_stats.received_audio_frames, 0);
>+  EXPECT_GE(second_stats.sent_audio_frames, first_stats.received_audio_frames);
>+
>+  EXPECT_GT(second_stats.received_audio_frames, 0);
>+  EXPECT_GE(first_stats.sent_audio_frames, second_stats.received_audio_frames);
>+}
>+
> // Test that the ICE connection and gathering states eventually reach
> // "complete".
> TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
>@@ -3557,6 +3755,17 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyIceStates) {
>               ElementsAre(PeerConnectionInterface::kIceConnectionChecking,
>                           PeerConnectionInterface::kIceConnectionConnected,
>                           PeerConnectionInterface::kIceConnectionCompleted));
>+  // After the ice transport transitions from checking to connected we revert
>+  // back to new as the standard requires, as at that point the DTLS transport
>+  // is in the "new" state while no transports are "connecting", "checking",
>+  // "failed" or disconnected. This is pretty unintuitive, and we might want to
>+  // amend the spec to handle this case more gracefully.
>+  EXPECT_THAT(
>+      caller()->peer_connection_state_history(),
>+      ElementsAre(PeerConnectionInterface::PeerConnectionState::kConnecting,
>+                  PeerConnectionInterface::PeerConnectionState::kNew,
>+                  PeerConnectionInterface::PeerConnectionState::kConnecting,
>+                  PeerConnectionInterface::PeerConnectionState::kConnected));
>   EXPECT_THAT(caller()->ice_gathering_state_history(),
>               ElementsAre(PeerConnectionInterface::kIceGatheringGathering,
>                           PeerConnectionInterface::kIceGatheringComplete));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc
>index 7de0a3ff14a4ab97a98cbc767a9398cc7fa888b1..fe878b05f78c231770068aeae2ccea61903524c5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc
>@@ -95,6 +95,7 @@ class PeerConnectionJsepTest : public ::testing::Test {
>       return nullptr;
>     }
> 
>+    observer->SetPeerConnectionInterface(pc.get());
>     return absl::make_unique<PeerConnectionWrapper>(pc_factory, pc,
>                                                     std::move(observer));
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc
>index 6f4fe1e3f489e9892d6492168b37ebdd5030e5bf..6af0c9887e6e2dfe4a1c07cad220ebd120e56486 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc
>@@ -15,6 +15,7 @@
> #include <tuple>
> 
> #include "api/call/callfactoryinterface.h"
>+#include "api/test/fake_media_transport.h"
> #include "logging/rtc_event_log/rtc_event_log_factory.h"
> #include "media/base/fakemediaengine.h"
> #include "p2p/base/fakeportallocator.h"
>@@ -71,13 +72,26 @@ class PeerConnectionMediaBaseTest : public ::testing::Test {
>     return CreatePeerConnection(RTCConfiguration());
>   }
> 
>+  // Creates PeerConnectionFactory and PeerConnection for given configuration.
>+  // Note that PeerConnectionFactory is created with MediaTransportFactory,
>+  // because some tests pass config.use_media_transport = true.
>   WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
>     auto media_engine = absl::make_unique<FakeMediaEngine>();
>     auto* media_engine_ptr = media_engine.get();
>-    auto pc_factory = CreateModularPeerConnectionFactory(
>-        rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
>-        std::move(media_engine), CreateCallFactory(),
>-        CreateRtcEventLogFactory());
>+
>+    PeerConnectionFactoryDependencies factory_dependencies;
>+
>+    factory_dependencies.network_thread = rtc::Thread::Current();
>+    factory_dependencies.worker_thread = rtc::Thread::Current();
>+    factory_dependencies.signaling_thread = rtc::Thread::Current();
>+    factory_dependencies.media_engine = std::move(media_engine);
>+    factory_dependencies.call_factory = CreateCallFactory();
>+    factory_dependencies.event_log_factory = CreateRtcEventLogFactory();
>+    factory_dependencies.media_transport_factory =
>+        absl::make_unique<FakeMediaTransportFactory>();
>+
>+    auto pc_factory =
>+        CreateModularPeerConnectionFactory(std::move(factory_dependencies));
> 
>     auto fake_port_allocator = absl::make_unique<cricket::FakePortAllocator>(
>         rtc::Thread::Current(), nullptr);
>@@ -91,12 +105,25 @@ class PeerConnectionMediaBaseTest : public ::testing::Test {
>       return nullptr;
>     }
> 
>+    observer->SetPeerConnectionInterface(pc.get());
>     auto wrapper = absl::make_unique<PeerConnectionWrapperForMediaTest>(
>         pc_factory, pc, std::move(observer));
>     wrapper->set_media_engine(media_engine_ptr);
>     return wrapper;
>   }
> 
>+  // Accepts the same arguments as CreatePeerConnection and adds default audio
>+  // track (but no video).
>+  template <typename... Args>
>+  WrapperPtr CreatePeerConnectionWithAudio(Args&&... args) {
>+    auto wrapper = CreatePeerConnection(std::forward<Args>(args)...);
>+    if (!wrapper) {
>+      return nullptr;
>+    }
>+    wrapper->AddAudioTrack("a");
>+    return wrapper;
>+  }
>+
>   // Accepts the same arguments as CreatePeerConnection and adds default audio
>   // and video tracks.
>   template <typename... Args>
>@@ -677,7 +704,7 @@ void AddComfortNoiseCodecsToSend(cricket::FakeMediaEngine* media_engine) {
>   const cricket::AudioCodec kComfortNoiseCodec8k(102, "CN", 8000, 0, 1);
>   const cricket::AudioCodec kComfortNoiseCodec16k(103, "CN", 16000, 0, 1);
> 
>-  auto codecs = media_engine->audio_send_codecs();
>+  auto codecs = media_engine->voice().send_codecs();
>   codecs.push_back(kComfortNoiseCodec8k);
>   codecs.push_back(kComfortNoiseCodec16k);
>   media_engine->SetAudioCodecs(codecs);
>@@ -1072,6 +1099,128 @@ TEST_P(PeerConnectionMediaTest,
>             audio_options.combined_audio_video_bwe);
> }
> 
>+TEST_P(PeerConnectionMediaTest, MediaTransportPropagatedToVoiceEngine) {
>+  RTCConfiguration config;
>+
>+  // Setup PeerConnection to use media transport.
>+  config.use_media_transport = true;
>+
>+  // Force SDES.
>+  config.enable_dtls_srtp = false;
>+
>+  auto caller = CreatePeerConnectionWithAudio(config);
>+  auto callee = CreatePeerConnectionWithAudio(config);
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+  auto answer = callee->CreateAnswer();
>+  ASSERT_TRUE(callee->SetLocalDescription(std::move(answer)));
>+
>+  auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
>+  auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
>+  ASSERT_TRUE(caller_voice);
>+  ASSERT_TRUE(callee_voice);
>+
>+  // Make sure media transport is propagated to voice channel.
>+  FakeMediaTransport* caller_voice_media_transport =
>+      static_cast<FakeMediaTransport*>(caller_voice->media_transport());
>+  FakeMediaTransport* callee_voice_media_transport =
>+      static_cast<FakeMediaTransport*>(callee_voice->media_transport());
>+  ASSERT_NE(nullptr, caller_voice_media_transport);
>+  ASSERT_NE(nullptr, callee_voice_media_transport);
>+
>+  // Make sure media transport is created with correct is_caller.
>+  EXPECT_TRUE(caller_voice_media_transport->is_caller());
>+  EXPECT_FALSE(callee_voice_media_transport->is_caller());
>+
>+  // TODO(sukhanov): Propagate media transport to video channel.
>+  // This test does NOT set up video channels, because currently it causes
>+  // us to create two media transports.
>+}
>+
>+TEST_P(PeerConnectionMediaTest, MediaTransportOnlyForDataChannels) {
>+  RTCConfiguration config;
>+
>+  // Setup PeerConnection to use media transport for data channels.
>+  config.use_media_transport_for_data_channels = true;
>+
>+  // Force SDES.
>+  config.enable_dtls_srtp = false;
>+
>+  auto caller = CreatePeerConnectionWithAudio(config);
>+  auto callee = CreatePeerConnectionWithAudio(config);
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+  ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
>+
>+  auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
>+  auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
>+  ASSERT_TRUE(caller_voice);
>+  ASSERT_TRUE(callee_voice);
>+
>+  // Make sure media transport is not propagated to voice channel.
>+  EXPECT_EQ(nullptr, caller_voice->media_transport());
>+  EXPECT_EQ(nullptr, callee_voice->media_transport());
>+}
>+
>+TEST_P(PeerConnectionMediaTest, MediaTransportForMediaAndDataChannels) {
>+  RTCConfiguration config;
>+
>+  // Setup PeerConnection to use media transport for both media and data
>+  // channels.
>+  config.use_media_transport = true;
>+  config.use_media_transport_for_data_channels = true;
>+
>+  // Force SDES.
>+  config.enable_dtls_srtp = false;
>+
>+  auto caller = CreatePeerConnectionWithAudio(config);
>+  auto callee = CreatePeerConnectionWithAudio(config);
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+  ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
>+
>+  auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
>+  auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
>+  ASSERT_TRUE(caller_voice);
>+  ASSERT_TRUE(callee_voice);
>+
>+  // Make sure media transport is propagated to voice channel.
>+  FakeMediaTransport* caller_voice_media_transport =
>+      static_cast<FakeMediaTransport*>(caller_voice->media_transport());
>+  FakeMediaTransport* callee_voice_media_transport =
>+      static_cast<FakeMediaTransport*>(callee_voice->media_transport());
>+  ASSERT_NE(nullptr, caller_voice_media_transport);
>+  ASSERT_NE(nullptr, callee_voice_media_transport);
>+
>+  // Make sure media transport is created with correct is_caller.
>+  EXPECT_TRUE(caller_voice_media_transport->is_caller());
>+  EXPECT_FALSE(callee_voice_media_transport->is_caller());
>+}
>+
>+TEST_P(PeerConnectionMediaTest, MediaTransportNotPropagatedToVoiceEngine) {
>+  auto caller = CreatePeerConnectionWithAudioVideo();
>+  auto callee = CreatePeerConnectionWithAudioVideo();
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+  auto answer = callee->CreateAnswer();
>+  ASSERT_TRUE(callee->SetLocalDescription(std::move(answer)));
>+
>+  auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
>+  auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
>+  ASSERT_TRUE(caller_voice);
>+  ASSERT_TRUE(callee_voice);
>+
>+  // Since we did not setup PeerConnection to use media transport, media
>+  // transport should not be created / propagated to the voice engine.
>+  ASSERT_EQ(nullptr, caller_voice->media_transport());
>+  ASSERT_EQ(nullptr, callee_voice->media_transport());
>+
>+  auto caller_video = caller->media_engine()->GetVideoChannel(0);
>+  auto callee_video = callee->media_engine()->GetVideoChannel(0);
>+  ASSERT_EQ(nullptr, caller_video->media_transport());
>+  ASSERT_EQ(nullptr, callee_video->media_transport());
>+}
>+
> INSTANTIATE_TEST_CASE_P(PeerConnectionMediaTest,
>                         PeerConnectionMediaTest,
>                         Values(SdpSemantics::kPlanB,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rampup_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rampup_tests.cc
>index 850212bf0dca4bd75282828869032ceda8c9330d..74f8018fb09b7e0da107f7834943227214e9c8c5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rampup_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rampup_tests.cc
>@@ -10,6 +10,7 @@
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/create_peerconnection_factory.h"
> #include "api/stats/rtcstats_objects.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc
>index bc3662cbbd43e341d1aae823c7d8168318ae282e..7098470baaeee98d0e74a6fc7074a87611f9596b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc
>@@ -14,9 +14,11 @@
> #include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/create_peerconnection_factory.h"
> #include "api/jsep.h"
> #include "api/mediastreaminterface.h"
> #include "api/peerconnectioninterface.h"
>+#include "api/umametrics.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "pc/mediasession.h"
>@@ -116,6 +118,8 @@ class PeerConnectionRtpBaseTest : public testing::Test {
>     auto observer = absl::make_unique<MockPeerConnectionObserver>();
>     auto pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr,
>                                                 observer.get());
>+    EXPECT_TRUE(pc.get());
>+    observer->SetPeerConnectionInterface(pc.get());
>     return absl::make_unique<PeerConnectionWrapper>(pc_factory_, pc,
>                                                     std::move(observer));
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_signaling_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_signaling_unittest.cc
>index 95f464b2ee249224042418d67b19cff9ba84c82b..f57f806b647f01339fbb0252cc1071ea3a242d4a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_signaling_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_signaling_unittest.cc
>@@ -15,6 +15,7 @@
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/create_peerconnection_factory.h"
> #include "api/peerconnectionproxy.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
>@@ -90,6 +91,7 @@ class PeerConnectionSignalingBaseTest : public ::testing::Test {
>       return nullptr;
>     }
> 
>+    observer->SetPeerConnectionInterface(pc.get());
>     return absl::make_unique<PeerConnectionWrapperForSignalingTest>(
>         pc_factory_, pc, std::move(observer));
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionendtoend_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionendtoend_unittest.cc
>index 60da2d242c1aa35dcf1cddef47fe6b323fede9ff..dee253b7c71870a728c22b98a0f67b5390cc94c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionendtoend_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionendtoend_unittest.cc
>@@ -11,6 +11,7 @@
> #include <memory>
> 
> #include "absl/memory/memory.h"
>+#include "absl/strings/match.h"
> #include "api/audio_codecs/L16/audio_decoder_L16.h"
> #include "api/audio_codecs/L16/audio_encoder_L16.h"
> #include "api/audio_codecs/audio_codec_pair_id.h"
>@@ -20,7 +21,6 @@
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/stringencode.h"
> #include "rtc_base/stringutils.h"
> 
> #ifdef WEBRTC_ANDROID
>@@ -31,6 +31,7 @@
> #include "pc/test/mockpeerconnectionobservers.h"
> #include "test/mock_audio_decoder.h"
> #include "test/mock_audio_decoder_factory.h"
>+#include "test/mock_audio_encoder_factory.h"
> 
> using testing::AtLeast;
> using testing::Invoke;
>@@ -300,7 +301,7 @@ CreateForwardingMockDecoderFactory(
> struct AudioEncoderUnicornSparklesRainbow {
>   using Config = webrtc::AudioEncoderL16::Config;
>   static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
>-    if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
>+    if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) {
>       const webrtc::SdpAudioFormat::Parameters expected_params = {
>           {"num_horns", "1"}};
>       EXPECT_EQ(expected_params, format.parameters);
>@@ -337,7 +338,7 @@ struct AudioEncoderUnicornSparklesRainbow {
> struct AudioDecoderUnicornSparklesRainbow {
>   using Config = webrtc::AudioDecoderL16::Config;
>   static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
>-    if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
>+    if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) {
>       const webrtc::SdpAudioFormat::Parameters expected_params = {
>           {"num_horns", "1"}};
>       EXPECT_EQ(expected_params, format.parameters);
>@@ -481,7 +482,7 @@ TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
> // Verifies that a DataChannel created before the negotiation can transition to
> // "OPEN" and transfer data.
> TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
>-  CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
>+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>   webrtc::DataChannelInit init;
>@@ -506,7 +507,7 @@ TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
> // Verifies that a DataChannel created after the negotiation can transition to
> // "OPEN" and transfer data.
> TEST_P(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
>-  CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
>+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>   webrtc::DataChannelInit init;
>@@ -538,7 +539,7 @@ TEST_P(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
> 
> // Verifies that a DataChannel created can transfer large messages.
> TEST_P(PeerConnectionEndToEndTest, CreateDataChannelLargeTransfer) {
>-  CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
>+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>   webrtc::DataChannelInit init;
>@@ -572,7 +573,7 @@ TEST_P(PeerConnectionEndToEndTest, CreateDataChannelLargeTransfer) {
> 
> // Verifies that DataChannel IDs are even/odd based on the DTLS roles.
> TEST_P(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
>-  CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
>+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>   webrtc::DataChannelInit init;
>@@ -600,7 +601,7 @@ TEST_P(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
> // there are multiple DataChannels.
> TEST_P(PeerConnectionEndToEndTest,
>        MessageTransferBetweenTwoPairsOfDataChannels) {
>-  CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
>+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>   webrtc::DataChannelInit init;
>@@ -640,7 +641,7 @@ TEST_P(PeerConnectionEndToEndTest,
> // channel, and the closed channel was incorrectly still assigned to the ID.
> TEST_P(PeerConnectionEndToEndTest,
>        DataChannelFromOpenWorksAfterPreviousChannelClosed) {
>-  CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
>+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>   webrtc::DataChannelInit init;
>@@ -673,7 +674,7 @@ TEST_P(PeerConnectionEndToEndTest,
> // closing before creating the second one.
> TEST_P(PeerConnectionEndToEndTest,
>        DataChannelFromOpenWorksWhilePreviousChannelClosing) {
>-  CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
>+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>   webrtc::DataChannelInit init;
>@@ -704,7 +705,7 @@ TEST_P(PeerConnectionEndToEndTest,
> // reference count), no memory access violation will occur.
> // See: https://code.google.com/p/chromium/issues/detail?id=565048
> TEST_P(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
>-  CreatePcs(webrtc::CreateBuiltinAudioEncoderFactory(),
>+  CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>   webrtc::DataChannelInit init;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>index 53f4da8516e46c8f588463ed9222f56edb202294..75261508b272d2e7e662a5f873f35af8bde538a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>@@ -15,6 +15,7 @@
> 
> #include "absl/memory/memory.h"
> #include "api/fec_controller.h"
>+#include "api/media_transport_interface.h"
> #include "api/mediaconstraintsinterface.h"
> #include "api/mediastreamproxy.h"
> #include "api/mediastreamtrackproxy.h"
>@@ -46,6 +47,7 @@
> #include "pc/videocapturertracksource.h"
> #include "pc/videotrack.h"
> #include "rtc_base/experiments/congestion_controller_experiment.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
>@@ -187,7 +189,9 @@ PeerConnectionFactory::PeerConnectionFactory(
>           std::move(dependencies.call_factory),
>           std::move(dependencies.event_log_factory),
>           std::move(dependencies.fec_controller_factory),
>-          std::move(dependencies.network_controller_factory)) {}
>+          std::move(dependencies.network_controller_factory)) {
>+  media_transport_factory_ = std::move(dependencies.media_transport_factory);
>+}
> 
> PeerConnectionFactory::~PeerConnectionFactory() {
>   RTC_DCHECK(signaling_thread_->IsCurrent());
>@@ -356,9 +360,12 @@ PeerConnectionFactory::CreatePeerConnection(
>                                                         network_thread_);
>   }
>   if (!dependencies.allocator) {
>-    dependencies.allocator.reset(new cricket::BasicPortAllocator(
>-        default_network_manager_.get(), default_socket_factory_.get(),
>-        configuration.turn_customizer));
>+    network_thread_->Invoke<void>(RTC_FROM_HERE, [this, &configuration,
>+                                                  &dependencies]() {
>+      dependencies.allocator = absl::make_unique<cricket::BasicPortAllocator>(
>+          default_network_manager_.get(), default_socket_factory_.get(),
>+          configuration.turn_customizer);
>+    });
>   }
> 
>   // TODO(zstein): Once chromium injects its own AsyncResolverFactory, set
>@@ -442,7 +449,10 @@ rtc::Thread* PeerConnectionFactory::network_thread() {
> 
> std::unique_ptr<RtcEventLog> PeerConnectionFactory::CreateRtcEventLog_w() {
>   RTC_DCHECK_RUN_ON(worker_thread_);
>-  const auto encoding_type = RtcEventLog::EncodingType::Legacy;
>+
>+  auto encoding_type = RtcEventLog::EncodingType::Legacy;
>+  if (field_trial::IsEnabled("WebRTC-RtcEventLogNewFormat"))
>+    encoding_type = RtcEventLog::EncodingType::NewFormat;
>   return event_log_factory_
>              ? event_log_factory_->CreateRtcEventLog(encoding_type)
>              : absl::make_unique<RtcEventLogNullImpl>();
>@@ -460,7 +470,8 @@ std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w(
>   if (!channel_manager_->media_engine() || !call_factory_) {
>     return nullptr;
>   }
>-  call_config.audio_state = channel_manager_->media_engine()->GetAudioState();
>+  call_config.audio_state =
>+      channel_manager_->media_engine()->voice().GetAudioState();
>   call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
>   call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
>   call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.h
>index f36a4e0c044bc445bb68ecc988477491f37052b1..1c1ea94fe28a73aaccb62471eae79bc0d5437b69 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.h
>@@ -15,6 +15,7 @@
> #include <memory>
> #include <string>
> 
>+#include "api/media_transport_interface.h"
> #include "api/mediastreaminterface.h"
> #include "api/peerconnectioninterface.h"
> #include "media/sctp/sctptransportinternal.h"
>@@ -96,6 +97,10 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
>   virtual rtc::Thread* network_thread();
>   const Options& options() const { return options_; }
> 
>+  MediaTransportFactory* media_transport_factory() {
>+    return media_transport_factory_.get();
>+  }
>+
>  protected:
>   PeerConnectionFactory(
>       rtc::Thread* network_thread,
>@@ -148,6 +153,7 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
>       injected_network_controller_factory_;
>   std::unique_ptr<NetworkControllerFactoryInterface>
>       bbr_network_controller_factory_;
>+  std::unique_ptr<MediaTransportFactory> media_transport_factory_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory_unittest.cc
>index 84c828f0961042a5d51cbcf12f6081b76462e93a..24441432d7a38b88409713406aa39fa13a5520af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory_unittest.cc
>@@ -15,6 +15,7 @@
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/create_peerconnection_factory.h"
> #include "api/mediastreaminterface.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectioninterface_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectioninterface_unittest.cc
>index 719fdbdd7f41cc80d8bb3b808219d69d3a0a5536..dd5d0d16a91986b0ec45f24b7df030e11ea40b30 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectioninterface_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectioninterface_unittest.cc
>@@ -17,6 +17,7 @@
> #include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/create_peerconnection_factory.h"
> #include "api/jsepsessiondescription.h"
> #include "api/mediastreaminterface.h"
> #include "api/peerconnectioninterface.h"
>@@ -1397,6 +1398,8 @@ TEST_P(PeerConnectionInterfaceTest,
>   rtc::scoped_refptr<PeerConnectionInterface> pc(
>       pc_factory->CreatePeerConnection(config, std::move(port_allocator),
>                                        nullptr, &observer_));
>+  EXPECT_TRUE(pc.get());
>+  observer_.SetPeerConnectionInterface(pc.get());
> 
>   // Now validate that the config fields set above were applied to the
>   // PortAllocator, as flags or otherwise.
>@@ -1424,15 +1427,22 @@ TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) {
> // Check that GetConfiguration returns the last configuration passed into
> // SetConfiguration.
> TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) {
>-  CreatePeerConnection();
>+  PeerConnectionInterface::RTCConfiguration starting_config;
>+  starting_config.bundle_policy =
>+      webrtc::PeerConnection::kBundlePolicyMaxBundle;
>+  CreatePeerConnection(starting_config);
> 
>   PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
>   config.type = PeerConnectionInterface::kRelay;
>+  config.use_media_transport = true;
>+  config.use_media_transport_for_data_channels = true;
>   EXPECT_TRUE(pc_->SetConfiguration(config));
> 
>   PeerConnectionInterface::RTCConfiguration returned_config =
>       pc_->GetConfiguration();
>   EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
>+  EXPECT_TRUE(returned_config.use_media_transport);
>+  EXPECT_TRUE(returned_config.use_media_transport_for_data_channels);
> }
> 
> TEST_P(PeerConnectionInterfaceTest, SetConfigurationFailsAfterClose) {
>@@ -3938,6 +3948,21 @@ TEST_P(PeerConnectionInterfaceTest,
>   EXPECT_FALSE(DoSetLocalDescription(std::move(offer)));
> }
> 
>+TEST_P(PeerConnectionInterfaceTest, ExtmapAllowMixedIsConfigurable) {
>+  RTCConfiguration config;
>+  // Default behavior is false.
>+  CreatePeerConnection(config);
>+  std::unique_ptr<SessionDescriptionInterface> offer;
>+  ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
>+  EXPECT_FALSE(offer->description()->extmap_allow_mixed());
>+  // Possible to set to true.
>+  config.offer_extmap_allow_mixed = true;
>+  CreatePeerConnection(config);
>+  offer.release();
>+  ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
>+  EXPECT_TRUE(offer->description()->extmap_allow_mixed());
>+}
>+
> INSTANTIATE_TEST_CASE_P(PeerConnectionInterfaceTest,
>                         PeerConnectionInterfaceTest,
>                         Values(SdpSemantics::kPlanB,
>@@ -3954,6 +3979,7 @@ class PeerConnectionMediaConfigTest : public testing::Test {
>     rtc::scoped_refptr<PeerConnectionInterface> pc(
>         pcf_->CreatePeerConnection(config, nullptr, nullptr, &observer_));
>     EXPECT_TRUE(pc.get());
>+    observer_.SetPeerConnectionInterface(pc.get());
>     return pc->GetConfiguration().media_config;
>   }
> 
>@@ -3961,6 +3987,17 @@ class PeerConnectionMediaConfigTest : public testing::Test {
>   MockPeerConnectionObserver observer_;
> };
> 
>+// This sanity check validates the test infrastructure itself.
>+TEST_F(PeerConnectionMediaConfigTest, TestCreateAndClose) {
>+  PeerConnectionInterface::RTCConfiguration config;
>+  rtc::scoped_refptr<PeerConnectionInterface> pc(
>+      pcf_->CreatePeerConnection(config, nullptr, nullptr, &observer_));
>+  EXPECT_TRUE(pc.get());
>+  observer_.SetPeerConnectionInterface(pc.get());  // Required.
>+  pc->Close();                                     // No abort -> ok.
>+  SUCCEED();
>+}
>+
> // This test verifies the default behaviour with no constraints and a
> // default RTCConfiguration.
> TEST_F(PeerConnectionMediaConfigTest, TestDefaults) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcpmuxfilter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcpmuxfilter_unittest.cc
>index 555ab5c4681054c40a71e1dce6c34aa970ce124a..a6ac4478fcd5ae75ba5716c036f90003fe08aa0b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcpmuxfilter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcpmuxfilter_unittest.cc
>@@ -9,7 +9,6 @@
>  */
> 
> #include "pc/rtcpmuxfilter.h"
>-#include "media/base/testutils.h"
> #include "rtc_base/gunit.h"
> 
> TEST(RtcpMuxFilterTest, IsActiveSender) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstats_integrationtest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstats_integrationtest.cc
>index 3c07e38b5a495aa93e7db9ffd1ef2f5594083ba8..49084de4d20717c22ddc891b7ea463204cb63b4c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstats_integrationtest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstats_integrationtest.cc
>@@ -604,11 +604,18 @@ class RTCStatsReportVerifier {
>           media_stream_track.concealed_samples);
>       verifier.TestMemberIsNonNegative<uint64_t>(
>           media_stream_track.concealment_events);
>+      verifier.TestMemberIsNonNegative<uint64_t>(
>+          media_stream_track.jitter_buffer_flushes);
>+      verifier.TestMemberIsNonNegative<uint64_t>(
>+          media_stream_track.delayed_packet_outage_samples);
>     } else {
>       verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_delay);
>       verifier.TestMemberIsUndefined(media_stream_track.total_samples_received);
>       verifier.TestMemberIsUndefined(media_stream_track.concealed_samples);
>       verifier.TestMemberIsUndefined(media_stream_track.concealment_events);
>+      verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_flushes);
>+      verifier.TestMemberIsUndefined(
>+          media_stream_track.delayed_packet_outage_samples);
>     }
>     return verifier.ExpectAllMembersSuccessfullyTested();
>   }
>@@ -806,7 +813,11 @@ TEST_F(RTCStatsIntegrationTest, GetStatsWithInvalidReceiverSelector) {
>   EXPECT_FALSE(report->size());
> }
> 
>-TEST_F(RTCStatsIntegrationTest, GetsStatsWhileDestroyingPeerConnections) {
>+// TODO(bugs.webrtc.org/10041) For now this is equivalent to the following
>+// test GetsStatsWhileClosingPeerConnection, because pc() is closed by
>+// PeerConnectionTestWrapper. See: bugs.webrtc.org/9847
>+TEST_F(RTCStatsIntegrationTest,
>+       DISABLED_GetStatsWhileDestroyingPeerConnection) {
>   StartCall();
> 
>   rtc::scoped_refptr<RTCStatsObtainer> stats_obtainer =
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc
>index d8839430ebef8423e2f98f583443c1e45bcf92d7..d48ecc01f359e51b900259828c085c8b5e1617f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc
>@@ -454,6 +454,10 @@ ProduceMediaStreamTrackStatsFromVoiceReceiverInfo(
>   audio_track_stats->concealed_samples = voice_receiver_info.concealed_samples;
>   audio_track_stats->concealment_events =
>       voice_receiver_info.concealment_events;
>+  audio_track_stats->jitter_buffer_flushes =
>+      voice_receiver_info.jitter_buffer_flushes;
>+  audio_track_stats->delayed_packet_outage_samples =
>+      voice_receiver_info.delayed_packet_outage_samples;
>   return audio_track_stats;
> }
> 
>@@ -1388,7 +1392,7 @@ RTCStatsCollector::PrepareTransportCertificateStats_n(
>     rtc::scoped_refptr<rtc::RTCCertificate> local_certificate;
>     if (pc_->GetLocalCertificate(transport_name, &local_certificate)) {
>       certificate_stats_pair.local =
>-          local_certificate->ssl_cert_chain().GetStats();
>+          local_certificate->GetSSLCertificateChain().GetStats();
>     }
> 
>     std::unique_ptr<rtc::SSLCertChain> remote_cert_chain =
>@@ -1426,7 +1430,7 @@ RTCStatsCollector::PrepareTransceiverStatsInfos_s() const {
>     stats.transceiver = transceiver->internal();
>     stats.media_type = media_type;
> 
>-    cricket::BaseChannel* channel = transceiver->internal()->channel();
>+    cricket::ChannelInterface* channel = transceiver->internal()->channel();
>     if (!channel) {
>       // The remaining fields require a BaseChannel.
>       continue;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector_unittest.cc
>index 7404d492ef37dd1aadcb7f423bd7b886d65adcb3..adca9ef6c4657ecddcf4f58d71e35d060c639505 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector_unittest.cc
>@@ -143,10 +143,10 @@ std::unique_ptr<CertificateInfo> CreateFakeCertificateAndInfoFromDers(
>   }
>   // Fingerprints for the whole certificate chain, starting with leaf
>   // certificate.
>-  const rtc::SSLCertChain& chain = info->certificate->ssl_cert_chain();
>+  const rtc::SSLCertChain& chain = info->certificate->GetSSLCertificateChain();
>   std::unique_ptr<rtc::SSLFingerprint> fp;
>   for (size_t i = 0; i < chain.GetSize(); i++) {
>-    fp.reset(rtc::SSLFingerprint::Create("sha-1", &chain.Get(i)));
>+    fp = rtc::SSLFingerprint::Create("sha-1", chain.Get(i));
>     EXPECT_TRUE(fp);
>     info->fingerprints.push_back(fp->GetRfc4572Fingerprint());
>   }
>@@ -704,7 +704,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsSingle) {
>           std::vector<std::string>({"(remote) single certificate"}));
>   pc_->SetRemoteCertChain(
>       kTransportName,
>-      remote_certinfo->certificate->ssl_cert_chain().UniqueCopy());
>+      remote_certinfo->certificate->GetSSLCertificateChain().Clone());
> 
>   rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>@@ -818,7 +818,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsMultiple) {
>           std::vector<std::string>({"(remote) audio"}));
>   pc_->SetRemoteCertChain(
>       kAudioTransport,
>-      audio_remote_certinfo->certificate->ssl_cert_chain().UniqueCopy());
>+      audio_remote_certinfo->certificate->GetSSLCertificateChain().Clone());
> 
>   pc_->AddVideoChannel("video", kVideoTransport);
>   std::unique_ptr<CertificateInfo> video_local_certinfo =
>@@ -830,7 +830,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsMultiple) {
>           std::vector<std::string>({"(remote) video"}));
>   pc_->SetRemoteCertChain(
>       kVideoTransport,
>-      video_remote_certinfo->certificate->ssl_cert_chain().UniqueCopy());
>+      video_remote_certinfo->certificate->GetSSLCertificateChain().Clone());
> 
>   rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
>   ExpectReportContainsCertificateInfo(report, *audio_local_certinfo);
>@@ -855,7 +855,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsChain) {
>                                             "(remote) chain"});
>   pc_->SetRemoteCertChain(
>       kTransportName,
>-      remote_certinfo->certificate->ssl_cert_chain().UniqueCopy());
>+      remote_certinfo->certificate->GetSSLCertificateChain().Clone());
> 
>   rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
>   ExpectReportContainsCertificateInfo(report, *local_certinfo);
>@@ -1426,6 +1426,8 @@ TEST_F(RTCStatsCollectorTest,
>   voice_receiver_info.concealed_samples = 123;
>   voice_receiver_info.concealment_events = 12;
>   voice_receiver_info.jitter_buffer_delay_seconds = 3456;
>+  voice_receiver_info.jitter_buffer_flushes = 7;
>+  voice_receiver_info.delayed_packet_outage_samples = 15;
> 
>   stats_->CreateMockRtpSendersReceiversAndChannels(
>       {}, {std::make_pair(remote_audio_track.get(), voice_receiver_info)}, {},
>@@ -1459,6 +1461,8 @@ TEST_F(RTCStatsCollectorTest,
>   expected_remote_audio_track.concealed_samples = 123;
>   expected_remote_audio_track.concealment_events = 12;
>   expected_remote_audio_track.jitter_buffer_delay = 3456;
>+  expected_remote_audio_track.jitter_buffer_flushes = 7;
>+  expected_remote_audio_track.delayed_packet_outage_samples = 15;
>   ASSERT_TRUE(report->Get(expected_remote_audio_track.id()));
>   EXPECT_EQ(expected_remote_audio_track,
>             report->Get(expected_remote_audio_track.id())
>@@ -1957,7 +1961,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
>           {"(remote) local", "(remote) chain"});
>   pc_->SetRemoteCertChain(
>       kTransportName,
>-      remote_certinfo->certificate->ssl_cert_chain().UniqueCopy());
>+      remote_certinfo->certificate->GetSSLCertificateChain().Clone());
> 
>   report = stats_->GetFreshStatsReport();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>index e519b9b06de6c33a026d4959e9b73ce6d4e2d790..865685d73c4324fb8f2f61fea06847db9c656328 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>@@ -396,6 +396,7 @@ RtpCapabilities ToRtpCapabilities(
>   bool have_red = false;
>   bool have_ulpfec = false;
>   bool have_flexfec = false;
>+  bool have_rtx = false;
>   for (const C& cricket_codec : cricket_codecs) {
>     if (cricket_codec.name == cricket::kRedCodecName) {
>       have_red = true;
>@@ -403,8 +404,19 @@ RtpCapabilities ToRtpCapabilities(
>       have_ulpfec = true;
>     } else if (cricket_codec.name == cricket::kFlexfecCodecName) {
>       have_flexfec = true;
>+    } else if (cricket_codec.name == cricket::kRtxCodecName) {
>+      if (have_rtx) {
>+        // There should only be one RTX codec entry
>+        continue;
>+      }
>+      have_rtx = true;
>+    }
>+    auto codec_capability = ToRtpCodecCapability(cricket_codec);
>+    if (cricket_codec.name == cricket::kRtxCodecName) {
>+      // RTX codec should not have any parameter
>+      codec_capability.parameters.clear();
>     }
>-    capabilities.codecs.push_back(ToRtpCodecCapability(cricket_codec));
>+    capabilities.codecs.push_back(codec_capability);
>   }
>   for (const RtpExtension& cricket_extension : cricket_extensions) {
>     capabilities.header_extensions.emplace_back(cricket_extension.uri,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion_unittest.cc
>index 257d26ebb50c3b8e6f25c99708a66137bf5c30c1..f4f1b9270c6c96afed9994780e9892b1e7d92d60 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion_unittest.cc
>@@ -574,11 +574,23 @@ TEST(RtpParametersConversionTest, ToRtpCapabilities) {
>   flexfec.id = 102;
>   flexfec.clockrate = 90000;
> 
>+  cricket::VideoCodec rtx;
>+  rtx.name = "rtx";
>+  rtx.id = 104;
>+  rtx.params.insert({"apt", "101"});
>+
>+  cricket::VideoCodec rtx2;
>+  rtx2.name = "rtx";
>+  rtx2.id = 105;
>+  rtx2.params.insert({"apt", "109"});
>+
>   RtpCapabilities capabilities = ToRtpCapabilities<cricket::VideoCodec>(
>-      {vp8, ulpfec}, {{"uri", 1}, {"uri2", 3}});
>-  ASSERT_EQ(2u, capabilities.codecs.size());
>+      {vp8, ulpfec, rtx, rtx2}, {{"uri", 1}, {"uri2", 3}});
>+  ASSERT_EQ(3u, capabilities.codecs.size());
>   EXPECT_EQ("VP8", capabilities.codecs[0].name);
>   EXPECT_EQ("ulpfec", capabilities.codecs[1].name);
>+  EXPECT_EQ("rtx", capabilities.codecs[2].name);
>+  EXPECT_EQ(0u, capabilities.codecs[2].parameters.size());
>   ASSERT_EQ(2u, capabilities.header_extensions.size());
>   EXPECT_EQ("uri", capabilities.header_extensions[0].uri);
>   EXPECT_EQ(1, capabilities.header_extensions[0].preferred_id);
>@@ -587,8 +599,8 @@ TEST(RtpParametersConversionTest, ToRtpCapabilities) {
>   EXPECT_EQ(0u, capabilities.fec.size());
> 
>   capabilities = ToRtpCapabilities<cricket::VideoCodec>(
>-      {vp8, red, ulpfec}, cricket::RtpHeaderExtensions());
>-  EXPECT_EQ(3u, capabilities.codecs.size());
>+      {vp8, red, ulpfec, rtx}, cricket::RtpHeaderExtensions());
>+  EXPECT_EQ(4u, capabilities.codecs.size());
>   EXPECT_EQ(2u, capabilities.fec.size());
>   EXPECT_NE(capabilities.fec.end(),
>             std::find(capabilities.fec.begin(), capabilities.fec.end(),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc
>index d4a1f62f83b84cca670faa0be58291e0c2a3bc4f..1916a7367032c4bb1f81a6510d2eef11c45a9279 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc
>@@ -50,9 +50,10 @@ void MaybeAttachFrameDecryptorToMediaChannel(
>     const absl::optional<uint32_t>& ssrc,
>     rtc::Thread* worker_thread,
>     rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor,
>-    cricket::MediaChannel* media_channel) {
>-  if (media_channel && ssrc.has_value()) {
>-    return worker_thread->Invoke<void>(RTC_FROM_HERE, [&] {
>+    cricket::MediaChannel* media_channel,
>+    bool stopped) {
>+  if (media_channel && frame_decryptor && ssrc.has_value() && !stopped) {
>+    worker_thread->Invoke<void>(RTC_FROM_HERE, [&] {
>       media_channel->SetFrameDecryptor(*ssrc, frame_decryptor);
>     });
>   }
>@@ -156,9 +157,12 @@ bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
> void AudioRtpReceiver::SetFrameDecryptor(
>     rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
>   frame_decryptor_ = std::move(frame_decryptor);
>-  // Attach the frame decryptor to the media channel if it exists.
>-  MaybeAttachFrameDecryptorToMediaChannel(ssrc_, worker_thread_,
>-                                          frame_decryptor_, media_channel_);
>+  // Special Case: Set the frame decryptor to any value on any existing channel.
>+  if (media_channel_ && ssrc_.has_value() && !stopped_) {
>+    worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
>+      media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_);
>+    });
>+  }
> }
> 
> rtc::scoped_refptr<FrameDecryptorInterface>
>@@ -252,8 +256,8 @@ void AudioRtpReceiver::Reconfigure() {
>     RTC_NOTREACHED();
>   }
>   // Reattach the frame decryptor if we were reconfigured.
>-  MaybeAttachFrameDecryptorToMediaChannel(ssrc_, worker_thread_,
>-                                          frame_decryptor_, media_channel_);
>+  MaybeAttachFrameDecryptorToMediaChannel(
>+      ssrc_, worker_thread_, frame_decryptor_, media_channel_, stopped_);
> }
> 
> void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
>@@ -264,9 +268,10 @@ void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
>   }
> }
> 
>-void AudioRtpReceiver::SetVoiceMediaChannel(
>-    cricket::VoiceMediaChannel* voice_media_channel) {
>-  media_channel_ = voice_media_channel;
>+void AudioRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
>+  RTC_DCHECK(media_channel == nullptr ||
>+             media_channel->media_type() == media_type());
>+  media_channel_ = static_cast<cricket::VoiceMediaChannel*>(media_channel);
> }
> 
> void AudioRtpReceiver::NotifyFirstPacketReceived() {
>@@ -347,9 +352,12 @@ bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
> void VideoRtpReceiver::SetFrameDecryptor(
>     rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
>   frame_decryptor_ = std::move(frame_decryptor);
>-  // Attach the new frame decryptor the media channel if it exists yet.
>-  MaybeAttachFrameDecryptorToMediaChannel(ssrc_, worker_thread_,
>-                                          frame_decryptor_, media_channel_);
>+  // Special Case: Set the frame decryptor to any value on any existing channel.
>+  if (media_channel_ && ssrc_.has_value() && !stopped_) {
>+    worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
>+      media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_);
>+    });
>+  }
> }
> 
> rtc::scoped_refptr<FrameDecryptorInterface>
>@@ -387,8 +395,8 @@ void VideoRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
>   ssrc_ = ssrc;
>   SetSink(source_->sink());
>   // Attach any existing frame decryptor to the media channel.
>-  MaybeAttachFrameDecryptorToMediaChannel(ssrc_, worker_thread_,
>-                                          frame_decryptor_, media_channel_);
>+  MaybeAttachFrameDecryptorToMediaChannel(
>+      ssrc_, worker_thread_, frame_decryptor_, media_channel_, stopped_);
> }
> 
> void VideoRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
>@@ -436,9 +444,10 @@ void VideoRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
>   }
> }
> 
>-void VideoRtpReceiver::SetVideoMediaChannel(
>-    cricket::VideoMediaChannel* video_media_channel) {
>-  media_channel_ = video_media_channel;
>+void VideoRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
>+  RTC_DCHECK(media_channel == nullptr ||
>+             media_channel->media_type() == media_type());
>+  media_channel_ = static_cast<cricket::VideoMediaChannel*>(media_channel);
> }
> 
> void VideoRtpReceiver::NotifyFirstPacketReceived() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.h
>index 85be8c258bfba2f5b092eff1bfcd52c4122c3f69..06331a66c4dd95ca2e5e1588462bb1efefd8727c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.h
>@@ -34,14 +34,10 @@ class RtpReceiverInternal : public RtpReceiverInterface {
>   virtual void Stop() = 0;
> 
>   // Sets the underlying MediaEngine channel associated with this RtpSender.
>-  // SetVoiceMediaChannel should be used for audio RtpSenders and
>-  // SetVideoMediaChannel should be used for video RtpSenders. Must call the
>-  // appropriate SetXxxMediaChannel(nullptr) before the media channel is
>-  // destroyed.
>-  virtual void SetVoiceMediaChannel(
>-      cricket::VoiceMediaChannel* voice_media_channel) = 0;
>-  virtual void SetVideoMediaChannel(
>-      cricket::VideoMediaChannel* video_media_channel) = 0;
>+  // A VoiceMediaChannel should be used for audio RtpSenders and
>+  // a VideoMediaChannel should be used for video RtpSenders.
>+  // Must call SetMediaChannel(nullptr) before the media channel is destroyed.
>+  virtual void SetMediaChannel(cricket::MediaChannel* media_channel) = 0;
> 
>   // Configures the RtpReceiver with the underlying media channel, with the
>   // given SSRC as the stream identifier. If |ssrc| is 0, the receiver will
>@@ -130,13 +126,8 @@ class AudioRtpReceiver : public ObserverInterface,
>   void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
>                       streams) override;
>   void SetObserver(RtpReceiverObserverInterface* observer) override;
>-  void SetVoiceMediaChannel(
>-      cricket::VoiceMediaChannel* voice_media_channel) override;
> 
>-  void SetVideoMediaChannel(
>-      cricket::VideoMediaChannel* video_media_channel) override {
>-    RTC_NOTREACHED();
>-  }
>+  void SetMediaChannel(cricket::MediaChannel* media_channel) override;
> 
>   std::vector<RtpSource> GetSources() const override;
>   int AttachmentId() const override { return attachment_id_; }
>@@ -217,13 +208,7 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
> 
>   void SetObserver(RtpReceiverObserverInterface* observer) override;
> 
>-  void SetVoiceMediaChannel(
>-      cricket::VoiceMediaChannel* voice_media_channel) override {
>-    RTC_NOTREACHED();
>-  }
>-
>-  void SetVideoMediaChannel(
>-      cricket::VideoMediaChannel* video_media_channel) override;
>+  void SetMediaChannel(cricket::MediaChannel* media_channel) override;
> 
>   int AttachmentId() const override { return attachment_id_; }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.cc
>index 3bb90f24da14b914d303f5ba8ec3c637cd4b52bd..76fdca6e94209483ea7673251dda42d5e9c6f88f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.cc
>@@ -58,7 +58,8 @@ bool UnimplementedRtpEncodingParameterHasValue(
> // layer.
> bool PerSenderRtpEncodingParameterHasValue(
>     const RtpEncodingParameters& encoding_params) {
>-  if (encoding_params.bitrate_priority != kDefaultBitratePriority) {
>+  if (encoding_params.bitrate_priority != kDefaultBitratePriority ||
>+      encoding_params.network_priority != kDefaultBitratePriority) {
>     return true;
>   }
>   return false;
>@@ -68,13 +69,14 @@ bool PerSenderRtpEncodingParameterHasValue(
> // correct worker thread only if both the media channel exists and a ssrc has
> // been allocated to the stream.
> void MaybeAttachFrameEncryptorToMediaChannel(
>-    const absl::optional<uint32_t> ssrc,
>+    const uint32_t ssrc,
>     rtc::Thread* worker_thread,
>     rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor,
>-    cricket::MediaChannel* media_channel) {
>-  if (media_channel && ssrc.has_value()) {
>-    return worker_thread->Invoke<void>(RTC_FROM_HERE, [&] {
>-      media_channel->SetFrameEncryptor(*ssrc, frame_encryptor);
>+    cricket::MediaChannel* media_channel,
>+    bool stopped) {
>+  if (media_channel && frame_encryptor && ssrc && !stopped) {
>+    worker_thread->Invoke<void>(RTC_FROM_HERE, [&] {
>+      media_channel->SetFrameEncryptor(ssrc, frame_encryptor);
>     });
>   }
> }
>@@ -305,8 +307,12 @@ rtc::scoped_refptr<DtmfSenderInterface> AudioRtpSender::GetDtmfSender() const {
> void AudioRtpSender::SetFrameEncryptor(
>     rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
>   frame_encryptor_ = std::move(frame_encryptor);
>-  MaybeAttachFrameEncryptorToMediaChannel(ssrc_, worker_thread_,
>-                                          frame_encryptor_, media_channel_);
>+  // Special Case: Set the frame encryptor to any value on any existing channel.
>+  if (media_channel_ && ssrc_ && !stopped_) {
>+    worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
>+      media_channel_->SetFrameEncryptor(ssrc_, frame_encryptor_);
>+    });
>+  }
> }
> 
> rtc::scoped_refptr<FrameEncryptorInterface> AudioRtpSender::GetFrameEncryptor()
>@@ -356,8 +362,8 @@ void AudioRtpSender::SetSsrc(uint32_t ssrc) {
>     });
>   }
>   // Each time there is an ssrc update.
>-  MaybeAttachFrameEncryptorToMediaChannel(ssrc_, worker_thread_,
>-                                          frame_encryptor_, media_channel_);
>+  MaybeAttachFrameEncryptorToMediaChannel(
>+      ssrc_, worker_thread_, frame_encryptor_, media_channel_, stopped_);
> }
> 
> void AudioRtpSender::Stop() {
>@@ -380,9 +386,10 @@ void AudioRtpSender::Stop() {
>   stopped_ = true;
> }
> 
>-void AudioRtpSender::SetVoiceMediaChannel(
>-    cricket::VoiceMediaChannel* voice_media_channel) {
>-  media_channel_ = voice_media_channel;
>+void AudioRtpSender::SetMediaChannel(cricket::MediaChannel* media_channel) {
>+  RTC_DCHECK(media_channel == nullptr ||
>+             media_channel->media_type() == media_type());
>+  media_channel_ = static_cast<cricket::VoiceMediaChannel*>(media_channel);
> }
> 
> void AudioRtpSender::SetAudioSend() {
>@@ -399,9 +406,7 @@ void AudioRtpSender::SetAudioSend() {
>   // options since it is also applied to all streams/channels, local or remote.
>   if (track_->enabled() && track_->GetSource() &&
>       !track_->GetSource()->remote()) {
>-    // TODO(xians): Remove this static_cast since we should be able to connect
>-    // a remote audio track to a peer connection.
>-    options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
>+    options = track_->GetSource()->options();
>   }
> #endif
> 
>@@ -557,8 +562,12 @@ rtc::scoped_refptr<DtmfSenderInterface> VideoRtpSender::GetDtmfSender() const {
> void VideoRtpSender::SetFrameEncryptor(
>     rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
>   frame_encryptor_ = std::move(frame_encryptor);
>-  MaybeAttachFrameEncryptorToMediaChannel(ssrc_, worker_thread_,
>-                                          frame_encryptor_, media_channel_);
>+  // Special Case: Set the frame encryptor to any value on any existing channel.
>+  if (media_channel_ && ssrc_ && !stopped_) {
>+    worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
>+      media_channel_->SetFrameEncryptor(ssrc_, frame_encryptor_);
>+    });
>+  }
> }
> 
> rtc::scoped_refptr<FrameEncryptorInterface> VideoRtpSender::GetFrameEncryptor()
>@@ -601,8 +610,8 @@ void VideoRtpSender::SetSsrc(uint32_t ssrc) {
>       init_parameters_.encodings.clear();
>     });
>   }
>-  MaybeAttachFrameEncryptorToMediaChannel(ssrc_, worker_thread_,
>-                                          frame_encryptor_, media_channel_);
>+  MaybeAttachFrameEncryptorToMediaChannel(
>+      ssrc_, worker_thread_, frame_encryptor_, media_channel_, stopped_);
> }
> 
> void VideoRtpSender::Stop() {
>@@ -621,9 +630,10 @@ void VideoRtpSender::Stop() {
>   stopped_ = true;
> }
> 
>-void VideoRtpSender::SetVideoMediaChannel(
>-    cricket::VideoMediaChannel* video_media_channel) {
>-  media_channel_ = video_media_channel;
>+void VideoRtpSender::SetMediaChannel(cricket::MediaChannel* media_channel) {
>+  RTC_DCHECK(media_channel == nullptr ||
>+             media_channel->media_type() == media_type());
>+  media_channel_ = static_cast<cricket::VideoMediaChannel*>(media_channel);
> }
> 
> void VideoRtpSender::SetVideoSend() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.h
>index 2a8289f417eac94e2483a98be7f78b6bbd7f827c..07fae6b9547c494a23ff3d758e2e3a62aeffc8d0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.h
>@@ -36,14 +36,10 @@ bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters);
> class RtpSenderInternal : public RtpSenderInterface {
>  public:
>   // Sets the underlying MediaEngine channel associated with this RtpSender.
>-  // SetVoiceMediaChannel should be used for audio RtpSenders and
>-  // SetVideoMediaChannel should be used for video RtpSenders. Must call the
>-  // appropriate SetXxxMediaChannel(nullptr) before the media channel is
>-  // destroyed.
>-  virtual void SetVoiceMediaChannel(
>-      cricket::VoiceMediaChannel* voice_media_channel) = 0;
>-  virtual void SetVideoMediaChannel(
>-      cricket::VideoMediaChannel* video_media_channel) = 0;
>+  // A VoiceMediaChannel should be used for audio RtpSenders and
>+  // a VideoMediaChannel should be used for video RtpSenders.
>+  // Must call SetMediaChannel(nullptr) before the media channel is destroyed.
>+  virtual void SetMediaChannel(cricket::MediaChannel* media_channel) = 0;
> 
>   // Used to set the SSRC of the sender, once a local description has been set.
>   // If |ssrc| is 0, this indiates that the sender should disconnect from the
>@@ -156,13 +152,7 @@ class AudioRtpSender : public DtmfProviderInterface,
> 
>   int AttachmentId() const override { return attachment_id_; }
> 
>-  void SetVoiceMediaChannel(
>-      cricket::VoiceMediaChannel* voice_media_channel) override;
>-
>-  void SetVideoMediaChannel(
>-      cricket::VideoMediaChannel* video_media_channel) override {
>-    RTC_NOTREACHED();
>-  }
>+  void SetMediaChannel(cricket::MediaChannel* media_channel) override;
> 
>  private:
>   // TODO(nisse): Since SSRC == 0 is technically valid, figure out
>@@ -253,13 +243,7 @@ class VideoRtpSender : public ObserverInterface,
>   void Stop() override;
>   int AttachmentId() const override { return attachment_id_; }
> 
>-  void SetVoiceMediaChannel(
>-      cricket::VoiceMediaChannel* voice_media_channel) override {
>-    RTC_NOTREACHED();
>-  }
>-
>-  void SetVideoMediaChannel(
>-      cricket::VideoMediaChannel* video_media_channel) override;
>+  void SetMediaChannel(cricket::MediaChannel* media_channel) override;
> 
>  private:
>   bool can_send_track() const { return track_ && ssrc_; }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsenderreceiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsenderreceiver_unittest.cc
>index 07ff6a3865446474a2186e2b691e8ffc9ca90344..3be69bafc9c01f5d6ef4d42f071882a23e23a66a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsenderreceiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsenderreceiver_unittest.cc
>@@ -80,12 +80,12 @@ class RtpSenderReceiverTest : public testing::Test,
> 
>     voice_channel_ = channel_manager_.CreateVoiceChannel(
>         &fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
>-        rtc::Thread::Current(), cricket::CN_AUDIO, srtp_required,
>-        rtc::CryptoOptions(), cricket::AudioOptions());
>+        /*media_transport=*/nullptr, rtc::Thread::Current(), cricket::CN_AUDIO,
>+        srtp_required, webrtc::CryptoOptions(), cricket::AudioOptions());
>     video_channel_ = channel_manager_.CreateVideoChannel(
>         &fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
>         rtc::Thread::Current(), cricket::CN_VIDEO, srtp_required,
>-        rtc::CryptoOptions(), cricket::VideoOptions());
>+        webrtc::CryptoOptions(), cricket::VideoOptions());
>     voice_channel_->Enable(true);
>     video_channel_->Enable(true);
>     voice_media_channel_ = media_engine_->GetVoiceChannel(0);
>@@ -154,7 +154,7 @@ class RtpSenderReceiverTest : public testing::Test,
>         new AudioRtpSender(worker_thread_, audio_track_->id(), nullptr);
>     ASSERT_TRUE(audio_rtp_sender_->SetTrack(audio_track_));
>     audio_rtp_sender_->set_stream_ids({local_stream_->id()});
>-    audio_rtp_sender_->SetVoiceMediaChannel(voice_media_channel_);
>+    audio_rtp_sender_->SetMediaChannel(voice_media_channel_);
>     audio_rtp_sender_->SetSsrc(kAudioSsrc);
>     audio_rtp_sender_->GetOnDestroyedSignal()->connect(
>         this, &RtpSenderReceiverTest::OnAudioSenderDestroyed);
>@@ -163,7 +163,7 @@ class RtpSenderReceiverTest : public testing::Test,
> 
>   void CreateAudioRtpSenderWithNoTrack() {
>     audio_rtp_sender_ = new AudioRtpSender(worker_thread_, /*id=*/"", nullptr);
>-    audio_rtp_sender_->SetVoiceMediaChannel(voice_media_channel_);
>+    audio_rtp_sender_->SetMediaChannel(voice_media_channel_);
>   }
> 
>   void OnAudioSenderDestroyed() { audio_sender_destroyed_signal_fired_ = true; }
>@@ -191,13 +191,13 @@ class RtpSenderReceiverTest : public testing::Test,
>     video_rtp_sender_ = new VideoRtpSender(worker_thread_, video_track_->id());
>     ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
>     video_rtp_sender_->set_stream_ids({local_stream_->id()});
>-    video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
>+    video_rtp_sender_->SetMediaChannel(video_media_channel_);
>     video_rtp_sender_->SetSsrc(ssrc);
>     VerifyVideoChannelInput(ssrc);
>   }
>   void CreateVideoRtpSenderWithNoTrack() {
>     video_rtp_sender_ = new VideoRtpSender(worker_thread_, /*id=*/"");
>-    video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
>+    video_rtp_sender_->SetMediaChannel(video_media_channel_);
>   }
> 
>   void DestroyAudioRtpSender() {
>@@ -214,7 +214,7 @@ class RtpSenderReceiverTest : public testing::Test,
>       std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
>     audio_rtp_receiver_ = new AudioRtpReceiver(
>         rtc::Thread::Current(), kAudioTrackId, std::move(streams));
>-    audio_rtp_receiver_->SetVoiceMediaChannel(voice_media_channel_);
>+    audio_rtp_receiver_->SetMediaChannel(voice_media_channel_);
>     audio_rtp_receiver_->SetupMediaChannel(kAudioSsrc);
>     audio_track_ = audio_rtp_receiver_->audio_track();
>     VerifyVoiceChannelOutput();
>@@ -224,12 +224,30 @@ class RtpSenderReceiverTest : public testing::Test,
>       std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
>     video_rtp_receiver_ = new VideoRtpReceiver(
>         rtc::Thread::Current(), kVideoTrackId, std::move(streams));
>-    video_rtp_receiver_->SetVideoMediaChannel(video_media_channel_);
>+    video_rtp_receiver_->SetMediaChannel(video_media_channel_);
>     video_rtp_receiver_->SetupMediaChannel(kVideoSsrc);
>     video_track_ = video_rtp_receiver_->video_track();
>     VerifyVideoChannelOutput();
>   }
> 
>+  void CreateVideoRtpReceiverWithSimulcast(
>+      std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {},
>+      int num_layers = kVideoSimulcastLayerCount) {
>+    std::vector<uint32_t> ssrcs;
>+    for (int i = 0; i < num_layers; ++i)
>+      ssrcs.push_back(kVideoSsrcSimulcast + i);
>+    cricket::StreamParams stream_params =
>+        cricket::CreateSimStreamParams("cname", ssrcs);
>+    video_media_channel_->AddRecvStream(stream_params);
>+    uint32_t primary_ssrc = stream_params.first_ssrc();
>+
>+    video_rtp_receiver_ = new VideoRtpReceiver(
>+        rtc::Thread::Current(), kVideoTrackId, std::move(streams));
>+    video_rtp_receiver_->SetMediaChannel(video_media_channel_);
>+    video_rtp_receiver_->SetupMediaChannel(primary_ssrc);
>+    video_track_ = video_rtp_receiver_->video_track();
>+  }
>+
>   void DestroyAudioRtpReceiver() {
>     audio_rtp_receiver_ = nullptr;
>     VerifyVoiceChannelNoOutput();
>@@ -664,7 +682,7 @@ TEST_F(RtpSenderReceiverTest, AudioSenderInitParametersMovedAfterNegotiation) {
>   cricket::StreamParams stream_params =
>       cricket::CreateSimStreamParams("cname", ssrcs);
>   voice_media_channel_->AddSendStream(stream_params);
>-  audio_rtp_sender_->SetVoiceMediaChannel(voice_media_channel_);
>+  audio_rtp_sender_->SetMediaChannel(voice_media_channel_);
>   audio_rtp_sender_->SetSsrc(1);
> 
>   params = audio_rtp_sender_->GetParameters();
>@@ -903,7 +921,7 @@ TEST_F(RtpSenderReceiverTest, VideoSenderInitParametersMovedAfterNegotiation) {
>   cricket::StreamParams stream_params =
>       cricket::CreateSimStreamParams("cname", ssrcs);
>   video_media_channel_->AddSendStream(stream_params);
>-  video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
>+  video_rtp_sender_->SetMediaChannel(video_media_channel_);
>   video_rtp_sender_->SetSsrc(kVideoSsrcSimulcast);
> 
>   params = video_rtp_sender_->GetParameters();
>@@ -938,7 +956,7 @@ TEST_F(RtpSenderReceiverTest,
>   cricket::StreamParams stream_params =
>       cricket::CreateSimStreamParams("cname", ssrcs);
>   video_media_channel_->AddSendStream(stream_params);
>-  video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
>+  video_rtp_sender_->SetMediaChannel(video_media_channel_);
>   video_rtp_sender_->SetSsrc(kVideoSsrcSimulcast);
> 
>   params = video_rtp_sender_->GetParameters();
>@@ -1263,6 +1281,15 @@ TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetParameters) {
>   DestroyVideoRtpReceiver();
> }
> 
>+TEST_F(RtpSenderReceiverTest, VideoReceiverCanGetParametersWithSimulcast) {
>+  CreateVideoRtpReceiverWithSimulcast({}, 2);
>+
>+  RtpParameters params = video_rtp_receiver_->GetParameters();
>+  EXPECT_EQ(2u, params.encodings.size());
>+
>+  DestroyVideoRtpReceiver();
>+}
>+
> // Test that makes sure that a video track content hint translates to the proper
> // value for sources that are not screencast.
> TEST_F(RtpSenderReceiverTest, PropagatesVideoTrackContentHint) {
>@@ -1332,7 +1359,7 @@ TEST_F(RtpSenderReceiverTest,
>   video_rtp_sender_ = new VideoRtpSender(worker_thread_, video_track_->id());
>   ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
>   video_rtp_sender_->set_stream_ids({local_stream_->id()});
>-  video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
>+  video_rtp_sender_->SetMediaChannel(video_media_channel_);
>   video_track_->set_enabled(true);
> 
>   // Sender is not ready to send (no SSRC) so no option should have been set.
>@@ -1424,6 +1451,18 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCanSetFrameEncryptor) {
>             audio_rtp_sender_->GetFrameEncryptor().get());
> }
> 
>+// Validate that setting a FrameEncryptor after the send stream is stopped does
>+// nothing.
>+TEST_F(RtpSenderReceiverTest, AudioSenderCannotSetFrameEncryptorAfterStop) {
>+  CreateAudioRtpSender();
>+  rtc::scoped_refptr<FrameEncryptorInterface> fake_frame_encryptor(
>+      new FakeFrameEncryptor());
>+  EXPECT_EQ(nullptr, audio_rtp_sender_->GetFrameEncryptor());
>+  audio_rtp_sender_->Stop();
>+  audio_rtp_sender_->SetFrameEncryptor(fake_frame_encryptor);
>+  // TODO(webrtc:9926) - Validate media channel not set once fakes updated.
>+}
>+
> // Validate that the default FrameEncryptor setting is nullptr.
> TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetFrameDecryptor) {
>   CreateAudioRtpReceiver();
>@@ -1435,4 +1474,60 @@ TEST_F(RtpSenderReceiverTest, AudioReceiverCanSetFrameDecryptor) {
>             audio_rtp_receiver_->GetFrameDecryptor().get());
> }
> 
>+// Validate that the default FrameEncryptor setting is nullptr.
>+TEST_F(RtpSenderReceiverTest, AudioReceiverCannotSetFrameDecryptorAfterStop) {
>+  CreateAudioRtpReceiver();
>+  rtc::scoped_refptr<FrameDecryptorInterface> fake_frame_decryptor(
>+      new FakeFrameDecryptor());
>+  EXPECT_EQ(nullptr, audio_rtp_receiver_->GetFrameDecryptor());
>+  audio_rtp_receiver_->Stop();
>+  audio_rtp_receiver_->SetFrameDecryptor(fake_frame_decryptor);
>+  // TODO(webrtc:9926) - Validate media channel not set once fakes updated.
>+}
>+
>+// Validate that the default FrameEncryptor setting is nullptr.
>+TEST_F(RtpSenderReceiverTest, VideoSenderCanSetFrameEncryptor) {
>+  CreateVideoRtpSender();
>+  rtc::scoped_refptr<FrameEncryptorInterface> fake_frame_encryptor(
>+      new FakeFrameEncryptor());
>+  EXPECT_EQ(nullptr, video_rtp_sender_->GetFrameEncryptor());
>+  video_rtp_sender_->SetFrameEncryptor(fake_frame_encryptor);
>+  EXPECT_EQ(fake_frame_encryptor.get(),
>+            video_rtp_sender_->GetFrameEncryptor().get());
>+}
>+
>+// Validate that setting a FrameEncryptor after the send stream is stopped does
>+// nothing.
>+TEST_F(RtpSenderReceiverTest, VideoSenderCannotSetFrameEncryptorAfterStop) {
>+  CreateVideoRtpSender();
>+  rtc::scoped_refptr<FrameEncryptorInterface> fake_frame_encryptor(
>+      new FakeFrameEncryptor());
>+  EXPECT_EQ(nullptr, video_rtp_sender_->GetFrameEncryptor());
>+  video_rtp_sender_->Stop();
>+  video_rtp_sender_->SetFrameEncryptor(fake_frame_encryptor);
>+  // TODO(webrtc:9926) - Validate media channel not set once fakes updated.
>+}
>+
>+// Validate that the default FrameEncryptor setting is nullptr.
>+TEST_F(RtpSenderReceiverTest, VideoReceiverCanSetFrameDecryptor) {
>+  CreateVideoRtpReceiver();
>+  rtc::scoped_refptr<FrameDecryptorInterface> fake_frame_decryptor(
>+      new FakeFrameDecryptor());
>+  EXPECT_EQ(nullptr, video_rtp_receiver_->GetFrameDecryptor());
>+  video_rtp_receiver_->SetFrameDecryptor(fake_frame_decryptor);
>+  EXPECT_EQ(fake_frame_decryptor.get(),
>+            video_rtp_receiver_->GetFrameDecryptor().get());
>+}
>+
>+// Validate that the default FrameEncryptor setting is nullptr.
>+TEST_F(RtpSenderReceiverTest, VideoReceiverCannotSetFrameDecryptorAfterStop) {
>+  CreateVideoRtpReceiver();
>+  rtc::scoped_refptr<FrameDecryptorInterface> fake_frame_decryptor(
>+      new FakeFrameDecryptor());
>+  EXPECT_EQ(nullptr, video_rtp_receiver_->GetFrameDecryptor());
>+  video_rtp_receiver_->Stop();
>+  video_rtp_receiver_->SetFrameDecryptor(fake_frame_decryptor);
>+  // TODO(webrtc:9926) - Validate media channel not set once fakes updated.
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc
>index 0fe8ea6f50a9930f3e8c8ab3151696d3befd2402..8b56b8b4f10a529f9b9a3b3e0814214d1664b769 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc
>@@ -38,52 +38,45 @@ RtpTransceiver::~RtpTransceiver() {
>   Stop();
> }
> 
>-void RtpTransceiver::SetChannel(cricket::BaseChannel* channel) {
>+void RtpTransceiver::SetChannel(cricket::ChannelInterface* channel) {
>+  // Cannot set a non-null channel on a stopped transceiver.
>+  if (stopped_ && channel) {
>+    return;
>+  }
>+
>   if (channel) {
>     RTC_DCHECK_EQ(media_type(), channel->media_type());
>   }
> 
>   if (channel_) {
>-    channel_->SignalFirstPacketReceived.disconnect(this);
>+    channel_->SignalFirstPacketReceived().disconnect(this);
>   }
> 
>   channel_ = channel;
> 
>   if (channel_) {
>-    channel_->SignalFirstPacketReceived.connect(
>+    channel_->SignalFirstPacketReceived().connect(
>         this, &RtpTransceiver::OnFirstPacketReceived);
>   }
> 
>   for (auto sender : senders_) {
>-    if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
>-      auto* voice_channel = static_cast<cricket::VoiceChannel*>(channel);
>-      sender->internal()->SetVoiceMediaChannel(
>-          voice_channel ? voice_channel->media_channel() : nullptr);
>-    } else {
>-      auto* video_channel = static_cast<cricket::VideoChannel*>(channel);
>-      sender->internal()->SetVideoMediaChannel(
>-          video_channel ? video_channel->media_channel() : nullptr);
>-    }
>+    sender->internal()->SetMediaChannel(channel_ ? channel_->media_channel()
>+                                                 : nullptr);
>   }
> 
>   for (auto receiver : receivers_) {
>-    if (!channel) {
>+    if (!channel_) {
>       receiver->internal()->Stop();
>     }
>-    if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
>-      auto* voice_channel = static_cast<cricket::VoiceChannel*>(channel);
>-      receiver->internal()->SetVoiceMediaChannel(
>-          voice_channel ? voice_channel->media_channel() : nullptr);
>-    } else {
>-      auto* video_channel = static_cast<cricket::VideoChannel*>(channel);
>-      receiver->internal()->SetVideoMediaChannel(
>-          video_channel ? video_channel->media_channel() : nullptr);
>-    }
>+
>+    receiver->internal()->SetMediaChannel(channel_ ? channel_->media_channel()
>+                                                   : nullptr);
>   }
> }
> 
> void RtpTransceiver::AddSender(
>     rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender) {
>+  RTC_DCHECK(!stopped_);
>   RTC_DCHECK(!unified_plan_);
>   RTC_DCHECK(sender);
>   RTC_DCHECK_EQ(media_type(), sender->media_type());
>@@ -109,6 +102,7 @@ bool RtpTransceiver::RemoveSender(RtpSenderInterface* sender) {
> void RtpTransceiver::AddReceiver(
>     rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
>         receiver) {
>+  RTC_DCHECK(!stopped_);
>   RTC_DCHECK(!unified_plan_);
>   RTC_DCHECK(receiver);
>   RTC_DCHECK_EQ(media_type(), receiver->media_type());
>@@ -152,7 +146,7 @@ absl::optional<std::string> RtpTransceiver::mid() const {
>   return mid_;
> }
> 
>-void RtpTransceiver::OnFirstPacketReceived(cricket::BaseChannel* channel) {
>+void RtpTransceiver::OnFirstPacketReceived(cricket::ChannelInterface*) {
>   for (auto receiver : receivers_) {
>     receiver->internal()->NotifyFirstPacketReceived();
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.h
>index 3e0d433782611576ca92e516e4e6b49543ae2950..07db196edd9abdca53cc8eee33c62a2bb4ea64bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.h
>@@ -70,11 +70,11 @@ class RtpTransceiver final
> 
>   // Returns the Voice/VideoChannel set for this transceiver. May be null if
>   // the transceiver is not in the currently set local/remote description.
>-  cricket::BaseChannel* channel() const { return channel_; }
>+  cricket::ChannelInterface* channel() const { return channel_; }
> 
>   // Sets the Voice/VideoChannel. The caller must pass in the correct channel
>   // implementation based on the type of the transceiver.
>-  void SetChannel(cricket::BaseChannel* channel);
>+  void SetChannel(cricket::ChannelInterface* channel);
> 
>   // Adds an RtpSender of the appropriate type to be owned by this transceiver.
>   // Must not be null.
>@@ -177,7 +177,7 @@ class RtpTransceiver final
>   void SetCodecPreferences(rtc::ArrayView<RtpCodecCapability> codecs) override;
> 
>  private:
>-  void OnFirstPacketReceived(cricket::BaseChannel* channel);
>+  void OnFirstPacketReceived(cricket::ChannelInterface* channel);
> 
>   const bool unified_plan_;
>   const cricket::MediaType media_type_;
>@@ -196,7 +196,7 @@ class RtpTransceiver final
>   bool created_by_addtrack_ = false;
>   bool has_ever_been_used_to_send_ = false;
> 
>-  cricket::BaseChannel* channel_ = nullptr;
>+  cricket::ChannelInterface* channel_ = nullptr;
> };
> 
> BEGIN_SIGNALING_PROXY_MAP(RtpTransceiver)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..b57d2123d152fb2bf335a896965e7fe61052f587
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver_unittest.cc
>@@ -0,0 +1,71 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// This file contains tests for |RtpTransceiver|.
>+
>+#include "pc/rtptransceiver.h"
>+#include "rtc_base/gunit.h"
>+#include "test/gmock.h"
>+#include "test/mock_channelinterface.h"
>+
>+using ::testing::Return;
>+using ::testing::ReturnRef;
>+
>+namespace webrtc {
>+
>+// Checks that a channel cannot be set on a stopped |RtpTransceiver|.
>+TEST(RtpTransceiverTest, CannotSetChannelOnStoppedTransceiver) {
>+  RtpTransceiver transceiver(cricket::MediaType::MEDIA_TYPE_AUDIO);
>+  cricket::MockChannelInterface channel1;
>+  sigslot::signal1<cricket::ChannelInterface*> signal;
>+  EXPECT_CALL(channel1, media_type())
>+      .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
>+  EXPECT_CALL(channel1, SignalFirstPacketReceived())
>+      .WillRepeatedly(ReturnRef(signal));
>+
>+  transceiver.SetChannel(&channel1);
>+  EXPECT_EQ(&channel1, transceiver.channel());
>+
>+  // Stop the transceiver.
>+  transceiver.Stop();
>+  EXPECT_EQ(&channel1, transceiver.channel());
>+
>+  cricket::MockChannelInterface channel2;
>+  EXPECT_CALL(channel2, media_type())
>+      .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_AUDIO));
>+
>+  // Channel can no longer be set, so this call should be a no-op.
>+  transceiver.SetChannel(&channel2);
>+  EXPECT_EQ(&channel1, transceiver.channel());
>+}
>+
>+// Checks that a channel can be unset on a stopped |RtpTransceiver|
>+TEST(RtpTransceiverTest, CanUnsetChannelOnStoppedTransceiver) {
>+  RtpTransceiver transceiver(cricket::MediaType::MEDIA_TYPE_VIDEO);
>+  cricket::MockChannelInterface channel;
>+  sigslot::signal1<cricket::ChannelInterface*> signal;
>+  EXPECT_CALL(channel, media_type())
>+      .WillRepeatedly(Return(cricket::MediaType::MEDIA_TYPE_VIDEO));
>+  EXPECT_CALL(channel, SignalFirstPacketReceived())
>+      .WillRepeatedly(ReturnRef(signal));
>+
>+  transceiver.SetChannel(&channel);
>+  EXPECT_EQ(&channel, transceiver.channel());
>+
>+  // Stop the transceiver.
>+  transceiver.Stop();
>+  EXPECT_EQ(&channel, transceiver.channel());
>+
>+  // Set the channel to |nullptr|.
>+  transceiver.SetChannel(nullptr);
>+  EXPECT_EQ(nullptr, transceiver.channel());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.cc
>index 9e994e9b9804a51ac56b6797a567d0af7136ecd0..4a573016f21e87bcc8b21e0fa71eeb846481d77b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.cc
>@@ -187,7 +187,7 @@ RtpTransportParameters RtpTransport::GetParameters() const {
> }
> 
> void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer* packet,
>-                               const rtc::PacketTime& time) {
>+                               int64_t packet_time_us) {
>   webrtc::RtpPacketReceived parsed_packet(&header_extension_map_);
>   if (!parsed_packet.Parse(std::move(*packet))) {
>     RTC_LOG(LS_ERROR)
>@@ -195,8 +195,8 @@ void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer* packet,
>     return;
>   }
> 
>-  if (time.timestamp != -1) {
>-    parsed_packet.set_arrival_time_ms((time.timestamp + 500) / 1000);
>+  if (packet_time_us != -1) {
>+    parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
>   }
>   rtp_demuxer_.OnRtpPacket(parsed_packet);
> }
>@@ -236,19 +236,19 @@ void RtpTransport::OnSentPacket(rtc::PacketTransportInternal* packet_transport,
> }
> 
> void RtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                       const rtc::PacketTime& packet_time) {
>-  DemuxPacket(packet, packet_time);
>+                                       int64_t packet_time_us) {
>+  DemuxPacket(packet, packet_time_us);
> }
> 
> void RtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                        const rtc::PacketTime& packet_time) {
>-  SignalRtcpPacketReceived(packet, packet_time);
>+                                        int64_t packet_time_us) {
>+  SignalRtcpPacketReceived(packet, packet_time_us);
> }
> 
> void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
>                                 const char* data,
>                                 size_t len,
>-                                const rtc::PacketTime& packet_time,
>+                                const int64_t& packet_time_us,
>                                 int flags) {
>   TRACE_EVENT0("webrtc", "RtpTransport::OnReadPacket");
> 
>@@ -272,9 +272,9 @@ void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
>   }
> 
>   if (rtcp) {
>-    OnRtcpPacketReceived(&packet, packet_time);
>+    OnRtcpPacketReceived(&packet, packet_time_us);
>   } else {
>-    OnRtpPacketReceived(&packet, packet_time);
>+    OnRtpPacketReceived(&packet, packet_time_us);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.h
>index 793ac901aa5c8cb48cf2cf33e5aeb9bd46b9d098..dab5903171558ef7793cf9cfe8f9b44d42793cf7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.h
>@@ -87,7 +87,7 @@ class RtpTransport : public RtpTransportInternal {
>   RtpTransportAdapter* GetInternal() override;
> 
>   // These methods will be used in the subclasses.
>-  void DemuxPacket(rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& time);
>+  void DemuxPacket(rtc::CopyOnWriteBuffer* packet, int64_t packet_time_us);
> 
>   bool SendPacket(bool rtcp,
>                   rtc::CopyOnWriteBuffer* packet,
>@@ -98,9 +98,9 @@ class RtpTransport : public RtpTransportInternal {
>   virtual void OnNetworkRouteChanged(
>       absl::optional<rtc::NetworkRoute> network_route);
>   virtual void OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                   const rtc::PacketTime& packet_time);
>+                                   int64_t packet_time_us);
>   virtual void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                    const rtc::PacketTime& packet_time);
>+                                    int64_t packet_time_us);
>   // Overridden by SrtpTransport and DtlsSrtpTransport.
>   virtual void OnWritableState(rtc::PacketTransportInternal* packet_transport);
> 
>@@ -111,7 +111,7 @@ class RtpTransport : public RtpTransportInternal {
>   void OnReadPacket(rtc::PacketTransportInternal* transport,
>                     const char* data,
>                     size_t len,
>-                    const rtc::PacketTime& packet_time,
>+                    const int64_t& packet_time_us,
>                     int flags);
> 
>   // Updates "ready to send" for an individual channel and fires
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternal.h
>index ca0568071086ce51ad1680b62cf29e7ec0e59edd..e6338366c0637ecf06d3b28fe31e19e657745d0a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternal.h
>@@ -24,7 +24,6 @@
> namespace rtc {
> class CopyOnWriteBuffer;
> struct PacketOptions;
>-struct PacketTime;
> }  // namespace rtc
> 
> namespace webrtc {
>@@ -59,8 +58,7 @@ class RtpTransportInternal : public SrtpTransportInterface,
>   // Called whenever an RTCP packet is received. There is no equivalent signal
>   // for RTP packets because they would be forwarded to the BaseChannel through
>   // the RtpDemuxer callback.
>-  sigslot::signal2<rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
>-      SignalRtcpPacketReceived;
>+  sigslot::signal2<rtc::CopyOnWriteBuffer*, int64_t> SignalRtcpPacketReceived;
> 
>   // Called whenever the network route of the P2P layer transport changes.
>   // The argument is an optional network route.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransporttestutil.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransporttestutil.h
>index eb337730d107f8c8d81a8d28a2aa04f32e602254..f2c35122aaf44a347267fad37f9f225c2447638f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransporttestutil.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransporttestutil.h
>@@ -39,7 +39,7 @@ class TransportObserver : public RtpPacketSinkInterface,
>   }
> 
>   void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                            const rtc::PacketTime& packet_time) {
>+                            int64_t packet_time_us) {
>     rtcp_count_++;
>     last_recv_rtcp_packet_ = *packet;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.cc
>index f3e89b47fadb66c000011dc888f2cc3634b9957d..3a9b18ee6919b37ff5940ee6064eccb7b46a8d4d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.cc
>@@ -84,21 +84,6 @@ bool ContentGroup::RemoveContentName(const std::string& content_name) {
> }
> 
> SessionDescription::SessionDescription() = default;
>-
>-SessionDescription::SessionDescription(const ContentInfos& contents)
>-    : contents_(contents) {}
>-
>-SessionDescription::SessionDescription(const ContentInfos& contents,
>-                                       const ContentGroups& groups)
>-    : contents_(contents), content_groups_(groups) {}
>-
>-SessionDescription::SessionDescription(const ContentInfos& contents,
>-                                       const TransportInfos& transports,
>-                                       const ContentGroups& groups)
>-    : contents_(contents),
>-      transport_infos_(transports),
>-      content_groups_(groups) {}
>-
> SessionDescription::SessionDescription(const SessionDescription&) = default;
> 
> SessionDescription::~SessionDescription() {
>@@ -162,7 +147,7 @@ void SessionDescription::AddContent(const std::string& name,
>   ContentInfo content(type);
>   content.name = name;
>   content.description = description;
>-  contents_.push_back(std::move(content));
>+  AddContent(&content);
> }
> 
> void SessionDescription::AddContent(const std::string& name,
>@@ -173,7 +158,7 @@ void SessionDescription::AddContent(const std::string& name,
>   content.name = name;
>   content.rejected = rejected;
>   content.description = description;
>-  contents_.push_back(std::move(content));
>+  AddContent(&content);
> }
> 
> void SessionDescription::AddContent(const std::string& name,
>@@ -186,7 +171,16 @@ void SessionDescription::AddContent(const std::string& name,
>   content.rejected = rejected;
>   content.bundle_only = bundle_only;
>   content.description = description;
>-  contents_.push_back(std::move(content));
>+  AddContent(&content);
>+}
>+
>+void SessionDescription::AddContent(ContentInfo* content) {
>+  if (extmap_allow_mixed()) {
>+    // Mixed support on session level overrides setting on media level.
>+    content->description->set_extmap_allow_mixed_enum(
>+        MediaContentDescription::kSession);
>+  }
>+  contents_.push_back(std::move(*content));
> }
> 
> bool SessionDescription::RemoveContentByName(const std::string& name) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h
>index 645ebc54f86e57974640af7bd59c86cea0776272..38291484cd8cb4e8888737a78d6b0f82154e80f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h
>@@ -183,6 +183,22 @@ class MediaContentDescription {
>     return connection_address_;
>   }
> 
>+  // Determines if it's allowed to mix one- and two-byte rtp header extensions
>+  // within the same rtp stream.
>+  enum ExtmapAllowMixed { kNo, kSession, kMedia };
>+  void set_extmap_allow_mixed_enum(ExtmapAllowMixed new_extmap_allow_mixed) {
>+    if (new_extmap_allow_mixed == kMedia &&
>+        extmap_allow_mixed_enum_ == kSession) {
>+      // Do not downgrade from session level to media level.
>+      return;
>+    }
>+    extmap_allow_mixed_enum_ = new_extmap_allow_mixed;
>+  }
>+  ExtmapAllowMixed extmap_allow_mixed_enum() const {
>+    return extmap_allow_mixed_enum_;
>+  }
>+  bool extmap_allow_mixed() const { return extmap_allow_mixed_enum_ != kNo; }
>+
>  protected:
>   bool rtcp_mux_ = false;
>   bool rtcp_reduced_size_ = false;
>@@ -196,6 +212,10 @@ class MediaContentDescription {
>   webrtc::RtpTransceiverDirection direction_ =
>       webrtc::RtpTransceiverDirection::kSendRecv;
>   rtc::SocketAddress connection_address_;
>+  // Mixed one- and two-byte header not included in offer on media level or
>+  // session level, but we will respond that we support it. The plan is to add
>+  // it to our offer on session level. See todo in SessionDescription.
>+  ExtmapAllowMixed extmap_allow_mixed_enum_ = kNo;
> };
> 
> // TODO(bugs.webrtc.org/8620): Remove this alias once downstream projects have
>@@ -378,11 +398,6 @@ enum MsidSignaling {
> class SessionDescription {
>  public:
>   SessionDescription();
>-  explicit SessionDescription(const ContentInfos& contents);
>-  SessionDescription(const ContentInfos& contents, const ContentGroups& groups);
>-  SessionDescription(const ContentInfos& contents,
>-                     const TransportInfos& transports,
>-                     const ContentGroups& groups);
>   ~SessionDescription();
> 
>   SessionDescription* Copy() const;
>@@ -412,6 +427,8 @@ class SessionDescription {
>                   bool rejected,
>                   bool bundle_only,
>                   MediaContentDescription* description);
>+  void AddContent(ContentInfo* content);
>+
>   bool RemoveContentByName(const std::string& name);
> 
>   // Transport accessors.
>@@ -455,6 +472,24 @@ class SessionDescription {
>   }
>   int msid_signaling() const { return msid_signaling_; }
> 
>+  // Determines if it's allowed to mix one- and two-byte rtp header extensions
>+  // within the same rtp stream.
>+  void set_extmap_allow_mixed(bool supported) {
>+    extmap_allow_mixed_ = supported;
>+    MediaContentDescription::ExtmapAllowMixed media_level_setting =
>+        supported ? MediaContentDescription::kSession
>+                  : MediaContentDescription::kNo;
>+    for (auto& content : contents_) {
>+      // Do not set to kNo if the current setting is kMedia.
>+      if (supported || content.media_description()->extmap_allow_mixed_enum() !=
>+                           MediaContentDescription::kMedia) {
>+        content.media_description()->set_extmap_allow_mixed_enum(
>+            media_level_setting);
>+      }
>+    }
>+  }
>+  bool extmap_allow_mixed() const { return extmap_allow_mixed_; }
>+
>  private:
>   SessionDescription(const SessionDescription&);
> 
>@@ -465,6 +500,12 @@ class SessionDescription {
>   // Default to what Plan B would do.
>   // TODO(bugs.webrtc.org/8530): Change default to kMsidSignalingMediaSection.
>   int msid_signaling_ = kMsidSignalingSsrcAttribute;
>+  // TODO(webrtc:9985): Activate mixed one- and two-byte header extension in
>+  // offer at session level. It's currently not included in offer by default
>+  // because clients prior to https://bugs.webrtc.org/9712 cannot parse this
>+  // correctly. If it's included in offer to us we will respond that we support
>+  // it.
>+  bool extmap_allow_mixed_ = false;
> };
> 
> // Indicates whether a session description was sent by the local client or
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..dcacf587f562b4f9a5267b4ac275add788e2731b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription_unittest.cc
>@@ -0,0 +1,130 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "pc/sessiondescription.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace cricket {
>+
>+TEST(MediaContentDescriptionTest, ExtmapAllowMixedDefaultValue) {
>+  VideoContentDescription video_desc;
>+  EXPECT_EQ(MediaContentDescription::kNo, video_desc.extmap_allow_mixed_enum());
>+}
>+
>+TEST(MediaContentDescriptionTest, SetExtmapAllowMixed) {
>+  VideoContentDescription video_desc;
>+  video_desc.set_extmap_allow_mixed_enum(MediaContentDescription::kNo);
>+  EXPECT_EQ(MediaContentDescription::kNo, video_desc.extmap_allow_mixed_enum());
>+  video_desc.set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
>+  EXPECT_EQ(MediaContentDescription::kMedia,
>+            video_desc.extmap_allow_mixed_enum());
>+  video_desc.set_extmap_allow_mixed_enum(MediaContentDescription::kSession);
>+  EXPECT_EQ(MediaContentDescription::kSession,
>+            video_desc.extmap_allow_mixed_enum());
>+
>+  // Not allowed to downgrade from kSession to kMedia.
>+  video_desc.set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
>+  EXPECT_EQ(MediaContentDescription::kSession,
>+            video_desc.extmap_allow_mixed_enum());
>+
>+  // Always okay to set not supported.
>+  video_desc.set_extmap_allow_mixed_enum(MediaContentDescription::kNo);
>+  EXPECT_EQ(MediaContentDescription::kNo, video_desc.extmap_allow_mixed_enum());
>+  video_desc.set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
>+  EXPECT_EQ(MediaContentDescription::kMedia,
>+            video_desc.extmap_allow_mixed_enum());
>+  video_desc.set_extmap_allow_mixed_enum(MediaContentDescription::kNo);
>+  EXPECT_EQ(MediaContentDescription::kNo, video_desc.extmap_allow_mixed_enum());
>+}
>+
>+TEST(MediaContentDescriptionTest, MixedOneTwoByteHeaderSupported) {
>+  VideoContentDescription video_desc;
>+  video_desc.set_extmap_allow_mixed_enum(MediaContentDescription::kNo);
>+  EXPECT_FALSE(video_desc.extmap_allow_mixed());
>+  video_desc.set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
>+  EXPECT_TRUE(video_desc.extmap_allow_mixed());
>+  video_desc.set_extmap_allow_mixed_enum(MediaContentDescription::kSession);
>+  EXPECT_TRUE(video_desc.extmap_allow_mixed());
>+}
>+
>+TEST(SessionDescriptionTest, SetExtmapAllowMixed) {
>+  SessionDescription session_desc;
>+  session_desc.set_extmap_allow_mixed(true);
>+  EXPECT_TRUE(session_desc.extmap_allow_mixed());
>+  session_desc.set_extmap_allow_mixed(false);
>+  EXPECT_FALSE(session_desc.extmap_allow_mixed());
>+}
>+
>+TEST(SessionDescriptionTest, SetExtmapAllowMixedPropagatesToMediaLevel) {
>+  SessionDescription session_desc;
>+  MediaContentDescription* video_desc = new VideoContentDescription();
>+  session_desc.AddContent("video", MediaProtocolType::kRtp, video_desc);
>+
>+  // Setting true on session level propagates to media level.
>+  session_desc.set_extmap_allow_mixed(true);
>+  EXPECT_EQ(MediaContentDescription::kSession,
>+            video_desc->extmap_allow_mixed_enum());
>+
>+  // Don't downgrade from session level to media level
>+  video_desc->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
>+  EXPECT_EQ(MediaContentDescription::kSession,
>+            video_desc->extmap_allow_mixed_enum());
>+
>+  // Setting false on session level propagates to media level if the current
>+  // state is kSession.
>+  session_desc.set_extmap_allow_mixed(false);
>+  EXPECT_EQ(MediaContentDescription::kNo,
>+            video_desc->extmap_allow_mixed_enum());
>+
>+  // Now possible to set at media level.
>+  video_desc->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
>+  EXPECT_EQ(MediaContentDescription::kMedia,
>+            video_desc->extmap_allow_mixed_enum());
>+
>+  // Setting false on session level does not override on media level if current
>+  // state is kMedia.
>+  session_desc.set_extmap_allow_mixed(false);
>+  EXPECT_EQ(MediaContentDescription::kMedia,
>+            video_desc->extmap_allow_mixed_enum());
>+
>+  // Setting true on session level overrides setting on media level.
>+  session_desc.set_extmap_allow_mixed(true);
>+  EXPECT_EQ(MediaContentDescription::kSession,
>+            video_desc->extmap_allow_mixed_enum());
>+}
>+
>+TEST(SessionDescriptionTest, AddContentTransfersExtmapAllowMixedSetting) {
>+  SessionDescription session_desc;
>+  session_desc.set_extmap_allow_mixed(false);
>+  MediaContentDescription* audio_desc = new AudioContentDescription();
>+  audio_desc->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
>+
>+  // If session setting is false, media level setting is preserved when new
>+  // content is added.
>+  session_desc.AddContent("audio", MediaProtocolType::kRtp, audio_desc);
>+  EXPECT_EQ(MediaContentDescription::kMedia,
>+            audio_desc->extmap_allow_mixed_enum());
>+
>+  // If session setting is true, it's transferred to media level when new
>+  // content is added.
>+  session_desc.set_extmap_allow_mixed(true);
>+  MediaContentDescription* video_desc = new VideoContentDescription();
>+  session_desc.AddContent("video", MediaProtocolType::kRtp, video_desc);
>+  EXPECT_EQ(MediaContentDescription::kSession,
>+            video_desc->extmap_allow_mixed_enum());
>+
>+  // Session level setting overrides media level when new content is added.
>+  MediaContentDescription* data_desc = new DataContentDescription;
>+  data_desc->set_extmap_allow_mixed_enum(MediaContentDescription::kMedia);
>+  session_desc.AddContent("data", MediaProtocolType::kRtp, data_desc);
>+  EXPECT_EQ(MediaContentDescription::kSession,
>+            data_desc->extmap_allow_mixed_enum());
>+}
>+
>+}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.cc
>index 5ffe34212e508ba73a00e3f86948ae507ff62e10..cd0393a03e331b1d73cff4a2259e23ae7a52f463 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.cc
>@@ -19,7 +19,6 @@
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/stringencode.h"
> #include "rtc_base/third_party/base64/base64.h"
> #include "rtc_base/timeutils.h"
> #include "rtc_base/zero_memory.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.h
>index a4dd54fa8bdebca68e19c0fddd0839b6910e69d5..4ab0dd72c1a412c1423c8c0245ade794ea8ef898 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.h
>@@ -78,6 +78,10 @@ class SrtpFilter {
> 
>   bool ResetParams();
> 
>+  static bool ParseKeyParams(const std::string& params,
>+                             uint8_t* key,
>+                             size_t len);
>+
>   absl::optional<int> send_cipher_suite() { return send_cipher_suite_; }
>   absl::optional<int> recv_cipher_suite() { return recv_cipher_suite_; }
> 
>@@ -104,10 +108,6 @@ class SrtpFilter {
> 
>   bool ApplyRecvParams(const CryptoParams& recv_params);
> 
>-  static bool ParseKeyParams(const std::string& params,
>-                             uint8_t* key,
>-                             size_t len);
>-
>   enum State {
>     ST_INIT,                    // SRTP filter unused.
>     ST_SENTOFFER,               // Offer with SRTP parameters sent.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.cc
>index a5fda8897feb9d726295d5e5378de456fcf0b537..aadd4711a503e1e74733f3f84625e0a811d82f27 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.cc
>@@ -138,7 +138,15 @@ bool SrtpSession::UnprotectRtp(void* p, int in_len, int* out_len) {
>   *out_len = in_len;
>   int err = srtp_unprotect(session_, p, out_len);
>   if (err != srtp_err_status_ok) {
>-    RTC_LOG(LS_WARNING) << "Failed to unprotect SRTP packet, err=" << err;
>+    // Limit the error logging to avoid excessive logs when there are lots of
>+    // bad packets.
>+    const int kFailureLogThrottleCount = 100;
>+    if (decryption_failure_count_ % kFailureLogThrottleCount == 0) {
>+      RTC_LOG(LS_WARNING) << "Failed to unprotect SRTP packet, err=" << err
>+                          << ", previous failure count: "
>+                          << decryption_failure_count_;
>+    }
>+    ++decryption_failure_count_;
>     RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtpUnprotectError",
>                               static_cast<int>(err), kSrtpErrorCodeBoundary);
>     return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.h
>index c3a01bd139bb0a536cbf8d9774c4e2bfa7e76193..ce05ce6d7258622c84973b2f771ddc6bb0f3d49c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.h
>@@ -122,6 +122,7 @@ class SrtpSession {
>   int last_send_seq_num_ = -1;
>   bool external_auth_active_ = false;
>   bool external_auth_enabled_ = false;
>+  int decryption_failure_count_ = 0;
>   RTC_DISALLOW_COPY_AND_ASSIGN(SrtpSession);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.cc
>index 6fd8be37c242481b9c9789930be43cdda1470f3b..0cf4106203e635150de7fe898fc4a917568d3e3b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.cc
>@@ -193,7 +193,7 @@ bool SrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
> }
> 
> void SrtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                        const rtc::PacketTime& packet_time) {
>+                                        int64_t packet_time_us) {
>   if (!IsSrtpActive()) {
>     RTC_LOG(LS_WARNING)
>         << "Inactive SRTP transport received an RTP packet. Drop it.";
>@@ -207,16 +207,25 @@ void SrtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>     uint32_t ssrc = 0;
>     cricket::GetRtpSeqNum(data, len, &seq_num);
>     cricket::GetRtpSsrc(data, len, &ssrc);
>-    RTC_LOG(LS_ERROR) << "Failed to unprotect RTP packet: size=" << len
>-                      << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
>+
>+    // Limit the error logging to avoid excessive logs when there are lots of
>+    // bad packets.
>+    const int kFailureLogThrottleCount = 100;
>+    if (decryption_failure_count_ % kFailureLogThrottleCount == 0) {
>+      RTC_LOG(LS_ERROR) << "Failed to unprotect RTP packet: size=" << len
>+                        << ", seqnum=" << seq_num << ", SSRC=" << ssrc
>+                        << ", previous failure count: "
>+                        << decryption_failure_count_;
>+    }
>+    ++decryption_failure_count_;
>     return;
>   }
>   packet->SetSize(len);
>-  DemuxPacket(packet, packet_time);
>+  DemuxPacket(packet, packet_time_us);
> }
> 
> void SrtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                                         const rtc::PacketTime& packet_time) {
>+                                         int64_t packet_time_us) {
>   if (!IsSrtpActive()) {
>     RTC_LOG(LS_WARNING)
>         << "Inactive SRTP transport received an RTCP packet. Drop it.";
>@@ -233,7 +242,7 @@ void SrtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>     return;
>   }
>   packet->SetSize(len);
>-  SignalRtcpPacketReceived(packet, packet_time);
>+  SignalRtcpPacketReceived(packet, packet_time_us);
> }
> 
> void SrtpTransport::OnNetworkRouteChanged(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.h
>index 198024f5a54ca4d9414575088384b46e98ad995d..c62359b944d48723990fa005039f1a3dc3421c86 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.h
>@@ -113,9 +113,9 @@ class SrtpTransport : public RtpTransport {
>   void CreateSrtpSessions();
> 
>   void OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                           const rtc::PacketTime& packet_time) override;
>+                           int64_t packet_time_us) override;
>   void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>-                            const rtc::PacketTime& packet_time) override;
>+                            int64_t packet_time_us) override;
>   void OnNetworkRouteChanged(
>       absl::optional<rtc::NetworkRoute> network_route) override;
> 
>@@ -160,6 +160,8 @@ class SrtpTransport : public RtpTransport {
>   bool external_auth_enabled_ = false;
> 
>   int rtp_abs_sendtime_extn_id_ = -1;
>+
>+  int decryption_failure_count_ = 0;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc
>index 23a449308c317e91110cad840400f2db35d78120..a1d52db04e92de8d457c93a0689a85221acd7140 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc
>@@ -783,8 +783,8 @@ void StatsCollector::ExtractSessionInfo() {
>     StatsReport::Id local_cert_report_id, remote_cert_report_id;
>     rtc::scoped_refptr<rtc::RTCCertificate> certificate;
>     if (pc_->GetLocalCertificate(transport_name, &certificate)) {
>-      StatsReport* r =
>-          AddCertificateReports(certificate->ssl_cert_chain().GetStats());
>+      StatsReport* r = AddCertificateReports(
>+          certificate->GetSSLCertificateChain().GetStats());
>       if (r)
>         local_cert_report_id = r->id();
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector_unittest.cc
>index fa4d3db81bd7c576f75cd1977294e36ddf251f89..53eb2eaa5eaba4677205bd3a9416a3420a6767d1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector_unittest.cc
>@@ -22,6 +22,7 @@
> #include "pc/test/fakevideotracksource.h"
> #include "pc/videotrack.h"
> #include "rtc_base/fakesslidentity.h"
>+#include "rtc_base/messagedigest.h"
> #include "rtc_base/third_party/base64/base64.h"
> #include "test/gtest.h"
> 
>@@ -56,14 +57,6 @@ class FakeAudioProcessor : public AudioProcessorInterface {
>   ~FakeAudioProcessor() {}
> 
>  private:
>-  void GetStats(AudioProcessorInterface::AudioProcessorStats* stats) override {
>-    stats->typing_noise_detected = true;
>-    stats->echo_return_loss = 2;
>-    stats->echo_return_loss_enhancement = 3;
>-    stats->echo_delay_median_ms = 4;
>-    stats->echo_delay_std_ms = 6;
>-  }
>-
>   AudioProcessorInterface::AudioProcessorStatistics GetStats(
>       bool has_recv_streams) override {
>     AudioProcessorStatistics stats;
>@@ -107,14 +100,6 @@ class FakeAudioProcessorWithInitValue : public AudioProcessorInterface {
>   ~FakeAudioProcessorWithInitValue() {}
> 
>  private:
>-  void GetStats(AudioProcessorInterface::AudioProcessorStats* stats) override {
>-    stats->typing_noise_detected = false;
>-    stats->echo_return_loss = -100;
>-    stats->echo_return_loss_enhancement = -100;
>-    stats->echo_delay_median_ms = -1;
>-    stats->echo_delay_std_ms = -1;
>-  }
>-
>   AudioProcessorInterface::AudioProcessorStatistics GetStats(
>       bool /*has_recv_streams*/) override {
>     AudioProcessorStatistics stats;
>@@ -494,10 +479,10 @@ void InitVoiceSenderInfo(cricket::VoiceSenderInfo* voice_sender_info) {
>   voice_sender_info->packets_lost = 105;
>   voice_sender_info->ext_seqnum = 106;
>   voice_sender_info->audio_level = 107;
>-  voice_sender_info->echo_return_loss = 108;
>-  voice_sender_info->echo_return_loss_enhancement = 109;
>-  voice_sender_info->echo_delay_median_ms = 110;
>-  voice_sender_info->echo_delay_std_ms = 111;
>+  voice_sender_info->apm_statistics.echo_return_loss = 108;
>+  voice_sender_info->apm_statistics.echo_return_loss_enhancement = 109;
>+  voice_sender_info->apm_statistics.delay_median_ms = 110;
>+  voice_sender_info->apm_statistics.delay_standard_deviation_ms = 111;
>   voice_sender_info->typing_noise_detected = false;
>   voice_sender_info->ana_statistics.bitrate_action_counter = 112;
>   voice_sender_info->ana_statistics.channel_action_counter = 113;
>@@ -641,7 +626,7 @@ class StatsCollectorTest : public testing::Test {
>             std::unique_ptr<rtc::SSLIdentity>(local_identity.GetReference())));
>     pc->SetLocalCertificate(kTransportName, local_certificate);
>     pc->SetRemoteCertChain(kTransportName,
>-                           remote_identity.cert_chain().UniqueCopy());
>+                           remote_identity.cert_chain().Clone());
> 
>     stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
> 
>@@ -1567,7 +1552,7 @@ TEST_P(StatsCollectorTrackTest, GetStatsAfterRemoveAudioStream) {
> 
>   // Verifies the values in the track report, no value will be changed by the
>   // AudioTrackInterface::GetSignalValue() and
>-  // AudioProcessorInterface::AudioProcessorStats::GetStats();
>+  // AudioProcessorInterface::GetStats();
>   VerifyVoiceSenderInfoReport(report, voice_sender_info);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakepeerconnectionforstats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakepeerconnectionforstats.h
>index 642fa011eb4e88096c8249e26c20a572fe9e77d2..af86639eca4d59849f6d5ee89a451e4b6400b217 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakepeerconnectionforstats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakepeerconnectionforstats.h
>@@ -126,7 +126,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
>     voice_channel_ = absl::make_unique<cricket::VoiceChannel>(
>         worker_thread_, network_thread_, signaling_thread_, nullptr,
>         std::move(voice_media_channel), mid, kDefaultSrtpRequired,
>-        rtc::CryptoOptions());
>+        webrtc::CryptoOptions());
>     voice_channel_->set_transport_name_for_testing(transport_name);
>     GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
>         ->internal()
>@@ -144,7 +144,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
>     video_channel_ = absl::make_unique<cricket::VideoChannel>(
>         worker_thread_, network_thread_, signaling_thread_,
>         std::move(video_media_channel), mid, kDefaultSrtpRequired,
>-        rtc::CryptoOptions());
>+        webrtc::CryptoOptions());
>     video_channel_->set_transport_name_for_testing(transport_name);
>     GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
>         ->internal()
>@@ -319,7 +319,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
>       const std::string& transport_name) override {
>     auto it = remote_cert_chains_by_transport_.find(transport_name);
>     if (it != remote_cert_chains_by_transport_.end()) {
>-      return it->second->UniqueCopy();
>+      return it->second->Clone();
>     } else {
>       return nullptr;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_channelinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_channelinterface.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..4c32a8fb8d979c05d99004cd1d44c42bbf4edd60
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_channelinterface.h
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_TEST_MOCK_CHANNELINTERFACE_H_
>+#define PC_TEST_MOCK_CHANNELINTERFACE_H_
>+
>+#include <string>
>+
>+#include "pc/channelinterface.h"
>+#include "test/gmock.h"
>+
>+namespace cricket {
>+
>+// Mock class for BaseChannel.
>+// Use this class in unit tests to avoid dependecy on a specific
>+// implementation of BaseChannel.
>+class MockChannelInterface : public cricket::ChannelInterface {
>+ public:
>+  MOCK_CONST_METHOD0(media_type, cricket::MediaType());
>+  MOCK_CONST_METHOD0(media_channel, MediaChannel*());
>+  MOCK_CONST_METHOD0(transport_name, const std::string&());
>+  MOCK_CONST_METHOD0(content_name, const std::string&());
>+  MOCK_CONST_METHOD0(enabled, bool());
>+  MOCK_METHOD1(Enable, bool(bool));
>+  MOCK_METHOD0(SignalFirstPacketReceived,
>+               sigslot::signal1<ChannelInterface*>&());
>+  MOCK_METHOD3(SetLocalContent,
>+               bool(const cricket::MediaContentDescription*,
>+                    webrtc::SdpType,
>+                    std::string*));
>+  MOCK_METHOD3(SetRemoteContent,
>+               bool(const cricket::MediaContentDescription*,
>+                    webrtc::SdpType,
>+                    std::string*));
>+  MOCK_METHOD1(SetRtpTransport, bool(webrtc::RtpTransportInternal*));
>+};
>+
>+}  // namespace cricket
>+
>+#endif  // PC_TEST_MOCK_CHANNELINTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpreceiverinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpreceiverinternal.h
>index 850d5a9f8e5990d010fd135c98e055101bccca1a..84b0d71766b9705e55b5eb408dbb81c82455664c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpreceiverinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpreceiverinternal.h
>@@ -41,8 +41,7 @@ class MockRtpReceiverInternal : public RtpReceiverInternal {
> 
>   // RtpReceiverInternal methods.
>   MOCK_METHOD0(Stop, void());
>-  MOCK_METHOD1(SetVoiceMediaChannel, void(cricket::VoiceMediaChannel*));
>-  MOCK_METHOD1(SetVideoMediaChannel, void(cricket::VideoMediaChannel*));
>+  MOCK_METHOD1(SetMediaChannel, void(cricket::MediaChannel*));
>   MOCK_METHOD1(SetupMediaChannel, void(uint32_t));
>   MOCK_CONST_METHOD0(ssrc, uint32_t());
>   MOCK_METHOD0(NotifyFirstPacketReceived, void());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpsenderinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpsenderinternal.h
>index 5e14f7ad7de625c48b2a8de2582662f465b08269..dd4f69497df41291610c157ac370dd9d3c0a75a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpsenderinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpsenderinternal.h
>@@ -39,8 +39,7 @@ class MockRtpSenderInternal : public RtpSenderInternal {
>                      rtc::scoped_refptr<FrameEncryptorInterface>());
> 
>   // RtpSenderInternal methods.
>-  MOCK_METHOD1(SetVoiceMediaChannel, void(cricket::VoiceMediaChannel*));
>-  MOCK_METHOD1(SetVideoMediaChannel, void(cricket::VideoMediaChannel*));
>+  MOCK_METHOD1(SetMediaChannel, void(cricket::MediaChannel*));
>   MOCK_METHOD1(SetSsrc, void(uint32_t));
>   MOCK_METHOD1(set_stream_ids, void(const std::vector<std::string>&));
>   MOCK_METHOD1(set_init_send_encodings,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mockpeerconnectionobservers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mockpeerconnectionobservers.h
>index 6f5e9bc665259e710b368cdb4d6e59908c0620f8..62c0c9415f6c4183bece1278870e2b067ad43903 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mockpeerconnectionobservers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mockpeerconnectionobservers.h
>@@ -67,6 +67,7 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
>   }
>   void OnSignalingChange(
>       PeerConnectionInterface::SignalingState new_state) override {
>+    RTC_DCHECK(pc_);
>     RTC_DCHECK(pc_->signaling_state() == new_state);
>     state_ = new_state;
>   }
>@@ -92,6 +93,7 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
> 
>   void OnIceConnectionChange(
>       PeerConnectionInterface::IceConnectionState new_state) override {
>+    RTC_DCHECK(pc_);
>     RTC_DCHECK(pc_->ice_connection_state() == new_state);
>     // When ICE is finished, the caller will get to a kIceConnectionCompleted
>     // state, because it has the ICE controlling role, while the callee
>@@ -104,12 +106,14 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
>   }
>   void OnIceGatheringChange(
>       PeerConnectionInterface::IceGatheringState new_state) override {
>+    RTC_DCHECK(pc_);
>     RTC_DCHECK(pc_->ice_gathering_state() == new_state);
>     ice_gathering_complete_ =
>         new_state == PeerConnectionInterface::kIceGatheringComplete;
>     callback_triggered_ = true;
>   }
>   void OnIceCandidate(const IceCandidateInterface* candidate) override {
>+    RTC_DCHECK(pc_);
>     RTC_DCHECK(PeerConnectionInterface::kIceGatheringNew !=
>                pc_->ice_gathering_state());
>     candidates_.push_back(absl::make_unique<JsepIceCandidate>(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.cc
>index 9be1596373c681ad420c5688d80c56003373a7f4..a1db9ed9e6f30c3e072695494e08c5a68f9eb094 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.cc
>@@ -12,6 +12,7 @@
> #include <utility>
> #include <vector>
> 
>+#include "api/create_peerconnection_factory.h"
> #include "api/video_codecs/builtin_video_decoder_factory.h"
> #include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "modules/audio_processing/include/audio_processing.h"
>@@ -22,6 +23,7 @@
> #include "pc/test/mockpeerconnectionobservers.h"
> #include "pc/test/peerconnectiontestwrapper.h"
> #include "rtc_base/gunit.h"
>+#include "rtc_base/thread_checker.h"
> #include "rtc_base/timeutils.h"
> 
> using webrtc::FakeConstraints;
>@@ -64,9 +66,20 @@ PeerConnectionTestWrapper::PeerConnectionTestWrapper(
>     rtc::Thread* worker_thread)
>     : name_(name),
>       network_thread_(network_thread),
>-      worker_thread_(worker_thread) {}
>+      worker_thread_(worker_thread) {
>+  pc_thread_checker_.DetachFromThread();
>+}
> 
>-PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {}
>+PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {
>+  RTC_DCHECK_RUN_ON(&pc_thread_checker_);
>+  // Either network_thread or worker_thread might be active at this point.
>+  // Relying on ~PeerConnection to properly wait for them doesn't work,
>+  // as a vptr race might occur (before we enter the destruction body).
>+  // See: bugs.webrtc.org/9847
>+  if (pc()) {
>+    pc()->Close();
>+  }
>+}
> 
> bool PeerConnectionTestWrapper::CreatePc(
>     const webrtc::PeerConnectionInterface::RTCConfiguration& config,
>@@ -75,6 +88,8 @@ bool PeerConnectionTestWrapper::CreatePc(
>   std::unique_ptr<cricket::PortAllocator> port_allocator(
>       new cricket::FakePortAllocator(network_thread_, nullptr));
> 
>+  RTC_DCHECK_RUN_ON(&pc_thread_checker_);
>+
>   fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
>   if (fake_audio_capture_module_ == NULL) {
>     return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.h
>index 21ba89acbf07305839a51d5198fad4dd89111f52..b6a57f35d3823311ba5f3d3d7f6646ff07d20888 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.h
>@@ -20,6 +20,7 @@
> #include "pc/test/fakeaudiocapturemodule.h"
> #include "pc/test/fakevideotrackrenderer.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
>+#include "rtc_base/thread_checker.h"
> 
> class PeerConnectionTestWrapper
>     : public webrtc::PeerConnectionObserver,
>@@ -106,6 +107,7 @@ class PeerConnectionTestWrapper
>   std::string name_;
>   rtc::Thread* const network_thread_;
>   rtc::Thread* const worker_thread_;
>+  rtc::ThreadChecker pc_thread_checker_;
>   rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
>   rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
>       peer_connection_factory_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource_unittest.cc
>index d2b8e1574cebd083d62372dc248ce7ec648ce88c..67bec345b697a4a725a332443fa50c3c961e1ec3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource_unittest.cc
>@@ -126,7 +126,7 @@ class VideoCapturerTrackSourceTest : public testing::Test {
>   }
> 
>   void CaptureSingleFrame() {
>-    rtc::Event event(false, false);
>+    rtc::Event event;
>     task_queue_.PostTask([this, &event]() {
>       ASSERT_TRUE(capturer_->CaptureFrame());
>       event.Set();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotrack.h
>index e669e0814c7253e0a873774f8933a34216bb550d..f119dd86d83de6806d63d072a7ea3e29882647b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotrack.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotrack.h
>@@ -17,6 +17,7 @@
> #include "media/base/videosourcebase.h"
> #include "pc/mediastreamtrack.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/thread.h"
> #include "rtc_base/thread_checker.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.h
>index 50488dd2c6011d336abe9bba277e10f646467e53..cedb75b3c88dd730f6387595e1b16a0a62ada94b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.h
>@@ -15,13 +15,14 @@
> #include "api/notifier.h"
> #include "api/video/video_sink_interface.h"
> #include "media/base/mediachannel.h"
>+#include "rtc_base/system/rtc_export.h"
> #include "rtc_base/thread_checker.h"
> 
> namespace webrtc {
> 
> // VideoTrackSource is a convenience base class for implementations of
> // VideoTrackSourceInterface.
>-class VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
>+class RTC_EXPORT VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
>  public:
>   explicit VideoTrackSource(bool remote);
>   void SetState(SourceState new_state);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc
>index 665b4d411a050a98430f26d4f4fab347f0ba47a8..0cd6beceefa377dffe7f934b9e5b96635046c545 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc
>@@ -20,13 +20,15 @@
> #include <set>
> #include <string>
> #include <unordered_map>
>+#include <utility>
> #include <vector>
> 
>-#include "api/mediatypes.h"
>+#include "absl/strings/match.h"
> #include "api/candidate.h"
> #include "api/cryptoparams.h"
> #include "api/jsepicecandidate.h"
> #include "api/jsepsessiondescription.h"
>+#include "api/mediatypes.h"
> // for RtpExtension
> #include "api/rtpparameters.h"
> #include "media/base/codec.h"
>@@ -119,6 +121,7 @@ static const char kAttributeRtcpMux[] = "rtcp-mux";
> static const char kAttributeRtcpReducedSize[] = "rtcp-rsize";
> static const char kAttributeSsrc[] = "ssrc";
> static const char kSsrcAttributeCname[] = "cname";
>+static const char kAttributeExtmapAllowMixed[] = "extmap-allow-mixed";
> static const char kAttributeExtmap[] = "extmap";
> // draft-alvestrand-mmusic-msid-01
> // a=msid-semantic: WMS
>@@ -334,9 +337,10 @@ static bool ParseIceOptions(const std::string& line,
> static bool ParseExtmap(const std::string& line,
>                         RtpExtension* extmap,
>                         SdpParseError* error);
>-static bool ParseFingerprintAttribute(const std::string& line,
>-                                      rtc::SSLFingerprint** fingerprint,
>-                                      SdpParseError* error);
>+static bool ParseFingerprintAttribute(
>+    const std::string& line,
>+    std::unique_ptr<rtc::SSLFingerprint>* fingerprint,
>+    SdpParseError* error);
> static bool ParseDtlsSetup(const std::string& line,
>                            cricket::ConnectionRole* role,
>                            SdpParseError* error);
>@@ -852,6 +856,12 @@ std::string SdpSerialize(const JsepSessionDescription& jdesc) {
>     AddLine(group_line, &message);
>   }
> 
>+  // Mixed one- and two-byte header extension.
>+  if (desc->extmap_allow_mixed()) {
>+    InitAttrLine(kAttributeExtmapAllowMixed, &os);
>+    AddLine(os.str(), &message);
>+  }
>+
>   // MediaStream semantics
>   InitAttrLine(kAttributeMsidSemantics, &os);
>   os << kSdpDelimiterColon << " " << kMediaStreamSemantic;
>@@ -1303,8 +1313,8 @@ void BuildMediaDescription(const ContentInfo* content_info,
> 
>       if (data_desc->use_sctpmap()) {
>         for (const cricket::DataCodec& codec : data_desc->codecs()) {
>-          if (cricket::CodecNamesEq(codec.name,
>-                                    cricket::kGoogleSctpDataCodecName) &&
>+          if (absl::EqualsIgnoreCase(codec.name,
>+                                     cricket::kGoogleSctpDataCodecName) &&
>               codec.GetParam(cricket::kCodecParamPort, &sctp_port)) {
>             break;
>           }
>@@ -1482,7 +1492,17 @@ void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
>                                int msid_signaling,
>                                std::string* message) {
>   rtc::StringBuilder os;
>-  // RFC 5285
>+  // RFC 8285
>+  // a=extmap-allow-mixed
>+  // The attribute MUST be either on session level or media level. We support
>+  // responding on both levels, however, we don't respond on media level if it's
>+  // set on session level.
>+  if (media_desc->extmap_allow_mixed_enum() ==
>+      MediaContentDescription::kMedia) {
>+    InitAttrLine(kAttributeExtmapAllowMixed, &os);
>+    AddLine(os.str(), message);
>+  }
>+  // RFC 8285
>   // a=extmap:<value>["/"<direction>] <URI> <extensionattributes>
>   // The definitions MUST be either all session level or all media level. This
>   // implementation uses all media level.
>@@ -1609,7 +1629,12 @@ void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
>         // Since a=ssrc msid signaling is used in Plan B SDP semantics, and
>         // multiple stream ids are not supported for Plan B, we are only adding
>         // a line for the first media stream id here.
>-        const std::string& stream_id = track.first_stream_id();
>+        const std::string& track_stream_id = track.first_stream_id();
>+        // We use a special msid-id value of "-" to represent no streams,
>+        // for Unified Plan compatibility. Plan B will always have a
>+        // track_stream_id.
>+        const std::string& stream_id =
>+            track_stream_id.empty() ? kNoStreamMsid : track_stream_id;
>         InitAttrLine(kAttributeSsrc, &os);
>         os << kSdpDelimiterColon << ssrc << kSdpDelimiterSpace
>            << kSsrcAttributeMsid << kSdpDelimiterColon << stream_id
>@@ -1727,7 +1752,7 @@ void AddRtcpFbLines(const T& codec, std::string* message) {
> 
> bool AddSctpDataCodec(DataContentDescription* media_desc, int sctp_port) {
>   for (const auto& codec : media_desc->codecs()) {
>-    if (cricket::CodecNamesEq(codec.name, cricket::kGoogleSctpDataCodecName)) {
>+    if (absl::EqualsIgnoreCase(codec.name, cricket::kGoogleSctpDataCodecName)) {
>       return ParseFailed("", "Can't have multiple sctp port attributes.", NULL);
>     }
>   }
>@@ -1996,7 +2021,7 @@ bool ParseSessionDescription(const std::string& message,
>   std::string line;
> 
>   desc->set_msid_supported(false);
>-
>+  desc->set_extmap_allow_mixed(false);
>   // RFC 4566
>   // v=  (protocol version)
>   if (!GetLineWithType(message, pos, &line, kLineTypeVersion)) {
>@@ -2117,11 +2142,11 @@ bool ParseSessionDescription(const std::string& message,
>             "Can't have multiple fingerprint attributes at the same level.",
>             error);
>       }
>-      rtc::SSLFingerprint* fingerprint = NULL;
>+      std::unique_ptr<rtc::SSLFingerprint> fingerprint;
>       if (!ParseFingerprintAttribute(line, &fingerprint, error)) {
>         return false;
>       }
>-      session_td->identity_fingerprint.reset(fingerprint);
>+      session_td->identity_fingerprint = std::move(fingerprint);
>     } else if (HasAttribute(line, kAttributeSetup)) {
>       if (!ParseDtlsSetup(line, &(session_td->connection_role), error)) {
>         return false;
>@@ -2133,6 +2158,8 @@ bool ParseSessionDescription(const std::string& message,
>       }
>       desc->set_msid_supported(
>           CaseInsensitiveFind(semantics, kMediaStreamSemantic));
>+    } else if (HasAttribute(line, kAttributeExtmapAllowMixed)) {
>+      desc->set_extmap_allow_mixed(true);
>     } else if (HasAttribute(line, kAttributeExtmap)) {
>       RtpExtension extmap;
>       if (!ParseExtmap(line, &extmap, error)) {
>@@ -2166,9 +2193,10 @@ bool ParseGroupAttribute(const std::string& line,
>   return true;
> }
> 
>-static bool ParseFingerprintAttribute(const std::string& line,
>-                                      rtc::SSLFingerprint** fingerprint,
>-                                      SdpParseError* error) {
>+static bool ParseFingerprintAttribute(
>+    const std::string& line,
>+    std::unique_ptr<rtc::SSLFingerprint>* fingerprint,
>+    SdpParseError* error) {
>   if (!IsLineType(line, kLineTypeAttributes) ||
>       !HasAttribute(line, kAttributeFingerprint)) {
>     return ParseFailedExpectLine(line, 0, kLineTypeAttributes,
>@@ -2194,7 +2222,8 @@ static bool ParseFingerprintAttribute(const std::string& line,
>                  ::tolower);
> 
>   // The second field is the digest value. De-hexify it.
>-  *fingerprint = rtc::SSLFingerprint::CreateFromRfc4572(algorithm, fields[1]);
>+  *fingerprint =
>+      rtc::SSLFingerprint::CreateUniqueFromRfc4572(algorithm, fields[1]);
>   if (!*fingerprint) {
>     return ParseFailed(line, "Failed to create fingerprint from the digest.",
>                        error);
>@@ -2834,12 +2863,11 @@ bool ParseContent(const std::string& message,
>         return false;
>       }
>     } else if (HasAttribute(line, kAttributeFingerprint)) {
>-      rtc::SSLFingerprint* fingerprint = NULL;
>-
>+      std::unique_ptr<rtc::SSLFingerprint> fingerprint;
>       if (!ParseFingerprintAttribute(line, &fingerprint, error)) {
>         return false;
>       }
>-      transport->identity_fingerprint.reset(fingerprint);
>+      transport->identity_fingerprint = std::move(fingerprint);
>     } else if (HasAttribute(line, kAttributeSetup)) {
>       if (!ParseDtlsSetup(line, &(transport->connection_role), error)) {
>         return false;
>@@ -2902,6 +2930,9 @@ bool ParseContent(const std::string& message,
>         media_desc->set_direction(RtpTransceiverDirection::kInactive);
>       } else if (HasAttribute(line, kAttributeSendRecv)) {
>         media_desc->set_direction(RtpTransceiverDirection::kSendRecv);
>+      } else if (HasAttribute(line, kAttributeExtmapAllowMixed)) {
>+        media_desc->set_extmap_allow_mixed_enum(
>+            MediaContentDescription::kMedia);
>       } else if (HasAttribute(line, kAttributeExtmap)) {
>         RtpExtension extmap;
>         if (!ParseExtmap(line, &extmap, error)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.h
>index 2d842a1d5d4b8091bb74a94f96efab609d077d29..f185c0893200898bd1706519f1b659d5f496b2b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.h
>@@ -22,6 +22,8 @@
> 
> #include <string>
> 
>+#include "rtc_base/system/rtc_export.h"
>+
> namespace cricket {
> class Candidate;
> }  // namespace cricket
>@@ -46,7 +48,8 @@ std::string SdpSerializeCandidate(const IceCandidateInterface& candidate);
> 
> // Serializes a cricket Candidate.
> // candidate - The candidate to be serialized.
>-std::string SdpSerializeCandidate(const cricket::Candidate& candidate);
>+RTC_EXPORT std::string SdpSerializeCandidate(
>+    const cricket::Candidate& candidate);
> 
> // Deserializes the passed in SDP string to a JsepSessionDescription.
> // message - SDP string to be Deserialized.
>@@ -76,10 +79,10 @@ bool SdpDeserializeCandidate(const std::string& message,
> // candidate - The cricket Candidate from the SDP string.
> // error - The detail error information when parsing fails.
> // return - true on success, false on failure.
>-bool SdpDeserializeCandidate(const std::string& transport_name,
>-                             const std::string& message,
>-                             cricket::Candidate* candidate,
>-                             SdpParseError* error);
>+RTC_EXPORT bool SdpDeserializeCandidate(const std::string& transport_name,
>+                                        const std::string& message,
>+                                        cricket::Candidate* candidate,
>+                                        SdpParseError* error);
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc
>index 9c1055463153be712254a0342127a80046ef66ee..e7c5c77bb61ade2b5f948926610257b10965d07a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc
>@@ -22,6 +22,7 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/messagedigest.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/stringutils.h"
> 
>@@ -89,6 +90,7 @@ static const char kAttributeCryptoVideo[] =
> static const char kFingerprint[] =
>     "a=fingerprint:sha-1 "
>     "4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n";
>+static const char kExtmapAllowMixed[] = "a=extmap-allow-mixed\r\n";
> static const int kExtmapId = 1;
> static const char kExtmapUri[] = "http://example.com/082005/ext.htm#ttime";
> static const char kExtmap[] =
>@@ -621,8 +623,8 @@ static const char kUnifiedPlanSdpFullStringWithSpecialMsid[] =
>     "generation 2\r\n"
>     "a=ice-ufrag:ufrag_voice\r\na=ice-pwd:pwd_voice\r\n"
>     "a=mid:audio_content_name\r\n"
>-    "a=msid:local_stream_1 audio_track_id_1\r\n"
>     "a=sendrecv\r\n"
>+    "a=msid:local_stream_1 audio_track_id_1\r\n"
>     "a=rtcp-mux\r\n"
>     "a=rtcp-rsize\r\n"
>     "a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
>@@ -641,9 +643,9 @@ static const char kUnifiedPlanSdpFullStringWithSpecialMsid[] =
>     "a=rtcp:9 IN IP4 0.0.0.0\r\n"
>     "a=ice-ufrag:ufrag_voice_2\r\na=ice-pwd:pwd_voice_2\r\n"
>     "a=mid:audio_content_name_2\r\n"
>+    "a=sendrecv\r\n"
>     "a=msid:local_stream_1 audio_track_id_2\r\n"
>     "a=msid:local_stream_2 audio_track_id_2\r\n"
>-    "a=sendrecv\r\n"
>     "a=rtcp-mux\r\n"
>     "a=rtcp-rsize\r\n"
>     "a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
>@@ -664,8 +666,8 @@ static const char kUnifiedPlanSdpFullStringWithSpecialMsid[] =
>     "a=rtcp:9 IN IP4 0.0.0.0\r\n"
>     "a=ice-ufrag:ufrag_voice_3\r\na=ice-pwd:pwd_voice_3\r\n"
>     "a=mid:audio_content_name_3\r\n"
>-    "a=msid:- audio_track_id_3\r\n"
>     "a=sendrecv\r\n"
>+    "a=msid:- audio_track_id_3\r\n"
>     "a=rtcp-mux\r\n"
>     "a=rtcp-rsize\r\n"
>     "a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
>@@ -674,7 +676,10 @@ static const char kUnifiedPlanSdpFullStringWithSpecialMsid[] =
>     "a=rtpmap:111 opus/48000/2\r\n"
>     "a=rtpmap:103 ISAC/16000\r\n"
>     "a=rtpmap:104 ISAC/32000\r\n"
>-    "a=ssrc:7 cname:stream_2_cname\r\n";
>+    "a=ssrc:7 cname:stream_2_cname\r\n"
>+    "a=ssrc:7 msid:- audio_track_id_3\r\n"
>+    "a=ssrc:7 mslabel:-\r\n"
>+    "a=ssrc:7 label:audio_track_id_3\r\n";
> 
> // One candidate reference string as per W3c spec.
> // candidate:<blah> not a=candidate:<blah>CRLF
>@@ -1160,7 +1165,7 @@ class WebRtcSdpTest : public testing::Test {
>   // with 3 audio MediaContentDescriptions with special StreamParams that
>   // contain 0 or multiple stream ids: - audio track 1 has 1 media stream id -
>   // audio track 2 has 2 media stream ids - audio track 3 has 0 media stream ids
>-  void MakeUnifiedPlanDescriptionMultipleStreamIds() {
>+  void MakeUnifiedPlanDescriptionMultipleStreamIds(const int msid_signaling) {
>     desc_.RemoveContentByName(kVideoContentName);
>     desc_.RemoveTransportInfoByName(kVideoContentName);
>     RemoveVideoCandidates();
>@@ -1188,9 +1193,7 @@ class WebRtcSdpTest : public testing::Test {
>     desc_.AddContent(kAudioContentName3, MediaProtocolType::kRtp, audio_desc_3);
>     EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo(
>         kAudioContentName3, TransportDescription(kUfragVoice3, kPwdVoice3))));
>-    // Make sure to create both a=msid lines.
>-    desc_.set_msid_signaling(cricket::kMsidSignalingMediaSection |
>-                             cricket::kMsidSignalingSsrcAttribute);
>+    desc_.set_msid_signaling(msid_signaling);
>     ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                   jdesc_.session_version()));
>   }
>@@ -1282,6 +1285,9 @@ class WebRtcSdpTest : public testing::Test {
>     // streams
>     EXPECT_EQ(cd1->streams(), cd2->streams());
> 
>+    // extmap-allow-mixed
>+    EXPECT_EQ(cd1->extmap_allow_mixed_enum(), cd2->extmap_allow_mixed_enum());
>+
>     // extmap
>     ASSERT_EQ(cd1->rtp_header_extensions().size(),
>               cd2->rtp_header_extensions().size());
>@@ -1398,6 +1404,7 @@ class WebRtcSdpTest : public testing::Test {
> 
>     // global attributes
>     EXPECT_EQ(desc1.msid_supported(), desc2.msid_supported());
>+    EXPECT_EQ(desc1.extmap_allow_mixed(), desc2.extmap_allow_mixed());
>   }
> 
>   bool CompareSessionDescription(const JsepSessionDescription& desc1,
>@@ -1495,8 +1502,7 @@ class WebRtcSdpTest : public testing::Test {
>   void AddFingerprint() {
>     desc_.RemoveTransportInfoByName(kAudioContentName);
>     desc_.RemoveTransportInfoByName(kVideoContentName);
>-    rtc::SSLFingerprint fingerprint(rtc::DIGEST_SHA_1, kIdentityDigest,
>-                                    sizeof(kIdentityDigest));
>+    rtc::SSLFingerprint fingerprint(rtc::DIGEST_SHA_1, kIdentityDigest);
>     EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo(
>         kAudioContentName,
>         TransportDescription(std::vector<std::string>(), kUfragVoice, kPwdVoice,
>@@ -1540,6 +1546,27 @@ class WebRtcSdpTest : public testing::Test {
>     }
>   }
> 
>+  // Removes all a=ssrc lines from the SDP string, except for the
>+  // "a=ssrc:... cname:..." lines.
>+  void RemoveSsrcMsidLinesFromSdpString(std::string* sdp_string) {
>+    const char kAttributeSsrc[] = "a=ssrc";
>+    const char kAttributeCname[] = "cname";
>+    size_t ssrc_line_pos = sdp_string->find(kAttributeSsrc);
>+    while (ssrc_line_pos != std::string::npos) {
>+      size_t beg_line_pos = sdp_string->rfind('\n', ssrc_line_pos);
>+      size_t end_line_pos = sdp_string->find('\n', ssrc_line_pos);
>+      size_t cname_pos = sdp_string->find(kAttributeCname, ssrc_line_pos);
>+      if (cname_pos == std::string::npos || cname_pos > end_line_pos) {
>+        // Only erase a=ssrc lines that don't contain "cname".
>+        sdp_string->erase(beg_line_pos, end_line_pos - beg_line_pos);
>+        ssrc_line_pos = sdp_string->find(kAttributeSsrc, beg_line_pos);
>+      } else {
>+        // Skip the "a=ssrc:... cname" line and find the next "a=ssrc" line.
>+        ssrc_line_pos = sdp_string->find(kAttributeSsrc, end_line_pos);
>+      }
>+    }
>+  }
>+
>   // Removes all a=ssrc lines from the SDP string.
>   void RemoveSsrcLinesFromSdpString(std::string* sdp_string) {
>     const char kAttributeSsrc[] = "a=ssrc";
>@@ -2070,9 +2097,8 @@ TEST_F(WebRtcSdpTest, SerializeWithSctpDataChannelAndNewPort) {
> 
>   char default_portstr[16];
>   char new_portstr[16];
>-  rtc::sprintfn(default_portstr, sizeof(default_portstr), "%d",
>-                kDefaultSctpPort);
>-  rtc::sprintfn(new_portstr, sizeof(new_portstr), "%d", kNewPort);
>+  snprintf(default_portstr, sizeof(default_portstr), "%d", kDefaultSctpPort);
>+  snprintf(new_portstr, sizeof(new_portstr), "%d", kNewPort);
>   rtc::replace_substrs(default_portstr, strlen(default_portstr), new_portstr,
>                        strlen(new_portstr), &expected_sdp);
> 
>@@ -2094,6 +2120,25 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) {
>   EXPECT_EQ(expected_sdp, message);
> }
> 
>+TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithExtmapAllowMixed) {
>+  jdesc_.description()->set_extmap_allow_mixed(true);
>+  TestSerialize(jdesc_);
>+}
>+
>+TEST_F(WebRtcSdpTest, SerializeMediaContentDescriptionWithExtmapAllowMixed) {
>+  cricket::MediaContentDescription* video_desc =
>+      jdesc_.description()->GetContentDescriptionByName(kVideoContentName);
>+  ASSERT_TRUE(video_desc);
>+  cricket::MediaContentDescription* audio_desc =
>+      jdesc_.description()->GetContentDescriptionByName(kAudioContentName);
>+  ASSERT_TRUE(audio_desc);
>+  video_desc->set_extmap_allow_mixed_enum(
>+      cricket::MediaContentDescription::kMedia);
>+  audio_desc->set_extmap_allow_mixed_enum(
>+      cricket::MediaContentDescription::kMedia);
>+  TestSerialize(jdesc_);
>+}
>+
> TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithExtmap) {
>   bool encrypted = false;
>   AddExtmap(encrypted);
>@@ -2431,6 +2476,53 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutMsid) {
>   EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc));
> }
> 
>+TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithExtmapAllowMixed) {
>+  jdesc_.description()->set_extmap_allow_mixed(true);
>+  std::string sdp_with_extmap_allow_mixed = kSdpFullString;
>+  InjectAfter("t=0 0\r\n", kExtmapAllowMixed, &sdp_with_extmap_allow_mixed);
>+  // Deserialize
>+  JsepSessionDescription jdesc_deserialized(kDummyType);
>+  EXPECT_TRUE(SdpDeserialize(sdp_with_extmap_allow_mixed, &jdesc_deserialized));
>+  // Verify
>+  EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_deserialized));
>+}
>+
>+TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutExtmapAllowMixed) {
>+  jdesc_.description()->set_extmap_allow_mixed(false);
>+  std::string sdp_without_extmap_allow_mixed = kSdpFullString;
>+  // Deserialize
>+  JsepSessionDescription jdesc_deserialized(kDummyType);
>+  EXPECT_TRUE(
>+      SdpDeserialize(sdp_without_extmap_allow_mixed, &jdesc_deserialized));
>+  // Verify
>+  EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_deserialized));
>+}
>+
>+TEST_F(WebRtcSdpTest, DeserializeMediaContentDescriptionWithExtmapAllowMixed) {
>+  cricket::MediaContentDescription* video_desc =
>+      jdesc_.description()->GetContentDescriptionByName(kVideoContentName);
>+  ASSERT_TRUE(video_desc);
>+  cricket::MediaContentDescription* audio_desc =
>+      jdesc_.description()->GetContentDescriptionByName(kAudioContentName);
>+  ASSERT_TRUE(audio_desc);
>+  video_desc->set_extmap_allow_mixed_enum(
>+      cricket::MediaContentDescription::kMedia);
>+  audio_desc->set_extmap_allow_mixed_enum(
>+      cricket::MediaContentDescription::kMedia);
>+
>+  std::string sdp_with_extmap_allow_mixed = kSdpFullString;
>+  InjectAfter("a=mid:audio_content_name\r\n", kExtmapAllowMixed,
>+              &sdp_with_extmap_allow_mixed);
>+  InjectAfter("a=mid:video_content_name\r\n", kExtmapAllowMixed,
>+              &sdp_with_extmap_allow_mixed);
>+
>+  // Deserialize
>+  JsepSessionDescription jdesc_deserialized(kDummyType);
>+  EXPECT_TRUE(SdpDeserialize(sdp_with_extmap_allow_mixed, &jdesc_deserialized));
>+  // Verify
>+  EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_deserialized));
>+}
>+
> TEST_F(WebRtcSdpTest, DeserializeCandidate) {
>   JsepIceCandidate jcandidate(kDummyMid, kDummyIndex);
> 
>@@ -3362,12 +3454,16 @@ TEST_F(WebRtcSdpTest, SerializeUnifiedPlanSessionDescription) {
> }
> 
> // This tests deserializing a Unified Plan SDP that is compatible with both
>-// Unified Plan and Plan B style SDP. It tests the case for audio/video tracks
>+// Unified Plan and Plan B style SDP, meaning that it contains both "a=ssrc
>+// msid" lines and "a=msid " lines. It tests the case for audio/video tracks
> // with no stream ids and multiple stream ids. For parsing this, the Unified
> // Plan a=msid lines should take priority, because the Plan B style a=ssrc msid
> // lines do not support multiple stream ids and no stream ids.
>-TEST_F(WebRtcSdpTest, DeserializeUnifiedPlanSessionDescriptionSpecialMsid) {
>-  MakeUnifiedPlanDescriptionMultipleStreamIds();
>+TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionSpecialMsid) {
>+  // Create both msid lines for Plan B and Unified Plan support.
>+  MakeUnifiedPlanDescriptionMultipleStreamIds(
>+      cricket::kMsidSignalingMediaSection |
>+      cricket::kMsidSignalingSsrcAttribute);
> 
>   JsepSessionDescription deserialized_description(kDummyType);
>   EXPECT_TRUE(SdpDeserialize(kUnifiedPlanSdpFullStringWithSpecialMsid,
>@@ -3376,8 +3472,50 @@ TEST_F(WebRtcSdpTest, DeserializeUnifiedPlanSessionDescriptionSpecialMsid) {
>   EXPECT_TRUE(CompareSessionDescription(jdesc_, deserialized_description));
> }
> 
>-TEST_F(WebRtcSdpTest, SerializeUnifiedPlanSessionDescriptionSpecialMsid) {
>-  MakeUnifiedPlanDescriptionMultipleStreamIds();
>+// Tests the serialization of a Unified Plan SDP that is compatible for both
>+// Unified Plan and Plan B style SDPs, meaning that it contains both "a=ssrc
>+// msid" lines and "a=msid " lines. It tests the case for no stream ids and
>+// multiple stream ids.
>+TEST_F(WebRtcSdpTest, SerializeSessionDescriptionSpecialMsid) {
>+  // Create both msid lines for Plan B and Unified Plan support.
>+  MakeUnifiedPlanDescriptionMultipleStreamIds(
>+      cricket::kMsidSignalingMediaSection |
>+      cricket::kMsidSignalingSsrcAttribute);
>+  std::string serialized_sdp = webrtc::SdpSerialize(jdesc_);
>+  // We explicitly test that the serialized SDP string is equal to the hard
>+  // coded SDP string. This is necessary, because in the parser "a=msid" lines
>+  // take priority over "a=ssrc msid" lines. This means if we just used
>+  // TestSerialize(), it could serialize an SDP that omits "a=ssrc msid" lines,
>+  // and still pass, because the deserialized version would be the same.
>+  EXPECT_EQ(kUnifiedPlanSdpFullStringWithSpecialMsid, serialized_sdp);
>+}
>+
>+// Tests that a Unified Plan style SDP (does not contain "a=ssrc msid" lines
>+// that signal stream IDs) is deserialized appropriately. It tests the case for
>+// no stream ids and multiple stream ids.
>+TEST_F(WebRtcSdpTest, UnifiedPlanDeserializeSessionDescriptionSpecialMsid) {
>+  // Only create a=msid lines for strictly Unified Plan stream ID support.
>+  MakeUnifiedPlanDescriptionMultipleStreamIds(
>+      cricket::kMsidSignalingMediaSection);
>+
>+  JsepSessionDescription deserialized_description(kDummyType);
>+  std::string unified_plan_sdp_string =
>+      kUnifiedPlanSdpFullStringWithSpecialMsid;
>+  RemoveSsrcMsidLinesFromSdpString(&unified_plan_sdp_string);
>+  EXPECT_TRUE(
>+      SdpDeserialize(unified_plan_sdp_string, &deserialized_description));
>+
>+  EXPECT_TRUE(CompareSessionDescription(jdesc_, deserialized_description));
>+}
>+
>+// Tests that a Unified Plan style SDP (does not contain "a=ssrc msid" lines
>+// that signal stream IDs) is serialized appropriately. It tests the case for no
>+// stream ids and multiple stream ids.
>+TEST_F(WebRtcSdpTest, UnifiedPlanSerializeSessionDescriptionSpecialMsid) {
>+  // Only create a=msid lines for strictly Unified Plan stream ID support.
>+  MakeUnifiedPlanDescriptionMultipleStreamIds(
>+      cricket::kMsidSignalingMediaSection);
>+
>   TestSerialize(jdesc_);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.cc
>index 808e411a39985852b963317666bc881b48d7a8c5..07b8e147cc8277407e26146585ea60a70c40a206 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.cc
>@@ -116,7 +116,6 @@ void WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
>   }
> }
> 
>-// Private constructor called by other constructors.
> WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
>     rtc::Thread* signaling_thread,
>     cricket::ChannelManager* channel_manager,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT20.sha1
>deleted file mode 100644
>index 5cb8e52cb25df314af960b1042980144525609d2..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT20.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-96fb5327ff7a1fe87bd4512773ce7347b4d72888
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT30.sha1
>deleted file mode 100644
>index 1470013b72af0deff58742f2cb9f7e593e8d0eaa..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT30.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-ea527e8e61241ea73265abba5765793511c42291
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT20.sha1
>deleted file mode 100644
>index d33eb7e1c7475b854d831cb4cbb82dbbc0e23a15..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT20.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-a5e8c268936d7c8d03edd708c675254474aed944
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT30.sha1
>deleted file mode 100644
>index 953656fe01c0b3383da368f054faf3d3290ef052..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT30.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-9893504bdaa560a4ddb67a99e6701e8d4244896a
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT20.sha1
>deleted file mode 100644
>index b8a52534b3ae8160c3c07242893856d9edc19a29..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT20.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-5e0053a0de7b4708402fcc0a1b5ae024815938a3
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT30.sha1
>deleted file mode 100644
>index d3bbc9866d15687c7268d754cfea81bce66ea798..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT30.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-d57986d3dc65c1cadc41419a77ed1472b73050b4
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT20.sha1
>deleted file mode 100644
>index a304fa41811a89de08c15e1af18237c152dd6552..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT20.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-8cc1f82ee0d1b7df39840895d3e99756920b85ad
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT30.sha1
>deleted file mode 100644
>index c6cbe13afb4b87a11ff967a1a501465b48c7a9c2..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT30.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-85ca3d9a5ca61c1bb9d798ba6d3551945ab7667d
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.INP.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.INP.sha1
>deleted file mode 100644
>index 53118fcb35c1d62f1e6422a004dee8f1786037f6..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.INP.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-730581035d5af832d1b05c8f3a623ec93863b3e3
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT20.sha1
>deleted file mode 100644
>index fd27f041e09eaa31a94a40c589c83ed5cef2c390..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT20.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-985e5ffa0b98718afadfb1f02b5595fb24a9476c
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT30.sha1
>deleted file mode 100644
>index 3ccec4effefb7feef0439de2a76c09fd0cc2b683..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT30.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-8b035fb0f9ce3bdc2ae6c983e232c257a515a781
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT20.sha1
>deleted file mode 100644
>index 39d82e6f521a72e0b93345c29033efdb6ec419bf..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT20.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-9069fecc3cd13bf901781af4ece8b374b5285984
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT30.sha1
>deleted file mode 100644
>index 513c9362a89119db2c62ff21e4f8eefc193477a4..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT30.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-44a04e133c15e013f6b4424877528a4445e28bef
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.INP.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.INP.sha1
>deleted file mode 100644
>index 0ee0a39f3ca412d0f06836812859a8f65759b7ad..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.INP.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-c3dde10c32f12da58181ecaccb7aeaa515239233
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT20.sha1
>deleted file mode 100644
>index df514526521a0589a4a4cb68674feeed8cbc2688..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT20.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-260f51f8320d92d3c8b834c1f38879baa12a7a5a
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT30.sha1
>deleted file mode 100644
>index bcc318c96f498b23581a7fb3349bc5f0d9d51d56..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT30.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-72abb4c1d84a4cff1efed4d3e01280d839a9f691
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT20.sha1
>deleted file mode 100644
>index 6b6e3dfb6845b51014347878513ad0ac3480eb42..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT20.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-b6530247acdfd7ae7b3fd80c677d71045351ad07
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT30.sha1
>deleted file mode 100644
>index f9f7975730e5c1d8495cf898718b42d826df0b52..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT30.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-0318de4dc48422bfe125fd6f827e6e7114dc61bc
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.INP.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.INP.sha1
>deleted file mode 100644
>index 1210209652c76d7d4b7e5a29039e757f7d16e5c0..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.INP.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-80359cff72cbdc04a337a51c2fcc0c74890710b4
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT20.sha1
>deleted file mode 100644
>index 7d50df8258d334d0f2e49447fdd0cce2e3b26255..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT20.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-c7bc71abc973440123a517f61e3dbcb96089ee3f
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT30.sha1
>deleted file mode 100644
>index ad9219dd0ad07f7b224057626538cf8c8ba226d7..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT30.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-5fde2defd2350b5ca7d3b8f8ceef75d2b7fd7f03
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/tlm10.chn.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/tlm10.chn.sha1
>deleted file mode 100644
>index 801f4487bc636578b94d9fe65ee74794da557f00..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/tlm10.chn.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-21c78516c2470667a75c7ed85fe37c53a3514456
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/BUILD.gn
>index e9efbd1360f124b2971104d368090ff12134d43a..75c4397ab7d48afc1775adaeb362f482722404a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/BUILD.gn
>@@ -67,9 +67,102 @@ rtc_source_set("compile_assert_c") {
>   ]
> }
> 
>+# The subset of rtc_base approved for use outside of libjingle.
>+# TODO(bugs.webrtc.org/9838): Create small and focused build targets and remove
>+# the old concept of rtc_base and rtc_base_approved.
> rtc_source_set("rtc_base_approved") {
>   visibility = [ "*" ]
>-  public_deps = [
>+  deps = [
>+    ":checks",
>+    ":rtc_task_queue",
>+    ":safe_compare",
>+    ":safe_minmax",
>+    ":type_traits",
>+    "../api:array_view",
>+    "../system_wrappers:field_trial",
>+    "experiments:field_trial_parser",
>+    "system:arch",
>+    "system:unused",
>+    "third_party/base64",
>+    "//third_party/abseil-cpp/absl/memory:memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+
>+  sources = [
>+    "bind.h",
>+    "bitbuffer.cc",
>+    "bitbuffer.h",
>+    "bitrateallocationstrategy.cc",
>+    "bitrateallocationstrategy.h",
>+    "buffer.h",
>+    "bufferqueue.cc",
>+    "bufferqueue.h",
>+    "bytebuffer.cc",
>+    "bytebuffer.h",
>+    "byteorder.h",
>+    "copyonwritebuffer.cc",
>+    "copyonwritebuffer.h",
>+    "event_tracer.cc",
>+    "event_tracer.h",
>+    "file.cc",
>+    "file.h",
>+    "flags.cc",
>+    "flags.h",
>+    "function_view.h",
>+    "ignore_wundef.h",
>+    "location.cc",
>+    "location.h",
>+    "message_buffer_reader.h",
>+    "numerics/histogram_percentile_counter.cc",
>+    "numerics/histogram_percentile_counter.h",
>+    "numerics/mod_ops.h",
>+    "numerics/moving_max_counter.h",
>+    "numerics/sample_counter.cc",
>+    "numerics/sample_counter.h",
>+    "onetimeevent.h",
>+    "platform_file.cc",
>+    "platform_file.h",
>+    "race_checker.cc",
>+    "race_checker.h",
>+    "random.cc",
>+    "random.h",
>+    "rate_statistics.cc",
>+    "rate_statistics.h",
>+    "ratetracker.cc",
>+    "ratetracker.h",
>+    "swap_queue.h",
>+    "template_util.h",
>+    "timestampaligner.cc",
>+    "timestampaligner.h",
>+    "trace_event.h",
>+    "zero_memory.cc",
>+    "zero_memory.h",
>+  ]
>+
>+  if (is_posix || is_fuchsia) {
>+    sources += [ "file_posix.cc" ]
>+  }
>+
>+  if (is_win) {
>+    sources += [
>+      "file_win.cc",
>+      "win/windows_version.cc",
>+      "win/windows_version.h",
>+    ]
>+    data_deps = [
>+      "//build/win:runtime_libs",
>+    ]
>+  }
>+
>+  if (is_nacl) {
>+    deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
>+  }
>+
>+  if (is_android) {
>+    libs = [ "log" ]
>+  }
>+
>+  public_deps = [  # no-presubmit-check TODO(webrtc:8603)
>     ":atomicops",
>     ":criticalsection",
>     ":logging",
>@@ -78,16 +171,12 @@ rtc_source_set("rtc_base_approved") {
>     ":platform_thread_types",
>     ":ptr_util",
>     ":refcount",
>-    ":rtc_base_approved_generic",
>     ":rtc_event",
>     ":safe_conversions",
>     ":stringutils",
>     ":thread_checker",
>     ":timeutils",
>   ]
>-  if (is_mac && !build_with_chromium) {
>-    public_deps += [ ":rtc_base_approved_objc" ]
>-  }
> }
> 
> rtc_source_set("macromagic") {
>@@ -117,6 +206,9 @@ rtc_source_set("ptr_util") {
>   sources = [
>     "scoped_ref_ptr.h",
>   ]
>+  deps = [
>+    "../api:scoped_refptr",
>+  ]
> }
> 
> rtc_source_set("refcount") {
>@@ -149,9 +241,9 @@ rtc_source_set("criticalsection") {
> rtc_source_set("platform_thread") {
>   visibility = [
>     ":rtc_base_approved",
>-    ":rtc_base_approved_generic",
>     ":rtc_task_queue_libevent",
>     ":rtc_task_queue_win",
>+    ":rtc_task_queue_stdlib",
>     ":sequenced_task_checker",
>   ]
>   sources = [
>@@ -172,7 +264,6 @@ rtc_source_set("platform_thread") {
> rtc_source_set("rtc_event") {
>   deps = [
>     ":checks",
>-    ":macromagic",
>   ]
> 
>   if (build_with_chromium) {
>@@ -192,7 +283,9 @@ rtc_source_set("rtc_event") {
> 
> rtc_source_set("logging") {
>   visibility = [ "*" ]
>+  libs = []
>   deps = [
>+    ":checks",
>     ":criticalsection",
>     ":macromagic",
>     ":platform_thread_types",
>@@ -219,9 +312,17 @@ rtc_source_set("logging") {
>     ]
>     deps += [ "system:inline" ]
> 
>+    if (is_mac) {
>+      deps += [ ":logging_mac" ]
>+    }
>+
>     # logging.h needs the deprecation header while downstream projects are
>     # removing code that depends on logging implementation details.
>     deps += [ ":deprecation" ]
>+
>+    if (is_android) {
>+      libs += [ "log" ]
>+    }
>   }
> }
> 
>@@ -248,6 +349,7 @@ rtc_source_set("atomicops") {
> rtc_source_set("checks") {
>   # TODO(bugs.webrtc.org/9607): This should not be public.
>   visibility = [ "*" ]
>+  libs = []
>   sources = [
>     "checks.cc",
>     "checks.h",
>@@ -256,6 +358,9 @@ rtc_source_set("checks") {
>     ":safe_compare",
>     "system:inline",
>   ]
>+  if (is_android) {
>+    libs += [ "log" ]
>+  }
> }
> 
> rtc_source_set("rate_limiter") {
>@@ -266,6 +371,7 @@ rtc_source_set("rate_limiter") {
>   deps = [
>     ":rtc_base_approved",
>     "../system_wrappers",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -273,6 +379,9 @@ rtc_source_set("sanitizer") {
>   sources = [
>     "sanitizer.h",
>   ]
>+  deps = [
>+    "//third_party/abseil-cpp/absl/meta:type_traits",
>+  ]
> }
> 
> rtc_source_set("safe_compare") {
>@@ -362,137 +471,14 @@ rtc_source_set("deprecation") {
>   ]
> }
> 
>-# The subset of rtc_base approved for use outside of libjingle.
>-rtc_source_set("rtc_base_approved_generic") {
>-  visibility = [
>-    ":rtc_base_approved",
>-    ":weak_ptr_unittests",
>-  ]
>-
>-  cflags = []
>-  defines = []
>-  libs = []
>-  data_deps = []
>-  deps = [
>-    ":atomicops",
>-    ":checks",
>-    ":criticalsection",
>-    ":logging",
>-    ":macromagic",
>-    ":platform_thread",
>-    ":platform_thread_types",
>-    ":ptr_util",
>-    ":refcount",
>-    ":rtc_event",
>-    ":rtc_task_queue",
>-    ":safe_compare",
>-    ":safe_conversions",
>-    ":stringutils",
>-    ":thread_checker",
>-    ":timeutils",
>-    ":type_traits",
>-    "system:arch",
>-    "system:unused",
>-    "third_party/base64",
>-  ]
>-
>-  sources = [
>-    "bind.h",
>-    "bitbuffer.cc",
>-    "bitbuffer.h",
>-    "bitrateallocationstrategy.cc",
>-    "bitrateallocationstrategy.h",
>-    "buffer.h",
>-    "bufferqueue.cc",
>-    "bufferqueue.h",
>-    "bytebuffer.cc",
>-    "bytebuffer.h",
>-    "byteorder.h",
>-    "copyonwritebuffer.cc",
>-    "copyonwritebuffer.h",
>-    "event_tracer.cc",
>-    "event_tracer.h",
>-    "file.cc",
>-    "file.h",
>-    "flags.cc",
>-    "flags.h",
>-    "function_view.h",
>-    "ignore_wundef.h",
>-    "location.cc",
>-    "location.h",
>-    "message_buffer_reader.h",
>-    "numerics/histogram_percentile_counter.cc",
>-    "numerics/histogram_percentile_counter.h",
>-    "numerics/mod_ops.h",
>-    "numerics/moving_max_counter.h",
>-    "numerics/sample_counter.cc",
>-    "numerics/sample_counter.h",
>-    "onetimeevent.h",
>-    "pathutils.cc",
>-    "pathutils.h",
>-    "platform_file.cc",
>-    "platform_file.h",
>-    "race_checker.cc",
>-    "race_checker.h",
>-    "random.cc",
>-    "random.h",
>-    "rate_statistics.cc",
>-    "rate_statistics.h",
>-    "ratetracker.cc",
>-    "ratetracker.h",
>-    "swap_queue.h",
>-    "template_util.h",
>-    "timestampaligner.cc",
>-    "timestampaligner.h",
>-    "trace_event.h",
>-    "zero_memory.cc",
>-    "zero_memory.h",
>-  ]
>-
>-  deps += [
>-    "..:webrtc_common",
>-    "../api:array_view",
>-    "//third_party/abseil-cpp/absl/memory:memory",
>-    "//third_party/abseil-cpp/absl/types:optional",
>-  ]
>-
>-  if (is_android) {
>-    libs += [ "log" ]
>-  }
>-
>-  if (is_posix || is_fuchsia) {
>-    sources += [ "file_posix.cc" ]
>-  }
>-
>-  if (is_win) {
>-    sources += [
>-      "file_win.cc",
>-      "win/windows_version.cc",
>-      "win/windows_version.h",
>-    ]
>-    data_deps += [ "//build/win:runtime_libs" ]
>-  }
>-
>-  if (is_nacl) {
>-    deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
>-  }
>-}
>-
> if (is_mac && !build_with_chromium) {
>-  config("rtc_base_approved_objc_all_dependent_config") {
>-    visibility = [ ":rtc_base_approved_objc" ]
>-    libs = [ "Foundation.framework" ]  # needed for logging_mac.mm
>-  }
>-
>-  rtc_source_set("rtc_base_approved_objc") {
>-    visibility = [ ":rtc_base_approved" ]
>-    all_dependent_configs = [ ":rtc_base_approved_objc_all_dependent_config" ]
>+  rtc_source_set("logging_mac") {
>+    visibility = [ ":logging" ]
>+    libs = [ "Foundation.framework" ]
>     sources = [
>+      "logging_mac.h",
>       "logging_mac.mm",
>     ]
>-    deps = [
>-      ":logging",
>-    ]
>   }
> }
> 
>@@ -529,6 +515,7 @@ rtc_source_set("rtc_task_queue_api") {
>   deps = [
>     ":macromagic",
>     ":ptr_util",
>+    "system:rtc_export",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
> }
>@@ -565,7 +552,9 @@ if (rtc_enable_libevent) {
>       ":checks",
>       ":criticalsection",
>       ":logging",
>+      ":macromagic",
>       ":platform_thread",
>+      ":platform_thread_types",
>       ":ptr_util",
>       ":refcount",
>       ":rtc_task_queue_api",
>@@ -619,6 +608,26 @@ if (is_win) {
>   }
> }
> 
>+rtc_source_set("rtc_task_queue_stdlib") {
>+  visibility = [ ":rtc_task_queue_impl" ]
>+  sources = [
>+    "task_queue_stdlib.cc",
>+  ]
>+  deps = [
>+    ":checks",
>+    ":criticalsection",
>+    ":logging",
>+    ":macromagic",
>+    ":platform_thread",
>+    ":ptr_util",
>+    ":refcount",
>+    ":rtc_event",
>+    ":rtc_task_queue_api",
>+    ":safe_conversions",
>+    ":timeutils",
>+  ]
>+}
>+
> rtc_source_set("rtc_task_queue_impl") {
>   visibility = [ "*" ]
>   if (rtc_enable_libevent) {
>@@ -632,9 +641,15 @@ rtc_source_set("rtc_task_queue_impl") {
>       ]
>     }
>     if (is_win) {
>-      deps = [
>-        ":rtc_task_queue_win",
>-      ]
>+      if (current_os == "winuwp") {
>+        deps = [
>+          ":rtc_task_queue_stdlib",
>+        ]
>+      } else {
>+        deps = [
>+          ":rtc_task_queue_win",
>+        ]
>+      }
>     }
>   }
> }
>@@ -670,6 +685,8 @@ rtc_static_library("rtc_numerics") {
>   sources = [
>     "numerics/exp_filter.cc",
>     "numerics/exp_filter.h",
>+    "numerics/moving_average.cc",
>+    "numerics/moving_average.h",
>     "numerics/moving_median_filter.h",
>     "numerics/percentile_filter.h",
>     "numerics/sequence_number_util.h",
>@@ -707,48 +724,23 @@ rtc_source_set("rtc_json") {
> 
> rtc_static_library("rtc_base") {
>   visibility = [ "*" ]
>-  public_deps = []
>-  if (!build_with_mozilla) {
>-    public_deps += [ ":rtc_base_generic" ]
>-  }
>-  if (is_win) {
>-    sources = [
>-      "noop.cc",
>-    ]
>-  }
>-  if (is_ios || is_mac) {
>-    sources = [
>-      "noop.mm",
>-    ]
>-    public_deps += [ ":rtc_base_objc" ]
>-  }
>-}
>-
>-if (is_ios || is_mac) {
>-  rtc_source_set("rtc_base_objc") {
>-    sources = [
>-      "thread_darwin.mm",
>-    ]
>-    deps = [
>-      ":rtc_base_generic",
>-    ]
>-    visibility = [ ":rtc_base" ]
>-  }
>-}
>-
>-rtc_static_library("rtc_base_generic") {
>   cflags = []
>   cflags_cc = []
>   libs = []
>   defines = []
>   deps = [
>     ":checks",
>+
>+    # For deprecation of rtc::PacketTime, in asyncpacketsocket.h.
>+    ":deprecation",
>     ":stringutils",
>     "..:webrtc_common",
>     "../api:array_view",
>+    "network:sent_packet",
>     "third_party/base64",
>     "third_party/sigslot",
>     "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/strings",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
>   public_deps = [
>@@ -781,8 +773,6 @@ rtc_static_library("rtc_base_generic") {
>     "dscp.h",
>     "filerotatingstream.cc",
>     "filerotatingstream.h",
>-    "fileutils.cc",
>-    "fileutils.h",
>     "gunit_prod.h",
>     "helpers.cc",
>     "helpers.h",
>@@ -791,6 +781,8 @@ rtc_static_library("rtc_base_generic") {
>     "ipaddress.cc",
>     "ipaddress.h",
>     "keep_ref_until_done.h",
>+    "key_derivation.cc",
>+    "key_derivation.h",
>     "mdns_responder_interface.h",
>     "messagedigest.cc",
>     "messagedigest.h",
>@@ -811,6 +803,8 @@ rtc_static_library("rtc_base_generic") {
>     "nullsocketserver.cc",
>     "nullsocketserver.h",
>     "openssl.h",
>+    "openssl_key_derivation_hkdf.cc",
>+    "openssl_key_derivation_hkdf.h",
>     "openssladapter.cc",
>     "openssladapter.h",
>     "opensslcertificate.cc",
>@@ -846,8 +840,6 @@ rtc_static_library("rtc_base_generic") {
>     "socketaddresspair.h",
>     "socketfactory.h",
>     "socketserver.h",
>-    "socketstream.cc",
>-    "socketstream.h",
>     "ssladapter.cc",
>     "ssladapter.h",
>     "sslcertificate.cc",
>@@ -864,11 +856,6 @@ rtc_static_library("rtc_base_generic") {
>     "thread.h",
>   ]
> 
>-  visibility = [
>-    ":rtc_base",
>-    ":rtc_base_objc",
>-  ]
>-
>   if (build_with_chromium) {
>     include_dirs = [ "../../boringssl/src/include" ]
>     public_configs += [ ":rtc_base_chromium_config" ]
>@@ -878,8 +865,6 @@ rtc_static_library("rtc_base_generic") {
>       "logsinks.cc",
>       "logsinks.h",
>       "numerics/mathutils.h",
>-      "optionsfile.cc",
>-      "optionsfile.h",
>       "rollingaccumulator.h",
>       "sslroots.h",
>     ]
>@@ -913,6 +898,7 @@ rtc_static_library("rtc_base_generic") {
> 
>   if (is_ios || is_mac) {
>     sources += [ "macifaddrs_converter.cc" ]
>+    deps += [ "system:cocoa_threading" ]
>   }
> 
>   if (rtc_use_x11) {
>@@ -938,18 +924,12 @@ rtc_static_library("rtc_base_generic") {
>       "macutils.cc",
>       "macutils.h",
>     ]
>-    libs += [
>-      # For ProcessInformationCopyDictionary in unixfilesystem.cc.
>-      "ApplicationServices.framework",
>-    ]
>   }
> 
>   if (is_win) {
>     sources += [
>       "win32.cc",
>       "win32.h",
>-      "win32filesystem.cc",
>-      "win32filesystem.h",
>       "win32window.cc",
>       "win32window.h",
>     ]
>@@ -967,8 +947,6 @@ rtc_static_library("rtc_base_generic") {
>     sources += [
>       "ifaddrs_converter.cc",
>       "ifaddrs_converter.h",
>-      "unixfilesystem.cc",
>-      "unixfilesystem.h",
>     ]
>   }
> 
>@@ -986,11 +964,41 @@ rtc_source_set("gtest_prod") {
>   ]
> }
> 
>+rtc_source_set("gunit_helpers") {
>+  testonly = true
>+  sources = [
>+    "gunit.cc",
>+    "gunit.h",
>+  ]
>+  deps = [
>+    ":logging",
>+    ":rtc_base",
>+    ":rtc_base_tests_utils",
>+    ":stringutils",
>+    "../test:test_support",
>+  ]
>+}
>+
>+rtc_source_set("testclient") {
>+  testonly = true
>+  sources = [
>+    "testclient.cc",
>+    "testclient.h",
>+  ]
>+  deps = [
>+    ":criticalsection",
>+    ":gunit_helpers",
>+    ":macromagic",
>+    ":rtc_base",
>+    ":rtc_base_tests_utils",
>+    ":timeutils",
>+    "//third_party/abseil-cpp/absl/memory:memory",
>+  ]
>+}
>+
> rtc_source_set("rtc_base_tests_utils") {
>   testonly = true
>   sources = [
>-    # Also use this as a convenient dumping ground for misc files that are
>-    # included by multiple targets below.
>     "cpu_time.cc",
>     "cpu_time.h",
>     "fake_mdns_responder.h",
>@@ -1001,8 +1009,8 @@ rtc_source_set("rtc_base_tests_utils") {
>     "fakesslidentity.h",
>     "firewallsocketserver.cc",
>     "firewallsocketserver.h",
>-    "gunit.cc",
>-    "gunit.h",
>+    "memory_stream.cc",
>+    "memory_stream.h",
>     "memory_usage.cc",
>     "memory_usage.h",
>     "natserver.cc",
>@@ -1015,10 +1023,10 @@ rtc_source_set("rtc_base_tests_utils") {
>     "proxyserver.h",
>     "sigslottester.h",
>     "sigslottester.h.pump",
>+    "socketstream.cc",
>+    "socketstream.h",
>     "testbase64.h",
>     "testcertificateverifier.h",
>-    "testclient.cc",
>-    "testclient.h",
>     "testechoserver.cc",
>     "testechoserver.h",
>     "testutils.cc",
>@@ -1029,16 +1037,10 @@ rtc_source_set("rtc_base_tests_utils") {
>   deps = [
>     ":checks",
>     ":rtc_base",
>-    ":stringutils",
>     "../api/units:time_delta",
>-    "../test:test_support",
>-    "system:fallthrough",
>     "third_party/sigslot",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
>-  public_deps = [
>-    "//testing/gtest",
>-  ]
> }
> 
> rtc_source_set("rtc_task_queue_for_test") {
>@@ -1063,6 +1065,7 @@ if (rtc_include_tests) {
>       "sigslot_unittest.cc",
>     ]
>     deps = [
>+      ":gunit_helpers",
>       ":rtc_base",
>       ":rtc_base_tests_utils",
>       "third_party/sigslot",
>@@ -1075,13 +1078,13 @@ if (rtc_include_tests) {
>       "unittest_main.cc",
>     ]
>     deps = [
>+      ":gunit_helpers",
>       ":rtc_base",
>       ":rtc_base_approved",
>       ":rtc_base_tests_utils",
>       "../system_wrappers:field_trial",
>       "../system_wrappers:metrics",
>       "../test:field_trial",
>-      "../test:fileutils",
>       "../test:test_support",
>     ]
> 
>@@ -1104,9 +1107,11 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       ":checks",
>+      ":gunit_helpers",
>       ":rtc_base",
>       ":rtc_base_tests_main",
>       ":rtc_base_tests_utils",
>+      ":testclient",
>       "../system_wrappers:system_wrappers",
>       "../test:fileutils",
>       "../test:test_support",
>@@ -1147,7 +1152,6 @@ if (rtc_include_tests) {
>       "numerics/safe_minmax_unittest.cc",
>       "numerics/sample_counter_unittest.cc",
>       "onetimeevent_unittest.cc",
>-      "pathutils_unittest.cc",
>       "platform_file_unittest.cc",
>       "platform_thread_unittest.cc",
>       "random_unittest.cc",
>@@ -1174,6 +1178,7 @@ if (rtc_include_tests) {
>     }
>     deps = [
>       ":checks",
>+      ":gunit_helpers",
>       ":rate_limiter",
>       ":rtc_base",
>       ":rtc_base_approved",
>@@ -1184,6 +1189,7 @@ if (rtc_include_tests) {
>       ":safe_minmax",
>       ":sanitizer",
>       ":stringutils",
>+      ":testclient",
>       "../api:array_view",
>       "../system_wrappers:system_wrappers",
>       "../test:fileutils",
>@@ -1203,6 +1209,7 @@ if (rtc_include_tests) {
>       "task_queue_unittest.cc",
>     ]
>     deps = [
>+      ":gunit_helpers",
>       ":rtc_base_approved",
>       ":rtc_base_tests_main",
>       ":rtc_base_tests_utils",
>@@ -1236,7 +1243,8 @@ if (rtc_include_tests) {
>       "weak_ptr_unittest.cc",
>     ]
>     deps = [
>-      ":rtc_base_approved_generic",
>+      ":gunit_helpers",
>+      ":rtc_base_approved",
>       ":rtc_base_tests_main",
>       ":rtc_base_tests_utils",
>       ":rtc_event",
>@@ -1251,6 +1259,7 @@ if (rtc_include_tests) {
> 
>     sources = [
>       "numerics/exp_filter_unittest.cc",
>+      "numerics/moving_average_unittest.cc",
>       "numerics/moving_median_filter_unittest.cc",
>       "numerics/percentile_filter_unittest.cc",
>       "numerics/sequence_number_util_unittest.cc",
>@@ -1270,6 +1279,7 @@ if (rtc_include_tests) {
>       "strings/json_unittest.cc",
>     ]
>     deps = [
>+      ":gunit_helpers",
>       ":rtc_base_tests_main",
>       ":rtc_base_tests_utils",
>       ":rtc_json",
>@@ -1292,7 +1302,6 @@ if (rtc_include_tests) {
>       "messagequeue_unittest.cc",
>       "nat_unittest.cc",
>       "network_unittest.cc",
>-      "optionsfile_unittest.cc",
>       "proxy_unittest.cc",
>       "rollingaccumulator_unittest.cc",
>       "rtccertificate_unittest.cc",
>@@ -1311,6 +1320,7 @@ if (rtc_include_tests) {
>     }
>     if (is_posix || is_fuchsia) {
>       sources += [
>+        "openssl_key_derivation_hkdf_unittest.cc",
>         "openssladapter_unittest.cc",
>         "opensslsessioncache_unittest.cc",
>         "opensslutility_unittest.cc",
>@@ -1321,9 +1331,11 @@ if (rtc_include_tests) {
>     }
>     deps = [
>       ":checks",
>+      ":gunit_helpers",
>       ":rtc_base_tests_main",
>       ":rtc_base_tests_utils",
>       ":stringutils",
>+      ":testclient",
>       "../api:array_view",
>       "../test:fileutils",
>       "../test:test_support",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/Dummy.java b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/Dummy.java
>deleted file mode 100644
>index d8f02c90dbf660dd058ee5d136e91a24eefa2e21..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/Dummy.java
>+++ /dev/null
>@@ -1,19 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-/**
>- * This class only exists as glue in a transition.
>- * TODO(kjellander): Remove.
>- * See https://bugs.webrtc.org/7634 for more details.
>- */
>-class Dummy {
>-  Dummy() {
>-  }
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/OWNERS
>index 69ce253087f59b17b76f2a5ce807290bcbf2bb14..e67e8340f7c7890f70e51b2368955a17e9615e8d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/OWNERS
>@@ -2,7 +2,6 @@ hta@webrtc.org
> juberti@webrtc.org
> kwiberg@webrtc.org
> mflodman@webrtc.org
>-perkj@webrtc.org
> pthatcher@webrtc.org
> qingsi@webrtc.org
> sergeyu@chromium.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.cc
>index e255fb98fccb7276b49f326e81777e7098045a00..f0dd1881ef3472b85c260d476d661fde83230712 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.cc
>@@ -17,7 +17,7 @@ namespace rtc {
> 
> AsyncInvoker::AsyncInvoker()
>     : pending_invocations_(0),
>-      invocation_complete_(new RefCountedObject<Event>(false, false)),
>+      invocation_complete_(new RefCountedObject<Event>()),
>       destroying_(false) {}
> 
> AsyncInvoker::~AsyncInvoker() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.cc
>index e1b1eae6a31d735b703af24b58cc8184b8c1dcb9..7e0cc8f81011e596c43805b20c9ed629aa4b1a88 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "rtc_base/asyncpacketsocket.h"
>+#include "rtc_base/nethelper.h"
> 
> namespace rtc {
> 
>@@ -33,9 +34,11 @@ void CopySocketInformationToPacketInfo(size_t packet_size_bytes,
>                                        bool is_connectionless,
>                                        rtc::PacketInfo* info) {
>   info->packet_size_bytes = packet_size_bytes;
>-  info->local_socket_address = socket_from.GetLocalAddress();
>-  if (!is_connectionless) {
>-    info->remote_socket_address = socket_from.GetRemoteAddress();
>+  // TODO(srte): Make sure that the family of the local socket is always set
>+  // in the VirtualSocket implementation and remove this check.
>+  int family = socket_from.GetLocalAddress().family();
>+  if (family != 0) {
>+    info->ip_overhead_bytes = cricket::GetIpOverhead(family);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.h
>index bb0b3bcc048e75fe486efdf6557f3ca7171cbb02..44d6c674267a6165f2b195f81f2d882f3364138f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.h
>@@ -12,6 +12,7 @@
> #define RTC_BASE_ASYNCPACKETSOCKET_H_
> 
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/deprecation.h"
> #include "rtc_base/dscp.h"
> #include "rtc_base/socket.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
>@@ -50,25 +51,9 @@ struct PacketOptions {
>   PacketInfo info_signaled_after_sent;
> };
> 
>-// This structure will have the information about when packet is actually
>-// received by socket.
>-struct PacketTime {
>-  PacketTime() : timestamp(-1), not_before(-1) {}
>-  PacketTime(int64_t timestamp, int64_t not_before)
>-      : timestamp(timestamp), not_before(not_before) {}
>-
>-  int64_t timestamp;  // Receive time after socket delivers the data.
>-
>-  // Earliest possible time the data could have arrived, indicating the
>-  // potential error in the |timestamp| value, in case the system, is busy. For
>-  // example, the time of the last select() call.
>-  // If unknown, this value will be set to zero.
>-  int64_t not_before;
>-};
>-
>-inline PacketTime CreatePacketTime(int64_t not_before) {
>-  return PacketTime(TimeMicros(), not_before);
>-}
>+// TODO(bugs.webrtc.org/9584): Compatibility alias, delete as soon as downstream
>+// code is updated.
>+typedef int64_t PacketTime;
> 
> // Provides the ability to receive packets asynchronously. Sends are not
> // buffered since it is acceptable to drop packets under high load.
>@@ -120,7 +105,9 @@ class AsyncPacketSocket : public sigslot::has_slots<> {
>                    const char*,
>                    size_t,
>                    const SocketAddress&,
>-                   const PacketTime&>
>+                   // TODO(bugs.webrtc.org/9584): Change to passing the int64_t
>+                   // timestamp by value.
>+                   const int64_t&>
>       SignalReadPacket;
> 
>   // Emitted each time a packet is sent.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.h
>index 5b2303f4d80286081b79a9b129f8d7f205e0051e..f3df8842493fde3c76c4ad5e99dc9e0691e4f7b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.h
>@@ -22,9 +22,12 @@ class AsyncResolverInterface {
>   AsyncResolverInterface();
>   virtual ~AsyncResolverInterface();
> 
>-  // Start address resolve process.
>+  // Start address resolution of the hostname in |addr|.
>   virtual void Start(const SocketAddress& addr) = 0;
>-  // Returns top most resolved address of |family|
>+  // Returns true iff the address from |Start| was successfully resolved.
>+  // If the address was successfully resolved, sets |addr| to a copy of the
>+  // address from |Start| with the IP address set to the top most resolved
>+  // address of |family| (|addr| will have both hostname and the resolved ip).
>   virtual bool GetResolvedAddress(int family, SocketAddress* addr) const = 0;
>   // Returns error from resolver.
>   virtual int GetError() const = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.cc
>index 3d68a2a46a6c637a8ed2fa7a2fbd9cb3e5ccf7dd..666b3357aaaf553fa30741ecb6507742866da2fb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.cc
>@@ -18,6 +18,7 @@
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/timeutils.h"  // for TimeMillis
> 
> #if defined(WEBRTC_POSIX)
> #include <errno.h>
>@@ -320,7 +321,7 @@ void AsyncTCPSocket::ProcessInput(char* data, size_t* len) {
>       return;
> 
>     SignalReadPacket(this, data + kPacketLenSize, pkt_len, remote_addr,
>-                     CreatePacketTime(0));
>+                     TimeMicros());
> 
>     *len -= kPacketLenSize + pkt_len;
>     if (*len > 0) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.h
>index 943e010f4dac019c1c812d8f406487cd2c1f5bda..9567dd91432b74c003e8d7b2d28278c3b2972d6b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.h
>@@ -11,12 +11,15 @@
> #ifndef RTC_BASE_ASYNCTCPSOCKET_H_
> #define RTC_BASE_ASYNCTCPSOCKET_H_
> 
>+#include <stddef.h>
> #include <memory>
> 
> #include "rtc_base/asyncpacketsocket.h"
>+#include "rtc_base/asyncsocket.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/socketfactory.h"
>+#include "rtc_base/socket.h"
>+#include "rtc_base/socketaddress.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.cc
>index ba5fa88aadb3e2f4d6f4124a934f144d502368eb..2f9011cab143aabd40acff0de9c551333ab3825d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.cc
>@@ -74,7 +74,6 @@ int AsyncUDPSocket::SendTo(const void* pv,
>   rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis(),
>                               options.info_signaled_after_sent);
>   CopySocketInformationToPacketInfo(cb, *this, true, &sent_packet.info);
>-  sent_packet.info.remote_socket_address = addr;
>   int ret = socket_->SendTo(pv, cb, addr);
>   SignalSentPacket(this, sent_packet);
>   return ret;
>@@ -123,9 +122,8 @@ void AsyncUDPSocket::OnReadEvent(AsyncSocket* socket) {
> 
>   // TODO: Make sure that we got all of the packet.
>   // If we did not, then we should resize our buffer to be large enough.
>-  SignalReadPacket(
>-      this, buf_, static_cast<size_t>(len), remote_addr,
>-      (timestamp > -1 ? PacketTime(timestamp, 0) : CreatePacketTime(0)));
>+  SignalReadPacket(this, buf_, static_cast<size_t>(len), remote_addr,
>+                   (timestamp > -1 ? timestamp : TimeMicros()));
> }
> 
> void AsyncUDPSocket::OnWriteEvent(AsyncSocket* socket) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.h
>index d814b4bfd0c5dbe5d7e4d4ac486c7b8fb56098f4..030946d8a03a5c6cf84b3f87b776c179bcc0927c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.h
>@@ -11,9 +11,13 @@
> #ifndef RTC_BASE_ASYNCUDPSOCKET_H_
> #define RTC_BASE_ASYNCUDPSOCKET_H_
> 
>+#include <stddef.h>
> #include <memory>
> 
> #include "rtc_base/asyncpacketsocket.h"
>+#include "rtc_base/asyncsocket.h"
>+#include "rtc_base/socket.h"
>+#include "rtc_base/socketaddress.h"
> #include "rtc_base/socketfactory.h"
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64_unittest.cc
>index 4b857f1aa94008419ed846fa5d33b88405f8440d..bdf8559c127f47b41af2582cdd5b537c98c94203 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64_unittest.cc
>@@ -11,7 +11,6 @@
> #include "rtc_base/third_party/base64/base64.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/stringutils.h"
> 
> #include "rtc_base/testbase64.h"
> 
>@@ -430,10 +429,10 @@ TEST(Base64, EncodeDecodeBattery) {
> 
>       // try putting some extra stuff after the equals signs, or in between them
>       if (equals == 2) {
>-        sprintfn(first_equals, 6, " = = ");
>+        snprintf(first_equals, 6, " = = ");
>         len = first_equals - encode_buffer + 5;
>       } else {
>-        sprintfn(first_equals, 6, " = ");
>+        snprintf(first_equals, 6, " = ");
>         len = first_equals - encode_buffer + 3;
>       }
>       decoded2.assign("this junk should be ignored");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy.cc
>index d2a06cd9efec81cc7a4eeab1036315d66dde7cd1..46e66741e0a0411555273dced76141a14dfd8a35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy.cc
>@@ -9,9 +9,31 @@
>  */
> 
> #include "rtc_base/bitrateallocationstrategy.h"
>+
> #include <algorithm>
>+#include <cstddef>
>+#include <cstdint>
>+#include <map>
> #include <utility>
> 
>+#include "rtc_base/numerics/safe_minmax.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+AudioPriorityConfig::AudioPriorityConfig()
>+    : min_rate("min"), max_rate("max"), target_rate("target") {
>+  std::string trial_string;
>+// TODO(bugs.webrtc.org/9889): Remove this when Chromium build has been fixed.
>+#if !defined(WEBRTC_CHROMIUM_BUILD)
>+  trial_string = field_trial::FindFullName("WebRTC-Bwe-AudioPriority");
>+#endif
>+  ParseFieldTrial({&min_rate, &max_rate, &target_rate}, trial_string);
>+}
>+AudioPriorityConfig::AudioPriorityConfig(const AudioPriorityConfig&) = default;
>+AudioPriorityConfig::~AudioPriorityConfig() = default;
>+
>+}  // namespace webrtc
>+
> namespace rtc {
> 
> // The purpose of this is to allow video streams to use extra bandwidth for FEC.
>@@ -21,31 +43,31 @@ namespace rtc {
> const int kTransmissionMaxBitrateMultiplier = 2;
> 
> std::vector<uint32_t> BitrateAllocationStrategy::SetAllBitratesToMinimum(
>-    const ArrayView<const TrackConfig*> track_configs) {
>+    const std::vector<BitrateAllocationStrategy::TrackConfig>& track_configs) {
>   std::vector<uint32_t> track_allocations;
>-  for (const auto* track_config : track_configs) {
>-    track_allocations.push_back(track_config->min_bitrate_bps);
>+  for (const auto& track_config : track_configs) {
>+    track_allocations.push_back(track_config.min_bitrate_bps);
>   }
>   return track_allocations;
> }
> 
> std::vector<uint32_t> BitrateAllocationStrategy::DistributeBitratesEvenly(
>-    const ArrayView<const TrackConfig*> track_configs,
>+    const std::vector<BitrateAllocationStrategy::TrackConfig>& track_configs,
>     uint32_t available_bitrate) {
>   std::vector<uint32_t> track_allocations =
>       SetAllBitratesToMinimum(track_configs);
>   uint32_t sum_min_bitrates = 0;
>   uint32_t sum_max_bitrates = 0;
>-  for (const auto* track_config : track_configs) {
>-    sum_min_bitrates += track_config->min_bitrate_bps;
>-    sum_max_bitrates += track_config->max_bitrate_bps;
>+  for (const auto& track_config : track_configs) {
>+    sum_min_bitrates += track_config.min_bitrate_bps;
>+    sum_max_bitrates += track_config.max_bitrate_bps;
>   }
>   if (sum_min_bitrates >= available_bitrate) {
>     return track_allocations;
>   } else if (available_bitrate >= sum_max_bitrates) {
>     auto track_allocations_it = track_allocations.begin();
>-    for (const auto* track_config : track_configs) {
>-      *track_allocations_it++ = track_config->max_bitrate_bps;
>+    for (const auto& track_config : track_configs) {
>+      *track_allocations_it++ = track_config.max_bitrate_bps;
>     }
>     return track_allocations;
>   } else {
>@@ -54,11 +76,10 @@ std::vector<uint32_t> BitrateAllocationStrategy::DistributeBitratesEvenly(
>     // lowest max_bitrate_bps. Remainder of available bitrate split evenly among
>     // remaining tracks.
>     std::multimap<uint32_t, size_t> max_bitrate_sorted_configs;
>-    for (const TrackConfig** track_configs_it = track_configs.begin();
>-         track_configs_it != track_configs.end(); ++track_configs_it) {
>+    for (const auto& track_config : track_configs) {
>       max_bitrate_sorted_configs.insert(
>-          std::make_pair((*track_configs_it)->max_bitrate_bps,
>-                         track_configs_it - track_configs.begin()));
>+          std::make_pair(track_config.max_bitrate_bps,
>+                         &track_config - &track_configs.front()));
>     }
>     uint32_t total_available_increase = available_bitrate - sum_min_bitrates;
>     int processed_configs = 0;
>@@ -67,8 +88,8 @@ std::vector<uint32_t> BitrateAllocationStrategy::DistributeBitratesEvenly(
>           total_available_increase /
>           (static_cast<uint32_t>(track_configs.size() - processed_configs));
>       uint32_t consumed_increase =
>-          std::min(track_configs[track_config_pair.second]->max_bitrate_bps -
>-                       track_configs[track_config_pair.second]->min_bitrate_bps,
>+          std::min(track_configs[track_config_pair.second].max_bitrate_bps -
>+                       track_configs[track_config_pair.second].min_bitrate_bps,
>                    available_increase);
>       track_allocations[track_config_pair.second] += consumed_increase;
>       total_available_increase -= consumed_increase;
>@@ -77,53 +98,50 @@ std::vector<uint32_t> BitrateAllocationStrategy::DistributeBitratesEvenly(
>     return track_allocations;
>   }
> }
>-
> AudioPriorityBitrateAllocationStrategy::AudioPriorityBitrateAllocationStrategy(
>     std::string audio_track_id,
>     uint32_t sufficient_audio_bitrate)
>     : audio_track_id_(audio_track_id),
>-      sufficient_audio_bitrate_(sufficient_audio_bitrate) {}
>+      sufficient_audio_bitrate_(sufficient_audio_bitrate) {
>+  if (config_.target_rate) {
>+    sufficient_audio_bitrate_ = config_.target_rate->bps();
>+  }
>+}
> 
> std::vector<uint32_t> AudioPriorityBitrateAllocationStrategy::AllocateBitrates(
>     uint32_t available_bitrate,
>-    const ArrayView<const TrackConfig*> track_configs) {
>-  const TrackConfig* audio_track_config = NULL;
>+    std::vector<BitrateAllocationStrategy::TrackConfig> track_configs) {
>+  TrackConfig* audio_track_config = nullptr;
>   size_t audio_config_index = 0;
>   uint32_t sum_min_bitrates = 0;
>   uint32_t sum_max_bitrates = 0;
> 
>-  for (const auto*& track_config : track_configs) {
>-    sum_min_bitrates += track_config->min_bitrate_bps;
>-    sum_max_bitrates += track_config->max_bitrate_bps;
>-    if (track_config->track_id == audio_track_id_) {
>-      audio_track_config = track_config;
>+  for (auto& track_config : track_configs) {
>+    if (track_config.track_id == audio_track_id_) {
>       audio_config_index = &track_config - &track_configs[0];
>+      audio_track_config = &track_config;
>+      if (config_.min_rate)
>+        audio_track_config->min_bitrate_bps = config_.min_rate->bps();
>+      if (config_.max_rate)
>+        audio_track_config->max_bitrate_bps = config_.max_rate->bps();
>     }
>+    sum_min_bitrates += track_config.min_bitrate_bps;
>+    sum_max_bitrates += track_config.max_bitrate_bps;
>   }
>   if (sum_max_bitrates < available_bitrate) {
>     // Allow non audio streams to go above max upto
>     // kTransmissionMaxBitrateMultiplier * max_bitrate_bps
>-    size_t track_configs_size = track_configs.size();
>-    std::vector<TrackConfig> increased_track_configs(track_configs_size);
>-    std::vector<const TrackConfig*> increased_track_configs_ptr(
>-        track_configs_size);
>-    for (unsigned long i = 0; i < track_configs_size; i++) {
>-      increased_track_configs[i] = (*track_configs[i]);
>-      increased_track_configs_ptr[i] = &increased_track_configs[i];
>-      if (track_configs[i]->track_id != audio_track_id_) {
>-        increased_track_configs[i].max_bitrate_bps =
>-            track_configs[i]->max_bitrate_bps *
>-            kTransmissionMaxBitrateMultiplier;
>-      }
>+    for (auto& track_config : track_configs) {
>+      if (&track_config != audio_track_config)
>+        track_config.max_bitrate_bps *= kTransmissionMaxBitrateMultiplier;
>     }
>-    return DistributeBitratesEvenly(increased_track_configs_ptr,
>-                                    available_bitrate);
>+    return DistributeBitratesEvenly(track_configs, available_bitrate);
>   }
>-  if (audio_track_config == nullptr) {
>+  if (!audio_track_config) {
>     return DistributeBitratesEvenly(track_configs, available_bitrate);
>   }
>-  auto safe_sufficient_audio_bitrate = std::min(
>-      std::max(audio_track_config->min_bitrate_bps, sufficient_audio_bitrate_),
>+  auto safe_sufficient_audio_bitrate = rtc::SafeClamp(
>+      sufficient_audio_bitrate_, audio_track_config->min_bitrate_bps,
>       audio_track_config->max_bitrate_bps);
>   if (available_bitrate <= sum_min_bitrates) {
>     return SetAllBitratesToMinimum(track_configs);
>@@ -139,9 +157,7 @@ std::vector<uint32_t> AudioPriorityBitrateAllocationStrategy::AllocateBitrates(
>       // Setting audio track minimum to safe_sufficient_audio_bitrate will
>       // allow using DistributeBitratesEvenly to allocate at least sufficient
>       // bitrate for audio and the rest evenly.
>-      TrackConfig sufficient_track_config(*track_configs[audio_config_index]);
>-      sufficient_track_config.min_bitrate_bps = safe_sufficient_audio_bitrate;
>-      track_configs[audio_config_index] = &sufficient_track_config;
>+      audio_track_config->min_bitrate_bps = safe_sufficient_audio_bitrate;
>       std::vector<uint32_t> track_allocations =
>           DistributeBitratesEvenly(track_configs, available_bitrate);
>       return track_allocations;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy.h
>index f711d1f18ffd814eeb16b8d349994f9aca1342c2..13a4eee524e2e98b1c767c95ddfa72174be0db06 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy.h
>@@ -11,12 +11,13 @@
> #ifndef RTC_BASE_BITRATEALLOCATIONSTRATEGY_H_
> #define RTC_BASE_BITRATEALLOCATIONSTRATEGY_H_
> 
>-#include <map>
>-#include <memory>
>+#include <stdint.h>
> #include <string>
> #include <vector>
>+
> #include "api/array_view.h"
>-#include "rtc_base/checks.h"
>+#include "rtc_base/experiments/field_trial_parser.h"
>+#include "rtc_base/experiments/field_trial_units.h"
> 
> namespace rtc {
> 
>@@ -56,10 +57,12 @@ class BitrateAllocationStrategy {
>     std::string track_id;
>   };
> 
>+  // These are only used by AudioPriorityBitrateAllocationStrategy. They are
>+  // exposed here to they can be unit tested.
>   static std::vector<uint32_t> SetAllBitratesToMinimum(
>-      const ArrayView<const TrackConfig*> track_configs);
>+      const std::vector<BitrateAllocationStrategy::TrackConfig>& track_configs);
>   static std::vector<uint32_t> DistributeBitratesEvenly(
>-      const ArrayView<const TrackConfig*> track_configs,
>+      const std::vector<BitrateAllocationStrategy::TrackConfig>& track_configs,
>       uint32_t available_bitrate);
> 
>   // Strategy is expected to allocate all available_bitrate up to the sum of
>@@ -74,11 +77,25 @@ class BitrateAllocationStrategy {
>   // available_bitrate decrease.
>   virtual std::vector<uint32_t> AllocateBitrates(
>       uint32_t available_bitrate,
>-      const ArrayView<const TrackConfig*> track_configs) = 0;
>+      std::vector<BitrateAllocationStrategy::TrackConfig> track_configs) = 0;
> 
>   virtual ~BitrateAllocationStrategy() = default;
> };
>+}  // namespace rtc
> 
>+namespace webrtc {
>+struct AudioPriorityConfig {
>+  FieldTrialOptional<DataRate> min_rate;
>+  FieldTrialOptional<DataRate> max_rate;
>+  FieldTrialOptional<DataRate> target_rate;
>+  AudioPriorityConfig();
>+  AudioPriorityConfig(const AudioPriorityConfig&);
>+  AudioPriorityConfig& operator=(const AudioPriorityConfig&) = default;
>+  ~AudioPriorityConfig();
>+};
>+}  // namespace webrtc
>+
>+namespace rtc {
> // Simple allocation strategy giving priority to audio until
> // sufficient_audio_bitrate is reached. Bitrate is distributed evenly between
> // the tracks after sufficient_audio_bitrate is reached. This implementation
>@@ -90,9 +107,11 @@ class AudioPriorityBitrateAllocationStrategy
>                                          uint32_t sufficient_audio_bitrate);
>   std::vector<uint32_t> AllocateBitrates(
>       uint32_t available_bitrate,
>-      const ArrayView<const TrackConfig*> track_configs) override;
>+      std::vector<BitrateAllocationStrategy::TrackConfig> track_configs)
>+      override;
> 
>  private:
>+  webrtc::AudioPriorityConfig config_;
>   std::string audio_track_id_;
>   uint32_t sufficient_audio_bitrate_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy_unittest.cc
>index bfc41f56e80102afd875096895d0a06d3cd2bc39..f4c7ee7043b4646a69f31b846c12d673ed784f67 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitrateallocationstrategy_unittest.cc
>@@ -43,11 +43,8 @@ TEST(BitrateAllocationStrategyTest, SetAllBitratesToMinimum) {
>       BitrateAllocationStrategy::TrackConfig(min_other_bitrate,
>                                              max_other_bitrate, false, "")};
> 
>-  std::vector<const rtc::BitrateAllocationStrategy::TrackConfig*>
>-      track_config_ptrs = MakeTrackConfigPtrsVector(track_configs);
>-
>   std::vector<uint32_t> allocations =
>-      BitrateAllocationStrategy::SetAllBitratesToMinimum(track_config_ptrs);
>+      BitrateAllocationStrategy::SetAllBitratesToMinimum(track_configs);
>   EXPECT_EQ(min_audio_bitrate, allocations[0]);
>   EXPECT_EQ(min_video_bitrate, allocations[1]);
>   EXPECT_EQ(min_other_bitrate, allocations[2]);
>@@ -76,11 +73,8 @@ TEST(BitrateAllocationStrategyTest, DistributeBitratesEvenly) {
>       BitrateAllocationStrategy::TrackConfig(min_other_bitrate,
>                                              max_other_bitrate, false, "")};
> 
>-  std::vector<const rtc::BitrateAllocationStrategy::TrackConfig*>
>-      track_config_ptrs = MakeTrackConfigPtrsVector(track_configs);
>-
>   std::vector<uint32_t> allocations =
>-      BitrateAllocationStrategy::DistributeBitratesEvenly(track_config_ptrs,
>+      BitrateAllocationStrategy::DistributeBitratesEvenly(track_configs,
>                                                           available_bitrate);
>   EXPECT_EQ(min_audio_bitrate + even_bitrate_increase, allocations[0]);
>   EXPECT_EQ(min_video_bitrate + even_bitrate_increase, allocations[1]);
>@@ -108,11 +102,7 @@ std::vector<uint32_t> RunAudioPriorityAllocation(
>       BitrateAllocationStrategy::TrackConfig(min_other_bitrate,
>                                              max_other_bitrate, false, "")};
> 
>-  std::vector<const rtc::BitrateAllocationStrategy::TrackConfig*>
>-      track_config_ptrs = MakeTrackConfigPtrsVector(track_configs);
>-
>-  return allocation_strategy.AllocateBitrates(available_bitrate,
>-                                              track_config_ptrs);
>+  return allocation_strategy.AllocateBitrates(available_bitrate, track_configs);
> }
> 
> // Test that when the available bitrate is less than the sum of the minimum
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer.h
>index 22f89372debc16d55d6c8c4c37f8f0dda3569114..f9291b99e5a103244c7d6c46cba82641e96eb176 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer.h
>@@ -149,11 +149,13 @@ class BufferT {
>   }
> 
>   BufferT& operator=(BufferT&& buf) {
>-    RTC_DCHECK(IsConsistent());
>     RTC_DCHECK(buf.IsConsistent());
>+    MaybeZeroCompleteBuffer();
>     size_ = buf.size_;
>     capacity_ = buf.capacity_;
>-    data_ = std::move(buf.data_);
>+    using std::swap;
>+    swap(data_, buf.data_);
>+    buf.data_.reset();
>     buf.OnMovedFrom();
>     return *this;
>   }
>@@ -374,10 +376,10 @@ class BufferT {
> 
>   // Zero the complete buffer if template argument "ZeroOnFree" is true.
>   void MaybeZeroCompleteBuffer() {
>-    if (ZeroOnFree && capacity_) {
>+    if (ZeroOnFree && capacity_ > 0) {
>       // It would be sufficient to only zero "size_" elements, as all other
>       // methods already ensure that the unused capacity contains no sensitive
>-      // data - but better safe than sorry.
>+      // data---but better safe than sorry.
>       ExplicitZeroMemory(data_.get(), capacity_ * sizeof(T));
>     }
>   }
>@@ -389,7 +391,7 @@ class BufferT {
>     ExplicitZeroMemory(data_.get() + size_, count * sizeof(T));
>   }
> 
>-  // Precondition for all methods except Clear and the destructor.
>+  // Precondition for all methods except Clear, operator= and the destructor.
>   // Postcondition for all methods except move construction and move
>   // assignment, which leave the moved-from object in a possibly inconsistent
>   // state.
>@@ -400,15 +402,16 @@ class BufferT {
>   // Called when *this has been moved from. Conceptually it's a no-op, but we
>   // can mutate the state slightly to help subsequent sanity checks catch bugs.
>   void OnMovedFrom() {
>+    RTC_DCHECK(!data_);  // Our heap block should have been stolen.
> #if RTC_DCHECK_IS_ON
>+    // Ensure that *this is always inconsistent, to provoke bugs.
>+    size_ = 1;
>+    capacity_ = 0;
>+#else
>     // Make *this consistent and empty. Shouldn't be necessary, but better safe
>     // than sorry.
>     size_ = 0;
>     capacity_ = 0;
>-#else
>-    // Ensure that *this is always inconsistent, to provoke bugs.
>-    size_ = 1;
>-    capacity_ = 0;
> #endif
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer_unittest.cc
>index ae976f1394322f11b1a9496466d7385e2bac94aa..b2f47c16ecbe3f3df8d1fc683f6da80ca3f8d02b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer_unittest.cc
>@@ -185,6 +185,17 @@ TEST(BufferTest, TestMoveAssign) {
>   EXPECT_TRUE(buf1.empty());
> }
> 
>+TEST(BufferTest, TestMoveAssignSelf) {
>+  // Move self-assignment isn't required to produce a meaningful state, but
>+  // should not leave the object in an inconsistent state. (Such inconsistent
>+  // state could be caught by the DCHECKs and/or by the leak checker.) We need
>+  // to be sneaky when testing this; if we're doing a too-obvious
>+  // move-assign-to-self, clang's -Wself-move triggers at compile time.
>+  Buffer buf(kTestData, 3, 40);
>+  Buffer* buf_ptr = &buf;
>+  buf = std::move(*buf_ptr);
>+}
>+
> TEST(BufferTest, TestSwap) {
>   Buffer buf1(kTestData, 3);
>   Buffer buf2(kTestData, 6, 40);
>@@ -434,6 +445,19 @@ TEST(BufferTest, TestStruct) {
>   EXPECT_EQ(kObsidian, buf[2].stone);
> }
> 
>+TEST(BufferTest, DieOnUseAfterMove) {
>+  Buffer buf(17);
>+  Buffer buf2 = std::move(buf);
>+  EXPECT_EQ(buf2.size(), 17u);
>+#if RTC_DCHECK_IS_ON
>+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>+  EXPECT_DEATH(buf.empty(), "");
>+#endif
>+#else
>+  EXPECT_TRUE(buf.empty());
>+#endif
>+}
>+
> TEST(ZeroOnFreeBufferTest, TestZeroOnSetData) {
>   ZeroOnFreeBuffer<uint8_t> buf(kTestData, 7);
>   const uint8_t* old_data = buf.data();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.cc
>index 48ff2e6d317d78208a6b330efb398add64e3df7c..74f7a502c0b910e61f5aaa8bc109e6334b8a918a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.cc
>@@ -10,6 +10,8 @@
> 
> #include "rtc_base/bufferqueue.h"
> 
>+#include <stdint.h>
>+#include <string.h>
> #include <algorithm>
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.h
>index 94ab0ca4c9eb0cf3bb3f964f7e6cc68a39e1d5d7..63f5182509f62204be01115798b90f1e0a8e2560 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.h
>@@ -11,12 +11,14 @@
> #ifndef RTC_BASE_BUFFERQUEUE_H_
> #define RTC_BASE_BUFFERQUEUE_H_
> 
>+#include <stddef.h>
> #include <deque>
> #include <vector>
> 
> #include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.cc
>index 94fc6acf08027412a25ed7367320f76e21ddad10..f8ce1a29efe16687ac654fe9afcb25edfaa308f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.cc
>@@ -12,8 +12,6 @@
> 
> #include <string.h>
> 
>-#include <algorithm>
>-
> namespace rtc {
> 
> ByteBufferWriter::ByteBufferWriter() : ByteBufferWriterT() {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.h
>index 9e08f025a84237e944a1ff3e538baf8972dce293..4d25c210507d731059f83bd53abd9737be4d0f84 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.h
>@@ -11,6 +11,8 @@
> #ifndef RTC_BASE_BYTEBUFFER_H_
> #define RTC_BASE_BYTEBUFFER_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <string>
> 
> #include "rtc_base/buffer.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cancelable_periodic_task_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cancelable_periodic_task_unittest.cc
>index fe27ea7d48b656ec5e98b9efaf56efc2668b9947..badd623415ad55319f72ddffffbf1b3bc2e4b768 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cancelable_periodic_task_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cancelable_periodic_task_unittest.cc
>@@ -68,7 +68,7 @@ TEST(CancelablePeriodicTaskTest, CanCallCancelOnEmptyHandle) {
> }
> 
> TEST(CancelablePeriodicTaskTest, CancelTaskBeforeItRuns) {
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   MockClosure mock;
>   EXPECT_CALL(mock, Call).Times(0);
>   EXPECT_CALL(mock, Delete).WillOnce(Invoke([&done] { done.Set(); }));
>@@ -84,7 +84,7 @@ TEST(CancelablePeriodicTaskTest, CancelTaskBeforeItRuns) {
> }
> 
> TEST(CancelablePeriodicTaskTest, CancelDelayedTaskBeforeItRuns) {
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   MockClosure mock;
>   EXPECT_CALL(mock, Call).Times(0);
>   EXPECT_CALL(mock, Delete).WillOnce(Invoke([&done] { done.Set(); }));
>@@ -100,7 +100,7 @@ TEST(CancelablePeriodicTaskTest, CancelDelayedTaskBeforeItRuns) {
> }
> 
> TEST(CancelablePeriodicTaskTest, CancelTaskAfterItRuns) {
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   MockClosure mock;
>   EXPECT_CALL(mock, Call).WillOnce(Return(100));
>   EXPECT_CALL(mock, Delete).WillOnce(Invoke([&done] { done.Set(); }));
>@@ -117,7 +117,7 @@ TEST(CancelablePeriodicTaskTest, CancelTaskAfterItRuns) {
> 
> TEST(CancelablePeriodicTaskTest, ZeroReturnValueRepostsTheTask) {
>   NiceMock<MockClosure> closure;
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   EXPECT_CALL(closure, Call()).WillOnce(Return(0)).WillOnce(Invoke([&done] {
>     done.Set();
>     return kTimeoutMs;
>@@ -130,7 +130,7 @@ TEST(CancelablePeriodicTaskTest, ZeroReturnValueRepostsTheTask) {
> 
> TEST(CancelablePeriodicTaskTest, StartPeriodicTask) {
>   MockFunction<int()> closure;
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   EXPECT_CALL(closure, Call())
>       .WillOnce(Return(20))
>       .WillOnce(Return(20))
>@@ -146,7 +146,7 @@ TEST(CancelablePeriodicTaskTest, StartPeriodicTask) {
> 
> // Validates perfect forwarding doesn't keep reference to deleted copy.
> TEST(CancelablePeriodicTaskTest, CreateWithCopyOfAClosure) {
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   MockClosure mock;
>   EXPECT_CALL(mock, Call).WillOnce(Invoke([&done] {
>     done.Set();
>@@ -166,7 +166,7 @@ TEST(CancelablePeriodicTaskTest, CreateWithCopyOfAClosure) {
> }
> 
> TEST(CancelablePeriodicTaskTest, DeletingHandleDoesntStopTheTask) {
>-  rtc::Event run(false, false);
>+  rtc::Event run;
>   rtc::TaskQueue task_queue("queue");
>   auto task = rtc::CreateCancelablePeriodicTask(([&] {
>     run.Set();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.cc
>index 6c48d52f17172ee8c4cfaefad3e9051ccb420933..8f5126a28268cf198acb4cda9ff4ac74981e77b7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.cc
>@@ -10,6 +10,8 @@
> 
> #include "rtc_base/copyonwritebuffer.h"
> 
>+#include <stddef.h>
>+
> namespace rtc {
> 
> CopyOnWriteBuffer::CopyOnWriteBuffer() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.h
>index 0514e2fe6611a25966fd633e8f692f4623a2f5cf..cc174dfc7c969e728995de3d34daa4ac8d2833cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.h
>@@ -11,12 +11,15 @@
> #ifndef RTC_BASE_COPYONWRITEBUFFER_H_
> #define RTC_BASE_COPYONWRITEBUFFER_H_
> 
>+#include <stdint.h>
> #include <algorithm>
>+#include <cstring>
>+#include <string>
>+#include <type_traits>
> #include <utility>
> 
> #include "rtc_base/buffer.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/refcount.h"
> #include "rtc_base/refcountedobject.h"
> #include "rtc_base/scoped_ref_ptr.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time.cc
>index de4a6bd6844b1eb4322b417d175fa0f166c8ee23..ad91ecace973195c495bb22791425929d33362ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time.cc
>@@ -65,6 +65,9 @@ int64_t GetProcessCpuTimeNanos() {
>   } else {
>     RTC_LOG_ERR(LS_ERROR) << "GetProcessTimes() failed.";
>   }
>+#elif defined(WEBRTC_FUCHSIA)
>+  RTC_LOG_ERR(LS_ERROR) << "GetProcessCpuTimeNanos() not implemented";
>+  return 0;
> #else
>   // Not implemented yet.
>   static_assert(
>@@ -107,10 +110,13 @@ int64_t GetThreadCpuTimeNanos() {
>   } else {
>     RTC_LOG_ERR(LS_ERROR) << "GetThreadTimes() failed.";
>   }
>+#elif defined(WEBRTC_FUCHSIA)
>+  RTC_LOG_ERR(LS_ERROR) << "GetThreadCpuTimeNanos() not implemented";
>+  return 0;
> #else
>   // Not implemented yet.
>   static_assert(
>-      false, "GetProcessCpuTimeNanos() platform support not yet implemented.");
>+      false, "GetThreadCpuTimeNanos() platform support not yet implemented.");
> #endif
>   return -1;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.cc
>index d8a5b48c0513902587ba21954f60847c516a395a..4e00be968de902ad2adfa7ee41020dfebf219ba0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.cc
>@@ -10,6 +10,8 @@
> 
> #include "rtc_base/criticalsection.h"
> 
>+#include <time.h>
>+
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/platform_thread_types.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection_unittest.cc
>index db4f9e7452f57c660c7d953241b38c9d20f09804..6016f85f898e5e48ea6505dda59ce8a6587bf23c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection_unittest.cc
>@@ -389,7 +389,7 @@ class PerfTestThread {
> // The test is disabled by default to avoid unecessarily loading the bots.
> TEST(CriticalSectionTest, DISABLED_Performance) {
>   PerfTestThread threads[8];
>-  Event event(false, false);
>+  Event event;
> 
>   static const int kThreadRepeats = 10000000;
>   static const int kExpectedCount = kThreadRepeats * arraysize(threads);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.cc
>index 6c9639b4fecc111a807eeaa18aef03ba71816baa..42c22a29ccb0968a837deee9db6bddec1e4e0ab2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.cc
>@@ -24,6 +24,8 @@
> 
> namespace rtc {
> 
>+Event::Event() : Event(false, false) {}
>+
> #if defined(WEBRTC_WIN)
> 
> Event::Event(bool manual_reset, bool initially_signaled) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.h
>index 7e61c4c8a6878fdea13eb700a39c025cdf1caa5b..2e11002066da91fc59184aa90f156e1f525daf1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.h
>@@ -11,7 +11,6 @@
> #ifndef RTC_BASE_EVENT_H_
> #define RTC_BASE_EVENT_H_
> 
>-#include "rtc_base/constructormagic.h"
> #if defined(WEBRTC_WIN)
> #include <windows.h>
> #elif defined(WEBRTC_POSIX)
>@@ -26,7 +25,10 @@ class Event {
>  public:
>   static const int kForever = -1;
> 
>+  Event();
>   Event(bool manual_reset, bool initially_signaled);
>+  Event(const Event&) = delete;
>+  Event& operator=(const Event&) = delete;
>   ~Event();
> 
>   void Set();
>@@ -45,8 +47,6 @@ class Event {
>   const bool is_manual_reset_;
>   bool event_status_;
> #endif
>-
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Event);
> };
> 
> // This class is provided for compatibility with Chromium.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.cc
>index 31f4271d31e2a39db7b88413dc1868ea37cb7f2b..af88c9ddd4c101cd89f1dcbf486c03bfb4982fc5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.cc
>@@ -10,7 +10,9 @@
> #include "rtc_base/event_tracer.h"
> 
> #include <inttypes.h>
>-
>+#include <stdint.h>
>+#include <stdio.h>
>+#include <string.h>
> #include <string>
> #include <vector>
> 
>@@ -20,7 +22,9 @@
> #include "rtc_base/event.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/platform_thread.h"
>-#include "rtc_base/stringutils.h"
>+#include "rtc_base/platform_thread_types.h"
>+#include "rtc_base/thread_annotations.h"
>+#include "rtc_base/thread_checker.h"
> #include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
> 
>@@ -86,8 +90,7 @@ class EventLogger final {
>       : logging_thread_(EventTracingThreadFunc,
>                         this,
>                         "EventTracingThread",
>-                        kLowPriority),
>-        shutdown_event_(false, false) {}
>+                        kLowPriority) {}
>   ~EventLogger() { RTC_DCHECK(thread_checker_.CalledOnValidThread()); }
> 
>   void AddTraceEvent(const char* name,
>@@ -286,19 +289,19 @@ class EventLogger final {
>           }
>           break;
>         case TRACE_VALUE_TYPE_UINT:
>-          print_length = sprintfn(&output[0], kTraceArgBufferLength, "%llu",
>+          print_length = snprintf(&output[0], kTraceArgBufferLength, "%llu",
>                                   arg.value.as_uint);
>           break;
>         case TRACE_VALUE_TYPE_INT:
>-          print_length = sprintfn(&output[0], kTraceArgBufferLength, "%lld",
>+          print_length = snprintf(&output[0], kTraceArgBufferLength, "%lld",
>                                   arg.value.as_int);
>           break;
>         case TRACE_VALUE_TYPE_DOUBLE:
>-          print_length = sprintfn(&output[0], kTraceArgBufferLength, "%f",
>+          print_length = snprintf(&output[0], kTraceArgBufferLength, "%f",
>                                   arg.value.as_double);
>           break;
>         case TRACE_VALUE_TYPE_POINTER:
>-          print_length = sprintfn(&output[0], kTraceArgBufferLength, "\"%p\"",
>+          print_length = snprintf(&output[0], kTraceArgBufferLength, "\"%p\"",
>                                   arg.value.as_pointer);
>           break;
>       }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_unittest.cc
>index 050619e849541cac07d090efd7bfc0f3b50b8d40..a65111bc90f7f3837e38d2378adcc40a8d896b75 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_unittest.cc
>@@ -32,7 +32,7 @@ TEST(EventTest, ManualReset) {
> }
> 
> TEST(EventTest, AutoReset) {
>-  Event event(false, false);
>+  Event event;
>   ASSERT_FALSE(event.Wait(0));
> 
>   event.Set();
>@@ -59,7 +59,7 @@ class SignalerThread {
>       me->reader_->Wait(Event::kForever);
>     }
>   }
>-  Event stop_event_{false, false};
>+  Event stop_event_;
>   Event* writer_;
>   Event* reader_;
>   PlatformThread thread_;
>@@ -68,7 +68,7 @@ class SignalerThread {
> // These tests are disabled by default and only intended to be run manually.
> TEST(EventTest, DISABLED_PerformanceSingleThread) {
>   static const int kNumIterations = 10000000;
>-  Event event(false, false);
>+  Event event;
>   for (int i = 0; i < kNumIterations; ++i) {
>     event.Set();
>     event.Wait(0);
>@@ -77,8 +77,8 @@ TEST(EventTest, DISABLED_PerformanceSingleThread) {
> 
> TEST(EventTest, DISABLED_PerformanceMultiThread) {
>   static const int kNumIterations = 10000;
>-  Event read(false, false);
>-  Event write(false, false);
>+  Event read;
>+  Event write;
>   SignalerThread thread;
>   thread.Start(&read, &write);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/BUILD.gn
>index b10d0c154592062671a9f2ac9d8a92f5036826da..d36a43e9469055edcc4df695a3ad15884ebf7317 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/BUILD.gn
>@@ -28,11 +28,11 @@ rtc_static_library("field_trial_parser") {
>     "field_trial_units.h",
>   ]
>   deps = [
>-    "../:rtc_base_approved",
>     "../../api/units:data_rate",
>     "../../api/units:data_size",
>     "../../api/units:time_delta",
>     "../../rtc_base:checks",
>+    "../../rtc_base:logging",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>@@ -56,13 +56,36 @@ rtc_static_library("quality_scaling_experiment") {
>   ]
>   deps = [
>     "../:rtc_base_approved",
>-    "../..:webrtc_common",
>     "../../api/video_codecs:video_codecs_api",
>     "../../system_wrappers:field_trial",
>     "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>+rtc_static_library("normalize_simulcast_size_experiment") {
>+  sources = [
>+    "normalize_simulcast_size_experiment.cc",
>+    "normalize_simulcast_size_experiment.h",
>+  ]
>+  deps = [
>+    "../:rtc_base_approved",
>+    "../../system_wrappers:field_trial",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_static_library("cpu_speed_experiment") {
>+  sources = [
>+    "cpu_speed_experiment.cc",
>+    "cpu_speed_experiment.h",
>+  ]
>+  deps = [
>+    "../:rtc_base_approved",
>+    "../../system_wrappers:field_trial",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
> rtc_static_library("rtt_mult_experiment") {
>   sources = [
>     "rtt_mult_experiment.cc",
>@@ -74,26 +97,44 @@ rtc_static_library("rtt_mult_experiment") {
>   ]
> }
> 
>+rtc_static_library("jitter_upper_bound_experiment") {
>+  sources = [
>+    "jitter_upper_bound_experiment.cc",
>+    "jitter_upper_bound_experiment.h",
>+  ]
>+  deps = [
>+    "../:rtc_base_approved",
>+    "../../system_wrappers:field_trial",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
> if (rtc_include_tests) {
>   rtc_source_set("experiments_unittests") {
>     testonly = true
> 
>     sources = [
>       "congestion_controller_experiment_unittest.cc",
>+      "cpu_speed_experiment_unittest.cc",
>       "field_trial_parser_unittest.cc",
>       "field_trial_units_unittest.cc",
>+      "normalize_simulcast_size_experiment_unittest.cc",
>       "quality_scaling_experiment_unittest.cc",
>       "rtt_mult_experiment_unittest.cc",
>     ]
>     deps = [
>       ":congestion_controller_experiment",
>+      ":cpu_speed_experiment",
>       ":field_trial_parser",
>+      ":normalize_simulcast_size_experiment",
>       ":quality_scaling_experiment",
>       ":rtt_mult_experiment",
>+      "..:gunit_helpers",
>       "../:rtc_base_tests_main",
>       "../:rtc_base_tests_utils",
>       "../../system_wrappers:field_trial",
>       "../../test:field_trial",
>+      "../../test:test_support",
>     ]
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/OWNERS
>new file mode 100644
>index 0000000000000000000000000000000000000000..9c5587226f8da22455e014ed961a112ce1fa8e0c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/OWNERS
>@@ -0,0 +1,8 @@
>+per-file alr_experiment*=sprang@webrtc.org
>+per-file congestion_controller_experiment*=srte@webrtc.org
>+per-file cpu_speed_experiment*=asapersson@webrtc.org
>+per-file field_trial*=srte@webrtc.org
>+per-file jitter_upper_bound_experiment*=sprang@webrtc.org
>+per-file normalize_simulcast_size_experiment*=asapersson@webrtc.org
>+per-file quality_scaling_experiment*=asapersson@webrtc.org
>+per-file rtt_mult_experiment*=mhoro@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.cc
>index dff5acecc8b7566a08c1569ee70ff1508e267444..25e948d98a0e7684bacd399db662a6985d57d37b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.cc
>@@ -10,9 +10,10 @@
> 
> #include "rtc_base/experiments/alr_experiment.h"
> 
>+#include <inttypes.h>
>+#include <stdio.h>
> #include <string>
> 
>-#include "rtc_base/format_macros.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/field_trial.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.h
>index 4d9fd001304965542fd558d144127ea3377842fe..876bd0251730b6a040cca3ae5f91031135ce9458 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.h
>@@ -11,6 +11,8 @@
> #ifndef RTC_BASE_EXPERIMENTS_ALR_EXPERIMENT_H_
> #define RTC_BASE_EXPERIMENTS_ALR_EXPERIMENT_H_
> 
>+#include <stdint.h>
>+
> #include "absl/types/optional.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/cpu_speed_experiment.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/cpu_speed_experiment.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..f39540cf0390f1b99894acd1e4f04f4176c4fd4f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/cpu_speed_experiment.cc
>@@ -0,0 +1,70 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/experiments/cpu_speed_experiment.h"
>+
>+#include <string>
>+
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+constexpr char kFieldTrial[] = "WebRTC-VP8-CpuSpeed-Arm";
>+constexpr int kMinSetting = -16;
>+constexpr int kMaxSetting = -1;
>+}  // namespace
>+
>+absl::optional<std::vector<CpuSpeedExperiment::Config>>
>+CpuSpeedExperiment::GetConfigs() {
>+  if (!webrtc::field_trial::IsEnabled(kFieldTrial))
>+    return absl::nullopt;
>+
>+  const std::string group = webrtc::field_trial::FindFullName(kFieldTrial);
>+  if (group.empty())
>+    return absl::nullopt;
>+
>+  std::vector<Config> configs(3);
>+  if (sscanf(group.c_str(), "Enabled-%d,%d,%d,%d,%d,%d", &(configs[0].pixels),
>+             &(configs[0].cpu_speed), &(configs[1].pixels),
>+             &(configs[1].cpu_speed), &(configs[2].pixels),
>+             &(configs[2].cpu_speed)) != 6) {
>+    RTC_LOG(LS_WARNING) << "Too few parameters provided.";
>+    return absl::nullopt;
>+  }
>+
>+  for (const auto& config : configs) {
>+    if (config.cpu_speed < kMinSetting || config.cpu_speed > kMaxSetting) {
>+      RTC_LOG(LS_WARNING) << "Unsupported cpu speed setting, value ignored.";
>+      return absl::nullopt;
>+    }
>+  }
>+
>+  for (size_t i = 1; i < configs.size(); ++i) {
>+    if (configs[i].pixels < configs[i - 1].pixels ||
>+        configs[i].cpu_speed > configs[i - 1].cpu_speed) {
>+      RTC_LOG(LS_WARNING) << "Invalid parameter value provided.";
>+      return absl::nullopt;
>+    }
>+  }
>+
>+  return absl::optional<std::vector<Config>>(configs);
>+}
>+
>+int CpuSpeedExperiment::GetValue(int pixels,
>+                                 const std::vector<Config>& configs) {
>+  for (const auto& config : configs) {
>+    if (pixels <= config.pixels)
>+      return config.cpu_speed;
>+  }
>+  return kMinSetting;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/cpu_speed_experiment.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/cpu_speed_experiment.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..e6c83409430141e5701ae20ba144c7d672ce9d9b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/cpu_speed_experiment.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_EXPERIMENTS_CPU_SPEED_EXPERIMENT_H_
>+#define RTC_BASE_EXPERIMENTS_CPU_SPEED_EXPERIMENT_H_
>+
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+
>+namespace webrtc {
>+
>+class CpuSpeedExperiment {
>+ public:
>+  struct Config {
>+    bool operator==(const Config& o) const {
>+      return pixels == o.pixels && cpu_speed == o.cpu_speed;
>+    }
>+
>+    int pixels;     // The video frame size.
>+    int cpu_speed;  // The |cpu_speed| to be used if the frame size is less
>+                    // than or equal to |pixels|.
>+  };
>+
>+  // Returns the configurations from field trial on success.
>+  static absl::optional<std::vector<Config>> GetConfigs();
>+
>+  // Gets the cpu speed from the |configs| based on |pixels|.
>+  static int GetValue(int pixels, const std::vector<Config>& configs);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // RTC_BASE_EXPERIMENTS_CPU_SPEED_EXPERIMENT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/cpu_speed_experiment_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/cpu_speed_experiment_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..edc782c0addbd694421e3bbb59cb478bb32eec2f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/cpu_speed_experiment_unittest.cc
>@@ -0,0 +1,85 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/experiments/cpu_speed_experiment.h"
>+
>+#include "rtc_base/gunit.h"
>+#include "test/field_trial.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+TEST(CpuSpeedExperimentTest, GetConfigsFailsIfNotEnabled) {
>+  EXPECT_FALSE(CpuSpeedExperiment::GetConfigs());
>+}
>+
>+TEST(CpuSpeedExperimentTest, GetConfigsFailsForTooFewParameters) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-VP8-CpuSpeed-Arm/Enabled-1000,-1,2000,-10,3000/");
>+  EXPECT_FALSE(CpuSpeedExperiment::GetConfigs());
>+}
>+
>+TEST(CpuSpeedExperimentTest, GetConfigs) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-VP8-CpuSpeed-Arm/Enabled-1000,-1,2000,-10,3000,-16/");
>+
>+  const absl::optional<std::vector<CpuSpeedExperiment::Config>> kConfigs =
>+      CpuSpeedExperiment::GetConfigs();
>+  ASSERT_TRUE(kConfigs);
>+  EXPECT_THAT(*kConfigs,
>+              ::testing::ElementsAre(CpuSpeedExperiment::Config{1000, -1},
>+                                     CpuSpeedExperiment::Config{2000, -10},
>+                                     CpuSpeedExperiment::Config{3000, -16}));
>+}
>+
>+TEST(CpuSpeedExperimentTest, GetValue) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-VP8-CpuSpeed-Arm/Enabled-1000,-5,2000,-10,3000,-12/");
>+
>+  const absl::optional<std::vector<CpuSpeedExperiment::Config>> kConfigs =
>+      CpuSpeedExperiment::GetConfigs();
>+  ASSERT_TRUE(kConfigs);
>+  ASSERT_EQ(3u, (*kConfigs).size());
>+  EXPECT_EQ(-5, CpuSpeedExperiment::GetValue(1, *kConfigs));
>+  EXPECT_EQ(-5, CpuSpeedExperiment::GetValue(1000, *kConfigs));
>+  EXPECT_EQ(-10, CpuSpeedExperiment::GetValue(1000 + 1, *kConfigs));
>+  EXPECT_EQ(-10, CpuSpeedExperiment::GetValue(2000, *kConfigs));
>+  EXPECT_EQ(-12, CpuSpeedExperiment::GetValue(2000 + 1, *kConfigs));
>+  EXPECT_EQ(-12, CpuSpeedExperiment::GetValue(3000, *kConfigs));
>+  EXPECT_EQ(-16, CpuSpeedExperiment::GetValue(3000 + 1, *kConfigs));
>+}
>+
>+TEST(CpuSpeedExperimentTest, GetConfigsFailsForTooSmallValue) {
>+  // Supported range: [-16, -1].
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-VP8-CpuSpeed-Arm/Enabled-1000,-1,2000,-10,3000,-17/");
>+  EXPECT_FALSE(CpuSpeedExperiment::GetConfigs());
>+}
>+
>+TEST(CpuSpeedExperimentTest, GetConfigsFailsForTooLargeValue) {
>+  // Supported range: [-16, -1].
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-VP8-CpuSpeed-Arm/Enabled-1000,0,2000,-10,3000,-16/");
>+  EXPECT_FALSE(CpuSpeedExperiment::GetConfigs());
>+}
>+
>+TEST(CpuSpeedExperimentTest, GetConfigsFailsIfPixelsDecreasing) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-VP8-CpuSpeed-Arm/Enabled-1000,-5,999,-10,3000,-16/");
>+  EXPECT_FALSE(CpuSpeedExperiment::GetConfigs());
>+}
>+
>+TEST(CpuSpeedExperimentTest, GetConfigsFailsIfCpuSpeedIncreasing) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-VP8-CpuSpeed-Arm/Enabled-1000,-5,2000,-4,3000,-16/");
>+  EXPECT_FALSE(CpuSpeedExperiment::GetConfigs());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.cc
>index a2d7f97bde1f4643f7e5efdd41a2a24aef29fb98..936487cb015e0262ef90a2802ac26dabf347350f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.cc
>@@ -169,6 +169,9 @@ template class FieldTrialParameter<double>;
> template class FieldTrialParameter<int>;
> template class FieldTrialParameter<std::string>;
> 
>+template class FieldTrialConstrained<double>;
>+template class FieldTrialConstrained<int>;
>+
> template class FieldTrialOptional<double>;
> template class FieldTrialOptional<int>;
> template class FieldTrialOptional<bool>;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.h
>index 22a88897170bc55072fd8d1e31a0dd7f037d8fd1..8bdd9b5d8ebe8601a44e6a22c9a768d9bae6ff9b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.h
>@@ -92,6 +92,43 @@ class FieldTrialParameter : public FieldTrialParameterInterface {
>   T value_;
> };
> 
>+// This class uses the ParseTypedParameter function to implement a parameter
>+// implementation with an enforced default value and a range constraint. Values
>+// outside the configured range will be ignored.
>+template <typename T>
>+class FieldTrialConstrained : public FieldTrialParameterInterface {
>+ public:
>+  FieldTrialConstrained(std::string key,
>+                        T default_value,
>+                        absl::optional<T> lower_limit,
>+                        absl::optional<T> upper_limit)
>+      : FieldTrialParameterInterface(key),
>+        value_(default_value),
>+        lower_limit_(lower_limit),
>+        upper_limit_(upper_limit) {}
>+  T Get() const { return value_; }
>+  operator T() const { return Get(); }
>+  const T* operator->() const { return &value_; }
>+
>+ protected:
>+  bool Parse(absl::optional<std::string> str_value) override {
>+    if (str_value) {
>+      absl::optional<T> value = ParseTypedParameter<T>(*str_value);
>+      if (value && (!lower_limit_ || *value >= *lower_limit_) &&
>+          (!upper_limit_ || *value <= *upper_limit_)) {
>+        value_ = *value;
>+        return true;
>+      }
>+    }
>+    return false;
>+  }
>+
>+ private:
>+  T value_;
>+  absl::optional<T> lower_limit_;
>+  absl::optional<T> upper_limit_;
>+};
>+
> class AbstractFieldTrialEnum : public FieldTrialParameterInterface {
>  public:
>   AbstractFieldTrialEnum(std::string key,
>@@ -191,6 +228,9 @@ extern template class FieldTrialParameter<int>;
> // Using the given value as is.
> extern template class FieldTrialParameter<std::string>;
> 
>+extern template class FieldTrialConstrained<double>;
>+extern template class FieldTrialConstrained<int>;
>+
> extern template class FieldTrialOptional<double>;
> extern template class FieldTrialOptional<int>;
> extern template class FieldTrialOptional<bool>;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser_unittest.cc
>index de977ec8ba809722030408d6597f57bac14ecbdc..0d067f5a49138a75cd8cde18724e6572bc9ad984 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser_unittest.cc
>@@ -99,6 +99,19 @@ TEST(FieldTrialParserTest, IgnoresInvalid) {
>   EXPECT_EQ(exp.ping.Get(), false);
>   EXPECT_EQ(exp.hash.Get(), "a80");
> }
>+TEST(FieldTrialParserTest, IgnoresOutOfRange) {
>+  FieldTrialConstrained<double> low("low", 10, absl::nullopt, 100);
>+  FieldTrialConstrained<double> high("high", 10, 5, absl::nullopt);
>+  ParseFieldTrial({&low, &high}, "low:1000,high:0");
>+  EXPECT_EQ(low.Get(), 10);
>+  EXPECT_EQ(high.Get(), 10);
>+  ParseFieldTrial({&low, &high}, "low:inf,high:nan");
>+  EXPECT_EQ(low.Get(), 10);
>+  EXPECT_EQ(high.Get(), 10);
>+  ParseFieldTrial({&low, &high}, "low:20,high:20");
>+  EXPECT_EQ(low.Get(), 20);
>+  EXPECT_EQ(high.Get(), 20);
>+}
> TEST(FieldTrialParserTest, ParsesOptionalParameters) {
>   FieldTrialOptional<int> max_count("c", absl::nullopt);
>   ParseFieldTrial({&max_count}, "");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.cc
>index f53978b21153d4fd5024e1c99564d57ebf91adf7..5311a3a941c780b9e035c0fb0b38781c437ba283 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.cc
>@@ -9,6 +9,7 @@
>  */
> #include "rtc_base/experiments/field_trial_units.h"
> 
>+#include <stdio.h>
> #include <limits>
> #include <string>
> 
>@@ -86,6 +87,10 @@ template class FieldTrialParameter<DataRate>;
> template class FieldTrialParameter<DataSize>;
> template class FieldTrialParameter<TimeDelta>;
> 
>+template class FieldTrialConstrained<DataRate>;
>+template class FieldTrialConstrained<DataSize>;
>+template class FieldTrialConstrained<TimeDelta>;
>+
> template class FieldTrialOptional<DataRate>;
> template class FieldTrialOptional<DataSize>;
> template class FieldTrialOptional<TimeDelta>;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.h
>index 932c5cb45001cf06e582e1b9dc660fb9312080df..af88f4a6c00df35106b9c5cda46339701259a6bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.h
>@@ -21,6 +21,10 @@ extern template class FieldTrialParameter<DataRate>;
> extern template class FieldTrialParameter<DataSize>;
> extern template class FieldTrialParameter<TimeDelta>;
> 
>+extern template class FieldTrialConstrained<DataRate>;
>+extern template class FieldTrialConstrained<DataSize>;
>+extern template class FieldTrialConstrained<TimeDelta>;
>+
> extern template class FieldTrialOptional<DataRate>;
> extern template class FieldTrialOptional<DataSize>;
> extern template class FieldTrialOptional<TimeDelta>;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units_unittest.cc
>index 80771d926abd37e7c80fde91804eff355d2740e8..57022c2f23f46e30b82313172b8392e3ba3a2267 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units_unittest.cc
>@@ -58,5 +58,25 @@ TEST(FieldTrialParserUnitsTest, ParsesOtherUnitParameters) {
>   EXPECT_EQ(*exp.max_buffer.GetOptional(), DataSize::bytes(8));
>   EXPECT_EQ(exp.period.Get(), TimeDelta::ms(300));
> }
>+TEST(FieldTrialParserUnitsTest, IgnoresOutOfRange) {
>+  FieldTrialConstrained<DataRate> rate("r", DataRate::kbps(30),
>+                                       DataRate::kbps(10), DataRate::kbps(100));
>+  FieldTrialConstrained<TimeDelta> delta("d", TimeDelta::ms(30),
>+                                         TimeDelta::ms(10), TimeDelta::ms(100));
>+  FieldTrialConstrained<DataSize> size(
>+      "s", DataSize::bytes(30), DataSize::bytes(10), DataSize::bytes(100));
>+  ParseFieldTrial({&rate, &delta, &size}, "r:0,d:0,s:0");
>+  EXPECT_EQ(rate->kbps(), 30);
>+  EXPECT_EQ(delta->ms(), 30);
>+  EXPECT_EQ(size->bytes(), 30);
>+  ParseFieldTrial({&rate, &delta, &size}, "r:300,d:300,s:300");
>+  EXPECT_EQ(rate->kbps(), 30);
>+  EXPECT_EQ(delta->ms(), 30);
>+  EXPECT_EQ(size->bytes(), 30);
>+  ParseFieldTrial({&rate, &delta, &size}, "r:50,d:50,s:50");
>+  EXPECT_EQ(rate->kbps(), 50);
>+  EXPECT_EQ(delta->ms(), 50);
>+  EXPECT_EQ(size->bytes(), 50);
>+}
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/jitter_upper_bound_experiment.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/jitter_upper_bound_experiment.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..b3e923019b3cfd82dc648b3ffa8934df406fdbf5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/jitter_upper_bound_experiment.cc
>@@ -0,0 +1,46 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/experiments/jitter_upper_bound_experiment.h"
>+
>+#include <algorithm>
>+#include <string>
>+
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+
>+const char JitterUpperBoundExperiment::kJitterUpperBoundExperimentName[] =
>+    "WebRTC-JitterUpperBound";
>+
>+absl::optional<double> JitterUpperBoundExperiment::GetUpperBoundSigmas() {
>+  if (!field_trial::IsEnabled(kJitterUpperBoundExperimentName)) {
>+    return absl::nullopt;
>+  }
>+  const std::string group =
>+      webrtc::field_trial::FindFullName(kJitterUpperBoundExperimentName);
>+
>+  double upper_bound_sigmas;
>+  if (sscanf(group.c_str(), "Enabled-%lf", &upper_bound_sigmas) != 1) {
>+    RTC_LOG(LS_WARNING) << "Invalid number of parameters provided.";
>+    return absl::nullopt;
>+  }
>+
>+  if (upper_bound_sigmas < 0) {
>+    RTC_LOG(LS_WARNING) << "Invalid jitter upper bound sigmas, must be >= 0.0: "
>+                        << upper_bound_sigmas;
>+    return absl::nullopt;
>+  }
>+
>+  return upper_bound_sigmas;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/jitter_upper_bound_experiment.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/jitter_upper_bound_experiment.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..262cd79efa6cccfd7ac1a9044ffd9700e78f20b0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/jitter_upper_bound_experiment.h
>@@ -0,0 +1,31 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_EXPERIMENTS_JITTER_UPPER_BOUND_EXPERIMENT_H_
>+#define RTC_BASE_EXPERIMENTS_JITTER_UPPER_BOUND_EXPERIMENT_H_
>+
>+#include "absl/types/optional.h"
>+
>+namespace webrtc {
>+
>+class JitterUpperBoundExperiment {
>+ public:
>+  // Returns nullopt if experiment is not on, otherwise returns the configured
>+  // upper bound for frame delay delta used in jitter estimation, expressed as
>+  // number of standard deviations of the current deviation from the expected
>+  // delay.
>+  static absl::optional<double> GetUpperBoundSigmas();
>+
>+  static const char kJitterUpperBoundExperimentName[];
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // RTC_BASE_EXPERIMENTS_JITTER_UPPER_BOUND_EXPERIMENT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/normalize_simulcast_size_experiment.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/normalize_simulcast_size_experiment.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..9ce5f57f8f0652d283f2ee7f37d3012670fd0dc1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/normalize_simulcast_size_experiment.cc
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/experiments/normalize_simulcast_size_experiment.h"
>+
>+#include <string>
>+
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+constexpr char kFieldTrial[] = "WebRTC-NormalizeSimulcastResolution";
>+constexpr int kMinSetting = 0;
>+constexpr int kMaxSetting = 5;
>+}  // namespace
>+
>+absl::optional<int> NormalizeSimulcastSizeExperiment::GetBase2Exponent() {
>+  if (!webrtc::field_trial::IsEnabled(kFieldTrial))
>+    return absl::nullopt;
>+
>+  const std::string group = webrtc::field_trial::FindFullName(kFieldTrial);
>+  if (group.empty())
>+    return absl::nullopt;
>+
>+  int exponent;
>+  if (sscanf(group.c_str(), "Enabled-%d", &exponent) != 1) {
>+    RTC_LOG(LS_WARNING) << "No parameter provided.";
>+    return absl::nullopt;
>+  }
>+
>+  if (exponent < kMinSetting || exponent > kMaxSetting) {
>+    RTC_LOG(LS_WARNING) << "Unsupported exp value provided, value ignored.";
>+    return absl::nullopt;
>+  }
>+
>+  return absl::optional<int>(exponent);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/normalize_simulcast_size_experiment.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/normalize_simulcast_size_experiment.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..6b358202b272ada414cc6874b360a25c7fb567cb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/normalize_simulcast_size_experiment.h
>@@ -0,0 +1,25 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_EXPERIMENTS_NORMALIZE_SIMULCAST_SIZE_EXPERIMENT_H_
>+#define RTC_BASE_EXPERIMENTS_NORMALIZE_SIMULCAST_SIZE_EXPERIMENT_H_
>+
>+#include "absl/types/optional.h"
>+
>+namespace webrtc {
>+class NormalizeSimulcastSizeExperiment {
>+ public:
>+  // Returns the base two exponent from field trial.
>+  static absl::optional<int> GetBase2Exponent();
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // RTC_BASE_EXPERIMENTS_NORMALIZE_SIMULCAST_SIZE_EXPERIMENT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/normalize_simulcast_size_experiment_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/normalize_simulcast_size_experiment_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..c37b80917fe3c60eb1e6698448c44f3e40e8dfce
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/normalize_simulcast_size_experiment_unittest.cc
>@@ -0,0 +1,59 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/experiments/normalize_simulcast_size_experiment.h"
>+
>+#include "rtc_base/gunit.h"
>+#include "test/field_trial.h"
>+
>+namespace webrtc {
>+
>+TEST(NormalizeSimulcastSizeExperimentTest, GetExponent) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-NormalizeSimulcastResolution/Enabled-2/");
>+  EXPECT_EQ(2, NormalizeSimulcastSizeExperiment::GetBase2Exponent());
>+}
>+
>+TEST(NormalizeSimulcastSizeExperimentTest, GetExponentWithTwoParameters) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-NormalizeSimulcastResolution/Enabled-3-4/");
>+  EXPECT_EQ(3, NormalizeSimulcastSizeExperiment::GetBase2Exponent());
>+}
>+
>+TEST(NormalizeSimulcastSizeExperimentTest, GetExponentFailsIfNotEnabled) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-NormalizeSimulcastResolution/Disabled/");
>+  EXPECT_FALSE(NormalizeSimulcastSizeExperiment::GetBase2Exponent());
>+}
>+
>+TEST(NormalizeSimulcastSizeExperimentTest,
>+     GetExponentFailsForInvalidFieldTrial) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-NormalizeSimulcastResolution/Enabled-invalid/");
>+  EXPECT_FALSE(NormalizeSimulcastSizeExperiment::GetBase2Exponent());
>+}
>+
>+TEST(NormalizeSimulcastSizeExperimentTest,
>+     GetExponentFailsForNegativeOutOfBoundValue) {
>+  // Supported range: [0, 5].
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-NormalizeSimulcastResolution/Enabled--1/");
>+  EXPECT_FALSE(NormalizeSimulcastSizeExperiment::GetBase2Exponent());
>+}
>+
>+TEST(NormalizeSimulcastSizeExperimentTest,
>+     GetExponentFailsForPositiveOutOfBoundValue) {
>+  // Supported range: [0, 5].
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-NormalizeSimulcastResolution/Enabled-6/");
>+  EXPECT_FALSE(NormalizeSimulcastSizeExperiment::GetBase2Exponent());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment.h
>index 80a25ef1eee58b191adc0f1bb5a015559b1e2351..14833c00b127978cb6b65e924c8c0449a9d323ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment.h
>@@ -12,7 +12,6 @@
> 
> #include "absl/types/optional.h"
> #include "api/video_codecs/video_encoder.h"
>-#include "common_types.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> class QualityScalingExperiment {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fake_mdns_responder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fake_mdns_responder.h
>index 32d69ba6836a7f9fc39e6976f6e66e26a39874d1..1e60a5d9de2d1ccc4fdf154bf6b30598d5b65bc5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fake_mdns_responder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fake_mdns_responder.h
>@@ -21,10 +21,10 @@
> 
> namespace webrtc {
> 
>-class FakeMDnsResponder : public MDnsResponderInterface {
>+class FakeMdnsResponder : public MdnsResponderInterface {
>  public:
>-  FakeMDnsResponder() = default;
>-  ~FakeMDnsResponder() = default;
>+  explicit FakeMdnsResponder(rtc::Thread* thread) : thread_(thread) {}
>+  ~FakeMdnsResponder() = default;
> 
>   void CreateNameForAddress(const rtc::IPAddress& addr,
>                             NameCreatedCallback callback) override {
>@@ -35,7 +35,9 @@ class FakeMDnsResponder : public MDnsResponderInterface {
>       name = std::to_string(next_available_id_++) + ".local";
>       addr_name_map_[addr] = name;
>     }
>-    callback(addr, name);
>+    invoker_.AsyncInvoke<void>(
>+        RTC_FROM_HERE, thread_,
>+        [callback, addr, name]() { callback(addr, name); });
>   }
>   void RemoveNameForAddress(const rtc::IPAddress& addr,
>                             NameRemovedCallback callback) override {
>@@ -43,12 +45,16 @@ class FakeMDnsResponder : public MDnsResponderInterface {
>     if (it != addr_name_map_.end()) {
>       addr_name_map_.erase(it);
>     }
>-    callback(it != addr_name_map_.end());
>+    bool result = it != addr_name_map_.end();
>+    invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_,
>+                               [callback, result]() { callback(result); });
>   }
> 
>  private:
>   uint32_t next_available_id_ = 0;
>   std::map<rtc::IPAddress, std::string> addr_name_map_;
>+  rtc::Thread* thread_;
>+  rtc::AsyncInvoker invoker_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakenetwork.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakenetwork.h
>index d5426a3e2c90955456df17c2d3e66b25c16eb077..cb890ec583dc876be6f4d44d2ebdf0803c05ab57 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakenetwork.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakenetwork.h
>@@ -82,16 +82,17 @@ class FakeNetworkManager : public NetworkManagerBase, public MessageHandler {
>   // MessageHandler interface.
>   virtual void OnMessage(Message* msg) { DoUpdateNetworks(); }
> 
>-  void CreateMDnsResponder() {
>+  void CreateMdnsResponder() {
>     if (mdns_responder_ == nullptr) {
>-      mdns_responder_ = absl::make_unique<webrtc::FakeMDnsResponder>();
>+      mdns_responder_ =
>+          absl::make_unique<webrtc::FakeMdnsResponder>(rtc::Thread::Current());
>     }
>   }
> 
>   using NetworkManagerBase::set_enumeration_permission;
>   using NetworkManagerBase::set_default_local_addresses;
> 
>-  webrtc::MDnsResponderInterface* GetMDnsResponder() const override {
>+  webrtc::MdnsResponderInterface* GetMdnsResponder() const override {
>     return mdns_responder_.get();
>   }
> 
>@@ -131,7 +132,7 @@ class FakeNetworkManager : public NetworkManagerBase, public MessageHandler {
>   IPAddress default_local_ipv4_address_;
>   IPAddress default_local_ipv6_address_;
> 
>-  std::unique_ptr<webrtc::FakeMDnsResponder> mdns_responder_;
>+  std::unique_ptr<webrtc::FakeMdnsResponder> mdns_responder_;
> };
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.cc
>index 80a3e788878441ed1dca6897863c3db9b121362b..62ac9dd0207d856a01b50e87a62ac1a309c9cd85 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.cc
>@@ -29,8 +29,8 @@ FakeSSLCertificate::FakeSSLCertificate(const FakeSSLCertificate&) = default;
> 
> FakeSSLCertificate::~FakeSSLCertificate() = default;
> 
>-FakeSSLCertificate* FakeSSLCertificate::GetReference() const {
>-  return new FakeSSLCertificate(*this);
>+std::unique_ptr<SSLCertificate> FakeSSLCertificate::Clone() const {
>+  return absl::make_unique<FakeSSLCertificate>(*this);
> }
> 
> std::string FakeSSLCertificate::ToPEMString() const {
>@@ -83,10 +83,10 @@ FakeSSLIdentity::FakeSSLIdentity(const std::vector<std::string>& pem_strings) {
> }
> 
> FakeSSLIdentity::FakeSSLIdentity(const FakeSSLCertificate& cert)
>-    : cert_chain_(absl::make_unique<SSLCertChain>(&cert)) {}
>+    : cert_chain_(absl::make_unique<SSLCertChain>(cert.Clone())) {}
> 
> FakeSSLIdentity::FakeSSLIdentity(const FakeSSLIdentity& o)
>-    : cert_chain_(o.cert_chain_->UniqueCopy()) {}
>+    : cert_chain_(o.cert_chain_->Clone()) {}
> 
> FakeSSLIdentity::~FakeSSLIdentity() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.h
>index 4494a524efab907916463fffb0c7178636974d1d..b19cbfbb492ca16e66ec1075b69fa5f95584ea56 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.h
>@@ -14,6 +14,7 @@
> #include <memory>
> #include <vector>
> 
>+#include "rtc_base/sslcertificate.h"
> #include "rtc_base/sslidentity.h"
> 
> namespace rtc {
>@@ -28,7 +29,7 @@ class FakeSSLCertificate : public SSLCertificate {
>   ~FakeSSLCertificate() override;
> 
>   // SSLCertificate implementation.
>-  FakeSSLCertificate* GetReference() const override;
>+  std::unique_ptr<SSLCertificate> Clone() const override;
>   std::string ToPEMString() const override;
>   void ToDER(Buffer* der_buffer) const override;
>   int64_t CertificateExpirationTime() const override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.cc
>index 62024119fcc17a49c5a7d331da42ef7ed22f6505..a793500646af623d603896b234a10d3162a6b611 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.cc
>@@ -10,8 +10,6 @@
> 
> #include "rtc_base/file.h"
> 
>-#include <utility>
>-
> namespace rtc {
> 
> File::File(PlatformFile file) : file_(file) {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.h
>index 75fd93d13101e25fcbdc7221dcf9b9a568d3af34..bc0974ae2a0bbf5f10840ed55aa1c9a29dd96de5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.h
>@@ -11,8 +11,8 @@
> #ifndef RTC_BASE_FILE_H_
> #define RTC_BASE_FILE_H_
> 
>+#include <stddef.h>
> #include <stdint.h>
>-
> #include <string>
> 
> #include "rtc_base/constructormagic.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file_posix.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file_posix.cc
>index b0fec9f9fcea0014cc76dd30f07705f8a42a179f..492019280c535af61327c652e4914e73fa971dea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file_posix.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file_posix.cc
>@@ -8,17 +8,16 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "rtc_base/file.h"
>-
> #include <errno.h>
> #include <fcntl.h>
>-#include <sys/stat.h>
>-#include <sys/types.h>
>+#include <stddef.h>
>+#include <stdint.h>
> #include <unistd.h>
>-
> #include <limits>
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/file.h"
>+#include "rtc_base/platform_file.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc
>index c9a663aafff869f3706f1fddde38502b4d840559..b1dc5ff99817851c1c9521b4e087b02abe509369 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc
>@@ -13,17 +13,159 @@
> #include <algorithm>
> #include <cstdio>
> #include <string>
>-
>+#include <utility>
>+
>+#if defined(WEBRTC_WIN)
>+#include <windows.h>
>+#include "rtc_base/stringutils.h"
>+#else
>+#include <dirent.h>
>+#include <sys/stat.h>
>+#include <unistd.h>
>+#endif  // WEBRTC_WIN
>+
>+#include "absl/strings/match.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/fileutils.h"
>-#include "rtc_base/pathutils.h"
>-#include "rtc_base/strings/string_builder.h"
>+#include "rtc_base/logging.h"
> 
> // Note: We use fprintf for logging in the write paths of this stream to avoid
> // infinite loops when logging.
> 
> namespace rtc {
> 
>+namespace {
>+
>+std::string AddTrailingPathDelimiterIfNeeded(std::string directory);
>+
>+// |dir| must have a trailing delimiter. |prefix| must not include wild card
>+// characters.
>+std::vector<std::string> GetFilesWithPrefix(const std::string& directory,
>+                                            const std::string& prefix);
>+bool DeleteFile(const std::string& file);
>+bool MoveFile(const std::string& old_file, const std::string& new_file);
>+bool IsFile(const std::string& file);
>+bool IsFolder(const std::string& file);
>+absl::optional<size_t> GetFileSize(const std::string& file);
>+
>+#if defined(WEBRTC_WIN)
>+
>+std::string AddTrailingPathDelimiterIfNeeded(std::string directory) {
>+  if (absl::EndsWith(directory, "\\")) {
>+    return directory;
>+  }
>+  return directory + "\\";
>+}
>+
>+std::vector<std::string> GetFilesWithPrefix(const std::string& directory,
>+                                            const std::string& prefix) {
>+  RTC_DCHECK(absl::EndsWith(directory, "\\"));
>+  WIN32_FIND_DATA data;
>+  HANDLE handle;
>+  handle = ::FindFirstFile(ToUtf16(directory + prefix + '*').c_str(), &data);
>+  if (handle == INVALID_HANDLE_VALUE)
>+    return {};
>+
>+  std::vector<std::string> file_list;
>+  do {
>+    file_list.emplace_back(directory + ToUtf8(data.cFileName));
>+  } while (::FindNextFile(handle, &data) == TRUE);
>+
>+  ::FindClose(handle);
>+  return file_list;
>+}
>+
>+bool DeleteFile(const std::string& file) {
>+  return ::DeleteFile(ToUtf16(file).c_str()) != 0;
>+}
>+
>+bool MoveFile(const std::string& old_file, const std::string& new_file) {
>+  return ::MoveFile(ToUtf16(old_file).c_str(), ToUtf16(new_file).c_str()) != 0;
>+}
>+
>+bool IsFile(const std::string& file) {
>+  WIN32_FILE_ATTRIBUTE_DATA data = {0};
>+  if (0 == ::GetFileAttributesEx(ToUtf16(file).c_str(), GetFileExInfoStandard,
>+                                 &data))
>+    return false;
>+  return (data.dwFileAttributes & FILE_ATTRIBUTE_DIRECTORY) == 0;
>+}
>+
>+bool IsFolder(const std::string& file) {
>+  WIN32_FILE_ATTRIBUTE_DATA data = {0};
>+  if (0 == ::GetFileAttributesEx(ToUtf16(file).c_str(), GetFileExInfoStandard,
>+                                 &data))
>+    return false;
>+  return (data.dwFileAttributes & FILE_ATTRIBUTE_DIRECTORY) ==
>+         FILE_ATTRIBUTE_DIRECTORY;
>+}
>+
>+absl::optional<size_t> GetFileSize(const std::string& file) {
>+  WIN32_FILE_ATTRIBUTE_DATA data = {0};
>+  if (::GetFileAttributesEx(ToUtf16(file).c_str(), GetFileExInfoStandard,
>+                            &data) == 0)
>+    return absl::nullopt;
>+  return data.nFileSizeLow;
>+}
>+
>+#else  // defined(WEBRTC_WIN)
>+
>+std::string AddTrailingPathDelimiterIfNeeded(std::string directory) {
>+  if (absl::EndsWith(directory, "/")) {
>+    return directory;
>+  }
>+  return directory + "/";
>+}
>+
>+std::vector<std::string> GetFilesWithPrefix(const std::string& directory,
>+                                            const std::string& prefix) {
>+  RTC_DCHECK(absl::EndsWith(directory, "/"));
>+  DIR* dir = ::opendir(directory.c_str());
>+  if (dir == nullptr)
>+    return {};
>+  std::vector<std::string> file_list;
>+  for (struct dirent* dirent = ::readdir(dir); dirent;
>+       dirent = ::readdir(dir)) {
>+    std::string name = dirent->d_name;
>+    if (name.compare(0, prefix.size(), prefix) == 0) {
>+      file_list.emplace_back(directory + name);
>+    }
>+  }
>+  ::closedir(dir);
>+  return file_list;
>+}
>+
>+bool DeleteFile(const std::string& file) {
>+  return ::unlink(file.c_str()) == 0;
>+}
>+
>+bool MoveFile(const std::string& old_file, const std::string& new_file) {
>+  return ::rename(old_file.c_str(), new_file.c_str()) == 0;
>+}
>+
>+bool IsFile(const std::string& file) {
>+  struct stat st;
>+  int res = ::stat(file.c_str(), &st);
>+  // Treat symlinks, named pipes, etc. all as files.
>+  return res == 0 && !S_ISDIR(st.st_mode);
>+}
>+
>+bool IsFolder(const std::string& file) {
>+  struct stat st;
>+  int res = ::stat(file.c_str(), &st);
>+  return res == 0 && S_ISDIR(st.st_mode);
>+}
>+
>+absl::optional<size_t> GetFileSize(const std::string& file) {
>+  struct stat st;
>+  if (::stat(file.c_str(), &st) != 0)
>+    return absl::nullopt;
>+  return st.st_size;
>+}
>+
>+#endif
>+
>+}  // namespace
>+
> FileRotatingStream::FileRotatingStream(const std::string& dir_path,
>                                        const std::string& file_prefix)
>     : FileRotatingStream(dir_path, file_prefix, 0, 0, kRead) {}
>@@ -46,7 +188,7 @@ FileRotatingStream::FileRotatingStream(const std::string& dir_path,
>                                        size_t max_file_size,
>                                        size_t num_files,
>                                        Mode mode)
>-    : dir_path_(dir_path),
>+    : dir_path_(AddTrailingPathDelimiterIfNeeded(dir_path)),
>       file_prefix_(file_prefix),
>       mode_(mode),
>       file_stream_(nullptr),
>@@ -55,7 +197,7 @@ FileRotatingStream::FileRotatingStream(const std::string& dir_path,
>       rotation_index_(0),
>       current_bytes_written_(0),
>       disable_buffering_(false) {
>-  RTC_DCHECK(Filesystem::IsFolder(dir_path));
>+  RTC_DCHECK(IsFolder(dir_path));
>   switch (mode) {
>     case kWrite: {
>       file_names_.clear();
>@@ -66,7 +208,7 @@ FileRotatingStream::FileRotatingStream(const std::string& dir_path,
>       break;
>     }
>     case kRead: {
>-      file_names_ = GetFilesWithPrefix();
>+      file_names_ = GetFilesWithPrefix(dir_path_, file_prefix_);
>       std::sort(file_names_.begin(), file_names_.end());
>       if (file_names_.size() > 0) {
>         // |file_names_| is sorted newest first, so read from the end.
>@@ -187,11 +329,7 @@ bool FileRotatingStream::GetSize(size_t* size) const {
>   *size = 0;
>   size_t total_size = 0;
>   for (auto file_name : file_names_) {
>-    Pathname pathname(file_name);
>-    size_t file_size = 0;
>-    if (Filesystem::GetFileSize(file_name, &file_size)) {
>-      total_size += file_size;
>-    }
>+    total_size += GetFileSize(file_name).value_or(0);
>   }
>   *size = total_size;
>   return true;
>@@ -209,9 +347,10 @@ bool FileRotatingStream::Open() {
>       return true;
>     case kWrite: {
>       // Delete existing files when opening for write.
>-      std::vector<std::string> matching_files = GetFilesWithPrefix();
>-      for (auto matching_file : matching_files) {
>-        if (!Filesystem::DeleteFile(matching_file)) {
>+      std::vector<std::string> matching_files =
>+          GetFilesWithPrefix(dir_path_, file_prefix_);
>+      for (const auto& matching_file : matching_files) {
>+        if (!DeleteFile(matching_file)) {
>           std::fprintf(stderr, "Failed to delete: %s\n", matching_file.c_str());
>         }
>       }
>@@ -282,16 +421,16 @@ void FileRotatingStream::RotateFiles() {
>   // See header file comments for example.
>   RTC_DCHECK_LT(rotation_index_, file_names_.size());
>   std::string file_to_delete = file_names_[rotation_index_];
>-  if (Filesystem::IsFile(file_to_delete)) {
>-    if (!Filesystem::DeleteFile(file_to_delete)) {
>+  if (IsFile(file_to_delete)) {
>+    if (!DeleteFile(file_to_delete)) {
>       std::fprintf(stderr, "Failed to delete: %s\n", file_to_delete.c_str());
>     }
>   }
>   for (auto i = rotation_index_; i > 0; --i) {
>     std::string rotated_name = file_names_[i];
>     std::string unrotated_name = file_names_[i - 1];
>-    if (Filesystem::IsFile(unrotated_name)) {
>-      if (!Filesystem::MoveFile(unrotated_name, rotated_name)) {
>+    if (IsFile(unrotated_name)) {
>+      if (!MoveFile(unrotated_name, rotated_name)) {
>         std::fprintf(stderr, "Failed to move: %s to %s\n",
>                      unrotated_name.c_str(), rotated_name.c_str());
>       }
>@@ -302,26 +441,6 @@ void FileRotatingStream::RotateFiles() {
>   OnRotation();
> }
> 
>-std::vector<std::string> FileRotatingStream::GetFilesWithPrefix() const {
>-  std::vector<std::string> files;
>-  // Iterate over the files in the directory.
>-  DirectoryIterator it;
>-  Pathname dir_path;
>-  dir_path.SetFolder(dir_path_);
>-  if (!it.Iterate(dir_path)) {
>-    return files;
>-  }
>-  do {
>-    std::string current_name = it.Name();
>-    if (current_name.size() && !it.IsDirectory() &&
>-        current_name.compare(0, file_prefix_.size(), file_prefix_) == 0) {
>-      Pathname path(dir_path_, current_name);
>-      files.push_back(path.pathname());
>-    }
>-  } while (it.Next());
>-  return files;
>-}
>-
> std::string FileRotatingStream::GetFilePath(size_t index,
>                                             size_t num_files) const {
>   RTC_DCHECK_LT(index, num_files);
>@@ -333,8 +452,7 @@ std::string FileRotatingStream::GetFilePath(size_t index,
>   RTC_DCHECK_LT(1 + max_digits, buffer_size);
>   std::snprintf(file_postfix, buffer_size, "_%0*zu", max_digits, index);
> 
>-  Pathname file_path(dir_path_, file_prefix_ + file_postfix);
>-  return file_path.pathname();
>+  return dir_path_ + file_prefix_ + file_postfix;
> }
> 
> CallSessionFileRotatingStream::CallSessionFileRotatingStream(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.h
>index 4dab3455db0193b21332ad5e13a0285042631e08..c75ee153b6e4a60e4c34f8754e0f16a213a875a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.h
>@@ -11,6 +11,7 @@
> #ifndef RTC_BASE_FILEROTATINGSTREAM_H_
> #define RTC_BASE_FILEROTATINGSTREAM_H_
> 
>+#include <stddef.h>
> #include <memory>
> #include <string>
> #include <vector>
>@@ -101,8 +102,6 @@ class FileRotatingStream : public StreamInterface {
>   // create new file_0
>   void RotateFiles();
> 
>-  // Returns a list of file names in the directory beginning with the prefix.
>-  std::vector<std::string> GetFilesWithPrefix() const;
>   // Private version of GetFilePath.
>   std::string GetFilePath(size_t index, size_t num_files) const;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream_unittest.cc
>index 19055162fd01945d5bcadf2c1f7d9ba09f536c89..172be57eacc69d48c9c25f0e58784186b9c71cd6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream_unittest.cc
>@@ -13,9 +13,7 @@
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/filerotatingstream.h"
>-#include "rtc_base/fileutils.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/pathutils.h"
> #include "test/testsupport/fileutils.h"
> 
> namespace rtc {
>@@ -46,12 +44,15 @@ class MAYBE_FileRotatingStreamTest : public ::testing::Test {
>   void Init(const std::string& dir_name,
>             const std::string& file_prefix,
>             size_t max_file_size,
>-            size_t num_log_files) {
>+            size_t num_log_files,
>+            bool ensure_trailing_delimiter = true) {
>     dir_path_ = webrtc::test::OutputPath();
> 
>     // Append per-test output path in order to run within gtest parallel.
>     dir_path_.append(dir_name);
>-    dir_path_.push_back(Pathname::DefaultFolderDelimiter());
>+    if (ensure_trailing_delimiter) {
>+      dir_path_.append(webrtc::test::kPathDelimiter);
>+    }
>     ASSERT_TRUE(webrtc::test::CreateDir(dir_path_));
>     stream_.reset(new FileRotatingStream(dir_path_, file_prefix, max_file_size,
>                                          num_log_files));
>@@ -159,12 +160,12 @@ TEST_F(MAYBE_FileRotatingStreamTest, WriteAndRead) {
>   }
>   // Check that exactly three files exist.
>   for (size_t i = 0; i < arraysize(messages); ++i) {
>-    EXPECT_TRUE(Filesystem::IsFile(stream_->GetFilePath(i)));
>+    EXPECT_TRUE(webrtc::test::FileExists(stream_->GetFilePath(i)));
>   }
>   std::string message("d");
>   WriteAndFlush(message.c_str(), message.size());
>   for (size_t i = 0; i < arraysize(messages); ++i) {
>-    EXPECT_TRUE(Filesystem::IsFile(stream_->GetFilePath(i)));
>+    EXPECT_TRUE(webrtc::test::FileExists(stream_->GetFilePath(i)));
>   }
>   // TODO(tkchin): Maybe check all the files in the dir.
> 
>@@ -174,6 +175,53 @@ TEST_F(MAYBE_FileRotatingStreamTest, WriteAndRead) {
>                    dir_path_, kFilePrefix);
> }
> 
>+// Tests that a write operation (with dir name without delimiter) followed by a
>+// read returns the expected data and writes to the expected files.
>+TEST_F(MAYBE_FileRotatingStreamTest, WriteWithoutDelimiterAndRead) {
>+  Init("FileRotatingStreamTestWriteWithoutDelimiterAndRead", kFilePrefix,
>+       kMaxFileSize, 3,
>+       /* ensure_trailing_delimiter*/ false);
>+
>+  ASSERT_TRUE(stream_->Open());
>+  // The test is set up to create three log files of length 2. Write and check
>+  // contents.
>+  std::string messages[3] = {"aa", "bb", "cc"};
>+  for (size_t i = 0; i < arraysize(messages); ++i) {
>+    const std::string& message = messages[i];
>+    WriteAndFlush(message.c_str(), message.size());
>+  }
>+  std::string message("d");
>+  WriteAndFlush(message.c_str(), message.size());
>+
>+  // Reopen for read.
>+  std::string expected_contents("bbccd");
>+  VerifyStreamRead(expected_contents.c_str(), expected_contents.size(),
>+                   dir_path_ + webrtc::test::kPathDelimiter, kFilePrefix);
>+}
>+
>+// Tests that a write operation followed by a read (without trailing delimiter)
>+// returns the expected data and writes to the expected files.
>+TEST_F(MAYBE_FileRotatingStreamTest, WriteAndReadWithoutDelimiter) {
>+  Init("FileRotatingStreamTestWriteAndReadWithoutDelimiter", kFilePrefix,
>+       kMaxFileSize, 3);
>+
>+  ASSERT_TRUE(stream_->Open());
>+  // The test is set up to create three log files of length 2. Write and check
>+  // contents.
>+  std::string messages[3] = {"aa", "bb", "cc"};
>+  for (size_t i = 0; i < arraysize(messages); ++i) {
>+    const std::string& message = messages[i];
>+    WriteAndFlush(message.c_str(), message.size());
>+  }
>+  std::string message("d");
>+  WriteAndFlush(message.c_str(), message.size());
>+
>+  // Reopen for read.
>+  std::string expected_contents("bbccd");
>+  VerifyStreamRead(expected_contents.c_str(), expected_contents.size(),
>+                   dir_path_.substr(0, dir_path_.size() - 1), kFilePrefix);
>+}
>+
> // Tests that writing data greater than the total capacity of the files
> // overwrites the files correctly and is read correctly after.
> TEST_F(MAYBE_FileRotatingStreamTest, WriteOverflowAndRead) {
>@@ -218,7 +266,7 @@ class MAYBE_CallSessionFileRotatingStreamTest : public ::testing::Test {
> 
>     // Append per-test output path in order to run within gtest parallel.
>     dir_path_.append(dir_name);
>-    dir_path_.push_back(Pathname::DefaultFolderDelimiter());
>+    dir_path_.append(webrtc::test::kPathDelimiter);
>     ASSERT_TRUE(webrtc::test::CreateDir(dir_path_));
>     stream_.reset(
>         new CallSessionFileRotatingStream(dir_path_, max_total_log_size));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.cc
>deleted file mode 100644
>index 7d83f97fae4c1ca06914745509bc047178ae93fa..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.cc
>+++ /dev/null
>@@ -1,130 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/fileutils.h"
>-
>-#include "rtc_base/arraysize.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/pathutils.h"
>-#include "rtc_base/stringutils.h"
>-
>-#if defined(WEBRTC_WIN)
>-#include "rtc_base/win32filesystem.h"
>-#else
>-#include "rtc_base/unixfilesystem.h"
>-#endif
>-
>-#if !defined(WEBRTC_WIN)
>-#define MAX_PATH 260
>-#endif
>-
>-namespace rtc {
>-
>-//////////////////////////
>-// Directory Iterator   //
>-//////////////////////////
>-
>-// A DirectoryIterator is created with a given directory. It originally points
>-// to the first file in the directory, and can be advanecd with Next(). This
>-// allows you to get information about each file.
>-
>-// Constructor
>-DirectoryIterator::DirectoryIterator()
>-#ifdef WEBRTC_WIN
>-    : handle_(INVALID_HANDLE_VALUE) {
>-#else
>-    : dir_(nullptr),
>-      dirent_(nullptr){
>-#endif
>-}
>-
>-// Destructor
>-DirectoryIterator::~DirectoryIterator() {
>-#if defined(WEBRTC_WIN)
>-  if (handle_ != INVALID_HANDLE_VALUE)
>-    ::FindClose(handle_);
>-#else
>-  if (dir_)
>-    closedir(dir_);
>-#endif
>-}
>-
>-// Starts traversing a directory.
>-// dir is the directory to traverse
>-// returns true if the directory exists and is valid
>-bool DirectoryIterator::Iterate(const Pathname& dir) {
>-  directory_ = dir.pathname();
>-#if defined(WEBRTC_WIN)
>-  if (handle_ != INVALID_HANDLE_VALUE)
>-    ::FindClose(handle_);
>-  std::string d = dir.pathname() + '*';
>-  handle_ = ::FindFirstFile(ToUtf16(d).c_str(), &data_);
>-  if (handle_ == INVALID_HANDLE_VALUE)
>-    return false;
>-#else
>-  if (dir_ != nullptr)
>-    closedir(dir_);
>-  dir_ = ::opendir(directory_.c_str());
>-  if (dir_ == nullptr)
>-    return false;
>-  dirent_ = readdir(dir_);
>-  if (dirent_ == nullptr)
>-    return false;
>-
>-  if (::stat(std::string(directory_ + Name()).c_str(), &stat_) != 0)
>-    return false;
>-#endif
>-  return true;
>-}
>-
>-// Advances to the next file
>-// returns true if there were more files in the directory.
>-bool DirectoryIterator::Next() {
>-#if defined(WEBRTC_WIN)
>-  return ::FindNextFile(handle_, &data_) == TRUE;
>-#else
>-  dirent_ = ::readdir(dir_);
>-  if (dirent_ == nullptr)
>-    return false;
>-
>-  return ::stat(std::string(directory_ + Name()).c_str(), &stat_) == 0;
>-#endif
>-}
>-
>-// returns true if the file currently pointed to is a directory
>-bool DirectoryIterator::IsDirectory() const {
>-#if defined(WEBRTC_WIN)
>-  return (data_.dwFileAttributes & FILE_ATTRIBUTE_DIRECTORY) != FALSE;
>-#else
>-  return S_ISDIR(stat_.st_mode);
>-#endif
>-}
>-
>-// returns the name of the file currently pointed to
>-std::string DirectoryIterator::Name() const {
>-#if defined(WEBRTC_WIN)
>-  return ToUtf8(data_.cFileName);
>-#else
>-  RTC_DCHECK(dirent_);
>-  return dirent_->d_name;
>-#endif
>-}
>-
>-FilesystemInterface* Filesystem::GetFilesystem() {
>-#if defined(WEBRTC_WIN)
>-  static FilesystemInterface* const filesystem = new Win32Filesystem();
>-#else
>-  static FilesystemInterface* const filesystem = new UnixFilesystem();
>-#endif
>-
>-  return filesystem;
>-}
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.h
>deleted file mode 100644
>index 132fd884401cff5ffb023efe85e8f08956751ff7..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.h
>+++ /dev/null
>@@ -1,133 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef RTC_BASE_FILEUTILS_H_
>-#define RTC_BASE_FILEUTILS_H_
>-
>-#include <string>
>-
>-#if defined(WEBRTC_WIN)
>-#include <windows.h>
>-#else
>-#include <dirent.h>
>-#include <stdio.h>
>-#include <sys/stat.h>
>-#include <sys/types.h>
>-#include <unistd.h>
>-#endif  // WEBRTC_WIN
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/platform_file.h"
>-
>-namespace rtc {
>-
>-class FileStream;
>-class Pathname;
>-
>-//////////////////////////
>-// Directory Iterator   //
>-//////////////////////////
>-
>-// A DirectoryIterator is created with a given directory. It originally points
>-// to the first file in the directory, and can be advanecd with Next(). This
>-// allows you to get information about each file.
>-
>-class DirectoryIterator {
>-  friend class Filesystem;
>-
>- public:
>-  // Constructor
>-  DirectoryIterator();
>-  // Destructor
>-  virtual ~DirectoryIterator();
>-
>-  // Starts traversing a directory
>-  // dir is the directory to traverse
>-  // returns true if the directory exists and is valid
>-  // The iterator will point to the first entry in the directory
>-  virtual bool Iterate(const Pathname& path);
>-
>-  // Advances to the next file
>-  // returns true if there were more files in the directory.
>-  virtual bool Next();
>-
>-  // returns true if the file currently pointed to is a directory
>-  virtual bool IsDirectory() const;
>-
>-  // returns the name of the file currently pointed to
>-  virtual std::string Name() const;
>-
>- private:
>-  std::string directory_;
>-#if defined(WEBRTC_WIN)
>-  WIN32_FIND_DATA data_;
>-  HANDLE handle_;
>-#else
>-  DIR* dir_;
>-  struct dirent* dirent_;
>-  struct stat stat_;
>-#endif
>-};
>-
>-class FilesystemInterface {
>- public:
>-  virtual ~FilesystemInterface() {}
>-
>-  // This will attempt to delete the path located at filename.
>-  // It DCHECKs and returns false if the path points to a folder or a
>-  // non-existent file.
>-  virtual bool DeleteFile(const Pathname& filename) = 0;
>-
>-  // This moves a file from old_path to new_path, where "old_path" is a
>-  // plain file. This DCHECKs and returns false if old_path points to a
>-  // directory, and returns true if the function succeeds.
>-  virtual bool MoveFile(const Pathname& old_path, const Pathname& new_path) = 0;
>-
>-  // Returns true if pathname refers to a directory
>-  virtual bool IsFolder(const Pathname& pathname) = 0;
>-
>-  // Returns true if pathname refers to a file
>-  virtual bool IsFile(const Pathname& pathname) = 0;
>-
>-  // Determines the size of the file indicated by path.
>-  virtual bool GetFileSize(const Pathname& path, size_t* size) = 0;
>-};
>-
>-class Filesystem {
>- public:
>-  static bool DeleteFile(const Pathname& filename) {
>-    return GetFilesystem()->DeleteFile(filename);
>-  }
>-
>-  static bool MoveFile(const Pathname& old_path, const Pathname& new_path) {
>-    return GetFilesystem()->MoveFile(old_path, new_path);
>-  }
>-
>-  static bool IsFolder(const Pathname& pathname) {
>-    return GetFilesystem()->IsFolder(pathname);
>-  }
>-
>-  static bool IsFile(const Pathname& pathname) {
>-    return GetFilesystem()->IsFile(pathname);
>-  }
>-
>-  static bool GetFileSize(const Pathname& path, size_t* size) {
>-    return GetFilesystem()->GetFileSize(path, size);
>-  }
>-
>- private:
>-  static FilesystemInterface* GetFilesystem();
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Filesystem);
>-};
>-
>-}  // namespace rtc
>-
>-#endif  // RTC_BASE_FILEUTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>index 4f294594bd4afcf29a3f89ac270e28abcbf75531..6b43b91e026acdc584215dd4eb42ddba5e5c04aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>@@ -15,7 +15,6 @@
> #include <string.h>
> 
> #include "rtc_base/checks.h"
>-#include "rtc_base/stringutils.h"
> 
> #if defined(WEBRTC_WIN)
> // clang-format off
>@@ -23,6 +22,8 @@
> #include <windows.h>
> #include <shellapi.h> // must come after windows.h
> // clang-format on
>+
>+#include "rtc_base/stringutils.h"  // For ToUtf8
> #endif
> 
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.h
>index 33f6e5bda1dbda54c6df8e5f268fd860fd4768ad..a08bfd29b85d5ff2b0d1daf511165ecb4f2103dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.h
>@@ -23,7 +23,10 @@
> #define RTC_BASE_FLAGS_H_
> 
> #include "rtc_base/checks.h"
>+
>+#if defined(WEBRTC_WIN)
> #include "rtc_base/constructormagic.h"
>+#endif
> 
> namespace rtc {
> 
>@@ -33,7 +36,7 @@ union FlagValue {
>   // bool values ('bool b = "false";' results in b == true!), we pass
>   // and int argument to New_BOOL as this appears to be safer - sigh.
>   // In particular, it prevents the (not uncommon!) bug where a bool
>-  // flag is defined via: DEFINE_bool(flag, "false", "some comment");.
>+  // flag is defined via: WEBRTC_DEFINE_bool(flag, "false", "some comment");.
>   static FlagValue New_BOOL(int b) {
>     FlagValue v;
>     v.b = (b != 0);
>@@ -152,7 +155,7 @@ class Flag {
> };
> 
> // Internal use only.
>-#define DEFINE_FLAG(type, c_type, name, default, comment)                   \
>+#define WEBRTC_DEFINE_FLAG(type, c_type, name, default, comment)            \
>   /* define and initialize the flag */                                      \
>   c_type FLAG_##name = (default);                                           \
>   /* register the flag */                                                   \
>@@ -161,25 +164,25 @@ class Flag {
>                                rtc::FlagValue::New_##type(default))
> 
> // Internal use only.
>-#define DECLARE_FLAG(c_type, name) \
>-  /* declare the external flag */  \
>+#define WEBRTC_DECLARE_FLAG(c_type, name) \
>+  /* declare the external flag */         \
>   extern c_type FLAG_##name
> 
> // Use the following macros to define a new flag:
>-#define DEFINE_bool(name, default, comment) \
>-  DEFINE_FLAG(BOOL, bool, name, default, comment)
>-#define DEFINE_int(name, default, comment) \
>-  DEFINE_FLAG(INT, int, name, default, comment)
>-#define DEFINE_float(name, default, comment) \
>-  DEFINE_FLAG(FLOAT, double, name, default, comment)
>-#define DEFINE_string(name, default, comment) \
>-  DEFINE_FLAG(STRING, const char*, name, default, comment)
>+#define WEBRTC_DEFINE_bool(name, default, comment) \
>+  WEBRTC_DEFINE_FLAG(BOOL, bool, name, default, comment)
>+#define WEBRTC_DEFINE_int(name, default, comment) \
>+  WEBRTC_DEFINE_FLAG(INT, int, name, default, comment)
>+#define WEBRTC_DEFINE_float(name, default, comment) \
>+  WEBRTC_DEFINE_FLAG(FLOAT, double, name, default, comment)
>+#define WEBRTC_DEFINE_string(name, default, comment) \
>+  WEBRTC_DEFINE_FLAG(STRING, const char*, name, default, comment)
> 
> // Use the following macros to declare a flag defined elsewhere:
>-#define DECLARE_bool(name) DECLARE_FLAG(bool, name)
>-#define DECLARE_int(name) DECLARE_FLAG(int, name)
>-#define DECLARE_float(name) DECLARE_FLAG(double, name)
>-#define DECLARE_string(name) DECLARE_FLAG(const char*, name)
>+#define WEBRTC_DECLARE_bool(name) WEBRTC_DECLARE_FLAG(bool, name)
>+#define WEBRTC_DECLARE_int(name) WEBRTC_DECLARE_FLAG(int, name)
>+#define WEBRTC_DECLARE_float(name) WEBRTC_DECLARE_FLAG(double, name)
>+#define WEBRTC_DECLARE_string(name) WEBRTC_DECLARE_FLAG(const char*, name)
> 
> // The global list of all flags.
> class FlagList {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.h
>index d3b09cfd48fb5bd3e5afd93bd6f854a11ace0ef7..a93b321eab92673de903b9e87ec2656366e31e45 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.h
>@@ -11,6 +11,8 @@
> #ifndef RTC_BASE_HELPERS_H_
> #define RTC_BASE_HELPERS_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <string>
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.cc
>index 716a41e16e93e72c6e90c162be625ee221458b35..7926f884f9d881b375f8c0d77c0465f08ea64db3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.cc
>@@ -18,17 +18,23 @@
> #include <security.h>
> #endif
> 
>+#include <ctype.h>  // for isspace
>+#include <stdio.h>  // for sprintf
> #include <algorithm>
>+#include <utility>  // for pair
>+#include <vector>
> 
>+#include "absl/strings/match.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/cryptstring.h"
>+#include "rtc_base/cryptstring.h"  // for CryptString
> #include "rtc_base/httpcommon.h"
>+#include "rtc_base/logging.h"
> #include "rtc_base/messagedigest.h"
> #include "rtc_base/socketaddress.h"
> #include "rtc_base/strings/string_builder.h"
>-#include "rtc_base/third_party/base64/base64.h"
>-#include "rtc_base/zero_memory.h"
>+#include "rtc_base/third_party/base64/base64.h"  // for Base64
>+#include "rtc_base/zero_memory.h"                // for ExplicitZeroMemory
> 
> namespace rtc {
> namespace {
>@@ -260,7 +266,7 @@ HttpAuthResult HttpAuthenticate(const char* challenge,
>     return HAR_IGNORE;
> 
>   // BASIC
>-  if (_stricmp(auth_method.c_str(), "basic") == 0) {
>+  if (absl::EqualsIgnoreCase(auth_method, "basic")) {
>     if (context)
>       return HAR_CREDENTIALS;  // Bad credentials
>     if (username.empty())
>@@ -288,7 +294,7 @@ HttpAuthResult HttpAuthenticate(const char* challenge,
>   }
> 
>   // DIGEST
>-  if (_stricmp(auth_method.c_str(), "digest") == 0) {
>+  if (absl::EqualsIgnoreCase(auth_method, "digest")) {
>     if (context)
>       return HAR_CREDENTIALS;  // Bad credentials
>     if (username.empty())
>@@ -355,8 +361,8 @@ HttpAuthResult HttpAuthenticate(const char* challenge,
> 
> #if defined(WEBRTC_WIN)
> #if 1
>-  bool want_negotiate = (_stricmp(auth_method.c_str(), "negotiate") == 0);
>-  bool want_ntlm = (_stricmp(auth_method.c_str(), "ntlm") == 0);
>+  bool want_negotiate = absl::EqualsIgnoreCase(auth_method, "negotiate");
>+  bool want_ntlm = absl::EqualsIgnoreCase(auth_method, "ntlm");
>   // SPNEGO & NTLM
>   if (want_negotiate || want_ntlm) {
>     const size_t MAX_MESSAGE = 12000, MAX_SPN = 256;
>@@ -371,7 +377,7 @@ HttpAuthResult HttpAuthenticate(const char* challenge,
>       return HAR_IGNORE;
>     }
> #else
>-    sprintfn(spn, MAX_SPN, "HTTP/%s", server.ToString().c_str());
>+    snprintf(spn, MAX_SPN, "HTTP/%s", server.ToString().c_str());
> #endif
> 
>     SecBuffer out_sec;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.h
>index 581bd1ee8a86c612cb62f6ab234153ad4d2c808d..fbad280e0250a0b7ad535d938381727dd7acd1c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.h
>@@ -11,14 +11,7 @@
> #ifndef RTC_BASE_HTTPCOMMON_H_
> #define RTC_BASE_HTTPCOMMON_H_
> 
>-#include <map>
>-#include <memory>
> #include <string>
>-#include <vector>
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/stream.h"
>-#include "rtc_base/stringutils.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.cc
>index c52c9a4adddac30dc4ed009dffae0aec55edb6b6..027a7b219e02a9e6c5370bc505f593ad8b40a47c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.cc
>@@ -11,24 +11,17 @@
> #if defined(WEBRTC_POSIX)
> #include <netinet/in.h>
> #include <sys/socket.h>
>-#include <sys/types.h>
> #ifdef OPENBSD
> #include <netinet/in_systm.h>
> #endif
> #ifndef __native_client__
> #include <netinet/ip.h>
> #endif
>-#include <arpa/inet.h>
> #include <netdb.h>
>-#include <unistd.h>
> #endif
> 
>-#include <stdio.h>
>-
> #include "rtc_base/byteorder.h"
>-#include "rtc_base/checks.h"
> #include "rtc_base/ipaddress.h"
>-#include "rtc_base/logging.h"
> #include "rtc_base/nethelpers.h"
> #include "rtc_base/stringutils.h"
> 
>@@ -162,11 +155,10 @@ std::string IPAddress::ToSensitiveString() const {
>       std::string result;
>       result.resize(INET6_ADDRSTRLEN);
>       in6_addr addr = ipv6_address();
>-      size_t len =
>-          rtc::sprintfn(&(result[0]), result.size(), "%x:%x:%x:x:x:x:x:x",
>-                        (addr.s6_addr[0] << 8) + addr.s6_addr[1],
>-                        (addr.s6_addr[2] << 8) + addr.s6_addr[3],
>-                        (addr.s6_addr[4] << 8) + addr.s6_addr[5]);
>+      size_t len = snprintf(&(result[0]), result.size(), "%x:%x:%x:x:x:x:x:x",
>+                            (addr.s6_addr[0] << 8) + addr.s6_addr[1],
>+                            (addr.s6_addr[2] << 8) + addr.s6_addr[3],
>+                            (addr.s6_addr[4] << 8) + addr.s6_addr[5]);
>       result.resize(len);
>       return result;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/key_derivation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/key_derivation.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..288e407ebc5a974310d840ba69cf405752978c2c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/key_derivation.cc
>@@ -0,0 +1,31 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/key_derivation.h"
>+
>+#include "absl/memory/memory.h"
>+#include "rtc_base/openssl_key_derivation_hkdf.h"
>+
>+namespace rtc {
>+
>+KeyDerivation::KeyDerivation() = default;
>+KeyDerivation::~KeyDerivation() = default;
>+
>+// static
>+std::unique_ptr<KeyDerivation> KeyDerivation::Create(
>+    KeyDerivationAlgorithm key_derivation_algorithm) {
>+  switch (key_derivation_algorithm) {
>+    case KeyDerivationAlgorithm::HKDF_SHA256:
>+      return absl::make_unique<OpenSSLKeyDerivationHKDF>();
>+  }
>+  RTC_NOTREACHED();
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/key_derivation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/key_derivation.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..fa329aeb4ce4c9d98eabe10061678bc91b7b6f39
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/key_derivation.h
>@@ -0,0 +1,70 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_KEY_DERIVATION_H_
>+#define RTC_BASE_KEY_DERIVATION_H_
>+
>+#include <memory>
>+
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "rtc_base/buffer.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace rtc {
>+
>+// Defines the set of key derivation algorithms that are supported. It is ideal
>+// to keep this list as small as possible.
>+enum class KeyDerivationAlgorithm {
>+  // This algorithm is not suitable to generate a key from a password. Please
>+  // only use with a cryptographically random master secret.
>+  HKDF_SHA256
>+};
>+
>+// KeyDerivation provides a generic interface for deriving keys in WebRTC. This
>+// class should be used over directly accessing openssl or boringssl primitives
>+// so that we can maintain seperate implementations.
>+// Example:
>+//   auto kd = KeyDerivation::Create(KeyDerivationAlgorithm::HDKF_SHA526);
>+//   if (kd == nullptr) return;
>+//   auto derived_key_or = kd->DeriveKey(secret, salt, label);
>+//   if (!derived_key_or.ok()) return;
>+//   DoSomethingWithKey(derived_key_or.value());
>+class KeyDerivation {
>+ public:
>+  KeyDerivation();
>+  virtual ~KeyDerivation();
>+
>+  // Derives a new key from existing key material.
>+  // secret - The random secret value you wish to derive a key from.
>+  // salt - Optional but recommended (non secret) cryptographically random.
>+  // label - A non secret but unique label value to determine the derivation.
>+  // derived_key_byte_size - This must be at least 128 bits.
>+  // return - An optional ZeroOnFreeBuffer containing the derived key or
>+  // absl::nullopt. Nullopt indicates a failure in derivation.
>+  virtual absl::optional<ZeroOnFreeBuffer<uint8_t>> DeriveKey(
>+      rtc::ArrayView<const uint8_t> secret,
>+      rtc::ArrayView<const uint8_t> salt,
>+      rtc::ArrayView<const uint8_t> label,
>+      size_t derived_key_byte_size) = 0;
>+
>+  // Static factory that will return an implementation that is capable of
>+  // handling the key derivation with the requested algorithm. If no
>+  // implementation is available nullptr will be returned.
>+  static std::unique_ptr<KeyDerivation> Create(
>+      KeyDerivationAlgorithm key_derivation_algorithm);
>+
>+ private:
>+  RTC_DISALLOW_COPY_AND_ASSIGN(KeyDerivation);
>+};
>+
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_KEY_DERIVATION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.cc
>index 9c90d9ec1797745b91c7396984efbca906473b5a..c95ad9c488a07d415b0e98e5ed5a7e25e190003b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.cc
>@@ -10,8 +10,6 @@
> 
> #include "rtc_base/location.h"
> 
>-#include "rtc_base/stringutils.h"
>-
> namespace rtc {
> 
> Location::Location(const char* function_name, const char* file_and_line)
>@@ -31,7 +29,7 @@ Location& Location::operator=(const Location& other) {
> 
> std::string Location::ToString() const {
>   char buf[256];
>-  sprintfn(buf, sizeof(buf), "%s@%s", function_name_, file_and_line_);
>+  snprintf(buf, sizeof(buf), "%s@%s", function_name_, file_and_line_);
>   return buf;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>index 44ec3f50c1961c799b896a43f9473be5ff3b618b..58b3345f9db52ef37f9ce8600a4aff0440569c77 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>@@ -75,7 +75,12 @@ CriticalSection& logCriticalScope() {
> void LogSink::OnLogMessage(const std::string& msg,
>                            LoggingSeverity severity,
>                            const char* tag) {
>-  OnLogMessage(tag + (": " + msg));
>+  OnLogMessage(tag + (": " + msg), severity);
>+}
>+
>+void LogSink::OnLogMessage(const std::string& msg,
>+                           LoggingSeverity /* severity */) {
>+  OnLogMessage(msg);
> }
> 
> /////////////////////////////////////////////////////////////////////////////
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h
>index 1bd0c7205e69267a1d1173817ff3419ee45305ac..c7d083e42d37a319126a2adf7e6037b7b0b6d8bc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h
>@@ -51,16 +51,15 @@
> #include <string>
> #include <utility>
> 
>-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
>-#include <CoreServices/CoreServices.h>
>-#endif
>-
> #include "absl/strings/string_view.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/deprecation.h"
> #include "rtc_base/strings/string_builder.h"
> #include "rtc_base/system/inline.h"
>-#include "rtc_base/thread_annotations.h"
>+
>+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
>+#include "rtc_base/logging_mac.h"
>+#endif  // WEBRTC_MAC
> 
> #if !defined(NDEBUG) || defined(DLOG_ALWAYS_ON)
> #define RTC_DLOG_IS_ON 1
>@@ -70,11 +69,6 @@
> 
> namespace rtc {
> 
>-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
>-// Returns a UTF8 description from an OS X Status error.
>-std::string DescriptionFromOSStatus(OSStatus err);
>-#endif
>-
> //////////////////////////////////////////////////////////////////////
> 
> // Note that the non-standard LoggingSeverity aliases exist because they are
>@@ -121,6 +115,8 @@ class LogSink {
>   virtual void OnLogMessage(const std::string& msg,
>                             LoggingSeverity severity,
>                             const char* tag);
>+  virtual void OnLogMessage(const std::string& message,
>+                            LoggingSeverity severity);
>   virtual void OnLogMessage(const std::string& message) = 0;
> };
> 
>@@ -523,10 +519,10 @@ class LogMessage {
>       ? static_cast<void>(0)               \
>       : rtc::webrtc_logging_impl::LogMessageVoidify()&
> 
>-#define RTC_LOG_FILE_LINE(sev, file, line)                                     \
>-  rtc::webrtc_logging_impl::LogCall() &                                        \
>-      rtc::webrtc_logging_impl::LogStreamer<>()                                \
>-          << rtc::webrtc_logging_impl::LogMetadata(__FILE__, __LINE__, sev)
>+#define RTC_LOG_FILE_LINE(sev, file, line)      \
>+  rtc::webrtc_logging_impl::LogCall() &         \
>+      rtc::webrtc_logging_impl::LogStreamer<>() \
>+          << rtc::webrtc_logging_impl::LogMetadata(file, line, sev)
> 
> #define RTC_LOG(sev) RTC_LOG_FILE_LINE(rtc::sev, __FILE__, __LINE__)
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_mac.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_mac.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..e65db560c1879f2b880377339419880cf6187c58
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_mac.h
>@@ -0,0 +1,28 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef RTC_BASE_LOGGING_MAC_H_
>+#define RTC_BASE_LOGGING_MAC_H_
>+
>+#if !defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
>+#error "Only include this header in macOS builds"
>+#endif
>+
>+#include <CoreServices/CoreServices.h>
>+
>+#include <string>
>+
>+namespace rtc {
>+
>+// Returns a UTF8 description from an OS X Status error.
>+std::string DescriptionFromOSStatus(OSStatus err);
>+
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_LOGGING_MAC_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_mac.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_mac.mm
>index 378cfbfcb9cd147701180b30b07b859297b5fb57..bd5f2b9d38fbf5e8d90075bfbc49a40036444b54 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_mac.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_mac.mm
>@@ -8,15 +8,15 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "rtc_base/logging.h"
>+#include "rtc_base/logging_mac.h"
> 
> #import <Foundation/Foundation.h>
> 
>-
> namespace rtc {
> std::string DescriptionFromOSStatus(OSStatus err) {
>   NSError* error =
>       [NSError errorWithDomain:NSOSStatusErrorDomain code:err userInfo:nil];
>   return error.description.UTF8String;
> }
>+
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>index 6b2263923fa8501b67b3bdeacc04677e04c6df36..4de1cf2fe4caab3e7edd44eb9bfd447a06dfb018 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>@@ -221,7 +221,7 @@ TEST(LogTest, SingleStream) {
> 
> #if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
> TEST(LogTest, Checks) {
>-  EXPECT_DEATH(RTC_FATAL() << "message",
>+  EXPECT_DEATH(FATAL() << "message",
>                "\n\n#\n"
>                "# Fatal error in: \\S+, line \\w+\n"
>                "# last system error: \\w+\n"
>@@ -292,7 +292,7 @@ class LogThread {
>   static void ThreadEntry(void* p) { static_cast<LogThread*>(p)->Run(); }
> 
>   PlatformThread thread_;
>-  Event event_{false, false};
>+  Event event_;
> };
> 
> // Ensure we don't crash when adding/removing streams while threads are going.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.cc
>index 662b1f20de9785c75360840aa5ee8b09c7d9e717..c01bafbdd2280f45988b2927cc75bcb04da4f86d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.cc
>@@ -10,10 +10,12 @@
> 
> #include "rtc_base/logsinks.h"
> 
>+#include <string.h>
> #include <cstdio>
> #include <string>
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/stream.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.h
>index caf4a5f728a37945250d0f89257290b5cbc54797..d0867a22024bbd15e7eddf8455c3c97dd0f9614c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.h
>@@ -11,6 +11,7 @@
> #ifndef RTC_BASE_LOGSINKS_H_
> #define RTC_BASE_LOGSINKS_H_
> 
>+#include <stddef.h>
> #include <memory>
> #include <string>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/mdns_responder_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/mdns_responder_interface.h
>index 9dbaf56ad88f1059bf3f3f0048aa1c0a6156a87b..71938b2576d79dd8a6c717e6a205d4365ac217ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/mdns_responder_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/mdns_responder_interface.h
>@@ -12,36 +12,35 @@
> #define RTC_BASE_MDNS_RESPONDER_INTERFACE_H_
> 
> #include <functional>
>-#include <map>
>-#include <memory>
>-#include <set>
> #include <string>
> 
> #include "rtc_base/ipaddress.h"
>-#include "rtc_base/socketaddress.h"
> 
> namespace webrtc {
> 
> // Defines an mDNS responder that can be used in ICE candidate gathering, where
>-// the local IP addresses of host candidates are obfuscated by mDNS hostnames.
>-class MDnsResponderInterface {
>+// the local IP addresses of host candidates are replaced by mDNS hostnames.
>+class MdnsResponderInterface {
>  public:
>   using NameCreatedCallback =
>       std::function<void(const rtc::IPAddress&, const std::string&)>;
>   using NameRemovedCallback = std::function<void(bool)>;
> 
>-  MDnsResponderInterface() = default;
>-  virtual ~MDnsResponderInterface() = default;
>+  MdnsResponderInterface() = default;
>+  virtual ~MdnsResponderInterface() = default;
> 
>-  // Asynchronously creates a type-4 UUID hostname for an IP address. The
>-  // created name should be given to |callback| with the address that it
>-  // represents.
>+  // Asynchronously creates and returns a new name via |callback| for |addr| if
>+  // there is no name mapped to it by this responder, and initializes the
>+  // reference count of this name to one. Otherwise the existing name mapped to
>+  // |addr| is returned and its reference count is incremented by one.
>   virtual void CreateNameForAddress(const rtc::IPAddress& addr,
>                                     NameCreatedCallback callback) = 0;
>-  // Removes the name mapped to the given address if there is such an
>-  // name-address mapping previously created via CreateNameForAddress. The
>-  // result of whether an associated name-address mapping is removed should be
>-  // given to |callback|.
>+  // Decrements the reference count of the mapped name of |addr|, if
>+  // there is a map created previously via CreateNameForAddress; asynchronously
>+  // removes the association between |addr| and its mapped name, and returns
>+  // true via |callback| if the decremented reference count reaches zero.
>+  // Otherwise no operation is done and false is returned via |callback|
>+  // asynchronously.
>   virtual void RemoveNameForAddress(const rtc::IPAddress& addr,
>                                     NameRemovedCallback callback) = 0;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc.cc
>index a1d54bdb785a8dfda5e3493d026b9ac4ed392119..c893c96ef2af0883c03fa120eff385a58e3acb48 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc.cc
>@@ -10,8 +10,8 @@
> 
> #include "rtc_base/memory/aligned_malloc.h"
> 
>-#include <stdlib.h>
>-#include <cstring>
>+#include <stdlib.h>  // for free, malloc
>+#include <string.h>  // for memcpy
> 
> #ifdef _WIN32
> #include <windows.h>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_stream.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..541de070c7281cb8f7f499576e6c6dc1087196de
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_stream.cc
>@@ -0,0 +1,143 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <algorithm>
>+
>+#include "rtc_base/memory_stream.h"
>+
>+namespace rtc {
>+
>+StreamState MemoryStream::GetState() const {
>+  return SS_OPEN;
>+}
>+
>+StreamResult MemoryStream::Read(void* buffer,
>+                                size_t bytes,
>+                                size_t* bytes_read,
>+                                int* error) {
>+  if (seek_position_ >= data_length_) {
>+    return SR_EOS;
>+  }
>+  size_t available = data_length_ - seek_position_;
>+  if (bytes > available) {
>+    // Read partial buffer
>+    bytes = available;
>+  }
>+  memcpy(buffer, &buffer_[seek_position_], bytes);
>+  seek_position_ += bytes;
>+  if (bytes_read) {
>+    *bytes_read = bytes;
>+  }
>+  return SR_SUCCESS;
>+}
>+
>+StreamResult MemoryStream::Write(const void* buffer,
>+                                 size_t bytes,
>+                                 size_t* bytes_written,
>+                                 int* error) {
>+  size_t available = buffer_length_ - seek_position_;
>+  if (0 == available) {
>+    // Increase buffer size to the larger of:
>+    // a) new position rounded up to next 256 bytes
>+    // b) double the previous length
>+    size_t new_buffer_length =
>+        std::max(((seek_position_ + bytes) | 0xFF) + 1, buffer_length_ * 2);
>+    StreamResult result = DoReserve(new_buffer_length, error);
>+    if (SR_SUCCESS != result) {
>+      return result;
>+    }
>+    RTC_DCHECK(buffer_length_ >= new_buffer_length);
>+    available = buffer_length_ - seek_position_;
>+  }
>+
>+  if (bytes > available) {
>+    bytes = available;
>+  }
>+  memcpy(&buffer_[seek_position_], buffer, bytes);
>+  seek_position_ += bytes;
>+  if (data_length_ < seek_position_) {
>+    data_length_ = seek_position_;
>+  }
>+  if (bytes_written) {
>+    *bytes_written = bytes;
>+  }
>+  return SR_SUCCESS;
>+}
>+
>+void MemoryStream::Close() {
>+  // nothing to do
>+}
>+
>+bool MemoryStream::SetPosition(size_t position) {
>+  if (position > data_length_)
>+    return false;
>+  seek_position_ = position;
>+  return true;
>+}
>+
>+bool MemoryStream::GetPosition(size_t* position) const {
>+  if (position)
>+    *position = seek_position_;
>+  return true;
>+}
>+
>+bool MemoryStream::GetSize(size_t* size) const {
>+  if (size)
>+    *size = data_length_;
>+  return true;
>+}
>+
>+bool MemoryStream::ReserveSize(size_t size) {
>+  return (SR_SUCCESS == DoReserve(size, nullptr));
>+}
>+
>+///////////////////////////////////////////////////////////////////////////////
>+
>+MemoryStream::MemoryStream() {}
>+
>+MemoryStream::MemoryStream(const char* data) {
>+  SetData(data, strlen(data));
>+}
>+
>+MemoryStream::MemoryStream(const void* data, size_t length) {
>+  SetData(data, length);
>+}
>+
>+MemoryStream::~MemoryStream() {
>+  delete[] buffer_;
>+}
>+
>+void MemoryStream::SetData(const void* data, size_t length) {
>+  data_length_ = buffer_length_ = length;
>+  delete[] buffer_;
>+  buffer_ = new char[buffer_length_];
>+  memcpy(buffer_, data, data_length_);
>+  seek_position_ = 0;
>+}
>+
>+StreamResult MemoryStream::DoReserve(size_t size, int* error) {
>+  if (buffer_length_ >= size)
>+    return SR_SUCCESS;
>+
>+  if (char* new_buffer = new char[size]) {
>+    memcpy(new_buffer, buffer_, data_length_);
>+    delete[] buffer_;
>+    buffer_ = new_buffer;
>+    buffer_length_ = size;
>+    return SR_SUCCESS;
>+  }
>+
>+  if (error) {
>+    *error = ENOMEM;
>+  }
>+  return SR_ERROR;
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_stream.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..936f71b34c010b2263e478029329b1ecfd270d70
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_stream.h
>@@ -0,0 +1,59 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_MEMORY_STREAM_H_
>+#define RTC_BASE_MEMORY_STREAM_H_
>+
>+#include "rtc_base/stream.h"
>+
>+namespace rtc {
>+
>+// MemoryStream dynamically resizes to accomodate written data.
>+
>+class MemoryStream final : public StreamInterface {
>+ public:
>+  MemoryStream();
>+  explicit MemoryStream(const char* data);  // Calls SetData(data, strlen(data))
>+  MemoryStream(const void* data, size_t length);  // Calls SetData(data, length)
>+  ~MemoryStream() override;
>+
>+  StreamState GetState() const override;
>+  StreamResult Read(void* buffer,
>+                    size_t bytes,
>+                    size_t* bytes_read,
>+                    int* error) override;
>+  StreamResult Write(const void* buffer,
>+                     size_t bytes,
>+                     size_t* bytes_written,
>+                     int* error) override;
>+  void Close() override;
>+  bool SetPosition(size_t position) override;
>+  bool GetPosition(size_t* position) const override;
>+  bool GetSize(size_t* size) const override;
>+  bool ReserveSize(size_t size) override;
>+
>+  char* GetBuffer() { return buffer_; }
>+  const char* GetBuffer() const { return buffer_; }
>+
>+  void SetData(const void* data, size_t length);
>+
>+ private:
>+  StreamResult DoReserve(size_t size, int* error);
>+
>+  // Invariant: 0 <= seek_position <= data_length_ <= buffer_length_
>+  char* buffer_ = nullptr;
>+  size_t buffer_length_ = 0;
>+  size_t data_length_ = 0;
>+  size_t seek_position_ = 0;
>+};
>+
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_MEMORY_STREAM_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage.cc
>index a70c54753fc4af7c857435c48d2a74eb1a244bf5..9cd36d372099aa497b0b43abc866ebbe23ca646e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage.cc
>@@ -61,6 +61,9 @@ int64_t GetProcessResidentSizeBytes() {
>     return -1;
>   }
>   return pmc.WorkingSetSize;
>+#elif defined(WEBRTC_FUCHSIA)
>+  RTC_LOG_ERR(LS_ERROR) << "GetProcessResidentSizeBytes() not implemented";
>+  return 0;
> #else
>   // Not implemented yet.
>   static_assert(false,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.cc
>index 9c10bcd486a761bbed6dfa3a39b6dba80da44f89..5a0d16a9a04b672ec69cff77fc98287450677a74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.cc
>@@ -10,9 +10,9 @@
> 
> #include "rtc_base/messagedigest.h"
> 
>-#include <memory>
>-
> #include <string.h>
>+#include <cstdint>
>+#include <memory>
> 
> #include "rtc_base/openssldigest.h"
> #include "rtc_base/stringencode.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.h
>index fc820885436f8ce71d45f6cb2153a7ef6173b398..757f914876b07e4130781e8e4bc17fa5faa97f97 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.h
>@@ -11,6 +11,7 @@
> #ifndef RTC_BASE_MESSAGEDIGEST_H_
> #define RTC_BASE_MESSAGEDIGEST_H_
> 
>+#include <stddef.h>
> #include <string>
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>index df2d1ada8ddcecc4493e5ba245a874270eb03af9..0c40853e214855c46478c5f7a90eff142264c70a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>@@ -11,7 +11,6 @@
> #ifndef RTC_BASE_MESSAGEHANDLER_H_
> #define RTC_BASE_MESSAGEHANDLER_H_
> 
>-#include <memory>
> #include <utility>
> 
> #include "rtc_base/constructormagic.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.cc
>index a561af42eb8084a74a39bebc9a448877b6233d8f..204952a7e2abfb42338834e0ec934c4ebcfe5d0b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.cc
>@@ -8,13 +8,15 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> #include <algorithm>
>+#include <string>
>+#include <utility>
> 
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/messagequeue.h"
>-#include "rtc_base/stringencode.h"
> #include "rtc_base/thread.h"
>+#include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
> 
> namespace rtc {
>@@ -348,8 +350,10 @@ void MessageQueue::Post(const Location& posted_from,
>                         uint32_t id,
>                         MessageData* pdata,
>                         bool time_sensitive) {
>-  if (IsQuitting())
>+  if (IsQuitting()) {
>+    delete pdata;
>     return;
>+  }
> 
>   // Keep thread safe
>   // Add the message to the end of the queue
>@@ -405,6 +409,7 @@ void MessageQueue::DoDelayPost(const Location& posted_from,
>                                uint32_t id,
>                                MessageData* pdata) {
>   if (IsQuitting()) {
>+    delete pdata;
>     return;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.h
>index e7e479285d20c8339a62ee63eaf3aa50d099e02e..c1b9b5a916a895ac06f2bdeb7b599cac63de78b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.h
>@@ -17,7 +17,6 @@
> #include <list>
> #include <memory>
> #include <queue>
>-#include <utility>
> #include <vector>
> 
> #include "rtc_base/constructormagic.h"
>@@ -28,7 +27,6 @@
> #include "rtc_base/socketserver.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread_annotations.h"
>-#include "rtc_base/timeutils.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.cc
>index 81193761a63c055c72691a91dfc63c033389e10f..b005ecae7a0d1a5bf6a0a6a4582d10a1c78d9a68 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.cc
>@@ -158,7 +158,7 @@ void NATServer::OnInternalUDPPacket(AsyncPacketSocket* socket,
>                                     const char* buf,
>                                     size_t size,
>                                     const SocketAddress& addr,
>-                                    const PacketTime& packet_time) {
>+                                    const int64_t& /* packet_time_us */) {
>   // Read the intended destination from the wire.
>   SocketAddress dest_addr;
>   size_t length = UnpackAddressFromNAT(buf, size, &dest_addr);
>@@ -184,7 +184,7 @@ void NATServer::OnExternalUDPPacket(AsyncPacketSocket* socket,
>                                     const char* buf,
>                                     size_t size,
>                                     const SocketAddress& remote_addr,
>-                                    const PacketTime& packet_time) {
>+                                    const int64_t& /* packet_time_us */) {
>   SocketAddress local_addr = socket->GetLocalAddress();
> 
>   // Find the translation for this addresses.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.h
>index f4cabcf0d9b7e130b5c8d299a95c48b89110de64..d16b5370f0507c6c84ee88813539cde9befc6a5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.h
>@@ -81,12 +81,12 @@ class NATServer : public sigslot::has_slots<> {
>                            const char* buf,
>                            size_t size,
>                            const SocketAddress& addr,
>-                           const PacketTime& packet_time);
>+                           const int64_t& packet_time_us);
>   void OnExternalUDPPacket(AsyncPacketSocket* socket,
>                            const char* buf,
>                            size_t size,
>                            const SocketAddress& remote_addr,
>-                           const PacketTime& packet_time);
>+                           const int64_t& packet_time_us);
> 
>  private:
>   typedef std::set<SocketAddress, AddrCmp> AddressSet;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelper.h
>index e86d126cbf45c6f79ea201515501588d1a583deb..f9561380fb7d23032aae9f6b06ce921933c0ee17 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelper.h
>@@ -10,7 +10,6 @@
> #ifndef RTC_BASE_NETHELPER_H_
> #define RTC_BASE_NETHELPER_H_
> 
>-#include <cstdlib>
> #include <string>
> 
> // This header contains helper functions and constants used by different types
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.cc
>index b1221c3d0014db634418bbdbfb1d7e65878ce85b..81cd1afcef4a069e6a76a076da20d1ce1071b315 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.cc
>@@ -25,10 +25,9 @@
> #endif
> #endif  // defined(WEBRTC_POSIX) && !defined(__native_client__)
> 
>-#include "rtc_base/byteorder.h"
>-#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/signalthread.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"  // for signal_with_thread...
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.h
>index f9d188f61d09a778251373c738dae7e869324a1e..138f9583ccca1aca8a808fdadc412c8fb4a80fc0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.h
>@@ -12,18 +12,17 @@
> #define RTC_BASE_NETHELPERS_H_
> 
> #if defined(WEBRTC_POSIX)
>-#include <netdb.h>
>-#include <stddef.h>
>+#include <sys/socket.h>
> #elif WEBRTC_WIN
> #include <winsock2.h>  // NOLINT
> #endif
> 
>-#include <list>
>+#include <vector>
> 
> #include "rtc_base/asyncresolverinterface.h"
>+#include "rtc_base/ipaddress.h"
> #include "rtc_base/signalthread.h"
> #include "rtc_base/socketaddress.h"
>-#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.cc
>index 67888ede8342310a0706cf13af824ae9b5f32992..5c7b019d66c2fdd3255d520ce3203f7c96b91bc8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.cc
>@@ -38,7 +38,6 @@
> #include "rtc_base/logging.h"
> #include "rtc_base/networkmonitor.h"
> #include "rtc_base/socket.h"  // includes something that makes windows happy
>-#include "rtc_base/stream.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/strings/string_builder.h"
> #include "rtc_base/stringutils.h"
>@@ -260,7 +259,7 @@ bool NetworkManager::GetDefaultLocalAddress(int family, IPAddress* addr) const {
>   return false;
> }
> 
>-webrtc::MDnsResponderInterface* NetworkManager::GetMDnsResponder() const {
>+webrtc::MdnsResponderInterface* NetworkManager::GetMdnsResponder() const {
>   return nullptr;
> }
> 
>@@ -286,7 +285,7 @@ void NetworkManagerBase::GetAnyAddressNetworks(NetworkList* networks) {
>         new rtc::Network("any", "any", ipv4_any_address, 0, ADAPTER_TYPE_ANY));
>     ipv4_any_address_network_->set_default_local_address_provider(this);
>     ipv4_any_address_network_->AddIP(ipv4_any_address);
>-    ipv4_any_address_network_->SetMDnsResponder(GetMDnsResponder());
>+    ipv4_any_address_network_->SetMdnsResponder(GetMdnsResponder());
>   }
>   networks->push_back(ipv4_any_address_network_.get());
> 
>@@ -297,7 +296,7 @@ void NetworkManagerBase::GetAnyAddressNetworks(NetworkList* networks) {
>           "any", "any", ipv6_any_address, 0, ADAPTER_TYPE_ANY));
>       ipv6_any_address_network_->set_default_local_address_provider(this);
>       ipv6_any_address_network_->AddIP(ipv6_any_address);
>-      ipv6_any_address_network_->SetMDnsResponder(GetMDnsResponder());
>+      ipv6_any_address_network_->SetMdnsResponder(GetMdnsResponder());
>     }
>     networks->push_back(ipv6_any_address_network_.get());
>   }
>@@ -387,7 +386,7 @@ void NetworkManagerBase::MergeNetworkList(const NetworkList& new_networks,
>         delete net;
>       }
>     }
>-    networks_map_[key]->SetMDnsResponder(GetMDnsResponder());
>+    networks_map_[key]->SetMdnsResponder(GetMdnsResponder());
>   }
>   // It may still happen that the merged list is a subset of |networks_|.
>   // To detect this change, we compare their sizes.
>@@ -775,26 +774,28 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
> 
> #if defined(WEBRTC_LINUX)
> bool IsDefaultRoute(const std::string& network_name) {
>-  FileStream fs;
>-  if (!fs.Open("/proc/net/route", "r", nullptr)) {
>+  FILE* f = fopen("/proc/net/route", "r");
>+  if (!f) {
>     RTC_LOG(LS_WARNING)
>         << "Couldn't read /proc/net/route, skipping default "
>         << "route check (assuming everything is a default route).";
>     return true;
>-  } else {
>-    std::string line;
>-    while (fs.ReadLine(&line) == SR_SUCCESS) {
>-      char iface_name[256];
>-      unsigned int iface_ip, iface_gw, iface_mask, iface_flags;
>-      if (sscanf(line.c_str(), "%255s %8X %8X %4X %*d %*u %*d %8X", iface_name,
>-                 &iface_ip, &iface_gw, &iface_flags, &iface_mask) == 5 &&
>-          network_name == iface_name && iface_mask == 0 &&
>-          (iface_flags & (RTF_UP | RTF_HOST)) == RTF_UP) {
>-        return true;
>-      }
>+  }
>+  bool is_default_route = false;
>+  char line[500];
>+  while (fgets(line, sizeof(line), f)) {
>+    char iface_name[256];
>+    unsigned int iface_ip, iface_gw, iface_mask, iface_flags;
>+    if (sscanf(line, "%255s %8X %8X %4X %*d %*u %*d %8X", iface_name, &iface_ip,
>+               &iface_gw, &iface_flags, &iface_mask) == 5 &&
>+        network_name == iface_name && iface_mask == 0 &&
>+        (iface_flags & (RTF_UP | RTF_HOST)) == RTF_UP) {
>+      is_default_route = true;
>+      break;
>     }
>   }
>-  return false;
>+  fclose(f);
>+  return is_default_route;
> }
> #endif
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>index 601e39f02771d03293e8d73008ac83d4b277b786..8b1a5fcc5c3ded1183ba86b8037dd8d86a90d4cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>@@ -141,7 +141,7 @@ class NetworkManager : public DefaultLocalAddressProvider {
> 
>   // Returns the mDNS responder that can be used to obfuscate the local IP
>   // addresses of ICE host candidates by mDNS hostnames.
>-  virtual webrtc::MDnsResponderInterface* GetMDnsResponder() const;
>+  virtual webrtc::MdnsResponderInterface* GetMdnsResponder() const;
> };
> 
> // Base class for NetworkManager implementations.
>@@ -367,11 +367,11 @@ class Network {
>   // created name will be resolved by the responder.
>   //
>   // The mDNS responder, if not null, should outlive this rtc::Network.
>-  void SetMDnsResponder(webrtc::MDnsResponderInterface* mdns_responder) {
>+  void SetMdnsResponder(webrtc::MdnsResponderInterface* mdns_responder) {
>     mdns_responder_ = mdns_responder;
>   }
>   // Returns the mDNS responder, which is null by default.
>-  webrtc::MDnsResponderInterface* GetMDnsResponder() const {
>+  webrtc::MdnsResponderInterface* GetMdnsResponder() const {
>     return mdns_responder_;
>   }
> 
>@@ -446,7 +446,7 @@ class Network {
>   int prefix_length_;
>   std::string key_;
>   std::vector<InterfaceAddress> ips_;
>-  webrtc::MDnsResponderInterface* mdns_responder_ = nullptr;
>+  webrtc::MdnsResponderInterface* mdns_responder_ = nullptr;
>   int scope_id_;
>   bool ignored_;
>   AdapterType type_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network/BUILD.gn
>new file mode 100644
>index 0000000000000000000000000000000000000000..0fbdbb1edbb0afc24e1f4bb7c135436a48ee842c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network/BUILD.gn
>@@ -0,0 +1,19 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../webrtc.gni")
>+
>+rtc_source_set("sent_packet") {
>+  sources = [
>+    "sent_packet.cc",
>+    "sent_packet.h",
>+  ]
>+  deps = [
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network/sent_packet.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network/sent_packet.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..8cc49737efdeca7081e155cdc7541e835c1c0bee
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network/sent_packet.cc
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/network/sent_packet.h"
>+
>+namespace rtc {
>+
>+PacketInfo::PacketInfo() = default;
>+PacketInfo::PacketInfo(const PacketInfo& info) = default;
>+PacketInfo::~PacketInfo() = default;
>+
>+SentPacket::SentPacket() = default;
>+SentPacket::SentPacket(int64_t packet_id, int64_t send_time_ms)
>+    : packet_id(packet_id), send_time_ms(send_time_ms) {}
>+SentPacket::SentPacket(int64_t packet_id,
>+                       int64_t send_time_ms,
>+                       const rtc::PacketInfo& info)
>+    : packet_id(packet_id), send_time_ms(send_time_ms), info(info) {}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network/sent_packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network/sent_packet.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..0cad31ca91c0f8902bb4ae4a8ddf478014c1a435
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network/sent_packet.h
>@@ -0,0 +1,68 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_NETWORK_SENT_PACKET_H_
>+#define RTC_BASE_NETWORK_SENT_PACKET_H_
>+
>+#include <stddef.h>
>+#include <stdint.h>
>+
>+#include "absl/types/optional.h"
>+
>+namespace rtc {
>+
>+enum class PacketType {
>+  kUnknown,
>+  kData,
>+  kIceConnectivityCheck,
>+  kIceConnectivityCheckResponse,
>+  kStunMessage,
>+  kTurnMessage,
>+};
>+
>+enum class PacketInfoProtocolType {
>+  kUnknown,
>+  kUdp,
>+  kTcp,
>+  kSsltcp,
>+  kTls,
>+};
>+
>+struct PacketInfo {
>+  PacketInfo();
>+  PacketInfo(const PacketInfo& info);
>+  ~PacketInfo();
>+
>+  bool included_in_feedback = false;
>+  bool included_in_allocation = false;
>+  PacketType packet_type = PacketType::kUnknown;
>+  PacketInfoProtocolType protocol = PacketInfoProtocolType::kUnknown;
>+  // A unique id assigned by the network manager, and absl::nullopt if not set.
>+  absl::optional<uint16_t> network_id;
>+  size_t packet_size_bytes = 0;
>+  size_t turn_overhead_bytes = 0;
>+  size_t ip_overhead_bytes = 0;
>+};
>+
>+struct SentPacket {
>+  SentPacket();
>+  SentPacket(int64_t packet_id, int64_t send_time_ms);
>+  SentPacket(int64_t packet_id,
>+             int64_t send_time_ms,
>+             const rtc::PacketInfo& info);
>+
>+  int64_t packet_id = -1;
>+  int64_t send_time_ms = -1;
>+  rtc::PacketInfo info;
>+};
>+
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_NETWORK_SENT_PACKET_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.cc
>index ad6805acec7e798ff8a9c6ea33dc9775580cbbb0..0185eab85613a3b19561a61fe4aaca165f3fece8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.cc
>@@ -10,7 +10,11 @@
> 
> #include "rtc_base/networkmonitor.h"
> 
>+#include <stdint.h>
>+
> #include "rtc_base/checks.h"
>+#include "rtc_base/location.h"
>+#include "rtc_base/logging.h"
> 
> namespace {
> const uint32_t UPDATE_NETWORKS_MESSAGE = 1;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.h
>index a84a30aed2bc3368665b7d4314775e63bf05561f..1ad76636696cc8e66fedbbeb01fb25e66cc77cac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.h
>@@ -11,7 +11,6 @@
> #ifndef RTC_BASE_NETWORKMONITOR_H_
> #define RTC_BASE_NETWORKMONITOR_H_
> 
>-#include "rtc_base/logging.h"
> #include "rtc_base/network_constants.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/noop.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/noop.cc
>deleted file mode 100644
>index 16a8e6d5c10b97134e68b08e2c65ef2fff16bd52..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/noop.cc
>+++ /dev/null
>@@ -1,13 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// This file is only needed to make ninja happy on some platforms.
>-// On some platforms it is not possible to link an rtc_static_library
>-// without any source file listed in the GN target.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/noop.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/noop.mm
>deleted file mode 100644
>index 16a8e6d5c10b97134e68b08e2c65ef2fff16bd52..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/noop.mm
>+++ /dev/null
>@@ -1,13 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// This file is only needed to make ninja happy on some platforms.
>-// On some platforms it is not possible to link an rtc_static_library
>-// without any source file listed in the GN target.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.cc
>index c890c6f3dcf653f9694efa6bbc5fc82991cac91d..ec042dd09517dc18e7e1e04b849ae517850a30d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.cc
>@@ -13,7 +13,7 @@
> 
> namespace rtc {
> 
>-NullSocketServer::NullSocketServer() : event_(false, false) {}
>+NullSocketServer::NullSocketServer() = default;
> NullSocketServer::~NullSocketServer() {}
> 
> bool NullSocketServer::Wait(int cms, bool process_io) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.h
>index 408bcd1b2f5163621f131015b6c5f6defab39c1a..47a7fa611871c4423d2516e398c9480c485f83ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.h
>@@ -11,7 +11,9 @@
> #ifndef RTC_BASE_NULLSOCKETSERVER_H_
> #define RTC_BASE_NULLSOCKETSERVER_H_
> 
>+#include "rtc_base/asyncsocket.h"
> #include "rtc_base/event.h"
>+#include "rtc_base/socket.h"
> #include "rtc_base/socketserver.h"
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_average.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_average.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..c825839227ebb8a4aac7c591138f60940508d27f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_average.cc
>@@ -0,0 +1,60 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/numerics/moving_average.h"
>+
>+#include <algorithm>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace rtc {
>+
>+MovingAverage::MovingAverage(size_t window_size) : history_(window_size, 0) {
>+  // Limit window size to avoid overflow.
>+  RTC_DCHECK_LE(window_size, (int64_t{1} << 32) - 1);
>+}
>+MovingAverage::~MovingAverage() = default;
>+
>+void MovingAverage::AddSample(int sample) {
>+  count_++;
>+  size_t index = count_ % history_.size();
>+  if (count_ > history_.size())
>+    sum_ -= history_[index];
>+  sum_ += sample;
>+  history_[index] = sample;
>+}
>+
>+absl::optional<int> MovingAverage::GetAverageRoundedDown() const {
>+  if (count_ == 0)
>+    return absl::nullopt;
>+  return sum_ / Size();
>+}
>+
>+absl::optional<int> MovingAverage::GetAverageRoundedToClosest() const {
>+  if (count_ == 0)
>+    return absl::nullopt;
>+  return (sum_ + Size() / 2) / Size();
>+}
>+
>+absl::optional<double> MovingAverage::GetUnroundedAverage() const {
>+  if (count_ == 0)
>+    return absl::nullopt;
>+  return sum_ / static_cast<double>(Size());
>+}
>+
>+void MovingAverage::Reset() {
>+  count_ = 0;
>+  sum_ = 0;
>+}
>+
>+size_t MovingAverage::Size() const {
>+  return std::min(count_, history_.size());
>+}
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_average.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_average.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..770e47d86f8128fcf7774d4bc4509676975c62ac
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_average.h
>@@ -0,0 +1,63 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_NUMERICS_MOVING_AVERAGE_H_
>+#define RTC_BASE_NUMERICS_MOVING_AVERAGE_H_
>+
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+
>+namespace rtc {
>+
>+// Calculates average over fixed size window. If there are less than window
>+// size elements, calculates average of all inserted so far elements.
>+//
>+class MovingAverage {
>+ public:
>+  // Maximum supported window size is 2^32 - 1.
>+  explicit MovingAverage(size_t window_size);
>+  ~MovingAverage();
>+  // MovingAverage is neither copyable nor movable.
>+  MovingAverage(const MovingAverage&) = delete;
>+  MovingAverage& operator=(const MovingAverage&) = delete;
>+
>+  // Adds new sample. If the window is full, the oldest element is pushed out.
>+  void AddSample(int sample);
>+
>+  // Returns rounded down average of last |window_size| elements or all
>+  // elements if there are not enough of them. Returns nullopt if there were
>+  // no elements added.
>+  absl::optional<int> GetAverageRoundedDown() const;
>+
>+  // Same as above but rounded to the closest integer.
>+  absl::optional<int> GetAverageRoundedToClosest() const;
>+
>+  // Returns unrounded average over the window.
>+  absl::optional<double> GetUnroundedAverage() const;
>+
>+  // Resets to the initial state before any elements were added.
>+  void Reset();
>+
>+  // Returns number of elements in the window.
>+  size_t Size() const;
>+
>+ private:
>+  // Total number of samples added to the class since last reset.
>+  size_t count_ = 0;
>+  // Sum of the samples in the moving window.
>+  int64_t sum_ = 0;
>+  // Circular buffer for all the samples in the moving window.
>+  // Size is always |window_size|
>+  std::vector<int> history_;
>+};
>+
>+}  // namespace rtc
>+#endif  // RTC_BASE_NUMERICS_MOVING_AVERAGE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_average_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_average_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..9bc9a1aef89a14ba9366be888fd22b2d13960959
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_average_unittest.cc
>@@ -0,0 +1,87 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/numerics/moving_average.h"
>+
>+#include "test/gtest.h"
>+
>+namespace test {
>+
>+TEST(MovingAverageTest, EmptyAverage) {
>+  rtc::MovingAverage moving_average(1);
>+  EXPECT_EQ(0u, moving_average.Size());
>+  EXPECT_EQ(absl::nullopt, moving_average.GetAverageRoundedDown());
>+}
>+
>+// Test single value.
>+TEST(MovingAverageTest, OneElement) {
>+  rtc::MovingAverage moving_average(1);
>+  moving_average.AddSample(3);
>+  EXPECT_EQ(1u, moving_average.Size());
>+  EXPECT_EQ(3, *moving_average.GetAverageRoundedDown());
>+}
>+
>+TEST(MovingAverageTest, GetAverage) {
>+  rtc::MovingAverage moving_average(1024);
>+  moving_average.AddSample(1);
>+  moving_average.AddSample(1);
>+  moving_average.AddSample(3);
>+  moving_average.AddSample(3);
>+  EXPECT_EQ(*moving_average.GetAverageRoundedDown(), 2);
>+  EXPECT_EQ(*moving_average.GetAverageRoundedToClosest(), 2);
>+}
>+
>+TEST(MovingAverageTest, GetAverageRoundedDownRounds) {
>+  rtc::MovingAverage moving_average(1024);
>+  moving_average.AddSample(1);
>+  moving_average.AddSample(2);
>+  moving_average.AddSample(2);
>+  moving_average.AddSample(2);
>+  EXPECT_EQ(*moving_average.GetAverageRoundedDown(), 1);
>+}
>+
>+TEST(MovingAverageTest, GetAverageRoundedToClosestRounds) {
>+  rtc::MovingAverage moving_average(1024);
>+  moving_average.AddSample(1);
>+  moving_average.AddSample(2);
>+  moving_average.AddSample(2);
>+  moving_average.AddSample(2);
>+  EXPECT_EQ(*moving_average.GetAverageRoundedToClosest(), 2);
>+}
>+
>+TEST(MovingAverageTest, Reset) {
>+  rtc::MovingAverage moving_average(5);
>+  moving_average.AddSample(1);
>+  EXPECT_EQ(1, *moving_average.GetAverageRoundedDown());
>+  EXPECT_EQ(1, *moving_average.GetAverageRoundedToClosest());
>+
>+  moving_average.Reset();
>+
>+  EXPECT_FALSE(moving_average.GetAverageRoundedDown());
>+  moving_average.AddSample(10);
>+  EXPECT_EQ(10, *moving_average.GetAverageRoundedDown());
>+  EXPECT_EQ(10, *moving_average.GetAverageRoundedToClosest());
>+}
>+
>+TEST(MovingAverageTest, ManySamples) {
>+  rtc::MovingAverage moving_average(10);
>+  for (int i = 1; i < 11; i++) {
>+    moving_average.AddSample(i);
>+  }
>+  EXPECT_EQ(*moving_average.GetAverageRoundedDown(), 5);
>+  EXPECT_EQ(*moving_average.GetAverageRoundedToClosest(), 6);
>+  for (int i = 1; i < 2001; i++) {
>+    moving_average.AddSample(i);
>+  }
>+  EXPECT_EQ(*moving_average.GetAverageRoundedDown(), 1995);
>+  EXPECT_EQ(*moving_average.GetAverageRoundedToClosest(), 1996);
>+}
>+
>+}  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter.h
>index 4fe71d19427a23ee229ecf4d21b61710d85080ee..18bd36b8f3052a652a1aa7be986d6db3c9717927 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter.h
>@@ -11,6 +11,8 @@
> #ifndef RTC_BASE_NUMERICS_SAMPLE_COUNTER_H_
> #define RTC_BASE_NUMERICS_SAMPLE_COUNTER_H_
> 
>+#include <stdint.h>
>+
> #include "absl/types/optional.h"
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl_key_derivation_hkdf.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl_key_derivation_hkdf.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..52af667645b39ee7703ce7b74ff20455b9e8acae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl_key_derivation_hkdf.cc
>@@ -0,0 +1,70 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/openssl_key_derivation_hkdf.h"
>+
>+#include <openssl/digest.h>
>+#include <openssl/err.h>
>+#include <openssl/hkdf.h>
>+#include <openssl/sha.h>
>+
>+#include <algorithm>
>+#include <utility>
>+
>+#include "rtc_base/buffer.h"
>+#include "rtc_base/openssl.h"
>+
>+namespace rtc {
>+
>+OpenSSLKeyDerivationHKDF::OpenSSLKeyDerivationHKDF() = default;
>+OpenSSLKeyDerivationHKDF::~OpenSSLKeyDerivationHKDF() = default;
>+
>+const size_t OpenSSLKeyDerivationHKDF::kMinKeyByteSize = 16;
>+const size_t OpenSSLKeyDerivationHKDF::kMaxKeyByteSize =
>+    255 * SHA256_DIGEST_LENGTH;
>+const size_t OpenSSLKeyDerivationHKDF::kMinSecretByteSize = 16;
>+
>+absl::optional<ZeroOnFreeBuffer<uint8_t>> OpenSSLKeyDerivationHKDF::DeriveKey(
>+    rtc::ArrayView<const uint8_t> secret,
>+    rtc::ArrayView<const uint8_t> salt,
>+    rtc::ArrayView<const uint8_t> label,
>+    size_t derived_key_byte_size) {
>+  // Prevent deriving less than 128 bits of key material or more than the max.
>+  if (derived_key_byte_size < kMinKeyByteSize ||
>+      derived_key_byte_size > kMaxKeyByteSize) {
>+    return absl::nullopt;
>+  }
>+  // The secret must reach the minimum number of bits to be secure.
>+  if (secret.data() == nullptr || secret.size() < kMinSecretByteSize) {
>+    return absl::nullopt;
>+  }
>+  // Empty labels are always invalid in derivation.
>+  if (label.data() == nullptr || label.size() == 0) {
>+    return absl::nullopt;
>+  }
>+  // If a random salt is not provided use all zeros.
>+  rtc::Buffer salt_buffer;
>+  if (salt.data() == nullptr || salt.size() == 0) {
>+    salt_buffer.SetSize(SHA256_DIGEST_LENGTH);
>+    std::fill(salt_buffer.begin(), salt_buffer.end(), 0);
>+    salt = salt_buffer;
>+  }
>+  // This buffer will erase itself on release.
>+  ZeroOnFreeBuffer<uint8_t> derived_key_buffer(derived_key_byte_size, 0);
>+  if (!HKDF(derived_key_buffer.data(), derived_key_buffer.size(), EVP_sha256(),
>+            secret.data(), secret.size(), salt.data(), salt.size(),
>+            label.data(), label.size())) {
>+    return absl::nullopt;
>+  }
>+  return absl::optional<ZeroOnFreeBuffer<uint8_t>>(
>+      std::move(derived_key_buffer));
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl_key_derivation_hkdf.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl_key_derivation_hkdf.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..ebf43cfaa7abad7a9a11aa5d5096e81efa1ee59f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl_key_derivation_hkdf.h
>@@ -0,0 +1,54 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_OPENSSL_KEY_DERIVATION_HKDF_H_
>+#define RTC_BASE_OPENSSL_KEY_DERIVATION_HKDF_H_
>+
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/key_derivation.h"
>+
>+namespace rtc {
>+
>+// OpenSSLKeyDerivationHKDF provides a concrete implementation of the
>+// KeyDerivation interface to support the HKDF algorithm using the
>+// OpenSSL/BoringSSL internal implementation.
>+class OpenSSLKeyDerivationHKDF final : public KeyDerivation {
>+ public:
>+  OpenSSLKeyDerivationHKDF();
>+  ~OpenSSLKeyDerivationHKDF() override;
>+
>+  // General users shouldn't be generating keys smaller than 128 bits.
>+  static const size_t kMinKeyByteSize;
>+  // The maximum available derivation size 255*DIGEST_LENGTH
>+  static const size_t kMaxKeyByteSize;
>+  // The minimum acceptable secret size.
>+  static const size_t kMinSecretByteSize;
>+
>+  // Derives a new key from existing key material using HKDF.
>+  // secret - The random secret value you wish to derive a key from.
>+  // salt - Optional (non secret) cryptographically random value.
>+  // label - A non secret but unique label value to determine the derivation.
>+  // derived_key_byte_size - The size of the derived key.
>+  // return - A ZeroOnFreeBuffer containing the derived key or an error
>+  // condition. Checking error codes is explicit in the API and error should
>+  // never be ignored.
>+  absl::optional<ZeroOnFreeBuffer<uint8_t>> DeriveKey(
>+      rtc::ArrayView<const uint8_t> secret,
>+      rtc::ArrayView<const uint8_t> salt,
>+      rtc::ArrayView<const uint8_t> label,
>+      size_t derived_key_byte_size) override;
>+
>+ private:
>+  RTC_DISALLOW_COPY_AND_ASSIGN(OpenSSLKeyDerivationHKDF);
>+};
>+
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_OPENSSL_KEY_DERIVATION_HKDF_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl_key_derivation_hkdf_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl_key_derivation_hkdf_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..92df42ffd30d6b4fd359c14b13d3883dea9af4db
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl_key_derivation_hkdf_unittest.cc
>@@ -0,0 +1,107 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/openssl_key_derivation_hkdf.h"
>+
>+#include <utility>
>+
>+#include "test/gmock.h"
>+
>+namespace rtc {
>+namespace {
>+
>+// Validates that a basic valid call works correctly.
>+TEST(OpenSSLKeyDerivationHKDF, DerivationBasicTest) {
>+  rtc::Buffer secret(32);
>+  rtc::Buffer salt(32);
>+  rtc::Buffer label(32);
>+  const size_t derived_key_byte_size = 16;
>+
>+  OpenSSLKeyDerivationHKDF hkdf;
>+  auto key_or = hkdf.DeriveKey(secret, salt, label, derived_key_byte_size);
>+  EXPECT_TRUE(key_or.has_value());
>+  ZeroOnFreeBuffer<uint8_t> key = std::move(key_or.value());
>+  EXPECT_EQ(derived_key_byte_size, key.size());
>+}
>+
>+// Derivation fails if output is too small.
>+TEST(OpenSSLKeyDerivationHKDF, DerivationFailsIfOutputIsTooSmall) {
>+  rtc::Buffer secret(32);
>+  rtc::Buffer salt(32);
>+  rtc::Buffer label(32);
>+  const size_t derived_key_byte_size = 15;
>+
>+  OpenSSLKeyDerivationHKDF hkdf;
>+  auto key_or = hkdf.DeriveKey(secret, salt, label, derived_key_byte_size);
>+  EXPECT_FALSE(key_or.has_value());
>+}
>+
>+// Derivation fails if output is too large.
>+TEST(OpenSSLKeyDerivationHKDF, DerivationFailsIfOutputIsTooLarge) {
>+  rtc::Buffer secret(32);
>+  rtc::Buffer salt(32);
>+  rtc::Buffer label(32);
>+  const size_t derived_key_byte_size = 256 * 32;
>+
>+  OpenSSLKeyDerivationHKDF hkdf;
>+  auto key_or = hkdf.DeriveKey(secret, salt, label, derived_key_byte_size);
>+  EXPECT_FALSE(key_or.has_value());
>+}
>+
>+// Validates that too little key material causes a failure.
>+TEST(OpenSSLKeyDerivationHKDF, DerivationFailsWithInvalidSecret) {
>+  rtc::Buffer secret(15);
>+  rtc::Buffer salt(32);
>+  rtc::Buffer label(32);
>+  const size_t derived_key_byte_size = 16;
>+
>+  OpenSSLKeyDerivationHKDF hkdf;
>+  auto key_or_0 = hkdf.DeriveKey(secret, salt, label, derived_key_byte_size);
>+  EXPECT_FALSE(key_or_0.has_value());
>+
>+  auto key_or_1 = hkdf.DeriveKey(nullptr, salt, label, derived_key_byte_size);
>+  EXPECT_FALSE(key_or_1.has_value());
>+
>+  rtc::Buffer secret_empty;
>+  auto key_or_2 =
>+      hkdf.DeriveKey(secret_empty, salt, label, derived_key_byte_size);
>+  EXPECT_FALSE(key_or_2.has_value());
>+}
>+
>+// Validates that HKDF works without a salt being set.
>+TEST(OpenSSLKeyDerivationHKDF, DerivationWorksWithNoSalt) {
>+  rtc::Buffer secret(32);
>+  rtc::Buffer label(32);
>+  const size_t derived_key_byte_size = 16;
>+
>+  OpenSSLKeyDerivationHKDF hkdf;
>+  auto key_or = hkdf.DeriveKey(secret, nullptr, label, derived_key_byte_size);
>+  EXPECT_TRUE(key_or.has_value());
>+}
>+
>+// Validates that a label is required to work correctly.
>+TEST(OpenSSLKeyDerivationHKDF, DerivationRequiresLabel) {
>+  rtc::Buffer secret(32);
>+  rtc::Buffer salt(32);
>+  rtc::Buffer label(1);
>+  const size_t derived_key_byte_size = 16;
>+
>+  OpenSSLKeyDerivationHKDF hkdf;
>+  auto key_or_0 = hkdf.DeriveKey(secret, salt, label, derived_key_byte_size);
>+  EXPECT_TRUE(key_or_0.has_value());
>+  ZeroOnFreeBuffer<uint8_t> key = std::move(key_or_0.value());
>+  EXPECT_EQ(key.size(), derived_key_byte_size);
>+
>+  auto key_or_1 = hkdf.DeriveKey(secret, salt, nullptr, derived_key_byte_size);
>+  EXPECT_FALSE(key_or_1.has_value());
>+}
>+
>+}  // namespace
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.cc
>index 50284a6719fe45146ace9266e0f9cbdb3bd633d8..fcfa53b7ee70d84ec1baad00b867454e4313901d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.cc
>@@ -10,25 +10,25 @@
> 
> #include "rtc_base/openssladapter.h"
> 
>-#if defined(WEBRTC_POSIX)
>-#include <unistd.h>
>-#endif
>+#include <errno.h>
> 
> #include <openssl/bio.h>
>-#include <openssl/crypto.h>
> #include <openssl/err.h>
>-#include <openssl/opensslv.h>
> #include <openssl/rand.h>
> #include <openssl/x509.h>
>-#include <openssl/x509v3.h>
> #include "rtc_base/openssl.h"
> 
>+#include <string.h>
>+#include <time.h>
>+
>+#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/location.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/opensslcertificate.h"
> #include "rtc_base/opensslutility.h"
> #include "rtc_base/stringencode.h"
>-#include "rtc_base/stringutils.h"
> #include "rtc_base/thread.h"
> 
> #ifndef OPENSSL_IS_BORINGSSL
>@@ -392,15 +392,17 @@ int OpenSSLAdapter::BeginSSL() {
> 
>   // Do the connect.
>   err = ContinueSSL();
>-  if (err != 0)
>+  if (err != 0) {
>     goto ssl_error;
>+  }
> 
>   return err;
> 
> ssl_error:
>   Cleanup();
>-  if (bio)
>+  if (bio) {
>     BIO_free(bio);
>+  }
> 
>   return err;
> }
>@@ -424,14 +426,13 @@ int OpenSSLAdapter::ContinueSSL() {
> 
>       state_ = SSL_CONNECTED;
>       AsyncSocketAdapter::OnConnectEvent(this);
>-#if 0  // TODO(benwright): worry about this
>-    // Don't let ourselves go away during the callbacks
>-    PRefPtr<OpenSSLAdapter> lock(this);
>-    RTC_LOG(LS_INFO) << " -- onStreamReadable";
>-    AsyncSocketAdapter::OnReadEvent(this);
>-    RTC_LOG(LS_INFO) << " -- onStreamWriteable";
>-    AsyncSocketAdapter::OnWriteEvent(this);
>-#endif
>+      // TODO(benwright): Refactor this code path.
>+      // Don't let ourselves go away during the callbacks
>+      // PRefPtr<OpenSSLAdapter> lock(this);
>+      // RTC_LOG(LS_INFO) << " -- onStreamReadable";
>+      // AsyncSocketAdapter::OnReadEvent(this);
>+      // RTC_LOG(LS_INFO) << " -- onStreamWriteable";
>+      // AsyncSocketAdapter::OnWriteEvent(this);
>       break;
> 
>     case SSL_ERROR_WANT_READ:
>@@ -462,8 +463,9 @@ void OpenSSLAdapter::Error(const char* context, int err, bool signal) {
>                       << ")";
>   state_ = SSL_ERROR;
>   SetError(err);
>-  if (signal)
>+  if (signal) {
>     AsyncSocketAdapter::OnCloseEvent(this, err);
>+  }
> }
> 
> void OpenSSLAdapter::Cleanup() {
>@@ -528,23 +530,20 @@ int OpenSSLAdapter::DoSslWrite(const void* pv, size_t cb, int* error) {
>   return SOCKET_ERROR;
> }
> 
>-//
>+///////////////////////////////////////////////////////////////////////////////
> // AsyncSocket Implementation
>-//
>+///////////////////////////////////////////////////////////////////////////////
> 
> int OpenSSLAdapter::Send(const void* pv, size_t cb) {
>   switch (state_) {
>     case SSL_NONE:
>       return AsyncSocketAdapter::Send(pv, cb);
>-
>     case SSL_WAIT:
>     case SSL_CONNECTING:
>       SetError(ENOTCONN);
>       return SOCKET_ERROR;
>-
>     case SSL_CONNECTED:
>       break;
>-
>     case SSL_ERROR:
>     default:
>       return SOCKET_ERROR;
>@@ -567,8 +566,9 @@ int OpenSSLAdapter::Send(const void* pv, size_t cb) {
>   }
> 
>   // OpenSSL will return an error if we try to write zero bytes
>-  if (cb == 0)
>+  if (cb == 0) {
>     return 0;
>+  }
> 
>   ret = DoSslWrite(pv, cb, &error);
> 
>@@ -595,7 +595,6 @@ int OpenSSLAdapter::Send(const void* pv, size_t cb) {
>     // size. The user of this class can consider it sent.
>     return rtc::dchecked_cast<int>(cb);
>   }
>-
>   return ret;
> }
> 
>@@ -608,7 +607,6 @@ int OpenSSLAdapter::SendTo(const void* pv,
>   }
> 
>   SetError(ENOTCONN);
>-
>   return SOCKET_ERROR;
> }
> 
>@@ -616,28 +614,26 @@ int OpenSSLAdapter::Recv(void* pv, size_t cb, int64_t* timestamp) {
>   switch (state_) {
>     case SSL_NONE:
>       return AsyncSocketAdapter::Recv(pv, cb, timestamp);
>-
>     case SSL_WAIT:
>     case SSL_CONNECTING:
>       SetError(ENOTCONN);
>       return SOCKET_ERROR;
>-
>     case SSL_CONNECTED:
>       break;
>-
>     case SSL_ERROR:
>     default:
>       return SOCKET_ERROR;
>   }
> 
>   // Don't trust OpenSSL with zero byte reads
>-  if (cb == 0)
>+  if (cb == 0) {
>     return 0;
>+  }
> 
>   ssl_read_needs_write_ = false;
>-
>   int code = SSL_read(ssl_, pv, checked_cast<int>(cb));
>   int error = SSL_get_error(ssl_, code);
>+
>   switch (error) {
>     case SSL_ERROR_NONE:
>       return code;
>@@ -660,7 +656,6 @@ int OpenSSLAdapter::Recv(void* pv, size_t cb, int64_t* timestamp) {
>       Error("SSL_read", (code ? code : -1), false);
>       break;
>   }
>-
>   return SOCKET_ERROR;
> }
> 
>@@ -670,14 +665,11 @@ int OpenSSLAdapter::RecvFrom(void* pv,
>                              int64_t* timestamp) {
>   if (socket_->GetState() == Socket::CS_CONNECTED) {
>     int ret = Recv(pv, cb, timestamp);
>-
>     *paddr = GetRemoteAddress();
>-
>     return ret;
>   }
> 
>   SetError(ENOTCONN);
>-
>   return SOCKET_ERROR;
> }
> 
>@@ -688,12 +680,11 @@ int OpenSSLAdapter::Close() {
> }
> 
> Socket::ConnState OpenSSLAdapter::GetState() const {
>-  // if (signal_close_)
>-  //  return CS_CONNECTED;
>   ConnState state = socket_->GetState();
>   if ((state == CS_CONNECTED) &&
>-      ((state_ == SSL_WAIT) || (state_ == SSL_CONNECTING)))
>+      ((state_ == SSL_WAIT) || (state_ == SSL_CONNECTING))) {
>     state = CS_CONNECTING;
>+  }
>   return state;
> }
> 
>@@ -736,8 +727,9 @@ void OpenSSLAdapter::OnReadEvent(AsyncSocket* socket) {
>     return;
>   }
> 
>-  if (state_ != SSL_CONNECTED)
>+  if (state_ != SSL_CONNECTED) {
>     return;
>+  }
> 
>   // Don't let ourselves go away during the callbacks
>   // PRefPtr<OpenSSLAdapter> lock(this); // TODO(benwright): fix this
>@@ -761,8 +753,9 @@ void OpenSSLAdapter::OnWriteEvent(AsyncSocket* socket) {
>     return;
>   }
> 
>-  if (state_ != SSL_CONNECTED)
>+  if (state_ != SSL_CONNECTED) {
>     return;
>+  }
> 
>   // Don't let ourselves go away during the callbacks
>   // PRefPtr<OpenSSLAdapter> lock(this); // TODO(benwright): fix this
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.h
>index 50a7c0845ca459dedd282105c773ac97ea8a2d6a..2e3a355f439e750e5c22c8aa9bc6ab696b85c8b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.h
>@@ -11,25 +11,28 @@
> #ifndef RTC_BASE_OPENSSLADAPTER_H_
> #define RTC_BASE_OPENSSLADAPTER_H_
> 
>-#include <openssl/ossl_typ.h>
>-
>-#include <map>
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> #include <string>
> #include <vector>
> 
>-#include "absl/memory/memory.h"
>+#include "rtc_base/asyncsocket.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/messagehandler.h"
> #include "rtc_base/messagequeue.h"
>-#include "rtc_base/opensslcertificate.h"
> #include "rtc_base/opensslidentity.h"
> #include "rtc_base/opensslsessioncache.h"
>+#include "rtc_base/socket.h"
>+#include "rtc_base/socketaddress.h"
> #include "rtc_base/ssladapter.h"
>+#include "rtc_base/sslcertificate.h"
>+#include "rtc_base/sslidentity.h"
>+#include "rtc_base/sslstreamadapter.h"
> 
> namespace rtc {
> 
>-class OpenSSLAdapter : public SSLAdapter, public MessageHandler {
>+class OpenSSLAdapter final : public SSLAdapter, public MessageHandler {
>  public:
>   static bool InitializeSSL();
>   static bool CleanupSSL();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc
>index 236bef55c8aac95a6c714374b8a2c0cea77b12dc..4e325dcec018fc92536a0ce5c828c32a98293830 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc
>@@ -10,10 +10,6 @@
> 
> #include "rtc_base/opensslcertificate.h"
> 
>-#include <memory>
>-#include <utility>
>-#include <vector>
>-
> #if defined(WEBRTC_WIN)
> // Must be included first before openssl headers.
> #include "rtc_base/win32.h"  // NOLINT
>@@ -21,63 +17,69 @@
> 
> #include <openssl/bio.h>
> #include <openssl/bn.h>
>-#include <openssl/crypto.h>
>-#include <openssl/err.h>
> #include <openssl/pem.h>
>-#include <openssl/rsa.h>
>+#include <time.h>
> 
> #include "absl/memory/memory.h"
>-#include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/openssl.h"
>+#include "rtc_base/messagedigest.h"
> #include "rtc_base/openssldigest.h"
> #include "rtc_base/opensslidentity.h"
> #include "rtc_base/opensslutility.h"
>-#ifndef WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
>-#include "rtc_base/sslroots.h"
>-#endif  // WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
> 
> namespace rtc {
>-
>-//////////////////////////////////////////////////////////////////////
>-// OpenSSLCertificate
>-//////////////////////////////////////////////////////////////////////
>-
>-// We could have exposed a myriad of parameters for the crypto stuff,
>-// but keeping it simple seems best.
>+namespace {
> 
> // Random bits for certificate serial number
> static const int SERIAL_RAND_BITS = 64;
> 
>+#if !defined(NDEBUG)
>+// Print a certificate to the log, for debugging.
>+static void PrintCert(X509* x509) {
>+  BIO* temp_memory_bio = BIO_new(BIO_s_mem());
>+  if (!temp_memory_bio) {
>+    RTC_DLOG_F(LS_ERROR) << "Failed to allocate temporary memory bio";
>+    return;
>+  }
>+  X509_print_ex(temp_memory_bio, x509, XN_FLAG_SEP_CPLUS_SPC, 0);
>+  BIO_write(temp_memory_bio, "\0", 1);
>+  char* buffer;
>+  BIO_get_mem_data(temp_memory_bio, &buffer);
>+  RTC_DLOG(LS_VERBOSE) << buffer;
>+  BIO_free(temp_memory_bio);
>+}
>+#endif
>+
> // Generate a self-signed certificate, with the public key from the
> // given key pair. Caller is responsible for freeing the returned object.
> static X509* MakeCertificate(EVP_PKEY* pkey, const SSLIdentityParams& params) {
>   RTC_LOG(LS_INFO) << "Making certificate for " << params.common_name;
>-  X509* x509 = nullptr;
>+
>+  ASN1_INTEGER* asn1_serial_number = nullptr;
>   BIGNUM* serial_number = nullptr;
>+  X509* x509 = nullptr;
>   X509_NAME* name = nullptr;
>   time_t epoch_off = 0;  // Time offset since epoch.
> 
>-  if ((x509 = X509_new()) == nullptr)
>+  if ((x509 = X509_new()) == nullptr) {
>     goto error;
>-
>-  if (!X509_set_pubkey(x509, pkey))
>+  }
>+  if (!X509_set_pubkey(x509, pkey)) {
>     goto error;
>-
>-  // serial number
>-  // temporary reference to serial number inside x509 struct
>-  ASN1_INTEGER* asn1_serial_number;
>+  }
>+  // serial number - temporary reference to serial number inside x509 struct
>   if ((serial_number = BN_new()) == nullptr ||
>       !BN_pseudo_rand(serial_number, SERIAL_RAND_BITS, 0, 0) ||
>       (asn1_serial_number = X509_get_serialNumber(x509)) == nullptr ||
>-      !BN_to_ASN1_INTEGER(serial_number, asn1_serial_number))
>+      !BN_to_ASN1_INTEGER(serial_number, asn1_serial_number)) {
>     goto error;
>-
>-  if (!X509_set_version(x509, 2L))  // version 3
>+  }
>+  // Set version to X509.V3
>+  if (!X509_set_version(x509, 2L)) {
>     goto error;
>+  }
> 
>   // There are a lot of possible components for the name entries. In
>   // our P2P SSL mode however, the certificates are pre-exchanged
>@@ -90,15 +92,16 @@ static X509* MakeCertificate(EVP_PKEY* pkey, const SSLIdentityParams& params) {
>       !X509_NAME_add_entry_by_NID(name, NID_commonName, MBSTRING_UTF8,
>                                   (unsigned char*)params.common_name.c_str(),
>                                   -1, -1, 0) ||
>-      !X509_set_subject_name(x509, name) || !X509_set_issuer_name(x509, name))
>+      !X509_set_subject_name(x509, name) || !X509_set_issuer_name(x509, name)) {
>     goto error;
>-
>+  }
>   if (!X509_time_adj(X509_get_notBefore(x509), params.not_before, &epoch_off) ||
>-      !X509_time_adj(X509_get_notAfter(x509), params.not_after, &epoch_off))
>+      !X509_time_adj(X509_get_notAfter(x509), params.not_after, &epoch_off)) {
>     goto error;
>-
>-  if (!X509_sign(x509, pkey, EVP_sha256()))
>+  }
>+  if (!X509_sign(x509, pkey, EVP_sha256())) {
>     goto error;
>+  }
> 
>   BN_free(serial_number);
>   X509_NAME_free(name);
>@@ -112,28 +115,14 @@ error:
>   return nullptr;
> }
> 
>-#if !defined(NDEBUG)
>-// Print a certificate to the log, for debugging.
>-static void PrintCert(X509* x509) {
>-  BIO* temp_memory_bio = BIO_new(BIO_s_mem());
>-  if (!temp_memory_bio) {
>-    RTC_DLOG_F(LS_ERROR) << "Failed to allocate temporary memory bio";
>-    return;
>-  }
>-  X509_print_ex(temp_memory_bio, x509, XN_FLAG_SEP_CPLUS_SPC, 0);
>-  BIO_write(temp_memory_bio, "\0", 1);
>-  char* buffer;
>-  BIO_get_mem_data(temp_memory_bio, &buffer);
>-  RTC_DLOG(LS_VERBOSE) << buffer;
>-  BIO_free(temp_memory_bio);
>-}
>-#endif
>+}  // namespace
> 
> OpenSSLCertificate::OpenSSLCertificate(X509* x509) : x509_(x509) {
>-  AddReference();
>+  RTC_DCHECK(x509_ != nullptr);
>+  X509_up_ref(x509_);
> }
> 
>-OpenSSLCertificate* OpenSSLCertificate::Generate(
>+std::unique_ptr<OpenSSLCertificate> OpenSSLCertificate::Generate(
>     OpenSSLKeyPair* key_pair,
>     const SSLIdentityParams& params) {
>   SSLIdentityParams actual_params(params);
>@@ -149,25 +138,27 @@ OpenSSLCertificate* OpenSSLCertificate::Generate(
> #if !defined(NDEBUG)
>   PrintCert(x509);
> #endif
>-  OpenSSLCertificate* ret = new OpenSSLCertificate(x509);
>+  auto ret = absl::make_unique<OpenSSLCertificate>(x509);
>   X509_free(x509);
>   return ret;
> }
> 
>-OpenSSLCertificate* OpenSSLCertificate::FromPEMString(
>+std::unique_ptr<OpenSSLCertificate> OpenSSLCertificate::FromPEMString(
>     const std::string& pem_string) {
>   BIO* bio = BIO_new_mem_buf(const_cast<char*>(pem_string.c_str()), -1);
>-  if (!bio)
>+  if (!bio) {
>     return nullptr;
>+  }
>+
>   BIO_set_mem_eof_return(bio, 0);
>   X509* x509 =
>       PEM_read_bio_X509(bio, nullptr, nullptr, const_cast<char*>("\0"));
>   BIO_free(bio);  // Frees the BIO, but not the pointed-to string.
> 
>-  if (!x509)
>+  if (!x509) {
>     return nullptr;
>-
>-  OpenSSLCertificate* ret = new OpenSSLCertificate(x509);
>+  }
>+  auto ret = absl::make_unique<OpenSSLCertificate>(x509);
>   X509_free(x509);
>   return ret;
> }
>@@ -229,19 +220,16 @@ bool OpenSSLCertificate::ComputeDigest(const X509* x509,
>                                        unsigned char* digest,
>                                        size_t size,
>                                        size_t* length) {
>-  const EVP_MD* md;
>-  unsigned int n;
>-
>-  if (!OpenSSLDigest::GetDigestEVP(algorithm, &md))
>+  const EVP_MD* md = nullptr;
>+  unsigned int n = 0;
>+  if (!OpenSSLDigest::GetDigestEVP(algorithm, &md)) {
>     return false;
>-
>-  if (size < static_cast<size_t>(EVP_MD_size(md)))
>+  }
>+  if (size < static_cast<size_t>(EVP_MD_size(md))) {
>     return false;
>-
>+  }
>   X509_digest(x509, md, digest, &n);
>-
>   *length = n;
>-
>   return true;
> }
> 
>@@ -249,18 +237,18 @@ OpenSSLCertificate::~OpenSSLCertificate() {
>   X509_free(x509_);
> }
> 
>-OpenSSLCertificate* OpenSSLCertificate::GetReference() const {
>-  return new OpenSSLCertificate(x509_);
>+std::unique_ptr<SSLCertificate> OpenSSLCertificate::Clone() const {
>+  return absl::make_unique<OpenSSLCertificate>(x509_);
> }
> 
> std::string OpenSSLCertificate::ToPEMString() const {
>   BIO* bio = BIO_new(BIO_s_mem());
>   if (!bio) {
>-    RTC_FATAL() << "unreachable code";
>+    RTC_FATAL() << "Unreachable code.";
>   }
>   if (!PEM_write_bio_X509(bio, x509_)) {
>     BIO_free(bio);
>-    RTC_FATAL() << "unreachable code";
>+    RTC_FATAL() << "Unreachable code.";
>   }
>   BIO_write(bio, "\0", 1);
>   char* buffer;
>@@ -273,27 +261,21 @@ std::string OpenSSLCertificate::ToPEMString() const {
> void OpenSSLCertificate::ToDER(Buffer* der_buffer) const {
>   // In case of failure, make sure to leave the buffer empty.
>   der_buffer->SetSize(0);
>-
>   // Calculates the DER representation of the certificate, from scratch.
>   BIO* bio = BIO_new(BIO_s_mem());
>   if (!bio) {
>-    RTC_FATAL() << "unreachable code";
>+    RTC_FATAL() << "Unreachable code.";
>   }
>   if (!i2d_X509_bio(bio, x509_)) {
>     BIO_free(bio);
>-    RTC_FATAL() << "unreachable code";
>+    RTC_FATAL() << "Unreachable code.";
>   }
>-  char* data;
>+  char* data = nullptr;
>   size_t length = BIO_get_mem_data(bio, &data);
>   der_buffer->SetData(data, length);
>   BIO_free(bio);
> }
> 
>-void OpenSSLCertificate::AddReference() const {
>-  RTC_DCHECK(x509_ != nullptr);
>-  X509_up_ref(x509_);
>-}
>-
> bool OpenSSLCertificate::operator==(const OpenSSLCertificate& other) const {
>   return X509_cmp(x509_, other.x509_) == 0;
> }
>@@ -302,11 +284,9 @@ bool OpenSSLCertificate::operator!=(const OpenSSLCertificate& other) const {
>   return !(*this == other);
> }
> 
>-// Documented in sslidentity.h.
> int64_t OpenSSLCertificate::CertificateExpirationTime() const {
>   ASN1_TIME* expire_time = X509_get_notAfter(x509_);
>   bool long_format;
>-
>   if (expire_time->type == V_ASN1_UTCTIME) {
>     long_format = false;
>   } else if (expire_time->type == V_ASN1_GENERALIZEDTIME) {
>@@ -314,7 +294,6 @@ int64_t OpenSSLCertificate::CertificateExpirationTime() const {
>   } else {
>     return -1;
>   }
>-
>   return ASN1TimeToSec(expire_time->data, expire_time->length, long_format);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.h
>index c730ffd0dc9e769c6bd0f97d4ee1e7bdc5429ff2..088725c0d755653d3b230365c56331cd042c069f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.h
>@@ -11,38 +11,38 @@
> #ifndef RTC_BASE_OPENSSLCERTIFICATE_H_
> #define RTC_BASE_OPENSSLCERTIFICATE_H_
> 
>-#include <openssl/evp.h>
>-#include <openssl/x509.h>
>+#include <openssl/ossl_typ.h>
> 
>-#include <memory>
>+#include <stddef.h>
>+#include <stdint.h>
> #include <string>
> 
>-#include "rtc_base/checks.h"
>+#include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/sslcertificate.h"
> #include "rtc_base/sslidentity.h"
> 
>-typedef struct ssl_ctx_st SSL_CTX;
>-
> namespace rtc {
> 
> class OpenSSLKeyPair;
> 
> // OpenSSLCertificate encapsulates an OpenSSL X509* certificate object,
> // which is also reference counted inside the OpenSSL library.
>-class OpenSSLCertificate : public SSLCertificate {
>+class OpenSSLCertificate final : public SSLCertificate {
>  public:
>   // X509 object has its reference count incremented. So the caller and
>   // OpenSSLCertificate share ownership.
>   explicit OpenSSLCertificate(X509* x509);
> 
>-  static OpenSSLCertificate* Generate(OpenSSLKeyPair* key_pair,
>-                                      const SSLIdentityParams& params);
>-  static OpenSSLCertificate* FromPEMString(const std::string& pem_string);
>+  static std::unique_ptr<OpenSSLCertificate> Generate(
>+      OpenSSLKeyPair* key_pair,
>+      const SSLIdentityParams& params);
>+  static std::unique_ptr<OpenSSLCertificate> FromPEMString(
>+      const std::string& pem_string);
> 
>   ~OpenSSLCertificate() override;
> 
>-  OpenSSLCertificate* GetReference() const override;
>+  std::unique_ptr<SSLCertificate> Clone() const override;
> 
>   X509* x509() const { return x509_; }
> 
>@@ -69,8 +69,6 @@ class OpenSSLCertificate : public SSLCertificate {
>   int64_t CertificateExpirationTime() const override;
> 
>  private:
>-  void AddReference() const;
>-
>   X509* x509_;  // NOT OWNED
>   RTC_DISALLOW_COPY_AND_ASSIGN(OpenSSLCertificate);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.cc
>index 9b644c460b5a5834b839fe68f1f74be90c7c5c93..da90b65f8221ca96c6a03339f2887d70b82b5a64 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.cc
>@@ -10,7 +10,7 @@
> 
> #include "rtc_base/openssldigest.h"
> 
>-#include "rtc_base/checks.h"
>+#include "rtc_base/checks.h"  // RTC_DCHECK, RTC_CHECK
> #include "rtc_base/openssl.h"
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.h
>index c4cd1e0314ed4e0abedcdbc0676c89006a1b6eb0..82dc9a96cd84bbaf0c7cdaf271ac22470ac58290 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.h
>@@ -11,14 +11,16 @@
> #ifndef RTC_BASE_OPENSSLDIGEST_H_
> #define RTC_BASE_OPENSSLDIGEST_H_
> 
>-#include <openssl/evp.h>
>+#include <openssl/base.h>
>+#include <stddef.h>
>+#include <string>
> 
> #include "rtc_base/messagedigest.h"
> 
> namespace rtc {
> 
> // An implementation of the digest class that uses OpenSSL.
>-class OpenSSLDigest : public MessageDigest {
>+class OpenSSLDigest final : public MessageDigest {
>  public:
>   // Creates an OpenSSLDigest with |algorithm| as the hash algorithm.
>   explicit OpenSSLDigest(const std::string& algorithm);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>index a8c6919779bede4790f9fdf0f115b572ced6e685..9850c855ea1679a413792690e65f99862c4d449b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>@@ -21,18 +21,17 @@
> 
> #include <openssl/bio.h>
> #include <openssl/bn.h>
>-#include <openssl/crypto.h>
> #include <openssl/err.h>
> #include <openssl/pem.h>
> #include <openssl/rsa.h>
> 
>+#include <stdint.h>
>+
> #include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/openssl.h"
>-#include "rtc_base/openssldigest.h"
> #include "rtc_base/opensslutility.h"
> 
> namespace rtc {
>@@ -316,7 +315,7 @@ const SSLCertChain& OpenSSLIdentity::cert_chain() const {
> 
> OpenSSLIdentity* OpenSSLIdentity::GetReference() const {
>   return new OpenSSLIdentity(absl::WrapUnique(key_pair_->GetReference()),
>-                             absl::WrapUnique(cert_chain_->Copy()));
>+                             cert_chain_->Clone());
> }
> 
> bool OpenSSLIdentity::ConfigureIdentity(SSL_CTX* ctx) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.h
>index 34044276ca7659bdd785545bd9bef11b8caf9c92..fcf7debf9072feb135f7472f24cd6b0c0cacec1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.h
>@@ -11,24 +11,23 @@
> #ifndef RTC_BASE_OPENSSLIDENTITY_H_
> #define RTC_BASE_OPENSSLIDENTITY_H_
> 
>-#include <openssl/evp.h>
>-#include <openssl/x509.h>
>+#include <openssl/ossl_typ.h>
> 
>+#include <ctime>
> #include <memory>
> #include <string>
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/opensslcertificate.h"
>+#include "rtc_base/sslcertificate.h"
> #include "rtc_base/sslidentity.h"
> 
>-typedef struct ssl_ctx_st SSL_CTX;
>-
> namespace rtc {
> 
> // OpenSSLKeyPair encapsulates an OpenSSL EVP_PKEY* keypair object,
> // which is reference counted inside the OpenSSL library.
>-class OpenSSLKeyPair {
>+class OpenSSLKeyPair final {
>  public:
>   explicit OpenSSLKeyPair(EVP_PKEY* pkey) : pkey_(pkey) {
>     RTC_DCHECK(pkey_ != nullptr);
>@@ -59,7 +58,7 @@ class OpenSSLKeyPair {
> 
> // Holds a keypair and certificate together, and a method to generate
> // them consistently.
>-class OpenSSLIdentity : public SSLIdentity {
>+class OpenSSLIdentity final : public SSLIdentity {
>  public:
>   static OpenSSLIdentity* GenerateWithExpiration(const std::string& common_name,
>                                                  const KeyParams& key_params,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.cc
>index fd54a082a7d738583e884c888eee5082f9434f8e..727cb84aae95da375c2474a92fbcc8eb8d91d8c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.cc
>@@ -22,6 +22,7 @@
> #endif
> 
> #include <memory>
>+#include <utility>
> #include <vector>
> 
> #include "rtc_base/checks.h"
>@@ -31,6 +32,7 @@
> #include "rtc_base/openssladapter.h"
> #include "rtc_base/openssldigest.h"
> #include "rtc_base/opensslidentity.h"
>+#include "rtc_base/sslcertificate.h"
> #include "rtc_base/stream.h"
> #include "rtc_base/stringutils.h"
> #include "rtc_base/thread.h"
>@@ -180,8 +182,9 @@ static BIO_METHOD* BIO_stream_method() {
> 
> static BIO* BIO_new_stream(StreamInterface* stream) {
>   BIO* ret = BIO_new(BIO_stream_method());
>-  if (ret == nullptr)
>+  if (ret == nullptr) {
>     return nullptr;
>+  }
>   BIO_set_data(ret, stream);
>   return ret;
> }
>@@ -196,14 +199,16 @@ static int stream_new(BIO* b) {
> }
> 
> static int stream_free(BIO* b) {
>-  if (b == nullptr)
>+  if (b == nullptr) {
>     return 0;
>+  }
>   return 1;
> }
> 
> static int stream_read(BIO* b, char* out, int outl) {
>-  if (!out)
>+  if (!out) {
>     return -1;
>+  }
>   StreamInterface* stream = static_cast<StreamInterface*>(BIO_get_data(b));
>   BIO_clear_retry_flags(b);
>   size_t read;
>@@ -218,8 +223,9 @@ static int stream_read(BIO* b, char* out, int outl) {
> }
> 
> static int stream_write(BIO* b, const char* in, int inl) {
>-  if (!in)
>+  if (!in) {
>     return -1;
>+  }
>   StreamInterface* stream = static_cast<StreamInterface*>(BIO_get_data(b));
>   BIO_clear_retry_flags(b);
>   size_t written;
>@@ -296,7 +302,7 @@ bool OpenSSLStreamAdapter::SetPeerCertificateDigest(
>     size_t digest_len,
>     SSLPeerCertificateDigestError* error) {
>   RTC_DCHECK(!peer_certificate_verified_);
>-  RTC_DCHECK(!has_peer_certificate_digest());
>+  RTC_DCHECK(!HasPeerCertificateDigest());
>   size_t expected_len;
>   if (error) {
>     *error = SSLPeerCertificateDigestError::NONE;
>@@ -362,8 +368,9 @@ std::string OpenSSLStreamAdapter::SslCipherSuiteToName(int cipher_suite) {
> }
> 
> bool OpenSSLStreamAdapter::GetSslCipherSuite(int* cipher_suite) {
>-  if (state_ != SSL_CONNECTED)
>+  if (state_ != SSL_CONNECTED) {
>     return false;
>+  }
> 
>   const SSL_CIPHER* current_cipher = SSL_get_current_cipher(ssl_);
>   if (current_cipher == nullptr) {
>@@ -380,17 +387,19 @@ int OpenSSLStreamAdapter::GetSslVersion() const {
> 
>   int ssl_version = SSL_version(ssl_);
>   if (ssl_mode_ == SSL_MODE_DTLS) {
>-    if (ssl_version == DTLS1_VERSION)
>+    if (ssl_version == DTLS1_VERSION) {
>       return SSL_PROTOCOL_DTLS_10;
>-    else if (ssl_version == DTLS1_2_VERSION)
>+    } else if (ssl_version == DTLS1_2_VERSION) {
>       return SSL_PROTOCOL_DTLS_12;
>+    }
>   } else {
>-    if (ssl_version == TLS1_VERSION)
>+    if (ssl_version == TLS1_VERSION) {
>       return SSL_PROTOCOL_TLS_10;
>-    else if (ssl_version == TLS1_1_VERSION)
>+    } else if (ssl_version == TLS1_1_VERSION) {
>       return SSL_PROTOCOL_TLS_11;
>-    else if (ssl_version == TLS1_2_VERSION)
>+    } else if (ssl_version == TLS1_2_VERSION) {
>       return SSL_PROTOCOL_TLS_12;
>+    }
>   }
> 
>   return -1;
>@@ -403,15 +412,11 @@ bool OpenSSLStreamAdapter::ExportKeyingMaterial(const std::string& label,
>                                                 bool use_context,
>                                                 uint8_t* result,
>                                                 size_t result_len) {
>-  int i;
>-
>-  i = SSL_export_keying_material(ssl_, result, result_len, label.c_str(),
>+  if (SSL_export_keying_material(ssl_, result, result_len, label.c_str(),
>                                  label.length(), const_cast<uint8_t*>(context),
>-                                 context_len, use_context);
>-
>-  if (i != 1)
>+                                 context_len, use_context) != 1) {
>     return false;
>-
>+  }
>   return true;
> }
> 
>@@ -419,8 +424,9 @@ bool OpenSSLStreamAdapter::SetDtlsSrtpCryptoSuites(
>     const std::vector<int>& ciphers) {
>   std::string internal_ciphers;
> 
>-  if (state_ != SSL_NONE)
>+  if (state_ != SSL_NONE) {
>     return false;
>+  }
> 
>   for (std::vector<int>::const_iterator cipher = ciphers.begin();
>        cipher != ciphers.end(); ++cipher) {
>@@ -429,8 +435,9 @@ bool OpenSSLStreamAdapter::SetDtlsSrtpCryptoSuites(
>          ++entry) {
>       if (*cipher == entry->id) {
>         found = true;
>-        if (!internal_ciphers.empty())
>+        if (!internal_ciphers.empty()) {
>           internal_ciphers += ":";
>+        }
>         internal_ciphers += entry->internal_name;
>         break;
>       }
>@@ -442,8 +449,9 @@ bool OpenSSLStreamAdapter::SetDtlsSrtpCryptoSuites(
>     }
>   }
> 
>-  if (internal_ciphers.empty())
>+  if (internal_ciphers.empty()) {
>     return false;
>+  }
> 
>   srtp_ciphers_ = internal_ciphers;
>   return true;
>@@ -451,14 +459,16 @@ bool OpenSSLStreamAdapter::SetDtlsSrtpCryptoSuites(
> 
> bool OpenSSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) {
>   RTC_DCHECK(state_ == SSL_CONNECTED);
>-  if (state_ != SSL_CONNECTED)
>+  if (state_ != SSL_CONNECTED) {
>     return false;
>+  }
> 
>   const SRTP_PROTECTION_PROFILE* srtp_profile =
>       SSL_get_selected_srtp_profile(ssl_);
> 
>-  if (!srtp_profile)
>+  if (!srtp_profile) {
>     return false;
>+  }
> 
>   *crypto_suite = srtp_profile->id;
>   RTC_DCHECK(!SrtpCryptoSuiteToName(*crypto_suite).empty());
>@@ -470,8 +480,8 @@ bool OpenSSLStreamAdapter::IsTlsConnected() {
> }
> 
> int OpenSSLStreamAdapter::StartSSL() {
>+  // Don't allow StartSSL to be called twice.
>   if (state_ != SSL_NONE) {
>-    // Don't allow StartSSL to be called twice.
>     return -1;
>   }
> 
>@@ -524,7 +534,7 @@ StreamResult OpenSSLStreamAdapter::Write(const void* data,
>       return SR_BLOCK;
> 
>     case SSL_CONNECTED:
>-      if (waiting_to_verify_peer_certificate()) {
>+      if (WaitingToVerifyPeerCertificate()) {
>         return SR_BLOCK;
>       }
>       break;
>@@ -532,15 +542,17 @@ StreamResult OpenSSLStreamAdapter::Write(const void* data,
>     case SSL_ERROR:
>     case SSL_CLOSED:
>     default:
>-      if (error)
>+      if (error) {
>         *error = ssl_error_code_;
>+      }
>       return SR_ERROR;
>   }
> 
>   // OpenSSL will return an error if we try to write zero bytes
>   if (data_len == 0) {
>-    if (written)
>+    if (written) {
>       *written = 0;
>+    }
>     return SR_SUCCESS;
>   }
> 
>@@ -567,8 +579,9 @@ StreamResult OpenSSLStreamAdapter::Write(const void* data,
>     case SSL_ERROR_ZERO_RETURN:
>     default:
>       Error("SSL_write", (ssl_error ? ssl_error : -1), 0, false);
>-      if (error)
>+      if (error) {
>         *error = ssl_error_code_;
>+      }
>       return SR_ERROR;
>   }
>   // not reached
>@@ -583,31 +596,29 @@ StreamResult OpenSSLStreamAdapter::Read(void* data,
>     case SSL_NONE:
>       // pass-through in clear text
>       return StreamAdapterInterface::Read(data, data_len, read, error);
>-
>     case SSL_WAIT:
>     case SSL_CONNECTING:
>       return SR_BLOCK;
>-
>     case SSL_CONNECTED:
>-      if (waiting_to_verify_peer_certificate()) {
>+      if (WaitingToVerifyPeerCertificate()) {
>         return SR_BLOCK;
>       }
>       break;
>-
>     case SSL_CLOSED:
>       return SR_EOS;
>-
>     case SSL_ERROR:
>     default:
>-      if (error)
>+      if (error) {
>         *error = ssl_error_code_;
>+      }
>       return SR_ERROR;
>   }
> 
>   // Don't trust OpenSSL with zero byte reads
>   if (data_len == 0) {
>-    if (read)
>+    if (read) {
>       *read = 0;
>+    }
>     return SR_SUCCESS;
>   }
> 
>@@ -620,8 +631,9 @@ StreamResult OpenSSLStreamAdapter::Read(void* data,
>       RTC_LOG(LS_VERBOSE) << " -- success";
>       RTC_DCHECK_GT(code, 0);
>       RTC_DCHECK_LE(code, data_len);
>-      if (read)
>+      if (read) {
>         *read = code;
>+      }
> 
>       if (ssl_mode_ == SSL_MODE_DTLS) {
>         // Enforce atomic reads -- this is a short read
>@@ -630,8 +642,9 @@ StreamResult OpenSSLStreamAdapter::Read(void* data,
>         if (pending) {
>           RTC_LOG(LS_INFO) << " -- short DTLS read. flushing";
>           FlushInput(pending);
>-          if (error)
>+          if (error) {
>             *error = SSE_MSG_TRUNC;
>+          }
>           return SR_ERROR;
>         }
>       }
>@@ -650,8 +663,9 @@ StreamResult OpenSSLStreamAdapter::Read(void* data,
>       break;
>     default:
>       Error("SSL_read", (ssl_error ? ssl_error : -1), 0, false);
>-      if (error)
>+      if (error) {
>         *error = ssl_error_code_;
>+      }
>       return SR_ERROR;
>   }
>   // not reached
>@@ -694,13 +708,13 @@ StreamState OpenSSLStreamAdapter::GetState() const {
>     case SSL_CONNECTING:
>       return SS_OPENING;
>     case SSL_CONNECTED:
>-      if (waiting_to_verify_peer_certificate()) {
>+      if (WaitingToVerifyPeerCertificate()) {
>         return SS_OPENING;
>       }
>       return SS_OPEN;
>     default:
>       return SS_CLOSED;
>-  };
>+  }
>   // not reached
> }
> 
>@@ -710,6 +724,7 @@ void OpenSSLStreamAdapter::OnEvent(StreamInterface* stream,
>   int events_to_signal = 0;
>   int signal_error = 0;
>   RTC_DCHECK(stream == this->stream());
>+
>   if ((events & SE_OPEN)) {
>     RTC_LOG(LS_VERBOSE) << "OpenSSLStreamAdapter::OnEvent SE_OPEN";
>     if (state_ != SSL_WAIT) {
>@@ -723,6 +738,7 @@ void OpenSSLStreamAdapter::OnEvent(StreamInterface* stream,
>       }
>     }
>   }
>+
>   if ((events & (SE_READ | SE_WRITE))) {
>     RTC_LOG(LS_VERBOSE) << "OpenSSLStreamAdapter::OnEvent"
>                         << ((events & SE_READ) ? " SE_READ" : "")
>@@ -747,6 +763,7 @@ void OpenSSLStreamAdapter::OnEvent(StreamInterface* stream,
>       }
>     }
>   }
>+
>   if ((events & SE_CLOSE)) {
>     RTC_LOG(LS_VERBOSE) << "OpenSSLStreamAdapter::OnEvent(SE_CLOSE, " << err
>                         << ")";
>@@ -756,8 +773,10 @@ void OpenSSLStreamAdapter::OnEvent(StreamInterface* stream,
>     RTC_DCHECK(signal_error == 0);
>     signal_error = err;
>   }
>-  if (events_to_signal)
>+
>+  if (events_to_signal) {
>     StreamAdapterInterface::OnEvent(stream, events_to_signal, signal_error);
>+  }
> }
> 
> int OpenSSLStreamAdapter::BeginSSL() {
>@@ -770,12 +789,14 @@ int OpenSSLStreamAdapter::BeginSSL() {
>   // First set up the context.
>   RTC_DCHECK(ssl_ctx_ == nullptr);
>   ssl_ctx_ = SetupSSLContext();
>-  if (!ssl_ctx_)
>+  if (!ssl_ctx_) {
>     return -1;
>+  }
> 
>   bio = BIO_new_stream(static_cast<StreamInterface*>(stream()));
>-  if (!bio)
>+  if (!bio) {
>     return -1;
>+  }
> 
>   ssl_ = SSL_new(ssl_ctx_);
>   if (!ssl_) {
>@@ -805,8 +826,9 @@ int OpenSSLStreamAdapter::BeginSSL() {
>   // commonly supported. BoringSSL doesn't need explicit configuration and has
>   // a reasonable default set.
>   EC_KEY* ecdh = EC_KEY_new_by_curve_name(NID_X9_62_prime256v1);
>-  if (ecdh == nullptr)
>+  if (ecdh == nullptr) {
>     return -1;
>+  }
>   SSL_set_options(ssl_, SSL_OP_SINGLE_ECDH_USE);
>   SSL_set_tmp_ecdh(ssl_, ecdh);
>   EC_KEY_free(ecdh);
>@@ -830,10 +852,10 @@ int OpenSSLStreamAdapter::ContinueSSL() {
>       RTC_LOG(LS_VERBOSE) << " -- success";
>       // By this point, OpenSSL should have given us a certificate, or errored
>       // out if one was missing.
>-      RTC_DCHECK(peer_cert_chain_ || !client_auth_enabled());
>+      RTC_DCHECK(peer_cert_chain_ || !GetClientAuthEnabled());
> 
>       state_ = SSL_CONNECTED;
>-      if (!waiting_to_verify_peer_certificate()) {
>+      if (!WaitingToVerifyPeerCertificate()) {
>         // We have everything we need to start the connection, so signal
>         // SE_OPEN. If we need a client certificate fingerprint and don't have
>         // it yet, we'll instead signal SE_OPEN in SetPeerCertificateDigest.
>@@ -886,8 +908,9 @@ void OpenSSLStreamAdapter::Error(const char* context,
>   state_ = SSL_ERROR;
>   ssl_error_code_ = err;
>   Cleanup(alert);
>-  if (signal)
>+  if (signal) {
>     StreamAdapterInterface::OnEvent(stream(), SE_CLOSE, err);
>+  }
> }
> 
> void OpenSSLStreamAdapter::Cleanup(uint8_t alert) {
>@@ -990,8 +1013,9 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
>   ctx = SSL_CTX_new(method);
> #endif  // OPENSSL_IS_BORINGSSL
> 
>-  if (ctx == nullptr)
>+  if (ctx == nullptr) {
>     return nullptr;
>+  }
> 
> #ifdef OPENSSL_IS_BORINGSSL
>   SSL_CTX_set_min_proto_version(
>@@ -1026,7 +1050,7 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
> #endif
> 
>   int mode = SSL_VERIFY_PEER;
>-  if (client_auth_enabled()) {
>+  if (GetClientAuthEnabled()) {
>     // Require a certificate from the client.
>     // Note: Normally this is always true in production, but it may be disabled
>     // for testing purposes (e.g. SSLAdapter unit tests).
>@@ -1058,7 +1082,7 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
> }
> 
> bool OpenSSLStreamAdapter::VerifyPeerCertificate() {
>-  if (!has_peer_certificate_digest() || !peer_cert_chain_ ||
>+  if (!HasPeerCertificateDigest() || !peer_cert_chain_ ||
>       !peer_cert_chain_->GetSize()) {
>     RTC_LOG(LS_WARNING) << "Missing digest or peer certificate.";
>     return false;
>@@ -1091,7 +1115,7 @@ bool OpenSSLStreamAdapter::VerifyPeerCertificate() {
> 
> std::unique_ptr<SSLCertChain> OpenSSLStreamAdapter::GetPeerSSLCertChain()
>     const {
>-  return peer_cert_chain_ ? peer_cert_chain_->UniqueCopy() : nullptr;
>+  return peer_cert_chain_ ? peer_cert_chain_->Clone() : nullptr;
> }
> 
> int OpenSSLStreamAdapter::SSLVerifyCallback(X509_STORE_CTX* store, void* arg) {
>@@ -1176,15 +1200,17 @@ static const cipher_list OK_ECDSA_ciphers[] = {
> bool OpenSSLStreamAdapter::IsAcceptableCipher(int cipher, KeyType key_type) {
>   if (key_type == KT_RSA) {
>     for (const cipher_list& c : OK_RSA_ciphers) {
>-      if (cipher == c.cipher)
>+      if (cipher == c.cipher) {
>         return true;
>+      }
>     }
>   }
> 
>   if (key_type == KT_ECDSA) {
>     for (const cipher_list& c : OK_ECDSA_ciphers) {
>-      if (cipher == c.cipher)
>+      if (cipher == c.cipher) {
>         return true;
>+      }
>     }
>   }
> 
>@@ -1195,22 +1221,24 @@ bool OpenSSLStreamAdapter::IsAcceptableCipher(const std::string& cipher,
>                                               KeyType key_type) {
>   if (key_type == KT_RSA) {
>     for (const cipher_list& c : OK_RSA_ciphers) {
>-      if (cipher == c.cipher_str)
>+      if (cipher == c.cipher_str) {
>         return true;
>+      }
>     }
>   }
> 
>   if (key_type == KT_ECDSA) {
>     for (const cipher_list& c : OK_ECDSA_ciphers) {
>-      if (cipher == c.cipher_str)
>+      if (cipher == c.cipher_str) {
>         return true;
>+      }
>     }
>   }
> 
>   return false;
> }
> 
>-void OpenSSLStreamAdapter::enable_time_callback_for_testing() {
>+void OpenSSLStreamAdapter::EnableTimeCallbackForTesting() {
>   g_use_time_callback_for_testing = true;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.h
>index 61ffc3dbfe2d6538bd3031fc970e752cea4f43ff..e012d172a62a0b4595c905cc61281c4989756ef3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.h
>@@ -13,13 +13,18 @@
> 
> #include <openssl/ossl_typ.h>
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> #include <string>
> #include <vector>
> 
> #include "rtc_base/buffer.h"
>+#include "rtc_base/messagequeue.h"
> #include "rtc_base/opensslidentity.h"
>+#include "rtc_base/sslidentity.h"
> #include "rtc_base/sslstreamadapter.h"
>+#include "rtc_base/stream.h"
> 
> namespace rtc {
> 
>@@ -47,11 +52,11 @@ namespace rtc {
> 
> // Look in sslstreamadapter.h for documentation of the methods.
> 
>-class OpenSSLIdentity;
>+class SSLCertChain;
> 
> ///////////////////////////////////////////////////////////////////////////////
> 
>-class OpenSSLStreamAdapter : public SSLStreamAdapter {
>+class OpenSSLStreamAdapter final : public SSLStreamAdapter {
>  public:
>   explicit OpenSSLStreamAdapter(StreamInterface* stream);
>   ~OpenSSLStreamAdapter() override;
>@@ -115,7 +120,7 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter {
> 
>   // Use our timeutils.h source of timing in BoringSSL, allowing us to test
>   // using a fake clock.
>-  static void enable_time_callback_for_testing();
>+  static void EnableTimeCallbackForTesting();
> 
>  protected:
>   void OnEvent(StreamInterface* stream, int events, int err) override;
>@@ -170,11 +175,11 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter {
>   // SSL_CTX_set_cert_verify_callback.
>   static int SSLVerifyCallback(X509_STORE_CTX* store, void* arg);
> 
>-  bool waiting_to_verify_peer_certificate() const {
>-    return client_auth_enabled() && !peer_certificate_verified_;
>+  bool WaitingToVerifyPeerCertificate() const {
>+    return GetClientAuthEnabled() && !peer_certificate_verified_;
>   }
> 
>-  bool has_peer_certificate_digest() const {
>+  bool HasPeerCertificateDigest() const {
>     return !peer_certificate_digest_algorithm_.empty() &&
>            !peer_certificate_digest_value_.empty();
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.cc
>index 46f4547436e1158a7b80308043f8f677d25e19e8..a3f33478ebc7b05100b73cb304281a0246b9256b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.cc
>@@ -9,29 +9,21 @@
>  */
> 
> #include "rtc_base/opensslutility.h"
>-
>-#include <memory>
>-
>-#if defined(WEBRTC_POSIX)
>-#include <unistd.h>
>-#endif
>-
> #if defined(WEBRTC_WIN)
> // Must be included first before openssl headers.
> #include "rtc_base/win32.h"  // NOLINT
> #endif                       // WEBRTC_WIN
> 
>-#include <openssl/bio.h>
>-#include <openssl/crypto.h>
> #include <openssl/err.h>
> #include <openssl/x509.h>
> #include <openssl/x509v3.h>
>+#include "rtc_base/openssl.h"
>+
>+#include <stddef.h>
> 
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/openssl.h"
> #include "rtc_base/opensslcertificate.h"
> #ifndef WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
> #include "rtc_base/sslroots.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.h
>index 7cb38b5b52763a51ddb40258936a656bd6721f9a..77ed0b1e7cca1cb7968a681acbf065a9fa3c3298 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.h
>@@ -13,7 +13,6 @@
> 
> #include <openssl/ossl_typ.h>
> #include <string>
>-#include "rtc_base/sslcertificate.h"
> 
> namespace rtc {
> // The openssl namespace holds static helper methods. All methods related
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.cc
>deleted file mode 100644
>index 535859b4cc075364b4bc62d1f85ec33de51c0c14..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.cc
>+++ /dev/null
>@@ -1,181 +0,0 @@
>-/*
>- *  Copyright 2008 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/optionsfile.h"
>-
>-#include <ctype.h>
>-
>-#include "rtc_base/logging.h"
>-#include "rtc_base/stream.h"
>-#include "rtc_base/stringencode.h"
>-
>-namespace rtc {
>-
>-OptionsFile::OptionsFile(const std::string& path) : path_(path) {}
>-
>-OptionsFile::~OptionsFile() = default;
>-
>-bool OptionsFile::Load() {
>-  options_.clear();
>-  // Open file.
>-  FileStream stream;
>-  int err;
>-  if (!stream.Open(path_, "r", &err)) {
>-    RTC_LOG_F(LS_WARNING) << "Could not open file, err=" << err;
>-    // We do not consider this an error because we expect there to be no file
>-    // until the user saves a setting.
>-    return true;
>-  }
>-  // Read in all its data.
>-  std::string line;
>-  StreamResult res;
>-  for (;;) {
>-    res = stream.ReadLine(&line);
>-    if (res != SR_SUCCESS) {
>-      break;
>-    }
>-    size_t equals_pos = line.find('=');
>-    if (equals_pos == std::string::npos) {
>-      // We do not consider this an error. Instead we ignore the line and
>-      // keep going.
>-      RTC_LOG_F(LS_WARNING) << "Ignoring malformed line in " << path_;
>-      continue;
>-    }
>-    std::string key(line, 0, equals_pos);
>-    std::string value(line, equals_pos + 1, line.length() - (equals_pos + 1));
>-    options_[key] = value;
>-  }
>-  if (res != SR_EOS) {
>-    RTC_LOG_F(LS_ERROR) << "Error when reading from file";
>-    return false;
>-  } else {
>-    return true;
>-  }
>-}
>-
>-bool OptionsFile::Save() {
>-  // Open file.
>-  FileStream stream;
>-  int err;
>-  if (!stream.Open(path_, "w", &err)) {
>-    RTC_LOG_F(LS_ERROR) << "Could not open file, err=" << err;
>-    return false;
>-  }
>-  // Write out all the data.
>-  StreamResult res = SR_SUCCESS;
>-  size_t written;
>-  int error;
>-  for (OptionsMap::const_iterator i = options_.begin(); i != options_.end();
>-       ++i) {
>-    res =
>-        stream.WriteAll(i->first.c_str(), i->first.length(), &written, &error);
>-    if (res != SR_SUCCESS) {
>-      break;
>-    }
>-    res = stream.WriteAll("=", 1, &written, &error);
>-    if (res != SR_SUCCESS) {
>-      break;
>-    }
>-    res = stream.WriteAll(i->second.c_str(), i->second.length(), &written,
>-                          &error);
>-    if (res != SR_SUCCESS) {
>-      break;
>-    }
>-    res = stream.WriteAll("\n", 1, &written, &error);
>-    if (res != SR_SUCCESS) {
>-      break;
>-    }
>-  }
>-  if (res != SR_SUCCESS) {
>-    RTC_LOG_F(LS_ERROR) << "Unable to write to file";
>-    return false;
>-  } else {
>-    return true;
>-  }
>-}
>-
>-bool OptionsFile::IsLegalName(const std::string& name) {
>-  for (size_t pos = 0; pos < name.length(); ++pos) {
>-    if (name[pos] == '\n' || name[pos] == '\\' || name[pos] == '=') {
>-      // Illegal character.
>-      RTC_LOG(LS_WARNING) << "Ignoring operation for illegal option " << name;
>-      return false;
>-    }
>-  }
>-  return true;
>-}
>-
>-bool OptionsFile::IsLegalValue(const std::string& value) {
>-  for (size_t pos = 0; pos < value.length(); ++pos) {
>-    if (value[pos] == '\n' || value[pos] == '\\') {
>-      // Illegal character.
>-      RTC_LOG(LS_WARNING) << "Ignoring operation for illegal value " << value;
>-      return false;
>-    }
>-  }
>-  return true;
>-}
>-
>-bool OptionsFile::GetStringValue(const std::string& option,
>-                                 std::string* out_val) const {
>-  RTC_LOG(LS_VERBOSE) << "OptionsFile::GetStringValue " << option;
>-  if (!IsLegalName(option)) {
>-    return false;
>-  }
>-  OptionsMap::const_iterator i = options_.find(option);
>-  if (i == options_.end()) {
>-    return false;
>-  }
>-  *out_val = i->second;
>-  return true;
>-}
>-
>-bool OptionsFile::GetIntValue(const std::string& option, int* out_val) const {
>-  RTC_LOG(LS_VERBOSE) << "OptionsFile::GetIntValue " << option;
>-  if (!IsLegalName(option)) {
>-    return false;
>-  }
>-  OptionsMap::const_iterator i = options_.find(option);
>-  if (i == options_.end()) {
>-    return false;
>-  }
>-  return FromString(i->second, out_val);
>-}
>-
>-bool OptionsFile::SetStringValue(const std::string& option,
>-                                 const std::string& value) {
>-  RTC_LOG(LS_VERBOSE) << "OptionsFile::SetStringValue " << option << ":"
>-                      << value;
>-  if (!IsLegalName(option) || !IsLegalValue(value)) {
>-    return false;
>-  }
>-  options_[option] = value;
>-  return true;
>-}
>-
>-bool OptionsFile::SetIntValue(const std::string& option, int value) {
>-  RTC_LOG(LS_VERBOSE) << "OptionsFile::SetIntValue " << option << ":" << value;
>-  if (!IsLegalName(option)) {
>-    return false;
>-  }
>-  options_[option] = ToString(value);
>-  return true;
>-}
>-
>-bool OptionsFile::RemoveValue(const std::string& option) {
>-  RTC_LOG(LS_VERBOSE) << "OptionsFile::RemoveValue " << option;
>-  if (!IsLegalName(option)) {
>-    return false;
>-  }
>-  options_.erase(option);
>-  return true;
>-}
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.h
>deleted file mode 100644
>index 55660ff2598be3decb2e56962e55e853be15e2b2..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.h
>+++ /dev/null
>@@ -1,50 +0,0 @@
>-/*
>- *  Copyright 2008 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef RTC_BASE_OPTIONSFILE_H_
>-#define RTC_BASE_OPTIONSFILE_H_
>-
>-#include <map>
>-#include <string>
>-
>-namespace rtc {
>-
>-// Implements storage of simple options in a text file on disk. This is
>-// cross-platform, but it is intended mostly for Linux where there is no
>-// first-class options storage system.
>-class OptionsFile {
>- public:
>-  OptionsFile(const std::string& path);
>-  ~OptionsFile();
>-
>-  // Loads the file from disk, overwriting the in-memory values.
>-  bool Load();
>-  // Saves the contents in memory, overwriting the on-disk values.
>-  bool Save();
>-
>-  bool GetStringValue(const std::string& option, std::string* out_val) const;
>-  bool GetIntValue(const std::string& option, int* out_val) const;
>-  bool SetStringValue(const std::string& option, const std::string& val);
>-  bool SetIntValue(const std::string& option, int val);
>-  bool RemoveValue(const std::string& option);
>-
>- private:
>-  typedef std::map<std::string, std::string> OptionsMap;
>-
>-  static bool IsLegalName(const std::string& name);
>-  static bool IsLegalValue(const std::string& value);
>-
>-  std::string path_;
>-  OptionsMap options_;
>-};
>-
>-}  // namespace rtc
>-
>-#endif  // RTC_BASE_OPTIONSFILE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile_unittest.cc
>deleted file mode 100644
>index fc5bc823ff4158f72db2c506e26f7b89d01d457a..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile_unittest.cc
>+++ /dev/null
>@@ -1,178 +0,0 @@
>-/*
>- *  Copyright 2008 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/optionsfile.h"
>-#include "test/testsupport/fileutils.h"
>-
>-namespace rtc {
>-
>-static const std::string kTestOptionA = "test-option-a";
>-static const std::string kTestOptionB = "test-option-b";
>-static const std::string kTestString1 = "a string";
>-static const std::string kTestString2 = "different string";
>-static const std::string kOptionWithEquals = "foo=bar";
>-static const std::string kOptionWithNewline = "foo\nbar";
>-static const std::string kValueWithEquals = "baz=quux";
>-static const std::string kValueWithNewline = "baz\nquux";
>-static const std::string kEmptyString = "";
>-static const char kOptionWithUtf8[] = {'O',    'p', 't', '\302', '\256',
>-                                       'i',    'o', 'n', '\342', '\204',
>-                                       '\242', '\0'};  // Opt(R)io(TM).
>-static const char kValueWithUtf8[] = {
>-    'V', 'a',    'l',    '\302', '\256', 'v',
>-    'e', '\342', '\204', '\242', '\0'};  // Val(R)ue(TM).
>-static int kTestInt1 = 12345;
>-static int kTestInt2 = 67890;
>-static int kNegInt = -634;
>-static int kZero = 0;
>-
>-#if defined(WEBRTC_ANDROID)
>-// Fails on Android: https://bugs.chromium.org/p/webrtc/issues/detail?id=4364.
>-#define MAYBE_OptionsFileTest DISABLED_OptionsFileTest
>-#else
>-#define MAYBE_OptionsFileTest OptionsFileTest
>-#endif
>-
>-class MAYBE_OptionsFileTest : public testing::Test {
>- public:
>-  MAYBE_OptionsFileTest() {
>-    test_file_ =
>-        webrtc::test::TempFilename(webrtc::test::OutputPath(), ".testfile");
>-    OpenStore();
>-  }
>-
>-  ~MAYBE_OptionsFileTest() override { webrtc::test::RemoveFile(test_file_); }
>-
>- protected:
>-  void OpenStore() { store_.reset(new OptionsFile(test_file_)); }
>-
>-  std::unique_ptr<OptionsFile> store_;
>-
>- private:
>-  std::string test_file_;
>-};
>-
>-TEST_F(MAYBE_OptionsFileTest, GetSetString) {
>-  // Clear contents of the file on disk.
>-  EXPECT_TRUE(store_->Save());
>-  std::string out1, out2;
>-  EXPECT_FALSE(store_->GetStringValue(kTestOptionA, &out1));
>-  EXPECT_FALSE(store_->GetStringValue(kTestOptionB, &out2));
>-  EXPECT_TRUE(store_->SetStringValue(kTestOptionA, kTestString1));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_TRUE(store_->SetStringValue(kTestOptionB, kTestString2));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_TRUE(store_->GetStringValue(kTestOptionA, &out1));
>-  EXPECT_TRUE(store_->GetStringValue(kTestOptionB, &out2));
>-  EXPECT_EQ(kTestString1, out1);
>-  EXPECT_EQ(kTestString2, out2);
>-  EXPECT_TRUE(store_->RemoveValue(kTestOptionA));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_TRUE(store_->RemoveValue(kTestOptionB));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_FALSE(store_->GetStringValue(kTestOptionA, &out1));
>-  EXPECT_FALSE(store_->GetStringValue(kTestOptionB, &out2));
>-}
>-
>-TEST_F(MAYBE_OptionsFileTest, GetSetInt) {
>-  // Clear contents of the file on disk.
>-  EXPECT_TRUE(store_->Save());
>-  int out1, out2;
>-  EXPECT_FALSE(store_->GetIntValue(kTestOptionA, &out1));
>-  EXPECT_FALSE(store_->GetIntValue(kTestOptionB, &out2));
>-  EXPECT_TRUE(store_->SetIntValue(kTestOptionA, kTestInt1));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_TRUE(store_->SetIntValue(kTestOptionB, kTestInt2));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_TRUE(store_->GetIntValue(kTestOptionA, &out1));
>-  EXPECT_TRUE(store_->GetIntValue(kTestOptionB, &out2));
>-  EXPECT_EQ(kTestInt1, out1);
>-  EXPECT_EQ(kTestInt2, out2);
>-  EXPECT_TRUE(store_->RemoveValue(kTestOptionA));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_TRUE(store_->RemoveValue(kTestOptionB));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_FALSE(store_->GetIntValue(kTestOptionA, &out1));
>-  EXPECT_FALSE(store_->GetIntValue(kTestOptionB, &out2));
>-  EXPECT_TRUE(store_->SetIntValue(kTestOptionA, kNegInt));
>-  EXPECT_TRUE(store_->GetIntValue(kTestOptionA, &out1));
>-  EXPECT_EQ(kNegInt, out1);
>-  EXPECT_TRUE(store_->SetIntValue(kTestOptionA, kZero));
>-  EXPECT_TRUE(store_->GetIntValue(kTestOptionA, &out1));
>-  EXPECT_EQ(kZero, out1);
>-}
>-
>-TEST_F(MAYBE_OptionsFileTest, Persist) {
>-  // Clear contents of the file on disk.
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->SetStringValue(kTestOptionA, kTestString1));
>-  EXPECT_TRUE(store_->SetIntValue(kTestOptionB, kNegInt));
>-  EXPECT_TRUE(store_->Save());
>-
>-  // Load the saved contents from above.
>-  OpenStore();
>-  EXPECT_TRUE(store_->Load());
>-  std::string out1;
>-  int out2;
>-  EXPECT_TRUE(store_->GetStringValue(kTestOptionA, &out1));
>-  EXPECT_TRUE(store_->GetIntValue(kTestOptionB, &out2));
>-  EXPECT_EQ(kTestString1, out1);
>-  EXPECT_EQ(kNegInt, out2);
>-}
>-
>-TEST_F(MAYBE_OptionsFileTest, SpecialCharacters) {
>-  // Clear contents of the file on disk.
>-  EXPECT_TRUE(store_->Save());
>-  std::string out;
>-  EXPECT_FALSE(store_->SetStringValue(kOptionWithEquals, kTestString1));
>-  EXPECT_FALSE(store_->GetStringValue(kOptionWithEquals, &out));
>-  EXPECT_FALSE(store_->SetStringValue(kOptionWithNewline, kTestString1));
>-  EXPECT_FALSE(store_->GetStringValue(kOptionWithNewline, &out));
>-  EXPECT_TRUE(store_->SetStringValue(kOptionWithUtf8, kValueWithUtf8));
>-  EXPECT_TRUE(store_->SetStringValue(kTestOptionA, kTestString1));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_TRUE(store_->GetStringValue(kTestOptionA, &out));
>-  EXPECT_EQ(kTestString1, out);
>-  EXPECT_TRUE(store_->GetStringValue(kOptionWithUtf8, &out));
>-  EXPECT_EQ(kValueWithUtf8, out);
>-  EXPECT_FALSE(store_->SetStringValue(kTestOptionA, kValueWithNewline));
>-  EXPECT_TRUE(store_->GetStringValue(kTestOptionA, &out));
>-  EXPECT_EQ(kTestString1, out);
>-  EXPECT_TRUE(store_->SetStringValue(kTestOptionA, kValueWithEquals));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_TRUE(store_->GetStringValue(kTestOptionA, &out));
>-  EXPECT_EQ(kValueWithEquals, out);
>-  EXPECT_TRUE(store_->SetStringValue(kEmptyString, kTestString2));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_TRUE(store_->GetStringValue(kEmptyString, &out));
>-  EXPECT_EQ(kTestString2, out);
>-  EXPECT_TRUE(store_->SetStringValue(kTestOptionB, kEmptyString));
>-  EXPECT_TRUE(store_->Save());
>-  EXPECT_TRUE(store_->Load());
>-  EXPECT_TRUE(store_->GetStringValue(kTestOptionB, &out));
>-  EXPECT_EQ(kEmptyString, out);
>-}
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.cc
>deleted file mode 100644
>index 594deb7f0fc132fa7c436ec1742e6829892a1a11..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.cc
>+++ /dev/null
>@@ -1,155 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#if defined(WEBRTC_WIN)
>-#include <windows.h>
>-#include <shellapi.h>
>-#include <shlobj.h>
>-#include <tchar.h>
>-#endif  // WEBRTC_WIN
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/pathutils.h"
>-#include "rtc_base/stringutils.h"
>-
>-namespace rtc {
>-
>-static const char EMPTY_STR[] = "";
>-
>-// EXT_DELIM separates a file basename from extension
>-const char EXT_DELIM = '.';
>-
>-// FOLDER_DELIMS separate folder segments and the filename
>-const char* const FOLDER_DELIMS = "/\\";
>-
>-// DEFAULT_FOLDER_DELIM is the preferred delimiter for this platform
>-#ifdef WEBRTC_WIN
>-const char DEFAULT_FOLDER_DELIM = '\\';
>-#else  // !WEBRTC_WIN
>-const char DEFAULT_FOLDER_DELIM = '/';
>-#endif  // !WEBRTC_WIN
>-
>-///////////////////////////////////////////////////////////////////////////////
>-// Pathname - parsing of pathnames into components, and vice versa
>-///////////////////////////////////////////////////////////////////////////////
>-
>-bool Pathname::IsFolderDelimiter(char ch) {
>-  return (nullptr != ::strchr(FOLDER_DELIMS, ch));
>-}
>-
>-char Pathname::DefaultFolderDelimiter() {
>-  return DEFAULT_FOLDER_DELIM;
>-}
>-
>-Pathname::Pathname()
>-    : folder_delimiter_(DEFAULT_FOLDER_DELIM) {
>-}
>-
>-Pathname::Pathname(const Pathname&) = default;
>-Pathname::Pathname(Pathname&&) = default;
>-
>-Pathname::Pathname(const std::string& pathname)
>-    : folder_delimiter_(DEFAULT_FOLDER_DELIM) {
>-  SetPathname(pathname);
>-}
>-
>-Pathname::Pathname(const std::string& folder, const std::string& filename)
>-    : folder_delimiter_(DEFAULT_FOLDER_DELIM) {
>-  SetPathname(folder, filename);
>-}
>-
>-Pathname& Pathname::operator=(const Pathname&) = default;
>-Pathname& Pathname::operator=(Pathname&&) = default;
>-
>-std::string Pathname::pathname() const {
>-  std::string pathname(folder_);
>-  pathname.append(basename_);
>-  pathname.append(extension_);
>-  if (pathname.empty()) {
>-    // Instead of the empty pathname, return the current working directory.
>-    pathname.push_back('.');
>-    pathname.push_back(folder_delimiter_);
>-  }
>-  return pathname;
>-}
>-
>-void Pathname::SetPathname(const std::string& pathname) {
>-  std::string::size_type pos = pathname.find_last_of(FOLDER_DELIMS);
>-  if (pos != std::string::npos) {
>-    SetFolder(pathname.substr(0, pos + 1));
>-    SetFilename(pathname.substr(pos + 1));
>-  } else {
>-    SetFolder(EMPTY_STR);
>-    SetFilename(pathname);
>-  }
>-}
>-
>-void Pathname::SetPathname(const std::string& folder,
>-                           const std::string& filename) {
>-  SetFolder(folder);
>-  SetFilename(filename);
>-}
>-
>-void Pathname::SetFolder(const std::string& folder) {
>-  folder_.assign(folder);
>-  // Ensure folder ends in a path delimiter
>-  if (!folder_.empty() && !IsFolderDelimiter(folder_[folder_.length()-1])) {
>-    folder_.push_back(folder_delimiter_);
>-  }
>-}
>-
>-void Pathname::AppendFolder(const std::string& folder) {
>-  folder_.append(folder);
>-  // Ensure folder ends in a path delimiter
>-  if (!folder_.empty() && !IsFolderDelimiter(folder_[folder_.length()-1])) {
>-    folder_.push_back(folder_delimiter_);
>-  }
>-}
>-
>-bool Pathname::SetBasename(const std::string& basename) {
>-  if(basename.find_first_of(FOLDER_DELIMS) != std::string::npos) {
>-    return false;
>-  }
>-  basename_.assign(basename);
>-  return true;
>-}
>-
>-bool Pathname::SetExtension(const std::string& extension) {
>-  if (extension.find_first_of(FOLDER_DELIMS) != std::string::npos ||
>-    extension.find_first_of(EXT_DELIM, 1) != std::string::npos) {
>-      return false;
>-  }
>-  extension_.assign(extension);
>-  // Ensure extension begins with the extension delimiter
>-  if (!extension_.empty() && (extension_[0] != EXT_DELIM)) {
>-    extension_.insert(extension_.begin(), EXT_DELIM);
>-  }
>-  return true;
>-}
>-
>-std::string Pathname::filename() const {
>-  std::string filename(basename_);
>-  filename.append(extension_);
>-  return filename;
>-}
>-
>-bool Pathname::SetFilename(const std::string& filename) {
>-  std::string::size_type pos = filename.rfind(EXT_DELIM);
>-  if ((pos == std::string::npos) || (pos == 0)) {
>-    return SetExtension(EMPTY_STR) && SetBasename(filename);
>-  } else {
>-    return SetExtension(filename.substr(pos)) && SetBasename(filename.substr(0, pos));
>-  }
>-}
>-
>-///////////////////////////////////////////////////////////////////////////////
>-
>-} // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.h
>deleted file mode 100644
>index 9543be073f245d3409fd690b5cf9cc97db6e5a7e..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.h
>+++ /dev/null
>@@ -1,81 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef RTC_BASE_PATHUTILS_H_
>-#define RTC_BASE_PATHUTILS_H_
>-
>-#include <string>
>-
>-#include "rtc_base/checks.h"
>-
>-namespace rtc {
>-
>-///////////////////////////////////////////////////////////////////////////////
>-// Pathname - parsing of pathnames into components, and vice versa.
>-//
>-// To establish consistent terminology, a filename never contains a folder
>-// component.  A folder never contains a filename.  A pathname may include
>-// a folder and/or filename component.  Here are some examples:
>-//
>-//   pathname()      /home/john/example.txt
>-//   folder()        /home/john/
>-//   filename()                 example.txt
>-//   parent_folder() /home/
>-//   folder_name()         john/
>-//   basename()                 example
>-//   extension()                       .txt
>-//
>-// Basename may begin, end, and/or include periods, but no folder delimiters.
>-// If extension exists, it consists of a period followed by zero or more
>-// non-period/non-delimiter characters, and basename is non-empty.
>-///////////////////////////////////////////////////////////////////////////////
>-
>-class Pathname {
>- public:
>-  // Folder delimiters are slash and backslash
>-  static bool IsFolderDelimiter(char ch);
>-  static char DefaultFolderDelimiter();
>-
>-  Pathname();
>-  Pathname(const Pathname&);
>-  Pathname(Pathname&&);
>-  Pathname(const std::string& pathname);
>-  Pathname(const std::string& folder, const std::string& filename);
>-
>-  Pathname& operator=(const Pathname&);
>-  Pathname& operator=(Pathname&&);
>-
>-  // Returns the folder and filename components.  If the pathname is empty,
>-  // returns a string representing the current directory (as a relative path,
>-  // i.e., ".").
>-  std::string pathname() const;
>-  void SetPathname(const std::string& pathname);
>-  void SetPathname(const std::string& folder, const std::string& filename);
>-
>-  // SetFolder and AppendFolder will append a folder delimiter, if needed.
>-  void SetFolder(const std::string& folder);
>-  void AppendFolder(const std::string& folder);
>-
>-  bool SetBasename(const std::string& basename);
>-
>-  // SetExtension will prefix a period, if needed.
>-  bool SetExtension(const std::string& extension);
>-
>-  std::string filename() const;
>-  bool SetFilename(const std::string& filename);
>-
>- private:
>-  std::string folder_, basename_, extension_;
>-  char folder_delimiter_;
>-};
>-
>-}  // namespace rtc
>-
>-#endif  // RTC_BASE_PATHUTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils_unittest.cc
>deleted file mode 100644
>index fae4f0aba59bf5e4f8e89bf9d9508d0f579dbee3..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils_unittest.cc
>+++ /dev/null
>@@ -1,37 +0,0 @@
>-/*
>- *  Copyright 2007 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/pathutils.h"
>-#include "rtc_base/gunit.h"
>-
>-TEST(Pathname, ReturnsDotForEmptyPathname) {
>-  const std::string kCWD =
>-      std::string(".") + rtc::Pathname::DefaultFolderDelimiter();
>-
>-  rtc::Pathname path("/", "");
>-  EXPECT_TRUE (path.filename().empty());
>-  EXPECT_FALSE(path.pathname().empty());
>-  EXPECT_EQ(std::string("/"), path.pathname());
>-
>-  path.SetPathname("", "foo");
>-  EXPECT_FALSE(path.filename().empty());
>-  EXPECT_FALSE(path.pathname().empty());
>-  EXPECT_EQ(std::string("foo"), path.pathname());
>-
>-  path.SetPathname("", "");
>-  EXPECT_TRUE (path.filename().empty());
>-  EXPECT_FALSE(path.pathname().empty());
>-  EXPECT_EQ(kCWD, path.pathname());
>-
>-  path.SetPathname(kCWD, "");
>-  EXPECT_TRUE (path.filename().empty());
>-  EXPECT_FALSE(path.pathname().empty());
>-  EXPECT_EQ(kCWD, path.pathname());
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver_unittest.cc
>index 1e046c057e5d8211c235d854f5d6ce712e6bbaa3..4b36cd5f608e8f87fe9e6d5b9d90d9e910e4eecd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver_unittest.cc
>@@ -547,7 +547,13 @@ Thread* PosixSignalDeliveryTest::signaled_thread_ = nullptr;
> 
> // Test receiving a synchronous signal while not in Wait() and then entering
> // Wait() afterwards.
>-TEST_F(PosixSignalDeliveryTest, RaiseThenWait) {
>+// TODO(webrtc:7864): Fails on real iOS devices
>+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM_FAMILY)
>+#define MAYBE_RaiseThenWait DISABLED_RaiseThenWait
>+#else
>+#define MAYBE_RaiseThenWait RaiseThenWait
>+#endif
>+TEST_F(PosixSignalDeliveryTest, MAYBE_RaiseThenWait) {
>   ASSERT_TRUE(ss_->SetPosixSignalHandler(SIGTERM, &RecordSignal));
>   raise(SIGTERM);
>   EXPECT_TRUE(ss_->Wait(0, true));
>@@ -557,7 +563,13 @@ TEST_F(PosixSignalDeliveryTest, RaiseThenWait) {
> 
> // Test that we can handle getting tons of repeated signals and that we see all
> // the different ones.
>-TEST_F(PosixSignalDeliveryTest, InsanelyManySignals) {
>+// TODO(webrtc:7864): Fails on real iOS devices
>+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM_FAMILY)
>+#define MAYBE_InsanelyManySignals DISABLED_InsanelyManySignals
>+#else
>+#define MAYBE_InsanelyManySignals InsanelyManySignals
>+#endif
>+TEST_F(PosixSignalDeliveryTest, MAYBE_InsanelyManySignals) {
>   ss_->SetPosixSignalHandler(SIGTERM, &RecordSignal);
>   ss_->SetPosixSignalHandler(SIGINT, &RecordSignal);
>   for (int i = 0; i < 10000; ++i) {
>@@ -597,7 +609,13 @@ class RaiseSigTermRunnable : public Runnable {
> 
> // Test that it works no matter what thread the kernel chooses to give the
> // signal to (since it's not guaranteed to be the one that Wait() runs on).
>-TEST_F(PosixSignalDeliveryTest, SignalOnDifferentThread) {
>+// TODO(webrtc:7864): Fails on real iOS devices
>+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM_FAMILY)
>+#define MAYBE_SignalOnDifferentThread DISABLED_SignalOnDifferentThread
>+#else
>+#define MAYBE_SignalOnDifferentThread SignalOnDifferentThread
>+#endif
>+TEST_F(PosixSignalDeliveryTest, DISABLED_SignalOnDifferentThread) {
>   ss_->SetPosixSignalHandler(SIGTERM, &RecordSignal);
>   // Mask out SIGTERM so that it can't be delivered to this thread.
>   sigset_t mask;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.cc
>index a4c906a1ed028adf6db0bff2528407b90b9bccc9..baefb229313bd2a73a34dd1324808a5929d8d035 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.cc
>@@ -10,14 +10,13 @@
> 
> #include "rtc_base/platform_file.h"
> 
>-#include "rtc_base/stringutils.h"
>-
> #if defined(WEBRTC_WIN)
> #include <io.h>
>+
>+#include "rtc_base/stringutils.h"  // For ToUtf16
> #else
> #include <fcntl.h>
> #include <sys/stat.h>
>-#include <sys/types.h>
> #include <unistd.h>
> #endif
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.cc
>index 79d9d53aeee02e0d8be0411b0e019b6075d07e1e..ba84b6ae8e049548773c7fed473155de5c6a343f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.cc
>@@ -10,15 +10,17 @@
> 
> #include "rtc_base/platform_thread.h"
> 
>+#if !defined(WEBRTC_WIN)
>+#include <sched.h>
>+#endif
>+#include <stdint.h>
>+#include <time.h>
>+#include <algorithm>
>+
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/timeutils.h"
> 
>-#if defined(WEBRTC_LINUX)
>-#include <sys/prctl.h>
>-#include <sys/syscall.h>
>-#endif
>-
> namespace rtc {
> namespace {
> #if defined(WEBRTC_WIN)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.h
>index 33921c209da0289b0747daa47b1545bca5446d54..47c23dc1b5b47f4b29885228f38b306e23ad2b0e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.h
>@@ -11,10 +11,12 @@
> #ifndef RTC_BASE_PLATFORM_THREAD_H_
> #define RTC_BASE_PLATFORM_THREAD_H_
> 
>+#ifndef WEBRTC_WIN
>+#include <pthread.h>
>+#endif
> #include <string>
> 
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/event.h"
> #include "rtc_base/platform_thread_types.h"
> #include "rtc_base/thread_checker.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxy_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxy_unittest.cc
>index f42039f50ba2118ddccaab845c1dc940e2a84c02..010189384f8b757099ae68bc8a36d0e98ae2bd60 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxy_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxy_unittest.cc
>@@ -18,7 +18,6 @@
> #include "rtc_base/virtualsocketserver.h"
> 
> using rtc::Socket;
>-using rtc::Thread;
> using rtc::SocketAddress;
> 
> static const SocketAddress kSocksProxyIntAddr("1.2.3.4", 1080);
>@@ -49,7 +48,8 @@ TEST_F(ProxyTest, TestSocks5Connect) {
>       socket, kSocksProxyIntAddr, "", rtc::CryptString());
>   // TODO: IPv6-ize these tests when proxy supports IPv6.
> 
>-  rtc::TestEchoServer server(Thread::Current(), SocketAddress(INADDR_ANY, 0));
>+  rtc::TestEchoServer server(rtc::Thread::Current(),
>+                             SocketAddress(INADDR_ANY, 0));
> 
>   std::unique_ptr<rtc::AsyncTCPSocket> packet_socket(
>       rtc::AsyncTCPSocket::Create(proxy_socket, SocketAddress(INADDR_ANY, 0),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.cc
>index 55cab80975c36fd1543e84eda512aff382bde74c..71c4879099f05648c48506096374bcba9d2f5305 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.cc
>@@ -13,6 +13,7 @@
> #include <algorithm>
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
> #include "rtc_base/socketfactory.h"
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/race_checker.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/race_checker.h
>index 73567e98d8ac07db766e81c897db34c0d3ad8dec..4d574601eb0b4ba62ace5cdb46d690a3413fa3ff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/race_checker.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/race_checker.h
>@@ -12,7 +12,7 @@
> #define RTC_BASE_RACE_CHECKER_H_
> 
> #include "rtc_base/checks.h"
>-#include "rtc_base/platform_thread.h"
>+#include "rtc_base/platform_thread_types.h"
> #include "rtc_base/thread_annotations.h"
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/random.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/random.h
>index 2faa9856dc6aee1105cfebb9ff6c8a5d609b6e09..e1c3bb7058e82697494994490086f02712dba044 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/random.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/random.h
>@@ -11,6 +11,7 @@
> #ifndef RTC_BASE_RANDOM_H_
> #define RTC_BASE_RANDOM_H_
> 
>+#include <stdint.h>
> #include <limits>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.cc
>index 0343f256b293f0a11448f167dca3190cf7505d56..5c7bdefb0c8b97a068cadd93477f5a05db726ca8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.cc
>@@ -9,6 +9,10 @@
>  */
> 
> #include "rtc_base/rate_limiter.h"
>+
>+#include <limits>
>+
>+#include "absl/types/optional.h"
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.h
>index 0bfde0da259bb7a4a2e5da0106753f0393e96f1e..43ef88d8fc356eb0e91c9782c1629cb8da6389c1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.h
>@@ -11,11 +11,13 @@
> #ifndef RTC_BASE_RATE_LIMITER_H_
> #define RTC_BASE_RATE_LIMITER_H_
> 
>-#include <limits>
>+#include <stddef.h>
>+#include <stdint.h>
> 
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/rate_statistics.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter_unittest.cc
>index 6efea5458e1b087cfa205ce1e5e1b4f37165086f..ac0625fa186f79ceb17cdc242648a07ebe919e3d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter_unittest.cc
>@@ -112,9 +112,7 @@ static const int64_t kMaxTimeoutMs = 30000;
> class ThreadTask {
>  public:
>   explicit ThreadTask(RateLimiter* rate_limiter)
>-      : rate_limiter_(rate_limiter),
>-        start_signal_(false, false),
>-        end_signal_(false, false) {}
>+      : rate_limiter_(rate_limiter) {}
>   virtual ~ThreadTask() {}
> 
>   void Run() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.h
>index 68035c9a68e98ba0032b57dd3da002d26cc1e976..d4ccc59bf5bdfd1c89388fd7dfef12bfd80be0a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.h
>@@ -11,6 +11,8 @@
> #ifndef RTC_BASE_RATE_STATISTICS_H_
> #define RTC_BASE_RATE_STATISTICS_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> 
> #include "absl/types/optional.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.cc
>index e31d26627ab19edb27c1dbe020b883e5df73d0e9..7c96ca9f60858892b95501ef2e6861b0b2178968 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.cc
>@@ -10,8 +10,6 @@
> 
> #include "rtc_base/ratetracker.h"
> 
>-#include <stddef.h>
>-
> #include <algorithm>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.cc
>index 7f027ba5f80f48886d73e50bb3945160f5a23bf9..875068f4fc988970e026f871f296f3d25f5ea2ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.cc
>@@ -14,6 +14,9 @@
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/refcountedobject.h"
>+#include "rtc_base/sslcertificate.h"
>+#include "rtc_base/sslidentity.h"
>+#include "rtc_base/timeutils.h"
> 
> namespace rtc {
> 
>@@ -29,7 +32,7 @@ RTCCertificate::RTCCertificate(SSLIdentity* identity) : identity_(identity) {
> RTCCertificate::~RTCCertificate() {}
> 
> uint64_t RTCCertificate::Expires() const {
>-  int64_t expires = ssl_certificate().CertificateExpirationTime();
>+  int64_t expires = GetSSLCertificate().CertificateExpirationTime();
>   if (expires != -1)
>     return static_cast<uint64_t>(expires) * kNumMillisecsPerSec;
>   // If the expiration time could not be retrieved return an expired timestamp.
>@@ -40,17 +43,22 @@ bool RTCCertificate::HasExpired(uint64_t now) const {
>   return Expires() <= now;
> }
> 
>+const SSLCertificate& RTCCertificate::GetSSLCertificate() const {
>+  return identity_->certificate();
>+}
>+
>+// Deprecated: TODO(benwright) - Remove once chromium is updated.
> const SSLCertificate& RTCCertificate::ssl_certificate() const {
>   return identity_->certificate();
> }
> 
>-const SSLCertChain& RTCCertificate::ssl_cert_chain() const {
>+const SSLCertChain& RTCCertificate::GetSSLCertificateChain() const {
>   return identity_->cert_chain();
> }
> 
> RTCCertificatePEM RTCCertificate::ToPEM() const {
>   return RTCCertificatePEM(identity_->PrivateKeyToPEMString(),
>-                           ssl_certificate().ToPEMString());
>+                           GetSSLCertificate().ToPEMString());
> }
> 
> scoped_refptr<RTCCertificate> RTCCertificate::FromPEM(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.h
>index d5422f8b4a47d2fe1928b6123c7b5c072e06364d..561ea0f9e7e906f4c78d0fa11999a9ba872c7037 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.h
>@@ -12,15 +12,18 @@
> #define RTC_BASE_RTCCERTIFICATE_H_
> 
> #include <stdint.h>
>-
> #include <memory>
>+#include <string>
> 
> #include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "rtc_base/sslidentity.h"
> 
> namespace rtc {
> 
>+class SSLCertChain;
>+class SSLCertificate;
>+class SSLIdentity;
>+
> // This class contains PEM strings of an RTCCertificate's private key and
> // certificate and acts as a text representation of RTCCertificate. Certificates
> // can be serialized and deserialized to and from this format, which allows for
>@@ -55,11 +58,15 @@ class RTCCertificate : public RefCountInterface {
>   // Checks if the certificate has expired, where |now| is expressed in ms
>   // relative to epoch, 1970-01-01T00:00:00Z.
>   bool HasExpired(uint64_t now) const;
>+
>+  const SSLCertificate& GetSSLCertificate() const;
>+  const SSLCertChain& GetSSLCertificateChain() const;
>+
>+  // Deprecated: TODO(benwright) - Remove once chromium is updated.
>   const SSLCertificate& ssl_certificate() const;
>-  const SSLCertChain& ssl_cert_chain() const;
> 
>   // TODO(hbos): If possible, remove once RTCCertificate and its
>-  // ssl_certificate() is used in all relevant places. Should not pass around
>+  // GetSSLCertificate() is used in all relevant places. Should not pass around
>   // raw SSLIdentity* for the sake of accessing SSLIdentity::certificate().
>   // However, some places might need SSLIdentity* for its public/private key...
>   SSLIdentity* identity() const { return identity_.get(); }
>@@ -77,7 +84,7 @@ class RTCCertificate : public RefCountInterface {
> 
>  private:
>   // The SSLIdentity is the owner of the SSLCertificate. To protect our
>-  // ssl_certificate() we take ownership of |identity_|.
>+  // GetSSLCertificate() we take ownership of |identity_|.
>   std::unique_ptr<SSLIdentity> identity_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.cc
>index 0b51c611badf537014bd204b3e5bab6853ffb2e2..114b35c6658fcc9e55725e7a3444549bc12df896 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.cc
>@@ -10,10 +10,15 @@
> 
> #include "rtc_base/rtccertificategenerator.h"
> 
>+#include <time.h>
> #include <algorithm>
> #include <memory>
>+#include <utility>
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/location.h"
>+#include "rtc_base/messagehandler.h"
>+#include "rtc_base/messagequeue.h"
> #include "rtc_base/refcountedobject.h"
> #include "rtc_base/sslidentity.h"
> 
>@@ -23,7 +28,6 @@ namespace {
> 
> // A certificates' subject and issuer name.
> const char kIdentityName[] = "WebRTC";
>-
> const uint64_t kYearInSeconds = 365 * 24 * 60 * 60;
> 
> enum {
>@@ -60,11 +64,9 @@ class RTCCertificateGenerationTask : public RefCountInterface,
>     switch (msg->message_id) {
>       case MSG_GENERATE:
>         RTC_DCHECK(worker_thread_->IsCurrent());
>-
>         // Perform the certificate generation work here on the worker thread.
>         certificate_ = RTCCertificateGenerator::GenerateCertificate(
>             key_params_, expires_ms_);
>-
>         // Handle callbacks on signaling thread. Pass on the |msg->pdata|
>         // (which references |this| with ref counting) to that thread.
>         signaling_thread_->Post(RTC_FROM_HERE, this, MSG_GENERATE_DONE,
>@@ -72,14 +74,12 @@ class RTCCertificateGenerationTask : public RefCountInterface,
>         break;
>       case MSG_GENERATE_DONE:
>         RTC_DCHECK(signaling_thread_->IsCurrent());
>-
>         // Perform callback with result here on the signaling thread.
>         if (certificate_) {
>           callback_->OnSuccess(certificate_);
>         } else {
>           callback_->OnFailure();
>         }
>-
>         // Destroy |msg->pdata| which references |this| with ref counting. This
>         // may result in |this| being deleted - do not touch member variables
>         // after this line.
>@@ -105,9 +105,11 @@ class RTCCertificateGenerationTask : public RefCountInterface,
> scoped_refptr<RTCCertificate> RTCCertificateGenerator::GenerateCertificate(
>     const KeyParams& key_params,
>     const absl::optional<uint64_t>& expires_ms) {
>-  if (!key_params.IsValid())
>+  if (!key_params.IsValid()) {
>     return nullptr;
>-  SSLIdentity* identity;
>+  }
>+
>+  SSLIdentity* identity = nullptr;
>   if (!expires_ms) {
>     identity = SSLIdentity::Generate(kIdentityName, key_params);
>   } else {
>@@ -124,8 +126,9 @@ scoped_refptr<RTCCertificate> RTCCertificateGenerator::GenerateCertificate(
>     identity = SSLIdentity::GenerateWithExpiration(kIdentityName, key_params,
>                                                    cert_lifetime_s);
>   }
>-  if (!identity)
>+  if (!identity) {
>     return nullptr;
>+  }
>   std::unique_ptr<SSLIdentity> identity_sptr(identity);
>   return RTCCertificate::Create(std::move(identity_sptr));
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.h
>index a6c503a9f29cf04d6d26a0572be5e97442ff45d2..fed075e6ddf4e67b434baaeab1c6329daac3bcc2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.h
>@@ -11,6 +11,8 @@
> #ifndef RTC_BASE_RTCCERTIFICATEGENERATOR_H_
> #define RTC_BASE_RTCCERTIFICATEGENERATOR_H_
> 
>+#include <stdint.h>
>+
> #include "absl/types/optional.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/rtccertificate.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>index 23a748f84fc027d4cdf5efbfb2430b2f56d82e93..8af0824b67c85bcdedb3efbc35e410e6e01ae99e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>@@ -14,7 +14,7 @@
> #include <stddef.h>  // For size_t.
> 
> #ifdef __cplusplus
>-#include <type_traits>
>+#include "absl/meta/type_traits.h"
> #endif
> 
> #if defined(__has_feature)
>@@ -98,10 +98,10 @@ namespace sanitizer_impl {
> 
> template <typename T>
> constexpr bool IsTriviallyCopyable() {
>-  return static_cast<bool>(std::is_trivially_copy_constructible<T>::value &&
>-                           (std::is_trivially_copy_assignable<T>::value ||
>+  return static_cast<bool>(absl::is_trivially_copy_constructible<T>::value &&
>+                           (absl::is_trivially_copy_assignable<T>::value ||
>                             !std::is_copy_assignable<T>::value) &&
>-                           std::is_trivially_destructible<T>::value);
>+                           absl::is_trivially_destructible<T>::value);
> }
> 
> }  // namespace sanitizer_impl
>@@ -123,9 +123,11 @@ inline void MsanMarkUninitialized(const T& mem) {
> 
> template <typename T>
> inline T MsanUninitialized(T t) {
>+#if RTC_HAS_MSAN
>   // TODO(bugs.webrtc.org/8762): Switch to std::is_trivially_copyable when it
>   // becomes available in downstream projects.
>   static_assert(sanitizer_impl::IsTriviallyCopyable<T>(), "");
>+#endif
>   rtc_MsanMarkUninitialized(&t, sizeof(T), 1);
>   return t;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/scoped_ref_ptr.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/scoped_ref_ptr.h
>index a583aa99b533bb29f6973b2ff297ae2fbba3fa1e..b961ff57e864a7393280934d3b2c812c71b9ee33 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/scoped_ref_ptr.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/scoped_ref_ptr.h
>@@ -8,154 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-// Originally these classes are from Chromium.
>-// http://src.chromium.org/viewvc/chrome/trunk/src/base/memory/ref_counted.h?view=markup
>-
>-//
>-// A smart pointer class for reference counted objects.  Use this class instead
>-// of calling AddRef and Release manually on a reference counted object to
>-// avoid common memory leaks caused by forgetting to Release an object
>-// reference.  Sample usage:
>-//
>-//   class MyFoo : public RefCounted<MyFoo> {
>-//    ...
>-//   };
>-//
>-//   void some_function() {
>-//     scoped_refptr<MyFoo> foo = new MyFoo();
>-//     foo->Method(param);
>-//     // |foo| is released when this function returns
>-//   }
>-//
>-//   void some_other_function() {
>-//     scoped_refptr<MyFoo> foo = new MyFoo();
>-//     ...
>-//     foo = nullptr;  // explicitly releases |foo|
>-//     ...
>-//     if (foo)
>-//       foo->Method(param);
>-//   }
>-//
>-// The above examples show how scoped_refptr<T> acts like a pointer to T.
>-// Given two scoped_refptr<T> classes, it is also possible to exchange
>-// references between the two objects, like so:
>-//
>-//   {
>-//     scoped_refptr<MyFoo> a = new MyFoo();
>-//     scoped_refptr<MyFoo> b;
>-//
>-//     b.swap(a);
>-//     // now, |b| references the MyFoo object, and |a| references null.
>-//   }
>-//
>-// To make both |a| and |b| in the above example reference the same MyFoo
>-// object, simply use the assignment operator:
>-//
>-//   {
>-//     scoped_refptr<MyFoo> a = new MyFoo();
>-//     scoped_refptr<MyFoo> b;
>-//
>-//     b = a;
>-//     // now, |a| and |b| each own a reference to the same MyFoo object.
>-//   }
>-//
>-
> #ifndef RTC_BASE_SCOPED_REF_PTR_H_
> #define RTC_BASE_SCOPED_REF_PTR_H_
> 
>-#include <memory>
>-
>-namespace rtc {
>-
>-template <class T>
>-class scoped_refptr {
>- public:
>-  scoped_refptr() : ptr_(nullptr) {}
>-
>-  scoped_refptr(T* p) : ptr_(p) {
>-    if (ptr_)
>-      ptr_->AddRef();
>-  }
>-
>-  scoped_refptr(const scoped_refptr<T>& r) : ptr_(r.ptr_) {
>-    if (ptr_)
>-      ptr_->AddRef();
>-  }
>-
>-  template <typename U>
>-  scoped_refptr(const scoped_refptr<U>& r) : ptr_(r.get()) {
>-    if (ptr_)
>-      ptr_->AddRef();
>-  }
>-
>-  // Move constructors.
>-  scoped_refptr(scoped_refptr<T>&& r) : ptr_(r.release()) {}
>-
>-  template <typename U>
>-  scoped_refptr(scoped_refptr<U>&& r) : ptr_(r.release()) {}
>-
>-  ~scoped_refptr() {
>-    if (ptr_)
>-      ptr_->Release();
>-  }
>-
>-  T* get() const { return ptr_; }
>-  operator T*() const { return ptr_; }
>-  T* operator->() const { return ptr_; }
>-
>-  // Returns the (possibly null) raw pointer, and makes the scoped_refptr hold a
>-  // null pointer, all without touching the reference count of the underlying
>-  // pointed-to object. The object is still reference counted, and the caller of
>-  // release() is now the proud owner of one reference, so it is responsible for
>-  // calling Release() once on the object when no longer using it.
>-  T* release() {
>-    T* retVal = ptr_;
>-    ptr_ = nullptr;
>-    return retVal;
>-  }
>-
>-  scoped_refptr<T>& operator=(T* p) {
>-    // AddRef first so that self assignment should work
>-    if (p)
>-      p->AddRef();
>-    if (ptr_)
>-      ptr_->Release();
>-    ptr_ = p;
>-    return *this;
>-  }
>-
>-  scoped_refptr<T>& operator=(const scoped_refptr<T>& r) {
>-    return *this = r.ptr_;
>-  }
>-
>-  template <typename U>
>-  scoped_refptr<T>& operator=(const scoped_refptr<U>& r) {
>-    return *this = r.get();
>-  }
>-
>-  scoped_refptr<T>& operator=(scoped_refptr<T>&& r) {
>-    scoped_refptr<T>(std::move(r)).swap(*this);
>-    return *this;
>-  }
>-
>-  template <typename U>
>-  scoped_refptr<T>& operator=(scoped_refptr<U>&& r) {
>-    scoped_refptr<T>(std::move(r)).swap(*this);
>-    return *this;
>-  }
>-
>-  void swap(T** pp) {
>-    T* p = ptr_;
>-    ptr_ = *pp;
>-    *pp = p;
>-  }
>-
>-  void swap(scoped_refptr<T>& r) { swap(&r.ptr_); }
>-
>- protected:
>-  T* ptr_;
>-};
>+// TODO(bugs.webrtc.org/9887): This is a forward header for backwards
>+// compatibility. Remove when downstream clients are updated.
> 
>-}  // namespace rtc
>+#include "api/scoped_refptr.h"
> 
> #endif  // RTC_BASE_SCOPED_REF_PTR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.cc
>index 16069c25f7ab064672d96f3bb0426b14790d2b47..717cb95040835a57b14805bab2025cfab9838575 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.cc
>@@ -14,6 +14,7 @@
> #include <dispatch/dispatch.h>
> #endif
> 
>+#include "rtc_base/checks.h"
> #include "rtc_base/sequenced_task_checker.h"
> #include "rtc_base/task_queue.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_unittest.cc
>index 96e655baad4f37c3706b23b72d14183ebfdf3826..7b7247c64c015fad3f1677f8ac712afec9b89021 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_unittest.cc
>@@ -12,6 +12,7 @@
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/platform_thread.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/thread_checker.h"
>@@ -43,7 +44,6 @@ class CallCalledSequentiallyOnThread {
>   CallCalledSequentiallyOnThread(bool expect_true,
>                                  SequencedTaskChecker* sequenced_task_checker)
>       : expect_true_(expect_true),
>-        thread_has_run_event_(false, false),
>         thread_(&Run, this, "call_do_stuff_on_thread"),
>         sequenced_task_checker_(sequenced_task_checker) {
>     thread_.Start();
>@@ -77,7 +77,6 @@ class DeleteSequencedCheckerOnThread {
>   explicit DeleteSequencedCheckerOnThread(
>       std::unique_ptr<SequencedTaskChecker> sequenced_task_checker)
>       : thread_(&Run, this, "delete_sequenced_task_checker_on_thread"),
>-        thread_has_run_event_(false, false),
>         sequenced_task_checker_(std::move(sequenced_task_checker)) {
>     thread_.Start();
>   }
>@@ -118,7 +117,7 @@ void RunMethodOnDifferentTaskQueue(bool expect_true) {
> 
>   static const char kQueueName[] = "MethodNotAllowedOnDifferentTq";
>   TaskQueue queue(kQueueName);
>-  Event done_event(false, false);
>+  Event done_event;
>   queue.PostTask([&sequenced_task_checker, &done_event, expect_true] {
>     if (expect_true)
>       EXPECT_TRUE(sequenced_task_checker->CalledSequentially());
>@@ -135,7 +134,7 @@ void DetachThenCallFromDifferentTaskQueue(bool expect_true) {
> 
>   sequenced_task_checker->Detach();
> 
>-  Event done_event(false, false);
>+  Event done_event;
>   TaskQueue queue1("DetachThenCallFromDifferentTaskQueueImpl1");
>   queue1.PostTask([&sequenced_task_checker, &done_event] {
>     EXPECT_TRUE(sequenced_task_checker->CalledSequentially());
>@@ -193,7 +192,7 @@ TEST(SequencedTaskCheckerTest, DetachFromThreadAndUseOnTaskQueue) {
>   sequenced_task_checker->Detach();
>   static const char kQueueName[] = "DetachFromThreadAndUseOnTaskQueue";
>   TaskQueue queue(kQueueName);
>-  Event done_event(false, false);
>+  Event done_event;
>   queue.PostTask([&sequenced_task_checker, &done_event] {
>     EXPECT_TRUE(sequenced_task_checker->CalledSequentially());
>     done_event.Set();
>@@ -203,7 +202,7 @@ TEST(SequencedTaskCheckerTest, DetachFromThreadAndUseOnTaskQueue) {
> 
> TEST(SequencedTaskCheckerTest, DetachFromTaskQueueAndUseOnThread) {
>   TaskQueue queue("DetachFromTaskQueueAndUseOnThread");
>-  Event done_event(false, false);
>+  Event done_event;
>   queue.PostTask([&done_event] {
>     std::unique_ptr<SequencedTaskChecker> sequenced_task_checker(
>         new SequencedTaskChecker());
>@@ -271,7 +270,7 @@ void TestAnnotationsOnWrongQueue() {
>   TestAnnotations annotations;
>   static const char kQueueName[] = "TestAnnotationsOnWrongQueueDebug";
>   TaskQueue queue(kQueueName);
>-  Event done_event(false, false);
>+  Event done_event;
>   queue.PostTask([&annotations, &done_event] {
>     annotations.ModifyTestVar();
>     done_event.Set();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.cc
>index eb79dc84fe8e5465dac1ee6e8944d0be71b4efb6..5dd93879315c39ead0ef065ff3c4f921b971f8fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.cc
>@@ -10,8 +10,13 @@
> 
> #include "rtc_base/signalthread.h"
> 
>+#include <memory>
>+
> #include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/location.h"
>+#include "rtc_base/nullsocketserver.h"
>+#include "rtc_base/socketserver.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.h
>index 021cf4d78df8e4a35db8d6b372787bfc107830d3..9208e2c2467787537e083b5ad9f73b5f8245d34a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.h
>@@ -15,9 +15,12 @@
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/nullsocketserver.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/messagehandler.h"
>+#include "rtc_base/messagequeue.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.cc
>index a9749a473ee1181fe5c9804e8314197bc4cb1b1f..f19b34412f4a58f96dc1a0155ee2102260fb5b0e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.cc
>@@ -12,16 +12,4 @@
> 
> namespace rtc {
> 
>-PacketInfo::PacketInfo() = default;
>-PacketInfo::PacketInfo(const PacketInfo& info) = default;
>-PacketInfo::~PacketInfo() = default;
>-
>-SentPacket::SentPacket() = default;
>-SentPacket::SentPacket(int64_t packet_id, int64_t send_time_ms)
>-    : packet_id(packet_id), send_time_ms(send_time_ms) {}
>-SentPacket::SentPacket(int64_t packet_id,
>-                       int64_t send_time_ms,
>-                       const rtc::PacketInfo& info)
>-    : packet_id(packet_id), send_time_ms(send_time_ms), info(info) {}
>-
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.h
>index b8290bb139b2e69a57b2074ea5763589d40669cf..e7e82108cd29d2abdb319bd236d9b647611ac518 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.h
>@@ -25,8 +25,8 @@
> #include "rtc_base/win32.h"
> #endif
> 
>-#include "absl/types/optional.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/network/sent_packet.h"
> #include "rtc_base/socketaddress.h"
> 
> // Rather than converting errors into a private namespace,
>@@ -123,50 +123,6 @@ inline bool IsBlockingError(int e) {
>   return (e == EWOULDBLOCK) || (e == EAGAIN) || (e == EINPROGRESS);
> }
> 
>-enum class PacketType {
>-  kUnknown,
>-  kData,
>-  kIceConnectivityCheck,
>-  kIceConnectivityCheckResponse,
>-  kStunMessage,
>-  kTurnMessage,
>-};
>-
>-enum class PacketInfoProtocolType {
>-  kUnknown,
>-  kUdp,
>-  kTcp,
>-  kSsltcp,
>-  kTls,
>-};
>-
>-struct PacketInfo {
>-  PacketInfo();
>-  PacketInfo(const PacketInfo& info);
>-  ~PacketInfo();
>-
>-  PacketType packet_type = PacketType::kUnknown;
>-  PacketInfoProtocolType protocol = PacketInfoProtocolType::kUnknown;
>-  // A unique id assigned by the network manager, and absl::nullopt if not set.
>-  absl::optional<uint16_t> network_id;
>-  size_t packet_size_bytes = 0;
>-  size_t turn_overhead_bytes = 0;
>-  SocketAddress local_socket_address;
>-  SocketAddress remote_socket_address;
>-};
>-
>-struct SentPacket {
>-  SentPacket();
>-  SentPacket(int64_t packet_id, int64_t send_time_ms);
>-  SentPacket(int64_t packet_id,
>-             int64_t send_time_ms,
>-             const rtc::PacketInfo& info);
>-
>-  int64_t packet_id = -1;
>-  int64_t send_time_ms = -1;
>-  rtc::PacketInfo info;
>-};
>-
> // General interface for the socket implementations of various networks.  The
> // methods match those of normal UNIX sockets very closely.
> class Socket {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.cc
>index acd9b633858c8814f6d4b8f86ca1d9d3d5184e81..945192889142a8e04ed3f9d3d78cab2def852b47 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.cc
>@@ -13,24 +13,29 @@
> #endif
> 
> #include <errno.h>
>+#include <stdio.h>
>+#include <stdlib.h>
>+#include <string.h>
> #include <time.h>
> 
> #if defined(WEBRTC_WIN)
> #include <windows.h>
> #include <winsock2.h>
> #include <ws2tcpip.h>
>+
> #define SECURITY_WIN32
> #include <security.h>
> #endif
> 
> #include <algorithm>
> 
>+#include "absl/strings/match.h"
>+#include "rtc_base/buffer.h"
> #include "rtc_base/bytebuffer.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/httpcommon.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/socketadapters.h"
>-#include "rtc_base/stringencode.h"
> #include "rtc_base/strings/string_builder.h"
> #include "rtc_base/stringutils.h"
> #include "rtc_base/zero_memory.h"
>@@ -457,7 +462,7 @@ void AsyncHttpsProxySocket::ProcessLine(char* data, size_t len) {
>         return;
>     }
>   } else if ((state_ == PS_AUTHENTICATE) &&
>-             (_strnicmp(data, "Proxy-Authenticate:", 19) == 0)) {
>+             absl::StartsWithIgnoreCase(data, "Proxy-Authenticate:")) {
>     std::string response, auth_method;
>     switch (HttpAuthenticate(data + 19, len - 19, proxy_, "CONNECT", "/", user_,
>                              pass_, context_, response, auth_method)) {
>@@ -485,12 +490,12 @@ void AsyncHttpsProxySocket::ProcessLine(char* data, size_t len) {
>         unknown_mechanisms_.clear();
>         break;
>     }
>-  } else if (_strnicmp(data, "Content-Length:", 15) == 0) {
>+  } else if (absl::StartsWithIgnoreCase(data, "Content-Length:")) {
>     content_length_ = strtoul(data + 15, 0, 0);
>-  } else if (_strnicmp(data, "Proxy-Connection: Keep-Alive", 28) == 0) {
>+  } else if (absl::StartsWithIgnoreCase(data, "Proxy-Connection: Keep-Alive")) {
>     expect_close_ = false;
>     /*
>-  } else if (_strnicmp(data, "Connection: close", 17) == 0) {
>+  } else if (absl::StartsWithIgnoreCase(data, "Connection: close") {
>     expect_close_ = true;
>     */
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.h
>index ad88fe6b6fb8cf2003beef81539c3e222382d10f..062f75c3feb1fffc300944144719eee5238a7942 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.h
>@@ -11,13 +11,11 @@
> #ifndef RTC_BASE_SOCKETADAPTERS_H_
> #define RTC_BASE_SOCKETADAPTERS_H_
> 
>-#include <map>
> #include <string>
> 
> #include "rtc_base/asyncsocket.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/cryptstring.h"
>-#include "rtc_base/logging.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.h
>index bff8e76495eb74bca589824b8ed93021ccf20b21..b1a52b93716a8e24ca28b7e5ded15ed995efd90a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.h
>@@ -15,7 +15,6 @@
> #ifdef UNIT_TEST
> #include <ostream>  // no-presubmit-check TODO(webrtc:8982)
> #endif              // UNIT_TEST
>-#include <vector>
> #include "rtc_base/ipaddress.h"
> 
> #undef SetPort
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.h
>index 5ff148a75d87604e1db1fc61d8490c1d2df722e9..6691386d3abff8b9e8e430c3b7355d29479f9ed4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.h
>@@ -11,6 +11,8 @@
> #ifndef RTC_BASE_SOCKETADDRESSPAIR_H_
> #define RTC_BASE_SOCKETADDRESSPAIR_H_
> 
>+#include <stddef.h>
>+
> #include "rtc_base/socketaddress.h"
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketstream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketstream.cc
>index 2ea1cec0ac736d35501103eda2ed06e8181574ec..8978404cf45f02d9d09016a6f7c15499d2abf479 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketstream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketstream.cc
>@@ -11,6 +11,7 @@
> #include "rtc_base/socketstream.h"
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/socket.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.h
>index 4843d264fcccd366d480f6438614103b9ea9785b..7ebedcadf026fef9ab9b1c133ad3598a3754ce0b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.h
>@@ -16,6 +16,7 @@
> 
> #include "rtc_base/asyncsocket.h"
> #include "rtc_base/sslcertificate.h"
>+#include "rtc_base/sslidentity.h"
> #include "rtc_base/sslstreamadapter.h"
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter_unittest.cc
>index ec532b1c442618ff2d74f558d56c602f6fde2809..c84c668c7e1a81281af894f26e05fb45afb85fbf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter_unittest.cc
>@@ -15,6 +15,7 @@
> #include "absl/memory/memory.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/ipaddress.h"
>+#include "rtc_base/messagedigest.h"
> #include "rtc_base/socketstream.h"
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/sslidentity.h"
>@@ -266,7 +267,7 @@ class SSLAdapterTestDummyServer : public sigslot::has_slots<> {
>     // (e.g. a WebRTC-based application and an RFC 5766 TURN server), where
>     // clients are not required to provide a certificate during handshake.
>     // Accordingly, we must disable client authentication here.
>-    ssl_stream_adapter_->set_client_auth_enabled(false);
>+    ssl_stream_adapter_->SetClientAuthEnabledForTesting(false);
> 
>     ssl_stream_adapter_->SetIdentity(ssl_identity_->GetReference());
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.cc
>index 9a38fc0384b704942a0f28e87a1370297bc29cd5..934848fb730457b3978f91a172cded67751eeb28 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.cc
>@@ -10,13 +10,12 @@
> 
> #include "rtc_base/sslcertificate.h"
> 
>-#include <ctime>
>+#include <algorithm>
> #include <string>
> #include <utility>
> 
> #include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
> #include "rtc_base/opensslcertificate.h"
> #include "rtc_base/sslfingerprint.h"
> #include "rtc_base/third_party/base64/base64.h"
>@@ -31,7 +30,7 @@ SSLCertificateStats::SSLCertificateStats(
>     std::string&& fingerprint,
>     std::string&& fingerprint_algorithm,
>     std::string&& base64_certificate,
>-    std::unique_ptr<SSLCertificateStats>&& issuer)
>+    std::unique_ptr<SSLCertificateStats> issuer)
>     : fingerprint(std::move(fingerprint)),
>       fingerprint_algorithm(std::move(fingerprint_algorithm)),
>       base64_certificate(std::move(base64_certificate)),
>@@ -55,8 +54,8 @@ std::unique_ptr<SSLCertificateStats> SSLCertificate::GetStats() const {
>   // |SSLCertificate::GetSignatureDigestAlgorithm| is not supported by the
>   // implementation of |SSLCertificate::ComputeDigest|. This currently happens
>   // with MD5- and SHA-224-signed certificates when linked to libNSS.
>-  std::unique_ptr<SSLFingerprint> ssl_fingerprint(
>-      SSLFingerprint::Create(digest_algorithm, this));
>+  std::unique_ptr<SSLFingerprint> ssl_fingerprint =
>+      SSLFingerprint::Create(digest_algorithm, *this);
>   if (!ssl_fingerprint)
>     return nullptr;
>   std::string fingerprint = ssl_fingerprint->GetRfc4572Fingerprint();
>@@ -71,49 +70,30 @@ std::unique_ptr<SSLCertificateStats> SSLCertificate::GetStats() const {
>                                                 std::move(der_base64), nullptr);
> }
> 
>-std::unique_ptr<SSLCertificate> SSLCertificate::GetUniqueReference() const {
>-  return absl::WrapUnique(GetReference());
>-}
>-
> //////////////////////////////////////////////////////////////////////
> // SSLCertChain
> //////////////////////////////////////////////////////////////////////
> 
>-SSLCertChain::SSLCertChain(std::vector<std::unique_ptr<SSLCertificate>> certs)
>-    : certs_(std::move(certs)) {}
>-
>-SSLCertChain::SSLCertChain(const std::vector<SSLCertificate*>& certs) {
>-  RTC_DCHECK(!certs.empty());
>-  certs_.resize(certs.size());
>-  std::transform(
>-      certs.begin(), certs.end(), certs_.begin(),
>-      [](const SSLCertificate* cert) -> std::unique_ptr<SSLCertificate> {
>-        return cert->GetUniqueReference();
>-      });
>+SSLCertChain::SSLCertChain(std::unique_ptr<SSLCertificate> single_cert) {
>+  certs_.push_back(std::move(single_cert));
> }
> 
>-SSLCertChain::SSLCertChain(const SSLCertificate* cert) {
>-  certs_.push_back(cert->GetUniqueReference());
>-}
>+SSLCertChain::SSLCertChain(std::vector<std::unique_ptr<SSLCertificate>> certs)
>+    : certs_(std::move(certs)) {}
> 
> SSLCertChain::SSLCertChain(SSLCertChain&& rhs) = default;
> 
> SSLCertChain& SSLCertChain::operator=(SSLCertChain&&) = default;
> 
>-SSLCertChain::~SSLCertChain() {}
>+SSLCertChain::~SSLCertChain() = default;
> 
>-SSLCertChain* SSLCertChain::Copy() const {
>+std::unique_ptr<SSLCertChain> SSLCertChain::Clone() const {
>   std::vector<std::unique_ptr<SSLCertificate>> new_certs(certs_.size());
>-  std::transform(certs_.begin(), certs_.end(), new_certs.begin(),
>-                 [](const std::unique_ptr<SSLCertificate>& cert)
>-                     -> std::unique_ptr<SSLCertificate> {
>-                   return cert->GetUniqueReference();
>-                 });
>-  return new SSLCertChain(std::move(new_certs));
>-}
>-
>-std::unique_ptr<SSLCertChain> SSLCertChain::UniqueCopy() const {
>-  return absl::WrapUnique(Copy());
>+  std::transform(
>+      certs_.begin(), certs_.end(), new_certs.begin(),
>+      [](const std::unique_ptr<SSLCertificate>& cert)
>+          -> std::unique_ptr<SSLCertificate> { return cert->Clone(); });
>+  return absl::make_unique<SSLCertChain>(std::move(new_certs));
> }
> 
> std::unique_ptr<SSLCertificateStats> SSLCertChain::GetStats() const {
>@@ -135,7 +115,8 @@ std::unique_ptr<SSLCertificateStats> SSLCertChain::GetStats() const {
> }
> 
> // static
>-SSLCertificate* SSLCertificate::FromPEMString(const std::string& pem_string) {
>+std::unique_ptr<SSLCertificate> SSLCertificate::FromPEMString(
>+    const std::string& pem_string) {
>   return OpenSSLCertificate::FromPEMString(pem_string);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.h
>index 29c4db58ef56c80035e832cd242cad72e9dc8dd4..eb81c209e4f0c3265664e015d7a733c4480600e3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.h
>@@ -15,15 +15,14 @@
> #ifndef RTC_BASE_SSLCERTIFICATE_H_
> #define RTC_BASE_SSLCERTIFICATE_H_
> 
>-#include <algorithm>
>+#include <stddef.h>
>+#include <stdint.h>
> #include <memory>
> #include <string>
> #include <vector>
> 
> #include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/messagedigest.h"
>-#include "rtc_base/timeutils.h"
> 
> namespace rtc {
> 
>@@ -31,7 +30,7 @@ struct SSLCertificateStats {
>   SSLCertificateStats(std::string&& fingerprint,
>                       std::string&& fingerprint_algorithm,
>                       std::string&& base64_certificate,
>-                      std::unique_ptr<SSLCertificateStats>&& issuer);
>+                      std::unique_ptr<SSLCertificateStats> issuer);
>   ~SSLCertificateStats();
>   std::string fingerprint;
>   std::string fingerprint_algorithm;
>@@ -54,17 +53,13 @@ class SSLCertificate {
>   // The length of the string representation of the certificate is
>   // stored in *pem_length if it is non-null, and only if
>   // parsing was successful.
>-  // Caller is responsible for freeing the returned object.
>-  static SSLCertificate* FromPEMString(const std::string& pem_string);
>-  virtual ~SSLCertificate() {}
>+  static std::unique_ptr<SSLCertificate> FromPEMString(
>+      const std::string& pem_string);
>+  virtual ~SSLCertificate() = default;
> 
>   // Returns a new SSLCertificate object instance wrapping the same
>-  // underlying certificate, including its chain if present.  Caller is
>-  // responsible for freeing the returned object. Use GetUniqueReference
>-  // instead.
>-  virtual SSLCertificate* GetReference() const = 0;
>-
>-  std::unique_ptr<SSLCertificate> GetUniqueReference() const;
>+  // underlying certificate, including its chain if present.
>+  virtual std::unique_ptr<SSLCertificate> Clone() const = 0;
> 
>   // Returns a PEM encoded string representation of the certificate.
>   virtual std::string ToPEMString() const = 0;
>@@ -95,13 +90,10 @@ class SSLCertificate {
> // SSLCertChain is a simple wrapper for a vector of SSLCertificates. It serves
> // primarily to ensure proper memory management (especially deletion) of the
> // SSLCertificate pointers.
>-class SSLCertChain {
>+class SSLCertChain final {
>  public:
>+  explicit SSLCertChain(std::unique_ptr<SSLCertificate> single_cert);
>   explicit SSLCertChain(std::vector<std::unique_ptr<SSLCertificate>> certs);
>-  // These constructors copy the provided SSLCertificate(s), so the caller
>-  // retains ownership.
>-  explicit SSLCertChain(const std::vector<SSLCertificate*>& certs);
>-  explicit SSLCertChain(const SSLCertificate* cert);
>   // Allow move semantics for the object.
>   SSLCertChain(SSLCertChain&&);
>   SSLCertChain& operator=(SSLCertChain&&);
>@@ -115,10 +107,8 @@ class SSLCertChain {
>   const SSLCertificate& Get(size_t pos) const { return *(certs_[pos]); }
> 
>   // Returns a new SSLCertChain object instance wrapping the same underlying
>-  // certificate chain.  Caller is responsible for freeing the returned object.
>-  SSLCertChain* Copy() const;
>-  // Same as above, but returning a unique_ptr for convenience.
>-  std::unique_ptr<SSLCertChain> UniqueCopy() const;
>+  // certificate chain.
>+  std::unique_ptr<SSLCertChain> Clone() const;
> 
>   // Gets information (fingerprint, etc.) about this certificate chain. This is
>   // used for certificate stats, see
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.cc
>index b651a3d6285205b55a2fb03677104b15a97b7662..b296d33ddd9c43123a4154b94b5f0f3a864a81e3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.cc
>@@ -11,67 +11,82 @@
> #include "rtc_base/sslfingerprint.h"
> 
> #include <ctype.h>
>+#include <algorithm>
>+#include <cstdint>
> #include <string>
> 
>-#include "rtc_base/helpers.h"
>+#include "absl/memory/memory.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/messagedigest.h"
>+#include "rtc_base/rtccertificate.h"
>+#include "rtc_base/sslcertificate.h"
>+#include "rtc_base/sslidentity.h"
> #include "rtc_base/stringencode.h"
> 
> namespace rtc {
> 
> SSLFingerprint* SSLFingerprint::Create(const std::string& algorithm,
>                                        const rtc::SSLIdentity* identity) {
>-  if (!identity) {
>-    return nullptr;
>-  }
>+  return CreateUnique(algorithm, *identity).release();
>+}
> 
>-  return Create(algorithm, &(identity->certificate()));
>+std::unique_ptr<SSLFingerprint> SSLFingerprint::CreateUnique(
>+    const std::string& algorithm,
>+    const rtc::SSLIdentity& identity) {
>+  return Create(algorithm, identity.certificate());
> }
> 
>-SSLFingerprint* SSLFingerprint::Create(const std::string& algorithm,
>-                                       const rtc::SSLCertificate* cert) {
>+std::unique_ptr<SSLFingerprint> SSLFingerprint::Create(
>+    const std::string& algorithm,
>+    const rtc::SSLCertificate& cert) {
>   uint8_t digest_val[64];
>   size_t digest_len;
>-  bool ret = cert->ComputeDigest(algorithm, digest_val, sizeof(digest_val),
>-                                 &digest_len);
>+  bool ret = cert.ComputeDigest(algorithm, digest_val, sizeof(digest_val),
>+                                &digest_len);
>   if (!ret) {
>     return nullptr;
>   }
>-
>-  return new SSLFingerprint(algorithm, digest_val, digest_len);
>+  return absl::make_unique<SSLFingerprint>(
>+      algorithm, ArrayView<const uint8_t>(digest_val, digest_len));
> }
> 
> SSLFingerprint* SSLFingerprint::CreateFromRfc4572(
>     const std::string& algorithm,
>     const std::string& fingerprint) {
>+  return CreateUniqueFromRfc4572(algorithm, fingerprint).release();
>+}
>+
>+std::unique_ptr<SSLFingerprint> SSLFingerprint::CreateUniqueFromRfc4572(
>+    const std::string& algorithm,
>+    const std::string& fingerprint) {
>   if (algorithm.empty() || !rtc::IsFips180DigestAlgorithm(algorithm))
>     return nullptr;
> 
>   if (fingerprint.empty())
>     return nullptr;
> 
>-  size_t value_len;
>   char value[rtc::MessageDigest::kMaxSize];
>-  value_len = rtc::hex_decode_with_delimiter(
>+  size_t value_len = rtc::hex_decode_with_delimiter(
>       value, sizeof(value), fingerprint.c_str(), fingerprint.length(), ':');
>   if (!value_len)
>     return nullptr;
> 
>-  return new SSLFingerprint(algorithm, reinterpret_cast<uint8_t*>(value),
>-                            value_len);
>+  return absl::make_unique<SSLFingerprint>(
>+      algorithm,
>+      ArrayView<const uint8_t>(reinterpret_cast<uint8_t*>(value), value_len));
> }
> 
>-SSLFingerprint* SSLFingerprint::CreateFromCertificate(
>-    const RTCCertificate* cert) {
>+std::unique_ptr<SSLFingerprint> SSLFingerprint::CreateFromCertificate(
>+    const RTCCertificate& cert) {
>   std::string digest_alg;
>-  if (!cert->ssl_certificate().GetSignatureDigestAlgorithm(&digest_alg)) {
>+  if (!cert.GetSSLCertificate().GetSignatureDigestAlgorithm(&digest_alg)) {
>     RTC_LOG(LS_ERROR)
>         << "Failed to retrieve the certificate's digest algorithm";
>     return nullptr;
>   }
> 
>-  SSLFingerprint* fingerprint = Create(digest_alg, cert->identity());
>+  std::unique_ptr<SSLFingerprint> fingerprint =
>+      CreateUnique(digest_alg, *cert.identity());
>   if (!fingerprint) {
>     RTC_LOG(LS_ERROR) << "Failed to create identity fingerprint, alg="
>                       << digest_alg;
>@@ -79,12 +94,14 @@ SSLFingerprint* SSLFingerprint::CreateFromCertificate(
>   return fingerprint;
> }
> 
>+SSLFingerprint::SSLFingerprint(const std::string& algorithm,
>+                               ArrayView<const uint8_t> digest_view)
>+    : algorithm(algorithm), digest(digest_view.data(), digest_view.size()) {}
>+
> SSLFingerprint::SSLFingerprint(const std::string& algorithm,
>                                const uint8_t* digest_in,
>                                size_t digest_len)
>-    : algorithm(algorithm) {
>-  digest.SetData(digest_in, digest_len);
>-}
>+    : SSLFingerprint(algorithm, MakeArrayView(digest_in, digest_len)) {}
> 
> SSLFingerprint::SSLFingerprint(const SSLFingerprint& from)
>     : algorithm(from.algorithm), digest(from.digest) {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.h
>index b204bc77f300211c73f4c47c216f38b3b91eeed9..ea10ede7206ba6f3691f97f3907bb02fae295998 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.h
>@@ -11,30 +11,47 @@
> #ifndef RTC_BASE_SSLFINGERPRINT_H_
> #define RTC_BASE_SSLFINGERPRINT_H_
> 
>+#include <stddef.h>
>+#include <stdint.h>
> #include <string>
> 
> #include "rtc_base/copyonwritebuffer.h"
>-#include "rtc_base/rtccertificate.h"
>-#include "rtc_base/sslidentity.h"
> 
> namespace rtc {
> 
>+class RTCCertificate;
> class SSLCertificate;
>+class SSLIdentity;
> 
> struct SSLFingerprint {
>+  // TODO(steveanton): Remove once downstream projects have moved off of this.
>   static SSLFingerprint* Create(const std::string& algorithm,
>                                 const rtc::SSLIdentity* identity);
>+  // TODO(steveanton): Rename to Create once projects have migrated.
>+  static std::unique_ptr<SSLFingerprint> CreateUnique(
>+      const std::string& algorithm,
>+      const rtc::SSLIdentity& identity);
> 
>-  static SSLFingerprint* Create(const std::string& algorithm,
>-                                const rtc::SSLCertificate* cert);
>+  static std::unique_ptr<SSLFingerprint> Create(
>+      const std::string& algorithm,
>+      const rtc::SSLCertificate& cert);
> 
>+  // TODO(steveanton): Remove once downstream projects have moved off of this.
>   static SSLFingerprint* CreateFromRfc4572(const std::string& algorithm,
>                                            const std::string& fingerprint);
>+  // TODO(steveanton): Rename to CreateFromRfc4572 once projects have migrated.
>+  static std::unique_ptr<SSLFingerprint> CreateUniqueFromRfc4572(
>+      const std::string& algorithm,
>+      const std::string& fingerprint);
> 
>   // Creates a fingerprint from a certificate, using the same digest algorithm
>   // as the certificate's signature.
>-  static SSLFingerprint* CreateFromCertificate(const RTCCertificate* cert);
>+  static std::unique_ptr<SSLFingerprint> CreateFromCertificate(
>+      const RTCCertificate& cert);
> 
>+  SSLFingerprint(const std::string& algorithm,
>+                 ArrayView<const uint8_t> digest_view);
>+  // TODO(steveanton): Remove once downstream projects have moved off of this.
>   SSLFingerprint(const std::string& algorithm,
>                  const uint8_t* digest_in,
>                  size_t digest_len);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.cc
>index 823fc388a3cef1b8150f60cf698b8736a1642b37..41eb35d5f52b6737e71b9336140eb73cc9771a2e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.cc
>@@ -11,20 +11,95 @@
> // Handling of certificates and keypairs for SSLStreamAdapter's peer mode.
> #include "rtc_base/sslidentity.h"
> 
>-#include <ctime>
>+#include <string.h>
>+#include <time.h>
> #include <string>
>-#include <utility>
> 
>-#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
> #include "rtc_base/opensslidentity.h"
>-#include "rtc_base/sslfingerprint.h"
>+#include "rtc_base/sslcertificate.h"
> #include "rtc_base/strings/string_builder.h"
> #include "rtc_base/third_party/base64/base64.h"
>+#include "rtc_base/timeutils.h"
> 
> namespace rtc {
> 
>+//////////////////////////////////////////////////////////////////////
>+// Helper Functions
>+//////////////////////////////////////////////////////////////////////
>+
>+namespace {
>+// Read |n| bytes from ASN1 number string at *|pp| and return the numeric value.
>+// Update *|pp| and *|np| to reflect number of read bytes.
>+// TODO(bugs.webrtc.org/9860) - Remove this code.
>+inline int ASN1ReadInt(const unsigned char** pp, size_t* np, size_t n) {
>+  const unsigned char* p = *pp;
>+  int x = 0;
>+  for (size_t i = 0; i < n; i++) {
>+    x = 10 * x + p[i] - '0';
>+  }
>+  *pp = p + n;
>+  *np = *np - n;
>+  return x;
>+}
>+
>+}  // namespace
>+
>+// TODO(bugs.webrtc.org/9860) - Remove this code.
>+int64_t ASN1TimeToSec(const unsigned char* s, size_t length, bool long_format) {
>+  size_t bytes_left = length;
>+  // Make sure the string ends with Z.  Doing it here protects the strspn call
>+  // from running off the end of the string in Z's absense.
>+  if (length == 0 || s[length - 1] != 'Z') {
>+    return -1;
>+  }
>+  // Make sure we only have ASCII digits so that we don't need to clutter the
>+  // code below and ASN1ReadInt with error checking.
>+  size_t n = strspn(reinterpret_cast<const char*>(s), "0123456789");
>+  if (n + 1 != length) {
>+    return -1;
>+  }
>+  // Read out ASN1 year, in either 2-char "UTCTIME" or 4-char "GENERALIZEDTIME"
>+  // format.  Both format use UTC in this context.
>+  int year = 0;
>+  if (long_format) {
>+    // ASN1 format: yyyymmddhh[mm[ss[.fff]]]Z where the Z is literal, but
>+    // RFC 5280 requires us to only support exactly yyyymmddhhmmssZ.
>+    if (bytes_left < 11) {
>+      return -1;
>+    }
>+    year = ASN1ReadInt(&s, &bytes_left, 4);
>+    year -= 1900;
>+  } else {
>+    // ASN1 format: yymmddhhmm[ss]Z where the Z is literal, but RFC 5280
>+    // requires us to only support exactly yymmddhhmmssZ.
>+    if (bytes_left < 9) {
>+      return -1;
>+    }
>+    year = ASN1ReadInt(&s, &bytes_left, 2);
>+    // Per RFC 5280 4.1.2.5.1
>+    if (year < 50) {
>+      year += 100;
>+    }
>+  }
>+
>+  // Read out remaining ASN1 time data and store it in |tm| in documented
>+  // std::tm format.
>+  tm tm;
>+  tm.tm_year = year;
>+  tm.tm_mon = ASN1ReadInt(&s, &bytes_left, 2) - 1;
>+  tm.tm_mday = ASN1ReadInt(&s, &bytes_left, 2);
>+  tm.tm_hour = ASN1ReadInt(&s, &bytes_left, 2);
>+  tm.tm_min = ASN1ReadInt(&s, &bytes_left, 2);
>+  tm.tm_sec = ASN1ReadInt(&s, &bytes_left, 2);
>+
>+  // Now just Z should remain.  Its existence was asserted above.
>+  if (bytes_left != 1) {
>+    return -1;
>+  }
>+  return TmToSeconds(tm);
>+}
>+
> //////////////////////////////////////////////////////////////////////
> // KeyParams
> //////////////////////////////////////////////////////////////////////
>@@ -93,22 +168,21 @@ KeyType IntKeyTypeFamilyToKeyType(int key_type_family) {
> bool SSLIdentity::PemToDer(const std::string& pem_type,
>                            const std::string& pem_string,
>                            std::string* der) {
>-  // Find the inner body. We need this to fulfill the contract of
>-  // returning pem_length.
>+  // Find the inner body. We need this to fulfill the contract of returning
>+  // pem_length.
>   size_t header = pem_string.find("-----BEGIN " + pem_type + "-----");
>-  if (header == std::string::npos)
>+  if (header == std::string::npos) {
>     return false;
>-
>+  }
>   size_t body = pem_string.find("\n", header);
>-  if (body == std::string::npos)
>+  if (body == std::string::npos) {
>     return false;
>-
>+  }
>   size_t trailer = pem_string.find("-----END " + pem_type + "-----");
>-  if (trailer == std::string::npos)
>+  if (trailer == std::string::npos) {
>     return false;
>-
>+  }
>   std::string inner = pem_string.substr(body + 1, trailer - (body + 1));
>-
>   *der = Base64::Decode(inner, Base64::DO_PARSE_WHITE | Base64::DO_PAD_ANY |
>                                    Base64::DO_TERM_BUFFER);
>   return true;
>@@ -118,14 +192,12 @@ std::string SSLIdentity::DerToPem(const std::string& pem_type,
>                                   const unsigned char* data,
>                                   size_t length) {
>   rtc::StringBuilder result;
>-
>   result << "-----BEGIN " << pem_type << "-----\n";
> 
>   std::string b64_encoded;
>   Base64::EncodeFromArray(data, length, &b64_encoded);
>-
>-  // Divide the Base-64 encoded data into 64-character chunks, as per
>-  // 4.3.2.4 of RFC 1421.
>+  // Divide the Base-64 encoded data into 64-character chunks, as per 4.3.2.4
>+  // of RFC 1421.
>   static const size_t kChunkSize = 64;
>   size_t chunks = (b64_encoded.size() + (kChunkSize - 1)) / kChunkSize;
>   for (size_t i = 0, chunk_offset = 0; i < chunks;
>@@ -133,9 +205,7 @@ std::string SSLIdentity::DerToPem(const std::string& pem_type,
>     result << b64_encoded.substr(chunk_offset, kChunkSize);
>     result << "\n";
>   }
>-
>   result << "-----END " << pem_type << "-----\n";
>-
>   return result.Release();
> }
> 
>@@ -186,78 +256,4 @@ bool operator!=(const SSLIdentity& a, const SSLIdentity& b) {
>   return !(a == b);
> }
> 
>-//////////////////////////////////////////////////////////////////////
>-// Helper Functions
>-//////////////////////////////////////////////////////////////////////
>-
>-// Read |n| bytes from ASN1 number string at *|pp| and return the numeric value.
>-// Update *|pp| and *|np| to reflect number of read bytes.
>-static inline int ASN1ReadInt(const unsigned char** pp, size_t* np, size_t n) {
>-  const unsigned char* p = *pp;
>-  int x = 0;
>-  for (size_t i = 0; i < n; i++)
>-    x = 10 * x + p[i] - '0';
>-  *pp = p + n;
>-  *np = *np - n;
>-  return x;
>-}
>-
>-int64_t ASN1TimeToSec(const unsigned char* s, size_t length, bool long_format) {
>-  size_t bytes_left = length;
>-
>-  // Make sure the string ends with Z.  Doing it here protects the strspn call
>-  // from running off the end of the string in Z's absense.
>-  if (length == 0 || s[length - 1] != 'Z')
>-    return -1;
>-
>-  // Make sure we only have ASCII digits so that we don't need to clutter the
>-  // code below and ASN1ReadInt with error checking.
>-  size_t n = strspn(reinterpret_cast<const char*>(s), "0123456789");
>-  if (n + 1 != length)
>-    return -1;
>-
>-  int year;
>-
>-  // Read out ASN1 year, in either 2-char "UTCTIME" or 4-char "GENERALIZEDTIME"
>-  // format.  Both format use UTC in this context.
>-  if (long_format) {
>-    // ASN1 format: yyyymmddhh[mm[ss[.fff]]]Z where the Z is literal, but
>-    // RFC 5280 requires us to only support exactly yyyymmddhhmmssZ.
>-
>-    if (bytes_left < 11)
>-      return -1;
>-
>-    year = ASN1ReadInt(&s, &bytes_left, 4);
>-    year -= 1900;
>-  } else {
>-    // ASN1 format: yymmddhhmm[ss]Z where the Z is literal, but RFC 5280
>-    // requires us to only support exactly yymmddhhmmssZ.
>-
>-    if (bytes_left < 9)
>-      return -1;
>-
>-    year = ASN1ReadInt(&s, &bytes_left, 2);
>-    if (year < 50)  // Per RFC 5280 4.1.2.5.1
>-      year += 100;
>-  }
>-
>-  std::tm tm;
>-  tm.tm_year = year;
>-
>-  // Read out remaining ASN1 time data and store it in |tm| in documented
>-  // std::tm format.
>-  tm.tm_mon = ASN1ReadInt(&s, &bytes_left, 2) - 1;
>-  tm.tm_mday = ASN1ReadInt(&s, &bytes_left, 2);
>-  tm.tm_hour = ASN1ReadInt(&s, &bytes_left, 2);
>-  tm.tm_min = ASN1ReadInt(&s, &bytes_left, 2);
>-  tm.tm_sec = ASN1ReadInt(&s, &bytes_left, 2);
>-
>-  if (bytes_left != 1) {
>-    // Now just Z should remain.  Its existence was asserted above.
>-    return -1;
>-  }
>-
>-  return TmToSeconds(tm);
>-}
>-
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.h
>index 1379d733be5f4f1a2f877c64f5d3fba5fe79ffb5..39feeabab312c14054d2653a53ac243f57ee02b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.h
>@@ -13,19 +13,15 @@
> #ifndef RTC_BASE_SSLIDENTITY_H_
> #define RTC_BASE_SSLIDENTITY_H_
> 
>-#include <algorithm>
>-#include <memory>
>+#include <stdint.h>
>+#include <ctime>
> #include <string>
>-#include <vector>
>-
>-#include "rtc_base/buffer.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/messagedigest.h"
>-#include "rtc_base/sslcertificate.h"
>-#include "rtc_base/timeutils.h"
> 
> namespace rtc {
> 
>+class SSLCertChain;
>+class SSLCertificate;
>+
> // KT_LAST is intended for vector declarations and loops over all key types;
> // it does not represent any key type in itself.
> // KT_DEFAULT is used as the default KeyType for KeyParams.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity_unittest.cc
>index 132e240c2f5a5c0dc48c77ac88e75e86a3ac28c0..ba53d17da431a2acf2ab6b18f41c8ed616514dd2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity_unittest.cc
>@@ -14,6 +14,7 @@
> #include "rtc_base/fakesslidentity.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/helpers.h"
>+#include "rtc_base/messagedigest.h"
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/sslfingerprint.h"
> #include "rtc_base/sslidentity.h"
>@@ -179,7 +180,7 @@ IdentityAndInfo CreateFakeIdentityAndInfoFromDers(
>   const rtc::SSLCertChain& chain = info.identity->cert_chain();
>   std::unique_ptr<rtc::SSLFingerprint> fp;
>   for (size_t i = 0; i < chain.GetSize(); i++) {
>-    fp.reset(rtc::SSLFingerprint::Create("sha-1", &chain.Get(i)));
>+    fp = rtc::SSLFingerprint::Create("sha-1", chain.Get(i));
>     EXPECT_TRUE(fp);
>     info.fingerprints.push_back(fp->GetRfc4572Fingerprint());
>   }
>@@ -200,7 +201,7 @@ class SSLIdentityTest : public testing::Test {
>     ASSERT_TRUE(identity_ecdsa1_);
>     ASSERT_TRUE(identity_ecdsa2_);
> 
>-    test_cert_.reset(rtc::SSLCertificate::FromPEMString(kTestCertificate));
>+    test_cert_ = rtc::SSLCertificate::FromPEMString(kTestCertificate);
>     ASSERT_TRUE(test_cert_);
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.cc
>index 746ebd50063dca8bfae68985dbdb6eed7f532e0e..9c33a9c933c98a904049fc86bcf0110a77dfb772 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.cc
>@@ -89,40 +89,12 @@ bool IsGcmCryptoSuiteName(const std::string& crypto_suite) {
>           crypto_suite == CS_AEAD_AES_128_GCM);
> }
> 
>-// static
>-CryptoOptions CryptoOptions::NoGcm() {
>-  CryptoOptions options;
>-  options.enable_gcm_crypto_suites = false;
>-  return options;
>-}
>-
>-std::vector<int> GetSupportedDtlsSrtpCryptoSuites(
>-    const rtc::CryptoOptions& crypto_options) {
>-  std::vector<int> crypto_suites;
>-  if (crypto_options.enable_gcm_crypto_suites) {
>-    crypto_suites.push_back(rtc::SRTP_AEAD_AES_256_GCM);
>-    crypto_suites.push_back(rtc::SRTP_AEAD_AES_128_GCM);
>-  }
>-  // Note: SRTP_AES128_CM_SHA1_80 is what is required to be supported (by
>-  // draft-ietf-rtcweb-security-arch), but SRTP_AES128_CM_SHA1_32 is allowed as
>-  // well, and saves a few bytes per packet if it ends up selected.
>-  // As the cipher suite is potentially insecure, it will only be used if
>-  // enabled by both peers.
>-  if (crypto_options.enable_aes128_sha1_32_crypto_cipher) {
>-    crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_32);
>-  }
>-  crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_80);
>-  return crypto_suites;
>-}
>-
> SSLStreamAdapter* SSLStreamAdapter::Create(StreamInterface* stream) {
>   return new OpenSSLStreamAdapter(stream);
> }
> 
> SSLStreamAdapter::SSLStreamAdapter(StreamInterface* stream)
>-    : StreamAdapterInterface(stream),
>-      ignore_bad_cert_(false),
>-      client_auth_enabled_(true) {}
>+    : StreamAdapterInterface(stream) {}
> 
> SSLStreamAdapter::~SSLStreamAdapter() {}
> 
>@@ -161,8 +133,13 @@ bool SSLStreamAdapter::IsAcceptableCipher(const std::string& cipher,
> std::string SSLStreamAdapter::SslCipherSuiteToName(int cipher_suite) {
>   return OpenSSLStreamAdapter::SslCipherSuiteToName(cipher_suite);
> }
>-void SSLStreamAdapter::enable_time_callback_for_testing() {
>-  OpenSSLStreamAdapter::enable_time_callback_for_testing();
>+
>+///////////////////////////////////////////////////////////////////////////////
>+// Test only settings
>+///////////////////////////////////////////////////////////////////////////////
>+
>+void SSLStreamAdapter::EnableTimeCallbackForTesting() {
>+  OpenSSLStreamAdapter::EnableTimeCallbackForTesting();
> }
> 
> ///////////////////////////////////////////////////////////////////////////////
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.h
>index 2d4e19f9b50fef30c08cf063adf65464133ffc85..25f4f33486dadd984c8a42be1b31d3f2aa9ab583 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.h
>@@ -70,34 +70,6 @@ bool IsGcmCryptoSuite(int crypto_suite);
> // Returns true if the given crypto suite name uses a GCM cipher.
> bool IsGcmCryptoSuiteName(const std::string& crypto_suite);
> 
>-struct CryptoOptions {
>-  CryptoOptions() {}
>-
>-  // Helper method to return an instance of the CryptoOptions with GCM crypto
>-  // suites disabled. This method should be used instead of depending on current
>-  // default values set by the constructor.
>-  static CryptoOptions NoGcm();
>-
>-  // Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
>-  // if both sides enable it.
>-  bool enable_gcm_crypto_suites = false;
>-
>-  // If set to true, the (potentially insecure) crypto cipher
>-  // SRTP_AES128_CM_SHA1_32 will be included in the list of supported ciphers
>-  // during negotiation. It will only be used if both peers support it and no
>-  // other ciphers get preferred.
>-  bool enable_aes128_sha1_32_crypto_cipher = false;
>-
>-  // If set to true, encrypted RTP header extensions as defined in RFC 6904
>-  // will be negotiated. They will only be used if both peers support them.
>-  bool enable_encrypted_rtp_header_extensions = false;
>-};
>-
>-// Returns supported crypto suites, given |crypto_options|.
>-// CS_AES_CM_128_HMAC_SHA1_32 will be preferred by default.
>-std::vector<int> GetSupportedDtlsSrtpCryptoSuites(
>-    const rtc::CryptoOptions& crypto_options);
>-
> // SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS.
> // After SSL has been started, the stream will only open on successful
> // SSL verification of certificates, and the communication is
>@@ -144,12 +116,6 @@ class SSLStreamAdapter : public StreamAdapterInterface {
>   explicit SSLStreamAdapter(StreamInterface* stream);
>   ~SSLStreamAdapter() override;
> 
>-  void set_ignore_bad_cert(bool ignore) { ignore_bad_cert_ = ignore; }
>-  bool ignore_bad_cert() const { return ignore_bad_cert_; }
>-
>-  void set_client_auth_enabled(bool enabled) { client_auth_enabled_ = enabled; }
>-  bool client_auth_enabled() const { return client_auth_enabled_; }
>-
>   // Specify our SSL identity: key and certificate. SSLStream takes ownership
>   // of the SSLIdentity object and will free it when appropriate. Should be
>   // called no more than once on a given SSLStream instance.
>@@ -263,22 +229,32 @@ class SSLStreamAdapter : public StreamAdapterInterface {
>   // depending on specific SSL implementation.
>   static std::string SslCipherSuiteToName(int cipher_suite);
> 
>+  ////////////////////////////////////////////////////////////////////////////
>+  // Testing only member functions
>+  ////////////////////////////////////////////////////////////////////////////
>+
>   // Use our timeutils.h source of timing in BoringSSL, allowing us to test
>   // using a fake clock.
>-  static void enable_time_callback_for_testing();
>+  static void EnableTimeCallbackForTesting();
>+
>+  // Deprecated. Do not use this API outside of testing.
>+  // Do not set this to false outside of testing.
>+  void SetClientAuthEnabledForTesting(bool enabled) {
>+    client_auth_enabled_ = enabled;
>+  }
>+
>+  // Deprecated. Do not use this API outside of testing.
>+  // Returns true by default, else false if explicitly set to disable client
>+  // authentication.
>+  bool GetClientAuthEnabled() const { return client_auth_enabled_; }
> 
>   sigslot::signal1<SSLHandshakeError> SignalSSLHandshakeError;
> 
>  private:
>-  // If true, the server certificate need not match the configured
>-  // server_name, and in fact missing certificate authority and other
>-  // verification errors are ignored.
>-  bool ignore_bad_cert_;
>-
>   // If true (default), the client is required to provide a certificate during
>   // handshake. If no certificate is given, handshake fails. This applies to
>   // server mode only.
>-  bool client_auth_enabled_;
>+  bool client_auth_enabled_ = true;
> };
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter_unittest.cc
>index 3403bdbe727ee33f0862860c9f71aad1db6d9fe7..6fbb1d7c2cc09822d26ba67e9a70ea52c315a71a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter_unittest.cc
>@@ -17,6 +17,8 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/helpers.h"
>+#include "rtc_base/memory_stream.h"
>+#include "rtc_base/messagedigest.h"
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/sslidentity.h"
> #include "rtc_base/sslstreamadapter.h"
>@@ -587,8 +589,7 @@ class SSLStreamAdapterTestBase : public testing::Test,
>       chain = client_ssl_->GetPeerSSLCertChain();
>     else
>       chain = server_ssl_->GetPeerSSLCertChain();
>-    return (chain && chain->GetSize()) ? chain->Get(0).GetUniqueReference()
>-                                       : nullptr;
>+    return (chain && chain->GetSize()) ? chain->Get(0).Clone() : nullptr;
>   }
> 
>   bool GetSslCipherSuite(bool client, int* retval) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.cc
>index d1ec1400de60b1de3feeda9f779a009672f2190d..783625c67a94602511937ae40d66343aa856e3b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.cc
>@@ -7,29 +7,23 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>-
>-#if defined(WEBRTC_POSIX)
>-#include <sys/file.h>
>-#endif  // WEBRTC_POSIX
> #include <errno.h>
>+#include <string.h>
> #include <sys/stat.h>
>-#include <sys/types.h>
>-
> #include <algorithm>
> #include <string>
> 
> #include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>+#include "rtc_base/location.h"
> #include "rtc_base/messagequeue.h"
> #include "rtc_base/stream.h"
>-#include "rtc_base/stringencode.h"
>-#include "rtc_base/stringutils.h"
> #include "rtc_base/thread.h"
>-#include "rtc_base/timeutils.h"
> 
> #if defined(WEBRTC_WIN)
> #include <windows.h>
>+
> #define fileno _fileno
>+#include "rtc_base/stringutils.h"
> #endif
> 
> namespace rtc {
>@@ -75,26 +69,6 @@ StreamResult StreamInterface::ReadAll(void* buffer,
>   return result;
> }
> 
>-StreamResult StreamInterface::ReadLine(std::string* line) {
>-  line->clear();
>-  StreamResult result = SR_SUCCESS;
>-  while (true) {
>-    char ch;
>-    result = Read(&ch, sizeof(ch), nullptr, nullptr);
>-    if (result != SR_SUCCESS) {
>-      break;
>-    }
>-    if (ch == '\n') {
>-      break;
>-    }
>-    line->push_back(ch);
>-  }
>-  if (!line->empty()) {   // give back the line we've collected so far with
>-    result = SR_SUCCESS;  // a success code.  Otherwise return the last code
>-  }
>-  return result;
>-}
>-
> void StreamInterface::PostEvent(Thread* t, int events, int err) {
>   t->Post(RTC_FROM_HERE, this, MSG_POST_EVENT,
>           new StreamEventData(events, err));
>@@ -104,14 +78,6 @@ void StreamInterface::PostEvent(int events, int err) {
>   PostEvent(Thread::Current(), events, err);
> }
> 
>-const void* StreamInterface::GetReadData(size_t* data_len) {
>-  return nullptr;
>-}
>-
>-void* StreamInterface::GetWriteBuffer(size_t* buf_len) {
>-  return nullptr;
>-}
>-
> bool StreamInterface::SetPosition(size_t position) {
>   return false;
> }
>@@ -124,10 +90,6 @@ bool StreamInterface::GetSize(size_t* size) const {
>   return false;
> }
> 
>-bool StreamInterface::GetWriteRemaining(size_t* size) const {
>-  return false;
>-}
>-
> bool StreamInterface::Flush() {
>   return false;
> }
>@@ -188,10 +150,6 @@ bool StreamAdapterInterface::GetSize(size_t* size) const {
>   return stream_->GetSize(size);
> }
> 
>-bool StreamAdapterInterface::GetWriteRemaining(size_t* size) const {
>-  return stream_->GetWriteRemaining(size);
>-}
>-
> bool StreamAdapterInterface::ReserveSize(size_t size) {
>   return stream_->ReserveSize(size);
> }
>@@ -389,143 +347,6 @@ void FileStream::DoClose() {
>   fclose(file_);
> }
> 
>-///////////////////////////////////////////////////////////////////////////////
>-// MemoryStream
>-///////////////////////////////////////////////////////////////////////////////
>-
>-MemoryStreamBase::MemoryStreamBase()
>-    : buffer_(nullptr), buffer_length_(0), data_length_(0), seek_position_(0) {}
>-
>-StreamState MemoryStreamBase::GetState() const {
>-  return SS_OPEN;
>-}
>-
>-StreamResult MemoryStreamBase::Read(void* buffer,
>-                                    size_t bytes,
>-                                    size_t* bytes_read,
>-                                    int* error) {
>-  if (seek_position_ >= data_length_) {
>-    return SR_EOS;
>-  }
>-  size_t available = data_length_ - seek_position_;
>-  if (bytes > available) {
>-    // Read partial buffer
>-    bytes = available;
>-  }
>-  memcpy(buffer, &buffer_[seek_position_], bytes);
>-  seek_position_ += bytes;
>-  if (bytes_read) {
>-    *bytes_read = bytes;
>-  }
>-  return SR_SUCCESS;
>-}
>-
>-StreamResult MemoryStreamBase::Write(const void* buffer,
>-                                     size_t bytes,
>-                                     size_t* bytes_written,
>-                                     int* error) {
>-  size_t available = buffer_length_ - seek_position_;
>-  if (0 == available) {
>-    // Increase buffer size to the larger of:
>-    // a) new position rounded up to next 256 bytes
>-    // b) double the previous length
>-    size_t new_buffer_length =
>-        std::max(((seek_position_ + bytes) | 0xFF) + 1, buffer_length_ * 2);
>-    StreamResult result = DoReserve(new_buffer_length, error);
>-    if (SR_SUCCESS != result) {
>-      return result;
>-    }
>-    RTC_DCHECK(buffer_length_ >= new_buffer_length);
>-    available = buffer_length_ - seek_position_;
>-  }
>-
>-  if (bytes > available) {
>-    bytes = available;
>-  }
>-  memcpy(&buffer_[seek_position_], buffer, bytes);
>-  seek_position_ += bytes;
>-  if (data_length_ < seek_position_) {
>-    data_length_ = seek_position_;
>-  }
>-  if (bytes_written) {
>-    *bytes_written = bytes;
>-  }
>-  return SR_SUCCESS;
>-}
>-
>-void MemoryStreamBase::Close() {
>-  // nothing to do
>-}
>-
>-bool MemoryStreamBase::SetPosition(size_t position) {
>-  if (position > data_length_)
>-    return false;
>-  seek_position_ = position;
>-  return true;
>-}
>-
>-bool MemoryStreamBase::GetPosition(size_t* position) const {
>-  if (position)
>-    *position = seek_position_;
>-  return true;
>-}
>-
>-bool MemoryStreamBase::GetSize(size_t* size) const {
>-  if (size)
>-    *size = data_length_;
>-  return true;
>-}
>-
>-bool MemoryStreamBase::ReserveSize(size_t size) {
>-  return (SR_SUCCESS == DoReserve(size, nullptr));
>-}
>-
>-StreamResult MemoryStreamBase::DoReserve(size_t size, int* error) {
>-  return (buffer_length_ >= size) ? SR_SUCCESS : SR_EOS;
>-}
>-
>-///////////////////////////////////////////////////////////////////////////////
>-
>-MemoryStream::MemoryStream() {}
>-
>-MemoryStream::MemoryStream(const char* data) {
>-  SetData(data, strlen(data));
>-}
>-
>-MemoryStream::MemoryStream(const void* data, size_t length) {
>-  SetData(data, length);
>-}
>-
>-MemoryStream::~MemoryStream() {
>-  delete[] buffer_;
>-}
>-
>-void MemoryStream::SetData(const void* data, size_t length) {
>-  data_length_ = buffer_length_ = length;
>-  delete[] buffer_;
>-  buffer_ = new char[buffer_length_];
>-  memcpy(buffer_, data, data_length_);
>-  seek_position_ = 0;
>-}
>-
>-StreamResult MemoryStream::DoReserve(size_t size, int* error) {
>-  if (buffer_length_ >= size)
>-    return SR_SUCCESS;
>-
>-  if (char* new_buffer = new char[size]) {
>-    memcpy(new_buffer, buffer_, data_length_);
>-    delete[] buffer_;
>-    buffer_ = new_buffer;
>-    buffer_length_ = size;
>-    return SR_SUCCESS;
>-  }
>-
>-  if (error) {
>-    *error = ENOMEM;
>-  }
>-  return SR_ERROR;
>-}
>-
> ///////////////////////////////////////////////////////////////////////////////
> // FifoBuffer
> ///////////////////////////////////////////////////////////////////////////////
>@@ -758,68 +579,4 @@ StreamResult FifoBuffer::WriteOffsetLocked(const void* buffer,
>   return SR_SUCCESS;
> }
> 
>-
>-///////////////////////////////////////////////////////////////////////////////
>-
>-StreamResult Flow(StreamInterface* source,
>-                  char* buffer,
>-                  size_t buffer_len,
>-                  StreamInterface* sink,
>-                  size_t* data_len /* = nullptr */) {
>-  RTC_DCHECK(buffer_len > 0);
>-
>-  StreamResult result;
>-  size_t count, read_pos, write_pos;
>-  if (data_len) {
>-    read_pos = *data_len;
>-  } else {
>-    read_pos = 0;
>-  }
>-
>-  bool end_of_stream = false;
>-  do {
>-    // Read until buffer is full, end of stream, or error
>-    while (!end_of_stream && (read_pos < buffer_len)) {
>-      result = source->Read(buffer + read_pos, buffer_len - read_pos, &count,
>-                            nullptr);
>-      if (result == SR_EOS) {
>-        end_of_stream = true;
>-      } else if (result != SR_SUCCESS) {
>-        if (data_len) {
>-          *data_len = read_pos;
>-        }
>-        return result;
>-      } else {
>-        read_pos += count;
>-      }
>-    }
>-
>-    // Write until buffer is empty, or error (including end of stream)
>-    write_pos = 0;
>-    while (write_pos < read_pos) {
>-      result = sink->Write(buffer + write_pos, read_pos - write_pos, &count,
>-                           nullptr);
>-      if (result != SR_SUCCESS) {
>-        if (data_len) {
>-          *data_len = read_pos - write_pos;
>-          if (write_pos > 0) {
>-            memmove(buffer, buffer + write_pos, *data_len);
>-          }
>-        }
>-        return result;
>-      }
>-      write_pos += count;
>-    }
>-
>-    read_pos = 0;
>-  } while (!end_of_stream);
>-
>-  if (data_len) {
>-    *data_len = 0;
>-  }
>-  return SR_SUCCESS;
>-}
>-
>-///////////////////////////////////////////////////////////////////////////////
>-
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>index 81c5e41801a5baad651b42b5dadd8453b5123378..43e7f58bc7c0961371ca9ce1ed865c00bac0be52 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>@@ -18,7 +18,6 @@
> #include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>-#include "rtc_base/logging.h"
> #include "rtc_base/messagehandler.h"
> #include "rtc_base/messagequeue.h"
> #include "rtc_base/third_party/sigslot/sigslot.h"
>@@ -108,59 +107,6 @@ class StreamInterface : public MessageHandler {
>   // Like the aforementioned method, but posts to the current thread.
>   void PostEvent(int events, int err);
> 
>-  //
>-  // OPTIONAL OPERATIONS
>-  //
>-  // Not all implementations will support the following operations.  In general,
>-  // a stream will only support an operation if it reasonably efficient to do
>-  // so.  For example, while a socket could buffer incoming data to support
>-  // seeking, it will not do so.  Instead, a buffering stream adapter should
>-  // be used.
>-  //
>-  // Even though several of these operations are related, you should
>-  // always use whichever operation is most relevant.
>-  //
>-  // The following four methods are used to avoid copying data multiple times.
>-
>-  // GetReadData returns a pointer to a buffer which is owned by the stream.
>-  // The buffer contains data_len bytes.  null is returned if no data is
>-  // available, or if the method fails.  If the caller processes the data, it
>-  // must call ConsumeReadData with the number of processed bytes.  GetReadData
>-  // does not require a matching call to ConsumeReadData if the data is not
>-  // processed.  Read and ConsumeReadData invalidate the buffer returned by
>-  // GetReadData.
>-  virtual const void* GetReadData(size_t* data_len);
>-  virtual void ConsumeReadData(size_t used) {}
>-
>-  // GetWriteBuffer returns a pointer to a buffer which is owned by the stream.
>-  // The buffer has a capacity of buf_len bytes.  null is returned if there is
>-  // no buffer available, or if the method fails.  The call may write data to
>-  // the buffer, and then call ConsumeWriteBuffer with the number of bytes
>-  // written.  GetWriteBuffer does not require a matching call to
>-  // ConsumeWriteData if no data is written.  Write, ForceWrite, and
>-  // ConsumeWriteData invalidate the buffer returned by GetWriteBuffer.
>-  // TODO: Allow the caller to specify a minimum buffer size.  If the specified
>-  // amount of buffer is not yet available, return null and Signal SE_WRITE
>-  // when it is available.  If the requested amount is too large, return an
>-  // error.
>-  virtual void* GetWriteBuffer(size_t* buf_len);
>-  virtual void ConsumeWriteBuffer(size_t used) {}
>-
>-  // Write data_len bytes found in data, circumventing any throttling which
>-  // would could cause SR_BLOCK to be returned.  Returns true if all the data
>-  // was written.  Otherwise, the method is unsupported, or an unrecoverable
>-  // error occurred, and the error value is set.  This method should be used
>-  // sparingly to write critical data which should not be throttled.  A stream
>-  // which cannot circumvent its blocking constraints should not implement this
>-  // method.
>-  // NOTE: This interface is being considered experimentally at the moment.  It
>-  // would be used by JUDP and BandwidthStream as a way to circumvent certain
>-  // soft limits in writing.
>-  // virtual bool ForceWrite(const void* data, size_t data_len, int* error) {
>-  //  if (error) *error = -1;
>-  //  return false;
>-  //}
>-
>   // Seek to a byte offset from the beginning of the stream.  Returns false if
>   // the stream does not support seeking, or cannot seek to the specified
>   // position.
>@@ -174,10 +120,6 @@ class StreamInterface : public MessageHandler {
>   // is not known.
>   virtual bool GetSize(size_t* size) const;
> 
>-  // Return the number of Write()-able bytes remaining before end-of-stream.
>-  // Returns false if not known.
>-  virtual bool GetWriteRemaining(size_t* size) const;
>-
>   // Return true if flush is successful.
>   virtual bool Flush();
> 
>@@ -214,12 +156,6 @@ class StreamInterface : public MessageHandler {
>                        size_t* read,
>                        int* error);
> 
>-  // ReadLine is a helper function which repeatedly calls Read until it hits
>-  // the end-of-line character, or something other than SR_SUCCESS.
>-  // TODO: this is too inefficient to keep here.  Break this out into a buffered
>-  // readline object or adapter
>-  StreamResult ReadLine(std::string* line);
>-
>  protected:
>   StreamInterface();
> 
>@@ -255,37 +191,9 @@ class StreamAdapterInterface : public StreamInterface,
>                      int* error) override;
>   void Close() override;
> 
>-  // Optional Stream Interface
>-  /*  Note: Many stream adapters were implemented prior to this Read/Write
>-      interface.  Therefore, a simple pass through of data in those cases may
>-      be broken.  At a later time, we should do a once-over pass of all
>-      adapters, and make them compliant with these interfaces, after which this
>-      code can be uncommented.
>-  virtual const void* GetReadData(size_t* data_len) {
>-    return stream_->GetReadData(data_len);
>-  }
>-  virtual void ConsumeReadData(size_t used) {
>-    stream_->ConsumeReadData(used);
>-  }
>-
>-  virtual void* GetWriteBuffer(size_t* buf_len) {
>-    return stream_->GetWriteBuffer(buf_len);
>-  }
>-  virtual void ConsumeWriteBuffer(size_t used) {
>-    stream_->ConsumeWriteBuffer(used);
>-  }
>-  */
>-
>-  /*  Note: This interface is currently undergoing evaluation.
>-  virtual bool ForceWrite(const void* data, size_t data_len, int* error) {
>-    return stream_->ForceWrite(data, data_len, error);
>-  }
>-  */
>-
>   bool SetPosition(size_t position) override;
>   bool GetPosition(size_t* position) const override;
>   bool GetSize(size_t* size) const override;
>-  bool GetWriteRemaining(size_t* size) const override;
>   bool ReserveSize(size_t size) override;
>   bool Flush() override;
> 
>@@ -353,68 +261,11 @@ class FileStream : public StreamInterface {
>   RTC_DISALLOW_COPY_AND_ASSIGN(FileStream);
> };
> 
>-///////////////////////////////////////////////////////////////////////////////
>-// MemoryStream is a simple implementation of a StreamInterface over in-memory
>-// data.  Data is read and written at the current seek position.  Reads return
>-// end-of-stream when they reach the end of data.  Writes actually extend the
>-// end of data mark.
>-///////////////////////////////////////////////////////////////////////////////
>-
>-class MemoryStreamBase : public StreamInterface {
>- public:
>-  StreamState GetState() const override;
>-  StreamResult Read(void* buffer,
>-                    size_t bytes,
>-                    size_t* bytes_read,
>-                    int* error) override;
>-  StreamResult Write(const void* buffer,
>-                     size_t bytes,
>-                     size_t* bytes_written,
>-                     int* error) override;
>-  void Close() override;
>-  bool SetPosition(size_t position) override;
>-  bool GetPosition(size_t* position) const override;
>-  bool GetSize(size_t* size) const override;
>-  bool ReserveSize(size_t size) override;
>-
>-  char* GetBuffer() { return buffer_; }
>-  const char* GetBuffer() const { return buffer_; }
>-
>- protected:
>-  MemoryStreamBase();
>-
>-  virtual StreamResult DoReserve(size_t size, int* error);
>-
>-  // Invariant: 0 <= seek_position <= data_length_ <= buffer_length_
>-  char* buffer_;
>-  size_t buffer_length_;
>-  size_t data_length_;
>-  size_t seek_position_;
>-
>- private:
>-  RTC_DISALLOW_COPY_AND_ASSIGN(MemoryStreamBase);
>-};
>-
>-// MemoryStream dynamically resizes to accomodate written data.
>-
>-class MemoryStream : public MemoryStreamBase {
>- public:
>-  MemoryStream();
>-  explicit MemoryStream(const char* data);  // Calls SetData(data, strlen(data))
>-  MemoryStream(const void* data, size_t length);  // Calls SetData(data, length)
>-  ~MemoryStream() override;
>-
>-  void SetData(const void* data, size_t length);
>-
>- protected:
>-  StreamResult DoReserve(size_t size, int* error) override;
>-};
>-
> // FifoBuffer allows for efficient, thread-safe buffering of data between
> // writer and reader. As the data can wrap around the end of the buffer,
> // MemoryStreamBase can't help us here.
> 
>-class FifoBuffer : public StreamInterface {
>+class FifoBuffer final : public StreamInterface {
>  public:
>   // Creates a FIFO buffer with the specified capacity.
>   explicit FifoBuffer(size_t length);
>@@ -455,11 +306,28 @@ class FifoBuffer : public StreamInterface {
>                      size_t* bytes_written,
>                      int* error) override;
>   void Close() override;
>-  const void* GetReadData(size_t* data_len) override;
>-  void ConsumeReadData(size_t used) override;
>-  void* GetWriteBuffer(size_t* buf_len) override;
>-  void ConsumeWriteBuffer(size_t used) override;
>-  bool GetWriteRemaining(size_t* size) const override;
>+  // GetReadData returns a pointer to a buffer which is owned by the stream.
>+  // The buffer contains data_len bytes.  null is returned if no data is
>+  // available, or if the method fails.  If the caller processes the data, it
>+  // must call ConsumeReadData with the number of processed bytes.  GetReadData
>+  // does not require a matching call to ConsumeReadData if the data is not
>+  // processed.  Read and ConsumeReadData invalidate the buffer returned by
>+  // GetReadData.
>+  const void* GetReadData(size_t* data_len);
>+  void ConsumeReadData(size_t used);
>+  // GetWriteBuffer returns a pointer to a buffer which is owned by the stream.
>+  // The buffer has a capacity of buf_len bytes.  null is returned if there is
>+  // no buffer available, or if the method fails.  The call may write data to
>+  // the buffer, and then call ConsumeWriteBuffer with the number of bytes
>+  // written.  GetWriteBuffer does not require a matching call to
>+  // ConsumeWriteData if no data is written.  Write and
>+  // ConsumeWriteData invalidate the buffer returned by GetWriteBuffer.
>+  void* GetWriteBuffer(size_t* buf_len);
>+  void ConsumeWriteBuffer(size_t used);
>+
>+  // Return the number of Write()-able bytes remaining before end-of-stream.
>+  // Returns false if not known.
>+  bool GetWriteRemaining(size_t* size) const;
> 
>  private:
>   // Helper method that implements ReadOffset. Caller must acquire a lock
>@@ -495,25 +363,6 @@ class FifoBuffer : public StreamInterface {
>   RTC_DISALLOW_COPY_AND_ASSIGN(FifoBuffer);
> };
> 
>-///////////////////////////////////////////////////////////////////////////////
>-
>-// Flow attempts to move bytes from source to sink via buffer of size
>-// buffer_len.  The function returns SR_SUCCESS when source reaches
>-// end-of-stream (returns SR_EOS), and all the data has been written successful
>-// to sink.  Alternately, if source returns SR_BLOCK or SR_ERROR, or if sink
>-// returns SR_BLOCK, SR_ERROR, or SR_EOS, then the function immediately returns
>-// with the unexpected StreamResult value.
>-// data_len is the length of the valid data in buffer. in case of error
>-// this is the data that read from source but can't move to destination.
>-// as a pass in parameter, it indicates data in buffer that should move to sink
>-StreamResult Flow(StreamInterface* source,
>-                  char* buffer,
>-                  size_t buffer_len,
>-                  StreamInterface* sink,
>-                  size_t* data_len = nullptr);
>-
>-///////////////////////////////////////////////////////////////////////////////
>-
> }  // namespace rtc
> 
> #endif  // RTC_BASE_STREAM_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream_unittest.cc
>index 616783cb4dc76011f60793a06a28b08ff798320e..2ca2526c3caaea70f4543c09862b4bbcfaf4cda7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream_unittest.cc
>@@ -105,56 +105,52 @@ TEST(FifoBufferTest, TestAll) {
>   const void* q;
>   size_t bytes;
>   FifoBuffer buf(kSize);
>-  StreamInterface* stream = &buf;
> 
>   // Test assumptions about base state
>-  EXPECT_EQ(SS_OPEN, stream->GetState());
>-  EXPECT_EQ(SR_BLOCK, stream->Read(out, kSize, &bytes, nullptr));
>-  EXPECT_TRUE(nullptr != stream->GetReadData(&bytes));
>-  EXPECT_EQ((size_t)0, bytes);
>-  stream->ConsumeReadData(0);
>-  EXPECT_TRUE(nullptr != stream->GetWriteBuffer(&bytes));
>+  EXPECT_EQ(SS_OPEN, buf.GetState());
>+  EXPECT_EQ(SR_BLOCK, buf.Read(out, kSize, &bytes, nullptr));
>+  EXPECT_TRUE(nullptr != buf.GetWriteBuffer(&bytes));
>   EXPECT_EQ(kSize, bytes);
>-  stream->ConsumeWriteBuffer(0);
>+  buf.ConsumeWriteBuffer(0);
> 
>   // Try a full write
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize, &bytes, nullptr));
>   EXPECT_EQ(kSize, bytes);
> 
>   // Try a write that should block
>-  EXPECT_EQ(SR_BLOCK, stream->Write(in, kSize, &bytes, nullptr));
>+  EXPECT_EQ(SR_BLOCK, buf.Write(in, kSize, &bytes, nullptr));
> 
>   // Try a full read
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize, &bytes, nullptr));
>   EXPECT_EQ(kSize, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize));
> 
>   // Try a read that should block
>-  EXPECT_EQ(SR_BLOCK, stream->Read(out, kSize, &bytes, nullptr));
>+  EXPECT_EQ(SR_BLOCK, buf.Read(out, kSize, &bytes, nullptr));
> 
>   // Try a too-big write
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize * 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize * 2, &bytes, nullptr));
>   EXPECT_EQ(bytes, kSize);
> 
>   // Try a too-big read
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize * 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize * 2, &bytes, nullptr));
>   EXPECT_EQ(kSize, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize));
> 
>   // Try some small writes and reads
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
> 
>@@ -166,51 +162,51 @@ TEST(FifoBufferTest, TestAll) {
>   // XXXXWWWWWWWWXXXX 4567012345670123
>   // RRRRXXXXXXXXRRRR ....01234567....
>   // ....RRRRRRRR.... ................
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize * 3 / 4, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize * 3 / 4, &bytes, nullptr));
>   EXPECT_EQ(kSize * 3 / 4, bytes);
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 4, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize / 4, &bytes, nullptr));
>   EXPECT_EQ(kSize / 4, bytes);
>   EXPECT_EQ(0, memcmp(in + kSize / 2, out, kSize / 4));
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
> 
>   // Use GetWriteBuffer to reset the read_position for the next tests
>-  stream->GetWriteBuffer(&bytes);
>-  stream->ConsumeWriteBuffer(0);
>+  buf.GetWriteBuffer(&bytes);
>+  buf.ConsumeWriteBuffer(0);
> 
>   // Try using GetReadData to do a full read
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize, &bytes, nullptr));
>-  q = stream->GetReadData(&bytes);
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize, &bytes, nullptr));
>+  q = buf.GetReadData(&bytes);
>   EXPECT_TRUE(nullptr != q);
>   EXPECT_EQ(kSize, bytes);
>   EXPECT_EQ(0, memcmp(q, in, kSize));
>-  stream->ConsumeReadData(kSize);
>-  EXPECT_EQ(SR_BLOCK, stream->Read(out, kSize, &bytes, nullptr));
>+  buf.ConsumeReadData(kSize);
>+  EXPECT_EQ(SR_BLOCK, buf.Read(out, kSize, &bytes, nullptr));
> 
>   // Try using GetReadData to do some small reads
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize, &bytes, nullptr));
>-  q = stream->GetReadData(&bytes);
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize, &bytes, nullptr));
>+  q = buf.GetReadData(&bytes);
>   EXPECT_TRUE(nullptr != q);
>   EXPECT_EQ(kSize, bytes);
>   EXPECT_EQ(0, memcmp(q, in, kSize / 2));
>-  stream->ConsumeReadData(kSize / 2);
>-  q = stream->GetReadData(&bytes);
>+  buf.ConsumeReadData(kSize / 2);
>+  q = buf.GetReadData(&bytes);
>   EXPECT_TRUE(nullptr != q);
>   EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(0, memcmp(q, in + kSize / 2, kSize / 2));
>-  stream->ConsumeReadData(kSize / 2);
>-  EXPECT_EQ(SR_BLOCK, stream->Read(out, kSize, &bytes, nullptr));
>+  buf.ConsumeReadData(kSize / 2);
>+  EXPECT_EQ(SR_BLOCK, buf.Read(out, kSize, &bytes, nullptr));
> 
>   // Try using GetReadData in a wraparound case
>   // WWWWWWWWWWWWWWWW 0123456789ABCDEF
>@@ -218,46 +214,46 @@ TEST(FifoBufferTest, TestAll) {
>   // WWWWWWWW....XXXX 01234567....CDEF
>   // ............RRRR 01234567........
>   // RRRRRRRR........ ................
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize, &bytes, nullptr));
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize * 3 / 4, &bytes, nullptr));
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize / 2, &bytes, nullptr));
>-  q = stream->GetReadData(&bytes);
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize * 3 / 4, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize / 2, &bytes, nullptr));
>+  q = buf.GetReadData(&bytes);
>   EXPECT_TRUE(nullptr != q);
>   EXPECT_EQ(kSize / 4, bytes);
>   EXPECT_EQ(0, memcmp(q, in + kSize * 3 / 4, kSize / 4));
>-  stream->ConsumeReadData(kSize / 4);
>-  q = stream->GetReadData(&bytes);
>+  buf.ConsumeReadData(kSize / 4);
>+  q = buf.GetReadData(&bytes);
>   EXPECT_TRUE(nullptr != q);
>   EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(0, memcmp(q, in, kSize / 2));
>-  stream->ConsumeReadData(kSize / 2);
>+  buf.ConsumeReadData(kSize / 2);
> 
>   // Use GetWriteBuffer to reset the read_position for the next tests
>-  stream->GetWriteBuffer(&bytes);
>-  stream->ConsumeWriteBuffer(0);
>+  buf.GetWriteBuffer(&bytes);
>+  buf.ConsumeWriteBuffer(0);
> 
>   // Try using GetWriteBuffer to do a full write
>-  p = stream->GetWriteBuffer(&bytes);
>+  p = buf.GetWriteBuffer(&bytes);
>   EXPECT_TRUE(nullptr != p);
>   EXPECT_EQ(kSize, bytes);
>   memcpy(p, in, kSize);
>-  stream->ConsumeWriteBuffer(kSize);
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize, &bytes, nullptr));
>+  buf.ConsumeWriteBuffer(kSize);
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize, &bytes, nullptr));
>   EXPECT_EQ(kSize, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize));
> 
>   // Try using GetWriteBuffer to do some small writes
>-  p = stream->GetWriteBuffer(&bytes);
>+  p = buf.GetWriteBuffer(&bytes);
>   EXPECT_TRUE(nullptr != p);
>   EXPECT_EQ(kSize, bytes);
>   memcpy(p, in, kSize / 2);
>-  stream->ConsumeWriteBuffer(kSize / 2);
>-  p = stream->GetWriteBuffer(&bytes);
>+  buf.ConsumeWriteBuffer(kSize / 2);
>+  p = buf.GetWriteBuffer(&bytes);
>   EXPECT_TRUE(nullptr != p);
>   EXPECT_EQ(kSize / 2, bytes);
>   memcpy(p, in + kSize / 2, kSize / 2);
>-  stream->ConsumeWriteBuffer(kSize / 2);
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize, &bytes, nullptr));
>+  buf.ConsumeWriteBuffer(kSize / 2);
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize, &bytes, nullptr));
>   EXPECT_EQ(kSize, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize));
> 
>@@ -267,53 +263,53 @@ TEST(FifoBufferTest, TestAll) {
>   // ........XXXXWWWW ........89AB0123
>   // WWWW....XXXXXXXX 4567....89AB0123
>   // RRRR....RRRRRRRR ................
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize * 3 / 4, &bytes, nullptr));
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 2, &bytes, nullptr));
>-  p = stream->GetWriteBuffer(&bytes);
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize * 3 / 4, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize / 2, &bytes, nullptr));
>+  p = buf.GetWriteBuffer(&bytes);
>   EXPECT_TRUE(nullptr != p);
>   EXPECT_EQ(kSize / 4, bytes);
>   memcpy(p, in, kSize / 4);
>-  stream->ConsumeWriteBuffer(kSize / 4);
>-  p = stream->GetWriteBuffer(&bytes);
>+  buf.ConsumeWriteBuffer(kSize / 4);
>+  p = buf.GetWriteBuffer(&bytes);
>   EXPECT_TRUE(nullptr != p);
>   EXPECT_EQ(kSize / 2, bytes);
>   memcpy(p, in + kSize / 4, kSize / 4);
>-  stream->ConsumeWriteBuffer(kSize / 4);
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize * 3 / 4, &bytes, nullptr));
>+  buf.ConsumeWriteBuffer(kSize / 4);
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize * 3 / 4, &bytes, nullptr));
>   EXPECT_EQ(kSize * 3 / 4, bytes);
>   EXPECT_EQ(0, memcmp(in + kSize / 2, out, kSize / 4));
>   EXPECT_EQ(0, memcmp(in, out + kSize / 4, kSize / 4));
> 
>   // Check that the stream is now empty
>-  EXPECT_EQ(SR_BLOCK, stream->Read(out, kSize, &bytes, nullptr));
>+  EXPECT_EQ(SR_BLOCK, buf.Read(out, kSize, &bytes, nullptr));
> 
>   // Try growing the buffer
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize, &bytes, nullptr));
>   EXPECT_EQ(kSize, bytes);
>   EXPECT_TRUE(buf.SetCapacity(kSize * 2));
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in + kSize, kSize, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in + kSize, kSize, &bytes, nullptr));
>   EXPECT_EQ(kSize, bytes);
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize * 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize * 2, &bytes, nullptr));
>   EXPECT_EQ(kSize * 2, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize * 2));
> 
>   // Try shrinking the buffer
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize, &bytes, nullptr));
>   EXPECT_EQ(kSize, bytes);
>   EXPECT_TRUE(buf.SetCapacity(kSize));
>-  EXPECT_EQ(SR_BLOCK, stream->Write(in, kSize, &bytes, nullptr));
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize, &bytes, nullptr));
>+  EXPECT_EQ(SR_BLOCK, buf.Write(in, kSize, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize, &bytes, nullptr));
>   EXPECT_EQ(kSize, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize));
> 
>   // Write to the stream, close it, read the remaining bytes
>-  EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize / 2, &bytes, nullptr));
>-  stream->Close();
>-  EXPECT_EQ(SS_CLOSED, stream->GetState());
>-  EXPECT_EQ(SR_EOS, stream->Write(in, kSize / 2, &bytes, nullptr));
>-  EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Write(in, kSize / 2, &bytes, nullptr));
>+  buf.Close();
>+  EXPECT_EQ(SS_CLOSED, buf.GetState());
>+  EXPECT_EQ(SR_EOS, buf.Write(in, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_SUCCESS, buf.Read(out, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
>-  EXPECT_EQ(SR_EOS, stream->Read(out, kSize / 2, &bytes, nullptr));
>+  EXPECT_EQ(SR_EOS, buf.Read(out, kSize / 2, &bytes, nullptr));
> }
> 
> TEST(FifoBufferTest, FullBufferCheck) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.cc
>index 9201242d5f578d7f558ef1f073ac35b558467d69..634652b83f6be2bea367fb9128c1da4dffd91304 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.cc
>@@ -10,6 +10,7 @@
> 
> #include "rtc_base/string_to_number.h"
> 
>+#include <ctype.h>
> #include <cerrno>
> #include <cstdlib>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.h
>index 7ea9f251ed468bbb6a8a0cf76c6888d491c0309e..4cb521595defadd2f10bfd6dab4a38d1553617b3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.h
>@@ -13,6 +13,7 @@
> 
> #include <limits>
> #include <string>
>+#include <type_traits>
> 
> #include "absl/types/optional.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.cc
>index a57845ffdad5a80b0bfc24f9b545922b064ef26b..fc4e3bc8e7315cc7235c1f32769a5a6135258522 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.cc
>@@ -10,8 +10,7 @@
> 
> #include "rtc_base/stringencode.h"
> 
>-#include <stdio.h>
>-#include <stdlib.h>
>+#include <cstdio>
> 
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
>@@ -51,67 +50,6 @@ size_t url_decode(char* buffer,
>   return bufpos;
> }
> 
>-size_t utf8_decode(const char* source, size_t srclen, unsigned long* value) {
>-  const unsigned char* s = reinterpret_cast<const unsigned char*>(source);
>-  if ((s[0] & 0x80) == 0x00) {  // Check s[0] == 0xxxxxxx
>-    *value = s[0];
>-    return 1;
>-  }
>-  if ((srclen < 2) || ((s[1] & 0xC0) != 0x80)) {  // Check s[1] != 10xxxxxx
>-    return 0;
>-  }
>-  // Accumulate the trailer byte values in value16, and combine it with the
>-  // relevant bits from s[0], once we've determined the sequence length.
>-  unsigned long value16 = (s[1] & 0x3F);
>-  if ((s[0] & 0xE0) == 0xC0) {  // Check s[0] == 110xxxxx
>-    *value = ((s[0] & 0x1F) << 6) | value16;
>-    return 2;
>-  }
>-  if ((srclen < 3) || ((s[2] & 0xC0) != 0x80)) {  // Check s[2] != 10xxxxxx
>-    return 0;
>-  }
>-  value16 = (value16 << 6) | (s[2] & 0x3F);
>-  if ((s[0] & 0xF0) == 0xE0) {  // Check s[0] == 1110xxxx
>-    *value = ((s[0] & 0x0F) << 12) | value16;
>-    return 3;
>-  }
>-  if ((srclen < 4) || ((s[3] & 0xC0) != 0x80)) {  // Check s[3] != 10xxxxxx
>-    return 0;
>-  }
>-  value16 = (value16 << 6) | (s[3] & 0x3F);
>-  if ((s[0] & 0xF8) == 0xF0) {  // Check s[0] == 11110xxx
>-    *value = ((s[0] & 0x07) << 18) | value16;
>-    return 4;
>-  }
>-  return 0;
>-}
>-
>-size_t utf8_encode(char* buffer, size_t buflen, unsigned long value) {
>-  if ((value <= 0x7F) && (buflen >= 1)) {
>-    buffer[0] = static_cast<unsigned char>(value);
>-    return 1;
>-  }
>-  if ((value <= 0x7FF) && (buflen >= 2)) {
>-    buffer[0] = 0xC0 | static_cast<unsigned char>(value >> 6);
>-    buffer[1] = 0x80 | static_cast<unsigned char>(value & 0x3F);
>-    return 2;
>-  }
>-  if ((value <= 0xFFFF) && (buflen >= 3)) {
>-    buffer[0] = 0xE0 | static_cast<unsigned char>(value >> 12);
>-    buffer[1] = 0x80 | static_cast<unsigned char>((value >> 6) & 0x3F);
>-    buffer[2] = 0x80 | static_cast<unsigned char>(value & 0x3F);
>-    return 3;
>-  }
>-  if ((value <= 0x1FFFFF) && (buflen >= 4)) {
>-    buffer[0] = 0xF0 | static_cast<unsigned char>(value >> 18);
>-    buffer[1] = 0x80 | static_cast<unsigned char>((value >> 12) & 0x3F);
>-    buffer[2] = 0x80 | static_cast<unsigned char>((value >> 6) & 0x3F);
>-    buffer[3] = 0x80 | static_cast<unsigned char>(value & 0x3F);
>-    return 4;
>-  }
>-  return 0;
>-}
>-
> static const char HEX[] = "0123456789abcdef";
> 
> char hex_encode(unsigned char val) {
>@@ -122,9 +60,9 @@ char hex_encode(unsigned char val) {
> bool hex_decode(char ch, unsigned char* val) {
>   if ((ch >= '0') && (ch <= '9')) {
>     *val = ch - '0';
>-  } else if ((ch >= 'A') && (ch <= 'Z')) {
>+  } else if ((ch >= 'A') && (ch <= 'F')) {
>     *val = (ch - 'A') + 10;
>-  } else if ((ch >= 'a') && (ch <= 'z')) {
>+  } else if ((ch >= 'a') && (ch <= 'f')) {
>     *val = (ch - 'a') + 10;
>   } else {
>     return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.h
>index 5d436dfd57ba08a57307f5ac3d3103e3fce6e081..09bf77f4461525a3bafc1fa1faca69c296e2d2c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.h
>@@ -11,9 +11,12 @@
> #ifndef RTC_BASE_STRINGENCODE_H_
> #define RTC_BASE_STRINGENCODE_H_
> 
>+#include <stddef.h>
> #include <string>
>+#include <type_traits>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/string_to_number.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode_unittest.cc
>index 9bdc592c29aa0663778ae2c7d86dca601819b1f4..f21c4cb78c6f8484f7853572840bea293292b9c1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode_unittest.cc
>@@ -141,7 +141,8 @@ TEST_F(HexEncodeTest, TestDecodeTooShort) {
> 
> // Test that decoding non-hex data fails.
> TEST_F(HexEncodeTest, TestDecodeBogusData) {
>-  dec_res_ = hex_decode_with_delimiter(decoded_, sizeof(decoded_), "xyz", 3, 0);
>+  dec_res_ =
>+      hex_decode_with_delimiter(decoded_, sizeof(decoded_), "axyz", 4, 0);
>   ASSERT_EQ(0U, dec_res_);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/audio_format_to_string.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/audio_format_to_string.cc
>index a1493441985fdc9d72ee254121460a258cf697f3..7e91c3b49dd415585c18dcf657e3393b921def10 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/audio_format_to_string.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/audio_format_to_string.cc
>@@ -10,6 +10,8 @@
> 
> #include "rtc_base/strings/audio_format_to_string.h"
> 
>+#include <utility>
>+
> #include "rtc_base/strings/string_builder.h"
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.cc
>index 2fd4aba19cdcdbd32dfe9e999a09b8ff92e3638d..adf4fa9e819a389dd5b1c2fe9edff9360ddc1b6d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.cc
>@@ -10,6 +10,12 @@
> 
> #include "rtc_base/strings/string_builder.h"
> 
>+#include <stdarg.h>
>+#include <cstring>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+
> namespace rtc {
> 
> SimpleStringBuilder::SimpleStringBuilder(rtc::ArrayView<char> buffer)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.h
>index 186c2f7b9d79816e31acc02ec50daea6ff6d8ff2..27001d1be5a3c12588830ab3dc60fca46a733584 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.h
>@@ -12,13 +12,11 @@
> #define RTC_BASE_STRINGS_STRING_BUILDER_H_
> 
> #include <cstdio>
>-#include <cstring>
> #include <string>
>+#include <utility>
> 
> #include "absl/strings/string_view.h"
> #include "api/array_view.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/numerics/safe_minmax.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/stringutils.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.cc
>index 6baddd11bdd36d196db9fe31cc294a8fea4ca53e..c808eb2cc41ee607c3b9e73a9ec60db762eef1ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.cc
>@@ -7,97 +7,29 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>-#include <algorithm>
>-#include <cstdio>
> 
>-#include "rtc_base/checks.h"
> #include "rtc_base/stringutils.h"
> 
> namespace rtc {
> 
>-bool memory_check(const void* memory, int c, size_t count) {
>-  const char* char_memory = static_cast<const char*>(memory);
>-  char char_c = static_cast<char>(c);
>-  for (size_t i = 0; i < count; ++i) {
>-    if (char_memory[i] != char_c) {
>-      return false;
>-    }
>-  }
>-  return true;
>-}
>-
>-bool string_match(const char* target, const char* pattern) {
>-  while (*pattern) {
>-    if (*pattern == '*') {
>-      if (!*++pattern) {
>-        return true;
>-      }
>-      while (*target) {
>-        if ((toupper(*pattern) == toupper(*target)) &&
>-            string_match(target + 1, pattern + 1)) {
>-          return true;
>-        }
>-        ++target;
>-      }
>-      return false;
>-    } else {
>-      if (toupper(*pattern) != toupper(*target)) {
>-        return false;
>-      }
>-      ++target;
>-      ++pattern;
>-    }
>-  }
>-  return !*target;
>-}
>-
>-#if defined(WEBRTC_WIN)
>-int ascii_string_compare(const wchar_t* s1,
>-                         const char* s2,
>-                         size_t n,
>-                         CharacterTransformation transformation) {
>-  wchar_t c1, c2;
>-  while (true) {
>-    if (n-- == 0)
>-      return 0;
>-    c1 = transformation(*s1);
>-    // Double check that characters are not UTF-8
>-    RTC_DCHECK_LT(*s2, 128);
>-    // Note: *s2 gets implicitly promoted to wchar_t
>-    c2 = transformation(*s2);
>-    if (c1 != c2)
>-      return (c1 < c2) ? -1 : 1;
>-    if (!c1)
>-      return 0;
>-    ++s1;
>-    ++s2;
>-  }
>-}
>-
>-size_t asccpyn(wchar_t* buffer,
>+size_t strcpyn(char* buffer,
>                size_t buflen,
>                const char* source,
>-               size_t srclen) {
>+               size_t srclen /* = SIZE_UNKNOWN */) {
>   if (buflen <= 0)
>     return 0;
> 
>   if (srclen == SIZE_UNKNOWN) {
>-    srclen = strlenn(source, buflen - 1);
>-  } else if (srclen >= buflen) {
>+    srclen = strlen(source);
>+  }
>+  if (srclen >= buflen) {
>     srclen = buflen - 1;
>   }
>-#if RTC_DCHECK_IS_ON
>-  // Double check that characters are not UTF-8
>-  for (size_t pos = 0; pos < srclen; ++pos)
>-    RTC_DCHECK_LT(source[pos], 128);
>-#endif
>-  std::copy(source, source + srclen, buffer);
>+  memcpy(buffer, source, srclen);
>   buffer[srclen] = 0;
>   return srclen;
> }
> 
>-#endif  // WEBRTC_WIN
>-
> void replace_substrs(const char* search,
>                      size_t search_len,
>                      const char* replace,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.h
>index 26c4627d5ca59803a4e3fb5da2977863bca97972..c9dabfb8aaafb85b5ebb28f2f2b9637e777d4326 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.h
>@@ -13,8 +13,7 @@
> 
> #if !defined(WEBRTC_WEBKIT_BUILD)
> #include <ctype.h>
>-#endif
>-
>+#endif // !defined(WEBRTC_WEBKIT_BUILD)
> #include <stdarg.h>
> #include <stdio.h>
> #include <string.h>
>@@ -23,6 +22,7 @@
> #include <malloc.h>
> #include <wchar.h>
> #include <windows.h>
>+
> #define alloca _alloca
> #endif  // WEBRTC_WIN
> 
>@@ -32,6 +32,7 @@
> #else  // BSD
> #include <alloca.h>
> #endif  // !BSD
>+#include <strings.h>
> #endif  // WEBRTC_POSIX
> 
> #include <string>
>@@ -43,49 +44,6 @@
> #define STACK_ARRAY(TYPE, LEN) \
>   static_cast<TYPE*>(::alloca((LEN) * sizeof(TYPE)))
> 
>-namespace rtc {
>-
>-// Determines whether the simple wildcard pattern matches target.
>-// Alpha characters in pattern match case-insensitively.
>-// Asterisks in pattern match 0 or more characters.
>-// Ex: string_match("www.TEST.GOOGLE.COM", "www.*.com") -> true
>-bool string_match(const char* target, const char* pattern);
>-
>-}  // namespace rtc
>-
>-#if !defined(WEBRTC_WEBKIT_BUILD)
>-///////////////////////////////////////////////////////////////////////////////
>-// Rename a few common string functions so they are consistent across platforms.
>-// tolowercase is like tolower, but not compatible with end-of-file value
>-//
>-// It's not clear if we will ever use wchar_t strings on unix.  In theory,
>-// all strings should be Utf8 all the time, except when interfacing with Win32
>-// APIs that require Utf16.
>-///////////////////////////////////////////////////////////////////////////////
>-inline char tolowercase(char c) {
>-  return static_cast<char>(tolower(c));
>-}
>-#endif // !defined(WEBRTC_WEBKIT_BUILD)
>-
>-#if defined(WEBRTC_WIN)
>-
>-inline wchar_t tolowercase(wchar_t c) {
>-  return static_cast<wchar_t>(towlower(c));
>-}
>-
>-#endif  // WEBRTC_WIN
>-
>-#if defined(WEBRTC_POSIX)
>-
>-inline int _stricmp(const char* s1, const char* s2) {
>-  return strcasecmp(s1, s2);
>-}
>-inline int _strnicmp(const char* s1, const char* s2, size_t n) {
>-  return strncasecmp(s1, s2, n);
>-}
>-
>-#endif  // WEBRTC_POSIX
>-
> ///////////////////////////////////////////////////////////////////////////////
> // Traits simplifies porting string functions to be CTYPE-agnostic
> ///////////////////////////////////////////////////////////////////////////////
>@@ -94,191 +52,12 @@ namespace rtc {
> 
> const size_t SIZE_UNKNOWN = static_cast<size_t>(-1);
> 
>-template <class CTYPE>
>-struct Traits {
>-  // STL string type
>-  // typedef XXX string;
>-  // Null-terminated string
>-  // inline static const CTYPE* empty_str();
>-};
>-
>-///////////////////////////////////////////////////////////////////////////////
>-// String utilities which work with char or wchar_t
>-///////////////////////////////////////////////////////////////////////////////
>-
>-template <class CTYPE>
>-inline const CTYPE* nonnull(const CTYPE* str, const CTYPE* def_str = nullptr) {
>-  return str ? str : (def_str ? def_str : Traits<CTYPE>::empty_str());
>-}
>-
>-template <class CTYPE>
>-const CTYPE* strchr(const CTYPE* str, const CTYPE* chs) {
>-  for (size_t i = 0; str[i]; ++i) {
>-    for (size_t j = 0; chs[j]; ++j) {
>-      if (str[i] == chs[j]) {
>-        return str + i;
>-      }
>-    }
>-  }
>-  return 0;
>-}
>-
>-template <class CTYPE>
>-const CTYPE* strchrn(const CTYPE* str, size_t slen, CTYPE ch) {
>-  for (size_t i = 0; i < slen && str[i]; ++i) {
>-    if (str[i] == ch) {
>-      return str + i;
>-    }
>-  }
>-  return 0;
>-}
>-
>-template <class CTYPE>
>-size_t strlenn(const CTYPE* buffer, size_t buflen) {
>-  size_t bufpos = 0;
>-  while (buffer[bufpos] && (bufpos < buflen)) {
>-    ++bufpos;
>-  }
>-  return bufpos;
>-}
>-
>-// Safe versions of strncpy, strncat, snprintf and vsnprintf that always
>-// null-terminate.
>-
>-template <class CTYPE>
>-size_t strcpyn(CTYPE* buffer,
>-               size_t buflen,
>-               const CTYPE* source,
>-               size_t srclen = SIZE_UNKNOWN) {
>-  if (buflen <= 0)
>-    return 0;
>-
>-  if (srclen == SIZE_UNKNOWN) {
>-    srclen = strlenn(source, buflen - 1);
>-  } else if (srclen >= buflen) {
>-    srclen = buflen - 1;
>-  }
>-  memcpy(buffer, source, srclen * sizeof(CTYPE));
>-  buffer[srclen] = 0;
>-  return srclen;
>-}
>-
>-template <class CTYPE>
>-size_t strcatn(CTYPE* buffer,
>-               size_t buflen,
>-               const CTYPE* source,
>-               size_t srclen = SIZE_UNKNOWN) {
>-  if (buflen <= 0)
>-    return 0;
>-
>-  size_t bufpos = strlenn(buffer, buflen - 1);
>-  return bufpos + strcpyn(buffer + bufpos, buflen - bufpos, source, srclen);
>-}
>-
>-// Some compilers (clang specifically) require vsprintfn be defined before
>-// sprintfn.
>-template <class CTYPE>
>-size_t vsprintfn(CTYPE* buffer,
>-                 size_t buflen,
>-                 const CTYPE* format,
>-                 va_list args) {
>-  int len = vsnprintf(buffer, buflen, format, args);
>-  if ((len < 0) || (static_cast<size_t>(len) >= buflen)) {
>-    len = static_cast<int>(buflen - 1);
>-    buffer[len] = 0;
>-  }
>-  return len;
>-}
>-
>-template <class CTYPE>
>-size_t sprintfn(CTYPE* buffer, size_t buflen, const CTYPE* format, ...);
>-template <class CTYPE>
>-size_t sprintfn(CTYPE* buffer, size_t buflen, const CTYPE* format, ...) {
>-  va_list args;
>-  va_start(args, format);
>-  size_t len = vsprintfn(buffer, buflen, format, args);
>-  va_end(args);
>-  return len;
>-}
>-
>-///////////////////////////////////////////////////////////////////////////////
>-// Allow safe comparing and copying ascii (not UTF-8) with both wide and
>-// non-wide character strings.
>-///////////////////////////////////////////////////////////////////////////////
>-
>-inline int asccmp(const char* s1, const char* s2) {
>-  return strcmp(s1, s2);
>-}
>-inline int ascicmp(const char* s1, const char* s2) {
>-  return _stricmp(s1, s2);
>-}
>-inline int ascncmp(const char* s1, const char* s2, size_t n) {
>-  return strncmp(s1, s2, n);
>-}
>-inline int ascnicmp(const char* s1, const char* s2, size_t n) {
>-  return _strnicmp(s1, s2, n);
>-}
>-inline size_t asccpyn(char* buffer,
>-                      size_t buflen,
>-                      const char* source,
>-                      size_t srclen = SIZE_UNKNOWN) {
>-  return strcpyn(buffer, buflen, source, srclen);
>-}
>-
>-#if defined(WEBRTC_WIN)
>-
>-typedef wchar_t (*CharacterTransformation)(wchar_t);
>-inline wchar_t identity(wchar_t c) {
>-  return c;
>-}
>-int ascii_string_compare(const wchar_t* s1,
>-                         const char* s2,
>-                         size_t n,
>-                         CharacterTransformation transformation);
>-
>-inline int asccmp(const wchar_t* s1, const char* s2) {
>-  return ascii_string_compare(s1, s2, static_cast<size_t>(-1), identity);
>-}
>-inline int ascicmp(const wchar_t* s1, const char* s2) {
>-  return ascii_string_compare(s1, s2, static_cast<size_t>(-1), tolowercase);
>-}
>-inline int ascncmp(const wchar_t* s1, const char* s2, size_t n) {
>-  return ascii_string_compare(s1, s2, n, identity);
>-}
>-inline int ascnicmp(const wchar_t* s1, const char* s2, size_t n) {
>-  return ascii_string_compare(s1, s2, n, tolowercase);
>-}
>-size_t asccpyn(wchar_t* buffer,
>+// Safe version of strncpy that always nul-terminate.
>+size_t strcpyn(char* buffer,
>                size_t buflen,
>                const char* source,
>                size_t srclen = SIZE_UNKNOWN);
> 
>-#endif  // WEBRTC_WIN
>-
>-///////////////////////////////////////////////////////////////////////////////
>-// Traits<char> specializations
>-///////////////////////////////////////////////////////////////////////////////
>-
>-template <>
>-struct Traits<char> {
>-  typedef std::string string;
>-  inline static const char* empty_str() { return ""; }
>-};
>-
>-///////////////////////////////////////////////////////////////////////////////
>-// Traits<wchar_t> specializations (Windows only, currently)
>-///////////////////////////////////////////////////////////////////////////////
>-
>-#if defined(WEBRTC_WIN)
>-
>-template <>
>-struct Traits<wchar_t> {
>-  typedef std::wstring string;
>-  inline static const wchar_t* empty_str() { return L""; }
>-};
>-
>-#endif  // WEBRTC_WIN
>-
> ///////////////////////////////////////////////////////////////////////////////
> // UTF helpers (Windows only)
> ///////////////////////////////////////////////////////////////////////////////
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils_unittest.cc
>index a6e64680bb7359d81ed1ae4effcb6396508b6cc6..663e9768f608a346c842045a11ced5704e76cf95 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils_unittest.cc
>@@ -13,70 +13,6 @@
> 
> namespace rtc {
> 
>-// Tests for string_match().
>-
>-TEST(string_matchTest, Matches) {
>-  EXPECT_TRUE(string_match("A.B.C.D", "a.b.c.d"));
>-  EXPECT_TRUE(string_match("www.TEST.GOOGLE.COM", "www.*.com"));
>-  EXPECT_TRUE(string_match("127.0.0.1", "12*.0.*1"));
>-  EXPECT_TRUE(string_match("127.1.0.21", "12*.0.*1"));
>-  EXPECT_FALSE(string_match("127.0.0.0", "12*.0.*1"));
>-  EXPECT_FALSE(string_match("127.0.0.0", "12*.0.*1"));
>-  EXPECT_FALSE(string_match("127.1.1.21", "12*.0.*1"));
>-}
>-
>-// It's not clear if we will ever use wchar_t strings on unix.  In theory,
>-// all strings should be Utf8 all the time, except when interfacing with Win32
>-// APIs that require Utf16.
>-
>-#if defined(WEBRTC_WIN)
>-
>-// Tests for ascii_string_compare().
>-
>-// Tests null input.
>-TEST(ascii_string_compareTest, NullInput) {
>-  // The following results in an access violation in
>-  // ascii_string_compare.  Is this a bug or by design?  stringutils.h
>-  // should document the expected behavior in this case.
>-
>-  // EXPECT_EQ(0, ascii_string_compare(nullptr, nullptr, 1, identity));
>-}
>-
>-// Tests comparing two strings of different lengths.
>-TEST(ascii_string_compareTest, DifferentLengths) {
>-  EXPECT_EQ(-1, ascii_string_compare(L"Test", "Test1", 5, identity));
>-}
>-
>-// Tests the case where the buffer size is smaller than the string
>-// lengths.
>-TEST(ascii_string_compareTest, SmallBuffer) {
>-  EXPECT_EQ(0, ascii_string_compare(L"Test", "Test1", 3, identity));
>-}
>-
>-// Tests the case where the buffer is not full.
>-TEST(ascii_string_compareTest, LargeBuffer) {
>-  EXPECT_EQ(0, ascii_string_compare(L"Test", "Test", 10, identity));
>-}
>-
>-// Tests comparing two eqaul strings.
>-TEST(ascii_string_compareTest, Equal) {
>-  EXPECT_EQ(0, ascii_string_compare(L"Test", "Test", 5, identity));
>-  EXPECT_EQ(0, ascii_string_compare(L"TeSt", "tEsT", 5, tolowercase));
>-}
>-
>-// Tests comparing a smller string to a larger one.
>-TEST(ascii_string_compareTest, LessThan) {
>-  EXPECT_EQ(-1, ascii_string_compare(L"abc", "abd", 4, identity));
>-  EXPECT_EQ(-1, ascii_string_compare(L"ABC", "abD", 5, tolowercase));
>-}
>-
>-// Tests comparing a larger string to a smaller one.
>-TEST(ascii_string_compareTest, GreaterThan) {
>-  EXPECT_EQ(1, ascii_string_compare(L"xyz", "xy", 5, identity));
>-  EXPECT_EQ(1, ascii_string_compare(L"abc", "ABB", 5, tolowercase));
>-}
>-#endif  // WEBRTC_WIN
>-
> TEST(string_trim_Test, Trimming) {
>   EXPECT_EQ("temp", string_trim("\n\r\t temp \n\r\t"));
>   EXPECT_EQ("temp\n\r\t temp", string_trim(" temp\n\r\t temp "));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_posix.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_posix.cc
>index 7e37dc924fdf81cb70916e31aa7507b62ce62003..15ef3d706e8aecadddbccfed6118eea30bdf4c0b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_posix.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_posix.cc
>@@ -10,6 +10,8 @@
> 
> #include "rtc_base/synchronization/rw_lock_posix.h"
> 
>+#include <stddef.h>
>+
> namespace webrtc {
> 
> RWLockPosix::RWLockPosix() : lock_() {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/BUILD.gn
>index f22efe14a0488e33b178d6cb6eb3522c43f6accc..e5aa32b4ef1d8c522a8243a16b44043be7bc215b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/BUILD.gn
>@@ -65,3 +65,15 @@ rtc_source_set("rtc_export") {
>     "rtc_export.h",
>   ]
> }
>+
>+if (is_mac || is_ios) {
>+  rtc_source_set("cocoa_threading") {
>+    sources = [
>+      "cocoa_threading.h",
>+      "cocoa_threading.mm",
>+    ]
>+    deps = [
>+      "..:checks",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/cocoa_threading.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/cocoa_threading.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..518cb71786baf34ac0e3d2830da8518398c2f729
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/cocoa_threading.h
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_SYSTEM_COCOA_THREADING_H_
>+#define RTC_BASE_SYSTEM_COCOA_THREADING_H_
>+
>+// If Cocoa is to be used on more than one thread, it must know that the
>+// application is multithreaded.  Since it's possible to enter Cocoa code
>+// from threads created by pthread_thread_create, Cocoa won't necessarily
>+// be aware that the application is multithreaded.  Spawning an NSThread is
>+// enough to get Cocoa to set up for multithreaded operation, so this is done
>+// if necessary before pthread_thread_create spawns any threads.
>+//
>+// http://developer.apple.com/documentation/Cocoa/Conceptual/Multithreading/CreatingThreads/chapter_4_section_4.html
>+void InitCocoaMultiThreading();
>+
>+#endif  // RTC_BASE_SYSTEM_COCOA_THREADING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/cocoa_threading.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/cocoa_threading.mm
>new file mode 100644
>index 0000000000000000000000000000000000000000..c09862e7e5f750db05ab884e05be24b015cd3dbc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/cocoa_threading.mm
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "rtc_base/system/cocoa_threading.h"
>+
>+#import <Foundation/Foundation.h>
>+
>+#include "rtc_base/checks.h"
>+
>+void InitCocoaMultiThreading() {
>+  static BOOL is_cocoa_multithreaded = [NSThread isMultiThreaded];
>+  if (!is_cocoa_multithreaded) {
>+    // +[NSObject class] is idempotent.
>+    [NSThread detachNewThreadSelector:@selector(class) toTarget:[NSObject class] withObject:nil];
>+    is_cocoa_multithreaded = YES;
>+    RTC_DCHECK([NSThread isMultiThreaded]);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.cc
>index 72f5f259dbdf689a03584bbcd7515b11b9ab5038..c033a792a50c3e9238896255ab6c657e3880943d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.cc
>@@ -13,14 +13,11 @@
> #ifdef _WIN32
> #include <Windows.h>
> #else
>-#include <stdarg.h>
> #include <string.h>
> #endif
> 
> #include <utility>
> 
>-#include "rtc_base/checks.h"
>-
> namespace webrtc {
> namespace {
> FILE* FileOpen(const char* file_name_utf8, bool read_only) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.h
>index 5411b04f6992d75d3eab9212690949b036cdc000..0bb86a397f9060bfe433d9e356f4f989f6555f00 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.h
>@@ -14,7 +14,6 @@
> #include <stddef.h>
> #include <stdio.h>
> 
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/criticalsection.h"
> 
> // Implementation that can read (exclusive) or write from/to a file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/rtc_export.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/rtc_export.h
>index b7a0fe416826082197b98462798f5caabdbddc58..d1eb60ad783a5f1c8a7a80742ebfded31ab735ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/rtc_export.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/rtc_export.h
>@@ -16,7 +16,7 @@
> // When WebRTC is built as a static library the RTC_EXPORT macro expands to
> // nothing.
> 
>-#ifdef COMPONENT_BUILD
>+#ifdef WEBRTC_ENABLE_SYMBOL_EXPORT
> 
> #ifdef WEBRTC_WIN
> 
>@@ -34,7 +34,7 @@
> 
> #endif  // WEBRTC_WIN
> 
>-#endif  // COMPONENT_BUILD
>+#endif  // WEBRTC_ENABLE_SYMBOL_EXPORT
> 
> #ifndef RTC_EXPORT
> #define RTC_EXPORT
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue.h
>index b8b307ca6a4058354e4b1b888698848b5394e793..888e203967cd365ad082b33e809dc1a749c16d8b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue.h
>@@ -18,6 +18,7 @@
> #include "absl/memory/memory.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/system/rtc_export.h"
> #include "rtc_base/thread_annotations.h"
> 
> namespace rtc {
>@@ -150,7 +151,7 @@ static std::unique_ptr<QueuedTask> NewClosure(Closure&& closure,
> // TaskQueue itself has been deleted or it may happen synchronously while the
> // TaskQueue instance is being deleted.  This may vary from one OS to the next
> // so assumptions about lifetimes of pending tasks should not be made.
>-class RTC_LOCKABLE TaskQueue {
>+class RTC_LOCKABLE RTC_EXPORT TaskQueue {
>  public:
>   // TaskQueue priority levels. On some platforms these will map to thread
>   // priorities, on others such as Mac and iOS, GCD queue priorities.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_for_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_for_test.h
>index 70c58fb516c1aa6ad43edfeaa93bd3cab2daca1a..9162e81b88ace9582b08d8f20d393cc1e0a24ca8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_for_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_for_test.h
>@@ -33,7 +33,7 @@ class RTC_LOCKABLE TaskQueueForTest : public TaskQueue {
>   template <class Closure>
>   void SendTask(Closure* task) {
>     RTC_DCHECK(!IsCurrent());
>-    rtc::Event event(false, false);
>+    rtc::Event event;
>     PostTask(rtc::NewClosure(
>         [&task]() {
>           RTC_CHECK_EQ(false, static_cast<QueuedTask*>(task)->Run());
>@@ -47,7 +47,7 @@ class RTC_LOCKABLE TaskQueueForTest : public TaskQueue {
>   template <class Closure>
>   void SendTask(Closure&& task) {
>     RTC_DCHECK(!IsCurrent());
>-    rtc::Event event(false, false);
>+    rtc::Event event;
>     PostTask(rtc::NewClosure(std::move(task), [&event]() { event.Set(); }));
>     event.Wait(rtc::Event::kForever);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>index cbed3334d77fd7aa8657ab895a8f13ceaeb56584..27b8801bb0de0483632f4b8a70c325851946dae2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>@@ -10,11 +10,17 @@
> 
> #include "rtc_base/task_queue.h"
> 
>+#include <errno.h>
> #include <fcntl.h>
>+#include <pthread.h>
> #include <signal.h>
>-#include <string.h>
>+#include <stdint.h>
>+#include <time.h>
> #include <unistd.h>
> #include <list>
>+#include <memory>
>+#include <type_traits>
>+#include <utility>
> 
> #if defined(WEBRTC_LINUX)
> #include <event2/event.h>
>@@ -23,17 +29,18 @@
> #else
> #include "base/third_party/libevent/event.h"
> #endif
>-
> #include "rtc_base/checks.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/platform_thread.h"
>+#include "rtc_base/platform_thread_types.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/refcountedobject.h"
>+#include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/system/unused.h"
>-#include "rtc_base/task_queue.h"
> #include "rtc_base/task_queue_posix.h"
>+#include "rtc_base/thread_annotations.h"
> #include "rtc_base/timeutils.h"
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_stdlib.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_stdlib.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..0fb0ed28b951ebc910e103878b1893ac7b8088a6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_stdlib.cc
>@@ -0,0 +1,399 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/task_queue.h"
>+
>+#include <string.h>
>+#include <algorithm>
>+#include <atomic>
>+#include <condition_variable>
>+#include <map>
>+#include <queue>
>+#include <utility>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/event.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/platform_thread.h"
>+#include "rtc_base/refcount.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "rtc_base/thread_annotations.h"
>+#include "rtc_base/timeutils.h"
>+
>+namespace rtc {
>+namespace {
>+
>+using Priority = TaskQueue::Priority;
>+
>+ThreadPriority TaskQueuePriorityToThreadPriority(Priority priority) {
>+  switch (priority) {
>+    case Priority::HIGH:
>+      return kRealtimePriority;
>+    case Priority::LOW:
>+      return kLowPriority;
>+    case Priority::NORMAL:
>+      return kNormalPriority;
>+    default:
>+      RTC_NOTREACHED();
>+      return kNormalPriority;
>+  }
>+  return kNormalPriority;
>+}
>+
>+}  // namespace
>+
>+class TaskQueue::Impl : public RefCountInterface {
>+ public:
>+  Impl(const char* queue_name, TaskQueue* queue, Priority priority);
>+  ~Impl() override;
>+
>+  static TaskQueue::Impl* Current();
>+  static TaskQueue* CurrentQueue();
>+
>+  // Used for DCHECKing the current queue.
>+  bool IsCurrent() const;
>+
>+  template <class Closure,
>+            typename std::enable_if<!std::is_convertible<
>+                Closure,
>+                std::unique_ptr<QueuedTask>>::value>::type* = nullptr>
>+  void PostTask(Closure&& closure) {
>+    PostTask(NewClosure(std::forward<Closure>(closure)));
>+  }
>+
>+  void PostTask(std::unique_ptr<QueuedTask> task);
>+  void PostTaskAndReply(std::unique_ptr<QueuedTask> task,
>+                        std::unique_ptr<QueuedTask> reply,
>+                        TaskQueue::Impl* reply_queue);
>+
>+  void PostDelayedTask(std::unique_ptr<QueuedTask> task, uint32_t milliseconds);
>+
>+  class WorkerThread : public PlatformThread {
>+   public:
>+    WorkerThread(ThreadRunFunction func,
>+                 void* obj,
>+                 const char* thread_name,
>+                 ThreadPriority priority)
>+        : PlatformThread(func, obj, thread_name, priority) {}
>+  };
>+
>+  using OrderId = uint64_t;
>+
>+  struct DelayedEntryTimeout {
>+    int64_t next_fire_at_ms_{};
>+    OrderId order_{};
>+
>+    bool operator<(const DelayedEntryTimeout& o) const {
>+      return std::tie(next_fire_at_ms_, order_) <
>+             std::tie(o.next_fire_at_ms_, o.order_);
>+    }
>+  };
>+
>+  struct NextTask {
>+    bool final_task_{false};
>+    std::unique_ptr<QueuedTask> run_task_;
>+    int64_t sleep_time_ms_{};
>+  };
>+
>+ protected:
>+  NextTask GetNextTask();
>+
>+ private:
>+  // The ThreadQueue::Current() method requires that the current thread
>+  // returns the task queue if the current thread is the active task
>+  // queue and this variable holds the value needed in thread_local to
>+  // on the initialized worker thread holding the queue.
>+  static thread_local TaskQueue::Impl* thread_context_;
>+
>+  static void ThreadMain(void* context);
>+
>+  void ProcessTasks();
>+
>+  void NotifyWake();
>+
>+  // The back pointer from the owner task queue object
>+  // from this implementation detail.
>+  TaskQueue* const queue_;
>+
>+  // Indicates if the thread has started.
>+  Event started_;
>+
>+  // Indicates if the thread has stopped.
>+  Event stopped_;
>+
>+  // Signaled whenever a new task is pending.
>+  Event flag_notify_;
>+
>+  // Contains the active worker thread assigned to processing
>+  // tasks (including delayed tasks).
>+  WorkerThread thread_;
>+
>+  rtc::CriticalSection pending_lock_;
>+
>+  // Indicates if the worker thread needs to shutdown now.
>+  bool thread_should_quit_ RTC_GUARDED_BY(pending_lock_){false};
>+
>+  // Holds the next order to use for the next task to be
>+  // put into one of the pending queues.
>+  OrderId thread_posting_order_ RTC_GUARDED_BY(pending_lock_){};
>+
>+  // The list of all pending tasks that need to be processed in the
>+  // FIFO queue ordering on the worker thread.
>+  std::queue<std::pair<OrderId, std::unique_ptr<QueuedTask>>> pending_queue_
>+      RTC_GUARDED_BY(pending_lock_);
>+
>+  // The list of all pending tasks that need to be processed at a future
>+  // time based upon a delay. On the off change the delayed task should
>+  // happen at exactly the same time interval as another task then the
>+  // task is processed based on FIFO ordering. std::priority_queue was
>+  // considered but rejected due to its inability to extract the
>+  // std::unique_ptr out of the queue without the presence of a hack.
>+  std::map<DelayedEntryTimeout, std::unique_ptr<QueuedTask>> delayed_queue_
>+      RTC_GUARDED_BY(pending_lock_);
>+};
>+
>+// static
>+thread_local TaskQueue::Impl* TaskQueue::Impl::thread_context_ = nullptr;
>+
>+TaskQueue::Impl::Impl(const char* queue_name,
>+                      TaskQueue* queue,
>+                      Priority priority)
>+    : queue_(queue),
>+      started_(/*manual_reset=*/false, /*initially_signaled=*/false),
>+      stopped_(/*manual_reset=*/false, /*initially_signaled=*/false),
>+      flag_notify_(/*manual_reset=*/false, /*initially_signaled=*/false),
>+      thread_(&TaskQueue::Impl::ThreadMain,
>+              this,
>+              queue_name,
>+              TaskQueuePriorityToThreadPriority(priority)) {
>+  RTC_DCHECK(queue_name);
>+  thread_.Start();
>+  started_.Wait(Event::kForever);
>+}
>+
>+TaskQueue::Impl::~Impl() {
>+  RTC_DCHECK(!IsCurrent());
>+
>+  {
>+    CritScope lock(&pending_lock_);
>+    thread_should_quit_ = true;
>+  }
>+
>+  NotifyWake();
>+
>+  stopped_.Wait(Event::kForever);
>+  thread_.Stop();
>+}
>+
>+// static
>+TaskQueue::Impl* TaskQueue::Impl::Current() {
>+  return thread_context_;
>+}
>+
>+// static
>+TaskQueue* TaskQueue::Impl::CurrentQueue() {
>+  TaskQueue::Impl* current = Current();
>+  return current ? current->queue_ : nullptr;
>+}
>+
>+bool TaskQueue::Impl::IsCurrent() const {
>+  return IsThreadRefEqual(thread_.GetThreadRef(), CurrentThreadRef());
>+}
>+
>+void TaskQueue::Impl::PostTask(std::unique_ptr<QueuedTask> task) {
>+  {
>+    CritScope lock(&pending_lock_);
>+    OrderId order = thread_posting_order_++;
>+
>+    pending_queue_.push(std::pair<OrderId, std::unique_ptr<QueuedTask>>(
>+        order, std::move(task)));
>+  }
>+
>+  NotifyWake();
>+}
>+
>+void TaskQueue::Impl::PostDelayedTask(std::unique_ptr<QueuedTask> task,
>+                                      uint32_t milliseconds) {
>+  auto fire_at = rtc::TimeMillis() + milliseconds;
>+
>+  DelayedEntryTimeout delay;
>+  delay.next_fire_at_ms_ = fire_at;
>+
>+  {
>+    CritScope lock(&pending_lock_);
>+    delay.order_ = ++thread_posting_order_;
>+    delayed_queue_[delay] = std::move(task);
>+  }
>+
>+  NotifyWake();
>+}
>+
>+void TaskQueue::Impl::PostTaskAndReply(std::unique_ptr<QueuedTask> task,
>+                                       std::unique_ptr<QueuedTask> reply,
>+                                       TaskQueue::Impl* reply_queue) {
>+  QueuedTask* task_ptr = task.release();
>+  QueuedTask* reply_task_ptr = reply.release();
>+  PostTask([task_ptr, reply_task_ptr, reply_queue]() {
>+    if (task_ptr->Run())
>+      delete task_ptr;
>+
>+    reply_queue->PostTask(std::unique_ptr<QueuedTask>(reply_task_ptr));
>+  });
>+}
>+
>+TaskQueue::Impl::NextTask TaskQueue::Impl::GetNextTask() {
>+  NextTask result{};
>+
>+  auto tick = rtc::TimeMillis();
>+
>+  CritScope lock(&pending_lock_);
>+
>+  if (thread_should_quit_) {
>+    result.final_task_ = true;
>+    return result;
>+  }
>+
>+  if (delayed_queue_.size() > 0) {
>+    auto delayed_entry = delayed_queue_.begin();
>+    const auto& delay_info = delayed_entry->first;
>+    auto& delay_run = delayed_entry->second;
>+    if (tick >= delay_info.next_fire_at_ms_) {
>+      if (pending_queue_.size() > 0) {
>+        auto& entry = pending_queue_.front();
>+        auto& entry_order = entry.first;
>+        auto& entry_run = entry.second;
>+        if (entry_order < delay_info.order_) {
>+          result.run_task_ = std::move(entry_run);
>+          pending_queue_.pop();
>+          return result;
>+        }
>+      }
>+
>+      result.run_task_ = std::move(delay_run);
>+      delayed_queue_.erase(delayed_entry);
>+      return result;
>+    }
>+
>+    result.sleep_time_ms_ = delay_info.next_fire_at_ms_ - tick;
>+  }
>+
>+  if (pending_queue_.size() > 0) {
>+    auto& entry = pending_queue_.front();
>+    result.run_task_ = std::move(entry.second);
>+    pending_queue_.pop();
>+  }
>+
>+  return result;
>+}
>+
>+// static
>+void TaskQueue::Impl::ThreadMain(void* context) {
>+  TaskQueue::Impl* me = static_cast<TaskQueue::Impl*>(context);
>+  me->ProcessTasks();
>+}
>+
>+void TaskQueue::Impl::ProcessTasks() {
>+  thread_context_ = this;
>+  started_.Set();
>+
>+  while (true) {
>+    auto task = GetNextTask();
>+
>+    if (task.final_task_)
>+      break;
>+
>+    if (task.run_task_) {
>+      // process entry immediately then try again
>+      QueuedTask* release_ptr = task.run_task_.release();
>+      if (release_ptr->Run())
>+        delete release_ptr;
>+
>+      // attempt to sleep again
>+      continue;
>+    }
>+
>+    if (0 == task.sleep_time_ms_)
>+      flag_notify_.Wait(Event::kForever);
>+    else
>+      flag_notify_.Wait(task.sleep_time_ms_);
>+  }
>+
>+  stopped_.Set();
>+}
>+
>+void TaskQueue::Impl::NotifyWake() {
>+  // The queue holds pending tasks to complete. Either tasks are to be
>+  // executed immediately or tasks are to be run at some future delayed time.
>+  // For immediate tasks the task queue's thread is busy running the task and
>+  // the thread will not be waiting on the flag_notify_ event. If no immediate
>+  // tasks are available but a delayed task is pending then the thread will be
>+  // waiting on flag_notify_ with a delayed time-out of the nearest timed task
>+  // to run. If no immediate or pending tasks are available, the thread will
>+  // wait on flag_notify_ until signaled that a task has been added (or the
>+  // thread to be told to shutdown).
>+
>+  // In all cases, when a new immediate task, delayed task, or request to
>+  // shutdown the thread is added the flag_notify_ is signaled after. If the
>+  // thread was waiting then the thread will wake up immediately and re-assess
>+  // what task needs to be run next (i.e. run a task now, wait for the nearest
>+  // timed delayed task, or shutdown the thread). If the thread was not waiting
>+  // then the thread will remained signaled to wake up the next time any
>+  // attempt to wait on the flag_notify_ event occurs.
>+
>+  // Any immediate or delayed pending task (or request to shutdown the thread)
>+  // must always be added to the queue prior to signaling flag_notify_ to wake
>+  // up the possibly sleeping thread. This prevents a race condition where the
>+  // thread is notified to wake up but the task queue's thread finds nothing to
>+  // do so it waits once again to be signaled where such a signal may never
>+  // happen.
>+  flag_notify_.Set();
>+}
>+
>+// Boilerplate for the PIMPL pattern.
>+TaskQueue::TaskQueue(const char* queue_name, Priority priority)
>+    : impl_(new RefCountedObject<TaskQueue::Impl>(queue_name, this, priority)) {
>+}
>+
>+TaskQueue::~TaskQueue() {}
>+
>+// static
>+TaskQueue* TaskQueue::Current() {
>+  return TaskQueue::Impl::CurrentQueue();
>+}
>+
>+// Used for DCHECKing the current queue.
>+bool TaskQueue::IsCurrent() const {
>+  return impl_->IsCurrent();
>+}
>+
>+void TaskQueue::PostTask(std::unique_ptr<QueuedTask> task) {
>+  return TaskQueue::impl_->PostTask(std::move(task));
>+}
>+
>+void TaskQueue::PostTaskAndReply(std::unique_ptr<QueuedTask> task,
>+                                 std::unique_ptr<QueuedTask> reply,
>+                                 TaskQueue* reply_queue) {
>+  return TaskQueue::impl_->PostTaskAndReply(std::move(task), std::move(reply),
>+                                            reply_queue->impl_.get());
>+}
>+
>+void TaskQueue::PostTaskAndReply(std::unique_ptr<QueuedTask> task,
>+                                 std::unique_ptr<QueuedTask> reply) {
>+  return TaskQueue::impl_->PostTaskAndReply(std::move(task), std::move(reply),
>+                                            impl_.get());
>+}
>+
>+void TaskQueue::PostDelayedTask(std::unique_ptr<QueuedTask> task,
>+                                uint32_t milliseconds) {
>+  return TaskQueue::impl_->PostDelayedTask(std::move(task), milliseconds);
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_unittest.cc
>index cedf68e2e851359c505e4c34b14b9cd89bdaee63..0af39a5e715fb1578c359be6c28cc41ad3dd8247 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_unittest.cc
>@@ -64,7 +64,7 @@ TEST(TaskQueueTest, Construct) {
> 
> TEST(TaskQueueTest, PostAndCheckCurrent) {
>   static const char kQueueName[] = "PostAndCheckCurrent";
>-  Event event(false, false);
>+  Event event;
>   TaskQueue queue(kQueueName);
> 
>   // We're not running a task, so there shouldn't be a current queue.
>@@ -106,7 +106,7 @@ TEST(TaskQueueTest, PostLambda) {
> 
> TEST(TaskQueueTest, PostDelayedZero) {
>   static const char kQueueName[] = "PostDelayedZero";
>-  Event event(false, false);
>+  Event event;
>   TaskQueue queue(kQueueName);
> 
>   queue.PostDelayedTask([&event]() { event.Set(); }, 0);
>@@ -115,7 +115,7 @@ TEST(TaskQueueTest, PostDelayedZero) {
> 
> TEST(TaskQueueTest, PostFromQueue) {
>   static const char kQueueName[] = "PostFromQueue";
>-  Event event(false, false);
>+  Event event;
>   TaskQueue queue(kQueueName);
> 
>   queue.PostTask(
>@@ -125,7 +125,7 @@ TEST(TaskQueueTest, PostFromQueue) {
> 
> TEST(TaskQueueTest, PostDelayed) {
>   static const char kQueueName[] = "PostDelayed";
>-  Event event(false, false);
>+  Event event;
>   TaskQueue queue(kQueueName, TaskQueue::Priority::HIGH);
> 
>   uint32_t start = Time();
>@@ -145,7 +145,7 @@ TEST(TaskQueueTest, DISABLED_PostDelayedHighRes) {
>   EnableHighResTimers high_res_scope;
> 
>   static const char kQueueName[] = "PostDelayedHighRes";
>-  Event event(false, false);
>+  Event event;
>   TaskQueue queue(kQueueName, TaskQueue::Priority::HIGH);
> 
>   uint32_t start = Time();
>@@ -165,7 +165,7 @@ TEST(TaskQueueTest, PostMultipleDelayed) {
> 
>   std::vector<std::unique_ptr<Event>> events;
>   for (int i = 0; i < 100; ++i) {
>-    events.push_back(std::unique_ptr<Event>(new Event(false, false)));
>+    events.push_back(absl::make_unique<Event>());
>     queue.PostDelayedTask(Bind(&CheckCurrent, events.back().get(), &queue), i);
>   }
> 
>@@ -175,8 +175,8 @@ TEST(TaskQueueTest, PostMultipleDelayed) {
> 
> TEST(TaskQueueTest, PostDelayedAfterDestruct) {
>   static const char kQueueName[] = "PostDelayedAfterDestruct";
>-  Event run(false, false);
>-  Event deleted(false, false);
>+  Event run;
>+  Event deleted;
>   {
>     TaskQueue queue(kQueueName);
>     queue.PostDelayedTask(
>@@ -191,7 +191,7 @@ TEST(TaskQueueTest, PostDelayedAfterDestruct) {
> TEST(TaskQueueTest, PostAndReply) {
>   static const char kPostQueue[] = "PostQueue";
>   static const char kReplyQueue[] = "ReplyQueue";
>-  Event event(false, false);
>+  Event event;
>   TaskQueue post_queue(kPostQueue);
>   TaskQueue reply_queue(kReplyQueue);
> 
>@@ -204,7 +204,7 @@ TEST(TaskQueueTest, PostAndReply) {
> TEST(TaskQueueTest, PostAndReuse) {
>   static const char kPostQueue[] = "PostQueue";
>   static const char kReplyQueue[] = "ReplyQueue";
>-  Event event(false, false);
>+  Event event;
>   TaskQueue post_queue(kPostQueue);
>   TaskQueue reply_queue(kReplyQueue);
> 
>@@ -251,7 +251,7 @@ TEST(TaskQueueTest, PostAndReuse) {
> TEST(TaskQueueTest, PostAndReplyLambda) {
>   static const char kPostQueue[] = "PostQueue";
>   static const char kReplyQueue[] = "ReplyQueue";
>-  Event event(false, false);
>+  Event event;
>   TaskQueue post_queue(kPostQueue);
>   TaskQueue reply_queue(kReplyQueue);
> 
>@@ -287,7 +287,7 @@ TEST(TaskQueueTest, PostCopyableClosure) {
> 
>   int num_copies = 0;
>   int num_moves = 0;
>-  Event event(false, false);
>+  Event event;
> 
>   static const char kPostQueue[] = "PostCopyableClosure";
>   TaskQueue post_queue(kPostQueue);
>@@ -323,7 +323,7 @@ TEST(TaskQueueTest, PostMoveOnlyClosure) {
>   };
> 
>   int num_moves = 0;
>-  Event event(false, false);
>+  Event event;
>   std::unique_ptr<SomeState> state(new SomeState(&event));
> 
>   static const char kPostQueue[] = "PostMoveOnlyClosure";
>@@ -346,8 +346,8 @@ TEST(TaskQueueTest, PostMoveOnlyCleanup) {
>     std::unique_ptr<SomeState> state;
>   };
> 
>-  Event event_run(false, false);
>-  Event event_cleanup(false, false);
>+  Event event_run;
>+  Event event_cleanup;
>   std::unique_ptr<SomeState> state_run(new SomeState(&event_run));
>   std::unique_ptr<SomeState> state_cleanup(new SomeState(&event_cleanup));
> 
>@@ -367,7 +367,7 @@ TEST(TaskQueueTest, PostMoveOnlyCleanup) {
> // written in a way that makes it likely and by running with --gtest_repeat=1000
> // the bug would occur. Alas, now it should be fixed.
> TEST(TaskQueueTest, PostAndReplyDeadlock) {
>-  Event event(false, false);
>+  Event event;
>   TaskQueue post_queue("PostQueue");
>   TaskQueue reply_queue("ReplyQueue");
> 
>@@ -384,8 +384,8 @@ TEST(TaskQueueTest, PostAndReplyDeadlock) {
> #define MAYBE_DeleteTaskQueueAfterPostAndReply DeleteTaskQueueAfterPostAndReply
> #endif
> TEST(TaskQueueTest, MAYBE_DeleteTaskQueueAfterPostAndReply) {
>-  Event task_deleted(false, false);
>-  Event reply_deleted(false, false);
>+  Event task_deleted;
>+  Event reply_deleted;
>   auto* task_queue = new TaskQueue("Queue");
> 
>   task_queue->PostTaskAndReply(
>@@ -413,7 +413,7 @@ void TestPostTaskAndReply(TaskQueue* work_queue, Event* event) {
> TEST(TaskQueueTest, PostAndReply2) {
>   static const char kQueueName[] = "PostAndReply2";
>   static const char kWorkQueueName[] = "PostAndReply2_Worker";
>-  Event event(false, false);
>+  Event event;
>   TaskQueue queue(kQueueName);
>   TaskQueue work_queue(kWorkQueueName);
> 
>@@ -425,7 +425,7 @@ TEST(TaskQueueTest, PostAndReply2) {
> // In situations like that, tasks will get dropped.
> TEST(TaskQueueTest, PostALot) {
>   // To destruct the event after the queue has gone out of scope.
>-  Event event(false, false);
>+  Event event;
> 
>   int tasks_executed = 0;
>   int tasks_cleaned_up = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.cc
>index 7c151c75101e7b0587faac802d56031da4d4254f..a5b90ddd2017dd6f280bfac7d7a5a62a796eaa7c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.cc
>@@ -97,7 +97,7 @@ bool TestClient::CheckNextPacket(const char* buf,
>   std::unique_ptr<Packet> packet = NextPacket(kTimeoutMs);
>   if (packet) {
>     res = (packet->size == size && memcmp(packet->buf, buf, size) == 0 &&
>-           CheckTimestamp(packet->packet_time.timestamp));
>+           CheckTimestamp(packet->packet_time_us));
>     if (addr)
>       *addr = packet->addr;
>   }
>@@ -144,10 +144,10 @@ void TestClient::OnPacket(AsyncPacketSocket* socket,
>                           const char* buf,
>                           size_t size,
>                           const SocketAddress& remote_addr,
>-                          const PacketTime& packet_time) {
>+                          const int64_t& packet_time_us) {
>   CritScope cs(&crit_);
>   packets_.push_back(
>-      absl::make_unique<Packet>(remote_addr, buf, size, packet_time));
>+      absl::make_unique<Packet>(remote_addr, buf, size, packet_time_us));
> }
> 
> void TestClient::OnReadyToSend(AsyncPacketSocket* socket) {
>@@ -157,14 +157,14 @@ void TestClient::OnReadyToSend(AsyncPacketSocket* socket) {
> TestClient::Packet::Packet(const SocketAddress& a,
>                            const char* b,
>                            size_t s,
>-                           const PacketTime& packet_time)
>-    : addr(a), buf(0), size(s), packet_time(packet_time) {
>+                           int64_t packet_time_us)
>+    : addr(a), buf(0), size(s), packet_time_us(packet_time_us) {
>   buf = new char[size];
>   memcpy(buf, b, size);
> }
> 
> TestClient::Packet::Packet(const Packet& p)
>-    : addr(p.addr), buf(0), size(p.size), packet_time(p.packet_time) {
>+    : addr(p.addr), buf(0), size(p.size), packet_time_us(p.packet_time_us) {
>   buf = new char[size];
>   memcpy(buf, p.buf, size);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.h
>index a0d98b30d1654bc60d2a3d9741c89da141db52ad..16fb6ba677c2226df898b09eb3db9b4a7c952689 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.h
>@@ -29,14 +29,14 @@ class TestClient : public sigslot::has_slots<> {
>     Packet(const SocketAddress& a,
>            const char* b,
>            size_t s,
>-           const PacketTime& packet_time);
>+           int64_t packet_time_us);
>     Packet(const Packet& p);
>     virtual ~Packet();
> 
>     SocketAddress addr;
>     char* buf;
>     size_t size;
>-    PacketTime packet_time;
>+    int64_t packet_time_us;
>   };
> 
>   // Default timeout for NextPacket reads.
>@@ -97,7 +97,7 @@ class TestClient : public sigslot::has_slots<> {
>                 const char* buf,
>                 size_t len,
>                 const SocketAddress& remote_addr,
>-                const PacketTime& packet_time);
>+                const int64_t& packet_time_us);
>   void OnReadyToSend(AsyncPacketSocket* socket);
>   bool CheckTimestamp(int64_t packet_timestamp);
>   void AdvanceTime(int ms);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testechoserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testechoserver.h
>index 5e714eb63dd30e0e69675b1e0df9fe479f2aff4a..9ffd78622a45f7202c89752b1e3292724da0b8ec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testechoserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testechoserver.h
>@@ -44,7 +44,7 @@ class TestEchoServer : public sigslot::has_slots<> {
>                 const char* buf,
>                 size_t size,
>                 const SocketAddress& remote_addr,
>-                const PacketTime& packet_time) {
>+                const int64_t& /* packet_time_us */) {
>     rtc::PacketOptions options;
>     socket->Send(buf, size, options);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.cc
>index b4a7433fbd7f183050b4626f558d7b32054c877f..f3292fd59f7cee0b20e7f81992ec21e3257e2400 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.cc
>@@ -14,75 +14,7 @@ namespace webrtc {
> namespace testing {
> 
> StreamSink::StreamSink() = default;
>-
> StreamSink::~StreamSink() = default;
> 
>-StreamSource::StreamSource() {
>-  Clear();
>-}
>-
>-StreamSource::~StreamSource() = default;
>-
>-StreamState StreamSource::GetState() const {
>-  return state_;
>-}
>-
>-StreamResult StreamSource::Read(void* buffer,
>-                                size_t buffer_len,
>-                                size_t* read,
>-                                int* error) {
>-  if (SS_CLOSED == state_) {
>-    if (error)
>-      *error = -1;
>-    return SR_ERROR;
>-  }
>-  if ((SS_OPENING == state_) || (readable_data_.size() <= read_block_)) {
>-    return SR_BLOCK;
>-  }
>-  size_t count = std::min(buffer_len, readable_data_.size() - read_block_);
>-  memcpy(buffer, &readable_data_[0], count);
>-  size_t new_size = readable_data_.size() - count;
>-  // Avoid undefined access beyond the last element of the vector.
>-  // This only happens when new_size is 0.
>-  if (count < readable_data_.size()) {
>-    memmove(&readable_data_[0], &readable_data_[count], new_size);
>-  }
>-  readable_data_.resize(new_size);
>-  if (read)
>-    *read = count;
>-  return SR_SUCCESS;
>-}
>-
>-StreamResult StreamSource::Write(const void* data,
>-                                 size_t data_len,
>-                                 size_t* written,
>-                                 int* error) {
>-  if (SS_CLOSED == state_) {
>-    if (error)
>-      *error = -1;
>-    return SR_ERROR;
>-  }
>-  if (SS_OPENING == state_) {
>-    return SR_BLOCK;
>-  }
>-  if (SIZE_UNKNOWN != write_block_) {
>-    if (written_data_.size() >= write_block_) {
>-      return SR_BLOCK;
>-    }
>-    if (data_len > (write_block_ - written_data_.size())) {
>-      data_len = write_block_ - written_data_.size();
>-    }
>-  }
>-  if (written)
>-    *written = data_len;
>-  const char* cdata = static_cast<const char*>(data);
>-  written_data_.insert(written_data_.end(), cdata, cdata + data_len);
>-  return SR_SUCCESS;
>-}
>-
>-void StreamSource::Close() {
>-  state_ = SS_CLOSED;
>-}
>-
> }  // namespace testing
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.h
>index 2ab4a35956a059eacb743915b5ae28a7254cb5b8..ac74203cf9f93cf7baa2a02e2d69ca6982dd0b61 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.h
>@@ -19,9 +19,7 @@
> #include <vector>
> #include "rtc_base/asyncsocket.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/gunit.h"
> #include "rtc_base/stream.h"
>-#include "rtc_base/stringutils.h"
> 
> namespace webrtc {
> namespace testing {
>@@ -137,91 +135,6 @@ class StreamSink : public sigslot::has_slots<> {
>   EventMap events_;
> };
> 
>-///////////////////////////////////////////////////////////////////////////////
>-// StreamSource - Implements stream interface and simulates asynchronous
>-// events on the stream, without a network.  Also buffers written data.
>-///////////////////////////////////////////////////////////////////////////////
>-
>-class StreamSource : public StreamInterface {
>- public:
>-  StreamSource();
>-  ~StreamSource() override;
>-
>-  void Clear() {
>-    readable_data_.clear();
>-    written_data_.clear();
>-    state_ = SS_CLOSED;
>-    read_block_ = 0;
>-    write_block_ = SIZE_UNKNOWN;
>-  }
>-  void QueueString(const char* data) { QueueData(data, strlen(data)); }
>-#if defined(__GNUC__)
>-  // Note: Implicit |this| argument counts as the first argument.
>-  __attribute__((__format__(__printf__, 2, 3)))
>-#endif
>-  void
>-  QueueStringF(const char* format, ...) {
>-    va_list args;
>-    va_start(args, format);
>-    char buffer[1024];
>-    size_t len = vsprintfn(buffer, sizeof(buffer), format, args);
>-    RTC_CHECK(len < sizeof(buffer) - 1);
>-    va_end(args);
>-    QueueData(buffer, len);
>-  }
>-  void QueueData(const char* data, size_t len) {
>-    readable_data_.insert(readable_data_.end(), data, data + len);
>-    if ((SS_OPEN == state_) && (readable_data_.size() == len)) {
>-      SignalEvent(this, SE_READ, 0);
>-    }
>-  }
>-  std::string ReadData() {
>-    std::string data;
>-    // avoid accessing written_data_[0] if it is undefined
>-    if (written_data_.size() > 0) {
>-      data.insert(0, &written_data_[0], written_data_.size());
>-    }
>-    written_data_.clear();
>-    return data;
>-  }
>-  void SetState(StreamState state) {
>-    int events = 0;
>-    if ((SS_OPENING == state_) && (SS_OPEN == state)) {
>-      events |= SE_OPEN;
>-      if (!readable_data_.empty()) {
>-        events |= SE_READ;
>-      }
>-    } else if ((SS_CLOSED != state_) && (SS_CLOSED == state)) {
>-      events |= SE_CLOSE;
>-    }
>-    state_ = state;
>-    if (events) {
>-      SignalEvent(this, events, 0);
>-    }
>-  }
>-  // Will cause Read to block when there are pos bytes in the read queue.
>-  void SetReadBlock(size_t pos) { read_block_ = pos; }
>-  // Will cause Write to block when there are pos bytes in the write queue.
>-  void SetWriteBlock(size_t pos) { write_block_ = pos; }
>-
>-  StreamState GetState() const override;
>-  StreamResult Read(void* buffer,
>-                    size_t buffer_len,
>-                    size_t* read,
>-                    int* error) override;
>-  StreamResult Write(const void* data,
>-                     size_t data_len,
>-                     size_t* written,
>-                     int* error) override;
>-  void Close() override;
>-
>- private:
>-  typedef std::vector<char> Buffer;
>-  Buffer readable_data_, written_data_;
>-  StreamState state_;
>-  size_t read_block_, write_block_;
>-};
>-
> }  // namespace testing
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/sigslot.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/sigslot.h
>index c77e4e675df504a084223932227eac8b54fbb435..8bd1c7064a3841e3aabb7cc5fd71ed4d7806548e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/sigslot.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/sigslot.h
>@@ -96,7 +96,6 @@
> #ifndef RTC_BASE_THIRD_PARTY_SIGSLOT_SIGSLOT_H_
> #define RTC_BASE_THIRD_PARTY_SIGSLOT_SIGSLOT_H_
> 
>-#include <stdlib.h>
> #include <cstring>
> #include <list>
> #include <set>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc
>index 2d5704e0be780a4748ee254f723423b42f1669f7..4dd5fd2224f12a6aaeb52ae81cfb845b2bdb4885 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc
>@@ -24,14 +24,43 @@
> #pragma warning(disable : 4722)
> #endif
> 
>+#include <stdio.h>
>+#include <utility>
>+
> #include "rtc_base/checks.h"
>+#include "rtc_base/criticalsection.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/nullsocketserver.h"
>-#include "rtc_base/platform_thread.h"
>-#include "rtc_base/stringutils.h"
> #include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
> 
>+#if defined(WEBRTC_MAC)
>+#include "rtc_base/system/cocoa_threading.h"
>+
>+/*
>+ * These are forward-declarations for methods that are part of the
>+ * ObjC runtime. They are declared in the private header objc-internal.h.
>+ * These calls are what clang inserts when using @autoreleasepool in ObjC,
>+ * but here they are used directly in order to keep this file C++.
>+ * https://clang.llvm.org/docs/AutomaticReferenceCounting.html#runtime-support
>+ */
>+extern "C" {
>+void* objc_autoreleasePoolPush(void);
>+void objc_autoreleasePoolPop(void* pool);
>+}
>+
>+namespace {
>+class ScopedAutoReleasePool {
>+ public:
>+  ScopedAutoReleasePool() : pool_(objc_autoreleasePoolPush()) {}
>+  ~ScopedAutoReleasePool() { objc_autoreleasePoolPop(pool_); }
>+
>+ private:
>+  void* const pool_;
>+};
>+}  // namespace
>+#endif
>+
> namespace rtc {
> 
> ThreadManager* ThreadManager::Instance() {
>@@ -61,11 +90,12 @@ Thread* Thread::Current() {
> }
> 
> #if defined(WEBRTC_POSIX)
>-#if !defined(WEBRTC_MAC)
> ThreadManager::ThreadManager() : main_thread_ref_(CurrentThreadRef()) {
>+#if defined(WEBRTC_MAC)
>+  InitCocoaMultiThreading();
>+#endif
>   pthread_key_create(&key_, nullptr);
> }
>-#endif
> 
> Thread* ThreadManager::CurrentThread() {
>   return static_cast<Thread*>(pthread_getspecific(key_));
>@@ -193,8 +223,10 @@ bool Thread::SetName(const std::string& name, const void* obj) {
> 
>   name_ = name;
>   if (obj) {
>-    char buf[16];
>-    sprintfn(buf, sizeof(buf), " 0x%p", obj);
>+    // The %p specifier typically produce at most 16 hex digits, possibly with a
>+    // 0x prefix. But format is implementation defined, so add some margin.
>+    char buf[30];
>+    snprintf(buf, sizeof(buf), " 0x%p", obj);
>     name_ += buf;
>   }
>   return true;
>@@ -300,7 +332,6 @@ void Thread::AssertBlockingIsAllowedOnCurrentThread() {
> }
> 
> // static
>-#if !defined(WEBRTC_MAC)
> #if defined(WEBRTC_WIN)
> DWORD WINAPI Thread::PreRun(LPVOID pv) {
> #else
>@@ -309,6 +340,9 @@ void* Thread::PreRun(void* pv) {
>   ThreadInit* init = static_cast<ThreadInit*>(pv);
>   ThreadManager::Instance()->SetCurrentThread(init->thread);
>   rtc::SetCurrentThreadName(init->thread->name_.c_str());
>+#if defined(WEBRTC_MAC)
>+  ScopedAutoReleasePool pool;
>+#endif
>   if (init->runnable) {
>     init->runnable->Run(init->thread);
>   } else {
>@@ -322,7 +356,6 @@ void* Thread::PreRun(void* pv) {
>   return nullptr;
> #endif
> }
>-#endif
> 
> void Thread::Run() {
>   ProcessMessages(kForever);
>@@ -485,9 +518,6 @@ void Thread::Clear(MessageHandler* phandler,
>   ClearInternal(phandler, id, removed);
> }
> 
>-#if !defined(WEBRTC_MAC)
>-// Note that these methods have a separate implementation for mac and ios
>-// defined in webrtc/rtc_base/thread_darwin.mm.
> bool Thread::ProcessMessages(int cmsLoop) {
>   // Using ProcessMessages with a custom clock for testing and a time greater
>   // than 0 doesn't work, since it's not guaranteed to advance the custom
>@@ -498,6 +528,9 @@ bool Thread::ProcessMessages(int cmsLoop) {
>   int cmsNext = cmsLoop;
> 
>   while (true) {
>+#if defined(WEBRTC_MAC)
>+    ScopedAutoReleasePool pool;
>+#endif
>     Message msg;
>     if (!Get(&msg, cmsNext))
>       return !IsQuitting();
>@@ -510,7 +543,6 @@ bool Thread::ProcessMessages(int cmsLoop) {
>     }
>   }
> }
>-#endif
> 
> bool Thread::WrapCurrentWithThreadManager(ThreadManager* thread_manager,
>                                           bool need_synchronize_access) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.h
>index 5a466104aa273e774adc7525cac59628e94e6236..039192ce7f674695ab2f1c758ec0c85497f2aa85 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.h
>@@ -11,19 +11,22 @@
> #ifndef RTC_BASE_THREAD_H_
> #define RTC_BASE_THREAD_H_
> 
>-#include <algorithm>
>+#include <stdint.h>
> #include <list>
> #include <memory>
> #include <string>
>-#include <utility>
>-#include <vector>
>+#include <type_traits>
> 
> #if defined(WEBRTC_POSIX)
> #include <pthread.h>
> #endif
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/location.h"
>+#include "rtc_base/messagehandler.h"
> #include "rtc_base/messagequeue.h"
> #include "rtc_base/platform_thread_types.h"
>+#include "rtc_base/socketserver.h"
>+#include "rtc_base/thread_annotations.h"
> 
> #if defined(WEBRTC_WIN)
> #include "rtc_base/win32.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_darwin.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_darwin.mm
>deleted file mode 100644
>index a404849c72b4cefd56f1f5457092cff0a23d7aa8..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_darwin.mm
>+++ /dev/null
>@@ -1,84 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/thread.h"
>-
>-#import <Foundation/Foundation.h>
>-
>-#include "rtc_base/platform_thread.h"
>-
>-/*
>- * This file contains platform-specific implementations for several
>- * methods in rtc::Thread.
>- */
>-
>-namespace {
>-void InitCocoaMultiThreading() {
>-  if ([NSThread isMultiThreaded] == NO) {
>-    // The sole purpose of this autorelease pool is to avoid a console
>-    // message on Leopard that tells us we're autoreleasing the thread
>-    // with no autorelease pool in place.
>-    @autoreleasepool {
>-      [NSThread detachNewThreadSelector:@selector(class)
>-                               toTarget:[NSObject class]
>-                             withObject:nil];
>-    }
>-  }
>-
>-  RTC_DCHECK([NSThread isMultiThreaded]);
>-}
>-}
>-
>-namespace rtc {
>-
>-ThreadManager::ThreadManager() : main_thread_ref_(CurrentThreadRef()) {
>-  pthread_key_create(&key_, nullptr);
>-  // This is necessary to alert the cocoa runtime of the fact that
>-  // we are running in a multithreaded environment.
>-  InitCocoaMultiThreading();
>-}
>-
>-// static
>-void* Thread::PreRun(void* pv) {
>-  ThreadInit* init = static_cast<ThreadInit*>(pv);
>-  ThreadManager::Instance()->SetCurrentThread(init->thread);
>-  rtc::SetCurrentThreadName(init->thread->name_.c_str());
>-  @autoreleasepool {
>-    if (init->runnable) {
>-      init->runnable->Run(init->thread);
>-    } else {
>-      init->thread->Run();
>-    }
>-  }
>-  ThreadManager::Instance()->SetCurrentThread(nullptr);
>-  delete init;
>-  return nullptr;
>-}
>-
>-bool Thread::ProcessMessages(int cmsLoop) {
>-  int64_t msEnd = (kForever == cmsLoop) ? 0 : TimeAfter(cmsLoop);
>-  int cmsNext = cmsLoop;
>-
>-  while (true) {
>-    @autoreleasepool {
>-      Message msg;
>-      if (!Get(&msg, cmsNext))
>-        return !IsQuitting();
>-      Dispatch(&msg);
>-
>-      if (cmsLoop != kForever) {
>-        cmsNext = static_cast<int>(TimeUntil(msEnd));
>-        if (cmsNext < 0)
>-          return true;
>-      }
>-    }
>-  }
>-}
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_unittest.cc
>index d5c53f8a7cc0a27fc4fa647bf97d1cf6f29cc2ec..7e392affd94a5b0a7bdbe3809e3b6a68d71d3d67 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_unittest.cc
>@@ -69,7 +69,7 @@ class SocketClient : public TestGenerator, public sigslot::has_slots<> {
>                 const char* buf,
>                 size_t size,
>                 const SocketAddress& remote_addr,
>-                const PacketTime& packet_time) {
>+                const int64_t& packet_time_us) {
>     EXPECT_EQ(size, sizeof(uint32_t));
>     uint32_t prev = reinterpret_cast<const uint32_t*>(buf)[0];
>     uint32_t result = Next(prev);
>@@ -470,9 +470,9 @@ TEST_F(AsyncInvokeTest, KillInvokerDuringExecute) {
>   // Use these events to get in a state where the functor is in the middle of
>   // executing, and then to wait for it to finish, ensuring the "EXPECT_FALSE"
>   // is run.
>-  Event functor_started(false, false);
>-  Event functor_continue(false, false);
>-  Event functor_finished(false, false);
>+  Event functor_started;
>+  Event functor_continue;
>+  Event functor_finished;
> 
>   auto thread = Thread::CreateWithSocketServer();
>   thread->Start();
>@@ -507,7 +507,7 @@ TEST_F(AsyncInvokeTest, KillInvokerDuringExecute) {
> // destroyed. This shouldn't deadlock or crash; this second invocation should
> // just be ignored.
> TEST_F(AsyncInvokeTest, KillInvokerDuringExecuteWithReentrantInvoke) {
>-  Event functor_started(false, false);
>+  Event functor_started;
>   // Flag used to verify that the recursively invoked task never actually runs.
>   bool reentrant_functor_run = false;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timestampaligner.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timestampaligner.cc
>index a9bcafbdc5f5adcbd2f6e65e6e85a39ec8a868e8..f2da101727110d47aeb99c0de2cc43b434ce61a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timestampaligner.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timestampaligner.cc
>@@ -8,6 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <cstdlib>
> #include <limits>
> 
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.cc
>index de8fc3259c28ec4e7c07df5a56f8e49733609737..dc5b611f647bec9452206de052d7e5fb69f4593c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.cc
>@@ -14,6 +14,7 @@
> #include <sys/time.h>
> #if defined(WEBRTC_MAC)
> #include <mach/mach_time.h>
>+#include "rtc_base/numerics/safe_conversions.h"
> #endif
> #endif
> 
>@@ -28,7 +29,6 @@
> #endif
> 
> #include "rtc_base/checks.h"
>-#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/timeutils.h"
> 
> namespace rtc {
>@@ -154,7 +154,7 @@ int64_t TimestampWrapAroundHandler::Unwrap(uint32_t ts) {
>   return ts + (num_wrap_ << 32);
> }
> 
>-int64_t TmToSeconds(const std::tm& tm) {
>+int64_t TmToSeconds(const tm& tm) {
>   static short int mdays[12] = {31, 28, 31, 30, 31, 30, 31, 31, 30, 31, 30, 31};
>   static short int cumul_mdays[12] = {0,   31,  59,  90,  120, 151,
>                                       181, 212, 243, 273, 304, 334};
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.h
>index 3a412a47cf8425bc9f1a4e3300d56d4501cca5ac..4e38a031409f0f05ab5838774bc7eef10a6907b3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.h
>@@ -13,8 +13,6 @@
> 
> #include <stdint.h>
> #include <time.h>
>-
>-#include <ctime>
> #include <string>
> 
> #include "rtc_base/checks.h"
>@@ -110,10 +108,10 @@ class TimestampWrapAroundHandler {
>   int64_t num_wrap_;
> };
> 
>-// Convert from std::tm, which is relative to 1900-01-01 00:00 to number of
>-// seconds from 1970-01-01 00:00 ("epoch").  Don't return time_t since that
>+// Convert from tm, which is relative to 1900-01-01 00:00 to number of
>+// seconds from 1970-01-01 00:00 ("epoch"). Don't return time_t since that
> // is still 32 bits on many systems.
>-int64_t TmToSeconds(const std::tm& tm);
>+int64_t TmToSeconds(const tm& tm);
> 
> // Return the number of microseconds since January 1, 1970, UTC.
> // Useful mainly when producing logs to be correlated with other
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils_unittest.cc
>index 0e1949aee31125bd36a721200338416e02b07c96..577efda0255b03cb679f4595bdebc5559cf482f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils_unittest.cc
>@@ -262,7 +262,7 @@ TEST(FakeClock, SettingTimeWakesThreads) {
>   worker->Start();
> 
>   // Post an event that won't be executed for 10 seconds.
>-  Event message_handler_dispatched(false, false);
>+  Event message_handler_dispatched;
>   auto functor = [&message_handler_dispatched] {
>     message_handler_dispatched.Set();
>   };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/units/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/units/BUILD.gn
>new file mode 100644
>index 0000000000000000000000000000000000000000..8c722bbc7b8e70fb206f806c112b61ddbfe985b3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/units/BUILD.gn
>@@ -0,0 +1,37 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../webrtc.gni")
>+
>+rtc_source_set("unit_base") {
>+  visibility = [
>+    "../../api/units:*",
>+    ":*",
>+  ]
>+  sources = [
>+    "unit_base.h",
>+  ]
>+
>+  deps = [
>+    "../../rtc_base:checks",
>+    "../../rtc_base:safe_conversions",
>+  ]
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("units_unittests") {
>+    testonly = true
>+    sources = [
>+      "unit_base_unittest.cc",
>+    ]
>+    deps = [
>+      ":unit_base",
>+      "../../test:test_support",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/units/unit_base.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/units/unit_base.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..5503a329937667642cfa70f7a17d29e333b4f064
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/units/unit_base.h
>@@ -0,0 +1,304 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef RTC_BASE_UNITS_UNIT_BASE_H_
>+#define RTC_BASE_UNITS_UNIT_BASE_H_
>+
>+#include <stdint.h>
>+#include <algorithm>
>+#include <cmath>
>+#include <limits>
>+#include <type_traits>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+
>+namespace webrtc {
>+namespace rtc_units_impl {
>+
>+// UnitBase is a base class for implementing custom value types with a specific
>+// unit. It provides type safety and sommonly useful operations. The undelying
>+// storage is always an int64_t, it's up to the unit implementation to choose
>+// what scale it represents.
>+//
>+// It's used like:
>+// class MyUnit: public UnitBase<MyUnit> {...};
>+//
>+// Unit_T is the subclass representing the specific unit.
>+template <class Unit_T>
>+class UnitBase {
>+ public:
>+  UnitBase() = delete;
>+  static constexpr Unit_T Zero() { return Unit_T(0); }
>+  static constexpr Unit_T PlusInfinity() { return Unit_T(PlusInfinityVal()); }
>+  static constexpr Unit_T MinusInfinity() { return Unit_T(MinusInfinityVal()); }
>+
>+  constexpr bool IsZero() const { return value_ == 0; }
>+  constexpr bool IsFinite() const { return !IsInfinite(); }
>+  constexpr bool IsInfinite() const {
>+    return value_ == PlusInfinityVal() || value_ == MinusInfinityVal();
>+  }
>+  constexpr bool IsPlusInfinity() const { return value_ == PlusInfinityVal(); }
>+  constexpr bool IsMinusInfinity() const {
>+    return value_ == MinusInfinityVal();
>+  }
>+
>+  constexpr bool operator==(const Unit_T& other) const {
>+    return value_ == other.value_;
>+  }
>+  constexpr bool operator!=(const Unit_T& other) const {
>+    return value_ != other.value_;
>+  }
>+  constexpr bool operator<=(const Unit_T& other) const {
>+    return value_ <= other.value_;
>+  }
>+  constexpr bool operator>=(const Unit_T& other) const {
>+    return value_ >= other.value_;
>+  }
>+  constexpr bool operator>(const Unit_T& other) const {
>+    return value_ > other.value_;
>+  }
>+  constexpr bool operator<(const Unit_T& other) const {
>+    return value_ < other.value_;
>+  }
>+
>+ protected:
>+  template <int64_t value>
>+  static constexpr Unit_T FromStaticValue() {
>+    static_assert(value >= 0 || !Unit_T::one_sided, "");
>+    static_assert(value > MinusInfinityVal(), "");
>+    static_assert(value < PlusInfinityVal(), "");
>+    return Unit_T(value);
>+  }
>+
>+  template <int64_t fraction_value, int64_t Denominator>
>+  static constexpr Unit_T FromStaticFraction() {
>+    static_assert(fraction_value >= 0 || !Unit_T::one_sided, "");
>+    static_assert(fraction_value > MinusInfinityVal() / Denominator, "");
>+    static_assert(fraction_value < PlusInfinityVal() / Denominator, "");
>+    return Unit_T(fraction_value * Denominator);
>+  }
>+
>+  template <
>+      typename T,
>+      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>+  static Unit_T FromValue(T value) {
>+    if (Unit_T::one_sided)
>+      RTC_DCHECK_GE(value, 0);
>+    RTC_DCHECK_GT(value, MinusInfinityVal());
>+    RTC_DCHECK_LT(value, PlusInfinityVal());
>+    return Unit_T(rtc::dchecked_cast<int64_t>(value));
>+  }
>+  template <typename T,
>+            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>+                nullptr>
>+  static Unit_T FromValue(T value) {
>+    if (value == std::numeric_limits<T>::infinity()) {
>+      return PlusInfinity();
>+    } else if (value == -std::numeric_limits<T>::infinity()) {
>+      return MinusInfinity();
>+    } else {
>+      RTC_DCHECK(!std::isnan(value));
>+      return FromValue(rtc::dchecked_cast<int64_t>(value));
>+    }
>+  }
>+
>+  template <
>+      int64_t Denominator,
>+      typename T,
>+      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>+  static Unit_T FromFraction(T value) {
>+    if (Unit_T::one_sided)
>+      RTC_DCHECK_GE(value, 0);
>+    RTC_DCHECK_GT(value, MinusInfinityVal() / Denominator);
>+    RTC_DCHECK_LT(value, PlusInfinityVal() / Denominator);
>+    return Unit_T(rtc::dchecked_cast<int64_t>(value * Denominator));
>+  }
>+  template <int64_t Denominator,
>+            typename T,
>+            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>+                nullptr>
>+  static Unit_T FromFraction(T value) {
>+    return FromValue(value * Denominator);
>+  }
>+
>+  template <typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type ToValue() const {
>+    RTC_DCHECK(IsFinite());
>+    return rtc::dchecked_cast<T>(value_);
>+  }
>+  template <typename T>
>+  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>+  ToValue() const {
>+    return IsPlusInfinity()
>+               ? std::numeric_limits<T>::infinity()
>+               : IsMinusInfinity() ? -std::numeric_limits<T>::infinity()
>+                                   : value_;
>+  }
>+  template <typename T>
>+  constexpr T ToValueOr(T fallback_value) const {
>+    return IsFinite() ? value_ : fallback_value;
>+  }
>+
>+  template <int64_t Denominator, typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type ToFraction()
>+      const {
>+    RTC_DCHECK(IsFinite());
>+    if (Unit_T::one_sided) {
>+      return rtc::dchecked_cast<T>(
>+          DivRoundPositiveToNearest(value_, Denominator));
>+    } else {
>+      return rtc::dchecked_cast<T>(DivRoundToNearest(value_, Denominator));
>+    }
>+  }
>+  template <int64_t Denominator, typename T>
>+  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>+  ToFraction() const {
>+    return ToValue<T>() * (1 / static_cast<T>(Denominator));
>+  }
>+
>+  template <int64_t Denominator>
>+  constexpr int64_t ToFractionOr(int64_t fallback_value) const {
>+    return IsFinite() ? Unit_T::one_sided
>+                            ? DivRoundPositiveToNearest(value_, Denominator)
>+                            : DivRoundToNearest(value_, Denominator)
>+                      : fallback_value;
>+  }
>+
>+  template <int64_t Factor, typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type ToMultiple()
>+      const {
>+    RTC_DCHECK_GE(ToValue(), std::numeric_limits<T>::min() / Factor);
>+    RTC_DCHECK_LE(ToValue(), std::numeric_limits<T>::max() / Factor);
>+    return rtc::dchecked_cast<T>(ToValue() * Factor);
>+  }
>+  template <int64_t Factor, typename T>
>+  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>+  ToMultiple() const {
>+    return ToValue<T>() * Factor;
>+  }
>+
>+  explicit constexpr UnitBase(int64_t value) : value_(value) {}
>+
>+ private:
>+  template <class RelativeUnit_T>
>+  friend class RelativeUnit;
>+
>+  static inline constexpr int64_t PlusInfinityVal() {
>+    return std::numeric_limits<int64_t>::max();
>+  }
>+  static inline constexpr int64_t MinusInfinityVal() {
>+    return std::numeric_limits<int64_t>::min();
>+  }
>+
>+  Unit_T& AsSubClassRef() { return reinterpret_cast<Unit_T&>(*this); }
>+  constexpr const Unit_T& AsSubClassRef() const {
>+    return reinterpret_cast<const Unit_T&>(*this);
>+  }
>+  // Assumes that n >= 0 and d > 0.
>+  static constexpr int64_t DivRoundPositiveToNearest(int64_t n, int64_t d) {
>+    return (n + d / 2) / d;
>+  }
>+  // Assumes that d > 0.
>+  static constexpr int64_t DivRoundToNearest(int64_t n, int64_t d) {
>+    return (n + (n >= 0 ? d / 2 : -d / 2)) / d;
>+  }
>+
>+  int64_t value_;
>+};
>+
>+// Extends UnitBase to provide operations for relative units, that is, units
>+// that have a meaningful relation between values such that a += b is a
>+// sensible thing to do. For a,b <- same unit.
>+template <class Unit_T>
>+class RelativeUnit : public UnitBase<Unit_T> {
>+ public:
>+  Unit_T Clamped(Unit_T min_value, Unit_T max_value) const {
>+    return std::max(min_value,
>+                    std::min(UnitBase<Unit_T>::AsSubClassRef(), max_value));
>+  }
>+  void Clamp(Unit_T min_value, Unit_T max_value) {
>+    *this = Clamped(min_value, max_value);
>+  }
>+  Unit_T operator+(const Unit_T other) const {
>+    if (this->IsPlusInfinity() || other.IsPlusInfinity()) {
>+      RTC_DCHECK(!this->IsMinusInfinity());
>+      RTC_DCHECK(!other.IsMinusInfinity());
>+      return this->PlusInfinity();
>+    } else if (this->IsMinusInfinity() || other.IsMinusInfinity()) {
>+      RTC_DCHECK(!this->IsPlusInfinity());
>+      RTC_DCHECK(!other.IsPlusInfinity());
>+      return this->MinusInfinity();
>+    }
>+    return UnitBase<Unit_T>::FromValue(this->ToValue() + other.ToValue());
>+  }
>+  Unit_T operator-(const Unit_T other) const {
>+    if (this->IsPlusInfinity() || other.IsMinusInfinity()) {
>+      RTC_DCHECK(!this->IsMinusInfinity());
>+      RTC_DCHECK(!other.IsPlusInfinity());
>+      return this->PlusInfinity();
>+    } else if (this->IsMinusInfinity() || other.IsPlusInfinity()) {
>+      RTC_DCHECK(!this->IsPlusInfinity());
>+      RTC_DCHECK(!other.IsMinusInfinity());
>+      return this->MinusInfinity();
>+    }
>+    return UnitBase<Unit_T>::FromValue(this->ToValue() - other.ToValue());
>+  }
>+  Unit_T& operator+=(const Unit_T other) {
>+    *this = *this + other;
>+    return this->AsSubClassRef();
>+  }
>+  Unit_T& operator-=(const Unit_T other) {
>+    *this = *this - other;
>+    return this->AsSubClassRef();
>+  }
>+  constexpr double operator/(const Unit_T other) const {
>+    return UnitBase<Unit_T>::template ToValue<double>() /
>+           other.template ToValue<double>();
>+  }
>+  template <typename T>
>+  typename std::enable_if<std::is_arithmetic<T>::value, Unit_T>::type operator/(
>+      const T& scalar) const {
>+    return UnitBase<Unit_T>::FromValue(
>+        std::round(UnitBase<Unit_T>::template ToValue<int64_t>() / scalar));
>+  }
>+  Unit_T operator*(const double scalar) const {
>+    return UnitBase<Unit_T>::FromValue(std::round(this->ToValue() * scalar));
>+  }
>+  Unit_T operator*(const int64_t scalar) const {
>+    return UnitBase<Unit_T>::FromValue(this->ToValue() * scalar);
>+  }
>+  Unit_T operator*(const int32_t scalar) const {
>+    return UnitBase<Unit_T>::FromValue(this->ToValue() * scalar);
>+  }
>+
>+ protected:
>+  using UnitBase<Unit_T>::UnitBase;
>+};
>+
>+template <class Unit_T>
>+inline Unit_T operator*(const double scalar, const RelativeUnit<Unit_T> other) {
>+  return other * scalar;
>+}
>+template <class Unit_T>
>+inline Unit_T operator*(const int64_t scalar,
>+                        const RelativeUnit<Unit_T> other) {
>+  return other * scalar;
>+}
>+template <class Unit_T>
>+inline Unit_T operator*(const int32_t& scalar,
>+                        const RelativeUnit<Unit_T> other) {
>+  return other * scalar;
>+}
>+
>+}  // namespace rtc_units_impl
>+
>+}  // namespace webrtc
>+
>+#endif  // RTC_BASE_UNITS_UNIT_BASE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/units/unit_base_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/units/unit_base_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..f8c8503decb47833e49d0df16d2039cb0ac04cfd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/units/unit_base_unittest.cc
>@@ -0,0 +1,234 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/units/unit_base.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace {
>+class TestUnit final : public rtc_units_impl::RelativeUnit<TestUnit> {
>+ public:
>+  TestUnit() = delete;
>+
>+  using UnitBase::FromStaticValue;
>+  using UnitBase::FromValue;
>+  using UnitBase::ToValue;
>+  using UnitBase::ToValueOr;
>+
>+  template <int64_t kilo>
>+  static constexpr TestUnit FromStaticKilo() {
>+    return FromStaticFraction<kilo, 1000>();
>+  }
>+  template <typename T>
>+  static TestUnit FromKilo(T kilo) {
>+    return FromFraction<1000>(kilo);
>+  }
>+  template <typename T = int64_t>
>+  T ToKilo() const {
>+    return UnitBase::ToFraction<1000, T>();
>+  }
>+  constexpr int64_t ToKiloOr(int64_t fallback) const {
>+    return UnitBase::ToFractionOr<1000>(fallback);
>+  }
>+  template <typename T>
>+  constexpr T ToMilli() const {
>+    return UnitBase::ToMultiple<1000, T>();
>+  }
>+
>+ private:
>+  friend class UnitBase<TestUnit>;
>+  static constexpr bool one_sided = false;
>+  using RelativeUnit<TestUnit>::RelativeUnit;
>+};
>+}  // namespace
>+namespace test {
>+TEST(UnitBaseTest, ConstExpr) {
>+  constexpr int64_t kValue = -12345;
>+  constexpr TestUnit kTestUnitZero = TestUnit::Zero();
>+  constexpr TestUnit kTestUnitPlusInf = TestUnit::PlusInfinity();
>+  constexpr TestUnit kTestUnitMinusInf = TestUnit::MinusInfinity();
>+  static_assert(kTestUnitZero.IsZero(), "");
>+  static_assert(kTestUnitPlusInf.IsPlusInfinity(), "");
>+  static_assert(kTestUnitMinusInf.IsMinusInfinity(), "");
>+  static_assert(kTestUnitPlusInf.ToKiloOr(-1) == -1, "");
>+
>+  static_assert(kTestUnitPlusInf > kTestUnitZero, "");
>+
>+  constexpr TestUnit kTestUnitKilo = TestUnit::FromStaticKilo<kValue>();
>+  constexpr TestUnit kTestUnitValue = TestUnit::FromStaticValue<kValue>();
>+
>+  static_assert(kTestUnitKilo.ToKiloOr(0) == kValue, "");
>+  static_assert(kTestUnitValue.ToValueOr(0) == kValue, "");
>+}
>+
>+TEST(UnitBaseTest, GetBackSameValues) {
>+  const int64_t kValue = 499;
>+  for (int sign = -1; sign <= 1; ++sign) {
>+    int64_t value = kValue * sign;
>+    EXPECT_EQ(TestUnit::FromKilo(value).ToKilo(), value);
>+    EXPECT_EQ(TestUnit::FromValue(value).ToValue<int64_t>(), value);
>+  }
>+  EXPECT_EQ(TestUnit::Zero().ToValue<int64_t>(), 0);
>+}
>+
>+TEST(UnitBaseTest, GetDifferentPrefix) {
>+  const int64_t kValue = 3000000;
>+  EXPECT_EQ(TestUnit::FromValue(kValue).ToKilo(), kValue / 1000);
>+  EXPECT_EQ(TestUnit::FromKilo(kValue).ToValue<int64_t>(), kValue * 1000);
>+}
>+
>+TEST(UnitBaseTest, IdentityChecks) {
>+  const int64_t kValue = 3000;
>+  EXPECT_TRUE(TestUnit::Zero().IsZero());
>+  EXPECT_FALSE(TestUnit::FromKilo(kValue).IsZero());
>+
>+  EXPECT_TRUE(TestUnit::PlusInfinity().IsInfinite());
>+  EXPECT_TRUE(TestUnit::MinusInfinity().IsInfinite());
>+  EXPECT_FALSE(TestUnit::Zero().IsInfinite());
>+  EXPECT_FALSE(TestUnit::FromKilo(-kValue).IsInfinite());
>+  EXPECT_FALSE(TestUnit::FromKilo(kValue).IsInfinite());
>+
>+  EXPECT_FALSE(TestUnit::PlusInfinity().IsFinite());
>+  EXPECT_FALSE(TestUnit::MinusInfinity().IsFinite());
>+  EXPECT_TRUE(TestUnit::FromKilo(-kValue).IsFinite());
>+  EXPECT_TRUE(TestUnit::FromKilo(kValue).IsFinite());
>+  EXPECT_TRUE(TestUnit::Zero().IsFinite());
>+
>+  EXPECT_TRUE(TestUnit::PlusInfinity().IsPlusInfinity());
>+  EXPECT_FALSE(TestUnit::MinusInfinity().IsPlusInfinity());
>+
>+  EXPECT_TRUE(TestUnit::MinusInfinity().IsMinusInfinity());
>+  EXPECT_FALSE(TestUnit::PlusInfinity().IsMinusInfinity());
>+}
>+
>+TEST(UnitBaseTest, ComparisonOperators) {
>+  const int64_t kSmall = 450;
>+  const int64_t kLarge = 451;
>+  const TestUnit small = TestUnit::FromKilo(kSmall);
>+  const TestUnit large = TestUnit::FromKilo(kLarge);
>+
>+  EXPECT_EQ(TestUnit::Zero(), TestUnit::FromKilo(0));
>+  EXPECT_EQ(TestUnit::PlusInfinity(), TestUnit::PlusInfinity());
>+  EXPECT_EQ(small, TestUnit::FromKilo(kSmall));
>+  EXPECT_LE(small, TestUnit::FromKilo(kSmall));
>+  EXPECT_GE(small, TestUnit::FromKilo(kSmall));
>+  EXPECT_NE(small, TestUnit::FromKilo(kLarge));
>+  EXPECT_LE(small, TestUnit::FromKilo(kLarge));
>+  EXPECT_LT(small, TestUnit::FromKilo(kLarge));
>+  EXPECT_GE(large, TestUnit::FromKilo(kSmall));
>+  EXPECT_GT(large, TestUnit::FromKilo(kSmall));
>+  EXPECT_LT(TestUnit::Zero(), small);
>+  EXPECT_GT(TestUnit::Zero(), TestUnit::FromKilo(-kSmall));
>+  EXPECT_GT(TestUnit::Zero(), TestUnit::FromKilo(-kSmall));
>+
>+  EXPECT_GT(TestUnit::PlusInfinity(), large);
>+  EXPECT_LT(TestUnit::MinusInfinity(), TestUnit::Zero());
>+}
>+
>+TEST(UnitBaseTest, Clamping) {
>+  const TestUnit upper = TestUnit::FromKilo(800);
>+  const TestUnit lower = TestUnit::FromKilo(100);
>+  const TestUnit under = TestUnit::FromKilo(100);
>+  const TestUnit inside = TestUnit::FromKilo(500);
>+  const TestUnit over = TestUnit::FromKilo(1000);
>+  EXPECT_EQ(under.Clamped(lower, upper), lower);
>+  EXPECT_EQ(inside.Clamped(lower, upper), inside);
>+  EXPECT_EQ(over.Clamped(lower, upper), upper);
>+
>+  TestUnit mutable_delta = lower;
>+  mutable_delta.Clamp(lower, upper);
>+  EXPECT_EQ(mutable_delta, lower);
>+  mutable_delta = inside;
>+  mutable_delta.Clamp(lower, upper);
>+  EXPECT_EQ(mutable_delta, inside);
>+  mutable_delta = over;
>+  mutable_delta.Clamp(lower, upper);
>+  EXPECT_EQ(mutable_delta, upper);
>+}
>+
>+TEST(UnitBaseTest, CanBeInititializedFromLargeInt) {
>+  const int kMaxInt = std::numeric_limits<int>::max();
>+  EXPECT_EQ(TestUnit::FromKilo(kMaxInt).ToValue<int64_t>(),
>+            static_cast<int64_t>(kMaxInt) * 1000);
>+}
>+
>+TEST(UnitBaseTest, ConvertsToAndFromDouble) {
>+  const int64_t kValue = 17017;
>+  const double kMilliDouble = kValue * 1e3;
>+  const double kValueDouble = kValue;
>+  const double kKiloDouble = kValue * 1e-3;
>+
>+  EXPECT_EQ(TestUnit::FromValue(kValue).ToKilo<double>(), kKiloDouble);
>+  EXPECT_EQ(TestUnit::FromKilo(kKiloDouble).ToValue<int64_t>(), kValue);
>+
>+  EXPECT_EQ(TestUnit::FromValue(kValue).ToValue<double>(), kValueDouble);
>+  EXPECT_EQ(TestUnit::FromValue(kValueDouble).ToValue<int64_t>(), kValue);
>+
>+  EXPECT_NEAR(TestUnit::FromValue(kValue).ToMilli<double>(), kMilliDouble, 1);
>+
>+  const double kPlusInfinity = std::numeric_limits<double>::infinity();
>+  const double kMinusInfinity = -kPlusInfinity;
>+
>+  EXPECT_EQ(TestUnit::PlusInfinity().ToKilo<double>(), kPlusInfinity);
>+  EXPECT_EQ(TestUnit::MinusInfinity().ToKilo<double>(), kMinusInfinity);
>+  EXPECT_EQ(TestUnit::PlusInfinity().ToValue<double>(), kPlusInfinity);
>+  EXPECT_EQ(TestUnit::MinusInfinity().ToValue<double>(), kMinusInfinity);
>+  EXPECT_EQ(TestUnit::PlusInfinity().ToMilli<double>(), kPlusInfinity);
>+  EXPECT_EQ(TestUnit::MinusInfinity().ToMilli<double>(), kMinusInfinity);
>+
>+  EXPECT_TRUE(TestUnit::FromKilo(kPlusInfinity).IsPlusInfinity());
>+  EXPECT_TRUE(TestUnit::FromKilo(kMinusInfinity).IsMinusInfinity());
>+  EXPECT_TRUE(TestUnit::FromValue(kPlusInfinity).IsPlusInfinity());
>+  EXPECT_TRUE(TestUnit::FromValue(kMinusInfinity).IsMinusInfinity());
>+}
>+
>+TEST(UnitBaseTest, MathOperations) {
>+  const int64_t kValueA = 267;
>+  const int64_t kValueB = 450;
>+  const TestUnit delta_a = TestUnit::FromKilo(kValueA);
>+  const TestUnit delta_b = TestUnit::FromKilo(kValueB);
>+  EXPECT_EQ((delta_a + delta_b).ToKilo(), kValueA + kValueB);
>+  EXPECT_EQ((delta_a - delta_b).ToKilo(), kValueA - kValueB);
>+
>+  const int32_t kInt32Value = 123;
>+  const double kFloatValue = 123.0;
>+  EXPECT_EQ((TestUnit::FromValue(kValueA) * kValueB).ToValue<int64_t>(),
>+            kValueA * kValueB);
>+  EXPECT_EQ((TestUnit::FromValue(kValueA) * kInt32Value).ToValue<int64_t>(),
>+            kValueA * kInt32Value);
>+  EXPECT_EQ((TestUnit::FromValue(kValueA) * kFloatValue).ToValue<int64_t>(),
>+            kValueA * kFloatValue);
>+
>+  EXPECT_EQ((delta_b / 10).ToKilo(), kValueB / 10);
>+  EXPECT_EQ(delta_b / delta_a, static_cast<double>(kValueB) / kValueA);
>+
>+  TestUnit mutable_delta = TestUnit::FromKilo(kValueA);
>+  mutable_delta += TestUnit::FromKilo(kValueB);
>+  EXPECT_EQ(mutable_delta, TestUnit::FromKilo(kValueA + kValueB));
>+  mutable_delta -= TestUnit::FromKilo(kValueB);
>+  EXPECT_EQ(mutable_delta, TestUnit::FromKilo(kValueA));
>+}
>+
>+TEST(UnitBaseTest, InfinityOperations) {
>+  const int64_t kValue = 267;
>+  const TestUnit finite = TestUnit::FromKilo(kValue);
>+  EXPECT_TRUE((TestUnit::PlusInfinity() + finite).IsPlusInfinity());
>+  EXPECT_TRUE((TestUnit::PlusInfinity() - finite).IsPlusInfinity());
>+  EXPECT_TRUE((finite + TestUnit::PlusInfinity()).IsPlusInfinity());
>+  EXPECT_TRUE((finite - TestUnit::MinusInfinity()).IsPlusInfinity());
>+
>+  EXPECT_TRUE((TestUnit::MinusInfinity() + finite).IsMinusInfinity());
>+  EXPECT_TRUE((TestUnit::MinusInfinity() - finite).IsMinusInfinity());
>+  EXPECT_TRUE((finite + TestUnit::MinusInfinity()).IsMinusInfinity());
>+  EXPECT_TRUE((finite - TestUnit::PlusInfinity()).IsMinusInfinity());
>+}
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unittest_main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unittest_main.cc
>index 8d4ff2df2d3930cf7ed4c859e197f91f43ce083c..5fd3a996876f8dca3544d0884412547747009999 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unittest_main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unittest_main.cc
>@@ -22,7 +22,6 @@
> #include "system_wrappers/include/field_trial.h"
> #include "system_wrappers/include/metrics.h"
> #include "test/field_trial.h"
>-#include "test/testsupport/fileutils.h"
> 
> #if defined(WEBRTC_WIN)
> #include "rtc_base/win32socketinit.h"
>@@ -32,19 +31,20 @@
> #include "test/ios/test_support.h"
> #endif
> 
>-DEFINE_bool(help, false, "prints this message");
>-DEFINE_string(log, "", "logging options to use");
>-DEFINE_string(
>+WEBRTC_DEFINE_bool(help, false, "prints this message");
>+WEBRTC_DEFINE_string(log, "", "logging options to use");
>+WEBRTC_DEFINE_string(
>     force_fieldtrials,
>     "",
>     "Field trials control experimental feature code which can be forced. "
>     "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
>     " will assign the group Enable to field trial WebRTC-FooFeature.");
> #if defined(WEBRTC_WIN)
>-DEFINE_int(crt_break_alloc, -1, "memory allocation to break on");
>-DEFINE_bool(default_error_handlers,
>-            false,
>-            "leave the default exception/dbg handler functions in place");
>+WEBRTC_DEFINE_int(crt_break_alloc, -1, "memory allocation to break on");
>+WEBRTC_DEFINE_bool(
>+    default_error_handlers,
>+    false,
>+    "leave the default exception/dbg handler functions in place");
> 
> void TestInvalidParameterHandler(const wchar_t* expression,
>                                  const wchar_t* function,
>@@ -81,7 +81,6 @@ int main(int argc, char* argv[]) {
>     return 0;
>   }
> 
>-  webrtc::test::SetExecutablePath(argv[0]);
>   webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
>   // InitFieldTrialsFromString stores the char*, so the char array must outlive
>   // the application.
>@@ -118,7 +117,7 @@ int main(int argc, char* argv[]) {
> 
>   // Initialize SSL which are used by several tests.
>   rtc::InitializeSSL();
>-  rtc::SSLStreamAdapter::enable_time_callback_for_testing();
>+  rtc::SSLStreamAdapter::EnableTimeCallbackForTesting();
> 
> #if defined(WEBRTC_IOS)
>   rtc::test::InitTestSuite(RUN_ALL_TESTS, argc, argv, false);
>@@ -138,5 +137,13 @@ int main(int argc, char* argv[]) {
>     _CrtSetReportHook2(_CRT_RPTHOOK_REMOVE, TestCrtReportHandler);
> #endif
> 
>+#if defined(ADDRESS_SANITIZER) || defined(LEAK_SANITIZER) ||  \
>+    defined(MEMORY_SANITIZER) || defined(THREAD_SANITIZER) || \
>+    defined(UNDEFINED_SANITIZER)
>+  // We want the test flagged as failed only for sanitizer defects,
>+  // in which case the sanitizer will override exit code with 66.
>+  return 0;
>+#endif
>+
>   return res;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unixfilesystem.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unixfilesystem.cc
>deleted file mode 100644
>index 023c34c8c943f873d8cbfaf125f5d71b89c8530e..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unixfilesystem.cc
>+++ /dev/null
>@@ -1,114 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/unixfilesystem.h"
>-
>-#include <errno.h>
>-#include <fcntl.h>
>-#include <stdlib.h>
>-#include <sys/stat.h>
>-#include <unistd.h>
>-
>-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
>-#include <CoreServices/CoreServices.h>
>-#include <IOKit/IOCFBundle.h>
>-#include <sys/statvfs.h>
>-#include "rtc_base/macutils.h"
>-#endif  // WEBRTC_MAC && !defined(WEBRTC_IOS)
>-
>-#if defined(WEBRTC_POSIX) && !defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
>-#include <sys/types.h>
>-#if defined(WEBRTC_ANDROID)
>-#include <sys/statfs.h>
>-#elif !defined(__native_client__)
>-#include <sys/statvfs.h>
>-#endif  //  !defined(__native_client__)
>-#include <limits.h>
>-#include <pwd.h>
>-#include <stdio.h>
>-#endif  // WEBRTC_POSIX && !WEBRTC_MAC || WEBRTC_IOS
>-
>-#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
>-#include <ctype.h>
>-#include <algorithm>
>-#endif
>-
>-#if defined(__native_client__) && !defined(__GLIBC__)
>-#include <sys/syslimits.h>
>-#endif
>-
>-#include "rtc_base/arraysize.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/fileutils.h"
>-#include "rtc_base/pathutils.h"
>-#include "rtc_base/stream.h"
>-#include "rtc_base/stringutils.h"
>-
>-namespace rtc {
>-
>-UnixFilesystem::UnixFilesystem() {}
>-
>-UnixFilesystem::~UnixFilesystem() {}
>-
>-bool UnixFilesystem::DeleteFile(const Pathname& filename) {
>-  RTC_LOG(LS_INFO) << "Deleting file:" << filename.pathname();
>-
>-  if (!IsFile(filename)) {
>-    RTC_DCHECK(IsFile(filename));
>-    return false;
>-  }
>-  return ::unlink(filename.pathname().c_str()) == 0;
>-}
>-
>-bool UnixFilesystem::MoveFile(const Pathname& old_path,
>-                              const Pathname& new_path) {
>-  if (!IsFile(old_path)) {
>-    RTC_DCHECK(IsFile(old_path));
>-    return false;
>-  }
>-  RTC_LOG(LS_VERBOSE) << "Moving " << old_path.pathname() << " to "
>-                      << new_path.pathname();
>-  if (rename(old_path.pathname().c_str(), new_path.pathname().c_str()) != 0) {
>-    return false;
>-  }
>-  return true;
>-}
>-
>-bool UnixFilesystem::IsFolder(const Pathname& path) {
>-  struct stat st;
>-  if (stat(path.pathname().c_str(), &st) < 0)
>-    return false;
>-  return S_ISDIR(st.st_mode);
>-}
>-
>-bool UnixFilesystem::IsFile(const Pathname& pathname) {
>-  struct stat st;
>-  int res = ::stat(pathname.pathname().c_str(), &st);
>-  // Treat symlinks, named pipes, etc. all as files.
>-  return res == 0 && !S_ISDIR(st.st_mode);
>-}
>-
>-bool UnixFilesystem::GetFileSize(const Pathname& pathname, size_t* size) {
>-  struct stat st;
>-  if (::stat(pathname.pathname().c_str(), &st) != 0)
>-    return false;
>-  *size = st.st_size;
>-  return true;
>-}
>-
>-}  // namespace rtc
>-
>-#if defined(__native_client__)
>-extern "C" int __attribute__((weak))
>-link(const char* oldpath, const char* newpath) {
>-  errno = EACCES;
>-  return -1;
>-}
>-#endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unixfilesystem.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unixfilesystem.h
>deleted file mode 100644
>index 711d7b3ea192e149c9111a0d94b9e8fdceeda276..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unixfilesystem.h
>+++ /dev/null
>@@ -1,45 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef RTC_BASE_UNIXFILESYSTEM_H_
>-#define RTC_BASE_UNIXFILESYSTEM_H_
>-
>-#include <sys/types.h>
>-
>-#include "rtc_base/fileutils.h"
>-
>-namespace rtc {
>-
>-class UnixFilesystem : public FilesystemInterface {
>- public:
>-  UnixFilesystem();
>-  ~UnixFilesystem() override;
>-
>-  // This will attempt to delete the file located at filename.
>-  // It will fail with VERIY if you pass it a non-existant file, or a directory.
>-  bool DeleteFile(const Pathname& filename) override;
>-
>-  // This moves a file from old_path to new_path, where "file" can be a plain
>-  // file or directory, which will be moved recursively.
>-  // Returns true if function succeeds.
>-  bool MoveFile(const Pathname& old_path, const Pathname& new_path) override;
>-
>-  // Returns true if a pathname is a directory
>-  bool IsFolder(const Pathname& pathname) override;
>-
>-  // Returns true of pathname represents an existing file
>-  bool IsFile(const Pathname& pathname) override;
>-
>-  bool GetFileSize(const Pathname& path, size_t* size) override;
>-};
>-
>-}  // namespace rtc
>-
>-#endif  // RTC_BASE_UNIXFILESYSTEM_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocket_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocket_unittest.cc
>index d2bff5468837bd16d9303831b58436fc37ad757b..d44f46a65e5c1483d55ccdc1082c3f535dfb9fc3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocket_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocket_unittest.cc
>@@ -104,7 +104,7 @@ struct Receiver : public MessageHandler, public sigslot::has_slots<> {
>                     const char* data,
>                     size_t size,
>                     const SocketAddress& remote_addr,
>-                    const PacketTime& packet_time) {
>+                    const int64_t& /* packet_time_us */) {
>     ASSERT_EQ(socket.get(), s);
>     ASSERT_GE(size, 4U);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>index ae40d7888f551bf7c5abcb88ba4a77f66c40b76a..bfce0f54541e9d1816c6d132b66e3729641702ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>@@ -525,7 +525,6 @@ VirtualSocketServer::VirtualSocketServer() : VirtualSocketServer(nullptr) {}
> 
> VirtualSocketServer::VirtualSocketServer(FakeClock* fake_clock)
>     : fake_clock_(fake_clock),
>-      wakeup_(/*manual_reset=*/false, /*initially_signaled=*/false),
>       msg_queue_(nullptr),
>       stop_on_idle_(false),
>       next_ipv4_(kInitialNextIPv4),
>@@ -1027,8 +1026,9 @@ void VirtualSocketServer::UpdateDelayDistribution() {
>   }
> }
> 
>+
> static double Normal(double x, double mean, double stddev) {
>-  static const double PI = 4 * atan(1.0);
>+  static double PI = 4 * atan(1.0);
>   double a = (x - mean) * (x - mean) / (2 * stddev * stddev);
>   return exp(-a) / (stddev * sqrt(2 * PI));
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr_unittest.cc
>index 954171834e67b29044e67448bb88ef5249b074f2..66f2b4df5b18fd8f6299b58334d6194c25c54b71 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr_unittest.cc
>@@ -202,7 +202,7 @@ template <class T>
> std::unique_ptr<T> NewObjectCreatedOnTaskQueue() {
>   std::unique_ptr<T> obj;
>   TaskQueue queue("NewObjectCreatedOnTaskQueue");
>-  Event event(false, false);
>+  Event event;
>   queue.PostTask([&event, &obj] {
>     obj.reset(new T());
>     event.Set();
>@@ -229,7 +229,7 @@ TEST(WeakPtrTest, WeakPtrInitiateAndUseOnDifferentThreads) {
>   // Create weak ptr on main thread
>   WeakPtr<Target> weak_ptr = target->factory.GetWeakPtr();
>   rtc::TaskQueue queue("queue");
>-  rtc::Event done(false, false);
>+  rtc::Event done;
>   queue.PostTask([&] {
>     // Dereference and invalide weak_ptr on another thread.
>     EXPECT_EQ(weak_ptr.get(), target.get());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.cc
>index d81d685f16c0f28c0dcd3e8a75ec7c9ebda8e010..e3482e3238448f6e5f2614e4659956c8b2a5d2cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.cc
>@@ -72,9 +72,9 @@ const char* inet_ntop_v4(const void* src, char* dst, socklen_t size) {
>   }
>   const struct in_addr* as_in_addr =
>       reinterpret_cast<const struct in_addr*>(src);
>-  rtc::sprintfn(dst, size, "%d.%d.%d.%d", as_in_addr->S_un.S_un_b.s_b1,
>-                as_in_addr->S_un.S_un_b.s_b2, as_in_addr->S_un.S_un_b.s_b3,
>-                as_in_addr->S_un.S_un_b.s_b4);
>+  snprintf(dst, size, "%d.%d.%d.%d", as_in_addr->S_un.S_un_b.s_b1,
>+           as_in_addr->S_un.S_un_b.s_b2, as_in_addr->S_un.S_un_b.s_b3,
>+           as_in_addr->S_un.S_un_b.s_b4);
>   return dst;
> }
> 
>@@ -127,7 +127,7 @@ const char* inet_ntop_v6(const void* src, char* dst, socklen_t size) {
>     *cursor++ = ':';
>     *cursor++ = ':';
>     if (maxpos == 4) {
>-      cursor += rtc::sprintfn(cursor, INET6_ADDRSTRLEN - 2, "ffff:");
>+      cursor += snprintf(cursor, INET6_ADDRSTRLEN - 2, "ffff:");
>     }
>     const struct in_addr* as_v4 =
>         reinterpret_cast<const struct in_addr*>(&(as_shorts[6]));
>@@ -136,8 +136,8 @@ const char* inet_ntop_v6(const void* src, char* dst, socklen_t size) {
>   } else {
>     for (int i = 0; i < run_array_size; ++i) {
>       if (runpos[i] == -1) {
>-        cursor += rtc::sprintfn(cursor, INET6_ADDRSTRLEN - (cursor - dst), "%x",
>-                                NetworkToHost16(as_shorts[i]));
>+        cursor += snprintf(cursor, INET6_ADDRSTRLEN - (cursor - dst), "%x",
>+                           NetworkToHost16(as_shorts[i]));
>         if (i != 7 && runpos[i + 1] != 1) {
>           *cursor++ = ':';
>         }
>@@ -224,8 +224,8 @@ int inet_pton_v6(const char* src, void* dst) {
>       *(readcursor + 2) != 0) {
>     // Check for periods, which we'll take as a sign of v4 addresses.
>     const char* addrstart = readcursor + 2;
>-    if (rtc::strchr(addrstart, ".")) {
>-      const char* colon = rtc::strchr(addrstart, "::");
>+    if (strchr(addrstart, '.')) {
>+      const char* colon = strchr(addrstart, ':');
>       if (colon) {
>         uint16_t a_short;
>         int bytesread = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32filesystem.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32filesystem.cc
>deleted file mode 100644
>index b500e5e629b8a6fc507a622ffa7ed5d92c6cd939..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32filesystem.cc
>+++ /dev/null
>@@ -1,83 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/win32filesystem.h"
>-
>-#include <shellapi.h>
>-#include <shlobj.h>
>-#include <tchar.h>
>-#include "rtc_base/win32.h"
>-
>-#include <memory>
>-
>-#include "rtc_base/arraysize.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/fileutils.h"
>-#include "rtc_base/pathutils.h"
>-#include "rtc_base/stream.h"
>-#include "rtc_base/stringutils.h"
>-
>-// In several places in this file, we test the integrity level of the process
>-// before calling GetLongPathName. We do this because calling GetLongPathName
>-// when running under protected mode IE (a low integrity process) can result in
>-// a virtualized path being returned, which is wrong if you only plan to read.
>-// TODO: Waiting to hear back from IE team on whether this is the
>-// best approach; IEIsProtectedModeProcess is another possible solution.
>-
>-namespace rtc {
>-
>-bool Win32Filesystem::DeleteFile(const Pathname& filename) {
>-  RTC_LOG(LS_INFO) << "Deleting file " << filename.pathname();
>-  if (!IsFile(filename)) {
>-    RTC_DCHECK(IsFile(filename));
>-    return false;
>-  }
>-  return ::DeleteFile(ToUtf16(filename.pathname()).c_str()) != 0;
>-}
>-
>-bool Win32Filesystem::MoveFile(const Pathname& old_path,
>-                               const Pathname& new_path) {
>-  if (!IsFile(old_path)) {
>-    RTC_DCHECK(IsFile(old_path));
>-    return false;
>-  }
>-  RTC_LOG(LS_INFO) << "Moving " << old_path.pathname() << " to "
>-                   << new_path.pathname();
>-  return ::MoveFile(ToUtf16(old_path.pathname()).c_str(),
>-                    ToUtf16(new_path.pathname()).c_str()) != 0;
>-}
>-
>-bool Win32Filesystem::IsFolder(const Pathname& path) {
>-  WIN32_FILE_ATTRIBUTE_DATA data = {0};
>-  if (0 == ::GetFileAttributesEx(ToUtf16(path.pathname()).c_str(),
>-                                 GetFileExInfoStandard, &data))
>-    return false;
>-  return (data.dwFileAttributes & FILE_ATTRIBUTE_DIRECTORY) ==
>-         FILE_ATTRIBUTE_DIRECTORY;
>-}
>-
>-bool Win32Filesystem::IsFile(const Pathname& path) {
>-  WIN32_FILE_ATTRIBUTE_DATA data = {0};
>-  if (0 == ::GetFileAttributesEx(ToUtf16(path.pathname()).c_str(),
>-                                 GetFileExInfoStandard, &data))
>-    return false;
>-  return (data.dwFileAttributes & FILE_ATTRIBUTE_DIRECTORY) == 0;
>-}
>-
>-bool Win32Filesystem::GetFileSize(const Pathname& pathname, size_t* size) {
>-  WIN32_FILE_ATTRIBUTE_DATA data = {0};
>-  if (::GetFileAttributesEx(ToUtf16(pathname.pathname()).c_str(),
>-                            GetFileExInfoStandard, &data) == 0)
>-    return false;
>-  *size = data.nFileSizeLow;
>-  return true;
>-}
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32filesystem.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32filesystem.h
>deleted file mode 100644
>index d26741e67b87048e181fcf0113208af5fad873da..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32filesystem.h
>+++ /dev/null
>@@ -1,41 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef RTC_BASE_WIN32FILESYSTEM_H_
>-#define RTC_BASE_WIN32FILESYSTEM_H_
>-
>-#include "fileutils.h"
>-
>-namespace rtc {
>-
>-class Win32Filesystem : public FilesystemInterface {
>- public:
>-  // This will attempt to delete the path located at filename.
>-  // If the path points to a folder, it will fail with VERIFY
>-  bool DeleteFile(const Pathname& filename) override;
>-
>-  // This moves a file from old_path to new_path. If the new path is on a
>-  // different volume than the old, it will attempt to copy and then delete
>-  // the folder
>-  // Returns true if the file is successfully moved
>-  bool MoveFile(const Pathname& old_path, const Pathname& new_path) override;
>-
>-  // Returns true if a pathname is a directory
>-  bool IsFolder(const Pathname& pathname) override;
>-
>-  // Returns true if a file exists at path
>-  bool IsFile(const Pathname& path) override;
>-
>-  bool GetFileSize(const Pathname& path, size_t* size) override;
>-};
>-
>-}  // namespace rtc
>-
>-#endif  // RTC_BASE_WIN32FILESYSTEM_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.cc
>index cab751a63173abfdc979feca2a3806cd761ff723..230f3ed46f9057e8ed6788ee6289ff6eacae185b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.cc
>@@ -16,6 +16,7 @@
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/timeutils.h"  // For Time, TimeSince
> #include "rtc_base/win32window.h"
> 
> namespace rtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory.h
>index cb4646c1fe4b359f71acba7cd0808f494e4634c9..f697bcbd6b81c762b1567bbce6c5615f25e32e8c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory.h
>@@ -11,6 +11,7 @@
> #ifndef RTC_BASE_ZERO_MEMORY_H_
> #define RTC_BASE_ZERO_MEMORY_H_
> 
>+#include <stddef.h>
> #include <type_traits>
> 
> #include "api/array_view.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/BUILD.gn
>index 252aec4547f43a12cd1ae590908404c61d86f732..5214bc3bd0f24fbf942dcbd0b4a92af7e3081681 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/BUILD.gn
>@@ -25,9 +25,6 @@ group("rtc_tools") {
>       ":psnr_ssim_analyzer",
>       ":rgba_to_i420_converter",
>     ]
>-    if (rtc_include_internal_audio_device) {
>-      deps += [ ":force_mic_volume_max" ]
>-    }
>     if (rtc_enable_protobuf) {
>       deps += [ ":chart_proto" ]
>     }
>@@ -39,8 +36,10 @@ group("rtc_tools") {
>       ":tools_unittests",
>     ]
>     if (rtc_enable_protobuf) {
>+      if (!build_with_chromium) {
>+        deps += [ ":event_log_visualizer" ]
>+      }
>       deps += [
>-        ":event_log_visualizer",
>         ":rtp_analyzer",
>         ":unpack_aecdump",
>         "network_tester",
>@@ -89,6 +88,12 @@ rtc_static_library("video_file_writer") {
> 
> rtc_static_library("video_quality_analysis") {
>   sources = [
>+    "frame_analyzer/linear_least_squares.cc",
>+    "frame_analyzer/linear_least_squares.h",
>+    "frame_analyzer/video_color_aligner.cc",
>+    "frame_analyzer/video_color_aligner.h",
>+    "frame_analyzer/video_geometry_aligner.cc",
>+    "frame_analyzer/video_geometry_aligner.h",
>     "frame_analyzer/video_quality_analysis.cc",
>     "frame_analyzer/video_quality_analysis.h",
>     "frame_analyzer/video_temporal_aligner.cc",
>@@ -96,6 +101,8 @@ rtc_static_library("video_quality_analysis") {
>   ]
>   deps = [
>     ":video_file_reader",
>+    "../api:array_view",
>+    "../api/video:video_frame",
>     "../api/video:video_frame_i420",
>     "../common_video",
>     "../rtc_base:checks",
>@@ -211,20 +218,6 @@ if (!build_with_chromium) {
>     ]
>   }
> 
>-  # It doesn't make sense to build this tool without the ADM enabled.
>-  if (rtc_include_internal_audio_device) {
>-    rtc_executable("force_mic_volume_max") {
>-      sources = [
>-        "force_mic_volume_max/force_mic_volume_max.cc",
>-      ]
>-
>-      deps = [
>-        "../modules/audio_device",
>-        "../modules/audio_device:audio_device_impl",
>-      ]
>-    }
>-  }
>-
>   if (rtc_enable_protobuf) {
>     proto_library("chart_proto") {
>       sources = [
>@@ -272,10 +265,13 @@ if (!build_with_chromium) {
> 
>         # TODO(kwiberg): Remove this dependency.
>         "../api/audio_codecs:audio_codecs_api",
>+        "../api/transport:goog_cc",
>         "../modules/congestion_controller",
>         "../modules/congestion_controller/goog_cc:delay_based_bwe",
>         "../modules/congestion_controller/goog_cc:estimators",
>+        "../modules/congestion_controller/rtp:transport_feedback",
>         "../modules/pacing",
>+        "../modules/remote_bitrate_estimator",
>         "../modules/rtp_rtcp",
>         "//third_party/abseil-cpp/absl/memory",
>       ]
>@@ -284,7 +280,7 @@ if (!build_with_chromium) {
> }
> 
> if (rtc_include_tests) {
>-  if (rtc_enable_protobuf) {
>+  if (rtc_enable_protobuf && !build_with_chromium) {
>     rtc_executable("event_log_visualizer") {
>       testonly = true
>       sources = [
>@@ -347,7 +343,10 @@ if (rtc_include_tests) {
>     testonly = true
> 
>     sources = [
>+      "frame_analyzer/linear_least_squares_unittest.cc",
>       "frame_analyzer/reference_less_video_analysis_unittest.cc",
>+      "frame_analyzer/video_color_aligner_unittest.cc",
>+      "frame_analyzer/video_geometry_aligner_unittest.cc",
>       "frame_analyzer/video_quality_analysis_unittest.cc",
>       "frame_analyzer/video_temporal_aligner_unittest.cc",
>       "frame_editing/frame_editing_unittest.cc",
>@@ -364,8 +363,6 @@ if (rtc_include_tests) {
> 
>     deps = [
>       ":command_line_parser",
>-      ":frame_editing_lib",
>-      ":reference_less_video_analysis_lib",
>       ":video_file_reader",
>       ":video_file_writer",
>       ":video_quality_analysis",
>@@ -374,10 +371,19 @@ if (rtc_include_tests) {
>       "../rtc_base:checks",
>       "../test:fileutils",
>       "../test:test_main",
>+      "../test:test_support",
>       "//testing/gtest",
>       "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/libyuv",
>     ]
> 
>+    if (!build_with_chromium) {
>+      deps += [
>+        ":frame_editing_lib",
>+        ":reference_less_video_analysis_lib",
>+      ]
>+    }
>+
>     if (rtc_enable_protobuf) {
>       deps += [ "network_tester:network_tester_unittests" ]
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/DEPS
>index d9b162cd15fb2c704597baad1aa20f4fd3b892ce..4015ae265ac8e49208774e7b96bb1fabab84d593 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/DEPS
>@@ -7,6 +7,7 @@ include_rules = [
>   "+modules/audio_coding/neteq/tools",
>   "+modules/audio_processing",
>   "+modules/bitrate_controller",
>+  "+modules/remote_bitrate_estimator",
>   "+modules/congestion_controller",
>   "+modules/pacing",
>   "+modules/rtp_rtcp",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/OWNERS
>index c1b87c9bdca37e0881d03bc681b30e7be78670b8..77385fcd316177a7c1c56584e1b1492305adae78 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/OWNERS
>@@ -3,3 +3,6 @@ per-file BUILD.gn=*
> phoglund@webrtc.org
> oprypin@webrtc.org
> mbonadei@webrtc.org
>+
>+# For video analysis tools
>+magjed@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/agc/activity_metric.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/agc/activity_metric.cc
>index b4ed3facb359b5a81ba83b3d4e52fca7534cd3fa..9b2276f16f1f5e27536f958d9d9e972f8599ee4a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/agc/activity_metric.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/agc/activity_metric.cc
>@@ -30,32 +30,34 @@
> static const int kAgcAnalWindowSamples = 100;
> static const float kDefaultActivityThreshold = 0.3f;
> 
>-DEFINE_bool(standalone_vad, true, "enable stand-alone VAD");
>-DEFINE_string(true_vad,
>-              "",
>-              "name of a file containing true VAD in 'int'"
>-              " format");
>-DEFINE_string(video_vad,
>-              "",
>-              "name of a file containing video VAD (activity"
>-              " probabilities) in double format. One activity per 10ms is"
>-              " required. If no file is given the video information is not"
>-              " incorporated. Negative activity is interpreted as video is"
>-              " not adapted and the statistics are not computed during"
>-              " the learning phase. Note that the negative video activities"
>-              " are ONLY allowed at the beginning.");
>-DEFINE_string(result,
>-              "",
>-              "name of a file to write the results. The results"
>-              " will be appended to the end of the file. This is optional.");
>-DEFINE_string(audio_content,
>-              "",
>-              "name of a file where audio content is written"
>-              " to, in double format.");
>-DEFINE_float(activity_threshold,
>-             kDefaultActivityThreshold,
>-             "Activity threshold");
>-DEFINE_bool(help, false, "prints this message");
>+WEBRTC_DEFINE_bool(standalone_vad, true, "enable stand-alone VAD");
>+WEBRTC_DEFINE_string(true_vad,
>+                     "",
>+                     "name of a file containing true VAD in 'int'"
>+                     " format");
>+WEBRTC_DEFINE_string(
>+    video_vad,
>+    "",
>+    "name of a file containing video VAD (activity"
>+    " probabilities) in double format. One activity per 10ms is"
>+    " required. If no file is given the video information is not"
>+    " incorporated. Negative activity is interpreted as video is"
>+    " not adapted and the statistics are not computed during"
>+    " the learning phase. Note that the negative video activities"
>+    " are ONLY allowed at the beginning.");
>+WEBRTC_DEFINE_string(
>+    result,
>+    "",
>+    "name of a file to write the results. The results"
>+    " will be appended to the end of the file. This is optional.");
>+WEBRTC_DEFINE_string(audio_content,
>+                     "",
>+                     "name of a file where audio content is written"
>+                     " to, in double format.");
>+WEBRTC_DEFINE_float(activity_threshold,
>+                    kDefaultActivityThreshold,
>+                    "Activity threshold");
>+WEBRTC_DEFINE_bool(help, false, "prints this message");
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/DEPS
>deleted file mode 100644
>index d0325a65aa1e344990120cf22f2d34a23808e205..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/DEPS
>+++ /dev/null
>@@ -1,13 +0,0 @@
>-# This is trimmed down version of the main tools DEPS file which is to be used
>-# in Chromiums PyAuto WebRTC video quality measurement test. We will only
>-# need the Zxing dependencies as we only use the barcode tools in this test.
>-
>-deps = {
>-  # Used by barcode_tools
>-  "barcode_tools/third_party/zxing/core":
>-    "http://zxing.googlecode.com/svn/trunk/core@2349",
>-
>-  # Used by barcode_tools
>-  "barcode_tools/third_party/zxing/javase":
>-    "http://zxing.googlecode.com/svn/trunk/javase@2349",
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/README b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/README
>deleted file mode 100644
>index a23e798064def22c99d668383fbd8ddccc435768..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/README
>+++ /dev/null
>@@ -1,34 +0,0 @@
>-This file explains how to get the dependencies needed for the barcode tools.
>-
>-barcode_encoder.py
>-==================
>-This script depends on:
>-* Zxing (Java version)
>-* Ant (must be installed manually)
>-* Java
>-
>-To automatically download Zxing for the encoder script, checkout this directory
>-as a separate gclient solution, like this:
>-gclient config http://webrtc.googlecode.com/svn/trunk/webrtc/rtc_tools/barcode_tools
>-gclient sync
>-Then the Zxing Java source code will be put in third_party/zxing.
>-
>-In order to run barcode_encoder.py you then need to build:
>-* zxing/core/core.jar
>-* zxing/javase/javase.jar
>-These are compiled using Ant by running build_zxing.py:
>-python build_zxing.py
>-
>-For more info about Zxing, see https://code.google.com/p/zxing/
>-
>-
>-barcode_decoder.py
>-==================
>-This script depends on:
>-* Zxing (C++ version). You need to checkout from Subversion and build the libs
>-  and zxing SCons targets. SVN URL: http://zxing.googlecode.com/svn/trunk/cpp
>-* FFMPEG fmpeg 0.11.1
>-
>-These dependencies must be precompiled separately before running the script.
>-Make sure to add FFMPEG to the PATH environment variable and provide the path
>-to the zxing executable using the mandatory command line flag to the script.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/barcode_decoder.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/barcode_decoder.py
>deleted file mode 100644
>index 078e47f1bac724307939e38ef846878e0eb14bce..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/barcode_decoder.py
>+++ /dev/null
>@@ -1,295 +0,0 @@
>-#!/usr/bin/env python
>-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-import optparse
>-import os
>-import sys
>-
>-if __name__ == '__main__':
>-  # Make sure we always can import helper_functions.
>-  sys.path.append(os.path.dirname(__file__))
>-
>-import helper_functions
>-
>-# Chrome browsertests will throw away stderr; avoid that output gets lost.
>-sys.stderr = sys.stdout
>-
>-
>-def ConvertYuvToPngFiles(yuv_file_name, yuv_frame_width, yuv_frame_height,
>-                         output_directory, ffmpeg_path):
>-  """Converts a YUV video file into PNG frames.
>-
>-  The function uses ffmpeg to convert the YUV file. The output of ffmpeg is in
>-  the form frame_xxxx.png, where xxxx is the frame number, starting from 0001.
>-
>-  Args:
>-    yuv_file_name(string): The name of the YUV file.
>-    yuv_frame_width(int): The width of one YUV frame.
>-    yuv_frame_height(int): The height of one YUV frame.
>-    output_directory(string): The output directory where the PNG frames will be
>-      stored.
>-    ffmpeg_path(string): The path to the ffmpeg executable. If None, the PATH
>-      will be searched for it.
>-
>-  Return:
>-    (bool): True if the conversion was OK.
>-  """
>-  size_string = str(yuv_frame_width) + 'x' + str(yuv_frame_height)
>-  output_files_pattern = os.path.join(output_directory, 'frame_%04d.png')
>-  if not ffmpeg_path:
>-    ffmpeg_path = 'ffmpeg.exe' if sys.platform == 'win32' else 'ffmpeg'
>-  if yuv_file_name.endswith('.yuv'):
>-    command = [ffmpeg_path, '-s', '%s' % size_string, '-i', '%s'
>-               % yuv_file_name, '-f', 'image2', '-vcodec', 'png',
>-               '%s' % output_files_pattern]
>-  else:
>-    command = [ffmpeg_path, '-i', '%s' % yuv_file_name, '-f', 'image2',
>-               '-vcodec', 'png', '%s' % output_files_pattern]
>-  try:
>-    print 'Converting YUV file to PNG images (may take a while)...'
>-    print ' '.join(command)
>-    helper_functions.RunShellCommand(
>-        command, fail_msg='Error during YUV to PNG conversion')
>-  except helper_functions.HelperError, err:
>-    print 'Error executing command: %s. Error: %s' % (command, err)
>-    return False
>-  except OSError:
>-    print 'Did not find %s. Have you installed it?' % ffmpeg_path
>-    return False
>-  return True
>-
>-
>-def DecodeFrames(input_directory, zxing_path):
>-  """Decodes the barcodes overlaid in each frame.
>-
>-  The function uses the Zxing command-line tool from the Zxing C++ distribution
>-  to decode the barcode in every PNG frame from the input directory. The frames
>-  should be named frame_xxxx.png, where xxxx is the frame number. The frame
>-  numbers should be consecutive and should start from 0001.
>-  The decoding results in a frame_xxxx.txt file for every successfully decoded
>-  barcode. This file contains the decoded barcode as 12-digit string (UPC-A
>-  format: 11 digits content + one check digit).
>-
>-  Args:
>-    input_directory(string): The input directory from where the PNG frames are
>-      read.
>-    zxing_path(string): The path to the zxing binary. If specified as None,
>-      the PATH will be searched for it.
>-  Return:
>-    (bool): True if the decoding succeeded.
>-  """
>-  if not zxing_path:
>-    zxing_path = 'zxing.exe' if sys.platform == 'win32' else 'zxing'
>-  print 'Decoding barcodes from PNG files with %s...' % zxing_path
>-  return helper_functions.PerformActionOnAllFiles(
>-      directory=input_directory, file_pattern='frame_',
>-      file_extension='png', start_number=1, action=_DecodeBarcodeInFile,
>-      command_line_decoder=zxing_path)
>-
>-
>-def _DecodeBarcodeInFile(file_name, command_line_decoder):
>-  """Decodes the barcode in the upper left corner of a PNG file.
>-
>-  Args:
>-    file_name(string): File name of the PNG file.
>-    command_line_decoder(string): The ZXing command-line decoding tool.
>-
>-  Return:
>-    (bool): True upon success, False otherwise.
>-  """
>-  command = [command_line_decoder, '--try-harder', '--dump-raw', file_name]
>-  try:
>-    out = helper_functions.RunShellCommand(
>-        command, fail_msg='Error during decoding of %s' % file_name)
>-    text_file = open('%s.txt' % file_name[:-4], 'w')
>-    text_file.write(out)
>-    text_file.close()
>-  except helper_functions.HelperError, err:
>-    print 'Barcode in %s cannot be decoded.' % file_name
>-    print err
>-    return False
>-  except OSError:
>-    print 'Did not find %s. Have you installed it?' % command_line_decoder
>-    return False
>-  return True
>-
>-
>-def _GenerateStatsFile(stats_file_name, input_directory='.'):
>-  """Generate statistics file.
>-
>-  The function generates a statistics file. The contents of the file are in the
>-  format <frame_name> <barcode>, where frame name is the name of every frame
>-  (effectively the frame number) and barcode is the decoded barcode. The frames
>-  and the helper .txt files are removed after they have been used.
>-  """
>-  file_prefix = os.path.join(input_directory, 'frame_')
>-  stats_file = open(stats_file_name, 'w')
>-
>-  print 'Generating stats file: %s' % stats_file_name
>-  for i in range(1, _CountFramesIn(input_directory=input_directory) + 1):
>-    frame_number = helper_functions.ZeroPad(i)
>-    barcode_file_name = file_prefix + frame_number + '.txt'
>-    png_frame = file_prefix + frame_number + '.png'
>-    entry_frame_number = helper_functions.ZeroPad(i-1)
>-    entry = 'frame_' + entry_frame_number + ' '
>-
>-    if os.path.isfile(barcode_file_name):
>-      barcode = _ReadBarcodeFromTextFile(barcode_file_name)
>-      os.remove(barcode_file_name)
>-
>-      if _CheckBarcode(barcode):
>-        entry += (helper_functions.ZeroPad(int(barcode[0:11])) + '\n')
>-      else:
>-        entry += 'Barcode error\n'  # Barcode is wrongly detected.
>-    else:  # Barcode file doesn't exist.
>-      entry += 'Barcode error\n'
>-
>-    stats_file.write(entry)
>-    os.remove(png_frame)
>-
>-  stats_file.close()
>-
>-
>-def _ReadBarcodeFromTextFile(barcode_file_name):
>-  """Reads the decoded barcode for a .txt file.
>-
>-  Args:
>-    barcode_file_name(string): The name of the .txt file.
>-  Return:
>-    (string): The decoded barcode.
>-  """
>-  barcode_file = open(barcode_file_name, 'r')
>-  barcode = barcode_file.read()
>-  barcode_file.close()
>-  return barcode
>-
>-
>-def _CheckBarcode(barcode):
>-  """Check weather the UPC-A barcode was decoded correctly.
>-
>-  This function calculates the check digit of the provided barcode and compares
>-  it to the check digit that was decoded.
>-
>-  Args:
>-    barcode(string): The barcode (12-digit).
>-  Return:
>-    (bool): True if the barcode was decoded correctly.
>-  """
>-  if len(barcode) != 12:
>-    return False
>-
>-  r1 = range(0, 11, 2)  # Odd digits
>-  r2 = range(1, 10, 2)  # Even digits except last
>-  dsum = 0
>-  # Sum all the even digits
>-  for i in r1:
>-    dsum += int(barcode[i])
>-  # Multiply the sum by 3
>-  dsum *= 3
>-  # Add all the even digits except the check digit (12th digit)
>-  for i in r2:
>-    dsum += int(barcode[i])
>-  # Get the modulo 10
>-  dsum = dsum % 10
>-  # If not 0 substract from 10
>-  if dsum != 0:
>-    dsum = 10 - dsum
>-  # Compare result and check digit
>-  return dsum == int(barcode[11])
>-
>-
>-def _CountFramesIn(input_directory='.'):
>-  """Calculates the number of frames in the input directory.
>-
>-  The function calculates the number of frames in the input directory. The
>-  frames should be named frame_xxxx.png, where xxxx is the number of the frame.
>-  The numbers should start from 1 and should be consecutive.
>-
>-  Args:
>-    input_directory(string): The input directory.
>-  Return:
>-    (int): The number of frames.
>-  """
>-  file_prefix = os.path.join(input_directory, 'frame_')
>-  file_exists = True
>-  num = 1
>-
>-  while file_exists:
>-    file_name = (file_prefix + helper_functions.ZeroPad(num) + '.png')
>-    if os.path.isfile(file_name):
>-      num += 1
>-    else:
>-      file_exists = False
>-  return num - 1
>-
>-
>-def _ParseArgs():
>-  """Registers the command-line options."""
>-  usage = "usage: %prog [options]"
>-  parser = optparse.OptionParser(usage=usage)
>-
>-  parser.add_option('--zxing_path', type='string',
>-                    help=('The path to where the zxing executable is located. '
>-                          'If omitted, it will be assumed to be present in the '
>-                          'PATH with the name zxing[.exe].'))
>-  parser.add_option('--ffmpeg_path', type='string',
>-                    help=('The path to where the ffmpeg executable is located. '
>-                          'If omitted, it will be assumed to be present in the '
>-                          'PATH with the name ffmpeg[.exe].'))
>-  parser.add_option('--yuv_frame_width', type='int', default=640,
>-                    help='Width of the YUV file\'s frames. Default: %default')
>-  parser.add_option('--yuv_frame_height', type='int', default=480,
>-                    help='Height of the YUV file\'s frames. Default: %default')
>-  parser.add_option('--yuv_file', type='string', default='output.yuv',
>-                    help='The YUV file to be decoded. Default: %default')
>-  parser.add_option('--stats_file', type='string', default='stats.txt',
>-                    help='The output stats file. Default: %default')
>-  parser.add_option('--png_working_dir', type='string', default='.',
>-                    help=('The directory for temporary PNG images to be stored '
>-                          'in when decoding from YUV before they\'re barcode '
>-                          'decoded. If using Windows and a Cygwin-compiled '
>-                          'zxing.exe, you should keep the default value to '
>-                          'avoid problems. Default: %default'))
>-  options, _ = parser.parse_args()
>-  return options
>-
>-
>-def main():
>-  """The main function.
>-
>-  A simple invocation is:
>-  ./webrtc/rtc_tools/barcode_tools/barcode_decoder.py
>-  --yuv_file=<path_and_name_of_overlaid_yuv_video>
>-  --yuv_frame_width=640 --yuv_frame_height=480
>-  --stats_file=<path_and_name_to_stats_file>
>-  """
>-  options = _ParseArgs()
>-
>-  # Convert the overlaid YUV video into a set of PNG frames.
>-  if not ConvertYuvToPngFiles(options.yuv_file, options.yuv_frame_width,
>-                              options.yuv_frame_height,
>-                              output_directory=options.png_working_dir,
>-                              ffmpeg_path=options.ffmpeg_path):
>-    print 'An error occurred converting from YUV to PNG frames.'
>-    return -1
>-
>-  # Decode the barcodes from the PNG frames.
>-  if not DecodeFrames(input_directory=options.png_working_dir,
>-                      zxing_path=options.zxing_path):
>-    print 'An error occurred decoding barcodes from PNG frames.'
>-    return -2
>-
>-  # Generate statistics file.
>-  _GenerateStatsFile(options.stats_file,
>-                     input_directory=options.png_working_dir)
>-  print 'Completed barcode decoding.'
>-  return 0
>-
>-if __name__ == '__main__':
>-  sys.exit(main())
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/barcode_encoder.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/barcode_encoder.py
>deleted file mode 100644
>index 9ab8b5075423992f92b8c7f87393ce02ce9e6b90..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/barcode_encoder.py
>+++ /dev/null
>@@ -1,372 +0,0 @@
>-#!/usr/bin/env python
>-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-import optparse
>-import os
>-import sys
>-
>-import helper_functions
>-
>-_DEFAULT_BARCODE_WIDTH = 352
>-_DEFAULT_BARCODES_FILE = 'barcodes.yuv'
>-
>-
>-def GenerateUpcaBarcodes(number_of_barcodes, barcode_width, barcode_height,
>-                         output_directory='.',
>-                         path_to_zxing='zxing-read-only'):
>-  """Generates UPC-A barcodes.
>-
>-  This function generates a number_of_barcodes UPC-A barcodes. The function
>-  calls an example Java encoder from the Zxing library. The barcodes are
>-  generated as PNG images. The width of the barcodes shouldn't be less than 102
>-  pixels because otherwise Zxing can't properly generate the barcodes.
>-
>-  Args:
>-    number_of_barcodes(int): The number of barcodes to generate.
>-    barcode_width(int): Width of barcode in pixels.
>-    barcode_height(int): Height of barcode in pixels.
>-    output_directory(string): Output directory where to store generated
>-      barcodes.
>-    path_to_zxing(string): The path to Zxing.
>-
>-  Return:
>-    (bool): True if the conversion is successful.
>-  """
>-  base_file_name = os.path.join(output_directory, "barcode_")
>-  jars = _FormJarsString(path_to_zxing)
>-  command_line_encoder = 'com.google.zxing.client.j2se.CommandLineEncoder'
>-  barcode_width = str(barcode_width)
>-  barcode_height = str(barcode_height)
>-
>-  errors = False
>-  for i in range(number_of_barcodes):
>-    suffix = helper_functions.ZeroPad(i)
>-    # Barcodes starting from 0
>-    content = helper_functions.ZeroPad(i, 11)
>-    output_file_name = base_file_name + suffix + ".png"
>-
>-    command = ["java", "-cp", jars, command_line_encoder,
>-               "--barcode_format=UPC_A", "--height=%s" % barcode_height,
>-               "--width=%s" % barcode_width,
>-               "--output=%s" % (output_file_name), "%s" % (content)]
>-    try:
>-      helper_functions.RunShellCommand(
>-          command, fail_msg=('Error during barcode %s generation' % content))
>-    except helper_functions.HelperError as err:
>-      print >> sys.stderr, err
>-      errors = True
>-  return not errors
>-
>-
>-def ConvertPngToYuvBarcodes(input_directory='.', output_directory='.'):
>-  """Converts PNG barcodes to YUV barcode images.
>-
>-  This function reads all the PNG files from the input directory which are in
>-  the format frame_xxxx.png, where xxxx is the number of the frame, starting
>-  from 0000. The frames should be consecutive numbers. The output YUV file is
>-  named frame_xxxx.yuv. The function uses ffmpeg to do the conversion.
>-
>-  Args:
>-    input_directory(string): The input direcotry to read the PNG barcodes from.
>-    output_directory(string): The putput directory to write the YUV files to.
>-  Return:
>-    (bool): True if the conversion was without errors.
>-  """
>-  return helper_functions.PerformActionOnAllFiles(
>-      input_directory, 'barcode_', 'png', 0, _ConvertToYuvAndDelete,
>-      output_directory=output_directory, pattern='barcode_')
>-
>-
>-def _ConvertToYuvAndDelete(output_directory, file_name, pattern):
>-  """Converts a PNG file to a YUV file and deletes the PNG file.
>-
>-  Args:
>-    output_directory(string): The output directory for the YUV file.
>-    file_name(string): The PNG file name.
>-    pattern(string): The file pattern of the PNG/YUV file. The PNG/YUV files are
>-      named patternxx..x.png/yuv, where xx..x are digits starting from 00..0.
>-  Return:
>-    (bool): True upon successful conversion, false otherwise.
>-  """
>-  # Pattern should be in file name
>-  if not pattern in file_name:
>-    return False
>-  pattern_position = file_name.rfind(pattern)
>-
>-  # Strip the path to the PNG file and replace the png extension with yuv
>-  yuv_file_name = file_name[pattern_position:-3] + 'yuv'
>-  yuv_file_name = os.path.join(output_directory, yuv_file_name)
>-
>-  command = ['ffmpeg', '-i', '%s' % (file_name), '-pix_fmt', 'yuv420p',
>-             '%s' % (yuv_file_name)]
>-  try:
>-    helper_functions.RunShellCommand(
>-        command, fail_msg=('Error during PNG to YUV conversion of %s' %
>-                           file_name))
>-    os.remove(file_name)
>-  except helper_functions.HelperError as err:
>-    print >> sys.stderr, err
>-    return False
>-  return True
>-
>-
>-def CombineYuvFramesIntoOneFile(output_file_name, input_directory='.'):
>-  """Combines several YUV frames into one YUV video file.
>-
>-  The function combines the YUV frames from input_directory into one YUV video
>-  file. The frames should be named in the format frame_xxxx.yuv where xxxx
>-  stands for the frame number. The numbers have to be consecutive and start from
>-  0000. The YUV frames are removed after they have been added to the video.
>-
>-  Args:
>-    output_file_name(string): The name of the file to produce.
>-    input_directory(string): The directory from which the YUV frames are read.
>-  Return:
>-    (bool): True if the frame stitching went OK.
>-  """
>-  output_file = open(output_file_name, "wb")
>-  success = helper_functions.PerformActionOnAllFiles(
>-      input_directory, 'barcode_', 'yuv', 0, _AddToFileAndDelete,
>-      output_file=output_file)
>-  output_file.close()
>-  return success
>-
>-def _AddToFileAndDelete(output_file, file_name):
>-  """Adds the contents of a file to a previously opened file.
>-
>-  Args:
>-    output_file(file): The ouput file, previously opened.
>-    file_name(string): The file name of the file to add to the output file.
>-
>-  Return:
>-    (bool): True if successful, False otherwise.
>-  """
>-  input_file = open(file_name, "rb")
>-  input_file_contents = input_file.read()
>-  output_file.write(input_file_contents)
>-  input_file.close()
>-  try:
>-    os.remove(file_name)
>-  except OSError as e:
>-    print >> sys.stderr, 'Error deleting file %s.\nError: %s' % (file_name, e)
>-    return False
>-  return True
>-
>-
>-def _OverlayBarcodeAndBaseFrames(barcodes_file, base_file, output_file,
>-                                 barcodes_component_sizes,
>-                                 base_component_sizes):
>-  """Overlays the next YUV frame from a file with a barcode.
>-
>-  Args:
>-    barcodes_file(FileObject): The YUV file containing the barcodes (opened).
>-    base_file(FileObject): The base YUV file (opened).
>-    output_file(FileObject): The output overlaid file (opened).
>-    barcodes_component_sizes(list of tuples): The width and height of each Y, U
>-      and V plane of the barcodes YUV file.
>-    base_component_sizes(list of tuples): The width and height of each Y, U and
>-      V plane of the base YUV file.
>-  Return:
>-    (bool): True if there are more planes (i.e. frames) in the base file, false
>-      otherwise.
>-  """
>-  # We will loop three times - once for the Y, U and V planes
>-  for ((barcode_comp_width, barcode_comp_height),
>-      (base_comp_width, base_comp_height)) in zip(barcodes_component_sizes,
>-                                                  base_component_sizes):
>-    for base_row in range(base_comp_height):
>-      barcode_plane_traversed = False
>-      if (base_row < barcode_comp_height) and not barcode_plane_traversed:
>-        barcode_plane = barcodes_file.read(barcode_comp_width)
>-        if barcode_plane == "":
>-          barcode_plane_traversed = True
>-      else:
>-        barcode_plane_traversed = True
>-      base_plane = base_file.read(base_comp_width)
>-
>-      if base_plane == "":
>-        return False
>-
>-      if not barcode_plane_traversed:
>-        # Substitute part of the base component with the top component
>-        output_file.write(barcode_plane)
>-        base_plane = base_plane[barcode_comp_width:]
>-      output_file.write(base_plane)
>-  return True
>-
>-
>-def OverlayYuvFiles(barcode_width, barcode_height, base_width, base_height,
>-                    barcodes_file_name, base_file_name, output_file_name):
>-  """Overlays two YUV files starting from the upper left corner of both.
>-
>-  Args:
>-    barcode_width(int): The width of the barcode (to be overlaid).
>-    barcode_height(int): The height of the barcode (to be overlaid).
>-    base_width(int): The width of a frame of the base file.
>-    base_height(int): The height of a frame of the base file.
>-    barcodes_file_name(string): The name of the YUV file containing the YUV
>-      barcodes.
>-    base_file_name(string): The name of the base YUV file.
>-    output_file_name(string): The name of the output file where the overlaid
>-      video will be written.
>-  """
>-  # Component sizes = [Y_sizes, U_sizes, V_sizes]
>-  barcodes_component_sizes = [(barcode_width, barcode_height),
>-                              (barcode_width/2, barcode_height/2),
>-                              (barcode_width/2, barcode_height/2)]
>-  base_component_sizes = [(base_width, base_height),
>-                          (base_width/2, base_height/2),
>-                          (base_width/2, base_height/2)]
>-
>-  barcodes_file = open(barcodes_file_name, 'rb')
>-  base_file = open(base_file_name, 'rb')
>-  output_file = open(output_file_name, 'wb')
>-
>-  data_left = True
>-  while data_left:
>-    data_left = _OverlayBarcodeAndBaseFrames(barcodes_file, base_file,
>-                                             output_file,
>-                                             barcodes_component_sizes,
>-                                             base_component_sizes)
>-
>-  barcodes_file.close()
>-  base_file.close()
>-  output_file.close()
>-
>-
>-def CalculateFramesNumberFromYuv(yuv_width, yuv_height, file_name):
>-  """Calculates the number of frames of a YUV video.
>-
>-  Args:
>-    yuv_width(int): Width of a frame of the yuv file.
>-    yuv_height(int): Height of a frame of the YUV file.
>-    file_name(string): The name of the YUV file.
>-  Return:
>-    (int): The number of frames in the YUV file.
>-  """
>-  file_size = os.path.getsize(file_name)
>-
>-  y_plane_size = yuv_width * yuv_height
>-  u_plane_size = (yuv_width/2) * (yuv_height/2)  # Equals to V plane size too
>-  frame_size = y_plane_size + (2 * u_plane_size)
>-  return int(file_size/frame_size)  # Should be int anyway
>-
>-
>-def _FormJarsString(path_to_zxing):
>-  """Forms the the Zxing core and javase jars argument.
>-
>-  Args:
>-    path_to_zxing(string): The path to the Zxing checkout folder.
>-  Return:
>-    (string): The newly formed jars argument.
>-  """
>-  javase_jar = os.path.join(path_to_zxing, "javase", "javase.jar")
>-  core_jar = os.path.join(path_to_zxing, "core", "core.jar")
>-  delimiter = ':'
>-  if os.name != 'posix':
>-    delimiter = ';'
>-  return javase_jar + delimiter + core_jar
>-
>-def _ParseArgs():
>-  """Registers the command-line options."""
>-  usage = "usage: %prog [options]"
>-  parser = optparse.OptionParser(usage=usage)
>-
>-  parser.add_option('--barcode_width', type='int',
>-                    default=_DEFAULT_BARCODE_WIDTH,
>-                    help=('Width of the barcodes to be overlaid on top of the'
>-                          ' base file. Default: %default'))
>-  parser.add_option('--barcode_height', type='int', default=32,
>-                    help=('Height of the barcodes to be overlaid on top of the'
>-                          ' base file. Default: %default'))
>-  parser.add_option('--base_frame_width', type='int', default=352,
>-                    help=('Width of the base YUV file\'s frames. '
>-                          'Default: %default'))
>-  parser.add_option('--base_frame_height', type='int', default=288,
>-                    help=('Height of the top YUV file\'s frames. '
>-                          'Default: %default'))
>-  parser.add_option('--barcodes_yuv', type='string',
>-                    default=_DEFAULT_BARCODES_FILE,
>-                    help=('The YUV file with the barcodes in YUV. '
>-                          'Default: %default'))
>-  parser.add_option('--base_yuv', type='string', default='base.yuv',
>-                    help=('The base YUV file to be overlaid. '
>-                          'Default: %default'))
>-  parser.add_option('--output_yuv', type='string', default='output.yuv',
>-                    help=('The output YUV file containing the base overlaid'
>-                          ' with the barcodes. Default: %default'))
>-  parser.add_option('--png_barcodes_output_dir', type='string', default='.',
>-                    help=('Output directory where the PNG barcodes will be '
>-                          'generated. Default: %default'))
>-  parser.add_option('--png_barcodes_input_dir', type='string', default='.',
>-                    help=('Input directory from where the PNG barcodes will be '
>-                          'read. Default: %default'))
>-  parser.add_option('--yuv_barcodes_output_dir', type='string', default='.',
>-                    help=('Output directory where the YUV barcodes will be '
>-                          'generated. Default: %default'))
>-  parser.add_option('--yuv_frames_input_dir', type='string', default='.',
>-                    help=('Input directory from where the YUV will be '
>-                          'read before combination. Default: %default'))
>-  parser.add_option('--zxing_dir', type='string', default='zxing',
>-                    help=('Path to the Zxing barcodes library. '
>-                          'Default: %default'))
>-  options = parser.parse_args()[0]
>-  return options
>-
>-
>-def main():
>-  """The main function.
>-
>-  A simple invocation will be:
>-  ./webrtc/rtc_tools/barcode_tools/barcode_encoder.py --barcode_height=32
>-  --base_frame_width=352 --base_frame_height=288
>-  --base_yuv=<path_and_name_of_base_file>
>-  --output_yuv=<path and name_of_output_file>
>-  """
>-  options = _ParseArgs()
>-  # The barcodes with will be different than the base frame width only if
>-  # explicitly specified at the command line.
>-  if options.barcode_width == _DEFAULT_BARCODE_WIDTH:
>-    options.barcode_width = options.base_frame_width
>-  # If the user provides a value for the barcodes YUV video file, we will keep
>-  # it. Otherwise we create a temp file which is removed after it has been used.
>-  keep_barcodes_yuv_file = False
>-  if options.barcodes_yuv != _DEFAULT_BARCODES_FILE:
>-    keep_barcodes_yuv_file = True
>-
>-  # Calculate the number of barcodes - it is equal to the number of frames in
>-  # the base file.
>-  number_of_barcodes = CalculateFramesNumberFromYuv(
>-      options.base_frame_width, options.base_frame_height, options.base_yuv)
>-
>-  script_dir = os.path.dirname(os.path.abspath(sys.argv[0]))
>-  zxing_dir = os.path.join(script_dir, 'third_party', 'zxing')
>-  # Generate barcodes - will generate them in PNG.
>-  GenerateUpcaBarcodes(number_of_barcodes, options.barcode_width,
>-                       options.barcode_height,
>-                       output_directory=options.png_barcodes_output_dir,
>-                       path_to_zxing=zxing_dir)
>-  # Convert the PNG barcodes to to YUV format.
>-  ConvertPngToYuvBarcodes(options.png_barcodes_input_dir,
>-                          options.yuv_barcodes_output_dir)
>-  # Combine the YUV barcodes into one YUV file.
>-  CombineYuvFramesIntoOneFile(options.barcodes_yuv,
>-                              input_directory=options.yuv_frames_input_dir)
>-  # Overlay the barcodes over the base file.
>-  OverlayYuvFiles(options.barcode_width, options.barcode_height,
>-                  options.base_frame_width, options.base_frame_height,
>-                  options.barcodes_yuv, options.base_yuv, options.output_yuv)
>-
>-  if not keep_barcodes_yuv_file:
>-    # Remove the temporary barcodes YUV file
>-    os.remove(options.barcodes_yuv)
>-
>-
>-if __name__ == '__main__':
>-  sys.exit(main())
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/build_zxing.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/build_zxing.py
>deleted file mode 100644
>index 92696a78c2c36fd123a9fc12531dc34c2ed0e9d6..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/build_zxing.py
>+++ /dev/null
>@@ -1,44 +0,0 @@
>-#!/usr/bin/env python
>-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-import os
>-import subprocess
>-import sys
>-
>-
>-def RunAntBuildCommand(path_to_ant_build_file):
>-  """Tries to build the passed build file with ant."""
>-  ant_executable = 'ant'
>-  if sys.platform == 'win32':
>-    if os.getenv('ANT_HOME'):
>-      ant_executable = os.path.join(os.getenv('ANT_HOME'), 'bin', 'ant.bat')
>-    else:
>-      ant_executable = 'ant.bat'
>-  cmd = [ant_executable, '-buildfile', path_to_ant_build_file]
>-  try:
>-    process = subprocess.Popen(cmd, stdout=sys.stdout, stderr=sys.stderr)
>-    process.wait()
>-    if process.returncode != 0:
>-      print >> sys.stderr, 'Failed to execute: %s' % ' '.join(cmd)
>-    return process.returncode
>-  except subprocess.CalledProcessError as e:
>-    print >> sys.stderr, 'Failed to execute: %s.\nCause: %s' % (' '.join(cmd),
>-                                                                e)
>-    return -1
>-
>-def main():
>-  core_build = os.path.join('third_party', 'zxing', 'core', 'build.xml')
>-  RunAntBuildCommand(core_build)
>-
>-  javase_build = os.path.join('third_party', 'zxing', 'javase', 'build.xml')
>-  return RunAntBuildCommand(javase_build)
>-
>-
>-if __name__ == '__main__':
>-  sys.exit(main())
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/helper_functions.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/helper_functions.py
>deleted file mode 100644
>index e27322f41f92670b711ae5f8a8a7b6071e1407c3..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/helper_functions.py
>+++ /dev/null
>@@ -1,105 +0,0 @@
>-#!/usr/bin/env python
>-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-import multiprocessing
>-import os
>-import subprocess
>-import sys
>-
>-_DEFAULT_PADDING = 4
>-
>-
>-class HelperError(Exception):
>-  """Exception raised for errors in the helper."""
>-  pass
>-
>-
>-def ZeroPad(number, padding=_DEFAULT_PADDING):
>-  """Converts an int into a zero padded string.
>-
>-  Args:
>-    number(int): The number to convert.
>-    padding(int): The number of chars in the output. Note that if you pass for
>-      example number=23456 and padding=4, the output will still be '23456',
>-      i.e. it will not be cropped. If you pass number=2 and padding=4, the
>-      return value will be '0002'.
>-  Return:
>-    (string): The zero padded number converted to string.
>-  """
>-  return str(number).zfill(padding)
>-
>-
>-def RunShellCommand(cmd_list, fail_msg=None):
>-  """Executes a command.
>-
>-  Args:
>-    cmd_list(list): Command list to execute.
>-    fail_msg(string): Message describing the error in case the command fails.
>-
>-  Return:
>-    (string): The standard output from running the command.
>-
>-  Raise:
>-    HelperError: If command fails.
>-  """
>-  process = subprocess.Popen(cmd_list, stdout=subprocess.PIPE,
>-                             stderr=subprocess.PIPE)
>-  output, error = process.communicate()
>-  if process.returncode != 0:
>-    if fail_msg:
>-      print >> sys.stderr, fail_msg
>-    raise HelperError('Failed to run %s: command returned %d and printed '
>-                      '%s and %s' % (' '.join(cmd_list), process.returncode,
>-                                     output, error))
>-  return output.strip()
>-
>-
>-def PerformActionOnAllFiles(directory, file_pattern, file_extension,
>-                            start_number, action, **kwargs):
>-  """Function that performs a given action on all files matching a pattern.
>-
>-  It is assumed that the files are named file_patternxxxx.file_extension, where
>-  xxxx are digits starting from start_number.
>-
>-  Args:
>-    directory(string): The directory where the files live.
>-    file_pattern(string): The name pattern of the files.
>-    file_extension(string): The files' extension.
>-    start_number(int): From where to start to count frames.
>-    action(function): The action to be performed over the files. Must return
>-        False if the action failed, True otherwise. It should take a file name
>-        as the first argument and **kwargs as arguments. The function must be
>-        possible to pickle, so it cannot be a bound function (for instance).
>-
>-  Return:
>-    (bool): Whether performing the action over all files was successful or not.
>-  """
>-  file_prefix = os.path.join(directory, file_pattern)
>-  file_number = start_number
>-
>-  process_pool = multiprocessing.Pool(processes=multiprocessing.cpu_count())
>-  results = []
>-  while True:
>-    zero_padded_file_number = ZeroPad(file_number)
>-    file_name = file_prefix + zero_padded_file_number + '.' + file_extension
>-    if not os.path.isfile(file_name):
>-      break
>-    future = process_pool.apply_async(action, args=(file_name,), kwds=kwargs)
>-    results.append(future)
>-    file_number += 1
>-
>-  successful = True
>-  for result in results:
>-    if not result.get():
>-      print "At least one action %s failed for files %sxxxx.%s." % (
>-          action, file_pattern, file_extension)
>-      successful = False
>-
>-  process_pool.close()
>-  return successful
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/yuv_cropper.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/yuv_cropper.py
>deleted file mode 100644
>index 609f07dc9cc7ad90ec2d429167f92b53a782cf3a..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/yuv_cropper.py
>+++ /dev/null
>@@ -1,125 +0,0 @@
>-#!/usr/bin/env python
>-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-import optparse
>-import os
>-import sys
>-
>-
>-def _CropOneFrame(yuv_file, output_file, component_sizes):
>-  """Crops one frame.
>-
>-  This function crops one frame going through all the YUV planes and cropping
>-  respective amount of rows.
>-
>-  Args:
>-    yuv_file(file): The opened (for binary reading) YUV file.
>-    output_file(file): The opened (for binary writing) file.
>-    component_sizes(list of 3 3-ples): The list contains the sizes for all the
>-      planes (Y, U, V) of the YUV file plus the crop_height scaled for every
>-      plane. The sizes equal width, height and crop_height for the Y plane,
>-      and are equal to width/2, height/2 and crop_height/2 for the U and V
>-      planes.
>-  Return:
>-    (bool): True if there are more frames to crop, False otherwise.
>-  """
>-  for comp_width, comp_height, comp_crop_height in component_sizes:
>-    for row in range(comp_height):
>-      # Read the plane data for this row.
>-      yuv_plane = yuv_file.read(comp_width)
>-
>-      # If the plane is empty, we have reached the end of the file.
>-      if yuv_plane == "":
>-        return False
>-
>-      # Only write the plane data for the rows bigger than crop_height.
>-      if row >= comp_crop_height:
>-        output_file.write(yuv_plane)
>-  return True
>-
>-
>-def CropFrames(yuv_file_name, output_file_name, width, height, crop_height):
>-  """Crops rows of pixels from the top of the YUV frames.
>-
>-  This function goes through all the frames in a video and crops the crop_height
>-  top pixel rows of every frame.
>-
>-  Args:
>-    yuv_file_name(string): The name of the YUV file to be cropped.
>-    output_file_name(string): The name of the output file where the result will
>-      be written.
>-    width(int): The width of the original YUV file.
>-    height(int): The height of the original YUV file.
>-    crop_height(int): The height (the number of pixel rows) to be cropped from
>-      the frames.
>-  """
>-  # Component sizes = [Y_sizes, U_sizes, V_sizes].
>-  component_sizes = [(width, height, crop_height),
>-                     (width/2, height/2, crop_height/2),
>-                     (width/2, height/2, crop_height/2)]
>-
>-  yuv_file = open(yuv_file_name, 'rb')
>-  output_file = open(output_file_name, 'wb')
>-
>-  data_left = True
>-  while data_left:
>-    data_left = _CropOneFrame(yuv_file, output_file, component_sizes)
>-
>-  yuv_file.close()
>-  output_file.close()
>-
>-
>-def _ParseArgs():
>-  """Registers the command-line options."""
>-  usage = "usage: %prog [options]"
>-  parser = optparse.OptionParser(usage=usage)
>-
>-  parser.add_option('--width', type='int',
>-                    default=352,
>-                    help=('Width of the YUV file\'s frames. '
>-                          'Default: %default'))
>-  parser.add_option('--height', type='int', default=288,
>-                    help=('Height of the YUV file\'s frames. '
>-                          'Default: %default'))
>-  parser.add_option('--crop_height', type='int', default=32,
>-                    help=('How much of the top of the YUV file to crop. '
>-                          'Has to be module of 2. Default: %default'))
>-  parser.add_option('--yuv_file', type='string',
>-                    help=('The YUV file to be cropped.'))
>-  parser.add_option('--output_file', type='string', default='output.yuv',
>-                    help=('The output YUV file containing the cropped YUV. '
>-                          'Default: %default'))
>-  options = parser.parse_args()[0]
>-  if not options.yuv_file:
>-    parser.error('yuv_file argument missing. Please specify input YUV file!')
>-  return options
>-
>-
>-def main():
>-  """A tool to crop rows of pixels from the top part of a YUV file.
>-
>-  A simple invocation will be:
>-  ./yuv_cropper.py --width=640 --height=480 --crop_height=32
>-  --yuv_file=<path_and_name_of_yuv_file>
>-  --output_yuv=<path and name_of_output_file>
>-  """
>-  options = _ParseArgs()
>-
>-  if os.path.getsize(options.yuv_file) == 0:
>-    sys.stderr.write('Error: The YUV file you have passed has size 0. The '
>-                     'produced output will also have size 0.\n')
>-    return -1
>-
>-  CropFrames(options.yuv_file, options.output_file, options.width,
>-             options.height, options.crop_height)
>-  return 0
>-
>-
>-if __name__ == '__main__':
>-  sys.exit(main())
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/compare_videos.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/compare_videos.py
>index 79790439b176daac191727cd0c217c946d52579d..0cb4a6ddb199f560f467c4ddd9b2551cf443c601 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/compare_videos.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/compare_videos.py
>@@ -7,6 +7,7 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>+import json
> import optparse
> import os
> import shutil
>@@ -45,25 +46,15 @@ def _ParseArgs():
>   parser.add_option('--vmaf_phone_model', action='store_true',
>                     help='Whether to use phone model in VMAF.')
>   parser.add_option('--barcode_decoder', type='string',
>-                    help=('Path to the barcode decoder script. By default, we '
>-                          'will assume we can find it in barcode_tools/'
>-                          'relative to this directory.'))
>+                    help=('DEPRECATED'))
>   parser.add_option('--ffmpeg_path', type='string',
>-                    help=('The path to where the ffmpeg executable is located. '
>-                          'If omitted, it will be assumed to be present in the '
>-                          'PATH with the name ffmpeg[.exe].'))
>+                    help=('DEPRECATED'))
>   parser.add_option('--zxing_path', type='string',
>-                    help=('The path to where the zxing executable is located. '
>-                          'If omitted, it will be assumed to be present in the '
>-                          'PATH with the name zxing[.exe].'))
>+                    help=('DEPRECATED'))
>   parser.add_option('--stats_file_ref', type='string', default='stats_ref.txt',
>-                    help=('Path to the temporary stats file to be created and '
>-                          'used for the reference video file. '
>-                          'Default: %default'))
>+                    help=('DEPRECATED'))
>   parser.add_option('--stats_file_test', type='string',
>-                    default='stats_test.txt',
>-                    help=('Path to the temporary stats file to be created and '
>-                          'used for the test video file. Default: %default'))
>+                    help=('DEPRECATED'))
>   parser.add_option('--stats_file', type='string',
>                     help=('DEPRECATED'))
>   parser.add_option('--yuv_frame_width', type='int', default=640,
>@@ -74,11 +65,6 @@ def _ParseArgs():
>                     help='Where to store perf results in chartjson format.')
>   options, _ = parser.parse_args()
> 
>-  if options.stats_file:
>-    options.stats_file_test = options.stats_file
>-    print ('WARNING: Using deprecated switch --stats_file. '
>-           'The new flag is --stats_file_test.')
>-
>   if not options.ref_video:
>     parser.error('You must provide a path to the reference video!')
>   if not os.path.exists(options.ref_video):
>@@ -107,34 +93,6 @@ def _DevNull():
>   """
>   return open(os.devnull, 'r')
> 
>-def DecodeBarcodesInVideo(options, path_to_decoder, video, stat_file):
>-  # Run barcode decoder on the test video to identify frame numbers.
>-  png_working_directory = tempfile.mkdtemp()
>-  cmd = [
>-    sys.executable,
>-    path_to_decoder,
>-    '--yuv_file=%s' % video,
>-    '--yuv_frame_width=%d' % options.yuv_frame_width,
>-    '--yuv_frame_height=%d' % options.yuv_frame_height,
>-    '--stats_file=%s' % stat_file,
>-    '--png_working_dir=%s' % png_working_directory,
>-  ]
>-  if options.zxing_path:
>-    cmd.append('--zxing_path=%s' % options.zxing_path)
>-  if options.ffmpeg_path:
>-    cmd.append('--ffmpeg_path=%s' % options.ffmpeg_path)
>-
>-
>-  barcode_decoder = subprocess.Popen(cmd, stdin=_DevNull(),
>-                                     stdout=sys.stdout, stderr=sys.stderr)
>-  barcode_decoder.wait()
>-
>-  shutil.rmtree(png_working_directory)
>-  if barcode_decoder.returncode != 0:
>-    print 'Failed to run barcode decoder script.'
>-    return 1
>-  return 0
>-
> 
> def _RunFrameAnalyzer(options, yuv_directory=None):
>   """Run frame analyzer to compare the videos and print output."""
>@@ -162,12 +120,9 @@ def _RunFrameAnalyzer(options, yuv_directory=None):
>   return frame_analyzer.returncode
> 
> 
>-def _RunVmaf(options, yuv_directory):
>+def _RunVmaf(options, yuv_directory, logfile):
>   """ Run VMAF to compare videos and print output.
> 
>-  The provided vmaf directory is assumed to contain a c++ wrapper executable
>-  and a model.
>-
>   The yuv_directory is assumed to have been populated with a reference and test
>   video in .yuv format, with names according to the label.
>   """
>@@ -179,26 +134,29 @@ def _RunVmaf(options, yuv_directory):
>     os.path.join(yuv_directory, "ref.yuv"),
>     os.path.join(yuv_directory, "test.yuv"),
>     options.vmaf_model,
>+    '--log',
>+    logfile,
>+    '--log-fmt',
>+    'json',
>   ]
>   if options.vmaf_phone_model:
>     cmd.append('--phone-model')
> 
>   vmaf = subprocess.Popen(cmd, stdin=_DevNull(),
>-                          stdout=subprocess.PIPE, stderr=sys.stderr)
>+                          stdout=sys.stdout, stderr=sys.stderr)
>   vmaf.wait()
>   if vmaf.returncode != 0:
>     print 'Failed to run VMAF.'
>     return 1
>-  output = vmaf.stdout.read()
>-  # Extract score from VMAF output.
>-  try:
>-    score = float(output.split('\n')[2].split()[3])
>-  except (ValueError, IndexError):
>-    print 'Error in VMAF output (expected "VMAF score = [float]" on line 3):'
>-    print output
>-    return 1
> 
>-  print 'RESULT Vmaf: %s= %f' % (options.label, score)
>+  # Read per-frame scores from VMAF output and print.
>+  with open(logfile) as f:
>+    vmaf_data = json.load(f)
>+    vmaf_scores = []
>+    for frame in vmaf_data['frames']:
>+      vmaf_scores.append(frame['metrics']['vmaf'])
>+    print 'RESULT VMAF: %s=' % options.label, vmaf_scores
>+
>   return 0
> 
> 
>@@ -213,40 +171,24 @@ def main():
> 
>   Running vmaf requires the following arguments:
>   --vmaf, --vmaf_model, --yuv_frame_width, --yuv_frame_height
>-
>-  Notice that the prerequisites for barcode_decoder.py also applies to this
>-  script. The means the following executables have to be available in the PATH:
>-  * zxing
>-  * ffmpeg
>   """
>   options = _ParseArgs()
> 
>-  if options.barcode_decoder:
>-    path_to_decoder = options.barcode_decoder
>-  else:
>-    path_to_decoder = os.path.join(SCRIPT_DIR, 'barcode_tools',
>-                                   'barcode_decoder.py')
>-
>-  if DecodeBarcodesInVideo(options, path_to_decoder,
>-                           options.ref_video, options.stats_file_ref) != 0:
>-    return 1
>-  if DecodeBarcodesInVideo(options, path_to_decoder,
>-                           options.test_video, options.stats_file_test) != 0:
>-    return 1
>-
>   if options.vmaf:
>     try:
>       # Directory to save temporary YUV files for VMAF in frame_analyzer.
>       yuv_directory = tempfile.mkdtemp()
>+      _, vmaf_logfile = tempfile.mkstemp()
> 
>       # Run frame analyzer to compare the videos and print output.
>       if _RunFrameAnalyzer(options, yuv_directory=yuv_directory) != 0:
>         return 1
> 
>       # Run VMAF for further video comparison and print output.
>-      return _RunVmaf(options, yuv_directory)
>+      return _RunVmaf(options, yuv_directory, vmaf_logfile)
>     finally:
>       shutil.rmtree(yuv_directory)
>+      os.remove(vmaf_logfile)
>   else:
>     return _RunFrameAnalyzer(options)
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
>index 43cbd69fc8e18c9461729d352fdaeb0b0d3185e5..3bdfbf32a0e1b516a54978451243252c0934c019 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
>@@ -18,6 +18,7 @@
> #include <utility>
> 
> #include "absl/memory/memory.h"
>+#include "api/transport/goog_cc_factory.h"
> #include "call/audio_receive_stream.h"
> #include "call/audio_send_stream.h"
> #include "call/call.h"
>@@ -36,8 +37,9 @@
> #include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
> #include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
> #include "modules/congestion_controller/include/receive_side_congestion_controller.h"
>-#include "modules/congestion_controller/include/send_side_congestion_controller.h"
>+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
> #include "modules/pacing/packet_router.h"
>+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
>@@ -175,13 +177,13 @@ constexpr float kBottomMargin = 0.02f;
> constexpr float kTopMargin = 0.05f;
> 
> absl::optional<double> NetworkDelayDiff_AbsSendTime(
>-    const LoggedRtpPacket& old_packet,
>-    const LoggedRtpPacket& new_packet) {
>-  if (old_packet.header.extension.hasAbsoluteSendTime &&
>-      new_packet.header.extension.hasAbsoluteSendTime) {
>+    const LoggedRtpPacketIncoming& old_packet,
>+    const LoggedRtpPacketIncoming& new_packet) {
>+  if (old_packet.rtp.header.extension.hasAbsoluteSendTime &&
>+      new_packet.rtp.header.extension.hasAbsoluteSendTime) {
>     int64_t send_time_diff = WrappingDifference(
>-        new_packet.header.extension.absoluteSendTime,
>-        old_packet.header.extension.absoluteSendTime, 1ul << 24);
>+        new_packet.rtp.header.extension.absoluteSendTime,
>+        old_packet.rtp.header.extension.absoluteSendTime, 1ul << 24);
>     int64_t recv_time_diff =
>         new_packet.log_time_us() - old_packet.log_time_us();
>     double delay_change_us =
>@@ -193,34 +195,31 @@ absl::optional<double> NetworkDelayDiff_AbsSendTime(
> }
> 
> absl::optional<double> NetworkDelayDiff_CaptureTime(
>-    const LoggedRtpPacket& old_packet,
>-    const LoggedRtpPacket& new_packet) {
>-  int64_t send_time_diff = WrappingDifference(
>-      new_packet.header.timestamp, old_packet.header.timestamp, 1ull << 32);
>+    const LoggedRtpPacketIncoming& old_packet,
>+    const LoggedRtpPacketIncoming& new_packet,
>+    const double sample_rate) {
>+  int64_t send_time_diff =
>+      WrappingDifference(new_packet.rtp.header.timestamp,
>+                         old_packet.rtp.header.timestamp, 1ull << 32);
>   int64_t recv_time_diff = new_packet.log_time_us() - old_packet.log_time_us();
> 
>-  const double kVideoSampleRate = 90000;
>-  // TODO(terelius): We treat all streams as video for now, even though
>-  // audio might be sampled at e.g. 16kHz, because it is really difficult to
>-  // figure out the true sampling rate of a stream. The effect is that the
>-  // delay will be scaled incorrectly for non-video streams.
>-
>   double delay_change =
>       static_cast<double>(recv_time_diff) / 1000 -
>-      static_cast<double>(send_time_diff) / kVideoSampleRate * 1000;
>+      static_cast<double>(send_time_diff) / sample_rate * 1000;
>   if (delay_change < -10000 || 10000 < delay_change) {
>     RTC_LOG(LS_WARNING) << "Very large delay change. Timestamps correct?";
>-    RTC_LOG(LS_WARNING) << "Old capture time " << old_packet.header.timestamp
>-                        << ", received time " << old_packet.log_time_us();
>-    RTC_LOG(LS_WARNING) << "New capture time " << new_packet.header.timestamp
>-                        << ", received time " << new_packet.log_time_us();
>+    RTC_LOG(LS_WARNING) << "Old capture time "
>+                        << old_packet.rtp.header.timestamp << ", received time "
>+                        << old_packet.log_time_us();
>+    RTC_LOG(LS_WARNING) << "New capture time "
>+                        << new_packet.rtp.header.timestamp << ", received time "
>+                        << new_packet.log_time_us();
>     RTC_LOG(LS_WARNING) << "Receive time difference " << recv_time_diff << " = "
>                         << static_cast<double>(recv_time_diff) /
>                                kNumMicrosecsPerSec
>                         << "s";
>     RTC_LOG(LS_WARNING) << "Send time difference " << send_time_diff << " = "
>-                        << static_cast<double>(send_time_diff) /
>-                               kVideoSampleRate
>+                        << static_cast<double>(send_time_diff) / sample_rate
>                         << "s";
>   }
>   return delay_change;
>@@ -274,9 +273,10 @@ void AccumulatePairs(
>   for (size_t i = 1; i < data.size(); i++) {
>     float x = fx(data[i]);
>     absl::optional<ResultType> y = fy(data[i - 1], data[i]);
>-    if (y)
>+    if (y) {
>       sum += *y;
>-    result->points.emplace_back(x, static_cast<float>(sum));
>+      result->points.emplace_back(x, static_cast<float>(sum));
>+    }
>   }
> }
> 
>@@ -486,17 +486,15 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLogNew& log,
>                    << " (LOG_START, LOG_END) segments in log.";
> }
> 
>-class BitrateObserver : public NetworkChangedObserver,
>-                        public RemoteBitrateObserver {
>+class BitrateObserver : public RemoteBitrateObserver {
>  public:
>   BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
> 
>-  void OnNetworkChanged(uint32_t bitrate_bps,
>-                        uint8_t fraction_lost,
>-                        int64_t rtt_ms,
>-                        int64_t probing_interval_ms) override {
>-    last_bitrate_bps_ = bitrate_bps;
>-    bitrate_updated_ = true;
>+  void Update(NetworkControlUpdate update) {
>+    if (update.target_rate) {
>+      last_bitrate_bps_ = update.target_rate->target_rate.bps();
>+      bitrate_updated_ = true;
>+    }
>   }
> 
>   void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
>@@ -737,71 +735,58 @@ void EventLogAnalyzer::CreateIncomingPacketLossGraph(Plot* plot) {
>   plot->SetTitle("Estimated incoming loss rate");
> }
> 
>-void EventLogAnalyzer::CreateIncomingDelayDeltaGraph(Plot* plot) {
>-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(kIncomingPacket)) {
>+void EventLogAnalyzer::CreateIncomingDelayGraph(Plot* plot) {
>+  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
>     // Filter on SSRC.
>     if (!MatchingSsrc(stream.ssrc, desired_ssrc_) ||
>-        IsAudioSsrc(kIncomingPacket, stream.ssrc) ||
>-        !IsVideoSsrc(kIncomingPacket, stream.ssrc) ||
>         IsRtxSsrc(kIncomingPacket, stream.ssrc)) {
>       continue;
>     }
> 
>-    TimeSeries capture_time_data(
>-        GetStreamName(kIncomingPacket, stream.ssrc) + " capture-time",
>-        LineStyle::kBar);
>-    auto ToCallTime = [this](const LoggedRtpPacket& packet) {
>-      return this->ToCallTimeSec(packet.log_time_us());
>-    };
>-    ProcessPairs<LoggedRtpPacket, double>(
>-        ToCallTime, NetworkDelayDiff_CaptureTime, stream.packet_view,
>-        &capture_time_data);
>-    plot->AppendTimeSeries(std::move(capture_time_data));
>-
>-    TimeSeries send_time_data(
>-        GetStreamName(kIncomingPacket, stream.ssrc) + " abs-send-time",
>-        LineStyle::kBar);
>-    ProcessPairs<LoggedRtpPacket, double>(ToCallTime,
>-                                          NetworkDelayDiff_AbsSendTime,
>-                                          stream.packet_view, &send_time_data);
>-    plot->AppendTimeSeries(std::move(send_time_data));
>-  }
>-
>-  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>-                 kLeftMargin, kRightMargin);
>-  plot->SetSuggestedYAxis(0, 1, "Latency change (ms)", kBottomMargin,
>-                          kTopMargin);
>-  plot->SetTitle("Network latency difference between consecutive packets");
>-}
>-
>-void EventLogAnalyzer::CreateIncomingDelayGraph(Plot* plot) {
>-  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(kIncomingPacket)) {
>-    // Filter on SSRC.
>-    if (!MatchingSsrc(stream.ssrc, desired_ssrc_) ||
>-        IsAudioSsrc(kIncomingPacket, stream.ssrc) ||
>-        !IsVideoSsrc(kIncomingPacket, stream.ssrc) ||
>-        IsRtxSsrc(kIncomingPacket, stream.ssrc)) {
>+    const std::vector<LoggedRtpPacketIncoming>& packets =
>+        stream.incoming_packets;
>+    if (packets.size() < 100) {
>+      RTC_LOG(LS_WARNING) << "Can't estimate the RTP clock frequency with "
>+                          << packets.size() << " packets in the stream.";
>+      continue;
>+    }
>+    int64_t end_time_us = log_segments_.empty()
>+                              ? std::numeric_limits<int64_t>::max()
>+                              : log_segments_.front().second;
>+    absl::optional<uint32_t> estimated_frequency =
>+        EstimateRtpClockFrequency(packets, end_time_us);
>+    if (!estimated_frequency)
>+      continue;
>+    const double frequency_hz = *estimated_frequency;
>+    if (IsVideoSsrc(kIncomingPacket, stream.ssrc) && frequency_hz != 90000) {
>+      RTC_LOG(LS_WARNING)
>+          << "Video stream should use a 90 kHz clock but appears to use "
>+          << frequency_hz / 1000 << ". Discarding.";
>       continue;
>     }
> 
>+    auto ToCallTime = [this](const LoggedRtpPacketIncoming& packet) {
>+      return this->ToCallTimeSec(packet.log_time_us());
>+    };
>+    auto ToNetworkDelay = [frequency_hz](
>+                              const LoggedRtpPacketIncoming& old_packet,
>+                              const LoggedRtpPacketIncoming& new_packet) {
>+      return NetworkDelayDiff_CaptureTime(old_packet, new_packet, frequency_hz);
>+    };
>+
>     TimeSeries capture_time_data(
>         GetStreamName(kIncomingPacket, stream.ssrc) + " capture-time",
>         LineStyle::kLine);
>-    auto ToCallTime = [this](const LoggedRtpPacket& packet) {
>-      return this->ToCallTimeSec(packet.log_time_us());
>-    };
>-    AccumulatePairs<LoggedRtpPacket, double>(
>-        ToCallTime, NetworkDelayDiff_CaptureTime, stream.packet_view,
>-        &capture_time_data);
>+    AccumulatePairs<LoggedRtpPacketIncoming, double>(
>+        ToCallTime, ToNetworkDelay, packets, &capture_time_data);
>     plot->AppendTimeSeries(std::move(capture_time_data));
> 
>     TimeSeries send_time_data(
>         GetStreamName(kIncomingPacket, stream.ssrc) + " abs-send-time",
>         LineStyle::kLine);
>-    AccumulatePairs<LoggedRtpPacket, double>(
>-        ToCallTime, NetworkDelayDiff_AbsSendTime, stream.packet_view,
>-        &send_time_data);
>-    plot->AppendTimeSeries(std::move(send_time_data));
>+    AccumulatePairs<LoggedRtpPacketIncoming, double>(
>+        ToCallTime, NetworkDelayDiff_AbsSendTime, packets, &send_time_data);
>+    plot->AppendTimeSeriesIfNotEmpty(std::move(send_time_data));
>   }
> 
>   plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>@@ -1081,10 +1066,15 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
>   RtcEventLogNullImpl null_event_log;
>   PacketRouter packet_router;
>   PacedSender pacer(&clock, &packet_router, &null_event_log);
>-  SendSideCongestionController cc(&clock, &observer, &null_event_log, &pacer);
>+  TransportFeedbackAdapter transport_feedback(&clock);
>+  auto factory = GoogCcNetworkControllerFactory(&null_event_log);
>+  TimeDelta process_interval = factory.GetProcessInterval();
>   // TODO(holmer): Log the call config and use that here instead.
>   static const uint32_t kDefaultStartBitrateBps = 300000;
>-  cc.SetBweBitrates(0, kDefaultStartBitrateBps, -1);
>+  NetworkControllerConfig cc_config;
>+  cc_config.constraints.at_time = Timestamp::us(clock.TimeInMicroseconds());
>+  cc_config.constraints.starting_rate = DataRate::bps(kDefaultStartBitrateBps);
>+  auto goog_cc = factory.Create(cc_config);
> 
>   TimeSeries time_series("Delay-based estimate", LineStyle::kStep,
>                          PointStyle::kHighlight);
>@@ -1107,12 +1097,12 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
>       return static_cast<int64_t>(rtcp_iterator->log_time_us());
>     return std::numeric_limits<int64_t>::max();
>   };
>+  int64_t next_process_time_us_ = clock.TimeInMicroseconds();
> 
>   auto NextProcessTime = [&]() {
>     if (rtcp_iterator != incoming_rtcp.end() ||
>         rtp_iterator != outgoing_rtp.end()) {
>-      return clock.TimeInMicroseconds() +
>-             std::max<int64_t>(cc.TimeUntilNextProcess() * 1000, 0);
>+      return next_process_time_us_;
>     }
>     return std::numeric_limits<int64_t>::max();
>   };
>@@ -1129,24 +1119,32 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
>   AcknowledgedBitrateEstimator acknowledged_bitrate_estimator(
>       absl::make_unique<BitrateEstimator>());
> #endif  // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
>-  int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
>+  int64_t time_us =
>+      std::min({NextRtpTime(), NextRtcpTime(), NextProcessTime()});
>   int64_t last_update_us = 0;
>   while (time_us != std::numeric_limits<int64_t>::max()) {
>     clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
>     if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
>       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
>-      cc.OnTransportFeedback(rtcp_iterator->transport_feedback);
>-      std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
>-      SortPacketFeedbackVector(&feedback);
>+
>+      auto feedback_msg = transport_feedback.ProcessTransportFeedback(
>+          rtcp_iterator->transport_feedback);
>       absl::optional<uint32_t> bitrate_bps;
>-      if (!feedback.empty()) {
>+      if (feedback_msg) {
>+        observer.Update(goog_cc->OnTransportPacketsFeedback(*feedback_msg));
>+        std::vector<PacketFeedback> feedback =
>+            transport_feedback.GetTransportFeedbackVector();
>+        SortPacketFeedbackVector(&feedback);
>+        if (!feedback.empty()) {
> #if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
>-        acknowledged_bitrate_estimator.IncomingPacketFeedbackVector(feedback);
>+          acknowledged_bitrate_estimator.IncomingPacketFeedbackVector(feedback);
> #endif  // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
>-        for (const PacketFeedback& packet : feedback)
>-          acked_bitrate.Update(packet.payload_size, packet.arrival_time_ms);
>-        bitrate_bps = acked_bitrate.Rate(feedback.back().arrival_time_ms);
>+          for (const PacketFeedback& packet : feedback)
>+            acked_bitrate.Update(packet.payload_size, packet.arrival_time_ms);
>+          bitrate_bps = acked_bitrate.Rate(feedback.back().arrival_time_ms);
>+        }
>       }
>+
>       float x = ToCallTimeSec(clock.TimeInMicroseconds());
>       float y = bitrate_bps.value_or(0) / 1000;
>       acked_time_series.points.emplace_back(x, y);
>@@ -1161,19 +1159,25 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
>       const RtpPacketType& rtp_packet = *rtp_iterator->second;
>       if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
>         RTC_DCHECK(rtp_packet.rtp.header.extension.hasTransportSequenceNumber);
>-        cc.AddPacket(rtp_packet.rtp.header.ssrc,
>-                     rtp_packet.rtp.header.extension.transportSequenceNumber,
>-                     rtp_packet.rtp.total_length, PacedPacketInfo());
>+        transport_feedback.AddPacket(
>+            rtp_packet.rtp.header.ssrc,
>+            rtp_packet.rtp.header.extension.transportSequenceNumber,
>+            rtp_packet.rtp.total_length, PacedPacketInfo());
>         rtc::SentPacket sent_packet(
>             rtp_packet.rtp.header.extension.transportSequenceNumber,
>             rtp_packet.rtp.log_time_us() / 1000);
>-        cc.OnSentPacket(sent_packet);
>+        auto sent_msg = transport_feedback.ProcessSentPacket(sent_packet);
>+        if (sent_msg)
>+          observer.Update(goog_cc->OnSentPacket(*sent_msg));
>       }
>       ++rtp_iterator;
>     }
>     if (clock.TimeInMicroseconds() >= NextProcessTime()) {
>       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
>-      cc.Process();
>+      ProcessInterval msg;
>+      msg.at_time = Timestamp::us(clock.TimeInMicroseconds());
>+      observer.Update(goog_cc->OnProcessInterval(msg));
>+      next_process_time_us_ += process_interval.us();
>     }
>     if (observer.GetAndResetBitrateUpdated() ||
>         time_us - last_update_us >= 1e6) {
>@@ -1277,9 +1281,10 @@ void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) {
> void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
>   TimeSeries late_feedback_series("Late feedback results.", LineStyle::kNone,
>                                   PointStyle::kHighlight);
>-  TimeSeries time_series("Network Delay Change", LineStyle::kLine,
>+  TimeSeries time_series("Network delay", LineStyle::kLine,
>                          PointStyle::kHighlight);
>-  int64_t estimated_base_delay_ms = std::numeric_limits<int64_t>::max();
>+  int64_t min_send_receive_diff_ms = std::numeric_limits<int64_t>::max();
>+  int64_t min_rtt_ms = std::numeric_limits<int64_t>::max();
> 
>   int64_t prev_y = 0;
>   for (auto packet : GetNetworkTrace(parsed_log_)) {
>@@ -1292,16 +1297,22 @@ void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
>     }
>     int64_t y = packet.arrival_time_ms - packet.send_time_ms;
>     prev_y = y;
>-    estimated_base_delay_ms = std::min(y, estimated_base_delay_ms);
>+    int64_t rtt_ms = packet.feedback_arrival_time_ms - packet.send_time_ms;
>+    min_rtt_ms = std::min(rtt_ms, min_rtt_ms);
>+    min_send_receive_diff_ms = std::min(y, min_send_receive_diff_ms);
>     time_series.points.emplace_back(x, y);
>   }
> 
>-  // We assume that the base network delay (w/o queues) is the min delay
>-  // observed during the call.
>+  // We assume that the base network delay (w/o queues) is equal to half
>+  // the minimum RTT. Therefore rescale the delays by subtracting the minimum
>+  // observed 1-ways delay and add half the minumum RTT.
>+  const int64_t estimated_clock_offset_ms =
>+      min_send_receive_diff_ms - min_rtt_ms / 2;
>   for (TimeSeriesPoint& point : time_series.points)
>-    point.y -= estimated_base_delay_ms;
>+    point.y -= estimated_clock_offset_ms;
>   for (TimeSeriesPoint& point : late_feedback_series.points)
>-    point.y -= estimated_base_delay_ms;
>+    point.y -= estimated_clock_offset_ms;
>+
>   // Add the data set to the plot.
>   plot->AppendTimeSeriesIfNotEmpty(std::move(time_series));
>   plot->AppendTimeSeriesIfNotEmpty(std::move(late_feedback_series));
>@@ -1309,7 +1320,7 @@ void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
>   plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>                  kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
>-  plot->SetTitle("Network Delay Change.");
>+  plot->SetTitle("Network delay (based on per-packet feedback)");
> }
> 
> void EventLogAnalyzer::CreatePacerDelayGraph(Plot* plot) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.h
>index 4b8c27c9517d0cbcc0401d48ab118cc23d90128a..c6606c2b1f16b122d35b2e1fd6f3ced3ebf9e60f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.h
>@@ -45,7 +45,6 @@ class EventLogAnalyzer {
> 
>   void CreateIncomingPacketLossGraph(Plot* plot);
> 
>-  void CreateIncomingDelayDeltaGraph(Plot* plot);
>   void CreateIncomingDelayGraph(Plot* plot);
> 
>   void CreateFractionLossGraph(Plot* plot);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/main.cc
>index 0c3689c1bcae3a0fdd003862b1d8dff44dab2ee1..0b4e7617e6068d756f390f4be2c74cc91ebca620 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/main.cc
>@@ -20,151 +20,169 @@
> #include "test/field_trial.h"
> #include "test/testsupport/fileutils.h"
> 
>-DEFINE_string(plot_profile,
>-              "default",
>-              "A profile that selects a certain subset of the plots. Currently "
>-              "defined profiles are \"all\", \"none\", \"sendside_bwe\","
>-              "\"receiveside_bwe\" and \"default\"");
>+WEBRTC_DEFINE_string(
>+    plot_profile,
>+    "default",
>+    "A profile that selects a certain subset of the plots. Currently "
>+    "defined profiles are \"all\", \"none\", \"sendside_bwe\","
>+    "\"receiveside_bwe\" and \"default\"");
> 
>-DEFINE_bool(plot_incoming_packet_sizes,
>-            false,
>-            "Plot bar graph showing the size of each incoming packet.");
>-DEFINE_bool(plot_outgoing_packet_sizes,
>-            false,
>-            "Plot bar graph showing the size of each outgoing packet.");
>-DEFINE_bool(plot_incoming_packet_count,
>-            false,
>-            "Plot the accumulated number of packets for each incoming stream.");
>-DEFINE_bool(plot_outgoing_packet_count,
>-            false,
>-            "Plot the accumulated number of packets for each outgoing stream.");
>-DEFINE_bool(plot_audio_playout,
>-            false,
>-            "Plot bar graph showing the time between each audio playout.");
>-DEFINE_bool(plot_audio_level,
>-            false,
>-            "Plot line graph showing the audio level of incoming audio.");
>-DEFINE_bool(plot_incoming_sequence_number_delta,
>-            false,
>-            "Plot the sequence number difference between consecutive incoming "
>-            "packets.");
>-DEFINE_bool(
>-    plot_incoming_delay_delta,
>+WEBRTC_DEFINE_bool(plot_incoming_packet_sizes,
>+                   false,
>+                   "Plot bar graph showing the size of each incoming packet.");
>+WEBRTC_DEFINE_bool(plot_outgoing_packet_sizes,
>+                   false,
>+                   "Plot bar graph showing the size of each outgoing packet.");
>+WEBRTC_DEFINE_bool(
>+    plot_incoming_packet_count,
>     false,
>-    "Plot the difference in 1-way path delay between consecutive packets.");
>-DEFINE_bool(plot_incoming_delay,
>-            true,
>-            "Plot the 1-way path delay for incoming packets, normalized so "
>-            "that the first packet has delay 0.");
>-DEFINE_bool(plot_incoming_loss_rate,
>-            true,
>-            "Compute the loss rate for incoming packets using a method that's "
>-            "similar to the one used for RTCP SR and RR fraction lost. Note "
>-            "that the loss rate can be negative if packets are duplicated or "
>-            "reordered.");
>-DEFINE_bool(plot_incoming_bitrate,
>-            true,
>-            "Plot the total bitrate used by all incoming streams.");
>-DEFINE_bool(plot_outgoing_bitrate,
>-            true,
>-            "Plot the total bitrate used by all outgoing streams.");
>-DEFINE_bool(plot_incoming_stream_bitrate,
>-            true,
>-            "Plot the bitrate used by each incoming stream.");
>-DEFINE_bool(plot_outgoing_stream_bitrate,
>-            true,
>-            "Plot the bitrate used by each outgoing stream.");
>-DEFINE_bool(plot_simulated_receiveside_bwe,
>-            false,
>-            "Run the receive-side bandwidth estimator with the incoming rtp "
>-            "packets and plot the resulting estimate.");
>-DEFINE_bool(plot_simulated_sendside_bwe,
>-            false,
>-            "Run the send-side bandwidth estimator with the outgoing rtp and "
>-            "incoming rtcp and plot the resulting estimate.");
>-DEFINE_bool(plot_network_delay_feedback,
>-            true,
>-            "Compute network delay based on sent packets and the received "
>-            "transport feedback.");
>-DEFINE_bool(plot_fraction_loss_feedback,
>-            true,
>-            "Plot packet loss in percent for outgoing packets (as perceived by "
>-            "the send-side bandwidth estimator).");
>-DEFINE_bool(plot_pacer_delay,
>-            false,
>-            "Plot the time each sent packet has spent in the pacer (based on "
>-            "the difference between the RTP timestamp and the send "
>-            "timestamp).");
>-DEFINE_bool(plot_timestamps,
>-            false,
>-            "Plot the rtp timestamps of all rtp and rtcp packets over time.");
>-DEFINE_bool(plot_rtcp_details,
>-            false,
>-            "Plot the contents of all report blocks in all sender and receiver "
>-            "reports. This includes fraction lost, cumulative number of lost "
>-            "packets, extended highest sequence number and time since last "
>-            "received SR.");
>-DEFINE_bool(plot_audio_encoder_bitrate_bps,
>-            false,
>-            "Plot the audio encoder target bitrate.");
>-DEFINE_bool(plot_audio_encoder_frame_length_ms,
>-            false,
>-            "Plot the audio encoder frame length.");
>-DEFINE_bool(
>+    "Plot the accumulated number of packets for each incoming stream.");
>+WEBRTC_DEFINE_bool(
>+    plot_outgoing_packet_count,
>+    false,
>+    "Plot the accumulated number of packets for each outgoing stream.");
>+WEBRTC_DEFINE_bool(
>+    plot_audio_playout,
>+    false,
>+    "Plot bar graph showing the time between each audio playout.");
>+WEBRTC_DEFINE_bool(
>+    plot_audio_level,
>+    false,
>+    "Plot line graph showing the audio level of incoming audio.");
>+WEBRTC_DEFINE_bool(
>+    plot_incoming_sequence_number_delta,
>+    false,
>+    "Plot the sequence number difference between consecutive incoming "
>+    "packets.");
>+WEBRTC_DEFINE_bool(
>+    plot_incoming_delay,
>+    true,
>+    "Plot the 1-way path delay for incoming packets, normalized so "
>+    "that the first packet has delay 0.");
>+WEBRTC_DEFINE_bool(
>+    plot_incoming_loss_rate,
>+    true,
>+    "Compute the loss rate for incoming packets using a method that's "
>+    "similar to the one used for RTCP SR and RR fraction lost. Note "
>+    "that the loss rate can be negative if packets are duplicated or "
>+    "reordered.");
>+WEBRTC_DEFINE_bool(plot_incoming_bitrate,
>+                   true,
>+                   "Plot the total bitrate used by all incoming streams.");
>+WEBRTC_DEFINE_bool(plot_outgoing_bitrate,
>+                   true,
>+                   "Plot the total bitrate used by all outgoing streams.");
>+WEBRTC_DEFINE_bool(plot_incoming_stream_bitrate,
>+                   true,
>+                   "Plot the bitrate used by each incoming stream.");
>+WEBRTC_DEFINE_bool(plot_outgoing_stream_bitrate,
>+                   true,
>+                   "Plot the bitrate used by each outgoing stream.");
>+WEBRTC_DEFINE_bool(
>+    plot_simulated_receiveside_bwe,
>+    false,
>+    "Run the receive-side bandwidth estimator with the incoming rtp "
>+    "packets and plot the resulting estimate.");
>+WEBRTC_DEFINE_bool(
>+    plot_simulated_sendside_bwe,
>+    false,
>+    "Run the send-side bandwidth estimator with the outgoing rtp and "
>+    "incoming rtcp and plot the resulting estimate.");
>+WEBRTC_DEFINE_bool(
>+    plot_network_delay_feedback,
>+    true,
>+    "Compute network delay based on sent packets and the received "
>+    "transport feedback.");
>+WEBRTC_DEFINE_bool(
>+    plot_fraction_loss_feedback,
>+    true,
>+    "Plot packet loss in percent for outgoing packets (as perceived by "
>+    "the send-side bandwidth estimator).");
>+WEBRTC_DEFINE_bool(
>+    plot_pacer_delay,
>+    false,
>+    "Plot the time each sent packet has spent in the pacer (based on "
>+    "the difference between the RTP timestamp and the send "
>+    "timestamp).");
>+WEBRTC_DEFINE_bool(
>+    plot_timestamps,
>+    false,
>+    "Plot the rtp timestamps of all rtp and rtcp packets over time.");
>+WEBRTC_DEFINE_bool(
>+    plot_rtcp_details,
>+    false,
>+    "Plot the contents of all report blocks in all sender and receiver "
>+    "reports. This includes fraction lost, cumulative number of lost "
>+    "packets, extended highest sequence number and time since last "
>+    "received SR.");
>+WEBRTC_DEFINE_bool(plot_audio_encoder_bitrate_bps,
>+                   false,
>+                   "Plot the audio encoder target bitrate.");
>+WEBRTC_DEFINE_bool(plot_audio_encoder_frame_length_ms,
>+                   false,
>+                   "Plot the audio encoder frame length.");
>+WEBRTC_DEFINE_bool(
>     plot_audio_encoder_packet_loss,
>     false,
>     "Plot the uplink packet loss fraction which is sent to the audio encoder.");
>-DEFINE_bool(plot_audio_encoder_fec, false, "Plot the audio encoder FEC.");
>-DEFINE_bool(plot_audio_encoder_dtx, false, "Plot the audio encoder DTX.");
>-DEFINE_bool(plot_audio_encoder_num_channels,
>-            false,
>-            "Plot the audio encoder number of channels.");
>-DEFINE_bool(plot_neteq_stats, false, "Plot the NetEq statistics.");
>-DEFINE_bool(plot_ice_candidate_pair_config,
>-            false,
>-            "Plot the ICE candidate pair config events.");
>-DEFINE_bool(plot_ice_connectivity_check,
>-            false,
>-            "Plot the ICE candidate pair connectivity checks.");
>+WEBRTC_DEFINE_bool(plot_audio_encoder_fec,
>+                   false,
>+                   "Plot the audio encoder FEC.");
>+WEBRTC_DEFINE_bool(plot_audio_encoder_dtx,
>+                   false,
>+                   "Plot the audio encoder DTX.");
>+WEBRTC_DEFINE_bool(plot_audio_encoder_num_channels,
>+                   false,
>+                   "Plot the audio encoder number of channels.");
>+WEBRTC_DEFINE_bool(plot_neteq_stats, false, "Plot the NetEq statistics.");
>+WEBRTC_DEFINE_bool(plot_ice_candidate_pair_config,
>+                   false,
>+                   "Plot the ICE candidate pair config events.");
>+WEBRTC_DEFINE_bool(plot_ice_connectivity_check,
>+                   false,
>+                   "Plot the ICE candidate pair connectivity checks.");
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     force_fieldtrials,
>     "",
>     "Field trials control experimental feature code which can be forced. "
>     "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/"
>     " will assign the group Enabled to field trial WebRTC-FooFeature. Multiple "
>     "trials are separated by \"/\"");
>-DEFINE_string(wav_filename,
>-              "",
>-              "Path to wav file used for simulation of jitter buffer");
>-DEFINE_bool(help, false, "prints this message");
>+WEBRTC_DEFINE_string(wav_filename,
>+                     "",
>+                     "Path to wav file used for simulation of jitter buffer");
>+WEBRTC_DEFINE_bool(help, false, "prints this message");
> 
>-DEFINE_bool(show_detector_state,
>-            false,
>-            "Show the state of the delay based BWE detector on the total "
>-            "bitrate graph");
>+WEBRTC_DEFINE_bool(
>+    show_detector_state,
>+    false,
>+    "Show the state of the delay based BWE detector on the total "
>+    "bitrate graph");
> 
>-DEFINE_bool(show_alr_state,
>-            false,
>-            "Show the state ALR state on the total bitrate graph");
>+WEBRTC_DEFINE_bool(show_alr_state,
>+                   false,
>+                   "Show the state ALR state on the total bitrate graph");
> 
>-DEFINE_bool(parse_unconfigured_header_extensions,
>-            true,
>-            "Attempt to parse unconfigured header extensions using the default "
>-            "WebRTC mapping. This can give very misleading results if the "
>-            "application negotiates a different mapping.");
>+WEBRTC_DEFINE_bool(
>+    parse_unconfigured_header_extensions,
>+    true,
>+    "Attempt to parse unconfigured header extensions using the default "
>+    "WebRTC mapping. This can give very misleading results if the "
>+    "application negotiates a different mapping.");
> 
>-DEFINE_bool(print_triage_alerts,
>-            false,
>-            "Print triage alerts, i.e. a list of potential problems.");
>+WEBRTC_DEFINE_bool(print_triage_alerts,
>+                   false,
>+                   "Print triage alerts, i.e. a list of potential problems.");
> 
>-DEFINE_bool(normalize_time,
>-            true,
>-            "Normalize the log timestamps so that the call starts at time 0.");
>+WEBRTC_DEFINE_bool(
>+    normalize_time,
>+    true,
>+    "Normalize the log timestamps so that the call starts at time 0.");
> 
>-DEFINE_bool(protobuf_output,
>-            false,
>-            "Output charts as protobuf instead of python code.");
>+WEBRTC_DEFINE_bool(protobuf_output,
>+                   false,
>+                   "Output charts as protobuf instead of python code.");
> 
> void SetAllPlotFlags(bool setting);
> 
>@@ -194,7 +212,6 @@ int main(int argc, char* argv[]) {
>   } else if (strcmp(FLAG_plot_profile, "receiveside_bwe") == 0) {
>     SetAllPlotFlags(false);
>     FLAG_plot_incoming_packet_sizes = true;
>-    FLAG_plot_incoming_delay_delta = true;
>     FLAG_plot_incoming_delay = true;
>     FLAG_plot_incoming_loss_rate = true;
>     FLAG_plot_incoming_bitrate = true;
>@@ -218,7 +235,6 @@ int main(int argc, char* argv[]) {
>     return 0;
>   }
> 
>-  webrtc::test::SetExecutablePath(argv[0]);
>   webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
>   // InitFieldTrialsFromString stores the char*, so the char array must outlive
>   // the application.
>@@ -236,9 +252,7 @@ int main(int argc, char* argv[]) {
> 
>   if (!parsed_log.ParseFile(filename)) {
>     std::cerr << "Could not parse the entire log file." << std::endl;
>-    std::cerr << "Proceeding to analyze the first "
>-              << parsed_log.GetNumberOfEvents() << " events in the file."
>-              << std::endl;
>+    std::cerr << "Only the parsable events will be analyzed." << std::endl;
>   }
> 
>   webrtc::EventLogAnalyzer analyzer(parsed_log, FLAG_normalize_time);
>@@ -277,9 +291,6 @@ int main(int argc, char* argv[]) {
>   if (FLAG_plot_incoming_sequence_number_delta) {
>     analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot());
>   }
>-  if (FLAG_plot_incoming_delay_delta) {
>-    analyzer.CreateIncomingDelayDeltaGraph(collection->AppendNewPlot());
>-  }
>   if (FLAG_plot_incoming_delay) {
>     analyzer.CreateIncomingDelayGraph(collection->AppendNewPlot());
>   }
>@@ -466,7 +477,6 @@ void SetAllPlotFlags(bool setting) {
>   FLAG_plot_audio_playout = setting;
>   FLAG_plot_audio_level = setting;
>   FLAG_plot_incoming_sequence_number_delta = setting;
>-  FLAG_plot_incoming_delay_delta = setting;
>   FLAG_plot_incoming_delay = setting;
>   FLAG_plot_incoming_loss_rate = setting;
>   FLAG_plot_incoming_bitrate = setting;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/force_mic_volume_max/force_mic_volume_max.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/force_mic_volume_max/force_mic_volume_max.cc
>deleted file mode 100644
>index 2cd1b36d9047b89ce959a68bfc659b1ef94fdb04..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/force_mic_volume_max/force_mic_volume_max.cc
>+++ /dev/null
>@@ -1,58 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// This utility will portably force the volume of the default microphone to max.
>-
>-#include <stdio.h>
>-
>-#include "modules/audio_device/include/audio_device.h"
>-
>-using webrtc::AudioDeviceModule;
>-
>-#if defined(_WIN32)
>-#define DEFAULT_INPUT_DEVICE (AudioDeviceModule::kDefaultCommunicationDevice)
>-#else
>-#define DEFAULT_INPUT_DEVICE (0)
>-#endif
>-
>-int main(int /*argc*/, char* /*argv*/ []) {
>-  // Create and initialize the ADM.
>-  rtc::scoped_refptr<AudioDeviceModule> adm(
>-      AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio));
>-  if (!adm.get()) {
>-    fprintf(stderr, "Failed to create Audio Device Module.\n");
>-    return 1;
>-  }
>-  if (adm->Init() != 0) {
>-    fprintf(stderr, "Failed to initialize Audio Device Module.\n");
>-    return 1;
>-  }
>-  if (adm->SetRecordingDevice(DEFAULT_INPUT_DEVICE) != 0) {
>-    fprintf(stderr, "Failed to set the default input device.\n");
>-    return 1;
>-  }
>-  if (adm->InitMicrophone() != 0) {
>-    fprintf(stderr, "Failed to to initialize the microphone.\n");
>-    return 1;
>-  }
>-
>-  // Set mic volume to max.
>-  uint32_t max_vol = 0;
>-  if (adm->MaxMicrophoneVolume(&max_vol) != 0) {
>-    fprintf(stderr, "Failed to get max volume.\n");
>-    return 1;
>-  }
>-  if (adm->SetMicrophoneVolume(max_vol) != 0) {
>-    fprintf(stderr, "Failed to set mic volume.\n");
>-    return 1;
>-  }
>-
>-  return 0;
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/frame_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/frame_analyzer.cc
>index 89440ea9845a25ef15898811db67d301affbcaf1..fb5f8de227afbc22949d05ddef4e0aeaca049099 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/frame_analyzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/frame_analyzer.cc
>@@ -15,7 +15,10 @@
> #include <string>
> #include <vector>
> 
>+#include "rtc_base/strings/string_builder.h"
> #include "rtc_base/stringutils.h"
>+#include "rtc_tools/frame_analyzer/video_color_aligner.h"
>+#include "rtc_tools/frame_analyzer/video_geometry_aligner.h"
> #include "rtc_tools/frame_analyzer/video_quality_analysis.h"
> #include "rtc_tools/frame_analyzer/video_temporal_aligner.h"
> #include "rtc_tools/simple_command_line_parser.h"
>@@ -133,8 +136,35 @@ int main(int argc, char* argv[]) {
>   const std::vector<size_t> matching_indices =
>       webrtc::test::FindMatchingFrameIndices(reference_video, test_video);
> 
>-  results.frames =
>-      webrtc::test::RunAnalysis(reference_video, test_video, matching_indices);
>+  // Align the reference video both temporally and geometrically. I.e. align the
>+  // frames to match up in order to the test video, and align a crop region of
>+  // the reference video to match up to the test video.
>+  const rtc::scoped_refptr<webrtc::test::Video> aligned_reference_video =
>+      AdjustCropping(ReorderVideo(reference_video, matching_indices),
>+                     test_video);
>+
>+  // Calculate if there is any systematic color difference between the reference
>+  // and test video.
>+  const webrtc::test::ColorTransformationMatrix color_transformation =
>+      CalculateColorTransformationMatrix(aligned_reference_video, test_video);
>+
>+  char buf[256];
>+  rtc::SimpleStringBuilder string_builder(buf);
>+  for (int i = 0; i < 3; ++i) {
>+    string_builder << "\n";
>+    for (int j = 0; j < 4; ++j)
>+      string_builder.AppendFormat("%6.2f ", color_transformation[i][j]);
>+  }
>+  printf("Adjusting test video with color transformation: %s\n",
>+         string_builder.str());
>+
>+  // Adjust all frames in the test video with the calculated color
>+  // transformation.
>+  const rtc::scoped_refptr<webrtc::test::Video> color_adjusted_test_video =
>+      AdjustColors(color_transformation, test_video);
>+
>+  results.frames = webrtc::test::RunAnalysis(
>+      aligned_reference_video, color_adjusted_test_video, matching_indices);
> 
>   const std::vector<webrtc::test::Cluster> clusters =
>       webrtc::test::CalculateFrameClusters(matching_indices);
>@@ -151,20 +181,17 @@ int main(int argc, char* argv[]) {
>   if (!chartjson_result_file.empty()) {
>     webrtc::test::WritePerfResults(chartjson_result_file);
>   }
>-  rtc::scoped_refptr<webrtc::test::Video> reordered_video =
>-      webrtc::test::GenerateAlignedReferenceVideo(reference_video,
>-                                                  matching_indices);
>   std::string aligned_output_file = parser.GetFlag("aligned_output_file");
>   if (!aligned_output_file.empty()) {
>-    webrtc::test::WriteVideoToFile(reordered_video, aligned_output_file,
>+    webrtc::test::WriteVideoToFile(aligned_reference_video, aligned_output_file,
>                                    /*fps=*/30);
>   }
>   std::string yuv_directory = parser.GetFlag("yuv_directory");
>   if (!yuv_directory.empty()) {
>-    webrtc::test::WriteVideoToFile(reordered_video,
>+    webrtc::test::WriteVideoToFile(aligned_reference_video,
>                                    JoinFilename(yuv_directory, "ref.yuv"),
>                                    /*fps=*/30);
>-    webrtc::test::WriteVideoToFile(test_video,
>+    webrtc::test::WriteVideoToFile(color_adjusted_test_video,
>                                    JoinFilename(yuv_directory, "test.yuv"),
>                                    /*fps=*/30);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/linear_least_squares.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/linear_least_squares.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..706d212d174abe9d76ce20039a0ba83e3fe842dd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/linear_least_squares.cc
>@@ -0,0 +1,200 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_tools/frame_analyzer/linear_least_squares.h"
>+
>+#include <numeric>
>+#include <utility>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+template <class T>
>+using Matrix = std::valarray<std::valarray<T>>;
>+
>+namespace {
>+
>+template <typename R, typename T>
>+R DotProduct(const std::valarray<T>& a, const std::valarray<T>& b) {
>+  RTC_CHECK_EQ(a.size(), b.size());
>+  return std::inner_product(std::begin(a), std::end(a), std::begin(b), R(0));
>+}
>+
>+// Calculates a^T * b.
>+template <typename R, typename T>
>+Matrix<R> MatrixMultiply(const Matrix<T>& a, const Matrix<T>& b) {
>+  Matrix<R> result(std::valarray<R>(a.size()), b.size());
>+  for (size_t i = 0; i < a.size(); ++i) {
>+    for (size_t j = 0; j < b.size(); ++j)
>+      result[j][i] = DotProduct<R>(a[i], b[j]);
>+  }
>+
>+  return result;
>+}
>+
>+template <typename T>
>+Matrix<T> Transpose(const Matrix<T>& matrix) {
>+  if (matrix.size() == 0)
>+    return Matrix<T>();
>+  const size_t rows = matrix.size();
>+  const size_t columns = matrix[0].size();
>+  Matrix<T> result(std::valarray<T>(rows), columns);
>+
>+  for (size_t i = 0; i < rows; ++i) {
>+    for (size_t j = 0; j < columns; ++j)
>+      result[j][i] = matrix[i][j];
>+  }
>+
>+  return result;
>+}
>+
>+// Convert valarray from type T to type R.
>+template <typename R, typename T>
>+std::valarray<R> ConvertTo(const std::valarray<T>& v) {
>+  std::valarray<R> result(v.size());
>+  for (size_t i = 0; i < v.size(); ++i)
>+    result[i] = static_cast<R>(v[i]);
>+  return result;
>+}
>+
>+// Convert valarray Matrix from type T to type R.
>+template <typename R, typename T>
>+Matrix<R> ConvertTo(const Matrix<T>& mat) {
>+  Matrix<R> result(mat.size());
>+  for (size_t i = 0; i < mat.size(); ++i)
>+    result[i] = ConvertTo<R>(mat[i]);
>+  return result;
>+}
>+
>+// Convert from valarray Matrix back to the more conventional std::vector.
>+template <typename T>
>+std::vector<std::vector<T>> ToVectorMatrix(const Matrix<T>& m) {
>+  std::vector<std::vector<T>> result;
>+  for (const std::valarray<T>& v : m)
>+    result.emplace_back(std::begin(v), std::end(v));
>+  return result;
>+}
>+
>+// Create a valarray Matrix from a conventional std::vector.
>+template <typename T>
>+Matrix<T> FromVectorMatrix(const std::vector<std::vector<T>>& mat) {
>+  Matrix<T> result(mat.size());
>+  for (size_t i = 0; i < mat.size(); ++i)
>+    result[i] = std::valarray<T>(mat[i].data(), mat[i].size());
>+  return result;
>+}
>+
>+// Returns |matrix_to_invert|^-1 * |right_hand_matrix|. |matrix_to_invert| must
>+// have square size.
>+Matrix<double> GaussianElimination(Matrix<double> matrix_to_invert,
>+                                   Matrix<double> right_hand_matrix) {
>+  // |n| is the width/height of |matrix_to_invert|.
>+  const size_t n = matrix_to_invert.size();
>+  // Make sure |matrix_to_invert| has square size.
>+  for (const std::valarray<double>& column : matrix_to_invert)
>+    RTC_CHECK_EQ(n, column.size());
>+  // Make sure |right_hand_matrix| has correct size.
>+  for (const std::valarray<double>& column : right_hand_matrix)
>+    RTC_CHECK_EQ(n, column.size());
>+
>+  // Transpose the matrices before and after so that we can perform Gaussian
>+  // elimination on the columns instead of the rows, since that is easier with
>+  // our representation.
>+  matrix_to_invert = Transpose(matrix_to_invert);
>+  right_hand_matrix = Transpose(right_hand_matrix);
>+
>+  // Loop over the diagonal of |matrix_to_invert| and perform column reduction.
>+  // Column reduction is a sequence of elementary column operations that is
>+  // performed on both |matrix_to_invert| and |right_hand_matrix| until
>+  // |matrix_to_invert| has been transformed to the identity matrix.
>+  for (size_t diagonal_index = 0; diagonal_index < n; ++diagonal_index) {
>+    // Make sure the diagonal element has the highest absolute value by
>+    // swapping columns if necessary.
>+    for (size_t column = diagonal_index + 1; column < n; ++column) {
>+      if (std::abs(matrix_to_invert[column][diagonal_index]) >
>+          std::abs(matrix_to_invert[diagonal_index][diagonal_index])) {
>+        std::swap(matrix_to_invert[column], matrix_to_invert[diagonal_index]);
>+        std::swap(right_hand_matrix[column], right_hand_matrix[diagonal_index]);
>+      }
>+    }
>+
>+    // Reduce the diagonal element to be 1, by dividing the column with that
>+    // value. If the diagonal element is 0, it means the system of equations has
>+    // many solutions, and in that case we will return an arbitrary solution.
>+    if (matrix_to_invert[diagonal_index][diagonal_index] == 0.0) {
>+      RTC_LOG(LS_WARNING) << "Matrix is not invertible, ignoring.";
>+      continue;
>+    }
>+    const double diagonal_element =
>+        matrix_to_invert[diagonal_index][diagonal_index];
>+    matrix_to_invert[diagonal_index] /= diagonal_element;
>+    right_hand_matrix[diagonal_index] /= diagonal_element;
>+
>+    // Eliminate the other entries in row |diagonal_index| by making them zero.
>+    for (size_t column = 0; column < n; ++column) {
>+      if (column == diagonal_index)
>+        continue;
>+      const double row_element = matrix_to_invert[column][diagonal_index];
>+      matrix_to_invert[column] -=
>+          row_element * matrix_to_invert[diagonal_index];
>+      right_hand_matrix[column] -=
>+          row_element * right_hand_matrix[diagonal_index];
>+    }
>+  }
>+
>+  // Transpose the result before returning it, explained in comment above.
>+  return Transpose(right_hand_matrix);
>+}
>+
>+}  // namespace
>+
>+IncrementalLinearLeastSquares::IncrementalLinearLeastSquares() = default;
>+IncrementalLinearLeastSquares::~IncrementalLinearLeastSquares() = default;
>+
>+void IncrementalLinearLeastSquares::AddObservations(
>+    const std::vector<std::vector<uint8_t>>& x,
>+    const std::vector<std::vector<uint8_t>>& y) {
>+  if (x.empty() || y.empty())
>+    return;
>+  // Make sure all columns are the same size.
>+  const size_t n = x[0].size();
>+  for (const std::vector<uint8_t>& column : x)
>+    RTC_CHECK_EQ(n, column.size());
>+  for (const std::vector<uint8_t>& column : y)
>+    RTC_CHECK_EQ(n, column.size());
>+
>+  // We will multiply the uint8_t values together, so we need to expand to a
>+  // type that can safely store those values, i.e. uint16_t.
>+  const Matrix<uint16_t> unpacked_x = ConvertTo<uint16_t>(FromVectorMatrix(x));
>+  const Matrix<uint16_t> unpacked_y = ConvertTo<uint16_t>(FromVectorMatrix(y));
>+
>+  const Matrix<uint64_t> xx = MatrixMultiply<uint64_t>(unpacked_x, unpacked_x);
>+  const Matrix<uint64_t> xy = MatrixMultiply<uint64_t>(unpacked_x, unpacked_y);
>+  if (sum_xx && sum_xy) {
>+    *sum_xx += xx;
>+    *sum_xy += xy;
>+  } else {
>+    sum_xx = xx;
>+    sum_xy = xy;
>+  }
>+}
>+
>+std::vector<std::vector<double>>
>+IncrementalLinearLeastSquares::GetBestSolution() const {
>+  RTC_CHECK(sum_xx && sum_xy) << "No observations have been added";
>+  return ToVectorMatrix(GaussianElimination(ConvertTo<double>(*sum_xx),
>+                                            ConvertTo<double>(*sum_xy)));
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/linear_least_squares.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/linear_least_squares.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..1b07dc1bde640afb792f1dc7625b3e63d8b582b3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/linear_least_squares.h
>@@ -0,0 +1,53 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_TOOLS_FRAME_ANALYZER_LINEAR_LEAST_SQUARES_H_
>+#define RTC_TOOLS_FRAME_ANALYZER_LINEAR_LEAST_SQUARES_H_
>+
>+#include <valarray>
>+#include <vector>
>+#include "absl/types/optional.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// This class is used for finding a matrix b that roughly solves the equation:
>+// y = x * b. This is generally impossible to do exactly, so the problem is
>+// rephrased as finding the matrix b that minimizes the difference:
>+// |y - x * b|^2. Calling multiple AddObservations() is equivalent to
>+// concatenating the observation vectors and calling AddObservations() once. The
>+// reason for doing it incrementally is that we can't store the raw YUV values
>+// for a whole video file in memory at once. This class has a constant memory
>+// footprint, regardless how may times AddObservations() is called.
>+class IncrementalLinearLeastSquares {
>+ public:
>+  IncrementalLinearLeastSquares();
>+  ~IncrementalLinearLeastSquares();
>+
>+  // Add a number of observations. The subvectors of x and y must have the same
>+  // length.
>+  void AddObservations(const std::vector<std::vector<uint8_t>>& x,
>+                       const std::vector<std::vector<uint8_t>>& y);
>+
>+  // Calculate and return the best linear solution, given the observations so
>+  // far.
>+  std::vector<std::vector<double>> GetBestSolution() const;
>+
>+ private:
>+  // Running sum of x^T * x.
>+  absl::optional<std::valarray<std::valarray<uint64_t>>> sum_xx;
>+  // Running sum of x^T * y.
>+  absl::optional<std::valarray<std::valarray<uint64_t>>> sum_xy;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // RTC_TOOLS_FRAME_ANALYZER_LINEAR_LEAST_SQUARES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/linear_least_squares_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/linear_least_squares_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..b074aacce856a8ff639696b8860624a9ed99c7cf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/linear_least_squares_unittest.cc
>@@ -0,0 +1,91 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_tools/frame_analyzer/linear_least_squares.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+TEST(LinearLeastSquares, ScalarIdentityOneObservation) {
>+  IncrementalLinearLeastSquares lls;
>+  lls.AddObservations({{1}}, {{1}});
>+  EXPECT_EQ(std::vector<std::vector<double>>({{1.0}}), lls.GetBestSolution());
>+}
>+
>+TEST(LinearLeastSquares, ScalarIdentityTwoObservationsOneCall) {
>+  IncrementalLinearLeastSquares lls;
>+  lls.AddObservations({{1, 2}}, {{1, 2}});
>+  EXPECT_EQ(std::vector<std::vector<double>>({{1.0}}), lls.GetBestSolution());
>+}
>+
>+TEST(LinearLeastSquares, ScalarIdentityTwoObservationsTwoCalls) {
>+  IncrementalLinearLeastSquares lls;
>+  lls.AddObservations({{1}}, {{1}});
>+  lls.AddObservations({{2}}, {{2}});
>+  EXPECT_EQ(std::vector<std::vector<double>>({{1.0}}), lls.GetBestSolution());
>+}
>+
>+TEST(LinearLeastSquares, MatrixIdentityOneObservation) {
>+  IncrementalLinearLeastSquares lls;
>+  lls.AddObservations({{1, 2}, {3, 4}}, {{1, 2}, {3, 4}});
>+  EXPECT_EQ(std::vector<std::vector<double>>({{1.0, 0.0}, {0.0, 1.0}}),
>+            lls.GetBestSolution());
>+}
>+
>+TEST(LinearLeastSquares, MatrixManyObservations) {
>+  IncrementalLinearLeastSquares lls;
>+  // Test that we can find the solution of the overspecified equation system:
>+  // [1, 2] [1, 3] = [5,  11]
>+  // [3, 4] [2, 4]   [11, 25]
>+  // [5, 6]          [17, 39]
>+  lls.AddObservations({{1}, {2}}, {{5}, {11}});
>+  lls.AddObservations({{3}, {4}}, {{11}, {25}});
>+  lls.AddObservations({{5}, {6}}, {{17}, {39}});
>+
>+  const std::vector<std::vector<double>> result = lls.GetBestSolution();
>+  // We allow some numerical flexibility here.
>+  EXPECT_DOUBLE_EQ(1.0, result[0][0]);
>+  EXPECT_DOUBLE_EQ(2.0, result[0][1]);
>+  EXPECT_DOUBLE_EQ(3.0, result[1][0]);
>+  EXPECT_DOUBLE_EQ(4.0, result[1][1]);
>+}
>+
>+TEST(LinearLeastSquares, MatrixVectorOneObservation) {
>+  IncrementalLinearLeastSquares lls;
>+  // Test that we can find the solution of the overspecified equation system:
>+  // [1, 2] [1] = [5]
>+  // [3, 4] [2]   [11]
>+  // [5, 6]       [17]
>+  lls.AddObservations({{1, 3, 5}, {2, 4, 6}}, {{5, 11, 17}});
>+
>+  const std::vector<std::vector<double>> result = lls.GetBestSolution();
>+  // We allow some numerical flexibility here.
>+  EXPECT_DOUBLE_EQ(1.0, result[0][0]);
>+  EXPECT_DOUBLE_EQ(2.0, result[0][1]);
>+}
>+
>+TEST(LinearLeastSquares, LinearLeastSquaresNonPerfectSolution) {
>+  IncrementalLinearLeastSquares lls;
>+  // Test that we can find the non-perfect solution of the overspecified
>+  // equation system:
>+  // [1] [20] = [21]
>+  // [2]        [39]
>+  // [3]        [60]
>+  // [2]        [41]
>+  // [1]        [19]
>+  lls.AddObservations({{1, 2, 3, 2, 1}}, {{21, 39, 60, 41, 19}});
>+
>+  EXPECT_DOUBLE_EQ(20.0, lls.GetBestSolution()[0][0]);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_color_aligner.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_color_aligner.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..7afb1e46950f9921f900aaa6a1d189e0a2a5d22b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_color_aligner.cc
>@@ -0,0 +1,239 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_tools/frame_analyzer/video_color_aligner.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+#include <deque>
>+#include <limits>
>+#include <vector>
>+
>+#include "api/array_view.h"
>+#include "api/video/i420_buffer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "rtc_tools/frame_analyzer/linear_least_squares.h"
>+#include "rtc_tools/frame_analyzer/video_quality_analysis.h"
>+#include "third_party/libyuv/include/libyuv/compare.h"
>+#include "third_party/libyuv/include/libyuv/planar_functions.h"
>+#include "third_party/libyuv/include/libyuv/scale.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+
>+// Helper function for AdjustColors(). This functions calculates a single output
>+// row for y with the given color coefficients. The u/v channels are assumed to
>+// be subsampled by a factor of 2, which is the case of I420.
>+void CalculateYChannel(rtc::ArrayView<const uint8_t> y_data,
>+                       rtc::ArrayView<const uint8_t> u_data,
>+                       rtc::ArrayView<const uint8_t> v_data,
>+                       const std::array<float, 4>& coeff,
>+                       rtc::ArrayView<uint8_t> output) {
>+  RTC_CHECK_EQ(y_data.size(), output.size());
>+  // Each u/v element represents two y elements. Make sure we have enough to
>+  // cover the Y values.
>+  RTC_CHECK_GE(u_data.size() * 2, y_data.size());
>+  RTC_CHECK_GE(v_data.size() * 2, y_data.size());
>+
>+  // Do two pixels at a time since u/v are subsampled.
>+  for (size_t i = 0; i * 2 < y_data.size() - 1; ++i) {
>+    const float uv_contribution =
>+        coeff[1] * u_data[i] + coeff[2] * v_data[i] + coeff[3];
>+
>+    const float val0 = coeff[0] * y_data[i * 2 + 0] + uv_contribution;
>+    const float val1 = coeff[0] * y_data[i * 2 + 1] + uv_contribution;
>+
>+    // Clamp result to a byte.
>+    output[i * 2 + 0] = static_cast<uint8_t>(
>+        std::round(std::max(0.0f, std::min(val0, 255.0f))));
>+    output[i * 2 + 1] = static_cast<uint8_t>(
>+        std::round(std::max(0.0f, std::min(val1, 255.0f))));
>+  }
>+
>+  // Handle the last pixel for odd widths.
>+  if (y_data.size() % 2 == 1) {
>+    const float val = coeff[0] * y_data[y_data.size() - 1] +
>+                      coeff[1] * u_data[(y_data.size() - 1) / 2] +
>+                      coeff[2] * v_data[(y_data.size() - 1) / 2] + coeff[3];
>+    output[y_data.size() - 1] =
>+        static_cast<uint8_t>(std::round(std::max(0.0f, std::min(val, 255.0f))));
>+  }
>+}
>+
>+// Helper function for AdjustColors(). This functions calculates a single output
>+// row for either u or v, with the given color coefficients. Y, U, and V are
>+// assumed to be the same size, i.e. no subsampling.
>+void CalculateUVChannel(rtc::ArrayView<const uint8_t> y_data,
>+                        rtc::ArrayView<const uint8_t> u_data,
>+                        rtc::ArrayView<const uint8_t> v_data,
>+                        const std::array<float, 4>& coeff,
>+                        rtc::ArrayView<uint8_t> output) {
>+  RTC_CHECK_EQ(y_data.size(), u_data.size());
>+  RTC_CHECK_EQ(y_data.size(), v_data.size());
>+  RTC_CHECK_EQ(y_data.size(), output.size());
>+
>+  for (size_t x = 0; x < y_data.size(); ++x) {
>+    const float val = coeff[0] * y_data[x] + coeff[1] * u_data[x] +
>+                      coeff[2] * v_data[x] + coeff[3];
>+    // Clamp result to a byte.
>+    output[x] =
>+        static_cast<uint8_t>(std::round(std::max(0.0f, std::min(val, 255.0f))));
>+  }
>+}
>+
>+// Convert a frame to four vectors consisting of [y, u, v, 1].
>+std::vector<std::vector<uint8_t>> FlattenYuvData(
>+    const rtc::scoped_refptr<I420BufferInterface>& frame) {
>+  std::vector<std::vector<uint8_t>> result(
>+      4, std::vector<uint8_t>(frame->ChromaWidth() * frame->ChromaHeight()));
>+
>+  // Downscale the Y plane so that all YUV planes are the same size.
>+  libyuv::ScalePlane(frame->DataY(), frame->StrideY(), frame->width(),
>+                     frame->height(), result[0].data(), frame->ChromaWidth(),
>+                     frame->ChromaWidth(), frame->ChromaHeight(),
>+                     libyuv::kFilterBox);
>+
>+  libyuv::CopyPlane(frame->DataU(), frame->StrideU(), result[1].data(),
>+                    frame->ChromaWidth(), frame->ChromaWidth(),
>+                    frame->ChromaHeight());
>+
>+  libyuv::CopyPlane(frame->DataV(), frame->StrideV(), result[2].data(),
>+                    frame->ChromaWidth(), frame->ChromaWidth(),
>+                    frame->ChromaHeight());
>+
>+  std::fill(result[3].begin(), result[3].end(), 1u);
>+
>+  return result;
>+}
>+
>+ColorTransformationMatrix VectorToColorMatrix(
>+    const std::vector<std::vector<double>>& v) {
>+  ColorTransformationMatrix color_transformation;
>+  for (int i = 0; i < 3; ++i) {
>+    for (int j = 0; j < 4; ++j)
>+      color_transformation[i][j] = v[i][j];
>+  }
>+  return color_transformation;
>+}
>+
>+}  // namespace
>+
>+ColorTransformationMatrix CalculateColorTransformationMatrix(
>+    const rtc::scoped_refptr<I420BufferInterface>& reference_frame,
>+    const rtc::scoped_refptr<I420BufferInterface>& test_frame) {
>+  IncrementalLinearLeastSquares incremental_lls;
>+  incremental_lls.AddObservations(FlattenYuvData(test_frame),
>+                                  FlattenYuvData(reference_frame));
>+  return VectorToColorMatrix(incremental_lls.GetBestSolution());
>+}
>+
>+ColorTransformationMatrix CalculateColorTransformationMatrix(
>+    const rtc::scoped_refptr<Video>& reference_video,
>+    const rtc::scoped_refptr<Video>& test_video) {
>+  RTC_CHECK_GE(reference_video->number_of_frames(),
>+               test_video->number_of_frames());
>+
>+  IncrementalLinearLeastSquares incremental_lls;
>+  for (size_t i = 0; i < test_video->number_of_frames(); ++i) {
>+    incremental_lls.AddObservations(
>+        FlattenYuvData(test_video->GetFrame(i)),
>+        FlattenYuvData(reference_video->GetFrame(i)));
>+  }
>+
>+  return VectorToColorMatrix(incremental_lls.GetBestSolution());
>+}
>+
>+rtc::scoped_refptr<Video> AdjustColors(
>+    const ColorTransformationMatrix& color_transformation,
>+    const rtc::scoped_refptr<Video>& video) {
>+  class ColorAdjustedVideo : public rtc::RefCountedObject<Video> {
>+   public:
>+    ColorAdjustedVideo(const ColorTransformationMatrix& color_transformation,
>+                       const rtc::scoped_refptr<Video>& video)
>+        : color_transformation_(color_transformation), video_(video) {}
>+
>+    int width() const override { return video_->width(); }
>+    int height() const override { return video_->height(); }
>+    size_t number_of_frames() const override {
>+      return video_->number_of_frames();
>+    }
>+
>+    rtc::scoped_refptr<I420BufferInterface> GetFrame(
>+        size_t index) const override {
>+      return AdjustColors(color_transformation_, video_->GetFrame(index));
>+    }
>+
>+   private:
>+    const ColorTransformationMatrix color_transformation_;
>+    const rtc::scoped_refptr<Video> video_;
>+  };
>+
>+  return new ColorAdjustedVideo(color_transformation, video);
>+}
>+
>+rtc::scoped_refptr<I420BufferInterface> AdjustColors(
>+    const ColorTransformationMatrix& color_matrix,
>+    const rtc::scoped_refptr<I420BufferInterface>& frame) {
>+  // Allocate I420 buffer that will hold the color adjusted frame.
>+  rtc::scoped_refptr<I420Buffer> adjusted_frame =
>+      I420Buffer::Create(frame->width(), frame->height());
>+
>+  // Create a downscaled Y plane with the same size as the U/V planes to
>+  // simplify converting the U/V planes.
>+  std::vector<uint8_t> downscaled_y_plane(frame->ChromaWidth() *
>+                                          frame->ChromaHeight());
>+  libyuv::ScalePlane(frame->DataY(), frame->StrideY(), frame->width(),
>+                     frame->height(), downscaled_y_plane.data(),
>+                     frame->ChromaWidth(), frame->ChromaWidth(),
>+                     frame->ChromaHeight(), libyuv::kFilterBox);
>+
>+  // Fill in the adjusted data row by row.
>+  for (int y = 0; y < frame->height(); ++y) {
>+    const int half_y = y / 2;
>+    rtc::ArrayView<const uint8_t> y_row(frame->DataY() + frame->StrideY() * y,
>+                                        frame->width());
>+    rtc::ArrayView<const uint8_t> u_row(
>+        frame->DataU() + frame->StrideU() * half_y, frame->ChromaWidth());
>+    rtc::ArrayView<const uint8_t> v_row(
>+        frame->DataV() + frame->StrideV() * half_y, frame->ChromaWidth());
>+    rtc::ArrayView<uint8_t> output_y_row(
>+        adjusted_frame->MutableDataY() + adjusted_frame->StrideY() * y,
>+        frame->width());
>+
>+    CalculateYChannel(y_row, u_row, v_row, color_matrix[0], output_y_row);
>+
>+    // Chroma channels only exist every second row for I420.
>+    if (y % 2 == 0) {
>+      rtc::ArrayView<const uint8_t> downscaled_y_row(
>+          downscaled_y_plane.data() + frame->ChromaWidth() * half_y,
>+          frame->ChromaWidth());
>+      rtc::ArrayView<uint8_t> output_u_row(
>+          adjusted_frame->MutableDataU() + adjusted_frame->StrideU() * half_y,
>+          frame->ChromaWidth());
>+      rtc::ArrayView<uint8_t> output_v_row(
>+          adjusted_frame->MutableDataV() + adjusted_frame->StrideV() * half_y,
>+          frame->ChromaWidth());
>+
>+      CalculateUVChannel(downscaled_y_row, u_row, v_row, color_matrix[1],
>+                         output_u_row);
>+      CalculateUVChannel(downscaled_y_row, u_row, v_row, color_matrix[2],
>+                         output_v_row);
>+    }
>+  }
>+
>+  return adjusted_frame;
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_color_aligner.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_color_aligner.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..997cbf65829f6826f6e21cb8291c336bfb7a4dae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_color_aligner.h
>@@ -0,0 +1,49 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_TOOLS_FRAME_ANALYZER_VIDEO_COLOR_ALIGNER_H_
>+#define RTC_TOOLS_FRAME_ANALYZER_VIDEO_COLOR_ALIGNER_H_
>+
>+#include <array>
>+
>+#include "rtc_tools/video_file_reader.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// Represents a linear color transformation from [y, u, v] to [y', u', v']
>+// through the equation: [y', u', v'] = [y, u, v, 1] * matrix.
>+using ColorTransformationMatrix = std::array<std::array<float, 4>, 3>;
>+
>+// Calculate the optimal color transformation that should be applied to the test
>+// video to match as closely as possible to the reference video.
>+ColorTransformationMatrix CalculateColorTransformationMatrix(
>+    const rtc::scoped_refptr<Video>& reference_video,
>+    const rtc::scoped_refptr<Video>& test_video);
>+
>+// Calculate color transformation for a single I420 frame.
>+ColorTransformationMatrix CalculateColorTransformationMatrix(
>+    const rtc::scoped_refptr<I420BufferInterface>& reference_frame,
>+    const rtc::scoped_refptr<I420BufferInterface>& test_frame);
>+
>+// Apply a color transformation to a video.
>+rtc::scoped_refptr<Video> AdjustColors(
>+    const ColorTransformationMatrix& color_matrix,
>+    const rtc::scoped_refptr<Video>& video);
>+
>+// Apply a color transformation to a single I420 frame.
>+rtc::scoped_refptr<I420BufferInterface> AdjustColors(
>+    const ColorTransformationMatrix& color_matrix,
>+    const rtc::scoped_refptr<I420BufferInterface>& frame);
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // RTC_TOOLS_FRAME_ANALYZER_VIDEO_COLOR_ALIGNER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_color_aligner_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_color_aligner_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..520d6f459e7052b5fadb0a4aabf9dfa465acc90e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_color_aligner_unittest.cc
>@@ -0,0 +1,174 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_tools/frame_analyzer/video_color_aligner.h"
>+
>+#include "rtc_tools/frame_analyzer/video_quality_analysis.h"
>+#include "rtc_tools/video_file_reader.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+#include "third_party/libyuv/include/libyuv/scale.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+
>+const ColorTransformationMatrix kIdentityColorMatrix = {
>+    {{1, 0, 0, 0}, {0, 1, 0, 0}, {0, 0, 1, 0}}};
>+
>+void ExpectNear(const ColorTransformationMatrix& expected,
>+                const ColorTransformationMatrix& actual) {
>+  // The scaling factor on y/u/v should be pretty precise.
>+  for (int i = 0; i < 3; ++i) {
>+    for (int j = 0; j < 3; ++j)
>+      EXPECT_NEAR(expected[i][j], actual[i][j], /* abs_error= */ 1.0e-3)
>+          << "at element i: " << i << ", j: " << j;
>+  }
>+  // The offset can be less precise since the range is [0, 255].
>+  for (int i = 0; i < 3; ++i)
>+    EXPECT_NEAR(expected[i][3], actual[i][3], /* abs_error= */ 0.1)
>+        << "at element i: " << i;
>+}
>+
>+}  // namespace
>+
>+class VideoColorAlignerTest : public ::testing::Test {
>+ protected:
>+  void SetUp() {
>+    reference_video_ =
>+        OpenYuvFile(ResourcePath("foreman_128x96", "yuv"), 128, 96);
>+    ASSERT_TRUE(reference_video_);
>+  }
>+
>+  rtc::scoped_refptr<Video> reference_video_;
>+};
>+
>+TEST_F(VideoColorAlignerTest, AdjustColorsFrameIdentity) {
>+  const rtc::scoped_refptr<I420BufferInterface> test_frame =
>+      reference_video_->GetFrame(0);
>+
>+  // Assume perfect match, i.e. ssim == 1.
>+  EXPECT_EQ(1.0,
>+            Ssim(test_frame, AdjustColors(kIdentityColorMatrix, test_frame)));
>+}
>+
>+TEST_F(VideoColorAlignerTest, AdjustColorsFrame1x1) {
>+  const ColorTransformationMatrix color_matrix = {
>+      {{1, 2, 3, 4}, {5, 6, 7, 8}, {9, 10, 11, 12}}};
>+
>+  const uint8_t data_y[] = {2};
>+  const uint8_t data_u[] = {6};
>+  const uint8_t data_v[] = {7};
>+  const rtc::scoped_refptr<I420BufferInterface> i420_buffer = I420Buffer::Copy(
>+      /* width= */ 1, /* height= */ 1, data_y, /* stride_y= */ 1, data_u,
>+      /* stride_u= */ 1, data_v, /* stride_v= */ 1);
>+
>+  const rtc::scoped_refptr<I420BufferInterface> adjusted_buffer =
>+      AdjustColors(color_matrix, i420_buffer);
>+
>+  EXPECT_EQ(2 * 1 + 6 * 2 + 7 * 3 + 4, adjusted_buffer->DataY()[0]);
>+  EXPECT_EQ(2 * 5 + 6 * 6 + 7 * 7 + 8, adjusted_buffer->DataU()[0]);
>+  EXPECT_EQ(2 * 9 + 6 * 10 + 7 * 11 + 12, adjusted_buffer->DataV()[0]);
>+}
>+
>+TEST_F(VideoColorAlignerTest, AdjustColorsFrame1x1Negative) {
>+  const ColorTransformationMatrix color_matrix = {
>+      {{-1, 0, 0, 255}, {0, -1, 0, 255}, {0, 0, -1, 255}}};
>+
>+  const uint8_t data_y[] = {2};
>+  const uint8_t data_u[] = {6};
>+  const uint8_t data_v[] = {7};
>+  const rtc::scoped_refptr<I420BufferInterface> i420_buffer = I420Buffer::Copy(
>+      /* width= */ 1, /* height= */ 1, data_y, /* stride_y= */ 1, data_u,
>+      /* stride_u= */ 1, data_v, /* stride_v= */ 1);
>+
>+  const rtc::scoped_refptr<I420BufferInterface> adjusted_buffer =
>+      AdjustColors(color_matrix, i420_buffer);
>+
>+  EXPECT_EQ(255 - 2, adjusted_buffer->DataY()[0]);
>+  EXPECT_EQ(255 - 6, adjusted_buffer->DataU()[0]);
>+  EXPECT_EQ(255 - 7, adjusted_buffer->DataV()[0]);
>+}
>+
>+TEST_F(VideoColorAlignerTest, AdjustColorsFrame2x2) {
>+  const ColorTransformationMatrix color_matrix = {
>+      {{1, 2, 3, 4}, {5, 6, 7, 8}, {9, 10, 11, 12}}};
>+
>+  const uint8_t data_y[] = {0, 1, 3, 4};
>+  const uint8_t data_u[] = {6};
>+  const uint8_t data_v[] = {7};
>+  const rtc::scoped_refptr<I420BufferInterface> i420_buffer = I420Buffer::Copy(
>+      /* width= */ 2, /* height= */ 2, data_y, /* stride_y= */ 2, data_u,
>+      /* stride_u= */ 1, data_v, /* stride_v= */ 1);
>+
>+  const rtc::scoped_refptr<I420BufferInterface> adjusted_buffer =
>+      AdjustColors(color_matrix, i420_buffer);
>+
>+  EXPECT_EQ(0 * 1 + 6 * 2 + 7 * 3 + 4, adjusted_buffer->DataY()[0]);
>+  EXPECT_EQ(1 * 1 + 6 * 2 + 7 * 3 + 4, adjusted_buffer->DataY()[1]);
>+  EXPECT_EQ(3 * 1 + 6 * 2 + 7 * 3 + 4, adjusted_buffer->DataY()[2]);
>+  EXPECT_EQ(4 * 1 + 6 * 2 + 7 * 3 + 4, adjusted_buffer->DataY()[3]);
>+
>+  EXPECT_EQ(2 * 5 + 6 * 6 + 7 * 7 + 8, adjusted_buffer->DataU()[0]);
>+  EXPECT_EQ(2 * 9 + 6 * 10 + 7 * 11 + 12, adjusted_buffer->DataV()[0]);
>+}
>+
>+TEST_F(VideoColorAlignerTest, CalculateColorTransformationMatrixIdentity) {
>+  EXPECT_EQ(kIdentityColorMatrix, CalculateColorTransformationMatrix(
>+                                      reference_video_, reference_video_));
>+}
>+
>+TEST_F(VideoColorAlignerTest, CalculateColorTransformationMatrixOffset) {
>+  const uint8_t small_data_y[] = {0, 1, 2,  3,  4,  5,  6,  7,
>+                                  8, 9, 10, 11, 12, 13, 14, 15};
>+  const uint8_t small_data_u[] = {15, 13, 17, 29};
>+  const uint8_t small_data_v[] = {3, 200, 170, 29};
>+  const rtc::scoped_refptr<I420BufferInterface> small_i420_buffer =
>+      I420Buffer::Copy(
>+          /* width= */ 4, /* height= */ 4, small_data_y, /* stride_y= */ 4,
>+          small_data_u, /* stride_u= */ 2, small_data_v, /* stride_v= */ 2);
>+
>+  uint8_t big_data_y[16];
>+  uint8_t big_data_u[4];
>+  uint8_t big_data_v[4];
>+  // Create another I420 frame where all values are 10 bigger.
>+  for (int i = 0; i < 16; ++i)
>+    big_data_y[i] = small_data_y[i] + 10;
>+  for (int i = 0; i < 4; ++i)
>+    big_data_u[i] = small_data_u[i] + 10;
>+  for (int i = 0; i < 4; ++i)
>+    big_data_v[i] = small_data_v[i] + 10;
>+
>+  const rtc::scoped_refptr<I420BufferInterface> big_i420_buffer =
>+      I420Buffer::Copy(
>+          /* width= */ 4, /* height= */ 4, big_data_y, /* stride_y= */ 4,
>+          big_data_u, /* stride_u= */ 2, big_data_v, /* stride_v= */ 2);
>+
>+  const ColorTransformationMatrix color_matrix =
>+      CalculateColorTransformationMatrix(big_i420_buffer, small_i420_buffer);
>+
>+  ExpectNear({{{1, 0, 0, 10}, {0, 1, 0, 10}, {0, 0, 1, 10}}}, color_matrix);
>+}
>+
>+TEST_F(VideoColorAlignerTest, CalculateColorTransformationMatrix) {
>+  // Arbitrary color transformation matrix.
>+  const ColorTransformationMatrix org_color_matrix = {
>+      {{0.8, 0.05, 0.04, -4}, {-0.2, 0.7, 0.1, 10}, {0.1, 0.2, 0.4, 20}}};
>+
>+  const ColorTransformationMatrix result_color_matrix =
>+      CalculateColorTransformationMatrix(
>+          AdjustColors(org_color_matrix, reference_video_), reference_video_);
>+
>+  ExpectNear(org_color_matrix, result_color_matrix);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_geometry_aligner.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_geometry_aligner.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..d3cbc051c9bf133c1fdf1e9cb7f52f506c06cc5f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_geometry_aligner.cc
>@@ -0,0 +1,178 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_tools/frame_analyzer/video_geometry_aligner.h"
>+
>+#include <map>
>+
>+#include "api/video/i420_buffer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "rtc_tools/frame_analyzer/video_quality_analysis.h"
>+#include "third_party/libyuv/include/libyuv/scale.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+
>+bool IsValidRegion(const CropRegion& region,
>+                   const rtc::scoped_refptr<I420BufferInterface>& frame) {
>+  return region.left >= 0 && region.right >= 0 && region.top >= 0 &&
>+         region.bottom >= 0 && region.left + region.right < frame->width() &&
>+         region.top + region.bottom < frame->height();
>+}
>+
>+}  // namespace
>+
>+rtc::scoped_refptr<I420BufferInterface> CropAndZoom(
>+    const CropRegion& crop_region,
>+    const rtc::scoped_refptr<I420BufferInterface>& frame) {
>+  RTC_CHECK(IsValidRegion(crop_region, frame));
>+
>+  const int uv_crop_left = crop_region.left / 2;
>+  const int uv_crop_top = crop_region.top / 2;
>+
>+  const int cropped_width =
>+      frame->width() - crop_region.left - crop_region.right;
>+  const int cropped_height =
>+      frame->height() - crop_region.top - crop_region.bottom;
>+
>+  // Crop by only adjusting pointers.
>+  const uint8_t* y_plane =
>+      frame->DataY() + frame->StrideY() * crop_region.top + crop_region.left;
>+  const uint8_t* u_plane =
>+      frame->DataU() + frame->StrideU() * uv_crop_top + uv_crop_left;
>+  const uint8_t* v_plane =
>+      frame->DataV() + frame->StrideV() * uv_crop_top + uv_crop_left;
>+
>+  // Stretch the cropped frame to the original size using libyuv.
>+  rtc::scoped_refptr<I420Buffer> adjusted_frame =
>+      I420Buffer::Create(frame->width(), frame->height());
>+  libyuv::I420Scale(y_plane, frame->StrideY(), u_plane, frame->StrideU(),
>+                    v_plane, frame->StrideV(), cropped_width, cropped_height,
>+                    adjusted_frame->MutableDataY(), adjusted_frame->StrideY(),
>+                    adjusted_frame->MutableDataU(), adjusted_frame->StrideU(),
>+                    adjusted_frame->MutableDataV(), adjusted_frame->StrideV(),
>+                    frame->width(), frame->height(), libyuv::kFilterBilinear);
>+
>+  return adjusted_frame;
>+}
>+
>+CropRegion CalculateCropRegion(
>+    const rtc::scoped_refptr<I420BufferInterface>& reference_frame,
>+    const rtc::scoped_refptr<I420BufferInterface>& test_frame) {
>+  RTC_CHECK_EQ(reference_frame->width(), test_frame->width());
>+  RTC_CHECK_EQ(reference_frame->height(), test_frame->height());
>+
>+  CropRegion best_region;
>+  double best_ssim = Ssim(reference_frame, test_frame);
>+
>+  typedef int CropRegion::*CropParameter;
>+  CropParameter crop_parameters[4] = {&CropRegion::left, &CropRegion::top,
>+                                      &CropRegion::right, &CropRegion::bottom};
>+
>+  while (true) {
>+    // Find the parameter in which direction SSIM improves the most.
>+    CropParameter best_parameter = nullptr;
>+    const CropRegion prev_best_region = best_region;
>+
>+    for (CropParameter crop_parameter : crop_parameters) {
>+      CropRegion test_region = prev_best_region;
>+      ++(test_region.*crop_parameter);
>+
>+      if (!IsValidRegion(test_region, reference_frame))
>+        continue;
>+
>+      const double ssim =
>+          Ssim(CropAndZoom(test_region, reference_frame), test_frame);
>+
>+      if (ssim > best_ssim) {
>+        best_ssim = ssim;
>+        best_parameter = crop_parameter;
>+        best_region = test_region;
>+      }
>+    }
>+
>+    // No improvement among any direction, stop iteration.
>+    if (best_parameter == nullptr)
>+      break;
>+
>+    // Iterate in the best direction as long as it improves SSIM.
>+    for (CropRegion test_region = best_region;
>+         IsValidRegion(test_region, reference_frame);
>+         ++(test_region.*best_parameter)) {
>+      const double ssim =
>+          Ssim(CropAndZoom(test_region, reference_frame), test_frame);
>+      if (ssim <= best_ssim)
>+        break;
>+
>+      best_ssim = ssim;
>+      best_region = test_region;
>+    }
>+  }
>+
>+  return best_region;
>+}
>+
>+rtc::scoped_refptr<I420BufferInterface> AdjustCropping(
>+    const rtc::scoped_refptr<I420BufferInterface>& reference_frame,
>+    const rtc::scoped_refptr<I420BufferInterface>& test_frame) {
>+  return CropAndZoom(CalculateCropRegion(reference_frame, test_frame),
>+                     reference_frame);
>+}
>+
>+rtc::scoped_refptr<Video> AdjustCropping(
>+    const rtc::scoped_refptr<Video>& reference_video,
>+    const rtc::scoped_refptr<Video>& test_video) {
>+  class CroppedVideo : public rtc::RefCountedObject<Video> {
>+   public:
>+    CroppedVideo(const rtc::scoped_refptr<Video>& reference_video,
>+                 const rtc::scoped_refptr<Video>& test_video)
>+        : reference_video_(reference_video), test_video_(test_video) {
>+      RTC_CHECK_EQ(reference_video->number_of_frames(),
>+                   test_video->number_of_frames());
>+      RTC_CHECK_EQ(reference_video->width(), test_video->width());
>+      RTC_CHECK_EQ(reference_video->height(), test_video->height());
>+    }
>+
>+    int width() const override { return test_video_->width(); }
>+    int height() const override { return test_video_->height(); }
>+    size_t number_of_frames() const override {
>+      return test_video_->number_of_frames();
>+    }
>+
>+    rtc::scoped_refptr<I420BufferInterface> GetFrame(
>+        size_t index) const override {
>+      const rtc::scoped_refptr<I420BufferInterface> reference_frame =
>+          reference_video_->GetFrame(index);
>+
>+      // Only calculate cropping region once per frame since it's expensive.
>+      if (!crop_regions_.count(index)) {
>+        crop_regions_[index] =
>+            CalculateCropRegion(reference_frame, test_video_->GetFrame(index));
>+      }
>+
>+      return CropAndZoom(crop_regions_[index], reference_frame);
>+    }
>+
>+   private:
>+    const rtc::scoped_refptr<Video> reference_video_;
>+    const rtc::scoped_refptr<Video> test_video_;
>+    // Mutable since this is a cache that affects performance and not logical
>+    // behavior.
>+    mutable std::map<size_t, CropRegion> crop_regions_;
>+  };
>+
>+  return new CroppedVideo(reference_video, test_video);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_geometry_aligner.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_geometry_aligner.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..47667b0d135da263ee40d0002f0b4937f1da6e6a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_geometry_aligner.h
>@@ -0,0 +1,57 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_TOOLS_FRAME_ANALYZER_VIDEO_GEOMETRY_ALIGNER_H_
>+#define RTC_TOOLS_FRAME_ANALYZER_VIDEO_GEOMETRY_ALIGNER_H_
>+
>+#include "api/video/video_frame_buffer.h"
>+#include "rtc_tools/video_file_reader.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+struct CropRegion {
>+  // Each value represents how much to crop from each side. Left is where x=0,
>+  // and top is where y=0. All values equal to zero represents no cropping.
>+  int left = 0;
>+  int right = 0;
>+  int top = 0;
>+  int bottom = 0;
>+};
>+
>+// Crops and zooms in on the cropped region so that the returned frame has the
>+// same resolution as the input frame.
>+rtc::scoped_refptr<I420BufferInterface> CropAndZoom(
>+    const CropRegion& crop_region,
>+    const rtc::scoped_refptr<I420BufferInterface>& frame);
>+
>+// Calculate the optimal cropping region on the reference frame to maximize SSIM
>+// to the test frame.
>+CropRegion CalculateCropRegion(
>+    const rtc::scoped_refptr<I420BufferInterface>& reference_frame,
>+    const rtc::scoped_refptr<I420BufferInterface>& test_frame);
>+
>+// Returns a cropped and zoomed version of the reference frame that matches up
>+// to the test frame. This is a simple helper function on top of
>+// CalculateCropRegion() and CropAndZoom().
>+rtc::scoped_refptr<I420BufferInterface> AdjustCropping(
>+    const rtc::scoped_refptr<I420BufferInterface>& reference_frame,
>+    const rtc::scoped_refptr<I420BufferInterface>& test_frame);
>+
>+// Returns a cropped and zoomed version of the reference video that matches up
>+// to the test video. Frames are individually adjusted for cropping.
>+rtc::scoped_refptr<Video> AdjustCropping(
>+    const rtc::scoped_refptr<Video>& reference_video,
>+    const rtc::scoped_refptr<Video>& test_video);
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // RTC_TOOLS_FRAME_ANALYZER_VIDEO_GEOMETRY_ALIGNER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_geometry_aligner_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_geometry_aligner_unittest.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..526f7772e1b92046e900dfb40a3afc6a5f7bd15b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_geometry_aligner_unittest.cc
>@@ -0,0 +1,150 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_tools/frame_analyzer/video_geometry_aligner.h"
>+
>+#include <vector>
>+
>+#include "rtc_tools/frame_analyzer/video_quality_analysis.h"
>+#include "rtc_tools/video_file_reader.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+class VideoGeometryAlignerTest : public ::testing::Test {
>+ protected:
>+  void SetUp() {
>+    reference_video_ =
>+        OpenYuvFile(ResourcePath("foreman_128x96", "yuv"), 128, 96);
>+    ASSERT_TRUE(reference_video_);
>+
>+    // Very simple 4x4 frame used for verying CropAndZoom.
>+    const uint8_t data_y[] = {0, 1, 2,  3,  4,  5,  6,  7,
>+                              8, 9, 10, 11, 12, 13, 14, 15};
>+    const uint8_t data_u[] = {0, 1, 2, 3};
>+    const uint8_t data_v[] = {0, 1, 2, 3};
>+    test_frame_ = I420Buffer::Copy(
>+        /* width= */ 4, /* height= */ 4, data_y, /* stride_y= */ 4, data_u,
>+        /* stride_u= */ 2, data_v, /* stride_v= */ 2);
>+  }
>+
>+  rtc::scoped_refptr<Video> reference_video_;
>+  rtc::scoped_refptr<I420BufferInterface> test_frame_;
>+};
>+
>+// Teach gtest how to compare CropRegions.
>+bool operator==(const CropRegion& a, const CropRegion& b) {
>+  return a.left == b.left && a.top == b.top && a.right == b.right &&
>+         a.bottom == b.bottom;
>+}
>+
>+TEST_F(VideoGeometryAlignerTest, CropAndZoomIdentity) {
>+  const rtc::scoped_refptr<I420BufferInterface> frame =
>+      reference_video_->GetFrame(0);
>+
>+  // Assume perfect match, i.e. SSIM == 1.
>+  CropRegion identity_region;
>+  EXPECT_EQ(1.0, Ssim(frame, CropAndZoom(identity_region, frame)));
>+}
>+
>+TEST_F(VideoGeometryAlignerTest, CropAndZoomLeft) {
>+  CropRegion region;
>+  region.left = 2;
>+  const rtc::scoped_refptr<I420BufferInterface> cropped_frame =
>+      CropAndZoom(region, test_frame_);
>+  EXPECT_EQ(std::vector<uint8_t>(
>+                {2, 2, 3, 3, 6, 6, 7, 7, 10, 10, 11, 11, 14, 14, 15, 15}),
>+            std::vector<uint8_t>(cropped_frame->DataY(),
>+                                 cropped_frame->DataY() + 16));
>+  EXPECT_EQ(
>+      std::vector<uint8_t>({1, 1, 3, 3}),
>+      std::vector<uint8_t>(cropped_frame->DataU(), cropped_frame->DataU() + 4));
>+  EXPECT_EQ(
>+      std::vector<uint8_t>({1, 1, 3, 3}),
>+      std::vector<uint8_t>(cropped_frame->DataV(), cropped_frame->DataV() + 4));
>+}
>+
>+TEST_F(VideoGeometryAlignerTest, CropAndZoomTop) {
>+  CropRegion region;
>+  region.top = 2;
>+  const rtc::scoped_refptr<I420BufferInterface> cropped_frame =
>+      CropAndZoom(region, test_frame_);
>+  EXPECT_EQ(std::vector<uint8_t>(
>+                {8, 9, 10, 11, 10, 11, 12, 13, 12, 13, 14, 15, 12, 13, 14, 15}),
>+            std::vector<uint8_t>(cropped_frame->DataY(),
>+                                 cropped_frame->DataY() + 16));
>+  EXPECT_EQ(
>+      std::vector<uint8_t>({2, 3, 2, 3}),
>+      std::vector<uint8_t>(cropped_frame->DataU(), cropped_frame->DataU() + 4));
>+  EXPECT_EQ(
>+      std::vector<uint8_t>({2, 3, 2, 3}),
>+      std::vector<uint8_t>(cropped_frame->DataV(), cropped_frame->DataV() + 4));
>+}
>+
>+TEST_F(VideoGeometryAlignerTest, CropAndZoomRight) {
>+  CropRegion region;
>+  region.right = 2;
>+  const rtc::scoped_refptr<I420BufferInterface> cropped_frame =
>+      CropAndZoom(region, test_frame_);
>+  EXPECT_EQ(std::vector<uint8_t>(
>+                {0, 0, 1, 1, 4, 4, 5, 5, 8, 8, 9, 9, 12, 12, 13, 13}),
>+            std::vector<uint8_t>(cropped_frame->DataY(),
>+                                 cropped_frame->DataY() + 16));
>+  EXPECT_EQ(
>+      std::vector<uint8_t>({0, 0, 2, 2}),
>+      std::vector<uint8_t>(cropped_frame->DataU(), cropped_frame->DataU() + 4));
>+  EXPECT_EQ(
>+      std::vector<uint8_t>({0, 0, 2, 2}),
>+      std::vector<uint8_t>(cropped_frame->DataV(), cropped_frame->DataV() + 4));
>+}
>+
>+TEST_F(VideoGeometryAlignerTest, CropAndZoomBottom) {
>+  CropRegion region;
>+  region.bottom = 2;
>+  const rtc::scoped_refptr<I420BufferInterface> cropped_frame =
>+      CropAndZoom(region, test_frame_);
>+  EXPECT_EQ(
>+      std::vector<uint8_t>({0, 1, 2, 3, 2, 3, 4, 5, 4, 5, 6, 7, 4, 5, 6, 7}),
>+      std::vector<uint8_t>(cropped_frame->DataY(),
>+                           cropped_frame->DataY() + 16));
>+  EXPECT_EQ(
>+      std::vector<uint8_t>({0, 1, 0, 1}),
>+      std::vector<uint8_t>(cropped_frame->DataU(), cropped_frame->DataU() + 4));
>+  EXPECT_EQ(
>+      std::vector<uint8_t>({0, 1, 0, 1}),
>+      std::vector<uint8_t>(cropped_frame->DataV(), cropped_frame->DataV() + 4));
>+}
>+
>+TEST_F(VideoGeometryAlignerTest, CalculateCropRegionIdentity) {
>+  const rtc::scoped_refptr<I420BufferInterface> frame =
>+      reference_video_->GetFrame(0);
>+  CropRegion identity_region;
>+  EXPECT_EQ(identity_region, CalculateCropRegion(frame, frame));
>+}
>+
>+TEST_F(VideoGeometryAlignerTest, CalculateCropRegionArbitrary) {
>+  // Arbitrary crop region.
>+  CropRegion crop_region;
>+  crop_region.left = 2;
>+  crop_region.top = 4;
>+  crop_region.right = 5;
>+  crop_region.bottom = 3;
>+
>+  const rtc::scoped_refptr<I420BufferInterface> frame =
>+      reference_video_->GetFrame(0);
>+
>+  EXPECT_EQ(crop_region,
>+            CalculateCropRegion(frame, CropAndZoom(crop_region, frame)));
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.cc
>index 1c30d08c56551ba3951cdc3cff4f0ed37616b329..40026802c7743da841294209bdaabf1ee8220abd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.cc
>@@ -13,6 +13,7 @@
> #include <algorithm>
> #include <numeric>
> 
>+#include "rtc_base/logging.h"
> #include "test/testsupport/perf_test.h"
> #include "third_party/libyuv/include/libyuv/compare.h"
> #include "third_party/libyuv/include/libyuv/convert.h"
>@@ -56,16 +57,11 @@ std::vector<AnalysisResult> RunAnalysis(
>     const rtc::scoped_refptr<webrtc::test::Video>& test_video,
>     const std::vector<size_t>& test_frame_indices) {
>   std::vector<AnalysisResult> results;
>-  for (size_t i = 0; i < test_frame_indices.size(); ++i) {
>-    // Ignore duplicated frames in the test video.
>-    if (i > 0 && test_frame_indices[i] == test_frame_indices[i - 1])
>-      continue;
>-
>+  for (size_t i = 0; i < test_video->number_of_frames(); ++i) {
>     const rtc::scoped_refptr<I420BufferInterface>& test_frame =
>         test_video->GetFrame(i);
>     const rtc::scoped_refptr<I420BufferInterface>& reference_frame =
>-        reference_video->GetFrame(test_frame_indices[i] %
>-                                  reference_video->number_of_frames());
>+        reference_video->GetFrame(i);
> 
>     // Fill in the result struct.
>     AnalysisResult result;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_temporal_aligner.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_temporal_aligner.cc
>index 2ebffbc4e75b3be400d9ed8d4d684481dab1ce80..fa25f73f8e1448c87d40d7799e2d9011e8d043f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_temporal_aligner.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_temporal_aligner.cc
>@@ -215,7 +215,7 @@ std::vector<size_t> FindMatchingFrameIndices(
> 
> rtc::scoped_refptr<Video> ReorderVideo(const rtc::scoped_refptr<Video>& video,
>                                        const std::vector<size_t>& indices) {
>-  return new ReorderedVideo(video, indices);
>+  return new ReorderedVideo(new LoopingVideo(video), indices);
> }
> 
> rtc::scoped_refptr<Video> GenerateAlignedReferenceVideo(
>@@ -228,7 +228,7 @@ rtc::scoped_refptr<Video> GenerateAlignedReferenceVideo(
> rtc::scoped_refptr<Video> GenerateAlignedReferenceVideo(
>     const rtc::scoped_refptr<Video>& reference_video,
>     const std::vector<size_t>& indices) {
>-  return ReorderVideo(new LoopingVideo(reference_video), indices);
>+  return ReorderVideo(reference_video, indices);
> }
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/BUILD.gn
>index a5036dcea9cd186001d8d493f4125750092e72b6..47b81430472b179574b056a33faed8b06624dbcc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/BUILD.gn
>@@ -6,8 +6,8 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>-import("../../webrtc.gni")
> import("//third_party/protobuf/proto_library.gni")
>+import("../../webrtc.gni")
> 
> if (rtc_enable_protobuf) {
>   proto_library("network_tester_config_proto") {
>@@ -44,6 +44,7 @@ if (rtc_enable_protobuf) {
>       "../../p2p",
>       "../../rtc_base:checks",
>       "../../rtc_base:protobuf_utils",
>+      "../../rtc_base:rtc_base",
>       "../../rtc_base:rtc_base_approved",
>       "../../rtc_base:rtc_task_queue",
>       "../../rtc_base:sequenced_task_checker",
>@@ -80,7 +81,7 @@ if (rtc_enable_protobuf) {
> 
>     deps = [
>       ":network_tester",
>-      "../../rtc_base:rtc_base_tests_utils",
>+      "../../rtc_base:gunit_helpers",
>       "../../test:fileutils",
>       "../../test:test_support",
>       "//testing/gtest",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.cc
>index e5bd92efa93c6db1550548ab628875f259b5b081..9bfdfa7732c820e86019fd913180834c1884c0e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.cc
>@@ -10,6 +10,9 @@
> 
> #include "rtc_tools/network_tester/test_controller.h"
> 
>+#include "absl/types/optional.h"
>+#include "rtc_base/thread.h"
>+
> namespace webrtc {
> 
> TestController::TestController(int min_port,
>@@ -24,17 +27,15 @@ TestController::TestController(int min_port,
>   RTC_DCHECK_RUN_ON(&test_controller_thread_checker_);
>   packet_sender_checker_.Detach();
>   send_data_.fill(42);
>-  auto socket =
>+  udp_socket_ =
>       std::unique_ptr<rtc::AsyncPacketSocket>(socket_factory_.CreateUdpSocket(
>           rtc::SocketAddress(rtc::GetAnyIP(AF_INET), 0), min_port, max_port));
>-  socket->SignalReadPacket.connect(this, &TestController::OnReadPacket);
>-  udp_transport_.reset(
>-      new cricket::UdpTransport("network tester transport", std::move(socket)));
>+  udp_socket_->SignalReadPacket.connect(this, &TestController::OnReadPacket);
> }
> 
> void TestController::SendConnectTo(const std::string& hostname, int port) {
>   RTC_DCHECK_RUN_ON(&test_controller_thread_checker_);
>-  udp_transport_->SetRemoteAddress(rtc::SocketAddress(hostname, port));
>+  remote_address_ = rtc::SocketAddress(hostname, port);
>   NetworkTesterPacket packet;
>   packet.set_type(NetworkTesterPacket::HAND_SHAKING);
>   SendData(packet, absl::nullopt);
>@@ -57,8 +58,8 @@ void TestController::SendData(const NetworkTesterPacket& packet,
>   packet.SerializeToArray(&send_data_[1], std::numeric_limits<char>::max());
>   if (data_size && *data_size > packet_size)
>     packet_size = *data_size;
>-  udp_transport_->SendPacket(send_data_.data(), packet_size,
>-                             rtc::PacketOptions(), 0);
>+  udp_socket_->SendTo((const void*)send_data_.data(), packet_size,
>+                      remote_address_, rtc::PacketOptions());
> }
> 
> void TestController::OnTestDone() {
>@@ -80,7 +81,7 @@ void TestController::OnReadPacket(rtc::AsyncPacketSocket* socket,
>                                   const char* data,
>                                   size_t len,
>                                   const rtc::SocketAddress& remote_addr,
>-                                  const rtc::PacketTime& packet_time) {
>+                                  const int64_t& packet_time_us) {
>   RTC_DCHECK_RUN_ON(&test_controller_thread_checker_);
>   size_t packet_size = data[0];
>   std::string receive_data(&data[1], packet_size);
>@@ -91,7 +92,7 @@ void TestController::OnReadPacket(rtc::AsyncPacketSocket* socket,
>     case NetworkTesterPacket::HAND_SHAKING: {
>       NetworkTesterPacket packet;
>       packet.set_type(NetworkTesterPacket::TEST_START);
>-      udp_transport_->SetRemoteAddress(remote_addr);
>+      remote_address_ = remote_addr;
>       SendData(packet, absl::nullopt);
>       packet_sender_.reset(new PacketSender(this, config_file_path_));
>       packet_sender_->StartSending();
>@@ -109,7 +110,7 @@ void TestController::OnReadPacket(rtc::AsyncPacketSocket* socket,
>       break;
>     }
>     case NetworkTesterPacket::TEST_DATA: {
>-      packet.set_arrival_timestamp(packet_time.timestamp);
>+      packet.set_arrival_timestamp(packet_time_us);
>       packet.set_packet_size(len);
>       packet_logger_.LogPacket(packet);
>       break;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.h
>index cf65e174595616995f3fd3873234e092b08fc38d..a65272a09d1eb40a5f0b90f537ef7b8bb622229c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.h
>@@ -18,7 +18,7 @@
> #include <utility>
> 
> #include "p2p/base/basicpacketsocketfactory.h"
>-#include "p2p/base/udptransport.h"
>+#include "rtc_base/asyncpacketsocket.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/ignore_wundef.h"
> #include "rtc_tools/network_tester/packet_logger.h"
>@@ -60,7 +60,7 @@ class TestController : public sigslot::has_slots<> {
>                     const char* data,
>                     size_t len,
>                     const rtc::SocketAddress& remote_addr,
>-                    const rtc::PacketTime& packet_time);
>+                    const int64_t& packet_time_us);
>   rtc::ThreadChecker test_controller_thread_checker_;
>   rtc::SequencedTaskChecker packet_sender_checker_;
>   rtc::BasicPacketSocketFactory socket_factory_;
>@@ -70,7 +70,8 @@ class TestController : public sigslot::has_slots<> {
>   bool local_test_done_ RTC_GUARDED_BY(local_test_done_lock_);
>   bool remote_test_done_;
>   std::array<char, kEthernetMtu> send_data_;
>-  std::unique_ptr<cricket::UdpTransport> udp_transport_;
>+  std::unique_ptr<rtc::AsyncPacketSocket> udp_socket_;
>+  rtc::SocketAddress remote_address_;
>   std::unique_ptr<PacketSender> packet_sender_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(TestController);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/unpack_aecdump/unpack.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/unpack_aecdump/unpack.cc
>index 4c75608e7488a60435fc11cf14489666a28c63ca..142b49730a36170b6576174a5b731228fd657ba3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/unpack_aecdump/unpack.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/unpack_aecdump/unpack.cc
>@@ -29,27 +29,34 @@ RTC_PUSH_IGNORING_WUNDEF()
> RTC_POP_IGNORING_WUNDEF()
> 
> // TODO(andrew): unpack more of the data.
>-DEFINE_string(input_file, "input", "The name of the input stream file.");
>-DEFINE_string(output_file,
>-              "ref_out",
>-              "The name of the reference output stream file.");
>-DEFINE_string(reverse_file,
>-              "reverse",
>-              "The name of the reverse input stream file.");
>-DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
>-DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
>-DEFINE_string(level_file, "level.int32", "The name of the level file.");
>-DEFINE_string(keypress_file, "keypress.bool", "The name of the keypress file.");
>-DEFINE_string(callorder_file,
>-              "callorder",
>-              "The name of the render/capture call order file.");
>-DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
>-DEFINE_bool(full, false, "Unpack the full set of files (normally not needed).");
>-DEFINE_bool(raw, false, "Write raw data instead of a WAV file.");
>-DEFINE_bool(text,
>-            false,
>-            "Write non-audio files as text files instead of binary files.");
>-DEFINE_bool(help, false, "Print this message.");
>+WEBRTC_DEFINE_string(input_file, "input", "The name of the input stream file.");
>+WEBRTC_DEFINE_string(output_file,
>+                     "ref_out",
>+                     "The name of the reference output stream file.");
>+WEBRTC_DEFINE_string(reverse_file,
>+                     "reverse",
>+                     "The name of the reverse input stream file.");
>+WEBRTC_DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
>+WEBRTC_DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
>+WEBRTC_DEFINE_string(level_file, "level.int32", "The name of the level file.");
>+WEBRTC_DEFINE_string(keypress_file,
>+                     "keypress.bool",
>+                     "The name of the keypress file.");
>+WEBRTC_DEFINE_string(callorder_file,
>+                     "callorder",
>+                     "The name of the render/capture call order file.");
>+WEBRTC_DEFINE_string(settings_file,
>+                     "settings.txt",
>+                     "The name of the settings file.");
>+WEBRTC_DEFINE_bool(full,
>+                   false,
>+                   "Unpack the full set of files (normally not needed).");
>+WEBRTC_DEFINE_bool(raw, false, "Write raw data instead of a WAV file.");
>+WEBRTC_DEFINE_bool(
>+    text,
>+    false,
>+    "Write non-audio files as text files instead of binary files.");
>+WEBRTC_DEFINE_bool(help, false, "Print this message.");
> 
> #define PRINT_CONFIG(field_name)                                         \
>   if (msg.has_##field_name()) {                                          \
>@@ -87,6 +94,10 @@ void WriteCallOrderData(const bool render_call,
>   WriteData(&call_type, sizeof(call_type), file, filename.c_str());
> }
> 
>+bool WritingCallOrderFile() {
>+  return FLAG_full;
>+}
>+
> }  // namespace
> 
> int do_main(int argc, char* argv[]) {
>@@ -125,7 +136,9 @@ int do_main(int argc, char* argv[]) {
> 
>   rtc::StringBuilder callorder_raw_name;
>   callorder_raw_name << FLAG_callorder_file << ".char";
>-  FILE* callorder_char_file = OpenFile(callorder_raw_name.str(), "wb");
>+  FILE* callorder_char_file = WritingCallOrderFile()
>+                                  ? OpenFile(callorder_raw_name.str(), "wb")
>+                                  : nullptr;
>   FILE* settings_file = OpenFile(FLAG_settings_file, "wb");
> 
>   while (ReadMessageFromFile(debug_file, &event_msg)) {
>@@ -163,8 +176,10 @@ int do_main(int argc, char* argv[]) {
>                        reverse_raw_file.get());
>       }
>       if (FLAG_full) {
>-        WriteCallOrderData(true /* render_call */, callorder_char_file,
>-                           FLAG_callorder_file);
>+        if (WritingCallOrderFile()) {
>+          WriteCallOrderData(true /* render_call */, callorder_char_file,
>+                             FLAG_callorder_file);
>+        }
>       }
>     } else if (event_msg.type() == Event::STREAM) {
>       frame_count++;
>@@ -222,8 +237,10 @@ int do_main(int argc, char* argv[]) {
>       }
> 
>       if (FLAG_full) {
>-        WriteCallOrderData(false /* render_call */, callorder_char_file,
>-                           FLAG_callorder_file);
>+        if (WritingCallOrderFile()) {
>+          WriteCallOrderData(false /* render_call */, callorder_char_file,
>+                             FLAG_callorder_file);
>+        }
>         if (msg.has_delay()) {
>           static FILE* delay_file = OpenFile(FLAG_delay_file, "wb");
>           int32_t delay = msg.delay();
>@@ -359,9 +376,11 @@ int do_main(int argc, char* argv[]) {
>         output_wav_file.reset(new WavWriter(
>             output_name.str(), output_sample_rate, num_output_channels));
> 
>-        rtc::StringBuilder callorder_name;
>-        callorder_name << FLAG_callorder_file << frame_count << ".char";
>-        callorder_char_file = OpenFile(callorder_name.str(), "wb");
>+        if (WritingCallOrderFile()) {
>+          rtc::StringBuilder callorder_name;
>+          callorder_name << FLAG_callorder_file << frame_count << ".char";
>+          callorder_char_file = OpenFile(callorder_name.str(), "wb");
>+        }
>       }
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/video_analysis.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/video_analysis.py
>index a5d267b0b29a3a306ce696b5dce7bfa964808ecd..0c22617ecf210ba787ae99825cb574706313435f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/video_analysis.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/video_analysis.py
>@@ -79,7 +79,7 @@ def _ParseArgs():
>                     help='Path to the frame analyzer executable.'
>                     'Default: %default')
>   parser.add_option('--zxing_path', type='string',
>-                    help='Path to the zebra xing barcode analyzer.')
>+                    help='DEPRECATED.')
>   parser.add_option('--ref_rec_dir', type='string', default='ref',
>                     help='Path to where reference recordings will be created.'
>                     'Ideally keep the ref and test directories on separate'
>@@ -118,8 +118,6 @@ def _ParseArgs():
>                    'generated!')
>   if not os.path.isfile(options.frame_analyzer):
>     parser.warning('Cannot find frame_analyzer, no metrics will be generated!')
>-  if not os.path.isfile(options.zxing_path):
>-    parser.warning('Cannot find Zebra Xing, no metrics will be generated!')
> 
>   return options
> 
>@@ -442,12 +440,6 @@ def CompareVideos(options, cropped_ref_file, cropped_test_file):
>     '--ref_video=%s' % cropped_ref_file,
>     '--test_video=%s' % cropped_test_file,
>     '--frame_analyzer=%s' % os.path.abspath(options.frame_analyzer),
>-    '--zxing_path=%s' % options.zxing_path,
>-    '--ffmpeg_path=%s' % options.ffmpeg,
>-    '--stats_file_ref=%s_stats.txt' %
>-        os.path.join(os.path.dirname(cropped_ref_file), cropped_ref_file),
>-    '--stats_file_test=%s_stats.txt' %
>-        os.path.join(os.path.dirname(cropped_test_file), cropped_test_file),
>     '--yuv_frame_height=%d' % crop_height,
>     '--yuv_frame_width=%d' % crop_width
>   ]
>@@ -472,7 +464,6 @@ def main():
>     --app_name AppRTCMobile \
>     --ffmpeg ./ffmpeg --ref_video_device=/dev/video0 \
>     --test_video_device=/dev/video1 \
>-    --zxing_path ./zxing \
>     --test_crop_parameters 'crop=950:420:130:56' \
>     --ref_crop_parameters 'hflip, crop=950:420:130:56' \
>     --ref_rec_dir /tmp/ref \
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/BUILD.gn
>index b4f193d1f326a6a550b32111a50a99c4e902b685..60ea4734e9520c181db51ace4a7e37a00621e229 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/BUILD.gn
>@@ -8,8 +8,8 @@
> 
> import("../webrtc.gni")
> if (is_ios) {
>-  import("//build/config/ios/rules.gni")
>   import("//build/config/ios/ios_sdk.gni")
>+  import("//build/config/ios/rules.gni")
> }
> if (is_mac) {
>   import("//build/config/mac/rules.gni")
>@@ -40,6 +40,7 @@ if (is_ios || is_mac) {
>       "objc/Framework/Headers",  # TODO(bugs.webrtc.org/9627): Remove this.
>     ]
>     cflags = [
>+      "-Wimplicit-retain-self",
>       "-Wstrict-overflow",
>       "-Wmissing-field-initializers",
>     ]
>@@ -262,6 +263,8 @@ if (is_ios || is_mac) {
>           "../system_wrappers:metrics",
>         ]
> 
>+        libs = [ "AudioToolbox.framework" ]
>+
>         if (is_clang) {
>           # Suppress warnings from the Chromium Clang plugin
>           # (bugs.webrtc.org/163).
>@@ -287,6 +290,8 @@ if (is_ios || is_mac) {
> 
>         public_configs = [ ":common_config_objc" ]
> 
>+        libs = [ "AVFoundation.framework" ]
>+
>         deps = [
>           ":audio_session_observer",
>           ":base_objc",
>@@ -756,8 +761,8 @@ if (is_ios || is_mac) {
>       deps = [
>         ":base_objc",
>         ":helpers_objc",
>+        "../api/video:encoded_image",
>         "../api/video_codecs:video_codecs_api",
>-        "../common_video",
>         "../modules:module_api",
>         "../modules/video_coding:video_codec_interface",
>         "../rtc_base:rtc_base",
>@@ -793,6 +798,8 @@ if (is_ios || is_mac) {
>         "objc/api/peerconnection/RTCConfiguration+Private.h",
>         "objc/api/peerconnection/RTCConfiguration.h",
>         "objc/api/peerconnection/RTCConfiguration.mm",
>+        "objc/api/peerconnection/RTCCryptoOptions.h",
>+        "objc/api/peerconnection/RTCCryptoOptions.mm",
>         "objc/api/peerconnection/RTCDataChannel+Private.h",
>         "objc/api/peerconnection/RTCDataChannel.h",
>         "objc/api/peerconnection/RTCDataChannel.mm",
>@@ -903,6 +910,7 @@ if (is_ios || is_mac) {
>         ":videorendereradapter_objc",
>         ":videosource_objc",
>         ":videotoolbox_objc",
>+        "../api:create_peerconnection_factory",
>         "../api:libjingle_peerconnection_api",
>         "../api/audio_codecs:audio_codecs_api",
>         "../api/audio_codecs:builtin_audio_decoder_factory",
>@@ -913,9 +921,9 @@ if (is_ios || is_mac) {
>         "../media:rtc_media_base",
>         "../modules:module_api",
>         "../modules/audio_device:audio_device_api",
>+        "../modules/audio_processing:api",
>         "../modules/audio_processing:audio_processing",
>         "../modules/video_coding:video_codec_interface",
>-        "../pc:create_pc_factory",
>         "../pc:peerconnection",
>         "../rtc_base:checks",
>         "../rtc_base:rtc_base",
>@@ -1031,6 +1039,7 @@ if (is_ios || is_mac) {
>             "objc/unittests/RTCPeerConnectionFactory_xctest.m",
>             "objc/unittests/frame_buffer_helpers.h",
>             "objc/unittests/frame_buffer_helpers.mm",
>+            "objc/unittests/nalu_rewriter_xctest.mm",
>           ]
> 
>           # TODO(peterhanspers): Reenable these tests on simulator.
>@@ -1053,6 +1062,7 @@ if (is_ios || is_mac) {
>             ":native_api_audio_device_module",
>             ":native_video",
>             ":peerconnectionfactory_base_objc",
>+            ":video_toolbox_cc",
>             ":videocapture_objc",
>             ":videocodec_objc",
>             ":videoframebuffer_objc",
>@@ -1063,8 +1073,8 @@ if (is_ios || is_mac) {
>             "../media:rtc_media_base",
>             "../media:rtc_media_tests_utils",
>             "../modules:module_api",
>+            "../rtc_base:gunit_helpers",
>             "../rtc_base:rtc_base",
>-            "../rtc_base:rtc_base_tests_utils",
>             "../system_wrappers:system_wrappers",
>             "//third_party/libyuv",
>           ]
>@@ -1098,7 +1108,7 @@ if (is_ios || is_mac) {
>         rtc_ios_xctest_test("sdk_unittests") {
>           info_plist = "//test/ios/Info.plist"
>           sources = [
>-            "objc/unittests/main.m",
>+            "objc/unittests/main.mm",
>           ]
> 
>           _bundle_id_suffix = ios_generic_test_bundle_id_suffix
>@@ -1107,6 +1117,7 @@ if (is_ios || is_mac) {
>             ":peerconnectionfactory_base_objc",
>             ":sdk_unittests_bundle_data",
>             ":sdk_unittests_sources",
>+            "//test:test_support",
>           ]
>           ldflags = [ "-all_load" ]
>         }
>@@ -1116,7 +1127,7 @@ if (is_ios || is_mac) {
>           info_plist = "//test/ios/Info.plist"
>           sources = [
>             "objc/unittests/RTCDoNotPutCPlusPlusInFrameworkHeaders_xctest.m",
>-            "objc/unittests/main.m",
>+            "objc/unittests/main.mm",
>           ]
> 
>           _bundle_id_suffix = ios_generic_test_bundle_id_suffix
>@@ -1124,6 +1135,7 @@ if (is_ios || is_mac) {
>           deps = [
>             ":framework_objc+link",
>             ":ios_framework_bundle",
>+            "//test:test_support",
>           ]
>         }
>       }
>@@ -1170,6 +1182,7 @@ if (is_ios || is_mac) {
>           ":videoframebuffer_objc",
>           ":videosource_objc",
>           ":videotoolbox_objc",
>+          "../api:libjingle_peerconnection_api",
>           "../api/audio_codecs:audio_codecs_api",
>           "../api/audio_codecs:builtin_audio_decoder_factory",
>           "../api/audio_codecs:builtin_audio_encoder_factory",
>@@ -1177,11 +1190,12 @@ if (is_ios || is_mac) {
>           "../media:rtc_media_base",
>           "../modules:module_api",
>           "../modules/audio_device:audio_device_api",
>+          "../modules/audio_processing:api",
>           "../modules/audio_processing:audio_processing",
>           "../modules/video_coding:video_codec_interface",
>-          "../rtc_base:rtc_base_tests_utils",
>+          "../rtc_base:gunit_helpers",
>           "../rtc_base/system:unused",
>-          "//test:test_support",
>+          "../test:test_support",
>           "//third_party/ocmock",
>         ]
> 
>@@ -1269,6 +1283,7 @@ if (is_ios || is_mac) {
>           "objc/api/peerconnection/RTCSessionDescription.h",
>           "objc/api/peerconnection/RTCTracing.h",
>           "objc/api/peerconnection/RTCCertificate.h",
>+          "objc/api/peerconnection/RTCCryptoOptions.h",
>           "objc/api/peerconnection/RTCVideoSource.h",
>           "objc/api/peerconnection/RTCVideoTrack.h",
>           "objc/api/video_codec/RTCVideoCodecConstants.h",
>@@ -1357,6 +1372,7 @@ if (is_ios || is_mac) {
>           "objc/api/peerconnection/RTCAudioTrack.h",
>           "objc/api/peerconnection/RTCCertificate.h",
>           "objc/api/peerconnection/RTCConfiguration.h",
>+          "objc/api/peerconnection/RTCCryptoOptions.h",
>           "objc/api/peerconnection/RTCDataChannel.h",
>           "objc/api/peerconnection/RTCDataChannelConfiguration.h",
>           "objc/api/peerconnection/RTCDtmfSender.h",
>@@ -1590,7 +1606,10 @@ if (is_ios || is_mac) {
>     }
> 
>     rtc_static_library("video_toolbox_cc") {
>-      visibility = [ ":videotoolbox_objc" ]
>+      visibility = [
>+        ":videotoolbox_objc",
>+        ":sdk_unittests_sources",
>+      ]
>       sources = [
>         "objc/components/video_codec/helpers.cc",
>         "objc/components/video_codec/helpers.h",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>index 2d958b76dbe78317279b470f4efb3fcad8b83bd9..65eefa36f2dfb2069ecca7caa49224b262955bba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>@@ -29,8 +29,9 @@
> #import "WebRTC/RTCVideoCodecH264.h"
> 
> #include "api/video/video_frame.h"
>-#include "third_party/libyuv/include/libyuv/convert_from.h"
> #include "native/src/objc_frame_buffer.h"
>+#include "third_party/libyuv/include/libyuv/convert_from.h"
>+#include "webrtc/rtc_base/checks.h"
> #include "Framework/Headers/WebRTC/RTCVideoCodecFactory.h"
> #include "Framework/Headers/WebRTC/RTCVideoFrame.h"
> #include "Framework/Headers/WebRTC/RTCVideoFrameBuffer.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/BUILD.gn
>index 63c5bd20c10436b000032b1025ef693597bab86c..edc5da30b79da763dbc74304df39acb797ccf04d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/BUILD.gn
>@@ -7,9 +7,9 @@
> # be found in the AUTHORS file in the root of the source tree.
> 
> if (is_android) {
>-  import("../../webrtc.gni")
>   import("//build/config/android/config.gni")
>   import("//build/config/android/rules.gni")
>+  import("../../webrtc.gni")
> 
>   group("android") {
>     if (!build_with_chromium && is_android) {
>@@ -38,6 +38,7 @@ if (is_android) {
>     deps = [
>       ":audio_api_java",
>       ":base_java",
>+      ":builtin_audio_codecs_java",
>       ":camera_java",
>       ":default_video_codec_factory_java",
>       ":filevideo_java",
>@@ -110,23 +111,22 @@ if (is_android) {
>     public_deps = [  # no-presubmit-check TODO(webrtc:8603)
>       ":audio_jni",
>       ":base_jni",
>+      ":builtin_audio_codecs_jni",
>+      ":default_video_codec_factory_jni",
>       ":java_audio_device_module_jni",
>+      ":legacy_hwcodecs_jni",
>       ":media_jni",
>       ":peerconnection_jni",
>       ":video_jni",
>-      "../../pc:create_pc_factory",
>+      "../../api:create_peerconnection_factory",
>     ]
>   }
> 
>   rtc_static_library("libjingle_peerconnection_metrics_default_jni") {
>     visibility = [ "*" ]
>-
>-    allow_poison = [ "software_video_codecs" ]  # TODO(bugs.webrtc.org/7925): Remove.
>-
>     sources = [
>       "src/jni/androidmetrics.cc",
>     ]
>-
>     deps = [
>       ":base_jni",
>       ":generated_metrics_jni",
>@@ -204,7 +204,11 @@ if (is_android) {
>   }
> 
>   rtc_android_library("audio_api_java") {
>-    java_files = [ "api/org/webrtc/audio/AudioDeviceModule.java" ]
>+    java_files = [
>+      "api/org/webrtc/audio/AudioDeviceModule.java",
>+      "api/org/webrtc/AudioDecoderFactoryFactory.java",
>+      "api/org/webrtc/AudioEncoderFactoryFactory.java",
>+    ]
> 
>     deps = [
>       ":base_java",
>@@ -250,6 +254,7 @@ if (is_android) {
>       "api/org/webrtc/VideoFrameDrawer.java",
>       "api/org/webrtc/YuvConverter.java",
>       "api/org/webrtc/YuvHelper.java",
>+      "api/org/webrtc/TimestampAligner.java",
>       "src/java/org/webrtc/EglBase10.java",
>       "src/java/org/webrtc/EglBase14.java",
>       "src/java/org/webrtc/GlGenericDrawer.java",
>@@ -276,9 +281,11 @@ if (is_android) {
>       "api/org/webrtc/AudioSource.java",
>       "api/org/webrtc/AudioTrack.java",
>       "api/org/webrtc/CallSessionFileRotatingLogSink.java",
>+      "api/org/webrtc/CryptoOptions.java",
>       "api/org/webrtc/DataChannel.java",
>       "api/org/webrtc/DtmfSender.java",
>       "api/org/webrtc/FecControllerFactoryFactoryInterface.java",
>+      "api/org/webrtc/MediaTransportFactoryFactory.java",
>       "api/org/webrtc/FrameDecryptor.java",
>       "api/org/webrtc/FrameEncryptor.java",
>       "api/org/webrtc/IceCandidate.java",
>@@ -295,6 +302,7 @@ if (is_android) {
>       "api/org/webrtc/PeerConnection.java",
>       "api/org/webrtc/PeerConnectionDependencies.java",
>       "api/org/webrtc/PeerConnectionFactory.java",
>+      "api/org/webrtc/RtcCertificatePem.java",
>       "api/org/webrtc/RTCStats.java",
>       "api/org/webrtc/RTCStatsCollectorCallback.java",
>       "api/org/webrtc/RTCStatsReport.java",
>@@ -317,6 +325,7 @@ if (is_android) {
>     deps = [
>       ":audio_api_java",
>       ":base_java",
>+      ":builtin_audio_codecs_java",
>       ":default_video_codec_factory_java",
>       ":logging_java",
>       ":swcodecs_java",
>@@ -426,6 +435,17 @@ if (is_android) {
>     ]
>   }
> 
>+  rtc_android_library("builtin_audio_codecs_java") {
>+    java_files = [
>+      "api/org/webrtc/BuiltinAudioDecoderFactoryFactory.java",
>+      "api/org/webrtc/BuiltinAudioEncoderFactoryFactory.java",
>+    ]
>+
>+    deps = [
>+      ":audio_api_java",
>+    ]
>+  }
>+
>   rtc_android_library("screencapturer_java") {
>     java_files = [ "api/org/webrtc/ScreenCapturerAndroid.java" ]
> 
>@@ -449,18 +469,44 @@ if (is_android) {
>     ]
>   }
> 
>+  rtc_android_library("libvpx_vp8_java") {
>+    visibility = [ "*" ]
>+    java_files = [
>+      "api/org/webrtc/LibvpxVp8Decoder.java",
>+      "api/org/webrtc/LibvpxVp8Encoder.java",
>+    ]
>+    deps = [
>+      ":base_java",
>+      ":video_api_java",
>+      ":video_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
>+
>+  rtc_android_library("libvpx_vp9_java") {
>+    visibility = [ "*" ]
>+    java_files = [
>+      "api/org/webrtc/LibvpxVp9Decoder.java",
>+      "api/org/webrtc/LibvpxVp9Encoder.java",
>+    ]
>+    deps = [
>+      ":base_java",
>+      ":video_api_java",
>+      ":video_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
>+
>   rtc_android_library("swcodecs_java") {
>     java_files = [
>       "api/org/webrtc/SoftwareVideoDecoderFactory.java",
>       "api/org/webrtc/SoftwareVideoEncoderFactory.java",
>-      "src/java/org/webrtc/VP8Encoder.java",
>-      "src/java/org/webrtc/VP8Decoder.java",
>-      "src/java/org/webrtc/VP9Encoder.java",
>-      "src/java/org/webrtc/VP9Decoder.java",
>     ]
> 
>     deps = [
>       ":base_java",
>+      ":libvpx_vp8_java",
>+      ":libvpx_vp9_java",
>       ":video_api_java",
>       ":video_java",
>       "//rtc_base:base_java",
>@@ -511,23 +557,66 @@ if (is_android) {
> 
>     deps = [
>       ":base_jni",
>+      ":builtin_audio_codecs_jni",
>       "../../api/audio_codecs:builtin_audio_decoder_factory",
>       "../../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../../modules/audio_processing:api",
>       "../../modules/audio_processing:audio_processing",
>       "../../rtc_base:rtc_base_approved",
>     ]
>   }
> 
>-  rtc_static_library("video_jni") {
>-    sources = []
>+  rtc_static_library("builtin_audio_codecs_jni") {
>+    sources = [
>+      "src/jni/builtinaudiodecoderfactoryfactory.cc",
>+      "src/jni/builtinaudioencoderfactoryfactory.cc",
>+    ]
>+
>     deps = [
>-      "../../system_wrappers:field_trial",
>+      ":base_jni",
>+      ":generated_builtin_audio_codecs_jni",
>+      ":native_api_jni",
>+      "../../api/audio_codecs:builtin_audio_decoder_factory",
>+      "../../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../../rtc_base:rtc_base_approved",
>     ]
>+  }
> 
>-    sources += [
>+  # Corresponds to MediaCodecVideoEncoder/Decoder in Java.
>+  rtc_static_library("legacy_hwcodecs_jni") {
>+    visibility = [ "*" ]
>+    allow_poison = [ "software_video_codecs" ]
>+    sources = [
>       "src/jni/androidmediacodeccommon.h",
>       "src/jni/androidmediadecoder.cc",
>       "src/jni/androidmediaencoder.cc",
>+    ]
>+    deps = [
>+      ":base_jni",
>+      ":default_video_codec_factory_jni",
>+      ":generated_video_jni",
>+      ":native_api_jni",
>+      ":video_jni",
>+      ":videoframe_jni",
>+      "../..:webrtc_common",
>+      "../../api/video_codecs:video_codecs_api",
>+      "../../common_video:common_video",
>+      "../../media:rtc_internal_video_codecs",
>+      "../../media:rtc_media_base",
>+      "../../modules/video_coding:video_codec_interface",
>+      "../../modules/video_coding:video_coding_utility",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base",
>+      "../../rtc_base:rtc_task_queue_api",
>+      "../../rtc_base:sequenced_task_checker",
>+      "../../rtc_base:weak_ptr",
>+      "../../system_wrappers:field_trial",
>+      "//third_party/libyuv",
>+    ]
>+  }
>+
>+  rtc_static_library("video_jni") {
>+    sources = [
>       "src/jni/androidvideotracksource.cc",
>       "src/jni/androidvideotracksource.h",
>       "src/jni/encodedimage.cc",
>@@ -540,6 +629,7 @@ if (is_android) {
>       "src/jni/nv12buffer.cc",
>       "src/jni/nv21buffer.cc",
>       "src/jni/pc/video.cc",
>+      "src/jni/timestampaligner.cc",
>       "src/jni/videocodecinfo.cc",
>       "src/jni/videocodecinfo.h",
>       "src/jni/videocodecstatus.cc",
>@@ -561,36 +651,25 @@ if (is_android) {
>       "src/jni/yuvhelper.cc",
>     ]
> 
>-    deps += [
>+    deps = [
>       ":base_jni",
>       ":generated_video_jni",
>       ":native_api_jni",
>       ":videoframe_jni",
>-      ":vp8_jni",  # TODO(bugs.webrtc.org/7925): Remove.
>-      ":vp9_jni",  # TODO(bugs.webrtc.org/7925): Remove.
>       "../..:webrtc_common",
>       "../../api:libjingle_peerconnection_api",
>+      "../../api/video:encoded_image",
>       "../../api/video:video_frame",
>-      "../../api/video_codecs:builtin_video_decoder_factory",
>-      "../../api/video_codecs:builtin_video_encoder_factory",
>       "../../api/video_codecs:rtc_software_fallback_wrappers",
>       "../../api/video_codecs:video_codecs_api",
>       "../../common_video:common_video",
>-      "../../media:rtc_audio_video",
>-      "../../media:rtc_h264_profile_id",
>-      "../../media:rtc_internal_video_codecs",
>       "../../media:rtc_media_base",
>-      "../../modules/utility:utility",
>       "../../modules/video_coding:codec_globals_headers",
>       "../../modules/video_coding:video_codec_interface",
>       "../../modules/video_coding:video_coding_utility",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base",
>-      "../../rtc_base:rtc_base_approved",
>-      "../../rtc_base:rtc_task_queue",
>-      "../../rtc_base:sequenced_task_checker",
>-      "../../rtc_base:weak_ptr",
>-      "../../rtc_base/memory:aligned_malloc",
>+      "../../rtc_base:rtc_task_queue_api",
>       "//third_party/libyuv",
>     ]
>   }
>@@ -606,6 +685,8 @@ if (is_android) {
>       "src/jni/pc/androidnetworkmonitor.h",
>       "src/jni/pc/audiotrack.cc",
>       "src/jni/pc/callsessionfilerotatinglogsink.cc",
>+      "src/jni/pc/cryptooptions.cc",
>+      "src/jni/pc/cryptooptions.h",
>       "src/jni/pc/datachannel.cc",
>       "src/jni/pc/datachannel.h",
>       "src/jni/pc/dtmfsender.cc",
>@@ -624,6 +705,8 @@ if (is_android) {
>       "src/jni/pc/peerconnection.h",
>       "src/jni/pc/peerconnectionfactory.cc",
>       "src/jni/pc/peerconnectionfactory.h",
>+      "src/jni/pc/rtccertificate.cc",
>+      "src/jni/pc/rtccertificate.h",
>       "src/jni/pc/rtcstatscollectorcallbackwrapper.cc",
>       "src/jni/pc/rtcstatscollectorcallbackwrapper.h",
>       "src/jni/pc/rtpparameters.cc",
>@@ -669,6 +752,7 @@ if (is_android) {
>       "../../rtc_base:stringutils",
>       "../../system_wrappers:field_trial",
>       "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>   }
> 
>@@ -685,6 +769,50 @@ if (is_android) {
>     ]
>   }
> 
>+  rtc_static_library("default_video_codec_factory_jni") {
>+    visibility = [ "*" ]
>+    allow_poison = [ "software_video_codecs" ]
>+    deps = [
>+      ":swcodecs_jni",
>+      ":video_jni",
>+    ]
>+  }
>+
>+  rtc_static_library("libvpx_vp8_jni") {
>+    visibility = [ "*" ]
>+    allow_poison = [ "software_video_codecs" ]
>+    sources = [
>+      "src/jni/vp8codec.cc",
>+    ]
>+    deps = [
>+      ":base_jni",
>+      ":generated_libvpx_vp8_jni",
>+      ":video_jni",
>+      "../../modules/video_coding:webrtc_vp8",
>+    ]
>+  }
>+
>+  rtc_static_library("libvpx_vp9_jni") {
>+    visibility = [ "*" ]
>+    allow_poison = [ "software_video_codecs" ]
>+    sources = [
>+      "src/jni/vp9codec.cc",
>+    ]
>+    deps = [
>+      ":base_jni",
>+      ":generated_libvpx_vp9_jni",
>+      ":video_jni",
>+      "../../modules/video_coding:webrtc_vp9",
>+    ]
>+  }
>+
>+  rtc_static_library("swcodecs_jni") {
>+    deps = [
>+      ":libvpx_vp8_jni",
>+      ":libvpx_vp9_jni",
>+    ]
>+  }
>+
>   ######################
>   # Native API targets #
>   ######################
>@@ -769,10 +897,7 @@ if (is_android) {
>   # objects.
>   rtc_static_library("native_api_codecs") {
>     visibility = [ "*" ]
>-    allow_poison = [
>-      "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>-      "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>-    ]
>+    allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
>     sources = [
>       "native_api/codecs/wrapper.cc",
>       "native_api/codecs/wrapper.h",
>@@ -792,14 +917,10 @@ if (is_android) {
>   # API for creating Java PeerConnectionFactory from C++ equivalents.
>   rtc_static_library("native_api_peerconnection") {
>     visibility = [ "*" ]
>-
>-    allow_poison = [ "software_video_codecs" ]  # TODO(bugs.webrtc.org/7925): Remove.
>-
>     sources = [
>       "native_api/peerconnection/peerconnectionfactory.cc",
>       "native_api/peerconnection/peerconnectionfactory.h",
>     ]
>-
>     deps = [
>       ":base_jni",
>       ":peerconnection_jni",
>@@ -813,10 +934,7 @@ if (is_android) {
>   # video interfaces from their Java equivalents.
>   rtc_static_library("native_api_video") {
>     visibility = [ "*" ]
>-    allow_poison = [
>-      "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>-      "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>-    ]
>+    allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
>     sources = [
>       "native_api/video/videosource.cc",
>       "native_api/video/videosource.h",
>@@ -829,6 +947,7 @@ if (is_android) {
>       ":videoframe_jni",
>       "//api:libjingle_peerconnection_api",
>       "//api/video:video_frame",
>+      "//rtc_base:rtc_base",
>       "//rtc_base:rtc_base_approved",
>       "//third_party/abseil-cpp/absl/memory",
>     ]
>@@ -891,6 +1010,7 @@ if (is_android) {
>       "../../logging:rtc_event_log_api",
>       "../../media:rtc_audio_video",
>       "../../modules/audio_device:audio_device",
>+      "../../modules/audio_processing:api",
>       "../../modules/audio_processing:audio_processing",
>     ]
>   }
>@@ -929,30 +1049,6 @@ if (is_android) {
>     ]
>   }
> 
>-  rtc_static_library("vp8_jni") {
>-    sources = [
>-      "src/jni/vp8codec.cc",
>-    ]
>-
>-    deps = [
>-      ":base_jni",
>-      ":generated_vp8_jni",
>-      "../../modules/video_coding:webrtc_vp8",
>-    ]
>-  }
>-
>-  rtc_static_library("vp9_jni") {
>-    sources = [
>-      "src/jni/vp9codec.cc",
>-    ]
>-
>-    deps = [
>-      ":base_jni",
>-      ":generated_vp9_jni",
>-      "../../modules/video_coding:webrtc_vp9",
>-    ]
>-  }
>-
>   rtc_static_library("logging_jni") {
>     visibility = [ "*" ]
>     sources = [
>@@ -981,6 +1077,7 @@ if (is_android) {
>       ":base_jni",
>       ":generated_audio_device_module_base_jni",
>       ":native_api_jni",
>+      "../../modules/audio_device:audio_device_api",
>       "../../modules/audio_device:audio_device_buffer",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>@@ -1114,6 +1211,7 @@ if (is_android) {
>       "api/org/webrtc/JavaI420Buffer.java",
>       "api/org/webrtc/MediaCodecVideoDecoder.java",
>       "api/org/webrtc/MediaCodecVideoEncoder.java",
>+      "api/org/webrtc/TimestampAligner.java",
>       "api/org/webrtc/VideoCodecInfo.java",
>       "api/org/webrtc/VideoCodecStatus.java",
>       "api/org/webrtc/VideoDecoder.java",
>@@ -1140,10 +1238,10 @@ if (is_android) {
>     jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>   }
> 
>-  generate_jni("generated_vp8_jni") {
>+  generate_jni("generated_libvpx_vp8_jni") {
>     sources = [
>-      "src/java/org/webrtc/VP8Decoder.java",
>-      "src/java/org/webrtc/VP8Encoder.java",
>+      "api/org/webrtc/LibvpxVp8Decoder.java",
>+      "api/org/webrtc/LibvpxVp8Encoder.java",
>     ]
> 
>     jni_package = ""
>@@ -1151,10 +1249,10 @@ if (is_android) {
>     jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>   }
> 
>-  generate_jni("generated_vp9_jni") {
>+  generate_jni("generated_libvpx_vp9_jni") {
>     sources = [
>-      "src/java/org/webrtc/VP9Decoder.java",
>-      "src/java/org/webrtc/VP9Encoder.java",
>+      "api/org/webrtc/LibvpxVp9Decoder.java",
>+      "api/org/webrtc/LibvpxVp9Encoder.java",
>     ]
> 
>     jni_package = ""
>@@ -1166,6 +1264,7 @@ if (is_android) {
>     sources = [
>       "api/org/webrtc/AudioTrack.java",
>       "api/org/webrtc/CallSessionFileRotatingLogSink.java",
>+      "api/org/webrtc/CryptoOptions.java",
>       "api/org/webrtc/DataChannel.java",
>       "api/org/webrtc/DtmfSender.java",
>       "api/org/webrtc/IceCandidate.java",
>@@ -1178,6 +1277,7 @@ if (is_android) {
>       "api/org/webrtc/RTCStats.java",
>       "api/org/webrtc/RTCStatsCollectorCallback.java",
>       "api/org/webrtc/RTCStatsReport.java",
>+      "api/org/webrtc/RtcCertificatePem.java",
>       "api/org/webrtc/RtpParameters.java",
>       "api/org/webrtc/RtpReceiver.java",
>       "api/org/webrtc/RtpSender.java",
>@@ -1203,6 +1303,16 @@ if (is_android) {
>     jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>   }
> 
>+  generate_jni("generated_builtin_audio_codecs_jni") {
>+    sources = [
>+      "api/org/webrtc/BuiltinAudioDecoderFactoryFactory.java",
>+      "api/org/webrtc/BuiltinAudioEncoderFactoryFactory.java",
>+    ]
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>   # Generated JNI for native API targets
> 
>   generate_jni("generated_native_api_jni") {
>@@ -1255,6 +1365,7 @@ if (is_android) {
> 
>       java_files = [
>         "instrumentationtests/src/org/webrtc/AndroidVideoDecoderInstrumentationTest.java",
>+        "instrumentationtests/src/org/webrtc/BuiltinAudioCodecsFactoryFactoryTest.java",
>         "instrumentationtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java",
>         "instrumentationtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java",
>         "instrumentationtests/src/org/webrtc/Camera2CapturerTest.java",
>@@ -1270,11 +1381,13 @@ if (is_android) {
>         "instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java",
>         "instrumentationtests/src/org/webrtc/PeerConnectionTest.java",
>         "instrumentationtests/src/org/webrtc/RendererCommonTest.java",
>+        "instrumentationtests/src/org/webrtc/RtcCertificatePemTest.java",
>         "instrumentationtests/src/org/webrtc/SurfaceTextureHelperTest.java",
>         "instrumentationtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java",
>         "instrumentationtests/src/org/webrtc/TestConstants.java",
>         "instrumentationtests/src/org/webrtc/VideoFileRendererTest.java",
>         "instrumentationtests/src/org/webrtc/VideoFrameBufferTest.java",
>+        "instrumentationtests/src/org/webrtc/TimestampAlignerTest.java",
>         "instrumentationtests/src/org/webrtc/WebRtcJniBootTest.java",
>         "instrumentationtests/src/org/webrtc/YuvHelperTest.java",
>       ]
>@@ -1294,6 +1407,7 @@ if (is_android) {
>         "//rtc_base:base_java",
>         "//third_party/android_support_test_runner:rules_java",
>         "//third_party/android_support_test_runner:runner_java",
>+        "//third_party/google-truth:google_truth_java",
>         "//third_party/junit",
>       ]
> 
>@@ -1371,23 +1485,23 @@ if (is_android) {
>       ":native_api_video",
>       ":opensles_audio_device_module",
>       ":video_jni",
>-      "../../system_wrappers:system_wrappers",
>-      "//api/audio_codecs:builtin_audio_decoder_factory",
>-      "//api/audio_codecs:builtin_audio_encoder_factory",
>-      "//api/video:video_frame",
>-      "//media:rtc_audio_video",
>-      "//media:rtc_internal_video_codecs",
>-      "//media:rtc_media_base",
>-      "//modules/audio_device:audio_device",
>-      "//modules/audio_device:mock_audio_device",
>-      "//modules/audio_processing:audio_processing",
>-      "//modules/utility:utility",
>-      "//pc:libjingle_peerconnection",
>-      "//rtc_base:checks",
>-      "//rtc_base:rtc_base_approved",
>-      "//test:fileutils",
>-      "//test:test_support",
>-      "//testing/gtest",
>+      "../../api/audio_codecs:builtin_audio_decoder_factory",
>+      "../../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../../api/video:video_frame",
>+      "../../media:rtc_audio_video",
>+      "../../media:rtc_internal_video_codecs",
>+      "../../media:rtc_media_base",
>+      "../../modules/audio_device:audio_device",
>+      "../../modules/audio_device:mock_audio_device",
>+      "../../modules/audio_processing:api",
>+      "../../modules/audio_processing:audio_processing",
>+      "../../modules/utility:utility",
>+      "../../pc:libjingle_peerconnection",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../test:fileutils",
>+      "../../test:test_support",
>+      "../../testing/gtest",
>       "//third_party/abseil-cpp/absl/memory",
>     ]
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/AudioDecoderFactoryFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/AudioDecoderFactoryFactory.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..dd3e262896cab84441e4a7d2b7f838fb59c3ab83
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/AudioDecoderFactoryFactory.java
>@@ -0,0 +1,21 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * Implementations of this interface can create a native {@code webrtc::AudioDecoderFactory}.
>+ */
>+public interface AudioDecoderFactoryFactory {
>+  /**
>+   * Returns a pointer to a {@code webrtc::AudioDecoderFactory}. The caller takes ownership.
>+   */
>+  long createNativeAudioDecoderFactory();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/AudioEncoderFactoryFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/AudioEncoderFactoryFactory.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..814b71aba1f060d66ee0c3a304f84c43eb39e653
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/AudioEncoderFactoryFactory.java
>@@ -0,0 +1,21 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * Implementations of this interface can create a native {@code webrtc::AudioEncoderFactory}.
>+ */
>+public interface AudioEncoderFactoryFactory {
>+  /**
>+   * Returns a pointer to a {@code webrtc::AudioEncoderFactory}. The caller takes ownership.
>+   */
>+  long createNativeAudioEncoderFactory();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/BuiltinAudioDecoderFactoryFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/BuiltinAudioDecoderFactoryFactory.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..5ebc19f25dceea3bfa4efa881158504459ebef2e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/BuiltinAudioDecoderFactoryFactory.java
>@@ -0,0 +1,23 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * Creates a native {@code webrtc::AudioDecoderFactory} with the builtin audio decoders.
>+ */
>+public class BuiltinAudioDecoderFactoryFactory implements AudioDecoderFactoryFactory {
>+  @Override
>+  public long createNativeAudioDecoderFactory() {
>+    return nativeCreateBuiltinAudioDecoderFactory();
>+  }
>+
>+  private static native long nativeCreateBuiltinAudioDecoderFactory();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/BuiltinAudioEncoderFactoryFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/BuiltinAudioEncoderFactoryFactory.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..e884d4c3b98d6cd7d3c2bb5d5a4a01b93b30c2e7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/BuiltinAudioEncoderFactoryFactory.java
>@@ -0,0 +1,23 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * This class creates a native {@code webrtc::AudioEncoderFactory} with the builtin audio encoders.
>+ */
>+public class BuiltinAudioEncoderFactoryFactory implements AudioEncoderFactoryFactory {
>+  @Override
>+  public long createNativeAudioEncoderFactory() {
>+    return nativeCreateBuiltinAudioEncoderFactory();
>+  }
>+
>+  private static native long nativeCreateBuiltinAudioEncoderFactory();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CryptoOptions.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CryptoOptions.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..77b9552aad87a718d621288108618ddf85631e90
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CryptoOptions.java
>@@ -0,0 +1,145 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * CryptoOptions defines advanced cryptographic settings for native WebRTC.
>+ * These settings must be passed into RTCConfiguration. WebRTC is secur by
>+ * default and you should not need to set any of these options unless you are
>+ * specifically looking for an additional crypto feature such as AES_GCM
>+ * support. This class is the Java binding of native api/crypto/cryptooptions.h
>+ */
>+public final class CryptoOptions {
>+  /**
>+   * SRTP Related Peer Connection Options.
>+   */
>+  public final class Srtp {
>+    /**
>+     * Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
>+     * if both sides enable it
>+     */
>+    private final boolean enableGcmCryptoSuites;
>+    /**
>+     * If set to true, the (potentially insecure) crypto cipher
>+     * SRTP_AES128_CM_SHA1_32 will be included in the list of supported ciphers
>+     * during negotiation. It will only be used if both peers support it and no
>+     * other ciphers get preferred.
>+     */
>+    private final boolean enableAes128Sha1_32CryptoCipher;
>+    /**
>+     * If set to true, encrypted RTP header extensions as defined in RFC 6904
>+     * will be negotiated. They will only be used if both peers support them.
>+     */
>+    private final boolean enableEncryptedRtpHeaderExtensions;
>+
>+    private Srtp(boolean enableGcmCryptoSuites, boolean enableAes128Sha1_32CryptoCipher,
>+        boolean enableEncryptedRtpHeaderExtensions) {
>+      this.enableGcmCryptoSuites = enableGcmCryptoSuites;
>+      this.enableAes128Sha1_32CryptoCipher = enableAes128Sha1_32CryptoCipher;
>+      this.enableEncryptedRtpHeaderExtensions = enableEncryptedRtpHeaderExtensions;
>+    }
>+
>+    @CalledByNative("Srtp")
>+    public boolean getEnableGcmCryptoSuites() {
>+      return enableGcmCryptoSuites;
>+    }
>+
>+    @CalledByNative("Srtp")
>+    public boolean getEnableAes128Sha1_32CryptoCipher() {
>+      return enableAes128Sha1_32CryptoCipher;
>+    }
>+
>+    @CalledByNative("Srtp")
>+    public boolean getEnableEncryptedRtpHeaderExtensions() {
>+      return enableEncryptedRtpHeaderExtensions;
>+    }
>+  }
>+
>+  /**
>+   * Options to be used when the FrameEncryptor / FrameDecryptor APIs are used.
>+   */
>+  public final class SFrame {
>+    /**
>+     * If set all RtpSenders must have an FrameEncryptor attached to them before
>+     * they are allowed to send packets. All RtpReceivers must have a
>+     * FrameDecryptor attached to them before they are able to receive packets.
>+     */
>+    private final boolean requireFrameEncryption;
>+
>+    private SFrame(boolean requireFrameEncryption) {
>+      this.requireFrameEncryption = requireFrameEncryption;
>+    }
>+
>+    @CalledByNative("SFrame")
>+    public boolean getRequireFrameEncryption() {
>+      return requireFrameEncryption;
>+    }
>+  }
>+
>+  private final Srtp srtp;
>+  private final SFrame sframe;
>+
>+  private CryptoOptions(boolean enableGcmCryptoSuites, boolean enableAes128Sha1_32CryptoCipher,
>+      boolean enableEncryptedRtpHeaderExtensions, boolean requireFrameEncryption) {
>+    this.srtp = new Srtp(
>+        enableGcmCryptoSuites, enableAes128Sha1_32CryptoCipher, enableEncryptedRtpHeaderExtensions);
>+    this.sframe = new SFrame(requireFrameEncryption);
>+  }
>+
>+  public static Builder builder() {
>+    return new Builder();
>+  }
>+
>+  @CalledByNative
>+  public Srtp getSrtp() {
>+    return srtp;
>+  }
>+
>+  @CalledByNative
>+  public SFrame getSFrame() {
>+    return sframe;
>+  }
>+
>+  public static class Builder {
>+    private boolean enableGcmCryptoSuites;
>+    private boolean enableAes128Sha1_32CryptoCipher;
>+    private boolean enableEncryptedRtpHeaderExtensions;
>+    private boolean requireFrameEncryption;
>+
>+    private Builder() {}
>+
>+    public Builder setEnableGcmCryptoSuites(boolean enableGcmCryptoSuites) {
>+      this.enableGcmCryptoSuites = enableGcmCryptoSuites;
>+      return this;
>+    }
>+
>+    public Builder setEnableAes128Sha1_32CryptoCipher(boolean enableAes128Sha1_32CryptoCipher) {
>+      this.enableAes128Sha1_32CryptoCipher = enableAes128Sha1_32CryptoCipher;
>+      return this;
>+    }
>+
>+    public Builder setEnableEncryptedRtpHeaderExtensions(
>+        boolean enableEncryptedRtpHeaderExtensions) {
>+      this.enableEncryptedRtpHeaderExtensions = enableEncryptedRtpHeaderExtensions;
>+      return this;
>+    }
>+
>+    public Builder setRequireFrameEncryption(boolean requireFrameEncryption) {
>+      this.requireFrameEncryption = requireFrameEncryption;
>+      return this;
>+    }
>+
>+    public CryptoOptions createCryptoOptions() {
>+      return new CryptoOptions(enableGcmCryptoSuites, enableAes128Sha1_32CryptoCipher,
>+          enableEncryptedRtpHeaderExtensions, requireFrameEncryption);
>+    }
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglRenderer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglRenderer.java
>index d0a1d98b2bdefca959dcc4d854c1f69427057c0c..e8100530036183cf33a781dcac2e2d371f9d0249 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglRenderer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglRenderer.java
>@@ -17,6 +17,7 @@ import android.opengl.GLES20;
> import android.os.Handler;
> import android.os.HandlerThread;
> import android.os.Looper;
>+import android.os.Message;
> import android.view.Surface;
> import java.nio.ByteBuffer;
> import java.text.DecimalFormat;
>@@ -79,6 +80,29 @@ public class EglRenderer implements VideoSink {
>     }
>   }
> 
>+  /**
>+   * Handler that triggers a callback when an uncaught exception happens when handling a message.
>+   */
>+  private static class HandlerWithExceptionCallback extends Handler {
>+    private final Runnable exceptionCallback;
>+
>+    public HandlerWithExceptionCallback(Looper looper, Runnable exceptionCallback) {
>+      super(looper);
>+      this.exceptionCallback = exceptionCallback;
>+    }
>+
>+    @Override
>+    public void dispatchMessage(Message msg) {
>+      try {
>+        super.dispatchMessage(msg);
>+      } catch (Exception e) {
>+        Logging.e(TAG, "Exception on EglRenderer thread", e);
>+        exceptionCallback.run();
>+        throw e;
>+      }
>+    }
>+  }
>+
>   protected final String name;
> 
>   // |renderThreadHandler| is a handler for communicating with |renderThread|, and is synchronized
>@@ -101,6 +125,7 @@ public class EglRenderer implements VideoSink {
>   @Nullable private EglBase eglBase;
>   private final VideoFrameDrawer frameDrawer = new VideoFrameDrawer();
>   @Nullable private RendererCommon.GlDrawer drawer;
>+  private boolean usePresentationTimeStamp;
>   private final Matrix drawMatrix = new Matrix();
> 
>   // Pending frame to render. Serves as a queue with size 1. Synchronized on |frameLock|.
>@@ -161,20 +186,31 @@ public class EglRenderer implements VideoSink {
>    * Initialize this class, sharing resources with |sharedContext|. The custom |drawer| will be used
>    * for drawing frames on the EGLSurface. This class is responsible for calling release() on
>    * |drawer|. It is allowed to call init() to reinitialize the renderer after a previous
>-   * init()/release() cycle.
>+   * init()/release() cycle. If usePresentationTimeStamp is true, eglPresentationTimeANDROID will be
>+   * set with the frame timestamps, which specifies desired presentation time and might be useful
>+   * for e.g. syncing audio and video.
>    */
>   public void init(@Nullable final EglBase.Context sharedContext, final int[] configAttributes,
>-      RendererCommon.GlDrawer drawer) {
>+      RendererCommon.GlDrawer drawer, boolean usePresentationTimeStamp) {
>     synchronized (handlerLock) {
>       if (renderThreadHandler != null) {
>         throw new IllegalStateException(name + "Already initialized");
>       }
>       logD("Initializing EglRenderer");
>       this.drawer = drawer;
>+      this.usePresentationTimeStamp = usePresentationTimeStamp;
> 
>       final HandlerThread renderThread = new HandlerThread(name + "EglRenderer");
>       renderThread.start();
>-      renderThreadHandler = new Handler(renderThread.getLooper());
>+      renderThreadHandler =
>+          new HandlerWithExceptionCallback(renderThread.getLooper(), new Runnable() {
>+            @Override
>+            public void run() {
>+              synchronized (handlerLock) {
>+                renderThreadHandler = null;
>+              }
>+            }
>+          });
>       // Create EGL context on the newly created render thread. It should be possibly to create the
>       // context on this thread and make it current on the render thread, but this causes failure on
>       // some Marvel based JB devices. https://bugs.chromium.org/p/webrtc/issues/detail?id=6350.
>@@ -198,6 +234,16 @@ public class EglRenderer implements VideoSink {
>     }
>   }
> 
>+  /**
>+   * Same as above with usePresentationTimeStamp set to false.
>+   *
>+   * @see #init(EglBase.Context, int[], RendererCommon.GlDrawer, boolean)
>+   */
>+  public void init(@Nullable final EglBase.Context sharedContext, final int[] configAttributes,
>+      RendererCommon.GlDrawer drawer) {
>+    init(sharedContext, configAttributes, drawer, /* usePresentationTimeStamp= */ false);
>+  }
>+
>   public void createEglSurface(Surface surface) {
>     createEglSurfaceInternal(surface);
>   }
>@@ -585,7 +631,11 @@ public class EglRenderer implements VideoSink {
>           eglBase.surfaceWidth(), eglBase.surfaceHeight());
> 
>       final long swapBuffersStartTimeNs = System.nanoTime();
>-      eglBase.swapBuffers();
>+      if (usePresentationTimeStamp) {
>+        eglBase.swapBuffers(frame.getTimestampNs());
>+      } else {
>+        eglBase.swapBuffers();
>+      }
> 
>       final long currentTimeNs = System.nanoTime();
>       synchronized (statisticsLock) {
>@@ -652,7 +702,7 @@ public class EglRenderer implements VideoSink {
>   }
> 
>   private String averageTimeAsString(long sumTimeNs, int count) {
>-    return (count <= 0) ? "NA" : TimeUnit.NANOSECONDS.toMicros(sumTimeNs / count) + " Î¼s";
>+    return (count <= 0) ? "NA" : TimeUnit.NANOSECONDS.toMicros(sumTimeNs / count) + " us";
>   }
> 
>   private void logStatistics() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp8Decoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp8Decoder.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..54ad0aa137c7520ebb5909e0ee34ab2da5cf5def
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp8Decoder.java
>@@ -0,0 +1,20 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+public class LibvpxVp8Decoder extends WrappedNativeVideoDecoder {
>+  @Override
>+  public long createNativeVideoDecoder() {
>+    return nativeCreateDecoder();
>+  }
>+
>+  static native long nativeCreateDecoder();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp8Encoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp8Encoder.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..4be9e52c1421f241c4ccc91899be159203e3a681
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp8Encoder.java
>@@ -0,0 +1,25 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+public class LibvpxVp8Encoder extends WrappedNativeVideoEncoder {
>+  @Override
>+  public long createNativeVideoEncoder() {
>+    return nativeCreateEncoder();
>+  }
>+
>+  static native long nativeCreateEncoder();
>+
>+  @Override
>+  public boolean isHardwareEncoder() {
>+    return false;
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp9Decoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp9Decoder.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..90a24433a3ef741cc9b81d45e076ec228bc4db6b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp9Decoder.java
>@@ -0,0 +1,22 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+public class LibvpxVp9Decoder extends WrappedNativeVideoDecoder {
>+  @Override
>+  public long createNativeVideoDecoder() {
>+    return nativeCreateDecoder();
>+  }
>+
>+  static native long nativeCreateDecoder();
>+
>+  static native boolean nativeIsSupported();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp9Encoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp9Encoder.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..1211ae93fb8ac5a179d3f77dabf7e3c9f2685f30
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/LibvpxVp9Encoder.java
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+public class LibvpxVp9Encoder extends WrappedNativeVideoEncoder {
>+  @Override
>+  public long createNativeVideoEncoder() {
>+    return nativeCreateEncoder();
>+  }
>+
>+  static native long nativeCreateEncoder();
>+
>+  @Override
>+  public boolean isHardwareEncoder() {
>+    return false;
>+  }
>+
>+  static native boolean nativeIsSupported();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaTransportFactoryFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaTransportFactoryFactory.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..c16a37a6d777539a460fc2946df1a220affa2365
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaTransportFactoryFactory.java
>@@ -0,0 +1,22 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * Factory for creating webrtc::MediaTransportFactory instances.
>+ */
>+public interface MediaTransportFactoryFactory {
>+  /**
>+   * Dynamically allocates a webrtc::MediaTransportFactory instance and returns a pointer to it.
>+   * The caller takes ownership of the object.
>+   */
>+  public long createNativeMediaTransportFactory();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitor.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitor.java
>index ba6bae0ed83498321656deb8251fd16b5d258751..9764a0cb354ff772aa3a3555479d41ab802ef5fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitor.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitor.java
>@@ -14,11 +14,10 @@ import static org.webrtc.NetworkMonitorAutoDetect.INVALID_NET_ID;
> 
> import android.content.Context;
> import android.os.Build;
>-import javax.annotation.Nullable;
> import java.util.ArrayList;
> import java.util.List;
>-import org.webrtc.NetworkMonitorAutoDetect.ConnectionType;
>-import org.webrtc.NetworkMonitorAutoDetect.NetworkInformation;
>+import javax.annotation.Nullable;
>+import org.webrtc.NetworkMonitorAutoDetect;
> 
> /**
>  * Borrowed from Chromium's
>@@ -33,7 +32,7 @@ public class NetworkMonitor {
>    * Alerted when the connection type of the network changes. The alert is fired on the UI thread.
>    */
>   public interface NetworkObserver {
>-    public void onConnectionTypeChanged(ConnectionType connectionType);
>+    public void onConnectionTypeChanged(NetworkMonitorAutoDetect.ConnectionType connectionType);
>   }
> 
>   private static final String TAG = "NetworkMonitor";
>@@ -55,13 +54,13 @@ public class NetworkMonitor {
>   // Also guarded by autoDetectLock.
>   private int numObservers;
> 
>-  private volatile ConnectionType currentConnectionType;
>+  private volatile NetworkMonitorAutoDetect.ConnectionType currentConnectionType;
> 
>   private NetworkMonitor() {
>     nativeNetworkObservers = new ArrayList<Long>();
>     networkObservers = new ArrayList<NetworkObserver>();
>     numObservers = 0;
>-    currentConnectionType = ConnectionType.CONNECTION_UNKNOWN;
>+    currentConnectionType = NetworkMonitorAutoDetect.ConnectionType.CONNECTION_UNKNOWN;
>   }
> 
>   // TODO(sakal): Remove once downstream dependencies have been updated.
>@@ -155,7 +154,7 @@ public class NetworkMonitor {
>     return Build.VERSION.SDK_INT;
>   }
> 
>-  private ConnectionType getCurrentConnectionType() {
>+  private NetworkMonitorAutoDetect.ConnectionType getCurrentConnectionType() {
>     return currentConnectionType;
>   }
> 
>@@ -169,12 +168,13 @@ public class NetworkMonitor {
>     return new NetworkMonitorAutoDetect(new NetworkMonitorAutoDetect.Observer() {
> 
>       @Override
>-      public void onConnectionTypeChanged(ConnectionType newConnectionType) {
>+      public void onConnectionTypeChanged(
>+          NetworkMonitorAutoDetect.ConnectionType newConnectionType) {
>         updateCurrentConnectionType(newConnectionType);
>       }
> 
>       @Override
>-      public void onNetworkConnect(NetworkInformation networkInfo) {
>+      public void onNetworkConnect(NetworkMonitorAutoDetect.NetworkInformation networkInfo) {
>         notifyObserversOfNetworkConnect(networkInfo);
>       }
> 
>@@ -185,13 +185,15 @@ public class NetworkMonitor {
>     }, appContext);
>   }
> 
>-  private void updateCurrentConnectionType(ConnectionType newConnectionType) {
>+  private void updateCurrentConnectionType(
>+      NetworkMonitorAutoDetect.ConnectionType newConnectionType) {
>     currentConnectionType = newConnectionType;
>     notifyObserversOfConnectionTypeChange(newConnectionType);
>   }
> 
>   /** Alerts all observers of a connection change. */
>-  private void notifyObserversOfConnectionTypeChange(ConnectionType newConnectionType) {
>+  private void notifyObserversOfConnectionTypeChange(
>+      NetworkMonitorAutoDetect.ConnectionType newConnectionType) {
>     List<Long> nativeObservers = getNativeNetworkObserversSync();
>     for (Long nativeObserver : nativeObservers) {
>       nativeNotifyConnectionTypeChanged(nativeObserver);
>@@ -206,7 +208,8 @@ public class NetworkMonitor {
>     }
>   }
> 
>-  private void notifyObserversOfNetworkConnect(NetworkInformation networkInfo) {
>+  private void notifyObserversOfNetworkConnect(
>+      NetworkMonitorAutoDetect.NetworkInformation networkInfo) {
>     List<Long> nativeObservers = getNativeNetworkObserversSync();
>     for (Long nativeObserver : nativeObservers) {
>       nativeNotifyOfNetworkConnect(nativeObserver, networkInfo);
>@@ -221,7 +224,7 @@ public class NetworkMonitor {
>   }
> 
>   private void updateObserverActiveNetworkList(long nativeObserver) {
>-    List<NetworkInformation> networkInfoList;
>+    List<NetworkMonitorAutoDetect.NetworkInformation> networkInfoList;
>     synchronized (autoDetectLock) {
>       networkInfoList = (autoDetect == null) ? null : autoDetect.getActiveNetworkList();
>     }
>@@ -229,7 +232,8 @@ public class NetworkMonitor {
>       return;
>     }
> 
>-    NetworkInformation[] networkInfos = new NetworkInformation[networkInfoList.size()];
>+    NetworkMonitorAutoDetect.NetworkInformation[] networkInfos =
>+        new NetworkMonitorAutoDetect.NetworkInformation[networkInfoList.size()];
>     networkInfos = networkInfoList.toArray(networkInfos);
>     nativeNotifyOfActiveNetworkList(nativeObserver, networkInfos);
>   }
>@@ -274,17 +278,18 @@ public class NetworkMonitor {
> 
>   /** Checks if there currently is connectivity. */
>   public static boolean isOnline() {
>-    ConnectionType connectionType = getInstance().getCurrentConnectionType();
>-    return connectionType != ConnectionType.CONNECTION_NONE;
>+    NetworkMonitorAutoDetect.ConnectionType connectionType =
>+        getInstance().getCurrentConnectionType();
>+    return connectionType != NetworkMonitorAutoDetect.ConnectionType.CONNECTION_NONE;
>   }
> 
>   private native void nativeNotifyConnectionTypeChanged(long nativeAndroidNetworkMonitor);
>   private native void nativeNotifyOfNetworkConnect(
>-      long nativeAndroidNetworkMonitor, NetworkInformation networkInfo);
>+      long nativeAndroidNetworkMonitor, NetworkMonitorAutoDetect.NetworkInformation networkInfo);
>   private native void nativeNotifyOfNetworkDisconnect(
>       long nativeAndroidNetworkMonitor, long networkHandle);
>   private native void nativeNotifyOfActiveNetworkList(
>-      long nativeAndroidNetworkMonitor, NetworkInformation[] networkInfos);
>+      long nativeAndroidNetworkMonitor, NetworkMonitorAutoDetect.NetworkInformation[] networkInfos);
> 
>   // For testing only.
>   @Nullable
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitorAutoDetect.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitorAutoDetect.java
>index 3d74f5fd7c5da4d82c94c2e9cdecdf4a01b9acd1..179f83d3761be1bc508b26ddc3d1be6fc77fd3f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitorAutoDetect.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitorAutoDetect.java
>@@ -297,7 +297,8 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>           // https://android.googlesource.com/platform/frameworks/base/+/d6a7980d
>           NetworkInfo underlyingActiveNetworkInfo = connectivityManager.getActiveNetworkInfo();
>           // We use the NetworkInfo of the underlying network if it is not of TYPE_VPN itself.
>-          if (underlyingActiveNetworkInfo.getType() != ConnectivityManager.TYPE_VPN) {
>+          if (underlyingActiveNetworkInfo != null
>+              && underlyingActiveNetworkInfo.getType() != ConnectivityManager.TYPE_VPN) {
>             return new NetworkState(networkInfo.isConnected(), ConnectivityManager.TYPE_VPN, -1,
>                 underlyingActiveNetworkInfo.getType(), underlyingActiveNetworkInfo.getSubtype());
>           }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnection.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnection.java
>index 84d49ebb8a7dcd412d4096b9c2cb0a172d39b2ca..47c54ee549c4f98a860e34013067dfbe8d968b05 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnection.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnection.java
>@@ -14,6 +14,9 @@ import java.util.ArrayList;
> import java.util.Collections;
> import java.util.List;
> import javax.annotation.Nullable;
>+import org.webrtc.DataChannel;
>+import org.webrtc.MediaStreamTrack;
>+import org.webrtc.RtpTransceiver;
> 
> /**
>  * Java-land version of the PeerConnection APIs; wraps the C++ API
>@@ -50,6 +53,21 @@ public class PeerConnection {
>     }
>   }
> 
>+  /** Tracks PeerConnectionInterface::PeerConnectionState */
>+  public enum PeerConnectionState {
>+    NEW,
>+    CONNECTING,
>+    CONNECTED,
>+    DISCONNECTED,
>+    FAILED,
>+    CLOSED;
>+
>+    @CalledByNative("PeerConnectionState")
>+    static PeerConnectionState fromNativeIndex(int nativeIndex) {
>+      return values()[nativeIndex];
>+    }
>+  }
>+
>   /** Tracks PeerConnectionInterface::TlsCertPolicy */
>   public enum TlsCertPolicy {
>     TLS_CERT_POLICY_SECURE,
>@@ -79,6 +97,10 @@ public class PeerConnection {
>     /** Triggered when the IceConnectionState changes. */
>     @CalledByNative("Observer") void onIceConnectionChange(IceConnectionState newState);
> 
>+    /** Triggered when the PeerConnectionState changes. */
>+    @CalledByNative("Observer")
>+    default void onConnectionChange(PeerConnectionState newState) {}
>+
>     /** Triggered when the ICE connection receiving status changes. */
>     @CalledByNative("Observer") void onIceConnectionReceivingChange(boolean receiving);
> 
>@@ -385,6 +407,7 @@ public class PeerConnection {
>     public IceTransportsType iceTransportsType;
>     public List<IceServer> iceServers;
>     public BundlePolicy bundlePolicy;
>+    @Nullable public RtcCertificatePem certificate;
>     public RtcpMuxPolicy rtcpMuxPolicy;
>     public TcpCandidatePolicy tcpCandidatePolicy;
>     public CandidateNetworkPolicy candidateNetworkPolicy;
>@@ -461,6 +484,25 @@ public class PeerConnection {
>     // every offer/answer negotiation.This is only intended to be a workaround for crbug.com/835958
>     public boolean activeResetSrtpParams;
> 
>+    /*
>+     * Experimental flag that enables a use of media transport. If this is true, the media transport
>+     * factory MUST be provided to the PeerConnectionFactory.
>+     */
>+    public boolean useMediaTransport;
>+
>+    /*
>+     * Experimental flag that enables a use of media transport for data channels. If this is true,
>+     * the media transport factory MUST be provided to the PeerConnectionFactory.
>+     */
>+    public boolean useMediaTransportForDataChannels;
>+
>+    /**
>+     * Defines advanced optional cryptographic settings related to SRTP and
>+     * frame encryption for native WebRTC. Setting this will overwrite any
>+     * options set through the PeerConnectionFactory (which is deprecated).
>+     */
>+    @Nullable public CryptoOptions cryptoOptions;
>+
>     // TODO(deadbeef): Instead of duplicating the defaults here, we should do
>     // something to pick up the defaults from C++. The Objective-C equivalent
>     // of RTCConfiguration does that.
>@@ -500,6 +542,9 @@ public class PeerConnection {
>       networkPreference = AdapterType.UNKNOWN;
>       sdpSemantics = SdpSemantics.PLAN_B;
>       activeResetSrtpParams = false;
>+      useMediaTransport = false;
>+      useMediaTransportForDataChannels = false;
>+      cryptoOptions = null;
>     }
> 
>     @CalledByNative("RTCConfiguration")
>@@ -517,6 +562,12 @@ public class PeerConnection {
>       return bundlePolicy;
>     }
> 
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    RtcCertificatePem getCertificate() {
>+      return certificate;
>+    }
>+
>     @CalledByNative("RTCConfiguration")
>     RtcpMuxPolicy getRtcpMuxPolicy() {
>       return rtcpMuxPolicy;
>@@ -692,6 +743,22 @@ public class PeerConnection {
>     boolean getActiveResetSrtpParams() {
>       return activeResetSrtpParams;
>     }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getUseMediaTransport() {
>+      return useMediaTransport;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getUseMediaTransportForDataChannels() {
>+      return useMediaTransportForDataChannels;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    CryptoOptions getCryptoOptions() {
>+      return cryptoOptions;
>+    }
>   };
> 
>   private final List<MediaStream> localStreams = new ArrayList<>();
>@@ -721,6 +788,10 @@ public class PeerConnection {
>     return nativeGetRemoteDescription();
>   }
> 
>+  public RtcCertificatePem getCertificate() {
>+    return nativeGetCertificate();
>+  }
>+
>   public DataChannel createDataChannel(String label, DataChannel.Init init) {
>     return nativeCreateDataChannel(label, init);
>   }
>@@ -1044,6 +1115,10 @@ public class PeerConnection {
>     return nativeIceConnectionState();
>   }
> 
>+  public PeerConnectionState connectionState() {
>+    return nativeConnectionState();
>+  }
>+
>   public IceGatheringState iceGatheringState() {
>     return nativeIceGatheringState();
>   }
>@@ -1107,6 +1182,7 @@ public class PeerConnection {
>   private native long nativeGetNativePeerConnection();
>   private native SessionDescription nativeGetLocalDescription();
>   private native SessionDescription nativeGetRemoteDescription();
>+  private native RtcCertificatePem nativeGetCertificate();
>   private native DataChannel nativeCreateDataChannel(String label, DataChannel.Init init);
>   private native void nativeCreateOffer(SdpObserver observer, MediaConstraints constraints);
>   private native void nativeCreateAnswer(SdpObserver observer, MediaConstraints constraints);
>@@ -1117,6 +1193,7 @@ public class PeerConnection {
>   private native boolean nativeSetBitrate(Integer min, Integer current, Integer max);
>   private native SignalingState nativeSignalingState();
>   private native IceConnectionState nativeIceConnectionState();
>+  private native PeerConnectionState nativeConnectionState();
>   private native IceGatheringState nativeIceGatheringState();
>   private native void nativeClose();
>   private static native long nativeCreatePeerConnectionObserver(Observer observer);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java
>index eb98e923f6cc856cf8060e5dec67e903b8ec9de0..018fb76e8570c4379c4f31a3da66ef9fb7234b9b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java
>@@ -14,6 +14,7 @@ import android.content.Context;
> import java.util.List;
> import javax.annotation.Nullable;
> import org.webrtc.Logging.Severity;
>+import org.webrtc.PeerConnection;
> import org.webrtc.audio.AudioDeviceModule;
> import org.webrtc.audio.LegacyAudioDeviceModule;
> 
>@@ -122,8 +123,6 @@ public class PeerConnectionFactory {
>     public int networkIgnoreMask;
>     public boolean disableEncryption;
>     public boolean disableNetworkMonitor;
>-    public boolean enableAes128Sha1_32CryptoCipher;
>-    public boolean enableGcmCryptoSuites;
> 
>     @CalledByNative("Options")
>     int getNetworkIgnoreMask() {
>@@ -139,25 +138,20 @@ public class PeerConnectionFactory {
>     boolean getDisableNetworkMonitor() {
>       return disableNetworkMonitor;
>     }
>-
>-    @CalledByNative("Options")
>-    boolean getEnableAes128Sha1_32CryptoCipher() {
>-      return enableAes128Sha1_32CryptoCipher;
>-    }
>-
>-    @CalledByNative("Options")
>-    boolean getEnableGcmCryptoSuites() {
>-      return enableGcmCryptoSuites;
>-    }
>   }
> 
>   public static class Builder {
>-    private @Nullable Options options;
>-    private @Nullable AudioDeviceModule audioDeviceModule = new LegacyAudioDeviceModule();
>-    private @Nullable VideoEncoderFactory encoderFactory;
>-    private @Nullable VideoDecoderFactory decoderFactory;
>-    private @Nullable AudioProcessingFactory audioProcessingFactory;
>-    private @Nullable FecControllerFactoryFactoryInterface fecControllerFactoryFactory;
>+    @Nullable private Options options;
>+    @Nullable private AudioDeviceModule audioDeviceModule = new LegacyAudioDeviceModule();
>+    private AudioEncoderFactoryFactory audioEncoderFactoryFactory =
>+        new BuiltinAudioEncoderFactoryFactory();
>+    private AudioDecoderFactoryFactory audioDecoderFactoryFactory =
>+        new BuiltinAudioDecoderFactoryFactory();
>+    @Nullable private VideoEncoderFactory videoEncoderFactory;
>+    @Nullable private VideoDecoderFactory videoDecoderFactory;
>+    @Nullable private AudioProcessingFactory audioProcessingFactory;
>+    @Nullable private FecControllerFactoryFactoryInterface fecControllerFactoryFactory;
>+    @Nullable private MediaTransportFactoryFactory mediaTransportFactoryFactory;
> 
>     private Builder() {}
> 
>@@ -171,13 +165,33 @@ public class PeerConnectionFactory {
>       return this;
>     }
> 
>-    public Builder setVideoEncoderFactory(VideoEncoderFactory encoderFactory) {
>-      this.encoderFactory = encoderFactory;
>+    public Builder setAudioEncoderFactoryFactory(
>+        AudioEncoderFactoryFactory audioEncoderFactoryFactory) {
>+      if (audioEncoderFactoryFactory == null) {
>+        throw new IllegalArgumentException(
>+            "PeerConnectionFactory.Builder does not accept a null AudioEncoderFactoryFactory.");
>+      }
>+      this.audioEncoderFactoryFactory = audioEncoderFactoryFactory;
>+      return this;
>+    }
>+
>+    public Builder setAudioDecoderFactoryFactory(
>+        AudioDecoderFactoryFactory audioDecoderFactoryFactory) {
>+      if (audioDecoderFactoryFactory == null) {
>+        throw new IllegalArgumentException(
>+            "PeerConnectionFactory.Builder does not accept a null AudioDecoderFactoryFactory.");
>+      }
>+      this.audioDecoderFactoryFactory = audioDecoderFactoryFactory;
>+      return this;
>+    }
>+
>+    public Builder setVideoEncoderFactory(VideoEncoderFactory videoEncoderFactory) {
>+      this.videoEncoderFactory = videoEncoderFactory;
>       return this;
>     }
> 
>-    public Builder setVideoDecoderFactory(VideoDecoderFactory decoderFactory) {
>-      this.decoderFactory = decoderFactory;
>+    public Builder setVideoDecoderFactory(VideoDecoderFactory videoDecoderFactory) {
>+      this.videoDecoderFactory = videoDecoderFactory;
>       return this;
>     }
> 
>@@ -196,9 +210,17 @@ public class PeerConnectionFactory {
>       return this;
>     }
> 
>+    /** Sets a MediaTransportFactoryFactory for a PeerConnectionFactory. */
>+    public Builder setMediaTransportFactoryFactory(
>+        MediaTransportFactoryFactory mediaTransportFactoryFactory) {
>+      this.mediaTransportFactoryFactory = mediaTransportFactoryFactory;
>+      return this;
>+    }
>+
>     public PeerConnectionFactory createPeerConnectionFactory() {
>-      return new PeerConnectionFactory(options, audioDeviceModule, encoderFactory, decoderFactory,
>-          audioProcessingFactory, fecControllerFactoryFactory);
>+      return new PeerConnectionFactory(options, audioDeviceModule, audioEncoderFactoryFactory,
>+          audioDecoderFactoryFactory, videoEncoderFactory, videoDecoderFactory,
>+          audioProcessingFactory, fecControllerFactoryFactory, mediaTransportFactoryFactory);
>     }
>   }
> 
>@@ -277,15 +299,24 @@ public class PeerConnectionFactory {
>   }
> 
>   private PeerConnectionFactory(Options options, @Nullable AudioDeviceModule audioDeviceModule,
>-      @Nullable VideoEncoderFactory encoderFactory, @Nullable VideoDecoderFactory decoderFactory,
>+      AudioEncoderFactoryFactory audioEncoderFactoryFactory,
>+      AudioDecoderFactoryFactory audioDecoderFactoryFactory,
>+      @Nullable VideoEncoderFactory videoEncoderFactory,
>+      @Nullable VideoDecoderFactory videoDecoderFactory,
>       @Nullable AudioProcessingFactory audioProcessingFactory,
>-      @Nullable FecControllerFactoryFactoryInterface fecControllerFactoryFactory) {
>+      @Nullable FecControllerFactoryFactoryInterface fecControllerFactoryFactory,
>+      @Nullable MediaTransportFactoryFactory mediaTransportFactoryFactory) {
>     checkInitializeHasBeenCalled();
>     nativeFactory = nativeCreatePeerConnectionFactory(ContextUtils.getApplicationContext(), options,
>         audioDeviceModule == null ? 0 : audioDeviceModule.getNativeAudioDeviceModulePointer(),
>-        encoderFactory, decoderFactory,
>+        audioEncoderFactoryFactory.createNativeAudioEncoderFactory(),
>+        audioDecoderFactoryFactory.createNativeAudioDecoderFactory(), videoEncoderFactory,
>+        videoDecoderFactory,
>         audioProcessingFactory == null ? 0 : audioProcessingFactory.createNative(),
>-        fecControllerFactoryFactory == null ? 0 : fecControllerFactoryFactory.createNative());
>+        fecControllerFactoryFactory == null ? 0 : fecControllerFactoryFactory.createNative(),
>+        mediaTransportFactoryFactory == null
>+            ? 0
>+            : mediaTransportFactoryFactory.createNativeMediaTransportFactory());
>     if (nativeFactory == 0) {
>       throw new RuntimeException("Failed to initialize PeerConnectionFactory!");
>     }
>@@ -369,9 +400,25 @@ public class PeerConnectionFactory {
>     return new MediaStream(nativeCreateLocalMediaStream(nativeFactory, label));
>   }
> 
>-  public VideoSource createVideoSource(boolean isScreencast) {
>+  /**
>+   * Create video source with given parameters. If alignTimestamps is false, the caller is
>+   * responsible for aligning the frame timestamps to rtc::TimeNanos(). This can be used to achieve
>+   * higher accuracy if there is a big delay between frame creation and frames being delivered to
>+   * the returned video source. If alignTimestamps is true, timestamps will be aligned to
>+   * rtc::TimeNanos() when they arrive to the returned video source.
>+   */
>+  public VideoSource createVideoSource(boolean isScreencast, boolean alignTimestamps) {
>     checkPeerConnectionFactoryExists();
>-    return new VideoSource(nativeCreateVideoSource(nativeFactory, isScreencast));
>+    return new VideoSource(nativeCreateVideoSource(nativeFactory, isScreencast, alignTimestamps));
>+  }
>+
>+  /**
>+   * Same as above with alignTimestamps set to true.
>+   *
>+   * @see #createVideoSource(boolean, boolean)
>+   */
>+  public VideoSource createVideoSource(boolean isScreencast) {
>+    return createVideoSource(isScreencast, /* alignTimestamps= */ true);
>   }
> 
>   public VideoTrack createVideoTrack(String id, VideoSource source) {
>@@ -486,15 +533,18 @@ public class PeerConnectionFactory {
>   private static native void nativeShutdownInternalTracer();
>   private static native boolean nativeStartInternalTracingCapture(String tracingFilename);
>   private static native void nativeStopInternalTracingCapture();
>+
>   private static native long nativeCreatePeerConnectionFactory(Context context, Options options,
>-      long nativeAudioDeviceModule, VideoEncoderFactory encoderFactory,
>-      VideoDecoderFactory decoderFactory, long nativeAudioProcessor,
>-      long nativeFecControllerFactory);
>+      long nativeAudioDeviceModule, long audioEncoderFactory, long audioDecoderFactory,
>+      VideoEncoderFactory encoderFactory, VideoDecoderFactory decoderFactory,
>+      long nativeAudioProcessor, long nativeFecControllerFactory, long mediaTransportFactory);
>+
>   private static native long nativeCreatePeerConnection(long factory,
>       PeerConnection.RTCConfiguration rtcConfig, MediaConstraints constraints, long nativeObserver,
>       SSLCertificateVerifier sslCertificateVerifier);
>   private static native long nativeCreateLocalMediaStream(long factory, String label);
>-  private static native long nativeCreateVideoSource(long factory, boolean is_screencast);
>+  private static native long nativeCreateVideoSource(
>+      long factory, boolean is_screencast, boolean alignTimestamps);
>   private static native long nativeCreateVideoTrack(
>       long factory, String id, long nativeVideoSource);
>   private static native long nativeCreateAudioSource(long factory, MediaConstraints constraints);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtcCertificatePem.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtcCertificatePem.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..6070135b3e12f27ef753596b45f9ce9ddf9415f0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtcCertificatePem.java
>@@ -0,0 +1,75 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import org.webrtc.PeerConnection;
>+
>+/**
>+ * Easily storable/serializable version of a native C++ RTCCertificatePEM.
>+ */
>+public class RtcCertificatePem {
>+  /** PEM string representation of the private key. */
>+  public final String privateKey;
>+  /** PEM string representation of the certificate. */
>+  public final String certificate;
>+  /** Default expiration time of 30 days. */
>+  private static final long DEFAULT_EXPIRY = 60 * 60 * 24 * 30;
>+
>+  /** Instantiate an RtcCertificatePem object from stored strings. */
>+  @CalledByNative
>+  public RtcCertificatePem(String privateKey, String certificate) {
>+    this.privateKey = privateKey;
>+    this.certificate = certificate;
>+  }
>+
>+  @CalledByNative
>+  String getPrivateKey() {
>+    return privateKey;
>+  }
>+
>+  @CalledByNative
>+  String getCertificate() {
>+    return certificate;
>+  }
>+
>+  /**
>+   * Generate a new RtcCertificatePem with the default settings of KeyType = ECDSA and
>+   * expires = 30 days.
>+   */
>+  public static RtcCertificatePem generateCertificate() {
>+    return nativeGenerateCertificate(PeerConnection.KeyType.ECDSA, DEFAULT_EXPIRY);
>+  }
>+
>+  /**
>+   * Generate a new RtcCertificatePem with a custom KeyType and the default setting of
>+   * expires = 30 days.
>+   */
>+  public static RtcCertificatePem generateCertificate(PeerConnection.KeyType keyType) {
>+    return nativeGenerateCertificate(keyType, DEFAULT_EXPIRY);
>+  }
>+
>+  /**
>+   * Generate a new RtcCertificatePem with a custom expires and the default setting of
>+   * KeyType = ECDSA.
>+   */
>+  public static RtcCertificatePem generateCertificate(long expires) {
>+    return nativeGenerateCertificate(PeerConnection.KeyType.ECDSA, expires);
>+  }
>+
>+  /** Generate a new RtcCertificatePem with a custom KeyType and a custom expires. */
>+  public static RtcCertificatePem generateCertificate(
>+      PeerConnection.KeyType keyType, long expires) {
>+    return nativeGenerateCertificate(keyType, expires);
>+  }
>+
>+  private static native RtcCertificatePem nativeGenerateCertificate(
>+      PeerConnection.KeyType keyType, long expires);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpTransceiver.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpTransceiver.java
>index a4a5aa0540581c611b017aaedf69ba0542cd0fff..7f875e6c8ead46d281c3131e96c72ba5549278eb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpTransceiver.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpTransceiver.java
>@@ -13,6 +13,7 @@ package org.webrtc;
> import java.util.ArrayList;
> import java.util.Collections;
> import java.util.List;
>+import org.webrtc.MediaStreamTrack;
> 
> /**
>  * Java wrapper for a C++ RtpTransceiverInterface.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java
>index 86dfb7a2c6f211eca41fbc3a9c41a63ea2007333..f421f16b1a1f8b97bff07e3b9c7380a6b9ce91bb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java
>@@ -27,10 +27,10 @@ public class SoftwareVideoDecoderFactory implements VideoDecoderFactory {
>   @Override
>   public VideoDecoder createDecoder(VideoCodecInfo codecType) {
>     if (codecType.getName().equalsIgnoreCase("VP8")) {
>-      return new VP8Decoder();
>+      return new LibvpxVp8Decoder();
>     }
>-    if (codecType.getName().equalsIgnoreCase("VP9") && VP9Decoder.nativeIsSupported()) {
>-      return new VP9Decoder();
>+    if (codecType.getName().equalsIgnoreCase("VP9") && LibvpxVp9Decoder.nativeIsSupported()) {
>+      return new LibvpxVp9Decoder();
>     }
> 
>     return null;
>@@ -45,7 +45,7 @@ public class SoftwareVideoDecoderFactory implements VideoDecoderFactory {
>     List<VideoCodecInfo> codecs = new ArrayList<VideoCodecInfo>();
> 
>     codecs.add(new VideoCodecInfo("VP8", new HashMap<>()));
>-    if (VP9Decoder.nativeIsSupported()) {
>+    if (LibvpxVp9Decoder.nativeIsSupported()) {
>       codecs.add(new VideoCodecInfo("VP9", new HashMap<>()));
>     }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java
>index a2ddf36c389d7c142939acbd638627fcabf6caeb..3c36c4d56a283313151fd9fedaf88376747a2351 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java
>@@ -20,10 +20,10 @@ public class SoftwareVideoEncoderFactory implements VideoEncoderFactory {
>   @Override
>   public VideoEncoder createEncoder(VideoCodecInfo info) {
>     if (info.name.equalsIgnoreCase("VP8")) {
>-      return new VP8Encoder();
>+      return new LibvpxVp8Encoder();
>     }
>-    if (info.name.equalsIgnoreCase("VP9") && VP9Encoder.nativeIsSupported()) {
>-      return new VP9Encoder();
>+    if (info.name.equalsIgnoreCase("VP9") && LibvpxVp9Encoder.nativeIsSupported()) {
>+      return new LibvpxVp9Encoder();
>     }
> 
>     return null;
>@@ -38,7 +38,7 @@ public class SoftwareVideoEncoderFactory implements VideoEncoderFactory {
>     List<VideoCodecInfo> codecs = new ArrayList<VideoCodecInfo>();
> 
>     codecs.add(new VideoCodecInfo("VP8", new HashMap<>()));
>-    if (VP9Encoder.nativeIsSupported()) {
>+    if (LibvpxVp9Encoder.nativeIsSupported()) {
>       codecs.add(new VideoCodecInfo("VP9", new HashMap<>()));
>     }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java
>index bb67d1f9fe0a62c70109a92482252c4c44de0e0b..29df688901e1626d1f3e046a6d41f99d7486bd32 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java
>@@ -36,10 +36,14 @@ public class SurfaceTextureHelper {
>   /**
>    * Construct a new SurfaceTextureHelper sharing OpenGL resources with |sharedContext|. A dedicated
>    * thread and handler is created for handling the SurfaceTexture. May return null if EGL fails to
>-   * initialize a pixel buffer surface and make it current.
>+   * initialize a pixel buffer surface and make it current. If alignTimestamps is true, the frame
>+   * timestamps will be aligned to rtc::TimeNanos(). If frame timestamps are aligned to
>+   * rtc::TimeNanos() there is no need for aligning timestamps again in
>+   * PeerConnectionFactory.createVideoSource(). This makes the timestamps more accurate and
>+   * closer to actual creation time.
>    */
>   public static SurfaceTextureHelper create(
>-      final String threadName, final EglBase.Context sharedContext) {
>+      final String threadName, final EglBase.Context sharedContext, boolean alignTimestamps) {
>     final HandlerThread thread = new HandlerThread(threadName);
>     thread.start();
>     final Handler handler = new Handler(thread.getLooper());
>@@ -53,7 +57,7 @@ public class SurfaceTextureHelper {
>       @Override
>       public SurfaceTextureHelper call() {
>         try {
>-          return new SurfaceTextureHelper(sharedContext, handler);
>+          return new SurfaceTextureHelper(sharedContext, handler, alignTimestamps);
>         } catch (RuntimeException e) {
>           Logging.e(TAG, threadName + " create failure", e);
>           return null;
>@@ -62,11 +66,22 @@ public class SurfaceTextureHelper {
>     });
>   }
> 
>+  /**
>+   * Same as above with alignTimestamps set to false.
>+   *
>+   * @see #create(String, EglBase.Context, boolean)
>+   */
>+  public static SurfaceTextureHelper create(
>+      final String threadName, final EglBase.Context sharedContext) {
>+    return create(threadName, sharedContext, /* alignTimestamps= */ false);
>+  }
>+
>   private final Handler handler;
>   private final EglBase eglBase;
>   private final SurfaceTexture surfaceTexture;
>   private final int oesTextureId;
>   private final YuvConverter yuvConverter = new YuvConverter();
>+  @Nullable private final TimestampAligner timestampAligner;
> 
>   // These variables are only accessed from the |handler| thread.
>   @Nullable private VideoSink listener;
>@@ -95,11 +110,13 @@ public class SurfaceTextureHelper {
>     }
>   };
> 
>-  private SurfaceTextureHelper(EglBase.Context sharedContext, Handler handler) {
>+  private SurfaceTextureHelper(
>+      EglBase.Context sharedContext, Handler handler, boolean alignTimestamps) {
>     if (handler.getLooper().getThread() != Thread.currentThread()) {
>       throw new IllegalStateException("SurfaceTextureHelper must be created on the handler thread");
>     }
>     this.handler = handler;
>+    this.timestampAligner = alignTimestamps ? new TimestampAligner() : null;
> 
>     eglBase = EglBase.create(sharedContext, EglBase.CONFIG_PIXEL_BUFFER);
>     try {
>@@ -264,7 +281,10 @@ public class SurfaceTextureHelper {
> 
>     final float[] transformMatrix = new float[16];
>     surfaceTexture.getTransformMatrix(transformMatrix);
>-    final long timestampNs = surfaceTexture.getTimestamp();
>+    long timestampNs = surfaceTexture.getTimestamp();
>+    if (timestampAligner != null) {
>+      timestampNs = timestampAligner.translateTimestamp(timestampNs);
>+    }
>     if (textureWidth == 0 || textureHeight == 0) {
>       throw new RuntimeException("Texture size has not been set.");
>     }
>@@ -289,5 +309,8 @@ public class SurfaceTextureHelper {
>     surfaceTexture.release();
>     eglBase.release();
>     handler.getLooper().quit();
>+    if (timestampAligner != null) {
>+      timestampAligner.dispose();
>+    }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/TimestampAligner.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/TimestampAligner.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..7c4bed4d8fba4484d5d1ea5e84a0a1a28f29954f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/TimestampAligner.java
>@@ -0,0 +1,59 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * The TimestampAligner class helps translating camera timestamps into the same timescale as is
>+ * used by rtc::TimeNanos(). Some cameras have built in timestamping which is more accurate than
>+ * reading the system clock, but using a different epoch and unknown clock drift. Frame timestamps
>+ * in webrtc should use rtc::TimeNanos (system monotonic time), and this class provides a filter
>+ * which lets us use the rtc::TimeNanos timescale, and at the same time take advantage of higher
>+ * accuracy of the camera clock. This class is a wrapper on top of rtc::TimestampAligner.
>+ */
>+public class TimestampAligner {
>+  /**
>+   * Wrapper around rtc::TimeNanos(). This is normally same as System.nanoTime(), but call this
>+   * function to be safe.
>+   */
>+  public static long getRtcTimeNanos() {
>+    return nativeRtcTimeNanos();
>+  }
>+
>+  private volatile long nativeTimestampAligner = nativeCreateTimestampAligner();
>+
>+  /**
>+   * Translates camera timestamps to the same timescale as is used by rtc::TimeNanos().
>+   * |cameraTimeNs| is assumed to be accurate, but with an unknown epoch and clock drift. Returns
>+   * the translated timestamp.
>+   */
>+  public long translateTimestamp(long cameraTimeNs) {
>+    checkNativeAlignerExists();
>+    return nativeTranslateTimestamp(nativeTimestampAligner, cameraTimeNs);
>+  }
>+
>+  /** Dispose native timestamp aligner. */
>+  public void dispose() {
>+    checkNativeAlignerExists();
>+    nativeReleaseTimestampAligner(nativeTimestampAligner);
>+    nativeTimestampAligner = 0;
>+  }
>+
>+  private void checkNativeAlignerExists() {
>+    if (nativeTimestampAligner == 0) {
>+      throw new IllegalStateException("TimestampAligner has been disposed.");
>+    }
>+  }
>+
>+  private static native long nativeRtcTimeNanos();
>+  private static native long nativeCreateTimestampAligner();
>+  private static native void nativeReleaseTimestampAligner(long timestampAligner);
>+  private static native long nativeTranslateTimestamp(long timestampAligner, long cameraTimeNs);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoder.java
>index 8dc7582926b243ab9d60d228e66b5c036cfafe2b..afc5cd0239d1a5a2e2082ca91dd7c8a34f58e0c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoder.java
>@@ -210,14 +210,6 @@ public interface VideoEncoder {
>    */
>   @CalledByNative VideoCodecStatus encode(VideoFrame frame, EncodeInfo info);
> 
>-  /**
>-   * Informs the encoder of the packet loss and the round-trip time of the network.
>-   *
>-   * @param packetLoss How many packets are lost on average per 255 packets.
>-   * @param roundTripTimeMs Round-trip time of the network in milliseconds.
>-   */
>-  @CalledByNative VideoCodecStatus setChannelParameters(short packetLoss, long roundTripTimeMs);
>-
>   /** Sets the bitrate allocation and the target framerate for the encoder. */
>   @CalledByNative VideoCodecStatus setRateAllocation(BitrateAllocation allocation, int framerate);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoSource.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoSource.java
>index 7f9f3e5323ff530f83f171b57b97fe41fa33cac7..a8ef6620acfcb4a43c9bcdef6afd284ed9d90875 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoSource.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoSource.java
>@@ -30,7 +30,20 @@ public class VideoSource extends MediaSource {
>    * maintain the input orientation, so it doesn't matter if e.g. 1280x720 or 720x1280 is requested.
>    */
>   public void adaptOutputFormat(int width, int height, int fps) {
>-    nativeAdaptOutputFormat(getNativeVideoTrackSource(), width, height, fps);
>+    final int maxSide = Math.max(width, height);
>+    final int minSide = Math.min(width, height);
>+    adaptOutputFormat(maxSide, minSide, minSide, maxSide, fps);
>+  }
>+
>+  /**
>+   * Same as above, but allows setting two different target resolutions depending on incoming
>+   * frame orientation. This gives more fine-grained control and can e.g. be used to force landscape
>+   * video to be cropped to portrait video.
>+   */
>+  public void adaptOutputFormat(
>+      int landscapeWidth, int landscapeHeight, int portraitWidth, int portraitHeight, int fps) {
>+    nativeAdaptOutputFormat(getNativeVideoTrackSource(), landscapeWidth, landscapeHeight,
>+        portraitWidth, portraitHeight, fps);
>   }
> 
>   public CapturerObserver getCapturerObserver() {
>@@ -44,5 +57,6 @@ public class VideoSource extends MediaSource {
> 
>   // Returns source->internal() from webrtc::VideoTrackSourceProxy.
>   private static native long nativeGetInternalSource(long source);
>-  private static native void nativeAdaptOutputFormat(long source, int width, int height, int fps);
>+  private static native void nativeAdaptOutputFormat(long source, int landscapeWidth,
>+      int landscapeHeight, int portraitWidth, int portraitHeight, int fps);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/BuiltinAudioCodecsFactoryFactoryTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/BuiltinAudioCodecsFactoryFactoryTest.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..36ee0e932e1cc202afddd1d9ba7fe34ee5775179
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/BuiltinAudioCodecsFactoryFactoryTest.java
>@@ -0,0 +1,58 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import static com.google.common.truth.Truth.assertThat;
>+
>+import android.support.test.filters.SmallTest;
>+
>+import org.junit.Before;
>+import org.junit.Test;
>+import org.junit.runner.RunWith;
>+import org.junit.runners.JUnit4;
>+
>+@RunWith(JUnit4.class)
>+public final class BuiltinAudioCodecsFactoryFactoryTest {
>+  @Before
>+  public void setUp() {
>+    System.loadLibrary(TestConstants.NATIVE_LIBRARY);
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testAudioEncoderFactoryFactoryTest() throws Exception {
>+    BuiltinAudioEncoderFactoryFactory factory = new BuiltinAudioEncoderFactoryFactory();
>+    long aef = 0;
>+    try {
>+      aef = factory.createNativeAudioEncoderFactory();
>+      assertThat(aef).isNotEqualTo(0);
>+    } finally {
>+      if (aef != 0) {
>+        JniCommon.nativeReleaseRef(aef);
>+      }
>+    }
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testAudioDecoderFactoryFactoryTest() throws Exception {
>+    BuiltinAudioDecoderFactoryFactory factory = new BuiltinAudioDecoderFactoryFactory();
>+    long adf = 0;
>+    try {
>+      adf = factory.createNativeAudioDecoderFactory();
>+      assertThat(adf).isNotEqualTo(0);
>+    } finally {
>+      if (adf != 0) {
>+        JniCommon.nativeReleaseRef(adf);
>+      }
>+    }
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java
>index ef7669102f9bb648b6db15421104c9b066e9b35d..17b0977d595fd27aff58661ef9aa7d1cc7d8759f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java
>@@ -169,6 +169,14 @@ public class Camera1CapturerUsingByteBufferTest {
>     fixtures.scaleCameraOutput();
>   }
> 
>+  // This test that frames forwarded to a renderer is cropped to a new orientation if
>+  // adaptOutputFormat is called in such a way. This test both Java and C++ parts of of the stack.
>+  @Test
>+  @MediumTest
>+  public void testCropCameraOutput() throws InterruptedException {
>+    fixtures.cropCameraOutput();
>+  }
>+
>   // This test that an error is reported if the camera is already opened
>   // when CameraVideoCapturer is started.
>   @Test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java
>index 28fa825ad74e61cc9406b32d24b9b6dbe60c8855..4dc003726a7196c626a47560f94918c4151155fb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java
>@@ -172,6 +172,14 @@ public class Camera1CapturerUsingTextureTest {
>     fixtures.scaleCameraOutput();
>   }
> 
>+  // This test that frames forwarded to a renderer is cropped to a new orientation if
>+  // adaptOutputFormat is called in such a way. This test both Java and C++ parts of of the stack.
>+  @Test
>+  @MediumTest
>+  public void testCropCameraOutput() throws InterruptedException {
>+    fixtures.cropCameraOutput();
>+  }
>+
>   // This test that an error is reported if the camera is already opened
>   // when CameraVideoCapturer is started.
>   @Test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera2CapturerTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera2CapturerTest.java
>index 77e8d77f0aef1d6568b1c7ada534dcb8c6411bc1..fba943176a8586c28ecf4918713922b4bfa4f44b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera2CapturerTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera2CapturerTest.java
>@@ -302,6 +302,14 @@ public class Camera2CapturerTest {
>     fixtures.scaleCameraOutput();
>   }
> 
>+  // This test that frames forwarded to a renderer is cropped to a new orientation if
>+  // adaptOutputFormat is called in such a way. This test both Java and C++ parts of of the stack.
>+  @Test
>+  @MediumTest
>+  public void testCropCameraOutput() throws InterruptedException {
>+    fixtures.cropCameraOutput();
>+  }
>+
>   // This test that an error is reported if the camera is already opened
>   // when CameraVideoCapturer is started.
>   @Test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java
>index dd0d780df08c7196260261ed182d9b83cc9e5b6d..b47fee08d15796f2349896123dca0ac43c3beeae 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java
>@@ -688,6 +688,49 @@ class CameraVideoCapturerTestFixtures {
>     assertTrue(gotExpectedResolution);
>   }
> 
>+  public void cropCameraOutput() throws InterruptedException {
>+    final CapturerInstance capturerInstance = createCapturer(false /* initialize */);
>+    final VideoTrackWithRenderer videoTrackWithRenderer =
>+        createVideoTrackWithRenderer(capturerInstance.capturer);
>+    assertTrue(videoTrackWithRenderer.rendererCallbacks.waitForNextFrameToRender() > 0);
>+
>+    final int startWidth = videoTrackWithRenderer.rendererCallbacks.frameWidth();
>+    final int startHeight = videoTrackWithRenderer.rendererCallbacks.frameHeight();
>+    final int frameRate = 30;
>+    final int cropWidth;
>+    final int cropHeight;
>+    if (startWidth > startHeight) {
>+      // Landscape input, request portrait output.
>+      cropWidth = 360;
>+      cropHeight = 640;
>+    } else {
>+      // Portrait input, request landscape output.
>+      cropWidth = 640;
>+      cropHeight = 630;
>+    }
>+
>+    // Request different output orientation than input.
>+    videoTrackWithRenderer.source.adaptOutputFormat(
>+        cropWidth, cropHeight, cropWidth, cropHeight, frameRate);
>+
>+    boolean gotExpectedOrientation = false;
>+    int numberOfInspectedFrames = 0;
>+
>+    do {
>+      videoTrackWithRenderer.rendererCallbacks.waitForNextFrameToRender();
>+      ++numberOfInspectedFrames;
>+
>+      gotExpectedOrientation = (cropWidth > cropHeight)
>+          == (videoTrackWithRenderer.rendererCallbacks.frameWidth()
>+                 > videoTrackWithRenderer.rendererCallbacks.frameHeight());
>+    } while (!gotExpectedOrientation && numberOfInspectedFrames < 30);
>+
>+    disposeCapturer(capturerInstance);
>+    disposeVideoTrackWithRenderer(videoTrackWithRenderer);
>+
>+    assertTrue(gotExpectedOrientation);
>+  }
>+
>   public void startWhileCameraIsAlreadyOpen() throws InterruptedException {
>     final String cameraName = testObjectFactory.getNameOfBackFacingDevice();
>     // At this point camera is not actually opened.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java
>index 7d0edc598e6c668c90b4db2259163b619c76d451..2063199e72dea8507326908e65acbf27e25b2abd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java
>@@ -11,6 +11,7 @@
> package org.webrtc;
> 
> import static org.junit.Assert.assertEquals;
>+import static org.junit.Assert.assertNotEquals;
> import static org.junit.Assert.assertFalse;
> import static org.junit.Assert.assertNotNull;
> import static org.junit.Assert.assertNull;
>@@ -44,6 +45,7 @@ import org.junit.runner.RunWith;
> import org.webrtc.Metrics.HistogramInfo;
> import org.webrtc.PeerConnection.IceConnectionState;
> import org.webrtc.PeerConnection.IceGatheringState;
>+import org.webrtc.PeerConnection.PeerConnectionState;
> import org.webrtc.PeerConnection.SignalingState;
> 
> /** End-to-end tests for PeerConnection.java. */
>@@ -73,6 +75,7 @@ public class PeerConnectionTest {
>     private int expectedTracksAdded;
>     private Queue<SignalingState> expectedSignalingChanges = new ArrayDeque<>();
>     private Queue<IceConnectionState> expectedIceConnectionChanges = new ArrayDeque<>();
>+    private Queue<PeerConnectionState> expectedConnectionChanges = new ArrayDeque<>();
>     private Queue<IceGatheringState> expectedIceGatheringChanges = new ArrayDeque<>();
>     private Queue<String> expectedAddStreamLabels = new ArrayDeque<>();
>     private Queue<String> expectedRemoveStreamLabels = new ArrayDeque<>();
>@@ -187,11 +190,29 @@ public class PeerConnectionTest {
>       assertEquals(expectedIceConnectionChanges.remove(), newState);
>     }
> 
>+    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>+    @SuppressWarnings("NoSynchronizedMethodCheck")
>+    public synchronized void expectConnectionChange(PeerConnectionState newState) {
>+      expectedConnectionChanges.add(newState);
>+    }
>+
>+    @Override
>+    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>+    @SuppressWarnings("NoSynchronizedMethodCheck")
>+    public synchronized void onConnectionChange(PeerConnectionState newState) {
>+      if (expectedConnectionChanges.isEmpty()) {
>+        System.out.println(name + " got an unexpected DTLS connection change " + newState);
>+        return;
>+      }
>+
>+      assertEquals(expectedConnectionChanges.remove(), newState);
>+    }
>+
>     @Override
>     // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>     @SuppressWarnings("NoSynchronizedMethodCheck")
>     public synchronized void onIceConnectionReceivingChange(boolean receiving) {
>-      System.out.println(name + "Got an ICE connection receiving change " + receiving);
>+      System.out.println(name + " got an ICE connection receiving change " + receiving);
>     }
> 
>     // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>@@ -663,6 +684,45 @@ public class PeerConnectionTest {
>     assertNotNull(offeringPC);
>   }
> 
>+  @Test
>+  @SmallTest
>+  public void testCreationWithCertificate() throws Exception {
>+    PeerConnectionFactory factory = PeerConnectionFactory.builder().createPeerConnectionFactory();
>+    PeerConnection.RTCConfiguration config = new PeerConnection.RTCConfiguration(Arrays.asList());
>+
>+    // Test certificate.
>+    RtcCertificatePem originalCert = RtcCertificatePem.generateCertificate();
>+    config.certificate = originalCert;
>+
>+    ObserverExpectations offeringExpectations = new ObserverExpectations("PCTest:offerer");
>+    PeerConnection offeringPC = factory.createPeerConnection(config, offeringExpectations);
>+
>+    RtcCertificatePem restoredCert = offeringPC.getCertificate();
>+    assertEquals(originalCert.privateKey, restoredCert.privateKey);
>+    assertEquals(originalCert.certificate, restoredCert.certificate);
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testCreationWithCryptoOptions() throws Exception {
>+    PeerConnectionFactory factory = PeerConnectionFactory.builder().createPeerConnectionFactory();
>+    PeerConnection.RTCConfiguration config = new PeerConnection.RTCConfiguration(Arrays.asList());
>+
>+    assertNull(config.cryptoOptions);
>+
>+    CryptoOptions cryptoOptions = CryptoOptions.builder()
>+                                      .setEnableGcmCryptoSuites(true)
>+                                      .setEnableAes128Sha1_32CryptoCipher(true)
>+                                      .setEnableEncryptedRtpHeaderExtensions(true)
>+                                      .setRequireFrameEncryption(true)
>+                                      .createCryptoOptions();
>+    config.cryptoOptions = cryptoOptions;
>+
>+    ObserverExpectations offeringExpectations = new ObserverExpectations("PCTest:offerer");
>+    PeerConnection offeringPC = factory.createPeerConnection(config, offeringExpectations);
>+    assertNotNull(offeringPC);
>+  }
>+
>   @Test
>   @MediumTest
>   public void testCompleteSession() throws Exception {
>@@ -759,12 +819,14 @@ public class PeerConnectionTest {
> 
>     sdpLatch = new SdpObserverLatch();
>     answeringExpectations.expectSignalingChange(SignalingState.STABLE);
>+    answeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTING);
>     answeringPC.setLocalDescription(sdpLatch, answerSdp);
>     assertTrue(sdpLatch.await());
>     assertNull(sdpLatch.getSdp());
> 
>     sdpLatch = new SdpObserverLatch();
>     offeringExpectations.expectSignalingChange(SignalingState.HAVE_LOCAL_OFFER);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTING);
>     offeringPC.setLocalDescription(sdpLatch, offerSdp);
>     assertTrue(sdpLatch.await());
>     assertNull(sdpLatch.getSdp());
>@@ -774,6 +836,9 @@ public class PeerConnectionTest {
> 
>     offeringExpectations.expectIceConnectionChange(IceConnectionState.CHECKING);
>     offeringExpectations.expectIceConnectionChange(IceConnectionState.CONNECTED);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.NEW);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTING);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTED);
>     // TODO(bemasc): uncomment once delivery of ICECompleted is reliable
>     // (https://code.google.com/p/webrtc/issues/detail?id=3021).
>     //
>@@ -781,6 +846,7 @@ public class PeerConnectionTest {
>     //     IceConnectionState.COMPLETED);
>     answeringExpectations.expectIceConnectionChange(IceConnectionState.CHECKING);
>     answeringExpectations.expectIceConnectionChange(IceConnectionState.CONNECTED);
>+    answeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTED);
> 
>     offeringPC.setRemoteDescription(sdpLatch, answerSdp);
>     assertTrue(sdpLatch.await());
>@@ -989,12 +1055,14 @@ public class PeerConnectionTest {
> 
>     sdpLatch = new SdpObserverLatch();
>     answeringExpectations.expectSignalingChange(SignalingState.STABLE);
>+    answeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTING);
>     answeringPC.setLocalDescription(sdpLatch, answerSdp);
>     assertTrue(sdpLatch.await());
>     assertNull(sdpLatch.getSdp());
> 
>     sdpLatch = new SdpObserverLatch();
>     offeringExpectations.expectSignalingChange(SignalingState.HAVE_LOCAL_OFFER);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTING);
>     offeringPC.setLocalDescription(sdpLatch, offerSdp);
>     assertTrue(sdpLatch.await());
>     assertNull(sdpLatch.getSdp());
>@@ -1003,10 +1071,14 @@ public class PeerConnectionTest {
> 
>     offeringExpectations.expectIceConnectionChange(IceConnectionState.CHECKING);
>     offeringExpectations.expectIceConnectionChange(IceConnectionState.CONNECTED);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.NEW);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTING);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTED);
>     // TODO(bemasc): uncomment once delivery of ICECompleted is reliable
>     // (https://code.google.com/p/webrtc/issues/detail?id=3021).
>     answeringExpectations.expectIceConnectionChange(IceConnectionState.CHECKING);
>     answeringExpectations.expectIceConnectionChange(IceConnectionState.CONNECTED);
>+    answeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTED);
> 
>     offeringPC.setRemoteDescription(sdpLatch, answerSdp);
>     assertTrue(sdpLatch.await());
>@@ -1139,6 +1211,7 @@ public class PeerConnectionTest {
>     offeringExpectations.expectSignalingChange(SignalingState.HAVE_LOCAL_OFFER);
>     offeringExpectations.expectIceCandidates(2);
>     offeringExpectations.expectIceGatheringChange(IceGatheringState.COMPLETE);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTING);
>     offeringPC.setLocalDescription(sdpLatch, offerSdp);
>     assertTrue(sdpLatch.await());
>     assertNull(sdpLatch.getSdp());
>@@ -1170,6 +1243,7 @@ public class PeerConnectionTest {
>     answeringExpectations.expectSignalingChange(SignalingState.STABLE);
>     answeringExpectations.expectIceCandidates(2);
>     answeringExpectations.expectIceGatheringChange(IceGatheringState.COMPLETE);
>+    answeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTING);
>     answeringPC.setLocalDescription(sdpLatch, answerSdp);
>     assertTrue(sdpLatch.await());
>     assertNull(sdpLatch.getSdp());
>@@ -1181,6 +1255,9 @@ public class PeerConnectionTest {
> 
>     offeringExpectations.expectIceConnectionChange(IceConnectionState.CHECKING);
>     offeringExpectations.expectIceConnectionChange(IceConnectionState.CONNECTED);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.NEW);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTING);
>+    offeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTED);
>     // TODO(bemasc): uncomment once delivery of ICECompleted is reliable
>     // (https://code.google.com/p/webrtc/issues/detail?id=3021).
>     //
>@@ -1188,6 +1265,7 @@ public class PeerConnectionTest {
>     //     IceConnectionState.COMPLETED);
>     answeringExpectations.expectIceConnectionChange(IceConnectionState.CHECKING);
>     answeringExpectations.expectIceConnectionChange(IceConnectionState.CONNECTED);
>+    answeringExpectations.expectConnectionChange(PeerConnectionState.CONNECTED);
> 
>     offeringPC.setRemoteDescription(sdpLatch, answerSdp);
>     assertTrue(sdpLatch.await());
>@@ -1573,6 +1651,7 @@ public class PeerConnectionTest {
>     assertTrue(expectations.waitForAllExpectationsToBeSatisfied(TIMEOUT_SECONDS));
> 
>     expectations.expectIceConnectionChange(IceConnectionState.CLOSED);
>+    expectations.expectConnectionChange(PeerConnectionState.CLOSED);
>     expectations.expectSignalingChange(SignalingState.CLOSED);
>     pc.close();
>     assertTrue(expectations.waitForAllExpectationsToBeSatisfied(TIMEOUT_SECONDS));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/RtcCertificatePemTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/RtcCertificatePemTest.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..00f295c942f57ff4f8bd3badc67c1287d9130db5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/RtcCertificatePemTest.java
>@@ -0,0 +1,73 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import static com.google.common.truth.Truth.assertThat;
>+
>+import android.support.test.filters.SmallTest;
>+import org.chromium.base.test.BaseJUnit4ClassRunner;
>+import org.junit.Before;
>+import org.junit.Test;
>+import org.junit.runner.RunWith;
>+import org.webrtc.PeerConnection;
>+import org.webrtc.RtcCertificatePem;
>+
>+/** Tests for RtcCertificatePem.java. */
>+@RunWith(BaseJUnit4ClassRunner.class)
>+public class RtcCertificatePemTest {
>+  @Before
>+  public void setUp() {
>+    System.loadLibrary(TestConstants.NATIVE_LIBRARY);
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testConstructor() {
>+    RtcCertificatePem original = RtcCertificatePem.generateCertificate();
>+    RtcCertificatePem recreated = new RtcCertificatePem(original.privateKey, original.certificate);
>+    assertThat(original.privateKey).isEqualTo(recreated.privateKey);
>+    assertThat(original.certificate).isEqualTo(recreated.certificate);
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testGenerateCertificateDefaults() {
>+    RtcCertificatePem rtcCertificate = RtcCertificatePem.generateCertificate();
>+    assertThat(rtcCertificate.privateKey).isNotEmpty();
>+    assertThat(rtcCertificate.certificate).isNotEmpty();
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testGenerateCertificateCustomKeyTypeDefaultExpires() {
>+    RtcCertificatePem rtcCertificate =
>+        RtcCertificatePem.generateCertificate(PeerConnection.KeyType.RSA);
>+    assertThat(rtcCertificate.privateKey).isNotEmpty();
>+    assertThat(rtcCertificate.certificate).isNotEmpty();
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testGenerateCertificateCustomExpiresDefaultKeyType() {
>+    RtcCertificatePem rtcCertificate = RtcCertificatePem.generateCertificate(60 * 60 * 24);
>+    assertThat(rtcCertificate.privateKey).isNotEmpty();
>+    assertThat(rtcCertificate.certificate).isNotEmpty();
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testGenerateCertificateCustomKeyTypeAndExpires() {
>+    RtcCertificatePem rtcCertificate =
>+        RtcCertificatePem.generateCertificate(PeerConnection.KeyType.RSA, 60 * 60 * 24);
>+    assertThat(rtcCertificate.privateKey).isNotEmpty();
>+    assertThat(rtcCertificate.certificate).isNotEmpty();
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/TimestampAlignerTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/TimestampAlignerTest.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..1d944b57f9d654d89db007e48407a6180c67b4a6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/TimestampAlignerTest.java
>@@ -0,0 +1,46 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import android.support.test.filters.SmallTest;
>+import org.chromium.base.test.params.BaseJUnit4RunnerDelegate;
>+import org.chromium.base.test.params.ParameterAnnotations.UseRunnerDelegate;
>+import org.junit.BeforeClass;
>+import org.junit.Test;
>+
>+@UseRunnerDelegate(BaseJUnit4RunnerDelegate.class)
>+public class TimestampAlignerTest {
>+  @BeforeClass
>+  public static void setUp() {
>+    System.loadLibrary(TestConstants.NATIVE_LIBRARY);
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testGetRtcTimeNanos() {
>+    TimestampAligner.getRtcTimeNanos();
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testDispose() {
>+    final TimestampAligner timestampAligner = new TimestampAligner();
>+    timestampAligner.dispose();
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testTranslateTimestamp() {
>+    final TimestampAligner timestampAligner = new TimestampAligner();
>+    timestampAligner.translateTimestamp(/* cameraTimeNs= */ 123);
>+    timestampAligner.dispose();
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.cc
>index 4c302da0ed9eb0b38c311f013f6443e4ff692236..aae1de7ff0342a0eb5ccc715dc7faea7c4e4d9a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.cc
>@@ -25,12 +25,14 @@ class JavaVideoTrackSourceImpl : public JavaVideoTrackSourceInterface {
>  public:
>   JavaVideoTrackSourceImpl(JNIEnv* env,
>                            rtc::Thread* signaling_thread,
>-                           bool is_screencast)
>+                           bool is_screencast,
>+                           bool align_timestamps)
>       : android_video_track_source_(
>             new rtc::RefCountedObject<jni::AndroidVideoTrackSource>(
>                 signaling_thread,
>                 env,
>-                is_screencast)),
>+                is_screencast,
>+                align_timestamps)),
>         native_capturer_observer_(jni::CreateJavaNativeCapturerObserver(
>             env,
>             android_video_track_source_)) {}
>@@ -96,9 +98,10 @@ class JavaVideoTrackSourceImpl : public JavaVideoTrackSourceInterface {
> rtc::scoped_refptr<JavaVideoTrackSourceInterface> CreateJavaVideoSource(
>     JNIEnv* jni,
>     rtc::Thread* signaling_thread,
>-    bool is_screencast) {
>+    bool is_screencast,
>+    bool align_timestamps) {
>   return new rtc::RefCountedObject<JavaVideoTrackSourceImpl>(
>-      jni, signaling_thread, is_screencast);
>+      jni, signaling_thread, is_screencast, align_timestamps);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.h
>index 0c4f33342ebad6d4b5979daf3caee97603b06f87..75ebdbce30263e3085540414f1283f610361a01b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.h
>@@ -14,6 +14,7 @@
> #include <jni.h>
> 
> #include "api/mediastreaminterface.h"
>+#include "rtc_base/thread.h"
> #include "sdk/android/native_api/jni/scoped_java_ref.h"
> 
> namespace webrtc {
>@@ -32,7 +33,8 @@ class JavaVideoTrackSourceInterface : public VideoTrackSourceInterface {
> rtc::scoped_refptr<JavaVideoTrackSourceInterface> CreateJavaVideoSource(
>     JNIEnv* env,
>     rtc::Thread* signaling_thread,
>-    bool is_screencast);
>+    bool is_screencast,
>+    bool align_timestamps);
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/audio_device/audio_device_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
>index 5e46052b90048156bf1c3d698a2513684df8d369..98d585387a04d8b66467fe2eacee460fd97b2ede 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
>@@ -15,6 +15,7 @@
> #include "modules/audio_device/include/mock_audio_transport.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/format_macros.h"
> #include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/timeutils.h"
>@@ -25,7 +26,6 @@
> #include "sdk/android/src/jni/audio_device/audio_device_module.h"
> #include "sdk/android/src/jni/audio_device/opensles_common.h"
> #include "sdk/android/src/jni/jni_helpers.h"
>-#include "system_wrappers/include/event_wrapper.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>@@ -364,7 +364,7 @@ class MockAudioTransportAndroid : public test::MockAudioTransport {
> 
>   // Set default actions of the mock object. We are delegating to fake
>   // implementations (of AudioStreamInterface) here.
>-  void HandleCallbacks(EventWrapper* test_is_done,
>+  void HandleCallbacks(rtc::Event* test_is_done,
>                        AudioStreamInterface* audio_stream,
>                        int num_callbacks) {
>     test_is_done_ = test_is_done;
>@@ -447,7 +447,7 @@ class MockAudioTransportAndroid : public test::MockAudioTransport {
>   bool rec_mode() const { return type_ & kRecording; }
> 
>  private:
>-  EventWrapper* test_is_done_;
>+  rtc::Event* test_is_done_;
>   size_t num_callbacks_;
>   int type_;
>   size_t play_count_;
>@@ -459,7 +459,7 @@ class MockAudioTransportAndroid : public test::MockAudioTransport {
> // AudioDeviceTest test fixture.
> class AudioDeviceTest : public ::testing::Test {
>  protected:
>-  AudioDeviceTest() : test_is_done_(EventWrapper::Create()) {
>+  AudioDeviceTest() {
>     // One-time initialization of JVM and application context. Ensures that we
>     // can do calls between C++ and Java. Initializes both Java and OpenSL ES
>     // implementations.
>@@ -683,7 +683,7 @@ class AudioDeviceTest : public ::testing::Test {
> 
>   JNIEnv* jni_;
>   std::unique_ptr<JavaParamRef<jobject>> context_;
>-  std::unique_ptr<EventWrapper> test_is_done_;
>+  rtc::Event test_is_done_;
>   rtc::scoped_refptr<AudioDeviceModule> audio_device_;
>   ScopedJavaLocalRef<jobject> audio_manager_;
>   AudioParameters output_parameters_;
>@@ -880,14 +880,14 @@ TEST_F(AudioDeviceTest, StopRecordingRequiresInitToRestart) {
> // audio samples to play out using the NeedMorePlayData callback.
> TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
>   MockAudioTransportAndroid mock(kPlayout);
>-  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
>   EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
>                                      kBytesPerSample, playout_channels(),
>                                      playout_sample_rate(), NotNull(), _, _, _))
>       .Times(AtLeast(kNumCallbacks));
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartPlayout();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopPlayout();
> }
> 
>@@ -895,7 +895,7 @@ TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
> // audio samples via the RecordedDataIsAvailable callback.
> TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
>   MockAudioTransportAndroid mock(kRecording);
>-  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
>   EXPECT_CALL(
>       mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
>                                     kBytesPerSample, record_channels(),
>@@ -904,7 +904,7 @@ TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
> 
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartRecording();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopRecording();
> }
> 
>@@ -912,7 +912,7 @@ TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
> // active in both directions.
> TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
>   MockAudioTransportAndroid mock(kPlayout | kRecording);
>-  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  mock.HandleCallbacks(&test_is_done_, nullptr, kNumCallbacks);
>   EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
>                                      kBytesPerSample, playout_channels(),
>                                      playout_sample_rate(), NotNull(), _, _, _))
>@@ -925,7 +925,7 @@ TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartPlayout();
>   StartRecording();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopRecording();
>   StopPlayout();
> }
>@@ -941,12 +941,11 @@ TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
>   std::string file_name = GetFileName(playout_sample_rate());
>   std::unique_ptr<FileAudioStream> file_audio_stream(
>       new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
>-  mock.HandleCallbacks(test_is_done_.get(), file_audio_stream.get(),
>-                       num_callbacks);
>+  mock.HandleCallbacks(&test_is_done_, file_audio_stream.get(), num_callbacks);
>   // SetMaxPlayoutVolume();
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartPlayout();
>-  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  test_is_done_.Wait(kTestTimeOutInMilliseconds);
>   StopPlayout();
> }
> 
>@@ -1084,13 +1083,13 @@ TEST_F(AudioDeviceTest, DISABLED_RunPlayoutAndRecordingInFullDuplex) {
>   NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
>   std::unique_ptr<FifoAudioStream> fifo_audio_stream(
>       new FifoAudioStream(playout_frames_per_10ms_buffer()));
>-  mock.HandleCallbacks(test_is_done_.get(), fifo_audio_stream.get(),
>+  mock.HandleCallbacks(&test_is_done_, fifo_audio_stream.get(),
>                        kFullDuplexTimeInSec * kNumCallbacksPerSecond);
>   SetMaxPlayoutVolume();
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartRecording();
>   StartPlayout();
>-  test_is_done_->Wait(
>+  test_is_done_.Wait(
>       std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
>   StopPlayout();
>   StopRecording();
>@@ -1117,14 +1116,14 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
>   NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
>   std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
>       new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
>-  mock.HandleCallbacks(test_is_done_.get(), latency_audio_stream.get(),
>+  mock.HandleCallbacks(&test_is_done_, latency_audio_stream.get(),
>                        kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   SetMaxPlayoutVolume();
>   DisableBuiltInAECIfAvailable();
>   StartRecording();
>   StartPlayout();
>-  test_is_done_->Wait(
>+  test_is_done_.Wait(
>       std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
>   StopPlayout();
>   StopRecording();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/JavaVideoSourceTestHelper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/JavaVideoSourceTestHelper.java
>index bde34e95cf6333b0d92501e66a220e98035955e4..309c78654ef366d655c581d5b93b36de50049ac8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/JavaVideoSourceTestHelper.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/JavaVideoSourceTestHelper.java
>@@ -24,8 +24,9 @@ public class JavaVideoSourceTestHelper {
>   }
> 
>   @CalledByNative
>-  public static void deliverFrame(int width, int height, int rotation, CapturerObserver observer) {
>+  public static void deliverFrame(
>+      int width, int height, int rotation, long timestampNs, CapturerObserver observer) {
>     observer.onFrameCaptured(
>-        new VideoFrame(JavaI420Buffer.allocate(width, height), rotation, 0 /* timestampNs= */));
>+        new VideoFrame(JavaI420Buffer.allocate(width, height), rotation, timestampNs));
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/video/videosource_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/video/videosource_unittest.cc
>index b58ed1a45a54ed07b9d87fc243082bd9de070c7b..7ecf9f8eb4b8a0f789653aef6ed679c33548b387 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/video/videosource_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/video/videosource_unittest.cc
>@@ -40,9 +40,9 @@ TEST(JavaVideoSourceTest, CreateJavaVideoSource) {
>   rtc::ThreadManager::Instance()->WrapCurrentThread();
> 
>   rtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
>-      CreateJavaVideoSource(env,
>-                            rtc::ThreadManager::Instance()->CurrentThread(),
>-                            false /* is_screencast */);
>+      CreateJavaVideoSource(
>+          env, rtc::ThreadManager::Instance()->CurrentThread(),
>+          false /* is_screencast */, true /* align_timestamps */);
> 
>   ASSERT_NE(nullptr, video_track_source);
>   EXPECT_NE(nullptr,
>@@ -57,9 +57,9 @@ TEST(JavaVideoSourceTest, OnFrameCapturedFrameIsDeliveredToSink) {
>   rtc::ThreadManager::Instance()->WrapCurrentThread();
> 
>   rtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
>-      CreateJavaVideoSource(env,
>-                            rtc::ThreadManager::Instance()->CurrentThread(),
>-                            false /* is_screencast */);
>+      CreateJavaVideoSource(
>+          env, rtc::ThreadManager::Instance()->CurrentThread(),
>+          false /* is_screencast */, true /* align_timestamps */);
>   video_track_source->AddOrUpdateSink(&test_video_sink, rtc::VideoSinkWants());
> 
>   jni::Java_JavaVideoSourceTestHelper_startCapture(
>@@ -68,8 +68,9 @@ TEST(JavaVideoSourceTest, OnFrameCapturedFrameIsDeliveredToSink) {
>   const int width = 20;
>   const int height = 32;
>   const int rotation = 180;
>+  const int64_t timestamp = 987654321;
>   jni::Java_JavaVideoSourceTestHelper_deliverFrame(
>-      env, width, height, rotation,
>+      env, width, height, rotation, timestamp,
>       video_track_source->GetJavaVideoCapturerObserver(env));
> 
>   std::vector<VideoFrame> frames = test_video_sink.GetFrames();
>@@ -80,15 +81,49 @@ TEST(JavaVideoSourceTest, OnFrameCapturedFrameIsDeliveredToSink) {
>   EXPECT_EQ(rotation, frame.rotation());
> }
> 
>+TEST(JavaVideoSourceTest,
>+     OnFrameCapturedFrameIsDeliveredToSinkWithPreservedTimestamp) {
>+  TestVideoSink test_video_sink;
>+
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  // Wrap test thread so it can be used as the signaling thread.
>+  rtc::ThreadManager::Instance()->WrapCurrentThread();
>+
>+  rtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
>+      CreateJavaVideoSource(
>+          env, rtc::ThreadManager::Instance()->CurrentThread(),
>+          false /* is_screencast */, false /* align_timestamps */);
>+  video_track_source->AddOrUpdateSink(&test_video_sink, rtc::VideoSinkWants());
>+
>+  jni::Java_JavaVideoSourceTestHelper_startCapture(
>+      env, video_track_source->GetJavaVideoCapturerObserver(env),
>+      true /* success */);
>+  const int width = 20;
>+  const int height = 32;
>+  const int rotation = 180;
>+  const int64_t timestamp = 987654321;
>+  jni::Java_JavaVideoSourceTestHelper_deliverFrame(
>+      env, width, height, rotation, 987654321,
>+      video_track_source->GetJavaVideoCapturerObserver(env));
>+
>+  std::vector<VideoFrame> frames = test_video_sink.GetFrames();
>+  ASSERT_EQ(1u, frames.size());
>+  webrtc::VideoFrame frame = frames[0];
>+  EXPECT_EQ(width, frame.width());
>+  EXPECT_EQ(height, frame.height());
>+  EXPECT_EQ(rotation, frame.rotation());
>+  EXPECT_EQ(timestamp / 1000, frame.timestamp_us());
>+}
>+
> TEST(JavaVideoSourceTest, CapturerStartedSuccessStateBecomesLive) {
>   JNIEnv* env = AttachCurrentThreadIfNeeded();
>   // Wrap test thread so it can be used as the signaling thread.
>   rtc::ThreadManager::Instance()->WrapCurrentThread();
> 
>   rtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
>-      CreateJavaVideoSource(env,
>-                            rtc::ThreadManager::Instance()->CurrentThread(),
>-                            false /* is_screencast */);
>+      CreateJavaVideoSource(
>+          env, rtc::ThreadManager::Instance()->CurrentThread(),
>+          false /* is_screencast */, true /* align_timestamps */);
> 
>   jni::Java_JavaVideoSourceTestHelper_startCapture(
>       env, video_track_source->GetJavaVideoCapturerObserver(env),
>@@ -104,9 +139,9 @@ TEST(JavaVideoSourceTest, CapturerStartedFailureStateBecomesEnded) {
>   rtc::ThreadManager::Instance()->WrapCurrentThread();
> 
>   rtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
>-      CreateJavaVideoSource(env,
>-                            rtc::ThreadManager::Instance()->CurrentThread(),
>-                            false /* is_screencast */);
>+      CreateJavaVideoSource(
>+          env, rtc::ThreadManager::Instance()->CurrentThread(),
>+          false /* is_screencast */, true /* align_timestamps */);
> 
>   jni::Java_JavaVideoSourceTestHelper_startCapture(
>       env, video_track_source->GetJavaVideoCapturerObserver(env),
>@@ -122,9 +157,9 @@ TEST(JavaVideoSourceTest, CapturerStoppedStateBecomesEnded) {
>   rtc::ThreadManager::Instance()->WrapCurrentThread();
> 
>   rtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
>-      CreateJavaVideoSource(env,
>-                            rtc::ThreadManager::Instance()->CurrentThread(),
>-                            false /* is_screencast */);
>+      CreateJavaVideoSource(
>+          env, rtc::ThreadManager::Instance()->CurrentThread(),
>+          false /* is_screencast */, true /* align_timestamps */);
> 
>   jni::Java_JavaVideoSourceTestHelper_startCapture(
>       env, video_track_source->GetJavaVideoCapturerObserver(env),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java
>index ebf9de374f14b01d93a181cf7be2765f6d4387d9..c36a68cd21b7f1b530fe148381f88365f4f290c3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java
>@@ -403,12 +403,6 @@ class HardwareVideoEncoder implements VideoEncoder {
>     return VideoCodecStatus.OK;
>   }
> 
>-  @Override
>-  public VideoCodecStatus setChannelParameters(short packetLoss, long roundTripTimeMs) {
>-    encodeThreadChecker.checkIsOnValidThread();
>-    return VideoCodecStatus.OK; // No op.
>-  }
>-
>   @Override
>   public VideoCodecStatus setRateAllocation(BitrateAllocation bitrateAllocation, int framerate) {
>     encodeThreadChecker.checkIsOnValidThread();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeCapturerObserver.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeCapturerObserver.java
>index 24869818a66a38e9c03e122a0a3a9042380c3c8e..f673b24885b2b62eedc20ab20eb65fe18a138796 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeCapturerObserver.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeCapturerObserver.java
>@@ -11,6 +11,7 @@
> package org.webrtc;
> 
> import javax.annotation.Nullable;
>+import org.webrtc.VideoFrame;
> 
> /**
>  * Implements VideoCapturer.CapturerObserver and feeds frames to
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Decoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Decoder.java
>deleted file mode 100644
>index ab29c349c7a10218717b9fc694fe04056ab13ec9..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Decoder.java
>+++ /dev/null
>@@ -1,20 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-package org.webrtc;
>-
>-class VP8Decoder extends WrappedNativeVideoDecoder {
>-  @Override
>-  public long createNativeVideoDecoder() {
>-    return nativeCreateDecoder();
>-  }
>-
>-  static native long nativeCreateDecoder();
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Encoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Encoder.java
>deleted file mode 100644
>index c2f8f504177deeef9c3ea7c775852da28e43b4e7..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Encoder.java
>+++ /dev/null
>@@ -1,25 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-package org.webrtc;
>-
>-class VP8Encoder extends WrappedNativeVideoEncoder {
>-  @Override
>-  public long createNativeVideoEncoder() {
>-    return nativeCreateEncoder();
>-  }
>-
>-  static native long nativeCreateEncoder();
>-
>-  @Override
>-  public boolean isHardwareEncoder() {
>-    return false;
>-  }
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Decoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Decoder.java
>deleted file mode 100644
>index b6fee2d362b2ae583758cbeb9dcd1ad7da9ee1b6..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Decoder.java
>+++ /dev/null
>@@ -1,22 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-package org.webrtc;
>-
>-class VP9Decoder extends WrappedNativeVideoDecoder {
>-  @Override
>-  public long createNativeVideoDecoder() {
>-    return nativeCreateDecoder();
>-  }
>-
>-  static native long nativeCreateDecoder();
>-
>-  static native boolean nativeIsSupported();
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Encoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Encoder.java
>deleted file mode 100644
>index bc705bd9c6017eb66449c727f3640715f71913df..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Encoder.java
>+++ /dev/null
>@@ -1,27 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-package org.webrtc;
>-
>-class VP9Encoder extends WrappedNativeVideoEncoder {
>-  @Override
>-  public long createNativeVideoEncoder() {
>-    return nativeCreateEncoder();
>-  }
>-
>-  static native long nativeCreateEncoder();
>-
>-  @Override
>-  public boolean isHardwareEncoder() {
>-    return false;
>-  }
>-
>-  static native boolean nativeIsSupported();
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoEncoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoEncoder.java
>index 07e842862101207107cfd42783bb9d6050a51d5b..959cafca56907c0a18bf51c66f8bfab261ccc26d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoEncoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoEncoder.java
>@@ -32,11 +32,6 @@ abstract class WrappedNativeVideoEncoder implements VideoEncoder {
>     throw new UnsupportedOperationException("Not implemented.");
>   }
> 
>-  @Override
>-  public VideoCodecStatus setChannelParameters(short packetLoss, long roundTripTimeMs) {
>-    throw new UnsupportedOperationException("Not implemented.");
>-  }
>-
>   @Override
>   public VideoCodecStatus setRateAllocation(BitrateAllocation allocation, int framerate) {
>     throw new UnsupportedOperationException("Not implemented.");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder.cc
>index 47ddaac4672eb01da0c24147479d5a3cb1e1cf41..4df50829c36b10e1c0e00992123c5368d9fd8fc1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder.cc
>@@ -35,7 +35,6 @@
> #include "third_party/libyuv/include/libyuv/video_common.h"
> 
> using rtc::Bind;
>-using rtc::Thread;
> using rtc::ThreadManager;
> namespace webrtc {
> namespace jni {
>@@ -121,7 +120,7 @@ class MediaCodecVideoDecoder : public VideoDecoder, public rtc::MessageHandler {
> 
>   // State that is constant for the lifetime of this object once the ctor
>   // returns.
>-  std::unique_ptr<Thread>
>+  std::unique_ptr<rtc::Thread>
>       codec_thread_;  // Thread on which to operate MediaCodec.
>   ScopedJavaGlobalRef<jobject> j_media_codec_video_decoder_;
> 
>@@ -137,7 +136,7 @@ MediaCodecVideoDecoder::MediaCodecVideoDecoder(JNIEnv* jni,
>       inited_(false),
>       sw_fallback_required_(false),
>       use_surface_(use_surface),
>-      codec_thread_(Thread::Create()),
>+      codec_thread_(rtc::Thread::Create()),
>       j_media_codec_video_decoder_(
>           jni,
>           Java_MediaCodecVideoDecoder_Constructor(jni)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder.cc
>index 435bc713a8580cd9ab4f1a9671680e14a89849bc..09b2ccf43f40cd491bea05618bb2babbe1167239 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder.cc
>@@ -47,7 +47,6 @@
> #include "third_party/libyuv/include/libyuv/video_common.h"
> 
> using rtc::Bind;
>-using rtc::Thread;
> using rtc::ThreadManager;
> 
> namespace webrtc {
>@@ -105,13 +104,9 @@ class MediaCodecVideoEncoder : public VideoEncoder {
>   int32_t RegisterEncodeCompleteCallback(
>       EncodedImageCallback* callback) override;
>   int32_t Release() override;
>-  int32_t SetChannelParameters(uint32_t /* packet_loss */,
>-                               int64_t /* rtt */) override;
>   int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
>                             uint32_t frame_rate) override;
>-
>-  bool SupportsNativeHandle() const override { return has_egl_context_; }
>-  const char* ImplementationName() const override;
>+  EncoderInfo GetEncoderInfo() const override;
> 
>   // Fills the input buffer with data from the buffers passed as parameters.
>   bool FillInputBuffer(JNIEnv* jni,
>@@ -180,7 +175,7 @@ class MediaCodecVideoEncoder : public VideoEncoder {
>   // true on success.
>   bool DeliverPendingOutputs(JNIEnv* jni);
> 
>-  VideoEncoder::ScalingSettings GetScalingSettings() const override;
>+  VideoEncoder::ScalingSettings GetScalingSettingsInternal() const;
> 
>   // Displays encoder statistics.
>   void LogStatistics(bool force_log);
>@@ -271,6 +266,7 @@ class MediaCodecVideoEncoder : public VideoEncoder {
>   size_t gof_idx_;
> 
>   const bool has_egl_context_;
>+  EncoderInfo encoder_info_;
> 
>   // Temporary fix for VP8.
>   // Sends a key frame if frames are largely spaced apart (possibly
>@@ -358,17 +354,16 @@ int32_t MediaCodecVideoEncoder::InitEncode(const VideoCodec* codec_settings,
>     ALOGD << "H.264 profile: " << profile_;
>   }
> 
>+  encoder_info_.supports_native_handle = has_egl_context_;
>+  encoder_info_.implementation_name = "MediaCodec";
>+  encoder_info_.scaling_settings = GetScalingSettingsInternal();
>+
>   return InitEncodeInternal(
>       init_width, init_height, codec_settings->startBitrate,
>       codec_settings->maxFramerate,
>       codec_settings->expect_encode_from_texture && has_egl_context_);
> }
> 
>-int32_t MediaCodecVideoEncoder::SetChannelParameters(uint32_t /* packet_loss */,
>-                                                     int64_t /* rtt */) {
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
> bool MediaCodecVideoEncoder::ResetCodec() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_queue_checker_);
>   ALOGE << "Reset";
>@@ -930,6 +925,10 @@ int32_t MediaCodecVideoEncoder::SetRateAllocation(
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>+VideoEncoder::EncoderInfo MediaCodecVideoEncoder::GetEncoderInfo() const {
>+  return encoder_info_;
>+}
>+
> bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_queue_checker_);
> 
>@@ -1148,8 +1147,8 @@ void MediaCodecVideoEncoder::LogStatistics(bool force_log) {
>   }
> }
> 
>-VideoEncoder::ScalingSettings MediaCodecVideoEncoder::GetScalingSettings()
>-    const {
>+VideoEncoder::ScalingSettings
>+MediaCodecVideoEncoder::GetScalingSettingsInternal() const {
>   if (!scale_)
>     return VideoEncoder::ScalingSettings::kOff;
> 
>@@ -1206,10 +1205,6 @@ VideoEncoder::ScalingSettings MediaCodecVideoEncoder::GetScalingSettings()
>   return VideoEncoder::ScalingSettings::kOff;
> }
> 
>-const char* MediaCodecVideoEncoder::ImplementationName() const {
>-  return "MediaCodec";
>-}
>-
> static void JNI_MediaCodecVideoEncoder_FillInputBuffer(
>     JNIEnv* jni,
>     const JavaParamRef<jclass>&,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor.cc
>index b1351ef1e1b745550497561f9bc0d192f9512f76..7ac17281f97caf9fce7361e547e6d9be2d70c8ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor.cc
>@@ -19,6 +19,7 @@
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/ipaddress.h"
>+#include "rtc_base/logging.h"
> #include "rtc_base/strings/string_builder.h"
> #include "sdk/android/generated_base_jni/jni/NetworkMonitorAutoDetect_jni.h"
> #include "sdk/android/generated_base_jni/jni/NetworkMonitor_jni.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.cc
>index 41d4278ac9cf3aad47c1faa6d6d2836005c1f89f..e72850f24988b0487a8850ba0511c7c4b5b1e4bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.cc
>@@ -25,10 +25,12 @@ const int kRequiredResolutionAlignment = 2;
> 
> AndroidVideoTrackSource::AndroidVideoTrackSource(rtc::Thread* signaling_thread,
>                                                  JNIEnv* jni,
>-                                                 bool is_screencast)
>+                                                 bool is_screencast,
>+                                                 bool align_timestamps)
>     : AdaptedVideoTrackSource(kRequiredResolutionAlignment),
>       signaling_thread_(signaling_thread),
>-      is_screencast_(is_screencast) {
>+      is_screencast_(is_screencast),
>+      align_timestamps_(align_timestamps) {
>   RTC_LOG(LS_INFO) << "AndroidVideoTrackSource ctor";
>   camera_thread_checker_.DetachFromThread();
> }
>@@ -75,7 +77,9 @@ void AndroidVideoTrackSource::OnFrameCaptured(
> 
>   int64_t camera_time_us = timestamp_ns / rtc::kNumNanosecsPerMicrosec;
>   int64_t translated_camera_time_us =
>-      timestamp_aligner_.TranslateTimestamp(camera_time_us, rtc::TimeMicros());
>+      align_timestamps_ ? timestamp_aligner_.TranslateTimestamp(
>+                              camera_time_us, rtc::TimeMicros())
>+                        : camera_time_us;
> 
>   int adapted_width;
>   int adapted_height;
>@@ -84,10 +88,19 @@ void AndroidVideoTrackSource::OnFrameCaptured(
>   int crop_x;
>   int crop_y;
> 
>-  if (!AdaptFrame(width, height, camera_time_us, &adapted_width,
>-                  &adapted_height, &crop_width, &crop_height, &crop_x,
>-                  &crop_y)) {
>-    return;
>+  if (rotation % 180 == 0) {
>+    if (!AdaptFrame(width, height, camera_time_us, &adapted_width,
>+                    &adapted_height, &crop_width, &crop_height, &crop_x,
>+                    &crop_y)) {
>+      return;
>+    }
>+  } else {
>+    // Swap all width/height and x/y.
>+    if (!AdaptFrame(height, width, camera_time_us, &adapted_height,
>+                    &adapted_width, &crop_height, &crop_width, &crop_y,
>+                    &crop_x)) {
>+      return;
>+    }
>   }
> 
>   rtc::scoped_refptr<VideoFrameBuffer> buffer =
>@@ -103,12 +116,16 @@ void AndroidVideoTrackSource::OnFrameCaptured(
>   OnFrame(VideoFrame(buffer, rotation, translated_camera_time_us));
> }
> 
>-void AndroidVideoTrackSource::OnOutputFormatRequest(int width,
>-                                                    int height,
>+void AndroidVideoTrackSource::OnOutputFormatRequest(int landscape_width,
>+                                                    int landscape_height,
>+                                                    int portrait_width,
>+                                                    int portrait_height,
>                                                     int fps) {
>-  cricket::VideoFormat format(width, height,
>-                              cricket::VideoFormat::FpsToInterval(fps), 0);
>-  video_adapter()->OnOutputFormatRequest(format);
>+  video_adapter()->OnOutputFormatRequest(
>+      std::make_pair(landscape_width, landscape_height),
>+      landscape_width * landscape_height,
>+      std::make_pair(portrait_width, portrait_height),
>+      portrait_width * portrait_height, fps);
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.h
>index 8f092c1b041d6aea3b4952b607b2c9a72182de4c..4f8442e57781580fdd19b287c4b8e703c5c54f4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.h
>@@ -29,7 +29,8 @@ class AndroidVideoTrackSource : public rtc::AdaptedVideoTrackSource {
>  public:
>   AndroidVideoTrackSource(rtc::Thread* signaling_thread,
>                           JNIEnv* jni,
>-                          bool is_screencast = false);
>+                          bool is_screencast,
>+                          bool align_timestamps);
>   ~AndroidVideoTrackSource() override;
> 
>   bool is_screencast() const override;
>@@ -53,15 +54,20 @@ class AndroidVideoTrackSource : public rtc::AdaptedVideoTrackSource {
>                        VideoRotation rotation,
>                        const JavaRef<jobject>& j_video_frame_buffer);
> 
>-  void OnOutputFormatRequest(int width, int height, int fps);
>+  void OnOutputFormatRequest(int landscape_width,
>+                             int landscape_height,
>+                             int portrait_width,
>+                             int portrait_height,
>+                             int fps);
> 
>  private:
>   rtc::Thread* signaling_thread_;
>   rtc::AsyncInvoker invoker_;
>   rtc::ThreadChecker camera_thread_checker_;
>   SourceState state_;
>-  rtc::TimestampAligner timestamp_aligner_;
>   const bool is_screencast_;
>+  rtc::TimestampAligner timestamp_aligner_;
>+  const bool align_timestamps_;
> };
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.cc
>index dc3a8fe4ba553a7dae1428c32e0441a664c26a7e..e04a01faead1bf60c3c6f4e1c18cbd52ef8ddef1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.cc
>@@ -12,6 +12,7 @@
> 
> #include <utility>
> 
>+#include "modules/audio_device/audio_device_buffer.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/refcountedobject.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.h
>index cddd3e0676978555940389b3492e61022ebe1f02..476da14ceb92825e9c334892441ebf405cd8be4a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.h
>@@ -14,10 +14,13 @@
> #include <memory>
> 
> #include "absl/types/optional.h"
>-#include "modules/audio_device/audio_device_buffer.h"
>+#include "modules/audio_device/include/audio_device.h"
> #include "sdk/android/native_api/jni/scoped_java_ref.h"
> 
> namespace webrtc {
>+
>+class AudioDeviceBuffer;
>+
> namespace jni {
> 
> class AudioInput {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/builtinaudiodecoderfactoryfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/builtinaudiodecoderfactoryfactory.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..68a2843637496764d418412c3c9f44df4699262f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/builtinaudiodecoderfactoryfactory.cc
>@@ -0,0 +1,28 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/generated_builtin_audio_codecs_jni/jni/BuiltinAudioDecoderFactoryFactory_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+static jlong
>+JNI_BuiltinAudioDecoderFactoryFactory_CreateBuiltinAudioDecoderFactory(
>+    JNIEnv* env,
>+    const JavaParamRef<jclass>& jcaller) {
>+  return NativeToJavaPointer(CreateBuiltinAudioDecoderFactory().release());
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/builtinaudioencoderfactoryfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/builtinaudioencoderfactoryfactory.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..c1aefba9ab278ef643db022d60e279e91ace82bc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/builtinaudioencoderfactoryfactory.cc
>@@ -0,0 +1,28 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/generated_builtin_audio_codecs_jni/jni/BuiltinAudioEncoderFactoryFactory_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+static jlong
>+JNI_BuiltinAudioEncoderFactoryFactory_CreateBuiltinAudioEncoderFactory(
>+    JNIEnv* env,
>+    const JavaParamRef<jclass>& jcaller) {
>+  return NativeToJavaPointer(CreateBuiltinAudioEncoderFactory().release());
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.cc
>index c0ea28e2dc323927a8e27b00d73df7c7e5d3dc61..21f83364d97d67a5f5949784c55ddaa06aa15fe4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.cc
>@@ -10,7 +10,7 @@
> 
> #include "sdk/android/src/jni/encodedimage.h"
> 
>-#include "common_video/include/video_frame.h"
>+#include "api/video/encoded_image.h"
> #include "rtc_base/timeutils.h"
> #include "sdk/android/generated_video_jni/jni/EncodedImage_jni.h"
> #include "sdk/android/native_api/jni/java_types.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.h
>index 53e1d84127d7b61b276bc3160662149e2b180e2b..e6d6f7ef159d68aa542540f131e047e534da410b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.h
>@@ -33,12 +33,6 @@
>   RTC_CHECK(!jni->ExceptionCheck()) \
>       << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
> 
>-namespace jni_generator {
>-inline void CheckException(JNIEnv* env) {
>-  CHECK_EXCEPTION(env);
>-}
>-}  // namespace jni_generator
>-
> namespace webrtc {
> 
> // This function will initialize |atomic_class_id| to contain a global ref to
>@@ -86,4 +80,80 @@ using webrtc::MethodID;
> }  // namespace android
> }  // namespace base
> 
>+namespace jni_generator {
>+inline void CheckException(JNIEnv* env) {
>+  CHECK_EXCEPTION(env);
>+}
>+
>+// A 32 bit number could be an address on stack. Random 64 bit marker on the
>+// stack is much less likely to be present on stack.
>+constexpr uint64_t kJniStackMarkerValue = 0xbdbdef1bebcade1b;
>+
>+// Context about the JNI call with exception checked to be stored in stack.
>+struct BASE_EXPORT JniJavaCallContextUnchecked {
>+  inline JniJavaCallContextUnchecked() {
>+// TODO(ssid): Implement for other architectures.
>+#if defined(__arm__) || defined(__aarch64__)
>+    // This assumes that this method does not increment the stack pointer.
>+    asm volatile("mov %0, sp" : "=r"(sp));
>+#else
>+    sp = 0;
>+#endif
>+  }
>+
>+  // Force no inline to reduce code size.
>+  template <base::android::MethodID::Type type>
>+  void Init(JNIEnv* env,
>+            jclass clazz,
>+            const char* method_name,
>+            const char* jni_signature,
>+            std::atomic<jmethodID>* atomic_method_id) {
>+    env1 = env;
>+
>+    // Make sure compiler doesn't optimize out the assignment.
>+    memcpy(&marker, &kJniStackMarkerValue, sizeof(kJniStackMarkerValue));
>+    // Gets PC of the calling function.
>+    pc = reinterpret_cast<uintptr_t>(__builtin_return_address(0));
>+
>+    method_id = base::android::MethodID::LazyGet<type>(
>+        env, clazz, method_name, jni_signature, atomic_method_id);
>+  }
>+
>+  ~JniJavaCallContextUnchecked() {
>+    // Reset so that spurious marker finds are avoided.
>+    memset(&marker, 0, sizeof(marker));
>+  }
>+
>+  uint64_t marker;
>+  uintptr_t sp;
>+  uintptr_t pc;
>+
>+  JNIEnv* env1;
>+  jmethodID method_id;
>+};
>+
>+// Context about the JNI call with exception unchecked to be stored in stack.
>+struct BASE_EXPORT JniJavaCallContextChecked {
>+  // Force no inline to reduce code size.
>+  template <base::android::MethodID::Type type>
>+  void Init(JNIEnv* env,
>+            jclass clazz,
>+            const char* method_name,
>+            const char* jni_signature,
>+            std::atomic<jmethodID>* atomic_method_id) {
>+    base.Init<type>(env, clazz, method_name, jni_signature, atomic_method_id);
>+    // Reset |pc| to correct caller.
>+    base.pc = reinterpret_cast<uintptr_t>(__builtin_return_address(0));
>+  }
>+
>+  ~JniJavaCallContextChecked() { jni_generator::CheckException(base.env1); }
>+
>+  JniJavaCallContextUnchecked base;
>+};
>+
>+static_assert(sizeof(JniJavaCallContextChecked) ==
>+                  sizeof(JniJavaCallContextUnchecked),
>+              "Stack unwinder cannot work with structs of different sizes.");
>+}  // namespace jni_generator
>+
> #endif  // SDK_ANDROID_SRC_JNI_JNI_GENERATOR_HELPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/cryptooptions.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/cryptooptions.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..0f38b13f78bdd733e962b85af69879d21491f7e7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/cryptooptions.cc
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/cryptooptions.h"
>+
>+#include "sdk/android/generated_peerconnection_jni/jni/CryptoOptions_jni.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+absl::optional<CryptoOptions> JavaToNativeOptionalCryptoOptions(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_crypto_options) {
>+  if (j_crypto_options.is_null()) {
>+    return absl::nullopt;
>+  }
>+
>+  ScopedJavaLocalRef<jobject> j_srtp =
>+      Java_CryptoOptions_getSrtp(jni, j_crypto_options);
>+  ScopedJavaLocalRef<jobject> j_sframe =
>+      Java_CryptoOptions_getSFrame(jni, j_crypto_options);
>+
>+  CryptoOptions native_crypto_options;
>+  native_crypto_options.srtp.enable_gcm_crypto_suites =
>+      Java_Srtp_getEnableGcmCryptoSuites(jni, j_srtp);
>+  native_crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
>+      Java_Srtp_getEnableAes128Sha1_32CryptoCipher(jni, j_srtp);
>+  native_crypto_options.srtp.enable_encrypted_rtp_header_extensions =
>+      Java_Srtp_getEnableEncryptedRtpHeaderExtensions(jni, j_srtp);
>+  native_crypto_options.sframe.require_frame_encryption =
>+      Java_SFrame_getRequireFrameEncryption(jni, j_sframe);
>+  return absl::optional<CryptoOptions>(native_crypto_options);
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/cryptooptions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/cryptooptions.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..9890264a55c6f2c388027918ebd912b74c3720e7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/cryptooptions.h
>@@ -0,0 +1,30 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_CRYPTOOPTIONS_H_
>+#define SDK_ANDROID_SRC_JNI_PC_CRYPTOOPTIONS_H_
>+
>+#include <jni.h>
>+
>+#include "absl/types/optional.h"
>+#include "api/crypto/cryptooptions.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+absl::optional<CryptoOptions> JavaToNativeOptionalCryptoOptions(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_crypto_options);
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_CRYPTOOPTIONS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_video.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_video.cc
>index 0e0b80d4612f25e874b90d8ea5a722403164674a..54a401d605c191283ae7fcffe464729420211ccb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_video.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_video.cc
>@@ -31,7 +31,8 @@ VideoDecoderFactory* CreateVideoDecoderFactory(
> void* CreateVideoSource(JNIEnv* env,
>                         rtc::Thread* signaling_thread,
>                         rtc::Thread* worker_thread,
>-                        jboolean is_screencast) {
>+                        jboolean is_screencast,
>+                        jboolean align_timestamps) {
>   return nullptr;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.cc
>index 9987f252275e327c2ddad4f38049d0c889e9499e..2d62ea32a131cc5e136b82a147b30f0d79772aae 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.cc
>@@ -21,9 +21,9 @@ PeerConnectionFactoryInterface* factoryFromJava(jlong j_p) {
> }
> 
> OwnedFactoryAndThreads::OwnedFactoryAndThreads(
>-    std::unique_ptr<Thread> network_thread,
>-    std::unique_ptr<Thread> worker_thread,
>-    std::unique_ptr<Thread> signaling_thread,
>+    std::unique_ptr<rtc::Thread> network_thread,
>+    std::unique_ptr<rtc::Thread> worker_thread,
>+    std::unique_ptr<rtc::Thread> signaling_thread,
>     rtc::NetworkMonitorFactory* network_monitor_factory,
>     PeerConnectionFactoryInterface* factory)
>     : network_thread_(std::move(network_thread)),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.h
>index 2ea8570282d955ca000618f103c30974c44cc9e8..bacd42dfcd6c17f790ceab5dfe84f90b349336a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.h
>@@ -18,8 +18,6 @@
> #include "api/peerconnectioninterface.h"
> #include "rtc_base/thread.h"
> 
>-using rtc::Thread;
>-
> namespace webrtc {
> namespace jni {
> 
>@@ -33,18 +31,18 @@ PeerConnectionFactoryInterface* factoryFromJava(jlong j_p);
> // single thing for Java to hold and eventually free.
> class OwnedFactoryAndThreads {
>  public:
>-  OwnedFactoryAndThreads(std::unique_ptr<Thread> network_thread,
>-                         std::unique_ptr<Thread> worker_thread,
>-                         std::unique_ptr<Thread> signaling_thread,
>+  OwnedFactoryAndThreads(std::unique_ptr<rtc::Thread> network_thread,
>+                         std::unique_ptr<rtc::Thread> worker_thread,
>+                         std::unique_ptr<rtc::Thread> signaling_thread,
>                          rtc::NetworkMonitorFactory* network_monitor_factory,
>                          PeerConnectionFactoryInterface* factory);
> 
>   ~OwnedFactoryAndThreads();
> 
>   PeerConnectionFactoryInterface* factory() { return factory_; }
>-  Thread* network_thread() { return network_thread_.get(); }
>-  Thread* signaling_thread() { return signaling_thread_.get(); }
>-  Thread* worker_thread() { return worker_thread_.get(); }
>+  rtc::Thread* network_thread() { return network_thread_.get(); }
>+  rtc::Thread* signaling_thread() { return signaling_thread_.get(); }
>+  rtc::Thread* worker_thread() { return worker_thread_.get(); }
>   rtc::NetworkMonitorFactory* network_monitor_factory() {
>     return network_monitor_factory_;
>   }
>@@ -52,9 +50,9 @@ class OwnedFactoryAndThreads {
>   void InvokeJavaCallbacksOnFactoryThreads();
> 
>  private:
>-  const std::unique_ptr<Thread> network_thread_;
>-  const std::unique_ptr<Thread> worker_thread_;
>-  const std::unique_ptr<Thread> signaling_thread_;
>+  const std::unique_ptr<rtc::Thread> network_thread_;
>+  const std::unique_ptr<rtc::Thread> worker_thread_;
>+  const std::unique_ptr<rtc::Thread> signaling_thread_;
>   rtc::NetworkMonitorFactory* network_monitor_factory_;
>   PeerConnectionFactoryInterface* factory_;  // Const after ctor except dtor.
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.cc
>index 79da797e38877d8bfee40b3d289e8e57820629ef..0515ce264e739d7cf5315dec9905f91785e01f5c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.cc
>@@ -43,10 +43,12 @@
> #include "sdk/android/generated_peerconnection_jni/jni/PeerConnection_jni.h"
> #include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/pc/cryptooptions.h"
> #include "sdk/android/src/jni/pc/datachannel.h"
> #include "sdk/android/src/jni/pc/icecandidate.h"
> #include "sdk/android/src/jni/pc/mediaconstraints.h"
> #include "sdk/android/src/jni/pc/mediastreamtrack.h"
>+#include "sdk/android/src/jni/pc/rtccertificate.h"
> #include "sdk/android/src/jni/pc/rtcstatscollectorcallbackwrapper.h"
> #include "sdk/android/src/jni/pc/rtpsender.h"
> #include "sdk/android/src/jni/pc/sdpobserver.h"
>@@ -129,6 +131,8 @@ void JavaToNativeRTCConfiguration(
>       Java_RTCConfiguration_getBundlePolicy(jni, j_rtc_config);
>   ScopedJavaLocalRef<jobject> j_rtcp_mux_policy =
>       Java_RTCConfiguration_getRtcpMuxPolicy(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_rtc_certificate =
>+      Java_RTCConfiguration_getCertificate(jni, j_rtc_config);
>   ScopedJavaLocalRef<jobject> j_tcp_candidate_policy =
>       Java_RTCConfiguration_getTcpCandidatePolicy(jni, j_rtc_config);
>   ScopedJavaLocalRef<jobject> j_candidate_network_policy =
>@@ -143,11 +147,20 @@ void JavaToNativeRTCConfiguration(
>       Java_RTCConfiguration_getNetworkPreference(jni, j_rtc_config);
>   ScopedJavaLocalRef<jobject> j_sdp_semantics =
>       Java_RTCConfiguration_getSdpSemantics(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_crypto_options =
>+      Java_RTCConfiguration_getCryptoOptions(jni, j_rtc_config);
> 
>   rtc_config->type = JavaToNativeIceTransportsType(jni, j_ice_transports_type);
>   rtc_config->bundle_policy = JavaToNativeBundlePolicy(jni, j_bundle_policy);
>   rtc_config->rtcp_mux_policy =
>       JavaToNativeRtcpMuxPolicy(jni, j_rtcp_mux_policy);
>+  if (!j_rtc_certificate.is_null()) {
>+    rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>+        rtc::RTCCertificate::FromPEM(
>+            JavaToNativeRTCCertificatePEM(jni, j_rtc_certificate));
>+    RTC_CHECK(certificate != nullptr) << "supplied certificate is malformed.";
>+    rtc_config->certificates.push_back(certificate);
>+  }
>   rtc_config->tcp_candidate_policy =
>       JavaToNativeTcpCandidatePolicy(jni, j_tcp_candidate_policy);
>   rtc_config->candidate_network_policy =
>@@ -234,6 +247,13 @@ void JavaToNativeRTCConfiguration(
>   rtc_config->sdp_semantics = JavaToNativeSdpSemantics(jni, j_sdp_semantics);
>   rtc_config->active_reset_srtp_params =
>       Java_RTCConfiguration_getActiveResetSrtpParams(jni, j_rtc_config);
>+  rtc_config->use_media_transport =
>+      Java_RTCConfiguration_getUseMediaTransport(jni, j_rtc_config);
>+  rtc_config->use_media_transport_for_data_channels =
>+      Java_RTCConfiguration_getUseMediaTransportForDataChannels(jni,
>+                                                                j_rtc_config);
>+  rtc_config->crypto_options =
>+      JavaToNativeOptionalCryptoOptions(jni, j_crypto_options);
> }
> 
> rtc::KeyType GetRtcConfigKeyType(JNIEnv* env,
>@@ -279,6 +299,14 @@ void PeerConnectionObserverJni::OnIceConnectionChange(
>       Java_IceConnectionState_fromNativeIndex(env, new_state));
> }
> 
>+void PeerConnectionObserverJni::OnConnectionChange(
>+    PeerConnectionInterface::PeerConnectionState new_state) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_Observer_onConnectionChange(env, j_observer_global_,
>+                                   Java_PeerConnectionState_fromNativeIndex(
>+                                       env, static_cast<int>(new_state)));
>+}
>+
> void PeerConnectionObserverJni::OnIceConnectionReceivingChange(bool receiving) {
>   JNIEnv* env = AttachCurrentThreadIfNeeded();
>   Java_Observer_onIceConnectionReceivingChange(env, j_observer_global_,
>@@ -429,6 +457,16 @@ static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetRemoteDescription(
>   return sdp ? NativeToJavaSessionDescription(jni, sdp) : nullptr;
> }
> 
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetCertificate(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc) {
>+  const PeerConnectionInterface::RTCConfiguration rtc_config =
>+      ExtractNativePC(jni, j_pc)->GetConfiguration();
>+  rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>+      rtc_config.certificates[0];
>+  return NativeToJavaRTCCertificatePEM(jni, certificate->ToPEM());
>+}
>+
> static ScopedJavaLocalRef<jobject> JNI_PeerConnection_CreateDataChannel(
>     JNIEnv* jni,
>     const JavaParamRef<jobject>& j_pc,
>@@ -722,6 +760,14 @@ static ScopedJavaLocalRef<jobject> JNI_PeerConnection_IceConnectionState(
>       env, ExtractNativePC(env, j_pc)->ice_connection_state());
> }
> 
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_ConnectionState(
>+    JNIEnv* env,
>+    const JavaParamRef<jobject>& j_pc) {
>+  return Java_PeerConnectionState_fromNativeIndex(
>+      env,
>+      static_cast<int>(ExtractNativePC(env, j_pc)->peer_connection_state()));
>+}
>+
> static ScopedJavaLocalRef<jobject> JNI_PeerConnection_IceGatheringState(
>     JNIEnv* env,
>     const JavaParamRef<jobject>& j_pc) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.h
>index f20166727ea0b4954b06a121eccc93cc6cac75c0..a9d7b39c3d4217ce2ec8257a1397fb68723e2989 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.h
>@@ -51,6 +51,8 @@ class PeerConnectionObserverJni : public PeerConnectionObserver {
>       PeerConnectionInterface::SignalingState new_state) override;
>   void OnIceConnectionChange(
>       PeerConnectionInterface::IceConnectionState new_state) override;
>+  void OnConnectionChange(
>+      PeerConnectionInterface::PeerConnectionState new_state) override;
>   void OnIceConnectionReceivingChange(bool receiving) override;
>   void OnIceGatheringChange(
>       PeerConnectionInterface::IceGatheringState new_state) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory.cc
>index 3de79555a63c1dd805cae135338db4aae38ecc99..c2b8b9c5d7dfa7bfe2eb8416a8e122f40166728b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory.cc
>@@ -50,10 +50,6 @@ JavaToNativePeerConnectionFactoryOptions(JNIEnv* jni,
>   bool disable_encryption = Java_Options_getDisableEncryption(jni, options);
>   bool disable_network_monitor =
>       Java_Options_getDisableNetworkMonitor(jni, options);
>-  bool enable_aes128_sha1_32_crypto_cipher =
>-      Java_Options_getEnableAes128Sha1_32CryptoCipher(jni, options);
>-  bool enable_gcm_crypto_suites =
>-      Java_Options_getEnableGcmCryptoSuites(jni, options);
> 
>   PeerConnectionFactoryInterface::Options native_options;
> 
>@@ -63,10 +59,6 @@ JavaToNativePeerConnectionFactoryOptions(JNIEnv* jni,
>   native_options.disable_encryption = disable_encryption;
>   native_options.disable_network_monitor = disable_network_monitor;
> 
>-  native_options.crypto_options.enable_aes128_sha1_32_crypto_cipher =
>-      enable_aes128_sha1_32_crypto_cipher;
>-  native_options.crypto_options.enable_gcm_crypto_suites =
>-      enable_gcm_crypto_suites;
>   return native_options;
> }
> 
>@@ -199,15 +191,21 @@ static void JNI_PeerConnectionFactory_ShutdownInternalTracer(
>   rtc::tracing::ShutdownInternalTracer();
> }
> 
>+// Following parameters are optional:
>+// |audio_device_module|, |jencoder_factory|, |jdecoder_factory|,
>+// |audio_processor|, |media_transport_factory|, |fec_controller_factory|.
> jlong CreatePeerConnectionFactoryForJava(
>     JNIEnv* jni,
>     const JavaParamRef<jobject>& jcontext,
>     const JavaParamRef<jobject>& joptions,
>     rtc::scoped_refptr<AudioDeviceModule> audio_device_module,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>     const JavaParamRef<jobject>& jencoder_factory,
>     const JavaParamRef<jobject>& jdecoder_factory,
>     rtc::scoped_refptr<AudioProcessing> audio_processor,
>-    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory) {
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
>+    std::unique_ptr<MediaTransportFactory> media_transport_factory) {
>   // talk/ assumes pretty widely that the current Thread is ThreadManager'd, but
>   // ThreadManager only WrapCurrentThread()s the thread where it is first
>   // created.  Since the semantics around when auto-wrapping happens in
>@@ -229,8 +227,6 @@ jlong CreatePeerConnectionFactoryForJava(
>   RTC_CHECK(signaling_thread->Start()) << "Failed to start thread";
> 
>   rtc::NetworkMonitorFactory* network_monitor_factory = nullptr;
>-  auto audio_encoder_factory = CreateAudioEncoderFactory();
>-  auto audio_decoder_factory = CreateAudioDecoderFactory();
> 
>   PeerConnectionFactoryInterface::Options options;
>   bool has_options = !joptions.is_null();
>@@ -257,12 +253,19 @@ jlong CreatePeerConnectionFactoryForJava(
>       std::unique_ptr<VideoDecoderFactory>(
>           CreateVideoDecoderFactory(jni, jdecoder_factory)),
>       audio_mixer, audio_processor));
>+  PeerConnectionFactoryDependencies dependencies;
>+  dependencies.network_thread = network_thread.get();
>+  dependencies.worker_thread = worker_thread.get();
>+  dependencies.signaling_thread = signaling_thread.get();
>+  dependencies.media_engine = std::move(media_engine);
>+  dependencies.call_factory = std::move(call_factory);
>+  dependencies.event_log_factory = std::move(rtc_event_log_factory);
>+  dependencies.fec_controller_factory = std::move(fec_controller_factory);
>+  dependencies.media_transport_factory = std::move(media_transport_factory);
> 
>   rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>-      CreateModularPeerConnectionFactory(
>-          network_thread.get(), worker_thread.get(), signaling_thread.get(),
>-          std::move(media_engine), std::move(call_factory),
>-          std::move(rtc_event_log_factory), std::move(fec_controller_factory)));
>+      CreateModularPeerConnectionFactory(std::move(dependencies)));
>+
>   RTC_CHECK(factory) << "Failed to create the peer connection factory; "
>                      << "WebRTC/libjingle init likely failed on this device";
>   // TODO(honghaiz): Maybe put the options as the argument of
>@@ -283,21 +286,39 @@ static jlong JNI_PeerConnectionFactory_CreatePeerConnectionFactory(
>     const JavaParamRef<jobject>& jcontext,
>     const JavaParamRef<jobject>& joptions,
>     jlong native_audio_device_module,
>+    jlong native_audio_encoder_factory,
>+    jlong native_audio_decoder_factory,
>     const JavaParamRef<jobject>& jencoder_factory,
>     const JavaParamRef<jobject>& jdecoder_factory,
>     jlong native_audio_processor,
>-    jlong native_fec_controller_factory) {
>+    jlong native_fec_controller_factory,
>+    jlong native_media_transport_factory) {
>   rtc::scoped_refptr<AudioProcessing> audio_processor =
>       reinterpret_cast<AudioProcessing*>(native_audio_processor);
>+  AudioEncoderFactory* audio_encoder_factory_ptr =
>+      reinterpret_cast<AudioEncoderFactory*>(native_audio_encoder_factory);
>+  rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory(
>+      audio_encoder_factory_ptr);
>+  // Release the caller's reference count.
>+  audio_encoder_factory->Release();
>+  AudioDecoderFactory* audio_decoder_factory_ptr =
>+      reinterpret_cast<AudioDecoderFactory*>(native_audio_decoder_factory);
>+  rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory(
>+      audio_decoder_factory_ptr);
>+  // Release the caller's reference count.
>+  audio_decoder_factory->Release();
>   std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory(
>       reinterpret_cast<FecControllerFactoryInterface*>(
>           native_fec_controller_factory));
>+  std::unique_ptr<MediaTransportFactory> media_transport_factory(
>+      reinterpret_cast<MediaTransportFactory*>(native_media_transport_factory));
>   return CreatePeerConnectionFactoryForJava(
>       jni, jcontext, joptions,
>       reinterpret_cast<AudioDeviceModule*>(native_audio_device_module),
>-      jencoder_factory, jdecoder_factory,
>+      audio_encoder_factory, audio_decoder_factory, jencoder_factory,
>+      jdecoder_factory,
>       audio_processor ? audio_processor : CreateAudioProcessing(),
>-      std::move(fec_controller_factory));
>+      std::move(fec_controller_factory), std::move(media_transport_factory));
> }
> 
> static void JNI_PeerConnectionFactory_FreeFactory(JNIEnv*,
>@@ -396,18 +417,20 @@ static jlong JNI_PeerConnectionFactory_CreatePeerConnection(
>       PeerConnectionInterface::RTCConfigurationType::kAggressive);
>   JavaToNativeRTCConfiguration(jni, j_rtc_config, &rtc_config);
> 
>-  // Generate non-default certificate.
>-  rtc::KeyType key_type = GetRtcConfigKeyType(jni, j_rtc_config);
>-  if (key_type != rtc::KT_DEFAULT) {
>-    rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>-        rtc::RTCCertificateGenerator::GenerateCertificate(
>-            rtc::KeyParams(key_type), absl::nullopt);
>-    if (!certificate) {
>-      RTC_LOG(LS_ERROR) << "Failed to generate certificate. KeyType: "
>-                        << key_type;
>-      return 0;
>+  if (rtc_config.certificates.empty()) {
>+    // Generate non-default certificate.
>+    rtc::KeyType key_type = GetRtcConfigKeyType(jni, j_rtc_config);
>+    if (key_type != rtc::KT_DEFAULT) {
>+      rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>+          rtc::RTCCertificateGenerator::GenerateCertificate(
>+              rtc::KeyParams(key_type), absl::nullopt);
>+      if (!certificate) {
>+        RTC_LOG(LS_ERROR) << "Failed to generate certificate. KeyType: "
>+                          << key_type;
>+        return 0;
>+      }
>+      rtc_config.certificates.push_back(certificate);
>     }
>-    rtc_config.certificates.push_back(certificate);
>   }
> 
>   std::unique_ptr<MediaConstraintsInterface> constraints;
>@@ -437,12 +460,13 @@ static jlong JNI_PeerConnectionFactory_CreateVideoSource(
>     JNIEnv* jni,
>     const JavaParamRef<jclass>&,
>     jlong native_factory,
>-    jboolean is_screencast) {
>+    jboolean is_screencast,
>+    jboolean align_timestamps) {
>   OwnedFactoryAndThreads* factory =
>       reinterpret_cast<OwnedFactoryAndThreads*>(native_factory);
>   return jlongFromPointer(CreateVideoSource(jni, factory->signaling_thread(),
>                                             factory->worker_thread(),
>-                                            is_screencast));
>+                                            is_screencast, align_timestamps));
> }
> 
> static jlong JNI_PeerConnectionFactory_CreateVideoTrack(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtccertificate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtccertificate.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..8c598cce09e35c6a4ac2f11f0dd5fcfd1af9e165
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtccertificate.cc
>@@ -0,0 +1,60 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/rtccertificate.h"
>+#include "sdk/android/src/jni/pc/icecandidate.h"
>+
>+#include "rtc_base/refcount.h"
>+#include "rtc_base/rtccertificate.h"
>+#include "rtc_base/rtccertificategenerator.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/RtcCertificatePem_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+rtc::RTCCertificatePEM JavaToNativeRTCCertificatePEM(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_rtc_certificate) {
>+  ScopedJavaLocalRef<jstring> privatekey_field =
>+      Java_RtcCertificatePem_getPrivateKey(jni, j_rtc_certificate);
>+  ScopedJavaLocalRef<jstring> certificate_field =
>+      Java_RtcCertificatePem_getCertificate(jni, j_rtc_certificate);
>+  return rtc::RTCCertificatePEM(JavaToNativeString(jni, privatekey_field),
>+                                JavaToNativeString(jni, certificate_field));
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRTCCertificatePEM(
>+    JNIEnv* jni,
>+    const rtc::RTCCertificatePEM& certificate) {
>+  return Java_RtcCertificatePem_Constructor(
>+      jni, NativeToJavaString(jni, certificate.private_key()),
>+      NativeToJavaString(jni, certificate.certificate()));
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_RtcCertificatePem_GenerateCertificate(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jobject>& j_key_type,
>+    jlong j_expires) {
>+  rtc::KeyType key_type = JavaToNativeKeyType(jni, j_key_type);
>+  uint64_t expires = (uint64_t)j_expires;
>+  rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>+      rtc::RTCCertificateGenerator::GenerateCertificate(
>+          rtc::KeyParams(key_type), expires);
>+  rtc::RTCCertificatePEM pem = certificate->ToPEM();
>+  return Java_RtcCertificatePem_Constructor(
>+      jni, NativeToJavaString(jni, pem.private_key()),
>+      NativeToJavaString(jni, pem.certificate()));
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtccertificate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtccertificate.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..f7d1f75d13993d50cbeaa6ed7f6156c829659adf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtccertificate.h
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_RTCCERTIFICATE_H_
>+#define SDK_ANDROID_SRC_JNI_PC_RTCCERTIFICATE_H_
>+
>+#include "rtc_base/refcount.h"
>+#include "rtc_base/rtccertificate.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+rtc::RTCCertificatePEM JavaToNativeRTCCertificatePEM(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_rtc_certificate);
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRTCCertificatePEM(
>+    JNIEnv* env,
>+    const rtc::RTCCertificatePEM& certificate);
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_RTCCERTIFICATE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.cc
>index dd10fd51cb6ad5f4f770f288b27fafbd7fc083dd..c7c6e76d5d9ba73e05b900d3bf5e0c0c10e55bc9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.cc
>@@ -44,10 +44,11 @@ VideoDecoderFactory* CreateVideoDecoderFactory(
> void* CreateVideoSource(JNIEnv* env,
>                         rtc::Thread* signaling_thread,
>                         rtc::Thread* worker_thread,
>-                        jboolean is_screencast) {
>+                        jboolean is_screencast,
>+                        jboolean align_timestamps) {
>   rtc::scoped_refptr<AndroidVideoTrackSource> source(
>-      new rtc::RefCountedObject<AndroidVideoTrackSource>(signaling_thread, env,
>-                                                         is_screencast));
>+      new rtc::RefCountedObject<AndroidVideoTrackSource>(
>+          signaling_thread, env, is_screencast, align_timestamps));
>   return VideoTrackSourceProxy::Create(signaling_thread, worker_thread, source)
>       .release();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.h
>index 183770b8206283902f79b35f6073b6eaf79e98fb..04390ea453ea0a3b8ce12b1b4e35fcb8e7e650d8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.h
>@@ -36,7 +36,8 @@ VideoDecoderFactory* CreateVideoDecoderFactory(
> void* CreateVideoSource(JNIEnv* env,
>                         rtc::Thread* signaling_thread,
>                         rtc::Thread* worker_thread,
>-                        jboolean is_screencast);
>+                        jboolean is_screencast,
>+                        jboolean align_timestamps);
> 
> }  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/timestampaligner.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/timestampaligner.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..abe9b0fdc513454f50dcb2b6bfd12b7bac10bbb4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/timestampaligner.cc
>@@ -0,0 +1,52 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <jni.h>
>+
>+#include "rtc_base/timestampaligner.h"
>+#include "rtc_base/timeutils.h"
>+#include "sdk/android/generated_video_jni/jni/TimestampAligner_jni.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+static jlong JNI_TimestampAligner_RtcTimeNanos(
>+    JNIEnv* env,
>+    const JavaParamRef<jclass>& j_caller) {
>+  return rtc::TimeNanos();
>+}
>+
>+static jlong JNI_TimestampAligner_CreateTimestampAligner(
>+    JNIEnv* env,
>+    const JavaParamRef<jclass>& j_caller) {
>+  return jlongFromPointer(new rtc::TimestampAligner());
>+}
>+
>+static void JNI_TimestampAligner_ReleaseTimestampAligner(
>+    JNIEnv* env,
>+    const JavaParamRef<jclass>& j_caller,
>+    jlong timestamp_aligner) {
>+  delete reinterpret_cast<rtc::TimestampAligner*>(timestamp_aligner);
>+}
>+
>+static jlong JNI_TimestampAligner_TranslateTimestamp(
>+    JNIEnv* env,
>+    const JavaParamRef<jclass>& j_caller,
>+    jlong timestamp_aligner,
>+    jlong camera_time_ns) {
>+  return reinterpret_cast<rtc::TimestampAligner*>(timestamp_aligner)
>+             ->TranslateTimestamp(camera_time_ns / rtc::kNumNanosecsPerMicrosec,
>+                                  rtc::TimeMicros()) *
>+         rtc::kNumNanosecsPerMicrosec;
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.cc
>index 6381c63cb06f9072968178dca09b144e90dc900d..1e4b84eac60f2b7c9badef1ac1cbe052b9aae95e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.cc
>@@ -32,8 +32,6 @@ namespace jni {
> VideoEncoderWrapper::VideoEncoderWrapper(JNIEnv* jni,
>                                          const JavaRef<jobject>& j_encoder)
>     : encoder_(jni, j_encoder), int_array_class_(GetClass(jni, "[I")) {
>-  implementation_name_ = GetImplementationName(jni);
>-
>   initialized_ = false;
>   num_resets_ = 0;
> }
>@@ -82,6 +80,10 @@ int32_t VideoEncoderWrapper::InitEncodeInternal(JNIEnv* jni) {
>       jni, Java_VideoEncoder_initEncode(jni, encoder_, settings, callback));
>   RTC_LOG(LS_INFO) << "initEncode: " << status;
> 
>+  encoder_info_.supports_native_handle = true;
>+  encoder_info_.implementation_name = GetImplementationName(jni);
>+  encoder_info_.scaling_settings = GetScalingSettingsInternal(jni);
>+
>   if (status == WEBRTC_VIDEO_CODEC_OK) {
>     initialized_ = true;
>   }
>@@ -136,14 +138,6 @@ int32_t VideoEncoderWrapper::Encode(
>   return HandleReturnCode(jni, ret, "encode");
> }
> 
>-int32_t VideoEncoderWrapper::SetChannelParameters(uint32_t packet_loss,
>-                                                  int64_t rtt) {
>-  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedJavaLocalRef<jobject> ret = Java_VideoEncoder_setChannelParameters(
>-      jni, encoder_, (jshort)packet_loss, (jlong)rtt);
>-  return HandleReturnCode(jni, ret, "setChannelParameters");
>-}
>-
> int32_t VideoEncoderWrapper::SetRateAllocation(
>     const VideoBitrateAllocation& allocation,
>     uint32_t framerate) {
>@@ -156,9 +150,12 @@ int32_t VideoEncoderWrapper::SetRateAllocation(
>   return HandleReturnCode(jni, ret, "setRateAllocation");
> }
> 
>-VideoEncoderWrapper::ScalingSettings VideoEncoderWrapper::GetScalingSettings()
>-    const {
>-  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>+VideoEncoder::EncoderInfo VideoEncoderWrapper::GetEncoderInfo() const {
>+  return encoder_info_;
>+}
>+
>+VideoEncoderWrapper::ScalingSettings
>+VideoEncoderWrapper::GetScalingSettingsInternal(JNIEnv* jni) const {
>   ScopedJavaLocalRef<jobject> j_scaling_settings =
>       Java_VideoEncoder_getScalingSettings(jni, encoder_);
>   bool isOn =
>@@ -206,14 +203,6 @@ VideoEncoderWrapper::ScalingSettings VideoEncoderWrapper::GetScalingSettings()
>   }
> }
> 
>-bool VideoEncoderWrapper::SupportsNativeHandle() const {
>-  return true;
>-}
>-
>-const char* VideoEncoderWrapper::ImplementationName() const {
>-  return implementation_name_.c_str();
>-}
>-
> void VideoEncoderWrapper::OnEncodedFrame(JNIEnv* jni,
>                                          const JavaRef<jobject>& j_caller,
>                                          const JavaRef<jobject>& j_buffer,
>@@ -385,7 +374,6 @@ CodecSpecificInfo VideoEncoderWrapper::ParseCodecSpecificInfo(
>   CodecSpecificInfo info;
>   memset(&info, 0, sizeof(info));
>   info.codecType = codec_settings_.codecType;
>-  info.codec_name = implementation_name_.c_str();
> 
>   switch (codec_settings_.codecType) {
>     case kVideoCodecVP8:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.h
>index c30560503cb75ef9aa542d181905539ae64d4dd8..3618f7847375e590345cf77c3089ec0d67cfd12b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.h
>@@ -45,14 +45,10 @@ class VideoEncoderWrapper : public VideoEncoder {
>                  const CodecSpecificInfo* codec_specific_info,
>                  const std::vector<FrameType>* frame_types) override;
> 
>-  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>-
>   int32_t SetRateAllocation(const VideoBitrateAllocation& allocation,
>                             uint32_t framerate) override;
> 
>-  ScalingSettings GetScalingSettings() const override;
>-
>-  bool SupportsNativeHandle() const override;
>+  EncoderInfo GetEncoderInfo() const override;
> 
>   // Should only be called by JNI.
>   void OnEncodedFrame(JNIEnv* jni,
>@@ -66,8 +62,6 @@ class VideoEncoderWrapper : public VideoEncoder {
>                       jboolean complete_frame,
>                       const JavaRef<jobject>& j_qp);
> 
>-  const char* ImplementationName() const override;
>-
>  private:
>   struct FrameExtraInfo {
>     int64_t capture_time_ns;  // Used as an identifier of the frame.
>@@ -92,11 +86,11 @@ class VideoEncoderWrapper : public VideoEncoder {
>       const VideoBitrateAllocation& allocation);
>   std::string GetImplementationName(JNIEnv* jni) const;
> 
>+  ScalingSettings GetScalingSettingsInternal(JNIEnv* jni) const;
>+
>   const ScopedJavaGlobalRef<jobject> encoder_;
>   const ScopedJavaGlobalRef<jclass> int_array_class_;
> 
>-  std::string implementation_name_;
>-
>   rtc::TaskQueue* encoder_queue_;
>   std::deque<FrameExtraInfo> frame_extra_infos_;
>   EncodedImageCallback* callback_;
>@@ -104,6 +98,7 @@ class VideoEncoderWrapper : public VideoEncoder {
>   int num_resets_;
>   int number_of_cores_;
>   VideoCodec codec_settings_;
>+  EncoderInfo encoder_info_;
>   H264BitstreamParser h264_bitstream_parser_;
> 
>   // VP9 variables to populate codec specific structure.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosource.cc
>index 09b538ebc265ba987e0c22c6bcc395131e85507f..37ba0b04b115101547e316457fbca968c5a9533c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosource.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosource.cc
>@@ -33,13 +33,16 @@ static jlong JNI_VideoSource_GetInternalSource(JNIEnv* jni,
> static void JNI_VideoSource_AdaptOutputFormat(JNIEnv* jni,
>                                               const JavaParamRef<jclass>&,
>                                               jlong j_source,
>-                                              jint j_width,
>-                                              jint j_height,
>+                                              jint j_landscape_width,
>+                                              jint j_landscape_height,
>+                                              jint j_portrait_width,
>+                                              jint j_portrait_height,
>                                               jint j_fps) {
>   RTC_LOG(LS_INFO) << "VideoSource_nativeAdaptOutputFormat";
>   AndroidVideoTrackSource* source =
>       AndroidVideoTrackSourceFromJavaProxy(j_source);
>-  source->OnOutputFormatRequest(j_width, j_height, j_fps);
>+  source->OnOutputFormatRequest(j_landscape_width, j_landscape_height,
>+                                j_portrait_width, j_portrait_height, j_fps);
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp8codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp8codec.cc
>index 295a03a610770f06b944511cfe003a2c275c9fc5..2756b102faf19324ec0d6b4e091c3fbd6c417f2f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp8codec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp8codec.cc
>@@ -11,20 +11,20 @@
> #include <jni.h>
> 
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>-#include "sdk/android/generated_vp8_jni/jni/VP8Decoder_jni.h"
>-#include "sdk/android/generated_vp8_jni/jni/VP8Encoder_jni.h"
>+#include "sdk/android/generated_libvpx_vp8_jni/jni/LibvpxVp8Decoder_jni.h"
>+#include "sdk/android/generated_libvpx_vp8_jni/jni/LibvpxVp8Encoder_jni.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-static jlong JNI_VP8Encoder_CreateEncoder(JNIEnv* jni,
>-                                          const JavaParamRef<jclass>&) {
>+static jlong JNI_LibvpxVp8Encoder_CreateEncoder(JNIEnv* jni,
>+                                                const JavaParamRef<jclass>&) {
>   return jlongFromPointer(VP8Encoder::Create().release());
> }
> 
>-static jlong JNI_VP8Decoder_CreateDecoder(JNIEnv* jni,
>-                                          const JavaParamRef<jclass>&) {
>+static jlong JNI_LibvpxVp8Decoder_CreateDecoder(JNIEnv* jni,
>+                                                const JavaParamRef<jclass>&) {
>   return jlongFromPointer(VP8Decoder::Create().release());
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp9codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp9codec.cc
>index ca712e6d8654d1ec5e90238d8f2b7b7b7f5942fa..92f9754d563311a073ec2b8a314c66e1a30fd508 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp9codec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp9codec.cc
>@@ -11,30 +11,30 @@
> #include <jni.h>
> 
> #include "modules/video_coding/codecs/vp9/include/vp9.h"
>-#include "sdk/android/generated_vp9_jni/jni/VP9Decoder_jni.h"
>-#include "sdk/android/generated_vp9_jni/jni/VP9Encoder_jni.h"
>+#include "sdk/android/generated_libvpx_vp9_jni/jni/LibvpxVp9Decoder_jni.h"
>+#include "sdk/android/generated_libvpx_vp9_jni/jni/LibvpxVp9Encoder_jni.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-static jlong JNI_VP9Encoder_CreateEncoder(JNIEnv* jni,
>-                                          const JavaParamRef<jclass>& w) {
>+static jlong JNI_LibvpxVp9Encoder_CreateEncoder(JNIEnv* jni,
>+                                                const JavaParamRef<jclass>& w) {
>   return jlongFromPointer(VP9Encoder::Create().release());
> }
> 
>-static jboolean JNI_VP9Encoder_IsSupported(JNIEnv* jni,
>-                                           const JavaParamRef<jclass>&) {
>+static jboolean JNI_LibvpxVp9Encoder_IsSupported(JNIEnv* jni,
>+                                                 const JavaParamRef<jclass>&) {
>   return !SupportedVP9Codecs().empty();
> }
> 
>-static jlong JNI_VP9Decoder_CreateDecoder(JNIEnv* jni,
>-                                          const JavaParamRef<jclass>& w) {
>+static jlong JNI_LibvpxVp9Decoder_CreateDecoder(JNIEnv* jni,
>+                                                const JavaParamRef<jclass>& w) {
>   return jlongFromPointer(VP9Decoder::Create().release());
> }
> 
>-static jboolean JNI_VP9Decoder_IsSupported(JNIEnv* jni,
>-                                           const JavaParamRef<jclass>&) {
>+static jboolean JNI_LibvpxVp9Decoder_IsSupported(JNIEnv* jni,
>+                                                 const JavaParamRef<jclass>&) {
>   return !SupportedVP9Codecs().empty();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/CryptoOptionsTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/CryptoOptionsTest.java
>new file mode 100644
>index 0000000000000000000000000000000000000000..f03811e6fcad52c8504d68c67e92579cb22f49d6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/CryptoOptionsTest.java
>@@ -0,0 +1,74 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import static com.google.common.truth.Truth.assertThat;
>+
>+import org.chromium.testing.local.LocalRobolectricTestRunner;
>+import org.junit.Test;
>+import org.junit.runner.RunWith;
>+import org.robolectric.annotation.Config;
>+import org.webrtc.CryptoOptions;
>+
>+@RunWith(LocalRobolectricTestRunner.class)
>+@Config(manifest = Config.NONE)
>+public class CryptoOptionsTest {
>+  // Validates the builder builds by default all false options.
>+  @Test
>+  public void testBuilderDefaultsAreFalse() {
>+    CryptoOptions cryptoOptions = CryptoOptions.builder().createCryptoOptions();
>+    assertThat(cryptoOptions.getSrtp().getEnableGcmCryptoSuites()).isFalse();
>+    assertThat(cryptoOptions.getSrtp().getEnableAes128Sha1_32CryptoCipher()).isFalse();
>+    assertThat(cryptoOptions.getSrtp().getEnableEncryptedRtpHeaderExtensions()).isFalse();
>+    assertThat(cryptoOptions.getSFrame().getRequireFrameEncryption()).isFalse();
>+  }
>+
>+  // Validates the builder sets the correct parameters.
>+  @Test
>+  public void testBuilderCorrectlyInitializingGcmCrypto() {
>+    CryptoOptions cryptoOptions =
>+        CryptoOptions.builder().setEnableGcmCryptoSuites(true).createCryptoOptions();
>+    assertThat(cryptoOptions.getSrtp().getEnableGcmCryptoSuites()).isTrue();
>+    assertThat(cryptoOptions.getSrtp().getEnableAes128Sha1_32CryptoCipher()).isFalse();
>+    assertThat(cryptoOptions.getSrtp().getEnableEncryptedRtpHeaderExtensions()).isFalse();
>+    assertThat(cryptoOptions.getSFrame().getRequireFrameEncryption()).isFalse();
>+  }
>+
>+  @Test
>+  public void testBuilderCorrectlyInitializingAes128Sha1_32CryptoCipher() {
>+    CryptoOptions cryptoOptions =
>+        CryptoOptions.builder().setEnableAes128Sha1_32CryptoCipher(true).createCryptoOptions();
>+    assertThat(cryptoOptions.getSrtp().getEnableGcmCryptoSuites()).isFalse();
>+    assertThat(cryptoOptions.getSrtp().getEnableAes128Sha1_32CryptoCipher()).isTrue();
>+    assertThat(cryptoOptions.getSrtp().getEnableEncryptedRtpHeaderExtensions()).isFalse();
>+    assertThat(cryptoOptions.getSFrame().getRequireFrameEncryption()).isFalse();
>+  }
>+
>+  @Test
>+  public void testBuilderCorrectlyInitializingEncryptedRtpHeaderExtensions() {
>+    CryptoOptions cryptoOptions =
>+        CryptoOptions.builder().setEnableEncryptedRtpHeaderExtensions(true).createCryptoOptions();
>+    assertThat(cryptoOptions.getSrtp().getEnableGcmCryptoSuites()).isFalse();
>+    assertThat(cryptoOptions.getSrtp().getEnableAes128Sha1_32CryptoCipher()).isFalse();
>+    assertThat(cryptoOptions.getSrtp().getEnableEncryptedRtpHeaderExtensions()).isTrue();
>+    assertThat(cryptoOptions.getSFrame().getRequireFrameEncryption()).isFalse();
>+  }
>+
>+  @Test
>+  public void testBuilderCorrectlyInitializingRequireFrameEncryption() {
>+    CryptoOptions cryptoOptions =
>+        CryptoOptions.builder().setRequireFrameEncryption(true).createCryptoOptions();
>+    assertThat(cryptoOptions.getSrtp().getEnableGcmCryptoSuites()).isFalse();
>+    assertThat(cryptoOptions.getSrtp().getEnableAes128Sha1_32CryptoCipher()).isFalse();
>+    assertThat(cryptoOptions.getSrtp().getEnableEncryptedRtpHeaderExtensions()).isFalse();
>+    assertThat(cryptoOptions.getSFrame().getRequireFrameEncryption()).isTrue();
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm
>deleted file mode 100644
>index 04a5689417c20737278a8b2b022f121ffe99662a..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm
>+++ /dev/null
>@@ -1,58 +0,0 @@
>-/*
>- *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- *
>- */
>-
>-#import "WebRTC/RTCVideoCodecH264.h"
>-
>-#include "media/base/h264_profile_level_id.h"
>-
>-@interface RTCH264ProfileLevelId ()
>-
>-@property(nonatomic, assign) RTCH264Profile profile;
>-@property(nonatomic, assign) RTCH264Level level;
>-@property(nonatomic, strong) NSString *hexString;
>-
>-@end
>-
>-@implementation RTCH264ProfileLevelId
>-
>-@synthesize profile = _profile;
>-@synthesize level = _level;
>-@synthesize hexString = _hexString;
>-
>-- (instancetype)initWithHexString:(NSString *)hexString {
>-  if (self = [super init]) {
>-    self.hexString = hexString;
>-
>-    absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
>-        webrtc::H264::ParseProfileLevelId([hexString cStringUsingEncoding:NSUTF8StringEncoding]);
>-    if (profile_level_id.has_value()) {
>-      self.profile = static_cast<RTCH264Profile>(profile_level_id->profile);
>-      self.level = static_cast<RTCH264Level>(profile_level_id->level);
>-    }
>-  }
>-  return self;
>-}
>-
>-- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level {
>-  if (self = [super init]) {
>-    self.profile = profile;
>-    self.level = level;
>-
>-    absl::optional<std::string> hex_string =
>-        webrtc::H264::ProfileLevelIdToString(webrtc::H264::ProfileLevelId(
>-            static_cast<webrtc::H264::Profile>(profile), static_cast<webrtc::H264::Level>(level)));
>-    self.hexString =
>-        [NSString stringWithCString:hex_string.value_or("").c_str() encoding:NSUTF8StringEncoding];
>-  }
>-  return self;
>-}
>-
>-@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/logging/RTCCallbackLogger.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/logging/RTCCallbackLogger.h
>index 4867b31e365cc1b83b47c0966774b655049eed8c..2bce03fe0f8ac5e0946e1698fd5d27ebee15cf2c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/logging/RTCCallbackLogger.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/logging/RTCCallbackLogger.h
>@@ -15,6 +15,10 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>+typedef void (^RTCCallbackLoggerMessageHandler)(NSString *message);
>+typedef void (^RTCCallbackLoggerMessageAndSeverityHandler)(NSString *message,
>+                                                           RTCLoggingSeverity severity);
>+
> // This class intercepts WebRTC logs and forwards them to a registered block.
> // This class is not threadsafe.
> RTC_OBJC_EXPORT
>@@ -23,10 +27,12 @@ RTC_OBJC_EXPORT
> // The severity level to capture. The default is kRTCLoggingSeverityInfo.
> @property(nonatomic, assign) RTCLoggingSeverity severity;
> 
>-// The callback will be called on the same thread that does the logging, so
>-// if the logging callback can be slow it may be a good idea to implement
>-// dispatching to some other queue.
>-- (void)start:(nullable void (^)(NSString*))callback;
>+// The callback handler will be called on the same thread that does the
>+// logging, so if the logging callback can be slow it may be a good idea
>+// to implement dispatching to some other queue.
>+- (void)start:(nullable RTCCallbackLoggerMessageHandler)handler;
>+- (void)startWithMessageAndSeverityHandler:
>+        (nullable RTCCallbackLoggerMessageAndSeverityHandler)handler;
> 
> - (void)stop;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/logging/RTCCallbackLogger.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/logging/RTCCallbackLogger.mm
>index 8440d63791ef52ad22061425785c4e44800c4650..9bbb53043409a27c56a64c5a17cb84a9dc0fb884 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/logging/RTCCallbackLogger.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/logging/RTCCallbackLogger.mm
>@@ -18,11 +18,8 @@
> 
> class CallbackLogSink : public rtc::LogSink {
>  public:
>-  CallbackLogSink(void (^callbackHandler)(NSString *message)) {
>-    callback_handler_ = callbackHandler;
>-  }
>-
>-  ~CallbackLogSink() override { callback_handler_ = nil; }
>+  CallbackLogSink(RTCCallbackLoggerMessageHandler callbackHandler)
>+      : callback_handler_(callbackHandler) {}
> 
>   void OnLogMessage(const std::string &message) override {
>     if (callback_handler_) {
>@@ -31,12 +28,47 @@ class CallbackLogSink : public rtc::LogSink {
>   }
> 
>  private:
>-  void (^callback_handler_)(NSString *message);
>+  RTCCallbackLoggerMessageHandler callback_handler_;
>+};
>+
>+class CallbackWithSeverityLogSink : public rtc::LogSink {
>+ public:
>+  CallbackWithSeverityLogSink(RTCCallbackLoggerMessageAndSeverityHandler callbackHandler)
>+      : callback_handler_(callbackHandler) {}
>+
>+  void OnLogMessage(const std::string& message) override { RTC_NOTREACHED(); }
>+
>+  void OnLogMessage(const std::string& message, rtc::LoggingSeverity severity) override {
>+    if (callback_handler_) {
>+      RTCLoggingSeverity loggingSeverity = NativeSeverityToObjcSeverity(severity);
>+      callback_handler_([NSString stringWithUTF8String:message.c_str()], loggingSeverity);
>+    }
>+  }
>+
>+ private:
>+  static RTCLoggingSeverity NativeSeverityToObjcSeverity(rtc::LoggingSeverity severity) {
>+    switch (severity) {
>+      case rtc::LS_SENSITIVE:
>+        return RTCLoggingSeveritySensitive;
>+      case rtc::LS_VERBOSE:
>+        return RTCLoggingSeverityVerbose;
>+      case rtc::LS_INFO:
>+        return RTCLoggingSeverityInfo;
>+      case rtc::LS_WARNING:
>+        return RTCLoggingSeverityWarning;
>+      case rtc::LS_ERROR:
>+        return RTCLoggingSeverityError;
>+      case rtc::LS_NONE:
>+        return RTCLoggingSeverityNone;
>+    }
>+  }
>+
>+  RTCCallbackLoggerMessageAndSeverityHandler callback_handler_;
> };
> 
> @implementation RTCCallbackLogger {
>   BOOL _hasStarted;
>-  std::unique_ptr<CallbackLogSink> _logSink;
>+  std::unique_ptr<rtc::LogSink> _logSink;
> }
> 
> @synthesize severity = _severity;
>@@ -53,12 +85,24 @@ - (void)dealloc {
>   [self stop];
> }
> 
>-- (void)start:(nullable void (^)(NSString *))callback {
>+- (void)start:(nullable RTCCallbackLoggerMessageHandler)handler {
>+  if (_hasStarted) {
>+    return;
>+  }
>+
>+  _logSink.reset(new CallbackLogSink(handler));
>+
>+  rtc::LogMessage::AddLogToStream(_logSink.get(), [self rtcSeverity]);
>+  _hasStarted = YES;
>+}
>+
>+- (void)startWithMessageAndSeverityHandler:
>+        (nullable RTCCallbackLoggerMessageAndSeverityHandler)handler {
>   if (_hasStarted) {
>     return;
>   }
> 
>-  _logSink.reset(new CallbackLogSink(callback));
>+  _logSink.reset(new CallbackWithSeverityLogSink(handler));
> 
>   rtc::LogMessage::AddLogToStream(_logSink.get(), [self rtcSeverity]);
>   _hasStarted = YES;
>@@ -78,6 +122,8 @@ - (void)stop {
> 
> - (rtc::LoggingSeverity)rtcSeverity {
>   switch (_severity) {
>+    case RTCLoggingSeveritySensitive:
>+      return rtc::LS_SENSITIVE;
>     case RTCLoggingSeverityVerbose:
>       return rtc::LS_VERBOSE;
>     case RTCLoggingSeverityInfo:
>@@ -86,6 +132,8 @@ - (rtc::LoggingSeverity)rtcSeverity {
>       return rtc::LS_WARNING;
>     case RTCLoggingSeverityError:
>       return rtc::LS_ERROR;
>+    case RTCLoggingSeverityNone:
>+      return rtc::LS_NONE;
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.h
>index 9fb8b4df3591d7a67a15d74ea9e175a5eac0bf8a..f9e6edfd97415a82190b3793999b317e07d25705 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.h
>@@ -11,6 +11,7 @@
> #import <Foundation/Foundation.h>
> 
> #import "RTCCertificate.h"
>+#import "RTCCryptoOptions.h"
> #import "RTCMacros.h"
> 
> @class RTCIceServer;
>@@ -91,6 +92,19 @@ RTC_OBJC_EXPORT
> @property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
> @property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
> 
>+/** If set to YES, don't gather IPv6 ICE candidates.
>+ *  Default is NO.
>+ */
>+@property(nonatomic, assign) BOOL disableIPV6;
>+
>+/** If set to YES, don't gather IPv6 ICE candidates on Wi-Fi.
>+ *  Only intended to be used on specific devices. Certain phones disable IPv6
>+ *  when the screen is turned off and it would be better to just disable the
>+ *  IPv6 ICE candidates on Wi-Fi in those cases.
>+ *  Default is NO.
>+ */
>+@property(nonatomic, assign) BOOL disableIPV6OnWiFi;
>+
> /** By default, the PeerConnection will use a limited number of IPv6 network
>  *  interfaces, in order to avoid too many ICE candidate pairs being created
>  *  and delaying ICE completion.
>@@ -168,6 +182,35 @@ RTC_OBJC_EXPORT
>  */
> @property(nonatomic, assign) BOOL activeResetSrtpParams;
> 
>+/**
>+ * If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
>+ * that it should use the MediaTransportInterface.
>+ */
>+@property(nonatomic, assign) BOOL useMediaTransport;
>+
>+/**
>+ * If MediaTransportFactory is provided in PeerConnectionFactory, this flag informs PeerConnection
>+ * that it should use the MediaTransportInterface for data channels.
>+ */
>+@property(nonatomic, assign) BOOL useMediaTransportForDataChannels;
>+
>+/**
>+ * Defines advanced optional cryptographic settings related to SRTP and
>+ * frame encryption for native WebRTC. Setting this will overwrite any
>+ * options set through the PeerConnectionFactory (which is deprecated).
>+ */
>+@property(nonatomic, nullable) RTCCryptoOptions *cryptoOptions;
>+
>+/**
>+ * Time interval between audio RTCP reports.
>+ */
>+@property(nonatomic, assign) int rtcpAudioReportIntervalMs;
>+
>+/**
>+ * Time interval between video RTCP reports.
>+ */
>+@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
>+
> - (instancetype)init;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.mm
>index 73a0fa9aa5fd80db4cd27cd293b650b7b800695f..4041f8399fc8b35567cbd989a25427eeb4ba1ebb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCConfiguration.mm
>@@ -31,6 +31,8 @@ @synthesize rtcpMuxPolicy = _rtcpMuxPolicy;
> @synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
> @synthesize candidateNetworkPolicy = _candidateNetworkPolicy;
> @synthesize continualGatheringPolicy = _continualGatheringPolicy;
>+@synthesize disableIPV6 = _disableIPV6;
>+@synthesize disableIPV6OnWiFi = _disableIPV6OnWiFi;
> @synthesize maxIPv6Networks = _maxIPv6Networks;
> @synthesize disableLinkLocalNetworks = _disableLinkLocalNetworks;
> @synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
>@@ -48,6 +50,11 @@ @synthesize iceRegatherIntervalRange = _iceRegatherIntervalRange;
> @synthesize sdpSemantics = _sdpSemantics;
> @synthesize turnCustomizer = _turnCustomizer;
> @synthesize activeResetSrtpParams = _activeResetSrtpParams;
>+@synthesize useMediaTransport = _useMediaTransport;
>+@synthesize useMediaTransportForDataChannels = _useMediaTransportForDataChannels;
>+@synthesize cryptoOptions = _cryptoOptions;
>+@synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
>+@synthesize rtcpVideoReportIntervalMs = _rtcpVideoReportIntervalMs;
> 
> - (instancetype)init {
>   // Copy defaults.
>@@ -86,6 +93,8 @@ - (instancetype)initWithNativeConfiguration:
>     config.continual_gathering_policy;
>     _continualGatheringPolicy =
>         [[self class] continualGatheringPolicyForNativePolicy:nativePolicy];
>+    _disableIPV6 = config.disable_ipv6;
>+    _disableIPV6OnWiFi = config.disable_ipv6_on_wifi;
>     _maxIPv6Networks = config.max_ipv6_networks;
>     _disableLinkLocalNetworks = config.disable_link_local_networks;
>     _audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
>@@ -93,6 +102,8 @@ - (instancetype)initWithNativeConfiguration:
>     _iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
>     _iceBackupCandidatePairPingInterval =
>         config.ice_backup_candidate_pair_ping_interval;
>+    _useMediaTransport = config.use_media_transport;
>+    _useMediaTransportForDataChannels = config.use_media_transport_for_data_channels;
>     _keyType = RTCEncryptionKeyTypeECDSA;
>     _iceCandidatePoolSize = config.ice_candidate_pool_size;
>     _shouldPruneTurnPorts = config.prune_turn_ports;
>@@ -110,14 +121,27 @@ - (instancetype)initWithNativeConfiguration:
>     _sdpSemantics = [[self class] sdpSemanticsForNativeSdpSemantics:config.sdp_semantics];
>     _turnCustomizer = config.turn_customizer;
>     _activeResetSrtpParams = config.active_reset_srtp_params;
>+    if (config.crypto_options) {
>+      _cryptoOptions = [[RTCCryptoOptions alloc]
>+               initWithSrtpEnableGcmCryptoSuites:config.crypto_options->srtp
>+                                                     .enable_gcm_crypto_suites
>+             srtpEnableAes128Sha1_32CryptoCipher:config.crypto_options->srtp
>+                                                     .enable_aes128_sha1_32_crypto_cipher
>+          srtpEnableEncryptedRtpHeaderExtensions:config.crypto_options->srtp
>+                                                     .enable_encrypted_rtp_header_extensions
>+                    sframeRequireFrameEncryption:config.crypto_options->sframe
>+                                                     .require_frame_encryption];
>+    }
>+    _rtcpAudioReportIntervalMs = config.audio_rtcp_report_interval_ms();
>+    _rtcpVideoReportIntervalMs = config.video_rtcp_report_interval_ms();
>   }
>   return self;
> }
> 
> - (NSString *)description {
>-  static NSString *formatString =
>-      @"RTCConfiguration: "
>-      @"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n%d\n%@\n%@\n%d\n%d\n%d\n}\n";
>+  static NSString *formatString = @"RTCConfiguration: "
>+                                  @"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n"
>+                                  @"%d\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%@\n}\n";
> 
>   return [NSString
>       stringWithFormat:formatString,
>@@ -139,8 +163,11 @@ - (NSString *)description {
>                        _iceCheckMinInterval,
>                        _iceRegatherIntervalRange,
>                        _disableLinkLocalNetworks,
>+                       _disableIPV6,
>+                       _disableIPV6OnWiFi,
>                        _maxIPv6Networks,
>-                       _activeResetSrtpParams];
>+                       _activeResetSrtpParams,
>+                       _useMediaTransport];
> }
> 
> #pragma mark - Private
>@@ -166,6 +193,8 @@ - (webrtc::PeerConnectionInterface::RTCConfiguration *)
>       nativeCandidateNetworkPolicyForPolicy:_candidateNetworkPolicy];
>   nativeConfig->continual_gathering_policy = [[self class]
>       nativeContinualGatheringPolicyForPolicy:_continualGatheringPolicy];
>+  nativeConfig->disable_ipv6 = _disableIPV6;
>+  nativeConfig->disable_ipv6_on_wifi = _disableIPV6OnWiFi;
>   nativeConfig->max_ipv6_networks = _maxIPv6Networks;
>   nativeConfig->disable_link_local_networks = _disableLinkLocalNetworks;
>   nativeConfig->audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
>@@ -175,6 +204,8 @@ - (webrtc::PeerConnectionInterface::RTCConfiguration *)
>       _iceConnectionReceivingTimeout;
>   nativeConfig->ice_backup_candidate_pair_ping_interval =
>       _iceBackupCandidatePairPingInterval;
>+  nativeConfig->use_media_transport = _useMediaTransport;
>+  nativeConfig->use_media_transport_for_data_channels = _useMediaTransportForDataChannels;
>   rtc::KeyType keyType =
>       [[self class] nativeEncryptionKeyTypeForKeyType:_keyType];
>   if (_certificate != nullptr) {
>@@ -222,6 +253,20 @@ - (webrtc::PeerConnectionInterface::RTCConfiguration *)
>     nativeConfig->turn_customizer = _turnCustomizer;
>   }
>   nativeConfig->active_reset_srtp_params = _activeResetSrtpParams ? true : false;
>+  if (_cryptoOptions) {
>+    webrtc::CryptoOptions nativeCryptoOptions;
>+    nativeCryptoOptions.srtp.enable_gcm_crypto_suites =
>+        _cryptoOptions.srtpEnableGcmCryptoSuites ? true : false;
>+    nativeCryptoOptions.srtp.enable_aes128_sha1_32_crypto_cipher =
>+        _cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher ? true : false;
>+    nativeCryptoOptions.srtp.enable_encrypted_rtp_header_extensions =
>+        _cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions ? true : false;
>+    nativeCryptoOptions.sframe.require_frame_encryption =
>+        _cryptoOptions.sframeRequireFrameEncryption ? true : false;
>+    nativeConfig->crypto_options = absl::optional<webrtc::CryptoOptions>(nativeCryptoOptions);
>+  }
>+  nativeConfig->set_audio_rtcp_report_interval_ms(_rtcpAudioReportIntervalMs);
>+  nativeConfig->set_video_rtcp_report_interval_ms(_rtcpVideoReportIntervalMs);
>   return nativeConfig.release();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..b465bb5a73bdfb8e37f882514a9cb1696c03cfcb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.h
>@@ -0,0 +1,63 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+
>+#import "RTCMacros.h"
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+/**
>+ * Objective-C bindings for webrtc::CryptoOptions. This API had to be flattened
>+ * as Objective-C doesn't support nested structures.
>+ */
>+RTC_OBJC_EXPORT
>+@interface RTCCryptoOptions : NSObject
>+
>+/**
>+ * Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
>+ * if both sides enable it
>+ */
>+@property(nonatomic, assign) BOOL srtpEnableGcmCryptoSuites;
>+/**
>+ * If set to true, the (potentially insecure) crypto cipher
>+ * SRTP_AES128_CM_SHA1_32 will be included in the list of supported ciphers
>+ * during negotiation. It will only be used if both peers support it and no
>+ * other ciphers get preferred.
>+ */
>+@property(nonatomic, assign) BOOL srtpEnableAes128Sha1_32CryptoCipher;
>+/**
>+ * If set to true, encrypted RTP header extensions as defined in RFC 6904
>+ * will be negotiated. They will only be used if both peers support them.
>+ */
>+@property(nonatomic, assign) BOOL srtpEnableEncryptedRtpHeaderExtensions;
>+
>+/**
>+ * If set all RtpSenders must have an FrameEncryptor attached to them before
>+ * they are allowed to send packets. All RtpReceivers must have a
>+ * FrameDecryptor attached to them before they are able to receive packets.
>+ */
>+@property(nonatomic, assign) BOOL sframeRequireFrameEncryption;
>+
>+/**
>+ * Initializes CryptoOptions with all possible options set explicitly. This
>+ * is done when converting from a native RTCConfiguration.crypto_options.
>+ */
>+- (instancetype)initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
>+              srtpEnableAes128Sha1_32CryptoCipher:(BOOL)srtpEnableAes128Sha1_32CryptoCipher
>+           srtpEnableEncryptedRtpHeaderExtensions:(BOOL)srtpEnableEncryptedRtpHeaderExtensions
>+                     sframeRequireFrameEncryption:(BOOL)sframeRequireFrameEncryption
>+    NS_DESIGNATED_INITIALIZER;
>+
>+- (instancetype)init NS_UNAVAILABLE;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.mm
>new file mode 100644
>index 0000000000000000000000000000000000000000..a059f755994e743030321da5f2e0432f0b211ff9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.mm
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "RTCCryptoOptions.h"
>+
>+@implementation RTCCryptoOptions
>+
>+@synthesize srtpEnableGcmCryptoSuites = _srtpEnableGcmCryptoSuites;
>+@synthesize srtpEnableAes128Sha1_32CryptoCipher = _srtpEnableAes128Sha1_32CryptoCipher;
>+@synthesize srtpEnableEncryptedRtpHeaderExtensions = _srtpEnableEncryptedRtpHeaderExtensions;
>+@synthesize sframeRequireFrameEncryption = _sframeRequireFrameEncryption;
>+
>+- (instancetype)initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
>+              srtpEnableAes128Sha1_32CryptoCipher:(BOOL)srtpEnableAes128Sha1_32CryptoCipher
>+           srtpEnableEncryptedRtpHeaderExtensions:(BOOL)srtpEnableEncryptedRtpHeaderExtensions
>+                     sframeRequireFrameEncryption:(BOOL)sframeRequireFrameEncryption {
>+  if (self = [super init]) {
>+    _srtpEnableGcmCryptoSuites = srtpEnableGcmCryptoSuites;
>+    _srtpEnableAes128Sha1_32CryptoCipher = srtpEnableAes128Sha1_32CryptoCipher;
>+    _srtpEnableEncryptedRtpHeaderExtensions = srtpEnableEncryptedRtpHeaderExtensions;
>+    _sframeRequireFrameEncryption = sframeRequireFrameEncryption;
>+  }
>+  return self;
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.h
>index 1b125f1039a22f229f092b9a76cf342b08c6bc41..c77a7ad85d7bb7a8dbb5f0a8cd7109577d582ef4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.h
>@@ -10,7 +10,7 @@
> 
> #import "base/RTCEncodedImage.h"
> 
>-#include "common_video/include/video_frame.h"
>+#include "api/video/encoded_image.h"
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
>index 94cb1f110f96cf03cd51ad7bed450e15b8218947..f9901bd327e8f24bcf0bf176b004ff9e1af5a748 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
>@@ -35,7 +35,6 @@ - (instancetype)initWithNativeEncodedImage:(webrtc::EncodedImage)encodedImage {
>     self.contentType = (encodedImage.content_type_ == webrtc::VideoContentType::SCREENSHARE) ?
>         RTCVideoContentTypeScreenshare :
>         RTCVideoContentTypeUnspecified;
>-    self.spatialIndex = encodedImage.SpatialIndex() ? *encodedImage.SpatialIndex() : 0;
>   }
> 
>   return self;
>@@ -57,11 +56,11 @@ - (webrtc::EncodedImage)nativeEncodedImage {
>   encodedImage.rotation_ = webrtc::VideoRotation(self.rotation);
>   encodedImage._completeFrame = self.completeFrame;
>   encodedImage.qp_ = self.qp ? self.qp.intValue : -1;
>+  encodedImage.SetSpatialIndex(self.spatialIndex);
>   encodedImage.content_type_ = (self.contentType == RTCVideoContentTypeScreenshare) ?
>       webrtc::VideoContentType::SCREENSHARE :
>       webrtc::VideoContentType::UNSPECIFIED;
>-  if (self.spatialIndex)
>-    encodedImage.SetSpatialIndex(self.spatialIndex);
>+
>   return encodedImage;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCFieldTrials.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCFieldTrials.h
>index effc470686f810b4015f23b51c4a4c1f391740eb..cf648d32c3a91081beb1d271f55c9d2ba221720b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCFieldTrials.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCFieldTrials.h
>@@ -20,24 +20,12 @@ RTC_EXTERN NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey;
> RTC_EXTERN NSString * const kRTCFieldTrialSendSideBweWithOverheadKey;
> RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03AdvertisedKey;
> RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03Key;
>-RTC_EXTERN NSString * const kRTCFieldTrialImprovedBitrateEstimateKey;
> RTC_EXTERN NSString * const kRTCFieldTrialH264HighProfileKey;
> RTC_EXTERN NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey;
> 
> /** The valid value for field trials above. */
> RTC_EXTERN NSString * const kRTCFieldTrialEnabledValue;
> 
>-/** Use a string returned by RTCFieldTrialMedianSlopeFilterValue as the value. */
>-RTC_EXTERN NSString * const kRTCFieldTrialMedianSlopeFilterKey;
>-RTC_EXTERN NSString *RTCFieldTrialMedianSlopeFilterValue(
>-    size_t windowSize, double thresholdGain);
>-
>-/** Use a string returned by RTCFieldTrialTrendlineFilterValue as the value. */
>-RTC_EXTERN NSString * const kRTCFieldTrialTrendlineFilterKey;
>-/** Returns a valid value for kRTCFieldTrialTrendlineFilterKey. */
>-RTC_EXTERN NSString *RTCFieldTrialTrendlineFilterValue(
>-    size_t windowSize, double smoothingCoeff, double thresholdGain);
>-
> /** Initialize field trials using a dictionary mapping field trial keys to their
>  * values. See above for valid keys and values. Must be called before any other
>  * call into WebRTC. See: webrtc/system_wrappers/include/field_trial.h
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCFieldTrials.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCFieldTrials.mm
>index 2212c7dec9714b799bd291f57ff136fd854c6206..127ce6feb89f37d3a414b55c907318a4cd5430aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCFieldTrials.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCFieldTrials.mm
>@@ -23,9 +23,6 @@ NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey = @"WebRTC-Audio-ABWENoTW
> NSString * const kRTCFieldTrialSendSideBweWithOverheadKey = @"WebRTC-SendSideBwe-WithOverhead";
> NSString * const kRTCFieldTrialFlexFec03AdvertisedKey = @"WebRTC-FlexFEC-03-Advertised";
> NSString * const kRTCFieldTrialFlexFec03Key = @"WebRTC-FlexFEC-03";
>-NSString * const kRTCFieldTrialImprovedBitrateEstimateKey = @"WebRTC-ImprovedBitrateEstimate";
>-NSString * const kRTCFieldTrialMedianSlopeFilterKey = @"WebRTC-BweMedianSlopeFilter";
>-NSString * const kRTCFieldTrialTrendlineFilterKey = @"WebRTC-BweTrendlineFilter";
> NSString * const kRTCFieldTrialH264HighProfileKey = @"WebRTC-H264HighProfile";
> NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey =
>     @"WebRTC-Audio-MinimizeResamplingOnMobile";
>@@ -33,18 +30,6 @@ NSString * const kRTCFieldTrialEnabledValue = @"Enabled";
> 
> static std::unique_ptr<char[]> gFieldTrialInitString;
> 
>-NSString *RTCFieldTrialMedianSlopeFilterValue(
>-    size_t windowSize, double thresholdGain) {
>-  NSString *format = @"Enabled-%zu,%lf";
>-  return [NSString stringWithFormat:format, windowSize, thresholdGain];
>-}
>-
>-NSString *RTCFieldTrialTrendlineFilterValue(
>-    size_t windowSize, double smoothingCoeff, double thresholdGain) {
>-  NSString *format = @"Enabled-%zu,%lf,%lf";
>-  return [NSString stringWithFormat:format, windowSize, smoothingCoeff, thresholdGain];
>-}
>-
> void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials) {
>   if (!fieldTrials) {
>     RTCLogWarning(@"No fieldTrials provided.");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection+Private.h
>index b6440cd434a857ebd7a3216f82fb333b977b24c1..d26eb2e592554bd23cfb4db8a18713dcc7321111 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection+Private.h
>@@ -39,6 +39,8 @@ class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
> 
>   void OnIceConnectionChange(PeerConnectionInterface::IceConnectionState new_state) override;
> 
>+  void OnConnectionChange(PeerConnectionInterface::PeerConnectionState new_state) override;
>+
>   void OnIceGatheringChange(PeerConnectionInterface::IceGatheringState new_state) override;
> 
>   void OnIceCandidate(const IceCandidateInterface *candidate) override;
>@@ -85,11 +87,19 @@ class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
> + (webrtc::PeerConnectionInterface::IceConnectionState)nativeIceConnectionStateForState:
>         (RTCIceConnectionState)state;
> 
>++ (webrtc::PeerConnectionInterface::PeerConnectionState)nativeConnectionStateForState:
>+        (RTCPeerConnectionState)state;
>+
> + (RTCIceConnectionState)iceConnectionStateForNativeState:
>         (webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
> 
>++ (RTCPeerConnectionState)connectionStateForNativeState:
>+        (webrtc::PeerConnectionInterface::PeerConnectionState)nativeState;
>+
> + (NSString *)stringForIceConnectionState:(RTCIceConnectionState)state;
> 
>++ (NSString *)stringForConnectionState:(RTCPeerConnectionState)state;
>+
> + (webrtc::PeerConnectionInterface::IceGatheringState)nativeIceGatheringStateForState:
>         (RTCIceGatheringState)state;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.h
>index 0179ec0e26df93ab51ee2b7df5722c1f2ff4cba1..393a50bc8984ec99c014b74e6341e5e7af26c532 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.h
>@@ -57,6 +57,16 @@ typedef NS_ENUM(NSInteger, RTCIceConnectionState) {
>   RTCIceConnectionStateCount,
> };
> 
>+/** Represents the combined ice+dtls connection state of the peer connection. */
>+typedef NS_ENUM(NSInteger, RTCPeerConnectionState) {
>+  RTCPeerConnectionStateNew,
>+  RTCPeerConnectionStateConnecting,
>+  RTCPeerConnectionStateConnected,
>+  RTCPeerConnectionStateDisconnected,
>+  RTCPeerConnectionStateFailed,
>+  RTCPeerConnectionStateClosed,
>+};
>+
> /** Represents the ice gathering state of the peer connection. */
> typedef NS_ENUM(NSInteger, RTCIceGatheringState) {
>   RTCIceGatheringStateNew,
>@@ -115,11 +125,14 @@ RTC_OBJC_EXPORT
>  *  This is only called with RTCSdpSemanticsUnifiedPlan specified.
>  */
> @optional
>+/** Called any time the PeerConnectionState changes. */
>+- (void)peerConnection:(RTCPeerConnection *)peerConnection
>+    didChangeConnectionState:(RTCPeerConnectionState)newState;
>+
> - (void)peerConnection:(RTCPeerConnection *)peerConnection
>     didStartReceivingOnTransceiver:(RTCRtpTransceiver *)transceiver;
> 
> /** Called when a receiver and its track are created. */
>-@optional
> - (void)peerConnection:(RTCPeerConnection *)peerConnection
>         didAddReceiver:(RTCRtpReceiver *)rtpReceiver
>                streams:(NSArray<RTCMediaStream *> *)mediaStreams;
>@@ -145,6 +158,7 @@ RTC_OBJC_EXPORT
> @property(nonatomic, readonly, nullable) RTCSessionDescription *remoteDescription;
> @property(nonatomic, readonly) RTCSignalingState signalingState;
> @property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
>+@property(nonatomic, readonly) RTCPeerConnectionState connectionState;
> @property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
> @property(nonatomic, readonly, copy) RTCConfiguration *configuration;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.mm
>index 5277489747b0b9d388a039c0f3c8070252772e2c..b59dafec35ff92254933a74e9ba150d77554d3b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnection.mm
>@@ -29,6 +29,7 @@
> #include <memory>
> 
> #include "api/jsepicecandidate.h"
>+#include "api/media_transport_interface.h"
> #include "rtc_base/checks.h"
> 
> NSString * const kRTCPeerConnectionErrorDomain =
>@@ -174,11 +175,17 @@ void PeerConnectionDelegateAdapter::OnRenegotiationNeeded() {
> 
> void PeerConnectionDelegateAdapter::OnIceConnectionChange(
>     PeerConnectionInterface::IceConnectionState new_state) {
>-  RTCIceConnectionState state =
>-      [[RTCPeerConnection class] iceConnectionStateForNativeState:new_state];
>-  RTCPeerConnection *peer_connection = peer_connection_;
>-  [peer_connection.delegate peerConnection:peer_connection
>-               didChangeIceConnectionState:state];
>+  RTCIceConnectionState state = [RTCPeerConnection iceConnectionStateForNativeState:new_state];
>+  [peer_connection_.delegate peerConnection:peer_connection_ didChangeIceConnectionState:state];
>+}
>+
>+void PeerConnectionDelegateAdapter::OnConnectionChange(
>+    PeerConnectionInterface::PeerConnectionState new_state) {
>+  if ([peer_connection_.delegate
>+          respondsToSelector:@selector(peerConnection:didChangeConnectionState:)]) {
>+    RTCPeerConnectionState state = [RTCPeerConnection connectionStateForNativeState:new_state];
>+    [peer_connection_.delegate peerConnection:peer_connection_ didChangeConnectionState:state];
>+  }
> }
> 
> void PeerConnectionDelegateAdapter::OnIceGatheringChange(
>@@ -321,6 +328,10 @@ - (RTCIceConnectionState)iceConnectionState {
>       _peerConnection->ice_connection_state()];
> }
> 
>+- (RTCPeerConnectionState)connectionState {
>+  return [[self class] connectionStateForNativeState:_peerConnection->peer_connection_state()];
>+}
>+
> - (RTCIceGatheringState)iceGatheringState {
>   return [[self class] iceGatheringStateForNativeState:
>       _peerConnection->ice_gathering_state()];
>@@ -631,6 +642,59 @@ + (NSString *)stringForSignalingState:(RTCSignalingState)state {
>   }
> }
> 
>++ (webrtc::PeerConnectionInterface::PeerConnectionState)nativeConnectionStateForState:
>+        (RTCPeerConnectionState)state {
>+  switch (state) {
>+    case RTCPeerConnectionStateNew:
>+      return webrtc::PeerConnectionInterface::PeerConnectionState::kNew;
>+    case RTCPeerConnectionStateConnecting:
>+      return webrtc::PeerConnectionInterface::PeerConnectionState::kConnecting;
>+    case RTCPeerConnectionStateConnected:
>+      return webrtc::PeerConnectionInterface::PeerConnectionState::kConnected;
>+    case RTCPeerConnectionStateFailed:
>+      return webrtc::PeerConnectionInterface::PeerConnectionState::kFailed;
>+    case RTCPeerConnectionStateDisconnected:
>+      return webrtc::PeerConnectionInterface::PeerConnectionState::kDisconnected;
>+    case RTCPeerConnectionStateClosed:
>+      return webrtc::PeerConnectionInterface::PeerConnectionState::kClosed;
>+  }
>+}
>+
>++ (RTCPeerConnectionState)connectionStateForNativeState:
>+        (webrtc::PeerConnectionInterface::PeerConnectionState)nativeState {
>+  switch (nativeState) {
>+    case webrtc::PeerConnectionInterface::PeerConnectionState::kNew:
>+      return RTCPeerConnectionStateNew;
>+    case webrtc::PeerConnectionInterface::PeerConnectionState::kConnecting:
>+      return RTCPeerConnectionStateConnecting;
>+    case webrtc::PeerConnectionInterface::PeerConnectionState::kConnected:
>+      return RTCPeerConnectionStateConnected;
>+    case webrtc::PeerConnectionInterface::PeerConnectionState::kFailed:
>+      return RTCPeerConnectionStateFailed;
>+    case webrtc::PeerConnectionInterface::PeerConnectionState::kDisconnected:
>+      return RTCPeerConnectionStateDisconnected;
>+    case webrtc::PeerConnectionInterface::PeerConnectionState::kClosed:
>+      return RTCPeerConnectionStateClosed;
>+  }
>+}
>+
>++ (NSString *)stringForConnectionState:(RTCPeerConnectionState)state {
>+  switch (state) {
>+    case RTCPeerConnectionStateNew:
>+      return @"NEW";
>+    case RTCPeerConnectionStateConnecting:
>+      return @"CONNECTING";
>+    case RTCPeerConnectionStateConnected:
>+      return @"CONNECTED";
>+    case RTCPeerConnectionStateFailed:
>+      return @"FAILED";
>+    case RTCPeerConnectionStateDisconnected:
>+      return @"DISCONNECTED";
>+    case RTCPeerConnectionStateClosed:
>+      return @"CLOSED";
>+  }
>+}
>+
> + (webrtc::PeerConnectionInterface::IceConnectionState)
>     nativeIceConnectionStateForState:(RTCIceConnectionState)state {
>   switch (state) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
>index 60aa08f7ca3a023faa03c3329fbceac1e71a4e2a..2752cf411766b9046f4123232f34f4a45a701a96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
>@@ -17,6 +17,7 @@ namespace webrtc {
> class AudioDeviceModule;
> class AudioEncoderFactory;
> class AudioDecoderFactory;
>+class MediaTransportFactory;
> class VideoEncoderFactory;
> class VideoDecoderFactory;
> class AudioProcessing;
>@@ -49,6 +50,25 @@ NS_ASSUME_NONNULL_BEGIN
>                             audioProcessingModule:
>                                 (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
> 
>+- (instancetype)
>+    initWithNativeAudioEncoderFactory:
>+        (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
>+            nativeAudioDecoderFactory:
>+                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
>+            nativeVideoEncoderFactory:
>+                (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
>+            nativeVideoDecoderFactory:
>+                (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
>+                    audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule
>+                audioProcessingModule:
>+                    (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
>+                mediaTransportFactory:
>+                    (std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory;
>+
>+- (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)encoderFactory
>+                        decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory
>+                 mediaTransportFactory:
>+                     (std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory;
> @end
> 
> NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
>index 403db040fb8622648d4832a0ec2086097ab37d78..48476f2368f0145240deb50e454ce6938979cfe4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
>@@ -50,6 +50,7 @@
> // TODO(zhihuang): Remove nogncheck once MediaEngineInterface is moved to C++
> // API layer.
> #include "absl/memory/memory.h"
>+#include "api/media_transport_interface.h"
> #include "media/engine/webrtcmediaengine.h"  // nogncheck
> 
> @implementation RTCPeerConnectionFactory {
>@@ -80,12 +81,15 @@ - (instancetype)init {
>                        nativeVideoDecoderFactory:webrtc::ObjCToNativeVideoDecoderFactory(
>                                                      [[RTCVideoDecoderFactoryH264 alloc] init])
>                                audioDeviceModule:[self audioDeviceModule]
>-                           audioProcessingModule:nullptr];
>+                           audioProcessingModule:nullptr
>+                           mediaTransportFactory:nullptr];
> #endif
> }
> 
> - (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)encoderFactory
>-                        decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory {
>+                        decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory
>+                 mediaTransportFactory:
>+                     (std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory {
> #ifdef HAVE_NO_MEDIA
>   return [self initWithNoMedia];
> #else
>@@ -102,9 +106,16 @@ - (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)enco
>                        nativeVideoEncoderFactory:std::move(native_encoder_factory)
>                        nativeVideoDecoderFactory:std::move(native_decoder_factory)
>                                audioDeviceModule:[self audioDeviceModule]
>-                           audioProcessingModule:nullptr];
>+                           audioProcessingModule:nullptr
>+                           mediaTransportFactory:std::move(mediaTransportFactory)];
> #endif
> }
>+- (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)encoderFactory
>+                        decoderFactory:(nullable id<RTCVideoDecoderFactory>)decoderFactory {
>+  return [self initWithEncoderFactory:encoderFactory
>+                       decoderFactory:decoderFactory
>+                mediaTransportFactory:nullptr];
>+}
> 
> - (instancetype)initNative {
>   if (self = [super init]) {
>@@ -152,20 +163,57 @@ - (instancetype)initWithNativeAudioEncoderFactory:
>                                     (nullable webrtc::AudioDeviceModule *)audioDeviceModule
>                             audioProcessingModule:
>                                 (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule {
>+  return [self initWithNativeAudioEncoderFactory:audioEncoderFactory
>+                       nativeAudioDecoderFactory:audioDecoderFactory
>+                       nativeVideoEncoderFactory:std::move(videoEncoderFactory)
>+                       nativeVideoDecoderFactory:std::move(videoDecoderFactory)
>+                               audioDeviceModule:audioDeviceModule
>+                           audioProcessingModule:audioProcessingModule
>+                           mediaTransportFactory:nullptr];
>+}
>+
>+- (instancetype)
>+    initWithNativeAudioEncoderFactory:
>+        (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
>+            nativeAudioDecoderFactory:
>+                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
>+            nativeVideoEncoderFactory:
>+                (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
>+            nativeVideoDecoderFactory:
>+                (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
>+                    audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule
>+                audioProcessingModule:
>+                    (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
>+                mediaTransportFactory:
>+                    (std::unique_ptr<webrtc::MediaTransportFactory>)mediaTransportFactory {
> #ifdef HAVE_NO_MEDIA
>   return [self initWithNoMedia];
> #else
>   if (self = [self initNative]) {
>-    _nativeFactory = webrtc::CreatePeerConnectionFactory(_networkThread.get(),
>-                                                         _workerThread.get(),
>-                                                         _signalingThread.get(),
>-                                                         audioDeviceModule,
>-                                                         audioEncoderFactory,
>-                                                         audioDecoderFactory,
>-                                                         std::move(videoEncoderFactory),
>-                                                         std::move(videoDecoderFactory),
>-                                                         nullptr,  // audio mixer
>-                                                         audioProcessingModule);
>+    if (!audioProcessingModule) audioProcessingModule = webrtc::AudioProcessingBuilder().Create();
>+
>+    std::unique_ptr<cricket::MediaEngineInterface> media_engine =
>+        cricket::WebRtcMediaEngineFactory::Create(audioDeviceModule,
>+                                                  audioEncoderFactory,
>+                                                  audioDecoderFactory,
>+                                                  std::move(videoEncoderFactory),
>+                                                  std::move(videoDecoderFactory),
>+                                                  nullptr,  // audio mixer
>+                                                  audioProcessingModule);
>+
>+    std::unique_ptr<webrtc::CallFactoryInterface> call_factory = webrtc::CreateCallFactory();
>+
>+    std::unique_ptr<webrtc::RtcEventLogFactoryInterface> event_log_factory =
>+        webrtc::CreateRtcEventLogFactory();
>+    webrtc::PeerConnectionFactoryDependencies dependencies;
>+    dependencies.network_thread = _networkThread.get();
>+    dependencies.worker_thread = _workerThread.get();
>+    dependencies.signaling_thread = _signalingThread.get();
>+    dependencies.media_engine = std::move(media_engine);
>+    dependencies.call_factory = std::move(call_factory);
>+    dependencies.event_log_factory = std::move(event_log_factory);
>+    dependencies.media_transport_factory = std::move(mediaTransportFactory);
>+    _nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
>     NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
>   }
>   return self;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
>index a26a639e28e44f21b8c2756d4fd0c2137d27115c..0adaa30e4f088946b6df47504fc5b91189fbeafa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
>@@ -13,6 +13,7 @@
> 
> #include "api/audio_codecs/audio_decoder_factory.h"
> #include "api/audio_codecs/audio_encoder_factory.h"
>+#include "api/media_transport_interface.h"
> #include "api/video_codecs/video_decoder_factory.h"
> #include "api/video_codecs/video_encoder_factory.h"
> #include "modules/audio_device/include/audio_device.h"
>@@ -25,6 +26,7 @@ @implementation RTCPeerConnectionFactoryBuilder {
>   rtc::scoped_refptr<webrtc::AudioDecoderFactory> _audioDecoderFactory;
>   rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
>   rtc::scoped_refptr<webrtc::AudioProcessing> _audioProcessingModule;
>+  std::unique_ptr<webrtc::MediaTransportFactory> _mediaTransportFactory;
> }
> 
> + (RTCPeerConnectionFactoryBuilder *)builder {
>@@ -38,7 +40,8 @@ - (RTCPeerConnectionFactory *)createPeerConnectionFactory {
>                           nativeVideoEncoderFactory:std::move(_videoEncoderFactory)
>                           nativeVideoDecoderFactory:std::move(_videoDecoderFactory)
>                                   audioDeviceModule:_audioDeviceModule
>-                              audioProcessingModule:_audioProcessingModule];
>+                              audioProcessingModule:_audioProcessingModule
>+                              mediaTransportFactory:std::move(_mediaTransportFactory)];
> }
> 
> - (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h
>index af327f987b27bca77b148d30f980d962646b3df7..4bec8695bdbf1979f50739ef95c64ade13590b08 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h
>@@ -31,10 +31,6 @@ RTC_OBJC_EXPORT
> 
> @property(nonatomic, assign) BOOL ignoreEthernetNetworkAdapter;
> 
>-@property(nonatomic, assign) BOOL enableAes128Sha1_32CryptoCipher;
>-
>-@property(nonatomic, assign) BOOL enableGcmCryptoSuites;
>-
> - (instancetype)init NS_DESIGNATED_INITIALIZER;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm
>index 103a130390c062bc6ec76ca0ae77c0f52fec3fe9..f0cc6a6c81f0c11ef622ce2b3e283e293bd20a35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm
>@@ -34,8 +34,6 @@ @synthesize ignoreVPNNetworkAdapter = _ignoreVPNNetworkAdapter;
> @synthesize ignoreCellularNetworkAdapter = _ignoreCellularNetworkAdapter;
> @synthesize ignoreWiFiNetworkAdapter = _ignoreWiFiNetworkAdapter;
> @synthesize ignoreEthernetNetworkAdapter = _ignoreEthernetNetworkAdapter;
>-@synthesize enableAes128Sha1_32CryptoCipher = _enableAes128Sha1_32CryptoCipher;
>-@synthesize enableGcmCryptoSuites = _enableGcmCryptoSuites;
> 
> - (instancetype)init {
>   return [super init];
>@@ -52,9 +50,6 @@ - (webrtc::PeerConnectionFactoryInterface::Options)nativeOptions {
>   setNetworkBit(&options, rtc::ADAPTER_TYPE_WIFI, self.ignoreWiFiNetworkAdapter);
>   setNetworkBit(&options, rtc::ADAPTER_TYPE_ETHERNET, self.ignoreEthernetNetworkAdapter);
> 
>-  options.crypto_options.enable_aes128_sha1_32_crypto_cipher = self.enableAes128Sha1_32CryptoCipher;
>-  options.crypto_options.enable_gcm_crypto_suites = self.enableGcmCryptoSuites;
>-
>   return options;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver.h
>index 8b962e4884b142f29b4626ba70d486e2b2f9b675..8ef3fc1d42e7a30178608e3641ae98dec7ba7f96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver.h
>@@ -28,6 +28,7 @@ typedef NS_ENUM(NSInteger, RTCRtpTransceiverDirection) {
>  *  RTCPeerConnection.addTransceiver.
>  *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
>  */
>+RTC_OBJC_EXPORT
> @interface RTCRtpTransceiverInit : NSObject
> 
> /** Direction of the RTCRtpTransceiver. See RTCRtpTransceiver.direction. */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoSource+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoSource+Private.h
>index 2441e0cdbe6248aaebb3e88ee409257d44d0efe6..828aad87bd07498ce942e2851dd95a0bd91ab966 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoSource+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoSource+Private.h
>@@ -13,6 +13,7 @@
> #import "RTCMediaSource+Private.h"
> 
> #include "api/mediastreaminterface.h"
>+#include "rtc_base/thread.h"
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCLogging.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCLogging.h
>index be62db25b569c8443ee0525cb285afdc56c61ffc..b20a4b4d428b580600e07464c3660241eed133ce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCLogging.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCLogging.h
>@@ -14,10 +14,12 @@
> 
> // Subset of rtc::LoggingSeverity.
> typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
>+  RTCLoggingSeveritySensitive,
>   RTCLoggingSeverityVerbose,
>   RTCLoggingSeverityInfo,
>   RTCLoggingSeverityWarning,
>   RTCLoggingSeverityError,
>+  RTCLoggingSeverityNone,
> };
> 
> // Wrapper for C++ RTC_LOG(sev) macros.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCLogging.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCLogging.mm
>index 5c8f9fb4f8bf944f1818f4484dafca5430f61a9a..3aae47a240cc400a2b2ff0b0bf1bf03454369385 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCLogging.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCLogging.mm
>@@ -14,6 +14,8 @@
> 
> rtc::LoggingSeverity RTCGetNativeLoggingSeverity(RTCLoggingSeverity severity) {
>   switch (severity) {
>+    case RTCLoggingSeveritySensitive:
>+      return rtc::LS_SENSITIVE;
>     case RTCLoggingSeverityVerbose:
>       return rtc::LS_VERBOSE;
>     case RTCLoggingSeverityInfo:
>@@ -22,6 +24,8 @@ rtc::LoggingSeverity RTCGetNativeLoggingSeverity(RTCLoggingSeverity severity) {
>       return rtc::LS_WARNING;
>     case RTCLoggingSeverityError:
>       return rtc::LS_ERROR;
>+    case RTCLoggingSeverityNone:
>+      return rtc::LS_NONE;
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/capturer/RTCCameraVideoCapturer.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/capturer/RTCCameraVideoCapturer.m
>index 2909405e7d8c511c580048c381dbf3fba32ede97..db833b398cbf1cc1211d996d3b21f9757224889b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/capturer/RTCCameraVideoCapturer.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/capturer/RTCCameraVideoCapturer.m
>@@ -26,18 +26,18 @@
> 
> @interface RTCCameraVideoCapturer ()<AVCaptureVideoDataOutputSampleBufferDelegate>
> @property(nonatomic, readonly) dispatch_queue_t frameQueue;
>+@property(nonatomic, strong) AVCaptureDevice *currentDevice;
>+@property(nonatomic, assign) BOOL hasRetriedOnFatalError;
>+@property(nonatomic, assign) BOOL isRunning;
>+// Will the session be running once all asynchronous operations have been completed?
>+@property(nonatomic, assign) BOOL willBeRunning;
> @end
> 
> @implementation RTCCameraVideoCapturer {
>   AVCaptureVideoDataOutput *_videoDataOutput;
>   AVCaptureSession *_captureSession;
>-  AVCaptureDevice *_currentDevice;
>   FourCharCode _preferredOutputPixelFormat;
>   FourCharCode _outputPixelFormat;
>-  BOOL _hasRetriedOnFatalError;
>-  BOOL _isRunning;
>-  // Will the session be running once all asynchronous operations have been completed?
>-  BOOL _willBeRunning;
>   RTCVideoRotation _rotation;
> #if TARGET_OS_IPHONE
>   UIDeviceOrientation _orientation;
>@@ -46,6 +46,10 @@ @implementation RTCCameraVideoCapturer {
> 
> @synthesize frameQueue = _frameQueue;
> @synthesize captureSession = _captureSession;
>+@synthesize currentDevice = _currentDevice;
>+@synthesize hasRetriedOnFatalError = _hasRetriedOnFatalError;
>+@synthesize isRunning = _isRunning;
>+@synthesize willBeRunning = _willBeRunning;
> 
> - (instancetype)init {
>   return [self initWithDelegate:nil captureSession:[[AVCaptureSession alloc] init]];
>@@ -157,25 +161,26 @@ - (void)startCaptureWithDevice:(AVCaptureDevice *)device
>                       [[UIDevice currentDevice] beginGeneratingDeviceOrientationNotifications];
> #endif
> 
>-                      _currentDevice = device;
>+                      self.currentDevice = device;
> 
>                       NSError *error = nil;
>-                      if (![_currentDevice lockForConfiguration:&error]) {
>-                        RTCLogError(
>-                            @"Failed to lock device %@. Error: %@", _currentDevice, error.userInfo);
>+                      if (![self.currentDevice lockForConfiguration:&error]) {
>+                        RTCLogError(@"Failed to lock device %@. Error: %@",
>+                                    self.currentDevice,
>+                                    error.userInfo);
>                         if (completionHandler) {
>                           completionHandler(error);
>                         }
>-                        _willBeRunning = NO;
>+                        self.willBeRunning = NO;
>                         return;
>                       }
>                       [self reconfigureCaptureSessionInput];
>                       [self updateOrientation];
>                       [self updateDeviceCaptureFormat:format fps:fps];
>                       [self updateVideoDataOutputPixelFormat:format];
>-                      [_captureSession startRunning];
>-                      [_currentDevice unlockForConfiguration];
>-                      _isRunning = YES;
>+                      [self.captureSession startRunning];
>+                      [self.currentDevice unlockForConfiguration];
>+                      self.isRunning = YES;
>                       if (completionHandler) {
>                         completionHandler(nil);
>                       }
>@@ -188,16 +193,16 @@ - (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHand
>       dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>                     block:^{
>                       RTCLogInfo("Stop");
>-                      _currentDevice = nil;
>-                      for (AVCaptureDeviceInput *oldInput in [_captureSession.inputs copy]) {
>-                        [_captureSession removeInput:oldInput];
>+                      self.currentDevice = nil;
>+                      for (AVCaptureDeviceInput *oldInput in [self.captureSession.inputs copy]) {
>+                        [self.captureSession removeInput:oldInput];
>                       }
>-                      [_captureSession stopRunning];
>+                      [self.captureSession stopRunning];
> 
> #if TARGET_OS_IPHONE
>                       [[UIDevice currentDevice] endGeneratingDeviceOrientationNotifications];
> #endif
>-                      _isRunning = NO;
>+                      self.isRunning = NO;
>                       if (completionHandler) {
>                         completionHandler();
>                       }
>@@ -340,7 +345,7 @@ - (void)handleCaptureSessionDidStartRunning:(NSNotification *)notification {
>                                block:^{
>                                  // If we successfully restarted after an unknown error,
>                                  // allow future retries on fatal errors.
>-                                 _hasRetriedOnFatalError = NO;
>+                                 self.hasRetriedOnFatalError = NO;
>                                }];
> }
> 
>@@ -352,10 +357,10 @@ - (void)handleFatalError {
>   [RTCDispatcher
>       dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>                     block:^{
>-                      if (!_hasRetriedOnFatalError) {
>+                      if (!self.hasRetriedOnFatalError) {
>                         RTCLogWarning(@"Attempting to recover from fatal capture error.");
>                         [self handleNonFatalError];
>-                        _hasRetriedOnFatalError = YES;
>+                        self.hasRetriedOnFatalError = YES;
>                       } else {
>                         RTCLogError(@"Previous fatal error recovery failed.");
>                       }
>@@ -366,8 +371,8 @@ - (void)handleNonFatalError {
>   [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>                                block:^{
>                                  RTCLog(@"Restarting capture session after error.");
>-                                 if (_isRunning) {
>-                                   [_captureSession startRunning];
>+                                 if (self.isRunning) {
>+                                   [self.captureSession startRunning];
>                                  }
>                                }];
> }
>@@ -379,9 +384,9 @@ - (void)handleNonFatalError {
> - (void)handleApplicationDidBecomeActive:(NSNotification *)notification {
>   [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>                                block:^{
>-                                 if (_isRunning && !_captureSession.isRunning) {
>+                                 if (self.isRunning && !self.captureSession.isRunning) {
>                                    RTCLog(@"Restarting capture session on active.");
>-                                   [_captureSession startRunning];
>+                                   [self.captureSession startRunning];
>                                  }
>                                }];
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/capturer/RTCFileVideoCapturer.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/capturer/RTCFileVideoCapturer.m
>index 00143e909fd0aa02fe771f80d790e9677848799a..207a21d8c0d3f6915d228af4cc21d2bd80500a15 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/capturer/RTCFileVideoCapturer.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/capturer/RTCFileVideoCapturer.m
>@@ -27,15 +27,21 @@ typedef NS_ENUM(NSInteger, RTCFileVideoCapturerStatus) {
>   RTCFileVideoCapturerStatusStopped
> };
> 
>+@interface RTCFileVideoCapturer ()
>+@property(nonatomic, assign) CMTime lastPresentationTime;
>+@property(nonatomic, strong) NSURL *fileURL;
>+@end
>+
> @implementation RTCFileVideoCapturer {
>   AVAssetReader *_reader;
>   AVAssetReaderTrackOutput *_outTrack;
>   RTCFileVideoCapturerStatus _status;
>-  CMTime _lastPresentationTime;
>   dispatch_queue_t _frameQueue;
>-  NSURL *_fileURL;
> }
> 
>+@synthesize lastPresentationTime = _lastPresentationTime;
>+@synthesize fileURL = _fileURL;
>+
> - (void)startCapturingFromFileNamed:(NSString *)nameOfFile
>                             onError:(RTCFileVideoCapturerErrorBlock)errorBlock {
>   if (_status == RTCFileVideoCapturerStatusStarted) {
>@@ -62,9 +68,9 @@ - (void)startCapturingFromFileNamed:(NSString *)nameOfFile
>       return;
>     }
> 
>-    _lastPresentationTime = CMTimeMake(0, 0);
>+    self.lastPresentationTime = CMTimeMake(0, 0);
> 
>-    _fileURL = [NSURL fileURLWithPath:pathForFile];
>+    self.fileURL = [NSURL fileURLWithPath:pathForFile];
>     [self setupReaderOnError:errorBlock];
>   });
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer.mm
>index 8f372bcdbd8152fbd25ccd4ba3ebecf11f1ade5a..fb478d25331e9ccec1494012f60209677f91f692 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer.mm
>@@ -176,7 +176,8 @@ - (BOOL)setupTexturesForFrame:(nonnull RTCVideoFrame *)frame {
>   RTCVideoRotation rotation;
>   NSValue *rotationOverride = self.rotationOverride;
>   if (rotationOverride) {
>-#if defined(__IPHONE_11_0) && (__IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_11_0)
>+#if defined(__IPHONE_11_0) && defined(__IPHONE_OS_VERSION_MAX_ALLOWED) && \
>+    (__IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_11_0)
>     if (@available(iOS 11, *)) {
>       [rotationOverride getValue:&rotation size:sizeof(rotation)];
>     } else
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/renderer/metal/RTCMTLVideoView.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/renderer/metal/RTCMTLVideoView.m
>index 7ad1d79d3e3f6d63be46dfe0861161811a9aa543..f8575c0cfed52fc4a55b8f6215fe8b3cb5865665 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/renderer/metal/RTCMTLVideoView.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/renderer/metal/RTCMTLVideoView.m
>@@ -141,6 +141,10 @@ - (void)drawInMTKView:(nonnull MTKView *)view {
>     return;
>   }
> 
>+  if (CGRectIsEmpty(view.bounds)) {
>+    return;
>+  }
>+
>   RTCMTLRenderer *renderer;
>   if ([videoFrame.buffer isKindOfClass:[RTCCVPixelBuffer class]]) {
>     RTCCVPixelBuffer *buffer = (RTCCVPixelBuffer*)videoFrame.buffer;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm
>index 6317013180daa3712ad72225e0d6f8a6059c5177..1c18fab17df964563eecd8e2e42edd0f080ee4e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm
>@@ -67,12 +67,23 @@ void decompressionOutputCallback(void *decoderRef,
> // Decoder.
> @implementation RTCVideoDecoderH264 {
>   CMVideoFormatDescriptionRef _videoFormat;
>+  CMMemoryPoolRef _memoryPool;
>   VTDecompressionSessionRef _decompressionSession;
>   RTCVideoDecoderCallback _callback;
>   OSStatus _error;
> }
> 
>+- (instancetype)init {
>+  self = [super init];
>+  if (self) {
>+    _memoryPool = CMMemoryPoolCreate(nil);
>+  }
>+  return self;
>+}
>+
> - (void)dealloc {
>+  CMMemoryPoolInvalidate(_memoryPool);
>+  CFRelease(_memoryPool);
>   [self destroyDecompressionSession];
>   [self setVideoFormat:nullptr];
> }
>@@ -125,7 +136,8 @@ - (NSInteger)decode:(RTCEncodedImage *)inputImage
>   if (!webrtc::H264AnnexBBufferToCMSampleBuffer((uint8_t *)inputImage.buffer.bytes,
>                                                 inputImage.buffer.length,
>                                                 _videoFormat,
>-                                                &sampleBuffer)) {
>+                                                &sampleBuffer,
>+                                                _memoryPool)) {
>     return WEBRTC_VIDEO_CODEC_ERROR;
>   }
>   RTC_DCHECK(sampleBuffer);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.cc
>index a849d3f053e39d21e3901ed4559a750862d46cfa..55ed1c2b4ca3700e0804ade2cbbea0e807b1e5ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.cc
>@@ -160,7 +160,8 @@ bool H264CMSampleBufferToAnnexBBuffer(
> bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
>                                       size_t annexb_buffer_size,
>                                       CMVideoFormatDescriptionRef video_format,
>-                                      CMSampleBufferRef* out_sample_buffer) {
>+                                      CMSampleBufferRef* out_sample_buffer,
>+                                      CMMemoryPoolRef memory_pool) {
>   RTC_DCHECK(annexb_buffer);
>   RTC_DCHECK(out_sample_buffer);
>   RTC_DCHECK(video_format);
>@@ -185,11 +186,11 @@ bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
>   }
> 
>   // Allocate memory as a block buffer.
>-  // TODO(tkchin): figure out how to use a pool.
>   CMBlockBufferRef block_buffer = nullptr;
>+  CFAllocatorRef block_allocator = CMMemoryPoolGetAllocator(memory_pool);
>   OSStatus status = CMBlockBufferCreateWithMemoryBlock(
>-      nullptr, nullptr, reader.BytesRemaining(), nullptr, nullptr, 0,
>-      reader.BytesRemaining(), kCMBlockBufferAssureMemoryNowFlag,
>+      kCFAllocatorDefault, nullptr, reader.BytesRemaining(), block_allocator,
>+      nullptr, 0, reader.BytesRemaining(), kCMBlockBufferAssureMemoryNowFlag,
>       &block_buffer);
>   if (status != kCMBlockBufferNoErr) {
>     RTC_LOG(LS_ERROR) << "Failed to create block buffer.";
>@@ -199,8 +200,9 @@ bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
>   // Make sure block buffer is contiguous.
>   CMBlockBufferRef contiguous_buffer = nullptr;
>   if (!CMBlockBufferIsRangeContiguous(block_buffer, 0, 0)) {
>-    status = CMBlockBufferCreateContiguous(
>-        nullptr, block_buffer, nullptr, nullptr, 0, 0, 0, &contiguous_buffer);
>+    status = CMBlockBufferCreateContiguous(kCFAllocatorDefault, block_buffer,
>+                                           block_allocator, nullptr, 0, 0, 0,
>+                                           &contiguous_buffer);
>     if (status != noErr) {
>       RTC_LOG(LS_ERROR) << "Failed to flatten non-contiguous block buffer: "
>                         << status;
>@@ -236,9 +238,9 @@ bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
>   }
> 
>   // Create sample buffer.
>-  status = CMSampleBufferCreate(nullptr, contiguous_buffer, true, nullptr,
>-                                nullptr, video_format, 1, 0, nullptr, 0,
>-                                nullptr, out_sample_buffer);
>+  status = CMSampleBufferCreate(kCFAllocatorDefault, contiguous_buffer, true,
>+                                nullptr, nullptr, video_format, 1, 0, nullptr,
>+                                0, nullptr, out_sample_buffer);
>   if (status != noErr) {
>     RTC_LOG(LS_ERROR) << "Failed to create sample buffer.";
>     CFRelease(contiguous_buffer);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.h
>index cd5a51079ab5b225877d00ce235c67d0b1eb0e98..a0c1aa90af5759de39b5d06c221d284294389876 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.h
>@@ -44,7 +44,8 @@ bool H264CMSampleBufferToAnnexBBuffer(
> bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
>                                       size_t annexb_buffer_size,
>                                       CMVideoFormatDescriptionRef video_format,
>-                                      CMSampleBufferRef* out_sample_buffer);
>+                                      CMSampleBufferRef* out_sample_buffer,
>+                                      CMMemoryPoolRef memory_pool);
> 
> // Returns a video format description created from the sps/pps information in
> // the Annex B buffer. If there is no such information, nullptr is returned.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h
>index abe9dfca93b04bf552ec0302b807597df261edb1..432a38257464b26cb53234c8a37af67d58ef80b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h
>@@ -17,7 +17,6 @@ NS_ASSUME_NONNULL_BEGIN
> 
> /** RTCVideoFrameBuffer containing a CVPixelBufferRef */
> RTC_OBJC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCCVPixelBuffer")))
> @interface RTCCVPixelBuffer : NSObject <RTCVideoFrameBuffer>
> 
> @property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/RTCCameraPreviewView.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/RTCCameraPreviewView.m
>index 2add46cf71f6a47db05f983c469e3f401b0e36ff..adc62cc30af497bdc16644a3181e59a45c8dd34f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/RTCCameraPreviewView.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/RTCCameraPreviewView.m
>@@ -47,19 +47,22 @@ - (void)setCaptureSession:(AVCaptureSession *)captureSession {
>   if (_captureSession == captureSession) {
>     return;
>   }
>-  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeMain
>-                               block:^{
>-    _captureSession = captureSession;
>-    AVCaptureVideoPreviewLayer *previewLayer = [self previewLayer];
>-    [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>-                                 block:^{
>-      previewLayer.session = captureSession;
>-      [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeMain
>-                               block:^{
>-        [self setCorrectVideoOrientation];
>-      }];
>-    }];
>-  }];
>+  [RTCDispatcher
>+      dispatchAsyncOnType:RTCDispatcherTypeMain
>+                    block:^{
>+                      self.captureSession = captureSession;
>+                      AVCaptureVideoPreviewLayer *previewLayer = [self previewLayer];
>+                      [RTCDispatcher
>+                          dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>+                                        block:^{
>+                                          previewLayer.session = captureSession;
>+                                          [RTCDispatcher
>+                                              dispatchAsyncOnType:RTCDispatcherTypeMain
>+                                                            block:^{
>+                                                              [self setCorrectVideoOrientation];
>+                                                            }];
>+                                        }];
>+                    }];
> }
> 
> - (void)layoutSubviews {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/UIDevice+RTCDevice.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/UIDevice+RTCDevice.h
>index eb39ea9cc4d767341a3f7aa396b98e706fb13e04..736237582fe828bd736e5c8c0bddeae4e8beba29 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/UIDevice+RTCDevice.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/UIDevice+RTCDevice.h
>@@ -34,6 +34,9 @@ typedef NS_ENUM(NSInteger, RTCDeviceType) {
>   RTCDeviceTypeIPhone8,
>   RTCDeviceTypeIPhone8Plus,
>   RTCDeviceTypeIPhoneX,
>+  RTCDeviceTypeIPhoneXS,
>+  RTCDeviceTypeIPhoneXSMax,
>+  RTCDeviceTypeIPhoneXR,
>   RTCDeviceTypeIPodTouch1G,
>   RTCDeviceTypeIPodTouch2G,
>   RTCDeviceTypeIPodTouch3G,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/UIDevice+RTCDevice.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/UIDevice+RTCDevice.mm
>index 96adb9af780e13f09882ead7d848aee7c616c090..6f68e9f7a0da46b04e5a979509874d6d2a3aa2f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/UIDevice+RTCDevice.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/helpers/UIDevice+RTCDevice.mm
>@@ -17,81 +17,85 @@ @implementation UIDevice (RTCDevice)
> 
> + (RTCDeviceType)deviceType {
>   NSDictionary *machineNameToType = @{
>-    @"iPhone1,1": @(RTCDeviceTypeIPhone1G),
>-    @"iPhone1,2": @(RTCDeviceTypeIPhone3G),
>-    @"iPhone2,1": @(RTCDeviceTypeIPhone3GS),
>-    @"iPhone3,1": @(RTCDeviceTypeIPhone4),
>-    @"iPhone3,2": @(RTCDeviceTypeIPhone4),
>-    @"iPhone3,3": @(RTCDeviceTypeIPhone4Verizon),
>-    @"iPhone4,1": @(RTCDeviceTypeIPhone4S),
>-    @"iPhone5,1": @(RTCDeviceTypeIPhone5GSM),
>-    @"iPhone5,2": @(RTCDeviceTypeIPhone5GSM_CDMA),
>-    @"iPhone5,3": @(RTCDeviceTypeIPhone5CGSM),
>-    @"iPhone5,4": @(RTCDeviceTypeIPhone5CGSM_CDMA),
>-    @"iPhone6,1": @(RTCDeviceTypeIPhone5SGSM),
>-    @"iPhone6,2": @(RTCDeviceTypeIPhone5SGSM_CDMA),
>-    @"iPhone7,1": @(RTCDeviceTypeIPhone6Plus),
>-    @"iPhone7,2": @(RTCDeviceTypeIPhone6),
>-    @"iPhone8,1": @(RTCDeviceTypeIPhone6S),
>-    @"iPhone8,2": @(RTCDeviceTypeIPhone6SPlus),
>-    @"iPhone8,4": @(RTCDeviceTypeIPhoneSE),
>-    @"iPhone9,1": @(RTCDeviceTypeIPhone7),
>-    @"iPhone9,2": @(RTCDeviceTypeIPhone7Plus),
>-    @"iPhone9,3": @(RTCDeviceTypeIPhone7),
>-    @"iPhone9,4": @(RTCDeviceTypeIPhone7Plus),
>-    @"iPhone10,1": @(RTCDeviceTypeIPhone8),
>-    @"iPhone10,2": @(RTCDeviceTypeIPhone8Plus),
>-    @"iPhone10,3": @(RTCDeviceTypeIPhoneX),
>-    @"iPhone10,4": @(RTCDeviceTypeIPhone8),
>-    @"iPhone10,5": @(RTCDeviceTypeIPhone8Plus),
>-    @"iPhone10,6": @(RTCDeviceTypeIPhoneX),
>-    @"iPod1,1": @(RTCDeviceTypeIPodTouch1G),
>-    @"iPod2,1": @(RTCDeviceTypeIPodTouch2G),
>-    @"iPod3,1": @(RTCDeviceTypeIPodTouch3G),
>-    @"iPod4,1": @(RTCDeviceTypeIPodTouch4G),
>-    @"iPod5,1": @(RTCDeviceTypeIPodTouch5G),
>-    @"iPod7,1": @(RTCDeviceTypeIPodTouch6G),
>-    @"iPad1,1": @(RTCDeviceTypeIPad),
>-    @"iPad2,1": @(RTCDeviceTypeIPad2Wifi),
>-    @"iPad2,2": @(RTCDeviceTypeIPad2GSM),
>-    @"iPad2,3": @(RTCDeviceTypeIPad2CDMA),
>-    @"iPad2,4": @(RTCDeviceTypeIPad2Wifi2),
>-    @"iPad2,5": @(RTCDeviceTypeIPadMiniWifi),
>-    @"iPad2,6": @(RTCDeviceTypeIPadMiniGSM),
>-    @"iPad2,7": @(RTCDeviceTypeIPadMiniGSM_CDMA),
>-    @"iPad3,1": @(RTCDeviceTypeIPad3Wifi),
>-    @"iPad3,2": @(RTCDeviceTypeIPad3GSM_CDMA),
>-    @"iPad3,3": @(RTCDeviceTypeIPad3GSM),
>-    @"iPad3,4": @(RTCDeviceTypeIPad4Wifi),
>-    @"iPad3,5": @(RTCDeviceTypeIPad4GSM),
>-    @"iPad3,6": @(RTCDeviceTypeIPad4GSM_CDMA),
>-    @"iPad4,1": @(RTCDeviceTypeIPadAirWifi),
>-    @"iPad4,2": @(RTCDeviceTypeIPadAirCellular),
>-    @"iPad4,3": @(RTCDeviceTypeIPadAirWifiCellular),
>-    @"iPad4,4": @(RTCDeviceTypeIPadMini2GWifi),
>-    @"iPad4,5": @(RTCDeviceTypeIPadMini2GCellular),
>-    @"iPad4,6": @(RTCDeviceTypeIPadMini2GWifiCellular),
>-    @"iPad4,7": @(RTCDeviceTypeIPadMini3),
>-    @"iPad4,8": @(RTCDeviceTypeIPadMini3),
>-    @"iPad4,9": @(RTCDeviceTypeIPadMini3),
>-    @"iPad5,1": @(RTCDeviceTypeIPadMini4),
>-    @"iPad5,2": @(RTCDeviceTypeIPadMini4),
>-    @"iPad5,3": @(RTCDeviceTypeIPadAir2),
>-    @"iPad5,4": @(RTCDeviceTypeIPadAir2),
>-    @"iPad6,3": @(RTCDeviceTypeIPadPro9Inch),
>-    @"iPad6,4": @(RTCDeviceTypeIPadPro9Inch),
>-    @"iPad6,7": @(RTCDeviceTypeIPadPro12Inch),
>-    @"iPad6,8": @(RTCDeviceTypeIPadPro12Inch),
>-    @"iPad6,11": @(RTCDeviceTypeIPad5),
>-    @"iPad6,12": @(RTCDeviceTypeIPad5),
>-    @"iPad7,1": @(RTCDeviceTypeIPadPro12Inch2),
>-    @"iPad7,2": @(RTCDeviceTypeIPadPro12Inch2),
>-    @"iPad7,3": @(RTCDeviceTypeIPadPro10Inch),
>-    @"iPad7,4": @(RTCDeviceTypeIPadPro10Inch),
>-    @"iPad7,5": @(RTCDeviceTypeIPad6),
>-    @"iPad7,6": @(RTCDeviceTypeIPad6),
>-    @"i386": @(RTCDeviceTypeSimulatori386),
>-    @"x86_64": @(RTCDeviceTypeSimulatorx86_64),
>+    @"iPhone1,1" : @(RTCDeviceTypeIPhone1G),
>+    @"iPhone1,2" : @(RTCDeviceTypeIPhone3G),
>+    @"iPhone2,1" : @(RTCDeviceTypeIPhone3GS),
>+    @"iPhone3,1" : @(RTCDeviceTypeIPhone4),
>+    @"iPhone3,2" : @(RTCDeviceTypeIPhone4),
>+    @"iPhone3,3" : @(RTCDeviceTypeIPhone4Verizon),
>+    @"iPhone4,1" : @(RTCDeviceTypeIPhone4S),
>+    @"iPhone5,1" : @(RTCDeviceTypeIPhone5GSM),
>+    @"iPhone5,2" : @(RTCDeviceTypeIPhone5GSM_CDMA),
>+    @"iPhone5,3" : @(RTCDeviceTypeIPhone5CGSM),
>+    @"iPhone5,4" : @(RTCDeviceTypeIPhone5CGSM_CDMA),
>+    @"iPhone6,1" : @(RTCDeviceTypeIPhone5SGSM),
>+    @"iPhone6,2" : @(RTCDeviceTypeIPhone5SGSM_CDMA),
>+    @"iPhone7,1" : @(RTCDeviceTypeIPhone6Plus),
>+    @"iPhone7,2" : @(RTCDeviceTypeIPhone6),
>+    @"iPhone8,1" : @(RTCDeviceTypeIPhone6S),
>+    @"iPhone8,2" : @(RTCDeviceTypeIPhone6SPlus),
>+    @"iPhone8,4" : @(RTCDeviceTypeIPhoneSE),
>+    @"iPhone9,1" : @(RTCDeviceTypeIPhone7),
>+    @"iPhone9,2" : @(RTCDeviceTypeIPhone7Plus),
>+    @"iPhone9,3" : @(RTCDeviceTypeIPhone7),
>+    @"iPhone9,4" : @(RTCDeviceTypeIPhone7Plus),
>+    @"iPhone10,1" : @(RTCDeviceTypeIPhone8),
>+    @"iPhone10,2" : @(RTCDeviceTypeIPhone8Plus),
>+    @"iPhone10,3" : @(RTCDeviceTypeIPhoneX),
>+    @"iPhone10,4" : @(RTCDeviceTypeIPhone8),
>+    @"iPhone10,5" : @(RTCDeviceTypeIPhone8Plus),
>+    @"iPhone10,6" : @(RTCDeviceTypeIPhoneX),
>+    @"iPhone11,2" : @(RTCDeviceTypeIPhoneXS),
>+    @"iPhone11,4" : @(RTCDeviceTypeIPhoneXSMax),
>+    @"iPhone11,6" : @(RTCDeviceTypeIPhoneXSMax),
>+    @"iPhone11,8" : @(RTCDeviceTypeIPhoneXR),
>+    @"iPod1,1" : @(RTCDeviceTypeIPodTouch1G),
>+    @"iPod2,1" : @(RTCDeviceTypeIPodTouch2G),
>+    @"iPod3,1" : @(RTCDeviceTypeIPodTouch3G),
>+    @"iPod4,1" : @(RTCDeviceTypeIPodTouch4G),
>+    @"iPod5,1" : @(RTCDeviceTypeIPodTouch5G),
>+    @"iPod7,1" : @(RTCDeviceTypeIPodTouch6G),
>+    @"iPad1,1" : @(RTCDeviceTypeIPad),
>+    @"iPad2,1" : @(RTCDeviceTypeIPad2Wifi),
>+    @"iPad2,2" : @(RTCDeviceTypeIPad2GSM),
>+    @"iPad2,3" : @(RTCDeviceTypeIPad2CDMA),
>+    @"iPad2,4" : @(RTCDeviceTypeIPad2Wifi2),
>+    @"iPad2,5" : @(RTCDeviceTypeIPadMiniWifi),
>+    @"iPad2,6" : @(RTCDeviceTypeIPadMiniGSM),
>+    @"iPad2,7" : @(RTCDeviceTypeIPadMiniGSM_CDMA),
>+    @"iPad3,1" : @(RTCDeviceTypeIPad3Wifi),
>+    @"iPad3,2" : @(RTCDeviceTypeIPad3GSM_CDMA),
>+    @"iPad3,3" : @(RTCDeviceTypeIPad3GSM),
>+    @"iPad3,4" : @(RTCDeviceTypeIPad4Wifi),
>+    @"iPad3,5" : @(RTCDeviceTypeIPad4GSM),
>+    @"iPad3,6" : @(RTCDeviceTypeIPad4GSM_CDMA),
>+    @"iPad4,1" : @(RTCDeviceTypeIPadAirWifi),
>+    @"iPad4,2" : @(RTCDeviceTypeIPadAirCellular),
>+    @"iPad4,3" : @(RTCDeviceTypeIPadAirWifiCellular),
>+    @"iPad4,4" : @(RTCDeviceTypeIPadMini2GWifi),
>+    @"iPad4,5" : @(RTCDeviceTypeIPadMini2GCellular),
>+    @"iPad4,6" : @(RTCDeviceTypeIPadMini2GWifiCellular),
>+    @"iPad4,7" : @(RTCDeviceTypeIPadMini3),
>+    @"iPad4,8" : @(RTCDeviceTypeIPadMini3),
>+    @"iPad4,9" : @(RTCDeviceTypeIPadMini3),
>+    @"iPad5,1" : @(RTCDeviceTypeIPadMini4),
>+    @"iPad5,2" : @(RTCDeviceTypeIPadMini4),
>+    @"iPad5,3" : @(RTCDeviceTypeIPadAir2),
>+    @"iPad5,4" : @(RTCDeviceTypeIPadAir2),
>+    @"iPad6,3" : @(RTCDeviceTypeIPadPro9Inch),
>+    @"iPad6,4" : @(RTCDeviceTypeIPadPro9Inch),
>+    @"iPad6,7" : @(RTCDeviceTypeIPadPro12Inch),
>+    @"iPad6,8" : @(RTCDeviceTypeIPadPro12Inch),
>+    @"iPad6,11" : @(RTCDeviceTypeIPad5),
>+    @"iPad6,12" : @(RTCDeviceTypeIPad5),
>+    @"iPad7,1" : @(RTCDeviceTypeIPadPro12Inch2),
>+    @"iPad7,2" : @(RTCDeviceTypeIPadPro12Inch2),
>+    @"iPad7,3" : @(RTCDeviceTypeIPadPro10Inch),
>+    @"iPad7,4" : @(RTCDeviceTypeIPadPro10Inch),
>+    @"iPad7,5" : @(RTCDeviceTypeIPad6),
>+    @"iPad7,6" : @(RTCDeviceTypeIPad6),
>+    @"i386" : @(RTCDeviceTypeSimulatori386),
>+    @"x86_64" : @(RTCDeviceTypeSimulatorx86_64),
>   };
> 
>   RTCDeviceType deviceType = RTCDeviceTypeUnknown;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/api/video_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/api/video_capturer.h
>index 5b6f3f939baa7c662335d791e6880693389a1d51..d6f396b4701cac5708b0f676da46636654f36a45 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/api/video_capturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/api/video_capturer.h
>@@ -15,6 +15,7 @@
> 
> #include "api/mediastreaminterface.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/thread.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm
>index c4c85cc6ab903535e5ef01e7cf5a81531d170e6f..ae45033a570f1c1989db31c54e1fbef19efc4a15 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm
>@@ -86,28 +86,26 @@ class ObjCVideoEncoder : public VideoEncoder {
>                  frameTypes:rtcFrameTypes];
>   }
> 
>-  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) { return WEBRTC_VIDEO_CODEC_OK; }
>-
>   int32_t SetRates(uint32_t bitrate, uint32_t framerate) {
>     return [encoder_ setBitrate:bitrate framerate:framerate];
>   }
> 
>   int32_t SetRateAllocation(const VideoBitrateAllocation& allocation, uint32_t framerate) {
>-    RTCVideoBitrateAllocation *bitrateAllocation =
>-      [[RTCVideoBitrateAllocation alloc] initWithNativeVideoBitrateAllocation:&allocation];
>-    return [encoder_ setRateAllocation: bitrateAllocation framerate:framerate];
>+    auto *rtcAllocation = [[RTCVideoBitrateAllocation alloc] initWithNativeVideoBitrateAllocation:&allocation];
>+    return [encoder_ setRateAllocation: rtcAllocation framerate:framerate];
>   }
> 
>-  bool SupportsNativeHandle() const { return true; }
>+  VideoEncoder::EncoderInfo GetEncoderInfo() const {
>+    EncoderInfo info;
>+    info.supports_native_handle = true;
>+    info.implementation_name = implementation_name_;
> 
>-  VideoEncoder::ScalingSettings GetScalingSettings() const {
>     RTCVideoEncoderQpThresholds *qp_thresholds = [encoder_ scalingSettings];
>-    return qp_thresholds ? ScalingSettings(qp_thresholds.low, qp_thresholds.high) :
>-                           ScalingSettings::kOff;
>+    info.scaling_settings = qp_thresholds ? ScalingSettings(qp_thresholds.low, qp_thresholds.high) :
>+                                            ScalingSettings::kOff;
>+    return info;
>   }
> 
>-  const char *ImplementationName() const { return implementation_name_.c_str(); }
>-
>  private:
>   id<RTCVideoEncoder> encoder_;
>   const std::string implementation_name_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm
>index 689de581d04eaf58ae473ccdfe52183b0256437e..55d48c28ba71e93f03f9deb7873f54113c0c1ea4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm
>@@ -128,17 +128,20 @@ static const NSUInteger kFullDuplexTimeInSec = 10;
> static const NSUInteger kNumIgnoreFirstCallbacks = 50;
> 
> @interface RTCAudioDeviceModuleTests : XCTestCase {
>-
>   rtc::scoped_refptr<webrtc::AudioDeviceModule> audioDeviceModule;
>-  webrtc::AudioParameters playoutParameters;
>-  webrtc::AudioParameters recordParameters;
>   MockAudioTransport mock;
> }
> 
>+@property(nonatomic, assign) webrtc::AudioParameters playoutParameters;
>+@property(nonatomic, assign) webrtc::AudioParameters recordParameters;
>+
> @end
> 
> @implementation RTCAudioDeviceModuleTests
> 
>+@synthesize playoutParameters;
>+@synthesize recordParameters;
>+
> - (void)setUp {
>   [super setUp];
>   audioDeviceModule = webrtc::CreateAudioDeviceModule();
>@@ -254,10 +257,10 @@ - (void)testStartPlayoutOnTwoInstances {
>                                        int64_t *elapsed_time_ms,
>                                        int64_t *ntp_time_ms) {
>     nSamplesOut = nSamples;
>-    XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nSamples, self.playoutParameters.frames_per_10ms_buffer());
>     XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>-    XCTAssertEqual(nChannels, playoutParameters.channels());
>-    XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
>+    XCTAssertEqual(nChannels, self.playoutParameters.channels());
>+    XCTAssertEqual((int)samplesPerSec, self.playoutParameters.sample_rate());
>     XCTAssertNotEqual((void*)NULL, audioSamples);
> 
>     return 0;
>@@ -291,10 +294,10 @@ - (void)testStartPlayoutOnTwoInstances {
>                                         int64_t *elapsed_time_ms,
>                                         int64_t *ntp_time_ms) {
>     nSamplesOut = nSamples;
>-    XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nSamples, self.playoutParameters.frames_per_10ms_buffer());
>     XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>-    XCTAssertEqual(nChannels, playoutParameters.channels());
>-    XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
>+    XCTAssertEqual(nChannels, self.playoutParameters.channels());
>+    XCTAssertEqual((int)samplesPerSec, self.playoutParameters.sample_rate());
>     XCTAssertNotEqual((void*)NULL, audioSamples);
>     if (++num_callbacks == kNumCallbacks) {
>       [playoutExpectation fulfill];
>@@ -330,10 +333,10 @@ - (void)testStartPlayoutVerifyCallbacks {
>                                        int64_t *elapsed_time_ms,
>                                        int64_t *ntp_time_ms) {
>     nSamplesOut = nSamples;
>-    XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nSamples, self.playoutParameters.frames_per_10ms_buffer());
>     XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>-    XCTAssertEqual(nChannels, playoutParameters.channels());
>-    XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
>+    XCTAssertEqual(nChannels, self.playoutParameters.channels());
>+    XCTAssertEqual((int)samplesPerSec, self.playoutParameters.sample_rate());
>     XCTAssertNotEqual((void*)NULL, audioSamples);
>     if (++num_callbacks == kNumCallbacks) {
>       [playoutExpectation fulfill];
>@@ -366,10 +369,10 @@ - (void)testStartRecordingVerifyCallbacks {
>                                        const bool keyPressed,
>                                        uint32_t& newMicLevel) {
>     XCTAssertNotEqual((void*)NULL, audioSamples);
>-    XCTAssertEqual(nSamples, recordParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nSamples, self.recordParameters.frames_per_10ms_buffer());
>     XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>-    XCTAssertEqual(nChannels, recordParameters.channels());
>-    XCTAssertEqual((int) samplesPerSec, recordParameters.sample_rate());
>+    XCTAssertEqual(nChannels, self.recordParameters.channels());
>+    XCTAssertEqual((int)samplesPerSec, self.recordParameters.sample_rate());
>     XCTAssertEqual(0, clockDrift);
>     XCTAssertEqual(0u, currentMicLevel);
>     XCTAssertFalse(keyPressed);
>@@ -405,10 +408,10 @@ - (void)testStartPlayoutAndRecordingVerifyCallbacks {
>                                        int64_t *elapsed_time_ms,
>                                        int64_t *ntp_time_ms) {
>     nSamplesOut = nSamples;
>-    XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nSamples, self.playoutParameters.frames_per_10ms_buffer());
>     XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>-    XCTAssertEqual(nChannels, playoutParameters.channels());
>-    XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
>+    XCTAssertEqual(nChannels, self.playoutParameters.channels());
>+    XCTAssertEqual((int)samplesPerSec, self.playoutParameters.sample_rate());
>     XCTAssertNotEqual((void*)NULL, audioSamples);
>     if (callbackCount++ >= kNumCallbacks) {
>       [playoutExpectation fulfill];
>@@ -428,10 +431,10 @@ - (void)testStartPlayoutAndRecordingVerifyCallbacks {
>                                        const bool keyPressed,
>                                        uint32_t& newMicLevel) {
>     XCTAssertNotEqual((void*)NULL, audioSamples);
>-    XCTAssertEqual(nSamples, recordParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nSamples, self.recordParameters.frames_per_10ms_buffer());
>     XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>-    XCTAssertEqual(nChannels, recordParameters.channels());
>-    XCTAssertEqual((int) samplesPerSec, recordParameters.sample_rate());
>+    XCTAssertEqual(nChannels, self.recordParameters.channels());
>+    XCTAssertEqual((int)samplesPerSec, self.recordParameters.sample_rate());
>     XCTAssertEqual(0, clockDrift);
>     XCTAssertEqual(0u, currentMicLevel);
>     XCTAssertFalse(keyPressed);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCCallbackLogger_xctest.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCCallbackLogger_xctest.m
>index 4a0a2b3e6d15bd71d96a592222acd5a3474c2a9e..ceaa762f1f8a1f097d51c55e09252d92ed99b71f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCCallbackLogger_xctest.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCCallbackLogger_xctest.m
>@@ -49,6 +49,23 @@ - (void)testCallbackGetsCalledForAppropriateLevel {
>   [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
> }
> 
>+- (void)testCallbackWithSeverityGetsCalledForAppropriateLevel {
>+  self.logger.severity = RTCLoggingSeverityWarning;
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"callbackWarning"];
>+
>+  [self.logger
>+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity severity) {
>+        XCTAssertTrue([message hasSuffix:@"Horrible error\n"]);
>+        XCTAssertEqual(severity, RTCLoggingSeverityError);
>+        [callbackExpectation fulfill];
>+      }];
>+
>+  RTCLogError("Horrible error");
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+}
>+
> - (void)testCallbackDoesNotGetCalledForOtherLevels {
>   self.logger.severity = RTCLoggingSeverityError;
> 
>@@ -66,12 +83,76 @@ - (void)testCallbackDoesNotGetCalledForOtherLevels {
>   [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
> }
> 
>+- (void)testCallbackWithSeverityDoesNotGetCalledForOtherLevels {
>+  self.logger.severity = RTCLoggingSeverityError;
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"callbackError"];
>+
>+  [self.logger
>+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity severity) {
>+        XCTAssertTrue([message hasSuffix:@"Horrible error\n"]);
>+        XCTAssertEqual(severity, RTCLoggingSeverityError);
>+        [callbackExpectation fulfill];
>+      }];
>+
>+  RTCLogInfo("Just some info");
>+  RTCLogWarning("Warning warning");
>+  RTCLogError("Horrible error");
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+}
>+
>+- (void)testCallbackDoesNotgetCalledForSeverityNone {
>+  self.logger.severity = RTCLoggingSeverityNone;
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"unexpectedCallback"];
>+
>+  [self.logger start:^(NSString *message) {
>+    [callbackExpectation fulfill];
>+    XCTAssertTrue(false);
>+  }];
>+
>+  RTCLogInfo("Just some info");
>+  RTCLogWarning("Warning warning");
>+  RTCLogError("Horrible error");
>+
>+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
>+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
>+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
>+}
>+
>+- (void)testCallbackWithSeverityDoesNotgetCalledForSeverityNone {
>+  self.logger.severity = RTCLoggingSeverityNone;
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"unexpectedCallback"];
>+
>+  [self.logger
>+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity severity) {
>+        [callbackExpectation fulfill];
>+        XCTAssertTrue(false);
>+      }];
>+
>+  RTCLogInfo("Just some info");
>+  RTCLogWarning("Warning warning");
>+  RTCLogError("Horrible error");
>+
>+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
>+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
>+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
>+}
>+
> - (void)testStartingWithNilCallbackDoesNotCrash {
>   [self.logger start:nil];
> 
>   RTCLogError("Horrible error");
> }
> 
>+- (void)testStartingWithNilCallbackWithSeverityDoesNotCrash {
>+  [self.logger startWithMessageAndSeverityHandler:nil];
>+
>+  RTCLogError("Horrible error");
>+}
>+
> - (void)testStopCallbackLogger {
>   XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"stopped"];
> 
>@@ -88,6 +169,23 @@ - (void)testStopCallbackLogger {
>   XCTAssertEqual(result, XCTWaiterResultTimedOut);
> }
> 
>+- (void)testStopCallbackWithSeverityLogger {
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"stopped"];
>+
>+  [self.logger
>+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity loggingServerity) {
>+        [callbackExpectation fulfill];
>+      }];
>+
>+  [self.logger stop];
>+
>+  RTCLogInfo("Just some info");
>+
>+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
>+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
>+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
>+}
>+
> - (void)testDestroyingCallbackLogger {
>   XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"destroyed"];
> 
>@@ -104,4 +202,43 @@ - (void)testDestroyingCallbackLogger {
>   XCTAssertEqual(result, XCTWaiterResultTimedOut);
> }
> 
>+- (void)testDestroyingCallbackWithSeverityLogger {
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"destroyed"];
>+
>+  [self.logger
>+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity loggingServerity) {
>+        [callbackExpectation fulfill];
>+      }];
>+
>+  self.logger = nil;
>+
>+  RTCLogInfo("Just some info");
>+
>+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
>+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
>+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
>+}
>+
>+- (void)testCallbackWithSeverityLoggerCannotStartTwice {
>+  self.logger.severity = RTCLoggingSeverityWarning;
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"callbackWarning"];
>+
>+  [self.logger
>+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity loggingServerity) {
>+        XCTAssertTrue([message hasSuffix:@"Horrible error\n"]);
>+        XCTAssertEqual(loggingServerity, RTCLoggingSeverityError);
>+        [callbackExpectation fulfill];
>+      }];
>+
>+  [self.logger start:^(NSString *message) {
>+    [callbackExpectation fulfill];
>+    XCTAssertTrue(false);
>+  }];
>+
>+  RTCLogError("Horrible error");
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+}
>+
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCCertificateTest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCCertificateTest.mm
>index 2585a278118e99bf375b401b13b84f0625d4e05f..5bf1eb3fe4707477ffb84223b554f46ae84d56c8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCCertificateTest.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCCertificateTest.mm
>@@ -71,7 +71,7 @@ - (void)testCertificateIsUsedInConfig {
> 
> @end
> 
>-TEST(CertificateTest, CertificateIsUsedInConfig) {
>+TEST(CertificateTest, DISABLED_CertificateIsUsedInConfig) {
>   RTCCertificateTest *test = [[RTCCertificateTest alloc] init];
>   [test testCertificateIsUsedInConfig];
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCConfigurationTest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCConfigurationTest.mm
>index 1ef718ad4958de7d776ff59bb7433d05703a99fc..f31fcfd858d929f6064f87c48ef107a46e8bf8ec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCConfigurationTest.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCConfigurationTest.mm
>@@ -50,6 +50,12 @@ - (void)testConversionToNativeConfiguration {
>       RTCContinualGatheringPolicyGatherContinually;
>   config.shouldPruneTurnPorts = YES;
>   config.iceRegatherIntervalRange = range;
>+  config.cryptoOptions = [[RTCCryptoOptions alloc] initWithSrtpEnableGcmCryptoSuites:YES
>+                                                 srtpEnableAes128Sha1_32CryptoCipher:YES
>+                                              srtpEnableEncryptedRtpHeaderExtensions:YES
>+                                                        sframeRequireFrameEncryption:YES];
>+  config.rtcpAudioReportIntervalMs = 2500;
>+  config.rtcpVideoReportIntervalMs = 3750;
> 
>   std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
>       nativeConfig([config createNativeConfiguration]);
>@@ -78,6 +84,12 @@ - (void)testConversionToNativeConfiguration {
>   EXPECT_EQ(true, nativeConfig->prune_turn_ports);
>   EXPECT_EQ(range.min, nativeConfig->ice_regather_interval_range->min());
>   EXPECT_EQ(range.max, nativeConfig->ice_regather_interval_range->max());
>+  EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_gcm_crypto_suites);
>+  EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_aes128_sha1_32_crypto_cipher);
>+  EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_encrypted_rtp_header_extensions);
>+  EXPECT_EQ(true, nativeConfig->crypto_options->sframe.require_frame_encryption);
>+  EXPECT_EQ(2500, nativeConfig->audio_rtcp_report_interval_ms());
>+  EXPECT_EQ(3750, nativeConfig->video_rtcp_report_interval_ms());
> }
> 
> - (void)testNativeConversionToConfiguration {
>@@ -103,6 +115,12 @@ - (void)testNativeConversionToConfiguration {
>       RTCContinualGatheringPolicyGatherContinually;
>   config.shouldPruneTurnPorts = YES;
>   config.iceRegatherIntervalRange = range;
>+  config.cryptoOptions = [[RTCCryptoOptions alloc] initWithSrtpEnableGcmCryptoSuites:YES
>+                                                 srtpEnableAes128Sha1_32CryptoCipher:NO
>+                                              srtpEnableEncryptedRtpHeaderExtensions:NO
>+                                                        sframeRequireFrameEncryption:NO];
>+  config.rtcpAudioReportIntervalMs = 1500;
>+  config.rtcpVideoReportIntervalMs = 2150;
> 
>   webrtc::PeerConnectionInterface::RTCConfiguration *nativeConfig =
>       [config createNativeConfiguration];
>@@ -130,6 +148,21 @@ - (void)testNativeConversionToConfiguration {
>   EXPECT_EQ(config.shouldPruneTurnPorts, newConfig.shouldPruneTurnPorts);
>   EXPECT_EQ(config.iceRegatherIntervalRange.min, newConfig.iceRegatherIntervalRange.min);
>   EXPECT_EQ(config.iceRegatherIntervalRange.max, newConfig.iceRegatherIntervalRange.max);
>+  EXPECT_EQ(config.cryptoOptions.srtpEnableGcmCryptoSuites,
>+            newConfig.cryptoOptions.srtpEnableGcmCryptoSuites);
>+  EXPECT_EQ(config.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher,
>+            newConfig.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher);
>+  EXPECT_EQ(config.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions,
>+            newConfig.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions);
>+  EXPECT_EQ(config.cryptoOptions.sframeRequireFrameEncryption,
>+            newConfig.cryptoOptions.sframeRequireFrameEncryption);
>+  EXPECT_EQ(config.rtcpAudioReportIntervalMs, newConfig.rtcpAudioReportIntervalMs);
>+  EXPECT_EQ(config.rtcpVideoReportIntervalMs, newConfig.rtcpVideoReportIntervalMs);
>+}
>+
>+- (void)testDefaultValues {
>+  RTCConfiguration *config = [[RTCConfiguration alloc] init];
>+  EXPECT_EQ(config.cryptoOptions, nil);
> }
> 
> @end
>@@ -139,5 +172,6 @@ TEST(RTCConfigurationTest, NativeConfigurationConversionTest) {
>     RTCConfigurationTest *test = [[RTCConfigurationTest alloc] init];
>     [test testConversionToNativeConfiguration];
>     [test testNativeConversionToConfiguration];
>+    [test testDefaultValues];
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
>index 9bb5179521f34a161cce3e15f7623ffd57ed5804..5f889a699da610851190043e3acfdfd767bd9a06 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
>@@ -22,6 +22,7 @@ extern "C" {
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/media_transport_interface.h"
> #include "api/video_codecs/video_decoder_factory.h"
> #include "api/video_codecs/video_encoder_factory.h"
> #include "modules/audio_device/include/audio_device.h"
>@@ -49,7 +50,8 @@ - (void)testBuilder {
>               nativeVideoEncoderFactory:nullptr
>               nativeVideoDecoderFactory:nullptr
>                       audioDeviceModule:nullptr
>-                  audioProcessingModule:nullptr]);
>+                  audioProcessingModule:nullptr
>+                  mediaTransportFactory:nullptr]);
> #endif
>   RTCPeerConnectionFactoryBuilder* builder = [[RTCPeerConnectionFactoryBuilder alloc] init];
>   RTCPeerConnectionFactory* peerConnectionFactory = [builder createPeerConnectionFactory];
>@@ -69,7 +71,8 @@ - (void)testDefaultComponentsBuilder {
>               nativeVideoEncoderFactory:nullptr
>               nativeVideoDecoderFactory:nullptr
>                       audioDeviceModule:nullptr
>-                  audioProcessingModule:nullptr]);
>+                  audioProcessingModule:nullptr
>+                  mediaTransportFactory:nullptr]);
> #endif
>   RTCPeerConnectionFactoryBuilder* builder = [RTCPeerConnectionFactoryBuilder defaultBuilder];
>   RTCPeerConnectionFactory* peerConnectionFactory = [builder createPeerConnectionFactory];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCPeerConnectionTest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCPeerConnectionTest.mm
>index 28fdee6a2bd6ed81e0700b4ec110a24b2ca0a3cd..35322587996d98ac6769b0655a648b19a3ad9d3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCPeerConnectionTest.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/RTCPeerConnectionTest.mm
>@@ -16,6 +16,7 @@
> 
> #import "api/peerconnection/RTCConfiguration+Private.h"
> #import "api/peerconnection/RTCConfiguration.h"
>+#import "api/peerconnection/RTCCryptoOptions.h"
> #import "api/peerconnection/RTCIceServer.h"
> #import "api/peerconnection/RTCMediaConstraints.h"
> #import "api/peerconnection/RTCPeerConnection.h"
>@@ -50,6 +51,10 @@ - (void)testConfigurationGetter {
>       RTCContinualGatheringPolicyGatherContinually;
>   config.shouldPruneTurnPorts = YES;
>   config.activeResetSrtpParams = YES;
>+  config.cryptoOptions = [[RTCCryptoOptions alloc] initWithSrtpEnableGcmCryptoSuites:YES
>+                                                 srtpEnableAes128Sha1_32CryptoCipher:YES
>+                                              srtpEnableEncryptedRtpHeaderExtensions:NO
>+                                                        sframeRequireFrameEncryption:NO];
> 
>   RTCMediaConstraints *contraints = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{}
>       optionalConstraints:nil];
>@@ -89,6 +94,14 @@ - (void)testConfigurationGetter {
>   EXPECT_EQ(config.continualGatheringPolicy, newConfig.continualGatheringPolicy);
>   EXPECT_EQ(config.shouldPruneTurnPorts, newConfig.shouldPruneTurnPorts);
>   EXPECT_EQ(config.activeResetSrtpParams, newConfig.activeResetSrtpParams);
>+  EXPECT_EQ(config.cryptoOptions.srtpEnableGcmCryptoSuites,
>+            newConfig.cryptoOptions.srtpEnableGcmCryptoSuites);
>+  EXPECT_EQ(config.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher,
>+            newConfig.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher);
>+  EXPECT_EQ(config.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions,
>+            newConfig.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions);
>+  EXPECT_EQ(config.cryptoOptions.sframeRequireFrameEncryption,
>+            newConfig.cryptoOptions.sframeRequireFrameEncryption);
> }
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/main.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/main.m
>deleted file mode 100644
>index 217fb4f479c056e7c2a3fa28d8ca57f586dca4ac..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/main.m
>+++ /dev/null
>@@ -1,18 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#import <Foundation/Foundation.h>
>-#import <UIKit/UIKit.h>
>-
>-int main(int argc, char* argv[]) {
>-  @autoreleasepool {
>-    return UIApplicationMain(argc, argv, nil, nil);
>-  }
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/main.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/main.mm
>new file mode 100644
>index 0000000000000000000000000000000000000000..77a88a64d3d74e35da8ee055e788010823d6fa6e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/main.mm
>@@ -0,0 +1,21 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+#import <UIKit/UIKit.h>
>+#include "test/ios/coverage_util_ios.h"
>+
>+int main(int argc, char* argv[]) {
>+  rtc::test::ConfigureCoverageReportPath();
>+
>+  @autoreleasepool {
>+    return UIApplicationMain(argc, argv, nil, nil);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/nalu_rewriter_xctest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/nalu_rewriter_xctest.mm
>new file mode 100644
>index 0000000000000000000000000000000000000000..4b049901c51ab77656b9347b2ed301f0d342061f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/nalu_rewriter_xctest.mm
>@@ -0,0 +1,414 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "common_video/h264/h264_common.h"
>+#include "components/video_codec/nalu_rewriter.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/gunit.h"
>+
>+#import <XCTest/XCTest.h>
>+
>+#if TARGET_OS_IPHONE
>+#import <AVFoundation/AVFoundation.h>
>+#import <UIKit/UIKit.h>
>+#endif
>+
>+@interface NaluRewriterTests : XCTestCase
>+
>+@end
>+
>+static const uint8_t NALU_TEST_DATA_0[] = {0xAA, 0xBB, 0xCC};
>+static const uint8_t NALU_TEST_DATA_1[] = {0xDE, 0xAD, 0xBE, 0xEF};
>+
>+// clang-format off
>+static const uint8_t SPS_PPS_BUFFER[] = {
>+  // SPS nalu.
>+  0x00, 0x00, 0x00, 0x01, 0x27, 0x42, 0x00, 0x1E, 0xAB, 0x40, 0xF0, 0x28,
>+  0xD3, 0x70, 0x20, 0x20, 0x20, 0x20,
>+  // PPS nalu.
>+  0x00, 0x00, 0x00, 0x01, 0x28, 0xCE, 0x3C, 0x30};
>+// clang-format on
>+
>+@implementation NaluRewriterTests
>+
>+- (void)testCreateVideoFormatDescription {
>+  CMVideoFormatDescriptionRef description =
>+      webrtc::CreateVideoFormatDescription(SPS_PPS_BUFFER, arraysize(SPS_PPS_BUFFER));
>+  XCTAssertTrue(description);
>+  if (description) {
>+    CFRelease(description);
>+    description = nullptr;
>+  }
>+
>+  // clang-format off
>+  const uint8_t sps_pps_not_at_start_buffer[] = {
>+    // Add some non-SPS/PPS NALUs at the beginning
>+    0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x01, 0xFF, 0x00, 0x00, 0x00, 0x01,
>+    0xAB, 0x33, 0x21,
>+    // SPS nalu.
>+    0x00, 0x00, 0x01, 0x27, 0x42, 0x00, 0x1E, 0xAB, 0x40, 0xF0, 0x28, 0xD3,
>+    0x70, 0x20, 0x20, 0x20, 0x20,
>+    // PPS nalu.
>+    0x00, 0x00, 0x01, 0x28, 0xCE, 0x3C, 0x30};
>+  // clang-format on
>+  description = webrtc::CreateVideoFormatDescription(sps_pps_not_at_start_buffer,
>+                                                     arraysize(sps_pps_not_at_start_buffer));
>+
>+  XCTAssertTrue(description);
>+
>+  if (description) {
>+    CFRelease(description);
>+    description = nullptr;
>+  }
>+
>+  const uint8_t other_buffer[] = {0x00, 0x00, 0x00, 0x01, 0x28};
>+  XCTAssertFalse(webrtc::CreateVideoFormatDescription(other_buffer, arraysize(other_buffer)));
>+}
>+
>+- (void)testReadEmptyInput {
>+  const uint8_t annex_b_test_data[] = {0x00};
>+  webrtc::AnnexBBufferReader reader(annex_b_test_data, 0);
>+  const uint8_t* nalu = nullptr;
>+  size_t nalu_length = 0;
>+  XCTAssertEqual(0u, reader.BytesRemaining());
>+  XCTAssertFalse(reader.ReadNalu(&nalu, &nalu_length));
>+  XCTAssertEqual(nullptr, nalu);
>+  XCTAssertEqual(0u, nalu_length);
>+}
>+
>+- (void)testReadSingleNalu {
>+  const uint8_t annex_b_test_data[] = {0x00, 0x00, 0x00, 0x01, 0xAA};
>+  webrtc::AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
>+  const uint8_t* nalu = nullptr;
>+  size_t nalu_length = 0;
>+  XCTAssertEqual(arraysize(annex_b_test_data), reader.BytesRemaining());
>+  XCTAssertTrue(reader.ReadNalu(&nalu, &nalu_length));
>+  XCTAssertEqual(annex_b_test_data + 4, nalu);
>+  XCTAssertEqual(1u, nalu_length);
>+  XCTAssertEqual(0u, reader.BytesRemaining());
>+  XCTAssertFalse(reader.ReadNalu(&nalu, &nalu_length));
>+  XCTAssertEqual(nullptr, nalu);
>+  XCTAssertEqual(0u, nalu_length);
>+}
>+
>+- (void)testReadSingleNalu3ByteHeader {
>+  const uint8_t annex_b_test_data[] = {0x00, 0x00, 0x01, 0xAA};
>+  webrtc::AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
>+  const uint8_t* nalu = nullptr;
>+  size_t nalu_length = 0;
>+  XCTAssertEqual(arraysize(annex_b_test_data), reader.BytesRemaining());
>+  XCTAssertTrue(reader.ReadNalu(&nalu, &nalu_length));
>+  XCTAssertEqual(annex_b_test_data + 3, nalu);
>+  XCTAssertEqual(1u, nalu_length);
>+  XCTAssertEqual(0u, reader.BytesRemaining());
>+  XCTAssertFalse(reader.ReadNalu(&nalu, &nalu_length));
>+  XCTAssertEqual(nullptr, nalu);
>+  XCTAssertEqual(0u, nalu_length);
>+}
>+
>+- (void)testReadMissingNalu {
>+  // clang-format off
>+  const uint8_t annex_b_test_data[] = {0x01,
>+                                       0x00, 0x01,
>+                                       0x00, 0x00, 0x00, 0xFF};
>+  // clang-format on
>+  webrtc::AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
>+  const uint8_t* nalu = nullptr;
>+  size_t nalu_length = 0;
>+  XCTAssertEqual(0u, reader.BytesRemaining());
>+  XCTAssertFalse(reader.ReadNalu(&nalu, &nalu_length));
>+  XCTAssertEqual(nullptr, nalu);
>+  XCTAssertEqual(0u, nalu_length);
>+}
>+
>+- (void)testReadMultipleNalus {
>+  // clang-format off
>+  const uint8_t annex_b_test_data[] = {0x00, 0x00, 0x00, 0x01, 0xFF,
>+                                       0x01,
>+                                       0x00, 0x01,
>+                                       0x00, 0x00, 0x00, 0xFF,
>+                                       0x00, 0x00, 0x01, 0xAA, 0xBB};
>+  // clang-format on
>+  webrtc::AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
>+  const uint8_t* nalu = nullptr;
>+  size_t nalu_length = 0;
>+  XCTAssertEqual(arraysize(annex_b_test_data), reader.BytesRemaining());
>+  XCTAssertTrue(reader.ReadNalu(&nalu, &nalu_length));
>+  XCTAssertEqual(annex_b_test_data + 4, nalu);
>+  XCTAssertEqual(8u, nalu_length);
>+  XCTAssertEqual(5u, reader.BytesRemaining());
>+  XCTAssertTrue(reader.ReadNalu(&nalu, &nalu_length));
>+  XCTAssertEqual(annex_b_test_data + 15, nalu);
>+  XCTAssertEqual(2u, nalu_length);
>+  XCTAssertEqual(0u, reader.BytesRemaining());
>+  XCTAssertFalse(reader.ReadNalu(&nalu, &nalu_length));
>+  XCTAssertEqual(nullptr, nalu);
>+  XCTAssertEqual(0u, nalu_length);
>+}
>+
>+- (void)testEmptyOutputBuffer {
>+  const uint8_t expected_buffer[] = {0x00};
>+  const size_t buffer_size = 1;
>+  std::unique_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
>+  memset(buffer.get(), 0, buffer_size);
>+  webrtc::AvccBufferWriter writer(buffer.get(), 0);
>+  XCTAssertEqual(0u, writer.BytesRemaining());
>+  XCTAssertFalse(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
>+  XCTAssertEqual(0, memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
>+}
>+
>+- (void)testWriteSingleNalu {
>+  const uint8_t expected_buffer[] = {
>+      0x00, 0x00, 0x00, 0x03, 0xAA, 0xBB, 0xCC,
>+  };
>+  const size_t buffer_size = arraysize(NALU_TEST_DATA_0) + 4;
>+  std::unique_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
>+  webrtc::AvccBufferWriter writer(buffer.get(), buffer_size);
>+  XCTAssertEqual(buffer_size, writer.BytesRemaining());
>+  XCTAssertTrue(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
>+  XCTAssertEqual(0u, writer.BytesRemaining());
>+  XCTAssertFalse(writer.WriteNalu(NALU_TEST_DATA_1, arraysize(NALU_TEST_DATA_1)));
>+  XCTAssertEqual(0, memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
>+}
>+
>+- (void)testWriteMultipleNalus {
>+  // clang-format off
>+  const uint8_t expected_buffer[] = {
>+    0x00, 0x00, 0x00, 0x03, 0xAA, 0xBB, 0xCC,
>+    0x00, 0x00, 0x00, 0x04, 0xDE, 0xAD, 0xBE, 0xEF
>+  };
>+  // clang-format on
>+  const size_t buffer_size = arraysize(NALU_TEST_DATA_0) + arraysize(NALU_TEST_DATA_1) + 8;
>+  std::unique_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
>+  webrtc::AvccBufferWriter writer(buffer.get(), buffer_size);
>+  XCTAssertEqual(buffer_size, writer.BytesRemaining());
>+  XCTAssertTrue(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
>+  XCTAssertEqual(buffer_size - (arraysize(NALU_TEST_DATA_0) + 4), writer.BytesRemaining());
>+  XCTAssertTrue(writer.WriteNalu(NALU_TEST_DATA_1, arraysize(NALU_TEST_DATA_1)));
>+  XCTAssertEqual(0u, writer.BytesRemaining());
>+  XCTAssertEqual(0, memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
>+}
>+
>+- (void)testOverflow {
>+  const uint8_t expected_buffer[] = {0x00, 0x00, 0x00};
>+  const size_t buffer_size = arraysize(NALU_TEST_DATA_0);
>+  std::unique_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
>+  memset(buffer.get(), 0, buffer_size);
>+  webrtc::AvccBufferWriter writer(buffer.get(), buffer_size);
>+  XCTAssertEqual(buffer_size, writer.BytesRemaining());
>+  XCTAssertFalse(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
>+  XCTAssertEqual(buffer_size, writer.BytesRemaining());
>+  XCTAssertEqual(0, memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
>+}
>+
>+- (void)testH264AnnexBBufferToCMSampleBuffer {
>+  // clang-format off
>+  const uint8_t annex_b_test_data[] = {
>+    0x00,
>+    0x00, 0x00, 0x01,
>+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
>+    0x00, 0x00, 0x01,
>+    0xAA, 0xFF, // second chunk, 2 bytes
>+    0x00, 0x00, 0x01,
>+    0xBB};  // third chunk, 1 byte, will not fit into output array
>+
>+  const uint8_t expected_cmsample_data[] = {
>+    0x00, 0x00, 0x00, 0x04,
>+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
>+    0x00, 0x00, 0x00, 0x02,
>+    0xAA, 0xFF}; // second chunk, 2 bytes
>+  // clang-format on
>+
>+  CMMemoryPoolRef memory_pool = CMMemoryPoolCreate(nil);
>+  CMSampleBufferRef out_sample_buffer = nil;
>+  CMVideoFormatDescriptionRef description = [self createDescription];
>+
>+  Boolean result = webrtc::H264AnnexBBufferToCMSampleBuffer(annex_b_test_data,
>+                                                            arraysize(annex_b_test_data),
>+                                                            description,
>+                                                            &out_sample_buffer,
>+                                                            memory_pool);
>+
>+  XCTAssertTrue(result);
>+
>+  XCTAssertEqual(description, CMSampleBufferGetFormatDescription(out_sample_buffer));
>+
>+  char* data_ptr = nullptr;
>+  CMBlockBufferRef block_buffer = CMSampleBufferGetDataBuffer(out_sample_buffer);
>+  size_t block_buffer_size = CMBlockBufferGetDataLength(block_buffer);
>+  CMBlockBufferGetDataPointer(block_buffer, 0, nullptr, nullptr, &data_ptr);
>+  XCTAssertEqual(block_buffer_size, arraysize(annex_b_test_data));
>+
>+  int data_comparison_result =
>+      memcmp(expected_cmsample_data, data_ptr, arraysize(expected_cmsample_data));
>+
>+  XCTAssertEqual(0, data_comparison_result);
>+
>+  if (description) {
>+    CFRelease(description);
>+    description = nullptr;
>+  }
>+
>+  CMMemoryPoolInvalidate(memory_pool);
>+  CFRelease(memory_pool);
>+}
>+
>+- (void)testH264CMSampleBufferToAnnexBBuffer {
>+  // clang-format off
>+  const uint8_t cmsample_data[] = {
>+    0x00, 0x00, 0x00, 0x04,
>+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
>+    0x00, 0x00, 0x00, 0x02,
>+    0xAA, 0xFF}; // second chunk, 2 bytes
>+
>+  const uint8_t expected_annex_b_data[] = {
>+    0x00, 0x00, 0x00, 0x01,
>+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
>+    0x00, 0x00, 0x00, 0x01,
>+    0xAA, 0xFF}; // second chunk, 2 bytes
>+  // clang-format on
>+
>+  rtc::Buffer annexb_buffer(arraysize(cmsample_data));
>+  std::unique_ptr<webrtc::RTPFragmentationHeader> out_header_ptr;
>+  CMSampleBufferRef sample_buffer =
>+      [self createCMSampleBufferRef:(void*)cmsample_data cmsampleSize:arraysize(cmsample_data)];
>+
>+  Boolean result = webrtc::H264CMSampleBufferToAnnexBBuffer(sample_buffer,
>+                                                            /* is_keyframe */ false,
>+                                                            &annexb_buffer,
>+                                                            &out_header_ptr);
>+
>+  XCTAssertTrue(result);
>+
>+  XCTAssertEqual(arraysize(expected_annex_b_data), annexb_buffer.size());
>+
>+  int data_comparison_result =
>+      memcmp(expected_annex_b_data, annexb_buffer.data(), arraysize(expected_annex_b_data));
>+
>+  XCTAssertEqual(0, data_comparison_result);
>+
>+  webrtc::RTPFragmentationHeader* out_header = out_header_ptr.get();
>+
>+  XCTAssertEqual(2, (int)out_header->Size());
>+
>+  XCTAssertEqual(4, (int)out_header->Offset(0));
>+  XCTAssertEqual(4, (int)out_header->Length(0));
>+  XCTAssertEqual(0, (int)out_header->TimeDiff(0));
>+  XCTAssertEqual(0, (int)out_header->PayloadType(0));
>+
>+  XCTAssertEqual(12, (int)out_header->Offset(1));
>+  XCTAssertEqual(2, (int)out_header->Length(1));
>+  XCTAssertEqual(0, (int)out_header->TimeDiff(1));
>+  XCTAssertEqual(0, (int)out_header->PayloadType(1));
>+}
>+
>+- (void)testH264CMSampleBufferToAnnexBBufferWithKeyframe {
>+  // clang-format off
>+  const uint8_t cmsample_data[] = {
>+    0x00, 0x00, 0x00, 0x04,
>+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
>+    0x00, 0x00, 0x00, 0x02,
>+    0xAA, 0xFF}; // second chunk, 2 bytes
>+
>+  const uint8_t expected_annex_b_data[] = {
>+    0x00, 0x00, 0x00, 0x01,
>+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
>+    0x00, 0x00, 0x00, 0x01,
>+    0xAA, 0xFF}; // second chunk, 2 bytes
>+  // clang-format on
>+
>+  rtc::Buffer annexb_buffer(arraysize(cmsample_data));
>+  std::unique_ptr<webrtc::RTPFragmentationHeader> out_header_ptr;
>+  CMSampleBufferRef sample_buffer =
>+      [self createCMSampleBufferRef:(void*)cmsample_data cmsampleSize:arraysize(cmsample_data)];
>+
>+  Boolean result = webrtc::H264CMSampleBufferToAnnexBBuffer(sample_buffer,
>+                                                            /* is_keyframe */ true,
>+                                                            &annexb_buffer,
>+                                                            &out_header_ptr);
>+
>+  XCTAssertTrue(result);
>+
>+  XCTAssertEqual(arraysize(SPS_PPS_BUFFER) + arraysize(expected_annex_b_data),
>+                 annexb_buffer.size());
>+
>+  XCTAssertEqual(0, memcmp(SPS_PPS_BUFFER, annexb_buffer.data(), arraysize(SPS_PPS_BUFFER)));
>+
>+  XCTAssertEqual(0,
>+                 memcmp(expected_annex_b_data,
>+                        annexb_buffer.data() + arraysize(SPS_PPS_BUFFER),
>+                        arraysize(expected_annex_b_data)));
>+
>+  webrtc::RTPFragmentationHeader* out_header = out_header_ptr.get();
>+
>+  XCTAssertEqual(4, (int)out_header->Size());
>+
>+  XCTAssertEqual(4, (int)out_header->Offset(0));
>+  XCTAssertEqual(14, (int)out_header->Length(0));
>+  XCTAssertEqual(0, (int)out_header->TimeDiff(0));
>+  XCTAssertEqual(0, (int)out_header->PayloadType(0));
>+
>+  XCTAssertEqual(22, (int)out_header->Offset(1));
>+  XCTAssertEqual(4, (int)out_header->Length(1));
>+  XCTAssertEqual(0, (int)out_header->TimeDiff(1));
>+  XCTAssertEqual(0, (int)out_header->PayloadType(1));
>+
>+  XCTAssertEqual(30, (int)out_header->Offset(2));
>+  XCTAssertEqual(4, (int)out_header->Length(2));
>+  XCTAssertEqual(0, (int)out_header->TimeDiff(2));
>+  XCTAssertEqual(0, (int)out_header->PayloadType(2));
>+
>+  XCTAssertEqual(38, (int)out_header->Offset(3));
>+  XCTAssertEqual(2, (int)out_header->Length(3));
>+  XCTAssertEqual(0, (int)out_header->TimeDiff(3));
>+  XCTAssertEqual(0, (int)out_header->PayloadType(3));
>+}
>+
>+- (CMVideoFormatDescriptionRef)createDescription {
>+  CMVideoFormatDescriptionRef description =
>+      webrtc::CreateVideoFormatDescription(SPS_PPS_BUFFER, arraysize(SPS_PPS_BUFFER));
>+  XCTAssertTrue(description);
>+  return description;
>+}
>+
>+- (CMSampleBufferRef)createCMSampleBufferRef:(void*)cmsampleData cmsampleSize:(size_t)cmsampleSize {
>+  CMSampleBufferRef sample_buffer = nil;
>+  OSStatus status;
>+
>+  CMVideoFormatDescriptionRef description = [self createDescription];
>+  CMBlockBufferRef block_buffer = nullptr;
>+
>+  status = CMBlockBufferCreateWithMemoryBlock(nullptr,
>+                                              cmsampleData,
>+                                              cmsampleSize,
>+                                              nullptr,
>+                                              nullptr,
>+                                              0,
>+                                              cmsampleSize,
>+                                              kCMBlockBufferAssureMemoryNowFlag,
>+                                              &block_buffer);
>+
>+  status = CMSampleBufferCreate(nullptr,
>+                                block_buffer,
>+                                true,
>+                                nullptr,
>+                                nullptr,
>+                                description,
>+                                1,
>+                                0,
>+                                nullptr,
>+                                0,
>+                                nullptr,
>+                                &sample_buffer);
>+
>+  return sample_buffer;
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/objc_video_encoder_factory_tests.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/objc_video_encoder_factory_tests.mm
>index d009d51bba3154547c2bc14eed2caf63066265b0..50c30c0c7442753371607a01fb0fe5deea2fe9fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/objc_video_encoder_factory_tests.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/unittests/objc_video_encoder_factory_tests.mm
>@@ -82,7 +82,6 @@ TEST(ObjCVideoEncoderFactoryTest, EncodeReturnsOKOnSuccess) {
>   webrtc::VideoFrame frame(buffer, webrtc::kVideoRotation_0, 0);
>   webrtc::CodecSpecificInfo info;
>   info.codecType = webrtc::kVideoCodecH264;
>-  info.codec_name = "H264";
>   std::vector<webrtc::FrameType> frame_types;
> 
>   EXPECT_EQ(encoder->Encode(frame, &info, &frame_types), WEBRTC_VIDEO_CODEC_OK);
>@@ -99,7 +98,6 @@ TEST(ObjCVideoEncoderFactoryTest, EncodeReturnsErrorOnFail) {
>   webrtc::VideoFrame frame(buffer, webrtc::kVideoRotation_0, 0);
>   webrtc::CodecSpecificInfo info;
>   info.codecType = webrtc::kVideoCodecH264;
>-  info.codec_name = "H264";
>   std::vector<webrtc::FrameType> frame_types;
> 
>   EXPECT_EQ(encoder->Encode(frame, &info, &frame_types), WEBRTC_VIDEO_CODEC_ERROR);
>@@ -117,12 +115,6 @@ TEST(ObjCVideoEncoderFactoryTest, ReleaseEncodeReturnsErrorOnFail) {
>   EXPECT_EQ(encoder->Release(), WEBRTC_VIDEO_CODEC_ERROR);
> }
> 
>-TEST(ObjCVideoEncoderFactoryTest, SetChannelParametersAlwaysReturnsOK) {
>-  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateErrorEncoderFactory());
>-
>-  EXPECT_EQ(encoder->SetChannelParameters(1, 1), WEBRTC_VIDEO_CODEC_OK);
>-}
>-
> TEST(ObjCVideoEncoderFactoryTest, SetRatesReturnsOKOnSuccess) {
>   std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateOKEncoderFactory());
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/BUILD.gn
>index de300f18cf2faf3b94b6f5900fed5f7bbd152778..c67a5be3654de6224442ea13f9548a70cc3558f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/BUILD.gn
>@@ -40,6 +40,7 @@ rtc_source_set("rtc_stats_test_utils") {
>   deps = [
>     ":rtc_stats",
>     "../api:rtc_stats_api",
>+    "../rtc_base/system:rtc_export",
>   ]
> }
> 
>@@ -56,9 +57,9 @@ if (rtc_include_tests) {
>       ":rtc_stats_test_utils",
>       "../api:rtc_stats_api",
>       "../rtc_base:checks",
>+      "../rtc_base:gunit_helpers",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_main",
>-      "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:rtc_json",
>     ]
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_objects.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_objects.cc
>index dcc1180f16f10ffc285adbfe6702f166c71ea851..cd52a55971beba3efff69057aae2e6206fd5723a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_objects.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_objects.cc
>@@ -374,7 +374,9 @@ WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track",
>                      &total_samples_received,
>                      &total_samples_duration,
>                      &concealed_samples,
>-                     &concealment_events);
>+                     &concealment_events,
>+                     &jitter_buffer_flushes,
>+                     &delayed_packet_outage_samples);
> // clang-format on
> 
> RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(const std::string& id,
>@@ -410,7 +412,9 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id,
>       total_samples_received("totalSamplesReceived"),
>       total_samples_duration("totalSamplesDuration"),
>       concealed_samples("concealedSamples"),
>-      concealment_events("concealmentEvents") {
>+      concealment_events("concealmentEvents"),
>+      jitter_buffer_flushes("jitterBufferFlushes"),
>+      delayed_packet_outage_samples("delayedPacketOutageSamples") {
>   RTC_DCHECK(kind == RTCMediaStreamTrackKind::kAudio ||
>              kind == RTCMediaStreamTrackKind::kVideo);
> }
>@@ -442,7 +446,9 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
>       total_samples_received(other.total_samples_received),
>       total_samples_duration(other.total_samples_duration),
>       concealed_samples(other.concealed_samples),
>-      concealment_events(other.concealment_events) {}
>+      concealment_events(other.concealment_events),
>+      jitter_buffer_flushes(other.jitter_buffer_flushes),
>+      delayed_packet_outage_samples(other.delayed_packet_outage_samples) {}
> 
> RTCMediaStreamTrackStats::~RTCMediaStreamTrackStats() {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/test/rtcteststats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/test/rtcteststats.h
>index 7c8e1051ff10ace4568c25749a92d796bbcad8a5..a7d2c91da061d2452251b860ff02c0a5d129c7a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/test/rtcteststats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/test/rtcteststats.h
>@@ -15,10 +15,11 @@
> #include <vector>
> 
> #include "api/stats/rtcstats.h"
>+#include "rtc_base/system/rtc_export.h"
> 
> namespace webrtc {
> 
>-class RTCTestStats : public RTCStats {
>+class RTC_EXPORT RTCTestStats : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide.md b/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide.md
>index 2a35fdc5d1a476ed75692e2101481e367683e7ab..391c45b644d041a8942401ad3debd6de84161cb4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide.md
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide.md
>@@ -77,6 +77,18 @@ instead of                          | use
> 
> See [the source](api/array_view.h) for more detailed docs.
> 
>+### `absl::optional<T>` as function argument
>+
>+`absl::optional<T>` is generally a good choice when you want to pass a
>+possibly missing `T` to a function&mdash;provided of course that `T`
>+is a type that it makes sense to pass by value.
>+
>+However, when you want to avoid pass-by-value, generally **do not pass
>+`const absl::optional<T>&`; use `const T*` instead.** `const
>+absl::optional<T>&` forces the caller to store the `T` in an
>+`absl::optional<T>`; `const T*`, on the other hand, makes no
>+assumptions about how the `T` is stored.
>+
> ### sigslot
> 
> sigslot is a lightweight library that adds a signal/slot language
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/BUILD.gn
>index eb6005226593b4eea44b19fd94afe830b8391dbc..da83d12e8db1919428a49391ab1146a40f8ef3b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/BUILD.gn
>@@ -33,7 +33,7 @@ rtc_static_library("system_wrappers") {
>   libs = []
>   deps = [
>     ":cpu_features_api",
>-    "..:webrtc_common",
>+    "../api:array_view",
>     "../modules:module_api_public",
>     "../rtc_base:checks",
>     "../rtc_base/synchronization:rw_lock_wrapper",
>@@ -89,9 +89,6 @@ rtc_source_set("cpu_features_api") {
>   sources = [
>     "include/cpu_features_wrapper.h",
>   ]
>-  deps = [
>-    "..:webrtc_common",
>-  ]
> }
> 
> rtc_source_set("field_trial") {
>@@ -105,6 +102,16 @@ rtc_source_set("field_trial") {
>   if (rtc_exclude_field_trial_default) {
>     defines = [ "WEBRTC_EXCLUDE_FIELD_TRIAL_DEFAULT" ]
>   }
>+  if (build_with_chromium) {
>+    # When WebRTC is built as part of Chromium it should exclude the default
>+    # implementation of field_trial unless it is building for NACL or
>+    # Chromecast.
>+    if (!is_nacl && !is_chromecast) {
>+      deps = [
>+        "../../webrtc_overrides:field_trial",
>+      ]
>+    }
>+  }
> }
> 
> rtc_source_set("metrics") {
>@@ -123,6 +130,9 @@ rtc_source_set("metrics") {
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>   ]
>+  if (build_with_chromium) {
>+    deps += [ "../../webrtc_overrides:metrics" ]
>+  }
> }
> 
> if (is_android && !build_with_mozilla) {
>@@ -166,6 +176,7 @@ if (rtc_include_tests) {
>       "..:webrtc_common",
>       "../rtc_base:rtc_base_approved",
>       "../test:test_main",
>+      "../test:test_support",
>       "//testing/gtest",
>     ]
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/OWNERS
>index 13b08a36ef8f482b49b6aff2a284af0d310275fd..a08f6114b9275df12de45c15efef24fef99d83df 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/OWNERS
>@@ -2,7 +2,6 @@ henrika@webrtc.org
> mflodman@webrtc.org
> niklas.enbom@webrtc.org
> nisse@webrtc.org
>-perkj@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
> # structural changes, please get a review from a reviewer in this file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/clock.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/clock.h
>index 4b6eab8edcebe598a3b8dcad83081ab243b2f2c5..f1fc11f17298e8762c93ccbdc6846043a21b14de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/clock.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/clock.h
>@@ -11,6 +11,7 @@
> #ifndef SYSTEM_WRAPPERS_INCLUDE_CLOCK_H_
> #define SYSTEM_WRAPPERS_INCLUDE_CLOCK_H_
> 
>+#include <stdint.h>
> #include <memory>
> 
> #include "rtc_base/synchronization/rw_lock_wrapper.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/event_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/event_wrapper.h
>index 0531ddb5638be0ece8456e98f6c3f421d59f570e..989e7929b39a08182301110ae4f2b522f991b4eb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/event_wrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/event_wrapper.h
>@@ -12,11 +12,7 @@
> #define SYSTEM_WRAPPERS_INCLUDE_EVENT_WRAPPER_H_
> 
> namespace webrtc {
>-enum EventTypeWrapper {
>-  kEventSignaled = 1,
>-  kEventError = 2,
>-  kEventTimeout = 3
>-};
>+enum EventTypeWrapper { kEventSignaled = 1, kEventTimeout = 2 };
> 
> #define WEBRTC_EVENT_INFINITE 0xffffffff
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h
>index 2a2cda02d03f59a34d621088ea8ea9c06c455012..f00ecf2622aefd817717086b63ab4eb4d9b33929 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h
>@@ -15,7 +15,6 @@
> #include <memory>
> #include <string>
> 
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/rtp_to_ntp_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/rtp_to_ntp_estimator.h
>index d7009d8b1f4504db5d7283bfa79d92e2889f445c..c244c4ff2745209bfd48ec45d9c8b6434aefce6a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/rtp_to_ntp_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/rtp_to_ntp_estimator.h
>@@ -11,10 +11,12 @@
> #ifndef SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_ESTIMATOR_H_
> #define SYSTEM_WRAPPERS_INCLUDE_RTP_TO_NTP_ESTIMATOR_H_
> 
>+#include <stdint.h>
> #include <list>
> 
> #include "absl/types/optional.h"
> #include "modules/include/module_common_types_public.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/numerics/moving_median_filter.h"
> #include "system_wrappers/include/ntp_time.h"
> 
>@@ -40,23 +42,13 @@ class RtpToNtpEstimator {
> 
>   // Estimated parameters from RTP and NTP timestamp pairs in |measurements_|.
>   struct Parameters {
>-    // Implicit conversion from int because MovingMedianFilter returns 0
>-    // internally if no samples are present. However, it should never happen as
>-    // we don't ask smoothing_filter_ to return anything if there were no
>-    // samples.
>-    Parameters(const int& value) {  // NOLINT
>-      RTC_NOTREACHED();
>-    }
>     Parameters() : frequency_khz(0.0), offset_ms(0.0) {}
> 
>+    Parameters(double frequency_khz, double offset_ms)
>+        : frequency_khz(frequency_khz), offset_ms(offset_ms) {}
>+
>     double frequency_khz;
>     double offset_ms;
>-
>-    // Needed to make it work inside MovingMedianFilter
>-    bool operator<(const Parameters& other) const;
>-    bool operator==(const Parameters& other) const;
>-    bool operator<=(const Parameters& other) const;
>-    bool operator!=(const Parameters& other) const;
>   };
> 
>   // Updates measurements with RTP/NTP timestamp pair from a RTCP sender report.
>@@ -68,7 +60,7 @@ class RtpToNtpEstimator {
> 
>   // Converts an RTP timestamp to the NTP domain in milliseconds.
>   // Returns true on success, false otherwise.
>-  bool Estimate(int64_t rtp_timestamp, int64_t* rtp_timestamp_ms) const;
>+  bool Estimate(int64_t rtp_timestamp, int64_t* ntp_timestamp_ms) const;
> 
>   // Returns estimated rtp to ntp linear transform parameters.
>   const absl::optional<Parameters> params() const;
>@@ -80,8 +72,7 @@ class RtpToNtpEstimator {
> 
>   int consecutive_invalid_samples_;
>   std::list<RtcpMeasurement> measurements_;
>-  MovingMedianFilter<Parameters> smoothing_filter_;
>-  bool params_calculated_;
>+  absl::optional<Parameters> params_;
>   mutable TimestampUnwrapper unwrapper_;
> };
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc
>index 35ab5f0878379ecf55c5bea85d9f1ac56e32036d..4f5d9cff1f3a3a73c39abb32adf7713da9b0e586 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc
>@@ -17,6 +17,8 @@
> 
> #include <mmsystem.h>
> 
>+#include "rtc_base/criticalsection.h"
>+
> #elif defined(WEBRTC_POSIX)
> 
> #include <sys/time.h>
>@@ -24,7 +26,6 @@
> 
> #endif  // defined(WEBRTC_POSIX)
> 
>-#include "rtc_base/criticalsection.h"
> #include "rtc_base/synchronization/rw_lock_wrapper.h"
> #include "rtc_base/timeutils.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event.cc
>index d1d2cdaff077cc3cbb3d4b27784b5a2d138c5d5e..0c4ce10157c21611169dbbeb73cd09318d590262 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event.cc
>@@ -14,9 +14,6 @@
> #include <windows.h>
> #elif defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
> #include <ApplicationServices/ApplicationServices.h>
>-#include <pthread.h>
>-#else
>-#include <pthread.h>
> #endif
> 
> #include "rtc_base/event.h"
>@@ -25,7 +22,6 @@ namespace webrtc {
> 
> class EventWrapperImpl : public EventWrapper {
>  public:
>-  EventWrapperImpl() : event_(false, false) {}
>   ~EventWrapperImpl() override {}
> 
>   bool Set() override {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/field_trial.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/field_trial.cc
>index 60158f48865bc51aafd7a371323524ab5c2feab3..ac7131144aef2b9713500fc363fa49bc361759e3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/field_trial.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/field_trial.cc
>@@ -9,6 +9,7 @@
> 
> #include "system_wrappers/include/field_trial.h"
> 
>+#include <stddef.h>
> #include <string>
> 
> // Simple field trial implementation, which allows client to
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator.cc
>index 730c4f6691fccc14dcf240041cb240fd1f235002..4bbf6096d90c5cc164c7eb4c76f83b3b90fe4561 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator.cc
>@@ -10,30 +10,23 @@
> 
> #include "system_wrappers/include/rtp_to_ntp_estimator.h"
> 
>+#include <stddef.h>
>+#include <cmath>
>+#include <vector>
>+
>+#include "api/array_view.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
> namespace {
>-// Number of RTCP SR reports to use to map between RTP and NTP.
>-const size_t kNumRtcpReportsToUse = 2;
>-// Number of parameters samples used to smooth.
>-const size_t kNumSamplesToSmooth = 20;
>-
>-// Calculates the RTP timestamp frequency from two pairs of NTP/RTP timestamps.
>-bool CalculateFrequency(int64_t ntp_ms1,
>-                        uint32_t rtp_timestamp1,
>-                        int64_t ntp_ms2,
>-                        uint32_t rtp_timestamp2,
>-                        double* frequency_khz) {
>-  if (ntp_ms1 <= ntp_ms2)
>-    return false;
>-
>-  *frequency_khz = static_cast<double>(rtp_timestamp1 - rtp_timestamp2) /
>-                   static_cast<double>(ntp_ms1 - ntp_ms2);
>-  return true;
>-}
>+// Maximum number of RTCP SR reports to use to map between RTP and NTP.
>+const size_t kNumRtcpReportsToUse = 20;
>+// Don't allow NTP timestamps to jump more than 1 hour. Chosen arbitrary as big
>+// enough to not affect normal use-cases. Yet it is smaller than RTP wrap-around
>+// half-period (90khz RTP clock wrap-arounds every 13.25 hours). After half of
>+// wrap-around period it is impossible to unwrap RTP timestamps correctly.
>+const int kMaxAllowedRtcpNtpIntervalMs = 60 * 60 * 1000;
> 
> bool Contains(const std::list<RtpToNtpEstimator::RtcpMeasurement>& measurements,
>               const RtpToNtpEstimator::RtcpMeasurement& other) {
>@@ -43,29 +36,47 @@ bool Contains(const std::list<RtpToNtpEstimator::RtcpMeasurement>& measurements,
>   }
>   return false;
> }
>-}  // namespace
> 
>-bool RtpToNtpEstimator::Parameters::operator<(const Parameters& other) const {
>-  if (frequency_khz < other.frequency_khz - 1e-6) {
>-    return true;
>-  } else if (frequency_khz > other.frequency_khz + 1e-6) {
>+// Given x[] and y[] writes out such k and b that line y=k*x+b approximates
>+// given points in the best way (Least Squares Method).
>+bool LinearRegression(rtc::ArrayView<const double> x,
>+                      rtc::ArrayView<const double> y,
>+                      double* k,
>+                      double* b) {
>+  size_t n = x.size();
>+  if (n < 2)
>+    return false;
>+
>+  if (y.size() != n)
>     return false;
>-  } else {
>-    return offset_ms < other.offset_ms - 1e-6;
>+
>+  double avg_x = 0;
>+  double avg_y = 0;
>+  for (size_t i = 0; i < n; ++i) {
>+    avg_x += x[i];
>+    avg_y += y[i];
>+  }
>+  avg_x /= n;
>+  avg_y /= n;
>+
>+  double variance_x = 0;
>+  double covariance_xy = 0;
>+  for (size_t i = 0; i < n; ++i) {
>+    double normalized_x = x[i] - avg_x;
>+    double normalized_y = y[i] - avg_y;
>+    variance_x += normalized_x * normalized_x;
>+    covariance_xy += normalized_x * normalized_y;
>   }
>-}
> 
>-bool RtpToNtpEstimator::Parameters::operator==(const Parameters& other) const {
>-  return !(other < *this || *this < other);
>-}
>+  if (std::fabs(variance_x) < 1e-8)
>+    return false;
> 
>-bool RtpToNtpEstimator::Parameters::operator!=(const Parameters& other) const {
>-  return other < *this || *this < other;
>+  *k = static_cast<double>(covariance_xy / variance_x);
>+  *b = static_cast<double>(avg_y - (*k) * avg_x);
>+  return true;
> }
> 
>-bool RtpToNtpEstimator::Parameters::operator<=(const Parameters& other) const {
>-  return !(other < *this);
>-}
>+}  // namespace
> 
> RtpToNtpEstimator::RtcpMeasurement::RtcpMeasurement(uint32_t ntp_secs,
>                                                     uint32_t ntp_frac,
>@@ -82,31 +93,29 @@ bool RtpToNtpEstimator::RtcpMeasurement::IsEqual(
> }
> 
> // Class for converting an RTP timestamp to the NTP domain.
>-RtpToNtpEstimator::RtpToNtpEstimator()
>-    : consecutive_invalid_samples_(0),
>-      smoothing_filter_(kNumSamplesToSmooth),
>-      params_calculated_(false) {}
>+RtpToNtpEstimator::RtpToNtpEstimator() : consecutive_invalid_samples_(0) {}
> 
> RtpToNtpEstimator::~RtpToNtpEstimator() {}
> 
> void RtpToNtpEstimator::UpdateParameters() {
>-  if (measurements_.size() != kNumRtcpReportsToUse)
>+  if (measurements_.size() < 2)
>     return;
> 
>-  Parameters params;
>-  int64_t timestamp_new = measurements_.front().unwrapped_rtp_timestamp;
>-  int64_t timestamp_old = measurements_.back().unwrapped_rtp_timestamp;
>-
>-  int64_t ntp_ms_new = measurements_.front().ntp_time.ToMs();
>-  int64_t ntp_ms_old = measurements_.back().ntp_time.ToMs();
>+  std::vector<double> x;
>+  std::vector<double> y;
>+  x.reserve(measurements_.size());
>+  y.reserve(measurements_.size());
>+  for (auto it = measurements_.begin(); it != measurements_.end(); ++it) {
>+    x.push_back(it->unwrapped_rtp_timestamp);
>+    y.push_back(it->ntp_time.ToMs());
>+  }
>+  double slope, offset;
> 
>-  if (!CalculateFrequency(ntp_ms_new, timestamp_new, ntp_ms_old, timestamp_old,
>-                          &params.frequency_khz)) {
>+  if (!LinearRegression(x, y, &slope, &offset)) {
>     return;
>   }
>-  params.offset_ms = timestamp_new - params.frequency_khz * ntp_ms_new;
>-  params_calculated_ = true;
>-  smoothing_filter_.Insert(params);
>+
>+  params_.emplace(1 / slope, offset);
> }
> 
> bool RtpToNtpEstimator::UpdateMeasurements(uint32_t ntp_secs,
>@@ -132,7 +141,8 @@ bool RtpToNtpEstimator::UpdateMeasurements(uint32_t ntp_secs,
>   if (!measurements_.empty()) {
>     int64_t old_rtp_timestamp = measurements_.front().unwrapped_rtp_timestamp;
>     int64_t old_ntp_ms = measurements_.front().ntp_time.ToMs();
>-    if (ntp_ms_new <= old_ntp_ms) {
>+    if (ntp_ms_new <= old_ntp_ms ||
>+        ntp_ms_new > old_ntp_ms + kMaxAllowedRtcpNtpIntervalMs) {
>       invalid_sample = true;
>     } else if (unwrapped_rtp_timestamp <= old_rtp_timestamp) {
>       RTC_LOG(LS_WARNING)
>@@ -152,8 +162,7 @@ bool RtpToNtpEstimator::UpdateMeasurements(uint32_t ntp_secs,
>     RTC_LOG(LS_WARNING) << "Multiple consecutively invalid RTCP SR reports, "
>                            "clearing measurements.";
>     measurements_.clear();
>-    smoothing_filter_.Reset();
>-    params_calculated_ = false;
>+    params_ = absl::nullopt;
>   }
>   consecutive_invalid_samples_ = 0;
> 
>@@ -170,35 +179,29 @@ bool RtpToNtpEstimator::UpdateMeasurements(uint32_t ntp_secs,
> }
> 
> bool RtpToNtpEstimator::Estimate(int64_t rtp_timestamp,
>-                                 int64_t* rtp_timestamp_ms) const {
>-  if (!params_calculated_)
>+                                 int64_t* ntp_timestamp_ms) const {
>+  if (!params_)
>     return false;
> 
>   int64_t rtp_timestamp_unwrapped = unwrapper_.Unwrap(rtp_timestamp);
> 
>-  Parameters params = smoothing_filter_.GetFilteredValue();
>-
>   // params_calculated_ should not be true unless ms params.frequency_khz has
>   // been calculated to something non zero.
>-  RTC_DCHECK_NE(params.frequency_khz, 0.0);
>+  RTC_DCHECK_NE(params_->frequency_khz, 0.0);
>   double rtp_ms =
>-      (static_cast<double>(rtp_timestamp_unwrapped) - params.offset_ms) /
>-          params.frequency_khz +
>-      0.5f;
>+      static_cast<double>(rtp_timestamp_unwrapped) / params_->frequency_khz +
>+      params_->offset_ms + 0.5f;
> 
>   if (rtp_ms < 0)
>     return false;
> 
>-  *rtp_timestamp_ms = rtp_ms;
>+  *ntp_timestamp_ms = rtp_ms;
>+
>   return true;
> }
> 
> const absl::optional<RtpToNtpEstimator::Parameters> RtpToNtpEstimator::params()
>     const {
>-  absl::optional<Parameters> res;
>-  if (params_calculated_) {
>-    res.emplace(smoothing_filter_.GetFilteredValue());
>-  }
>-  return res;
>+  return params_;
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator_unittest.cc
>index 0647ec8cad6398c26c5bbbdbef5dac4557bb3d40..b0b83bbbb1c7666b4e61553825480cc269cf336c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator_unittest.cc
>@@ -9,11 +9,13 @@
>  */
> 
> #include "system_wrappers/include/rtp_to_ntp_estimator.h"
>+#include "rtc_base/random.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> namespace {
> const uint32_t kOneMsInNtpFrac = 4294967;
>+const uint32_t kOneHourInNtpSec = 60 * 60;
> const uint32_t kTimestampTicksPerMs = 90;
> }  // namespace
> 
>@@ -224,6 +226,22 @@ TEST(UpdateRtcpMeasurementTests, FailsForOldNtp) {
>       estimator.UpdateMeasurements(ntp_sec, ntp_frac, timestamp, &new_sr));
> }
> 
>+TEST(UpdateRtcpMeasurementTests, FailsForTooNewNtp) {
>+  RtpToNtpEstimator estimator;
>+  uint32_t ntp_sec = 1;
>+  uint32_t ntp_frac = 699925050;
>+  uint32_t timestamp = 0x12345678;
>+  bool new_sr;
>+  EXPECT_TRUE(
>+      estimator.UpdateMeasurements(ntp_sec, ntp_frac, timestamp, &new_sr));
>+  EXPECT_TRUE(new_sr);
>+  // Ntp time from far future, list not updated.
>+  ntp_sec += kOneHourInNtpSec * 2;
>+  timestamp += kTimestampTicksPerMs * 10;
>+  EXPECT_FALSE(
>+      estimator.UpdateMeasurements(ntp_sec, ntp_frac, timestamp, &new_sr));
>+}
>+
> TEST(UpdateRtcpMeasurementTests, FailsForEqualTimestamp) {
>   RtpToNtpEstimator estimator;
>   uint32_t ntp_sec = 0;
>@@ -292,4 +310,37 @@ TEST(RtpToNtpTests, FailsForNoParameters) {
>   EXPECT_FALSE(estimator.Estimate(timestamp, &timestamp_ms));
> }
> 
>+TEST(RtpToNtpTests, AveragesErrorOut) {
>+  RtpToNtpEstimator estimator;
>+  uint32_t ntp_sec = 1;
>+  uint32_t ntp_frac = 90000000;  // More than 1 ms.
>+  uint32_t timestamp = 0x12345678;
>+  const int kNtpSecStep = 1;  // 1 second.
>+  const int kRtpTicksPerMs = 90;
>+  const int kRtpStep = kRtpTicksPerMs * 1000;
>+  bool new_sr;
>+  EXPECT_TRUE(
>+      estimator.UpdateMeasurements(ntp_sec, ntp_frac, timestamp, &new_sr));
>+  EXPECT_TRUE(new_sr);
>+
>+  Random rand(1123536L);
>+  for (size_t i = 0; i < 1000; i++) {
>+    // Advance both timestamps by exactly 1 second.
>+    ntp_sec += kNtpSecStep;
>+    timestamp += kRtpStep;
>+    // Add upto 1ms of errors to NTP and RTP timestamps passed to estimator.
>+    EXPECT_TRUE(estimator.UpdateMeasurements(
>+        ntp_sec,
>+        ntp_frac + rand.Rand(-static_cast<int>(kOneMsInNtpFrac),
>+                             static_cast<int>(kOneMsInNtpFrac)),
>+        timestamp + rand.Rand(-kRtpTicksPerMs, kRtpTicksPerMs), &new_sr));
>+    EXPECT_TRUE(new_sr);
>+
>+    int64_t estimated_ntp_ms;
>+    EXPECT_TRUE(estimator.Estimate(timestamp, &estimated_ntp_ms));
>+    // Allow upto 2 ms of error.
>+    EXPECT_NEAR(NtpTime(ntp_sec, ntp_frac).ToMs(), estimated_ntp_ms, 2);
>+  }
>+}
>+
> };  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/test/BUILD.gn
>index 28146e0599cde9b55200b3d6fa9e771b109d5f0d..7cd7b3eaa1fb93724a26d74cdd69265a15cf0537 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/BUILD.gn
>@@ -137,6 +137,8 @@ if (is_ios) {
>     testonly = true
>     visibility = [ ":test_support" ]
>     sources = [
>+      "ios/coverage_util_ios.h",
>+      "ios/coverage_util_ios.mm",
>       "ios/test_support.h",
>       "ios/test_support.mm",
>     ]
>@@ -144,6 +146,15 @@ if (is_ios) {
>       ":perf_test",
>       "../sdk:helpers_objc",
>     ]
>+    configs += [ ":test_support_objc_config" ]
>+  }
>+
>+  config("test_support_objc_config") {
>+    defines = []
>+
>+    if (use_clang_coverage) {
>+      defines += [ "WEBRTC_IOS_ENABLE_COVERAGE" ]
>+    }
>   }
> }
> 
>@@ -196,24 +207,41 @@ rtc_source_set("test_support") {
> }
> 
> if (rtc_include_tests) {
>-  rtc_source_set("test_main") {
>+  rtc_source_set("test_main_lib") {
>     visibility = [ "*" ]
>     testonly = true
>     sources = [
>-      "test_main.cc",
>+      "test_main_lib.cc",
>+      "test_main_lib.h",
>     ]
> 
>-    public_deps = [
>-      ":test_support",
>-    ]
>     deps = [
>       ":field_trial",
>-      ":fileutils",
>       ":perf_test",
>+      ":test_support",
>       "../rtc_base:rtc_base",
>       "../system_wrappers:field_trial",
>       "../system_wrappers:metrics",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory:memory",
>+
>+      # TODO(bugs.webrtc.org/9792): This is needed for downstream projects on
>+      # Android, where it's replaced by an internal version of fileutils that
>+      # has a certain flag. Remove this once the internal fileutils has been
>+      # eliminated.
>+      "../test:fileutils",
>+    ]
>+  }
>+
>+  rtc_source_set("test_main") {
>+    visibility = [ "*" ]
>+    testonly = true
>+    sources = [
>+      "test_main.cc",
>+    ]
>+
>+    deps = [
>+      ":test_main_lib",
>     ]
>   }
> 
>@@ -309,6 +337,8 @@ if (rtc_include_tests) {
>       ":video_test_support",
>       "../api:create_simulcast_test_fixture_api",
>       "../api:simulcast_test_fixture_api",
>+      "../api/test/video:function_video_factory",
>+      "../api/video:builtin_video_bitrate_allocator_factory",
>       "../api/video:video_frame_i420",
>       "../modules/rtp_rtcp:rtp_rtcp",
>       "../modules/video_capture",
>@@ -360,7 +390,10 @@ if (rtc_include_tests) {
> 
> if (is_ios) {
>   rtc_source_set("fileutils_ios_objc") {
>-    visibility = [ ":fileutils" ]
>+    visibility = [
>+      ":fileutils",
>+      ":fileutils_override_impl",
>+    ]
>     sources = [
>       "testsupport/iosfileutils.h",
>       "testsupport/iosfileutils.mm",
>@@ -376,7 +409,10 @@ if (is_ios) {
> 
> if (is_mac) {
>   rtc_source_set("fileutils_mac_objc") {
>-    visibility = [ ":fileutils" ]
>+    visibility = [
>+      ":fileutils",
>+      ":fileutils_override_impl",
>+    ]
>     sources = [
>       "testsupport/macfileutils.h",
>       "testsupport/macfileutils.mm",
>@@ -395,6 +431,42 @@ rtc_source_set("fileutils") {
>     "testsupport/fileutils.h",
>   ]
>   deps = [
>+    ":fileutils_override_api",
>+    ":fileutils_override_impl",
>+    "..:webrtc_common",
>+    "../rtc_base:checks",
>+    "../rtc_base:rtc_base_approved",
>+    "../rtc_base/system:arch",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+  if (is_ios) {
>+    deps += [ ":fileutils_ios_objc" ]
>+  }
>+  if (is_mac) {
>+    deps += [ ":fileutils_mac_objc" ]
>+  }
>+  if (is_win) {
>+    deps += [ "../rtc_base:rtc_base" ]
>+  }
>+}
>+
>+# We separate header into own target to make it possible for downstream
>+# projects to override implementation.
>+rtc_source_set("fileutils_override_api") {
>+  testonly = true
>+  sources = [
>+    "testsupport/fileutils_override.h",
>+  ]
>+}
>+
>+rtc_source_set("fileutils_override_impl") {
>+  testonly = true
>+  visibility = [ ":fileutils" ]
>+  sources = [
>+    "testsupport/fileutils_override.cc",
>+  ]
>+  deps = [
>+    ":fileutils_override_api",
>     "..:webrtc_common",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>@@ -523,6 +595,8 @@ rtc_source_set("fake_video_codecs") {
>     "fake_decoder.h",
>     "fake_encoder.cc",
>     "fake_encoder.h",
>+    "fake_vp8_decoder.cc",
>+    "fake_vp8_decoder.h",
>     "fake_vp8_encoder.cc",
>     "fake_vp8_encoder.h",
>   ]
>@@ -532,7 +606,9 @@ rtc_source_set("fake_video_codecs") {
>   }
>   deps = [
>     "..:webrtc_common",
>+    "../api/video:encoded_image",
>     "../api/video:video_frame_i420",
>+    "../api/video_codecs:create_vp8_temporal_layers",
>     "../api/video_codecs:video_codecs_api",
>     "../common_video:common_video",
>     "../modules/video_coding:video_codec_interface",
>@@ -560,8 +636,6 @@ rtc_source_set("test_common") {
>     "encoder_settings.cc",
>     "encoder_settings.h",
>     "fake_videorenderer.h",
>-    "function_video_decoder_factory.h",
>-    "function_video_encoder_factory.h",
>     "layer_filtering_transport.cc",
>     "layer_filtering_transport.h",
>     "mock_transport.cc",
>@@ -599,6 +673,9 @@ rtc_source_set("test_common") {
>     "../api:transport_api",
>     "../api/audio_codecs:builtin_audio_decoder_factory",
>     "../api/audio_codecs:builtin_audio_encoder_factory",
>+    "../api/test/video:function_video_factory",
>+    "../api/video:builtin_video_bitrate_allocator_factory",
>+    "../api/video:video_bitrate_allocator_factory",
>     "../api/video:video_frame",
>     "../api/video_codecs:video_codecs_api",
>     "../audio",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.cc
>index f4fbf221be90f54e87fdd8650ea82b684f68035a..6e25b85a56c725385e35d2e8b3e5dc84b46a10da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.cc
>@@ -15,6 +15,7 @@
> #include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "api/video_codecs/video_encoder_config.h"
> #include "call/fake_network_pipe.h"
> #include "call/rtp_transport_controller_send.h"
>@@ -39,7 +40,8 @@ CallTest::CallTest()
>       send_event_log_(RtcEventLog::CreateNull()),
>       recv_event_log_(RtcEventLog::CreateNull()),
>       sender_call_transport_controller_(nullptr),
>-      audio_send_config_(nullptr),
>+      audio_send_config_(/*send_transport=*/nullptr,
>+                         /*media_transport=*/nullptr),
>       audio_send_stream_(nullptr),
>       bbr_network_controller_factory_(new BbrNetworkControllerFactory()),
>       fake_encoder_factory_([this]() {
>@@ -53,6 +55,7 @@ CallTest::CallTest()
>         return fake_encoder;
>       }),
>       fake_decoder_factory_([]() { return absl::make_unique<FakeDecoder>(); }),
>+      bitrate_allocator_factory_(CreateBuiltinVideoBitrateAllocatorFactory()),
>       num_video_streams_(1),
>       num_audio_streams_(0),
>       num_flexfec_streams_(0),
>@@ -75,7 +78,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
>     num_flexfec_streams_ = test->GetNumFlexfecStreams();
>     RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
>     Call::Config send_config(send_event_log_.get());
>-    test->ModifySenderCallConfig(&send_config);
>+    test->ModifySenderBitrateConfig(&send_config.bitrate_config);
>     if (num_audio_streams_ > 0) {
>       CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
>       test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
>@@ -99,7 +102,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
>     }
>     if (test->ShouldCreateReceivers()) {
>       Call::Config recv_config(recv_event_log_.get());
>-      test->ModifyReceiverCallConfig(&recv_config);
>+      test->ModifyReceiverBitrateConfig(&recv_config.bitrate_config);
>       if (num_audio_streams_ > 0) {
>         AudioState::Config audio_state_config;
>         audio_state_config.audio_mixer = AudioMixerImpl::Create();
>@@ -163,6 +166,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
>       int height = kDefaultHeight;
>       int frame_rate = kDefaultFramerate;
>       test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate);
>+      test->ModifyVideoDegradationPreference(&degradation_preference_);
>       CreateFrameGeneratorCapturer(frame_rate, width, height);
>       test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_);
>     }
>@@ -232,6 +236,8 @@ void CallTest::CreateVideoSendConfig(VideoSendStream::Config* video_config,
>   RTC_DCHECK_LE(num_video_streams + num_used_ssrcs, kNumSsrcs);
>   *video_config = VideoSendStream::Config(send_transport);
>   video_config->encoder_settings.encoder_factory = &fake_encoder_factory_;
>+  video_config->encoder_settings.bitrate_allocator_factory =
>+      bitrate_allocator_factory_.get();
>   video_config->rtp.payload_name = "FAKE";
>   video_config->rtp.payload_type = kFakeVideoSendPayloadType;
>   video_config->rtp.extensions.push_back(
>@@ -259,7 +265,8 @@ void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams,
>   RTC_DCHECK_LE(num_audio_streams, 1);
>   RTC_DCHECK_LE(num_flexfec_streams, 1);
>   if (num_audio_streams > 0) {
>-    AudioSendStream::Config audio_send_config(send_transport);
>+    AudioSendStream::Config audio_send_config(send_transport,
>+                                              /*media_transport=*/nullptr);
>     audio_send_config.rtp.ssrc = kAudioSendSsrc;
>     audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
>         kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
>@@ -726,9 +733,10 @@ void BaseTest::OnFakeAudioDevicesCreated(
>     TestAudioDeviceModule* send_audio_device,
>     TestAudioDeviceModule* recv_audio_device) {}
> 
>-void BaseTest::ModifySenderCallConfig(Call::Config* config) {}
>+void BaseTest::ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) {}
> 
>-void BaseTest::ModifyReceiverCallConfig(Call::Config* config) {}
>+void BaseTest::ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config) {
>+}
> 
> void BaseTest::OnRtpTransportControllerSendCreated(
>     RtpTransportControllerSend* controller) {}
>@@ -742,8 +750,8 @@ test::PacketTransport* BaseTest::CreateSendTransport(
>       task_queue, sender_call, this, test::PacketTransport::kSender,
>       CallTest::payload_type_map_,
>       absl::make_unique<FakeNetworkPipe>(
>-          Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                         DefaultNetworkSimulationConfig())));
>+          Clock::GetRealTimeClock(),
>+          absl::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())));
> }
> 
> test::PacketTransport* BaseTest::CreateReceiveTransport(
>@@ -752,8 +760,8 @@ test::PacketTransport* BaseTest::CreateReceiveTransport(
>       task_queue, nullptr, this, test::PacketTransport::kReceiver,
>       CallTest::payload_type_map_,
>       absl::make_unique<FakeNetworkPipe>(
>-          Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                         DefaultNetworkSimulationConfig())));
>+          Clock::GetRealTimeClock(),
>+          absl::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())));
> }
> 
> size_t BaseTest::GetNumVideoStreams() const {
>@@ -777,6 +785,9 @@ void BaseTest::ModifyVideoCaptureStartResolution(int* width,
>                                                  int* heigt,
>                                                  int* frame_rate) {}
> 
>+void BaseTest::ModifyVideoDegradationPreference(
>+    DegradationPreference* degradation_preference) {}
>+
> void BaseTest::OnVideoStreamsCreated(
>     VideoSendStream* send_stream,
>     const std::vector<VideoReceiveStream*>& receive_streams) {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.h
>index 7a5c3e9b8c1a00ed706f8042d192210677371663..8961b7fccb3f9390e12691279914ebc6425389df 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.h
>@@ -13,6 +13,9 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/test/video/function_video_decoder_factory.h"
>+#include "api/test/video/function_video_encoder_factory.h"
>+#include "api/video/video_bitrate_allocator_factory.h"
> #include "call/call.h"
> #include "call/rtp_transport_controller_send.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
>@@ -22,8 +25,6 @@
> #include "test/fake_videorenderer.h"
> #include "test/fake_vp8_encoder.h"
> #include "test/frame_generator_capturer.h"
>-#include "test/function_video_decoder_factory.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/rtp_rtcp_observer.h"
> #include "test/single_threaded_task_queue.h"
> 
>@@ -205,6 +206,7 @@ class CallTest : public ::testing::Test {
>   test::FunctionVideoEncoderFactory fake_encoder_factory_;
>   int fake_encoder_max_bitrate_ = -1;
>   test::FunctionVideoDecoderFactory fake_decoder_factory_;
>+  std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
>   // Number of simulcast substreams.
>   size_t num_video_streams_;
>   size_t num_audio_streams_;
>@@ -241,8 +243,8 @@ class BaseTest : public RtpRtcpObserver {
>       TestAudioDeviceModule* send_audio_device,
>       TestAudioDeviceModule* recv_audio_device);
> 
>-  virtual void ModifySenderCallConfig(Call::Config* config);
>-  virtual void ModifyReceiverCallConfig(Call::Config* config);
>+  virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
>+  virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);
> 
>   virtual void OnRtpTransportControllerSendCreated(
>       RtpTransportControllerSend* controller);
>@@ -261,6 +263,9 @@ class BaseTest : public RtpRtcpObserver {
>   virtual void ModifyVideoCaptureStartResolution(int* width,
>                                                  int* heigt,
>                                                  int* frame_rate);
>+  virtual void ModifyVideoDegradationPreference(
>+      DegradationPreference* degradation_preference);
>+
>   virtual void OnVideoStreamsCreated(
>       VideoSendStream* send_stream,
>       const std::vector<VideoReceiveStream*>& receive_streams);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.cc
>index c0fe2966e85ff21b712ad48282a94aea95885835..bdd61d032fc834a1d8d99e6c2416bfc438bd65fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.cc
>@@ -12,7 +12,7 @@
> 
> #include <string.h>
> 
>-#include "common_video/include/video_frame.h"
>+#include "api/video/encoded_image.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "rtc_base/checks.h"
> 
>@@ -54,7 +54,9 @@ int32_t ConfigurableFrameSizeEncoder::Encode(
>   CodecSpecificInfo specific{};
>   specific.codecType = codec_type_;
>   callback_->OnEncodedImage(encodedImage, &specific, fragmentation);
>-
>+  if (post_encode_callback_) {
>+    (*post_encode_callback_)();
>+  }
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>@@ -68,11 +70,6 @@ int32_t ConfigurableFrameSizeEncoder::Release() {
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-int32_t ConfigurableFrameSizeEncoder::SetChannelParameters(uint32_t packet_loss,
>-                                                           int64_t rtt) {
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
> int32_t ConfigurableFrameSizeEncoder::SetRateAllocation(
>     const VideoBitrateAllocation& allocation,
>     uint32_t framerate) {
>@@ -89,5 +86,10 @@ void ConfigurableFrameSizeEncoder::SetCodecType(VideoCodecType codec_type) {
>   codec_type_ = codec_type;
> }
> 
>+void ConfigurableFrameSizeEncoder::RegisterPostEncodeCallback(
>+    std::function<void(void)> post_encode_callback) {
>+  post_encode_callback_ = std::move(post_encode_callback);
>+}
>+
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.h
>index 1dc9e48f47f5ab7bf0c478038e8bdf5831bc59dd..f881513fcc27361f7a04b4bf68ae5570d3ea8709 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.h
>@@ -37,8 +37,6 @@ class ConfigurableFrameSizeEncoder : public VideoEncoder {
> 
>   int32_t Release() override;
> 
>-  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>-
>   int32_t SetRateAllocation(const VideoBitrateAllocation& allocation,
>                             uint32_t framerate) override;
> 
>@@ -46,8 +44,13 @@ class ConfigurableFrameSizeEncoder : public VideoEncoder {
> 
>   void SetCodecType(VideoCodecType codec_type_);
> 
>+  void RegisterPostEncodeCallback(
>+      std::function<void(void)> post_encode_callback);
>+
>  private:
>   EncodedImageCallback* callback_;
>+  absl::optional<std::function<void(void)>> post_encode_callback_;
>+
>   const size_t max_frame_size_;
>   size_t current_frame_size_;
>   std::unique_ptr<uint8_t[]> buffer_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.cc
>index 6aa8c9158f0de6720e225bb48eae463a79a53313..fd7369171a2200c714d49727da1f60f57385cb99 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.cc
>@@ -72,6 +72,8 @@ bool DirectTransport::SendRtp(const uint8_t* data,
>   if (send_call_) {
>     rtc::SentPacket sent_packet(options.packet_id,
>                                 clock_->TimeInMilliseconds());
>+    sent_packet.info.included_in_feedback = options.included_in_feedback;
>+    sent_packet.info.included_in_allocation = options.included_in_allocation;
>     sent_packet.info.packet_size_bytes = length;
>     sent_packet.info.packet_type = rtc::PacketType::kData;
>     send_call_->OnSentPacket(sent_packet);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.cc
>index 945559b4ee02ee85519f733ba85d689c7129970e..bfbd2bdec47a9834991a137c3e9521ff7590ada7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.cc
>@@ -61,6 +61,14 @@ std::vector<VideoStream> CreateVideoStreams(
>               ? stream.target_bitrate_bps
>               : DefaultVideoStreamFactory::kMaxBitratePerStream[i];
>       target_bitrate_bps = std::min(max_bitrate_bps, target_bitrate_bps);
>+
>+      if (stream.max_framerate > 0) {
>+        stream_settings[i].max_framerate = stream.max_framerate;
>+      }
>+      if (stream.num_temporal_layers) {
>+        RTC_DCHECK_GE(*stream.num_temporal_layers, 1);
>+        stream_settings[i].num_temporal_layers = stream.num_temporal_layers;
>+      }
>     } else {
>       max_bitrate_bps = std::min(
>           bitrate_left_bps, DefaultVideoStreamFactory::kMaxBitratePerStream[i]);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.cc
>index bc3c7b1087b4ea0f6a95879c7ae4726a3119179c..ddaba77a4f45eba13708db8ab6e9b9a443eee063 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.cc
>@@ -15,16 +15,33 @@
> #include <algorithm>
> #include <memory>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "rtc_base/checks.h"
> #include "system_wrappers/include/sleep.h"
> 
> namespace webrtc {
> namespace test {
>+namespace {
>+const int kKeyframeSizeFactor = 5;
>+
>+// Inverse of proportion of frames assigned to each temporal layer for all
>+// possible temporal layers numbers.
>+const int kTemporalLayerRateFactor[4][4] = {
>+    {1, 0, 0, 0},  // 1/1
>+    {2, 2, 0, 0},  // 1/2 + 1/2
>+    {4, 4, 2, 0},  // 1/4 + 1/4 + 1/2
>+    {8, 8, 4, 2},  // 1/8 + 1/8 + 1/4 + 1/2
>+};
>+
>+void WriteCounter(unsigned char* payload, uint32_t counter) {
>+  payload[0] = (counter & 0x00FF);
>+  payload[1] = (counter & 0xFF00) >> 8;
>+  payload[2] = (counter & 0xFF0000) >> 16;
>+  payload[3] = (counter & 0xFF000000) >> 24;
>+}
> 
>-const int kKeyframeSizeFactor = 10;
>+};  // namespace
> 
> FakeEncoder::FakeEncoder(Clock* clock)
>     : clock_(clock),
>@@ -32,6 +49,7 @@ FakeEncoder::FakeEncoder(Clock* clock)
>       configured_input_framerate_(-1),
>       max_target_bitrate_kbps_(-1),
>       pending_keyframe_(true),
>+      counter_(0),
>       debt_bytes_(0) {
>   // Generate some arbitrary not-all-zero data
>   for (size_t i = 0; i < sizeof(encoded_buffer_); ++i) {
>@@ -46,6 +64,7 @@ void FakeEncoder::SetMaxBitrate(int max_kbps) {
>   RTC_DCHECK_GE(max_kbps, -1);  // max_kbps == -1 disables it.
>   rtc::CritScope cs(&crit_sect_);
>   max_target_bitrate_kbps_ = max_kbps;
>+  SetRateAllocation(target_bitrate_, configured_input_framerate_);
> }
> 
> int32_t FakeEncoder::InitEncode(const VideoCodec* config,
>@@ -71,6 +90,7 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image,
>   int framerate;
>   VideoCodecMode mode;
>   bool keyframe;
>+  uint32_t counter;
>   {
>     rtc::CritScope cs(&crit_sect_);
>     max_framerate = config_.maxFramerate;
>@@ -88,13 +108,15 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image,
>     }
>     keyframe = pending_keyframe_;
>     pending_keyframe_ = false;
>+    counter = counter_++;
>   }
> 
>   FrameInfo frame_info =
>       NextFrame(frame_types, keyframe, num_simulcast_streams, target_bitrate,
>                 simulcast_streams, framerate);
>   for (uint8_t i = 0; i < frame_info.layers.size(); ++i) {
>-    if (frame_info.layers[i].size == 0) {
>+    constexpr int kMinPayLoadLength = 14;
>+    if (frame_info.layers[i].size < kMinPayLoadLength) {
>       // Drop this temporal layer.
>       continue;
>     }
>@@ -104,7 +126,10 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image,
>     specifics.codecType = kVideoCodecGeneric;
>     std::unique_ptr<uint8_t[]> encoded_buffer(
>         new uint8_t[frame_info.layers[i].size]);
>-    memcpy(encoded_buffer.get(), encoded_buffer_, frame_info.layers[i].size);
>+    memcpy(encoded_buffer.get(), encoded_buffer_,
>+           frame_info.layers[i].size - 4);
>+    // Write a counter to the image to make each frame unique.
>+    WriteCounter(encoded_buffer.get() + frame_info.layers[i].size - 4, counter);
>     EncodedImage encoded(encoded_buffer.get(), frame_info.layers[i].size,
>                          sizeof(encoded_buffer_));
>     encoded.SetTimestamp(input_image.timestamp());
>@@ -118,7 +143,6 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image,
>                                 ? VideoContentType::SCREENSHARE
>                                 : VideoContentType::UNSPECIFIED;
>     encoded.SetSpatialIndex(i);
>-    specifics.codec_name = ImplementationName();
>     if (callback->OnEncodedImage(encoded, &specifics, nullptr).error !=
>         EncodedImageCallback::Result::OK) {
>       return -1;
>@@ -146,6 +170,7 @@ FakeEncoder::FrameInfo FakeEncoder::NextFrame(
>     }
>   }
> 
>+  rtc::CritScope cs(&crit_sect_);
>   for (uint8_t i = 0; i < num_simulcast_streams; ++i) {
>     if (target_bitrate.GetBitrate(i, 0) > 0) {
>       int temporal_id = last_frame_info_.layers.size() > i
>@@ -166,7 +191,9 @@ FakeEncoder::FrameInfo FakeEncoder::NextFrame(
>     if (frame_info.keyframe) {
>       layer_info.temporal_id = 0;
>       size_t avg_frame_size =
>-          (target_bitrate.GetBitrate(i, 0) + 7) / (8 * framerate);
>+          (target_bitrate.GetBitrate(i, 0) + 7) *
>+          kTemporalLayerRateFactor[frame_info.layers.size() - 1][i] /
>+          (8 * framerate);
> 
>       // The first frame is a key frame and should be larger.
>       // Store the overshoot bytes and distribute them over the coming frames,
>@@ -175,7 +202,8 @@ FakeEncoder::FrameInfo FakeEncoder::NextFrame(
>       layer_info.size = kKeyframeSizeFactor * avg_frame_size;
>     } else {
>       size_t avg_frame_size =
>-          (target_bitrate.GetBitrate(i, layer_info.temporal_id) + 7) /
>+          (target_bitrate.GetBitrate(i, layer_info.temporal_id) + 7) *
>+          kTemporalLayerRateFactor[frame_info.layers.size() - 1][i] /
>           (8 * framerate);
>       layer_info.size = avg_frame_size;
>       if (debt_bytes_ > 0) {
>@@ -201,10 +229,6 @@ int32_t FakeEncoder::Release() {
>   return 0;
> }
> 
>-int32_t FakeEncoder::SetChannelParameters(uint32_t packet_loss, int64_t rtt) {
>-  return 0;
>-}
>-
> int32_t FakeEncoder::SetRateAllocation(
>     const VideoBitrateAllocation& rate_allocation,
>     uint32_t framerate) {
>@@ -236,8 +260,10 @@ int32_t FakeEncoder::SetRateAllocation(
> }
> 
> const char* FakeEncoder::kImplementationName = "fake_encoder";
>-const char* FakeEncoder::ImplementationName() const {
>-  return kImplementationName;
>+VideoEncoder::EncoderInfo FakeEncoder::GetEncoderInfo() const {
>+  EncoderInfo info;
>+  info.implementation_name = kImplementationName;
>+  return info;
> }
> 
> int FakeEncoder::GetConfiguredInputFramerate() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.h
>index 67acebec69c94b7270f6fd28713ffef78df3895f..da38f11b456cfabaa47e4646d50ffcb36abeb39c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.h
>@@ -15,7 +15,6 @@
> #include <vector>
> 
> #include "api/video_codecs/video_encoder.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/sequenced_task_checker.h"
> #include "rtc_base/task_queue.h"
>@@ -41,11 +40,10 @@ class FakeEncoder : public VideoEncoder {
>   int32_t RegisterEncodeCompleteCallback(
>       EncodedImageCallback* callback) override;
>   int32_t Release() override;
>-  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>   int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
>                             uint32_t framerate) override;
>-  const char* ImplementationName() const override;
>   int GetConfiguredInputFramerate() const;
>+  EncoderInfo GetEncoderInfo() const override;
> 
>   static const char* kImplementationName;
> 
>@@ -71,7 +69,7 @@ class FakeEncoder : public VideoEncoder {
>                       SimulcastStream simulcast_streams[kMaxSimulcastStreams],
>                       int framerate);
> 
>-  FrameInfo last_frame_info_;
>+  FrameInfo last_frame_info_ RTC_GUARDED_BY(crit_sect_);
>   Clock* const clock_;
> 
>   VideoCodec config_ RTC_GUARDED_BY(crit_sect_);
>@@ -80,6 +78,7 @@ class FakeEncoder : public VideoEncoder {
>   int configured_input_framerate_ RTC_GUARDED_BY(crit_sect_);
>   int max_target_bitrate_kbps_ RTC_GUARDED_BY(crit_sect_);
>   bool pending_keyframe_ RTC_GUARDED_BY(crit_sect_);
>+  uint32_t counter_ RTC_GUARDED_BY(crit_sect_);
>   rtc::CriticalSection crit_sect_;
> 
>   uint8_t encoded_buffer_[100000];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_decoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_decoder.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..f15bb2149dea4f2ef6ec79e9bb358db238bbb2b0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_decoder.cc
>@@ -0,0 +1,77 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "test/fake_vp8_decoder.h"
>+
>+#include "api/video/i420_buffer.h"
>+#include "rtc_base/timeutils.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+// Read width and height from the payload of the frame if it is a key frame the
>+// same way as the real VP8 decoder.
>+// FakeEncoder writes width, height and frame type.
>+void ParseFakeVp8(const unsigned char* data, int* width, int* height) {
>+  bool key_frame = data[0] == 0;
>+  if (key_frame) {
>+    *width = ((data[7] << 8) + data[6]) & 0x3FFF;
>+    *height = ((data[9] << 8) + data[8]) & 0x3FFF;
>+  }
>+}
>+}  // namespace
>+
>+FakeVp8Decoder::FakeVp8Decoder() : callback_(nullptr), width_(0), height_(0) {}
>+
>+int32_t FakeVp8Decoder::InitDecode(const VideoCodec* config,
>+                                   int32_t number_of_cores) {
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int32_t FakeVp8Decoder::Decode(const EncodedImage& input,
>+                               bool missing_frames,
>+                               const CodecSpecificInfo* codec_specific_info,
>+                               int64_t render_time_ms) {
>+  constexpr size_t kMinPayLoadHeaderLength = 10;
>+  if (input._length < kMinPayLoadHeaderLength) {
>+    return WEBRTC_VIDEO_CODEC_ERROR;
>+  }
>+  ParseFakeVp8(input._buffer, &width_, &height_);
>+
>+  VideoFrame frame(I420Buffer::Create(width_, height_),
>+                   webrtc::kVideoRotation_0,
>+                   render_time_ms * rtc::kNumMicrosecsPerMillisec);
>+  frame.set_timestamp(input.Timestamp());
>+  frame.set_ntp_time_ms(input.ntp_time_ms_);
>+
>+  callback_->Decoded(frame, /*decode_time_ms=*/absl::nullopt,
>+                     /*qp=*/absl::nullopt);
>+
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int32_t FakeVp8Decoder::RegisterDecodeCompleteCallback(
>+    DecodedImageCallback* callback) {
>+  callback_ = callback;
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int32_t FakeVp8Decoder::Release() {
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+const char* FakeVp8Decoder::kImplementationName = "fake_vp8_decoder";
>+const char* FakeVp8Decoder::ImplementationName() const {
>+  return kImplementationName;
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_decoder.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..974af403d692040d6a56d546a23c05a64c5aba08
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_decoder.h
>@@ -0,0 +1,51 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef TEST_FAKE_VP8_DECODER_H_
>+#define TEST_FAKE_VP8_DECODER_H_
>+
>+#include "modules/video_coding/include/video_codec_interface.h"
>+#include "system_wrappers/include/clock.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+class FakeVp8Decoder : public VideoDecoder {
>+ public:
>+  FakeVp8Decoder();
>+  ~FakeVp8Decoder() override {}
>+
>+  int32_t InitDecode(const VideoCodec* config,
>+                     int32_t number_of_cores) override;
>+
>+  int32_t Decode(const EncodedImage& input,
>+                 bool missing_frames,
>+                 const CodecSpecificInfo* codec_specific_info,
>+                 int64_t render_time_ms) override;
>+
>+  int32_t RegisterDecodeCompleteCallback(
>+      DecodedImageCallback* callback) override;
>+
>+  int32_t Release() override;
>+
>+  const char* ImplementationName() const override;
>+
>+  static const char* kImplementationName;
>+
>+ private:
>+  DecodedImageCallback* callback_;
>+  int width_;
>+  int height_;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // TEST_FAKE_VP8_DECODER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder.cc
>index 4038837798315601ad812ecb63ed9aaec5bc4fac..9180bec04996b688583fea6567e1fa0478ca14d8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder.cc
>@@ -10,8 +10,8 @@
> 
> #include "test/fake_vp8_encoder.h"
> 
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
>+#include "api/video_codecs/create_vp8_temporal_layers.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/include/video_error_codes.h"
> #include "modules/video_coding/utility/simulcast_utility.h"
>@@ -20,6 +20,26 @@
> #include "rtc_base/random.h"
> #include "rtc_base/timeutils.h"
> 
>+namespace {
>+
>+// Write width and height to the payload the same way as the real encoder does.
>+// It requires that |payload| has a size of at least kMinPayLoadHeaderLength.
>+void WriteFakeVp8(unsigned char* payload,
>+                  int width,
>+                  int height,
>+                  bool key_frame) {
>+  payload[0] = key_frame ? 0 : 0x01;
>+
>+  if (key_frame) {
>+    payload[9] = (height & 0x3F00) >> 8;
>+    payload[8] = (height & 0x00FF);
>+
>+    payload[7] = (width & 0x3F00) >> 8;
>+    payload[6] = (width & 0x00FF);
>+  }
>+}
>+}  // namespace
>+
> namespace webrtc {
> 
> namespace test {
>@@ -63,18 +83,18 @@ void FakeVP8Encoder::SetupTemporalLayers(const VideoCodec& codec) {
> 
>   int num_streams = SimulcastUtility::NumberOfSimulcastStreams(codec);
>   for (int i = 0; i < num_streams; ++i) {
>-    TemporalLayersType type;
>+    Vp8TemporalLayersType type;
>     int num_temporal_layers =
>         SimulcastUtility::NumberOfTemporalLayers(codec, i);
>     if (SimulcastUtility::IsConferenceModeScreenshare(codec) && i == 0) {
>-      type = TemporalLayersType::kBitrateDynamic;
>+      type = Vp8TemporalLayersType::kBitrateDynamic;
>       // Legacy screenshare layers supports max 2 layers.
>       num_temporal_layers = std::max<int>(2, num_temporal_layers);
>     } else {
>-      type = TemporalLayersType::kFixedPattern;
>+      type = Vp8TemporalLayersType::kFixedPattern;
>     }
>     temporal_layers_.emplace_back(
>-        TemporalLayers::CreateTemporalLayers(type, num_temporal_layers));
>+        CreateVp8TemporalLayers(type, num_temporal_layers));
>   }
> }
> 
>@@ -85,7 +105,6 @@ void FakeVP8Encoder::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>                                            uint32_t timestamp) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
>   codec_specific->codecType = kVideoCodecVP8;
>-  codec_specific->codec_name = ImplementationName();
>   CodecSpecificInfoVP8* vp8Info = &(codec_specific->codecSpecific.VP8);
>   vp8Info->keyIdx = kNoKeyIdx;
>   vp8Info->nonReference = false;
>@@ -105,9 +124,20 @@ EncodedImageCallback::Result FakeVP8Encoder::OnEncodedImage(
>                         encoded_image._frameType, stream_idx,
>                         encoded_image.Timestamp());
> 
>+  // Write width and height to the payload the same way as the real encoder
>+  // does.
>+  WriteFakeVp8(encoded_image._buffer, encoded_image._encodedWidth,
>+               encoded_image._encodedHeight,
>+               encoded_image._frameType == kVideoFrameKey);
>   return callback_->OnEncodedImage(encoded_image, &overrided_specific_info,
>                                    fragments);
> }
> 
>+VideoEncoder::EncoderInfo FakeVP8Encoder::GetEncoderInfo() const {
>+  EncoderInfo info;
>+  info.implementation_name = "FakeVp8Encoder";
>+  return info;
>+}
>+
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder.h
>index 7518e3a2db5d358405ffc16c86a73d02571c8ce4..a4d63a9ae30a6701ca0bb2a6eb39c495f07b11af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder.h
>@@ -14,7 +14,7 @@
> #include <memory>
> #include <vector>
> 
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "test/fake_encoder.h"
> 
> #include "rtc_base/criticalsection.h"
>@@ -37,12 +37,12 @@ class FakeVP8Encoder : public FakeEncoder, public EncodedImageCallback {
> 
>   int32_t Release() override;
> 
>-  const char* ImplementationName() const override { return "FakeVp8Encoder"; }
>-
>   Result OnEncodedImage(const EncodedImage& encodedImage,
>                         const CodecSpecificInfo* codecSpecificInfo,
>                         const RTPFragmentationHeader* fragments) override;
> 
>+  EncoderInfo GetEncoderInfo() const override;
>+
>  private:
>   void SetupTemporalLayers(const VideoCodec& codec);
>   void PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>@@ -54,7 +54,7 @@ class FakeVP8Encoder : public FakeEncoder, public EncodedImageCallback {
>   rtc::SequencedTaskChecker sequence_checker_;
>   EncodedImageCallback* callback_ RTC_GUARDED_BY(sequence_checker_);
> 
>-  std::vector<std::unique_ptr<TemporalLayers>> temporal_layers_
>+  std::vector<std::unique_ptr<Vp8TemporalLayers>> temporal_layers_
>       RTC_GUARDED_BY(sequence_checker_);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder_unittest.cc
>index c79ba0c5515c5fd65404bda058cee0c82daeb438..4e576fe9c9c25364f1489c9f199885faa289915d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_vp8_encoder_unittest.cc
>@@ -14,11 +14,11 @@
> #include "absl/memory/memory.h"
> #include "api/test/create_simulcast_test_fixture.h"
> #include "api/test/simulcast_test_fixture.h"
>+#include "api/test/video/function_video_decoder_factory.h"
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "modules/video_coding/utility/simulcast_test_fixture_impl.h"
>-#include "test/fake_decoder.h"
>+#include "test/fake_vp8_decoder.h"
> #include "test/fake_vp8_encoder.h"
>-#include "test/function_video_decoder_factory.h"
>-#include "test/function_video_encoder_factory.h"
> 
> namespace webrtc {
> namespace test {
>@@ -32,7 +32,7 @@ std::unique_ptr<SimulcastTestFixture> CreateSpecificSimulcastTestFixture() {
>       });
>   std::unique_ptr<VideoDecoderFactory> decoder_factory =
>       absl::make_unique<FunctionVideoDecoderFactory>(
>-          []() { return absl::make_unique<FakeDecoder>(); });
>+          []() { return absl::make_unique<FakeVp8Decoder>(); });
>   return CreateSimulcastTestFixture(std::move(encoder_factory),
>                                     std::move(decoder_factory),
>                                     SdpVideoFormat("VP8"));
>@@ -99,5 +99,10 @@ TEST(TestFakeVp8Codec, TestSpatioTemporalLayers333PatternEncoder) {
>   fixture->TestSpatioTemporalLayers333PatternEncoder();
> }
> 
>+TEST(TestFakeVp8Codec, TestDecodeWidthHeightSet) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestDecodeWidthHeightSet();
>+}
>+
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.cc
>index acd63f931220b58836b1f5cbb03e6dcf4da6b060..f6fd285c1a13eaeedaf0853e3080dd22af55515a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.cc
>@@ -390,11 +390,11 @@ class ScrollingImageFrameGenerator : public FrameGenerator {
>                    (pixels_scrolled_x / 2);
> 
>     current_frame_ = webrtc::VideoFrame(
>-        new rtc::RefCountedObject<webrtc::WrappedI420Buffer>(
>-            target_width_, target_height_, &i420_buffer->DataY()[offset_y],
>-            i420_buffer->StrideY(), &i420_buffer->DataU()[offset_u],
>-            i420_buffer->StrideU(), &i420_buffer->DataV()[offset_v],
>-            i420_buffer->StrideV(), KeepRefUntilDone(i420_buffer)),
>+        WrapI420Buffer(target_width_, target_height_,
>+                       &i420_buffer->DataY()[offset_y], i420_buffer->StrideY(),
>+                       &i420_buffer->DataU()[offset_u], i420_buffer->StrideU(),
>+                       &i420_buffer->DataV()[offset_v], i420_buffer->StrideV(),
>+                       KeepRefUntilDone(i420_buffer)),
>         kVideoRotation_0, 0);
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_decoder_factory.h
>deleted file mode 100644
>index 984f8534be00bb5901e9ab799423018f3039be01..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_decoder_factory.h
>+++ /dev/null
>@@ -1,56 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef TEST_FUNCTION_VIDEO_DECODER_FACTORY_H_
>-#define TEST_FUNCTION_VIDEO_DECODER_FACTORY_H_
>-
>-#include <functional>
>-#include <memory>
>-#include <utility>
>-#include <vector>
>-
>-#include "api/video_codecs/sdp_video_format.h"
>-#include "api/video_codecs/video_decoder_factory.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-// A decoder factory producing decoders by calling a supplied create function.
>-class FunctionVideoDecoderFactory final : public VideoDecoderFactory {
>- public:
>-  explicit FunctionVideoDecoderFactory(
>-      std::function<std::unique_ptr<VideoDecoder>()> create)
>-      : create_([create](const SdpVideoFormat&) { return create(); }) {}
>-  explicit FunctionVideoDecoderFactory(
>-      std::function<std::unique_ptr<VideoDecoder>(const SdpVideoFormat&)>
>-          create)
>-      : create_(std::move(create)) {}
>-
>-  // Unused by tests.
>-  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
>-    RTC_NOTREACHED();
>-    return {};
>-  }
>-
>-  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
>-      const SdpVideoFormat& format) override {
>-    return create_(format);
>-  }
>-
>- private:
>-  const std::function<std::unique_ptr<VideoDecoder>(const SdpVideoFormat&)>
>-      create_;
>-};
>-
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // TEST_FUNCTION_VIDEO_DECODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_encoder_factory.h
>deleted file mode 100644
>index 567397f6c0c697994c60b5149b1d6dc37aa40d04..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_encoder_factory.h
>+++ /dev/null
>@@ -1,65 +0,0 @@
>-/*
>- *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef TEST_FUNCTION_VIDEO_ENCODER_FACTORY_H_
>-#define TEST_FUNCTION_VIDEO_ENCODER_FACTORY_H_
>-
>-#include <functional>
>-#include <memory>
>-#include <utility>
>-#include <vector>
>-
>-#include "api/video_codecs/sdp_video_format.h"
>-#include "api/video_codecs/video_encoder_factory.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-// An encoder factory producing encoders by calling a supplied create
>-// function.
>-class FunctionVideoEncoderFactory final : public VideoEncoderFactory {
>- public:
>-  explicit FunctionVideoEncoderFactory(
>-      std::function<std::unique_ptr<VideoEncoder>()> create)
>-      : create_([create](const SdpVideoFormat&) { return create(); }) {}
>-  explicit FunctionVideoEncoderFactory(
>-      std::function<std::unique_ptr<VideoEncoder>(const SdpVideoFormat&)>
>-          create)
>-      : create_(std::move(create)) {}
>-
>-  // Unused by tests.
>-  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
>-    RTC_NOTREACHED();
>-    return {};
>-  }
>-
>-  CodecInfo QueryVideoEncoder(
>-      const SdpVideoFormat& /* format */) const override {
>-    CodecInfo codec_info;
>-    codec_info.is_hardware_accelerated = false;
>-    codec_info.has_internal_source = false;
>-    return codec_info;
>-  }
>-
>-  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>-      const SdpVideoFormat& format) override {
>-    return create_(format);
>-  }
>-
>- private:
>-  const std::function<std::unique_ptr<VideoEncoder>(const SdpVideoFormat&)>
>-      create_;
>-};
>-
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // TEST_FUNCTION_VIDEO_ENCODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/BUILD.gn
>index ccc15a61005e9d3ae862f00c62d2d5a689f3a497..43de5f998cef6108000418d0e7fc5c9504daf6c3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/BUILD.gn
>@@ -6,9 +6,9 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>-import("../../webrtc.gni")
> import("//build/config/features.gni")
> import("//testing/libfuzzer/fuzzer_test.gni")
>+import("../../webrtc.gni")
> 
> rtc_static_library("webrtc_fuzzer_main") {
>   sources = [
>@@ -125,7 +125,6 @@ webrtc_fuzzer_test("forward_error_correction_fuzzer") {
>     "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../rtc_base:rtc_base_approved",
>   ]
>-  libfuzzer_options = [ "max_len=5000" ]
> }
> 
> webrtc_fuzzer_test("flexfec_header_reader_fuzzer") {
>@@ -148,7 +147,6 @@ webrtc_fuzzer_test("flexfec_sender_fuzzer") {
>     "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../system_wrappers",
>   ]
>-  libfuzzer_options = [ "max_len=200" ]
> }
> 
> webrtc_fuzzer_test("ulpfec_header_reader_fuzzer") {
>@@ -186,7 +184,6 @@ webrtc_fuzzer_test("ulpfec_receiver_fuzzer") {
>     "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../rtc_base:rtc_base_approved",
>   ]
>-  libfuzzer_options = [ "max_len=2000" ]
> }
> 
> webrtc_fuzzer_test("flexfec_receiver_fuzzer") {
>@@ -198,7 +195,6 @@ webrtc_fuzzer_test("flexfec_receiver_fuzzer") {
>     "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../rtc_base:rtc_base_approved",
>   ]
>-  libfuzzer_options = [ "max_len=2000" ]
> }
> 
> webrtc_fuzzer_test("packet_buffer_fuzzer") {
>@@ -209,7 +205,6 @@ webrtc_fuzzer_test("packet_buffer_fuzzer") {
>     "../../modules/video_coding/",
>     "../../system_wrappers",
>   ]
>-  libfuzzer_options = [ "max_len=200000" ]
> }
> 
> webrtc_fuzzer_test("rtcp_receiver_fuzzer") {
>@@ -283,8 +278,6 @@ webrtc_fuzzer_test("audio_decoder_ilbc_fuzzer") {
>     ":audio_decoder_fuzzer",
>     "../../modules/audio_coding:ilbc",
>   ]
>-
>-  libfuzzer_options = [ "max_len=10000" ]
> }
> 
> webrtc_fuzzer_test("audio_decoder_isac_fuzzer") {
>@@ -295,8 +288,6 @@ webrtc_fuzzer_test("audio_decoder_isac_fuzzer") {
>     ":audio_decoder_fuzzer",
>     "../../modules/audio_coding:isac",
>   ]
>-
>-  libfuzzer_options = [ "max_len=20000" ]
> }
> 
> webrtc_fuzzer_test("audio_decoder_isac_incoming_packet_fuzzer") {
>@@ -307,8 +298,6 @@ webrtc_fuzzer_test("audio_decoder_isac_incoming_packet_fuzzer") {
>     ":audio_decoder_fuzzer",
>     "../../modules/audio_coding:isac",
>   ]
>-
>-  libfuzzer_options = [ "max_len=20000" ]
> }
> 
> webrtc_fuzzer_test("audio_decoder_isacfix_fuzzer") {
>@@ -319,8 +308,6 @@ webrtc_fuzzer_test("audio_decoder_isacfix_fuzzer") {
>     ":audio_decoder_fuzzer",
>     "../../modules/audio_coding:isac_fix",
>   ]
>-
>-  libfuzzer_options = [ "max_len=20000" ]
> }
> 
> webrtc_fuzzer_test("audio_decoder_opus_fuzzer") {
>@@ -377,10 +364,7 @@ webrtc_fuzzer_test("neteq_rtp_fuzzer") {
>     "../../modules/audio_coding:pcm16b",
>     "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../rtc_base:rtc_base_approved",
>-    "../../rtc_base:rtc_base_tests_utils",
>   ]
>-
>-  libfuzzer_options = [ "max_len=100000" ]
> }
> 
> webrtc_fuzzer_test("neteq_signal_fuzzer") {
>@@ -394,10 +378,7 @@ webrtc_fuzzer_test("neteq_signal_fuzzer") {
>     "../../modules/audio_coding:neteq_tools_minimal",
>     "../../modules/audio_coding:pcm16b",
>     "../../rtc_base:rtc_base_approved",
>-    "../../rtc_base:rtc_base_tests_utils",
>   ]
>-
>-  libfuzzer_options = [ "max_len=100000" ]
> }
> 
> webrtc_fuzzer_test("residual_echo_detector_fuzzer") {
>@@ -420,7 +401,6 @@ webrtc_fuzzer_test("sdp_parser_fuzzer") {
>     "../../pc:libjingle_peerconnection",
>   ]
>   seed_corpus = "corpora/sdp-corpus"
>-  libfuzzer_options = [ "max_len=16384" ]
> }
> 
> webrtc_fuzzer_test("stun_parser_fuzzer") {
>@@ -487,6 +467,7 @@ rtc_static_library("audio_processing_fuzzer_helper") {
>     ":fuzz_data_helper",
>     "../../api/audio:audio_frame_api",
>     "../../modules/audio_processing",
>+    "../../modules/audio_processing:api",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "//third_party/abseil-cpp/absl/types:optional",
>@@ -501,10 +482,13 @@ webrtc_fuzzer_test("audio_processing_fuzzer") {
>     ":audio_processing_fuzzer_helper",
>     "../../api/audio:aec3_factory",
>     "../../modules/audio_processing",
>+    "../../modules/audio_processing:api",
>     "../../modules/audio_processing/aec3",
>-    "../../modules/audio_processing/aec_dump:mock_aec_dump",
>+    "../../modules/audio_processing/aec_dump",
>+    "../../modules/audio_processing/aec_dump:aec_dump_impl",
>     "../../rtc_base:ptr_util",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:rtc_task_queue",
>     "../../rtc_base:safe_minmax",
>     "../../system_wrappers:field_trial",
>     "//third_party/abseil-cpp/absl/memory",
>@@ -519,13 +503,13 @@ webrtc_fuzzer_test("agc_fuzzer") {
>   deps = [
>     ":fuzz_data_helper",
>     "../../modules/audio_processing",
>+    "../../modules/audio_processing:api",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:safe_minmax",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
> 
>   seed_corpus = "corpora/agc-corpus"
>-  libfuzzer_options = [ "max_len=200000" ]
> }
> 
> webrtc_fuzzer_test("comfort_noise_decoder_fuzzer") {
>@@ -534,11 +518,9 @@ webrtc_fuzzer_test("comfort_noise_decoder_fuzzer") {
>   ]
>   deps = [
>     "../../api:array_view",
>-    "../../modules/audio_coding:cng",
>+    "../../modules/audio_coding:webrtc_cng",
>     "../../rtc_base:rtc_base_approved",
>   ]
>-
>-  libfuzzer_options = [ "max_len=5000" ]
> }
> 
> webrtc_fuzzer_test("rtp_frame_reference_finder_fuzzer") {
>@@ -551,7 +533,6 @@ webrtc_fuzzer_test("rtp_frame_reference_finder_fuzzer") {
>     "../../system_wrappers",
>     "//third_party/abseil-cpp/absl/memory",
>   ]
>-  libfuzzer_options = [ "max_len=20000" ]
> }
> 
> webrtc_fuzzer_test("frame_buffer2_fuzzer") {
>@@ -562,5 +543,4 @@ webrtc_fuzzer_test("frame_buffer2_fuzzer") {
>     "../../modules/video_coding/",
>     "../../system_wrappers:system_wrappers",
>   ]
>-  libfuzzer_options = [ "max_len=10000" ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/agc_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/agc_fuzzer.cc
>index 5bd921e74dae3b99859ccf52214cc11faba09f40..f2c90480c835eadee3db2289222a5ccadeb99e87 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/agc_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/agc_fuzzer.cc
>@@ -109,6 +109,9 @@ void FuzzGainController(test::FuzzDataHelper* fuzz_data, GainControlImpl* gci) {
> }  // namespace
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 200000) {
>+    return;
>+  }
>   test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
>   rtc::CriticalSection crit_capture;
>   rtc::CriticalSection crit_render;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc
>index a68725d6aacd162e2a2fb40f69a1110fc74027b3..8548645c63f9b767a5952b7601cb2c0f1c3dabd5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc
>@@ -13,6 +13,9 @@
> 
> namespace webrtc {
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 10000) {
>+    return;
>+  }
>   AudioDecoderIlbcImpl dec;
>   static const int kSampleRateHz = 8000;
>   static const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc
>index e79996e05b28a34bf11d1b89697406b46d120e70..b579083956c6bce064be4f98b75e95f003b67e2c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isac_fuzzer.cc
>@@ -13,6 +13,9 @@
> 
> namespace webrtc {
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 20000) {
>+    return;
>+  }
>   const int sample_rate_hz = size % 2 == 0 ? 16000 : 32000;     // 16 or 32 kHz.
>   static const size_t kAllocatedOuputSizeSamples = 32000 / 10;  // 100 ms.
>   int16_t output[kAllocatedOuputSizeSamples];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isac_incoming_packet_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isac_incoming_packet_fuzzer.cc
>index 5645142a034fa2326ca153e2393194bec9462684..9bd6234fa1495e7175fc01221e36eae24a6303aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isac_incoming_packet_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isac_incoming_packet_fuzzer.cc
>@@ -13,6 +13,9 @@
> 
> namespace webrtc {
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 20000) {
>+    return;
>+  }
>   AudioDecoderIsacFloatImpl dec(16000);
>   FuzzAudioDecoderIncomingPacket(data, size, &dec);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc
>index 444395b17df20c5c9d90c9208e1d0cf205255354..6477dc361bddd777f60b830baf21050c3df07a1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_isacfix_fuzzer.cc
>@@ -13,6 +13,9 @@
> 
> namespace webrtc {
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 20000) {
>+    return;
>+  }
>   static const int kSampleRateHz = 16000;
>   static const size_t kAllocatedOuputSizeSamples = 16000 / 10;  // 100 ms.
>   int16_t output[kAllocatedOuputSizeSamples];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_configs_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_configs_fuzzer.cc
>index 8117c52aaf2b315f7f066b4b192919891a41d333..dd0013684edbcbcf34081faf38ae21421bfb2378 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_configs_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_configs_fuzzer.cc
>@@ -13,10 +13,11 @@
> 
> #include "absl/memory/memory.h"
> #include "api/audio/echo_canceller3_factory.h"
>-#include "modules/audio_processing/aec_dump/mock_aec_dump.h"
>+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/numerics/safe_minmax.h"
>+#include "rtc_base/task_queue.h"
> #include "system_wrappers/include/field_trial.h"
> #include "test/fuzzers/audio_processing_fuzzer_helper.h"
> #include "test/fuzzers/fuzz_data_helper.h"
>@@ -25,36 +26,45 @@ namespace webrtc {
> namespace {
> 
> const std::string kFieldTrialNames[] = {
>-    "WebRTC-Aec3TransparentModeKillSwitch",
>-    "WebRTC-Aec3StationaryRenderImprovementsKillSwitch",
>-    "WebRTC-Aec3EnforceDelayAfterRealignmentKillSwitch",
>-    "WebRTC-Aec3UseShortDelayEstimatorWindow",
>-    "WebRTC-Aec3ReverbBasedOnRenderKillSwitch",
>-    "WebRTC-Aec3ReverbModellingKillSwitch",
>-    "WebRTC-Aec3FilterAnalyzerPreprocessorKillSwitch",
>-    "WebRTC-Aec3TransparencyImprovementsKillSwitch",
>-    "WebRTC-Aec3SoftTransparentModeKillSwitch",
>-    "WebRTC-Aec3OverrideEchoPathGainKillSwitch",
>-    "WebRTC-Aec3ZeroExternalDelayHeadroomKillSwitch",
>-    "WebRTC-Aec3DownSamplingFactor8KillSwitch",
>+    "WebRTC-Aec3AdaptErleOnLowRenderKillSwitch",
>+    "WebRTC-Aec3AgcGainChangeResponseKillSwitch",
>+    "WebRTC-Aec3BoundedNearendKillSwitch",
>+    "WebRTC-Aec3EarlyShadowFilterJumpstartKillSwitch",
>+    "WebRTC-Aec3EnableAdaptiveEchoReverbEstimation",
>+    "WebRTC-Aec3EnableLegacyDominantNearend",
>+    "WebRTC-Aec3EnableUnityInitialRampupGain",
>+    "WebRTC-Aec3EnableUnityNonZeroRampupGain",
>     "WebRTC-Aec3EnforceSkewHysteresis1",
>     "WebRTC-Aec3EnforceSkewHysteresis2",
>-    "WebRTC-Aec3NewSuppressionKillSwitch",
>-    "WebRTC-Aec3LinearModeWithDivergedFilterKillSwitch",
>+    "WebRTC-Aec3FilterAnalyzerPreprocessorKillSwitch",
>     "WebRTC-Aec3MisadjustmentEstimatorKillSwitch",
>+    "WebRTC-Aec3NewFilterParamsKillSwitch",
>+    "WebRTC-Aec3NewRenderBufferingKillSwitch",
>+    "WebRTC-Aec3OverrideEchoPathGainKillSwitch",
>     "WebRTC-Aec3RapidAgcGainRecoveryKillSwitch",
>-    "WebRTC-Aec3SlowFilterAdaptationKillSwitch",
>-    "WebRTC-Aec3SmoothUpdatesTailFreqRespKillSwitch",
>-    "WebRTC-Aec3SuppressorNearendAveragingKillSwitch",
>-    "WebRTC-Aec3AgcGainChangeResponseKillSwitch",
>+    "WebRTC-Aec3ResetErleAtGainChangesKillSwitch",
>+    "WebRTC-Aec3ReverbBasedOnRenderKillSwitch",
>+    "WebRTC-Aec3ReverbModellingKillSwitch",
>+    "WebRTC-Aec3ShadowFilterBoostedJumpstartKillSwitch",
>     "WebRTC-Aec3ShadowFilterJumpstartKillSwitch",
>-    "WebRTC-Aec3EarlyLinearFilterUsageKillSwitch",
>-    "WebRTC-Aec3ShortInitialStateKillSwitch",
>+    "WebRTC-Aec3ShortReverbKillSwitch",
>+    "WebRTC-Aec3SmoothSignalTransitionsKillSwitch",
>+    "WebRTC-Aec3SmoothUpdatesTailFreqRespKillSwitch",
>+    "WebRTC-Aec3SoftTransparentModeKillSwitch",
>     "WebRTC-Aec3StandardNonlinearReverbModelKillSwitch",
>-    "WebRTC-Aec3EnableAdaptiveEchoReverbEstimation"};
>+    "WebRTC-Aec3StrictDivergenceCheckKillSwitch",
>+    "WebRTC-Aec3UseLegacyNormalSuppressorTuning",
>+    "WebRTC-Aec3UseOffsetBlocks",
>+    "WebRTC-Aec3UseShortDelayEstimatorWindow",
>+    "WebRTC-Aec3UseStationarityPropertiesKillSwitch",
>+    "WebRTC-Aec3UtilizeShadowFilterOutputKillSwitch",
>+    "WebRTC-Aec3ZeroExternalDelayHeadroomKillSwitch",
>+    "WebRTC-Aec3EarlyDelayDetectionKillSwitch",
>+};
> 
> std::unique_ptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
>-                                           std::string* field_trial_string) {
>+                                           std::string* field_trial_string,
>+                                           rtc::TaskQueue* worker_queue) {
>   // Parse boolean values for optionally enabling different
>   // configurable public components of APM.
>   bool exp_agc = fuzz_data->ReadOrDefaultValue(true);
>@@ -79,12 +89,13 @@ std::unique_ptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
> 
>   // Read an int8 value, but don't let it be too large or small.
>   const float gain_controller2_gain_db =
>-      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(0), -50, 50);
>+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(0), -40, 40);
> 
>   constexpr size_t kNumFieldTrials = arraysize(kFieldTrialNames);
>   // Verify that the read data type has enough bits to fuzz the field trials.
>-  using FieldTrialBitmaskType = uint32_t;
>-  RTC_DCHECK_LE(kNumFieldTrials, sizeof(FieldTrialBitmaskType) * 8);
>+  using FieldTrialBitmaskType = uint64_t;
>+  static_assert(kNumFieldTrials <= sizeof(FieldTrialBitmaskType) * 8,
>+                "FieldTrialBitmaskType is not large enough.");
>   std::bitset<kNumFieldTrials> field_trial_bitmask(
>       fuzz_data->ReadOrDefaultValue<FieldTrialBitmaskType>(0));
>   for (size_t i = 0; i < kNumFieldTrials; ++i) {
>@@ -129,8 +140,9 @@ std::unique_ptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
>           .SetEchoControlFactory(std::move(echo_control_factory))
>           .Create(config));
> 
>-  apm->AttachAecDump(
>-      absl::make_unique<testing::NiceMock<webrtc::test::MockAecDump>>());
>+#ifdef WEBRTC_LINUX
>+  apm->AttachAecDump(AecDumpFactory::Create("/dev/null", -1, worker_queue));
>+#endif
> 
>   webrtc::AudioProcessing::Config apm_config;
>   apm_config.echo_canceller.enabled = use_aec || use_aecm;
>@@ -139,7 +151,7 @@ std::unique_ptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
>   apm_config.high_pass_filter.enabled = hpf;
>   apm_config.gain_controller2.enabled = use_agc2_limiter;
> 
>-  apm_config.gain_controller2.fixed_gain_db = gain_controller2_gain_db;
>+  apm_config.gain_controller2.fixed_digital.gain_db = gain_controller2_gain_db;
> 
>   apm->ApplyConfig(apm_config);
> 
>@@ -158,7 +170,10 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
>   // This string must be in scope during execution, according to documentation
>   // for field_trial.h. Hence it's created here and not in CreateApm.
>   std::string field_trial_string = "";
>-  auto apm = CreateApm(&fuzz_data, &field_trial_string);
>+
>+  std::unique_ptr<rtc::TaskQueue> worker_queue(
>+      new rtc::TaskQueue("rtc-low-prio", rtc::TaskQueue::Priority::LOW));
>+  auto apm = CreateApm(&fuzz_data, &field_trial_string, worker_queue.get());
> 
>   if (apm) {
>     FuzzAudioProcessing(&fuzz_data, std::move(apm));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_helper.cc
>index 073ab17bd931c4bbec133d0e90ad5c1b3cbaf4c3..dec4e1f05e4d422a4dfef86a680fa430be076f75 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_helper.cc
>@@ -130,10 +130,8 @@ void FuzzAudioProcessing(test::FuzzDataHelper* fuzz_data,
>       }
>     }
> 
>-    // Make calls to stats gathering functions to cover these
>-    // codeways.
>-    static_cast<void>(apm->GetStatistics());
>-    static_cast<void>(apm->GetStatistics(true));
>+    // Cover stats gathering code paths.
>+    static_cast<void>(apm->GetStatistics(true /*has_remote_tracks*/));
>     static_cast<void>(apm->UpdateHistogramsOnCallEnd());
> 
>     RTC_DCHECK_NE(apm_return_code, AudioProcessing::kBadDataLengthError);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/comfort_noise_decoder_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/comfort_noise_decoder_fuzzer.cc
>index 50166d7df778250041e52dc577e174b4a48b94ce..7f44af99fb53035d43c725fba93779800bf92fcd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/comfort_noise_decoder_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/comfort_noise_decoder_fuzzer.cc
>@@ -50,6 +50,9 @@ void FuzzOneInputTest(rtc::ArrayView<const uint8_t> data) {
> }  // namespace test
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 5000) {
>+    return;
>+  }
>   test::FuzzOneInputTest(rtc::ArrayView<const uint8_t>(data, size));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_receiver_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_receiver_fuzzer.cc
>index d96a328fa2c58bfef0a5ea54c431cd202a115d4f..c5034bb933ee83b46a18fc40a1a8f69b953b9867 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_receiver_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_receiver_fuzzer.cc
>@@ -25,7 +25,7 @@ class DummyCallback : public RecoveredPacketReceiver {
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>   constexpr size_t kMinDataNeeded = 12;
>-  if (size < kMinDataNeeded) {
>+  if (size < kMinDataNeeded || size > 2000) {
>     return;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_sender_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_sender_fuzzer.cc
>index 8e79f95cf3fee323ac502e3dd2cd0231a1aaa0d4..4882f7df51a5c6bfff3824dc0004b70b351d76c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_sender_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_sender_fuzzer.cc
>@@ -31,10 +31,9 @@ const std::vector<RtpExtensionSize> kNoRtpHeaderExtensionSizes;
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>   size_t i = 0;
>-  if (size < 5) {
>+  if (size < 5 || size > 200) {
>     return;
>   }
>-
>   SimulatedClock clock(1 + data[i++]);
>   FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/forward_error_correction_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/forward_error_correction_fuzzer.cc
>index 9d5b872434ad3163c65980924cd9c8873abc7e2e..2eb357b74a81426bd3c5f26bd849bc8b51575c07 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/forward_error_correction_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/forward_error_correction_fuzzer.cc
>@@ -26,6 +26,9 @@ constexpr size_t kMaxPacketsInBuffer = 48;
> }  // namespace
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 5000) {
>+    return;
>+  }
>   // Object under test.
>   std::unique_ptr<ForwardErrorCorrection> fec =
>       ForwardErrorCorrection::CreateFlexfec(kFecSsrc, kMediaSsrc);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/frame_buffer2_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/frame_buffer2_fuzzer.cc
>index 2d5830979c124e409c7f4aa19a23ef15719053ea..a5591041ff69b091a31a35dd35be32fa4c04d988 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/frame_buffer2_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/frame_buffer2_fuzzer.cc
>@@ -63,6 +63,9 @@ class FuzzyFrameObject : public video_coding::EncodedFrame {
> }  // namespace
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 10000) {
>+    return;
>+  }
>   DataReader reader(data, size);
>   Clock* clock = Clock::GetRealTimeClock();
>   VCMJitterEstimator jitter_estimator(clock, 0, 0);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/mdns_parser_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/mdns_parser_fuzzer.cc
>index c3f765ef787b482b098c0e881982345c3a9da7e8..294f683226c07603f8e3fb887505c976e7b4fc86 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/mdns_parser_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/mdns_parser_fuzzer.cc
>@@ -19,7 +19,7 @@ namespace webrtc {
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>   MessageBufferReader buf(reinterpret_cast<const char*>(data), size);
>-  auto mdns_msg = absl::make_unique<MDnsMessage>();
>+  auto mdns_msg = absl::make_unique<MdnsMessage>();
>   mdns_msg->Read(&buf);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_rtp_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_rtp_fuzzer.cc
>index 8aa6be59dd0e1c8d74cf9bc1d8735103e03b2798..94dbef39ebe5edeec46a3c2e5b0c1460946ad854 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_rtp_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_rtp_fuzzer.cc
>@@ -146,6 +146,9 @@ void FuzzOneInputTest(const uint8_t* data, size_t size) {
> }  // namespace test
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 100000) {
>+    return;
>+  }
>   test::FuzzOneInputTest(data, size);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_signal_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_signal_fuzzer.cc
>index 16e776072bc2f1cc62e6c1149091bf74546ad896..25302c31a8857a10ac0228584384f9368adb99c5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_signal_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_signal_fuzzer.cc
>@@ -140,8 +140,9 @@ class FuzzSignalInput : public NetEqInput {
> }  // namespace
> 
> void FuzzOneInputTest(const uint8_t* data, size_t size) {
>-  if (size < 1)
>+  if (size < 1 || size > 90000) {
>     return;
>+  }
> 
>   FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/packet_buffer_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/packet_buffer_fuzzer.cc
>index d226764cb25d82bff85fde48de87895b9750bead..56d1557072f47f52516f095f7e8d2b06efe1b4ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/packet_buffer_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/packet_buffer_fuzzer.cc
>@@ -21,6 +21,9 @@ class NullCallback : public video_coding::OnReceivedFrameCallback {
> }  // namespace
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 200000) {
>+    return;
>+  }
>   VCMPacket packet;
>   NullCallback callback;
>   SimulatedClock clock(0);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
>index cd763d0f605880416302b50a2ba3224dd21524e9..7f8cabafd68ea48beb9a4b87e9570cc4e303def3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
>@@ -15,6 +15,8 @@
> namespace webrtc {
> namespace {
> 
>+constexpr int kRtcpIntervalMs = 1000;
>+
> class NullModuleRtpRtcp : public RTCPReceiver::ModuleRtpRtcp {
>  public:
>   void SetTmmbn(std::vector<rtcp::TmmbItem>) override {}
>@@ -30,7 +32,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
>   SimulatedClock clock(1234);
> 
>   RTCPReceiver receiver(&clock, false, nullptr, nullptr, nullptr, nullptr,
>-                        nullptr, &rtp_rtcp_module);
>+                        nullptr, kRtcpIntervalMs, &rtp_rtcp_module);
> 
>   receiver.IncomingPacket(data, size);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
>index fd98892a0480fd9cf0eb40efbe8a488174d21c0d..57d7a9623a07a0d12887ee7618433e2a94e1787a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
>@@ -101,6 +101,9 @@ class FuzzyPacketBuffer : public video_coding::PacketBuffer {
> }  // namespace
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 20000) {
>+    return;
>+  }
>   DataReader reader(data, size);
>   rtc::scoped_refptr<FuzzyPacketBuffer> pb(new FuzzyPacketBuffer(&reader));
>   NullCallback cb;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packet_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packet_fuzzer.cc
>index 8bf8e74aba94cf7345a5dff1ec160ba59ebcc4fc..f774c0cb8b0184343601a8f2ffbc6230902794d0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packet_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packet_fuzzer.cc
>@@ -121,6 +121,11 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
>         packet.GetExtension<RtpGenericFrameDescriptorExtension>(&descriptor);
>         break;
>       }
>+      case kRtpExtensionColorSpace: {
>+        ColorSpace color_space;
>+        packet.GetExtension<ColorSpaceExtension>(&color_space);
>+        break;
>+      }
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/sdp_parser_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/sdp_parser_fuzzer.cc
>index e47156c571e5cd901530c0db6d8d87f510a78714..763dbc594acb74125bc328ecce881ab9afe6e368 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/sdp_parser_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/sdp_parser_fuzzer.cc
>@@ -15,6 +15,9 @@
> 
> namespace webrtc {
> void FuzzOneInput(const uint8_t* data, size_t size) {
>+  if (size > 16384) {
>+    return;
>+  }
>   std::string message(reinterpret_cast<const char*>(data), size);
>   webrtc::SdpParseError error;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_receiver_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_receiver_fuzzer.cc
>index 6b74f3ac95d4d3ff3ad29074a12f67410c174298..7cb0fc61b5d2f20bcd5c42b709d1c8c7bd5d2c1a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_receiver_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_receiver_fuzzer.cc
>@@ -25,7 +25,7 @@ class DummyCallback : public RecoveredPacketReceiver {
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>   constexpr size_t kMinDataNeeded = 12;
>-  if (size < kMinDataNeeded) {
>+  if (size < kMinDataNeeded || size > 2000) {
>     return;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/coverage_util_ios.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/coverage_util_ios.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..a17b69dca8011b1d216e3cbb8dde86d8779fabbf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/coverage_util_ios.h
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef TEST_IOS_COVERAGE_UTIL_IOS_H_
>+#define TEST_IOS_COVERAGE_UTIL_IOS_H_
>+
>+namespace rtc {
>+namespace test {
>+
>+// In debug builds, if IOS_ENABLE_COVERAGE is defined, sets the filename of the
>+// coverage file. Otherwise, it does nothing.
>+void ConfigureCoverageReportPath();
>+
>+}  // namespace test
>+}  // namespace rtc
>+
>+#endif  // TEST_IOS_COVERAGE_UTIL_IOS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/coverage_util_ios.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/coverage_util_ios.mm
>new file mode 100644
>index 0000000000000000000000000000000000000000..c21a16def24746a8ffd0fd22d9ac72b2d2484ae0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/coverage_util_ios.mm
>@@ -0,0 +1,42 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+
>+#ifdef WEBRTC_IOS_ENABLE_COVERAGE
>+extern "C" void __llvm_profile_set_filename(const char* name);
>+#endif
>+
>+namespace rtc {
>+namespace test {
>+
>+void ConfigureCoverageReportPath() {
>+#ifdef WEBRTC_IOS_ENABLE_COVERAGE
>+  static dispatch_once_t once_token;
>+  dispatch_once(&once_token, ^{
>+    // Writes the profraw file to the Documents directory, where the app has
>+    // write rights.
>+    NSArray* paths =
>+        NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
>+    NSString* documents_directory = [paths firstObject];
>+    NSString* file_name = [documents_directory stringByAppendingPathComponent:@"coverage.profraw"];
>+
>+    // For documentation, see:
>+    // http://clang.llvm.org/docs/SourceBasedCodeCoverage.html
>+    __llvm_profile_set_filename([file_name cStringUsingEncoding:NSUTF8StringEncoding]);
>+
>+    // Print the path for easier retrieval.
>+    NSLog(@"Coverage data at %@.", file_name);
>+  });
>+#endif  // ifdef WEBRTC_IOS_ENABLE_COVERAGE
>+}
>+
>+}  // namespace test
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.mm
>index fec597885497d6ba2996cdebb6bd96bd220cf06b..86005974fb2216c6365cc50c167b3968a8244d06 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.mm
>@@ -10,6 +10,7 @@
> 
> #import <UIKit/UIKit.h>
> 
>+#include "test/ios/coverage_util_ios.h"
> #include "test/ios/test_support.h"
> #include "test/testsupport/perf_test.h"
> 
>@@ -70,6 +71,8 @@ - (BOOL)application:(UIApplication *)application
> }
> 
> - (void)runTests {
>+  rtc::test::ConfigureCoverageReportPath();
>+
>   int exitStatus = g_test_suite();
> 
>   if (g_save_chartjson_result) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.h
>index 4913522ba701a6ad8100477b5ce3ce959bc80bf3..805044f52f9f9c7399c824910f19d12113eff9d1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.h
>@@ -13,8 +13,6 @@
> #include <set>
> #include <string>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> namespace test {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_writer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_writer.h
>index 4496ca4a25fdc0867f177a49bb7a5e0257990be6..5e560d737560a4772be39d93215da1925d1c055a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_writer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_writer.h
>@@ -12,7 +12,6 @@
> 
> #include <string>
> 
>-#include "common_types.h"  // NOLINT(build/include)
> #include "test/rtp_file_reader.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_rtcp_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_rtcp_observer.h
>index c9744109553742e7fd96513ccfaa3beb0ab4b7e2..d4aadfe0dadaf0f93fe3086e3985746beaa46d20 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_rtcp_observer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_rtcp_observer.h
>@@ -71,9 +71,7 @@ class RtpRtcpObserver {
>  protected:
>   RtpRtcpObserver() : RtpRtcpObserver(0) {}
>   explicit RtpRtcpObserver(int event_timeout_ms)
>-      : observation_complete_(false, false),
>-        parser_(RtpHeaderParser::Create()),
>-        timeout_ms_(event_timeout_ms) {
>+      : parser_(RtpHeaderParser::Create()), timeout_ms_(event_timeout_ms) {
>     parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
>                                         kTOffsetExtensionId);
>     parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.cc
>index dc2ec18eea79f4d46ce7ec39cb4fb80cec24008e..052662a464e94229d4621db74ce60555b3bf92a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.cc
>@@ -14,7 +14,7 @@
> namespace webrtc {
> namespace test {
> 
>-void PressEnterToContinue() {
>+void PressEnterToContinue(SingleThreadedTaskQueueForTesting&) {
>   puts(">> Press ENTER to continue...");
>   while (getc(stdin) != '\n' && !feof(stdin))
>     ;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.h
>index 90063dc3e873d7ff249812fc50be079f8cffa731..36bfa0669708abe8abbbe031e76ca170b92f8b34 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.h
>@@ -10,11 +10,13 @@
> #ifndef TEST_RUN_LOOP_H_
> #define TEST_RUN_LOOP_H_
> 
>+#include "test/single_threaded_task_queue.h"
>+
> namespace webrtc {
> namespace test {
> 
> // Blocks until the user presses enter.
>-void PressEnterToContinue();
>+void PressEnterToContinue(SingleThreadedTaskQueueForTesting &task_queue);
> 
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/BUILD.gn
>index 3703aa84e72e6c5cb5639607a3c633c308df8613..e9bda5112cb8ac9bd62f80b68242b97dc201981c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/BUILD.gn
>@@ -37,14 +37,15 @@ if (rtc_include_tests) {
>       "../:test_common",
>       "../:test_support",
>       "../:video_test_common",
>-      "../..:webrtc_common",
>       "../../api:libjingle_peerconnection_api",
>       "../../api:transport_api",
>       "../../api/audio_codecs:builtin_audio_decoder_factory",
>       "../../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../../api/test/video:function_video_factory",
>       "../../api/units:data_rate",
>       "../../api/units:time_delta",
>       "../../api/units:timestamp",
>+      "../../api/video:builtin_video_bitrate_allocator_factory",
>       "../../api/video:video_frame",
>       "../../api/video:video_frame_i420",
>       "../../api/video_codecs:video_codecs_api",
>@@ -82,6 +83,7 @@ if (rtc_include_tests) {
>       "../../rtc_base:rtc_base_approved",
>       "../../rtc_base:rtc_base_tests_utils",
>       "../../rtc_base:rtc_task_queue",
>+      "../../rtc_base:safe_minmax",
>       "../../rtc_base:sequenced_task_checker",
>       "../../rtc_base:stringutils",
>       "../../system_wrappers",
>@@ -94,6 +96,9 @@ if (rtc_include_tests) {
>     } else if (is_ios || is_mac) {
>       deps += [ "../../modules/video_coding:objc_codec_factory_helper" ]
>     }
>+    if (rtc_enable_protobuf) {
>+      deps += [ "../../modules/audio_coding:ana_config_proto" ]
>+    }
>     if (!build_with_chromium && is_clang) {
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>     }
>@@ -111,11 +116,10 @@ if (rtc_include_tests) {
>       "../../logging:mocks",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>-      "../../rtc_base:rtc_base_tests_utils",
>       "../../system_wrappers",
>+      "../../system_wrappers:field_trial",
>       "../../test:field_trial",
>       "../../test:test_support",
>-      "//system_wrappers:field_trial",
>       "//testing/gmock",
>       "//third_party/abseil-cpp/absl/memory",
>     ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/audio_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/audio_stream.cc
>index bd317d97d8b7c48819a0dfe65a1cb868788c6e1e..8359bd09ec58ece19ef9bcfd26b0df5bca087430 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/audio_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/audio_stream.cc
>@@ -9,10 +9,58 @@
>  */
> #include "test/scenario/audio_stream.h"
> 
>+#include "rtc_base/bitrateallocationstrategy.h"
> #include "test/call_test.h"
> 
>+#if WEBRTC_ENABLE_PROTOBUF
>+RTC_PUSH_IGNORING_WUNDEF()
>+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
>+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
>+#else
>+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
>+#endif
>+RTC_POP_IGNORING_WUNDEF()
>+#endif
>+
> namespace webrtc {
> namespace test {
>+namespace {
>+absl::optional<std::string> CreateAdaptationString(
>+    AudioStreamConfig::NetworkAdaptation config) {
>+#if WEBRTC_ENABLE_PROTOBUF
>+
>+  audio_network_adaptor::config::ControllerManager cont_conf;
>+  if (config.frame.max_rate_for_60_ms.IsFinite()) {
>+    auto controller =
>+        cont_conf.add_controllers()->mutable_frame_length_controller();
>+    controller->set_fl_decreasing_packet_loss_fraction(
>+        config.frame.min_packet_loss_for_decrease);
>+    controller->set_fl_increasing_packet_loss_fraction(
>+        config.frame.max_packet_loss_for_increase);
>+
>+    controller->set_fl_20ms_to_60ms_bandwidth_bps(
>+        config.frame.min_rate_for_20_ms.bps<int32_t>());
>+    controller->set_fl_60ms_to_20ms_bandwidth_bps(
>+        config.frame.max_rate_for_60_ms.bps<int32_t>());
>+
>+    if (config.frame.max_rate_for_120_ms.IsFinite()) {
>+      controller->set_fl_60ms_to_120ms_bandwidth_bps(
>+          config.frame.min_rate_for_60_ms.bps<int32_t>());
>+      controller->set_fl_120ms_to_60ms_bandwidth_bps(
>+          config.frame.max_rate_for_120_ms.bps<int32_t>());
>+    }
>+  }
>+  cont_conf.add_controllers()->mutable_bitrate_controller();
>+  std::string config_string = cont_conf.SerializeAsString();
>+  return config_string;
>+#else
>+  RTC_LOG(LS_ERROR) << "audio_network_adaptation is enabled"
>+                       " but WEBRTC_ENABLE_PROTOBUF is false.\n"
>+                       "Ignoring settings.";
>+  return absl::nullopt;
>+#endif  // WEBRTC_ENABLE_PROTOBUF
>+}
>+}  // namespace
> 
> SendAudioStream::SendAudioStream(
>     CallClient* sender,
>@@ -20,7 +68,8 @@ SendAudioStream::SendAudioStream(
>     rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
>     Transport* send_transport)
>     : sender_(sender), config_(config) {
>-  AudioSendStream::Config send_config(send_transport);
>+  AudioSendStream::Config send_config(send_transport,
>+                                      /*media_transport=*/nullptr);
>   ssrc_ = sender->GetNextAudioSsrc();
>   send_config.rtp.ssrc = ssrc_;
>   SdpAudioFormat::Parameters sdp_params;
>@@ -42,6 +91,10 @@ SendAudioStream::SendAudioStream(
>     send_config.send_codec_spec->target_bitrate_bps =
>         config.encoder.fixed_rate->bps();
> 
>+  if (config.network_adaptation) {
>+    send_config.audio_network_adaptor_config =
>+        CreateAdaptationString(config.adapt);
>+  }
>   if (config.encoder.allocate_bitrate ||
>       config.stream.in_bandwidth_estimation) {
>     DataRate min_rate = DataRate::Infinity();
>@@ -54,11 +107,20 @@ SendAudioStream::SendAudioStream(
>       max_rate = *config.encoder.max_rate;
>     }
>     if (field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
>-      TimeDelta frame_length = config.encoder.initial_frame_length;
>+      TimeDelta min_frame_length = TimeDelta::ms(20);
>+      // Note, depends on WEBRTC_OPUS_SUPPORT_120MS_PTIME being set, which is
>+      // the default.
>+      TimeDelta max_frame_length = TimeDelta::ms(120);
>       DataSize rtp_overhead = DataSize::bytes(12);
>-      DataSize total_overhead = config.stream.packet_overhead + rtp_overhead;
>-      min_rate += total_overhead / frame_length;
>-      max_rate += total_overhead / frame_length;
>+      // Note that this does not include rtp extension overhead and will not
>+      // follow updates in the transport overhead over time.
>+      DataSize total_overhead =
>+          sender_->transport_.packet_overhead() + rtp_overhead;
>+
>+      min_rate += total_overhead / max_frame_length;
>+      // In WebRTCVoiceEngine the max rate is also based on the max frame
>+      // length.
>+      max_rate += total_overhead / min_frame_length;
>     }
>     send_config.min_bitrate_bps = min_rate.bps();
>     send_config.max_bitrate_bps = max_rate.bps();
>@@ -70,13 +132,17 @@ SendAudioStream::SendAudioStream(
>         {RtpExtension::kTransportSequenceNumberUri, 8}};
>   }
> 
>-  if (config.stream.rate_allocation_priority) {
>+  if (config.encoder.priority_rate) {
>     send_config.track_id = sender->GetNextPriorityId();
>+    sender_->call_->SetBitrateAllocationStrategy(
>+        absl::make_unique<rtc::AudioPriorityBitrateAllocationStrategy>(
>+            send_config.track_id,
>+            config.encoder.priority_rate->bps<uint32_t>()));
>   }
>   send_stream_ = sender_->call_->CreateAudioSendStream(send_config);
>   if (field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
>     sender->call_->OnAudioTransportOverheadChanged(
>-        config.stream.packet_overhead.bytes());
>+        sender_->transport_.packet_overhead().bytes());
>   }
> }
> 
>@@ -86,17 +152,19 @@ SendAudioStream::~SendAudioStream() {
> 
> void SendAudioStream::Start() {
>   send_stream_->Start();
>+  sender_->call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
> }
> 
>-bool SendAudioStream::TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
>-                                       uint64_t receiver,
>-                                       Timestamp at_time) {
>-  // Removes added overhead before delivering RTCP packet to sender.
>-  RTC_DCHECK_GE(packet.size(), config_.stream.packet_overhead.bytes());
>-  packet.SetSize(packet.size() - config_.stream.packet_overhead.bytes());
>-  sender_->DeliverPacket(MediaType::AUDIO, packet, at_time);
>-  return true;
>+ColumnPrinter SendAudioStream::StatsPrinter() {
>+  return ColumnPrinter::Lambda(
>+      "audio_target_rate",
>+      [this](rtc::SimpleStringBuilder& sb) {
>+        AudioSendStream::Stats stats = send_stream_->GetStats();
>+        sb.AppendFormat("%.0lf", stats.target_bitrate_bps / 8.0);
>+      },
>+      64);
> }
>+
> ReceiveAudioStream::ReceiveAudioStream(
>     CallClient* receiver,
>     AudioStreamConfig config,
>@@ -108,11 +176,13 @@ ReceiveAudioStream::ReceiveAudioStream(
>   recv_config.rtp.local_ssrc = CallTest::kReceiverLocalAudioSsrc;
>   recv_config.rtcp_send_transport = feedback_transport;
>   recv_config.rtp.remote_ssrc = send_stream->ssrc_;
>+  receiver->ssrc_media_types_[recv_config.rtp.remote_ssrc] = MediaType::AUDIO;
>   if (config.stream.in_bandwidth_estimation) {
>     recv_config.rtp.transport_cc = true;
>     recv_config.rtp.extensions = {
>         {RtpExtension::kTransportSequenceNumberUri, 8}};
>   }
>+  receiver_->AddExtensions(recv_config.rtp.extensions);
>   recv_config.decoder_factory = decoder_factory;
>   recv_config.decoder_map = {
>       {CallTest::kAudioSendPayloadType, {"opus", 48000, 2}}};
>@@ -123,49 +193,26 @@ ReceiveAudioStream::~ReceiveAudioStream() {
>   receiver_->call_->DestroyAudioReceiveStream(receive_stream_);
> }
> 
>-bool ReceiveAudioStream::TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
>-                                          uint64_t receiver,
>-                                          Timestamp at_time) {
>-  RTC_DCHECK_GE(packet.size(), config_.stream.packet_overhead.bytes());
>-  packet.SetSize(packet.size() - config_.stream.packet_overhead.bytes());
>-  receiver_->DeliverPacket(MediaType::AUDIO, packet, at_time);
>-  return true;
>+void ReceiveAudioStream::Start() {
>+  receive_stream_->Start();
>+  receiver_->call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
> }
> 
> AudioStreamPair::~AudioStreamPair() = default;
> 
> AudioStreamPair::AudioStreamPair(
>     CallClient* sender,
>-    std::vector<NetworkNode*> send_link,
>-    uint64_t send_receiver_id,
>     rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
>     CallClient* receiver,
>-    std::vector<NetworkNode*> return_link,
>-    uint64_t return_receiver_id,
>     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
>     AudioStreamConfig config)
>     : config_(config),
>-      send_link_(send_link),
>-      return_link_(return_link),
>-      send_transport_(sender,
>-                      send_link.front(),
>-                      send_receiver_id,
>-                      config.stream.packet_overhead),
>-      return_transport_(receiver,
>-                        return_link.front(),
>-                        return_receiver_id,
>-                        config.stream.packet_overhead),
>-      send_stream_(sender, config, encoder_factory, &send_transport_),
>+      send_stream_(sender, config, encoder_factory, &sender->transport_),
>       receive_stream_(receiver,
>                       config,
>                       &send_stream_,
>                       decoder_factory,
>-                      &return_transport_) {
>-  NetworkNode::Route(send_transport_.ReceiverId(), send_link_,
>-                     &receive_stream_);
>-  NetworkNode::Route(return_transport_.ReceiverId(), return_link_,
>-                     &send_stream_);
>-}
>+                      &receiver->transport_) {}
> 
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/audio_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/audio_stream.h
>index 17b8bc370efd3d1921598b860fb0f87b30238926..3ab0a1f2def74c9ebd87d35c9ecb578a23d2ffc0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/audio_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/audio_stream.h
>@@ -24,11 +24,12 @@ namespace test {
> 
> // SendAudioStream represents sending of audio. It can be used for starting the
> // stream if neccessary.
>-class SendAudioStream : public NetworkReceiverInterface {
>+class SendAudioStream {
>  public:
>   RTC_DISALLOW_COPY_AND_ASSIGN(SendAudioStream);
>   ~SendAudioStream();
>   void Start();
>+  ColumnPrinter StatsPrinter();
> 
>  private:
>   friend class Scenario;
>@@ -38,11 +39,6 @@ class SendAudioStream : public NetworkReceiverInterface {
>                   AudioStreamConfig config,
>                   rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
>                   Transport* send_transport);
>-  // Handles RTCP feedback for this stream.
>-  bool TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
>-                        uint64_t receiver,
>-                        Timestamp at_time) override;
>-
>   AudioSendStream* send_stream_ = nullptr;
>   CallClient* const sender_;
>   const AudioStreamConfig config_;
>@@ -50,10 +46,11 @@ class SendAudioStream : public NetworkReceiverInterface {
> };
> 
> // ReceiveAudioStream represents an audio receiver. It can't be used directly.
>-class ReceiveAudioStream : public NetworkReceiverInterface {
>+class ReceiveAudioStream {
>  public:
>   RTC_DISALLOW_COPY_AND_ASSIGN(ReceiveAudioStream);
>   ~ReceiveAudioStream();
>+  void Start();
> 
>  private:
>   friend class Scenario;
>@@ -63,9 +60,6 @@ class ReceiveAudioStream : public NetworkReceiverInterface {
>                      SendAudioStream* send_stream,
>                      rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
>                      Transport* feedback_transport);
>-  bool TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
>-                        uint64_t receiver,
>-                        Timestamp at_time) override;
>   AudioReceiveStream* receive_stream_ = nullptr;
>   CallClient* const receiver_;
>   const AudioStreamConfig config_;
>@@ -84,23 +78,13 @@ class AudioStreamPair {
>  private:
>   friend class Scenario;
>   AudioStreamPair(CallClient* sender,
>-                  std::vector<NetworkNode*> send_link,
>-                  uint64_t send_receiver_id,
>                   rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
>-
>                   CallClient* receiver,
>-                  std::vector<NetworkNode*> return_link,
>-                  uint64_t return_receiver_id,
>                   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
>                   AudioStreamConfig config);
> 
>  private:
>   const AudioStreamConfig config_;
>-  std::vector<NetworkNode*> send_link_;
>-  std::vector<NetworkNode*> return_link_;
>-  NetworkNodeTransport send_transport_;
>-  NetworkNodeTransport return_transport_;
>-
>   SendAudioStream send_stream_;
>   ReceiveAudioStream receive_stream_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/call_client.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/call_client.cc
>index 47cc3bc57b82d2536e22fcde9e62c37ba43d260e..86eb4dc1db6a40a5f85169a36fe2e46411ce53a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/call_client.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/call_client.cc
>@@ -21,6 +21,39 @@ namespace webrtc {
> namespace test {
> namespace {
> const char* kPriorityStreamId = "priority-track";
>+
>+CallClientFakeAudio InitAudio() {
>+  CallClientFakeAudio setup;
>+  auto capturer = TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000);
>+  auto renderer = TestAudioDeviceModule::CreateDiscardRenderer(48000);
>+  setup.fake_audio_device = TestAudioDeviceModule::CreateTestAudioDeviceModule(
>+      std::move(capturer), std::move(renderer), 1.f);
>+  setup.apm = AudioProcessingBuilder().Create();
>+  setup.fake_audio_device->Init();
>+  AudioState::Config audio_state_config;
>+  audio_state_config.audio_mixer = AudioMixerImpl::Create();
>+  audio_state_config.audio_processing = setup.apm;
>+  audio_state_config.audio_device_module = setup.fake_audio_device;
>+  setup.audio_state = AudioState::Create(audio_state_config);
>+  setup.fake_audio_device->RegisterAudioCallback(
>+      setup.audio_state->audio_transport());
>+  return setup;
>+}
>+
>+Call* CreateCall(CallClientConfig config,
>+                 LoggingNetworkControllerFactory* network_controller_factory_,
>+                 rtc::scoped_refptr<AudioState> audio_state) {
>+  CallConfig call_config(network_controller_factory_->GetEventLog());
>+  call_config.bitrate_config.max_bitrate_bps =
>+      config.transport.rates.max_rate.bps_or(-1);
>+  call_config.bitrate_config.min_bitrate_bps =
>+      config.transport.rates.min_rate.bps();
>+  call_config.bitrate_config.start_bitrate_bps =
>+      config.transport.rates.start_rate.bps();
>+  call_config.network_controller_factory = network_controller_factory_;
>+  call_config.audio_state = audio_state;
>+  return Call::Create(call_config);
>+}
> }
> 
> LoggingNetworkControllerFactory::LoggingNetworkControllerFactory(
>@@ -107,31 +140,17 @@ CallClient::CallClient(Clock* clock,
>                        std::string log_filename,
>                        CallClientConfig config)
>     : clock_(clock),
>-      network_controller_factory_(log_filename, config.transport) {
>-  CallConfig call_config(network_controller_factory_.GetEventLog());
>-  call_config.bitrate_config.max_bitrate_bps =
>-      config.transport.rates.max_rate.bps_or(-1);
>-  call_config.bitrate_config.min_bitrate_bps =
>-      config.transport.rates.min_rate.bps();
>-  call_config.bitrate_config.start_bitrate_bps =
>-      config.transport.rates.start_rate.bps();
>-  call_config.network_controller_factory = &network_controller_factory_;
>-  call_config.audio_state = InitAudio();
>-  call_.reset(Call::Create(call_config));
>-  if (!config.priority_target_rate.IsZero() &&
>-      config.priority_target_rate.IsFinite()) {
>-    call_->SetBitrateAllocationStrategy(
>-        absl::make_unique<rtc::AudioPriorityBitrateAllocationStrategy>(
>-            kPriorityStreamId, config.priority_target_rate.bps()));
>-  }
>+      network_controller_factory_(log_filename, config.transport),
>+      fake_audio_setup_(InitAudio()),
>+      call_(CreateCall(config,
>+                       &network_controller_factory_,
>+                       fake_audio_setup_.audio_state)),
>+      transport_(clock_, call_.get()),
>+      header_parser_(RtpHeaderParser::Create()) {
> }  // namespace test
> 
>-CallClient::~CallClient() {}
>-
>-void CallClient::DeliverPacket(MediaType media_type,
>-                               rtc::CopyOnWriteBuffer packet,
>-                               Timestamp at_time) {
>-  call_->Receiver()->DeliverPacket(media_type, packet, at_time.us());
>+CallClient::~CallClient() {
>+  delete header_parser_;
> }
> 
> ColumnPrinter CallClient::StatsPrinter() {
>@@ -149,6 +168,26 @@ Call::Stats CallClient::GetStats() {
>   return call_->GetStats();
> }
> 
>+bool CallClient::TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
>+                                  uint64_t receiver,
>+                                  Timestamp at_time) {
>+  // Removes added overhead before delivering packet to sender.
>+  RTC_DCHECK_GE(packet.size(), route_overhead_.at(receiver).bytes());
>+  packet.SetSize(packet.size() - route_overhead_.at(receiver).bytes());
>+
>+  MediaType media_type = MediaType::ANY;
>+  if (!RtpHeaderParser::IsRtcp(packet.cdata(), packet.size())) {
>+    RTPHeader header;
>+    bool success =
>+        header_parser_->Parse(packet.cdata(), packet.size(), &header);
>+    if (!success)
>+      return false;
>+    media_type = ssrc_media_types_[header.ssrc];
>+  }
>+  call_->Receiver()->DeliverPacket(media_type, packet, at_time.us());
>+  return true;
>+}
>+
> uint32_t CallClient::GetNextVideoSsrc() {
>   RTC_CHECK_LT(next_video_ssrc_index_, CallTest::kNumSsrcs);
>   return CallTest::kVideoSendSsrcs[next_video_ssrc_index_++];
>@@ -170,21 +209,12 @@ std::string CallClient::GetNextPriorityId() {
>   return kPriorityStreamId;
> }
> 
>-rtc::scoped_refptr<AudioState> CallClient::InitAudio() {
>-  auto capturer = TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000);
>-  auto renderer = TestAudioDeviceModule::CreateDiscardRenderer(48000);
>-  fake_audio_device_ = TestAudioDeviceModule::CreateTestAudioDeviceModule(
>-      std::move(capturer), std::move(renderer), 1.f);
>-  apm_ = AudioProcessingBuilder().Create();
>-  fake_audio_device_->Init();
>-  AudioState::Config audio_state_config;
>-  audio_state_config.audio_mixer = AudioMixerImpl::Create();
>-  audio_state_config.audio_processing = apm_;
>-  audio_state_config.audio_device_module = fake_audio_device_;
>-  auto audio_state = AudioState::Create(audio_state_config);
>-  fake_audio_device_->RegisterAudioCallback(audio_state->audio_transport());
>-  return audio_state;
>+void CallClient::AddExtensions(std::vector<RtpExtension> extensions) {
>+  for (const auto& extension : extensions)
>+    header_parser_->RegisterRtpHeaderExtension(extension);
> }
> 
>+CallClientPair::~CallClientPair() = default;
>+
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/call_client.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/call_client.h
>index 80e2faf1626afd1d0b7ac4d4f7cf88b1d9988776..a12e06a065648979940bb841d24c658041edd615 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/call_client.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/call_client.h
>@@ -17,8 +17,10 @@
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/audio_device/include/test_audio_device.h"
> #include "modules/congestion_controller/test/controller_printer.h"
>+#include "modules/rtp_rtcp/include/rtp_header_parser.h"
> #include "rtc_base/constructormagic.h"
> #include "test/scenario/column_printer.h"
>+#include "test/scenario/network_node.h"
> #include "test/scenario/scenario_config.h"
> 
> namespace webrtc {
>@@ -45,10 +47,15 @@ class LoggingNetworkControllerFactory
>   FILE* cc_out_ = nullptr;
> };
> 
>+struct CallClientFakeAudio {
>+  rtc::scoped_refptr<AudioProcessing> apm;
>+  rtc::scoped_refptr<TestAudioDeviceModule> fake_audio_device;
>+  rtc::scoped_refptr<AudioState> audio_state;
>+};
> // CallClient represents a participant in a call scenario. It is created by the
> // Scenario class and is used as sender and receiver when setting up a media
> // stream session.
>-class CallClient {
>+class CallClient : public NetworkReceiverInterface {
>  public:
>   CallClient(Clock* clock, std::string log_filename, CallClientConfig config);
>   RTC_DISALLOW_COPY_AND_ASSIGN(CallClient);
>@@ -56,37 +63,63 @@ class CallClient {
>   ~CallClient();
>   ColumnPrinter StatsPrinter();
>   Call::Stats GetStats();
>+  DataRate send_bandwidth() {
>+    return DataRate::bps(GetStats().send_bandwidth_bps);
>+  }
>+
>+  bool TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
>+                        uint64_t receiver,
>+                        Timestamp at_time) override;
> 
>  private:
>   friend class Scenario;
>+  friend class CallClientPair;
>   friend class SendVideoStream;
>+  friend class VideoStreamPair;
>   friend class ReceiveVideoStream;
>   friend class SendAudioStream;
>   friend class ReceiveAudioStream;
>+  friend class AudioStreamPair;
>   friend class NetworkNodeTransport;
>-  // TODO(srte): Consider using the Columnprinter interface for this.
>-  void DeliverPacket(MediaType media_type,
>-                     rtc::CopyOnWriteBuffer packet,
>-                     Timestamp at_time);
>   uint32_t GetNextVideoSsrc();
>   uint32_t GetNextAudioSsrc();
>   uint32_t GetNextRtxSsrc();
>   std::string GetNextPriorityId();
>+  void AddExtensions(std::vector<RtpExtension> extensions);
> 
>   Clock* clock_;
>   LoggingNetworkControllerFactory network_controller_factory_;
>+  CallClientFakeAudio fake_audio_setup_;
>   std::unique_ptr<Call> call_;
>-
>-  rtc::scoped_refptr<AudioState> InitAudio();
>-
>-  rtc::scoped_refptr<AudioProcessing> apm_;
>-  rtc::scoped_refptr<TestAudioDeviceModule> fake_audio_device_;
>+  NetworkNodeTransport transport_;
>+  RtpHeaderParser* const header_parser_;
> 
>   std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
>+  // Stores the configured overhead per known incomming route. This is used to
>+  // subtract the overhead before processing.
>+  std::map<uint64_t, DataSize> route_overhead_;
>   int next_video_ssrc_index_ = 0;
>   int next_rtx_ssrc_index_ = 0;
>   int next_audio_ssrc_index_ = 0;
>   int next_priority_index_ = 0;
>+  std::map<uint32_t, MediaType> ssrc_media_types_;
>+};
>+
>+class CallClientPair {
>+ public:
>+  RTC_DISALLOW_COPY_AND_ASSIGN(CallClientPair);
>+  ~CallClientPair();
>+  CallClient* first() { return first_; }
>+  CallClient* second() { return second_; }
>+  std::pair<CallClient*, CallClient*> forward() { return {first(), second()}; }
>+  std::pair<CallClient*, CallClient*> reverse() { return {second(), first()}; }
>+
>+ private:
>+  friend class Scenario;
>+  CallClientPair(CallClient* first, CallClient* second)
>+      : first_(first), second_(second) {}
>+  CallClient* const first_;
>+  CallClient* const second_;
> };
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/network_node.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/network_node.cc
>index ff138939abb1efd51c1ae4a58ba179500dddbaca..b39d67bd25c8909a32035d3aeaa8a3e3dc3e1c9f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/network_node.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/network_node.cc
>@@ -12,6 +12,8 @@
> #include <algorithm>
> #include <vector>
> 
>+#include "rtc_base/numerics/safe_minmax.h"
>+
> namespace webrtc {
> namespace test {
> namespace {
>@@ -168,29 +170,29 @@ SimulationNode::SimulationNode(
>       simulated_network_(simulation),
>       config_(config) {}
> 
>-NetworkNodeTransport::NetworkNodeTransport(CallClient* sender,
>-                                           NetworkNode* send_net,
>-                                           uint64_t receiver,
>-                                           DataSize packet_overhead)
>-    : sender_(sender),
>-      send_net_(send_net),
>-      receiver_id_(receiver),
>-      packet_overhead_(packet_overhead) {}
>+NetworkNodeTransport::NetworkNodeTransport(const Clock* sender_clock,
>+                                           Call* sender_call)
>+    : sender_clock_(sender_clock), sender_call_(sender_call) {}
> 
> NetworkNodeTransport::~NetworkNodeTransport() = default;
> 
> bool NetworkNodeTransport::SendRtp(const uint8_t* packet,
>                                    size_t length,
>                                    const PacketOptions& options) {
>-  int64_t send_time_ms = sender_->clock_->TimeInMilliseconds();
>+  int64_t send_time_ms = sender_clock_->TimeInMilliseconds();
>   rtc::SentPacket sent_packet;
>   sent_packet.packet_id = options.packet_id;
>+  sent_packet.info.included_in_feedback = options.included_in_feedback;
>+  sent_packet.info.included_in_allocation = options.included_in_allocation;
>   sent_packet.send_time_ms = send_time_ms;
>   sent_packet.info.packet_size_bytes = length;
>   sent_packet.info.packet_type = rtc::PacketType::kData;
>-  sender_->call_->OnSentPacket(sent_packet);
>+  sender_call_->OnSentPacket(sent_packet);
> 
>   Timestamp send_time = Timestamp::ms(send_time_ms);
>+  rtc::CritScope crit(&crit_sect_);
>+  if (!send_net_)
>+    return false;
>   rtc::CopyOnWriteBuffer buffer(packet, length,
>                                 length + packet_overhead_.bytes());
>   buffer.SetSize(length + packet_overhead_.bytes());
>@@ -199,13 +201,29 @@ bool NetworkNodeTransport::SendRtp(const uint8_t* packet,
> 
> bool NetworkNodeTransport::SendRtcp(const uint8_t* packet, size_t length) {
>   rtc::CopyOnWriteBuffer buffer(packet, length);
>-  Timestamp send_time = Timestamp::ms(sender_->clock_->TimeInMilliseconds());
>+  Timestamp send_time = Timestamp::ms(sender_clock_->TimeInMilliseconds());
>+  rtc::CritScope crit(&crit_sect_);
>   buffer.SetSize(length + packet_overhead_.bytes());
>+  if (!send_net_)
>+    return false;
>   return send_net_->TryDeliverPacket(buffer, receiver_id_, send_time);
> }
> 
>-uint64_t NetworkNodeTransport::ReceiverId() const {
>-  return receiver_id_;
>+void NetworkNodeTransport::Connect(NetworkNode* send_node,
>+                                   uint64_t receiver_id,
>+                                   DataSize packet_overhead) {
>+  rtc::CritScope crit(&crit_sect_);
>+  send_net_ = send_node;
>+  receiver_id_ = receiver_id;
>+  packet_overhead_ = packet_overhead;
>+
>+  rtc::NetworkRoute route;
>+  route.connected = true;
>+  route.local_network_id = receiver_id;
>+  route.remote_network_id = receiver_id;
>+  std::string transport_name = "dummy";
>+  sender_call_->GetTransportControllerSend()->OnNetworkRouteChanged(
>+      transport_name, route);
> }
> 
> CrossTrafficSource::CrossTrafficSource(NetworkReceiverInterface* target,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/network_node.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/network_node.h
>index af6db0a0099cc387fdb4708b98de1ba595f4ceb2..4e90f6f252a2c7cf5de76b88dd09a7e907a8686c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/network_node.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/network_node.h
>@@ -18,10 +18,10 @@
> 
> #include "api/call/transport.h"
> #include "api/units/timestamp.h"
>+#include "call/call.h"
> #include "call/simulated_network.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/copyonwritebuffer.h"
>-#include "test/scenario/call_client.h"
> #include "test/scenario/column_printer.h"
> #include "test/scenario/scenario_config.h"
> 
>@@ -119,23 +119,30 @@ class SimulationNode : public NetworkNode {
> 
> class NetworkNodeTransport : public Transport {
>  public:
>-  NetworkNodeTransport(CallClient* sender,
>-                       NetworkNode* send_net,
>-                       uint64_t receiver,
>-                       DataSize packet_overhead);
>+  NetworkNodeTransport(const Clock* sender_clock, Call* sender_call);
>   ~NetworkNodeTransport() override;
> 
>   bool SendRtp(const uint8_t* packet,
>                size_t length,
>                const PacketOptions& options) override;
>   bool SendRtcp(const uint8_t* packet, size_t length) override;
>-  uint64_t ReceiverId() const;
>+
>+  void Connect(NetworkNode* send_node,
>+               uint64_t receiver_id,
>+               DataSize packet_overhead);
>+
>+  DataSize packet_overhead() {
>+    rtc::CritScope crit(&crit_sect_);
>+    return packet_overhead_;
>+  }
> 
>  private:
>-  CallClient* const sender_;
>-  NetworkNode* const send_net_;
>-  const uint64_t receiver_id_;
>-  const DataSize packet_overhead_;
>+  rtc::CriticalSection crit_sect_;
>+  const Clock* const sender_clock_;
>+  Call* const sender_call_;
>+  NetworkNode* send_net_ RTC_GUARDED_BY(crit_sect_) = nullptr;
>+  uint64_t receiver_id_ RTC_GUARDED_BY(crit_sect_) = 0;
>+  DataSize packet_overhead_ RTC_GUARDED_BY(crit_sect_) = DataSize::Zero();
> };
> 
> // CrossTrafficSource is created by a Scenario and generates cross traffic. It
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario.cc
>index 7eb09e8bb7a1d1248624327ab4baae2eaea391c5..7589878d5ba353978f530cc2fbc0669bd5e64ba2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario.cc
>@@ -16,7 +16,7 @@
> #include "rtc_base/flags.h"
> #include "test/testsupport/fileutils.h"
> 
>-DEFINE_bool(scenario_logs, false, "Save logs from scenario framework.");
>+WEBRTC_DEFINE_bool(scenario_logs, false, "Save logs from scenario framework.");
> 
> namespace webrtc {
> namespace test {
>@@ -75,6 +75,8 @@ Scenario::Scenario(std::string file_name, bool real_time)
> }
> 
> Scenario::~Scenario() {
>+  if (start_time_.IsFinite())
>+    Stop();
>   if (!real_time_mode_)
>     rtc::SetClockForTesting(nullptr);
> }
>@@ -123,14 +125,50 @@ CallClient* Scenario::CreateClient(
>   return CreateClient(name, config);
> }
> 
>+CallClientPair* Scenario::CreateRoutes(CallClient* first,
>+                                       std::vector<NetworkNode*> send_link,
>+                                       CallClient* second,
>+                                       std::vector<NetworkNode*> return_link) {
>+  return CreateRoutes(first, send_link,
>+                      DataSize::bytes(PacketOverhead::kDefault), second,
>+                      return_link, DataSize::bytes(PacketOverhead::kDefault));
>+}
>+
>+CallClientPair* Scenario::CreateRoutes(CallClient* first,
>+                                       std::vector<NetworkNode*> send_link,
>+                                       DataSize first_overhead,
>+                                       CallClient* second,
>+                                       std::vector<NetworkNode*> return_link,
>+                                       DataSize second_overhead) {
>+  CallClientPair* client_pair = new CallClientPair(first, second);
>+  ChangeRoute(client_pair->forward(), send_link, first_overhead);
>+  ChangeRoute(client_pair->reverse(), return_link, second_overhead);
>+  client_pairs_.emplace_back(client_pair);
>+  return client_pair;
>+}
>+
>+void Scenario::ChangeRoute(std::pair<CallClient*, CallClient*> clients,
>+                           std::vector<NetworkNode*> over_nodes) {
>+  ChangeRoute(clients, over_nodes, DataSize::bytes(PacketOverhead::kDefault));
>+}
>+
>+void Scenario::ChangeRoute(std::pair<CallClient*, CallClient*> clients,
>+                           std::vector<NetworkNode*> over_nodes,
>+                           DataSize overhead) {
>+  uint64_t route_id = next_route_id_++;
>+  clients.second->route_overhead_.insert({route_id, overhead});
>+  NetworkNode::Route(route_id, over_nodes, clients.second);
>+  clients.first->transport_.Connect(over_nodes.front(), route_id, overhead);
>+}
>+
> SimulatedTimeClient* Scenario::CreateSimulatedTimeClient(
>     std::string name,
>     SimulatedTimeClientConfig config,
>     std::vector<PacketStreamConfig> stream_configs,
>     std::vector<NetworkNode*> send_link,
>     std::vector<NetworkNode*> return_link) {
>-  uint64_t send_id = next_receiver_id_++;
>-  uint64_t return_id = next_receiver_id_++;
>+  uint64_t send_id = next_route_id_++;
>+  uint64_t return_id = next_route_id_++;
>   SimulatedTimeClient* client = new SimulatedTimeClient(
>       GetFullPathOrEmpty(name), config, stream_configs, send_link, return_link,
>       send_id, return_id, Now());
>@@ -179,21 +217,21 @@ NetworkNode* Scenario::CreateNetworkNode(
> void Scenario::TriggerPacketBurst(std::vector<NetworkNode*> over_nodes,
>                                   size_t num_packets,
>                                   size_t packet_size) {
>-  int64_t receiver_id = next_receiver_id_++;
>-  NetworkNode::Route(receiver_id, over_nodes, &null_receiver_);
>+  int64_t route_id = next_route_id_++;
>+  NetworkNode::Route(route_id, over_nodes, &null_receiver_);
>   for (size_t i = 0; i < num_packets; ++i)
>     over_nodes[0]->TryDeliverPacket(rtc::CopyOnWriteBuffer(packet_size),
>-                                    receiver_id, Now());
>+                                    route_id, Now());
> }
> 
> void Scenario::NetworkDelayedAction(std::vector<NetworkNode*> over_nodes,
>                                     size_t packet_size,
>                                     std::function<void()> action) {
>-  int64_t receiver_id = next_receiver_id_++;
>+  int64_t route_id = next_route_id_++;
>   action_receivers_.emplace_back(new ActionReceiver(action));
>-  NetworkNode::Route(receiver_id, over_nodes, action_receivers_.back().get());
>-  over_nodes[0]->TryDeliverPacket(rtc::CopyOnWriteBuffer(packet_size),
>-                                  receiver_id, Now());
>+  NetworkNode::Route(route_id, over_nodes, action_receivers_.back().get());
>+  over_nodes[0]->TryDeliverPacket(rtc::CopyOnWriteBuffer(packet_size), route_id,
>+                                  Now());
> }
> 
> CrossTrafficSource* Scenario::CreateCrossTraffic(
>@@ -207,65 +245,46 @@ CrossTrafficSource* Scenario::CreateCrossTraffic(
> CrossTrafficSource* Scenario::CreateCrossTraffic(
>     std::vector<NetworkNode*> over_nodes,
>     CrossTrafficConfig config) {
>-  int64_t receiver_id = next_receiver_id_++;
>+  int64_t route_id = next_route_id_++;
>   cross_traffic_sources_.emplace_back(
>-      new CrossTrafficSource(over_nodes.front(), receiver_id, config));
>+      new CrossTrafficSource(over_nodes.front(), route_id, config));
>   CrossTrafficSource* node = cross_traffic_sources_.back().get();
>-  NetworkNode::Route(receiver_id, over_nodes, &null_receiver_);
>+  NetworkNode::Route(route_id, over_nodes, &null_receiver_);
>   Every(config.min_packet_interval,
>         [this, node](TimeDelta delta) { node->Process(Now(), delta); });
>   return node;
> }
> 
> VideoStreamPair* Scenario::CreateVideoStream(
>-    CallClient* sender,
>-    std::vector<NetworkNode*> send_link,
>-    CallClient* receiver,
>-    std::vector<NetworkNode*> return_link,
>+    std::pair<CallClient*, CallClient*> clients,
>     std::function<void(VideoStreamConfig*)> config_modifier) {
>   VideoStreamConfig config;
>   config_modifier(&config);
>-  return CreateVideoStream(sender, send_link, receiver, return_link, config);
>+  return CreateVideoStream(clients, config);
> }
> 
> VideoStreamPair* Scenario::CreateVideoStream(
>-    CallClient* sender,
>-    std::vector<NetworkNode*> send_link,
>-    CallClient* receiver,
>-    std::vector<NetworkNode*> return_link,
>+    std::pair<CallClient*, CallClient*> clients,
>     VideoStreamConfig config) {
>-  uint64_t send_receiver_id = next_receiver_id_++;
>-  uint64_t return_receiver_id = next_receiver_id_++;
>-
>   video_streams_.emplace_back(
>-      new VideoStreamPair(sender, send_link, send_receiver_id, receiver,
>-                          return_link, return_receiver_id, config));
>+      new VideoStreamPair(clients.first, clients.second, config));
>   return video_streams_.back().get();
> }
> 
> AudioStreamPair* Scenario::CreateAudioStream(
>-    CallClient* sender,
>-    std::vector<NetworkNode*> send_link,
>-    CallClient* receiver,
>-    std::vector<NetworkNode*> return_link,
>+    std::pair<CallClient*, CallClient*> clients,
>     std::function<void(AudioStreamConfig*)> config_modifier) {
>   AudioStreamConfig config;
>   config_modifier(&config);
>-  return CreateAudioStream(sender, send_link, receiver, return_link, config);
>+  return CreateAudioStream(clients, config);
> }
> 
> AudioStreamPair* Scenario::CreateAudioStream(
>-    CallClient* sender,
>-    std::vector<NetworkNode*> send_link,
>-    CallClient* receiver,
>-    std::vector<NetworkNode*> return_link,
>+    std::pair<CallClient*, CallClient*> clients,
>     AudioStreamConfig config) {
>-  uint64_t send_receiver_id = next_receiver_id_++;
>-  uint64_t return_receiver_id = next_receiver_id_++;
>-
>-  audio_streams_.emplace_back(new AudioStreamPair(
>-      sender, send_link, send_receiver_id, audio_encoder_factory_, receiver,
>-      return_link, return_receiver_id, audio_decoder_factory_, config));
>+  audio_streams_.emplace_back(
>+      new AudioStreamPair(clients.first, audio_encoder_factory_, clients.second,
>+                          audio_decoder_factory_, config));
>   return audio_streams_.back().get();
> }
> 
>@@ -288,37 +307,20 @@ void Scenario::At(TimeDelta offset, std::function<void()> function) {
> }
> 
> void Scenario::RunFor(TimeDelta duration) {
>-  RunUntil(duration, TimeDelta::PlusInfinity(), []() { return false; });
>+  RunUntil(Duration() + duration);
>+}
>+
>+void Scenario::RunUntil(TimeDelta max_duration) {
>+  RunUntil(max_duration, TimeDelta::PlusInfinity(), []() { return false; });
> }
> 
> void Scenario::RunUntil(TimeDelta max_duration,
>                         TimeDelta poll_interval,
>                         std::function<bool()> exit_function) {
>-  start_time_ = Timestamp::us(clock_->TimeInMicroseconds());
>-  for (auto& activity : repeated_activities_) {
>-    activity->SetStartTime(start_time_);
>-  }
>+  if (start_time_.IsInfinite())
>+    Start();
> 
>-  for (auto& stream_pair : video_streams_)
>-    stream_pair->receive()->receive_stream_->Start();
>-  for (auto& stream_pair : audio_streams_)
>-    stream_pair->receive()->receive_stream_->Start();
>-  for (auto& stream_pair : video_streams_) {
>-    if (stream_pair->config_.autostart) {
>-      stream_pair->send()->Start();
>-    }
>-  }
>-  for (auto& stream_pair : audio_streams_) {
>-    if (stream_pair->config_.autostart) {
>-      stream_pair->send()->Start();
>-    }
>-  }
>-  for (auto& call : clients_) {
>-    call->call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
>-    call->call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>-  }
>-
>-  rtc::Event done_(false, false);
>+  rtc::Event done_;
>   while (!exit_function() && Duration() < max_duration) {
>     Timestamp current_time = Now();
>     TimeDelta duration = current_time - start_time_;
>@@ -346,6 +348,32 @@ void Scenario::RunUntil(TimeDelta max_duration,
>                                            1000);
>     }
>   }
>+}
>+
>+void Scenario::Start() {
>+  start_time_ = Timestamp::us(clock_->TimeInMicroseconds());
>+  for (auto& activity : repeated_activities_) {
>+    activity->SetStartTime(start_time_);
>+  }
>+
>+  for (auto& stream_pair : video_streams_)
>+    stream_pair->receive()->Start();
>+  for (auto& stream_pair : audio_streams_)
>+    stream_pair->receive()->Start();
>+  for (auto& stream_pair : video_streams_) {
>+    if (stream_pair->config_.autostart) {
>+      stream_pair->send()->Start();
>+    }
>+  }
>+  for (auto& stream_pair : audio_streams_) {
>+    if (stream_pair->config_.autostart) {
>+      stream_pair->send()->Start();
>+    }
>+  }
>+}
>+
>+void Scenario::Stop() {
>+  RTC_DCHECK(start_time_.IsFinite());
>   for (auto& stream_pair : video_streams_) {
>     stream_pair->send()->video_capturer_->Stop();
>     stream_pair->send()->send_stream_->Stop();
>@@ -356,6 +384,7 @@ void Scenario::RunUntil(TimeDelta max_duration,
>     stream_pair->receive()->receive_stream_->Stop();
>   for (auto& stream_pair : audio_streams_)
>     stream_pair->receive()->receive_stream_->Stop();
>+  start_time_ = Timestamp::PlusInfinity();
> }
> 
> Timestamp Scenario::Now() {
>@@ -363,6 +392,8 @@ Timestamp Scenario::Now() {
> }
> 
> TimeDelta Scenario::Duration() {
>+  if (start_time_.IsInfinite())
>+    return TimeDelta::Zero();
>   return Now() - start_time_;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario.h
>index 05876ac49c1563872f26ed38dc006839e8490379..38bef442ecdd53029513768dcea12eb55737949a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario.h
>@@ -78,6 +78,25 @@ class Scenario {
>       std::string name,
>       std::function<void(CallClientConfig*)> config_modifier);
> 
>+  CallClientPair* CreateRoutes(CallClient* first,
>+                               std::vector<NetworkNode*> send_link,
>+                               CallClient* second,
>+                               std::vector<NetworkNode*> return_link);
>+
>+  CallClientPair* CreateRoutes(CallClient* first,
>+                               std::vector<NetworkNode*> send_link,
>+                               DataSize first_overhead,
>+                               CallClient* second,
>+                               std::vector<NetworkNode*> return_link,
>+                               DataSize second_overhead);
>+
>+  void ChangeRoute(std::pair<CallClient*, CallClient*> clients,
>+                   std::vector<NetworkNode*> over_nodes);
>+
>+  void ChangeRoute(std::pair<CallClient*, CallClient*> clients,
>+                   std::vector<NetworkNode*> over_nodes,
>+                   DataSize overhead);
>+
>   SimulatedTimeClient* CreateSimulatedTimeClient(
>       std::string name,
>       SimulatedTimeClientConfig config,
>@@ -86,28 +105,18 @@ class Scenario {
>       std::vector<NetworkNode*> return_link);
> 
>   VideoStreamPair* CreateVideoStream(
>-      CallClient* sender,
>-      std::vector<NetworkNode*> send_link,
>-      CallClient* receiver,
>-      std::vector<NetworkNode*> return_link,
>+      std::pair<CallClient*, CallClient*> clients,
>       std::function<void(VideoStreamConfig*)> config_modifier);
>-  VideoStreamPair* CreateVideoStream(CallClient* sender,
>-                                     std::vector<NetworkNode*> send_link,
>-                                     CallClient* receiver,
>-                                     std::vector<NetworkNode*> return_link,
>-                                     VideoStreamConfig config);
>+  VideoStreamPair* CreateVideoStream(
>+      std::pair<CallClient*, CallClient*> clients,
>+      VideoStreamConfig config);
> 
>   AudioStreamPair* CreateAudioStream(
>-      CallClient* sender,
>-      std::vector<NetworkNode*> send_link,
>-      CallClient* receiver,
>-      std::vector<NetworkNode*> return_link,
>+      std::pair<CallClient*, CallClient*> clients,
>       std::function<void(AudioStreamConfig*)> config_modifier);
>-  AudioStreamPair* CreateAudioStream(CallClient* sender,
>-                                     std::vector<NetworkNode*> send_link,
>-                                     CallClient* receiver,
>-                                     std::vector<NetworkNode*> return_link,
>-                                     AudioStreamConfig config);
>+  AudioStreamPair* CreateAudioStream(
>+      std::pair<CallClient*, CallClient*> clients,
>+      AudioStreamConfig config);
> 
>   CrossTrafficSource* CreateCrossTraffic(
>       std::vector<NetworkNode*> over_nodes,
>@@ -131,9 +140,12 @@ class Scenario {
>   // Runs the scenario for the given time or until the exit function returns
>   // true.
>   void RunFor(TimeDelta duration);
>+  void RunUntil(TimeDelta max_duration);
>   void RunUntil(TimeDelta max_duration,
>                 TimeDelta probe_interval,
>                 std::function<bool()> exit_function);
>+  void Start();
>+  void Stop();
> 
>   // Triggers sending of dummy packets over the given nodes.
>   void TriggerPacketBurst(std::vector<NetworkNode*> over_nodes,
>@@ -167,6 +179,7 @@ class Scenario {
>   rtc::FakeClock event_log_fake_clock_;
> 
>   std::vector<std::unique_ptr<CallClient>> clients_;
>+  std::vector<std::unique_ptr<CallClientPair>> client_pairs_;
>   std::vector<std::unique_ptr<NetworkNode>> network_nodes_;
>   std::vector<std::unique_ptr<CrossTrafficSource>> cross_traffic_sources_;
>   std::vector<std::unique_ptr<VideoStreamPair>> video_streams_;
>@@ -179,7 +192,7 @@ class Scenario {
>   std::vector<std::unique_ptr<PendingActivity>> pending_activities_;
>   std::vector<std::unique_ptr<StatesPrinter>> printers_;
> 
>-  int64_t next_receiver_id_ = 40000;
>+  int64_t next_route_id_ = 40000;
>   rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
>   rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_config.h
>index e652e2ce3921df9d634957c1d05a1dc8443e97a1..88fdb8af01735cbc0567c1b1548a531cd3cfe02c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_config.h
>@@ -28,7 +28,12 @@ struct PacketOverhead {
>   static constexpr size_t kIpv6 = 40;
>   static constexpr size_t kUdp = 8;
>   static constexpr size_t kSrtp = 10;
>-  static constexpr size_t kTurn = 4;
>+  static constexpr size_t kStun = 4;
>+  // TURN messages can be sent either with or without an establieshed channel.
>+  // In the latter case, a TURN Send/Data Indication is sent which has
>+  // significantly more overhead.
>+  static constexpr size_t kTurnChannelMessage = 4;
>+  static constexpr size_t kTurnIndicationMessage = 36;
>   static constexpr size_t kDefault = kIpv4 + kUdp + kSrtp;
> };
> struct TransportControllerConfig {
>@@ -46,7 +51,6 @@ struct TransportControllerConfig {
> 
> struct CallClientConfig {
>   TransportControllerConfig transport;
>-  DataRate priority_target_rate = DataRate::Zero();
> };
> 
> struct SimulatedTimeClientConfig {
>@@ -71,6 +75,10 @@ struct PacketStreamConfig {
> struct VideoStreamConfig {
>   bool autostart = true;
>   struct Source {
>+    enum class ContentType {
>+      kVideo,
>+      kScreen,
>+    } content_type = ContentType::kVideo;
>     enum Capture {
>       kGenerator,
>       kVideoFile,
>@@ -103,6 +111,7 @@ struct VideoStreamConfig {
>     absl::optional<int> key_frame_interval = 3000;
> 
>     absl::optional<DataRate> max_data_rate;
>+    absl::optional<int> max_framerate;
>     size_t num_simulcast_streams = 1;
>     using DegradationPreference = DegradationPreference;
>     DegradationPreference degradation_preference =
>@@ -114,10 +123,10 @@ struct VideoStreamConfig {
>     ~Stream();
>     bool packet_feedback = true;
>     bool use_rtx = true;
>+    DataRate pad_to_rate = DataRate::Zero();
>     TimeDelta nack_history_time = TimeDelta::ms(1000);
>     bool use_flexfec = false;
>     bool use_ulpfec = false;
>-    DataSize packet_overhead = DataSize::bytes(PacketOverhead::kDefault);
>   } stream;
>   struct Renderer {
>     enum Type { kFake } type = kFake;
>@@ -132,6 +141,17 @@ struct AudioStreamConfig {
>   struct Source {
>     int channels = 1;
>   } source;
>+  bool network_adaptation = false;
>+  struct NetworkAdaptation {
>+    struct FrameLength {
>+      double min_packet_loss_for_decrease = 0;
>+      double max_packet_loss_for_increase = 1;
>+      DataRate min_rate_for_20_ms = DataRate::Zero();
>+      DataRate max_rate_for_60_ms = DataRate::Infinity();
>+      DataRate min_rate_for_60_ms = DataRate::Zero();
>+      DataRate max_rate_for_120_ms = DataRate::Infinity();
>+    } frame;
>+  } adapt;
>   struct Encoder {
>     Encoder();
>     Encoder(const Encoder&);
>@@ -140,6 +160,7 @@ struct AudioStreamConfig {
>     absl::optional<DataRate> fixed_rate;
>     absl::optional<DataRate> min_rate;
>     absl::optional<DataRate> max_rate;
>+    absl::optional<DataRate> priority_rate;
>     TimeDelta initial_frame_length = TimeDelta::ms(20);
>   } encoder;
>   struct Stream {
>@@ -147,8 +168,6 @@ struct AudioStreamConfig {
>     Stream(const Stream&);
>     ~Stream();
>     bool in_bandwidth_estimation = false;
>-    bool rate_allocation_priority = false;
>-    DataSize packet_overhead = DataSize::bytes(PacketOverhead::kDefault);
>   } stream;
>   struct Render {
>     std::string sync_group;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_tests/bbr_performance.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_tests/bbr_performance.cc
>index e87cc68dd31f70c8f8fce3f0dda757a6a2759d29..28cabc070fda07fb7663a130e4f87795c187f549 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_tests/bbr_performance.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_tests/bbr_performance.cc
>@@ -167,31 +167,31 @@ TEST_P(BbrScenarioTest, ReceivesVideo) {
>   net_conf.simulation.delay_std_dev = conf_.scenario.delay_noise;
>   SimulationNode* send_net = s.CreateSimulationNode(net_conf);
>   SimulationNode* ret_net = s.CreateSimulationNode(net_conf);
>-  VideoStreamPair* alice_video = s.CreateVideoStream(
>-      alice, {send_net}, bob, {ret_net}, [&](VideoStreamConfig* c) {
>+  auto route = s.CreateRoutes(alice, {send_net}, bob, {ret_net});
>+
>+  VideoStreamPair* alice_video =
>+      s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
>         c->encoder.fake.max_rate = DataRate::kbps(1800);
>       });
>-  s.CreateAudioStream(alice, {send_net}, bob, {ret_net},
>-                      [&](AudioStreamConfig* c) {
>-                        if (conf_.tuning.use_bbr) {
>-                          c->stream.in_bandwidth_estimation = true;
>-                          c->encoder.fixed_rate = DataRate::kbps(31);
>-                        }
>-                      });
>+  s.CreateAudioStream(route->forward(), [&](AudioStreamConfig* c) {
>+    if (conf_.tuning.use_bbr) {
>+      c->stream.in_bandwidth_estimation = true;
>+      c->encoder.fixed_rate = DataRate::kbps(31);
>+    }
>+  });
> 
>   VideoStreamPair* bob_video = nullptr;
>   if (conf_.scenario.return_traffic) {
>-    bob_video = s.CreateVideoStream(
>-        bob, {ret_net}, alice, {send_net}, [&](VideoStreamConfig* c) {
>+    bob_video =
>+        s.CreateVideoStream(route->reverse(), [&](VideoStreamConfig* c) {
>           c->encoder.fake.max_rate = DataRate::kbps(1800);
>         });
>-    s.CreateAudioStream(bob, {ret_net}, alice, {send_net},
>-                        [&](AudioStreamConfig* c) {
>-                          if (conf_.tuning.use_bbr) {
>-                            c->stream.in_bandwidth_estimation = true;
>-                            c->encoder.fixed_rate = DataRate::kbps(31);
>-                          }
>-                        });
>+    s.CreateAudioStream(route->reverse(), [&](AudioStreamConfig* c) {
>+      if (conf_.tuning.use_bbr) {
>+        c->stream.in_bandwidth_estimation = true;
>+        c->encoder.fixed_rate = DataRate::kbps(31);
>+      }
>+    });
>   }
>   CrossTrafficConfig cross_config;
>   cross_config.peak_rate = conf_.scenario.cross_traffic;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_unittest.cc
>index 305f8660ccd744e9e7f563d8f33e2bfdc24c8ce3..97285d391459f2cfc8c3afc665b381c8151f7cc1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/scenario_unittest.cc
>@@ -21,14 +21,19 @@ TEST(ScenarioTest, StartsAndStopsWithoutErrors) {
>   NetworkNodeConfig network_config;
>   auto alice_net = s.CreateSimulationNode(network_config);
>   auto bob_net = s.CreateSimulationNode(network_config);
>+  auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net});
> 
>   VideoStreamConfig video_stream_config;
>-  s.CreateVideoStream(alice, {alice_net}, bob, {bob_net}, video_stream_config);
>-  s.CreateVideoStream(bob, {bob_net}, alice, {alice_net}, video_stream_config);
>+  s.CreateVideoStream(route->forward(), video_stream_config);
>+  s.CreateVideoStream(route->reverse(), video_stream_config);
> 
>   AudioStreamConfig audio_stream_config;
>-  s.CreateAudioStream(alice, {alice_net}, bob, {bob_net}, audio_stream_config);
>-  s.CreateAudioStream(bob, {bob_net}, alice, {alice_net}, audio_stream_config);
>+  audio_stream_config.encoder.min_rate = DataRate::kbps(6);
>+  audio_stream_config.encoder.max_rate = DataRate::kbps(64);
>+  audio_stream_config.encoder.allocate_bitrate = true;
>+  audio_stream_config.stream.in_bandwidth_estimation = false;
>+  s.CreateAudioStream(route->forward(), audio_stream_config);
>+  s.CreateAudioStream(route->reverse(), audio_stream_config);
> 
>   CrossTrafficConfig cross_traffic_config;
>   s.CreateCrossTraffic({alice_net}, cross_traffic_config);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/simulated_time.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/simulated_time.cc
>index f29a82c455029f239614f3c1eaa42dcd16f9b313..a5977e7d965d236e9954e3801de4e1451dc9813f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/simulated_time.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/simulated_time.cc
>@@ -144,7 +144,7 @@ TransportPacketsFeedback SimulatedSender::PullFeedbackReport(
>         }
>       }
> 
>-      data_in_flight_ -= feedback.sent_packet->size;
>+      data_in_flight_ -= feedback.sent_packet.size;
>       report.packet_feedbacks.push_back(feedback);
>     }
>   }
>@@ -252,11 +252,12 @@ SimulatedTimeClient::SimulatedTimeClient(
>       return_link_(return_link),
>       sender_(send_link.front(), send_receiver_id),
>       feedback_(config, return_receiver_id, return_link.front()) {
>+  current_contraints_.at_time = at_time;
>+  current_contraints_.starting_rate = config.transport.rates.start_rate;
>+  current_contraints_.min_data_rate = config.transport.rates.min_rate;
>+  current_contraints_.max_data_rate = config.transport.rates.max_rate;
>   NetworkControllerConfig initial_config;
>-  initial_config.constraints.at_time = at_time;
>-  initial_config.constraints.starting_rate = config.transport.rates.start_rate;
>-  initial_config.constraints.min_data_rate = config.transport.rates.min_rate;
>-  initial_config.constraints.max_data_rate = config.transport.rates.max_rate;
>+  initial_config.constraints = current_contraints_;
>   congestion_controller_ = network_controller_factory_.Create(initial_config);
>   for (auto& stream_config : stream_configs)
>     packet_streams_.emplace_back(new PacketStream(stream_config));
>@@ -283,9 +284,9 @@ bool SimulatedTimeClient::TryDeliverPacket(rtc::CopyOnWriteBuffer raw_buffer,
>   for (PacketResult& feedback : report.packet_feedbacks) {
>     if (packet_log_)
>       fprintf(packet_log_, "%" PRId64 " %" PRId64 " %.3lf %.3lf %.3lf\n",
>-              feedback.sent_packet->sequence_number,
>-              feedback.sent_packet->size.bytes(),
>-              feedback.sent_packet->send_time.seconds<double>(),
>+              feedback.sent_packet.sequence_number,
>+              feedback.sent_packet.size.bytes(),
>+              feedback.sent_packet.send_time.seconds<double>(),
>               feedback.receive_time.seconds<double>(),
>               at_time.seconds<double>());
>   }
>@@ -307,6 +308,7 @@ void SimulatedTimeClient::Update(NetworkControlUpdate update) {
>     DataRate rate_per_stream =
>         update.target_rate->target_rate * ratio_per_stream;
>     target_rate_ = update.target_rate->target_rate;
>+    link_capacity_ = update.target_rate->network_estimate.bandwidth;
>     for (auto& stream : packet_streams_)
>       stream->OnTargetRateUpdate(rate_per_stream);
>   }
>@@ -336,10 +338,22 @@ void SimulatedTimeClient::ProcessFrames(Timestamp at_time) {
>   }
> }
> 
>+void SimulatedTimeClient::TriggerFakeReroute(Timestamp at_time) {
>+  NetworkRouteChange msg;
>+  msg.at_time = at_time;
>+  msg.constraints = current_contraints_;
>+  msg.constraints.at_time = at_time;
>+  Update(congestion_controller_->OnNetworkRouteChange(msg));
>+}
>+
> TimeDelta SimulatedTimeClient::GetNetworkControllerProcessInterval() const {
>   return network_controller_factory_.GetProcessInterval();
> }
> 
>+DataRate SimulatedTimeClient::link_capacity() const {
>+  return link_capacity_;
>+}
>+
> double SimulatedTimeClient::target_rate_kbps() const {
>   return target_rate_.kbps<double>();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/simulated_time.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/simulated_time.h
>index b9c6de124d41f5d53abe92819b5ea65dd412bd4b..79fdf845736d26a442f17e1b62c680289e30d58a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/simulated_time.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/simulated_time.h
>@@ -129,8 +129,10 @@ class SimulatedTimeClient : NetworkReceiverInterface {
>   void CongestionProcess(Timestamp at_time);
>   void PacerProcess(Timestamp at_time);
>   void ProcessFrames(Timestamp at_time);
>+  void TriggerFakeReroute(Timestamp at_time);
>   TimeDelta GetNetworkControllerProcessInterval() const;
>   double target_rate_kbps() const;
>+  DataRate link_capacity() const;
> 
>   bool TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
>                         uint64_t receiver,
>@@ -144,7 +146,9 @@ class SimulatedTimeClient : NetworkReceiverInterface {
>   std::vector<NetworkNode*> return_link_;
>   SimulatedSender sender_;
>   SimulatedFeedback feedback_;
>+  TargetRateConstraints current_contraints_;
>   DataRate target_rate_ = DataRate::Infinity();
>+  DataRate link_capacity_ = DataRate::Infinity();
>   FILE* packet_log_ = nullptr;
> 
>   std::vector<std::unique_ptr<PacketStream>> packet_streams_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/video_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/video_stream.cc
>index b85034ffb20223302b69232358aeb3597bdfd03c..36399234adcaf98d2ab6c46818614b3f5dd6c1cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/video_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/video_stream.cc
>@@ -12,13 +12,14 @@
> #include <algorithm>
> #include <utility>
> 
>+#include "api/test/video/function_video_encoder_factory.h"
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "media/base/mediaconstants.h"
> #include "media/engine/internaldecoderfactory.h"
> #include "media/engine/internalencoderfactory.h"
> #include "media/engine/webrtcvideoengine.h"
> #include "test/call_test.h"
> #include "test/fake_encoder.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/scenario/hardware_codecs.h"
> #include "test/testsupport/fileutils.h"
> 
>@@ -131,11 +132,20 @@ VideoEncoderConfig CreateVideoEncoderConfig(VideoStreamConfig config) {
>   size_t num_streams = config.encoder.num_simulcast_streams;
>   VideoEncoderConfig encoder_config;
>   encoder_config.codec_type = config.encoder.codec;
>-  encoder_config.content_type = VideoEncoderConfig::ContentType::kRealtimeVideo;
>+  switch (config.source.content_type) {
>+    case VideoStreamConfig::Source::ContentType::kVideo:
>+      encoder_config.content_type =
>+          VideoEncoderConfig::ContentType::kRealtimeVideo;
>+      break;
>+    case VideoStreamConfig::Source::ContentType::kScreen:
>+      encoder_config.content_type = VideoEncoderConfig::ContentType::kScreen;
>+      break;
>+  }
>   encoder_config.video_format =
>       SdpVideoFormat(CodecTypeToPayloadString(config.encoder.codec), {});
>   encoder_config.number_of_streams = num_streams;
>   encoder_config.simulcast_layers = std::vector<VideoStream>(num_streams);
>+  encoder_config.min_transmit_bitrate_bps = config.stream.pad_to_rate.bps();
> 
>   std::string cricket_codec = CodecTypeToCodecName(config.encoder.codec);
>   if (!cricket_codec.empty()) {
>@@ -153,6 +163,12 @@ VideoEncoderConfig CreateVideoEncoderConfig(VideoStreamConfig config) {
>   }
>   encoder_config.encoder_specific_settings =
>       CreateEncoderSpecificSettings(config);
>+  if (config.encoder.max_framerate) {
>+    for (auto& layer : encoder_config.simulcast_layers) {
>+      layer.max_framerate = *config.encoder.max_framerate;
>+    }
>+  }
>+
>   return encoder_config;
> }
> }  // namespace
>@@ -193,11 +209,13 @@ SendVideoStream::SendVideoStream(CallClient* sender,
>     case Encoder::Implementation::kFake:
>       if (config.encoder.codec == Codec::kVideoCodecGeneric) {
>         encoder_factory_ =
>-            absl::make_unique<FunctionVideoEncoderFactory>([this, config]() {
>+            absl::make_unique<FunctionVideoEncoderFactory>([this]() {
>+              rtc::CritScope cs(&crit_);
>               auto encoder =
>                   absl::make_unique<test::FakeEncoder>(sender_->clock_);
>-              if (config.encoder.fake.max_rate.IsFinite())
>-                encoder->SetMaxBitrate(config.encoder.fake.max_rate.kbps());
>+              fake_encoders_.push_back(encoder.get());
>+              if (config_.encoder.fake.max_rate.IsFinite())
>+                encoder->SetMaxBitrate(config_.encoder.fake.max_rate.kbps());
>               return encoder;
>             });
>       } else {
>@@ -213,9 +231,15 @@ SendVideoStream::SendVideoStream(CallClient* sender,
>   }
>   RTC_CHECK(encoder_factory_);
> 
>+  bitrate_allocator_factory_ = CreateBuiltinVideoBitrateAllocatorFactory();
>+  RTC_CHECK(bitrate_allocator_factory_);
>+
>   VideoSendStream::Config send_config =
>       CreateVideoSendStreamConfig(config, ssrcs_, send_transport);
>   send_config.encoder_settings.encoder_factory = encoder_factory_.get();
>+  send_config.encoder_settings.bitrate_allocator_factory =
>+      bitrate_allocator_factory_.get();
>+
>   VideoEncoderConfig encoder_config = CreateVideoEncoderConfig(config);
> 
>   send_stream_ = sender_->call_->CreateVideoSendStream(
>@@ -232,29 +256,34 @@ SendVideoStream::~SendVideoStream() {
> void SendVideoStream::Start() {
>   send_stream_->Start();
>   video_capturer_->Start();
>+  sender_->call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
> }
> 
>-bool SendVideoStream::TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
>-                                       uint64_t receiver,
>-                                       Timestamp at_time) {
>-  // Removes added overhead before delivering RTCP packet to sender.
>-  RTC_DCHECK_GE(packet.size(), config_.stream.packet_overhead.bytes());
>-  packet.SetSize(packet.size() - config_.stream.packet_overhead.bytes());
>-  sender_->DeliverPacket(MediaType::VIDEO, packet, at_time);
>-  return true;
>+void SendVideoStream::UpdateConfig(
>+    std::function<void(VideoStreamConfig*)> modifier) {
>+  rtc::CritScope cs(&crit_);
>+  VideoStreamConfig prior_config = config_;
>+  modifier(&config_);
>+  if (prior_config.encoder.fake.max_rate != config_.encoder.fake.max_rate) {
>+    for (auto* encoder : fake_encoders_) {
>+      encoder->SetMaxBitrate(config_.encoder.fake.max_rate.kbps());
>+    }
>+  }
>+  // TODO(srte): Add more conditions that should cause reconfiguration.
>+  if (prior_config.encoder.max_framerate != config_.encoder.max_framerate) {
>+    VideoEncoderConfig encoder_config = CreateVideoEncoderConfig(config_);
>+    send_stream_->ReconfigureVideoEncoder(std::move(encoder_config));
>+  }
>+  if (prior_config.source.framerate != config_.source.framerate) {
>+    SetCaptureFramerate(config_.source.framerate);
>+  }
> }
> 
> void SendVideoStream::SetCaptureFramerate(int framerate) {
>   RTC_CHECK(frame_generator_)
>       << "Framerate change only implemented for generators";
>   frame_generator_->ChangeFramerate(framerate);
>-}
> 
>-void SendVideoStream::SetMaxFramerate(absl::optional<int> max_framerate) {
>-  VideoEncoderConfig encoder_config = CreateVideoEncoderConfig(config_);
>-  RTC_DCHECK_EQ(encoder_config.simulcast_layers.size(), 1);
>-  encoder_config.simulcast_layers[0].max_framerate = max_framerate.value_or(-1);
>-  send_stream_->ReconfigureVideoEncoder(std::move(encoder_config));
> }
> 
> VideoSendStream::Stats SendVideoStream::GetStats() const {
>@@ -284,15 +313,14 @@ ReceiveVideoStream::ReceiveVideoStream(CallClient* receiver,
>                                        SendVideoStream* send_stream,
>                                        size_t chosen_stream,
>                                        Transport* feedback_transport)
>-    : receiver_(receiver),
>-      config_(config),
>-      decoder_factory_(absl::make_unique<InternalDecoderFactory>()) {
>+    : receiver_(receiver), config_(config) {
>   renderer_ = absl::make_unique<FakeVideoRenderer>();
>   VideoReceiveStream::Config recv_config(feedback_transport);
>   recv_config.rtp.remb = !config.stream.packet_feedback;
>   recv_config.rtp.transport_cc = config.stream.packet_feedback;
>   recv_config.rtp.local_ssrc = CallTest::kReceiverLocalVideoSsrc;
>   recv_config.rtp.extensions = GetVideoRtpExtensions(config);
>+  receiver_->AddExtensions(recv_config.rtp.extensions);
>   RTC_DCHECK(!config.stream.use_rtx ||
>              config.stream.nack_history_time > TimeDelta::Zero());
>   recv_config.rtp.nack.rtp_history_ms = config.stream.nack_history_time.ms();
>@@ -300,14 +328,24 @@ ReceiveVideoStream::ReceiveVideoStream(CallClient* receiver,
>   recv_config.renderer = renderer_.get();
>   if (config.stream.use_rtx) {
>     recv_config.rtp.rtx_ssrc = send_stream->rtx_ssrcs_[chosen_stream];
>+    receiver->ssrc_media_types_[recv_config.rtp.rtx_ssrc] = MediaType::VIDEO;
>     recv_config.rtp
>         .rtx_associated_payload_types[CallTest::kSendRtxPayloadType] =
>         CodecTypeToPayloadType(config.encoder.codec);
>   }
>   recv_config.rtp.remote_ssrc = send_stream->ssrcs_[chosen_stream];
>+  receiver->ssrc_media_types_[recv_config.rtp.remote_ssrc] = MediaType::VIDEO;
>+
>   VideoReceiveStream::Decoder decoder =
>       CreateMatchingDecoder(CodecTypeToPayloadType(config.encoder.codec),
>                             CodecTypeToPayloadString(config.encoder.codec));
>+  if (config.encoder.codec ==
>+      VideoStreamConfig::Encoder::Codec::kVideoCodecGeneric) {
>+    decoder_factory_ = absl::make_unique<FunctionVideoDecoderFactory>(
>+        []() { return absl::make_unique<FakeDecoder>(); });
>+  } else {
>+    decoder_factory_ = absl::make_unique<InternalDecoderFactory>();
>+  }
>   decoder.decoder_factory = decoder_factory_.get();
>   recv_config.decoders.push_back(decoder);
> 
>@@ -316,6 +354,7 @@ ReceiveVideoStream::ReceiveVideoStream(CallClient* receiver,
>     FlexfecReceiveStream::Config flexfec_config(feedback_transport);
>     flexfec_config.payload_type = CallTest::kFlexfecPayloadType;
>     flexfec_config.remote_ssrc = CallTest::kFlexfecSendSsrc;
>+    receiver->ssrc_media_types_[flexfec_config.remote_ssrc] = MediaType::VIDEO;
>     flexfec_config.protected_media_ssrcs = send_stream->rtx_ssrcs_;
>     flexfec_config.local_ssrc = recv_config.rtp.local_ssrc;
>     flecfec_stream_ =
>@@ -337,46 +376,23 @@ ReceiveVideoStream::~ReceiveVideoStream() {
>     receiver_->call_->DestroyFlexfecReceiveStream(flecfec_stream_);
> }
> 
>-bool ReceiveVideoStream::TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
>-                                          uint64_t receiver,
>-                                          Timestamp at_time) {
>-  RTC_DCHECK_GE(packet.size(), config_.stream.packet_overhead.bytes());
>-  packet.SetSize(packet.size() - config_.stream.packet_overhead.bytes());
>-  receiver_->DeliverPacket(MediaType::VIDEO, packet, at_time);
>-  return true;
>+void ReceiveVideoStream::Start() {
>+  receive_stream_->Start();
>+  receiver_->call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
> }
> 
> VideoStreamPair::~VideoStreamPair() = default;
> 
> VideoStreamPair::VideoStreamPair(CallClient* sender,
>-                                 std::vector<NetworkNode*> send_link,
>-                                 uint64_t send_receiver_id,
>                                  CallClient* receiver,
>-                                 std::vector<NetworkNode*> return_link,
>-                                 uint64_t return_receiver_id,
>                                  VideoStreamConfig config)
>     : config_(config),
>-      send_link_(send_link),
>-      return_link_(return_link),
>-      send_transport_(sender,
>-                      send_link.front(),
>-                      send_receiver_id,
>-                      config.stream.packet_overhead),
>-      return_transport_(receiver,
>-                        return_link.front(),
>-                        return_receiver_id,
>-                        config.stream.packet_overhead),
>-      send_stream_(sender, config, &send_transport_),
>+      send_stream_(sender, config, &sender->transport_),
>       receive_stream_(receiver,
>                       config,
>                       &send_stream_,
>                       /*chosen_stream=*/0,
>-                      &return_transport_) {
>-  NetworkNode::Route(send_transport_.ReceiverId(), send_link_,
>-                     &receive_stream_);
>-  NetworkNode::Route(return_transport_.ReceiverId(), return_link_,
>-                     &send_stream_);
>-}
>+                      &receiver->transport_) {}
> 
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/video_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/video_stream.h
>index e6696183a7ebd503cfee0527e78cac048ec9b601..ebf9c951b706923596518058068dc64bebc3bd1a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/video_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/video_stream.h
>@@ -14,6 +14,7 @@
> #include <vector>
> 
> #include "rtc_base/constructormagic.h"
>+#include "test/fake_encoder.h"
> #include "test/frame_generator_capturer.h"
> #include "test/scenario/call_client.h"
> #include "test/scenario/column_printer.h"
>@@ -25,15 +26,15 @@ namespace webrtc {
> namespace test {
> // SendVideoStream provides an interface for changing parameters and retrieving
> // states at run time.
>-class SendVideoStream : public NetworkReceiverInterface {
>+class SendVideoStream {
>  public:
>   RTC_DISALLOW_COPY_AND_ASSIGN(SendVideoStream);
>   ~SendVideoStream();
>   void SetCaptureFramerate(int framerate);
>-  void SetMaxFramerate(absl::optional<int> max_framerate);
>   VideoSendStream::Stats GetStats() const;
>   ColumnPrinter StatsPrinter();
>   void Start();
>+  void UpdateConfig(std::function<void(VideoStreamConfig*)> modifier);
> 
>  private:
>   friend class Scenario;
>@@ -43,25 +44,28 @@ class SendVideoStream : public NetworkReceiverInterface {
>   SendVideoStream(CallClient* sender,
>                   VideoStreamConfig config,
>                   Transport* send_transport);
>-  bool TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
>-                        uint64_t receiver,
>-                        Timestamp at_time) override;
> 
>+  rtc::CriticalSection crit_;
>   std::vector<uint32_t> ssrcs_;
>   std::vector<uint32_t> rtx_ssrcs_;
>   VideoSendStream* send_stream_ = nullptr;
>   CallClient* const sender_;
>-  const VideoStreamConfig config_;
>+  VideoStreamConfig config_ RTC_GUARDED_BY(crit_);
>   std::unique_ptr<VideoEncoderFactory> encoder_factory_;
>+  std::vector<test::FakeEncoder*> fake_encoders_ RTC_GUARDED_BY(crit_);
>+  std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
>   std::unique_ptr<TestVideoCapturer> video_capturer_;
>   FrameGeneratorCapturer* frame_generator_ = nullptr;
>+  int next_local_network_id_ = 0;
>+  int next_remote_network_id_ = 0;
> };
> 
> // ReceiveVideoStream represents a video receiver. It can't be used directly.
>-class ReceiveVideoStream : public NetworkReceiverInterface {
>+class ReceiveVideoStream {
>  public:
>   RTC_DISALLOW_COPY_AND_ASSIGN(ReceiveVideoStream);
>   ~ReceiveVideoStream();
>+  void Start();
> 
>  private:
>   friend class Scenario;
>@@ -71,9 +75,7 @@ class ReceiveVideoStream : public NetworkReceiverInterface {
>                      SendVideoStream* send_stream,
>                      size_t chosen_stream,
>                      Transport* feedback_transport);
>-  bool TryDeliverPacket(rtc::CopyOnWriteBuffer packet,
>-                        uint64_t receiver,
>-                        Timestamp at_time) override;
>+
>   VideoReceiveStream* receive_stream_ = nullptr;
>   FlexfecReceiveStream* flecfec_stream_ = nullptr;
>   std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> renderer_;
>@@ -95,18 +97,10 @@ class VideoStreamPair {
>  private:
>   friend class Scenario;
>   VideoStreamPair(CallClient* sender,
>-                  std::vector<NetworkNode*> send_link,
>-                  uint64_t send_receiver_id,
>                   CallClient* receiver,
>-                  std::vector<NetworkNode*> return_link,
>-                  uint64_t return_receiver_id,
>                   VideoStreamConfig config);
> 
>   const VideoStreamConfig config_;
>-  std::vector<NetworkNode*> send_link_;
>-  std::vector<NetworkNode*> return_link_;
>-  NetworkNodeTransport send_transport_;
>-  NetworkNodeTransport return_transport_;
> 
>   SendVideoStream send_stream_;
>   ReceiveVideoStream receive_stream_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue.cc
>index ad7a50327b900bf250ab231381437ef6c52a9288..5c35c2e43b0939f916c27843141684dc2985e434 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue.cc
>@@ -32,10 +32,7 @@ SingleThreadedTaskQueueForTesting::QueuedTask::~QueuedTask() = default;
> 
> SingleThreadedTaskQueueForTesting::SingleThreadedTaskQueueForTesting(
>     const char* name)
>-    : thread_(Run, this, name),
>-      running_(true),
>-      next_task_id_(0),
>-      wake_up_(false, false) {
>+    : thread_(Run, this, name), running_(true), next_task_id_(0) {
>   thread_.Start();
> }
> 
>@@ -83,7 +80,7 @@ SingleThreadedTaskQueueForTesting::PostDelayedTask(Task task,
> }
> 
> void SingleThreadedTaskQueueForTesting::SendTask(Task task) {
>-  rtc::Event done(true, false);
>+  rtc::Event done;
>   PostTask([&task, &done]() {
>     task();
>     done.Set();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue_unittest.cc
>index 084f0d7e8cbb8212cfe2716532066bc95cd310c3..04961650c760e14ab2cb407868bfa2edeae036b3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue_unittest.cc
>@@ -37,7 +37,7 @@ TEST(SingleThreadedTaskQueueForTestingTest, ExecutesPostedTasks) {
>   SingleThreadedTaskQueueForTesting task_queue("task_queue");
> 
>   std::atomic<bool> executed(false);
>-  rtc::Event done(true, false);
>+  rtc::Event done;
> 
>   task_queue.PostTask([&executed, &done]() {
>     executed.store(true);
>@@ -60,14 +60,14 @@ TEST(SingleThreadedTaskQueueForTestingTest,
> 
>   std::vector<std::unique_ptr<rtc::Event>> done_events;
>   for (size_t i = 0; i < kCount; i++) {
>-    done_events.emplace_back(absl::make_unique<rtc::Event>(false, false));
>+    done_events.emplace_back(absl::make_unique<rtc::Event>());
>   }
> 
>   // To avoid the tasks which comprise the actual test from running before they
>   // have all be posted, which could result in only one task ever being in the
>   // queue at any given time, post one waiting task that would block the
>   // task-queue, and unblock only after all tasks have been posted.
>-  rtc::Event rendezvous(true, false);
>+  rtc::Event rendezvous;
>   task_queue.PostTask(
>       [&rendezvous]() { ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs)); });
> 
>@@ -95,7 +95,7 @@ TEST(SingleThreadedTaskQueueForTestingTest, PostToTaskQueueFromOwnThread) {
>   SingleThreadedTaskQueueForTesting task_queue("task_queue");
> 
>   std::atomic<bool> executed(false);
>-  rtc::Event done(true, false);
>+  rtc::Event done;
> 
>   auto internally_posted_task = [&executed, &done]() {
>     executed.store(true);
>@@ -124,7 +124,7 @@ TEST(SingleThreadedTaskQueueForTestingTest, TasksExecutedInSequence) {
> 
>   // Prevent the chain from being set in motion before we've had time to
>   // schedule it all, lest the queue only contain one task at a time.
>-  rtc::Event rendezvous(true, false);
>+  rtc::Event rendezvous;
>   task_queue.PostTask(
>       [&rendezvous]() { ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs)); });
> 
>@@ -136,7 +136,7 @@ TEST(SingleThreadedTaskQueueForTestingTest, TasksExecutedInSequence) {
>   }
> 
>   // The test will wait for the task-queue to finish.
>-  rtc::Event done(true, false);
>+  rtc::Event done;
>   task_queue.PostTask([&done]() { done.Set(); });
> 
>   rendezvous.Set();  // Set the chain in motion.
>@@ -150,7 +150,7 @@ TEST(SingleThreadedTaskQueueForTestingTest, ExecutesPostedDelayedTask) {
>   SingleThreadedTaskQueueForTesting task_queue("task_queue");
> 
>   std::atomic<bool> executed(false);
>-  rtc::Event done(true, false);
>+  rtc::Event done;
> 
>   constexpr int64_t delay_ms = 20;
>   static_assert(delay_ms < kMaxWaitTimeMs / 2, "Delay too long for tests.");
>@@ -177,7 +177,7 @@ TEST(SingleThreadedTaskQueueForTestingTest, DoesNotExecuteDelayedTaskTooSoon) {
>   task_queue.PostDelayedTask([&executed]() { executed.store(true); }, delay_ms);
> 
>   // Wait less than is enough, make sure the task was not yet executed.
>-  rtc::Event not_done(true, false);
>+  rtc::Event not_done;
>   ASSERT_FALSE(not_done.Wait(delay_ms / 2));
>   EXPECT_FALSE(executed.load());
> }
>@@ -195,7 +195,7 @@ TEST(SingleThreadedTaskQueueForTestingTest,
>   static_assert(earlier_delay_ms + later_delay_ms < kMaxWaitTimeMs / 2,
>                 "Delay too long for tests.");
> 
>-  rtc::Event done(true, false);
>+  rtc::Event done;
> 
>   auto earlier_task = [&earlier_executed, &later_executed]() {
>     EXPECT_FALSE(later_executed.load());
>@@ -229,7 +229,7 @@ TEST(SingleThreadedTaskQueueForTestingTest,
>   static_assert(earlier_delay_ms + later_delay_ms < kMaxWaitTimeMs / 2,
>                 "Delay too long for tests.");
> 
>-  rtc::Event done(true, false);
>+  rtc::Event done;
> 
>   auto earlier_task = [&earlier_executed, &later_executed]() {
>     EXPECT_FALSE(later_executed.load());
>@@ -253,11 +253,11 @@ TEST(SingleThreadedTaskQueueForTestingTest,
> TEST(SingleThreadedTaskQueueForTestingTest, ExternalThreadCancelsTask) {
>   SingleThreadedTaskQueueForTesting task_queue("task_queue");
> 
>-  rtc::Event done(true, false);
>+  rtc::Event done;
> 
>   // Prevent the to-be-cancelled task from being executed before we've had
>   // time to cancel it.
>-  rtc::Event rendezvous(true, false);
>+  rtc::Event rendezvous;
>   task_queue.PostTask(
>       [&rendezvous]() { ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs)); });
> 
>@@ -279,10 +279,10 @@ TEST(SingleThreadedTaskQueueForTestingTest, ExternalThreadCancelsTask) {
> TEST(SingleThreadedTaskQueueForTestingTest, InternalThreadCancelsTask) {
>   SingleThreadedTaskQueueForTesting task_queue("task_queue");
> 
>-  rtc::Event done(true, false);
>+  rtc::Event done;
> 
>   // Prevent the chain from being set-off before we've set everything up.
>-  rtc::Event rendezvous(true, false);
>+  rtc::Event rendezvous;
>   task_queue.PostTask(
>       [&rendezvous]() { ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs)); });
> 
>@@ -316,7 +316,7 @@ TEST(SingleThreadedTaskQueueForTestingTest, SendTask) {
>   task_queue.SendTask([&executed]() {
>     // Intentionally delay, so that if SendTask didn't block, the sender thread
>     // would have time to read |executed|.
>-    rtc::Event delay(true, false);
>+    rtc::Event delay;
>     ASSERT_FALSE(delay.Wait(1000));
>     executed.store(true);
>   });
>@@ -335,7 +335,7 @@ TEST(SingleThreadedTaskQueueForTestingTest,
>   for (size_t i = 0; i < tasks; i++) {
>     task_queue->PostTask([&counter]() {
>       std::atomic_fetch_add(&counter, static_cast<size_t>(1));
>-      rtc::Event delay(true, false);
>+      rtc::Event delay;
>       ASSERT_FALSE(delay.Wait(500));
>     });
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main.cc
>index c8d0b1b584e6a75c95f076a6df75a7c309e065cf..1a311f63c279cc9b8829decfa9b5ccd42901af90 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main.cc
>@@ -8,129 +8,15 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <fstream>
>+#include <memory>
> 
>-#include "rtc_base/flags.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/thread.h"
>-#include "system_wrappers/include/field_trial.h"
>-#include "system_wrappers/include/metrics.h"
>-#include "test/field_trial.h"
>-#include "test/gmock.h"
>-#include "test/gtest.h"
>-#include "test/testsupport/fileutils.h"
>-#include "test/testsupport/perf_test.h"
>-
>-#if defined(WEBRTC_WIN)
>-#include "rtc_base/win32socketinit.h"
>-#endif
>-
>-#if defined(WEBRTC_IOS)
>-#include "test/ios/test_support.h"
>-
>-DEFINE_string(NSTreatUnknownArgumentsAsOpen,
>-              "",
>-              "Intentionally ignored flag intended for iOS simulator.");
>-DEFINE_string(ApplePersistenceIgnoreState,
>-              "",
>-              "Intentionally ignored flag intended for iOS simulator.");
>-DEFINE_bool(
>-    save_chartjson_result,
>-    false,
>-    "Store the perf results in Documents/perf_result.json in the format "
>-    "described by "
>-    "https://github.com/catapult-project/catapult/blob/master/dashboard/docs/"
>-    "data-format.md.");
>-
>-#else
>-
>-DEFINE_string(
>-    isolated_script_test_output,
>-    "",
>-    "Path to output an empty JSON file which Chromium infra requires.");
>-
>-DEFINE_string(
>-    isolated_script_test_perf_output,
>-    "",
>-    "Path where the perf results should be stored in the JSON format described "
>-    "by "
>-    "https://github.com/catapult-project/catapult/blob/master/dashboard/docs/"
>-    "data-format.md.");
>-
>-#endif
>-
>-DEFINE_bool(logs, false, "print logs to stderr");
>-
>-DEFINE_string(
>-    force_fieldtrials,
>-    "",
>-    "Field trials control experimental feature code which can be forced. "
>-    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
>-    " will assign the group Enable to field trial WebRTC-FooFeature.");
>-
>-DEFINE_bool(help, false, "Print this message.");
>+#include "test/test_main_lib.h"
> 
> int main(int argc, char* argv[]) {
>-  ::testing::InitGoogleMock(&argc, argv);
>-
>-  // Default to LS_INFO, even for release builds to provide better test logging.
>-  // TODO(pbos): Consider adding a command-line override.
>-  if (rtc::LogMessage::GetLogToDebug() > rtc::LS_INFO)
>-    rtc::LogMessage::LogToDebug(rtc::LS_INFO);
>-
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, false)) {
>-    return 1;
>-  }
>-  if (FLAG_help) {
>-    rtc::FlagList::Print(nullptr, false);
>-    return 0;
>+  std::unique_ptr<webrtc::TestMain> main = webrtc::TestMain::Create();
>+  int err_code = main->Init(&argc, argv);
>+  if (err_code != 0) {
>+    return err_code;
>   }
>-
>-  webrtc::test::SetExecutablePath(argv[0]);
>-  webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
>-  // InitFieldTrialsFromString stores the char*, so the char array must outlive
>-  // the application.
>-  webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
>-  webrtc::metrics::Enable();
>-
>-#if defined(WEBRTC_WIN)
>-  rtc::WinsockInitializer winsock_init;
>-#endif
>-
>-  rtc::LogMessage::SetLogToStderr(FLAG_logs);
>-
>-  // Ensure that main thread gets wrapped as an rtc::Thread.
>-  // TODO(bugs.webrt.org/9714): It might be better to avoid wrapping the main
>-  // thread, or leave it to individual tests that need it. But as long as we
>-  // have automatic thread wrapping, we need this to avoid that some other
>-  // random thread (which one depending on which tests are run) gets
>-  // automatically wrapped.
>-  rtc::ThreadManager::Instance()->WrapCurrentThread();
>-  RTC_CHECK(rtc::Thread::Current());
>-
>-#if defined(WEBRTC_IOS)
>-
>-  rtc::test::InitTestSuite(RUN_ALL_TESTS, argc, argv,
>-                           FLAG_save_chartjson_result);
>-  rtc::test::RunTestsFromIOSApp();
>-
>-#else
>-
>-  int exit_code = RUN_ALL_TESTS();
>-
>-  std::string chartjson_result_file = FLAG_isolated_script_test_perf_output;
>-  if (!chartjson_result_file.empty()) {
>-    webrtc::test::WritePerfResults(chartjson_result_file);
>-  }
>-
>-  std::string result_filename = FLAG_isolated_script_test_output;
>-  if (!result_filename.empty()) {
>-    std::ofstream result_file(result_filename);
>-    result_file << "{\"version\": 3}";
>-    result_file.close();
>-  }
>-
>-  return exit_code;
>-
>-#endif
>+  return main->Run(argc, argv);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main_lib.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main_lib.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..95144c98fcb9347a07d64fe7117895f6135abe85
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main_lib.cc
>@@ -0,0 +1,177 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "test/test_main_lib.h"
>+
>+#include <fstream>
>+#include <string>
>+
>+#include "absl/memory/memory.h"
>+#include "rtc_base/flags.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/thread.h"
>+#include "system_wrappers/include/field_trial.h"
>+#include "system_wrappers/include/metrics.h"
>+#include "test/field_trial.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+#include "test/testsupport/perf_test.h"
>+
>+#if defined(WEBRTC_WIN)
>+#include "rtc_base/win32socketinit.h"
>+#endif
>+
>+#if defined(WEBRTC_IOS)
>+#include "test/ios/test_support.h"
>+
>+WEBRTC_DEFINE_string(NSTreatUnknownArgumentsAsOpen,
>+                     "",
>+                     "Intentionally ignored flag intended for iOS simulator.");
>+WEBRTC_DEFINE_string(ApplePersistenceIgnoreState,
>+                     "",
>+                     "Intentionally ignored flag intended for iOS simulator.");
>+WEBRTC_DEFINE_bool(
>+    save_chartjson_result,
>+    false,
>+    "Store the perf results in Documents/perf_result.json in the format "
>+    "described by "
>+    "https://github.com/catapult-project/catapult/blob/master/dashboard/docs/"
>+    "data-format.md.");
>+
>+#else
>+
>+WEBRTC_DEFINE_string(
>+    isolated_script_test_output,
>+    "",
>+    "Path to output an empty JSON file which Chromium infra requires.");
>+
>+WEBRTC_DEFINE_string(
>+    isolated_script_test_perf_output,
>+    "",
>+    "Path where the perf results should be stored in the JSON format described "
>+    "by "
>+    "https://github.com/catapult-project/catapult/blob/master/dashboard/docs/"
>+    "data-format.md.");
>+
>+#endif
>+
>+WEBRTC_DEFINE_bool(logs, false, "print logs to stderr");
>+
>+WEBRTC_DEFINE_string(
>+    force_fieldtrials,
>+    "",
>+    "Field trials control experimental feature code which can be forced. "
>+    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
>+    " will assign the group Enable to field trial WebRTC-FooFeature.");
>+
>+WEBRTC_DEFINE_bool(help, false, "Print this message.");
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+class TestMainImpl : public TestMain {
>+ public:
>+  int Init(int* argc, char* argv[]) override {
>+    ::testing::InitGoogleMock(argc, argv);
>+
>+    // Default to LS_INFO, even for release builds to provide better test
>+    // logging.
>+    // TODO(pbos): Consider adding a command-line override.
>+    if (rtc::LogMessage::GetLogToDebug() > rtc::LS_INFO)
>+      rtc::LogMessage::LogToDebug(rtc::LS_INFO);
>+
>+    if (rtc::FlagList::SetFlagsFromCommandLine(argc, argv, false)) {
>+      return 1;
>+    }
>+    if (FLAG_help) {
>+      rtc::FlagList::Print(nullptr, false);
>+      return 0;
>+    }
>+
>+    // TODO(bugs.webrtc.org/9792): we need to reference something from
>+    // fileutils.h so that our downstream hack where we replace fileutils.cc
>+    // works. Otherwise the downstream flag implementation will take over and
>+    // botch the flag introduced by the hack. Remove this awful thing once the
>+    // downstream implementation has been eliminated.
>+    (void)webrtc::test::JoinFilename("horrible", "hack");
>+
>+    webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
>+    // InitFieldTrialsFromString stores the char*, so the char array must
>+    // outlive the application.
>+    webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
>+    webrtc::metrics::Enable();
>+
>+#if defined(WEBRTC_WIN)
>+    winsock_init_ = absl::make_unique<rtc::WinsockInitializer>();
>+#endif
>+
>+    rtc::LogMessage::SetLogToStderr(FLAG_logs);
>+
>+    // Ensure that main thread gets wrapped as an rtc::Thread.
>+    // TODO(bugs.webrt.org/9714): It might be better to avoid wrapping the main
>+    // thread, or leave it to individual tests that need it. But as long as we
>+    // have automatic thread wrapping, we need this to avoid that some other
>+    // random thread (which one depending on which tests are run) gets
>+    // automatically wrapped.
>+    rtc::ThreadManager::Instance()->WrapCurrentThread();
>+    RTC_CHECK(rtc::Thread::Current());
>+    return 0;
>+  }
>+
>+  int Run(int argc, char* argv[]) override {
>+#if defined(WEBRTC_IOS)
>+    rtc::test::InitTestSuite(RUN_ALL_TESTS, argc, argv,
>+                             FLAG_save_chartjson_result);
>+    rtc::test::RunTestsFromIOSApp();
>+    return 0;
>+#else
>+    int exit_code = RUN_ALL_TESTS();
>+
>+    std::string chartjson_result_file = FLAG_isolated_script_test_perf_output;
>+    if (!chartjson_result_file.empty()) {
>+      webrtc::test::WritePerfResults(chartjson_result_file);
>+    }
>+
>+    std::string result_filename = FLAG_isolated_script_test_output;
>+    if (!result_filename.empty()) {
>+      std::ofstream result_file(result_filename);
>+      result_file << "{\"version\": 3}";
>+      result_file.close();
>+    }
>+
>+#if defined(ADDRESS_SANITIZER) || defined(LEAK_SANITIZER) ||  \
>+    defined(MEMORY_SANITIZER) || defined(THREAD_SANITIZER) || \
>+    defined(UNDEFINED_SANITIZER)
>+    // We want the test flagged as failed only for sanitizer defects,
>+    // in which case the sanitizer will override exit code with 66.
>+    return 0;
>+#endif
>+
>+    return exit_code;
>+#endif
>+  }
>+
>+  ~TestMainImpl() override = default;
>+
>+ private:
>+#if defined(WEBRTC_WIN)
>+  std::unique_ptr<rtc::WinsockInitializer> winsock_init_;
>+#endif
>+};
>+
>+}  // namespace
>+
>+std::unique_ptr<TestMain> TestMain::Create() {
>+  return absl::make_unique<TestMainImpl>();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main_lib.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main_lib.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..44dd764ae24d3dfb863f6ec134a4d693e8dd030f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main_lib.h
>@@ -0,0 +1,38 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef TEST_TEST_MAIN_LIB_H_
>+#define TEST_TEST_MAIN_LIB_H_
>+
>+#include <memory>
>+
>+namespace webrtc {
>+
>+// Class to initialize test environment and run tests.
>+class TestMain {
>+ public:
>+  virtual ~TestMain() {}
>+
>+  static std::unique_ptr<TestMain> Create();
>+
>+  // Initializes test environment. Clients can add their own initialization
>+  // steps after call to this method and before running tests.
>+  // Returns 0 if initialization was successful and non 0 otherwise.
>+  virtual int Init(int* argc, char* argv[]) = 0;
>+
>+  // Runs test end return result error code. 0 - no errors.
>+  virtual int Run(int argc, char* argv[]) = 0;
>+
>+ protected:
>+  TestMain() = default;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // TEST_TEST_MAIN_LIB_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.cc
>index cb88472e425bf2afc6a898864c9309cbd4ebb535..a71d8f8db57d57ee94034114db4b68e0288e68ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.cc
>@@ -59,35 +59,17 @@
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/stringutils.h"
>+#include "test/testsupport/fileutils_override.h"
> 
> namespace webrtc {
> namespace test {
> 
>-namespace {
>-
> #if defined(WEBRTC_WIN)
> const char* kPathDelimiter = "\\";
> #else
> const char* kPathDelimiter = "/";
> #endif
> 
>-#if defined(WEBRTC_ANDROID)
>-// This is a special case in Chrome infrastructure. See
>-// base/test/test_support_android.cc.
>-const char* kAndroidChromiumTestsRoot = "/sdcard/chromium_tests_root/";
>-#endif
>-
>-#if !defined(WEBRTC_IOS)
>-const char* kResourcesDirName = "resources";
>-#endif
>-
>-char relative_dir_path[FILENAME_MAX];
>-bool relative_dir_path_set = false;
>-
>-}  // namespace
>-
>-const char* kCannotFindProjectRootDir = "ERROR_CANNOT_FIND_PROJECT_ROOT_DIR";
>-
> std::string DirName(const std::string& path) {
>   if (path.empty())
>     return "";
>@@ -101,27 +83,6 @@ std::string DirName(const std::string& path) {
>   return result.substr(0, result.find_last_of(kPathDelimiter));
> }
> 
>-void SetExecutablePath(const std::string& path) {
>-  std::string working_dir = WorkingDir();
>-  std::string temp_path = path;
>-
>-  // Handle absolute paths; convert them to relative paths to the working dir.
>-  if (path.find(working_dir) != std::string::npos) {
>-    temp_path = path.substr(working_dir.length() + 1);
>-  }
>-// On Windows, when tests are run under memory tools like DrMemory and TSan,
>-// slashes occur in the path as directory separators. Make sure we replace
>-// such cases with backslashes in order for the paths to be correct.
>-#ifdef WIN32
>-  std::replace(temp_path.begin(), temp_path.end(), '/', '\\');
>-#endif
>-
>-  // Trim away the executable name; only store the relative dir path.
>-  temp_path = DirName(temp_path);
>-  strncpy(relative_dir_path, temp_path.c_str(), FILENAME_MAX);
>-  relative_dir_path_set = true;
>-}
>-
> bool FileExists(const std::string& file_name) {
>   struct stat file_info = {0};
>   return stat(file_name.c_str(), &file_info) == 0;
>@@ -133,69 +94,12 @@ bool DirExists(const std::string& directory_name) {
>          S_ISDIR(directory_info.st_mode);
> }
> 
>-// Finds the WebRTC src dir.
>-// The returned path always ends with a path separator.
>-std::string ProjectRootPath() {
>-#if defined(WEBRTC_ANDROID)
>-  return kAndroidChromiumTestsRoot;
>-#elif defined WEBRTC_IOS
>-  return IOSRootPath();
>-#elif defined(WEBRTC_MAC)
>-  std::string path;
>-  GetNSExecutablePath(&path);
>-  std::string exe_dir = DirName(path);
>-  // On Mac, tests execute in out/Whatever, so src is two levels up except if
>-  // the test is bundled (which our tests are not), in which case it's 5 levels.
>-  return DirName(DirName(exe_dir)) + kPathDelimiter;
>-#elif defined(WEBRTC_POSIX)
>-  char buf[PATH_MAX];
>-  ssize_t count = ::readlink("/proc/self/exe", buf, arraysize(buf));
>-  if (count <= 0) {
>-    RTC_NOTREACHED() << "Unable to resolve /proc/self/exe.";
>-    return kCannotFindProjectRootDir;
>-  }
>-  // On POSIX, tests execute in out/Whatever, so src is two levels up.
>-  std::string exe_dir = DirName(std::string(buf, count));
>-  return DirName(DirName(exe_dir)) + kPathDelimiter;
>-#elif defined(WEBRTC_WIN)
>-  wchar_t buf[MAX_PATH];
>-  buf[0] = 0;
>-  if (GetModuleFileName(NULL, buf, MAX_PATH) == 0)
>-    return kCannotFindProjectRootDir;
>-
>-  std::string exe_path = rtc::ToUtf8(std::wstring(buf));
>-  std::string exe_dir = DirName(exe_path);
>-  return DirName(DirName(exe_dir)) + kPathDelimiter;
>-#endif
>-}
>-
> std::string OutputPath() {
>-#if defined(WEBRTC_IOS)
>-  return IOSOutputPath();
>-#elif defined(WEBRTC_ANDROID)
>-  return kAndroidChromiumTestsRoot;
>-#else
>-  std::string path = ProjectRootPath();
>-  RTC_DCHECK_NE(path, kCannotFindProjectRootDir);
>-  path += "out";
>-  if (!CreateDir(path)) {
>-    return "./";
>-  }
>-  return path + kPathDelimiter;
>-#endif
>+  return webrtc::test::internal::OutputPath();
> }
> 
> std::string WorkingDir() {
>-#if defined(WEBRTC_ANDROID)
>-  return kAndroidChromiumTestsRoot;
>-#endif
>-  char path_buffer[FILENAME_MAX];
>-  if (!GET_CURRENT_DIR(path_buffer, sizeof(path_buffer))) {
>-    fprintf(stderr, "Cannot get current directory!\n");
>-    return "./";
>-  } else {
>-    return std::string(path_buffer);
>-  }
>+  return webrtc::test::internal::WorkingDir();
> }
> 
> // Generate a temporary filename in a safe way.
>@@ -322,30 +226,7 @@ bool RemoveFile(const std::string& file_name) {
> 
> std::string ResourcePath(const std::string& name,
>                          const std::string& extension) {
>-#if defined(WEBRTC_IOS)
>-  return IOSResourcePath(name, extension);
>-#else
>-  std::string platform = "win";
>-#ifdef WEBRTC_LINUX
>-  platform = "linux";
>-#endif  // WEBRTC_LINUX
>-#ifdef WEBRTC_MAC
>-  platform = "mac";
>-#endif  // WEBRTC_MAC
>-#ifdef WEBRTC_ANDROID
>-  platform = "android";
>-#endif  // WEBRTC_ANDROID
>-
>-  std::string resources_path =
>-      ProjectRootPath() + kResourcesDirName + kPathDelimiter;
>-  std::string resource_file =
>-      resources_path + name + "_" + platform + "." + extension;
>-  if (FileExists(resource_file)) {
>-    return resource_file;
>-  }
>-  // Fall back on name without platform.
>-  return resources_path + name + "." + extension;
>-#endif  // defined (WEBRTC_IOS)
>+  return webrtc::test::internal::ResourcePath(name, extension);
> }
> 
> std::string JoinFilename(const std::string& dir, const std::string& name) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.h
>index 5ff019600ed5981ecfef25461169e975d492296a..8a186988d638086d16445be21f5fe14c1db567f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.h
>@@ -25,6 +25,9 @@ namespace test {
> // to find the project root.
> extern const char* kCannotFindProjectRootDir;
> 
>+// Slash or backslash, depending on platform. NUL-terminated string.
>+extern const char* kPathDelimiter;
>+
> // Returns the absolute path to the output directory where log files and other
> // test artifacts should be put. The output directory is generally a directory
> // named "out" at the project root. This root is assumed to be two levels above
>@@ -51,18 +54,8 @@ std::string TempFilename(const std::string& dir, const std::string& prefix);
> std::string GenerateTempFilename(const std::string& dir,
>                                  const std::string& prefix);
> 
>-// Returns a path to a resource file for the currently executing platform.
>-// Adapts to what filenames are currently present in the
>-// [project-root]/resources/ dir.
>-// Returns an absolute path according to this priority list (the directory
>-// part of the path is left out for readability):
>-// 1. [name]_[platform]_[architecture].[extension]
>-// 2. [name]_[platform].[extension]
>-// 3. [name]_[architecture].[extension]
>-// 4. [name].[extension]
>-// Where
>-// * platform is either of "win", "mac" or "linux".
>-// * architecture is either of "32" or "64".
>+// Returns a path to a resource file in [project-root]/resources/ dir.
>+// Returns an absolute path
> //
> // Arguments:
> //    name - Name of the resource file. If a plain filename (no directory path)
>@@ -110,15 +103,6 @@ std::string DirName(const std::string& path);
> // empty or if the file does not exist/is readable.
> size_t GetFileSize(const std::string& filename);
> 
>-// Sets the executable path, i.e. the path to the executable that is being used
>-// when launching it. This is usually the path relative to the working directory
>-// but can also be an absolute path. The intention with this function is to pass
>-// the argv[0] being sent into the main function to make it possible for
>-// fileutils.h to find the correct project paths even when the working directory
>-// is outside the project tree (which happens in some cases).
>-// TODO(bugs.webrtc.org/9792): Deprecated - going away soon.
>-void SetExecutablePath(const std::string& path_to_executable);
>-
> }  // namespace test
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_override.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_override.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..884069c10dd4f610cdbea906acf61aa323b2a0ac
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_override.cc
>@@ -0,0 +1,151 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "test/testsupport/fileutils_override.h"
>+
>+#if defined(WEBRTC_WIN)
>+#include <direct.h>
>+#include <tchar.h>
>+#include <windows.h>
>+#include <algorithm>
>+#include <codecvt>
>+#include <locale>
>+
>+#include "Shlwapi.h"
>+#include "WinDef.h"
>+
>+#include "rtc_base/win32.h"
>+#define GET_CURRENT_DIR _getcwd
>+#else
>+#include <dirent.h>
>+#include <unistd.h>
>+#define GET_CURRENT_DIR getcwd
>+#endif
>+
>+#if defined(WEBRTC_IOS)
>+#include "test/testsupport/iosfileutils.h"
>+#endif
>+
>+#if defined(WEBRTC_MAC)
>+#include "test/testsupport/macfileutils.h"
>+#endif
>+
>+#include "absl/types/optional.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/stringutils.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+std::string DirName(const std::string& path);
>+bool CreateDir(const std::string& directory_name);
>+
>+namespace internal {
>+
>+namespace {
>+#if defined(WEBRTC_WIN)
>+const char* kPathDelimiter = "\\";
>+#elif !defined(WEBRTC_IOS)
>+const char* kPathDelimiter = "/";
>+#endif
>+
>+#if defined(WEBRTC_ANDROID)
>+// This is a special case in Chrome infrastructure. See
>+// base/test/test_support_android.cc.
>+const char* kAndroidChromiumTestsRoot = "/sdcard/chromium_tests_root/";
>+#endif
>+
>+#if !defined(WEBRTC_IOS)
>+const char* kResourcesDirName = "resources";
>+#endif
>+
>+}  // namespace
>+
>+// Finds the WebRTC src dir.
>+// The returned path always ends with a path separator.
>+absl::optional<std::string> ProjectRootPath() {
>+#if defined(WEBRTC_ANDROID)
>+  return kAndroidChromiumTestsRoot;
>+#elif defined WEBRTC_IOS
>+  return IOSRootPath();
>+#elif defined(WEBRTC_MAC)
>+  std::string path;
>+  GetNSExecutablePath(&path);
>+  std::string exe_dir = DirName(path);
>+  // On Mac, tests execute in out/Whatever, so src is two levels up except if
>+  // the test is bundled (which our tests are not), in which case it's 5 levels.
>+  return DirName(DirName(exe_dir)) + kPathDelimiter;
>+#elif defined(WEBRTC_POSIX)
>+  char buf[PATH_MAX];
>+  ssize_t count = ::readlink("/proc/self/exe", buf, arraysize(buf));
>+  if (count <= 0) {
>+    RTC_NOTREACHED() << "Unable to resolve /proc/self/exe.";
>+    return absl::nullopt;
>+  }
>+  // On POSIX, tests execute in out/Whatever, so src is two levels up.
>+  std::string exe_dir = DirName(std::string(buf, count));
>+  return DirName(DirName(exe_dir)) + kPathDelimiter;
>+#elif defined(WEBRTC_WIN)
>+  wchar_t buf[MAX_PATH];
>+  buf[0] = 0;
>+  if (GetModuleFileName(NULL, buf, MAX_PATH) == 0)
>+    return absl::nullopt;
>+
>+  std::string exe_path = rtc::ToUtf8(std::wstring(buf));
>+  std::string exe_dir = DirName(exe_path);
>+  return DirName(DirName(exe_dir)) + kPathDelimiter;
>+#endif
>+}
>+
>+std::string OutputPath() {
>+#if defined(WEBRTC_IOS)
>+  return IOSOutputPath();
>+#elif defined(WEBRTC_ANDROID)
>+  return kAndroidChromiumTestsRoot;
>+#else
>+  absl::optional<std::string> path_opt = ProjectRootPath();
>+  RTC_DCHECK(path_opt);
>+  std::string path = *path_opt + "out";
>+  if (!CreateDir(path)) {
>+    return "./";
>+  }
>+  return path + kPathDelimiter;
>+#endif
>+}
>+
>+std::string WorkingDir() {
>+#if defined(WEBRTC_ANDROID)
>+  return kAndroidChromiumTestsRoot;
>+#endif
>+  char path_buffer[FILENAME_MAX];
>+  if (!GET_CURRENT_DIR(path_buffer, sizeof(path_buffer))) {
>+    fprintf(stderr, "Cannot get current directory!\n");
>+    return "./";
>+  } else {
>+    return std::string(path_buffer);
>+  }
>+}
>+
>+std::string ResourcePath(const std::string& name,
>+                         const std::string& extension) {
>+#if defined(WEBRTC_IOS)
>+  return IOSResourcePath(name, extension);
>+#else
>+  absl::optional<std::string> path_opt = ProjectRootPath();
>+  RTC_DCHECK(path_opt);
>+  std::string resources_path = *path_opt + kResourcesDirName + kPathDelimiter;
>+  return resources_path + name + "." + extension;
>+#endif
>+}
>+
>+}  // namespace internal
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_override.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_override.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..141c76860716ad7892edddbd8359931cf20b026b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_override.h
>@@ -0,0 +1,57 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <stdio.h>
>+#include <string>
>+
>+#ifndef TEST_TESTSUPPORT_FILEUTILS_OVERRIDE_H_
>+#define TEST_TESTSUPPORT_FILEUTILS_OVERRIDE_H_
>+
>+namespace webrtc {
>+namespace test {
>+namespace internal {
>+
>+// Returns the absolute path to the output directory where log files and other
>+// test artifacts should be put. The output directory is generally a directory
>+// named "out" at the project root. This root is assumed to be two levels above
>+// where the test binary is located; this is because tests execute in a dir
>+// out/Whatever relative to the project root. This convention is also followed
>+// in Chromium.
>+//
>+// The exception is Android where we use /sdcard/ instead.
>+//
>+// If symbolic links occur in the path they will be resolved and the actual
>+// directory will be returned.
>+//
>+// Returns the path WITH a trailing path delimiter. If the project root is not
>+// found, the current working directory ("./") is returned as a fallback.
>+std::string OutputPath();
>+
>+// Gets the current working directory for the executing program.
>+// Returns "./" if for some reason it is not possible to find the working
>+// directory.
>+std::string WorkingDir();
>+
>+// Returns a path to a resource file in [project-root]/resources/ dir.
>+// Returns an absolute path
>+//
>+// Arguments:
>+//    name - Name of the resource file. If a plain filename (no directory path)
>+//           is supplied, the file is assumed to be located in resources/
>+//           If a directory path is prepended to the filename, a subdirectory
>+//           hierarchy reflecting that path is assumed to be present.
>+//    extension - File extension, without the dot, i.e. "bmp" or "yuv".
>+std::string ResourcePath(const std::string& name, const std::string& extension);
>+
>+}  // namespace internal
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // TEST_TESTSUPPORT_FILEUTILS_OVERRIDE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_unittest.cc
>index c6dc86deb200a3c3c02d28d104e265266870982a..d39f46800418da7c376768fad123ee71969604b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_unittest.cc
>@@ -24,9 +24,6 @@
> 
> #ifdef WIN32
> #define chdir _chdir
>-static const char* kPathDelimiter = "\\";
>-#else
>-static const char* kPathDelimiter = "/";
> #endif
> 
> using ::testing::EndsWith;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer.cc
>index 537e85cfa5e55f10029114432a25e977f68b5f98..8bf1ee4630e1f3b84a7da43fd171c9d1b1263457 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer.cc
>@@ -10,7 +10,6 @@
> 
> #include <stdio.h>
> 
>-#include "common_types.h"  // NOLINT(build/include)
> #include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts.cc
>index 0f7e0cd4ff86606d0b9ab2ccfbcca352a390d7ee..9438cefbc4fd7e186bbc22efb7dabca2849a44ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts.cc
>@@ -24,9 +24,9 @@ const std::string& DefaultArtifactPath() {
> }
> }  // namespace
> 
>-DEFINE_string(test_artifacts_dir,
>-              DefaultArtifactPath().c_str(),
>-              "The output folder where test output should be saved.");
>+WEBRTC_DEFINE_string(test_artifacts_dir,
>+                     DefaultArtifactPath().c_str(),
>+                     "The output folder where test output should be saved.");
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts_unittest.cc
>index c423cd90eada908af3c92ffa9de2e1f86196ddb9..df02d27c6725d6167226ef2410f57040513c395b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts_unittest.cc
>@@ -16,12 +16,11 @@
> 
> #include "rtc_base/file.h"
> #include "rtc_base/flags.h"
>-#include "rtc_base/pathutils.h"
> #include "rtc_base/platform_file.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
> 
>-DECLARE_string(test_artifacts_dir);
>+WEBRTC_DECLARE_string(test_artifacts_dir);
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.h
>index 59df4c71e2aa863e1bcd89f6f0013aff4ab93420..5f4707f911551f449cb8cf42f46b64036180f7fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.h
>@@ -12,7 +12,6 @@
> 
> #include <memory>
> 
>-#include "common_types.h"  // NOLINT(build/include)
> #include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "modules/video_capture/video_capture.h"
> #include "rtc_base/criticalsection.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_encoder_proxy_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_encoder_proxy_factory.h
>index e7289c0399e2667ee8f13256648d603755e345f4..9e5ea12ce2bcbb9382f335381b559b95740dd72a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_encoder_proxy_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_encoder_proxy_factory.h
>@@ -71,23 +71,17 @@ class VideoEncoderProxyFactory final : public VideoEncoderFactory {
>                        size_t max_payload_size) override {
>       return encoder_->InitEncode(config, number_of_cores, max_payload_size);
>     }
>-    VideoEncoder::ScalingSettings GetScalingSettings() const override {
>-      return encoder_->GetScalingSettings();
>-    }
>     int32_t RegisterEncodeCompleteCallback(
>         EncodedImageCallback* callback) override {
>       return encoder_->RegisterEncodeCompleteCallback(callback);
>     }
>     int32_t Release() override { return encoder_->Release(); }
>-    int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override {
>-      return encoder_->SetChannelParameters(packet_loss, rtt);
>-    }
>     int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
>                               uint32_t framerate) override {
>       return encoder_->SetRateAllocation(rate_allocation, framerate);
>     }
>-    const char* ImplementationName() const override {
>-      return encoder_->ImplementationName();
>+    VideoEncoder::EncoderInfo GetEncoderInfo() const override {
>+      return encoder_->GetEncoderInfo();
>     }
> 
>     VideoEncoder* const encoder_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/run_loop_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/run_loop_win.cc
>index 28dc9334e086ff5919c639bd660275f56851b9e4..ee936358fa6d2be03fa7598bd84870b47b5f90b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/run_loop_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/run_loop_win.cc
>@@ -18,15 +18,18 @@
> namespace webrtc {
> namespace test {
> 
>-void PressEnterToContinue() {
>+void PressEnterToContinue(SingleThreadedTaskQueueForTesting &task_queue) {
>   puts(">> Press ENTER to continue...");
> 
>-  MSG msg;
>-  BOOL ret;
>-  while ((ret = GetMessage(&msg, NULL, 0, 0)) != 0) {
>-    assert(ret != -1);
>-    TranslateMessage(&msg);
>-    DispatchMessage(&msg);
>+  while (!_kbhit() || _getch() != '\r') {
>+    // Drive the message loop for the thread running the task_queue
>+    task_queue.SendTask([&]() {
>+      MSG msg;
>+      if (PeekMessage(&msg, NULL, 0, 0, PM_REMOVE)) {
>+        TranslateMessage(&msg);
>+        DispatchMessage(&msg);
>+      }
>+    });
>   }
> }
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/suppressions.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/suppressions.xml
>index 67e5b4165302fe83c1bfecca6ec86a23522da19a..45375c2fe71c8d3626ccf9fd01e6305d5eca3211 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/suppressions.xml
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/suppressions.xml
>@@ -22,7 +22,10 @@
> 
>   <issue id="GoogleAppIndexingWarning" severity="ignore"/>
>   <issue id="MissingRegistered" severity="ignore"/>
>-
>+  <issue id="LintError">
>+    
>+    <ignore regexp="No `.class` files were found in project"/>
>+  </issue>
>   
>   <issue id="MissingApplicationIcon" severity="ignore"/>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/roll_deps.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/roll_deps.py
>index 9999c6f6afbf6df1da110d152039f2da20c38df5..d3a8bdbca8f9fb27f2f1134f88dd051d9ee1de8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/roll_deps.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/roll_deps.py
>@@ -70,6 +70,8 @@ ANDROID_DEPS_END = r'=== ANDROID_DEPS Generated Code End ==='
> # Location of automically gathered android deps.
> ANDROID_DEPS_PATH = 'src/third_party/android_deps/'
> 
>+NOTIFY_EMAIL = 'webrtc-trooper@grotations.appspotmail.com'
>+
> 
> sys.path.append(os.path.join(CHECKOUT_SRC_DIR, 'build'))
> import find_depot_tools
>@@ -256,8 +258,9 @@ def BuildDepsentryDict(deps_dict):
> 
> 
> def _FindChangedCipdPackages(path, old_pkgs, new_pkgs):
>-  assert ({p['package'] for p in old_pkgs} ==
>-          {p['package'] for p in new_pkgs})
>+  pkgs_equal = ({p['package'] for p in old_pkgs} ==
>+      {p['package'] for p in new_pkgs})
>+  assert pkgs_equal, 'Old: %s\n New: %s' % (old_pkgs, new_pkgs)
>   for old_pkg in old_pkgs:
>     for new_pkg in new_pkgs:
>       old_version = old_pkg['version']
>@@ -592,10 +595,11 @@ def _UploadCL(commit_queue_mode):
>     - 1: Run trybots but do not submit to CQ.
>     - 0: Skip CQ, upload only.
>   """
>-  cmd = ['git', 'cl', 'upload', '--force', '--bypass-hooks']
>+  cmd = ['git', 'cl', 'upload', '--force', '--bypass-hooks', '--send-mail']
>+  cmd.extend(['--cc', NOTIFY_EMAIL])
>   if commit_queue_mode >= 2:
>     logging.info('Sending the CL to the CQ...')
>-    cmd.extend(['--use-commit-queue', '--send-mail'])
>+    cmd.extend(['--use-commit-queue'])
>   elif commit_queue_mode >= 1:
>     logging.info('Starting CQ dry run...')
>     cmd.extend(['--cq-dry-run'])
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/coverage/generate_coverage_command.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/coverage/generate_coverage_command.py
>index a402f38da7d45893dba77d32adc83eacb312d073..856666816dff1659d2f681a62807f477278c3e91 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/coverage/generate_coverage_command.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/coverage/generate_coverage_command.py
>@@ -22,13 +22,13 @@ TESTS = [
>   'video_engine_tests',
>   'tools_unittests',
>   'test_support_unittests',
>+  'slow_tests',
>   'system_wrappers_unittests',
>   'rtc_unittests',
>   'rtc_stats_unittests',
>   'rtc_pc_unittests',
>   'rtc_media_unittests',
>   'peerconnection_unittests',
>-  'ortc_unittests',
>   'modules_unittests',
>   'modules_tests',
>   'low_bandwidth_audio_test',
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/coverage/generate_ios_coverage_command.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/coverage/generate_ios_coverage_command.py
>new file mode 100644
>index 0000000000000000000000000000000000000000..f81ee2c62b0a75d2e9ace9c2afc95ff6db62c8d4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/coverage/generate_ios_coverage_command.py
>@@ -0,0 +1,174 @@
>+#!/usr/bin/env python
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+"""Generates command-line instructions to produce one-time iOS coverage using
>+coverage.py.
>+
>+This script is usable for both real devices and simulator.
>+But for real devices actual execution should be done manually from Xcode
>+and coverage.profraw files should be also copied manually from the device.
>+
>+Additional prerequisites:
>+
>+1. Xcode 10+ with iPhone Simulator SDK. Can be installed by command:
>+   $ mac_toolchain install -kind ios -xcode-version 10l232m \
>+     -output-dir build/mac_files/Xcode.app
>+
>+2. For computing coverage on real device you probably also need to apply
>+following patch to code_coverage/coverage.py script:
>+
>+========== BEGINNING OF PATCH ==========
>+--- a/code_coverage/coverage.py
>++++ b/code_coverage/coverage.py
>+@@ -693,8 +693,7 @@ def _AddArchArgumentForIOSIfNeeded(cmd_list, num_archs):
>+   to use, and one architecture needs to be specified for each binary.
>+   "" "
>+if _IsIOS():
>+-    cmd_list.extend(['-arch=x86_64'] * num_archs)
>++    cmd_list.extend(['-arch=arm64'] * num_archs)
>+
>+
>+def _GetBinaryPath(command):
>+@@ -836,8 +835,8 @@ def _GetBinaryPathsFromTargets(targets, build_dir):
>+  binary_path = os.path.join(build_dir, target)
>+  if coverage_utils.GetHostPlatform() == 'win':
>+    binary_path += '.exe'
>++    elif coverage_utils.GetHostPlatform() == 'mac':
>++      binary_path += '.app/%s' % target
>+
>+if os.path.exists(binary_path):
>+  binary_paths.append(binary_path)
>+========== ENDING OF PATCH ==========
>+
>+"""
>+
>+import sys
>+
>+DIRECTORY = 'out/coverage'
>+
>+TESTS = [
>+  'audio_decoder_unittests',
>+  'common_audio_unittests',
>+  'common_video_unittests',
>+  'modules_tests',
>+  'modules_unittests',
>+  'rtc_media_unittests',
>+  'rtc_pc_unittests',
>+  'rtc_stats_unittests',
>+  'rtc_unittests',
>+  'slow_tests',
>+  'system_wrappers_unittests',
>+  'test_support_unittests',
>+  'tools_unittests',
>+  'video_capture_tests',
>+  'video_engine_tests',
>+  'webrtc_nonparallel_tests',
>+]
>+
>+XC_TESTS = [
>+  'apprtcmobile_tests',
>+  'sdk_framework_unittests',
>+  'sdk_unittests',
>+]
>+
>+
>+def FormatIossimTest(test_name, is_xctest=False):
>+  args = ['%s/%s.app' % (DIRECTORY, test_name)]
>+  if is_xctest:
>+    args += ['%s/%s_module.xctest' % (DIRECTORY, test_name)]
>+
>+  return '-c \'%s/iossim %s\'' % (DIRECTORY, ' '.join(args))
>+
>+
>+def GetGNArgs(is_simulator):
>+  target_cpu = 'x64' if is_simulator else 'arm64'
>+  return ([] +
>+          ['target_os="ios"'] +
>+          ['target_cpu="%s"' % target_cpu] +
>+          ['use_clang_coverage=true'] +
>+          ['is_component_build=false'] +
>+          ['dcheck_always_on=true'])
>+
>+
>+def GenerateIOSSimulatorCommand():
>+  gn_args_string = ' '.join(GetGNArgs(is_simulator=True))
>+  gn_cmd = ['gn', 'gen', DIRECTORY, '--args=\'%s\'' % gn_args_string]
>+
>+  coverage_cmd = (
>+      [sys.executable, 'tools/code_coverage/coverage.py'] +
>+      ["%s.app" % t for t in XC_TESTS + TESTS] +
>+      ['-b %s' % DIRECTORY, '-o out/report'] +
>+      ['-i=\'.*/out/.*|.*/third_party/.*|.*test.*\''] +
>+      [FormatIossimTest(t, is_xctest=True) for t in XC_TESTS] +
>+      [FormatIossimTest(t, is_xctest=False) for t in TESTS]
>+  )
>+
>+  print 'To get code coverage using iOS simulator just run following commands:'
>+  print ''
>+  print ' '.join(gn_cmd)
>+  print ''
>+  print ' '.join(coverage_cmd)
>+  return 0
>+
>+
>+def GenerateIOSDeviceCommand():
>+  gn_args_string = ' '.join(GetGNArgs(is_simulator=False))
>+
>+  coverage_report_cmd = (
>+      [sys.executable, 'tools/code_coverage/coverage.py'] +
>+      ['%s.app' % t for t in TESTS] +
>+      ['-b %s' % DIRECTORY] +
>+      ['-o out/report'] +
>+      ['-p %s/merged.profdata' % DIRECTORY] +
>+      ['-i=\'.*/out/.*|.*/third_party/.*|.*test.*\'']
>+  )
>+
>+  print 'Computing code coverage for real iOS device is a little bit tedious.'
>+  print ''
>+  print 'You will need:'
>+  print ''
>+  print '1. Generate xcode project and open it with Xcode 10+:'
>+  print '  gn gen %s --ide=xcode --args=\'%s\'' % (DIRECTORY, gn_args_string)
>+  print '  open %s/all.xcworkspace' % DIRECTORY
>+  print ''
>+  print '2. Execute these Run targets manually with Xcode Run button and '
>+  print 'manually save generated coverage.profraw file to %s:' % DIRECTORY
>+  print '\n'.join('- %s' % t for t in TESTS)
>+  print ''
>+  print '3. Execute these Test targets manually with Xcode Test button and '
>+  print 'manually save generated coverage.profraw file to %s:' % DIRECTORY
>+  print '\n'.join('- %s' % t for t in XC_TESTS)
>+  print ''
>+  print '4. Merge *.profraw files to *.profdata using llvm-profdata tool:'
>+  print ('  build/mac_files/Xcode.app/Contents/Developer/Toolchains/' +
>+         'XcodeDefault.xctoolchain/usr/bin/llvm-profdata merge ' +
>+         '-o %s/merged.profdata ' % DIRECTORY +
>+         '-sparse=true %s/*.profraw' % DIRECTORY)
>+  print ''
>+  print '5. Generate coverage report:'
>+  print '  ' + ' '.join(coverage_report_cmd)
>+  return 0
>+
>+
>+def Main():
>+  if len(sys.argv) < 2:
>+    print 'Please specify type of coverage:'
>+    print '  %s simulator' % sys.argv[0]
>+    print '  %s device' % sys.argv[0]
>+  elif sys.argv[1] == 'simulator':
>+    GenerateIOSSimulatorCommand()
>+  elif sys.argv[1] == 'device':
>+    GenerateIOSDeviceCommand()
>+  else:
>+    print 'Unsupported type of coverage'
>+
>+  return 0
>+
>+if __name__ == '__main__':
>+  sys.exit(Main())
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/gn_check_autofix.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/gn_check_autofix.py
>index b939636d0c831e928772389ab5f19155eddefa6f..e44f7122dfff9ef037cbc9d46062ccc9462a759d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/gn_check_autofix.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/gn_check_autofix.py
>@@ -31,6 +31,9 @@ from collections import defaultdict
> 
> SCRIPT_DIR = os.path.dirname(os.path.abspath(__file__))
> 
>+CHROMIUM_DIRS = ['base', 'build', 'buildtools',
>+                 'testing', 'third_party', 'tools']
>+
> TARGET_RE = re.compile(
>     r'(?P<indentation_level>\s*)\w*$"(?P<target_name>\w*)"$ {$')
> 
>@@ -88,22 +91,48 @@ def FixErrors(filename, missing_deps, deleted_sources):
> 
>   Run(['gn', 'format', filename])
> 
>+def FirstNonEmpty(iterable):
>+  """Return first item which evaluates to True, or fallback to None."""
>+  return next((x for x in iterable if x), None)
>+
> def Rebase(base_path, dependency_path, dependency):
>-  base_path = base_path.split(os.path.sep)
>-  dependency_path = dependency_path.split(os.path.sep)
>-
>-  first_difference = None
>-  shortest_length = min(len(dependency_path), len(base_path))
>-  for i in range(shortest_length):
>-    if dependency_path[i] != base_path[i]:
>-      first_difference = i
>-      break
>-
>-  first_difference = first_difference or shortest_length
>-  base_path = base_path[first_difference:]
>-  dependency_path = dependency_path[first_difference:]
>-  return (os.path.sep.join((['..'] * len(base_path)) + dependency_path) +
>-          ':' + dependency)
>+  """Adapt paths so they work both in stand-alone WebRTC and Chromium tree.
>+
>+  To cope with varying top-level directory (WebRTC VS Chromium), we use:
>+    * relative paths for WebRTC modules.
>+    * absolute paths for shared ones.
>+  E.g. '//common_audio/...' -> '../../common_audio/'
>+       '//third_party/...' remains as is.
>+
>+  Args:
>+    base_path: current module path  (E.g. '//video')
>+    dependency_path: path from root (E.g. '//rtc_base/time')
>+    dependency: target itself       (E.g. 'timestamp_extrapolator')
>+
>+  Returns:
>+    Full target path (E.g. '../rtc_base/time:timestamp_extrapolator').
>+  """
>+
>+  root = FirstNonEmpty(dependency_path.split('/'))
>+  if root in CHROMIUM_DIRS:
>+    # Chromium paths must remain absolute. E.g. //third_party//abseil-cpp...
>+    rebased = dependency_path
>+  else:
>+    base_path = base_path.split(os.path.sep)
>+    dependency_path = dependency_path.split(os.path.sep)
>+
>+    first_difference = None
>+    shortest_length = min(len(dependency_path), len(base_path))
>+    for i in range(shortest_length):
>+      if dependency_path[i] != base_path[i]:
>+        first_difference = i
>+        break
>+
>+    first_difference = first_difference or shortest_length
>+    base_path = base_path[first_difference:]
>+    dependency_path = dependency_path[first_difference:]
>+    rebased = os.path.sep.join((['..'] * len(base_path)) + dependency_path)
>+  return rebased + ':' + dependency
> 
> def main():
>   deleted_sources = set()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Debug.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Debug.json
>index d2966ec3ad4d67ece0e87edf2bc5bd5b6aa52fad..b14809ff3d4effc3dc773be9bae433c457a38134 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Debug.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Debug.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Builder for 32-bit devices."
>+    "Builder for 32-bit devices.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Release.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Release.json
>index e72a2291787c70f9e301b78a66df56a14cf3e4a5..97263256de9919c9b1f7e932bbf4dd4cdef8f12b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Release.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Release.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Builder for 32-bit devices."
>+    "Builder for 32-bit devices.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Sim_Debug_(iOS_9.0).json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Sim_Debug_(iOS_9.0).json
>index 7a56d4618075068ceb74b114418c24fa66b7cb6a..03b272bb5347b3cdd2890469e3138ad4fac9157a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Sim_Debug_(iOS_9.0).json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Sim_Debug_(iOS_9.0).json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Tests for 32-bit iOS simulators."
>+    "Tests for 32-bit iOS simulators.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Debug.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Debug.json
>index 6532c90f5097bab8e6e3c0cd66d6d5f6cc7f5cf2..f1fb7fa71901a80d3338939830610501928b8195 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Debug.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Debug.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Builder for 64-bit devices."
>+    "Builder for 64-bit devices.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Release.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Release.json
>index 8ca464c3c4300555fc8a689658aa9298e2d0da74..c2555c6a170d420a9cb46c4ff603de29172d4641 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Release.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Release.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Builder for 64-bit devices."
>+    "Builder for 64-bit devices.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_10.0).json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_10.0).json
>index e759d95e3686b972c524d98c9a2d01d8075bb561..d8a71f665a342ad913537af80b630d489769125d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_10.0).json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_10.0).json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Tests for 64-bit iOS simulators."
>+    "Tests for 64-bit iOS simulators.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_11.0).json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_11.0).json
>index 3dad91ba64f7e3faf78d4188624ad4823e48bc25..c303088b44977e10856fd394ee099a54ecfc89de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_11.0).json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_11.0).json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Tests for 64-bit iOS simulators."
>+    "Tests for 64-bit iOS simulators.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_9.0).json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_9.0).json
>index d27cdaa45f540c56a0827cda6aaa6f939dfbed79..bcf52d728326ecb278c5d12446e7b9135920de19 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_9.0).json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_9.0).json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Tests for 64-bit iOS simulators."
>+    "Tests for 64-bit iOS simulators.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/generate_modulemap.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/generate_modulemap.py
>new file mode 100644
>index 0000000000000000000000000000000000000000..45bd3d875e7cd0a062c75fc0a2ff13fc7d1fe66d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/generate_modulemap.py
>@@ -0,0 +1,32 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import argparse
>+import sys
>+
>+def GenerateModulemap():
>+  parser = argparse.ArgumentParser(description='Generate modulemap')
>+  parser.add_argument("-o", "--out", type=str, help="Output file.")
>+  parser.add_argument("-n", "--name", type=str, help="Name of binary.")
>+
>+  args = parser.parse_args()
>+
>+  with open(args.out, "w") as outfile:
>+    module_template = 'framework module %s {\n' \
>+                      '  umbrella header "%s.h"\n' \
>+                      '\n' \
>+                      '  export *\n' \
>+                      '  module * { export * }\n' \
>+                      '}\n' % (args.name, args.name)
>+    outfile.write(module_template)
>+  return 0
>+
>+
>+if __name__ == '__main__':
>+  sys.exit(GenerateModulemap())
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Debug.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Debug.json
>index 7c3b3cee1da128adc6b3bca0853a184e5e4e3b33..a97d8b706f82150fd236e11855ab4bd37a77ad83 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Debug.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Debug.json
>@@ -21,6 +21,7 @@
>   "tests": [
>     {
>       "include": "real_device_tests.json",
>+      "device type": "iPhone 6s",
>       "os": "11.4.1",
>       "dimensions": [
>         {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Perf.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Perf.json
>index b5357661a8427b21cf3eb37226c7267498f8d9ab..88bfe75264ad56b2552afd01b34bfab9a9a4b9ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Perf.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Perf.json
>@@ -21,13 +21,8 @@
>   "tests": [
>     {
>       "include": "perf_tests.json",
>+      "device type": "iPhone 7",
>       "os": "11.4.1",
>-      "dimensions": [
>-        {
>-          "os": "Mac",
>-          "pool": "Chrome"
>-        }
>-      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Release.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Release.json
>index ac84db10bf5c6228f37c0f992cd80826f777c544..d75cde07ff681ba69d0291e4cb0968262c0b6986 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Release.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Release.json
>@@ -22,6 +22,7 @@
>   "tests": [
>     {
>       "include": "real_device_tests.json",
>+      "device type": "iPhone 6s",
>       "os": "11.4.1",
>       "dimensions": [
>         {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_perf.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_perf.json
>index 0230c74a92ad07d158ef048423a7695378e6d237..0c30581da1df01076d88bf838b507c45db3e2c7f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_perf.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_perf.json
>@@ -23,12 +23,6 @@
>       "include": "perf_trybot_tests.json",
>       "device type": "iPhone 7",
>       "os": "11.0.3",
>-      "dimensions": [
>-        {
>-          "os": "Mac",
>-          "pool": "Chrome"
>-        }
>-      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/common_tests.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/common_tests.json
>index 5eeb408387fec473712d4c679eaba83b1e52f06c..7459503d7f06832539baf473f40c8fac43922d59 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/common_tests.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/common_tests.json
>@@ -27,9 +27,6 @@
>     {
>       "app": "modules_unittests"
>     },
>-    {
>-      "app": "ortc_unittests"
>-    },
>     {
>       "app": "rtc_media_unittests"
>     },
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/real_device_tests.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/real_device_tests.json
>index 8417470ddb882e29852c9048ae0c79b1bbed9e3b..dcee51688d6cf9fd55faa0ae471ef2bd4f514363 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/real_device_tests.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/real_device_tests.json
>@@ -12,9 +12,6 @@
>     {
>       "app": "modules_unittests"
>     },
>-    {
>-      "app": "ortc_unittests"
>-    },
>     {
>       "app": "rtc_pc_unittests"
>     },
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios32_sim_ios9_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios32_sim_ios9_dbg.json
>index 29a113d41388c7feb6eb9eb358c34075fc025ff1..1eacc9295fb1a72bfcce36eae273929dcc825dfd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios32_sim_ios9_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios32_sim_ios9_dbg.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Tests for 32-bit iOS simulators."
>+    "Tests for 32-bit iOS simulators.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios10_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios10_dbg.json
>index 6484ca1050de0eaa4f19d81005bf05d0b67f6a0a..8cecf4a41ef599e55a8ea63cb66cfd72165f793a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios10_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios10_dbg.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Tests for 64-bit iOS simulators."
>+    "Tests for 64-bit iOS simulators.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios11_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios11_dbg.json
>index b54f212acbd68adc1ed426eb41e9f6ac490f9f97..3a5f7c0cf6b0f7dd04132d5b4ffdb94229e66259 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios11_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios11_dbg.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Tests for 64-bit iOS simulators."
>+    "Tests for 64-bit iOS simulators.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios9_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios9_dbg.json
>index fdfa7b382d5ae2a09f29f29e531e03ff1194c31f..6c9848ee6dd968715672810cab0d5c1ec5b73837 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios9_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios9_dbg.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Tests for 64-bit iOS simulators."
>+    "Tests for 64-bit iOS simulators.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_dbg.json
>index 7594cd2a36d66e6c8dceb6099227ca45fbe80d02..43add6710e6c5729a322dc92bbb185597707f655 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_dbg.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Builder for 64-bit devices."
>+    "Builder for 64-bit devices.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_rel.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_rel.json
>index 08ae7e198332124b31e4536467e7f826739cf080..95a57e52262e52ae6feee95a35cfbad76de5e451 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_rel.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_rel.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Builder for 64-bit devices."
>+    "Builder for 64-bit devices.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_dbg.json
>index e9d06c2a76140048809ebb235a467f02d7024f78..8be9e95c0f0b5aa1cd32b8c96e66aad5252ccd8d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_dbg.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Builder for 32-bit devices."
>+    "Builder for 32-bit devices.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_rel.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_rel.json
>index 4521ef515372155759376a73abecc51f520b4205..f644940c7c400d8d18fefc44aa8453573afc0acf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_rel.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_rel.json
>@@ -1,6 +1,7 @@
> {
>   "comments": [
>-    "Builder for 32-bit devices."
>+    "Builder for 32-bit devices.",
>+    "NOTE: Update cache entry in cr-buildbucket.cfg when changing xcode version."
>   ],
>   "xcode build version": "10l232m",
>   "gn_args": [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses.py
>index 892fb46c05178f4e5b49576ab4de4e37762e364a..d1a76a64eec1a35467f1ede98eef7f9627339dc9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses.py
>@@ -36,7 +36,6 @@ LIB_TO_LICENSES_DICT = {
>     'bazel': ['third_party/bazel/LICENSE'],
>     'boringssl': ['third_party/boringssl/src/LICENSE'],
>     'errorprone': ['third_party/errorprone/LICENSE'],
>-    'expat': ['third_party/expat/files/COPYING'],
>     'fiat': ['third_party/boringssl/src/third_party/fiat/LICENSE'],
>     'guava': ['third_party/guava/LICENSE'],
>     'ijar': ['third_party/ijar/LICENSE'],
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/gn_isolate_map.pyl b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/gn_isolate_map.pyl
>index fecc56c38e1d1ea694889b00b6615407eceb0d13..d72bbdc0090af848bf89c26c9ee3b32d17af582b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/gn_isolate_map.pyl
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/gn_isolate_map.pyl
>@@ -82,10 +82,6 @@
>     "label": "//modules:modules_unittests",
>     "type": "windowed_test_launcher",
>   },
>-  "ortc_unittests": {
>-    "label": "//ortc:ortc_unittests",
>-    "type": "console_test_launcher",
>-  },
>   "peerconnection_unittests": {
>     "label": "//pc:peerconnection_unittests",
>     "type": "console_test_launcher",
>@@ -106,6 +102,10 @@
>     "label": "//:rtc_unittests",
>     "type": "console_test_launcher",
>   },
>+  "slow_tests": {
>+    "label": "//:slow_tests",
>+    "type": "console_test_launcher",
>+  },
>   "system_wrappers_unittests": {
>     "label": "//system_wrappers:system_wrappers_unittests",
>     "type": "console_test_launcher",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb.py
>index 7f196bb733846c8cf0031af11782f25939699eb9..485fceb858c98aa39d2e9ffe23c6614f7e0dd0b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb.py
>@@ -833,6 +833,7 @@ class MetaBuildWrapper(object):
>       '../../testing/test_env.py',
>     ]
> 
>+    must_retry = False
>     if test_type == 'script':
>       cmdline = ['../../' + self.ToSrcRelPath(isolate_map[target]['script'])]
>     elif is_android:
>@@ -874,12 +875,21 @@ class MetaBuildWrapper(object):
>             # so it can exit cleanly and report results, instead of being
>             # interrupted by swarming and not reporting anything.
>             '--timeout=%s' % timeout,
>-            '--retry_failed=3',
>         ]
>         if test_type == 'non_parallel_console_test_launcher':
>           # Still use the gtest-parallel-wrapper.py script since we need it to
>           # run tests on swarming, but don't execute tests in parallel.
>           cmdline.append('--workers=1')
>+        must_retry = True
>+
>+      asan = 'is_asan=true' in vals['gn_args']
>+      lsan = 'is_lsan=true' in vals['gn_args']
>+      msan = 'is_msan=true' in vals['gn_args']
>+      tsan = 'is_tsan=true' in vals['gn_args']
>+      sanitizer = asan or lsan or msan or tsan
>+      if must_retry and not sanitizer:
>+        # Retry would hide most sanitizers detections.
>+        cmdline.append('--retry_failed=3')
> 
>       executable_prefix = '.\\' if self.platform == 'win32' else './'
>       executable_suffix = '.exe' if self.platform == 'win32' else ''
>@@ -887,11 +897,6 @@ class MetaBuildWrapper(object):
> 
>       cmdline.append(executable)
> 
>-      asan = 'is_asan=true' in vals['gn_args']
>-      lsan = 'is_lsan=true' in vals['gn_args']
>-      msan = 'is_msan=true' in vals['gn_args']
>-      tsan = 'is_tsan=true' in vals['gn_args']
>-
>       cmdline.extend([
>           '--asan=%d' % asan,
>           '--lsan=%d' % lsan,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_config.pyl b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_config.pyl
>index 56c85837fe6a5a10ab12139193fc7c40c47de838..549ee4cb68ecbe1a8e9d6a40d9308637d941a0a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_config.pyl
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_config.pyl
>@@ -24,7 +24,6 @@
>       'iOS32 Release': 'ios',
>       'iOS64 Debug': 'ios',
>       'iOS64 Release': 'ios',
>-      'iOS32 Sim Debug (iOS 9.0)': 'ios',
>       'iOS64 Sim Debug (iOS 9.0)': 'ios',
>       'iOS64 Sim Debug (iOS 10.0)': 'ios',
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_unittest.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_unittest.py
>index 09093c06bae98511f5a5f40b82db01c537d71c4a..0e7173abb4ce66a6755686e3512d66a01a746eb2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_unittest.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_unittest.py
>@@ -382,8 +382,8 @@ class UnitTest(unittest.TestCase):
>         '--output_dir=${ISOLATED_OUTDIR}/test_logs',
>         '--gtest_color=no',
>         '--timeout=500',
>-        '--retry_failed=3',
>         '--workers=1',
>+        '--retry_failed=3',
>         './base_unittests',
>         '--asan=0',
>         '--lsan=0',
>@@ -502,8 +502,8 @@ class UnitTest(unittest.TestCase):
>         '--output_dir=${ISOLATED_OUTDIR}/test_logs',
>         '--gtest_color=no',
>         '--timeout=900',
>-        '--retry_failed=3',
>         '--workers=1',
>+        '--retry_failed=3',
>         './base_unittests',
>         '--asan=0',
>         '--lsan=0',
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/linux/zxing.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/linux/zxing.sha1
>deleted file mode 100644
>index f69a6ac4371f3ce5cc710cffe63a43961a55f5bb..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/linux/zxing.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-3e2a25eaa2e207a1e569c6585e50addb20bca89a
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/mac/zxing.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/mac/zxing.sha1
>deleted file mode 100644
>index 1b496e7ce6a904e61680c20cb12bc4917277837b..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/mac/zxing.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-6778e3ac489f4fc18a1ed1252bbd6eca54c9960e
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/win/zxing.exe.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/win/zxing.exe.sha1
>deleted file mode 100644
>index 0e5290fb346c53af866f39c4084d717e9d27ffd5..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/win/zxing.exe.sha1
>+++ /dev/null
>@@ -1 +0,0 @@
>-86ca3148778079665ad5b9459311d343a1865943
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn
>index 3273fda25d552868eb95805f2e54b27be079ca5a..b53d730ae07e06f8004e35536cae722a1a5a6ec6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn
>@@ -52,7 +52,7 @@ rtc_static_library("video") {
>   }
> 
>   deps = [
>-    "..:webrtc_common",
>+    ":frame_dumping_decoder",
>     "../api:fec_controller_api",
>     "../api:libjingle_peerconnection_api",
>     "../api:transport_api",
>@@ -133,6 +133,25 @@ rtc_source_set("video_stream_decoder_impl") {
>   ]
> }
> 
>+rtc_source_set("frame_dumping_decoder") {
>+  visibility = [ "*" ]
>+
>+  sources = [
>+    "frame_dumping_decoder.cc",
>+    "frame_dumping_decoder.h",
>+  ]
>+
>+  deps = [
>+    "../api/video:encoded_frame",
>+    "../api/video_codecs:video_codecs_api",
>+    "../modules/video_coding:video_codec_interface",
>+    "../modules/video_coding:video_coding",
>+    "../modules/video_coding:video_coding_utility",
>+    "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
> rtc_source_set("video_stream_encoder_impl") {
>   visibility = [ "*" ]
> 
>@@ -152,7 +171,9 @@ rtc_source_set("video_stream_encoder_impl") {
>   }
> 
>   deps = [
>+    "../api/video:encoded_image",
>     "../api/video:video_bitrate_allocator",
>+    "../api/video:video_bitrate_allocator_factory",
>     "../api/video:video_frame",
>     "../api/video:video_frame_i420",
>     "../api/video:video_stream_encoder",
>@@ -203,15 +224,19 @@ if (rtc_include_tests) {
>       "video_quality_test.h",
>     ]
>     deps = [
>+      ":frame_dumping_decoder",
>       "../api:fec_controller_api",
>       "../api:test_dependency_factory",
>       "../api:video_quality_test_fixture_api",
>+      "../api/video:builtin_video_bitrate_allocator_factory",
>+      "../api/video:video_bitrate_allocator_factory",
>       "../call:fake_network",
>       "../call:simulated_network",
>       "../logging:rtc_event_log_api",
>       "../logging:rtc_event_log_impl_output",
>       "../media:rtc_audio_video",
>       "../media:rtc_internal_video_codecs",
>+      "../media:rtc_vp8_encoder_simulcast_proxy",
>       "../modules/audio_device:audio_device_api",
>       "../modules/audio_device:audio_device_module_from_input_and_output",
>       "../modules/audio_device:windows_core_audio_utility",
>@@ -350,7 +375,7 @@ if (rtc_include_tests) {
>       "replay.cc",
>     ]
>     deps = [
>-      "..:webrtc_common",
>+      "../api/test/video:function_video_factory",
>       "../api/video_codecs:video_codecs_api",
>       "../call:call_interfaces",
>       "../common_video",
>@@ -384,6 +409,7 @@ if (rtc_include_tests) {
>     defines = []
>     sources = [
>       "call_stats_unittest.cc",
>+      "cpu_scaling_tests.cc",
>       "encoder_rtcp_feedback_unittest.cc",
>       "end_to_end_tests/bandwidth_tests.cc",
>       "end_to_end_tests/call_operation_tests.cc",
>@@ -391,6 +417,7 @@ if (rtc_include_tests) {
>       "end_to_end_tests/config_tests.cc",
>       "end_to_end_tests/extended_reports_tests.cc",
>       "end_to_end_tests/fec_tests.cc",
>+      "end_to_end_tests/frame_encryption_tests.cc",
>       "end_to_end_tests/histogram_tests.cc",
>       "end_to_end_tests/multi_codec_receive_tests.cc",
>       "end_to_end_tests/multi_stream_tester.cc",
>@@ -398,7 +425,6 @@ if (rtc_include_tests) {
>       "end_to_end_tests/multi_stream_tests.cc",
>       "end_to_end_tests/network_state_tests.cc",
>       "end_to_end_tests/probing_tests.cc",
>-      "end_to_end_tests/receive_time_tests.cc",
>       "end_to_end_tests/retransmission_tests.cc",
>       "end_to_end_tests/rtp_rtcp_tests.cc",
>       "end_to_end_tests/ssrc_tests.cc",
>@@ -424,9 +450,15 @@ if (rtc_include_tests) {
>       ":video",
>       ":video_mocks",
>       ":video_stream_encoder_impl",
>+      "../api:fake_frame_decryptor",
>+      "../api:fake_frame_encryptor",
>       "../api:simulated_network_api",
>+      "../api/test/video:function_video_factory",
>+      "../api/video:builtin_video_bitrate_allocator_factory",
>+      "../api/video:encoded_image",
>       "../api/video:video_frame",
>       "../api/video:video_frame_i420",
>+      "../api/video_codecs:create_vp8_temporal_layers",
>       "../api/video_codecs:video_codecs_api",
>       "../call:call_interfaces",
>       "../call:fake_network",
>@@ -443,6 +475,7 @@ if (rtc_include_tests) {
>       "../media:rtc_media",
>       "../media:rtc_media_base",
>       "../media:rtc_media_tests_utils",
>+      "../media:rtc_simulcast_encoder_adapter",
>       "../modules:module_api",
>       "../modules/pacing",
>       "../modules/rtp_rtcp",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats_unittest.cc
>index 71faa692a578a96ebe32e9a5d449fdeb04252ab1..dc6c69af9e98c71b5638b2dd34bb235245bbe7a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats_unittest.cc
>@@ -55,7 +55,7 @@ class CallStatsTest : public ::testing::Test {
> };
> 
> TEST_F(CallStatsTest, AddAndTriggerCallback) {
>-  rtc::Event event(false, false);
>+  rtc::Event event;
> 
>   static constexpr const int64_t kRtt = 25;
> 
>@@ -77,7 +77,7 @@ TEST_F(CallStatsTest, AddAndTriggerCallback) {
> }
> 
> TEST_F(CallStatsTest, ProcessTime) {
>-  rtc::Event event(false, false);
>+  rtc::Event event;
> 
>   static constexpr const int64_t kRtt = 100;
>   static constexpr const int64_t kRtt2 = 80;
>@@ -128,8 +128,8 @@ TEST_F(CallStatsTest, MultipleObservers) {
>   static constexpr const int64_t kRtt = 100;
> 
>   // Verify both observers are updated.
>-  rtc::Event ev1(false, false);
>-  rtc::Event ev2(false, false);
>+  rtc::Event ev1;
>+  rtc::Event ev2;
>   EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt))
>       .Times(AnyNumber())
>       .WillOnce(InvokeWithoutArgs([&ev1] { ev1.Set(); }))
>@@ -167,7 +167,7 @@ TEST_F(CallStatsTest, MultipleObservers) {
> 
>   // Flush the queue on the process thread to make sure we return after
>   // Process() has been called.
>-  rtc::Event event(false, false);
>+  rtc::Event event;
>   process_thread_->PostTask(rtc::NewClosure([&event]() { event.Set(); }));
>   event.Wait(rtc::Event::kForever);
> }
>@@ -183,7 +183,7 @@ TEST_F(CallStatsTest, ChangeRtt) {
>   call_stats_.RegisterStatsObserver(&stats_observer);
>   RtcpRttStats* rtcp_rtt_stats = &call_stats_;
> 
>-  rtc::Event event(false, false);
>+  rtc::Event event;
> 
>   static constexpr const int64_t kFirstRtt = 100;
>   static constexpr const int64_t kLowRtt = kFirstRtt - 20;
>@@ -240,7 +240,7 @@ TEST_F(CallStatsTest, ChangeRtt) {
> }
> 
> TEST_F(CallStatsTest, LastProcessedRtt) {
>-  rtc::Event event(false, false);
>+  rtc::Event event;
>   MockStatsObserver stats_observer;
>   call_stats_.RegisterStatsObserver(&stats_observer);
>   RtcpRttStats* rtcp_rtt_stats = &call_stats_;
>@@ -289,7 +289,7 @@ TEST_F(CallStatsTest, LastProcessedRtt) {
> 
> TEST_F(CallStatsTest, ProducesHistogramMetrics) {
>   metrics::Reset();
>-  rtc::Event event(false, false);
>+  rtc::Event event;
>   static constexpr const int64_t kRtt = 123;
>   RtcpRttStats* rtcp_rtt_stats = &call_stats_;
>   MockStatsObserver stats_observer;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/cpu_scaling_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/cpu_scaling_tests.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..608f05e05247db9f236c3f2d08108e39b1d37bdd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/cpu_scaling_tests.cc
>@@ -0,0 +1,131 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "test/call_test.h"
>+#include "test/field_trial.h"
>+#include "test/frame_generator_capturer.h"
>+
>+namespace webrtc {
>+namespace {
>+constexpr int kWidth = 1280;
>+constexpr int kHeight = 720;
>+constexpr int kFps = 28;
>+}  // namespace
>+
>+// Minimal normal usage at start, then 60s overuse.
>+class CpuOveruseTest : public test::CallTest {
>+ protected:
>+  CpuOveruseTest()
>+      : field_trials_("WebRTC-ForceSimulatedOveruseIntervalMs/1-60000-60000/") {
>+  }
>+
>+  void RunTestAndCheckForAdaptation(
>+      const DegradationPreference& degradation_preference,
>+      bool expect_adaptation);
>+
>+  test::ScopedFieldTrials field_trials_;
>+};
>+
>+void CpuOveruseTest::RunTestAndCheckForAdaptation(
>+    const DegradationPreference& degradation_preference,
>+    bool expect_adaptation) {
>+  class OveruseObserver
>+      : public test::SendTest,
>+        public test::FrameGeneratorCapturer::SinkWantsObserver {
>+   public:
>+    OveruseObserver(const DegradationPreference& degradation_preference,
>+                    bool expect_adaptation)
>+        : SendTest(expect_adaptation ? kLongTimeoutMs : kDefaultTimeoutMs),
>+          degradation_preference_(degradation_preference),
>+          expect_adaptation_(expect_adaptation) {}
>+
>+   private:
>+    void OnFrameGeneratorCapturerCreated(
>+        test::FrameGeneratorCapturer* frame_generator_capturer) override {
>+      frame_generator_capturer->SetSinkWantsObserver(this);
>+      // Set initial resolution.
>+      frame_generator_capturer->ChangeResolution(kWidth, kHeight);
>+    }
>+
>+    // Called when FrameGeneratorCapturer::AddOrUpdateSink is called.
>+    void OnSinkWantsChanged(rtc::VideoSinkInterface<VideoFrame>* sink,
>+                            const rtc::VideoSinkWants& wants) override {
>+      if (wants.max_pixel_count == std::numeric_limits<int>::max() &&
>+          wants.max_framerate_fps == kFps) {
>+        // Max configured framerate is initially set.
>+        return;
>+      }
>+      switch (degradation_preference_) {
>+        case DegradationPreference::MAINTAIN_FRAMERATE:
>+          EXPECT_LT(wants.max_pixel_count, kWidth * kHeight);
>+          observation_complete_.Set();
>+          break;
>+        case DegradationPreference::MAINTAIN_RESOLUTION:
>+          EXPECT_LT(wants.max_framerate_fps, kFps);
>+          observation_complete_.Set();
>+          break;
>+        case DegradationPreference::BALANCED:
>+          if (wants.max_pixel_count == std::numeric_limits<int>::max() &&
>+              wants.max_framerate_fps == std::numeric_limits<int>::max()) {
>+            // |adapt_counters_| map in VideoStreamEncoder is reset when
>+            // balanced mode is set.
>+            break;
>+          }
>+          EXPECT_TRUE(wants.max_pixel_count < kWidth * kHeight ||
>+                      wants.max_framerate_fps < kFps);
>+          observation_complete_.Set();
>+          break;
>+        default:
>+          RTC_NOTREACHED();
>+      }
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      EXPECT_FALSE(encoder_config->simulcast_layers.empty());
>+      encoder_config->simulcast_layers[0].max_framerate = kFps;
>+    }
>+
>+    void ModifyVideoDegradationPreference(
>+        DegradationPreference* degradation_preference) override {
>+      *degradation_preference = degradation_preference_;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_EQ(expect_adaptation_, Wait())
>+          << "Timed out while waiting for a scale down.";
>+    }
>+
>+    const DegradationPreference degradation_preference_;
>+    const bool expect_adaptation_;
>+  } test(degradation_preference, expect_adaptation);
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(CpuOveruseTest, AdaptsDownInResolutionOnOveruse) {
>+  RunTestAndCheckForAdaptation(DegradationPreference::MAINTAIN_FRAMERATE, true);
>+}
>+
>+TEST_F(CpuOveruseTest, AdaptsDownInFpsOnOveruse) {
>+  RunTestAndCheckForAdaptation(DegradationPreference::MAINTAIN_RESOLUTION,
>+                               true);
>+}
>+
>+TEST_F(CpuOveruseTest, AdaptsDownInResolutionOrFpsOnOveruse) {
>+  RunTestAndCheckForAdaptation(DegradationPreference::BALANCED, true);
>+}
>+
>+TEST_F(CpuOveruseTest, NoAdaptDownOnOveruse) {
>+  RunTestAndCheckForAdaptation(DegradationPreference::DISABLED, false);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/bandwidth_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/bandwidth_tests.cc
>index ab5e5687b997babebb420c37f6f5166f04b397c3..a35f08c3f5f6be3e6b2621418a1929059a8b2a27 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/bandwidth_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/bandwidth_tests.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "api/test/simulated_network.h"
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "call/fake_network_pipe.h"
> #include "call/simulated_network.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
>@@ -145,7 +146,6 @@ TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) {
>           sender_ssrc_(0),
>           remb_bitrate_bps_(1000000),
>           receive_transport_(nullptr),
>-          stop_event_(false, false),
>           poller_thread_(&BitrateStatsPollingThread,
>                          this,
>                          "BitrateStatsPollingThread"),
>@@ -160,15 +160,15 @@ TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) {
>           task_queue, nullptr, this, test::PacketTransport::kReceiver,
>           payload_type_map_,
>           absl::make_unique<FakeNetworkPipe>(
>-              Clock::GetRealTimeClock(),
>-              absl::make_unique<SimulatedNetwork>(
>-                  DefaultNetworkSimulationConfig())));
>+              Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>+                                             BuiltInNetworkBehaviorConfig())));
>       return receive_transport_;
>     }
> 
>-    void ModifySenderCallConfig(Call::Config* config) override {
>+    void ModifySenderBitrateConfig(
>+        BitrateConstraints* bitrate_config) override {
>       // Set a high start bitrate to reduce the test completion time.
>-      config->bitrate_config.start_bitrate_bps = remb_bitrate_bps_;
>+      bitrate_config->start_bitrate_bps = remb_bitrate_bps_;
>     }
> 
>     void ModifyVideoConfigs(
>@@ -274,6 +274,8 @@ TEST_F(BandwidthEndToEndTest, ReportsSetEncoderRates) {
>           task_queue_(task_queue),
>           send_stream_(nullptr),
>           encoder_factory_(this),
>+          bitrate_allocator_factory_(
>+              CreateBuiltinVideoBitrateAllocatorFactory()),
>           bitrate_kbps_(0) {}
> 
>     void OnVideoStreamsCreated(
>@@ -287,6 +289,8 @@ TEST_F(BandwidthEndToEndTest, ReportsSetEncoderRates) {
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>       send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+      send_config->encoder_settings.bitrate_allocator_factory =
>+          bitrate_allocator_factory_.get();
>       RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
>     }
> 
>@@ -345,6 +349,7 @@ TEST_F(BandwidthEndToEndTest, ReportsSetEncoderRates) {
>     rtc::CriticalSection crit_;
>     VideoSendStream* send_stream_;
>     test::VideoEncoderProxyFactory encoder_factory_;
>+    std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
>     uint32_t bitrate_kbps_ RTC_GUARDED_BY(crit_);
>   } test(&task_queue_);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/call_operation_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/call_operation_tests.cc
>index 2acd1b9fe812918701ad6f6f6c80f6d57132c51f..ec3ca84bfb547c10d6439c9bc3e317c36b060f6d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/call_operation_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/call_operation_tests.cc
>@@ -92,8 +92,6 @@ TEST_P(CallOperationEndToEndTest, RendersSingleDelayedFrame) {
> 
>   class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
>    public:
>-    Renderer() : event_(false, false) {}
>-
>     void OnFrame(const VideoFrame& video_frame) override {
>       SleepMs(kRenderDelayMs);
>       event_.Set();
>@@ -114,15 +112,15 @@ TEST_P(CallOperationEndToEndTest, RendersSingleDelayedFrame) {
> 
>     sender_transport = absl::make_unique<test::DirectTransport>(
>         &task_queue_,
>-        absl::make_unique<FakeNetworkPipe>(
>-            Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())),
>+        absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
>+                                           absl::make_unique<SimulatedNetwork>(
>+                                               BuiltInNetworkBehaviorConfig())),
>         sender_call_.get(), payload_type_map_);
>     receiver_transport = absl::make_unique<test::DirectTransport>(
>         &task_queue_,
>-        absl::make_unique<FakeNetworkPipe>(
>-            Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())),
>+        absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
>+                                           absl::make_unique<SimulatedNetwork>(
>+                                               BuiltInNetworkBehaviorConfig())),
>         receiver_call_.get(), payload_type_map_);
>     sender_transport->SetReceiver(receiver_call_->Receiver());
>     receiver_transport->SetReceiver(sender_call_->Receiver());
>@@ -161,8 +159,6 @@ TEST_P(CallOperationEndToEndTest, RendersSingleDelayedFrame) {
> TEST_P(CallOperationEndToEndTest, TransmitsFirstFrame) {
>   class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
>    public:
>-    Renderer() : event_(false, false) {}
>-
>     void OnFrame(const VideoFrame& video_frame) override { event_.Set(); }
> 
>     bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
>@@ -182,15 +178,15 @@ TEST_P(CallOperationEndToEndTest, TransmitsFirstFrame) {
> 
>     sender_transport = absl::make_unique<test::DirectTransport>(
>         &task_queue_,
>-        absl::make_unique<FakeNetworkPipe>(
>-            Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())),
>+        absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
>+                                           absl::make_unique<SimulatedNetwork>(
>+                                               BuiltInNetworkBehaviorConfig())),
>         sender_call_.get(), payload_type_map_);
>     receiver_transport = absl::make_unique<test::DirectTransport>(
>         &task_queue_,
>-        absl::make_unique<FakeNetworkPipe>(
>-            Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())),
>+        absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
>+                                           absl::make_unique<SimulatedNetwork>(
>+                                               BuiltInNetworkBehaviorConfig())),
>         receiver_call_.get(), payload_type_map_);
>     sender_transport->SetReceiver(receiver_call_->Receiver());
>     receiver_transport->SetReceiver(sender_call_->Receiver());
>@@ -221,79 +217,4 @@ TEST_P(CallOperationEndToEndTest, TransmitsFirstFrame) {
>   });
> }
> 
>-TEST_P(CallOperationEndToEndTest, ObserversEncodedFrames) {
>-  class EncodedFrameTestObserver : public EncodedFrameObserver {
>-   public:
>-    EncodedFrameTestObserver()
>-        : length_(0), frame_type_(kEmptyFrame), called_(false, false) {}
>-    virtual ~EncodedFrameTestObserver() {}
>-
>-    virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
>-      frame_type_ = encoded_frame.frame_type_;
>-      length_ = encoded_frame.length_;
>-      buffer_.reset(new uint8_t[length_]);
>-      memcpy(buffer_.get(), encoded_frame.data_, length_);
>-      called_.Set();
>-    }
>-
>-    bool Wait() { return called_.Wait(kDefaultTimeoutMs); }
>-
>-   private:
>-    std::unique_ptr<uint8_t[]> buffer_;
>-    size_t length_;
>-    FrameType frame_type_;
>-    rtc::Event called_;
>-  };
>-
>-  EncodedFrameTestObserver post_encode_observer;
>-  test::FrameForwarder forwarder;
>-  std::unique_ptr<test::FrameGenerator> frame_generator;
>-
>-  std::unique_ptr<test::DirectTransport> sender_transport;
>-  std::unique_ptr<test::DirectTransport> receiver_transport;
>-
>-  task_queue_.SendTask([&]() {
>-    CreateCalls();
>-
>-    sender_transport = absl::make_unique<test::DirectTransport>(
>-        &task_queue_,
>-        absl::make_unique<FakeNetworkPipe>(
>-            Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())),
>-        sender_call_.get(), payload_type_map_);
>-    receiver_transport = absl::make_unique<test::DirectTransport>(
>-        &task_queue_,
>-        absl::make_unique<FakeNetworkPipe>(
>-            Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())),
>-        receiver_call_.get(), payload_type_map_);
>-    sender_transport->SetReceiver(receiver_call_->Receiver());
>-    receiver_transport->SetReceiver(sender_call_->Receiver());
>-
>-    CreateSendConfig(1, 0, 0, sender_transport.get());
>-    CreateMatchingReceiveConfigs(receiver_transport.get());
>-    GetVideoSendConfig()->post_encode_callback = &post_encode_observer;
>-
>-    CreateVideoStreams();
>-    Start();
>-
>-    frame_generator = test::FrameGenerator::CreateSquareGenerator(
>-        kDefaultWidth, kDefaultHeight, absl::nullopt, absl::nullopt);
>-    GetVideoSendStream()->SetSource(&forwarder,
>-                                    DegradationPreference::MAINTAIN_FRAMERATE);
>-    forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
>-  });
>-
>-  EXPECT_TRUE(post_encode_observer.Wait())
>-      << "Timed out while waiting for send-side encoded-frame callback.";
>-
>-  task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>-    Stop();
>-    DestroyStreams();
>-    sender_transport.reset();
>-    receiver_transport.reset();
>-    DestroyCalls();
>-  });
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/codec_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/codec_tests.cc
>index fad3ac7edef3d0dabc3d2a15f888669f8131fbb8..67ef611a7088823d7d2ddc43332e48a6a0dfe6ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/codec_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/codec_tests.cc
>@@ -8,6 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "media/engine/internaldecoderfactory.h"
> #include "media/engine/internalencoderfactory.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
>@@ -18,7 +19,6 @@
> #include "test/call_test.h"
> #include "test/encoder_settings.h"
> #include "test/field_trial.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -118,7 +118,7 @@ TEST_P(CodecEndToEndTest, SendsAndReceivesVP8Rotation90) {
>   RunBaseTest(&test);
> }
> 
>-#if !defined(RTC_DISABLE_VP9)
>+#if defined(RTC_ENABLE_VP9)
> TEST_P(CodecEndToEndTest, SendsAndReceivesVP9) {
>   test::FunctionVideoEncoderFactory encoder_factory(
>       []() { return VP9Encoder::Create(); });
>@@ -177,7 +177,7 @@ TEST_P(CodecEndToEndTest, SendsAndReceivesMultiplexVideoRotation90) {
>   RunBaseTest(&test);
> }
> 
>-#endif  // !defined(RTC_DISABLE_VP9)
>+#endif  // defined(RTC_ENABLE_VP9)
> 
> #if defined(WEBRTC_USE_H264)
> class EndToEndTestH264 : public test::CallTest,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/config_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/config_tests.cc
>index d32b1118b7c321747124e88792e66b7116299a48..1d33589c9d1ced7cf028207330e32d5732081f38 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/config_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/config_tests.cc
>@@ -49,6 +49,12 @@ TEST_F(ConfigEndToEndTest, VerifyDefaultSendConfigParameters) {
>       << "Enabling RTX requires rtpmap: rtx negotiation.";
>   EXPECT_TRUE(default_send_config.rtp.extensions.empty())
>       << "Enabling RTP extensions require negotiation.";
>+  EXPECT_EQ(nullptr, default_send_config.frame_encryptor)
>+      << "Enabling Frame Encryption requires a frame encryptor to be attached";
>+  EXPECT_FALSE(
>+      default_send_config.crypto_options.sframe.require_frame_encryption)
>+      << "Enabling Require Frame Encryption means an encryptor must be "
>+         "attached";
> 
>   VerifyEmptyNackConfig(default_send_config.rtp.nack);
>   VerifyEmptyUlpfecConfig(default_send_config.rtp.ulpfec);
>@@ -70,12 +76,16 @@ TEST_F(ConfigEndToEndTest, VerifyDefaultVideoReceiveConfigParameters) {
>       << "Enabling RTX requires rtpmap: rtx negotiation.";
>   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
>       << "Enabling RTP extensions require negotiation.";
>-
>   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
>   EXPECT_EQ(-1, default_receive_config.rtp.ulpfec_payload_type)
>       << "Enabling ULPFEC requires rtpmap: ulpfec negotiation.";
>   EXPECT_EQ(-1, default_receive_config.rtp.red_payload_type)
>       << "Enabling ULPFEC requires rtpmap: red negotiation.";
>+  EXPECT_EQ(nullptr, default_receive_config.frame_decryptor)
>+      << "Enabling Frame Decryption requires a frame decryptor to be attached";
>+  EXPECT_FALSE(
>+      default_receive_config.crypto_options.sframe.require_frame_encryption)
>+      << "Enabling Require Frame Encryption means a decryptor must be attached";
> }
> 
> TEST_F(ConfigEndToEndTest, VerifyDefaultFlexfecReceiveConfigParameters) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/extended_reports_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/extended_reports_tests.cc
>index ba5ca693f2b5e8d5ba83fb6d3ea1c1edd61d1822..4de0441b3dfc444ffe3e022d7ebe9483d074ef2d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/extended_reports_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/extended_reports_tests.cc
>@@ -8,9 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "api/video_codecs/video_encoder_config.h"
> #include "call/fake_network_pipe.h"
> #include "call/simulated_network.h"
> #include "test/call_test.h"
>+#include "test/field_trial.h"
> #include "test/gtest.h"
> #include "test/rtcp_packet_parser.h"
> 
>@@ -21,12 +23,14 @@ class ExtendedReportsEndToEndTest : public test::CallTest {};
> class RtcpXrObserver : public test::EndToEndTest {
>  public:
>   RtcpXrObserver(bool enable_rrtr,
>-                 bool enable_target_bitrate,
>-                 bool enable_zero_target_bitrate)
>+                 bool expect_target_bitrate,
>+                 bool enable_zero_target_bitrate,
>+                 VideoEncoderConfig::ContentType content_type)
>       : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
>         enable_rrtr_(enable_rrtr),
>-        enable_target_bitrate_(enable_target_bitrate),
>+        expect_target_bitrate_(expect_target_bitrate),
>         enable_zero_target_bitrate_(enable_zero_target_bitrate),
>+        content_type_(content_type),
>         sent_rtcp_sr_(0),
>         sent_rtcp_rr_(0),
>         sent_rtcp_rrtr_(0),
>@@ -95,7 +99,7 @@ class RtcpXrObserver : public test::EndToEndTest {
> 
>     if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
>         sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve &&
>-        (sent_rtcp_target_bitrate_ || !enable_target_bitrate_) &&
>+        (sent_rtcp_target_bitrate_ || !expect_target_bitrate_) &&
>         (sent_zero_rtcp_target_bitrate_ || !enable_zero_target_bitrate_)) {
>       if (enable_rrtr_) {
>         EXPECT_GT(sent_rtcp_rrtr_, 0);
>@@ -104,7 +108,7 @@ class RtcpXrObserver : public test::EndToEndTest {
>         EXPECT_EQ(sent_rtcp_rrtr_, 0);
>         EXPECT_EQ(sent_rtcp_dlrr_, 0);
>       }
>-      EXPECT_EQ(enable_target_bitrate_, sent_rtcp_target_bitrate_);
>+      EXPECT_EQ(expect_target_bitrate_, sent_rtcp_target_bitrate_);
>       EXPECT_EQ(enable_zero_target_bitrate_, sent_zero_rtcp_target_bitrate_);
>       observation_complete_.Set();
>     }
>@@ -144,10 +148,7 @@ class RtcpXrObserver : public test::EndToEndTest {
>       (*receive_configs)[0].decoders[0].video_format =
>           SdpVideoFormat(send_config->rtp.payload_name);
>     }
>-    if (enable_target_bitrate_) {
>-      // TargetBitrate only signaled for screensharing.
>-      encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
>-    }
>+    encoder_config->content_type = content_type_;
>     (*receive_configs)[0].rtp.rtcp_mode = RtcpMode::kReducedSize;
>     (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
>         enable_rrtr_;
>@@ -162,50 +163,65 @@ class RtcpXrObserver : public test::EndToEndTest {
> 
>   rtc::CriticalSection crit_;
>   const bool enable_rrtr_;
>-  const bool enable_target_bitrate_;
>+  const bool expect_target_bitrate_;
>   const bool enable_zero_target_bitrate_;
>+  const VideoEncoderConfig::ContentType content_type_;
>   int sent_rtcp_sr_;
>   int sent_rtcp_rr_ RTC_GUARDED_BY(&crit_);
>   int sent_rtcp_rrtr_ RTC_GUARDED_BY(&crit_);
>   bool sent_rtcp_target_bitrate_ RTC_GUARDED_BY(&crit_);
>   bool sent_zero_rtcp_target_bitrate_ RTC_GUARDED_BY(&crit_);
>   int sent_rtcp_dlrr_;
>-  DefaultNetworkSimulationConfig forward_transport_config_;
>+  BuiltInNetworkBehaviorConfig forward_transport_config_;
>   SimulatedNetwork* send_simulated_network_;
> };
> 
> TEST_F(ExtendedReportsEndToEndTest,
>        TestExtendedReportsWithRrtrWithoutTargetBitrate) {
>-  RtcpXrObserver test(/*enable_rrtr=*/true, /*enable_target_bitrate=*/false,
>-                      /*enable_zero_target_bitrate=*/false);
>+  RtcpXrObserver test(/*enable_rrtr=*/true, /*expect_target_bitrate=*/false,
>+                      /*enable_zero_target_bitrate=*/false,
>+                      VideoEncoderConfig::ContentType::kRealtimeVideo);
>   RunBaseTest(&test);
> }
> 
> TEST_F(ExtendedReportsEndToEndTest,
>        TestExtendedReportsWithoutRrtrWithoutTargetBitrate) {
>-  RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/false,
>-                      /*enable_zero_target_bitrate=*/false);
>+  RtcpXrObserver test(/*enable_rrtr=*/false, /*expect_target_bitrate=*/false,
>+                      /*enable_zero_target_bitrate=*/false,
>+                      VideoEncoderConfig::ContentType::kRealtimeVideo);
>   RunBaseTest(&test);
> }
> 
> TEST_F(ExtendedReportsEndToEndTest,
>        TestExtendedReportsWithRrtrWithTargetBitrate) {
>-  RtcpXrObserver test(/*enable_rrtr=*/true, /*enable_target_bitrate=*/true,
>-                      /*enable_zero_target_bitrate=*/false);
>+  RtcpXrObserver test(/*enable_rrtr=*/true, /*expect_target_bitrate=*/true,
>+                      /*enable_zero_target_bitrate=*/false,
>+                      VideoEncoderConfig::ContentType::kScreen);
>   RunBaseTest(&test);
> }
> 
> TEST_F(ExtendedReportsEndToEndTest,
>        TestExtendedReportsWithoutRrtrWithTargetBitrate) {
>-  RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/true,
>-                      /*enable_zero_target_bitrate=*/false);
>+  RtcpXrObserver test(/*enable_rrtr=*/false, /*expect_target_bitrate=*/true,
>+                      /*enable_zero_target_bitrate=*/false,
>+                      VideoEncoderConfig::ContentType::kScreen);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(ExtendedReportsEndToEndTest,
>+       TestExtendedReportsWithoutRrtrWithTargetBitrateFromFieldTrial) {
>+  test::ScopedFieldTrials field_trials("WebRTC-Target-Bitrate-Rtcp/Enabled/");
>+  RtcpXrObserver test(/*enable_rrtr=*/false, /*expect_target_bitrate=*/true,
>+                      /*enable_zero_target_bitrate=*/false,
>+                      VideoEncoderConfig::ContentType::kRealtimeVideo);
>   RunBaseTest(&test);
> }
> 
> TEST_F(ExtendedReportsEndToEndTest,
>        TestExtendedReportsCanSignalZeroTargetBitrate) {
>-  RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/true,
>-                      /*enable_zero_target_bitrate=*/true);
>+  RtcpXrObserver test(/*enable_rrtr=*/false, /*expect_target_bitrate=*/true,
>+                      /*enable_zero_target_bitrate=*/true,
>+                      VideoEncoderConfig::ContentType::kScreen);
>   RunBaseTest(&test);
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/fec_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/fec_tests.cc
>index 0300d12f8bcac6d2aa28bebffd125186894383f3..2c53ccc7cbec34099d78809fd57f66f441a693be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/fec_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/fec_tests.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "api/test/simulated_network.h"
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "call/fake_network_pipe.h"
> #include "call/simulated_network.h"
> #include "media/engine/internaldecoderfactory.h"
>@@ -16,7 +17,6 @@
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "test/call_test.h"
> #include "test/field_trial.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/gtest.h"
> #include "test/rtcp_packet_parser.h"
> 
>@@ -234,7 +234,7 @@ class FlexfecRenderObserver : public test::EndToEndTest,
>       Call* sender_call) override {
>     // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
>     const int kNetworkDelayMs = 100;
>-    DefaultNetworkSimulationConfig config;
>+    BuiltInNetworkBehaviorConfig config;
>     config.queue_delay_ms = kNetworkDelayMs;
>     return new test::PacketTransport(
>         task_queue, sender_call, this, test::PacketTransport::kSender,
>@@ -421,7 +421,7 @@ TEST_F(FecEndToEndTest, ReceivedUlpfecPacketsNotNacked) {
>       // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
>       // Configure some network delay.
>       const int kNetworkDelayMs = 50;
>-      DefaultNetworkSimulationConfig config;
>+      BuiltInNetworkBehaviorConfig config;
>       config.queue_delay_ms = kNetworkDelayMs;
>       return new test::PacketTransport(
>           task_queue, sender_call, this, test::PacketTransport::kSender,
>@@ -433,9 +433,10 @@ TEST_F(FecEndToEndTest, ReceivedUlpfecPacketsNotNacked) {
> 
>     // TODO(holmer): Investigate why we don't send FEC packets when the bitrate
>     // is 10 kbps.
>-    void ModifySenderCallConfig(Call::Config* config) override {
>+    void ModifySenderBitrateConfig(
>+        BitrateConstraints* bitrate_config) override {
>       const int kMinBitrateBps = 30000;
>-      config->bitrate_config.min_bitrate_bps = kMinBitrateBps;
>+      bitrate_config->min_bitrate_bps = kMinBitrateBps;
>     }
> 
>     void ModifyVideoConfigs(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/frame_encryption_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/frame_encryption_tests.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..8cd7b478493594012afac17cf3563e97ec4148f6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/frame_encryption_tests.cc
>@@ -0,0 +1,81 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/test/fake_frame_decryptor.h"
>+#include "api/test/fake_frame_encryptor.h"
>+#include "media/engine/internaldecoderfactory.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "test/call_test.h"
>+#include "test/field_trial.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+class FrameEncryptionEndToEndTest : public test::CallTest {
>+ public:
>+  FrameEncryptionEndToEndTest() = default;
>+
>+ private:
>+  // GenericDescriptor is required for FrameEncryption to work.
>+  test::ScopedFieldTrials field_trials_{"WebRTC-GenericDescriptor/Enabled/"};
>+};
>+
>+// Validates that payloads cannot be sent without a frame encryptor and frame
>+// decryptor attached.
>+TEST_F(FrameEncryptionEndToEndTest, RequireFrameEncryptionEnforced) {
>+  class DecryptedFrameObserver : public test::EndToEndTest,
>+                                 public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    DecryptedFrameObserver()
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          encoder_factory_([]() { return VP8Encoder::Create(); }) {}
>+
>+   private:
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      // Use VP8 instead of FAKE.
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+      send_config->rtp.payload_name = "VP8";
>+      send_config->rtp.payload_type = kVideoSendPayloadType;
>+      send_config->frame_encryptor = new FakeFrameEncryptor();
>+      send_config->crypto_options.sframe.require_frame_encryption = true;
>+      encoder_config->codec_type = kVideoCodecVP8;
>+      VideoReceiveStream::Decoder decoder =
>+          test::CreateMatchingDecoder(*send_config);
>+      decoder.decoder_factory = &decoder_factory_;
>+      for (auto& recv_config : *receive_configs) {
>+        recv_config.decoders.clear();
>+        recv_config.decoders.push_back(decoder);
>+        recv_config.renderer = this;
>+        recv_config.frame_decryptor = new FakeFrameDecryptor();
>+        recv_config.crypto_options.sframe.require_frame_encryption = true;
>+      }
>+    }
>+
>+    // Validate that rotation is preserved.
>+    void OnFrame(const VideoFrame& video_frame) override {
>+      observation_complete_.Set();
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait())
>+          << "Timed out waiting for decrypted frames to be rendered.";
>+    }
>+
>+    std::unique_ptr<VideoEncoder> encoder_;
>+    test::FunctionVideoEncoderFactory encoder_factory_;
>+    InternalDecoderFactory decoder_factory_;
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/histogram_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/histogram_tests.cc
>index a4aa0c810a96295a9f00c8f3bec08d7498ff622e..312924c986b862c81aae600ca067a6a91e5e4fa0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/histogram_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/histogram_tests.cc
>@@ -9,10 +9,10 @@
>  */
> 
> #include "absl/types/optional.h"
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "system_wrappers/include/metrics.h"
> #include "test/call_test.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_codec_receive_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_codec_receive_tests.cc
>index e13bc351ca1cc70738a41bc30a332fd1dc7b987a..c58f0fe16a8e2ead05709a86127b39d574c23d9c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_codec_receive_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_codec_receive_tests.cc
>@@ -9,13 +9,13 @@
>  */
> 
> #include "api/test/simulated_network.h"
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "call/fake_network_pipe.h"
> #include "call/simulated_network.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "modules/video_coding/codecs/vp9/include/vp9.h"
> #include "test/call_test.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -51,29 +51,6 @@ int RemoveOlderOrEqual(uint32_t timestamp, std::vector<uint32_t>* timestamps) {
>   return num_removed;
> }
> 
>-class VideoStreamFactoryTest
>-    : public VideoEncoderConfig::VideoStreamFactoryInterface {
>- public:
>-  explicit VideoStreamFactoryTest(size_t num_temporal_layers)
>-      : num_temporal_layers_(num_temporal_layers) {}
>-
>- private:
>-  std::vector<VideoStream> CreateEncoderStreams(
>-      int width,
>-      int height,
>-      const VideoEncoderConfig& encoder_config) override {
>-    std::vector<VideoStream> streams =
>-        test::CreateVideoStreams(width, height, encoder_config);
>-
>-    for (size_t i = 0; i < encoder_config.number_of_streams; ++i)
>-      streams[i].num_temporal_layers = num_temporal_layers_;
>-
>-    return streams;
>-  }
>-
>-  const size_t num_temporal_layers_;
>-};
>-
> class FrameObserver : public test::RtpRtcpObserver,
>                       public rtc::VideoSinkInterface<VideoFrame> {
>  public:
>@@ -153,18 +130,16 @@ class MultiCodecReceiveTest : public test::CallTest {
>           &task_queue_, sender_call_.get(), &observer_,
>           test::PacketTransport::kSender, kPayloadTypeMap,
>           absl::make_unique<FakeNetworkPipe>(
>-              Clock::GetRealTimeClock(),
>-              absl::make_unique<SimulatedNetwork>(
>-                  DefaultNetworkSimulationConfig()))));
>+              Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>+                                             BuiltInNetworkBehaviorConfig()))));
>       send_transport_->SetReceiver(receiver_call_->Receiver());
> 
>       receive_transport_.reset(new test::PacketTransport(
>           &task_queue_, receiver_call_.get(), &observer_,
>           test::PacketTransport::kReceiver, kPayloadTypeMap,
>           absl::make_unique<FakeNetworkPipe>(
>-              Clock::GetRealTimeClock(),
>-              absl::make_unique<SimulatedNetwork>(
>-                  DefaultNetworkSimulationConfig()))));
>+              Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>+                                             BuiltInNetworkBehaviorConfig()))));
>       receive_transport_->SetReceiver(sender_call_->Receiver());
>     });
>   }
>@@ -219,9 +194,9 @@ void MultiCodecReceiveTest::ConfigureEncoder(const CodecConfig& config) {
>       PayloadNameToPayloadType(config.payload_name);
>   GetVideoEncoderConfig()->codec_type =
>       PayloadStringToCodecType(config.payload_name);
>-  GetVideoEncoderConfig()->video_stream_factory =
>-      new rtc::RefCountedObject<VideoStreamFactoryTest>(
>-          config.num_temporal_layers);
>+  EXPECT_EQ(1u, GetVideoEncoderConfig()->simulcast_layers.size());
>+  GetVideoEncoderConfig()->simulcast_layers[0].num_temporal_layers =
>+      config.num_temporal_layers;
> }
> 
> void MultiCodecReceiveTest::RunTestWithCodecs(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tester.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tester.cc
>index 7d66976001adb2fd456467ce7f4cb76ebbf252a2..5d5da65ca8f7f80512ecc0f6a80dd56ea5efbf3f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tester.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tester.cc
>@@ -15,6 +15,8 @@
> #include <vector>
> 
> #include "api/test/simulated_network.h"
>+#include "api/test/video/function_video_encoder_factory.h"
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "call/fake_network_pipe.h"
> #include "call/simulated_network.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
>@@ -22,7 +24,6 @@
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "test/call_test.h"
> #include "test/encoder_settings.h"
>-#include "test/function_video_encoder_factory.h"
> 
> namespace webrtc {
> 
>@@ -54,6 +55,8 @@ void MultiStreamTester::RunTest() {
>   test::FrameGeneratorCapturer* frame_generators[kNumStreams];
>   test::FunctionVideoEncoderFactory encoder_factory(
>       []() { return VP8Encoder::Create(); });
>+  std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory =
>+      CreateBuiltinVideoBitrateAllocatorFactory();
>   InternalDecoderFactory decoder_factory;
> 
>   task_queue_->SendTask([&]() {
>@@ -75,6 +78,8 @@ void MultiStreamTester::RunTest() {
>       VideoSendStream::Config send_config(sender_transport.get());
>       send_config.rtp.ssrcs.push_back(ssrc);
>       send_config.encoder_settings.encoder_factory = &encoder_factory;
>+      send_config.encoder_settings.bitrate_allocator_factory =
>+          bitrate_allocator_factory.get();
>       send_config.rtp.payload_name = "VP8";
>       send_config.rtp.payload_type = kVideoPayloadType;
>       VideoEncoderConfig encoder_config;
>@@ -143,9 +148,9 @@ test::DirectTransport* MultiStreamTester::CreateSendTransport(
>     Call* sender_call) {
>   return new test::DirectTransport(
>       task_queue,
>-      absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
>-                                         absl::make_unique<SimulatedNetwork>(
>-                                             DefaultNetworkSimulationConfig())),
>+      absl::make_unique<FakeNetworkPipe>(
>+          Clock::GetRealTimeClock(),
>+          absl::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())),
>       sender_call, payload_type_map_);
> }
> 
>@@ -154,9 +159,9 @@ test::DirectTransport* MultiStreamTester::CreateReceiveTransport(
>     Call* receiver_call) {
>   return new test::DirectTransport(
>       task_queue,
>-      absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
>-                                         absl::make_unique<SimulatedNetwork>(
>-                                             DefaultNetworkSimulationConfig())),
>+      absl::make_unique<FakeNetworkPipe>(
>+          Clock::GetRealTimeClock(),
>+          absl::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())),
>       receiver_call, payload_type_map_);
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tests.cc
>index dbead24564fe0d70be0001faf12592639cab4c5b..9c754fb70a57de67fd02a04420e87f1965c35d0f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tests.cc
>@@ -30,10 +30,7 @@ TEST_F(MultiStreamEndToEndTest, SendsAndReceivesMultipleStreams) {
>     VideoOutputObserver(const MultiStreamTester::CodecSettings& settings,
>                         uint32_t ssrc,
>                         test::FrameGeneratorCapturer** frame_generator)
>-        : settings_(settings),
>-          ssrc_(ssrc),
>-          frame_generator_(frame_generator),
>-          done_(false, false) {}
>+        : settings_(settings), ssrc_(ssrc), frame_generator_(frame_generator) {}
> 
>     void OnFrame(const VideoFrame& video_frame) override {
>       EXPECT_EQ(settings_.width, video_frame.width());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/network_state_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/network_state_tests.cc
>index 937777f6b3479276506739d25a3869c0cea6c0c8..d66634a135f32aa66bb032fa13ba9f7b13d2241e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/network_state_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/network_state_tests.cc
>@@ -118,9 +118,9 @@ void NetworkStateEndToEndTest::VerifyNewVideoReceiveStreamsRespectNetworkState(
>     receiver_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
>     sender_transport = absl::make_unique<test::DirectTransport>(
>         &task_queue_,
>-        absl::make_unique<FakeNetworkPipe>(
>-            Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())),
>+        absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
>+                                           absl::make_unique<SimulatedNetwork>(
>+                                               BuiltInNetworkBehaviorConfig())),
>         sender_call_.get(), payload_type_map_);
>     sender_transport->SetReceiver(receiver_call_->Receiver());
>     CreateSendConfig(1, 0, 0, sender_transport.get());
>@@ -158,8 +158,6 @@ TEST_F(NetworkStateEndToEndTest, RespectsNetworkState) {
>         : EndToEndTest(kDefaultTimeoutMs),
>           FakeEncoder(Clock::GetRealTimeClock()),
>           task_queue_(task_queue),
>-          encoded_frames_(false, false),
>-          packet_event_(false, false),
>           sender_call_(nullptr),
>           receiver_call_(nullptr),
>           encoder_factory_(this),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/probing_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/probing_tests.cc
>index 7dff657ad0397cbf96340a94cf437f60620c0c94..7970d4a1bf7989b95ffc69d3f76cb9b296beb742 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/probing_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/probing_tests.cc
>@@ -41,8 +41,8 @@ class ProbingTest : public test::EndToEndTest {
>         state_(0),
>         sender_call_(nullptr) {}
> 
>-  void ModifySenderCallConfig(Call::Config* config) override {
>-    config->bitrate_config.start_bitrate_bps = start_bitrate_bps_;
>+  void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) override {
>+    bitrate_config->start_bitrate_bps = start_bitrate_bps_;
>   }
> 
>   void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>@@ -221,7 +221,7 @@ TEST_P(ProbingEndToEndTest, ProbeOnVideoEncoderReconfiguration) {
>         test::SingleThreadedTaskQueueForTesting* task_queue,
>         Call* sender_call) override {
>       auto network =
>-          absl::make_unique<SimulatedNetwork>(DefaultNetworkSimulationConfig());
>+          absl::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig());
>       send_simulated_network_ = network.get();
>       return new test::PacketTransport(
>           task_queue, sender_call, this, test::PacketTransport::kSender,
>@@ -245,7 +245,7 @@ TEST_P(ProbingEndToEndTest, ProbeOnVideoEncoderReconfiguration) {
>             // bitrate).
>             if (stats.send_bandwidth_bps >= 250000 &&
>                 stats.send_bandwidth_bps <= 350000) {
>-              DefaultNetworkSimulationConfig config;
>+              BuiltInNetworkBehaviorConfig config;
>               config.link_capacity_kbps = 200;
>               send_simulated_network_->SetConfig(config);
> 
>@@ -260,7 +260,7 @@ TEST_P(ProbingEndToEndTest, ProbeOnVideoEncoderReconfiguration) {
>             break;
>           case 1:
>             if (stats.send_bandwidth_bps <= 210000) {
>-              DefaultNetworkSimulationConfig config;
>+              BuiltInNetworkBehaviorConfig config;
>               config.link_capacity_kbps = 5000;
>               send_simulated_network_->SetConfig(config);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/receive_time_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/receive_time_tests.cc
>deleted file mode 100644
>index 3ca66f1d3d67e276e57b0a88308cf18470bab4d0..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/receive_time_tests.cc
>+++ /dev/null
>@@ -1,149 +0,0 @@
>-/*
>- *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/test/simulated_network.h"
>-#include "call/fake_network_pipe.h"
>-#include "call/simulated_network.h"
>-#include "rtc_base/criticalsection.h"
>-#include "rtc_base/timeutils.h"
>-#include "test/call_test.h"
>-#include "test/field_trial.h"
>-#include "test/rtcp_packet_parser.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-// This tester simulates a series of clock reset events where different offsets
>-// are added to the receive time. It detects jumps in the resulting reported
>-// receive times of more than 200 ms.
>-class ReportedReceiveTimeTester : public test::EndToEndTest {
>- public:
>-  struct TimeJump {
>-    int64_t at_send_time_ms;
>-    int64_t add_offset_ms;
>-    static constexpr int64_t kStop = 0;
>-  };
>-
>-  ReportedReceiveTimeTester()
>-      : EndToEndTest(test::CallTest::kDefaultTimeoutMs) {
>-    // These should be let trough without correction and filtered if correction
>-    // is enabled.
>-    jumps_.push({500, 2000});
>-    jumps_.push({1000, -400});
>-    jumps_.push({1500, 2000000});
>-    jumps_.push({1700, TimeJump::kStop});
>-  }
>-  bool JumpInReportedTimes() { return jump_in_reported_times_; }
>-
>- protected:
>-  Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
>-    test::RtcpPacketParser parser;
>-    EXPECT_TRUE(parser.Parse(data, length));
>-    const auto& fb = parser.transport_feedback();
>-    if (fb->num_packets() > 0) {
>-      int64_t arrival_time_us = fb->GetBaseTimeUs();
>-      for (const auto& pkt : fb->GetReceivedPackets()) {
>-        arrival_time_us += pkt.delta_us();
>-        if (last_arrival_time_us_ != 0) {
>-          int64_t delta_us = arrival_time_us - last_arrival_time_us_;
>-          rtc::CritScope crit(&send_times_crit_);
>-          if (send_times_us_.size() >= 2) {
>-            int64_t ground_truth_delta_us =
>-                send_times_us_[1] - send_times_us_[0];
>-            send_times_us_.pop_front();
>-            int64_t delta_diff_ms = (delta_us - ground_truth_delta_us) / 1000;
>-            if (std::abs(delta_diff_ms) > 200) {
>-              jump_in_reported_times_ = true;
>-              observation_complete_.Set();
>-            }
>-          }
>-        }
>-        last_arrival_time_us_ = arrival_time_us;
>-      }
>-    }
>-    return SEND_PACKET;
>-  }
>-  Action OnSendRtp(const uint8_t* data, size_t length) override {
>-    {
>-      rtc::CritScope crit(&send_times_crit_);
>-      send_times_us_.push_back(rtc::TimeMicros());
>-    }
>-    int64_t now_ms = rtc::TimeMillis();
>-    if (!first_send_time_ms_)
>-      first_send_time_ms_ = now_ms;
>-    int64_t send_time_ms = now_ms - first_send_time_ms_;
>-    if (send_time_ms >= jumps_.front().at_send_time_ms) {
>-      if (jumps_.front().add_offset_ms == TimeJump::kStop) {
>-        observation_complete_.Set();
>-        jumps_.pop();
>-        return SEND_PACKET;
>-      }
>-      clock_offset_ms_ += jumps_.front().add_offset_ms;
>-      send_pipe_->SetClockOffset(clock_offset_ms_);
>-      jumps_.pop();
>-    }
>-    return SEND_PACKET;
>-  }
>-  test::PacketTransport* CreateSendTransport(
>-      test::SingleThreadedTaskQueueForTesting* task_queue,
>-      Call* sender_call) override {
>-    auto pipe = absl::make_unique<FakeNetworkPipe>(
>-        Clock::GetRealTimeClock(),
>-        absl::make_unique<SimulatedNetwork>(DefaultNetworkSimulationConfig()));
>-    send_pipe_ = pipe.get();
>-    return send_transport_ = new test::PacketTransport(
>-               task_queue, sender_call, this, test::PacketTransport::kSender,
>-               test::CallTest::payload_type_map_, std::move(pipe));
>-  }
>-  void PerformTest() override {
>-    observation_complete_.Wait(test::CallTest::kDefaultTimeoutMs);
>-  }
>-  size_t GetNumVideoStreams() const override { return 1; }
>-  size_t GetNumAudioStreams() const override { return 0; }
>-
>- private:
>-  int64_t last_arrival_time_us_ = 0;
>-  int64_t first_send_time_ms_ = 0;
>-  rtc::CriticalSection send_times_crit_;
>-  std::deque<int64_t> send_times_us_ RTC_GUARDED_BY(send_times_crit_);
>-  bool jump_in_reported_times_ = false;
>-  FakeNetworkPipe* send_pipe_;
>-  test::PacketTransport* send_transport_;
>-  int64_t clock_offset_ms_ = 0;
>-  std::queue<TimeJump> jumps_;
>-};
>-}  // namespace
>-
>-class ReceiveTimeEndToEndTest : public test::CallTest {
>- public:
>-  ReceiveTimeEndToEndTest() {}
>-
>-  virtual ~ReceiveTimeEndToEndTest() {}
>-};
>-
>-TEST_F(ReceiveTimeEndToEndTest, ReceiveTimeJumpsWithoutFieldTrial) {
>-  // Without the field trial, the jumps in clock offset should be let trough and
>-  // be detected.
>-  ReportedReceiveTimeTester test;
>-  RunBaseTest(&test);
>-  EXPECT_TRUE(test.JumpInReportedTimes());
>-}
>-
>-TEST_F(ReceiveTimeEndToEndTest, ReceiveTimeSteadyWithFieldTrial) {
>-  // Since all the added jumps by the tester are outside the interval of -100 ms
>-  // to 1000 ms, they should all be filtered by the field trial below, and no
>-  // jumps should be detected.
>-  test::ScopedFieldTrials field_trial(
>-      "WebRTC-BweReceiveTimeCorrection/Enabled,-100,1000/");
>-  ReportedReceiveTimeTester test;
>-  RunBaseTest(&test);
>-  EXPECT_FALSE(test.JumpInReportedTimes());
>-}
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/retransmission_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/retransmission_tests.cc
>index 73a5585e747191b2e385867abf27257475728ef1..d1a69e3ed3181c9f5bc5da78b976e6f876598f1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/retransmission_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/retransmission_tests.cc
>@@ -9,13 +9,13 @@
>  */
> 
> #include "api/test/simulated_network.h"
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "call/fake_network_pipe.h"
> #include "call/simulated_network.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "system_wrappers/include/sleep.h"
> #include "test/call_test.h"
> #include "test/field_trial.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/gtest.h"
> #include "test/rtcp_packet_parser.h"
> 
>@@ -131,9 +131,8 @@ TEST_F(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
>           task_queue, nullptr, this, test::PacketTransport::kReceiver,
>           payload_type_map_,
>           absl::make_unique<FakeNetworkPipe>(
>-              Clock::GetRealTimeClock(),
>-              absl::make_unique<SimulatedNetwork>(
>-                  DefaultNetworkSimulationConfig())));
>+              Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>+                                             BuiltInNetworkBehaviorConfig())));
>       receive_transport_ = receive_transport;
>       return receive_transport;
>     }
>@@ -166,7 +165,6 @@ TEST_F(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
>     void ModifyAudioConfigs(
>         AudioSendStream::Config* send_config,
>         std::vector<AudioReceiveStream::Config>* receive_configs) override {
>-      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>       local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
>       remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/rtp_rtcp_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/rtp_rtcp_tests.cc
>index 7d96dcca7610eda2ba258ba4ccf28121c8bbbcbb..f19c6ad3faa5f6a41870c3858da8b540ab2bda63 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/rtp_rtcp_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/rtp_rtcp_tests.cc
>@@ -282,13 +282,13 @@ void RtpRtcpEndToEndTest::TestRtpStatePreservation(
>         test::PacketTransport::kSender, payload_type_map_,
>         absl::make_unique<FakeNetworkPipe>(
>             Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())));
>+                                           BuiltInNetworkBehaviorConfig())));
>     receive_transport = absl::make_unique<test::PacketTransport>(
>         &task_queue_, nullptr, &observer, test::PacketTransport::kReceiver,
>         payload_type_map_,
>         absl::make_unique<FakeNetworkPipe>(
>             Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())));
>+                                           BuiltInNetworkBehaviorConfig())));
>     send_transport->SetReceiver(receiver_call_->Receiver());
>     receive_transport->SetReceiver(sender_call_->Receiver());
> 
>@@ -474,7 +474,7 @@ TEST_F(RtpRtcpEndToEndTest, DISABLED_TestFlexfecRtpStatePreservation) {
>   task_queue_.SendTask([&]() {
>     CreateCalls();
> 
>-    DefaultNetworkSimulationConfig lossy_delayed_link;
>+    BuiltInNetworkBehaviorConfig lossy_delayed_link;
>     lossy_delayed_link.loss_percent = 2;
>     lossy_delayed_link.queue_delay_ms = 50;
> 
>@@ -486,7 +486,7 @@ TEST_F(RtpRtcpEndToEndTest, DISABLED_TestFlexfecRtpStatePreservation) {
>             absl::make_unique<SimulatedNetwork>(lossy_delayed_link)));
>     send_transport->SetReceiver(receiver_call_->Receiver());
> 
>-    DefaultNetworkSimulationConfig flawless_link;
>+    BuiltInNetworkBehaviorConfig flawless_link;
>     receive_transport = absl::make_unique<test::PacketTransport>(
>         &task_queue_, nullptr, &observer, test::PacketTransport::kReceiver,
>         payload_type_map_,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/ssrc_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/ssrc_tests.cc
>index c03b49477d60a8be8285519d3eaf962df292abff..86faa0465b5fb3c1d8dda882ab39cefd9a929957 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/ssrc_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/ssrc_tests.cc
>@@ -48,7 +48,7 @@ TEST_F(SsrcEndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
>   class PacketInputObserver : public PacketReceiver {
>    public:
>     explicit PacketInputObserver(PacketReceiver* receiver)
>-        : receiver_(receiver), delivered_packet_(false, false) {}
>+        : receiver_(receiver) {}
> 
>     bool Wait() { return delivered_packet_.Wait(kDefaultTimeoutMs); }
> 
>@@ -82,15 +82,15 @@ TEST_F(SsrcEndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
> 
>     send_transport = absl::make_unique<test::DirectTransport>(
>         &task_queue_,
>-        absl::make_unique<FakeNetworkPipe>(
>-            Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())),
>+        absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
>+                                           absl::make_unique<SimulatedNetwork>(
>+                                               BuiltInNetworkBehaviorConfig())),
>         sender_call_.get(), payload_type_map_);
>     receive_transport = absl::make_unique<test::DirectTransport>(
>         &task_queue_,
>-        absl::make_unique<FakeNetworkPipe>(
>-            Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig())),
>+        absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
>+                                           absl::make_unique<SimulatedNetwork>(
>+                                               BuiltInNetworkBehaviorConfig())),
>         receiver_call_.get(), payload_type_map_);
>     input_observer =
>         absl::make_unique<PacketInputObserver>(receiver_call_->Receiver());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/stats_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/stats_tests.cc
>index af6bbda168e244dec7760fe8a51405ea45980379..d6e92c8af203aef5cbef54417fbd2800946b9ef5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/stats_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/stats_tests.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "api/test/simulated_network.h"
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "call/fake_network_pipe.h"
> #include "call/simulated_network.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
>@@ -18,7 +19,6 @@
> #include "system_wrappers/include/sleep.h"
> #include "test/call_test.h"
> #include "test/fake_encoder.h"
>-#include "test/function_video_encoder_factory.h"
> #include "test/gtest.h"
> #include "test/rtcp_packet_parser.h"
> 
>@@ -47,8 +47,7 @@ TEST_F(StatsEndToEndTest, GetStats) {
>                 Clock::GetRealTimeClock(), 10);
>           }),
>           send_stream_(nullptr),
>-          expected_send_ssrcs_(),
>-          check_stats_event_(false, false) {}
>+          expected_send_ssrcs_() {}
> 
>    private:
>     Action OnSendRtp(const uint8_t* packet, size_t length) override {
>@@ -231,7 +230,7 @@ TEST_F(StatsEndToEndTest, GetStats) {
>     test::PacketTransport* CreateSendTransport(
>         test::SingleThreadedTaskQueueForTesting* task_queue,
>         Call* sender_call) override {
>-      DefaultNetworkSimulationConfig network_config;
>+      BuiltInNetworkBehaviorConfig network_config;
>       network_config.loss_percent = 5;
>       return new test::PacketTransport(
>           task_queue, sender_call, this, test::PacketTransport::kSender,
>@@ -240,8 +239,9 @@ TEST_F(StatsEndToEndTest, GetStats) {
>               Clock::GetRealTimeClock(),
>               absl::make_unique<SimulatedNetwork>(network_config)));
>     }
>-    void ModifySenderCallConfig(Call::Config* config) override {
>-      config->bitrate_config.start_bitrate_bps = kStartBitrateBps;
>+    void ModifySenderBitrateConfig(
>+        BitrateConstraints* bitrate_config) override {
>+      bitrate_config->start_bitrate_bps = kStartBitrateBps;
>     }
> 
>     // This test use other VideoStream settings than the the default settings
>@@ -515,9 +515,9 @@ TEST_F(StatsEndToEndTest, MAYBE_ContentTypeSwitches) {
>   metrics::Reset();
> 
>   Call::Config send_config(send_event_log_.get());
>-  test.ModifySenderCallConfig(&send_config);
>+  test.ModifySenderBitrateConfig(&send_config.bitrate_config);
>   Call::Config recv_config(recv_event_log_.get());
>-  test.ModifyReceiverCallConfig(&recv_config);
>+  test.ModifyReceiverBitrateConfig(&recv_config.bitrate_config);
> 
>   VideoEncoderConfig encoder_config_with_screenshare;
> 
>@@ -713,7 +713,7 @@ TEST_F(StatsEndToEndTest, CallReportsRttForSender) {
>   std::unique_ptr<test::DirectTransport> receiver_transport;
> 
>   task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>-    DefaultNetworkSimulationConfig config;
>+    BuiltInNetworkBehaviorConfig config;
>     config.queue_delay_ms = kSendDelayMs;
>     CreateCalls();
>     sender_transport = absl::make_unique<test::DirectTransport>(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/transport_feedback_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/transport_feedback_tests.cc
>index 1437cd46e485eb4f3699071239ede28bce2efa14..94db0383305198e6016f567d824c547811d0d446 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/transport_feedback_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/transport_feedback_tests.cc
>@@ -55,10 +55,9 @@ TEST_P(TransportFeedbackEndToEndTest, AssignsTransportSequenceNumbers) {
>                           absl::make_unique<FakeNetworkPipe>(
>                               Clock::GetRealTimeClock(),
>                               absl::make_unique<SimulatedNetwork>(
>-                                  DefaultNetworkSimulationConfig())),
>+                                  BuiltInNetworkBehaviorConfig())),
>                           sender_call,
>                           payload_type_map),
>-          done_(false, false),
>           parser_(RtpHeaderParser::Create()),
>           first_media_ssrc_(first_media_ssrc),
>           rtx_to_media_ssrcs_(ssrc_map),
>@@ -415,8 +414,9 @@ TEST_P(TransportFeedbackEndToEndTest,
>       EXPECT_TRUE(parser.Parse(data, length));
>       return parser.transport_feedback()->num_packets() > 0;
>     }
>-    void ModifySenderCallConfig(Call::Config* config) override {
>-      config->bitrate_config.max_bitrate_bps = 300000;
>+    void ModifySenderBitrateConfig(
>+        BitrateConstraints* bitrate_config) override {
>+      bitrate_config->max_bitrate_bps = 300000;
>     }
> 
>     void PerformTest() override {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/frame_dumping_decoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/frame_dumping_decoder.cc
>new file mode 100644
>index 0000000000000000000000000000000000000000..7798511b5439a3ba71cfc8ecc3062ebd03794e2e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/frame_dumping_decoder.cc
>@@ -0,0 +1,61 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "video/frame_dumping_decoder.h"
>+
>+#include <utility>
>+
>+#include "modules/video_coding/include/video_codec_interface.h"
>+
>+namespace webrtc {
>+
>+FrameDumpingDecoder::FrameDumpingDecoder(std::unique_ptr<VideoDecoder> decoder,
>+                                         rtc::PlatformFile file)
>+    : decoder_(std::move(decoder)),
>+      writer_(IvfFileWriter::Wrap(rtc::File(file),
>+                                  /* byte_limit= */ 100000000)) {}
>+
>+FrameDumpingDecoder::~FrameDumpingDecoder() = default;
>+
>+int32_t FrameDumpingDecoder::InitDecode(const VideoCodec* codec_settings,
>+                                        int32_t number_of_cores) {
>+  return decoder_->InitDecode(codec_settings, number_of_cores);
>+}
>+
>+int32_t FrameDumpingDecoder::Decode(
>+    const EncodedImage& input_image,
>+    bool missing_frames,
>+    const CodecSpecificInfo* codec_specific_info,
>+    int64_t render_time_ms) {
>+  int32_t ret = decoder_->Decode(input_image, missing_frames,
>+                                 codec_specific_info, render_time_ms);
>+  writer_->WriteFrame(input_image, codec_specific_info->codecType);
>+
>+  return ret;
>+}
>+
>+int32_t FrameDumpingDecoder::RegisterDecodeCompleteCallback(
>+    DecodedImageCallback* callback) {
>+  return decoder_->RegisterDecodeCompleteCallback(callback);
>+}
>+
>+int32_t FrameDumpingDecoder::Release() {
>+  return decoder_->Release();
>+}
>+
>+bool FrameDumpingDecoder::PrefersLateDecoding() const {
>+  return decoder_->PrefersLateDecoding();
>+}
>+
>+const char* FrameDumpingDecoder::ImplementationName() const {
>+  return decoder_->ImplementationName();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/frame_dumping_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/frame_dumping_decoder.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..e19e9b4db089d835a2f3eb8255940e271fec090b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/frame_dumping_decoder.h
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef VIDEO_FRAME_DUMPING_DECODER_H_
>+#define VIDEO_FRAME_DUMPING_DECODER_H_
>+
>+#include <memory>
>+
>+#include "api/video_codecs/video_decoder.h"
>+#include "modules/video_coding/utility/ivf_file_writer.h"
>+#include "rtc_base/platform_file.h"
>+
>+namespace webrtc {
>+
>+// A decoder wrapper that writes the encoded frames to a file.
>+class FrameDumpingDecoder : public VideoDecoder {
>+ public:
>+  FrameDumpingDecoder(std::unique_ptr<VideoDecoder> decoder,
>+                      rtc::PlatformFile file);
>+  ~FrameDumpingDecoder() override;
>+
>+  int32_t InitDecode(const VideoCodec* codec_settings,
>+                     int32_t number_of_cores) override;
>+  int32_t Decode(const EncodedImage& input_image,
>+                 bool missing_frames,
>+                 const CodecSpecificInfo* codec_specific_info,
>+                 int64_t render_time_ms) override;
>+  int32_t RegisterDecodeCompleteCallback(
>+      DecodedImageCallback* callback) override;
>+  int32_t Release() override;
>+  bool PrefersLateDecoding() const override;
>+  const char* ImplementationName() const override;
>+
>+ private:
>+  std::unique_ptr<VideoDecoder> decoder_;
>+  std::unique_ptr<IvfFileWriter> writer_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // VIDEO_FRAME_DUMPING_DECODER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests.cc
>index 77da2534489a2aae8027dc79ea838a12d1eb967c..9a8d252bfcc51c6fad13b5416901618702104e94 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests.cc
>@@ -22,21 +22,24 @@
> namespace webrtc {
> namespace flags {
> 
>-DEFINE_string(rtc_event_log_name,
>-              "",
>-              "Filename for rtc event log. Two files "
>-              "with \"_send\" and \"_recv\" suffixes will be created.");
>+WEBRTC_DEFINE_string(rtc_event_log_name,
>+                     "",
>+                     "Filename for rtc event log. Two files "
>+                     "with \"_send\" and \"_recv\" suffixes will be created.");
> std::string RtcEventLogName() {
>   return static_cast<std::string>(FLAG_rtc_event_log_name);
> }
>-DEFINE_string(rtp_dump_name, "", "Filename for dumped received RTP stream.");
>+WEBRTC_DEFINE_string(rtp_dump_name,
>+                     "",
>+                     "Filename for dumped received RTP stream.");
> std::string RtpDumpName() {
>   return static_cast<std::string>(FLAG_rtp_dump_name);
> }
>-DEFINE_string(encoded_frame_path,
>-              "",
>-              "The base path for encoded frame logs. Created files will have "
>-              "the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf");
>+WEBRTC_DEFINE_string(
>+    encoded_frame_path,
>+    "",
>+    "The base path for encoded frame logs. Created files will have "
>+    "the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf");
> std::string EncodedFramePath() {
>   return static_cast<std::string>(FLAG_encoded_frame_path);
> }
>@@ -47,10 +50,10 @@ namespace webrtc {
> 
> namespace {
> static const int kFullStackTestDurationSecs = 45;
>-const char kScreenshareSimulcastExperiment[] =
>-    "WebRTC-SimulcastScreenshare/Enabled/";
> const char kPacerPushBackExperiment[] =
>     "WebRTC-PacerPushbackExperiment/Enabled/";
>+const char kVp8TrustedRateControllerFieldTrial[] =
>+    "WebRTC-LibvpxVp8TrustedRateController/Enabled/";
> 
> struct ParamsWithLogging : public VideoQualityTest::Params {
>  public:
>@@ -58,7 +61,7 @@ struct ParamsWithLogging : public VideoQualityTest::Params {
>     // Use these logging flags by default, for everything.
>     logging = {flags::RtcEventLogName(), flags::RtpDumpName(),
>                flags::EncodedFramePath()};
>-    this->config = DefaultNetworkSimulationConfig();
>+    this->config = BuiltInNetworkBehaviorConfig();
>   }
> };
> 
>@@ -106,7 +109,7 @@ class GenericDescriptorTest : public ::testing::TestWithParam<std::string> {
>   bool generic_descriptor_enabled_;
> };
> 
>-#if !defined(RTC_DISABLE_VP9)
>+#if defined(RTC_ENABLE_VP9)
> TEST(FullStackTest, ForemanCifWithoutPacketLossVp9) {
>   auto fixture = CreateVideoQualityTestFixture();
>   ParamsWithLogging foreman_cif;
>@@ -187,7 +190,7 @@ TEST(FullStackTest, GeneratorWithoutPacketLossMultiplexI420AFrame) {
>   fixture->RunWithAnalyzer(generator);
> }
> 
>-#endif  // !defined(RTC_DISABLE_VP9)
>+#endif  // defined(RTC_ENABLE_VP9)
> 
> #if defined(WEBRTC_LINUX)
> // Crashes on the linux trusty perf bot: bugs.webrtc.org/9129.
>@@ -234,6 +237,25 @@ TEST_P(GenericDescriptorTest, ForemanCif30kbpsWithoutPacketLoss) {
>   fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>+// TODO(webrtc:9722): Remove when experiment is cleaned up.
>+TEST_P(GenericDescriptorTest,
>+       ForemanCif30kbpsWithoutPacketLossTrustedRateControl) {
>+  test::ScopedFieldTrials override_field_trials(
>+      AppendFieldTrials(kVp8TrustedRateControllerFieldTrial));
>+  auto fixture = CreateVideoQualityTestFixture();
>+
>+  ParamsWithLogging foreman_cif;
>+  foreman_cif.call.send_side_bwe = true;
>+  foreman_cif.video[0] = {true,  352,          288, 10, 30000, 30000, 30000,
>+                          false, "VP8",        1,   0,  0,     false, false,
>+                          false, "foreman_cif"};
>+  foreman_cif.analyzer = {
>+      GetTestName("foreman_cif_30kbps_net_delay_0_0_plr_0_trusted_rate_ctrl"),
>+      0.0, 0.0, kFullStackTestDurationSecs};
>+  foreman_cif.call.generic_descriptor = GenericDescriptorEnabled();
>+  fixture->RunWithAnalyzer(foreman_cif);
>+}
>+
> // Link capacity below default start rate. Automatic down scaling enabled.
> TEST(FullStackTest, ForemanCifLink150kbpsWithoutPacketLoss) {
>   auto fixture = CreateVideoQualityTestFixture();
>@@ -365,9 +387,9 @@ TEST_P(GenericDescriptorTest, ForemanCifPlr5H264) {
> }
> 
> TEST(FullStackTest, ForemanCifPlr5H264SpsPpsIdrIsKeyframe) {
>-  auto fixture = CreateVideoQualityTestFixture();
>   test::ScopedFieldTrials override_field_trials(
>       AppendFieldTrials("WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/"));
>+  auto fixture = CreateVideoQualityTestFixture();
> 
>   ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>@@ -521,16 +543,21 @@ TEST(FullStackTest, ConferenceMotionHd2000kbps100msLimitedQueue) {
>   fixture->RunWithAnalyzer(conf_motion_hd);
> }
> 
>-TEST(FullStackTest, ConferenceMotionHd1TLModerateLimits) {
>+// TODO(webrtc:9722): Remove when experiment is cleaned up.
>+TEST(FullStackTest, ConferenceMotionHd1TLModerateLimitsWhitelistVp8) {
>+  test::ScopedFieldTrials override_field_trials(
>+      AppendFieldTrials(kVp8TrustedRateControllerFieldTrial));
>   auto fixture = CreateVideoQualityTestFixture();
>+
>   ParamsWithLogging conf_motion_hd;
>   conf_motion_hd.call.send_side_bwe = true;
>   conf_motion_hd.video[0] = {
>       true,    1280,    720,   50,    30000,
>       3000000, 3000000, false, "VP8", 1,
>       -1,      0,       false, false, false, "ConferenceMotion_1280_720_50"};
>-  conf_motion_hd.analyzer = {"conference_motion_hd_1tl_moderate_limits", 0.0,
>-                             0.0, kFullStackTestDurationSecs};
>+  conf_motion_hd.analyzer = {
>+      "conference_motion_hd_1tl_moderate_limits_trusted_rate_ctrl", 0.0, 0.0,
>+      kFullStackTestDurationSecs};
>   conf_motion_hd.config->queue_length_packets = 50;
>   conf_motion_hd.config->loss_percent = 3;
>   conf_motion_hd.config->queue_delay_ms = 100;
>@@ -592,17 +619,40 @@ TEST(FullStackTest, ConferenceMotionHd4TLModerateLimits) {
> }
> 
> TEST(FullStackTest, ConferenceMotionHd3TLModerateLimitsAltTLPattern) {
>-  auto fixture = CreateVideoQualityTestFixture();
>   test::ScopedFieldTrials field_trial(
>       AppendFieldTrials("WebRTC-UseShortVP8TL3Pattern/Enabled/"));
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging conf_motion_hd;
>+  conf_motion_hd.call.send_side_bwe = true;
>+  conf_motion_hd.video[0] = {
>+      true,  1280,    720,     50,
>+      30000, 3000000, 3000000, false,
>+      "VP8", 3,       -1,      0,
>+      false, false,   false,   "ConferenceMotion_1280_720_50"};
>+  conf_motion_hd.analyzer = {"conference_motion_hd_3tl_alt_moderate_limits",
>+                             0.0, 0.0, kFullStackTestDurationSecs};
>+  conf_motion_hd.config->queue_length_packets = 50;
>+  conf_motion_hd.config->loss_percent = 3;
>+  conf_motion_hd.config->queue_delay_ms = 100;
>+  conf_motion_hd.config->link_capacity_kbps = 2000;
>+  fixture->RunWithAnalyzer(conf_motion_hd);
>+}
>+
>+TEST(FullStackTest,
>+     ConferenceMotionHd3TLModerateLimitsAltTLPatternAndBaseHeavyTLAllocation) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  test::ScopedFieldTrials field_trial(
>+      AppendFieldTrials("WebRTC-UseShortVP8TL3Pattern/Enabled/"
>+                        "WebRTC-UseBaseHeavyVP8TL3RateAllocation/Enabled/"));
>   ParamsWithLogging conf_motion_hd;
>   conf_motion_hd.call.send_side_bwe = true;
>   conf_motion_hd.video[0] = {
>       true,    1280,    720,   50,    30000,
>       3000000, 3000000, false, "VP8", 3,
>       -1,      0,       false, false, false, "ConferenceMotion_1280_720_50"};
>-  conf_motion_hd.analyzer = {"conference_motion_hd_3tl_alt_moderate_limits",
>-                             0.0, 0.0, kFullStackTestDurationSecs};
>+  conf_motion_hd.analyzer = {
>+      "conference_motion_hd_3tl_alt_heavy_moderate_limits", 0.0, 0.0,
>+      kFullStackTestDurationSecs};
>   conf_motion_hd.config->queue_length_packets = 50;
>   conf_motion_hd.config->loss_percent = 3;
>   conf_motion_hd.config->queue_delay_ms = 100;
>@@ -610,7 +660,7 @@ TEST(FullStackTest, ConferenceMotionHd3TLModerateLimitsAltTLPattern) {
>   fixture->RunWithAnalyzer(conf_motion_hd);
> }
> 
>-#if !defined(RTC_DISABLE_VP9)
>+#if defined(RTC_ENABLE_VP9)
> TEST(FullStackTest, ConferenceMotionHd2000kbps100msLimitedQueueVP9) {
>   auto fixture = CreateVideoQualityTestFixture();
>   ParamsWithLogging conf_motion_hd;
>@@ -642,9 +692,15 @@ TEST(FullStackTest, ScreenshareSlidesVP8_2TL) {
>   fixture->RunWithAnalyzer(screenshare);
> }
> 
>+// TODO(bugs.webrtc.org/9840): Investigate why is this test flaky on MAC.
>+#if !defined(WEBRTC_MAC)
>+const char kScreenshareSimulcastExperiment[] =
>+    "WebRTC-SimulcastScreenshare/Enabled/";
>+
> TEST(FullStackTest, ScreenshareSlidesVP8_3TL_Simulcast) {
>+  test::ScopedFieldTrials field_trial(
>+      AppendFieldTrials(kScreenshareSimulcastExperiment));
>   auto fixture = CreateVideoQualityTestFixture();
>-  test::ScopedFieldTrials field_trial(kScreenshareSimulcastExperiment);
>   ParamsWithLogging screenshare;
>   screenshare.call.send_side_bwe = true;
>   screenshare.screenshare[0] = {true, false, 10};
>@@ -670,6 +726,7 @@ TEST(FullStackTest, ScreenshareSlidesVP8_3TL_Simulcast) {
>       false};
>   fixture->RunWithAnalyzer(screenshare);
> }
>+#endif  // !defined(WEBRTC_MAC)
> 
> TEST(FullStackTest, ScreenshareSlidesVP8_2TL_Scroll) {
>   auto fixture = CreateVideoQualityTestFixture();
>@@ -771,7 +828,7 @@ const ParamsWithLogging::Video kSimulcastVp8VideoLow = {
>     150000, 200000, false, "VP8", 3,
>     2,      400000, false, false, false, "ConferenceMotion_1280_720_50"};
> 
>-#if !defined(RTC_DISABLE_VP9)
>+#if defined(RTC_ENABLE_VP9)
> TEST(FullStackTest, ScreenshareSlidesVP9_2SL) {
>   auto fixture = CreateVideoQualityTestFixture();
>   ParamsWithLogging screenshare;
>@@ -891,9 +948,28 @@ TEST(FullStackTest, VP9KSVC_3SL_Medium_Network_Restricted) {
>   simulcast.config->queue_delay_ms = 100;
>   fixture->RunWithAnalyzer(simulcast);
> }
>+
>+// TODO(webrtc:9722): Remove when experiment is cleaned up.
>+TEST(FullStackTest, VP9KSVC_3SL_Medium_Network_Restricted_Trusted_Rate) {
>+  webrtc::test::ScopedFieldTrials override_trials(
>+      AppendFieldTrials("WebRTC-Vp9IssueKeyFrameOnLayerDeactivation/Enabled/"
>+                        "WebRTC-LibvpxVp9TrustedRateController/Enabled/"));
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging simulcast;
>+  simulcast.call.send_side_bwe = true;
>+  simulcast.video[0] = kSvcVp9Video;
>+  simulcast.analyzer = {"vp9ksvc_3sl_medium_network_restricted_trusted_rate",
>+                        0.0, 0.0, kFullStackTestDurationSecs};
>+  simulcast.ss[0] = {
>+      std::vector<VideoStream>(),  0,    3, -1, InterLayerPredMode::kOnKeyPic,
>+      std::vector<SpatialLayer>(), false};
>+  simulcast.config->link_capacity_kbps = 1000;
>+  simulcast.config->queue_delay_ms = 100;
>+  fixture->RunWithAnalyzer(simulcast);
>+}
> #endif  // !defined(WEBRTC_MAC)
> 
>-#endif  // !defined(RTC_DISABLE_VP9)
>+#endif  // defined(RTC_ENABLE_VP9)
> 
> // Android bots can't handle FullHD, so disable the test.
> // TODO(bugs.webrtc.org/9220): Investigate source of flakiness on Mac.
>@@ -1104,7 +1180,8 @@ INSTANTIATE_TEST_CASE_P(FullStackTest,
> class DualStreamsTest : public ::testing::TestWithParam<int> {};
> 
> // Disable dual video test on mobile device becuase it's too heavy.
>-#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
>+// TODO(bugs.webrtc.org/9840): Investigate why is this test flaky on MAC.
>+#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) && !defined(WEBRTC_MAC)
> TEST_P(DualStreamsTest,
>        ModeratelyRestricted_SlidesVp8_3TL_Simulcast_Video_Simulcast_High) {
>   test::ScopedFieldTrials field_trial(
>@@ -1168,7 +1245,8 @@ TEST_P(DualStreamsTest,
>   auto fixture = CreateVideoQualityTestFixture();
>   fixture->RunWithAnalyzer(dual_streams);
> }
>-#endif  // !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
>+#endif  // !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS) &&
>+        // !defined(WEBRTC_MAC)
> 
> TEST_P(DualStreamsTest, Conference_Restricted) {
>   test::ScopedFieldTrials field_trial(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector_unittest.cc
>index ac019754c54e261da17fab89c96401d67ec61cb8..91050d5056f01b4cfe798d2883741cf43b3c8cb3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector_unittest.cc
>@@ -10,8 +10,8 @@
> 
> #include <memory>
> 
>+#include "api/video/encoded_image.h"
> #include "api/video/i420_buffer.h"
>-#include "common_video/include/video_frame.h"
> #include "modules/video_coding/utility/quality_scaler.h"
> #include "rtc_base/event.h"
> #include "rtc_base/fakeclock.h"
>@@ -406,7 +406,7 @@ TEST_F(OveruseFrameDetectorTest, UpdatesExistingSamples) {
> TEST_F(OveruseFrameDetectorTest, RunOnTqNormalUsage) {
>   rtc::TaskQueue queue("OveruseFrameDetectorTestQueue");
> 
>-  rtc::Event event(false, false);
>+  rtc::Event event;
>   queue.PostTask([this, &event] {
>     overuse_detector_->StartCheckForOveruse(options_, observer_);
>     event.Set();
>@@ -870,7 +870,7 @@ TEST_F(OveruseFrameDetectorTest2, UpdatesExistingSamples) {
> TEST_F(OveruseFrameDetectorTest2, RunOnTqNormalUsage) {
>   rtc::TaskQueue queue("OveruseFrameDetectorTestQueue");
> 
>-  rtc::Event event(false, false);
>+  rtc::Event event;
>   queue.PostTask([this, &event] {
>     overuse_detector_->StartCheckForOveruse(options_, observer_);
>     event.Set();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/picture_id_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/picture_id_tests.cc
>index 155da13ae49dc4967aba9ab27690f383c5ff37f4..6ce0c99ffe7b32701aa25289c10edde57d22912b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/picture_id_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/picture_id_tests.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "api/test/simulated_network.h"
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "call/fake_network_pipe.h"
> #include "call/simulated_network.h"
> #include "media/engine/internalencoderfactory.h"
>@@ -19,7 +20,6 @@
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/numerics/sequence_number_util.h"
> #include "test/call_test.h"
>-#include "test/function_video_encoder_factory.h"
> 
> namespace webrtc {
> namespace {
>@@ -299,7 +299,7 @@ void PictureIdTest::SetupEncoder(VideoEncoderFactory* encoder_factory,
>         test::PacketTransport::kSender, payload_type_map_,
>         absl::make_unique<FakeNetworkPipe>(
>             Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
>-                                           DefaultNetworkSimulationConfig()))));
>+                                           BuiltInNetworkBehaviorConfig()))));
> 
>     CreateSendConfig(kNumSimulcastStreams, 0, 0, send_transport_.get());
>     GetVideoSendConfig()->encoder_settings.encoder_factory = encoder_factory;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_scaling_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_scaling_tests.cc
>index 9b1d0faeff2128c7eaccc24e72b1768670277904..01328f81925725f2062b0dfc462c7411bc99ccfe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_scaling_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_scaling_tests.cc
>@@ -10,6 +10,7 @@
> 
> #include <string>
> 
>+#include "api/test/video/function_video_encoder_factory.h"
> #include "media/engine/internalencoderfactory.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>@@ -17,7 +18,6 @@
> #include "test/call_test.h"
> #include "test/field_trial.h"
> #include "test/frame_generator_capturer.h"
>-#include "test/function_video_encoder_factory.h"
> 
> namespace webrtc {
> namespace {
>@@ -103,8 +103,9 @@ void QualityScalingTest::RunTest(VideoEncoderFactory* encoder_factory,
>       if (wants.max_pixel_count < kWidth * kHeight)
>         observation_complete_.Set();
>     }
>-    void ModifySenderCallConfig(Call::Config* config) override {
>-      config->bitrate_config.start_bitrate_bps = start_bps_;
>+    void ModifySenderBitrateConfig(
>+        BitrateConstraints* bitrate_config) override {
>+      bitrate_config->start_bitrate_bps = start_bps_;
>     }
> 
>     void ModifyVideoConfigs(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc
>index ccd2c20f65f8a729eed1dd56d177cceaace8ed7c..e20b7d294fd6224942c7798ae66403b5d8357ddd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc
>@@ -759,6 +759,9 @@ void ReceiveStatisticsProxy::OnRenderedFrame(const VideoFrame& frame) {
>   RTC_DCHECK_GT(height, 0);
>   int64_t now_ms = clock_->TimeInMilliseconds();
>   rtc::CritScope lock(&crit_);
>+
>+  video_quality_observer_->OnRenderedFrame(now_ms);
>+
>   ContentSpecificStats* content_specific_stats =
>       &content_specific_stats_[last_content_type_];
>   renders_fps_estimator_.Update(1, now_ms);
>@@ -813,8 +816,15 @@ void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
>     ++stats_.frame_counts.delta_frames;
>   }
> 
>+  // Content type extension is set only for keyframes and should be propagated
>+  // for all the following delta frames. Here we may receive frames out of order
>+  // and miscategorise some delta frames near the layer switch.
>+  // This may slightly offset calculated bitrate and keyframes permille metrics.
>+  VideoContentType propagated_content_type =
>+      is_keyframe ? content_type : last_content_type_;
>+
>   ContentSpecificStats* content_specific_stats =
>-      &content_specific_stats_[content_type];
>+      &content_specific_stats_[propagated_content_type];
> 
>   content_specific_stats->total_media_bytes += size_bytes;
>   if (is_keyframe) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.h
>index 4daefe047d658994e893bd8353320c3c281fc6de..100e75116c47360020c294036df4e18b59912855 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.h
>@@ -18,8 +18,6 @@
> 
> #include "absl/types/optional.h"
> #include "call/video_receive_stream.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/frame_callback.h"
> #include "modules/video_coding/include/video_coding_defines.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/numerics/histogram_percentile_counter.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy_unittest.cc
>index b8a96b2f5e824248351285606e093acf65e018c2..b74e4601aab0a916e8496ac3ea024be1ea11041d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy_unittest.cc
>@@ -1057,15 +1057,19 @@ TEST_P(ReceiveStatisticsProxyTest, FreezesAreReported) {
>   const int kFreezeDelayMs = 200;
>   const int kCallDurationMs =
>       kMinRequiredSamples * kInterFrameDelayMs + kFreezeDelayMs;
>+  webrtc::VideoFrame frame(webrtc::I420Buffer::Create(1, 1), 0, 0,
>+                           webrtc::kVideoRotation_0);
>   for (int i = 0; i < kMinRequiredSamples; ++i) {
>     statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>                                       content_type);
>+    statistics_proxy_->OnRenderedFrame(frame);
>     fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>   }
>   // Add extra freeze.
>   fake_clock_.AdvanceTimeMilliseconds(kFreezeDelayMs);
>   statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>                                     content_type);
>+  statistics_proxy_->OnRenderedFrame(frame);
> 
>   statistics_proxy_.reset();
>   const int kExpectedTimeBetweenFreezes =
>@@ -1095,9 +1099,12 @@ TEST_P(ReceiveStatisticsProxyTest, PausesAreIgnored) {
>   const VideoContentType content_type = GetParam();
>   const int kInterFrameDelayMs = 33;
>   const int kPauseDurationMs = 10000;
>+  webrtc::VideoFrame frame(webrtc::I420Buffer::Create(1, 1), 0, 0,
>+                           webrtc::kVideoRotation_0);
>   for (int i = 0; i <= kMinRequiredSamples; ++i) {
>     statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>                                       content_type);
>+    statistics_proxy_->OnRenderedFrame(frame);
>     fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>   }
>   // Add a pause.
>@@ -1108,6 +1115,7 @@ TEST_P(ReceiveStatisticsProxyTest, PausesAreIgnored) {
>   for (int i = 0; i <= kMinRequiredSamples * 3; ++i) {
>     statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>                                       content_type);
>+    statistics_proxy_->OnRenderedFrame(frame);
>     fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/replay.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/replay.cc
>index 5c17e5b3b7d947bd846e9f6967bf1d1a23a0b453..2c1c8a55ad4f5f8527cb162e8555d19b28acb649 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/replay.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/replay.cc
>@@ -15,6 +15,7 @@
> #include <memory>
> #include <sstream>
> 
>+#include "api/test/video/function_video_decoder_factory.h"
> #include "api/video_codecs/video_decoder.h"
> #include "call/call.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>@@ -32,7 +33,6 @@
> #include "test/call_test.h"
> #include "test/encoder_settings.h"
> #include "test/fake_decoder.h"
>-#include "test/function_video_decoder_factory.h"
> #include "test/gtest.h"
> #include "test/null_transport.h"
> #include "test/rtp_file_reader.h"
>@@ -72,39 +72,39 @@ namespace flags {
> // TODO(pbos): Multiple receivers.
> 
> // Flag for payload type.
>-DEFINE_int(media_payload_type,
>-           test::CallTest::kPayloadTypeVP8,
>-           "Media payload type");
>+WEBRTC_DEFINE_int(media_payload_type,
>+                  test::CallTest::kPayloadTypeVP8,
>+                  "Media payload type");
> static int MediaPayloadType() {
>   return static_cast<int>(FLAG_media_payload_type);
> }
> 
> // Flag for RED payload type.
>-DEFINE_int(red_payload_type,
>-           test::CallTest::kRedPayloadType,
>-           "RED payload type");
>+WEBRTC_DEFINE_int(red_payload_type,
>+                  test::CallTest::kRedPayloadType,
>+                  "RED payload type");
> static int RedPayloadType() {
>   return static_cast<int>(FLAG_red_payload_type);
> }
> 
> // Flag for ULPFEC payload type.
>-DEFINE_int(ulpfec_payload_type,
>-           test::CallTest::kUlpfecPayloadType,
>-           "ULPFEC payload type");
>+WEBRTC_DEFINE_int(ulpfec_payload_type,
>+                  test::CallTest::kUlpfecPayloadType,
>+                  "ULPFEC payload type");
> static int UlpfecPayloadType() {
>   return static_cast<int>(FLAG_ulpfec_payload_type);
> }
> 
>-DEFINE_int(media_payload_type_rtx,
>-           test::CallTest::kSendRtxPayloadType,
>-           "Media over RTX payload type");
>+WEBRTC_DEFINE_int(media_payload_type_rtx,
>+                  test::CallTest::kSendRtxPayloadType,
>+                  "Media over RTX payload type");
> static int MediaPayloadTypeRtx() {
>   return static_cast<int>(FLAG_media_payload_type_rtx);
> }
> 
>-DEFINE_int(red_payload_type_rtx,
>-           test::CallTest::kRtxRedPayloadType,
>-           "RED over RTX payload type");
>+WEBRTC_DEFINE_int(red_payload_type_rtx,
>+                  test::CallTest::kRtxRedPayloadType,
>+                  "RED over RTX payload type");
> static int RedPayloadTypeRtx() {
>   return static_cast<int>(FLAG_red_payload_type_rtx);
> }
>@@ -115,7 +115,7 @@ const std::string& DefaultSsrc() {
>       std::to_string(test::CallTest::kVideoSendSsrcs[0]);
>   return ssrc;
> }
>-DEFINE_string(ssrc, DefaultSsrc().c_str(), "Incoming SSRC");
>+WEBRTC_DEFINE_string(ssrc, DefaultSsrc().c_str(), "Incoming SSRC");
> static uint32_t Ssrc() {
>   return rtc::StringToNumber<uint32_t>(FLAG_ssrc).value();
> }
>@@ -125,54 +125,56 @@ const std::string& DefaultSsrcRtx() {
>       std::to_string(test::CallTest::kSendRtxSsrcs[0]);
>   return ssrc_rtx;
> }
>-DEFINE_string(ssrc_rtx, DefaultSsrcRtx().c_str(), "Incoming RTX SSRC");
>+WEBRTC_DEFINE_string(ssrc_rtx, DefaultSsrcRtx().c_str(), "Incoming RTX SSRC");
> static uint32_t SsrcRtx() {
>   return rtc::StringToNumber<uint32_t>(FLAG_ssrc_rtx).value();
> }
> 
> // Flag for abs-send-time id.
>-DEFINE_int(abs_send_time_id, -1, "RTP extension ID for abs-send-time");
>+WEBRTC_DEFINE_int(abs_send_time_id, -1, "RTP extension ID for abs-send-time");
> static int AbsSendTimeId() {
>   return static_cast<int>(FLAG_abs_send_time_id);
> }
> 
> // Flag for transmission-offset id.
>-DEFINE_int(transmission_offset_id,
>-           -1,
>-           "RTP extension ID for transmission-offset");
>+WEBRTC_DEFINE_int(transmission_offset_id,
>+                  -1,
>+                  "RTP extension ID for transmission-offset");
> static int TransmissionOffsetId() {
>   return static_cast<int>(FLAG_transmission_offset_id);
> }
> 
> // Flag for rtpdump input file.
>-DEFINE_string(input_file, "", "input file");
>+WEBRTC_DEFINE_string(input_file, "", "input file");
> static std::string InputFile() {
>   return static_cast<std::string>(FLAG_input_file);
> }
> 
>-DEFINE_string(config_file, "", "config file");
>+WEBRTC_DEFINE_string(config_file, "", "config file");
> static std::string ConfigFile() {
>   return static_cast<std::string>(FLAG_config_file);
> }
> 
> // Flag for raw output files.
>-DEFINE_string(out_base, "", "Basename (excluding .jpg) for raw output");
>+WEBRTC_DEFINE_string(out_base, "", "Basename (excluding .jpg) for raw output");
> static std::string OutBase() {
>   return static_cast<std::string>(FLAG_out_base);
> }
> 
>-DEFINE_string(decoder_bitstream_filename, "", "Decoder bitstream output file");
>+WEBRTC_DEFINE_string(decoder_bitstream_filename,
>+                     "",
>+                     "Decoder bitstream output file");
> static std::string DecoderBitstreamFilename() {
>   return static_cast<std::string>(FLAG_decoder_bitstream_filename);
> }
> 
> // Flag for video codec.
>-DEFINE_string(codec, "VP8", "Video codec");
>+WEBRTC_DEFINE_string(codec, "VP8", "Video codec");
> static std::string Codec() {
>   return static_cast<std::string>(FLAG_codec);
> }
> 
>-DEFINE_bool(help, false, "Print this message.");
>+WEBRTC_DEFINE_bool(help, false, "Print this message.");
> }  // namespace flags
> 
> static const uint32_t kReceiverLocalSsrc = 0x123456;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.h
>index 241fec76a69923842959dcac864e860c0c47375a..90badf70862fc1ab70d6a4b1296754988c451577 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.h
>@@ -14,7 +14,7 @@
> #include <map>
> #include <vector>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.cc
>index 64ec544b70a6f1b3288840bac4006506cf903c40..fdf02ff329d741faf289084e38ad8aa7e738b44b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.cc
>@@ -16,7 +16,6 @@
> 
> #include "absl/memory/memory.h"
> 
>-#include "common_types.h"  // NOLINT(build/include)
> #include "media/base/mediaconstants.h"
> #include "modules/pacing/packet_router.h"
> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
>@@ -50,7 +49,7 @@ namespace {
> // TODO(philipel): Change kPacketBufferStartSize back to 32 in M63 see:
> //                 crbug.com/752886
> constexpr int kPacketBufferStartSize = 512;
>-constexpr int kPacketBufferMaxSixe = 2048;
>+constexpr int kPacketBufferMaxSize = 2048;
> }  // namespace
> 
> std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
>@@ -95,7 +94,8 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
>     ProcessThread* process_thread,
>     NackSender* nack_sender,
>     KeyFrameRequestSender* keyframe_request_sender,
>-    video_coding::OnCompleteFrameCallback* complete_frame_callback)
>+    video_coding::OnCompleteFrameCallback* complete_frame_callback,
>+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor)
>     : clock_(Clock::GetRealTimeClock()),
>       config_(*config),
>       packet_router_(packet_router),
>@@ -113,11 +113,10 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
>                                     packet_router)),
>       complete_frame_callback_(complete_frame_callback),
>       keyframe_request_sender_(keyframe_request_sender),
>-      has_received_frame_(false) {
>+      has_received_frame_(false),
>+      frame_decryptor_(frame_decryptor) {
>   constexpr bool remb_candidate = true;
>   packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
>-  rtp_receive_statistics_->RegisterRtpStatisticsCallback(receive_stats_proxy);
>-  rtp_receive_statistics_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
> 
>   RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
>       << "A stream should not be configured with RTCP disabled. This value is "
>@@ -147,14 +146,29 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
>   process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
> 
>   if (config_.rtp.nack.rtp_history_ms != 0) {
>-    nack_module_.reset(
>-        new NackModule(clock_, nack_sender, keyframe_request_sender));
>+    nack_module_ = absl::make_unique<NackModule>(clock_, nack_sender,
>+                                                 keyframe_request_sender);
>     process_thread_->RegisterModule(nack_module_.get(), RTC_FROM_HERE);
>   }
> 
>+  // The group here must be a positive power of 2, in which case that is used as
>+  // size. All other values shall result in the default value being used.
>+  const std::string group_name =
>+      webrtc::field_trial::FindFullName("WebRTC-PacketBufferMaxSize");
>+  int packet_buffer_max_size = kPacketBufferMaxSize;
>+  if (!group_name.empty() &&
>+      (sscanf(group_name.c_str(), "%d", &packet_buffer_max_size) != 1 ||
>+       packet_buffer_max_size <= 0 ||
>+       // Verify that the number is a positive power of 2.
>+       (packet_buffer_max_size & (packet_buffer_max_size - 1)) != 0)) {
>+    RTC_LOG(LS_WARNING) << "Invalid packet buffer max size: " << group_name;
>+    packet_buffer_max_size = kPacketBufferMaxSize;
>+  }
>+
>   packet_buffer_ = video_coding::PacketBuffer::Create(
>-      clock_, kPacketBufferStartSize, kPacketBufferMaxSixe, this);
>-  reference_finder_.reset(new video_coding::RtpFrameReferenceFinder(this));
>+      clock_, kPacketBufferStartSize, packet_buffer_max_size, this);
>+  reference_finder_ =
>+      absl::make_unique<video_coding::RtpFrameReferenceFinder>(this);
> }
> 
> RtpVideoStreamReceiver::~RtpVideoStreamReceiver() {
>@@ -202,14 +216,15 @@ int32_t RtpVideoStreamReceiver::OnReceivedPayloadData(
>     size_t payload_size,
>     const WebRtcRTPHeader* rtp_header) {
>   return OnReceivedPayloadData(payload_data, payload_size, rtp_header,
>-                               absl::nullopt);
>+                               absl::nullopt, false);
> }
> 
> int32_t RtpVideoStreamReceiver::OnReceivedPayloadData(
>     const uint8_t* payload_data,
>     size_t payload_size,
>     const WebRtcRTPHeader* rtp_header,
>-    const absl::optional<RtpGenericFrameDescriptor>& generic_descriptor) {
>+    const absl::optional<RtpGenericFrameDescriptor>& generic_descriptor,
>+    bool is_recovered) {
>   WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
>   rtp_header_with_ntp.ntp_time_ms =
>       ntp_estimator_.Estimate(rtp_header->header.timestamp);
>@@ -221,7 +236,8 @@ int32_t RtpVideoStreamReceiver::OnReceivedPayloadData(
>         rtp_header->frameType == kVideoFrameKey;
> 
>     packet.timesNacked = nack_module_->OnReceivedPacket(
>-        rtp_header->header.sequenceNumber, is_keyframe);
>+        rtp_header->header.sequenceNumber, is_keyframe, is_recovered);
>+
>   } else {
>     packet.timesNacked = -1;
>   }
>@@ -302,7 +318,8 @@ void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
>       last_received_rtp_system_time_ms_ = now_ms;
> 
>       std::vector<uint32_t> csrcs = packet.Csrcs();
>-      contributing_sources_.Update(now_ms, csrcs);
>+      contributing_sources_.Update(now_ms, csrcs,
>+                                   /* audio level */absl::nullopt);
>     }
>     // Periodically log the RTP header of incoming packets.
>     if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
>@@ -363,10 +380,61 @@ int32_t RtpVideoStreamReceiver::ResendPackets(const uint16_t* sequence_numbers,
> 
> void RtpVideoStreamReceiver::OnReceivedFrame(
>     std::unique_ptr<video_coding::RtpFrameObject> frame) {
>+  // Request a key frame as soon as possible.
>+  bool key_frame_requested = false;
>   if (!has_received_frame_) {
>     has_received_frame_ = true;
>-    if (frame->FrameType() != kVideoFrameKey)
>+    if (frame->FrameType() != kVideoFrameKey) {
>+      key_frame_requested = true;
>       keyframe_request_sender_->RequestKeyFrame();
>+    }
>+  }
>+
>+  // Optionally attempt to decrypt the raw video frame if it was provided.
>+  if (frame_decryptor_ != nullptr) {
>+    // When using encryption we expect the frame to have the generic descriptor.
>+    absl::optional<RtpGenericFrameDescriptor> descriptor =
>+        frame->GetGenericFrameDescriptor();
>+    if (!descriptor) {
>+      RTC_LOG(LS_ERROR) << "No generic frame descriptor found dropping frame.";
>+      return;
>+    }
>+
>+    // Retrieve the bitstream of the encrypted video frame.
>+    rtc::ArrayView<const uint8_t> encrypted_frame_bitstream(frame->Buffer(),
>+                                                            frame->size());
>+    // Retrieve the maximum possible size of the decrypted payload.
>+    const size_t max_plaintext_byte_size =
>+        frame_decryptor_->GetMaxPlaintextByteSize(cricket::MEDIA_TYPE_VIDEO,
>+                                                  frame->size());
>+    RTC_CHECK(max_plaintext_byte_size <= frame->size());
>+    // Place the decrypted frame inline into the existing frame.
>+    rtc::ArrayView<uint8_t> inline_decrypted_bitstream(frame->MutableBuffer(),
>+                                                       max_plaintext_byte_size);
>+
>+    // Attempt to decrypt the video frame.
>+    size_t bytes_written = 0;
>+    if (frame_decryptor_->Decrypt(
>+            cricket::MEDIA_TYPE_VIDEO, /*csrcs=*/{},
>+            /*additional_data=*/nullptr, encrypted_frame_bitstream,
>+            inline_decrypted_bitstream, &bytes_written) != 0) {
>+      return;
>+    }
>+
>+    if (!has_received_decrypted_frame_ && !key_frame_requested) {
>+      has_received_decrypted_frame_ = true;
>+      if (frame->FrameType() != kVideoFrameKey) {
>+        keyframe_request_sender_->RequestKeyFrame();
>+      }
>+    }
>+
>+    RTC_CHECK(bytes_written <= max_plaintext_byte_size);
>+    // Update the frame to contain just the written bytes.
>+    frame->SetLength(bytes_written);
>+  } else if (config_.crypto_options.sframe.require_frame_encryption) {
>+    RTC_LOG(LS_WARNING) << "Frame decryption required but not attached to this "
>+                           "stream. Dropping  frame.";
>+    return;
>   }
> 
>   reference_finder_->ManageFrame(std::move(frame));
>@@ -459,8 +527,6 @@ void RtpVideoStreamReceiver::ReceivePacket(const RtpPacketReceived& packet) {
>   webrtc_rtp_header.video_header().content_type = VideoContentType::UNSPECIFIED;
>   webrtc_rtp_header.video_header().video_timing.flags =
>       VideoSendTiming::kInvalid;
>-  webrtc_rtp_header.video_header().playout_delay.min_ms = -1;
>-  webrtc_rtp_header.video_header().playout_delay.max_ms = -1;
>   webrtc_rtp_header.video_header().is_last_packet_in_frame =
>       webrtc_rtp_header.header.markerBit;
> 
>@@ -486,12 +552,23 @@ void RtpVideoStreamReceiver::ReceivePacket(const RtpPacketReceived& packet) {
>         webrtc_rtp_header.header.markerBit ||
>         (generic_descriptor_wire->LastSubFrameInFrame() &&
>          generic_descriptor_wire->LastPacketInSubFrame());
>+
>+    if (generic_descriptor_wire->FirstPacketInSubFrame()) {
>+      webrtc_rtp_header.frameType =
>+          generic_descriptor_wire->FrameDependenciesDiffs().empty()
>+              ? kVideoFrameKey
>+              : kVideoFrameDelta;
>+    }
>+
>+    webrtc_rtp_header.video_header().width = generic_descriptor_wire->Width();
>+    webrtc_rtp_header.video_header().height = generic_descriptor_wire->Height();
>   } else {
>     generic_descriptor_wire.reset();
>   }
> 
>   OnReceivedPayloadData(parsed_payload.payload, parsed_payload.payload_length,
>-                        &webrtc_rtp_header, generic_descriptor_wire);
>+                        &webrtc_rtp_header, generic_descriptor_wire,
>+                        packet.recovered());
> }
> 
> void RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(
>@@ -523,7 +600,8 @@ void RtpVideoStreamReceiver::NotifyReceiverOfEmptyPacket(uint16_t seq_num) {
>   reference_finder_->PaddingReceived(seq_num);
>   packet_buffer_->PaddingReceived(seq_num);
>   if (nack_module_) {
>-    nack_module_->OnReceivedPacket(seq_num, /* is_keyframe = */ false);
>+    nack_module_->OnReceivedPacket(seq_num, /* is_keyframe = */ false,
>+                                   /* is _recovered = */ false);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.h
>index ca5d7a0abd94258147b1f6b9b2aabc59ada553b0..a2006d1bb2a087246a680b6eed057e2d24af0b4c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.h
>@@ -19,6 +19,7 @@
> 
> #include "absl/types/optional.h"
> 
>+#include "api/crypto/framedecryptorinterface.h"
> #include "api/video_codecs/video_codec.h"
> #include "call/rtp_packet_sink_interface.h"
> #include "call/syncable.h"
>@@ -67,7 +68,8 @@ class RtpVideoStreamReceiver : public RecoveredPacketReceiver,
>       ProcessThread* process_thread,
>       NackSender* nack_sender,
>       KeyFrameRequestSender* keyframe_request_sender,
>-      video_coding::OnCompleteFrameCallback* complete_frame_callback);
>+      video_coding::OnCompleteFrameCallback* complete_frame_callback,
>+      rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor);
>   ~RtpVideoStreamReceiver() override;
> 
>   void AddReceiveCodec(const VideoCodec& video_codec,
>@@ -102,7 +104,8 @@ class RtpVideoStreamReceiver : public RecoveredPacketReceiver,
>       const uint8_t* payload_data,
>       size_t payload_size,
>       const WebRtcRTPHeader* rtp_header,
>-      const absl::optional<RtpGenericFrameDescriptor>& generic_descriptor);
>+      const absl::optional<RtpGenericFrameDescriptor>& generic_descriptor,
>+      bool is_recovered);
> 
>   // Implements RecoveredPacketReceiver.
>   void OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
>@@ -203,6 +206,11 @@ class RtpVideoStreamReceiver : public RecoveredPacketReceiver,
>       RTC_GUARDED_BY(rtp_sources_lock_);
>   absl::optional<int64_t> last_received_rtp_system_time_ms_
>       RTC_GUARDED_BY(rtp_sources_lock_);
>+
>+  // E2EE Video Frame Decryptor (Optional)
>+  rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
>+  // Set to true on the first successsfully decrypted frame.
>+  bool has_received_decrypted_frame_ = false;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver_unittest.cc
>index ec2bf334e9b6d0e7984fe8c665100749a4169653..10c5e7eebce2af1fcb0ccc9c9f4e26fae35ce76e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver_unittest.cc
>@@ -133,7 +133,7 @@ class RtpVideoStreamReceiverTest : public testing::Test {
>         &mock_transport_, nullptr, &packet_router_, &config_,
>         rtp_receive_statistics_.get(), nullptr, process_thread_.get(),
>         &mock_nack_sender_, &mock_key_frame_request_sender_,
>-        &mock_on_complete_frame_callback_);
>+        &mock_on_complete_frame_callback_, nullptr);
>   }
> 
>   WebRtcRTPHeader GetDefaultPacket() {
>@@ -568,10 +568,8 @@ TEST_F(RtpVideoStreamReceiverTest, ParseGenericDescriptorTwoPackets) {
>   first_packet_descriptor.SetFirstSubFrameInFrame(true);
>   first_packet_descriptor.SetLastSubFrameInFrame(true);
>   first_packet_descriptor.SetFrameId(100);
>-  first_packet_descriptor.SetTemporalLayer(1);
>   first_packet_descriptor.SetSpatialLayersBitmask(1 << kSpatialIndex);
>-  first_packet_descriptor.AddFrameDependencyDiff(90);
>-  first_packet_descriptor.AddFrameDependencyDiff(80);
>+  first_packet_descriptor.SetResolution(480, 360);
>   EXPECT_TRUE(first_packet.SetExtension<RtpGenericFrameDescriptorExtension>(
>       first_packet_descriptor));
> 
>@@ -606,10 +604,10 @@ TEST_F(RtpVideoStreamReceiverTest, ParseGenericDescriptorTwoPackets) {
> 
>   EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame)
>       .WillOnce(Invoke([kSpatialIndex](video_coding::EncodedFrame* frame) {
>-        EXPECT_EQ(frame->num_references, 2U);
>-        EXPECT_EQ(frame->references[0], frame->id.picture_id - 90);
>-        EXPECT_EQ(frame->references[1], frame->id.picture_id - 80);
>+        EXPECT_EQ(frame->num_references, 0U);
>         EXPECT_EQ(frame->id.spatial_layer, kSpatialIndex);
>+        EXPECT_EQ(frame->EncodedImage()._encodedWidth, 480u);
>+        EXPECT_EQ(frame->EncodedImage()._encodedHeight, 360u);
>       }));
> 
>   rtp_video_stream_receiver_->OnRtpPacket(second_packet);
>@@ -628,4 +626,13 @@ TEST_F(RtpVideoStreamReceiverTest, RepeatedSecondarySinkDisallowed) {
> }
> #endif
> 
>+// Initialization of WebRtcRTPHeader is a bit convoluted, with some fields
>+// zero-initialized. RtpVideoStreamReceiver depends on proper default values for
>+// the playout delay.
>+TEST(WebRtcRTPHeader, DefaultPlayoutDelayIsUnspecified) {
>+  WebRtcRTPHeader webrtc_rtp_header = {};
>+  EXPECT_EQ(webrtc_rtp_header.video_header().playout_delay.min_ms, -1);
>+  EXPECT_EQ(webrtc_rtp_header.video_header().playout_delay.max_ms, -1);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/screenshare_loopback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/screenshare_loopback.cc
>index 1d6e87e71ff4ba92e47cc74d9ae8b020b3e39e01..4696666908707d450e2a3fb3a17e62a20636cadc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/screenshare_loopback.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/screenshare_loopback.cc
>@@ -12,7 +12,6 @@
> 
> #include "rtc_base/flags.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/stringencode.h"
> #include "system_wrappers/include/field_trial.h"
> #include "test/field_trial.h"
> #include "test/gtest.h"
>@@ -23,73 +22,76 @@ namespace webrtc {
> namespace flags {
> 
> // Flags common with video loopback, with different default values.
>-DEFINE_int(width, 1850, "Video width (crops source).");
>+WEBRTC_DEFINE_int(width, 1850, "Video width (crops source).");
> size_t Width() {
>   return static_cast<size_t>(FLAG_width);
> }
> 
>-DEFINE_int(height, 1110, "Video height (crops source).");
>+WEBRTC_DEFINE_int(height, 1110, "Video height (crops source).");
> size_t Height() {
>   return static_cast<size_t>(FLAG_height);
> }
> 
>-DEFINE_int(fps, 5, "Frames per second.");
>+WEBRTC_DEFINE_int(fps, 5, "Frames per second.");
> int Fps() {
>   return static_cast<int>(FLAG_fps);
> }
> 
>-DEFINE_int(min_bitrate, 50, "Call and stream min bitrate in kbps.");
>+WEBRTC_DEFINE_int(min_bitrate, 50, "Call and stream min bitrate in kbps.");
> int MinBitrateKbps() {
>   return static_cast<int>(FLAG_min_bitrate);
> }
> 
>-DEFINE_int(start_bitrate, 300, "Call start bitrate in kbps.");
>+WEBRTC_DEFINE_int(start_bitrate, 300, "Call start bitrate in kbps.");
> int StartBitrateKbps() {
>   return static_cast<int>(FLAG_start_bitrate);
> }
> 
>-DEFINE_int(target_bitrate, 200, "Stream target bitrate in kbps.");
>+WEBRTC_DEFINE_int(target_bitrate, 200, "Stream target bitrate in kbps.");
> int TargetBitrateKbps() {
>   return static_cast<int>(FLAG_target_bitrate);
> }
> 
>-DEFINE_int(max_bitrate, 1000, "Call and stream max bitrate in kbps.");
>+WEBRTC_DEFINE_int(max_bitrate, 1000, "Call and stream max bitrate in kbps.");
> int MaxBitrateKbps() {
>   return static_cast<int>(FLAG_max_bitrate);
> }
> 
>-DEFINE_int(num_temporal_layers, 2, "Number of temporal layers to use.");
>+WEBRTC_DEFINE_int(num_temporal_layers, 2, "Number of temporal layers to use.");
> int NumTemporalLayers() {
>   return static_cast<int>(FLAG_num_temporal_layers);
> }
> 
> // Flags common with video loopback, with equal default values.
>-DEFINE_string(codec, "VP8", "Video codec to use.");
>+WEBRTC_DEFINE_string(codec, "VP8", "Video codec to use.");
> std::string Codec() {
>   return static_cast<std::string>(FLAG_codec);
> }
> 
>-DEFINE_string(rtc_event_log_name,
>-              "",
>-              "Filename for rtc event log. Two files "
>-              "with \"_send\" and \"_recv\" suffixes will be created.");
>+WEBRTC_DEFINE_string(rtc_event_log_name,
>+                     "",
>+                     "Filename for rtc event log. Two files "
>+                     "with \"_send\" and \"_recv\" suffixes will be created.");
> std::string RtcEventLogName() {
>   return static_cast<std::string>(FLAG_rtc_event_log_name);
> }
> 
>-DEFINE_string(rtp_dump_name, "", "Filename for dumped received RTP stream.");
>+WEBRTC_DEFINE_string(rtp_dump_name,
>+                     "",
>+                     "Filename for dumped received RTP stream.");
> std::string RtpDumpName() {
>   return static_cast<std::string>(FLAG_rtp_dump_name);
> }
> 
>-DEFINE_int(selected_tl,
>-           -1,
>-           "Temporal layer to show or analyze. -1 to disable filtering.");
>+WEBRTC_DEFINE_int(
>+    selected_tl,
>+    -1,
>+    "Temporal layer to show or analyze. -1 to disable filtering.");
> int SelectedTL() {
>   return static_cast<int>(FLAG_selected_tl);
> }
> 
>-DEFINE_int(
>+WEBRTC_DEFINE_int(
>     duration,
>     0,
>     "Duration of the test in seconds. If 0, rendered will be shown instead.");
>@@ -97,71 +99,74 @@ int DurationSecs() {
>   return static_cast<int>(FLAG_duration);
> }
> 
>-DEFINE_string(output_filename, "", "Target graph data filename.");
>+WEBRTC_DEFINE_string(output_filename, "", "Target graph data filename.");
> std::string OutputFilename() {
>   return static_cast<std::string>(FLAG_output_filename);
> }
> 
>-DEFINE_string(graph_title,
>-              "",
>-              "If empty, title will be generated automatically.");
>+WEBRTC_DEFINE_string(graph_title,
>+                     "",
>+                     "If empty, title will be generated automatically.");
> std::string GraphTitle() {
>   return static_cast<std::string>(FLAG_graph_title);
> }
> 
>-DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost.");
>+WEBRTC_DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost.");
> int LossPercent() {
>   return static_cast<int>(FLAG_loss_percent);
> }
> 
>-DEFINE_int(link_capacity,
>-           0,
>-           "Capacity (kbps) of the fake link. 0 means infinite.");
>+WEBRTC_DEFINE_int(link_capacity,
>+                  0,
>+                  "Capacity (kbps) of the fake link. 0 means infinite.");
> int LinkCapacityKbps() {
>   return static_cast<int>(FLAG_link_capacity);
> }
> 
>-DEFINE_int(queue_size, 0, "Size of the bottleneck link queue in packets.");
>+WEBRTC_DEFINE_int(queue_size,
>+                  0,
>+                  "Size of the bottleneck link queue in packets.");
> int QueueSize() {
>   return static_cast<int>(FLAG_queue_size);
> }
> 
>-DEFINE_int(avg_propagation_delay_ms,
>-           0,
>-           "Average link propagation delay in ms.");
>+WEBRTC_DEFINE_int(avg_propagation_delay_ms,
>+                  0,
>+                  "Average link propagation delay in ms.");
> int AvgPropagationDelayMs() {
>   return static_cast<int>(FLAG_avg_propagation_delay_ms);
> }
> 
>-DEFINE_int(std_propagation_delay_ms,
>-           0,
>-           "Link propagation delay standard deviation in ms.");
>+WEBRTC_DEFINE_int(std_propagation_delay_ms,
>+                  0,
>+                  "Link propagation delay standard deviation in ms.");
> int StdPropagationDelayMs() {
>   return static_cast<int>(FLAG_std_propagation_delay_ms);
> }
> 
>-DEFINE_int(num_streams, 0, "Number of streams to show or analyze.");
>+WEBRTC_DEFINE_int(num_streams, 0, "Number of streams to show or analyze.");
> int NumStreams() {
>   return static_cast<int>(FLAG_num_streams);
> }
> 
>-DEFINE_int(selected_stream,
>-           0,
>-           "ID of the stream to show or analyze. "
>-           "Set to the number of streams to show them all.");
>+WEBRTC_DEFINE_int(selected_stream,
>+                  0,
>+                  "ID of the stream to show or analyze. "
>+                  "Set to the number of streams to show them all.");
> int SelectedStream() {
>   return static_cast<int>(FLAG_selected_stream);
> }
> 
>-DEFINE_int(num_spatial_layers, 1, "Number of spatial layers to use.");
>+WEBRTC_DEFINE_int(num_spatial_layers, 1, "Number of spatial layers to use.");
> int NumSpatialLayers() {
>   return static_cast<int>(FLAG_num_spatial_layers);
> }
> 
>-DEFINE_int(inter_layer_pred,
>-           0,
>-           "Inter-layer prediction mode. "
>-           "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
>+WEBRTC_DEFINE_int(
>+    inter_layer_pred,
>+    0,
>+    "Inter-layer prediction mode. "
>+    "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
> InterLayerPredMode InterLayerPred() {
>   if (FLAG_inter_layer_pred == 0) {
>     return InterLayerPredMode::kOn;
>@@ -173,58 +178,65 @@ InterLayerPredMode InterLayerPred() {
>   }
> }
> 
>-DEFINE_int(selected_sl,
>-           -1,
>-           "Spatial layer to show or analyze. -1 to disable filtering.");
>+WEBRTC_DEFINE_int(selected_sl,
>+                  -1,
>+                  "Spatial layer to show or analyze. -1 to disable filtering.");
> int SelectedSL() {
>   return static_cast<int>(FLAG_selected_sl);
> }
> 
>-DEFINE_string(stream0,
>-              "",
>-              "Comma separated values describing VideoStream for stream #0.");
>+WEBRTC_DEFINE_string(
>+    stream0,
>+    "",
>+    "Comma separated values describing VideoStream for stream #0.");
> std::string Stream0() {
>   return static_cast<std::string>(FLAG_stream0);
> }
> 
>-DEFINE_string(stream1,
>-              "",
>-              "Comma separated values describing VideoStream for stream #1.");
>+WEBRTC_DEFINE_string(
>+    stream1,
>+    "",
>+    "Comma separated values describing VideoStream for stream #1.");
> std::string Stream1() {
>   return static_cast<std::string>(FLAG_stream1);
> }
> 
>-DEFINE_string(sl0,
>-              "",
>-              "Comma separated values describing SpatialLayer for layer #0.");
>+WEBRTC_DEFINE_string(
>+    sl0,
>+    "",
>+    "Comma separated values describing SpatialLayer for layer #0.");
> std::string SL0() {
>   return static_cast<std::string>(FLAG_sl0);
> }
> 
>-DEFINE_string(sl1,
>-              "",
>-              "Comma separated values describing SpatialLayer for layer #1.");
>+WEBRTC_DEFINE_string(
>+    sl1,
>+    "",
>+    "Comma separated values describing SpatialLayer for layer #1.");
> std::string SL1() {
>   return static_cast<std::string>(FLAG_sl1);
> }
> 
>-DEFINE_string(encoded_frame_path,
>-              "",
>-              "The base path for encoded frame logs. Created files will have "
>-              "the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf");
>+WEBRTC_DEFINE_string(
>+    encoded_frame_path,
>+    "",
>+    "The base path for encoded frame logs. Created files will have "
>+    "the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf");
> std::string EncodedFramePath() {
>   return static_cast<std::string>(FLAG_encoded_frame_path);
> }
> 
>-DEFINE_bool(logs, false, "print logs to stderr");
>+WEBRTC_DEFINE_bool(logs, false, "print logs to stderr");
> 
>-DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
>+WEBRTC_DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
> 
>-DEFINE_bool(generic_descriptor, false, "Use the generic frame descriptor.");
>+WEBRTC_DEFINE_bool(generic_descriptor,
>+                   false,
>+                   "Use the generic frame descriptor.");
> 
>-DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur");
>+WEBRTC_DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur");
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     force_fieldtrials,
>     "",
>     "Field trials control experimental feature code which can be forced. "
>@@ -233,12 +245,14 @@ DEFINE_string(
>     "trials are separated by \"/\"");
> 
> // Screenshare-specific flags.
>-DEFINE_int(min_transmit_bitrate, 400, "Min transmit bitrate incl. padding.");
>+WEBRTC_DEFINE_int(min_transmit_bitrate,
>+                  400,
>+                  "Min transmit bitrate incl. padding.");
> int MinTransmitBitrateKbps() {
>   return FLAG_min_transmit_bitrate;
> }
> 
>-DEFINE_bool(
>+WEBRTC_DEFINE_bool(
>     generate_slides,
>     false,
>     "Whether to use randomly generated slides or read them from files.");
>@@ -246,14 +260,14 @@ bool GenerateSlides() {
>   return static_cast<int>(FLAG_generate_slides);
> }
> 
>-DEFINE_int(slide_change_interval,
>-           10,
>-           "Interval (in seconds) between simulated slide changes.");
>+WEBRTC_DEFINE_int(slide_change_interval,
>+                  10,
>+                  "Interval (in seconds) between simulated slide changes.");
> int SlideChangeInterval() {
>   return static_cast<int>(FLAG_slide_change_interval);
> }
> 
>-DEFINE_int(
>+WEBRTC_DEFINE_int(
>     scroll_duration,
>     0,
>     "Duration (in seconds) during which a slide will be scrolled into place.");
>@@ -261,9 +275,10 @@ int ScrollDuration() {
>   return static_cast<int>(FLAG_scroll_duration);
> }
> 
>-DEFINE_string(slides,
>-              "",
>-              "Comma-separated list of *.yuv files to display as slides.");
>+WEBRTC_DEFINE_string(
>+    slides,
>+    "",
>+    "Comma-separated list of *.yuv files to display as slides.");
> std::vector<std::string> Slides() {
>   std::vector<std::string> slides;
>   std::string slides_list = FLAG_slides;
>@@ -271,12 +286,12 @@ std::vector<std::string> Slides() {
>   return slides;
> }
> 
>-DEFINE_bool(help, false, "prints this message");
>+WEBRTC_DEFINE_bool(help, false, "prints this message");
> 
> }  // namespace flags
> 
> void Loopback() {
>-  DefaultNetworkSimulationConfig pipe_config;
>+  BuiltInNetworkBehaviorConfig pipe_config;
>   pipe_config.loss_percent = flags::LossPercent();
>   pipe_config.link_capacity_kbps = flags::LinkCapacityKbps();
>   pipe_config.queue_length_packets = flags::QueueSize();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats.h
>index 3337899069a006097660030ba005decfda02a576..b44516b285541aba081f110b316fe22de9c0bab9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats.h
>@@ -16,7 +16,6 @@
> #include <set>
> 
> #include "call/video_send_stream.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/clock.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.cc
>index 9839ed7a3b7094bdffe74de834d1f77315ecbafb..24e964194714503748f2055331ccb59597f7d1b3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.cc
>@@ -15,7 +15,6 @@
> #include <limits>
> #include <utility>
> 
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>@@ -823,10 +822,13 @@ void SendStatisticsProxy::UpdateEncoderFallbackStats(
> 
>   const int64_t now_ms = clock_->TimeInMilliseconds();
>   bool is_active = fallback_info->is_active;
>-  if (codec_info->codec_name != stats_.encoder_implementation_name) {
>+  if (encoder_changed_) {
>     // Implementation changed.
>-    is_active = strcmp(codec_info->codec_name, kVp8SwCodecName) == 0;
>-    if (!is_active && stats_.encoder_implementation_name != kVp8SwCodecName) {
>+    const bool last_was_vp8_software =
>+        encoder_changed_->previous_encoder_implementation == kVp8SwCodecName;
>+    is_active = encoder_changed_->new_encoder_implementation == kVp8SwCodecName;
>+    encoder_changed_.reset();
>+    if (!is_active && !last_was_vp8_software) {
>       // First or not a VP8 SW change, update stats on next call.
>       return;
>     }
>@@ -865,7 +867,7 @@ void SendStatisticsProxy::UpdateFallbackDisabledStats(
>   }
> 
>   if (!IsForcedFallbackPossible(codec_info, simulcast_index) ||
>-      strcmp(codec_info->codec_name, kVp8SwCodecName) == 0) {
>+      stats_.encoder_implementation_name == kVp8SwCodecName) {
>     uma_container_->fallback_info_disabled_.is_possible = false;
>     return;
>   }
>@@ -895,13 +897,9 @@ void SendStatisticsProxy::OnSendEncodedImage(
>   rtc::CritScope lock(&crit_);
>   ++stats_.frames_encoded;
>   if (codec_info) {
>-    if (codec_info->codec_name) {
>-      UpdateEncoderFallbackStats(
>-          codec_info,
>-          encoded_image._encodedWidth * encoded_image._encodedHeight,
>-          simulcast_idx);
>-      stats_.encoder_implementation_name = codec_info->codec_name;
>-    }
>+    UpdateEncoderFallbackStats(
>+        codec_info, encoded_image._encodedWidth * encoded_image._encodedHeight,
>+        simulcast_idx);
>   }
> 
>   if (static_cast<size_t>(simulcast_idx) >= rtp_config_.ssrcs.size()) {
>@@ -981,6 +979,14 @@ void SendStatisticsProxy::OnSendEncodedImage(
>   }
> }
> 
>+void SendStatisticsProxy::OnEncoderImplementationChanged(
>+    const std::string& implementation_name) {
>+  rtc::CritScope lock(&crit_);
>+  encoder_changed_ = EncoderChangeEvent{stats_.encoder_implementation_name,
>+                                        implementation_name};
>+  stats_.encoder_implementation_name = implementation_name;
>+}
>+
> int SendStatisticsProxy::GetInputFrameRate() const {
>   rtc::CritScope lock(&crit_);
>   return round(uma_container_->input_frame_rate_tracker_.ComputeRate());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.h
>index b5868d0890727f05260e9030b4570909fd2f4186..7316f3b55aec0d34fd5f5aa092e0f19ca889bc41 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.h
>@@ -18,7 +18,6 @@
> 
> #include "api/video/video_stream_encoder_observer.h"
> #include "call/video_send_stream.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/include/video_coding_defines.h"
> #include "rtc_base/criticalsection.h"
>@@ -53,6 +52,10 @@ class SendStatisticsProxy : public VideoStreamEncoderObserver,
> 
>   void OnSendEncodedImage(const EncodedImage& encoded_image,
>                           const CodecSpecificInfo* codec_info) override;
>+
>+  void OnEncoderImplementationChanged(
>+      const std::string& implementation_name) override;
>+
>   // Used to update incoming frame rate.
>   void OnIncomingFrame(int width, int height) override;
> 
>@@ -241,6 +244,14 @@ class SendStatisticsProxy : public VideoStreamEncoderObserver,
> 
>   absl::optional<int64_t> last_outlier_timestamp_ RTC_GUARDED_BY(crit_);
> 
>+  struct EncoderChangeEvent {
>+    std::string previous_encoder_implementation;
>+    std::string new_encoder_implementation;
>+  };
>+  // Stores the last change in encoder implementation in an optional, so that
>+  // the event can be consumed.
>+  absl::optional<EncoderChangeEvent> encoder_changed_;
>+
>   // Contains stats used for UMA histograms. These stats will be reset if
>   // content type changes between real-time video and screenshare, since these
>   // will be reported separately.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy_unittest.cc
>index 3d27c712c6b6a3727f9f993bdb16d1a312daf443..586d57fb284880fff1733ec9e7c82055b48f339b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy_unittest.cc
>@@ -2167,13 +2167,9 @@ TEST_F(SendStatisticsProxyTest, FecBitrateNotReportedWhenNotEnabled) {
> }
> 
> TEST_F(SendStatisticsProxyTest, GetStatsReportsEncoderImplementationName) {
>-  const char* kName = "encoderName";
>-  EncodedImage encoded_image;
>-  CodecSpecificInfo codec_info;
>-  codec_info.codec_name = kName;
>-  statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
>-  EXPECT_STREQ(
>-      kName, statistics_proxy_->GetStats().encoder_implementation_name.c_str());
>+  const std::string kName = "encoderName";
>+  statistics_proxy_->OnEncoderImplementationChanged(kName);
>+  EXPECT_EQ(kName, statistics_proxy_->GetStats().encoder_implementation_name);
> }
> 
> TEST_F(SendStatisticsProxyTest, Vp9SvcLowSpatialLayerDoesNotUpdateResolution) {
>@@ -2219,7 +2215,6 @@ class ForcedFallbackTest : public SendStatisticsProxyTest {
>       : SendStatisticsProxyTest(field_trials) {
>     codec_info_.codecType = kVideoCodecVP8;
>     codec_info_.codecSpecific.VP8.temporalIdx = 0;
>-    codec_info_.codec_name = "fake_codec";
>     encoded_image_._encodedWidth = kWidth;
>     encoded_image_._encodedHeight = kHeight;
>     encoded_image_.SetSpatialIndex(0);
>@@ -2229,6 +2224,8 @@ class ForcedFallbackTest : public SendStatisticsProxyTest {
> 
>  protected:
>   void InsertEncodedFrames(int num_frames, int interval_ms) {
>+    statistics_proxy_->OnEncoderImplementationChanged(codec_name_);
>+
>     // First frame is not updating stats, insert initial frame.
>     if (statistics_proxy_->GetStats().frames_encoded == 0) {
>       statistics_proxy_->OnSendEncodedImage(encoded_image_, &codec_info_);
>@@ -2243,6 +2240,7 @@ class ForcedFallbackTest : public SendStatisticsProxyTest {
> 
>   EncodedImage encoded_image_;
>   CodecSpecificInfo codec_info_;
>+  std::string codec_name_;
>   const std::string kPrefix = "WebRTC.Video.Encoder.ForcedSw";
>   const int kFrameIntervalMs = 1000;
>   const int kMinFrames = 20;  // Min run time 20 sec.
>@@ -2321,7 +2319,7 @@ TEST_F(ForcedFallbackEnabled, EnteredLowResolutionNotSetIfNotLibvpx) {
> }
> 
> TEST_F(ForcedFallbackEnabled, EnteredLowResolutionSetIfLibvpx) {
>-  codec_info_.codec_name = "libvpx";
>+  codec_name_ = "libvpx";
>   InsertEncodedFrames(1, kFrameIntervalMs);
>   EXPECT_TRUE(statistics_proxy_->GetStats().has_entered_low_resolution);
> }
>@@ -2333,7 +2331,7 @@ TEST_F(ForcedFallbackDisabled, EnteredLowResolutionNotSetIfAboveMaxPixels) {
> }
> 
> TEST_F(ForcedFallbackDisabled, EnteredLowResolutionNotSetIfLibvpx) {
>-  codec_info_.codec_name = "libvpx";
>+  codec_name_ = "libvpx";
>   InsertEncodedFrames(1, kFrameIntervalMs);
>   EXPECT_FALSE(statistics_proxy_->GetStats().has_entered_low_resolution);
> }
>@@ -2351,7 +2349,7 @@ TEST_F(ForcedFallbackEnabled, OneFallbackEvent) {
>   EXPECT_FALSE(statistics_proxy_->GetStats().has_entered_low_resolution);
>   InsertEncodedFrames(15, 1000);
>   EXPECT_FALSE(statistics_proxy_->GetStats().has_entered_low_resolution);
>-  codec_info_.codec_name = "libvpx";
>+  codec_name_ = "libvpx";
>   InsertEncodedFrames(5, 1000);
>   EXPECT_TRUE(statistics_proxy_->GetStats().has_entered_low_resolution);
> 
>@@ -2369,18 +2367,18 @@ TEST_F(ForcedFallbackEnabled, ThreeFallbackEvents) {
>   // Three changes. Video: 60000 ms, fallback: 15000 ms (25%).
>   InsertEncodedFrames(10, 1000);
>   EXPECT_FALSE(statistics_proxy_->GetStats().has_entered_low_resolution);
>-  codec_info_.codec_name = "libvpx";
>+  codec_name_ = "libvpx";
>   InsertEncodedFrames(15, 500);
>   EXPECT_TRUE(statistics_proxy_->GetStats().has_entered_low_resolution);
>-  codec_info_.codec_name = "notlibvpx";
>+  codec_name_ = "notlibvpx";
>   InsertEncodedFrames(20, 1000);
>   InsertEncodedFrames(3, kMaxFrameDiffMs);  // Should not be included.
>   InsertEncodedFrames(10, 1000);
>   EXPECT_TRUE(statistics_proxy_->GetStats().has_entered_low_resolution);
>-  codec_info_.codec_name = "notlibvpx2";
>+  codec_name_ = "notlibvpx2";
>   InsertEncodedFrames(10, 500);
>   EXPECT_TRUE(statistics_proxy_->GetStats().has_entered_low_resolution);
>-  codec_info_.codec_name = "libvpx";
>+  codec_name_ = "libvpx";
>   InsertEncodedFrames(15, 500);
>   EXPECT_TRUE(statistics_proxy_->GetStats().has_entered_low_resolution);
> 
>@@ -2393,7 +2391,7 @@ TEST_F(ForcedFallbackEnabled, ThreeFallbackEvents) {
> 
> TEST_F(ForcedFallbackEnabled, NoFallbackIfAboveMaxPixels) {
>   encoded_image_._encodedWidth = kWidth + 1;
>-  codec_info_.codec_name = "libvpx";
>+  codec_name_ = "libvpx";
>   InsertEncodedFrames(kMinFrames, kFrameIntervalMs);
> 
>   EXPECT_FALSE(statistics_proxy_->GetStats().has_entered_low_resolution);
>@@ -2404,7 +2402,7 @@ TEST_F(ForcedFallbackEnabled, NoFallbackIfAboveMaxPixels) {
> 
> TEST_F(ForcedFallbackEnabled, FallbackIfAtMaxPixels) {
>   encoded_image_._encodedWidth = kWidth;
>-  codec_info_.codec_name = "libvpx";
>+  codec_name_ = "libvpx";
>   InsertEncodedFrames(kMinFrames, kFrameIntervalMs);
> 
>   EXPECT_TRUE(statistics_proxy_->GetStats().has_entered_low_resolution);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/sv_loopback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/sv_loopback.cc
>index fd72c999124d155a5e404a9d6c22624149118d1b..1ff4956ca2950cebf179723044d9aed1003c95fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/sv_loopback.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/sv_loopback.cc
>@@ -12,7 +12,6 @@
> 
> #include "rtc_base/flags.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/stringencode.h"
> #include "system_wrappers/include/field_trial.h"
> #include "test/field_trial.h"
> #include "test/gtest.h"
>@@ -34,73 +33,79 @@ InterLayerPredMode IntToInterLayerPredMode(int inter_layer_pred) {
> }
> 
> // Flags for video.
>-DEFINE_int(vwidth, 640, "Video width.");
>+WEBRTC_DEFINE_int(vwidth, 640, "Video width.");
> size_t VideoWidth() {
>   return static_cast<size_t>(FLAG_vwidth);
> }
> 
>-DEFINE_int(vheight, 480, "Video height.");
>+WEBRTC_DEFINE_int(vheight, 480, "Video height.");
> size_t VideoHeight() {
>   return static_cast<size_t>(FLAG_vheight);
> }
> 
>-DEFINE_int(vfps, 30, "Video frames per second.");
>+WEBRTC_DEFINE_int(vfps, 30, "Video frames per second.");
> int VideoFps() {
>   return static_cast<int>(FLAG_vfps);
> }
> 
>-DEFINE_int(capture_device_index,
>-           0,
>-           "Capture device to select for video stream");
>+WEBRTC_DEFINE_int(capture_device_index,
>+                  0,
>+                  "Capture device to select for video stream");
> size_t GetCaptureDevice() {
>   return static_cast<size_t>(FLAG_capture_device_index);
> }
> 
>-DEFINE_int(vtarget_bitrate, 400, "Video stream target bitrate in kbps.");
>+WEBRTC_DEFINE_int(vtarget_bitrate, 400, "Video stream target bitrate in kbps.");
> int VideoTargetBitrateKbps() {
>   return static_cast<int>(FLAG_vtarget_bitrate);
> }
> 
>-DEFINE_int(vmin_bitrate, 100, "Video stream min bitrate in kbps.");
>+WEBRTC_DEFINE_int(vmin_bitrate, 100, "Video stream min bitrate in kbps.");
> int VideoMinBitrateKbps() {
>   return static_cast<int>(FLAG_vmin_bitrate);
> }
> 
>-DEFINE_int(vmax_bitrate, 2000, "Video stream max bitrate in kbps.");
>+WEBRTC_DEFINE_int(vmax_bitrate, 2000, "Video stream max bitrate in kbps.");
> int VideoMaxBitrateKbps() {
>   return static_cast<int>(FLAG_vmax_bitrate);
> }
> 
>-DEFINE_bool(suspend_below_min_bitrate,
>-            false,
>-            "Suspends video below the configured min bitrate.");
>+WEBRTC_DEFINE_bool(suspend_below_min_bitrate,
>+                   false,
>+                   "Suspends video below the configured min bitrate.");
> 
>-DEFINE_int(vnum_temporal_layers,
>-           1,
>-           "Number of temporal layers for video. Set to 1-4 to override.");
>+WEBRTC_DEFINE_int(
>+    vnum_temporal_layers,
>+    1,
>+    "Number of temporal layers for video. Set to 1-4 to override.");
> int VideoNumTemporalLayers() {
>   return static_cast<int>(FLAG_vnum_temporal_layers);
> }
> 
>-DEFINE_int(vnum_streams, 0, "Number of video streams to show or analyze.");
>+WEBRTC_DEFINE_int(vnum_streams,
>+                  0,
>+                  "Number of video streams to show or analyze.");
> int VideoNumStreams() {
>   return static_cast<int>(FLAG_vnum_streams);
> }
> 
>-DEFINE_int(vnum_spatial_layers, 1, "Number of video spatial layers to use.");
>+WEBRTC_DEFINE_int(vnum_spatial_layers,
>+                  1,
>+                  "Number of video spatial layers to use.");
> int VideoNumSpatialLayers() {
>   return static_cast<int>(FLAG_vnum_spatial_layers);
> }
> 
>-DEFINE_int(vinter_layer_pred,
>-           2,
>-           "Video inter-layer prediction mode. "
>-           "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
>+WEBRTC_DEFINE_int(
>+    vinter_layer_pred,
>+    2,
>+    "Video inter-layer prediction mode. "
>+    "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
> InterLayerPredMode VideoInterLayerPred() {
>   return IntToInterLayerPredMode(FLAG_vinter_layer_pred);
> }
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     vstream0,
>     "",
>     "Comma separated values describing VideoStream for video stream #0.");
>@@ -108,7 +113,7 @@ std::string VideoStream0() {
>   return static_cast<std::string>(FLAG_vstream0);
> }
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     vstream1,
>     "",
>     "Comma separated values describing VideoStream for video stream #1.");
>@@ -116,7 +121,7 @@ std::string VideoStream1() {
>   return static_cast<std::string>(FLAG_vstream1);
> }
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     vsl0,
>     "",
>     "Comma separated values describing SpatialLayer for video layer #0.");
>@@ -124,7 +129,7 @@ std::string VideoSL0() {
>   return static_cast<std::string>(FLAG_vsl0);
> }
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     vsl1,
>     "",
>     "Comma separated values describing SpatialLayer for video layer #1.");
>@@ -132,98 +137,105 @@ std::string VideoSL1() {
>   return static_cast<std::string>(FLAG_vsl1);
> }
> 
>-DEFINE_int(vselected_tl,
>-           -1,
>-           "Temporal layer to show or analyze for screenshare. -1 to disable "
>-           "filtering.");
>+WEBRTC_DEFINE_int(
>+    vselected_tl,
>+    -1,
>+    "Temporal layer to show or analyze for screenshare. -1 to disable "
>+    "filtering.");
> int VideoSelectedTL() {
>   return static_cast<int>(FLAG_vselected_tl);
> }
> 
>-DEFINE_int(vselected_stream,
>-           0,
>-           "ID of the stream to show or analyze for screenshare."
>-           "Set to the number of streams to show them all.");
>+WEBRTC_DEFINE_int(vselected_stream,
>+                  0,
>+                  "ID of the stream to show or analyze for screenshare."
>+                  "Set to the number of streams to show them all.");
> int VideoSelectedStream() {
>   return static_cast<int>(FLAG_vselected_stream);
> }
> 
>-DEFINE_int(vselected_sl,
>-           -1,
>-           "Spatial layer to show or analyze for screenshare. -1 to disable "
>-           "filtering.");
>+WEBRTC_DEFINE_int(
>+    vselected_sl,
>+    -1,
>+    "Spatial layer to show or analyze for screenshare. -1 to disable "
>+    "filtering.");
> int VideoSelectedSL() {
>   return static_cast<int>(FLAG_vselected_sl);
> }
> 
> // Flags for screenshare.
>-DEFINE_int(min_transmit_bitrate,
>-           400,
>-           "Min transmit bitrate incl. padding for screenshare.");
>+WEBRTC_DEFINE_int(min_transmit_bitrate,
>+                  400,
>+                  "Min transmit bitrate incl. padding for screenshare.");
> int ScreenshareMinTransmitBitrateKbps() {
>   return FLAG_min_transmit_bitrate;
> }
> 
>-DEFINE_int(swidth, 1850, "Screenshare width (crops source).");
>+WEBRTC_DEFINE_int(swidth, 1850, "Screenshare width (crops source).");
> size_t ScreenshareWidth() {
>   return static_cast<size_t>(FLAG_swidth);
> }
> 
>-DEFINE_int(sheight, 1110, "Screenshare height (crops source).");
>+WEBRTC_DEFINE_int(sheight, 1110, "Screenshare height (crops source).");
> size_t ScreenshareHeight() {
>   return static_cast<size_t>(FLAG_sheight);
> }
> 
>-DEFINE_int(sfps, 5, "Frames per second for screenshare.");
>+WEBRTC_DEFINE_int(sfps, 5, "Frames per second for screenshare.");
> int ScreenshareFps() {
>   return static_cast<int>(FLAG_sfps);
> }
> 
>-DEFINE_int(starget_bitrate, 100, "Screenshare stream target bitrate in kbps.");
>+WEBRTC_DEFINE_int(starget_bitrate,
>+                  100,
>+                  "Screenshare stream target bitrate in kbps.");
> int ScreenshareTargetBitrateKbps() {
>   return static_cast<int>(FLAG_starget_bitrate);
> }
> 
>-DEFINE_int(smin_bitrate, 100, "Screenshare stream min bitrate in kbps.");
>+WEBRTC_DEFINE_int(smin_bitrate, 100, "Screenshare stream min bitrate in kbps.");
> int ScreenshareMinBitrateKbps() {
>   return static_cast<int>(FLAG_smin_bitrate);
> }
> 
>-DEFINE_int(smax_bitrate, 2000, "Screenshare stream max bitrate in kbps.");
>+WEBRTC_DEFINE_int(smax_bitrate,
>+                  2000,
>+                  "Screenshare stream max bitrate in kbps.");
> int ScreenshareMaxBitrateKbps() {
>   return static_cast<int>(FLAG_smax_bitrate);
> }
> 
>-DEFINE_int(snum_temporal_layers,
>-           2,
>-           "Number of temporal layers to use in screenshare.");
>+WEBRTC_DEFINE_int(snum_temporal_layers,
>+                  2,
>+                  "Number of temporal layers to use in screenshare.");
> int ScreenshareNumTemporalLayers() {
>   return static_cast<int>(FLAG_snum_temporal_layers);
> }
> 
>-DEFINE_int(snum_streams,
>-           0,
>-           "Number of screenshare streams to show or analyze.");
>+WEBRTC_DEFINE_int(snum_streams,
>+                  0,
>+                  "Number of screenshare streams to show or analyze.");
> int ScreenshareNumStreams() {
>   return static_cast<int>(FLAG_snum_streams);
> }
> 
>-DEFINE_int(snum_spatial_layers,
>-           1,
>-           "Number of screenshare spatial layers to use.");
>+WEBRTC_DEFINE_int(snum_spatial_layers,
>+                  1,
>+                  "Number of screenshare spatial layers to use.");
> int ScreenshareNumSpatialLayers() {
>   return static_cast<int>(FLAG_snum_spatial_layers);
> }
> 
>-DEFINE_int(sinter_layer_pred,
>-           0,
>-           "Screenshare inter-layer prediction mode. "
>-           "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
>+WEBRTC_DEFINE_int(
>+    sinter_layer_pred,
>+    0,
>+    "Screenshare inter-layer prediction mode. "
>+    "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
> InterLayerPredMode ScreenshareInterLayerPred() {
>   return IntToInterLayerPredMode(FLAG_sinter_layer_pred);
> }
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     sstream0,
>     "",
>     "Comma separated values describing VideoStream for screenshare stream #0.");
>@@ -231,7 +243,7 @@ std::string ScreenshareStream0() {
>   return static_cast<std::string>(FLAG_sstream0);
> }
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     sstream1,
>     "",
>     "Comma separated values describing VideoStream for screenshare stream #1.");
>@@ -239,7 +251,7 @@ std::string ScreenshareStream1() {
>   return static_cast<std::string>(FLAG_sstream1);
> }
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     ssl0,
>     "",
>     "Comma separated values describing SpatialLayer for screenshare layer #0.");
>@@ -247,7 +259,7 @@ std::string ScreenshareSL0() {
>   return static_cast<std::string>(FLAG_ssl0);
> }
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     ssl1,
>     "",
>     "Comma separated values describing SpatialLayer for screenshare layer #1.");
>@@ -255,31 +267,33 @@ std::string ScreenshareSL1() {
>   return static_cast<std::string>(FLAG_ssl1);
> }
> 
>-DEFINE_int(sselected_tl,
>-           -1,
>-           "Temporal layer to show or analyze for screenshare. -1 to disable "
>-           "filtering.");
>+WEBRTC_DEFINE_int(
>+    sselected_tl,
>+    -1,
>+    "Temporal layer to show or analyze for screenshare. -1 to disable "
>+    "filtering.");
> int ScreenshareSelectedTL() {
>   return static_cast<int>(FLAG_sselected_tl);
> }
> 
>-DEFINE_int(sselected_stream,
>-           0,
>-           "ID of the stream to show or analyze for screenshare."
>-           "Set to the number of streams to show them all.");
>+WEBRTC_DEFINE_int(sselected_stream,
>+                  0,
>+                  "ID of the stream to show or analyze for screenshare."
>+                  "Set to the number of streams to show them all.");
> int ScreenshareSelectedStream() {
>   return static_cast<int>(FLAG_sselected_stream);
> }
> 
>-DEFINE_int(sselected_sl,
>-           -1,
>-           "Spatial layer to show or analyze for screenshare. -1 to disable "
>-           "filtering.");
>+WEBRTC_DEFINE_int(
>+    sselected_sl,
>+    -1,
>+    "Spatial layer to show or analyze for screenshare. -1 to disable "
>+    "filtering.");
> int ScreenshareSelectedSL() {
>   return static_cast<int>(FLAG_sselected_sl);
> }
> 
>-DEFINE_bool(
>+WEBRTC_DEFINE_bool(
>     generate_slides,
>     false,
>     "Whether to use randomly generated slides or read them from files.");
>@@ -287,14 +301,14 @@ bool GenerateSlides() {
>   return static_cast<int>(FLAG_generate_slides);
> }
> 
>-DEFINE_int(slide_change_interval,
>-           10,
>-           "Interval (in seconds) between simulated slide changes.");
>+WEBRTC_DEFINE_int(slide_change_interval,
>+                  10,
>+                  "Interval (in seconds) between simulated slide changes.");
> int SlideChangeInterval() {
>   return static_cast<int>(FLAG_slide_change_interval);
> }
> 
>-DEFINE_int(
>+WEBRTC_DEFINE_int(
>     scroll_duration,
>     0,
>     "Duration (in seconds) during which a slide will be scrolled into place.");
>@@ -302,9 +316,10 @@ int ScrollDuration() {
>   return static_cast<int>(FLAG_scroll_duration);
> }
> 
>-DEFINE_string(slides,
>-              "",
>-              "Comma-separated list of *.yuv files to display as slides.");
>+WEBRTC_DEFINE_string(
>+    slides,
>+    "",
>+    "Comma-separated list of *.yuv files to display as slides.");
> std::vector<std::string> Slides() {
>   std::vector<std::string> slides;
>   std::string slides_list = FLAG_slides;
>@@ -313,31 +328,31 @@ std::vector<std::string> Slides() {
> }
> 
> // Flags common with screenshare and video loopback, with equal default values.
>-DEFINE_int(start_bitrate, 600, "Call start bitrate in kbps.");
>+WEBRTC_DEFINE_int(start_bitrate, 600, "Call start bitrate in kbps.");
> int StartBitrateKbps() {
>   return static_cast<int>(FLAG_start_bitrate);
> }
> 
>-DEFINE_string(codec, "VP8", "Video codec to use.");
>+WEBRTC_DEFINE_string(codec, "VP8", "Video codec to use.");
> std::string Codec() {
>   return static_cast<std::string>(FLAG_codec);
> }
> 
>-DEFINE_bool(analyze_video,
>-            false,
>-            "Analyze video stream (if --duration is present)");
>+WEBRTC_DEFINE_bool(analyze_video,
>+                   false,
>+                   "Analyze video stream (if --duration is present)");
> bool AnalyzeVideo() {
>   return static_cast<bool>(FLAG_analyze_video);
> }
> 
>-DEFINE_bool(analyze_screenshare,
>-            false,
>-            "Analyze screenshare stream (if --duration is present)");
>+WEBRTC_DEFINE_bool(analyze_screenshare,
>+                   false,
>+                   "Analyze screenshare stream (if --duration is present)");
> bool AnalyzeScreenshare() {
>   return static_cast<bool>(FLAG_analyze_screenshare);
> }
> 
>-DEFINE_int(
>+WEBRTC_DEFINE_int(
>     duration,
>     0,
>     "Duration of the test in seconds. If 0, rendered will be shown instead.");
>@@ -345,100 +360,113 @@ int DurationSecs() {
>   return static_cast<int>(FLAG_duration);
> }
> 
>-DEFINE_string(output_filename, "", "Target graph data filename.");
>+WEBRTC_DEFINE_string(output_filename, "", "Target graph data filename.");
> std::string OutputFilename() {
>   return static_cast<std::string>(FLAG_output_filename);
> }
> 
>-DEFINE_string(graph_title,
>-              "",
>-              "If empty, title will be generated automatically.");
>+WEBRTC_DEFINE_string(graph_title,
>+                     "",
>+                     "If empty, title will be generated automatically.");
> std::string GraphTitle() {
>   return static_cast<std::string>(FLAG_graph_title);
> }
> 
>-DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost.");
>+WEBRTC_DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost.");
> int LossPercent() {
>   return static_cast<int>(FLAG_loss_percent);
> }
> 
>-DEFINE_int(avg_burst_loss_length, -1, "Average burst length of lost packets.");
>+WEBRTC_DEFINE_int(avg_burst_loss_length,
>+                  -1,
>+                  "Average burst length of lost packets.");
> int AvgBurstLossLength() {
>   return static_cast<int>(FLAG_avg_burst_loss_length);
> }
> 
>-DEFINE_int(link_capacity,
>-           0,
>-           "Capacity (kbps) of the fake link. 0 means infinite.");
>+WEBRTC_DEFINE_int(link_capacity,
>+                  0,
>+                  "Capacity (kbps) of the fake link. 0 means infinite.");
> int LinkCapacityKbps() {
>   return static_cast<int>(FLAG_link_capacity);
> }
> 
>-DEFINE_int(queue_size, 0, "Size of the bottleneck link queue in packets.");
>+WEBRTC_DEFINE_int(queue_size,
>+                  0,
>+                  "Size of the bottleneck link queue in packets.");
> int QueueSize() {
>   return static_cast<int>(FLAG_queue_size);
> }
> 
>-DEFINE_int(avg_propagation_delay_ms,
>-           0,
>-           "Average link propagation delay in ms.");
>+WEBRTC_DEFINE_int(avg_propagation_delay_ms,
>+                  0,
>+                  "Average link propagation delay in ms.");
> int AvgPropagationDelayMs() {
>   return static_cast<int>(FLAG_avg_propagation_delay_ms);
> }
> 
>-DEFINE_string(rtc_event_log_name,
>-              "",
>-              "Filename for rtc event log. Two files "
>-              "with \"_send\" and \"_recv\" suffixes will be created. "
>-              "Works only when --duration is set.");
>+WEBRTC_DEFINE_string(rtc_event_log_name,
>+                     "",
>+                     "Filename for rtc event log. Two files "
>+                     "with \"_send\" and \"_recv\" suffixes will be created. "
>+                     "Works only when --duration is set.");
> std::string RtcEventLogName() {
>   return static_cast<std::string>(FLAG_rtc_event_log_name);
> }
> 
>-DEFINE_string(rtp_dump_name, "", "Filename for dumped received RTP stream.");
>+WEBRTC_DEFINE_string(rtp_dump_name,
>+                     "",
>+                     "Filename for dumped received RTP stream.");
> std::string RtpDumpName() {
>   return static_cast<std::string>(FLAG_rtp_dump_name);
> }
> 
>-DEFINE_int(std_propagation_delay_ms,
>-           0,
>-           "Link propagation delay standard deviation in ms.");
>+WEBRTC_DEFINE_int(std_propagation_delay_ms,
>+                  0,
>+                  "Link propagation delay standard deviation in ms.");
> int StdPropagationDelayMs() {
>   return static_cast<int>(FLAG_std_propagation_delay_ms);
> }
> 
>-DEFINE_string(encoded_frame_path,
>-              "",
>-              "The base path for encoded frame logs. Created files will have "
>-              "the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf");
>+WEBRTC_DEFINE_string(
>+    encoded_frame_path,
>+    "",
>+    "The base path for encoded frame logs. Created files will have "
>+    "the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf");
> std::string EncodedFramePath() {
>   return static_cast<std::string>(FLAG_encoded_frame_path);
> }
> 
>-DEFINE_bool(logs, false, "print logs to stderr");
>+WEBRTC_DEFINE_bool(logs, false, "print logs to stderr");
> 
>-DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
>+WEBRTC_DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
> 
>-DEFINE_bool(generic_descriptor, false, "Use the generic frame descriptor.");
>+WEBRTC_DEFINE_bool(generic_descriptor,
>+                   false,
>+                   "Use the generic frame descriptor.");
> 
>-DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur");
>+WEBRTC_DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur");
> 
>-DEFINE_bool(use_ulpfec, false, "Use RED+ULPFEC forward error correction.");
>+WEBRTC_DEFINE_bool(use_ulpfec,
>+                   false,
>+                   "Use RED+ULPFEC forward error correction.");
> 
>-DEFINE_bool(use_flexfec, false, "Use FlexFEC forward error correction.");
>+WEBRTC_DEFINE_bool(use_flexfec, false, "Use FlexFEC forward error correction.");
> 
>-DEFINE_bool(audio, false, "Add audio stream");
>+WEBRTC_DEFINE_bool(audio, false, "Add audio stream");
> 
>-DEFINE_bool(audio_video_sync,
>-            false,
>-            "Sync audio and video stream (no effect if"
>-            " audio is false)");
>+WEBRTC_DEFINE_bool(audio_video_sync,
>+                   false,
>+                   "Sync audio and video stream (no effect if"
>+                   " audio is false)");
> 
>-DEFINE_bool(audio_dtx, false, "Enable audio DTX (no effect if audio is false)");
>+WEBRTC_DEFINE_bool(audio_dtx,
>+                   false,
>+                   "Enable audio DTX (no effect if audio is false)");
> 
>-DEFINE_bool(video, true, "Add video stream");
>+WEBRTC_DEFINE_bool(video, true, "Add video stream");
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     force_fieldtrials,
>     "",
>     "Field trials control experimental feature code which can be forced. "
>@@ -447,15 +475,16 @@ DEFINE_string(
>     "trials are separated by \"/\"");
> 
> // Video-specific flags.
>-DEFINE_string(vclip,
>-              "",
>-              "Name of the clip to show. If empty, the camera is used. Use "
>-              "\"Generator\" for chroma generator.");
>+WEBRTC_DEFINE_string(
>+    vclip,
>+    "",
>+    "Name of the clip to show. If empty, the camera is used. Use "
>+    "\"Generator\" for chroma generator.");
> std::string VideoClip() {
>   return static_cast<std::string>(FLAG_vclip);
> }
> 
>-DEFINE_bool(help, false, "prints this message");
>+WEBRTC_DEFINE_bool(help, false, "prints this message");
> 
> }  // namespace flags
> 
>@@ -475,7 +504,7 @@ void Loopback() {
>     screenshare_idx = 0;
>   }
> 
>-  DefaultNetworkSimulationConfig pipe_config;
>+  BuiltInNetworkBehaviorConfig pipe_config;
>   pipe_config.loss_percent = flags::LossPercent();
>   pipe_config.avg_burst_loss_length = flags::AvgBurstLossLength();
>   pipe_config.link_capacity_kbps = flags::LinkCapacityKbps();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/transport_adapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/transport_adapter.h
>index 8ec53fdf25a916c71029c7deae24b5887fe272af..786858b0e21c74103410ec4feb2d44ec4899ae9a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/transport_adapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/transport_adapter.h
>@@ -13,7 +13,6 @@
> #include <atomic>
> 
> #include "api/call/transport.h"
>-#include "common_types.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace internal {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.cc
>index 05abe1fcf3b240b229e23d2dd02137411ae2d831..1fd22ff8060b094a5dd8541b89d4b5b4e8c2e60e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.cc
>@@ -18,17 +18,18 @@
> #include "rtc_base/flags.h"
> #include "rtc_base/format_macros.h"
> #include "rtc_base/memory_usage.h"
>-#include "rtc_base/pathutils.h"
> #include "system_wrappers/include/cpu_info.h"
> #include "test/call_test.h"
>+#include "test/testsupport/fileutils.h"
> #include "test/testsupport/frame_writer.h"
> #include "test/testsupport/perf_test.h"
> #include "test/testsupport/test_artifacts.h"
> 
>-DEFINE_bool(save_worst_frame,
>-            false,
>-            "Enable saving a frame with the lowest PSNR to a jpeg file in the "
>-            "test_artifacts_dir");
>+WEBRTC_DEFINE_bool(
>+    save_worst_frame,
>+    false,
>+    "Enable saving a frame with the lowest PSNR to a jpeg file in the "
>+    "test_artifacts_dir");
> 
> namespace webrtc {
> namespace {
>@@ -70,7 +71,6 @@ VideoAnalyzer::VideoAnalyzer(test::LayerFilteringTransport* transport,
>       selected_stream_(selected_stream),
>       selected_sl_(selected_sl),
>       selected_tl_(selected_tl),
>-      pre_encode_proxy_(this),
>       last_fec_bytes_(0),
>       frames_to_process_(duration_frames),
>       frames_recorded_(0),
>@@ -91,7 +91,6 @@ VideoAnalyzer::VideoAnalyzer(test::LayerFilteringTransport* transport,
>       avg_ssim_threshold_(avg_ssim_threshold),
>       is_quick_test_enabled_(is_quick_test_enabled),
>       stats_polling_thread_(&PollStatsThread, this, "StatsPoller"),
>-      comparison_available_event_(false, false),
>       done_(true, false),
>       clock_(clock),
>       start_ms_(clock->TimeInMilliseconds()) {
>@@ -231,10 +230,10 @@ void VideoAnalyzer::PreEncodeOnFrame(const VideoFrame& video_frame) {
>   }
> }
> 
>-void VideoAnalyzer::EncodedFrameCallback(const EncodedFrame& encoded_frame) {
>+void VideoAnalyzer::PostEncodeOnFrame(size_t stream_id, uint32_t timestamp) {
>   rtc::CritScope lock(&crit_);
>-  if (!first_sent_timestamp_ && encoded_frame.stream_id_ == selected_stream_) {
>-    first_sent_timestamp_ = encoded_frame.timestamp_;
>+  if (!first_sent_timestamp_ && stream_id == selected_stream_) {
>+    first_sent_timestamp_ = timestamp;
>   }
> }
> 
>@@ -366,10 +365,6 @@ void VideoAnalyzer::Wait() {
>   stats_polling_thread_.Stop();
> }
> 
>-rtc::VideoSinkInterface<VideoFrame>* VideoAnalyzer::pre_encode_proxy() {
>-  return &pre_encode_proxy_;
>-}
>-
> void VideoAnalyzer::StartMeasuringCpuProcessTime() {
>   rtc::CritScope lock(&cpu_measurement_lock_);
>   cpu_time_ -= rtc::GetProcessCpuTimeNanos();
>@@ -611,7 +606,7 @@ void VideoAnalyzer::PrintResults() {
>     std::string output_dir;
>     test::GetTestArtifactsDir(&output_dir);
>     std::string output_path =
>-        rtc::Pathname(output_dir, test_label_ + ".jpg").pathname();
>+        test::JoinFilename(output_dir, test_label_ + ".jpg");
>     RTC_LOG(LS_INFO) << "Saving worst frame to " << output_path;
>     test::JpegFrameWriter frame_writer(output_path);
>     RTC_CHECK(
>@@ -841,13 +836,6 @@ VideoAnalyzer::Sample::Sample(int dropped,
>       psnr(psnr),
>       ssim(ssim) {}
> 
>-VideoAnalyzer::PreEncodeProxy::PreEncodeProxy(VideoAnalyzer* parent)
>-    : parent_(parent) {}
>-
>-void VideoAnalyzer::PreEncodeProxy::OnFrame(const VideoFrame& video_frame) {
>-  parent_->PreEncodeOnFrame(video_frame);
>-}
>-
> VideoAnalyzer::CapturedFrameForwarder::CapturedFrameForwarder(
>     VideoAnalyzer* analyzer,
>     Clock* clock)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.h
>index 681d1f464f481c6e7baf3f1ff61fdcae2f40d7cd..d2c4fa84cda3a15b37e6007698b7bcc9f97fae7b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.h
>@@ -25,8 +25,7 @@ namespace webrtc {
> 
> class VideoAnalyzer : public PacketReceiver,
>                       public Transport,
>-                      public rtc::VideoSinkInterface<VideoFrame>,
>-                      public EncodedFrameObserver {
>+                      public rtc::VideoSinkInterface<VideoFrame> {
>  public:
>   VideoAnalyzer(test::LayerFilteringTransport* transport,
>                 const std::string& test_label,
>@@ -61,9 +60,7 @@ class VideoAnalyzer : public PacketReceiver,
>                                int64_t packet_time_us) override;
> 
>   void PreEncodeOnFrame(const VideoFrame& video_frame);
>-
>-  // EncodedFrameObserver implementation, wired to post_encode_callback.
>-  void EncodedFrameCallback(const EncodedFrame& encoded_frame) override;
>+  void PostEncodeOnFrame(size_t stream_id, uint32_t timestamp);
> 
>   bool SendRtp(const uint8_t* packet,
>                size_t length,
>@@ -73,8 +70,6 @@ class VideoAnalyzer : public PacketReceiver,
>   void OnFrame(const VideoFrame& video_frame) override;
>   void Wait();
> 
>-  rtc::VideoSinkInterface<VideoFrame>* pre_encode_proxy();
>-
>   void StartMeasuringCpuProcessTime();
>   void StopMeasuringCpuProcessTime();
>   void StartExcludingCpuThreadTime();
>@@ -132,17 +127,6 @@ class VideoAnalyzer : public PacketReceiver,
>     double ssim;
>   };
> 
>-  // This class receives the send-side OnFrame callback and is provided to not
>-  // conflict with the receiver-side renderer callback.
>-  class PreEncodeProxy : public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    explicit PreEncodeProxy(VideoAnalyzer* parent);
>-    void OnFrame(const VideoFrame& video_frame) override;
>-
>-   private:
>-    VideoAnalyzer* const parent_;
>-  };
>-
>   // Implements VideoSinkInterface to receive captured frames from a
>   // FrameGeneratorCapturer. Implements VideoSourceInterface to be able to act
>   // as a source to VideoSendStream.
>@@ -221,7 +205,6 @@ class VideoAnalyzer : public PacketReceiver,
>   const size_t selected_stream_;
>   const int selected_sl_;
>   const int selected_tl_;
>-  PreEncodeProxy pre_encode_proxy_;
> 
>   rtc::CriticalSection comparison_lock_;
>   std::vector<Sample> samples_ RTC_GUARDED_BY(comparison_lock_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_loopback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_loopback.cc
>index 57590ed144954756e9097366a7c09a76eb03c1f1..4cdddb94e5285850d93d93b098ec813d72b0e363 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_loopback.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_loopback.cc
>@@ -22,61 +22,62 @@ namespace webrtc {
> namespace flags {
> 
> // Flags common with screenshare loopback, with different default values.
>-DEFINE_int(width, 640, "Video width.");
>+WEBRTC_DEFINE_int(width, 640, "Video width.");
> size_t Width() {
>   return static_cast<size_t>(FLAG_width);
> }
> 
>-DEFINE_int(height, 480, "Video height.");
>+WEBRTC_DEFINE_int(height, 480, "Video height.");
> size_t Height() {
>   return static_cast<size_t>(FLAG_height);
> }
> 
>-DEFINE_int(fps, 30, "Frames per second.");
>+WEBRTC_DEFINE_int(fps, 30, "Frames per second.");
> int Fps() {
>   return static_cast<int>(FLAG_fps);
> }
> 
>-DEFINE_int(capture_device_index, 0, "Capture device to select");
>+WEBRTC_DEFINE_int(capture_device_index, 0, "Capture device to select");
> size_t GetCaptureDevice() {
>   return static_cast<size_t>(FLAG_capture_device_index);
> }
> 
>-DEFINE_int(min_bitrate, 50, "Call and stream min bitrate in kbps.");
>+WEBRTC_DEFINE_int(min_bitrate, 50, "Call and stream min bitrate in kbps.");
> int MinBitrateKbps() {
>   return static_cast<int>(FLAG_min_bitrate);
> }
> 
>-DEFINE_int(start_bitrate, 300, "Call start bitrate in kbps.");
>+WEBRTC_DEFINE_int(start_bitrate, 300, "Call start bitrate in kbps.");
> int StartBitrateKbps() {
>   return static_cast<int>(FLAG_start_bitrate);
> }
> 
>-DEFINE_int(target_bitrate, 800, "Stream target bitrate in kbps.");
>+WEBRTC_DEFINE_int(target_bitrate, 800, "Stream target bitrate in kbps.");
> int TargetBitrateKbps() {
>   return static_cast<int>(FLAG_target_bitrate);
> }
> 
>-DEFINE_int(max_bitrate, 800, "Call and stream max bitrate in kbps.");
>+WEBRTC_DEFINE_int(max_bitrate, 800, "Call and stream max bitrate in kbps.");
> int MaxBitrateKbps() {
>   return static_cast<int>(FLAG_max_bitrate);
> }
> 
>-DEFINE_bool(suspend_below_min_bitrate,
>-            false,
>-            "Suspends video below the configured min bitrate.");
>+WEBRTC_DEFINE_bool(suspend_below_min_bitrate,
>+                   false,
>+                   "Suspends video below the configured min bitrate.");
> 
>-DEFINE_int(num_temporal_layers,
>-           1,
>-           "Number of temporal layers. Set to 1-4 to override.");
>+WEBRTC_DEFINE_int(num_temporal_layers,
>+                  1,
>+                  "Number of temporal layers. Set to 1-4 to override.");
> int NumTemporalLayers() {
>   return static_cast<int>(FLAG_num_temporal_layers);
> }
> 
>-DEFINE_int(inter_layer_pred,
>-           2,
>-           "Inter-layer prediction mode. "
>-           "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
>+WEBRTC_DEFINE_int(
>+    inter_layer_pred,
>+    2,
>+    "Inter-layer prediction mode. "
>+    "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
> InterLayerPredMode InterLayerPred() {
>   if (FLAG_inter_layer_pred == 0) {
>     return InterLayerPredMode::kOn;
>@@ -89,19 +90,20 @@ InterLayerPredMode InterLayerPred() {
> }
> 
> // Flags common with screenshare loopback, with equal default values.
>-DEFINE_string(codec, "VP8", "Video codec to use.");
>+WEBRTC_DEFINE_string(codec, "VP8", "Video codec to use.");
> std::string Codec() {
>   return static_cast<std::string>(FLAG_codec);
> }
> 
>-DEFINE_int(selected_tl,
>-           -1,
>-           "Temporal layer to show or analyze. -1 to disable filtering.");
>+WEBRTC_DEFINE_int(
>+    selected_tl,
>+    -1,
>+    "Temporal layer to show or analyze. -1 to disable filtering.");
> int SelectedTL() {
>   return static_cast<int>(FLAG_selected_tl);
> }
> 
>-DEFINE_int(
>+WEBRTC_DEFINE_int(
>     duration,
>     0,
>     "Duration of the test in seconds. If 0, rendered will be shown instead.");
>@@ -109,177 +111,196 @@ int DurationSecs() {
>   return static_cast<int>(FLAG_duration);
> }
> 
>-DEFINE_string(output_filename, "", "Target graph data filename.");
>+WEBRTC_DEFINE_string(output_filename, "", "Target graph data filename.");
> std::string OutputFilename() {
>   return static_cast<std::string>(FLAG_output_filename);
> }
> 
>-DEFINE_string(graph_title,
>-              "",
>-              "If empty, title will be generated automatically.");
>+WEBRTC_DEFINE_string(graph_title,
>+                     "",
>+                     "If empty, title will be generated automatically.");
> std::string GraphTitle() {
>   return static_cast<std::string>(FLAG_graph_title);
> }
> 
>-DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost.");
>+WEBRTC_DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost.");
> int LossPercent() {
>   return static_cast<int>(FLAG_loss_percent);
> }
> 
>-DEFINE_int(avg_burst_loss_length, -1, "Average burst length of lost packets.");
>+WEBRTC_DEFINE_int(avg_burst_loss_length,
>+                  -1,
>+                  "Average burst length of lost packets.");
> int AvgBurstLossLength() {
>   return static_cast<int>(FLAG_avg_burst_loss_length);
> }
> 
>-DEFINE_int(link_capacity,
>-           0,
>-           "Capacity (kbps) of the fake link. 0 means infinite.");
>+WEBRTC_DEFINE_int(link_capacity,
>+                  0,
>+                  "Capacity (kbps) of the fake link. 0 means infinite.");
> int LinkCapacityKbps() {
>   return static_cast<int>(FLAG_link_capacity);
> }
> 
>-DEFINE_int(queue_size, 0, "Size of the bottleneck link queue in packets.");
>+WEBRTC_DEFINE_int(queue_size,
>+                  0,
>+                  "Size of the bottleneck link queue in packets.");
> int QueueSize() {
>   return static_cast<int>(FLAG_queue_size);
> }
> 
>-DEFINE_int(avg_propagation_delay_ms,
>-           0,
>-           "Average link propagation delay in ms.");
>+WEBRTC_DEFINE_int(avg_propagation_delay_ms,
>+                  0,
>+                  "Average link propagation delay in ms.");
> int AvgPropagationDelayMs() {
>   return static_cast<int>(FLAG_avg_propagation_delay_ms);
> }
> 
>-DEFINE_string(rtc_event_log_name,
>-              "",
>-              "Filename for rtc event log. Two files "
>-              "with \"_send\" and \"_recv\" suffixes will be created.");
>+WEBRTC_DEFINE_string(rtc_event_log_name,
>+                     "",
>+                     "Filename for rtc event log. Two files "
>+                     "with \"_send\" and \"_recv\" suffixes will be created.");
> std::string RtcEventLogName() {
>   return static_cast<std::string>(FLAG_rtc_event_log_name);
> }
> 
>-DEFINE_string(rtp_dump_name, "", "Filename for dumped received RTP stream.");
>+WEBRTC_DEFINE_string(rtp_dump_name,
>+                     "",
>+                     "Filename for dumped received RTP stream.");
> std::string RtpDumpName() {
>   return static_cast<std::string>(FLAG_rtp_dump_name);
> }
> 
>-DEFINE_int(std_propagation_delay_ms,
>-           0,
>-           "Link propagation delay standard deviation in ms.");
>+WEBRTC_DEFINE_int(std_propagation_delay_ms,
>+                  0,
>+                  "Link propagation delay standard deviation in ms.");
> int StdPropagationDelayMs() {
>   return static_cast<int>(FLAG_std_propagation_delay_ms);
> }
> 
>-DEFINE_int(num_streams, 0, "Number of streams to show or analyze.");
>+WEBRTC_DEFINE_int(num_streams, 0, "Number of streams to show or analyze.");
> int NumStreams() {
>   return static_cast<int>(FLAG_num_streams);
> }
> 
>-DEFINE_int(selected_stream,
>-           0,
>-           "ID of the stream to show or analyze. "
>-           "Set to the number of streams to show them all.");
>+WEBRTC_DEFINE_int(selected_stream,
>+                  0,
>+                  "ID of the stream to show or analyze. "
>+                  "Set to the number of streams to show them all.");
> int SelectedStream() {
>   return static_cast<int>(FLAG_selected_stream);
> }
> 
>-DEFINE_int(num_spatial_layers, 1, "Number of spatial layers to use.");
>+WEBRTC_DEFINE_int(num_spatial_layers, 1, "Number of spatial layers to use.");
> int NumSpatialLayers() {
>   return static_cast<int>(FLAG_num_spatial_layers);
> }
> 
>-DEFINE_int(selected_sl,
>-           -1,
>-           "Spatial layer to show or analyze. -1 to disable filtering.");
>+WEBRTC_DEFINE_int(selected_sl,
>+                  -1,
>+                  "Spatial layer to show or analyze. -1 to disable filtering.");
> int SelectedSL() {
>   return static_cast<int>(FLAG_selected_sl);
> }
> 
>-DEFINE_string(stream0,
>-              "",
>-              "Comma separated values describing VideoStream for stream #0.");
>+WEBRTC_DEFINE_string(
>+    stream0,
>+    "",
>+    "Comma separated values describing VideoStream for stream #0.");
> std::string Stream0() {
>   return static_cast<std::string>(FLAG_stream0);
> }
> 
>-DEFINE_string(stream1,
>-              "",
>-              "Comma separated values describing VideoStream for stream #1.");
>+WEBRTC_DEFINE_string(
>+    stream1,
>+    "",
>+    "Comma separated values describing VideoStream for stream #1.");
> std::string Stream1() {
>   return static_cast<std::string>(FLAG_stream1);
> }
> 
>-DEFINE_string(sl0,
>-              "",
>-              "Comma separated values describing SpatialLayer for layer #0.");
>+WEBRTC_DEFINE_string(
>+    sl0,
>+    "",
>+    "Comma separated values describing SpatialLayer for layer #0.");
> std::string SL0() {
>   return static_cast<std::string>(FLAG_sl0);
> }
> 
>-DEFINE_string(sl1,
>-              "",
>-              "Comma separated values describing SpatialLayer for layer #1.");
>+WEBRTC_DEFINE_string(
>+    sl1,
>+    "",
>+    "Comma separated values describing SpatialLayer for layer #1.");
> std::string SL1() {
>   return static_cast<std::string>(FLAG_sl1);
> }
> 
>-DEFINE_string(encoded_frame_path,
>-              "",
>-              "The base path for encoded frame logs. Created files will have "
>-              "the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf");
>+WEBRTC_DEFINE_string(
>+    encoded_frame_path,
>+    "",
>+    "The base path for encoded frame logs. Created files will have "
>+    "the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf");
> std::string EncodedFramePath() {
>   return static_cast<std::string>(FLAG_encoded_frame_path);
> }
> 
>-DEFINE_bool(logs, false, "print logs to stderr");
>+WEBRTC_DEFINE_bool(logs, false, "print logs to stderr");
> 
>-DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
>+WEBRTC_DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
> 
>-DEFINE_bool(generic_descriptor, false, "Use the generic frame descriptor.");
>+WEBRTC_DEFINE_bool(generic_descriptor,
>+                   false,
>+                   "Use the generic frame descriptor.");
> 
>-DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur");
>+WEBRTC_DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur");
> 
>-DEFINE_bool(use_ulpfec, false, "Use RED+ULPFEC forward error correction.");
>+WEBRTC_DEFINE_bool(use_ulpfec,
>+                   false,
>+                   "Use RED+ULPFEC forward error correction.");
> 
>-DEFINE_bool(use_flexfec, false, "Use FlexFEC forward error correction.");
>+WEBRTC_DEFINE_bool(use_flexfec, false, "Use FlexFEC forward error correction.");
> 
>-DEFINE_bool(audio, false, "Add audio stream");
>+WEBRTC_DEFINE_bool(audio, false, "Add audio stream");
> 
>-DEFINE_bool(use_real_adm,
>-            false,
>-            "Use real ADM instead of fake (no effect if audio is false)");
>+WEBRTC_DEFINE_bool(
>+    use_real_adm,
>+    false,
>+    "Use real ADM instead of fake (no effect if audio is false)");
> 
>-DEFINE_bool(audio_video_sync,
>-            false,
>-            "Sync audio and video stream (no effect if"
>-            " audio is false)");
>+WEBRTC_DEFINE_bool(audio_video_sync,
>+                   false,
>+                   "Sync audio and video stream (no effect if"
>+                   " audio is false)");
> 
>-DEFINE_bool(audio_dtx, false, "Enable audio DTX (no effect if audio is false)");
>+WEBRTC_DEFINE_bool(audio_dtx,
>+                   false,
>+                   "Enable audio DTX (no effect if audio is false)");
> 
>-DEFINE_bool(video, true, "Add video stream");
>+WEBRTC_DEFINE_bool(video, true, "Add video stream");
> 
>-DEFINE_string(
>+WEBRTC_DEFINE_string(
>     force_fieldtrials,
>     "",
>     "Field trials control experimental feature code which can be forced. "
>-    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
>+    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/"
>     " will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
>     "trials are separated by \"/\"");
> 
> // Video-specific flags.
>-DEFINE_string(clip,
>-              "",
>-              "Name of the clip to show. If empty, using chroma generator.");
>+WEBRTC_DEFINE_string(
>+    clip,
>+    "",
>+    "Name of the clip to show. If empty, using chroma generator.");
> std::string Clip() {
>   return static_cast<std::string>(FLAG_clip);
> }
> 
>-DEFINE_bool(help, false, "prints this message");
>+WEBRTC_DEFINE_bool(help, false, "prints this message");
> 
> }  // namespace flags
> 
> void Loopback() {
>-  DefaultNetworkSimulationConfig pipe_config;
>+  BuiltInNetworkBehaviorConfig pipe_config;
>   pipe_config.loss_percent = flags::LossPercent();
>   pipe_config.avg_burst_loss_length = flags::AvgBurstLossLength();
>   pipe_config.link_capacity_kbps = flags::LinkCapacityKbps();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.cc
>index b05c3535e0ce717152321f4a6b0a6fa514478d09..c4a574abfe0e60a182879c5349f1336878ef4359 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.cc
>@@ -34,7 +34,9 @@ const int kBlockyQpThresholdVp9 = 60;  // TODO(ilnik): tune this value.
> 
> VideoQualityObserver::VideoQualityObserver(VideoContentType content_type)
>     : last_frame_decoded_ms_(-1),
>+      last_frame_rendered_ms_(-1),
>       num_frames_decoded_(0),
>+      num_frames_rendered_(0),
>       first_frame_decoded_ms_(-1),
>       last_frame_pixels_(0),
>       last_frame_qp_(0),
>@@ -119,6 +121,46 @@ void VideoQualityObserver::UpdateHistograms() {
>   RTC_LOG(LS_INFO) << log_stream.str();
> }
> 
>+void VideoQualityObserver::OnRenderedFrame(int64_t now_ms) {
>+  if (num_frames_rendered_ == 0) {
>+    last_unfreeze_time_ = now_ms;
>+  }
>+
>+  ++num_frames_rendered_;
>+
>+  if (!is_paused_ && num_frames_rendered_ > 1) {
>+    // Process inter-frame delay.
>+    int64_t interframe_delay_ms = now_ms - last_frame_rendered_ms_;
>+    render_interframe_delays_.Add(interframe_delay_ms);
>+    absl::optional<int> avg_interframe_delay =
>+        render_interframe_delays_.Avg(kMinFrameSamplesToDetectFreeze);
>+    // Check if it was a freeze.
>+    if (avg_interframe_delay &&
>+        interframe_delay_ms >=
>+            std::max(3 * *avg_interframe_delay,
>+                     *avg_interframe_delay + kMinIncreaseForFreezeMs)) {
>+      freezes_durations_.Add(interframe_delay_ms);
>+      smooth_playback_durations_.Add(last_frame_rendered_ms_ -
>+                                     last_unfreeze_time_);
>+      last_unfreeze_time_ = now_ms;
>+    }
>+  }
>+
>+  if (is_paused_) {
>+    // If the stream was paused since the previous frame, do not count the
>+    // pause toward smooth playback. Explicitly count the part before it and
>+    // start the new smooth playback interval from this frame.
>+    is_paused_ = false;
>+    if (last_frame_rendered_ms_ > last_unfreeze_time_) {
>+      smooth_playback_durations_.Add(last_frame_rendered_ms_ -
>+                                     last_unfreeze_time_);
>+    }
>+    last_unfreeze_time_ = now_ms;
>+  }
>+
>+  last_frame_rendered_ms_ = now_ms;
>+}
>+
> void VideoQualityObserver::OnDecodedFrame(absl::optional<uint8_t> qp,
>                                           int width,
>                                           int height,
>@@ -126,7 +168,6 @@ void VideoQualityObserver::OnDecodedFrame(absl::optional<uint8_t> qp,
>                                           VideoCodecType codec) {
>   if (num_frames_decoded_ == 0) {
>     first_frame_decoded_ms_ = now_ms;
>-    last_unfreeze_time_ = now_ms;
>   }
> 
>   ++num_frames_decoded_;
>@@ -134,21 +175,14 @@ void VideoQualityObserver::OnDecodedFrame(absl::optional<uint8_t> qp,
>   if (!is_paused_ && num_frames_decoded_ > 1) {
>     // Process inter-frame delay.
>     int64_t interframe_delay_ms = now_ms - last_frame_decoded_ms_;
>-    interframe_delays_.Add(interframe_delay_ms);
>+    decode_interframe_delays_.Add(interframe_delay_ms);
>     absl::optional<int> avg_interframe_delay =
>-        interframe_delays_.Avg(kMinFrameSamplesToDetectFreeze);
>-    // Check if it was a freeze.
>-    if (avg_interframe_delay &&
>-        interframe_delay_ms >=
>+        decode_interframe_delays_.Avg(kMinFrameSamplesToDetectFreeze);
>+    // Count spatial metrics if there were no freeze.
>+    if (!avg_interframe_delay ||
>+        interframe_delay_ms <
>             std::max(3 * *avg_interframe_delay,
>                      *avg_interframe_delay + kMinIncreaseForFreezeMs)) {
>-      freezes_durations_.Add(interframe_delay_ms);
>-      smooth_playback_durations_.Add(last_frame_decoded_ms_ -
>-                                     last_unfreeze_time_);
>-      last_unfreeze_time_ = now_ms;
>-    } else {
>-      // Only count inter-frame delay as playback time if there
>-      // was no freeze.
>       time_in_resolution_ms_[current_resolution_] += interframe_delay_ms;
>       absl::optional<int> qp_blocky_threshold;
>       // TODO(ilnik): add other codec types when we have QP for them.
>@@ -168,18 +202,6 @@ void VideoQualityObserver::OnDecodedFrame(absl::optional<uint8_t> qp,
>     }
>   }
> 
>-  if (is_paused_) {
>-    // If the stream was paused since the previous frame, do not count the
>-    // pause toward smooth playback. Explicitly count the part before it and
>-    // start the new smooth playback interval from this frame.
>-    is_paused_ = false;
>-    if (last_frame_decoded_ms_ > last_unfreeze_time_) {
>-      smooth_playback_durations_.Add(last_frame_decoded_ms_ -
>-                                     last_unfreeze_time_);
>-    }
>-    last_unfreeze_time_ = now_ms;
>-  }
>-
>   int64_t pixels = width * height;
>   if (pixels >= kPixelsInHighResolution) {
>     current_resolution_ = Resolution::High;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.h
>index 8ba4af39842a8de010683e1cda7448c1f6d3dbb1..74f201e8c45f53f567ee0c9759f8598cb7344f7d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.h
>@@ -15,8 +15,8 @@
> #include <vector>
> 
> #include "absl/types/optional.h"
>+#include "api/video/video_codec_type.h"
> #include "api/video/video_content_type.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/numerics/sample_counter.h"
> 
> namespace webrtc {
>@@ -35,6 +35,7 @@ class VideoQualityObserver {
>                       int height,
>                       int64_t now_ms,
>                       VideoCodecType codec);
>+  void OnRenderedFrame(int64_t now_ms);
> 
>   void OnStreamInactive();
> 
>@@ -48,13 +49,16 @@ class VideoQualityObserver {
>   };
> 
>   int64_t last_frame_decoded_ms_;
>+  int64_t last_frame_rendered_ms_;
>   int64_t num_frames_decoded_;
>+  int64_t num_frames_rendered_;
>   int64_t first_frame_decoded_ms_;
>   int64_t last_frame_pixels_;
>   uint8_t last_frame_qp_;
>   // Decoded timestamp of the last delayed frame.
>   int64_t last_unfreeze_time_;
>-  rtc::SampleCounter interframe_delays_;
>+  rtc::SampleCounter render_interframe_delays_;
>+  rtc::SampleCounter decode_interframe_delays_;
>   // An inter-frame delay is counted as a freeze if it's significantly longer
>   // than average inter-frame delay.
>   rtc::SampleCounter freezes_durations_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.cc
>index 2a3be672d76068a141543e890ba1d8b61b5bc9ab..d6ccb6522e5832177fc5b485329dadd91a027672 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.cc
>@@ -16,6 +16,7 @@
> #include <string>
> #include <vector>
> 
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "call/fake_network_pipe.h"
> #include "call/simulated_network.h"
> #include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
>@@ -35,7 +36,7 @@
> #include "test/run_loop.h"
> #include "test/testsupport/fileutils.h"
> #include "test/video_renderer.h"
>-#include "video/video_analyzer.h"
>+#include "video/frame_dumping_decoder.h"
> #ifdef WEBRTC_WIN
> #include "modules/audio_device/include/audio_device_factory.h"
> #endif
>@@ -75,61 +76,16 @@ class VideoStreamFactory
>   std::vector<VideoStream> streams_;
> };
> 
>-// A decoder wrapper that writes the encoded frames to a file.
>-class FrameDumpingDecoder : public VideoDecoder {
>+// This wrapper provides two features needed by the video quality tests:
>+//  1. Invoke VideoAnalyzer callbacks before and after encoding each frame.
>+//  2. Write the encoded frames to file, one file per simulcast layer.
>+class QualityTestVideoEncoder : public VideoEncoder,
>+                                private EncodedImageCallback {
>  public:
>-  FrameDumpingDecoder(std::unique_ptr<VideoDecoder> decoder,
>-                      rtc::PlatformFile file)
>-      : decoder_(std::move(decoder)),
>-        writer_(IvfFileWriter::Wrap(rtc::File(file),
>-                                    /* byte_limit= */ 100000000)) {}
>-
>-  int32_t InitDecode(const VideoCodec* codec_settings,
>-                     int32_t number_of_cores) override {
>-    return decoder_->InitDecode(codec_settings, number_of_cores);
>-  }
>-
>-  int32_t Decode(const EncodedImage& input_image,
>-                 bool missing_frames,
>-                 const CodecSpecificInfo* codec_specific_info,
>-                 int64_t render_time_ms) override {
>-    int32_t ret = decoder_->Decode(input_image, missing_frames,
>-                                   codec_specific_info, render_time_ms);
>-    writer_->WriteFrame(input_image, codec_specific_info->codecType);
>-
>-    return ret;
>-  }
>-
>-  int32_t RegisterDecodeCompleteCallback(
>-      DecodedImageCallback* callback) override {
>-    return decoder_->RegisterDecodeCompleteCallback(callback);
>-  }
>-
>-  int32_t Release() override { return decoder_->Release(); }
>-
>-  // Returns true if the decoder prefer to decode frames late.
>-  // That is, it can not decode infinite number of frames before the decoded
>-  // frame is consumed.
>-  bool PrefersLateDecoding() const override {
>-    return decoder_->PrefersLateDecoding();
>-  }
>-
>-  const char* ImplementationName() const override {
>-    return decoder_->ImplementationName();
>-  }
>-
>- private:
>-  std::unique_ptr<VideoDecoder> decoder_;
>-  std::unique_ptr<IvfFileWriter> writer_;
>-};
>-
>-// An encoder wrapper that writes the encoded frames to file, one per simulcast
>-// layer.
>-class FrameDumpingEncoder : public VideoEncoder, private EncodedImageCallback {
>- public:
>-  FrameDumpingEncoder(std::unique_ptr<VideoEncoder> encoder,
>-                      std::vector<rtc::PlatformFile> files)
>-      : encoder_(std::move(encoder)) {
>+  QualityTestVideoEncoder(std::unique_ptr<VideoEncoder> encoder,
>+                          VideoAnalyzer* analyzer,
>+                          std::vector<rtc::PlatformFile> files)
>+      : encoder_(std::move(encoder)), analyzer_(analyzer) {
>     for (rtc::PlatformFile file : files) {
>       writers_.push_back(
>           IvfFileWriter::Wrap(rtc::File(file), /* byte_limit= */ 100000000));
>@@ -151,11 +107,11 @@ class FrameDumpingEncoder : public VideoEncoder, private EncodedImageCallback {
>   int32_t Encode(const VideoFrame& frame,
>                  const CodecSpecificInfo* codec_specific_info,
>                  const std::vector<FrameType>* frame_types) {
>+    if (analyzer_) {
>+      analyzer_->PreEncodeOnFrame(frame);
>+    }
>     return encoder_->Encode(frame, codec_specific_info, frame_types);
>   }
>-  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override {
>-    return encoder_->SetChannelParameters(packet_loss, rtt);
>-  }
>   int32_t SetRates(uint32_t bitrate, uint32_t framerate) override {
>     return encoder_->SetRates(bitrate, framerate);
>   }
>@@ -163,14 +119,8 @@ class FrameDumpingEncoder : public VideoEncoder, private EncodedImageCallback {
>                             uint32_t framerate) override {
>     return encoder_->SetRateAllocation(allocation, framerate);
>   }
>-  ScalingSettings GetScalingSettings() const override {
>-    return encoder_->GetScalingSettings();
>-  }
>-  bool SupportsNativeHandle() const override {
>-    return encoder_->SupportsNativeHandle();
>-  }
>-  const char* ImplementationName() const override {
>-    return encoder_->ImplementationName();
>+  EncoderInfo GetEncoderInfo() const override {
>+    return encoder_->GetEncoderInfo();
>   }
> 
>  private:
>@@ -186,6 +136,10 @@ class FrameDumpingEncoder : public VideoEncoder, private EncodedImageCallback {
>         simulcast_index = encoded_image.SpatialIndex().value_or(0);
>       }
>       RTC_DCHECK_GE(simulcast_index, 0);
>+      if (analyzer_) {
>+        analyzer_->PostEncodeOnFrame(simulcast_index,
>+                                     encoded_image.Timestamp());
>+      }
>       if (static_cast<size_t>(simulcast_index) < writers_.size()) {
>         writers_[simulcast_index]->WriteFrame(encoded_image,
>                                               codec_specific_info->codecType);
>@@ -202,6 +156,7 @@ class FrameDumpingEncoder : public VideoEncoder, private EncodedImageCallback {
> 
>   std::unique_ptr<VideoEncoder> encoder_;
>   EncodedImageCallback* callback_ = nullptr;
>+  VideoAnalyzer* const analyzer_;
>   std::vector<std::unique_ptr<IvfFileWriter>> writers_;
> };
> 
>@@ -228,7 +183,8 @@ std::unique_ptr<VideoDecoder> VideoQualityTest::CreateVideoDecoder(
> }
> 
> std::unique_ptr<VideoEncoder> VideoQualityTest::CreateVideoEncoder(
>-    const SdpVideoFormat& format) {
>+    const SdpVideoFormat& format,
>+    VideoAnalyzer* analyzer) {
>   std::unique_ptr<VideoEncoder> encoder;
>   if (format.name == "VP8") {
>     encoder = absl::make_unique<VP8EncoderSimulcastProxy>(
>@@ -245,12 +201,17 @@ std::unique_ptr<VideoEncoder> VideoQualityTest::CreateVideoEncoder(
>     sb << send_logs_++;
>     std::string prefix =
>         params_.logging.encoded_frame_base_path + "." + sb.str() + ".send.";
>-    encoder = absl::make_unique<FrameDumpingEncoder>(
>-        std::move(encoder), std::vector<rtc::PlatformFile>(
>-                                {rtc::CreatePlatformFile(prefix + "1.ivf"),
>-                                 rtc::CreatePlatformFile(prefix + "2.ivf"),
>-                                 rtc::CreatePlatformFile(prefix + "3.ivf")}));
>+    encoder = absl::make_unique<QualityTestVideoEncoder>(
>+        std::move(encoder), analyzer,
>+        std::vector<rtc::PlatformFile>(
>+            {rtc::CreatePlatformFile(prefix + "1.ivf"),
>+             rtc::CreatePlatformFile(prefix + "2.ivf"),
>+             rtc::CreatePlatformFile(prefix + "3.ivf")}));
>+  } else if (analyzer) {
>+    encoder = absl::make_unique<QualityTestVideoEncoder>(
>+        std::move(encoder), analyzer, std::vector<rtc::PlatformFile>());
>   }
>+
>   return encoder;
> }
> 
>@@ -261,11 +222,18 @@ VideoQualityTest::VideoQualityTest(
>         return this->CreateVideoDecoder(format);
>       }),
>       video_encoder_factory_([this](const SdpVideoFormat& format) {
>-        return this->CreateVideoEncoder(format);
>+        return this->CreateVideoEncoder(format, nullptr);
>       }),
>+      video_encoder_factory_with_analyzer_(
>+          [this](const SdpVideoFormat& format) {
>+            return this->CreateVideoEncoder(format, analyzer_.get());
>+          }),
>+      video_bitrate_allocator_factory_(
>+          CreateBuiltinVideoBitrateAllocatorFactory()),
>       receive_logs_(0),
>       send_logs_(0),
>-      injection_components_(std::move(injection_components)) {
>+      injection_components_(std::move(injection_components)),
>+      num_video_streams_(0) {
>   if (injection_components_ == nullptr) {
>     injection_components_ = absl::make_unique<InjectionComponents>();
>   }
>@@ -294,7 +262,6 @@ VideoQualityTest::Params::Params()
>       audio({false, false, false, false}),
>       screenshare{{false, false, 10, 0}, {false, false, 10, 0}},
>       analyzer({"", 0.0, 0.0, 0, "", ""}),
>-      pipe(),
>       config(absl::nullopt),
>       ss{{std::vector<VideoStream>(), 0, 0, -1, InterLayerPredMode::kOn,
>           std::vector<SpatialLayer>()},
>@@ -331,9 +298,7 @@ void VideoQualityTest::CheckParamsAndInjectionComponents() {
>   }
>   if (!params_.config && injection_components_->sender_network == nullptr &&
>       injection_components_->receiver_network == nullptr) {
>-    // TODO(titovartem) replace with default config creation when removing
>-    // pipe.
>-    params_.config = params_.pipe;
>+    params_.config = BuiltInNetworkBehaviorConfig();
>   }
>   RTC_CHECK(
>       (params_.config && injection_components_->sender_network == nullptr &&
>@@ -594,7 +559,10 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
>       return;
>     }
>     video_send_configs_[video_idx].encoder_settings.encoder_factory =
>-        &video_encoder_factory_;
>+        (video_idx == 0) ? &video_encoder_factory_with_analyzer_
>+                         : &video_encoder_factory_;
>+    video_send_configs_[video_idx].encoder_settings.bitrate_allocator_factory =
>+        video_bitrate_allocator_factory_.get();
> 
>     video_send_configs_[video_idx].rtp.payload_name =
>         params_.video[video_idx].codec;
>@@ -702,6 +670,10 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
>         vp9_settings.numberOfSpatialLayers = static_cast<unsigned char>(
>             params_.ss[video_idx].num_spatial_layers);
>         vp9_settings.interLayerPred = params_.ss[video_idx].inter_layer_pred;
>+        // High FPS vp9 screenshare requires flexible mode.
>+        if (params_.video[video_idx].fps > 5) {
>+          vp9_settings.flexibleMode = true;
>+        }
>         video_encoder_configs_[video_idx].encoder_specific_settings =
>             new rtc::RefCountedObject<
>                 VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
>@@ -783,6 +755,8 @@ void VideoQualityTest::SetupThumbnails(Transport* send_transport,
>     // TODO(nisse): Could use a simpler VP8-only encoder factory.
>     thumbnail_send_config.encoder_settings.encoder_factory =
>         &video_encoder_factory_;
>+    thumbnail_send_config.encoder_settings.bitrate_allocator_factory =
>+        video_bitrate_allocator_factory_.get();
>     thumbnail_send_config.rtp.payload_name = params_.video[0].codec;
>     thumbnail_send_config.rtp.payload_type = kPayloadTypeVP8;
>     thumbnail_send_config.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>@@ -1043,7 +1017,6 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>   std::unique_ptr<test::LayerFilteringTransport> send_transport;
>   std::unique_ptr<test::DirectTransport> recv_transport;
>   FILE* graph_data_output_file = nullptr;
>-  std::unique_ptr<VideoAnalyzer> analyzer;
> 
>   params_ = params;
>   // TODO(ivica): Merge with RunWithRenderer and use a flag / argument to
>@@ -1100,7 +1073,7 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>   if (graph_title.empty())
>     graph_title = VideoQualityTest::GenerateGraphTitle();
>   bool is_quick_test_enabled = field_trial::IsEnabled("WebRTC-QuickPerfTest");
>-  analyzer = absl::make_unique<VideoAnalyzer>(
>+  analyzer_ = absl::make_unique<VideoAnalyzer>(
>       send_transport.get(), params_.analyzer.test_label,
>       params_.analyzer.avg_psnr_threshold, params_.analyzer.avg_ssim_threshold,
>       is_quick_test_enabled
>@@ -1114,26 +1087,23 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>       is_quick_test_enabled, clock_, params_.logging.rtp_dump_name);
> 
>   task_queue_.SendTask([&]() {
>-    analyzer->SetCall(sender_call_.get());
>-    analyzer->SetReceiver(receiver_call_->Receiver());
>-    send_transport->SetReceiver(analyzer.get());
>+    analyzer_->SetCall(sender_call_.get());
>+    analyzer_->SetReceiver(receiver_call_->Receiver());
>+    send_transport->SetReceiver(analyzer_.get());
>     recv_transport->SetReceiver(sender_call_->Receiver());
> 
>-    SetupVideo(analyzer.get(), recv_transport.get());
>-    SetupThumbnails(analyzer.get(), recv_transport.get());
>+    SetupVideo(analyzer_.get(), recv_transport.get());
>+    SetupThumbnails(analyzer_.get(), recv_transport.get());
>     video_receive_configs_[params_.ss[0].selected_stream].renderer =
>-        analyzer.get();
>-    GetVideoSendConfig()->pre_encode_callback = analyzer->pre_encode_proxy();
>-    RTC_DCHECK(!GetVideoSendConfig()->post_encode_callback);
>-    GetVideoSendConfig()->post_encode_callback = analyzer.get();
>+        analyzer_.get();
> 
>     CreateFlexfecStreams();
>     CreateVideoStreams();
>-    analyzer->SetSendStream(video_send_streams_[0]);
>+    analyzer_->SetSendStream(video_send_streams_[0]);
>     if (video_receive_streams_.size() == 1)
>-      analyzer->SetReceiveStream(video_receive_streams_[0]);
>+      analyzer_->SetReceiveStream(video_receive_streams_[0]);
> 
>-    GetVideoSendStream()->SetSource(analyzer->OutputInterface(),
>+    GetVideoSendStream()->SetSource(analyzer_->OutputInterface(),
>                                     degradation_preference_);
>     SetupThumbnailCapturers(params_.call.num_thumbnails);
>     for (size_t i = 0; i < thumbnail_send_streams_.size(); ++i) {
>@@ -1143,7 +1113,8 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
> 
>     CreateCapturers();
> 
>-    analyzer->SetSource(video_capturers_[0].get(), params_.ss[0].infer_streams);
>+    analyzer_->SetSource(video_capturers_[0].get(),
>+                         params_.ss[0].infer_streams);
> 
>     for (size_t video_idx = 1; video_idx < num_video_streams_; ++video_idx) {
>       video_send_streams_[video_idx]->SetSource(
>@@ -1153,16 +1124,16 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>     if (params_.audio.enabled) {
>       SetupAudio(send_transport.get());
>       StartAudioStreams();
>-      analyzer->SetAudioReceiveStream(audio_receive_streams_[0]);
>+      analyzer_->SetAudioReceiveStream(audio_receive_streams_[0]);
>     }
>     StartVideoStreams();
>     StartThumbnails();
>-    analyzer->StartMeasuringCpuProcessTime();
>+    analyzer_->StartMeasuringCpuProcessTime();
>     StartVideoCapture();
>     StartThumbnailCapture();
>   });
> 
>-  analyzer->Wait();
>+  analyzer_->Wait();
> 
>   task_queue_.SendTask([&]() {
>     StopThumbnailCapture();
>@@ -1181,6 +1152,7 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
> 
>     DestroyCalls();
>   });
>+  analyzer_ = nullptr;
> }
> 
> rtc::scoped_refptr<AudioDeviceModule> VideoQualityTest::CreateAudioDevice() {
>@@ -1239,7 +1211,8 @@ void VideoQualityTest::InitializeAudioDevice(Call::Config* send_call_config,
> }
> 
> void VideoQualityTest::SetupAudio(Transport* transport) {
>-  AudioSendStream::Config audio_send_config(transport);
>+  AudioSendStream::Config audio_send_config(transport,
>+                                            /*media_transport=*/nullptr);
>   audio_send_config.rtp.ssrc = kAudioSendSsrc;
> 
>   // Add extension to enable audio send side BWE, and allow audio bit rate
>@@ -1278,7 +1251,28 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
>   std::unique_ptr<test::DirectTransport> recv_transport;
>   std::unique_ptr<test::VideoRenderer> local_preview;
>   std::vector<std::unique_ptr<test::VideoRenderer>> loopback_renderers;
>-  RtcEventLogNullImpl null_event_log;
>+
>+  if (!params.logging.rtc_event_log_name.empty()) {
>+    send_event_log_ = RtcEventLog::Create(RtcEventLog::EncodingType::Legacy);
>+    recv_event_log_ = RtcEventLog::Create(RtcEventLog::EncodingType::Legacy);
>+    std::unique_ptr<RtcEventLogOutputFile> send_output(
>+        absl::make_unique<RtcEventLogOutputFile>(
>+            params.logging.rtc_event_log_name + "_send",
>+            RtcEventLog::kUnlimitedOutput));
>+    std::unique_ptr<RtcEventLogOutputFile> recv_output(
>+        absl::make_unique<RtcEventLogOutputFile>(
>+            params.logging.rtc_event_log_name + "_recv",
>+            RtcEventLog::kUnlimitedOutput));
>+    bool event_log_started =
>+        send_event_log_->StartLogging(std::move(send_output),
>+                                      /*output_period_ms=*/5000) &&
>+        recv_event_log_->StartLogging(std::move(recv_output),
>+                                      /*output_period_ms=*/5000);
>+    RTC_DCHECK(event_log_started);
>+  } else {
>+    send_event_log_ = RtcEventLog::CreateNull();
>+    recv_event_log_ = RtcEventLog::CreateNull();
>+  }
> 
>   task_queue_.SendTask([&]() {
>     params_ = params;
>@@ -1286,9 +1280,9 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
> 
>     // TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
>     // match the full stack tests.
>-    Call::Config send_call_config(&null_event_log);
>+    Call::Config send_call_config(send_event_log_.get());
>     send_call_config.bitrate_config = params_.call.call_bitrate_config;
>-    Call::Config recv_call_config(&null_event_log);
>+    Call::Config recv_call_config(recv_event_log_.get());
> 
>     if (params_.audio.enabled)
>       InitializeAudioDevice(
>@@ -1364,7 +1358,7 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
>     Start();
>   });
> 
>-  test::PressEnterToContinue();
>+  test::PressEnterToContinue(task_queue_);
> 
>   task_queue_.SendTask([&]() {
>     Stop();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.h
>index 7ec7e8da7e44b627ffba871124a01c558de72c7b..f67e65bc13797428c0cc7cc61ca678a526870ebc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.h
>@@ -17,12 +17,14 @@
> 
> #include "api/fec_controller.h"
> #include "api/test/video_quality_test_fixture.h"
>+#include "api/video/video_bitrate_allocator_factory.h"
> #include "call/fake_network_pipe.h"
> #include "media/engine/internaldecoderfactory.h"
> #include "media/engine/internalencoderfactory.h"
> #include "test/call_test.h"
> #include "test/frame_generator.h"
> #include "test/layer_filtering_transport.h"
>+#include "video/video_analyzer.h"
> #ifdef WEBRTC_WIN
> #include "modules/audio_device/win/core_audio_utility_win.h"
> #endif
>@@ -75,8 +77,8 @@ class VideoQualityTest :
>   void SetupThumbnailCapturers(size_t num_thumbnail_streams);
>   std::unique_ptr<VideoDecoder> CreateVideoDecoder(
>       const SdpVideoFormat& format);
>-  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>-      const SdpVideoFormat& format);
>+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(const SdpVideoFormat& format,
>+                                                   VideoAnalyzer* analyzer);
>   void SetupVideo(Transport* send_transport, Transport* recv_transport);
>   void SetupThumbnails(Transport* send_transport, Transport* recv_transport);
>   void StartAudioStreams();
>@@ -103,6 +105,9 @@ class VideoQualityTest :
>   test::FunctionVideoDecoderFactory video_decoder_factory_;
>   InternalDecoderFactory internal_decoder_factory_;
>   test::FunctionVideoEncoderFactory video_encoder_factory_;
>+  test::FunctionVideoEncoderFactory video_encoder_factory_with_analyzer_;
>+  std::unique_ptr<VideoBitrateAllocatorFactory>
>+      video_bitrate_allocator_factory_;
>   InternalEncoderFactory internal_encoder_factory_;
>   std::vector<VideoSendStream::Config> thumbnail_send_configs_;
>   std::vector<VideoEncoderConfig> thumbnail_encoder_configs_;
>@@ -116,6 +121,9 @@ class VideoQualityTest :
>   Params params_;
>   std::unique_ptr<InjectionComponents> injection_components_;
> 
>+  // Set non-null when running with analyzer.
>+  std::unique_ptr<VideoAnalyzer> analyzer_;
>+
>   // Note: not same as similarly named member in CallTest. This is the number of
>   // separate send streams, the one in CallTest is the number of substreams for
>   // a single send stream.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.cc
>index 86b93686ca5f9733920456d9b7189eb380fe45d5..bba33aa363e8a695dac6cf4480cbdc4777615300 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.cc
>@@ -21,7 +21,6 @@
> #include "api/video_codecs/video_decoder_factory.h"
> #include "call/rtp_stream_receiver_controller_interface.h"
> #include "call/rtx_receive_stream.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "common_video/h264/profile_level_id.h"
> #include "common_video/include/incoming_video_stream.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>@@ -35,10 +34,13 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/location.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/platform_file.h"
>+#include "rtc_base/strings/string_builder.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/field_trial.h"
> #include "video/call_stats.h"
>+#include "video/frame_dumping_decoder.h"
> #include "video/receive_statistics_proxy.h"
> 
> namespace webrtc {
>@@ -107,6 +109,11 @@ class NullVideoDecoder : public webrtc::VideoDecoder {
>   const char* ImplementationName() const override { return "NullVideoDecoder"; }
> };
> 
>+// TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
>+// Maximum time between frames before resetting the FrameBuffer to avoid RTP
>+// timestamps wraparound to affect FrameBuffer.
>+constexpr int kInactiveStreamThresholdMs = 600000;  //  10 minutes.
>+
> }  // namespace
> 
> namespace internal {
>@@ -128,10 +135,11 @@ VideoReceiveStream::VideoReceiveStream(
>                      "DecodingThread",
>                      rtc::kHighestPriority),
>       call_stats_(call_stats),
>-      rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
>-      timing_(new VCMTiming(clock_)),
>-      video_receiver_(clock_, nullptr, timing_.get(), this, this),
>       stats_proxy_(&config_, clock_),
>+      rtp_receive_statistics_(
>+          ReceiveStatistics::Create(clock_, &stats_proxy_, &stats_proxy_)),
>+      timing_(new VCMTiming(clock_)),
>+      video_receiver_(clock_, timing_.get(), this, this),
>       rtp_video_stream_receiver_(&transport_adapter_,
>                                  call_stats,
>                                  packet_router,
>@@ -139,9 +147,10 @@ VideoReceiveStream::VideoReceiveStream(
>                                  rtp_receive_statistics_.get(),
>                                  &stats_proxy_,
>                                  process_thread_,
>-                                 this,   // NackSender
>-                                 this,   // KeyFrameRequestSender
>-                                 this),  // OnCompleteFrameCallback
>+                                 this,  // NackSender
>+                                 this,  // KeyFrameRequestSender
>+                                 this,  // OnCompleteFrameCallback
>+                                 config_.frame_decryptor),
>       rtp_stream_sync_(this) {
>   RTC_LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
> 
>@@ -245,6 +254,18 @@ void VideoReceiveStream::Start() {
>     if (!video_decoder) {
>       video_decoder = absl::make_unique<NullVideoDecoder>();
>     }
>+
>+    std::string decoded_output_file =
>+        field_trial::FindFullName("WebRTC-DecoderDataDumpDirectory");
>+    if (!decoded_output_file.empty()) {
>+      char filename_buffer[256];
>+      rtc::SimpleStringBuilder ssb(filename_buffer);
>+      ssb << decoded_output_file << "/webrtc_receive_stream_"
>+          << this->config_.rtp.local_ssrc << ".ivf";
>+      video_decoder = absl::make_unique<FrameDumpingDecoder>(
>+          std::move(video_decoder), rtc::CreatePlatformFile(ssb.str()));
>+    }
>+
>     video_decoders_.push_back(std::move(video_decoder));
> 
>     video_receiver_.RegisterExternalDecoder(video_decoders_.back().get(),
>@@ -357,6 +378,14 @@ void VideoReceiveStream::OnCompleteFrame(
>   // partially enabled inter-layer prediction.
>   frame->id.spatial_layer = 0;
> 
>+  // TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
>+  int64_t time_now_ms = rtc::TimeMillis();
>+  if (last_complete_frame_time_ms_ > 0 &&
>+      time_now_ms - last_complete_frame_time_ms_ > kInactiveStreamThresholdMs) {
>+    frame_buffer_->Clear();
>+  }
>+  last_complete_frame_time_ms_ = time_now_ms;
>+
>   int64_t last_continuous_pid = frame_buffer_->InsertFrame(std::move(frame));
>   if (last_continuous_pid != -1)
>     rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.h
>index ae5ced51c3b7b19fea184326b0912df97aba088e..5c9fd83c62a9ed1153bb20e5108d4eb3b3b2ba85 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.h
>@@ -17,7 +17,6 @@
> #include "call/rtp_packet_sink_interface.h"
> #include "call/syncable.h"
> #include "call/video_receive_stream.h"
>-#include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "modules/rtp_rtcp/include/flexfec_receiver.h"
> #include "modules/video_coding/frame_buffer2.h"
> #include "modules/video_coding/video_coding_impl.h"
>@@ -115,6 +114,7 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
> 
>   CallStats* const call_stats_;
> 
>+  ReceiveStatisticsProxy stats_proxy_;
>   // Shared by media and rtx stream receivers, since the latter has no RtpRtcp
>   // module of its own.
>   const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
>@@ -122,7 +122,6 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
>   std::unique_ptr<VCMTiming> timing_;  // Jitter buffer experiment.
>   vcm::VideoReceiver video_receiver_;
>   std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
>-  ReceiveStatisticsProxy stats_proxy_;
>   RtpVideoStreamReceiver rtp_video_stream_receiver_;
>   std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
>   RtpStreamsSynchronizer rtp_stream_sync_;
>@@ -147,6 +146,7 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
>   bool frame_decoded_ = false;
> 
>   int64_t last_keyframe_request_ms_ = 0;
>+  int64_t last_complete_frame_time_ms_ = 0;
> };
> }  // namespace internal
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream_unittest.cc
>index e4ba69b7b92be4bef077dcb6c9b24c7acef0f67a..0ec9e1773384d3e72b661ce468ae5794d56a774b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream_unittest.cc
>@@ -118,7 +118,7 @@ TEST_F(VideoReceiveStreamTest, CreateFrameFromH264FmtpSpropAndIdr) {
>   rtppacket.SetPayloadType(99);
>   rtppacket.SetSequenceNumber(1);
>   rtppacket.SetTimestamp(0);
>-  rtc::Event init_decode_event_(false, false);
>+  rtc::Event init_decode_event_;
>   EXPECT_CALL(mock_h264_video_decoder_, InitDecode(_, _))
>       .WillOnce(Invoke([&init_decode_event_](const VideoCodec* config,
>                                              int32_t number_of_cores) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.cc
>index 054d27e421c00179d2cfad96cf018a2a8cfa11ce..059a6150af9b9ce3ed4f879ac4c97abf41223c67 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.cc
>@@ -15,12 +15,15 @@
> #include "modules/rtp_rtcp/source/rtp_header_extension_size.h"
> #include "modules/rtp_rtcp/source/rtp_sender.h"
> #include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
> #include "video/video_send_stream_impl.h"
> 
> namespace webrtc {
> 
> namespace {
> 
>+constexpr char kTargetBitrateRtcpFieldTrial[] = "WebRTC-Target-Bitrate-Rtcp";
>+
> size_t CalculateMaxHeaderSize(const RtpConfig& config) {
>   size_t header_size = kRtpHeaderSize;
>   size_t extensions_size = 0;
>@@ -62,7 +65,7 @@ VideoSendStream::VideoSendStream(
>     rtc::TaskQueue* worker_queue,
>     CallStats* call_stats,
>     RtpTransportControllerSendInterface* transport,
>-    BitrateAllocator* bitrate_allocator,
>+    BitrateAllocatorInterface* bitrate_allocator,
>     SendDelayStats* send_delay_stats,
>     RtcEventLog* event_log,
>     VideoSendStream::Config config,
>@@ -71,13 +74,13 @@ VideoSendStream::VideoSendStream(
>     const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
>     std::unique_ptr<FecController> fec_controller)
>     : worker_queue_(worker_queue),
>-      thread_sync_event_(false /* manual_reset */, false),
>       stats_proxy_(Clock::GetRealTimeClock(),
>                    config,
>                    encoder_config.content_type),
>       config_(std::move(config)),
>       content_type_(encoder_config.content_type) {
>   RTC_DCHECK(config_.encoder_settings.encoder_factory);
>+  RTC_DCHECK(config_.encoder_settings.bitrate_allocator_factory);
> 
>   video_stream_encoder_ = CreateVideoStreamEncoder(num_cpu_cores, &stats_proxy_,
>                                                    config_.encoder_settings,
>@@ -104,9 +107,10 @@ VideoSendStream::VideoSendStream(
>   // it was created on.
>   thread_sync_event_.Wait(rtc::Event::kForever);
>   send_stream_->RegisterProcessThread(module_process_thread);
>-  // TODO(sprang): Enable this also for regular video calls if it works well.
>-  if (encoder_config.content_type == VideoEncoderConfig::ContentType::kScreen) {
>-    // Only signal target bitrate for screenshare streams, for now.
>+  // TODO(sprang): Enable this also for regular video calls by default, if it
>+  // works well.
>+  if (encoder_config.content_type == VideoEncoderConfig::ContentType::kScreen ||
>+      field_trial::IsEnabled(kTargetBitrateRtcpFieldTrial)) {
>     video_stream_encoder_->SetBitrateAllocationObserver(send_stream_.get());
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.h
>index 0945203df31b5a579819a1b1d5bb3252b57577dd..22aa4b1c6d5dbd36275efa7a61af95181d638a1e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.h
>@@ -20,7 +20,6 @@
> #include "call/bitrate_allocator.h"
> #include "call/video_receive_stream.h"
> #include "call/video_send_stream.h"
>-#include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/event.h"
> #include "rtc_base/task_queue.h"
>@@ -60,7 +59,7 @@ class VideoSendStream : public webrtc::VideoSendStream {
>       rtc::TaskQueue* worker_queue,
>       CallStats* call_stats,
>       RtpTransportControllerSendInterface* transport,
>-      BitrateAllocator* bitrate_allocator,
>+      BitrateAllocatorInterface* bitrate_allocator,
>       SendDelayStats* send_delay_stats,
>       RtcEventLog* event_log,
>       VideoSendStream::Config config,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.cc
>index 58caf4777462d7595ca00af6c3f58a0e9741457b..5dcf18243d5f876f01206750464b197fb3c2d149 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.cc
>@@ -111,6 +111,14 @@ int CalculateMaxPadBitrateBps(const std::vector<VideoStream>& streams,
>   return pad_up_to_bitrate_bps;
> }
> 
>+RtpSenderFrameEncryptionConfig CreateFrameEncryptionConfig(
>+    const VideoSendStream::Config* config) {
>+  RtpSenderFrameEncryptionConfig frame_encryption_config;
>+  frame_encryption_config.frame_encryptor = config->frame_encryptor;
>+  frame_encryption_config.crypto_options = config->crypto_options;
>+  return frame_encryption_config;
>+}
>+
> RtpSenderObservers CreateObservers(CallStats* call_stats,
>                                    EncoderRtcpFeedback* encoder_feedback,
>                                    SendStatisticsProxy* stats_proxy,
>@@ -151,6 +159,16 @@ bool SameStreamsEnabled(const VideoBitrateAllocation& lhs,
> }
> }  // namespace
> 
>+PacingConfig::PacingConfig()
>+    : pacing_factor("factor", PacedSender::kDefaultPaceMultiplier),
>+      max_pacing_delay("max_delay",
>+                       TimeDelta::ms(PacedSender::kMaxQueueLengthMs)) {
>+  ParseFieldTrial({&pacing_factor, &max_pacing_delay},
>+                  field_trial::FindFullName("WebRTC-Video-Pacing"));
>+}
>+PacingConfig::PacingConfig(const PacingConfig&) = default;
>+PacingConfig::~PacingConfig() = default;
>+
> // CheckEncoderActivityTask is used for tracking when the encoder last produced
> // and encoded video frame. If the encoder has not produced anything the last
> // kEncoderTimeOutMs we also want to stop sending padding.
>@@ -221,6 +239,7 @@ VideoSendStreamImpl::VideoSendStreamImpl(
>     std::unique_ptr<FecController> fec_controller)
>     : has_alr_probing_(config->periodic_alr_bandwidth_probing ||
>                        GetAlrSettings(content_type)),
>+      pacing_config_(PacingConfig()),
>       stats_proxy_(stats_proxy),
>       config_(config),
>       worker_queue_(worker_queue),
>@@ -238,19 +257,20 @@ VideoSendStreamImpl::VideoSendStreamImpl(
>                         config_->rtp.ssrcs,
>                         video_stream_encoder),
>       bandwidth_observer_(transport->GetBandwidthObserver()),
>-      rtp_video_sender_(
>-          transport_->CreateRtpVideoSender(config_->rtp.ssrcs,
>-                                           suspended_ssrcs,
>-                                           suspended_payload_states,
>-                                           config_->rtp,
>-                                           config_->rtcp,
>-                                           config_->send_transport,
>-                                           CreateObservers(call_stats,
>-                                                           &encoder_feedback_,
>-                                                           stats_proxy_,
>-                                                           send_delay_stats),
>-                                           event_log,
>-                                           std::move(fec_controller))),
>+      rtp_video_sender_(transport_->CreateRtpVideoSender(
>+          config_->rtp.ssrcs,
>+          suspended_ssrcs,
>+          suspended_payload_states,
>+          config_->rtp,
>+          config_->rtcp_report_interval_ms,
>+          config_->send_transport,
>+          CreateObservers(call_stats,
>+                          &encoder_feedback_,
>+                          stats_proxy_,
>+                          send_delay_stats),
>+          event_log,
>+          std::move(fec_controller),
>+          CreateFrameEncryptionConfig(config_))),
>       weak_ptr_factory_(this) {
>   RTC_DCHECK_RUN_ON(worker_queue_);
>   RTC_LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString();
>@@ -293,9 +313,9 @@ VideoSendStreamImpl::VideoSendStreamImpl(
>       transport->SetQueueTimeLimit(alr_settings->max_paced_queue_time);
>     } else {
>       transport->EnablePeriodicAlrProbing(false);
>-      transport->SetPacingFactor(PacedSender::kDefaultPaceMultiplier);
>-      configured_pacing_factor_ = PacedSender::kDefaultPaceMultiplier;
>-      transport->SetQueueTimeLimit(PacedSender::kMaxQueueLengthMs);
>+      transport->SetPacingFactor(pacing_config_.pacing_factor);
>+      configured_pacing_factor_ = pacing_config_.pacing_factor;
>+      transport->SetQueueTimeLimit(pacing_config_.max_pacing_delay.Get().ms());
>     }
>   }
> 
>@@ -563,17 +583,6 @@ EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage(
>   // Encoded is called on whatever thread the real encoder implementation run
>   // on. In the case of hardware encoders, there might be several encoders
>   // running in parallel on different threads.
>-  const size_t simulcast_idx =
>-      (codec_specific_info->codecType != kVideoCodecVP9)
>-          ? encoded_image.SpatialIndex().value_or(0)
>-          : 0;
>-  if (config_->post_encode_callback) {
>-    // TODO(nisse): Delete webrtc::EncodedFrame class, pass EncodedImage
>-    // instead.
>-    config_->post_encode_callback->EncodedFrameCallback(EncodedFrame(
>-        encoded_image._buffer, encoded_image._length, encoded_image._frameType,
>-        simulcast_idx, encoded_image.Timestamp()));
>-  }
>   {
>     rtc::CritScope lock(&encoder_activity_crit_sect_);
>     if (check_encoder_activity_task_)
>@@ -613,21 +622,22 @@ std::map<uint32_t, RtpPayloadState> VideoSendStreamImpl::GetRtpPayloadStates()
>   return rtp_video_sender_->GetRtpPayloadStates();
> }
> 
>-uint32_t VideoSendStreamImpl::OnBitrateUpdated(uint32_t bitrate_bps,
>-                                               uint8_t fraction_loss,
>-                                               int64_t rtt,
>-                                               int64_t probing_interval_ms) {
>+uint32_t VideoSendStreamImpl::OnBitrateUpdated(BitrateAllocationUpdate update) {
>   RTC_DCHECK_RUN_ON(worker_queue_);
>   RTC_DCHECK(rtp_video_sender_->IsActive())
>       << "VideoSendStream::Start has not been called.";
> 
>-  rtp_video_sender_->OnBitrateUpdated(bitrate_bps, fraction_loss, rtt,
>-                                      stats_proxy_->GetSendFrameRate());
>+  rtp_video_sender_->OnBitrateUpdated(
>+      update.target_bitrate.bps(),
>+      rtc::dchecked_cast<uint8_t>(update.packet_loss_ratio * 256),
>+      update.round_trip_time.ms(), stats_proxy_->GetSendFrameRate());
>   encoder_target_rate_bps_ = rtp_video_sender_->GetPayloadBitrateBps();
>   encoder_target_rate_bps_ =
>       std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
>-  video_stream_encoder_->OnBitrateUpdated(encoder_target_rate_bps_,
>-                                          fraction_loss, rtt);
>+  video_stream_encoder_->OnBitrateUpdated(
>+      encoder_target_rate_bps_,
>+      rtc::dchecked_cast<uint8_t>(update.packet_loss_ratio * 256),
>+      update.round_trip_time.ms());
>   stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_);
>   return rtp_video_sender_->GetProtectionBitrateBps();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.h
>index eed9f567fb08186fa851dd29c42c54170ba33993..9caabacdab92448d13677caf910517f1b2affeb0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.h
>@@ -18,7 +18,6 @@
> #include "api/video/video_stream_encoder_interface.h"
> #include "call/bitrate_allocator.h"
> #include "call/rtp_video_sender_interface.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/utility/include/process_thread.h"
> #include "rtc_base/weak_ptr.h"
> #include "video/call_stats.h"
>@@ -30,6 +29,16 @@
> namespace webrtc {
> namespace internal {
> 
>+// Pacing buffer config; overridden by ALR config if provided.
>+struct PacingConfig {
>+  PacingConfig();
>+  PacingConfig(const PacingConfig&);
>+  PacingConfig& operator=(const PacingConfig&) = default;
>+  ~PacingConfig();
>+  FieldTrialParameter<double> pacing_factor;
>+  FieldTrialParameter<TimeDelta> max_pacing_delay;
>+};
>+
> // VideoSendStreamImpl implements internal::VideoSendStream.
> // It is created and destroyed on |worker_queue|. The intent is to decrease the
> // need for locking and to ensure methods are called in sequence.
>@@ -83,10 +92,7 @@ class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver,
>   class CheckEncoderActivityTask;
> 
>   // Implements BitrateAllocatorObserver.
>-  uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
>-                            uint8_t fraction_loss,
>-                            int64_t rtt,
>-                            int64_t probing_interval_ms) override;
>+  uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override;
> 
>   void OnEncoderConfigurationChanged(std::vector<VideoStream> streams,
>                                      int min_transmit_bitrate_bps) override;
>@@ -115,6 +121,7 @@ class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver,
>   void SignalEncoderActive();
> 
>   const bool has_alr_probing_;
>+  const PacingConfig pacing_config_;
> 
>   SendStatisticsProxy* const stats_proxy_;
>   const VideoSendStream::Config* const config_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl_unittest.cc
>index dd036909a6af1ec56798d48b958a169d0b7cdb18..32d0df1c2e61b7dc5efcdae83f63a41881071371 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl_unittest.cc
>@@ -69,6 +69,14 @@ class MockRtpVideoSender : public RtpVideoSenderInterface {
>   MOCK_CONST_METHOD0(GetProtectionBitrateBps, uint32_t());
>   MOCK_METHOD3(SetEncodingData, void(size_t, size_t, size_t));
> };
>+
>+BitrateAllocationUpdate CreateAllocation(int bitrate_bps) {
>+  BitrateAllocationUpdate update;
>+  update.target_bitrate = DataRate::bps(bitrate_bps);
>+  update.packet_loss_ratio = 0;
>+  update.round_trip_time = TimeDelta::Zero();
>+  return update;
>+}
> }  // namespace
> 
> class VideoSendStreamImplTest : public ::testing::Test {
>@@ -91,7 +99,7 @@ class VideoSendStreamImplTest : public ::testing::Test {
>     EXPECT_CALL(transport_controller_, packet_router())
>         .WillRepeatedly(Return(&packet_router_));
>     EXPECT_CALL(transport_controller_,
>-                CreateRtpVideoSender(_, _, _, _, _, _, _, _, _))
>+                CreateRtpVideoSender(_, _, _, _, _, _, _, _, _, _))
>         .WillRepeatedly(Return(&rtp_video_sender_));
>     EXPECT_CALL(rtp_video_sender_, SetActive(_))
>         .WillRepeatedly(testing::Invoke(
>@@ -367,7 +375,7 @@ TEST_F(VideoSendStreamImplTest, ForwardsVideoBitrateAllocationWhenEnabled) {
>         .Times(1)
>         .WillOnce(Return(kBitrateBps));
>     static_cast<BitrateAllocatorObserver*>(vss_impl.get())
>-        ->OnBitrateUpdated(kBitrateBps, 0, 0, 0);
>+        ->OnBitrateUpdated(CreateAllocation(kBitrateBps));
>     EXPECT_CALL(rtp_video_sender_, OnBitrateAllocationUpdated(alloc)).Times(1);
>     observer->OnBitrateAllocationUpdated(alloc);
> 
>@@ -376,7 +384,7 @@ TEST_F(VideoSendStreamImplTest, ForwardsVideoBitrateAllocationWhenEnabled) {
>         .Times(1)
>         .WillOnce(Return(0));
>     static_cast<BitrateAllocatorObserver*>(vss_impl.get())
>-        ->OnBitrateUpdated(0, 0, 0, 0);
>+        ->OnBitrateUpdated(CreateAllocation(0));
>     EXPECT_CALL(rtp_video_sender_, OnBitrateAllocationUpdated(alloc)).Times(0);
>     observer->OnBitrateAllocationUpdated(alloc);
> 
>@@ -396,7 +404,7 @@ TEST_F(VideoSendStreamImplTest, ThrottlesVideoBitrateAllocationWhenTooSimilar) {
>         .Times(1)
>         .WillOnce(Return(kBitrateBps));
>     static_cast<BitrateAllocatorObserver*>(vss_impl.get())
>-        ->OnBitrateUpdated(kBitrateBps, 0, 0, 0);
>+        ->OnBitrateUpdated(CreateAllocation(kBitrateBps));
>     VideoBitrateAllocationObserver* const observer =
>         static_cast<VideoBitrateAllocationObserver*>(vss_impl.get());
> 
>@@ -450,7 +458,7 @@ TEST_F(VideoSendStreamImplTest, ForwardsVideoBitrateAllocationOnLayerChange) {
>         .Times(1)
>         .WillOnce(Return(kBitrateBps));
>     static_cast<BitrateAllocatorObserver*>(vss_impl.get())
>-        ->OnBitrateUpdated(kBitrateBps, 0, 0, 0);
>+        ->OnBitrateUpdated(CreateAllocation(kBitrateBps));
>     VideoBitrateAllocationObserver* const observer =
>         static_cast<VideoBitrateAllocationObserver*>(vss_impl.get());
> 
>@@ -494,7 +502,7 @@ TEST_F(VideoSendStreamImplTest, ForwardsVideoBitrateAllocationAfterTimeout) {
>         .Times(1)
>         .WillRepeatedly(Return(kBitrateBps));
>     static_cast<BitrateAllocatorObserver*>(vss_impl.get())
>-        ->OnBitrateUpdated(kBitrateBps, 0, 0, 0);
>+        ->OnBitrateUpdated(CreateAllocation(kBitrateBps));
>     VideoBitrateAllocationObserver* const observer =
>         static_cast<VideoBitrateAllocationObserver*>(vss_impl.get());
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_tests.cc
>index 09ccf1c49893a53cf3bc739ddffb70fc2175ac99..ea77cd65291493795c9d21ea4c27801f33d8955f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_tests.cc
>@@ -12,12 +12,11 @@
> #include <vector>
> 
> #include "api/test/simulated_network.h"
>+#include "api/video/encoded_image.h"
> #include "call/call.h"
> #include "call/fake_network_pipe.h"
> #include "call/rtp_transport_controller_send.h"
> #include "call/simulated_network.h"
>-#include "common_video/include/frame_callback.h"
>-#include "common_video/include/video_frame.h"
> #include "modules/rtp_rtcp/include/rtp_header_parser.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "modules/rtp_rtcp/source/rtcp_sender.h"
>@@ -70,10 +69,14 @@ class VideoSendStreamPeer {
> };
> }  // namespace test
> 
>+namespace {
>+constexpr int64_t kRtcpIntervalMs = 1000;
>+
> enum VideoFormat {
>   kGeneric,
>   kVP8,
> };
>+}  // namespace
> 
> void ExpectEqualFramesVector(const std::vector<VideoFrame>& frames1,
>                              const std::vector<VideoFrame>& frames2);
>@@ -541,7 +544,7 @@ class UlpfecObserver : public test::EndToEndTest {
>     // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
>     // Configure some network delay.
>     const int kNetworkDelayMs = 100;
>-    DefaultNetworkSimulationConfig config;
>+    BuiltInNetworkBehaviorConfig config;
>     config.loss_percent = 5;
>     config.queue_delay_ms = kNetworkDelayMs;
>     return new test::PacketTransport(
>@@ -655,7 +658,7 @@ TEST_P(VideoSendStreamTest, DISABLED_DoesUtilizeUlpfecForVp8WithNackEnabled) {
>   RunBaseTest(&test);
> }
> 
>-#if !defined(RTC_DISABLE_VP9)
>+#if defined(RTC_ENABLE_VP9)
> // Disabled as flaky, see https://crbug.com/webrtc/7285 for details.
> TEST_P(VideoSendStreamTest, DISABLED_DoesUtilizeUlpfecForVp9WithNackEnabled) {
>   test::FunctionVideoEncoderFactory encoder_factory(
>@@ -663,7 +666,7 @@ TEST_P(VideoSendStreamTest, DISABLED_DoesUtilizeUlpfecForVp9WithNackEnabled) {
>   UlpfecObserver test(false, true, true, true, "VP9", &encoder_factory);
>   RunBaseTest(&test);
> }
>-#endif  // !defined(RTC_DISABLE_VP9)
>+#endif  // defined(RTC_ENABLE_VP9)
> 
> TEST_P(VideoSendStreamTest, SupportsUlpfecWithMultithreadedH264) {
>   test::FunctionVideoEncoderFactory encoder_factory([]() {
>@@ -731,7 +734,7 @@ class FlexfecObserver : public test::EndToEndTest {
>     // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
>     // Therefore we need some network delay.
>     const int kNetworkDelayMs = 100;
>-    DefaultNetworkSimulationConfig config;
>+    BuiltInNetworkBehaviorConfig config;
>     config.loss_percent = 5;
>     config.queue_delay_ms = kNetworkDelayMs;
>     return new test::PacketTransport(
>@@ -806,7 +809,7 @@ TEST_P(VideoSendStreamTest, SupportsFlexfecWithRtpExtensionsVp8) {
>   RunBaseTest(&test);
> }
> 
>-#if !defined(RTC_DISABLE_VP9)
>+#if defined(RTC_ENABLE_VP9)
> TEST_P(VideoSendStreamTest, SupportsFlexfecVp9) {
>   test::FunctionVideoEncoderFactory encoder_factory(
>       []() { return VP9Encoder::Create(); });
>@@ -820,7 +823,7 @@ TEST_P(VideoSendStreamTest, SupportsFlexfecWithNackVp9) {
>   FlexfecObserver test(false, true, "VP9", &encoder_factory, 1);
>   RunBaseTest(&test);
> }
>-#endif  // defined(RTC_DISABLE_VP9)
>+#endif  // defined(RTC_ENABLE_VP9)
> 
> TEST_P(VideoSendStreamTest, SupportsFlexfecH264) {
>   test::FunctionVideoEncoderFactory encoder_factory([]() {
>@@ -866,19 +869,27 @@ void VideoSendStreamTest::TestNackRetransmission(
>       RTPHeader header;
>       EXPECT_TRUE(parser_->Parse(packet, length, &header));
> 
>-      int kRetransmitTarget = 6;
>+      // NACK packets two times at some arbitrary points.
>+      const int kNackedPacketsAtOnceCount = 3;
>+      const int kRetransmitTarget = kNackedPacketsAtOnceCount * 2;
>+
>+      // Skip padding packets because they will never be retransmitted.
>+      if (header.paddingLength + header.headerLength == length) {
>+        return SEND_PACKET;
>+      }
>+
>       ++send_count_;
>+
>+      // NACK packets at arbitrary points.
>       if (send_count_ == 5 || send_count_ == 25) {
>-        nacked_sequence_numbers_.push_back(
>-            static_cast<uint16_t>(header.sequenceNumber - 3));
>-        nacked_sequence_numbers_.push_back(
>-            static_cast<uint16_t>(header.sequenceNumber - 2));
>-        nacked_sequence_numbers_.push_back(
>-            static_cast<uint16_t>(header.sequenceNumber - 1));
>+        nacked_sequence_numbers_.insert(
>+            nacked_sequence_numbers_.end(),
>+            non_padding_sequence_numbers_.end() - kNackedPacketsAtOnceCount,
>+            non_padding_sequence_numbers_.end());
> 
>         RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), nullptr,
>                                nullptr, nullptr, transport_adapter_.get(),
>-                               RtcpIntervalConfig{});
>+                               kRtcpIntervalMs);
> 
>         rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
>         rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
>@@ -892,7 +903,6 @@ void VideoSendStreamTest::TestNackRetransmission(
>       }
> 
>       uint16_t sequence_number = header.sequenceNumber;
>-
>       if (header.ssrc == retransmit_ssrc_ &&
>           retransmit_ssrc_ != kVideoSendSsrcs[0]) {
>         // Not kVideoSendSsrcs[0], assume correct RTX packet. Extract sequence
>@@ -900,6 +910,7 @@ void VideoSendStreamTest::TestNackRetransmission(
>         const uint8_t* rtx_header = packet + header.headerLength;
>         sequence_number = (rtx_header[0] << 8) + rtx_header[1];
>       }
>+
>       auto found = std::find(nacked_sequence_numbers_.begin(),
>                              nacked_sequence_numbers_.end(), sequence_number);
>       if (found != nacked_sequence_numbers_.end()) {
>@@ -910,6 +921,8 @@ void VideoSendStreamTest::TestNackRetransmission(
>           EXPECT_EQ(retransmit_payload_type_, header.payloadType);
>           observation_complete_.Set();
>         }
>+      } else {
>+        non_padding_sequence_numbers_.push_back(sequence_number);
>       }
> 
>       return SEND_PACKET;
>@@ -938,6 +951,7 @@ void VideoSendStreamTest::TestNackRetransmission(
>     uint32_t retransmit_ssrc_;
>     uint8_t retransmit_payload_type_;
>     std::vector<uint16_t> nacked_sequence_numbers_;
>+    std::vector<uint16_t> non_padding_sequence_numbers_;
>   } test(retransmit_ssrc, retransmit_payload_type);
> 
>   RunBaseTest(&test);
>@@ -964,8 +978,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
> 
>   // Observer that verifies that the expected number of packets and bytes
>   // arrive for each frame size, from start_size to stop_size.
>-  class FrameFragmentationTest : public test::SendTest,
>-                                 public EncodedFrameObserver {
>+  class FrameFragmentationTest : public test::SendTest {
>    public:
>     FrameFragmentationTest(size_t max_packet_size,
>                            size_t start_size,
>@@ -1090,7 +1103,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
>             loss_ratio);    // Loss percent.
>         RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
>                                &lossy_receive_stats, nullptr, nullptr,
>-                               transport_adapter_.get(), RtcpIntervalConfig{});
>+                               transport_adapter_.get(), kRtcpIntervalMs);
> 
>         rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
>         rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
>@@ -1101,7 +1114,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
>       }
>     }
> 
>-    void EncodedFrameCallback(const EncodedFrame& encoded_frame) override {
>+    void UpdateConfiguration() {
>       rtc::CritScope lock(&mutex_);
>       // Increase frame size for next encoded frame, in the context of the
>       // encoder thread.
>@@ -1110,9 +1123,10 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
>       }
>       encoder_.SetFrameSize(static_cast<size_t>(current_size_frame_));
>     }
>-    void ModifySenderCallConfig(Call::Config* config) override {
>+    void ModifySenderBitrateConfig(
>+        BitrateConstraints* bitrate_config) override {
>       const int kMinBitrateBps = 30000;
>-      config->bitrate_config.min_bitrate_bps = kMinBitrateBps;
>+      bitrate_config->min_bitrate_bps = kMinBitrateBps;
>     }
> 
>     void ModifyVideoConfigs(
>@@ -1132,38 +1146,15 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
> 
>       send_config->encoder_settings.encoder_factory = &encoder_factory_;
>       send_config->rtp.max_packet_size = kMaxPacketSize;
>-      send_config->post_encode_callback = this;
>+      encoder_.RegisterPostEncodeCallback([this]() { UpdateConfiguration(); });
> 
>       // Make sure there is at least one extension header, to make the RTP
>       // header larger than the base length of 12 bytes.
>       EXPECT_FALSE(send_config->rtp.extensions.empty());
> 
>       // Setup screen content disables frame dropping which makes this easier.
>-      class VideoStreamFactory
>-          : public VideoEncoderConfig::VideoStreamFactoryInterface {
>-       public:
>-        explicit VideoStreamFactory(size_t num_temporal_layers)
>-            : num_temporal_layers_(num_temporal_layers) {
>-          EXPECT_GT(num_temporal_layers, 0u);
>-        }
>-
>-       private:
>-        std::vector<VideoStream> CreateEncoderStreams(
>-            int width,
>-            int height,
>-            const VideoEncoderConfig& encoder_config) override {
>-          std::vector<VideoStream> streams =
>-              test::CreateVideoStreams(width, height, encoder_config);
>-          for (VideoStream& stream : streams) {
>-            stream.num_temporal_layers = num_temporal_layers_;
>-          }
>-          return streams;
>-        }
>-        const size_t num_temporal_layers_;
>-      };
>-
>-      encoder_config->video_stream_factory =
>-          new rtc::RefCountedObject<VideoStreamFactory>(2);
>+      EXPECT_EQ(1u, encoder_config->simulcast_layers.size());
>+      encoder_config->simulcast_layers[0].num_temporal_layers = 2;
>       encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
>     }
> 
>@@ -1347,7 +1338,7 @@ TEST_P(VideoSendStreamTest, SuspendBelowMinBitrate) {
>       FakeReceiveStatistics receive_stats(kVideoSendSsrcs[0],
>                                           last_sequence_number_, rtp_count_, 0);
>       RTCPSender rtcp_sender(false, clock_, &receive_stats, nullptr, nullptr,
>-                             transport_adapter_.get(), RtcpIntervalConfig{});
>+                             transport_adapter_.get(), kRtcpIntervalMs);
> 
>       rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
>       rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
>@@ -1396,12 +1387,20 @@ TEST_P(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
>           header.headerLength + header.paddingLength == length;
> 
>       if (test_state_ == kBeforeStopCapture) {
>+        // Packets are flowing, stop camera.
>         capturer_->Stop();
>         test_state_ = kWaitingForPadding;
>       } else if (test_state_ == kWaitingForPadding && only_padding) {
>+        // We're still getting padding, after stopping camera.
>         test_state_ = kWaitingForNoPackets;
>-      } else if (test_state_ == kWaitingForPaddingAfterCameraRestart &&
>+      } else if (test_state_ == kWaitingForMediaAfterCameraRestart &&
>+                 !only_padding) {
>+        // Media packets are flowing again, stop camera a second time.
>+        capturer_->Stop();
>+        test_state_ = kWaitingForPaddingAfterCameraStopsAgain;
>+      } else if (test_state_ == kWaitingForPaddingAfterCameraStopsAgain &&
>                  only_padding) {
>+        // Padding is still flowing, test ok.
>         observation_complete_.Set();
>       }
>       return SEND_PACKET;
>@@ -1414,13 +1413,20 @@ TEST_P(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
>           (last_packet_time_ms_ > 0 &&
>            clock_->TimeInMilliseconds() - last_packet_time_ms_ >
>                kNoPacketsThresholdMs)) {
>+        // No packets seen for |kNoPacketsThresholdMs|, restart camera.
>         capturer_->Start();
>-        test_state_ = kWaitingForPaddingAfterCameraRestart;
>+        test_state_ = kWaitingForMediaAfterCameraRestart;
>       }
>       return SEND_PACKET;
>     }
> 
>-    size_t GetNumVideoStreams() const override { return 3; }
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      // Make sure padding is sent if encoder is not producing media.
>+      encoder_config->min_transmit_bitrate_bps = 50000;
>+    }
> 
>     void OnFrameGeneratorCapturerCreated(
>         test::FrameGeneratorCapturer* frame_generator_capturer) override {
>@@ -1437,7 +1443,8 @@ TEST_P(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
>       kBeforeStopCapture,
>       kWaitingForPadding,
>       kWaitingForNoPackets,
>-      kWaitingForPaddingAfterCameraRestart
>+      kWaitingForMediaAfterCameraRestart,
>+      kWaitingForPaddingAfterCameraStopsAgain
>     };
> 
>     TestState test_state_ = kBeforeStopCapture;
>@@ -1481,7 +1488,7 @@ TEST_P(VideoSendStreamTest, PaddingIsPrimarilyRetransmissions) {
>         test::SingleThreadedTaskQueueForTesting* task_queue,
>         Call* sender_call) override {
>       const int kNetworkDelayMs = 50;
>-      DefaultNetworkSimulationConfig config;
>+      BuiltInNetworkBehaviorConfig config;
>       config.loss_percent = 10;
>       config.link_capacity_kbps = kCapacityKbps;
>       config.queue_delay_ms = kNetworkDelayMs;
>@@ -1780,7 +1787,6 @@ class MaxPaddingSetTest : public test::SendTest {
> 
>   explicit MaxPaddingSetTest(bool test_switch_content_type, T* stream_reset_fun)
>       : SendTest(test::CallTest::kDefaultTimeoutMs),
>-        content_switch_event_(false, false),
>         call_(nullptr),
>         send_stream_(nullptr),
>         send_stream_config_(nullptr),
>@@ -1910,7 +1916,6 @@ TEST_P(VideoSendStreamTest,
>    public:
>     EncoderObserver()
>         : FakeEncoder(Clock::GetRealTimeClock()),
>-          init_encode_called_(false, false),
>           number_of_initializations_(0),
>           last_initialized_frame_width_(0),
>           last_initialized_frame_height_(0) {}
>@@ -1993,7 +1998,6 @@ TEST_P(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
>    public:
>     StartBitrateObserver()
>         : FakeEncoder(Clock::GetRealTimeClock()),
>-          start_bitrate_changed_(false, false),
>           start_bitrate_kbps_(0) {}
>     int32_t InitEncode(const VideoCodec* config,
>                        int32_t number_of_cores,
>@@ -2068,10 +2072,7 @@ TEST_P(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
> 
> class StartStopBitrateObserver : public test::FakeEncoder {
>  public:
>-  StartStopBitrateObserver()
>-      : FakeEncoder(Clock::GetRealTimeClock()),
>-        encoder_init_(false, false),
>-        bitrate_changed_(false, false) {}
>+  StartStopBitrateObserver() : FakeEncoder(Clock::GetRealTimeClock()) {}
>   int32_t InitEncode(const VideoCodec* config,
>                      int32_t number_of_cores,
>                      size_t max_payload_size) override {
>@@ -2228,8 +2229,6 @@ TEST_P(VideoSendStreamTest, VideoSendStreamUpdateActiveSimulcastLayers) {
> TEST_P(VideoSendStreamTest, CapturesTextureAndVideoFrames) {
>   class FrameObserver : public rtc::VideoSinkInterface<VideoFrame> {
>    public:
>-    FrameObserver() : output_frame_event_(false, false) {}
>-
>     void OnFrame(const VideoFrame& video_frame) override {
>       output_frames_.push_back(video_frame);
>       output_frame_event_.Set();
>@@ -2392,11 +2391,6 @@ TEST_P(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
>       return 0;
>     }
> 
>-    int32_t SetChannelParameters(uint32_t packetLoss, int64_t rtt) override {
>-      EXPECT_TRUE(IsReadyForEncode());
>-      return 0;
>-    }
>-
>     int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) override {
>       EXPECT_TRUE(IsReadyForEncode());
>       return 0;
>@@ -2458,7 +2452,6 @@ TEST_P(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) {
>     VideoCodecConfigObserver()
>         : SendTest(kDefaultTimeoutMs),
>           FakeEncoder(Clock::GetRealTimeClock()),
>-          init_encode_event_(false, false),
>           num_initializations_(0),
>           stream_(nullptr),
>           encoder_factory_(this) {}
>@@ -2530,7 +2523,6 @@ class VideoCodecConfigObserver : public test::SendTest,
>         FakeEncoder(Clock::GetRealTimeClock()),
>         video_codec_type_(video_codec_type),
>         codec_name_(codec_name),
>-        init_encode_event_(false, false),
>         num_initializations_(0),
>         stream_(nullptr),
>         encoder_factory_(this) {
>@@ -2538,26 +2530,6 @@ class VideoCodecConfigObserver : public test::SendTest,
>   }
> 
>  private:
>-  class VideoStreamFactory
>-      : public VideoEncoderConfig::VideoStreamFactoryInterface {
>-   public:
>-    VideoStreamFactory() {}
>-
>-   private:
>-    std::vector<VideoStream> CreateEncoderStreams(
>-        int width,
>-        int height,
>-        const VideoEncoderConfig& encoder_config) override {
>-      std::vector<VideoStream> streams =
>-          test::CreateVideoStreams(width, height, encoder_config);
>-      for (size_t i = 0; i < streams.size(); ++i) {
>-        streams[i].num_temporal_layers =
>-            kVideoCodecConfigObserverNumberOfTemporalLayers;
>-      }
>-      return streams;
>-    }
>-  };
>-
>   void ModifyVideoConfigs(
>       VideoSendStream::Config* send_config,
>       std::vector<VideoReceiveStream::Config>* receive_configs,
>@@ -2567,8 +2539,9 @@ class VideoCodecConfigObserver : public test::SendTest,
> 
>     encoder_config->codec_type = video_codec_type_;
>     encoder_config->encoder_specific_settings = GetEncoderSpecificSettings();
>-    encoder_config->video_stream_factory =
>-        new rtc::RefCountedObject<VideoStreamFactory>();
>+    EXPECT_EQ(1u, encoder_config->simulcast_layers.size());
>+    encoder_config->simulcast_layers[0].num_temporal_layers =
>+        kVideoCodecConfigObserverNumberOfTemporalLayers;
>     encoder_config_ = encoder_config->Copy();
>   }
> 
>@@ -2794,8 +2767,6 @@ TEST_P(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
>         const VideoEncoderConfig& encoder_config) override {
>       std::vector<VideoStream> streams =
>           test::CreateVideoStreams(width, height, encoder_config);
>-      EXPECT_FALSE(streams[0].num_temporal_layers.has_value());
>-      streams[0].num_temporal_layers = 2;
>       RTC_CHECK_GT(streams[0].max_bitrate_bps,
>                    kScreencastMaxTargetBitrateDeltaKbps);
>       streams[0].target_bitrate_bps =
>@@ -2833,6 +2804,8 @@ TEST_P(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
>       EXPECT_EQ(1u, encoder_config->number_of_streams);
>       encoder_config->video_stream_factory =
>           new rtc::RefCountedObject<VideoStreamFactory>();
>+      EXPECT_EQ(1u, encoder_config->simulcast_layers.size());
>+      encoder_config->simulcast_layers[0].num_temporal_layers = 2;
>       encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
>     }
> 
>@@ -2863,8 +2836,6 @@ TEST_P(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
>         : SendTest(kDefaultTimeoutMs),
>           FakeEncoder(Clock::GetRealTimeClock()),
>           task_queue_(task_queue),
>-          init_encode_event_(false, false),
>-          bitrate_changed_event_(false, false),
>           target_bitrate_(0),
>           num_initializations_(0),
>           call_(nullptr),
>@@ -2927,10 +2898,11 @@ TEST_P(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
>       EXPECT_EQ(expected_bitrate, target_bitrate_);
>     }
> 
>-    void ModifySenderCallConfig(Call::Config* config) override {
>-      config->bitrate_config.min_bitrate_bps = kMinBitrateKbps * 1000;
>-      config->bitrate_config.start_bitrate_bps = kStartBitrateKbps * 1000;
>-      config->bitrate_config.max_bitrate_bps = kMaxBitrateKbps * 1000;
>+    void ModifySenderBitrateConfig(
>+        BitrateConstraints* bitrate_config) override {
>+      bitrate_config->min_bitrate_bps = kMinBitrateKbps * 1000;
>+      bitrate_config->start_bitrate_bps = kStartBitrateKbps * 1000;
>+      bitrate_config->max_bitrate_bps = kMaxBitrateKbps * 1000;
>     }
> 
>     class VideoStreamFactory
>@@ -3115,7 +3087,7 @@ TEST_P(VideoSendStreamTest, ReportsSentResolution) {
>   RunBaseTest(&test);
> }
> 
>-#if !defined(RTC_DISABLE_VP9)
>+#if defined(RTC_ENABLE_VP9)
> class Vp9HeaderObserver : public test::SendTest {
>  public:
>   Vp9HeaderObserver()
>@@ -3137,26 +3109,6 @@ class Vp9HeaderObserver : public test::SendTest {
>  private:
>   const int kVp9PayloadType = test::CallTest::kVideoSendPayloadType;
> 
>-  class VideoStreamFactory
>-      : public VideoEncoderConfig::VideoStreamFactoryInterface {
>-   public:
>-    explicit VideoStreamFactory(size_t number_of_temporal_layers)
>-        : number_of_temporal_layers_(number_of_temporal_layers) {}
>-
>-   private:
>-    std::vector<VideoStream> CreateEncoderStreams(
>-        int width,
>-        int height,
>-        const VideoEncoderConfig& encoder_config) override {
>-      std::vector<VideoStream> streams =
>-          test::CreateVideoStreams(width, height, encoder_config);
>-      streams.back().num_temporal_layers = number_of_temporal_layers_;
>-      return streams;
>-    }
>-
>-    const size_t number_of_temporal_layers_;
>-  };
>-
>   void ModifyVideoConfigs(
>       VideoSendStream::Config* send_config,
>       std::vector<VideoReceiveStream::Config>* receive_configs,
>@@ -3168,9 +3120,9 @@ class Vp9HeaderObserver : public test::SendTest {
>     encoder_config->encoder_specific_settings = new rtc::RefCountedObject<
>         VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings_);
>     EXPECT_EQ(1u, encoder_config->number_of_streams);
>-    encoder_config->video_stream_factory =
>-        new rtc::RefCountedObject<VideoStreamFactory>(
>-            vp9_settings_.numberOfTemporalLayers);
>+    EXPECT_EQ(1u, encoder_config->simulcast_layers.size());
>+    encoder_config->simulcast_layers[0].num_temporal_layers =
>+        vp9_settings_.numberOfTemporalLayers;
>     encoder_config_ = encoder_config->Copy();
>   }
> 
>@@ -3628,7 +3580,7 @@ TEST_P(VideoSendStreamTest, MAYBE_Vp9FlexModeRefCount) {
> 
>   RunBaseTest(&test);
> }
>-#endif  // !defined(RTC_DISABLE_VP9)
>+#endif  // defined(RTC_ENABLE_VP9)
> 
> void VideoSendStreamTest::TestRequestSourceRotateVideo(
>     bool support_orientation_ext) {
>@@ -3663,6 +3615,39 @@ TEST_P(VideoSendStreamTest,
>   TestRequestSourceRotateVideo(true);
> }
> 
>+TEST_P(VideoSendStreamTest, EncoderConfigMaxFramerateReportedToSource) {
>+  static const int kMaxFps = 22;
>+  class FpsObserver : public test::SendTest,
>+                      public test::FrameGeneratorCapturer::SinkWantsObserver {
>+   public:
>+    FpsObserver() : SendTest(kDefaultTimeoutMs) {}
>+
>+    void OnFrameGeneratorCapturerCreated(
>+        test::FrameGeneratorCapturer* frame_generator_capturer) override {
>+      frame_generator_capturer->SetSinkWantsObserver(this);
>+    }
>+
>+    void OnSinkWantsChanged(rtc::VideoSinkInterface<VideoFrame>* sink,
>+                            const rtc::VideoSinkWants& wants) override {
>+      if (wants.max_framerate_fps == kMaxFps)
>+        observation_complete_.Set();
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      encoder_config->simulcast_layers[0].max_framerate = kMaxFps;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait()) << "Timed out while waiting for fps to be reported.";
>+    }
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+
> // This test verifies that overhead is removed from the bandwidth estimate by
> // testing that the maximum possible target payload rate is smaller than the
> // maximum bandwidth estimate by the overhead rate.
>@@ -3680,8 +3665,7 @@ TEST_P(VideoSendStreamTest, RemoveOverheadFromBandwidth) {
>           encoder_factory_(this),
>           call_(nullptr),
>           max_bitrate_bps_(0),
>-          first_packet_sent_(false),
>-          bitrate_changed_event_(false, false) {}
>+          first_packet_sent_(false) {}
> 
>     int32_t SetRateAllocation(const VideoBitrateAllocation& bitrate,
>                               uint32_t frameRate) override {
>@@ -3894,7 +3878,6 @@ class ContentSwitchTest : public test::SendTest {
> 
>   explicit ContentSwitchTest(T* stream_reset_fun)
>       : SendTest(test::CallTest::kDefaultTimeoutMs),
>-        content_switch_event_(false, false),
>         call_(nullptr),
>         state_(StreamState::kBeforeSwitch),
>         send_stream_(nullptr),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.cc
>index b7b5a326113f57a3ff14d9db2b01722800d6298f..a22d58adf1eaadf887bc54b387f0e4d62e6cee4e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.cc
>@@ -15,8 +15,9 @@
> #include <numeric>
> #include <utility>
> 
>+#include "api/video/encoded_image.h"
> #include "api/video/i420_buffer.h"
>-#include "common_video/include/video_frame.h"
>+#include "api/video/video_bitrate_allocator_factory.h"
> #include "modules/video_coding/include/video_codec_initializer.h"
> #include "modules/video_coding/include/video_coding.h"
> #include "rtc_base/arraysize.h"
>@@ -363,6 +364,8 @@ VideoStreamEncoder::VideoStreamEncoder(
>       max_framerate_(-1),
>       pending_encoder_reconfiguration_(false),
>       pending_encoder_creation_(false),
>+      crop_width_(0),
>+      crop_height_(0),
>       encoder_start_bitrate_bps_(0),
>       max_data_payload_length_(0),
>       last_observed_bitrate_bps_(0),
>@@ -376,6 +379,7 @@ VideoStreamEncoder::VideoStreamEncoder(
>       last_frame_log_ms_(clock_->TimeInMilliseconds()),
>       captured_frame_count_(0),
>       dropped_frame_count_(0),
>+      pending_frame_post_time_us_(0),
>       bitrate_observer_(nullptr),
>       encoder_queue_("EncoderQueue") {
>   RTC_DCHECK(encoder_stats_observer);
>@@ -531,11 +535,14 @@ void VideoStreamEncoder::ReconfigureEncoder() {
>   crop_height_ = last_frame_info_->height - highest_stream_height;
> 
>   VideoCodec codec;
>-  if (!VideoCodecInitializer::SetupCodec(encoder_config_, streams, &codec,
>-                                         &rate_allocator_)) {
>+  if (!VideoCodecInitializer::SetupCodec(encoder_config_, streams, &codec)) {
>     RTC_LOG(LS_ERROR) << "Failed to create encoder configuration.";
>   }
> 
>+  rate_allocator_ =
>+      settings_.bitrate_allocator_factory->CreateVideoBitrateAllocator(codec);
>+  RTC_CHECK(rate_allocator_) << "Failed to create bitrate allocator.";
>+
>   // Set min_bitrate_bps, max_bitrate_bps, and max padding bit rate for VP9.
>   if (encoder_config_.codec_type == kVideoCodecVP9) {
>     RTC_DCHECK_EQ(1U, streams.size());
>@@ -629,7 +636,7 @@ void VideoStreamEncoder::ReconfigureEncoder() {
> 
> void VideoStreamEncoder::ConfigureQualityScaler() {
>   RTC_DCHECK_RUN_ON(&encoder_queue_);
>-  const auto scaling_settings = encoder_->GetScalingSettings();
>+  const auto scaling_settings = encoder_->GetEncoderInfo().scaling_settings;
>   const bool quality_scaling_allowed =
>       IsResolutionScalingEnabled(degradation_preference_) &&
>       scaling_settings.thresholds;
>@@ -887,7 +894,15 @@ void VideoStreamEncoder::EncodeVideoFrame(const VideoFrame& video_frame,
> 
>   overuse_detector_->FrameCaptured(out_frame, time_when_posted_us);
> 
>-  video_sender_.AddVideoFrame(out_frame, nullptr);
>+  // Encoder metadata needs to be updated before encode complete callback.
>+  VideoEncoder::EncoderInfo info = encoder_->GetEncoderInfo();
>+  if (info.implementation_name != encoder_info_.implementation_name) {
>+    encoder_stats_observer_->OnEncoderImplementationChanged(
>+        info.implementation_name);
>+  }
>+  encoder_info_ = info;
>+
>+  video_sender_.AddVideoFrame(out_frame, nullptr, encoder_info_);
> }
> 
> void VideoStreamEncoder::SendKeyFrame() {
>@@ -1085,7 +1100,7 @@ void VideoStreamEncoder::AdaptDown(AdaptReason reason) {
>       bool min_pixels_reached = false;
>       if (!source_proxy_->RequestResolutionLowerThan(
>               adaptation_request.input_pixel_count_,
>-              encoder_->GetScalingSettings().min_pixels_per_frame,
>+              encoder_->GetEncoderInfo().scaling_settings.min_pixels_per_frame,
>               &min_pixels_reached)) {
>         if (min_pixels_reached)
>           encoder_stats_observer_->OnMinPixelLimitReached();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.h
>index 4008426bac9cf0e6dc28e92864484713001256d5..8c5efbe5c369991b39bc3e9cf8278e364164a831 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.h
>@@ -268,6 +268,8 @@ class VideoStreamEncoder : public VideoStreamEncoderInterface,
>   absl::optional<int64_t> last_parameters_update_ms_
>       RTC_GUARDED_BY(&encoder_queue_);
> 
>+  VideoEncoder::EncoderInfo encoder_info_ RTC_GUARDED_BY(&encoder_queue_);
>+
>   // All public methods are proxied to |encoder_queue_|. It must must be
>   // destroyed first to make sure no tasks are run that use other members.
>   rtc::TaskQueue encoder_queue_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder_unittest.cc
>index 80f8d5a592175f3c22579033de06fcda3ab390d7..05a23c05b45e736d07dd8761b549be134a2280aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder_unittest.cc
>@@ -8,13 +8,17 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "video/video_stream_encoder.h"
>+
> #include <algorithm>
> #include <limits>
> #include <utility>
> 
>+#include "api/video/builtin_video_bitrate_allocator_factory.h"
> #include "api/video/i420_buffer.h"
>+#include "api/video_codecs/create_vp8_temporal_layers.h"
>+#include "api/video_codecs/vp8_temporal_layers.h"
> #include "media/base/videoadapter.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_temporal_layers.h"
> #include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
> #include "modules/video_coding/utility/default_video_bitrate_allocator.h"
> #include "rtc_base/fakeclock.h"
>@@ -30,7 +34,6 @@
> #include "test/gtest.h"
> #include "test/video_encoder_proxy_factory.h"
> #include "video/send_statistics_proxy.h"
>-#include "video/video_stream_encoder.h"
> 
> namespace webrtc {
> 
>@@ -102,7 +105,7 @@ class VideoStreamEncoderUnderTest : public VideoStreamEncoder {
>                                    new CpuOveruseDetectorProxy(stats_proxy))) {}
> 
>   void PostTaskAndWait(bool down, AdaptReason reason) {
>-    rtc::Event event(false, false);
>+    rtc::Event event;
>     encoder_queue()->PostTask([this, &event, reason, down] {
>       down ? AdaptDown(reason) : AdaptUp(reason);
>       event.Set();
>@@ -113,7 +116,7 @@ class VideoStreamEncoderUnderTest : public VideoStreamEncoder {
>   // This is used as a synchronisation mechanism, to make sure that the
>   // encoder queue is not blocked before we start sending it frames.
>   void WaitUntilTaskQueueIsIdle() {
>-    rtc::Event event(false, false);
>+    rtc::Event event;
>     encoder_queue()->PostTask([&event] { event.Set(); });
>     ASSERT_TRUE(event.Wait(5000));
>   }
>@@ -277,6 +280,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
>         max_framerate_(kDefaultFramerate),
>         fake_encoder_(),
>         encoder_factory_(&fake_encoder_),
>+        bitrate_allocator_factory_(CreateBuiltinVideoBitrateAllocatorFactory()),
>         stats_proxy_(new MockableSendStatisticsProxy(
>             Clock::GetRealTimeClock(),
>             video_send_config_,
>@@ -287,6 +291,8 @@ class VideoStreamEncoderTest : public ::testing::Test {
>     metrics::Reset();
>     video_send_config_ = VideoSendStream::Config(nullptr);
>     video_send_config_.encoder_settings.encoder_factory = &encoder_factory_;
>+    video_send_config_.encoder_settings.bitrate_allocator_factory =
>+        bitrate_allocator_factory_.get();
>     video_send_config_.rtp.payload_name = "FAKE";
>     video_send_config_.rtp.payload_type = 125;
> 
>@@ -492,9 +498,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
> 
>   class TestEncoder : public test::FakeEncoder {
>    public:
>-    TestEncoder()
>-        : FakeEncoder(Clock::GetRealTimeClock()),
>-          continue_encode_event_(false, false) {}
>+    TestEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
> 
>     VideoCodec codec_config() const {
>       rtc::CritScope lock(&crit_sect_);
>@@ -506,11 +510,14 @@ class VideoStreamEncoderTest : public ::testing::Test {
>       block_next_encode_ = true;
>     }
> 
>-    VideoEncoder::ScalingSettings GetScalingSettings() const override {
>+    VideoEncoder::EncoderInfo GetEncoderInfo() const override {
>+      EncoderInfo info;
>       rtc::CritScope lock(&local_crit_sect_);
>-      if (quality_scaling_)
>-        return VideoEncoder::ScalingSettings(1, 2, kMinPixelsPerFrame);
>-      return VideoEncoder::ScalingSettings::kOff;
>+      if (quality_scaling_) {
>+        info.scaling_settings =
>+            VideoEncoder::ScalingSettings(1, 2, kMinPixelsPerFrame);
>+      }
>+      return info;
>     }
> 
>     void ContinueEncode() { continue_encode_event_.Set(); }
>@@ -569,9 +576,8 @@ class VideoStreamEncoderTest : public ::testing::Test {
>         int num_streams = std::max<int>(1, config->numberOfSimulcastStreams);
>         for (int i = 0; i < num_streams; ++i) {
>           allocated_temporal_layers_.emplace_back(
>-              TemporalLayers::CreateTemporalLayers(
>-                  TemporalLayersType::kFixedPattern,
>-                  config->VP8().numberOfTemporalLayers));
>+              CreateVp8TemporalLayers(Vp8TemporalLayersType::kFixedPattern,
>+                                      config->VP8().numberOfTemporalLayers));
>         }
>       }
>       if (force_init_encode_failed_)
>@@ -587,7 +593,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
>     int last_input_width_ RTC_GUARDED_BY(local_crit_sect_) = 0;
>     int last_input_height_ RTC_GUARDED_BY(local_crit_sect_) = 0;
>     bool quality_scaling_ RTC_GUARDED_BY(local_crit_sect_) = true;
>-    std::vector<std::unique_ptr<TemporalLayers>> allocated_temporal_layers_
>+    std::vector<std::unique_ptr<Vp8TemporalLayers>> allocated_temporal_layers_
>         RTC_GUARDED_BY(local_crit_sect_);
>     bool force_init_encode_failed_ RTC_GUARDED_BY(local_crit_sect_) = false;
>   };
>@@ -595,7 +601,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
>   class TestSink : public VideoStreamEncoder::EncoderSink {
>    public:
>     explicit TestSink(TestEncoder* test_encoder)
>-        : test_encoder_(test_encoder), encoded_frame_event_(false, false) {}
>+        : test_encoder_(test_encoder) {}
> 
>     void WaitForEncodedFrame(int64_t expected_ntp_time) {
>       EXPECT_TRUE(
>@@ -694,6 +700,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
>   int max_framerate_;
>   TestEncoder fake_encoder_;
>   test::VideoEncoderProxyFactory encoder_factory_;
>+  std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
>   std::unique_ptr<MockableSendStatisticsProxy> stats_proxy_;
>   TestSink sink_;
>   AdaptingFrameForwarder video_source_;
>@@ -703,7 +710,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
> 
> TEST_F(VideoStreamEncoderTest, EncodeOneFrame) {
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-  rtc::Event frame_destroyed_event(false, false);
>+  rtc::Event frame_destroyed_event;
>   video_source_.IncomingCapturedFrame(CreateFrame(1, &frame_destroyed_event));
>   WaitForEncodedFrame(1);
>   EXPECT_TRUE(frame_destroyed_event.Wait(kDefaultTimeoutMs));
>@@ -712,7 +719,7 @@ TEST_F(VideoStreamEncoderTest, EncodeOneFrame) {
> 
> TEST_F(VideoStreamEncoderTest, DropsFramesBeforeFirstOnBitrateUpdated) {
>   // Dropped since no target bitrate has been set.
>-  rtc::Event frame_destroyed_event(false, false);
>+  rtc::Event frame_destroyed_event;
>   // The encoder will cache up to one frame for a short duration. Adding two
>   // frames means that the first frame will be dropped and the second frame will
>   // be sent when the encoder is enabled.
>@@ -770,7 +777,7 @@ TEST_F(VideoStreamEncoderTest, DropsFrameAfterStop) {
> 
>   video_stream_encoder_->Stop();
>   sink_.SetExpectNoFrames();
>-  rtc::Event frame_destroyed_event(false, false);
>+  rtc::Event frame_destroyed_event;
>   video_source_.IncomingCapturedFrame(CreateFrame(2, &frame_destroyed_event));
>   EXPECT_TRUE(frame_destroyed_event.Wait(kDefaultTimeoutMs));
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/webrtc.gni b/Source/ThirdParty/libwebrtc/Source/webrtc/webrtc.gni
>index 6ecf859ba103d0f68607659d298eee7c6a3c7184..5c5214e8a157f16fca8b83803782b523b5947ad6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/webrtc.gni
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/webrtc.gni
>@@ -9,6 +9,7 @@ import("//build/config/arm.gni")
> import("//build/config/features.gni")
> import("//build/config/mips.gni")
> import("//build/config/sanitizers/sanitizers.gni")
>+import("//build/config/sysroot.gni")
> import("//build/config/ui.gni")
> import("//build_overrides/build.gni")
> 
>@@ -35,6 +36,10 @@ if (is_mac) {
> }
> 
> declare_args() {
>+  # Setting this to true will make RTC_EXPORT (see rtc_base/system/rtc_export.h)
>+  # expand to code that will manage symbols visibility.
>+  rtc_enable_symbol_export = false
>+
>   # Setting this to true will define WEBRTC_EXCLUDE_FIELD_TRIAL_DEFAULT which
>   # will tell the pre-processor to remove the default definition of symbols
>   # needed to use field_trial. In that case a new implementation needs to be
>@@ -103,6 +108,9 @@ declare_args() {
>   # Set this to false to skip building code that requires X11.
>   rtc_use_x11 = use_x11
> 
>+  # Set this to use PipeWire on the Wayland display server.
>+  rtc_use_pipewire = false
>+
>   # Enable to use the Mozilla internal settings.
>   build_with_mozilla = false
> 
>@@ -225,7 +233,8 @@ rtc_libvpx_dir = "//third_party/libvpx"
> rtc_opus_dir = "//third_party/opus"
> 
> # Desktop capturer is supported only on Windows, OSX and Linux.
>-rtc_desktop_capture_supported = is_win || is_mac || (is_linux && rtc_use_x11)
>+rtc_desktop_capture_supported =
>+    is_win || is_mac || (is_linux && (rtc_use_x11 || rtc_use_pipewire))
> 
> ###############################################################################
> # Templates
>@@ -649,6 +658,8 @@ if (is_ios) {
>     }
>   }
> 
>+  # TODO: Generate module.modulemap file to enable use in Swift
>+  # projects. See "mac_framework_bundle_with_umbrella_header".
>   template("ios_framework_bundle_with_umbrella_header") {
>     forward_variables_from(invoker, [ "output_name" ])
>     umbrella_header_path =
>@@ -699,12 +710,17 @@ if (is_mac) {
>     forward_variables_from(invoker, [ "output_name" ])
>     this_target_name = target_name
>     umbrella_header_path = "$target_gen_dir/umbrella_header/$output_name.h"
>+    modulemap_path = "$target_gen_dir/Modules/module.modulemap"
> 
>     mac_framework_bundle(target_name) {
>       forward_variables_from(invoker, "*", [])
> 
>       framework_version = "A"
>-      framework_contents = [ "Headers" ]
>+      framework_contents = [
>+        "Headers",
>+        "Modules",
>+        "Resources",
>+      ]
> 
>       ldflags = [
>         "-all_load",
>@@ -714,7 +730,9 @@ if (is_mac) {
> 
>       deps += [
>         ":copy_framework_headers_$this_target_name",
>+        ":copy_modulemap_$this_target_name",
>         ":copy_umbrella_header_$this_target_name",
>+        ":modulemap_$this_target_name",
>         ":umbrella_header_$this_target_name",
>       ]
>     }
>@@ -727,6 +745,31 @@ if (is_mac) {
>       ]
>     }
> 
>+    action("modulemap_$this_target_name") {
>+      script = "//tools_webrtc/ios/generate_modulemap.py"
>+      args = [
>+        "--out",
>+        rebase_path(modulemap_path, root_build_dir),
>+        "--name",
>+        output_name,
>+      ]
>+      outputs = [
>+        modulemap_path,
>+      ]
>+    }
>+
>+    bundle_data("copy_modulemap_$this_target_name") {
>+      sources = [
>+        modulemap_path,
>+      ]
>+      outputs = [
>+        "{{bundle_contents_dir}}/Modules/module.modulemap",
>+      ]
>+      deps = [
>+        ":modulemap_$this_target_name",
>+      ]
>+    }
>+
>     action("umbrella_header_$this_target_name") {
>       forward_variables_from(invoker, [ "sources" ])
> 
>@@ -769,14 +812,14 @@ if (is_android) {
>                                "visibility",
>                              ])
> 
>-      javac_args = []
>+      errorprone_args = []
> 
>       # Treat warnings as errors.
>-      javac_args += [ "-Werror" ]
>+      errorprone_args += [ "-Werror" ]
> 
>       # TODO(crbug.com/824679): Find out why this fails in Chromium
>       if (!build_with_chromium) {
>-        javac_args += [
>+        errorprone_args += [
>           "-Xep:ParameterNotNullable:ERROR",
>           "-Xep:FieldMissingNullable:ERROR",
>           "-Xep:ReturnMissingNullable:ERROR",
>@@ -784,8 +827,8 @@ if (is_android) {
>       }
> 
>       # Add any arguments defined by the invoker.
>-      if (defined(invoker.javac_args)) {
>-        javac_args += invoker.javac_args
>+      if (defined(invoker.errorprone_args)) {
>+        errorprone_args += invoker.errorprone_args
>       }
> 
>       if (!defined(deps)) {
>@@ -809,11 +852,11 @@ if (is_android) {
>                              ])
> 
>       # Treat warnings as errors.
>-      javac_args = [ "-Werror" ]
>+      errorprone_args = [ "-Werror" ]
> 
>       # TODO(crbug.com/824679): Find out why this fails in Chromium
>       if (!build_with_chromium) {
>-        javac_args += [
>+        errorprone_args += [
>           "-Xep:ParameterNotNullable:ERROR",
>           "-Xep:FieldMissingNullable:ERROR",
>           "-Xep:ReturnMissingNullable:ERROR",
>@@ -841,11 +884,11 @@ if (is_android) {
>                              ])
> 
>       # Treat warnings as errors.
>-      javac_args = [ "-Werror" ]
>+      errorprone_args = [ "-Werror" ]
> 
>       # TODO(crbug.com/824679): Find out why this fails in Chromium
>       if (!build_with_chromium) {
>-        javac_args += [
>+        errorprone_args += [
>           "-Xep:ParameterNotNullable:ERROR",
>           "-Xep:FieldMissingNullable:ERROR",
>           "-Xep:ReturnMissingNullable:ERROR",
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0001-libwebrtc-changes.patch b/Source/ThirdParty/libwebrtc/WebKit/0001-libwebrtc-changes.patch
>deleted file mode 100644
>index 6b0134bb2d1a0f04461678e1f4bd58bd74faefd2..0000000000000000000000000000000000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0001-libwebrtc-changes.patch
>+++ /dev/null
>@@ -1,2082 +0,0 @@
>-From 38a838284552711d6228db9bab2dec728402098d Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Mon, 8 Oct 2018 10:38:12 -0700
>-Subject: [PATCH 1/2] libwebrtc changes
>-
>----
>- .../audio_codecs/ilbc/audio_encoder_ilbc.cc   |   2 +-
>- .../libwebrtc/Source/webrtc/api/mediatypes.cc |   4 +-
>- .../Source/webrtc/audio/remix_resample.cc     |   4 +-
>- .../Source/webrtc/common_audio/wav_file.cc    |   4 +-
>- .../webrtc/common_video/video_frame_buffer.cc |   2 +-
>- .../modules/audio_coding/acm2/rent_a_codec.cc |   4 +-
>- .../codecs/cng/audio_encoder_cng.cc           |   2 +-
>- .../codecs/ilbc/audio_encoder_ilbc.cc         |   4 +-
>- .../neteq/tools/rtp_file_source.cc            |   2 +-
>- .../Source/webrtc/p2p/base/tcpport.cc         |  10 +
>- .../libwebrtc/Source/webrtc/pc/bundlefilter.h |   2 +-
>- .../Source/webrtc/pc/peerconnectionfactory.cc |   4 +-
>- .../webrtc/pc/rtpparametersconversion.cc      |   2 +-
>- .../libwebrtc/Source/webrtc/rtc_base/checks.h |   6 +-
>- .../libwebrtc/Source/webrtc/rtc_base/flags.cc |   6 +-
>- .../Source/webrtc/rtc_base/location.h         |   2 +-
>- .../Source/webrtc/rtc_base/logging.cc         |  54 ++-
>- .../webrtc/rtc_base/logging_unittest.cc       |   2 +-
>- .../Source/webrtc/rtc_base/never_destroyed.h  |  69 ++++
>- .../rtc_base/numerics/safe_conversions.h      |   4 +-
>- .../webrtc/rtc_base/opensslcertificate.cc     |   8 +-
>- .../Source/webrtc/rtc_base/stringize_macros.h |   4 +-
>- .../webrtc/rtc_base/virtualsocketserver.cc    |   3 +-
>- .../PeerConnection/RTCH264ProfileLevelId.mm   |  58 +++
>- .../peerconnection/RTCEncodedImage+Private.mm |   4 +-
>- .../peerconnection/RTCRtpEncodingParameters.h |   1 +
>- .../RTCVideoEncoderSettings+Private.h         |   7 +
>- .../RTCVideoEncoderSettings+Private.mm        |  52 ++-
>- .../api/video_codec/RTCVideoCodecConstants.mm |   4 +-
>- .../objc/api/video_codec/RTCVideoEncoderVP8.h |   1 +
>- .../RTCWrappedNativeVideoEncoder.mm           |   5 +
>- .../video_frame_buffer/RTCNativeI420Buffer.h  |   1 +
>- .../RTCNativeMutableI420Buffer.h              |   1 +
>- .../sdk/objc/base/RTCCodecSpecificInfo.h      |   1 +
>- .../webrtc/sdk/objc/base/RTCEncodedImage.h    |   2 +
>- .../webrtc/sdk/objc/base/RTCEncodedImage.m    |   1 +
>- .../webrtc/sdk/objc/base/RTCI420Buffer.h      |   1 +
>- .../sdk/objc/base/RTCRtpFragmentationHeader.h |   1 +
>- .../sdk/objc/base/RTCRtpFragmentationHeader.m |   3 +-
>- .../webrtc/sdk/objc/base/RTCVideoCodecInfo.h  |   1 +
>- .../webrtc/sdk/objc/base/RTCVideoDecoder.h    |   1 +
>- .../sdk/objc/base/RTCVideoDecoderFactory.h    |   1 +
>- .../webrtc/sdk/objc/base/RTCVideoEncoder.h    |   8 +-
>- .../sdk/objc/base/RTCVideoEncoderFactory.h    |   1 +
>- .../objc/base/RTCVideoEncoderQpThresholds.h   |   1 +
>- .../sdk/objc/base/RTCVideoEncoderSettings.h   |   6 +
>- .../webrtc/sdk/objc/base/RTCVideoFrame.h      |   1 +
>- .../sdk/objc/base/RTCVideoFrameBuffer.h       |   1 +
>- .../video_codec/RTCCodecSpecificInfoH264.h    |   3 +-
>- .../RTCDefaultVideoDecoderFactory.h           |   1 +
>- .../RTCDefaultVideoEncoderFactory.h           |   1 +
>- .../video_codec/RTCH264ProfileLevelId.h       |   1 +
>- .../video_codec/RTCH264ProfileLevelId.mm      |  16 +-
>- .../video_codec/RTCVideoDecoderFactoryH264.h  |   1 +
>- .../video_codec/RTCVideoDecoderH264.h         |   1 +
>- .../video_codec/RTCVideoEncoderFactoryH264.h  |   1 +
>- .../video_codec/RTCVideoEncoderH264.h         |   1 +
>- .../video_codec/RTCVideoEncoderH264.mm        | 382 ++++++++++++++----
>- .../objc/components/video_codec/helpers.cc    |  16 +-
>- .../sdk/objc/components/video_codec/helpers.h |  10 +-
>- .../video_frame_buffer/RTCCVPixelBuffer.h     |   1 +
>- .../native/src/objc_video_encoder_factory.mm  |   6 +
>- 62 files changed, 637 insertions(+), 172 deletions(-)
>- create mode 100644 Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/never_destroyed.h
>- create mode 100644 Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>-index efcef382ff7..2ae75474ccd 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>-@@ -33,7 +33,7 @@ int GetIlbcBitrate(int ptime) {
>-       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
>-       return 13333;
>-     default:
>--      FATAL();
>-+      RTC_FATAL();
>-   }
>- }
>- }  // namespace
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc
>-index 599542db08f..140d0ae9625 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc
>-@@ -28,7 +28,7 @@ std::string MediaTypeToString(MediaType type) {
>-     case MEDIA_TYPE_DATA:
>-       return kMediaTypeData;
>-   }
>--  FATAL();
>-+  RTC_FATAL();
>-   // Not reachable; avoids compile warning.
>-   return "";
>- }
>-@@ -41,7 +41,7 @@ MediaType MediaTypeFromString(const std::string& type_str) {
>-   } else if (type_str == kMediaTypeData) {
>-     return MEDIA_TYPE_DATA;
>-   }
>--  FATAL();
>-+  RTC_FATAL();
>-   // Not reachable; avoids compile warning.
>-   return static_cast<MediaType>(-1);
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>-index eda70c7c378..97222e947a4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>-@@ -58,7 +58,7 @@ void RemixAndResample(const int16_t* src_data,
>- 
>-   if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
>-                                     audio_ptr_num_channels) == -1) {
>--    FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
>-+    RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
>-             << ", dst_frame->sample_rate_hz_ = " << dst_frame->sample_rate_hz_
>-             << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
>-   }
>-@@ -72,7 +72,7 @@ void RemixAndResample(const int16_t* src_data,
>-       resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(),
>-                           AudioFrame::kMaxDataSizeSamples);
>-   if (out_length == -1) {
>--    FATAL() << "Resample failed: audio_ptr = " << audio_ptr
>-+    RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
>-             << ", src_length = " << src_length
>-             << ", dst_frame->mutable_data() = " << dst_frame->mutable_data();
>-   }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>-index 008891f9cfb..0209f52d2c7 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>-@@ -53,7 +53,7 @@ WavReader::WavReader(rtc::PlatformFile file) {
>-     if (!rtc::ClosePlatformFile(file)) {
>-       RTC_LOG(LS_ERROR) << "Can't close file.";
>-     }
>--    FATAL() << "Could not open wav file for reading.";
>-+    RTC_FATAL() << "Could not open wav file for reading.";
>-   }
>- 
>-   ReadableWavFile readable(file_handle_);
>-@@ -138,7 +138,7 @@ WavWriter::WavWriter(rtc::PlatformFile file,
>-     if (!rtc::ClosePlatformFile(file)) {
>-       RTC_LOG(LS_ERROR) << "Can't close file.";
>-     }
>--    FATAL() << "Could not open wav file for writing.";
>-+    RTC_FATAL() << "Could not open wav file for writing.";
>-   }
>- 
>-   RTC_CHECK(CheckWavParameters(num_channels_, sample_rate_, kWavFormat,
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>-index 2cd0b290f15..be0b4927bbe 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>-@@ -323,7 +323,7 @@ rtc::scoped_refptr<PlanarYuvBuffer> WrapYuvBuffer(
>-       return WrapI444Buffer(width, height, y_plane, y_stride, u_plane, u_stride,
>-                             v_plane, v_stride, no_longer_used);
>-     default:
>--      FATAL() << "Unexpected frame buffer type.";
>-+      RTC_FATAL() << "Unexpected frame buffer type.";
>-       return nullptr;
>-   }
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>-index 0a9ce117e2b..45d78d74a5d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>-@@ -211,7 +211,7 @@ std::unique_ptr<AudioEncoder> CreateCngEncoder(
>-       config.vad_mode = Vad::kVadVeryAggressive;
>-       break;
>-     default:
>--      FATAL();
>-+      RTC_FATAL();
>-   }
>-   return std::unique_ptr<AudioEncoder>(new AudioEncoderCng(std::move(config)));
>- }
>-@@ -226,7 +226,7 @@ std::unique_ptr<AudioDecoder> CreateIsacDecoder(
>-   return std::unique_ptr<AudioDecoder>(
>-       new AudioDecoderIsacFloatImpl(sample_rate_hz, bwinfo));
>- #else
>--  FATAL() << "iSAC is not supported.";
>-+  RTC_FATAL() << "iSAC is not supported.";
>-   return std::unique_ptr<AudioDecoder>();
>- #endif
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>-index 4cda3401fa2..055190497a5 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>-@@ -139,7 +139,7 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl(
>-       break;
>-     }
>-     case Vad::kError: {
>--      FATAL();  // Fails only if fed invalid data.
>-+      RTC_FATAL();  // Fails only if fed invalid data.
>-       break;
>-     }
>-   }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>-index 84695e37630..dae956b30c0 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>-@@ -40,7 +40,7 @@ int GetIlbcBitrate(int ptime) {
>-       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
>-       return 13333;
>-     default:
>--      FATAL();
>-+      RTC_FATAL();
>-   }
>- }
>- 
>-@@ -147,7 +147,7 @@ size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
>-     case 6:
>-       return 2 * 50;
>-     default:
>--      FATAL();
>-+      RTC_FATAL();
>-   }
>- }
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>-index 806bba7a9fe..7bf7d52083c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>-@@ -89,7 +89,7 @@ bool RtpFileSource::OpenFile(const std::string& file_name) {
>-     return true;
>-   rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name));
>-   if (!rtp_reader_) {
>--    FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
>-+    RTC_FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
>-                "that .pcapng is not supported.";
>-   }
>-   return true;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>-index d070f6ecee8..c66c7307375 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>-@@ -351,11 +351,21 @@ TCPConnection::TCPConnection(TCPPort* port,
>-                         << ", port() Network:" << port->Network()->ToString();
>-     const std::vector<rtc::InterfaceAddress>& desired_addresses =
>-         port_->Network()->GetIPs();
>-+
>-+#if defined(WEBRTC_WEBKIT_BUILD)
>-+     RTC_DCHECK(socket->GetLocalAddress().IsLoopbackIP() ||
>-+                (std::find_if(desired_addresses.begin(), desired_addresses.end(),
>-+                             [this](const rtc::InterfaceAddress& addr) {
>-+                               return socket_->GetLocalAddress().ipaddr() ==
>-+                                      addr;
>-+                             }) != desired_addresses.end()));
>-+ #else
>-     RTC_DCHECK(std::find_if(desired_addresses.begin(), desired_addresses.end(),
>-                             [this](const rtc::InterfaceAddress& addr) {
>-                               return socket_->GetLocalAddress().ipaddr() ==
>-                                      addr;
>-                             }) != desired_addresses.end());
>-+#endif
>-     ConnectSocketSignals(socket);
>-   }
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.h
>-index 7decbba8a4f..de477f457f2 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.h
>-@@ -17,7 +17,7 @@
>- #include <vector>
>- 
>- #include "media/base/streamparams.h"
>--#include "rtc_base/basictypes.h"
>-+//#include "rtc_base/basictypes.h"
>- 
>- namespace cricket {
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>-index b78ed8db1ee..53f4da8516e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>-@@ -257,7 +257,7 @@ RtpCapabilities PeerConnectionFactory::GetRtpSenderCapabilities(
>-       return RtpCapabilities();
>-   }
>-   // Not reached; avoids compile warning.
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities(
>-@@ -284,7 +284,7 @@ RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities(
>-       return RtpCapabilities();
>-   }
>-   // Not reached; avoids compile warning.
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- rtc::scoped_refptr<AudioSourceInterface>
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>-index a57066e4db0..e519b9b06de 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>-@@ -61,7 +61,7 @@ RTCErrorOr<cricket::FeedbackParam> ToCricketFeedbackParam(
>-       return cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc);
>-   }
>-   // Not reached; avoids compile warning.
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- template <typename C>
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>-index 9de6d47573c..5cf0679910e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>-@@ -80,7 +80,7 @@ RTC_NORETURN void rtc_FatalMessage(const char* file, int line, const char* msg);
>- //   messages if the condition doesn't hold. Prefer them to raw RTC_CHECK and
>- //   RTC_DCHECK.
>- //
>--// - FATAL() aborts unconditionally.
>-+// - RTC_FATAL() aborts unconditionally.
>- //
>- // TODO(ajm): Ideally, checks.h would be combined with logging.h, but
>- // consolidation with system_wrappers/logging.h should happen first.
>-@@ -348,9 +348,9 @@ class FatalLogCall final {
>- #define RTC_NOTREACHED() RTC_DCHECK(RTC_UNREACHABLE_CODE_HIT)
>- 
>- // TODO(bugs.webrtc.org/8454): Add an RTC_ prefix or rename differently.
>--#define FATAL()                                                    \
>-+#define RTC_FATAL()                                                    \
>-   rtc::webrtc_checks_impl::FatalLogCall<false>(__FILE__, __LINE__, \
>--                                               "FATAL()") &        \
>-+                                               "RTC_FATAL()") &        \
>-       rtc::webrtc_checks_impl::LogStreamer<>()
>- 
>- // Performs the integer division a/b and returns the result. CHECKs that the
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>-index 5b28794f6ca..4f294594bd4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>-@@ -80,7 +80,7 @@ void Flag::SetToDefault() {
>-       variable_->s = default_.s;
>-       return;
>-   }
>--  FATAL() << "unreachable code";
>-+  RTC_FATAL() << "unreachable code";
>- }
>- 
>- static const char* Type2String(Flag::Type type) {
>-@@ -94,7 +94,7 @@ static const char* Type2String(Flag::Type type) {
>-     case Flag::STRING:
>-       return "string";
>-   }
>--  FATAL() << "unreachable code";
>-+  RTC_FATAL() << "unreachable code";
>- }
>- 
>- static void PrintFlagValue(Flag::Type type, FlagValue* p) {
>-@@ -112,7 +112,7 @@ static void PrintFlagValue(Flag::Type type, FlagValue* p) {
>-       printf("%s", p->s);
>-       return;
>-   }
>--  FATAL() << "unreachable code";
>-+  RTC_FATAL() << "unreachable code";
>- }
>- 
>- void Flag::Print(bool print_current_value) {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h
>-index 513bc263651..718d9589348 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h
>-@@ -50,7 +50,7 @@ class Location {
>- #define RTC_FROM_HERE RTC_FROM_HERE_WITH_FUNCTION(__FUNCTION__)
>- 
>- #define RTC_FROM_HERE_WITH_FUNCTION(function_name) \
>--  ::rtc::Location(function_name, __FILE__ ":" STRINGIZE(__LINE__))
>-+  ::rtc::Location(function_name, __FILE__ ":" RTC_STRINGIZE(__LINE__))
>- 
>- }  // namespace rtc
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>-index 129a8974397..58456c599ec 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>-@@ -36,6 +36,7 @@ static const int kMaxLogLineSize = 1024 - 60;
>- #include "rtc_base/criticalsection.h"
>- #include "rtc_base/logging.h"
>- #include "rtc_base/platform_thread_types.h"
>-+#include "rtc_base/never_destroyed.h"
>- #include "rtc_base/stringencode.h"
>- #include "rtc_base/strings/string_builder.h"
>- #include "rtc_base/stringutils.h"
>-@@ -62,8 +63,12 @@ const char* FilenameFromPath(const char* file) {
>-     return (end1 > end2) ? end1 + 1 : end2 + 1;
>- }
>- 
>--// Global lock for log subsystem, only needed to serialize access to streams_.
>--CriticalSection g_log_crit;
>-+// Global lock for log subsystem, only needed to serialize access to streamList().
>-+CriticalSection& logCriticalScope() {
>-+  static auto scope = makeNeverDestroyed<>(CriticalSection { });
>-+  return scope.get();
>-+}
>-+
>- }  // namespace
>- 
>- // Inefficient default implementation, override is recommended.
>-@@ -83,7 +88,15 @@ bool LogMessage::log_to_stderr_ = true;
>- // Note: we explicitly do not clean this up, because of the uncertain ordering
>- // of destructors at program exit.  Let the person who sets the stream trigger
>- // cleanup by setting to null, or let it leak (safe at program exit).
>--LogMessage::StreamList LogMessage::streams_ RTC_GUARDED_BY(g_log_crit);
>-+typedef std::pair<LogSink*, LoggingSeverity> StreamAndSeverity;
>-+typedef std::list<StreamAndSeverity> StreamList;
>-+
>-+// The output streams and their associated severities
>-+StreamList& streamList()
>-+    RTC_EXCLUSIVE_LOCKS_REQUIRED(logCriticalScope()) {
>-+        static auto stream_list = makeNeverDestroyed<>(StreamList { });
>-+  return stream_list.get();
>-+}
>- 
>- // Boolean options default to false (0)
>- bool LogMessage::thread_, LogMessage::timestamp_;
>-@@ -97,6 +110,10 @@ LogMessage::LogMessage(const char* file,
>-                        LogErrorContext err_ctx,
>-                        int err)
>-     : severity_(sev) {
>-+
>-+  static std::once_flag callLogCriticalScopeOnce;
>-+  std::call_once(callLogCriticalScopeOnce,[] { logCriticalScope(); });
>-+
>-   if (timestamp_) {
>-     // Use SystemTimeMillis so that even if tests use fake clocks, the timestamp
>-     // in log messages represents the real system time.
>-@@ -197,8 +214,8 @@ LogMessage::~LogMessage() {
>- #endif
>-   }
>- 
>--  CritScope cs(&g_log_crit);
>--  for (auto& kv : streams_) {
>-+  CritScope cs(&logCriticalScope());
>-+  for (auto& kv : streamList()) {
>-     if (severity_ >= kv.second) {
>- #if defined(WEBRTC_ANDROID)
>-       kv.first->OnLogMessage(str, severity_, tag_);
>-@@ -246,7 +263,7 @@ void LogMessage::LogTimestamps(bool on) {
>- 
>- void LogMessage::LogToDebug(LoggingSeverity min_sev) {
>-   g_dbg_sev = min_sev;
>--  CritScope cs(&g_log_crit);
>-+  CritScope cs(&logCriticalScope());
>-   UpdateMinLogSeverity();
>- }
>- 
>-@@ -255,9 +272,9 @@ void LogMessage::SetLogToStderr(bool log_to_stderr) {
>- }
>- 
>- int LogMessage::GetLogToStream(LogSink* stream) {
>--  CritScope cs(&g_log_crit);
>-+  CritScope cs(&logCriticalScope());
>-   LoggingSeverity sev = LS_NONE;
>--  for (auto& kv : streams_) {
>-+  for (auto& kv : streamList()) {
>-     if (!stream || stream == kv.first) {
>-       sev = std::min(sev, kv.second);
>-     }
>-@@ -266,16 +283,16 @@ int LogMessage::GetLogToStream(LogSink* stream) {
>- }
>- 
>- void LogMessage::AddLogToStream(LogSink* stream, LoggingSeverity min_sev) {
>--  CritScope cs(&g_log_crit);
>--  streams_.push_back(std::make_pair(stream, min_sev));
>-+  CritScope cs(&logCriticalScope());
>-+  streamList().push_back(std::make_pair(stream, min_sev));
>-   UpdateMinLogSeverity();
>- }
>- 
>- void LogMessage::RemoveLogToStream(LogSink* stream) {
>--  CritScope cs(&g_log_crit);
>--  for (StreamList::iterator it = streams_.begin(); it != streams_.end(); ++it) {
>-+  CritScope cs(&logCriticalScope());
>-+  for (StreamList::iterator it = streamList().begin(); it != streamList().end(); ++it) {
>-     if (stream == it->first) {
>--      streams_.erase(it);
>-+      streamList().erase(it);
>-       break;
>-     }
>-   }
>-@@ -334,9 +351,9 @@ void LogMessage::ConfigureLogging(const char* params) {
>- }
>- 
>- void LogMessage::UpdateMinLogSeverity()
>--    RTC_EXCLUSIVE_LOCKS_REQUIRED(g_log_crit) {
>-+    RTC_EXCLUSIVE_LOCKS_REQUIRED(logCriticalScope()) {
>-   LoggingSeverity min_sev = g_dbg_sev;
>--  for (const auto& kv : streams_) {
>-+  for (const auto& kv : streamList()) {
>-     const LoggingSeverity sev = kv.second;
>-     min_sev = std::min(min_sev, sev);
>-   }
>-@@ -450,11 +467,8 @@ bool LogMessage::IsNoop(LoggingSeverity severity) {
>-   // TODO(tommi): We're grabbing this lock for every LogMessage instance that
>-   // is going to be logged. This introduces unnecessary synchronization for
>-   // a feature that's mostly used for testing.
>--  CritScope cs(&g_log_crit);
>--  if (streams_.size() > 0)
>--    return false;
>--
>--  return true;
>-+  CritScope cs(&logCriticalScope());
>-+  return streamList().size() == 0;
>- }
>- 
>- void LogMessage::FinishPrintStream() {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>-index a475e52d465..6b2263923fa 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>-@@ -221,7 +221,7 @@ TEST(LogTest, SingleStream) {
>- 
>- #if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>- TEST(LogTest, Checks) {
>--  EXPECT_DEATH(FATAL() << "message",
>-+  EXPECT_DEATH(RTC_FATAL() << "message",
>-                "\n\n#\n"
>-                "# Fatal error in: \\S+, line \\w+\n"
>-                "# last system error: \\w+\n"
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/never_destroyed.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/never_destroyed.h
>-new file mode 100644
>-index 00000000000..fcc62e35534
>---- /dev/null
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/never_destroyed.h
>-@@ -0,0 +1,69 @@
>-+/*
>-+ * Copyright (C) 2018 Apple Inc. All rights reserved.
>-+ *
>-+ * Redistribution and use in source and binary forms, with or without
>-+ * modification, are permitted provided that the following conditions
>-+ * are met:
>-+ * 1. Redistributions of source code must retain the above copyright
>-+ *    notice, this list of conditions and the following disclaimer.
>-+ * 2. Redistributions in binary form must reproduce the above copyright
>-+ *    notice, this list of conditions and the following disclaimer in the
>-+ *    documentation and/or other materials provided with the distribution.
>-+ *
>-+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
>-+ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
>-+ * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
>-+ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
>-+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
>-+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
>-+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
>-+ * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
>-+ * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
>-+ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
>-+ * THE POSSIBILITY OF SUCH DAMAGE.
>-+ */
>-+
>-+#pragma once
>-+
>-+#include <type_traits>
>-+#include <utility>
>-+
>-+namespace rtc {
>-+
>-+template<typename T> class NeverDestroyed {
>-+public:
>-+    template<typename... Args> NeverDestroyed(Args&&... args)
>-+    {
>-+        new (storagePointer()) T(std::forward<Args>(args)...);
>-+    }
>-+
>-+    NeverDestroyed(NeverDestroyed&& other)
>-+    {
>-+        new (storagePointer()) T(std::move(*other.storagePointer()));
>-+    }
>-+
>-+    operator T&() { return *storagePointer(); }
>-+    T& get() { return *storagePointer(); }
>-+
>-+    operator const T&() const { return *storagePointer(); }
>-+    const T& get() const { return *storagePointer(); }
>-+
>-+private:
>-+    NeverDestroyed(const NeverDestroyed&) = delete;
>-+    NeverDestroyed& operator=(const NeverDestroyed&) = delete;
>-+
>-+    using PointerType = typename std::remove_const<T>::type*;
>-+
>-+    PointerType storagePointer() const { return const_cast<PointerType>(reinterpret_cast<const T*>(&m_storage)); }
>-+
>-+    // FIXME: Investigate whether we should allocate a hunk of virtual memory
>-+    // and hand out chunks of it to NeverDestroyed instead, to reduce fragmentation.
>-+    typename std::aligned_storage<sizeof(T), std::alignment_of<T>::value>::type m_storage;
>-+};
>-+
>-+template<typename T> inline NeverDestroyed<T> makeNeverDestroyed(T&& argument)
>-+{
>-+    return NeverDestroyed<T>(std::move(argument));
>-+}
>-+
>-+} // namespace rtc
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h
>-index 58efcaa746a..48c212e4d49 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h
>-@@ -63,11 +63,11 @@ inline Dst saturated_cast(Src value) {
>- 
>-     // Should fail only on attempting to assign NaN to a saturated integer.
>-     case internal::TYPE_INVALID:
>--      FATAL();
>-+      RTC_FATAL();
>-       return std::numeric_limits<Dst>::max();
>-   }
>- 
>--  FATAL();
>-+  RTC_FATAL();
>-   return static_cast<Dst>(value);
>- }
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc
>-index ed67a8938e1..236bef55c8a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc
>-@@ -256,11 +256,11 @@ OpenSSLCertificate* OpenSSLCertificate::GetReference() const {
>- std::string OpenSSLCertificate::ToPEMString() const {
>-   BIO* bio = BIO_new(BIO_s_mem());
>-   if (!bio) {
>--    FATAL() << "unreachable code";
>-+    RTC_FATAL() << "unreachable code";
>-   }
>-   if (!PEM_write_bio_X509(bio, x509_)) {
>-     BIO_free(bio);
>--    FATAL() << "unreachable code";
>-+    RTC_FATAL() << "unreachable code";
>-   }
>-   BIO_write(bio, "\0", 1);
>-   char* buffer;
>-@@ -277,11 +277,11 @@ void OpenSSLCertificate::ToDER(Buffer* der_buffer) const {
>-   // Calculates the DER representation of the certificate, from scratch.
>-   BIO* bio = BIO_new(BIO_s_mem());
>-   if (!bio) {
>--    FATAL() << "unreachable code";
>-+    RTC_FATAL() << "unreachable code";
>-   }
>-   if (!i2d_X509_bio(bio, x509_)) {
>-     BIO_free(bio);
>--    FATAL() << "unreachable code";
>-+    RTC_FATAL() << "unreachable code";
>-   }
>-   char* data;
>-   size_t length = BIO_get_mem_data(bio, &data);
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>-index aee8d14551d..38c6f1836bd 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>-@@ -8,7 +8,7 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--// Modified from the Chromium original:
>-+// Modified from the Chxsromium original:
>- // src/base/strings/stringize_macros.h
>- 
>- // This file defines preprocessor macros for stringizing preprocessor
>-@@ -33,6 +33,6 @@
>- // Then:
>- //   STRINGIZE(A) produces "FOO"
>- //   STRINGIZE(B(y)) produces "myobj->FunctionCall(y)"
>--#define STRINGIZE(x) STRINGIZE_NO_EXPANSION(x)
>-+#define RTC_STRINGIZE(x) STRINGIZE_NO_EXPANSION(x)
>- 
>- #endif  // RTC_BASE_STRINGIZE_MACROS_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>-index 53dad46fced..ae40d7888f5 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>-@@ -1027,9 +1027,8 @@ void VirtualSocketServer::UpdateDelayDistribution() {
>-   }
>- }
>- 
>--static double PI = 4 * atan(1.0);
>--
>- static double Normal(double x, double mean, double stddev) {
>-+  static const double PI = 4 * atan(1.0);
>-   double a = (x - mean) * (x - mean) / (2 * stddev * stddev);
>-   return exp(-a) / (stddev * sqrt(2 * PI));
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm
>-new file mode 100644
>-index 00000000000..04a5689417c
>---- /dev/null
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm
>-@@ -0,0 +1,58 @@
>-+/*
>-+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>-+ *
>-+ *  Use of this source code is governed by a BSD-style license
>-+ *  that can be found in the LICENSE file in the root of the source
>-+ *  tree. An additional intellectual property rights grant can be found
>-+ *  in the file PATENTS.  All contributing project authors may
>-+ *  be found in the AUTHORS file in the root of the source tree.
>-+ *
>-+ */
>-+
>-+#import "WebRTC/RTCVideoCodecH264.h"
>-+
>-+#include "media/base/h264_profile_level_id.h"
>-+
>-+@interface RTCH264ProfileLevelId ()
>-+
>-+@property(nonatomic, assign) RTCH264Profile profile;
>-+@property(nonatomic, assign) RTCH264Level level;
>-+@property(nonatomic, strong) NSString *hexString;
>-+
>-+@end
>-+
>-+@implementation RTCH264ProfileLevelId
>-+
>-+@synthesize profile = _profile;
>-+@synthesize level = _level;
>-+@synthesize hexString = _hexString;
>-+
>-+- (instancetype)initWithHexString:(NSString *)hexString {
>-+  if (self = [super init]) {
>-+    self.hexString = hexString;
>-+
>-+    absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
>-+        webrtc::H264::ParseProfileLevelId([hexString cStringUsingEncoding:NSUTF8StringEncoding]);
>-+    if (profile_level_id.has_value()) {
>-+      self.profile = static_cast<RTCH264Profile>(profile_level_id->profile);
>-+      self.level = static_cast<RTCH264Level>(profile_level_id->level);
>-+    }
>-+  }
>-+  return self;
>-+}
>-+
>-+- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level {
>-+  if (self = [super init]) {
>-+    self.profile = profile;
>-+    self.level = level;
>-+
>-+    absl::optional<std::string> hex_string =
>-+        webrtc::H264::ProfileLevelIdToString(webrtc::H264::ProfileLevelId(
>-+            static_cast<webrtc::H264::Profile>(profile), static_cast<webrtc::H264::Level>(level)));
>-+    self.hexString =
>-+        [NSString stringWithCString:hex_string.value_or("").c_str() encoding:NSUTF8StringEncoding];
>-+  }
>-+  return self;
>-+}
>-+
>-+@end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
>-index 6f2d1f46d9c..94cb1f110f9 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
>-@@ -35,6 +35,7 @@ - (instancetype)initWithNativeEncodedImage:(webrtc::EncodedImage)encodedImage {
>-     self.contentType = (encodedImage.content_type_ == webrtc::VideoContentType::SCREENSHARE) ?
>-         RTCVideoContentTypeScreenshare :
>-         RTCVideoContentTypeUnspecified;
>-+    self.spatialIndex = encodedImage.SpatialIndex() ? *encodedImage.SpatialIndex() : 0;
>-   }
>- 
>-   return self;
>-@@ -59,7 +60,8 @@ - (webrtc::EncodedImage)nativeEncodedImage {
>-   encodedImage.content_type_ = (self.contentType == RTCVideoContentTypeScreenshare) ?
>-       webrtc::VideoContentType::SCREENSHARE :
>-       webrtc::VideoContentType::UNSPECIFIED;
>--
>-+  if (self.spatialIndex)
>-+    encodedImage.SetSpatialIndex(self.spatialIndex);
>-   return encodedImage;
>- }
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h
>-index ba50bde649f..dabff725abb 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h
>-@@ -15,6 +15,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpEncodingParameters")))
>- @interface RTCRtpEncodingParameters : NSObject
>- 
>- /** Controls whether the encoding is currently transmitted. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h
>-index 5b062455bc0..72af732b23b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h
>-@@ -22,4 +22,11 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- @end
>- 
>-+@interface RTCVideoBitrateAllocation (Private)
>-+
>-+- (instancetype)initWithNativeVideoBitrateAllocation:(const webrtc::VideoBitrateAllocation *__nullable)videoBitrateAllocation;
>-+- (webrtc::VideoBitrateAllocation)nativeVideoBitrateAllocation;
>-+
>-+@end
>-+
>- NS_ASSUME_NONNULL_END
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm
>-index 6fb81dbb8af..82b5ba7bc0f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm
>-@@ -12,6 +12,23 @@
>- 
>- #import "helpers/NSString+StdString.h"
>- 
>-+@implementation RTCVideoEncoderSettings {
>-+  webrtc::VideoCodec _nativeVideoCodec;
>-+}
>-+
>-+@synthesize name = _name;
>-+@synthesize width = _width;
>-+@synthesize height = _height;
>-+@synthesize startBitrate = _startBitrate;
>-+@synthesize maxBitrate = _maxBitrate;
>-+@synthesize minBitrate = _minBitrate;
>-+@synthesize targetBitrate = _targetBitrate;
>-+@synthesize maxFramerate = _maxFramerate;
>-+@synthesize qpMax = _qpMax;
>-+@synthesize mode = _mode;
>-+
>-+@end
>-+
>- @implementation RTCVideoEncoderSettings (Private)
>- 
>- - (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *)videoCodec {
>-@@ -20,6 +37,7 @@ - (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *)videoCodec
>-       const char *codecName = CodecTypeToPayloadString(videoCodec->codecType);
>-       self.name = [NSString stringWithUTF8String:codecName];
>- 
>-+      _nativeVideoCodec = *videoCodec;
>-       self.width = videoCodec->width;
>-       self.height = videoCodec->height;
>-       self.startBitrate = videoCodec->startBitrate;
>-@@ -36,18 +54,28 @@ - (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *)videoCodec
>- }
>- 
>- - (webrtc::VideoCodec)nativeVideoCodec {
>--  webrtc::VideoCodec videoCodec;
>--  videoCodec.width = self.width;
>--  videoCodec.height = self.height;
>--  videoCodec.startBitrate = self.startBitrate;
>--  videoCodec.maxBitrate = self.maxBitrate;
>--  videoCodec.minBitrate = self.minBitrate;
>--  videoCodec.targetBitrate = self.targetBitrate;
>--  videoCodec.maxBitrate = self.maxBitrate;
>--  videoCodec.qpMax = self.qpMax;
>--  videoCodec.mode = (webrtc::VideoCodecMode)self.mode;
>--
>--  return videoCodec;
>-+  return _nativeVideoCodec;
>-+}
>-+
>-+@end
>-+
>-+@implementation RTCVideoBitrateAllocation {
>-+  webrtc::VideoBitrateAllocation _nativeVideoBitrateAllocation;
>-+}
>-+
>-+@end
>-+
>-+@implementation RTCVideoBitrateAllocation (Private)
>-+
>-+- (instancetype)initWithNativeVideoBitrateAllocation:(const webrtc::VideoBitrateAllocation *)videoBitrateAllocation {
>-+  if (self = [super init]) {
>-+    _nativeVideoBitrateAllocation = *videoBitrateAllocation;
>-+  }
>-+  return self;
>-+}
>-+
>-+- (webrtc::VideoBitrateAllocation)nativeVideoBitrateAllocation {
>-+  return _nativeVideoBitrateAllocation;
>- }
>- 
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.mm
>-index acbf126170c..1447d3b27e1 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.mm
>-@@ -13,5 +13,5 @@
>- 
>- #include "media/base/mediaconstants.h"
>- 
>--NSString *const kRTCVideoCodecVp8Name = @(cricket::kVp8CodecName);
>--NSString *const kRTCVideoCodecVp9Name = @(cricket::kVp9CodecName);
>-+NSString *const kRTCVideoCodecVp8Name = @"VP8";
>-+NSString *const kRTCVideoCodecVp9Name = @"VP9";
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.h
>-index 8d87a898931..08d18504222 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.h
>-@@ -14,6 +14,7 @@
>- #import "RTCVideoEncoder.h"
>- 
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderVP8")))
>- @interface RTCVideoEncoderVP8 : NSObject
>- 
>- /* This returns a VP8 encoder that can be returned from a RTCVideoEncoderFactory injected into
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm
>-index 9afd54f55f5..b9634779f1f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm
>-@@ -58,6 +58,11 @@ - (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
>-   return 0;
>- }
>- 
>-+- (int)setRateAllocation:(nonnull RTCVideoBitrateAllocation *)allocation framerate:(uint32_t)framerate {
>-+  RTC_NOTREACHED();
>-+  return 0;
>-+}
>-+
>- - (NSString *)implementationName {
>-   RTC_NOTREACHED();
>-   return nil;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h
>-index 9a904f5396a..dc903bc4bca 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h
>-@@ -17,6 +17,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** RTCI420Buffer implements the RTCI420Buffer protocol */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
>- @interface RTCI420Buffer : NSObject<RTCI420Buffer>
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h
>-index 6cd5110460b..7cfa950abb6 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h
>-@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** Mutable version of RTCI420Buffer */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMutableI420Buffer")))
>- @interface RTCMutableI420Buffer : RTCI420Buffer<RTCMutableI420Buffer>
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCCodecSpecificInfo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCCodecSpecificInfo.h
>-index e2ae4cafa11..9dc44c12af8 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCCodecSpecificInfo.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCCodecSpecificInfo.h
>-@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  *  Corresponds to webrtc::CodecSpecificInfo.
>-  */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCCodecSpecificInfo")))
>- @protocol RTCCodecSpecificInfo <NSObject>
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.h
>-index 670c7276ff7..7994ce64a1f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.h
>-@@ -31,6 +31,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoContentType) {
>- 
>- /** Represents an encoded frame. Corresponds to webrtc::EncodedImage. */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCEncodedImage")))
>- @interface RTCEncodedImage : NSObject
>- 
>- @property(nonatomic, strong) NSData *buffer;
>-@@ -47,6 +48,7 @@ RTC_OBJC_EXPORT
>- @property(nonatomic, assign) BOOL completeFrame;
>- @property(nonatomic, strong) NSNumber *qp;
>- @property(nonatomic, assign) RTCVideoContentType contentType;
>-+@property(nonatomic, assign) int spatialIndex;
>- 
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.m
>-index 024a57c541b..85cd5e298e4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.m
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.m
>-@@ -26,5 +26,6 @@ @implementation RTCEncodedImage
>- @synthesize completeFrame = _completeFrame;
>- @synthesize qp = _qp;
>- @synthesize contentType = _contentType;
>-+@synthesize spatialIndex = _spatialIndex;
>- 
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCI420Buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCI420Buffer.h
>-index a6c7e41bcba..cfa14f8af23 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCI420Buffer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCI420Buffer.h
>-@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** Protocol for RTCYUVPlanarBuffers containing I420 data */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
>- @protocol RTCI420Buffer <RTCYUVPlanarBuffer>
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.h
>-index 2e26b08b8af..fbbc68a8ef6 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.h
>-@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** Information for header. Corresponds to webrtc::RTPFragmentationHeader. */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpFragmentationHeader")))
>- @interface RTCRtpFragmentationHeader : NSObject
>- 
>- @property(nonatomic, strong) NSArray<NSNumber *> *fragmentationOffset;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.m
>-index 8049abc411e..6cf21ddc2a1 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.m
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.m
>-@@ -17,4 +17,5 @@ @implementation RTCRtpFragmentationHeader
>- @synthesize fragmentationTimeDiff = _fragmentationTimeDiff;
>- @synthesize fragmentationPlType = _fragmentationPlType;
>- 
>--@end
>-\ No newline at end of file
>-+@end
>-+
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoCodecInfo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoCodecInfo.h
>-index 2162caaa21f..4e643d94347 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoCodecInfo.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoCodecInfo.h
>-@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** Holds information to identify a codec. Corresponds to webrtc::SdpVideoFormat. */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoCodecInfo")))
>- @interface RTCVideoCodecInfo : NSObject <NSCoding>
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoder.h
>-index 18c6f6b0006..534420d3955 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoder.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoder.h
>-@@ -23,6 +23,7 @@ typedef void (^RTCVideoDecoderCallback)(RTCVideoFrame *frame);
>- 
>- /** Protocol for decoder implementations. */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoder")))
>- @protocol RTCVideoDecoder <NSObject>
>- 
>- - (void)setCallback:(RTCVideoDecoderCallback)callback;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoderFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoderFactory.h
>-index 3e24153b82c..cb95fa3f134 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoderFactory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoderFactory.h
>-@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** RTCVideoDecoderFactory is an Objective-C version of webrtc::VideoDecoderFactory. */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactory")))
>- @protocol RTCVideoDecoderFactory <NSObject>
>- 
>- - (nullable id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoder.h
>-index c5257674d83..5291bb1342b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoder.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoder.h
>-@@ -19,7 +19,11 @@
>- #import "RTCVideoFrame.h"
>- 
>- NS_ASSUME_NONNULL_BEGIN
>--
>-+/*
>-+namespace webrtc {
>-+class VideoBitrateAllocation;
>-+};
>-+*/
>- /** Callback block for encoder. */
>- typedef BOOL (^RTCVideoEncoderCallback)(RTCEncodedImage *frame,
>-                                         id<RTCCodecSpecificInfo> info,
>-@@ -27,6 +31,7 @@ typedef BOOL (^RTCVideoEncoderCallback)(RTCEncodedImage *frame,
>- 
>- /** Protocol for encoder implementations. */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoder")))
>- @protocol RTCVideoEncoder <NSObject>
>- 
>- - (void)setCallback:(RTCVideoEncoderCallback)callback;
>-@@ -37,6 +42,7 @@ RTC_OBJC_EXPORT
>-     codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
>-            frameTypes:(NSArray<NSNumber *> *)frameTypes;
>- - (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate;
>-+- (int)setRateAllocation: (RTCVideoBitrateAllocation *)allocation framerate:(uint32_t)framerate;
>- - (NSString *)implementationName;
>- 
>- /** Returns QP scaling settings for encoder. The quality scaler adjusts the resolution in order to
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderFactory.h
>-index 20c603d6fe6..cbe9ea492ae 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderFactory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderFactory.h
>-@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** RTCVideoEncoderFactory is an Objective-C version of webrtc::VideoEncoderFactory. */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
>- @protocol RTCVideoEncoderFactory <NSObject>
>- 
>- - (nullable id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.h
>-index 2b48f45ce0a..68cfd66df47 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.h
>-@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** QP thresholds for encoder. Corresponds to webrtc::VideoEncoder::QpThresholds. */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderQpThresholds")))
>- @interface RTCVideoEncoderQpThresholds : NSObject
>- 
>- - (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderSettings.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderSettings.h
>-index 69e04cac70f..ae3bf5fb5f1 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderSettings.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderSettings.h
>-@@ -21,6 +21,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
>- 
>- /** Settings for encoder. Corresponds to webrtc::VideoCodec. */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderSettings")))
>- @interface RTCVideoEncoderSettings : NSObject
>- 
>- @property(nonatomic, strong) NSString *name;
>-@@ -40,4 +41,9 @@ RTC_OBJC_EXPORT
>- 
>- @end
>- 
>-+RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoBitrateAllocation")))
>-+@interface RTCVideoBitrateAllocation : NSObject
>-+@end
>-+
>- NS_ASSUME_NONNULL_END
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrame.h
>-index 9aca7433f34..bde09c50c9d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrame.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrame.h
>-@@ -26,6 +26,7 @@ typedef NS_ENUM(NSInteger, RTCVideoRotation) {
>- 
>- // RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoFrame")))
>- @interface RTCVideoFrame : NSObject
>- 
>- /** Width without rotation applied. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrameBuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrameBuffer.h
>-index bb9e6fba631..186ad0889bf 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrameBuffer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrameBuffer.h
>-@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- // RTCVideoFrameBuffer is an ObjectiveC version of webrtc::VideoFrameBuffer.
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
>- @protocol RTCVideoFrameBuffer <NSObject>
>- 
>- @property(nonatomic, readonly) int width;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.h
>-index ece9570a13b..e947dc6e623 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.h
>-@@ -10,7 +10,7 @@
>- 
>- #import <Foundation/Foundation.h>
>- 
>--#import "RTCCodecSpecificInfo.h"
>-+#import "base/RTCCodecSpecificInfo.h"
>- #import "RTCMacros.h"
>- 
>- /** Class for H264 specific config. */
>-@@ -20,6 +20,7 @@ typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
>- };
>- 
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCCodecSpecificInfoH264")))
>- @interface RTCCodecSpecificInfoH264 : NSObject <RTCCodecSpecificInfo>
>- 
>- @property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h
>-index 7ca9463a593..d11460300d5 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h
>-@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  *  codecs, create custom implementations of RTCVideoEncoderFactory and RTCVideoDecoderFactory.
>-  */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDefaultVideoDecoderFactory")))
>- @interface RTCDefaultVideoDecoderFactory : NSObject <RTCVideoDecoderFactory>
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h
>-index c45e54362b2..9323256cead 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h
>-@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  *  codecs, create custom implementations of RTCVideoEncoderFactory and RTCVideoDecoderFactory.
>-  */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDefaultVideoEncoderFactory")))
>- @interface RTCDefaultVideoEncoderFactory : NSObject <RTCVideoEncoderFactory>
>- 
>- @property(nonatomic, retain) RTCVideoCodecInfo *preferredCodec;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.h
>-index 56b353215a2..d022297674e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.h
>-@@ -48,6 +48,7 @@ typedef NS_ENUM(NSUInteger, RTCH264Level) {
>- };
>- 
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCH264ProfileLevelId")))
>- @interface RTCH264ProfileLevelId : NSObject
>- 
>- @property(nonatomic, readonly) RTCH264Profile profile;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm
>-index 359656cb97b..28f52139c17 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm
>-@@ -19,20 +19,30 @@
>- #include "media/base/h264_profile_level_id.h"
>- #include "media/base/mediaconstants.h"
>- 
>-+#if !defined(WEBRTC_WEBKIT_BUILD)
>- namespace {
>- 
>- NSString *MaxSupportedProfileLevelConstrainedHigh();
>- NSString *MaxSupportedProfileLevelConstrainedBaseline();
>- 
>- }  // namespace
>-+#endif
>- 
>--NSString *const kRTCVideoCodecH264Name = @(cricket::kH264CodecName);
>-+NSString *const kRTCVideoCodecH264Name = @"H264";
>- NSString *const kRTCLevel31ConstrainedHigh = @"640c1f";
>- NSString *const kRTCLevel31ConstrainedBaseline = @"42e01f";
>-+
>-+#if defined(WEBRTC_WEBKIT_BUILD)
>-+NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh =
>-+    @"640c1f";
>-+NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline =
>-+    @"42e01f";
>-+#else
>- NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh =
>-     MaxSupportedProfileLevelConstrainedHigh();
>- NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline =
>-     MaxSupportedProfileLevelConstrainedBaseline();
>-+#endif
>- 
>- namespace {
>- 
>-@@ -53,6 +63,8 @@ NSString *MaxSupportedLevelForProfile(Profile profile) {
>- }
>- #endif
>- 
>-+#if !defined(WEBRTC_WEBKIT_BUILD)
>-+
>- NSString *MaxSupportedProfileLevelConstrainedBaseline() {
>- #if defined(WEBRTC_IOS)
>-   NSString *profile = MaxSupportedLevelForProfile(webrtc::H264::kProfileConstrainedBaseline);
>-@@ -73,6 +85,8 @@ NSString *MaxSupportedProfileLevelConstrainedHigh() {
>-   return kRTCLevel31ConstrainedHigh;
>- }
>- 
>-+#endif
>-+
>- }  // namespace
>- 
>- @interface RTCH264ProfileLevelId ()
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h
>-index 4fcff1dff79..fdabd807480 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h
>-@@ -14,5 +14,6 @@
>- #import "RTCVideoDecoderFactory.h"
>- 
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactoryH264")))
>- @interface RTCVideoDecoderFactoryH264 : NSObject <RTCVideoDecoderFactory>
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.h
>-index b860276206c..b10d1e6a099 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.h
>-@@ -14,5 +14,6 @@
>- #import "RTCVideoDecoder.h"
>- 
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderH264")))
>- @interface RTCVideoDecoderH264 : NSObject <RTCVideoDecoder>
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h
>-index c64405e4dac..5f6f78dfa9a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h
>-@@ -14,5 +14,6 @@
>- #import "RTCVideoEncoderFactory.h"
>- 
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactoryH264")))
>- @interface RTCVideoEncoderFactoryH264 : NSObject <RTCVideoEncoderFactory>
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.h
>-index a9c05580a41..7714b25eaa4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.h
>-@@ -15,6 +15,7 @@
>- #import "RTCVideoEncoder.h"
>- 
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderH264")))
>- @interface RTCVideoEncoderH264 : NSObject <RTCVideoEncoder>
>- 
>- - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
>-index 16ad1ff47aa..9013ba962fb 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
>-@@ -9,39 +9,72 @@
>-  *
>-  */
>- 
>--#import "RTCVideoEncoderH264.h"
>-+#import "WebRTC/RTCVideoCodecH264.h"
>- 
>- #import <VideoToolbox/VideoToolbox.h>
>- #include <vector>
>- 
>- #if defined(WEBRTC_IOS)
>--#import "helpers/UIDevice+RTCDevice.h"
>-+#import "Common/RTCUIApplicationStatusObserver.h"
>-+#import "WebRTC/UIDevice+RTCDevice.h"
>- #endif
>--#import "RTCCodecSpecificInfoH264.h"
>--#import "RTCH264ProfileLevelId.h"
>--#import "api/peerconnection/RTCRtpFragmentationHeader+Private.h"
>--#import "api/peerconnection/RTCVideoCodecInfo+Private.h"
>--#import "base/RTCCodecSpecificInfo.h"
>--#import "base/RTCI420Buffer.h"
>--#import "base/RTCVideoEncoder.h"
>--#import "base/RTCVideoFrame.h"
>--#import "base/RTCVideoFrameBuffer.h"
>--#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
>--#import "helpers.h"
>--
>-+#import "PeerConnection/RTCVideoCodec+Private.h"
>-+#import "WebRTC/RTCVideoCodec.h"
>-+#import "WebRTC/RTCVideoFrame.h"
>-+#import "WebRTC/RTCVideoFrameBuffer.h"
>- #include "common_video/h264/h264_bitstream_parser.h"
>- #include "common_video/h264/profile_level_id.h"
>- #include "common_video/include/bitrate_adjuster.h"
>-+#import "helpers.h"
>- #include "modules/include/module_common_types.h"
>- #include "modules/video_coding/include/video_error_codes.h"
>- #include "rtc_base/buffer.h"
>- #include "rtc_base/logging.h"
>- #include "rtc_base/timeutils.h"
>--#include "sdk/objc/components/video_codec/nalu_rewriter.h"
>-+#include "sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h"
>-+#include "system_wrappers/include/clock.h"
>- #include "third_party/libyuv/include/libyuv/convert_from.h"
>- 
>--@interface RTCVideoEncoderH264 ()
>-+#include "sdk/WebKit/EncoderUtilities.h"
>-+#include "sdk/WebKit/WebKitUtilities.h"
>-+
>-+#import <dlfcn.h>
>-+#import <objc/runtime.h>
>-+
>-+SOFT_LINK_FRAMEWORK_OPTIONAL(VideoToolBox)
>-+SOFT_LINK_POINTER_OPTIONAL(VideoToolBox, kVTVideoEncoderSpecification_Usage, NSString *)
>-+
>-+#if !ENABLE_VCP_ENCODER && !defined(WEBRTC_IOS)
>-+static inline bool isStandardFrameSize(int32_t width, int32_t height)
>-+{
>-+    // FIXME: Envision relaxing this rule, something like width and height dividable by 4 or 8 should be good enough.
>-+    if (width == 1280)
>-+        return height == 720;
>-+    if (width == 720)
>-+        return height == 1280;
>-+    if (width == 960)
>-+        return height == 540;
>-+    if (width == 540)
>-+        return height == 960;
>-+    if (width == 640)
>-+        return height == 480;
>-+    if (width == 480)
>-+        return height == 640;
>-+    if (width == 288)
>-+        return height == 352;
>-+    if (width == 352)
>-+        return height == 288;
>-+    if (width == 320)
>-+        return height == 240;
>-+    if (width == 240)
>-+        return height == 320;
>-+    return false;
>-+}
>-+#endif
>- 
>-+__attribute__((objc_runtime_name("WK_RTCSingleVideoEncoderH264")))
>-+@interface RTCSingleVideoEncoderH264 : NSObject <RTCVideoEncoder>
>-+- (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo simulcastIndex: (int)index;
>- - (void)frameWasEncoded:(OSStatus)status
>-                   flags:(VTEncodeInfoFlags)infoFlags
>-            sampleBuffer:(CMSampleBufferRef)sampleBuffer
>-@@ -51,7 +84,6 @@ - (void)frameWasEncoded:(OSStatus)status
>-            renderTimeMs:(int64_t)renderTimeMs
>-               timestamp:(uint32_t)timestamp
>-                rotation:(RTCVideoRotation)rotation;
>--
>- @end
>- 
>- namespace {  // anonymous namespace
>-@@ -70,7 +102,7 @@ const OSType kNV12PixelFormat = kCVPixelFormatType_420YpCbCr8BiPlanarFullRange;
>- // Struct that we pass to the encoder per frame to encode. We receive it again
>- // in the encoder callback.
>- struct RTCFrameEncodeParams {
>--  RTCFrameEncodeParams(RTCVideoEncoderH264 *e,
>-+  RTCFrameEncodeParams(RTCSingleVideoEncoderH264 *e,
>-                        RTCCodecSpecificInfoH264 *csi,
>-                        int32_t w,
>-                        int32_t h,
>-@@ -85,7 +117,7 @@ struct RTCFrameEncodeParams {
>-     }
>-   }
>- 
>--  RTCVideoEncoderH264 *encoder;
>-+  RTCSingleVideoEncoderH264 *encoder;
>-   RTCCodecSpecificInfoH264 *codecSpecificInfo;
>-   int32_t width;
>-   int32_t height;
>-@@ -173,8 +205,8 @@ void compressionOutputCallback(void *encoder,
>-                                 rotation:encodeParams->rotation];
>- }
>- 
>--// Extract VideoToolbox profile out of the webrtc::SdpVideoFormat. If there is
>--// no specific VideoToolbox profile for the specified level, AutoLevel will be
>-+// Extract VideoToolbox profile out of the cricket::VideoCodec. If there is no
>-+// specific VideoToolbox profile for the specified level, AutoLevel will be
>- // returned. The user must initialize the encoder with a resolution and
>- // framerate conforming to the selected H264 level regardless.
>- CFStringRef ExtractProfile(webrtc::SdpVideoFormat videoFormat) {
>-@@ -280,7 +312,7 @@ CFStringRef ExtractProfile(webrtc::SdpVideoFormat videoFormat) {
>- }
>- }  // namespace
>- 
>--@implementation RTCVideoEncoderH264 {
>-+@implementation RTCSingleVideoEncoderH264 {
>-   RTCVideoCodecInfo *_codecInfo;
>-   std::unique_ptr<webrtc::BitrateAdjuster> _bitrateAdjuster;
>-   uint32_t _targetBitrateBps;
>-@@ -290,12 +322,14 @@ @implementation RTCVideoEncoderH264 {
>-   RTCVideoEncoderCallback _callback;
>-   int32_t _width;
>-   int32_t _height;
>--  VTCompressionSessionRef _compressionSession;
>--  CVPixelBufferPoolRef _pixelBufferPool;
>-+  CompressionSessionRef _compressionSession;
>-   RTCVideoCodecMode _mode;
>- 
>-   webrtc::H264BitstreamParser _h264BitstreamParser;
>-   std::vector<uint8_t> _frameScaleBuffer;
>-+
>-+  webrtc::VideoCodec _nativeVideoCodec;
>-+  int _simulcastIndex;
>- }
>- 
>- // .5 is set as a mininum to prevent overcompensating for large temporary
>-@@ -305,14 +339,19 @@ @implementation RTCVideoEncoderH264 {
>- // drastically reduced bitrate, so we want to avoid that. In steady state
>- // conditions, 0.95 seems to give us better overall bitrate over long periods
>- // of time.
>--- (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo {
>-+- (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo simulcastIndex:(int)index {
>-   if (self = [super init]) {
>-     _codecInfo = codecInfo;
>-     _bitrateAdjuster.reset(new webrtc::BitrateAdjuster(.5, .95));
>-     _packetizationMode = RTCH264PacketizationModeNonInterleaved;
>-     _profile = ExtractProfile([codecInfo nativeSdpVideoFormat]);
>-+    _simulcastIndex = index;
>-     RTC_LOG(LS_INFO) << "Using profile " << CFStringToString(_profile);
>-     RTC_CHECK([codecInfo.name isEqualToString:kRTCVideoCodecH264Name]);
>-+
>-+#if defined(WEBRTC_IOS)
>-+    [RTCUIApplicationStatusObserver prepareForUse];
>-+#endif
>-   }
>-   return self;
>- }
>-@@ -329,9 +368,12 @@ - (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings
>-   _width = settings.width;
>-   _height = settings.height;
>-   _mode = settings.mode;
>-+  _nativeVideoCodec = settings.nativeVideoCodec;
>-+
>-+  RTC_DCHECK(_nativeVideoCodec.numberOfSimulcastStreams != 1);
>- 
>-   // We can only set average bitrate on the HW encoder.
>--  _targetBitrateBps = settings.startBitrate * 1000;  // startBitrate is in kbps.
>-+  _targetBitrateBps = settings.startBitrate;
>-   _bitrateAdjuster->SetTargetBitrateBps(_targetBitrateBps);
>- 
>-   // TODO(tkchin): Try setting payload size via
>-@@ -348,10 +390,20 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>-   if (!_callback || !_compressionSession) {
>-     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>-   }
>-+#if defined(WEBRTC_IOS)
>-+  if (![[RTCUIApplicationStatusObserver sharedInstance] isApplicationActive]) {
>-+    // Ignore all encode requests when app isn't active. In this state, the
>-+    // hardware encoder has been invalidated by the OS.
>-+    return WEBRTC_VIDEO_CODEC_OK;
>-+  }
>-+#endif
>-   BOOL isKeyframeRequired = NO;
>- 
>-   // Get a pixel buffer from the pool and copy frame data over.
>--  if ([self resetCompressionSessionIfNeededWithFrame:frame]) {
>-+  CVPixelBufferPoolRef pixelBufferPool =
>-+      CompressionSessionGetPixelBufferPool(_compressionSession);
>-+  if ([self resetCompressionSessionIfNeededForPool:pixelBufferPool withFrame:frame]) {
>-+    pixelBufferPool = CompressionSessionGetPixelBufferPool(_compressionSession);
>-     isKeyframeRequired = YES;
>-   }
>- 
>-@@ -368,7 +420,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>-       CVBufferRetain(pixelBuffer);
>-     } else {
>-       // Cropping required, we need to crop and scale to a new pixel buffer.
>--      pixelBuffer = CreatePixelBuffer(_pixelBufferPool);
>-+      pixelBuffer = CreatePixelBuffer(pixelBufferPool);
>-       if (!pixelBuffer) {
>-         return WEBRTC_VIDEO_CODEC_ERROR;
>-       }
>-@@ -391,7 +443,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>- 
>-   if (!pixelBuffer) {
>-     // We did not have a native frame buffer
>--    pixelBuffer = CreatePixelBuffer(_pixelBufferPool);
>-+    pixelBuffer = CreatePixelBuffer(pixelBufferPool);
>-     if (!pixelBuffer) {
>-       return WEBRTC_VIDEO_CODEC_ERROR;
>-     }
>-@@ -434,7 +486,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>-   // Update the bitrate if needed.
>-   [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps()];
>- 
>--  OSStatus status = VTCompressionSessionEncodeFrame(_compressionSession,
>-+  OSStatus status = CompressionSessionEncodeFrame(_compressionSession,
>-                                                     pixelBuffer,
>-                                                     presentationTimeStamp,
>-                                                     kCMTimeInvalid,
>-@@ -447,14 +499,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>-   if (pixelBuffer) {
>-     CVBufferRelease(pixelBuffer);
>-   }
>--
>--  if (status == kVTInvalidSessionErr) {
>--    // This error occurs when entering foreground after backgrounding the app.
>--    RTC_LOG(LS_ERROR) << "Invalid compression session, resetting.";
>--    [self resetCompressionSessionWithPixelFormat:[self pixelFormatOfFrame:frame]];
>--
>--    return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>--  } else if (status != noErr) {
>-+  if (status != noErr) {
>-     RTC_LOG(LS_ERROR) << "Failed to encode frame with code: " << status;
>-     return WEBRTC_VIDEO_CODEC_ERROR;
>-   }
>-@@ -465,8 +510,8 @@ - (void)setCallback:(RTCVideoEncoderCallback)callback {
>-   _callback = callback;
>- }
>- 
>--- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
>--  _targetBitrateBps = 1000 * bitrateKbit;
>-+- (int)setBitrate:(uint32_t)bitrateBps framerate:(uint32_t)framerate {
>-+  _targetBitrateBps = bitrateBps;
>-   _bitrateAdjuster->SetTargetBitrateBps(_targetBitrateBps);
>-   [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps()];
>-   return WEBRTC_VIDEO_CODEC_OK;
>-@@ -483,44 +528,51 @@ - (NSInteger)releaseEncoder {
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>- 
>--- (OSType)pixelFormatOfFrame:(RTCVideoFrame *)frame {
>--  // Use NV12 for non-native frames.
>--  if ([frame.buffer isKindOfClass:[RTCCVPixelBuffer class]]) {
>--    RTCCVPixelBuffer *rtcPixelBuffer = (RTCCVPixelBuffer *)frame.buffer;
>--    return CVPixelBufferGetPixelFormatType(rtcPixelBuffer.pixelBuffer);
>--  }
>--
>--  return kNV12PixelFormat;
>--}
>--
>--- (BOOL)resetCompressionSessionIfNeededWithFrame:(RTCVideoFrame *)frame {
>-+- (BOOL)resetCompressionSessionIfNeededForPool:(CVPixelBufferPoolRef)pixelBufferPool
>-+                                     withFrame:(RTCVideoFrame *)frame {
>-   BOOL resetCompressionSession = NO;
>- 
>-+#if defined(WEBRTC_IOS)
>-+  if (!pixelBufferPool) {
>-+    // Kind of a hack. On backgrounding, the compression session seems to get
>-+    // invalidated, which causes this pool call to fail when the application
>-+    // is foregrounded and frames are being sent for encoding again.
>-+    // Resetting the session when this happens fixes the issue.
>-+    // In addition we request a keyframe so video can recover quickly.
>-+    resetCompressionSession = YES;
>-+    RTC_LOG(LS_INFO) << "Resetting compression session due to invalid pool.";
>-+  }
>-+#endif
>-+
>-   // If we're capturing native frames in another pixel format than the compression session is
>-   // configured with, make sure the compression session is reset using the correct pixel format.
>--  OSType framePixelFormat = [self pixelFormatOfFrame:frame];
>--
>--  if (_compressionSession) {
>-+  // If we're capturing non-native frames and the compression session is configured with a non-NV12
>-+  // format, reset it to NV12.
>-+  OSType framePixelFormat = kNV12PixelFormat;
>-+  if (pixelBufferPool) {
>-     // The pool attribute `kCVPixelBufferPixelFormatTypeKey` can contain either an array of pixel
>-     // formats or a single pixel format.
>-     NSDictionary *poolAttributes =
>--        (__bridge NSDictionary *)CVPixelBufferPoolGetPixelBufferAttributes(_pixelBufferPool);
>-+        (__bridge NSDictionary *)CVPixelBufferPoolGetPixelBufferAttributes(pixelBufferPool);
>-     id pixelFormats =
>-         [poolAttributes objectForKey:(__bridge NSString *)kCVPixelBufferPixelFormatTypeKey];
>-     NSArray<NSNumber *> *compressionSessionPixelFormats = nil;
>-     if ([pixelFormats isKindOfClass:[NSArray class]]) {
>-       compressionSessionPixelFormats = (NSArray *)pixelFormats;
>--    } else if ([pixelFormats isKindOfClass:[NSNumber class]]) {
>-+    } else {
>-       compressionSessionPixelFormats = @[ (NSNumber *)pixelFormats ];
>-     }
>- 
>-+    if ([frame.buffer isKindOfClass:[RTCCVPixelBuffer class]]) {
>-+      RTCCVPixelBuffer *rtcPixelBuffer = (RTCCVPixelBuffer *)frame.buffer;
>-+      framePixelFormat = CVPixelBufferGetPixelFormatType(rtcPixelBuffer.pixelBuffer);
>-+    }
>-+
>-     if (![compressionSessionPixelFormats
>-             containsObject:[NSNumber numberWithLong:framePixelFormat]]) {
>-       resetCompressionSession = YES;
>-       RTC_LOG(LS_INFO) << "Resetting compression session due to non-matching pixel format.";
>-     }
>--  } else {
>--    resetCompressionSession = YES;
>-   }
>- 
>-   if (resetCompressionSession) {
>-@@ -557,22 +609,26 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>-     CFRelease(pixelFormat);
>-     pixelFormat = nullptr;
>-   }
>--  CFMutableDictionaryRef encoder_specs = nullptr;
>--#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
>-+  CFDictionaryRef encoderSpecs = nullptr;
>-+#if !defined(WEBRTC_IOS)
>-+  auto useHardwareEncoder = webrtc::isH264HardwareEncoderAllowed() ? kCFBooleanTrue : kCFBooleanFalse;
>-   // Currently hw accl is supported above 360p on mac, below 360p
>-   // the compression session will be created with hw accl disabled.
>--  encoder_specs = CFDictionaryCreateMutable(
>--      nullptr, 1, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>--  CFDictionarySetValue(encoder_specs,
>--                       kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder,
>--                       kCFBooleanTrue);
>-+  CFTypeRef sessionKeys[] = { kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder, kVTVideoEncoderSpecification_RequireHardwareAcceleratedVideoEncoder, kVTCompressionPropertyKey_RealTime };
>-+  CFTypeRef sessionValues[] = { useHardwareEncoder, useHardwareEncoder, kCFBooleanTrue };
>-+  encoderSpecs = CFDictionaryCreate(kCFAllocatorDefault, sessionKeys, sessionValues, 3, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>-+#else
>-+  CFTypeRef sessionKeys[] = { kVTCompressionPropertyKey_RealTime };
>-+  CFTypeRef sessionValues[] = { kCFBooleanTrue };
>-+  encoderSpecs = CFDictionaryCreate(kCFAllocatorDefault, sessionKeys, sessionValues, 1, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>- #endif
>-+
>-   OSStatus status =
>--      VTCompressionSessionCreate(nullptr,  // use default allocator
>-+      CompressionSessionCreate(nullptr,  // use default allocator
>-                                  _width,
>-                                  _height,
>--                                 kCMVideoCodecType_H264,
>--                                 encoder_specs,  // use hardware accelerated encoder if available
>-+                                 kCodecTypeH264,
>-+                                 encoderSpecs,  // use hardware accelerated encoder if available
>-                                  sourceAttributes,
>-                                  nullptr,  // use default compressed data allocator
>-                                  compressionOutputCallback,
>-@@ -582,32 +638,105 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>-     CFRelease(sourceAttributes);
>-     sourceAttributes = nullptr;
>-   }
>--  if (encoder_specs) {
>--    CFRelease(encoder_specs);
>--    encoder_specs = nullptr;
>-+  if (encoderSpecs) {
>-+    CFRelease(encoderSpecs);
>-+    encoderSpecs = nullptr;
>-   }
>-+
>-+#if ENABLE_VCP_ENCODER || defined(WEBRTC_IOS)
>-   if (status != noErr) {
>-     RTC_LOG(LS_ERROR) << "Failed to create compression session: " << status;
>-     return WEBRTC_VIDEO_CODEC_ERROR;
>-   }
>-+#endif
>- #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
>-   CFBooleanRef hwaccl_enabled = nullptr;
>--  status = VTSessionCopyProperty(_compressionSession,
>-+  if (status == noErr) {
>-+    status = VTSessionCopyProperty(_compressionSession,
>-                                  kVTCompressionPropertyKey_UsingHardwareAcceleratedVideoEncoder,
>-                                  nullptr,
>-                                  &hwaccl_enabled);
>-+  }
>-   if (status == noErr && (CFBooleanGetValue(hwaccl_enabled))) {
>-     RTC_LOG(LS_INFO) << "Compression session created with hw accl enabled";
>-   } else {
>-     RTC_LOG(LS_INFO) << "Compression session created with hw accl disabled";
>-+
>-+#if !ENABLE_VCP_ENCODER && !defined(WEBRTC_IOS)
>-+    if (!isStandardFrameSize(_width, _height)) {
>-+      RTC_LOG(LS_ERROR) << "Using H264 software encoder with non standard size is not supported";
>-+      return WEBRTC_VIDEO_CODEC_ERROR;
>-+    }
>-+
>-+    if (!getkVTVideoEncoderSpecification_Usage()) {
>-+      RTC_LOG(LS_ERROR) << "RTCSingleVideoEncoderH264 cannot create a H264 software encoder";
>-+      return WEBRTC_VIDEO_CODEC_ERROR;
>-+    }
>-+
>-+    CFDictionaryRef ioSurfaceValue = CreateCFTypeDictionary(nullptr, nullptr, 0);
>-+    int64_t pixelFormatType = framePixelFormat;
>-+    CFNumberRef pixelFormat = CFNumberCreate(nullptr, kCFNumberLongType, &pixelFormatType);
>-+
>-+    const size_t attributesSize = 3;
>-+    CFTypeRef keys[attributesSize] = {
>-+      kCVPixelBufferOpenGLCompatibilityKey,
>-+      kCVPixelBufferIOSurfacePropertiesKey,
>-+      kCVPixelBufferPixelFormatTypeKey
>-+    };
>-+    CFTypeRef values[attributesSize] = {
>-+      kCFBooleanTrue,
>-+      ioSurfaceValue,
>-+      pixelFormat};
>-+    CFDictionaryRef sourceAttributes = CreateCFTypeDictionary(keys, values, attributesSize);
>-+
>-+    if (ioSurfaceValue) {
>-+      CFRelease(ioSurfaceValue);
>-+      ioSurfaceValue = nullptr;
>-+    }
>-+    if (pixelFormat) {
>-+      CFRelease(pixelFormat);
>-+      pixelFormat = nullptr;
>-+    }
>-+
>-+    CFMutableDictionaryRef encoderSpecs = CFDictionaryCreateMutable(nullptr, 2, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>-+    CFDictionarySetValue(encoderSpecs, kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder, kCFBooleanFalse);
>-+    int usageValue = 1;
>-+    CFNumberRef usage = CFNumberCreate(nullptr, kCFNumberIntType, &usageValue);
>-+    CFDictionarySetValue(encoderSpecs, (__bridge CFStringRef)getkVTVideoEncoderSpecification_Usage(), usage);
>-+
>-+    if (usage) {
>-+      CFRelease(usage);
>-+      usage = nullptr;
>-+    }
>-+
>-+    [self destroyCompressionSession];
>-+
>-+    OSStatus status =
>-+      CompressionSessionCreate(nullptr,  // use default allocator
>-+                                 _width,
>-+                                 _height,
>-+                                 kCodecTypeH264,
>-+                                 encoderSpecs,
>-+                                 sourceAttributes,
>-+                                 nullptr,  // use default compressed data allocator
>-+                                 compressionOutputCallback,
>-+                                 nullptr,
>-+                                 &_compressionSession);
>-+    if (sourceAttributes) {
>-+      CFRelease(sourceAttributes);
>-+      sourceAttributes = nullptr;
>-+    }
>-+    if (encoderSpecs) {
>-+      CFRelease(encoderSpecs);
>-+      encoderSpecs = nullptr;
>-+    }
>-+    if (status != noErr) {
>-+      return WEBRTC_VIDEO_CODEC_ERROR;
>-+    }
>-+#endif
>-   }
>- #endif
>-   [self configureCompressionSession];
>--
>--  // The pixel buffer pool is dependent on the compression session so if the session is reset, the
>--  // pool should be reset as well.
>--  _pixelBufferPool = VTCompressionSessionGetPixelBufferPool(_compressionSession);
>--
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>- 
>-@@ -616,6 +745,10 @@ - (void)configureCompressionSession {
>-   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_RealTime, true);
>-   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_ProfileLevel, _profile);
>-   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_AllowFrameReordering, false);
>-+#if ENABLE_VCP_ENCODER
>-+  if (auto key = getkVTVideoEncoderSpecification_Usage())
>-+      SetVTSessionProperty(_compressionSession, (__bridge CFStringRef)key, 1);
>-+#endif
>-   [self setEncoderBitrateBps:_targetBitrateBps];
>-   // TODO(tkchin): Look at entropy mode and colorspace matrices.
>-   // TODO(tkchin): Investigate to see if there's any way to make this work.
>-@@ -633,10 +766,9 @@ - (void)configureCompressionSession {
>- 
>- - (void)destroyCompressionSession {
>-   if (_compressionSession) {
>--    VTCompressionSessionInvalidate(_compressionSession);
>-+    CompressionSessionInvalidate(_compressionSession);
>-     CFRelease(_compressionSession);
>-     _compressionSession = nullptr;
>--    _pixelBufferPool = nullptr;
>-   }
>- }
>- 
>-@@ -664,7 +796,7 @@ - (void)setEncoderBitrateBps:(uint32_t)bitrateBps {
>-         CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt64Type, &oneSecondValue);
>-     const void *nums[2] = {bytesPerSecond, oneSecond};
>-     CFArrayRef dataRateLimits = CFArrayCreate(nullptr, nums, 2, &kCFTypeArrayCallBacks);
>--    OSStatus status = VTSessionSetProperty(
>-+    OSStatus status = CompressionSessionSetProperty(
>-         _compressionSession, kVTCompressionPropertyKey_DataRateLimits, dataRateLimits);
>-     if (bytesPerSecond) {
>-       CFRelease(bytesPerSecond);
>-@@ -676,7 +808,7 @@ - (void)setEncoderBitrateBps:(uint32_t)bitrateBps {
>-       CFRelease(dataRateLimits);
>-     }
>-     if (status != noErr) {
>--      RTC_LOG(LS_ERROR) << "Failed to set data rate limit with code: " << status;
>-+      RTC_LOG(LS_ERROR) << "Failed to set data rate limit";
>-     }
>- 
>-     _encoderBitrateBps = bitrateBps;
>-@@ -693,7 +825,7 @@ - (void)frameWasEncoded:(OSStatus)status
>-               timestamp:(uint32_t)timestamp
>-                rotation:(RTCVideoRotation)rotation {
>-   if (status != noErr) {
>--    RTC_LOG(LS_ERROR) << "H264 encode failed with code: " << status;
>-+    RTC_LOG(LS_ERROR) << "H264 encode failed: " << status;
>-     return;
>-   }
>-   if (infoFlags & kVTEncodeInfo_FrameDropped) {
>-@@ -738,6 +870,7 @@ - (void)frameWasEncoded:(OSStatus)status
>-   frame.rotation = rotation;
>-   frame.contentType = (_mode == RTCVideoCodecModeScreensharing) ? RTCVideoContentTypeScreenshare :
>-                                                                   RTCVideoContentTypeUnspecified;
>-+  frame.spatialIndex = _simulcastIndex;
>-   frame.flags = webrtc::VideoSendTiming::kInvalid;
>- 
>-   int qp;
>-@@ -753,9 +886,88 @@ - (void)frameWasEncoded:(OSStatus)status
>-   _bitrateAdjuster->Update(frame.buffer.length);
>- }
>- 
>--- (nullable RTCVideoEncoderQpThresholds *)scalingSettings {
>--  return [[RTCVideoEncoderQpThresholds alloc] initWithThresholdsLow:kLowH264QpThreshold
>--                                                               high:kHighH264QpThreshold];
>-+- (RTCVideoEncoderQpThresholds *)scalingSettings {
>-+  return [[RTCVideoEncoderQpThresholds alloc] initWithThresholdsLow:kLowH264QpThreshold high:kHighH264QpThreshold];
>-+}
>-+
>-+- (int)setRateAllocation:(RTCVideoBitrateAllocation *)allocation framerate:(uint32_t)framerate {
>-+  return 0;
>-+}
>-+
>-+@end
>-+
>-+@implementation RTCVideoEncoderH264 {
>-+  NSMutableArray<RTCSingleVideoEncoderH264*> *_codecs;
>-+  RTCVideoCodecInfo *_codecInfo;
>-+}
>-+- (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo {
>-+  if (self = [super init]) {
>-+    _codecInfo = codecInfo;
>-+  }
>-+  return self;
>-+}
>-+
>-+- (void)setCallback:(RTCVideoEncoderCallback)callback {
>-+  for (RTCSingleVideoEncoderH264 *codec : _codecs)
>-+    [codec setCallback:callback];
>-+}
>-+
>-+- (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings numberOfCores:(int)numberOfCores {
>-+    auto nativeCodecSettings = settings.nativeVideoCodec;
>-+
>-+  _codecs = [[NSMutableArray alloc] init];
>-+  for (unsigned index = 0 ; index < nativeCodecSettings.numberOfSimulcastStreams; ++index) {
>-+    auto codec = [[RTCSingleVideoEncoderH264 alloc] initWithCodecInfo:_codecInfo simulcastIndex:index];
>-+    [_codecs addObject:codec];
>-+
>-+    auto codecSettings = nativeCodecSettings;
>-+    codecSettings.width = nativeCodecSettings.simulcastStream[index].width;
>-+    codecSettings.height = nativeCodecSettings.simulcastStream[index].height;
>-+    codecSettings.maxBitrate = nativeCodecSettings.simulcastStream[index].maxBitrate;
>-+    codecSettings.targetBitrate = nativeCodecSettings.simulcastStream[index].targetBitrate;
>-+    codecSettings.minBitrate = nativeCodecSettings.simulcastStream[index].minBitrate;
>-+    codecSettings.qpMax = nativeCodecSettings.simulcastStream[index].qpMax;
>-+    codecSettings.active = true;
>-+
>-+    auto *settings = [[RTCVideoEncoderSettings alloc] initWithNativeVideoCodec:&codecSettings];
>-+    [codec startEncodeWithSettings:settings numberOfCores:numberOfCores];
>-+  }
>-+  return 0;
>-+}
>-+
>-+- (NSInteger)releaseEncoder {
>-+  for (RTCSingleVideoEncoderH264 *codec : _codecs)
>-+    [codec releaseEncoder];
>-+  _codecs = nil;
>-+  return 0;
>-+}
>-+
>-+- (NSInteger)encode:(RTCVideoFrame *)frame codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info frameTypes:(NSArray<NSNumber *> *)frameTypes {
>-+  int result = 0;
>-+  for (RTCSingleVideoEncoderH264 *codec : _codecs)
>-+    result |= [codec encode:frame codecSpecificInfo:info frameTypes:frameTypes];
>-+  return result;
>-+}
>-+
>-+- (int)setRateAllocation:(RTCVideoBitrateAllocation *)bitRateAllocation framerate:(uint32_t) framerate {
>-+  int result = 0;
>-+  unsigned counter = 0;
>-+  auto nativeBitRateAllocation = bitRateAllocation.nativeVideoBitrateAllocation;
>-+  for (RTCSingleVideoEncoderH264 *codec : _codecs)
>-+    result |= [codec setBitrate:nativeBitRateAllocation.GetSpatialLayerSum(counter++) framerate:framerate];
>-+  return result;
>-+}
>-+
>-+- (NSString *)implementationName {
>-+  return @"VideoToolbox";
>-+}
>-+
>-+- (RTCVideoEncoderQpThresholds *)scalingSettings {
>-+  return [[RTCVideoEncoderQpThresholds alloc] initWithThresholdsLow:kLowH264QpThreshold high:kHighH264QpThreshold];
>-+}
>-+
>-+- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
>-+  return 0;
>- }
>- 
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.cc
>-index ac957f1b497..46daf44ecac 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.cc
>-@@ -35,12 +35,12 @@ std::string CFStringToString(const CFStringRef cf_string) {
>- }
>- 
>- // Convenience function for setting a VT property.
>--void SetVTSessionProperty(VTSessionRef session,
>-+void SetVTSessionProperty(CompressionSessionRef session,
>-                           CFStringRef key,
>-                           int32_t value) {
>-   CFNumberRef cfNum =
>-       CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt32Type, &value);
>--  OSStatus status = VTSessionSetProperty(session, key, cfNum);
>-+  OSStatus status = CompressionSessionSetProperty(session, key, cfNum);
>-   CFRelease(cfNum);
>-   if (status != noErr) {
>-     std::string key_string = CFStringToString(key);
>-@@ -50,13 +50,13 @@ void SetVTSessionProperty(VTSessionRef session,
>- }
>- 
>- // Convenience function for setting a VT property.
>--void SetVTSessionProperty(VTSessionRef session,
>-+void SetVTSessionProperty(CompressionSessionRef session,
>-                           CFStringRef key,
>-                           uint32_t value) {
>-   int64_t value_64 = value;
>-   CFNumberRef cfNum =
>-       CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt64Type, &value_64);
>--  OSStatus status = VTSessionSetProperty(session, key, cfNum);
>-+  OSStatus status = CompressionSessionSetProperty(session, key, cfNum);
>-   CFRelease(cfNum);
>-   if (status != noErr) {
>-     std::string key_string = CFStringToString(key);
>-@@ -66,9 +66,9 @@ void SetVTSessionProperty(VTSessionRef session,
>- }
>- 
>- // Convenience function for setting a VT property.
>--void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value) {
>-+void SetVTSessionProperty(CompressionSessionRef session, CFStringRef key, bool value) {
>-   CFBooleanRef cf_bool = (value) ? kCFBooleanTrue : kCFBooleanFalse;
>--  OSStatus status = VTSessionSetProperty(session, key, cf_bool);
>-+  OSStatus status = CompressionSessionSetProperty(session, key, cf_bool);
>-   if (status != noErr) {
>-     std::string key_string = CFStringToString(key);
>-     RTC_LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
>-@@ -77,10 +77,10 @@ void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value) {
>- }
>- 
>- // Convenience function for setting a VT property.
>--void SetVTSessionProperty(VTSessionRef session,
>-+void SetVTSessionProperty(CompressionSessionRef session,
>-                           CFStringRef key,
>-                           CFStringRef value) {
>--  OSStatus status = VTSessionSetProperty(session, key, value);
>-+  OSStatus status = CompressionSessionSetProperty(session, key, value);
>-   if (status != noErr) {
>-     std::string key_string = CFStringToString(key);
>-     std::string val_string = CFStringToString(value);
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.h
>-index 0683ea79e56..0c669e8bce6 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.h
>-@@ -16,6 +16,8 @@
>- #include <VideoToolbox/VideoToolbox.h>
>- #include <string>
>- 
>-+#include "sdk/WebKit/EncoderUtilities.h"
>-+
>- // Convenience function for creating a dictionary.
>- inline CFDictionaryRef CreateCFTypeDictionary(CFTypeRef* keys,
>-                                               CFTypeRef* values,
>-@@ -29,18 +31,18 @@ inline CFDictionaryRef CreateCFTypeDictionary(CFTypeRef* keys,
>- std::string CFStringToString(const CFStringRef cf_string);
>- 
>- // Convenience function for setting a VT property.
>--void SetVTSessionProperty(VTSessionRef session, CFStringRef key, int32_t value);
>-+void SetVTSessionProperty(CompressionSessionRef session, CFStringRef key, int32_t value);
>- 
>- // Convenience function for setting a VT property.
>--void SetVTSessionProperty(VTSessionRef session,
>-+void SetVTSessionProperty(CompressionSessionRef session,
>-                           CFStringRef key,
>-                           uint32_t value);
>- 
>- // Convenience function for setting a VT property.
>--void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value);
>-+void SetVTSessionProperty(CompressionSessionRef session, CFStringRef key, bool value);
>- 
>- // Convenience function for setting a VT property.
>--void SetVTSessionProperty(VTSessionRef session,
>-+void SetVTSessionProperty(CompressionSessionRef session,
>-                           CFStringRef key,
>-                           CFStringRef value);
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h
>-index 432a3825746..abe9dfca93b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h
>-@@ -17,6 +17,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** RTCVideoFrameBuffer containing a CVPixelBufferRef */
>- RTC_OBJC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCCVPixelBuffer")))
>- @interface RTCCVPixelBuffer : NSObject <RTCVideoFrameBuffer>
>- 
>- @property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm
>-index ba7c3e96754..c4c85cc6ab9 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm
>-@@ -92,6 +92,12 @@ class ObjCVideoEncoder : public VideoEncoder {
>-     return [encoder_ setBitrate:bitrate framerate:framerate];
>-   }
>- 
>-+  int32_t SetRateAllocation(const VideoBitrateAllocation& allocation, uint32_t framerate) {
>-+    RTCVideoBitrateAllocation *bitrateAllocation =
>-+      [[RTCVideoBitrateAllocation alloc] initWithNativeVideoBitrateAllocation:&allocation];
>-+    return [encoder_ setRateAllocation: bitrateAllocation framerate:framerate];
>-+  }
>-+
>-   bool SupportsNativeHandle() const { return true; }
>- 
>-   VideoEncoder::ScalingSettings GetScalingSettings() const {
>--- 
>-2.17.1 (Apple Git-112)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/001-GTK-changes.patch b/Source/ThirdParty/libwebrtc/WebKit/001-GTK-changes.patch
>new file mode 100644
>index 0000000000000000000000000000000000000000..2c4e16555c8df6cb151200e9ab2a84da46a88d0a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/WebKit/001-GTK-changes.patch
>@@ -0,0 +1,36 @@
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>+index 44ec3f50c19..58b3345f9db 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>+@@ -75,7 +75,12 @@ CriticalSection& logCriticalScope() {
>+ void LogSink::OnLogMessage(const std::string& msg,
>+                            LoggingSeverity severity,
>+                            const char* tag) {
>+-  OnLogMessage(tag + (": " + msg));
>++  OnLogMessage(tag + (": " + msg), severity);
>++}
>++
>++void LogSink::OnLogMessage(const std::string& msg,
>++                           LoggingSeverity /* severity */) {
>++  OnLogMessage(msg);
>+ }
>+ 
>+ /////////////////////////////////////////////////////////////////////////////
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>+index 905bbdac0e2..27b8801bb0d 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>+@@ -22,7 +22,13 @@
>+ #include <type_traits>
>+ #include <utility>
>+ 
>++#if defined(WEBRTC_LINUX)
>++#include <event2/event.h>
>++#include <event2/event_compat.h>
>++#include <event2/event_struct.h>
>++#else
>+ #include "base/third_party/libevent/event.h"
>++#endif
>+ #include "rtc_base/checks.h"
>+ #include "rtc_base/criticalsection.h"
>+ #include "rtc_base/logging.h"
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/libwebrtc.diff b/Source/ThirdParty/libwebrtc/WebKit/libwebrtc.diff
>new file mode 100644
>index 0000000000000000000000000000000000000000..92428e415dca86fc0744ecd9813eecf0d2c077e6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/WebKit/libwebrtc.diff
>@@ -0,0 +1,2568 @@
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>+index 896ed238cc3..4d028c9aaf4 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>+@@ -33,7 +33,7 @@ int GetIlbcBitrate(int ptime) {
>+       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
>+       return 13333;
>+     default:
>+-      FATAL();
>++      RTC_FATAL();
>+   }
>+ }
>+ }  // namespace
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc
>+index 599542db08f..140d0ae9625 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc
>+@@ -28,7 +28,7 @@ std::string MediaTypeToString(MediaType type) {
>+     case MEDIA_TYPE_DATA:
>+       return kMediaTypeData;
>+   }
>+-  FATAL();
>++  RTC_FATAL();
>+   // Not reachable; avoids compile warning.
>+   return "";
>+ }
>+@@ -41,7 +41,7 @@ MediaType MediaTypeFromString(const std::string& type_str) {
>+   } else if (type_str == kMediaTypeData) {
>+     return MEDIA_TYPE_DATA;
>+   }
>+-  FATAL();
>++  RTC_FATAL();
>+   // Not reachable; avoids compile warning.
>+   return static_cast<MediaType>(-1);
>+ }
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>+index cc59e2a823c..b1158da213f 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>+@@ -55,7 +55,7 @@ void RemixAndResample(const int16_t* src_data,
>+ 
>+   if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
>+                                     audio_ptr_num_channels) == -1) {
>+-    FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
>++    RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
>+             << ", dst_frame->sample_rate_hz_ = " << dst_frame->sample_rate_hz_
>+             << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
>+   }
>+@@ -69,7 +69,7 @@ void RemixAndResample(const int16_t* src_data,
>+       resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(),
>+                           AudioFrame::kMaxDataSizeSamples);
>+   if (out_length == -1) {
>+-    FATAL() << "Resample failed: audio_ptr = " << audio_ptr
>++    RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
>+             << ", src_length = " << src_length
>+             << ", dst_frame->mutable_data() = " << dst_frame->mutable_data();
>+   }
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>+index 1b8bcbd923c..90a89281e5c 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>+@@ -63,7 +63,7 @@ WavReader::WavReader(rtc::PlatformFile file) {
>+     if (!rtc::ClosePlatformFile(file)) {
>+       RTC_LOG(LS_ERROR) << "Can't close file.";
>+     }
>+-    FATAL() << "Could not open wav file for reading.";
>++    RTC_FATAL() << "Could not open wav file for reading.";
>+   }
>+ 
>+   ReadableWavFile readable(file_handle_);
>+@@ -148,7 +148,7 @@ WavWriter::WavWriter(rtc::PlatformFile file,
>+     if (!rtc::ClosePlatformFile(file)) {
>+       RTC_LOG(LS_ERROR) << "Can't close file.";
>+     }
>+-    FATAL() << "Could not open wav file for writing.";
>++    RTC_FATAL() << "Could not open wav file for writing.";
>+   }
>+ 
>+   RTC_CHECK(CheckWavParameters(num_channels_, sample_rate_, kWavFormat,
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>+index f4f138fccb5..5f89b4acb96 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>+@@ -272,7 +272,7 @@ rtc::scoped_refptr<PlanarYuvBuffer> WrapYuvBuffer(
>+       return WrapI444Buffer(width, height, y_plane, y_stride, u_plane, u_stride,
>+                             v_plane, v_stride, no_longer_used);
>+     default:
>+-      FATAL() << "Unexpected frame buffer type.";
>++      RTC_FATAL() << "Unexpected frame buffer type.";
>+       return nullptr;
>+   }
>+ }
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>+index 9a2926143d0..476954fe60d 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>+@@ -171,7 +171,7 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl(
>+       break;
>+     }
>+     case Vad::kError: {
>+-      FATAL();  // Fails only if fed invalid data.
>++      RTC_FATAL();  // Fails only if fed invalid data.
>+       break;
>+     }
>+   }
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>+index 8801fd55590..35e6fc64a62 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>+@@ -41,7 +41,7 @@ int GetIlbcBitrate(int ptime) {
>+       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
>+       return 13333;
>+     default:
>+-      FATAL();
>++      RTC_FATAL();
>+   }
>+ }
>+ 
>+@@ -148,7 +148,7 @@ size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
>+     case 6:
>+       return 2 * 50;
>+     default:
>+-      FATAL();
>++      RTC_FATAL();
>+   }
>+ }
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc
>+new file mode 100644
>+index 00000000000..b058c2d6728
>+--- /dev/null
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc
>+@@ -0,0 +1,43 @@
>++/*
>++ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>++ *
>++ *  Use of this source code is governed by a BSD-style license
>++ *  that can be found in the LICENSE file in the root of the source
>++ *  tree. An additional intellectual property rights grant can be found
>++ *  in the file PATENTS.  All contributing project authors may
>++ *  be found in the AUTHORS file in the root of the source tree.
>++ *
>++ */
>++
>++#if !defined(RTC_DISABLE_VP9)
>++#error
>++#endif  // !defined(RTC_DISABLE_VP9)
>++
>++#include "modules/video_coding/codecs/vp9/include/vp9.h"
>++
>++#include "api/video_codecs/sdp_video_format.h"
>++#include "rtc_base/checks.h"
>++
>++namespace webrtc {
>++
>++std::vector<SdpVideoFormat> SupportedVP9Codecs() {
>++  return std::vector<SdpVideoFormat>();
>++}
>++
>++std::unique_ptr<VP9Encoder> VP9Encoder::Create() {
>++  RTC_NOTREACHED();
>++  return nullptr;
>++}
>++
>++std::unique_ptr<VP9Encoder> VP9Encoder::Create(
>++    const cricket::VideoCodec& codec) {
>++  RTC_NOTREACHED();
>++  return nullptr;
>++}
>++
>++std::unique_ptr<VP9Decoder> VP9Decoder::Create() {
>++  RTC_NOTREACHED();
>++  return nullptr;
>++}
>++
>++}  // namespace webrtc
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>+index e018ea755d6..0d7aea9d60a 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>+@@ -351,11 +351,21 @@ TCPConnection::TCPConnection(TCPPort* port,
>+                         << ", port() Network:" << port->Network()->ToString();
>+     const std::vector<rtc::InterfaceAddress>& desired_addresses =
>+         port_->Network()->GetIPs();
>++
>++#if defined(WEBRTC_WEBKIT_BUILD)
>++     RTC_DCHECK(socket->GetLocalAddress().IsLoopbackIP() ||
>++                (std::find_if(desired_addresses.begin(), desired_addresses.end(),
>++                             [this](const rtc::InterfaceAddress& addr) {
>++                               return socket_->GetLocalAddress().ipaddr() ==
>++                                      addr;
>++                             }) != desired_addresses.end()));
>++ #else
>+     RTC_DCHECK(std::find_if(desired_addresses.begin(), desired_addresses.end(),
>+                             [this](const rtc::InterfaceAddress& addr) {
>+                               return socket_->GetLocalAddress().ipaddr() ==
>+                                      addr;
>+                             }) != desired_addresses.end());
>++#endif
>+     ConnectSocketSignals(socket);
>+   }
>+ }
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>+index 37c6a0b0bcf..75261508b27 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>+@@ -261,7 +261,7 @@ RtpCapabilities PeerConnectionFactory::GetRtpSenderCapabilities(
>+       return RtpCapabilities();
>+   }
>+   // Not reached; avoids compile warning.
>+-  FATAL();
>++  RTC_FATAL();
>+ }
>+ 
>+ RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities(
>+@@ -288,7 +288,7 @@ RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities(
>+       return RtpCapabilities();
>+   }
>+   // Not reached; avoids compile warning.
>+-  FATAL();
>++  RTC_FATAL();
>+ }
>+ 
>+ rtc::scoped_refptr<AudioSourceInterface>
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>+index bf3bce3daa3..865685d73c4 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>+@@ -61,7 +61,7 @@ RTCErrorOr<cricket::FeedbackParam> ToCricketFeedbackParam(
>+       return cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc);
>+   }
>+   // Not reached; avoids compile warning.
>+-  FATAL();
>++  RTC_FATAL();
>+ }
>+ 
>+ template <typename C>
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>+index 9de6d47573c..5cf0679910e 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>+@@ -80,7 +80,7 @@ RTC_NORETURN void rtc_FatalMessage(const char* file, int line, const char* msg);
>+ //   messages if the condition doesn't hold. Prefer them to raw RTC_CHECK and
>+ //   RTC_DCHECK.
>+ //
>+-// - FATAL() aborts unconditionally.
>++// - RTC_FATAL() aborts unconditionally.
>+ //
>+ // TODO(ajm): Ideally, checks.h would be combined with logging.h, but
>+ // consolidation with system_wrappers/logging.h should happen first.
>+@@ -348,9 +348,9 @@ class FatalLogCall final {
>+ #define RTC_NOTREACHED() RTC_DCHECK(RTC_UNREACHABLE_CODE_HIT)
>+ 
>+ // TODO(bugs.webrtc.org/8454): Add an RTC_ prefix or rename differently.
>+-#define FATAL()                                                    \
>++#define RTC_FATAL()                                                    \
>+   rtc::webrtc_checks_impl::FatalLogCall<false>(__FILE__, __LINE__, \
>+-                                               "FATAL()") &        \
>++                                               "RTC_FATAL()") &        \
>+       rtc::webrtc_checks_impl::LogStreamer<>()
>+ 
>+ // Performs the integer division a/b and returns the result. CHECKs that the
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>+index bcce0dafbc0..6b43b91e026 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>+@@ -81,7 +81,7 @@ void Flag::SetToDefault() {
>+       variable_->s = default_.s;
>+       return;
>+   }
>+-  FATAL() << "unreachable code";
>++  RTC_FATAL() << "unreachable code";
>+ }
>+ 
>+ static const char* Type2String(Flag::Type type) {
>+@@ -95,7 +95,7 @@ static const char* Type2String(Flag::Type type) {
>+     case Flag::STRING:
>+       return "string";
>+   }
>+-  FATAL() << "unreachable code";
>++  RTC_FATAL() << "unreachable code";
>+ }
>+ 
>+ static void PrintFlagValue(Flag::Type type, FlagValue* p) {
>+@@ -113,7 +113,7 @@ static void PrintFlagValue(Flag::Type type, FlagValue* p) {
>+       printf("%s", p->s);
>+       return;
>+   }
>+-  FATAL() << "unreachable code";
>++  RTC_FATAL() << "unreachable code";
>+ }
>+ 
>+ void Flag::Print(bool print_current_value) {
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h
>+index 513bc263651..718d9589348 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h
>+@@ -50,7 +50,7 @@ class Location {
>+ #define RTC_FROM_HERE RTC_FROM_HERE_WITH_FUNCTION(__FUNCTION__)
>+ 
>+ #define RTC_FROM_HERE_WITH_FUNCTION(function_name) \
>+-  ::rtc::Location(function_name, __FILE__ ":" STRINGIZE(__LINE__))
>++  ::rtc::Location(function_name, __FILE__ ":" RTC_STRINGIZE(__LINE__))
>+ 
>+ }  // namespace rtc
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>+index bb4fbfaae25..44ec3f50c19 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>+@@ -20,28 +20,26 @@
>+ #include <CoreServices/CoreServices.h>
>+ #elif defined(WEBRTC_ANDROID)
>+ #include <android/log.h>
>+-
>+ // Android has a 1024 limit on log inputs. We use 60 chars as an
>+ // approx for the header/tag portion.
>+ // See android/system/core/liblog/logd_write.c
>+ static const int kMaxLogLineSize = 1024 - 60;
>+ #endif  // WEBRTC_MAC && !defined(WEBRTC_IOS) || WEBRTC_ANDROID
>+ 
>+-#include <stdio.h>
>+-#include <string.h>
>++#include <limits.h>
>+ #include <time.h>
>++
>+ #include <algorithm>
>+ #include <cstdarg>
>+ #include <vector>
>+ 
>+-#include "rtc_base/checks.h"
>+ #include "rtc_base/criticalsection.h"
>+ #include "rtc_base/logging.h"
>+ #include "rtc_base/platform_thread_types.h"
>++#include "rtc_base/never_destroyed.h"
>+ #include "rtc_base/stringencode.h"
>+ #include "rtc_base/strings/string_builder.h"
>+ #include "rtc_base/stringutils.h"
>+-#include "rtc_base/thread_annotations.h"
>+ #include "rtc_base/timeutils.h"
>+ 
>+ namespace rtc {
>+@@ -65,20 +63,19 @@ const char* FilenameFromPath(const char* file) {
>+     return (end1 > end2) ? end1 + 1 : end2 + 1;
>+ }
>+ 
>+-// Global lock for log subsystem, only needed to serialize access to streams_.
>+-CriticalSection g_log_crit;
>++// Global lock for log subsystem, only needed to serialize access to streamList().
>++CriticalSection& logCriticalScope() {
>++  static auto scope = makeNeverDestroyed<>(CriticalSection { });
>++  return scope.get();
>++}
>++
>+ }  // namespace
>+ 
>+ // Inefficient default implementation, override is recommended.
>+ void LogSink::OnLogMessage(const std::string& msg,
>+                            LoggingSeverity severity,
>+                            const char* tag) {
>+-  OnLogMessage(tag + (": " + msg), severity);
>+-}
>+-
>+-void LogSink::OnLogMessage(const std::string& msg,
>+-                           LoggingSeverity /* severity */) {
>+-  OnLogMessage(msg);
>++  OnLogMessage(tag + (": " + msg));
>+ }
>+ 
>+ /////////////////////////////////////////////////////////////////////////////
>+@@ -91,7 +88,15 @@ bool LogMessage::log_to_stderr_ = true;
>+ // Note: we explicitly do not clean this up, because of the uncertain ordering
>+ // of destructors at program exit.  Let the person who sets the stream trigger
>+ // cleanup by setting to null, or let it leak (safe at program exit).
>+-LogMessage::StreamList LogMessage::streams_ RTC_GUARDED_BY(g_log_crit);
>++typedef std::pair<LogSink*, LoggingSeverity> StreamAndSeverity;
>++typedef std::list<StreamAndSeverity> StreamList;
>++
>++// The output streams and their associated severities
>++StreamList& streamList()
>++    RTC_EXCLUSIVE_LOCKS_REQUIRED(logCriticalScope()) {
>++        static auto stream_list = makeNeverDestroyed<>(StreamList { });
>++  return stream_list.get();
>++}
>+ 
>+ // Boolean options default to false (0)
>+ bool LogMessage::thread_, LogMessage::timestamp_;
>+@@ -105,6 +110,7 @@ LogMessage::LogMessage(const char* file,
>+                        LogErrorContext err_ctx,
>+                        int err)
>+     : severity_(sev) {
>++
>+   if (timestamp_) {
>+     // Use SystemTimeMillis so that even if tests use fake clocks, the timestamp
>+     // in log messages represents the real system time.
>+@@ -205,13 +211,13 @@ LogMessage::~LogMessage() {
>+ #endif
>+   }
>+ 
>+-  CritScope cs(&g_log_crit);
>+-  for (auto& kv : streams_) {
>++  CritScope cs(&logCriticalScope());
>++  for (auto& kv : streamList()) {
>+     if (severity_ >= kv.second) {
>+ #if defined(WEBRTC_ANDROID)
>+       kv.first->OnLogMessage(str, severity_, tag_);
>+ #else
>+-      kv.first->OnLogMessage(str, severity_);
>++      kv.first->OnLogMessage(str);
>+ #endif
>+     }
>+   }
>+@@ -254,7 +260,7 @@ void LogMessage::LogTimestamps(bool on) {
>+ 
>+ void LogMessage::LogToDebug(LoggingSeverity min_sev) {
>+   g_dbg_sev = min_sev;
>+-  CritScope cs(&g_log_crit);
>++  CritScope cs(&logCriticalScope());
>+   UpdateMinLogSeverity();
>+ }
>+ 
>+@@ -263,9 +269,9 @@ void LogMessage::SetLogToStderr(bool log_to_stderr) {
>+ }
>+ 
>+ int LogMessage::GetLogToStream(LogSink* stream) {
>+-  CritScope cs(&g_log_crit);
>++  CritScope cs(&logCriticalScope());
>+   LoggingSeverity sev = LS_NONE;
>+-  for (auto& kv : streams_) {
>++  for (auto& kv : streamList()) {
>+     if (!stream || stream == kv.first) {
>+       sev = std::min(sev, kv.second);
>+     }
>+@@ -274,16 +280,16 @@ int LogMessage::GetLogToStream(LogSink* stream) {
>+ }
>+ 
>+ void LogMessage::AddLogToStream(LogSink* stream, LoggingSeverity min_sev) {
>+-  CritScope cs(&g_log_crit);
>+-  streams_.push_back(std::make_pair(stream, min_sev));
>++  CritScope cs(&logCriticalScope());
>++  streamList().push_back(std::make_pair(stream, min_sev));
>+   UpdateMinLogSeverity();
>+ }
>+ 
>+ void LogMessage::RemoveLogToStream(LogSink* stream) {
>+-  CritScope cs(&g_log_crit);
>+-  for (StreamList::iterator it = streams_.begin(); it != streams_.end(); ++it) {
>++  CritScope cs(&logCriticalScope());
>++  for (StreamList::iterator it = streamList().begin(); it != streamList().end(); ++it) {
>+     if (stream == it->first) {
>+-      streams_.erase(it);
>++      streamList().erase(it);
>+       break;
>+     }
>+   }
>+@@ -342,9 +348,9 @@ void LogMessage::ConfigureLogging(const char* params) {
>+ }
>+ 
>+ void LogMessage::UpdateMinLogSeverity()
>+-    RTC_EXCLUSIVE_LOCKS_REQUIRED(g_log_crit) {
>++    RTC_EXCLUSIVE_LOCKS_REQUIRED(logCriticalScope()) {
>+   LoggingSeverity min_sev = g_dbg_sev;
>+-  for (const auto& kv : streams_) {
>++  for (const auto& kv : streamList()) {
>+     const LoggingSeverity sev = kv.second;
>+     min_sev = std::min(min_sev, sev);
>+   }
>+@@ -458,11 +464,8 @@ bool LogMessage::IsNoop(LoggingSeverity severity) {
>+   // TODO(tommi): We're grabbing this lock for every LogMessage instance that
>+   // is going to be logged. This introduces unnecessary synchronization for
>+   // a feature that's mostly used for testing.
>+-  CritScope cs(&g_log_crit);
>+-  if (streams_.size() > 0)
>+-    return false;
>+-
>+-  return true;
>++  CritScope cs(&logCriticalScope());
>++  return streamList().size() == 0;
>+ }
>+ 
>+ void LogMessage::FinishPrintStream() {
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/never_destroyed.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/never_destroyed.h
>+new file mode 100644
>+index 00000000000..fcc62e35534
>+--- /dev/null
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/never_destroyed.h
>+@@ -0,0 +1,69 @@
>++/*
>++ * Copyright (C) 2018 Apple Inc. All rights reserved.
>++ *
>++ * Redistribution and use in source and binary forms, with or without
>++ * modification, are permitted provided that the following conditions
>++ * are met:
>++ * 1. Redistributions of source code must retain the above copyright
>++ *    notice, this list of conditions and the following disclaimer.
>++ * 2. Redistributions in binary form must reproduce the above copyright
>++ *    notice, this list of conditions and the following disclaimer in the
>++ *    documentation and/or other materials provided with the distribution.
>++ *
>++ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
>++ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
>++ * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
>++ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
>++ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
>++ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
>++ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
>++ * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
>++ * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
>++ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
>++ * THE POSSIBILITY OF SUCH DAMAGE.
>++ */
>++
>++#pragma once
>++
>++#include <type_traits>
>++#include <utility>
>++
>++namespace rtc {
>++
>++template<typename T> class NeverDestroyed {
>++public:
>++    template<typename... Args> NeverDestroyed(Args&&... args)
>++    {
>++        new (storagePointer()) T(std::forward<Args>(args)...);
>++    }
>++
>++    NeverDestroyed(NeverDestroyed&& other)
>++    {
>++        new (storagePointer()) T(std::move(*other.storagePointer()));
>++    }
>++
>++    operator T&() { return *storagePointer(); }
>++    T& get() { return *storagePointer(); }
>++
>++    operator const T&() const { return *storagePointer(); }
>++    const T& get() const { return *storagePointer(); }
>++
>++private:
>++    NeverDestroyed(const NeverDestroyed&) = delete;
>++    NeverDestroyed& operator=(const NeverDestroyed&) = delete;
>++
>++    using PointerType = typename std::remove_const<T>::type*;
>++
>++    PointerType storagePointer() const { return const_cast<PointerType>(reinterpret_cast<const T*>(&m_storage)); }
>++
>++    // FIXME: Investigate whether we should allocate a hunk of virtual memory
>++    // and hand out chunks of it to NeverDestroyed instead, to reduce fragmentation.
>++    typename std::aligned_storage<sizeof(T), std::alignment_of<T>::value>::type m_storage;
>++};
>++
>++template<typename T> inline NeverDestroyed<T> makeNeverDestroyed(T&& argument)
>++{
>++    return NeverDestroyed<T>(std::move(argument));
>++}
>++
>++} // namespace rtc
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h
>+index 58efcaa746a..48c212e4d49 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h
>+@@ -63,11 +63,11 @@ inline Dst saturated_cast(Src value) {
>+ 
>+     // Should fail only on attempting to assign NaN to a saturated integer.
>+     case internal::TYPE_INVALID:
>+-      FATAL();
>++      RTC_FATAL();
>+       return std::numeric_limits<Dst>::max();
>+   }
>+ 
>+-  FATAL();
>++  RTC_FATAL();
>+   return static_cast<Dst>(value);
>+ }
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc
>+index 4e61b86f3ba..4e325dcec01 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc
>+@@ -244,11 +244,11 @@ std::unique_ptr<SSLCertificate> OpenSSLCertificate::Clone() const {
>+ std::string OpenSSLCertificate::ToPEMString() const {
>+   BIO* bio = BIO_new(BIO_s_mem());
>+   if (!bio) {
>+-    FATAL() << "Unreachable code.";
>++    RTC_FATAL() << "Unreachable code.";
>+   }
>+   if (!PEM_write_bio_X509(bio, x509_)) {
>+     BIO_free(bio);
>+-    FATAL() << "Unreachable code.";
>++    RTC_FATAL() << "Unreachable code.";
>+   }
>+   BIO_write(bio, "\0", 1);
>+   char* buffer;
>+@@ -264,11 +264,11 @@ void OpenSSLCertificate::ToDER(Buffer* der_buffer) const {
>+   // Calculates the DER representation of the certificate, from scratch.
>+   BIO* bio = BIO_new(BIO_s_mem());
>+   if (!bio) {
>+-    FATAL() << "Unreachable code.";
>++    RTC_FATAL() << "Unreachable code.";
>+   }
>+   if (!i2d_X509_bio(bio, x509_)) {
>+     BIO_free(bio);
>+-    FATAL() << "Unreachable code.";
>++    RTC_FATAL() << "Unreachable code.";
>+   }
>+   char* data = nullptr;
>+   size_t length = BIO_get_mem_data(bio, &data);
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>+index 4ad2857c00c..7448a7e0427 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>+@@ -596,7 +596,7 @@ bool SocketDispatcher::Initialize() {
>+ #elif defined(WEBRTC_POSIX)
>+   fcntl(s_, F_SETFL, fcntl(s_, F_GETFL, 0) | O_NONBLOCK);
>+ #endif
>+-#if defined(WEBRTC_IOS)
>++#if defined(WEBRTC_IOS) || (defined(WEBRTC_MAC) && defined(WEBRTC_WEBKIT_BUILD))
>+   // iOS may kill sockets when the app is moved to the background
>+   // (specifically, if the app doesn't use the "voip" UIBackgroundMode). When
>+   // we attempt to write to such a socket, SIGPIPE will be raised, which by
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>+index aee8d14551d..38c6f1836bd 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>+@@ -8,7 +8,7 @@
>+  *  be found in the AUTHORS file in the root of the source tree.
>+  */
>+ 
>+-// Modified from the Chromium original:
>++// Modified from the Chxsromium original:
>+ // src/base/strings/stringize_macros.h
>+ 
>+ // This file defines preprocessor macros for stringizing preprocessor
>+@@ -33,6 +33,6 @@
>+ // Then:
>+ //   STRINGIZE(A) produces "FOO"
>+ //   STRINGIZE(B(y)) produces "myobj->FunctionCall(y)"
>+-#define STRINGIZE(x) STRINGIZE_NO_EXPANSION(x)
>++#define RTC_STRINGIZE(x) STRINGIZE_NO_EXPANSION(x)
>+ 
>+ #endif  // RTC_BASE_STRINGIZE_MACROS_H_
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>+index 79694112c81..bfce0f54541 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>+@@ -1026,9 +1026,9 @@ void VirtualSocketServer::UpdateDelayDistribution() {
>+   }
>+ }
>+ 
>+-static double PI = 4 * atan(1.0);
>+ 
>+ static double Normal(double x, double mean, double stddev) {
>++  static double PI = 4 * atan(1.0);
>+   double a = (x - mean) * (x - mean) / (2 * stddev * stddev);
>+   return exp(-a) / (stddev * sqrt(2 * PI));
>+ }
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/EncoderUtilities.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/EncoderUtilities.h
>+new file mode 100644
>+index 00000000000..5b9bdfda114
>+--- /dev/null
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/EncoderUtilities.h
>+@@ -0,0 +1,50 @@
>++/*
>++ * Copyright (C) 2018 Apple Inc. All rights reserved.
>++ *
>++ * Redistribution and use in source and binary forms, with or without
>++ * modification, are permitted provided that the following conditions
>++ * are met:
>++ * 1. Redistributions of source code must retain the above copyright
>++ *    notice, this list of conditions and the following disclaimer.
>++ * 2. Redistributions in binary form must reproduce the above copyright
>++ *    notice, this list of conditions and the following disclaimer in the
>++ *    documentation and/or other materials provided with the distribution.
>++ *
>++ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
>++ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
>++ * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
>++ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
>++ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
>++ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
>++ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
>++ * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
>++ * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
>++ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
>++ * THE POSSIBILITY OF SUCH DAMAGE.
>++ */
>++
>++#pragma once
>++
>++#include "VideoProcessingSoftLink.h"
>++
>++#if ENABLE_VCP_ENCODER
>++
>++#define CompressionSessionRef VCPCompressionSessionRef
>++#define CompressionSessionSetProperty webrtc::VCPCompressionSessionSetProperty
>++#define CompressionSessionGetPixelBufferPool webrtc::VCPCompressionSessionGetPixelBufferPool
>++#define CompressionSessionEncodeFrame webrtc::VCPCompressionSessionEncodeFrame
>++#define CompressionSessionCreate webrtc::VCPCompressionSessionCreate
>++#define kCodecTypeH264 kVCPCodecType4CC_H264
>++#define CompressionSessionInvalidate webrtc::VCPCompressionSessionInvalidate
>++
>++#else
>++
>++#define CompressionSessionRef VTCompressionSessionRef
>++#define CompressionSessionSetProperty VTSessionSetProperty
>++#define CompressionSessionGetPixelBufferPool VTCompressionSessionGetPixelBufferPool
>++#define CompressionSessionEncodeFrame VTCompressionSessionEncodeFrame
>++#define CompressionSessionCreate VTCompressionSessionCreate
>++#define kCodecTypeH264 kCMVideoCodecType_H264
>++#define CompressionSessionInvalidate VTCompressionSessionInvalidate
>++
>++#endif
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.cpp b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.cpp
>+new file mode 100644
>+index 00000000000..101e87c6677
>+--- /dev/null
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.cpp
>+@@ -0,0 +1,78 @@
>++/*
>++ * Copyright (C) 2018 Apple Inc. All rights reserved.
>++ *
>++ * Redistribution and use in source and binary forms, with or without
>++ * modification, are permitted provided that the following conditions
>++ * are met:
>++ * 1. Redistributions of source code must retain the above copyright
>++ *    notice, this list of conditions and the following disclaimer.
>++ * 2. Redistributions in binary form must reproduce the above copyright
>++ *    notice, this list of conditions and the following disclaimer in the
>++ *    documentation and/or other materials provided with the distribution.
>++ *
>++ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
>++ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
>++ * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
>++ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
>++ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
>++ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
>++ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
>++ * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
>++ * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
>++ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
>++ * THE POSSIBILITY OF SUCH DAMAGE.
>++ */
>++
>++#include "VideoProcessingSoftLink.h"
>++
>++#if ENABLE_VCP_ENCODER
>++
>++#include "rtc_base/logging.h"
>++#import <dlfcn.h>
>++#import <objc/runtime.h>
>++
>++// Macros copied from <wtf/cocoa/SoftLinking.h>
>++#define SOFT_LINK_PRIVATE_FRAMEWORK_FOR_SOURCE(functionNamespace, framework) \
>++    namespace functionNamespace { \
>++    void* framework##Library(bool isOptional) \
>++    { \
>++        static void* frameworkLibrary; \
>++        static dispatch_once_t once; \
>++        dispatch_once(&once, ^{ \
>++            frameworkLibrary = dlopen("/System/Library/PrivateFrameworks/" #framework ".framework/" #framework, RTLD_NOW); \
>++            if (!isOptional && !frameworkLibrary) \
>++                RTC_LOG(LS_ERROR) << "Cannot open framework: " << dlerror(); \
>++        }); \
>++        return frameworkLibrary; \
>++    } \
>++    }
>++
>++#define SOFT_LINK_FUNCTION_FOR_SOURCE(functionNamespace, framework, functionName, resultType, parameterDeclarations, parameterNames) \
>++    extern "C" { \
>++    resultType functionName parameterDeclarations; \
>++    } \
>++    namespace functionNamespace { \
>++    static resultType init##framework##functionName parameterDeclarations; \
>++    resultType (*softLink##framework##functionName) parameterDeclarations = init##framework##functionName; \
>++    static resultType init##framework##functionName parameterDeclarations \
>++    { \
>++        static dispatch_once_t once; \
>++        dispatch_once(&once, ^{ \
>++            softLink##framework##functionName = (resultType (*) parameterDeclarations) dlsym(framework##Library(), #functionName); \
>++            if (!softLink##framework##functionName) \
>++                RTC_LOG(LS_ERROR) << "Cannot find function ##functionName: " << dlerror(); \
>++        }); \
>++        return softLink##framework##functionName parameterNames; \
>++    } \
>++}
>++
>++SOFT_LINK_PRIVATE_FRAMEWORK_FOR_SOURCE(webrtc, VideoProcessing)
>++
>++SOFT_LINK_FUNCTION_FOR_SOURCE(webrtc, VideoProcessing, VCPCompressionSessionSetProperty, OSStatus, (VCPCompressionSessionRef session, CFStringRef key, CFTypeRef value), (session, key, value))
>++SOFT_LINK_FUNCTION_FOR_SOURCE(webrtc, VideoProcessing, VCPCompressionSessionGetPixelBufferPool, CVPixelBufferPoolRef, (VCPCompressionSessionRef session), (session))
>++SOFT_LINK_FUNCTION_FOR_SOURCE(webrtc, VideoProcessing, VCPCompressionSessionEncodeFrame, OSStatus, (VCPCompressionSessionRef session, CVImageBufferRef buffer, CMTime timestamp, CMTime time, CFDictionaryRef dictionary, void* data, VTEncodeInfoFlags* flags), (session, buffer, timestamp, time, dictionary, data, flags))
>++SOFT_LINK_FUNCTION_FOR_SOURCE(webrtc, VideoProcessing, VCPCompressionSessionCreate, OSStatus, (CFAllocatorRef allocator1, int32_t value1 , int32_t value2, CMVideoCodecType type, CFDictionaryRef dictionary1, CFDictionaryRef dictionary2, CFAllocatorRef allocator3, VTCompressionOutputCallback callback, void* data, VCPCompressionSessionRef* session), (allocator1, value1, value2, type, dictionary1, dictionary2, allocator3, callback, data, session))
>++SOFT_LINK_FUNCTION_FOR_SOURCE(webrtc, VideoProcessing, VCPCompressionSessionInvalidate, void, (VCPCompressionSessionRef session), (session))
>++SOFT_LINK_FUNCTION_FOR_SOURCE(webrtc, VideoProcessing, VPModuleInitialize, void, (), ())
>++
>++#endif // ENABLE_VCP_ENCODER
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h
>+new file mode 100644
>+index 00000000000..cf866d8a8db
>+--- /dev/null
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h
>+@@ -0,0 +1,140 @@
>++/*
>++ * Copyright (C) 2018 Apple Inc. All rights reserved.
>++ *
>++ * Redistribution and use in source and binary forms, with or without
>++ * modification, are permitted provided that the following conditions
>++ * are met:
>++ * 1. Redistributions of source code must retain the above copyright
>++ *    notice, this list of conditions and the following disclaimer.
>++ * 2. Redistributions in binary form must reproduce the above copyright
>++ *    notice, this list of conditions and the following disclaimer in the
>++ *    documentation and/or other materials provided with the distribution.
>++ *
>++ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
>++ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
>++ * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
>++ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
>++ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
>++ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
>++ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
>++ * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
>++ * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
>++ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
>++ * THE POSSIBILITY OF SUCH DAMAGE.
>++ */
>++
>++#pragma once
>++
>++#ifdef __APPLE__
>++#include <Availability.h>
>++#include <AvailabilityMacros.h>
>++#include <TargetConditionals.h>
>++
>++// Macro taken from WTF/wtf/Platform.h
>++#if defined __has_include && __has_include(<CoreFoundation/CFPriv.h>)
>++
>++#if (defined(TARGET_IPHONE_SIMULATOR)  && TARGET_IPHONE_SIMULATOR)
>++#define ENABLE_VCP_ENCODER 0
>++#elif (defined(TARGET_OS_IPHONE)  && TARGET_OS_IPHONE)
>++#define ENABLE_VCP_ENCODER 1
>++#elif (defined(TARGET_OS_MAC) && TARGET_OS_MAC)
>++#define ENABLE_VCP_ENCODER (__MAC_OS_X_VERSION_MIN_REQUIRED >= 101300 && __MAC_OS_X_VERSION_MAX_ALLOWED >= 101304)
>++#endif
>++
>++#endif
>++
>++#if !defined(ENABLE_VCP_ENCODER)
>++#define ENABLE_VCP_ENCODER 0
>++#endif
>++
>++#if !defined(ALWAYS_INLINE)
>++#define ALWAYS_INLINE inline
>++#endif
>++
>++#ifdef __cplusplus
>++#define WTF_EXTERN_C_BEGIN extern "C" {
>++#define WTF_EXTERN_C_END }
>++#else
>++#define WTF_EXTERN_C_BEGIN
>++#define WTF_EXTERN_C_END
>++#endif
>++
>++// Macros copied from <wtf/cocoa/SoftLinking.h>
>++#define SOFT_LINK_FRAMEWORK_FOR_HEADER(functionNamespace, framework) \
>++    namespace functionNamespace { \
>++    extern void* framework##Library(bool isOptional = false); \
>++    bool is##framework##FrameworkAvailable(); \
>++    inline bool is##framework##FrameworkAvailable() { \
>++        return framework##Library(true) != nullptr; \
>++    } \
>++    }
>++
>++#define SOFT_LINK_FUNCTION_FOR_HEADER(functionNamespace, framework, functionName, resultType, parameterDeclarations, parameterNames) \
>++    WTF_EXTERN_C_BEGIN \
>++    resultType functionName parameterDeclarations; \
>++    WTF_EXTERN_C_END \
>++    namespace functionNamespace { \
>++    extern resultType (*softLink##framework##functionName) parameterDeclarations; \
>++    inline resultType softLink_##framework##_##functionName parameterDeclarations \
>++    { \
>++        return softLink##framework##functionName parameterNames; \
>++    } \
>++    } \
>++    ALWAYS_INLINE resultType functionName parameterDeclarations \
>++    {\
>++        return functionNamespace::softLink##framework##functionName parameterNames; \
>++    }
>++
>++#define SOFT_LINK_FRAMEWORK_OPTIONAL(framework) \
>++    static void* framework##Library() \
>++    { \
>++        static void* frameworkLibrary = dlopen("/System/Library/Frameworks/" #framework ".framework/" #framework, RTLD_NOW); \
>++        return frameworkLibrary; \
>++    }
>++
>++#define SOFT_LINK_POINTER_OPTIONAL(framework, name, type) \
>++    static type init##name(); \
>++    static type (*get##name)() = init##name; \
>++    static type pointer##name; \
>++    \
>++    static type name##Function() \
>++    { \
>++        return pointer##name; \
>++    } \
>++    \
>++    static type init##name() \
>++    { \
>++        void** pointer = static_cast<void**>(dlsym(framework##Library(), #name)); \
>++        if (pointer) \
>++            pointer##name = (__bridge type)(*pointer); \
>++        get##name = name##Function; \
>++        return pointer##name; \
>++    }
>++
>++#if ENABLE_VCP_ENCODER
>++
>++#include <VideoProcessing/VideoProcessing.h>
>++
>++SOFT_LINK_FRAMEWORK_FOR_HEADER(webrtc, VideoProcessing)
>++
>++SOFT_LINK_FUNCTION_FOR_HEADER(webrtc, VideoProcessing, VCPCompressionSessionSetProperty, OSStatus, (VCPCompressionSessionRef session, CFStringRef key, CFTypeRef value), (session, key, value))
>++#define VCPCompressionSessionSetProperty softLink_VideoProcessing_VCPCompressionSessionSetProperty
>++
>++SOFT_LINK_FUNCTION_FOR_HEADER(webrtc, VideoProcessing, VCPCompressionSessionGetPixelBufferPool, CVPixelBufferPoolRef, (VCPCompressionSessionRef session), (session))
>++#define VCPCompressionSessionGetPixelBufferPool softLink_VideoProcessing_VCPCompressionSessionGetPixelBufferPool
>++
>++SOFT_LINK_FUNCTION_FOR_HEADER(webrtc, VideoProcessing, VCPCompressionSessionEncodeFrame, OSStatus, (VCPCompressionSessionRef session, CVImageBufferRef buffer, CMTime timestamp, CMTime time, CFDictionaryRef dictionary, void * data, VTEncodeInfoFlags * flags), (session, buffer, timestamp, time, dictionary, data, flags))
>++#define VCPCompressionSessionEncodeFrame softLink_VideoProcessing_VCPCompressionSessionEncodeFrame
>++
>++SOFT_LINK_FUNCTION_FOR_HEADER(webrtc, VideoProcessing, VCPCompressionSessionCreate, OSStatus, (CFAllocatorRef allocator1, int32_t value1 , int32_t value2, CMVideoCodecType type, CFDictionaryRef dictionary1, CFDictionaryRef dictionary2, CFAllocatorRef allocator3, VTCompressionOutputCallback callback, void * data, VCPCompressionSessionRef *session), (allocator1, value1, value2, type, dictionary1, dictionary2, allocator3, callback, data, session))
>++#define VCPCompressionSessionCreate softLink_VideoProcessing_VCPCompressionSessionCreate
>++
>++SOFT_LINK_FUNCTION_FOR_HEADER(webrtc, VideoProcessing, VCPCompressionSessionInvalidate, void, (VCPCompressionSessionRef session), (session))
>++#define VCPCompressionSessionInvalidate softLink_VideoProcessing_VCPCompressionSessionInvalidate
>++
>++SOFT_LINK_FUNCTION_FOR_HEADER(webrtc, VideoProcessing, VPModuleInitialize, void, (), ())
>++#define VPModuleInitialize softLink_VideoProcessing_VPModuleInitialize
>++
>++#endif // ENABLE_VCP_ENCODER
>++
>++#endif // __APPLE__
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.h
>+new file mode 100644
>+index 00000000000..ef1456eddd3
>+--- /dev/null
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.h
>+@@ -0,0 +1,54 @@
>++/*
>++ * Copyright (C) 2018 Apple Inc. All rights reserved.
>++ *
>++ * Redistribution and use in source and binary forms, with or without
>++ * modification, are permitted provided that the following conditions
>++ * are met:
>++ * 1. Redistributions of source code must retain the above copyright
>++ *    notice, this list of conditions and the following disclaimer.
>++ * 2. Redistributions in binary form must reproduce the above copyright
>++ *    notice, this list of conditions and the following disclaimer in the
>++ *    documentation and/or other materials provided with the distribution.
>++ *
>++ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
>++ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
>++ * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
>++ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
>++ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
>++ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
>++ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
>++ * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
>++ * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
>++ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
>++ * THE POSSIBILITY OF SUCH DAMAGE.
>++ */
>++
>++#pragma once
>++
>++#include "api/video/video_frame_buffer.h"
>++#include "rtc_base/scoped_ref_ptr.h"
>++#include "webrtc/media/engine/webrtcvideodecoderfactory.h"
>++#include "webrtc/media/engine/webrtcvideoencoderfactory.h"
>++
>++typedef struct __CVBuffer* CVPixelBufferRef;
>++
>++namespace webrtc {
>++
>++class VideoDecoderFactory;
>++class VideoEncoderFactory;
>++class VideoFrame;
>++
>++enum class WebKitCodecSupport { H264, H264AndVP8 };
>++
>++std::unique_ptr<webrtc::VideoEncoderFactory> createWebKitEncoderFactory(WebKitCodecSupport);
>++std::unique_ptr<webrtc::VideoDecoderFactory> createWebKitDecoderFactory(WebKitCodecSupport);
>++
>++void setApplicationStatus(bool isActive);
>++
>++void setH264HardwareEncoderAllowed(bool);
>++bool isH264HardwareEncoderAllowed();
>++
>++CVPixelBufferRef pixelBufferFromFrame(const VideoFrame&, const std::function<CVPixelBufferRef(size_t, size_t)>&);
>++rtc::scoped_refptr<webrtc::VideoFrameBuffer> pixelBufferToFrame(CVPixelBufferRef);
>++
>++}
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>+new file mode 100644
>+index 00000000000..65eefa36f2d
>+--- /dev/null
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>+@@ -0,0 +1,185 @@
>++/*
>++ * Copyright (C) 2018 Apple Inc. All rights reserved.
>++ *
>++ * Redistribution and use in source and binary forms, with or without
>++ * modification, are permitted provided that the following conditions
>++ * are met:
>++ * 1. Redistributions of source code must retain the above copyright
>++ *    notice, this list of conditions and the following disclaimer.
>++ * 2. Redistributions in binary form must reproduce the above copyright
>++ *    notice, this list of conditions and the following disclaimer in the
>++ *    documentation and/or other materials provided with the distribution.
>++ *
>++ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
>++ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
>++ * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
>++ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
>++ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
>++ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
>++ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
>++ * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
>++ * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
>++ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
>++ * THE POSSIBILITY OF SUCH DAMAGE.
>++ */
>++
>++#include "WebKitUtilities.h"
>++
>++//#include "Common/RTCUIApplicationStatusObserver.h"
>++#import "WebRTC/RTCVideoCodecH264.h"
>++
>++#include "api/video/video_frame.h"
>++#include "native/src/objc_frame_buffer.h"
>++#include "third_party/libyuv/include/libyuv/convert_from.h"
>++#include "webrtc/rtc_base/checks.h"
>++#include "Framework/Headers/WebRTC/RTCVideoCodecFactory.h"
>++#include "Framework/Headers/WebRTC/RTCVideoFrame.h"
>++#include "Framework/Headers/WebRTC/RTCVideoFrameBuffer.h"
>++#include "Framework/Native/api/video_decoder_factory.h"
>++#include "Framework/Native/api/video_encoder_factory.h"
>++/*
>++#if !defined(WEBRTC_IOS)
>++__attribute__((objc_runtime_name("WK_RTCUIApplicationStatusObserver")))
>++@interface RTCUIApplicationStatusObserver : NSObject
>++
>+++ (instancetype)sharedInstance;
>+++ (void)prepareForUse;
>++
>++- (BOOL)isApplicationActive;
>++
>++@end
>++#endif
>++
>++@implementation RTCUIApplicationStatusObserver {
>++    BOOL _isActive;
>++}
>++
>+++ (instancetype)sharedInstance {
>++    static id sharedInstance;
>++    static dispatch_once_t onceToken;
>++    dispatch_once(&onceToken, ^{
>++        sharedInstance = [[self alloc] init];
>++    });
>++
>++    return sharedInstance;
>++}
>++
>+++ (void)prepareForUse {
>++    __unused RTCUIApplicationStatusObserver *observer = [self sharedInstance];
>++}
>++
>++- (id)init {
>++    _isActive = YES;
>++    return self;
>++}
>++
>++- (void)setActive {
>++    _isActive = YES;
>++}
>++
>++- (void)setInactive {
>++    _isActive = NO;
>++}
>++
>++- (BOOL)isApplicationActive {
>++    return _isActive;
>++}
>++
>++@end
>++*/
>++namespace webrtc {
>++
>++void setApplicationStatus(bool isActive)
>++{
>++/*
>++    if (isActive)
>++        [[RTCUIApplicationStatusObserver sharedInstance] setActive];
>++    else
>++        [[RTCUIApplicationStatusObserver sharedInstance] setInactive];
>++ */
>++}
>++
>++std::unique_ptr<webrtc::VideoEncoderFactory> createWebKitEncoderFactory(WebKitCodecSupport codecSupport)
>++{
>++#if ENABLE_VCP_ENCODER
>++    static std::once_flag onceFlag;
>++    std::call_once(onceFlag, [] {
>++        webrtc::VPModuleInitialize();
>++    });
>++#endif
>++    return ObjCToNativeVideoEncoderFactory(codecSupport == WebKitCodecSupport::H264AndVP8 ? [[RTCDefaultVideoEncoderFactory alloc] init] : [[RTCVideoEncoderFactoryH264 alloc] init]);
>++}
>++
>++std::unique_ptr<webrtc::VideoDecoderFactory> createWebKitDecoderFactory(WebKitCodecSupport codecSupport)
>++{
>++    return ObjCToNativeVideoDecoderFactory(codecSupport == WebKitCodecSupport::H264AndVP8 ? [[RTCDefaultVideoDecoderFactory alloc] init] : [[RTCVideoDecoderFactoryH264 alloc] init]);
>++}
>++
>++static bool h264HardwareEncoderAllowed = true;
>++void setH264HardwareEncoderAllowed(bool allowed)
>++{
>++    h264HardwareEncoderAllowed = allowed;
>++}
>++
>++bool isH264HardwareEncoderAllowed()
>++{
>++    return h264HardwareEncoderAllowed;
>++}
>++
>++rtc::scoped_refptr<webrtc::VideoFrameBuffer> pixelBufferToFrame(CVPixelBufferRef pixelBuffer)
>++{
>++    RTCCVPixelBuffer *frameBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer];
>++    return new rtc::RefCountedObject<ObjCFrameBuffer>(frameBuffer);
>++}
>++
>++static bool CopyVideoFrameToPixelBuffer(const rtc::scoped_refptr<webrtc::I420BufferInterface>& frame, CVPixelBufferRef pixel_buffer) {
>++    RTC_DCHECK(pixel_buffer);
>++    RTC_DCHECK(CVPixelBufferGetPixelFormatType(pixel_buffer) == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange || CVPixelBufferGetPixelFormatType(pixel_buffer) == kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange);
>++    RTC_DCHECK_EQ(CVPixelBufferGetHeightOfPlane(pixel_buffer, 0), static_cast<size_t>(frame->height()));
>++    RTC_DCHECK_EQ(CVPixelBufferGetWidthOfPlane(pixel_buffer, 0), static_cast<size_t>(frame->width()));
>++
>++    if (CVPixelBufferLockBaseAddress(pixel_buffer, 0) != kCVReturnSuccess)
>++        return false;
>++
>++    uint8_t* dst_y = reinterpret_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(pixel_buffer, 0));
>++    int dst_stride_y = CVPixelBufferGetBytesPerRowOfPlane(pixel_buffer, 0);
>++
>++    uint8_t* dst_uv = reinterpret_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(pixel_buffer, 1));
>++    int dst_stride_uv = CVPixelBufferGetBytesPerRowOfPlane(pixel_buffer, 1);
>++
>++    int result = libyuv::I420ToNV12(
>++        frame->DataY(), frame->StrideY(),
>++        frame->DataU(), frame->StrideU(),
>++        frame->DataV(), frame->StrideV(),
>++        dst_y, dst_stride_y, dst_uv, dst_stride_uv,
>++        frame->width(), frame->height());
>++
>++    CVPixelBufferUnlockBaseAddress(pixel_buffer, 0);
>++
>++    if (result)
>++        return false;
>++
>++    return true;
>++}
>++
>++
>++CVPixelBufferRef pixelBufferFromFrame(const VideoFrame& frame, const std::function<CVPixelBufferRef(size_t, size_t)>& makePixelBuffer)
>++{
>++    if (frame.video_frame_buffer()->type() != VideoFrameBuffer::Type::kNative) {
>++        rtc::scoped_refptr<const I420BufferInterface> buffer = frame.video_frame_buffer()->GetI420();
>++
>++        auto pixelBuffer = makePixelBuffer(buffer->width(), buffer->height());
>++        if (pixelBuffer)
>++            CopyVideoFrameToPixelBuffer(frame.video_frame_buffer()->GetI420(), pixelBuffer);
>++        return pixelBuffer;
>++    }
>++
>++    auto *frameBuffer = static_cast<ObjCFrameBuffer*>(frame.video_frame_buffer().get())->wrapped_frame_buffer();
>++    if (![frameBuffer isKindOfClass:[RTCCVPixelBuffer class]])
>++        return nullptr;
>++
>++    auto *rtcPixelBuffer = (RTCCVPixelBuffer *)frameBuffer;
>++    return rtcPixelBuffer.pixelBuffer;
>++}
>++
>++}
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
>+index 6f2d1f46d9c..f9901bd327e 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
>+@@ -56,6 +56,7 @@ - (webrtc::EncodedImage)nativeEncodedImage {
>+   encodedImage.rotation_ = webrtc::VideoRotation(self.rotation);
>+   encodedImage._completeFrame = self.completeFrame;
>+   encodedImage.qp_ = self.qp ? self.qp.intValue : -1;
>++  encodedImage.SetSpatialIndex(self.spatialIndex);
>+   encodedImage.content_type_ = (self.contentType == RTCVideoContentTypeScreenshare) ?
>+       webrtc::VideoContentType::SCREENSHARE :
>+       webrtc::VideoContentType::UNSPECIFIED;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h
>+index ba50bde649f..dabff725abb 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h
>+@@ -15,6 +15,7 @@
>+ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCRtpEncodingParameters")))
>+ @interface RTCRtpEncodingParameters : NSObject
>+ 
>+ /** Controls whether the encoding is currently transmitted. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h
>+index 5b062455bc0..72af732b23b 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h
>+@@ -22,4 +22,11 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ @end
>+ 
>++@interface RTCVideoBitrateAllocation (Private)
>++
>++- (instancetype)initWithNativeVideoBitrateAllocation:(const webrtc::VideoBitrateAllocation *__nullable)videoBitrateAllocation;
>++- (webrtc::VideoBitrateAllocation)nativeVideoBitrateAllocation;
>++
>++@end
>++
>+ NS_ASSUME_NONNULL_END
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm
>+index 6fb81dbb8af..82b5ba7bc0f 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm
>+@@ -12,6 +12,23 @@
>+ 
>+ #import "helpers/NSString+StdString.h"
>+ 
>++@implementation RTCVideoEncoderSettings {
>++  webrtc::VideoCodec _nativeVideoCodec;
>++}
>++
>++@synthesize name = _name;
>++@synthesize width = _width;
>++@synthesize height = _height;
>++@synthesize startBitrate = _startBitrate;
>++@synthesize maxBitrate = _maxBitrate;
>++@synthesize minBitrate = _minBitrate;
>++@synthesize targetBitrate = _targetBitrate;
>++@synthesize maxFramerate = _maxFramerate;
>++@synthesize qpMax = _qpMax;
>++@synthesize mode = _mode;
>++
>++@end
>++
>+ @implementation RTCVideoEncoderSettings (Private)
>+ 
>+ - (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *)videoCodec {
>+@@ -20,6 +37,7 @@ - (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *)videoCodec
>+       const char *codecName = CodecTypeToPayloadString(videoCodec->codecType);
>+       self.name = [NSString stringWithUTF8String:codecName];
>+ 
>++      _nativeVideoCodec = *videoCodec;
>+       self.width = videoCodec->width;
>+       self.height = videoCodec->height;
>+       self.startBitrate = videoCodec->startBitrate;
>+@@ -36,18 +54,28 @@ - (instancetype)initWithNativeVideoCodec:(const webrtc::VideoCodec *)videoCodec
>+ }
>+ 
>+ - (webrtc::VideoCodec)nativeVideoCodec {
>+-  webrtc::VideoCodec videoCodec;
>+-  videoCodec.width = self.width;
>+-  videoCodec.height = self.height;
>+-  videoCodec.startBitrate = self.startBitrate;
>+-  videoCodec.maxBitrate = self.maxBitrate;
>+-  videoCodec.minBitrate = self.minBitrate;
>+-  videoCodec.targetBitrate = self.targetBitrate;
>+-  videoCodec.maxBitrate = self.maxBitrate;
>+-  videoCodec.qpMax = self.qpMax;
>+-  videoCodec.mode = (webrtc::VideoCodecMode)self.mode;
>+-
>+-  return videoCodec;
>++  return _nativeVideoCodec;
>++}
>++
>++@end
>++
>++@implementation RTCVideoBitrateAllocation {
>++  webrtc::VideoBitrateAllocation _nativeVideoBitrateAllocation;
>++}
>++
>++@end
>++
>++@implementation RTCVideoBitrateAllocation (Private)
>++
>++- (instancetype)initWithNativeVideoBitrateAllocation:(const webrtc::VideoBitrateAllocation *)videoBitrateAllocation {
>++  if (self = [super init]) {
>++    _nativeVideoBitrateAllocation = *videoBitrateAllocation;
>++  }
>++  return self;
>++}
>++
>++- (webrtc::VideoBitrateAllocation)nativeVideoBitrateAllocation {
>++  return _nativeVideoBitrateAllocation;
>+ }
>+ 
>+ @end
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.mm
>+index acbf126170c..1447d3b27e1 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.mm
>+@@ -13,5 +13,5 @@
>+ 
>+ #include "media/base/mediaconstants.h"
>+ 
>+-NSString *const kRTCVideoCodecVp8Name = @(cricket::kVp8CodecName);
>+-NSString *const kRTCVideoCodecVp9Name = @(cricket::kVp9CodecName);
>++NSString *const kRTCVideoCodecVp8Name = @"VP8";
>++NSString *const kRTCVideoCodecVp9Name = @"VP9";
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.h
>+index 8d87a898931..08d18504222 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.h
>+@@ -14,6 +14,7 @@
>+ #import "RTCVideoEncoder.h"
>+ 
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoEncoderVP8")))
>+ @interface RTCVideoEncoderVP8 : NSObject
>+ 
>+ /* This returns a VP8 encoder that can be returned from a RTCVideoEncoderFactory injected into
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm
>+index 9afd54f55f5..b9634779f1f 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm
>+@@ -58,6 +58,11 @@ - (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
>+   return 0;
>+ }
>+ 
>++- (int)setRateAllocation:(nonnull RTCVideoBitrateAllocation *)allocation framerate:(uint32_t)framerate {
>++  RTC_NOTREACHED();
>++  return 0;
>++}
>++
>+ - (NSString *)implementationName {
>+   RTC_NOTREACHED();
>+   return nil;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h
>+index 9a904f5396a..dc903bc4bca 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h
>+@@ -17,6 +17,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ /** RTCI420Buffer implements the RTCI420Buffer protocol */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
>+ @interface RTCI420Buffer : NSObject<RTCI420Buffer>
>+ @end
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h
>+index 6cd5110460b..7cfa950abb6 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h
>+@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ /** Mutable version of RTCI420Buffer */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCMutableI420Buffer")))
>+ @interface RTCMutableI420Buffer : RTCI420Buffer<RTCMutableI420Buffer>
>+ @end
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCCodecSpecificInfo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCCodecSpecificInfo.h
>+index e2ae4cafa11..9dc44c12af8 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCCodecSpecificInfo.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCCodecSpecificInfo.h
>+@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>+  *  Corresponds to webrtc::CodecSpecificInfo.
>+  */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCCodecSpecificInfo")))
>+ @protocol RTCCodecSpecificInfo <NSObject>
>+ @end
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.h
>+index 670c7276ff7..7994ce64a1f 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.h
>+@@ -31,6 +31,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoContentType) {
>+ 
>+ /** Represents an encoded frame. Corresponds to webrtc::EncodedImage. */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCEncodedImage")))
>+ @interface RTCEncodedImage : NSObject
>+ 
>+ @property(nonatomic, strong) NSData *buffer;
>+@@ -47,6 +48,7 @@ RTC_OBJC_EXPORT
>+ @property(nonatomic, assign) BOOL completeFrame;
>+ @property(nonatomic, strong) NSNumber *qp;
>+ @property(nonatomic, assign) RTCVideoContentType contentType;
>++@property(nonatomic, assign) int spatialIndex;
>+ 
>+ @end
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.m
>+index 024a57c541b..85cd5e298e4 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.m
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCEncodedImage.m
>+@@ -26,5 +26,6 @@ @implementation RTCEncodedImage
>+ @synthesize completeFrame = _completeFrame;
>+ @synthesize qp = _qp;
>+ @synthesize contentType = _contentType;
>++@synthesize spatialIndex = _spatialIndex;
>+ 
>+ @end
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCI420Buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCI420Buffer.h
>+index a6c7e41bcba..cfa14f8af23 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCI420Buffer.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCI420Buffer.h
>+@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ /** Protocol for RTCYUVPlanarBuffers containing I420 data */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
>+ @protocol RTCI420Buffer <RTCYUVPlanarBuffer>
>+ @end
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.h
>+index 2e26b08b8af..fbbc68a8ef6 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.h
>+@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ /** Information for header. Corresponds to webrtc::RTPFragmentationHeader. */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCRtpFragmentationHeader")))
>+ @interface RTCRtpFragmentationHeader : NSObject
>+ 
>+ @property(nonatomic, strong) NSArray<NSNumber *> *fragmentationOffset;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.m
>+index 8049abc411e..6cf21ddc2a1 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.m
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCRtpFragmentationHeader.m
>+@@ -17,4 +17,5 @@ @implementation RTCRtpFragmentationHeader
>+ @synthesize fragmentationTimeDiff = _fragmentationTimeDiff;
>+ @synthesize fragmentationPlType = _fragmentationPlType;
>+ 
>+-@end
>+\ No newline at end of file
>++@end
>++
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoCodecInfo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoCodecInfo.h
>+index 2162caaa21f..4e643d94347 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoCodecInfo.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoCodecInfo.h
>+@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ /** Holds information to identify a codec. Corresponds to webrtc::SdpVideoFormat. */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoCodecInfo")))
>+ @interface RTCVideoCodecInfo : NSObject <NSCoding>
>+ 
>+ - (instancetype)init NS_UNAVAILABLE;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoder.h
>+index 18c6f6b0006..534420d3955 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoder.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoder.h
>+@@ -23,6 +23,7 @@ typedef void (^RTCVideoDecoderCallback)(RTCVideoFrame *frame);
>+ 
>+ /** Protocol for decoder implementations. */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoDecoder")))
>+ @protocol RTCVideoDecoder <NSObject>
>+ 
>+ - (void)setCallback:(RTCVideoDecoderCallback)callback;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoderFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoderFactory.h
>+index 3e24153b82c..cb95fa3f134 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoderFactory.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoDecoderFactory.h
>+@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ /** RTCVideoDecoderFactory is an Objective-C version of webrtc::VideoDecoderFactory. */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactory")))
>+ @protocol RTCVideoDecoderFactory <NSObject>
>+ 
>+ - (nullable id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoder.h
>+index c5257674d83..5291bb1342b 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoder.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoder.h
>+@@ -19,7 +19,11 @@
>+ #import "RTCVideoFrame.h"
>+ 
>+ NS_ASSUME_NONNULL_BEGIN
>+-
>++/*
>++namespace webrtc {
>++class VideoBitrateAllocation;
>++};
>++*/
>+ /** Callback block for encoder. */
>+ typedef BOOL (^RTCVideoEncoderCallback)(RTCEncodedImage *frame,
>+                                         id<RTCCodecSpecificInfo> info,
>+@@ -27,6 +31,7 @@ typedef BOOL (^RTCVideoEncoderCallback)(RTCEncodedImage *frame,
>+ 
>+ /** Protocol for encoder implementations. */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoEncoder")))
>+ @protocol RTCVideoEncoder <NSObject>
>+ 
>+ - (void)setCallback:(RTCVideoEncoderCallback)callback;
>+@@ -37,6 +42,7 @@ RTC_OBJC_EXPORT
>+     codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
>+            frameTypes:(NSArray<NSNumber *> *)frameTypes;
>+ - (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate;
>++- (int)setRateAllocation: (RTCVideoBitrateAllocation *)allocation framerate:(uint32_t)framerate;
>+ - (NSString *)implementationName;
>+ 
>+ /** Returns QP scaling settings for encoder. The quality scaler adjusts the resolution in order to
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderFactory.h
>+index 20c603d6fe6..cbe9ea492ae 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderFactory.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderFactory.h
>+@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ /** RTCVideoEncoderFactory is an Objective-C version of webrtc::VideoEncoderFactory. */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
>+ @protocol RTCVideoEncoderFactory <NSObject>
>+ 
>+ - (nullable id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.h
>+index 2b48f45ce0a..68cfd66df47 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.h
>+@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ /** QP thresholds for encoder. Corresponds to webrtc::VideoEncoder::QpThresholds. */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoEncoderQpThresholds")))
>+ @interface RTCVideoEncoderQpThresholds : NSObject
>+ 
>+ - (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderSettings.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderSettings.h
>+index 69e04cac70f..ae3bf5fb5f1 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderSettings.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoEncoderSettings.h
>+@@ -21,6 +21,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
>+ 
>+ /** Settings for encoder. Corresponds to webrtc::VideoCodec. */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoEncoderSettings")))
>+ @interface RTCVideoEncoderSettings : NSObject
>+ 
>+ @property(nonatomic, strong) NSString *name;
>+@@ -40,4 +41,9 @@ RTC_OBJC_EXPORT
>+ 
>+ @end
>+ 
>++RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoBitrateAllocation")))
>++@interface RTCVideoBitrateAllocation : NSObject
>++@end
>++
>+ NS_ASSUME_NONNULL_END
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrame.h
>+index 9aca7433f34..bde09c50c9d 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrame.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrame.h
>+@@ -26,6 +26,7 @@ typedef NS_ENUM(NSInteger, RTCVideoRotation) {
>+ 
>+ // RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoFrame")))
>+ @interface RTCVideoFrame : NSObject
>+ 
>+ /** Width without rotation applied. */
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrameBuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrameBuffer.h
>+index bb9e6fba631..186ad0889bf 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrameBuffer.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/base/RTCVideoFrameBuffer.h
>+@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>+ 
>+ // RTCVideoFrameBuffer is an ObjectiveC version of webrtc::VideoFrameBuffer.
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
>+ @protocol RTCVideoFrameBuffer <NSObject>
>+ 
>+ @property(nonatomic, readonly) int width;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.h
>+index ece9570a13b..e947dc6e623 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.h
>+@@ -10,7 +10,7 @@
>+ 
>+ #import <Foundation/Foundation.h>
>+ 
>+-#import "RTCCodecSpecificInfo.h"
>++#import "base/RTCCodecSpecificInfo.h"
>+ #import "RTCMacros.h"
>+ 
>+ /** Class for H264 specific config. */
>+@@ -20,6 +20,7 @@ typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
>+ };
>+ 
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCCodecSpecificInfoH264")))
>+ @interface RTCCodecSpecificInfoH264 : NSObject <RTCCodecSpecificInfo>
>+ 
>+ @property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.mm
>+index 57f2411e3bb..63e8bd7f42a 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.mm
>+@@ -20,6 +20,7 @@ @synthesize packetizationMode = _packetizationMode;
>+ - (webrtc::CodecSpecificInfo)nativeCodecSpecificInfo {
>+   webrtc::CodecSpecificInfo codecSpecificInfo;
>+   codecSpecificInfo.codecType = webrtc::kVideoCodecH264;
>++  codecSpecificInfo.codec_name = [kRTCVideoCodecH264Name cStringUsingEncoding:NSUTF8StringEncoding];
>+   codecSpecificInfo.codecSpecific.H264.packetization_mode =
>+       (webrtc::H264PacketizationMode)_packetizationMode;
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h
>+index 7ca9463a593..d11460300d5 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h
>+@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>+  *  codecs, create custom implementations of RTCVideoEncoderFactory and RTCVideoDecoderFactory.
>+  */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCDefaultVideoDecoderFactory")))
>+ @interface RTCDefaultVideoDecoderFactory : NSObject <RTCVideoDecoderFactory>
>+ @end
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.m
>+index bdb18517caf..c7d08caa9c9 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.m
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.m
>+@@ -15,7 +15,7 @@
>+ #import "api/video_codec/RTCVideoCodecConstants.h"
>+ #import "api/video_codec/RTCVideoDecoderVP8.h"
>+ #import "base/RTCVideoCodecInfo.h"
>+-#if defined(RTC_ENABLE_VP9)
>++#if !defined(RTC_DISABLE_VP9)
>+ #import "api/video_codec/RTCVideoDecoderVP9.h"
>+ #endif
>+ 
>+@@ -26,7 +26,7 @@ @implementation RTCDefaultVideoDecoderFactory
>+     return [[RTCVideoDecoderH264 alloc] init];
>+   } else if ([info.name isEqualToString:kRTCVideoCodecVp8Name]) {
>+     return [RTCVideoDecoderVP8 vp8Decoder];
>+-#if defined(RTC_ENABLE_VP9)
>++#if !defined(RTC_DISABLE_VP9)
>+   } else if ([info.name isEqualToString:kRTCVideoCodecVp9Name]) {
>+     return [RTCVideoDecoderVP9 vp9Decoder];
>+ #endif
>+@@ -39,7 +39,7 @@ @implementation RTCDefaultVideoDecoderFactory
>+   return @[
>+     [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecH264Name],
>+     [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp8Name],
>+-#if defined(RTC_ENABLE_VP9)
>++#if !defined(RTC_DISABLE_VP9)
>+     [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp9Name],
>+ #endif
>+   ];
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h
>+index c45e54362b2..9323256cead 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h
>+@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>+  *  codecs, create custom implementations of RTCVideoEncoderFactory and RTCVideoDecoderFactory.
>+  */
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCDefaultVideoEncoderFactory")))
>+ @interface RTCDefaultVideoEncoderFactory : NSObject <RTCVideoEncoderFactory>
>+ 
>+ @property(nonatomic, retain) RTCVideoCodecInfo *preferredCodec;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.m
>+index b72296b64f3..b01c0746792 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.m
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.m
>+@@ -15,7 +15,7 @@
>+ #import "api/video_codec/RTCVideoCodecConstants.h"
>+ #import "api/video_codec/RTCVideoEncoderVP8.h"
>+ #import "base/RTCVideoCodecInfo.h"
>+-#if defined(RTC_ENABLE_VP9)
>++#if !defined(RTC_DISABLE_VP9)
>+ #import "api/video_codec/RTCVideoEncoderVP9.h"
>+ #endif
>+ 
>+@@ -44,7 +44,7 @@ @implementation RTCDefaultVideoEncoderFactory
>+ 
>+   RTCVideoCodecInfo *vp8Info = [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp8Name];
>+ 
>+-#if defined(RTC_ENABLE_VP9)
>++#if !defined(RTC_DISABLE_VP9)
>+   RTCVideoCodecInfo *vp9Info = [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp9Name];
>+ #endif
>+ 
>+@@ -52,7 +52,7 @@ @implementation RTCDefaultVideoEncoderFactory
>+     constrainedHighInfo,
>+     constrainedBaselineInfo,
>+     vp8Info,
>+-#if defined(RTC_ENABLE_VP9)
>++#if !defined(RTC_DISABLE_VP9)
>+     vp9Info,
>+ #endif
>+   ];
>+@@ -63,7 +63,7 @@ @implementation RTCDefaultVideoEncoderFactory
>+     return [[RTCVideoEncoderH264 alloc] initWithCodecInfo:info];
>+   } else if ([info.name isEqualToString:kRTCVideoCodecVp8Name]) {
>+     return [RTCVideoEncoderVP8 vp8Encoder];
>+-#if defined(RTC_ENABLE_VP9)
>++#if !defined(RTC_DISABLE_VP9)
>+   } else if ([info.name isEqualToString:kRTCVideoCodecVp9Name]) {
>+     return [RTCVideoEncoderVP9 vp9Encoder];
>+ #endif
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.h
>+index 56b353215a2..d022297674e 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.h
>+@@ -48,6 +48,7 @@ typedef NS_ENUM(NSUInteger, RTCH264Level) {
>+ };
>+ 
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCH264ProfileLevelId")))
>+ @interface RTCH264ProfileLevelId : NSObject
>+ 
>+ @property(nonatomic, readonly) RTCH264Profile profile;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm
>+index 359656cb97b..3d5c8337246 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm
>+@@ -19,23 +19,35 @@
>+ #include "media/base/h264_profile_level_id.h"
>+ #include "media/base/mediaconstants.h"
>+ 
>++#if !defined(WEBRTC_WEBKIT_BUILD)
>+ namespace {
>+ 
>+ NSString *MaxSupportedProfileLevelConstrainedHigh();
>+ NSString *MaxSupportedProfileLevelConstrainedBaseline();
>+ 
>+ }  // namespace
>++#endif
>+ 
>+-NSString *const kRTCVideoCodecH264Name = @(cricket::kH264CodecName);
>++NSString *const kRTCVideoCodecH264Name = @"H264";
>+ NSString *const kRTCLevel31ConstrainedHigh = @"640c1f";
>+ NSString *const kRTCLevel31ConstrainedBaseline = @"42e01f";
>++
>++#if defined(WEBRTC_WEBKIT_BUILD)
>++NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh =
>++    @"640c1f";
>++NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline =
>++    @"42e01f";
>++#else
>+ NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh =
>+     MaxSupportedProfileLevelConstrainedHigh();
>+ NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline =
>+     MaxSupportedProfileLevelConstrainedBaseline();
>++#endif
>+ 
>+ namespace {
>+ 
>++#if !defined(WEBRTC_WEBKIT_BUILD)
>++
>+ #if defined(WEBRTC_IOS)
>+ 
>+ using namespace webrtc::H264;
>+@@ -73,6 +85,8 @@ NSString *MaxSupportedProfileLevelConstrainedHigh() {
>+   return kRTCLevel31ConstrainedHigh;
>+ }
>+ 
>++#endif
>++
>+ }  // namespace
>+ 
>+ @interface RTCH264ProfileLevelId ()
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h
>+index 4fcff1dff79..fdabd807480 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h
>+@@ -14,5 +14,6 @@
>+ #import "RTCVideoDecoderFactory.h"
>+ 
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactoryH264")))
>+ @interface RTCVideoDecoderFactoryH264 : NSObject <RTCVideoDecoderFactory>
>+ @end
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.h
>+index b860276206c..b10d1e6a099 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.h
>+@@ -14,5 +14,6 @@
>+ #import "RTCVideoDecoder.h"
>+ 
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoDecoderH264")))
>+ @interface RTCVideoDecoderH264 : NSObject <RTCVideoDecoder>
>+ @end
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm
>+index 3bfb918bf61..1c18fab17df 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm
>+@@ -19,10 +19,6 @@
>+ #import "helpers.h"
>+ #import "helpers/scoped_cftyperef.h"
>+ 
>+-#if defined(WEBRTC_IOS)
>+-#import "helpers/UIDevice+RTCDevice.h"
>+-#endif
>+-
>+ #include "modules/video_coding/include/video_error_codes.h"
>+ #include "rtc_base/checks.h"
>+ #include "rtc_base/logging.h"
>+@@ -251,9 +247,7 @@ - (void)configureDecompressionSession {
>+ - (void)destroyDecompressionSession {
>+   if (_decompressionSession) {
>+ #if defined(WEBRTC_IOS)
>+-    if ([UIDevice isIOS11OrLater]) {
>+-      VTDecompressionSessionWaitForAsynchronousFrames(_decompressionSession);
>+-    }
>++    VTDecompressionSessionWaitForAsynchronousFrames(_decompressionSession);
>+ #endif
>+     VTDecompressionSessionInvalidate(_decompressionSession);
>+     CFRelease(_decompressionSession);
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h
>+index c64405e4dac..5f6f78dfa9a 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h
>+@@ -14,5 +14,6 @@
>+ #import "RTCVideoEncoderFactory.h"
>+ 
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactoryH264")))
>+ @interface RTCVideoEncoderFactoryH264 : NSObject <RTCVideoEncoderFactory>
>+ @end
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.h
>+index a9c05580a41..7714b25eaa4 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.h
>+@@ -15,6 +15,7 @@
>+ #import "RTCVideoEncoder.h"
>+ 
>+ RTC_OBJC_EXPORT
>++__attribute__((objc_runtime_name("WK_RTCVideoEncoderH264")))
>+ @interface RTCVideoEncoderH264 : NSObject <RTCVideoEncoder>
>+ 
>+ - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo;
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
>+index f6690b6d861..a14357fb80c 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
>+@@ -9,39 +9,68 @@
>+  *
>+  */
>+ 
>+-#import "RTCVideoEncoderH264.h"
>++#import "WebRTC/RTCVideoCodecH264.h"
>+ 
>+ #import <VideoToolbox/VideoToolbox.h>
>+ #include <vector>
>+ 
>+-#if defined(WEBRTC_IOS)
>+-#import "helpers/UIDevice+RTCDevice.h"
>+-#endif
>+-#import "RTCCodecSpecificInfoH264.h"
>+-#import "RTCH264ProfileLevelId.h"
>+-#import "api/peerconnection/RTCRtpFragmentationHeader+Private.h"
>+-#import "api/peerconnection/RTCVideoCodecInfo+Private.h"
>+-#import "base/RTCCodecSpecificInfo.h"
>+-#import "base/RTCI420Buffer.h"
>+-#import "base/RTCVideoEncoder.h"
>+-#import "base/RTCVideoFrame.h"
>+-#import "base/RTCVideoFrameBuffer.h"
>+-#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
>+-#import "helpers.h"
>+-
>++#import "PeerConnection/RTCVideoCodec+Private.h"
>++#import "WebRTC/RTCVideoCodec.h"
>++#import "WebRTC/RTCVideoFrame.h"
>++#import "WebRTC/RTCVideoFrameBuffer.h"
>+ #include "common_video/h264/h264_bitstream_parser.h"
>+ #include "common_video/h264/profile_level_id.h"
>+ #include "common_video/include/bitrate_adjuster.h"
>++#import "helpers.h"
>+ #include "modules/include/module_common_types.h"
>+ #include "modules/video_coding/include/video_error_codes.h"
>+ #include "rtc_base/buffer.h"
>+ #include "rtc_base/logging.h"
>+ #include "rtc_base/timeutils.h"
>+-#include "sdk/objc/components/video_codec/nalu_rewriter.h"
>++#include "sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h"
>++#include "system_wrappers/include/clock.h"
>+ #include "third_party/libyuv/include/libyuv/convert_from.h"
>+ 
>+-@interface RTCVideoEncoderH264 ()
>++#include "sdk/WebKit/EncoderUtilities.h"
>++#include "sdk/WebKit/WebKitUtilities.h"
>++
>++#import <dlfcn.h>
>++#import <objc/runtime.h>
>++
>++SOFT_LINK_FRAMEWORK_OPTIONAL(VideoToolBox)
>++SOFT_LINK_POINTER_OPTIONAL(VideoToolBox, kVTVideoEncoderSpecification_Usage, NSString *)
>++
>++#if !ENABLE_VCP_ENCODER && !defined(WEBRTC_IOS)
>++static inline bool isStandardFrameSize(int32_t width, int32_t height)
>++{
>++    // FIXME: Envision relaxing this rule, something like width and height dividable by 4 or 8 should be good enough.
>++    if (width == 1280)
>++        return height == 720;
>++    if (width == 720)
>++        return height == 1280;
>++    if (width == 960)
>++        return height == 540;
>++    if (width == 540)
>++        return height == 960;
>++    if (width == 640)
>++        return height == 480;
>++    if (width == 480)
>++        return height == 640;
>++    if (width == 288)
>++        return height == 352;
>++    if (width == 352)
>++        return height == 288;
>++    if (width == 320)
>++        return height == 240;
>++    if (width == 240)
>++        return height == 320;
>++    return false;
>++}
>++#endif
>+ 
>++__attribute__((objc_runtime_name("WK_RTCSingleVideoEncoderH264")))
>++@interface RTCSingleVideoEncoderH264 : NSObject <RTCVideoEncoder>
>++- (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo simulcastIndex: (int)index;
>+ - (void)frameWasEncoded:(OSStatus)status
>+                   flags:(VTEncodeInfoFlags)infoFlags
>+            sampleBuffer:(CMSampleBufferRef)sampleBuffer
>+@@ -51,7 +80,6 @@ - (void)frameWasEncoded:(OSStatus)status
>+            renderTimeMs:(int64_t)renderTimeMs
>+               timestamp:(uint32_t)timestamp
>+                rotation:(RTCVideoRotation)rotation;
>+-
>+ @end
>+ 
>+ namespace {  // anonymous namespace
>+@@ -70,7 +98,7 @@ const OSType kNV12PixelFormat = kCVPixelFormatType_420YpCbCr8BiPlanarFullRange;
>+ // Struct that we pass to the encoder per frame to encode. We receive it again
>+ // in the encoder callback.
>+ struct RTCFrameEncodeParams {
>+-  RTCFrameEncodeParams(RTCVideoEncoderH264 *e,
>++  RTCFrameEncodeParams(RTCSingleVideoEncoderH264 *e,
>+                        RTCCodecSpecificInfoH264 *csi,
>+                        int32_t w,
>+                        int32_t h,
>+@@ -85,7 +113,7 @@ struct RTCFrameEncodeParams {
>+     }
>+   }
>+ 
>+-  RTCVideoEncoderH264 *encoder;
>++  RTCSingleVideoEncoderH264 *encoder;
>+   RTCCodecSpecificInfoH264 *codecSpecificInfo;
>+   int32_t width;
>+   int32_t height;
>+@@ -173,8 +201,8 @@ void compressionOutputCallback(void *encoder,
>+                                 rotation:encodeParams->rotation];
>+ }
>+ 
>+-// Extract VideoToolbox profile out of the webrtc::SdpVideoFormat. If there is
>+-// no specific VideoToolbox profile for the specified level, AutoLevel will be
>++// Extract VideoToolbox profile out of the cricket::VideoCodec. If there is no
>++// specific VideoToolbox profile for the specified level, AutoLevel will be
>+ // returned. The user must initialize the encoder with a resolution and
>+ // framerate conforming to the selected H264 level regardless.
>+ CFStringRef ExtractProfile(webrtc::SdpVideoFormat videoFormat) {
>+@@ -280,23 +308,26 @@ CFStringRef ExtractProfile(webrtc::SdpVideoFormat videoFormat) {
>+ }
>+ }  // namespace
>+ 
>+-@implementation RTCVideoEncoderH264 {
>++@implementation RTCSingleVideoEncoderH264 {
>+   RTCVideoCodecInfo *_codecInfo;
>+   std::unique_ptr<webrtc::BitrateAdjuster> _bitrateAdjuster;
>+   uint32_t _targetBitrateBps;
>+-  uint32_t _encoderFrameRate;
>+   uint32_t _encoderBitrateBps;
>+   RTCH264PacketizationMode _packetizationMode;
>+   CFStringRef _profile;
>+   RTCVideoEncoderCallback _callback;
>+   int32_t _width;
>+   int32_t _height;
>+-  VTCompressionSessionRef _compressionSession;
>++  CompressionSessionRef _compressionSession;
>+   CVPixelBufferPoolRef _pixelBufferPool;
>+   RTCVideoCodecMode _mode;
>+ 
>+   webrtc::H264BitstreamParser _h264BitstreamParser;
>+   std::vector<uint8_t> _frameScaleBuffer;
>++
>++  webrtc::VideoCodec _nativeVideoCodec;
>++  int _simulcastIndex;
>++  bool _disableEncoding;
>+ }
>+ 
>+ // .5 is set as a mininum to prevent overcompensating for large temporary
>+@@ -306,12 +337,13 @@ @implementation RTCVideoEncoderH264 {
>+ // drastically reduced bitrate, so we want to avoid that. In steady state
>+ // conditions, 0.95 seems to give us better overall bitrate over long periods
>+ // of time.
>+-- (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo {
>++- (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo simulcastIndex:(int)index {
>+   if (self = [super init]) {
>+     _codecInfo = codecInfo;
>+     _bitrateAdjuster.reset(new webrtc::BitrateAdjuster(.5, .95));
>+     _packetizationMode = RTCH264PacketizationModeNonInterleaved;
>+     _profile = ExtractProfile([codecInfo nativeSdpVideoFormat]);
>++    _simulcastIndex = index;
>+     RTC_LOG(LS_INFO) << "Using profile " << CFStringToString(_profile);
>+     RTC_CHECK([codecInfo.name isEqualToString:kRTCVideoCodecH264Name]);
>+   }
>+@@ -330,11 +362,13 @@ - (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings
>+   _width = settings.width;
>+   _height = settings.height;
>+   _mode = settings.mode;
>++  _nativeVideoCodec = settings.nativeVideoCodec;
>++
>++  RTC_DCHECK(_nativeVideoCodec.numberOfSimulcastStreams != 1);
>+ 
>+   // We can only set average bitrate on the HW encoder.
>+-  _targetBitrateBps = settings.startBitrate * 1000;  // startBitrate is in kbps.
>++  _targetBitrateBps = settings.startBitrate;
>+   _bitrateAdjuster->SetTargetBitrateBps(_targetBitrateBps);
>+-  _encoderFrameRate = settings.maxFramerate;
>+ 
>+   // TODO(tkchin): Try setting payload size via
>+   // kVTCompressionPropertyKey_MaxH264SliceBytes.
>+@@ -357,6 +391,10 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>+     isKeyframeRequired = YES;
>+   }
>+ 
>++  if (_disableEncoding) {
>++    return WEBRTC_VIDEO_CODEC_ERROR;
>++  }
>++
>+   CVPixelBufferRef pixelBuffer = nullptr;
>+   if ([frame.buffer isKindOfClass:[RTCCVPixelBuffer class]]) {
>+     // Native frame buffer
>+@@ -434,9 +472,9 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>+   encodeParams->codecSpecificInfo.packetizationMode = _packetizationMode;
>+ 
>+   // Update the bitrate if needed.
>+-  [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps() frameRate:_encoderFrameRate];
>++  [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps()];
>+ 
>+-  OSStatus status = VTCompressionSessionEncodeFrame(_compressionSession,
>++  OSStatus status = CompressionSessionEncodeFrame(_compressionSession,
>+                                                     pixelBuffer,
>+                                                     presentationTimeStamp,
>+                                                     kCMTimeInvalid,
>+@@ -467,10 +505,10 @@ - (void)setCallback:(RTCVideoEncoderCallback)callback {
>+   _callback = callback;
>+ }
>+ 
>+-- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
>+-  _targetBitrateBps = 1000 * bitrateKbit;
>++- (int)setBitrate:(uint32_t)bitrateBps framerate:(uint32_t)framerate {
>++  _targetBitrateBps = bitrateBps;
>+   _bitrateAdjuster->SetTargetBitrateBps(_targetBitrateBps);
>+-  [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps() frameRate:framerate];
>++  [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps()];
>+   return WEBRTC_VIDEO_CODEC_OK;
>+ }
>+ 
>+@@ -503,6 +541,11 @@ - (BOOL)resetCompressionSessionIfNeededWithFrame:(RTCVideoFrame *)frame {
>+   OSType framePixelFormat = [self pixelFormatOfFrame:frame];
>+ 
>+   if (_compressionSession) {
>++#if defined(WEBRTC_WEBKIT_BUILD)
>++    if (!_pixelBufferPool) {
>++      return NO;
>++    }
>++#endif
>+     // The pool attribute `kCVPixelBufferPixelFormatTypeKey` can contain either an array of pixel
>+     // formats or a single pixel format.
>+     NSDictionary *poolAttributes =
>+@@ -516,6 +559,11 @@ - (BOOL)resetCompressionSessionIfNeededWithFrame:(RTCVideoFrame *)frame {
>+       compressionSessionPixelFormats = @[ (NSNumber *)pixelFormats ];
>+     }
>+ 
>++    if ([frame.buffer isKindOfClass:[RTCCVPixelBuffer class]]) {
>++      RTCCVPixelBuffer *rtcPixelBuffer = (RTCCVPixelBuffer *)frame.buffer;
>++      framePixelFormat = CVPixelBufferGetPixelFormatType(rtcPixelBuffer.pixelBuffer);
>++    }
>++
>+     if (![compressionSessionPixelFormats
>+             containsObject:[NSNumber numberWithLong:framePixelFormat]]) {
>+       resetCompressionSession = YES;
>+@@ -533,6 +581,7 @@ - (BOOL)resetCompressionSessionIfNeededWithFrame:(RTCVideoFrame *)frame {
>+ 
>+ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>+   [self destroyCompressionSession];
>++  _disableEncoding = false;
>+ 
>+   // Set source image buffer attributes. These attributes will be present on
>+   // buffers retrieved from the encoder's pixel buffer pool.
>+@@ -559,22 +608,26 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>+     CFRelease(pixelFormat);
>+     pixelFormat = nullptr;
>+   }
>+-  CFMutableDictionaryRef encoder_specs = nullptr;
>+-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
>++  CFDictionaryRef encoderSpecs = nullptr;
>++#if !defined(WEBRTC_IOS)
>++  auto useHardwareEncoder = webrtc::isH264HardwareEncoderAllowed() ? kCFBooleanTrue : kCFBooleanFalse;
>+   // Currently hw accl is supported above 360p on mac, below 360p
>+   // the compression session will be created with hw accl disabled.
>+-  encoder_specs = CFDictionaryCreateMutable(
>+-      nullptr, 1, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>+-  CFDictionarySetValue(encoder_specs,
>+-                       kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder,
>+-                       kCFBooleanTrue);
>++  CFTypeRef sessionKeys[] = { kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder, kVTVideoEncoderSpecification_RequireHardwareAcceleratedVideoEncoder, kVTCompressionPropertyKey_RealTime };
>++  CFTypeRef sessionValues[] = { useHardwareEncoder, useHardwareEncoder, kCFBooleanTrue };
>++  encoderSpecs = CFDictionaryCreate(kCFAllocatorDefault, sessionKeys, sessionValues, 3, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>++#else
>++  CFTypeRef sessionKeys[] = { kVTCompressionPropertyKey_RealTime };
>++  CFTypeRef sessionValues[] = { kCFBooleanTrue };
>++  encoderSpecs = CFDictionaryCreate(kCFAllocatorDefault, sessionKeys, sessionValues, 1, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>+ #endif
>++
>+   OSStatus status =
>+-      VTCompressionSessionCreate(nullptr,  // use default allocator
>++      CompressionSessionCreate(nullptr,  // use default allocator
>+                                  _width,
>+                                  _height,
>+-                                 kCMVideoCodecType_H264,
>+-                                 encoder_specs,  // use hardware accelerated encoder if available
>++                                 kCodecTypeH264,
>++                                 encoderSpecs,  // use hardware accelerated encoder if available
>+                                  sourceAttributes,
>+                                  nullptr,  // use default compressed data allocator
>+                                  compressionOutputCallback,
>+@@ -584,32 +637,113 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>+     CFRelease(sourceAttributes);
>+     sourceAttributes = nullptr;
>+   }
>+-  if (encoder_specs) {
>+-    CFRelease(encoder_specs);
>+-    encoder_specs = nullptr;
>++  if (encoderSpecs) {
>++    CFRelease(encoderSpecs);
>++    encoderSpecs = nullptr;
>+   }
>++
>++#if ENABLE_VCP_ENCODER || defined(WEBRTC_IOS)
>+   if (status != noErr) {
>+     RTC_LOG(LS_ERROR) << "Failed to create compression session: " << status;
>+     return WEBRTC_VIDEO_CODEC_ERROR;
>+   }
>++#endif
>+ #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
>+   CFBooleanRef hwaccl_enabled = nullptr;
>+-  status = VTSessionCopyProperty(_compressionSession,
>++  if (status == noErr) {
>++    status = VTSessionCopyProperty(_compressionSession,
>+                                  kVTCompressionPropertyKey_UsingHardwareAcceleratedVideoEncoder,
>+                                  nullptr,
>+                                  &hwaccl_enabled);
>++  }
>+   if (status == noErr && (CFBooleanGetValue(hwaccl_enabled))) {
>+     RTC_LOG(LS_INFO) << "Compression session created with hw accl enabled";
>+   } else {
>+     RTC_LOG(LS_INFO) << "Compression session created with hw accl disabled";
>++
>++#if !ENABLE_VCP_ENCODER && !defined(WEBRTC_IOS)
>++    if (!isStandardFrameSize(_width, _height)) {
>++      _disableEncoding = true;
>++      RTC_LOG(LS_ERROR) << "Using H264 software encoder with non standard size is not supported";
>++      return WEBRTC_VIDEO_CODEC_ERROR;
>++    }
>++
>++    if (!getkVTVideoEncoderSpecification_Usage()) {
>++      _disableEncoding = true;
>++      RTC_LOG(LS_ERROR) << "RTCSingleVideoEncoderH264 cannot create a H264 software encoder";
>++      return WEBRTC_VIDEO_CODEC_ERROR;
>++    }
>++
>++    CFDictionaryRef ioSurfaceValue = CreateCFTypeDictionary(nullptr, nullptr, 0);
>++    int64_t pixelFormatType = framePixelFormat;
>++    CFNumberRef pixelFormat = CFNumberCreate(nullptr, kCFNumberLongType, &pixelFormatType);
>++
>++    const size_t attributesSize = 3;
>++    CFTypeRef keys[attributesSize] = {
>++      kCVPixelBufferOpenGLCompatibilityKey,
>++      kCVPixelBufferIOSurfacePropertiesKey,
>++      kCVPixelBufferPixelFormatTypeKey
>++    };
>++    CFTypeRef values[attributesSize] = {
>++      kCFBooleanTrue,
>++      ioSurfaceValue,
>++      pixelFormat};
>++    CFDictionaryRef sourceAttributes = CreateCFTypeDictionary(keys, values, attributesSize);
>++
>++    if (ioSurfaceValue) {
>++      CFRelease(ioSurfaceValue);
>++      ioSurfaceValue = nullptr;
>++    }
>++    if (pixelFormat) {
>++      CFRelease(pixelFormat);
>++      pixelFormat = nullptr;
>++    }
>++
>++    CFMutableDictionaryRef encoderSpecs = CFDictionaryCreateMutable(nullptr, 2, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>++    CFDictionarySetValue(encoderSpecs, kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder, kCFBooleanFalse);
>++    int usageValue = 1;
>++    CFNumberRef usage = CFNumberCreate(nullptr, kCFNumberIntType, &usageValue);
>++    CFDictionarySetValue(encoderSpecs, (__bridge CFStringRef)getkVTVideoEncoderSpecification_Usage(), usage);
>++
>++    if (usage) {
>++      CFRelease(usage);
>++      usage = nullptr;
>++    }
>++
>++    [self destroyCompressionSession];
>++
>++    OSStatus status =
>++      CompressionSessionCreate(nullptr,  // use default allocator
>++                                 _width,
>++                                 _height,
>++                                 kCodecTypeH264,
>++                                 encoderSpecs,
>++                                 sourceAttributes,
>++                                 nullptr,  // use default compressed data allocator
>++                                 compressionOutputCallback,
>++                                 nullptr,
>++                                 &_compressionSession);
>++    if (sourceAttributes) {
>++      CFRelease(sourceAttributes);
>++      sourceAttributes = nullptr;
>++    }
>++    if (encoderSpecs) {
>++      CFRelease(encoderSpecs);
>++      encoderSpecs = nullptr;
>++    }
>++    if (status != noErr) {
>++      return WEBRTC_VIDEO_CODEC_ERROR;
>++    }
>++#endif
>+   }
>+ #endif
>+   [self configureCompressionSession];
>+ 
>++#if !defined(WEBRTC_WEBKIT_BUILD)
>+   // The pixel buffer pool is dependent on the compression session so if the session is reset, the
>+   // pool should be reset as well.
>+-  _pixelBufferPool = VTCompressionSessionGetPixelBufferPool(_compressionSession);
>+-
>++  _pixelBufferPool = CompressionSessionGetPixelBufferPool(_compressionSession);
>++#endif
>+   return WEBRTC_VIDEO_CODEC_OK;
>+ }
>+ 
>+@@ -618,7 +752,11 @@ - (void)configureCompressionSession {
>+   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_RealTime, true);
>+   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_ProfileLevel, _profile);
>+   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_AllowFrameReordering, false);
>+-  [self setEncoderBitrateBps:_targetBitrateBps frameRate:_encoderFrameRate];
>++#if ENABLE_VCP_ENCODER
>++  if (auto key = getkVTVideoEncoderSpecification_Usage())
>++      SetVTSessionProperty(_compressionSession, (__bridge CFStringRef)key, 1);
>++#endif
>++  [self setEncoderBitrateBps:_targetBitrateBps];
>+   // TODO(tkchin): Look at entropy mode and colorspace matrices.
>+   // TODO(tkchin): Investigate to see if there's any way to make this work.
>+   // May need it to interop with Android. Currently this call just fails.
>+@@ -635,7 +773,7 @@ - (void)configureCompressionSession {
>+ 
>+ - (void)destroyCompressionSession {
>+   if (_compressionSession) {
>+-    VTCompressionSessionInvalidate(_compressionSession);
>++    CompressionSessionInvalidate(_compressionSession);
>+     CFRelease(_compressionSession);
>+     _compressionSession = nullptr;
>+     _pixelBufferPool = nullptr;
>+@@ -646,17 +784,15 @@ - (NSString *)implementationName {
>+   return @"VideoToolbox";
>+ }
>+ 
>+-- (void)setBitrateBps:(uint32_t)bitrateBps frameRate:(uint32_t)frameRate {
>+-  if (_encoderBitrateBps != bitrateBps || _encoderFrameRate != frameRate) {
>+-    [self setEncoderBitrateBps:bitrateBps frameRate:frameRate];
>++- (void)setBitrateBps:(uint32_t)bitrateBps {
>++  if (_encoderBitrateBps != bitrateBps) {
>++    [self setEncoderBitrateBps:bitrateBps];
>+   }
>+ }
>+ 
>+-- (void)setEncoderBitrateBps:(uint32_t)bitrateBps frameRate:(uint32_t)frameRate {
>++- (void)setEncoderBitrateBps:(uint32_t)bitrateBps {
>+   if (_compressionSession) {
>+     SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_AverageBitRate, bitrateBps);
>+-    SetVTSessionProperty(
>+-        _compressionSession, kVTCompressionPropertyKey_ExpectedFrameRate, frameRate);
>+ 
>+     // TODO(tkchin): Add a helper method to set array value.
>+     int64_t dataLimitBytesPerSecondValue =
>+@@ -668,7 +804,7 @@ - (void)setEncoderBitrateBps:(uint32_t)bitrateBps frameRate:(uint32_t)frameRate
>+         CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt64Type, &oneSecondValue);
>+     const void *nums[2] = {bytesPerSecond, oneSecond};
>+     CFArrayRef dataRateLimits = CFArrayCreate(nullptr, nums, 2, &kCFTypeArrayCallBacks);
>+-    OSStatus status = VTSessionSetProperty(
>++    OSStatus status = CompressionSessionSetProperty(
>+         _compressionSession, kVTCompressionPropertyKey_DataRateLimits, dataRateLimits);
>+     if (bytesPerSecond) {
>+       CFRelease(bytesPerSecond);
>+@@ -680,11 +816,10 @@ - (void)setEncoderBitrateBps:(uint32_t)bitrateBps frameRate:(uint32_t)frameRate
>+       CFRelease(dataRateLimits);
>+     }
>+     if (status != noErr) {
>+-      RTC_LOG(LS_ERROR) << "Failed to set data rate limit with code: " << status;
>++      RTC_LOG(LS_ERROR) << "Failed to set data rate limit";
>+     }
>+ 
>+     _encoderBitrateBps = bitrateBps;
>+-    _encoderFrameRate = frameRate;
>+   }
>+ }
>+ 
>+@@ -698,7 +833,7 @@ - (void)frameWasEncoded:(OSStatus)status
>+               timestamp:(uint32_t)timestamp
>+                rotation:(RTCVideoRotation)rotation {
>+   if (status != noErr) {
>+-    RTC_LOG(LS_ERROR) << "H264 encode failed with code: " << status;
>++    RTC_LOG(LS_ERROR) << "H264 encode failed: " << status;
>+     return;
>+   }
>+   if (infoFlags & kVTEncodeInfo_FrameDropped) {
>+@@ -743,6 +878,7 @@ - (void)frameWasEncoded:(OSStatus)status
>+   frame.rotation = rotation;
>+   frame.contentType = (_mode == RTCVideoCodecModeScreensharing) ? RTCVideoContentTypeScreenshare :
>+                                                                   RTCVideoContentTypeUnspecified;
>++  frame.spatialIndex = _simulcastIndex;
>+   frame.flags = webrtc::VideoSendTiming::kInvalid;
>+ 
>+   int qp;
>+@@ -758,9 +894,88 @@ - (void)frameWasEncoded:(OSStatus)status
>+   _bitrateAdjuster->Update(frame.buffer.length);
>+ }
>+ 
>+-- (nullable RTCVideoEncoderQpThresholds *)scalingSettings {
>+-  return [[RTCVideoEncoderQpThresholds alloc] initWithThresholdsLow:kLowH264QpThreshold
>+-                                                               high:kHighH264QpThreshold];
>++- (RTCVideoEncoderQpThresholds *)scalingSettings {
>++  return [[RTCVideoEncoderQpThresholds alloc] initWithThresholdsLow:kLowH264QpThreshold high:kHighH264QpThreshold];
>++}
>++
>++- (int)setRateAllocation:(RTCVideoBitrateAllocation *)allocation framerate:(uint32_t)framerate {
>++  return 0;
>++}
>++
>++@end
>++
>++@implementation RTCVideoEncoderH264 {
>++  NSMutableArray<RTCSingleVideoEncoderH264*> *_codecs;
>++  RTCVideoCodecInfo *_codecInfo;
>++}
>++- (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo {
>++  if (self = [super init]) {
>++    _codecInfo = codecInfo;
>++  }
>++  return self;
>++}
>++
>++- (void)setCallback:(RTCVideoEncoderCallback)callback {
>++  for (RTCSingleVideoEncoderH264 *codec : _codecs)
>++    [codec setCallback:callback];
>++}
>++
>++- (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings numberOfCores:(int)numberOfCores {
>++    auto nativeCodecSettings = settings.nativeVideoCodec;
>++
>++  _codecs = [[NSMutableArray alloc] init];
>++  for (unsigned index = 0 ; index < nativeCodecSettings.numberOfSimulcastStreams; ++index) {
>++    auto codec = [[RTCSingleVideoEncoderH264 alloc] initWithCodecInfo:_codecInfo simulcastIndex:index];
>++    [_codecs addObject:codec];
>++
>++    auto codecSettings = nativeCodecSettings;
>++    codecSettings.width = nativeCodecSettings.simulcastStream[index].width;
>++    codecSettings.height = nativeCodecSettings.simulcastStream[index].height;
>++    codecSettings.maxBitrate = nativeCodecSettings.simulcastStream[index].maxBitrate;
>++    codecSettings.targetBitrate = nativeCodecSettings.simulcastStream[index].targetBitrate;
>++    codecSettings.minBitrate = nativeCodecSettings.simulcastStream[index].minBitrate;
>++    codecSettings.qpMax = nativeCodecSettings.simulcastStream[index].qpMax;
>++    codecSettings.active = true;
>++
>++    auto *settings = [[RTCVideoEncoderSettings alloc] initWithNativeVideoCodec:&codecSettings];
>++    [codec startEncodeWithSettings:settings numberOfCores:numberOfCores];
>++  }
>++  return 0;
>++}
>++
>++- (NSInteger)releaseEncoder {
>++  for (RTCSingleVideoEncoderH264 *codec : _codecs)
>++    [codec releaseEncoder];
>++  _codecs = nil;
>++  return 0;
>++}
>++
>++- (NSInteger)encode:(RTCVideoFrame *)frame codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info frameTypes:(NSArray<NSNumber *> *)frameTypes {
>++  int result = 0;
>++  for (RTCSingleVideoEncoderH264 *codec : _codecs)
>++    result |= [codec encode:frame codecSpecificInfo:info frameTypes:frameTypes];
>++  return result;
>++}
>++
>++- (int)setRateAllocation:(RTCVideoBitrateAllocation *)bitRateAllocation framerate:(uint32_t) framerate {
>++  int result = 0;
>++  unsigned counter = 0;
>++  auto nativeBitRateAllocation = bitRateAllocation.nativeVideoBitrateAllocation;
>++  for (RTCSingleVideoEncoderH264 *codec : _codecs)
>++    result |= [codec setBitrate:nativeBitRateAllocation.GetSpatialLayerSum(counter++) framerate:framerate];
>++  return result;
>++}
>++
>++- (NSString *)implementationName {
>++  return @"VideoToolbox";
>++}
>++
>++- (RTCVideoEncoderQpThresholds *)scalingSettings {
>++  return [[RTCVideoEncoderQpThresholds alloc] initWithThresholdsLow:kLowH264QpThreshold high:kHighH264QpThreshold];
>++}
>++
>++- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
>++  return 0;
>+ }
>+ 
>+ @end
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/UIDevice+H264Profile.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/UIDevice+H264Profile.mm
>+index 852c3d4702f..a015f7ca15c 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/UIDevice+H264Profile.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/UIDevice+H264Profile.mm
>+@@ -24,9 +24,6 @@ struct SupportedH264Profile {
>+ 
>+ constexpr SupportedH264Profile kH264MaxSupportedProfiles[] = {
>+     // iPhones with at least iOS 9
>+-    {RTCDeviceTypeIPhoneXS, {kProfileHigh, kLevel5_2}},      // https://support.apple.com/kb/SP779
>+-    {RTCDeviceTypeIPhoneXSMax, {kProfileHigh, kLevel5_2}},   // https://support.apple.com/kb/SP780
>+-    {RTCDeviceTypeIPhoneXR, {kProfileHigh, kLevel5_2}},      // https://support.apple.com/kb/SP781
>+     {RTCDeviceTypeIPhoneX, {kProfileHigh, kLevel5_2}},       // https://support.apple.com/kb/SP770
>+     {RTCDeviceTypeIPhone8, {kProfileHigh, kLevel5_2}},       // https://support.apple.com/kb/SP767
>+     {RTCDeviceTypeIPhone8Plus, {kProfileHigh, kLevel5_2}},   // https://support.apple.com/kb/SP768
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.cc
>+index ac957f1b497..46daf44ecac 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.cc
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.cc
>+@@ -35,12 +35,12 @@ std::string CFStringToString(const CFStringRef cf_string) {
>+ }
>+ 
>+ // Convenience function for setting a VT property.
>+-void SetVTSessionProperty(VTSessionRef session,
>++void SetVTSessionProperty(CompressionSessionRef session,
>+                           CFStringRef key,
>+                           int32_t value) {
>+   CFNumberRef cfNum =
>+       CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt32Type, &value);
>+-  OSStatus status = VTSessionSetProperty(session, key, cfNum);
>++  OSStatus status = CompressionSessionSetProperty(session, key, cfNum);
>+   CFRelease(cfNum);
>+   if (status != noErr) {
>+     std::string key_string = CFStringToString(key);
>+@@ -50,13 +50,13 @@ void SetVTSessionProperty(VTSessionRef session,
>+ }
>+ 
>+ // Convenience function for setting a VT property.
>+-void SetVTSessionProperty(VTSessionRef session,
>++void SetVTSessionProperty(CompressionSessionRef session,
>+                           CFStringRef key,
>+                           uint32_t value) {
>+   int64_t value_64 = value;
>+   CFNumberRef cfNum =
>+       CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt64Type, &value_64);
>+-  OSStatus status = VTSessionSetProperty(session, key, cfNum);
>++  OSStatus status = CompressionSessionSetProperty(session, key, cfNum);
>+   CFRelease(cfNum);
>+   if (status != noErr) {
>+     std::string key_string = CFStringToString(key);
>+@@ -66,9 +66,9 @@ void SetVTSessionProperty(VTSessionRef session,
>+ }
>+ 
>+ // Convenience function for setting a VT property.
>+-void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value) {
>++void SetVTSessionProperty(CompressionSessionRef session, CFStringRef key, bool value) {
>+   CFBooleanRef cf_bool = (value) ? kCFBooleanTrue : kCFBooleanFalse;
>+-  OSStatus status = VTSessionSetProperty(session, key, cf_bool);
>++  OSStatus status = CompressionSessionSetProperty(session, key, cf_bool);
>+   if (status != noErr) {
>+     std::string key_string = CFStringToString(key);
>+     RTC_LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
>+@@ -77,10 +77,10 @@ void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value) {
>+ }
>+ 
>+ // Convenience function for setting a VT property.
>+-void SetVTSessionProperty(VTSessionRef session,
>++void SetVTSessionProperty(CompressionSessionRef session,
>+                           CFStringRef key,
>+                           CFStringRef value) {
>+-  OSStatus status = VTSessionSetProperty(session, key, value);
>++  OSStatus status = CompressionSessionSetProperty(session, key, value);
>+   if (status != noErr) {
>+     std::string key_string = CFStringToString(key);
>+     std::string val_string = CFStringToString(value);
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.h
>+index 0683ea79e56..0c669e8bce6 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.h
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/helpers.h
>+@@ -16,6 +16,8 @@
>+ #include <VideoToolbox/VideoToolbox.h>
>+ #include <string>
>+ 
>++#include "sdk/WebKit/EncoderUtilities.h"
>++
>+ // Convenience function for creating a dictionary.
>+ inline CFDictionaryRef CreateCFTypeDictionary(CFTypeRef* keys,
>+                                               CFTypeRef* values,
>+@@ -29,18 +31,18 @@ inline CFDictionaryRef CreateCFTypeDictionary(CFTypeRef* keys,
>+ std::string CFStringToString(const CFStringRef cf_string);
>+ 
>+ // Convenience function for setting a VT property.
>+-void SetVTSessionProperty(VTSessionRef session, CFStringRef key, int32_t value);
>++void SetVTSessionProperty(CompressionSessionRef session, CFStringRef key, int32_t value);
>+ 
>+ // Convenience function for setting a VT property.
>+-void SetVTSessionProperty(VTSessionRef session,
>++void SetVTSessionProperty(CompressionSessionRef session,
>+                           CFStringRef key,
>+                           uint32_t value);
>+ 
>+ // Convenience function for setting a VT property.
>+-void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value);
>++void SetVTSessionProperty(CompressionSessionRef session, CFStringRef key, bool value);
>+ 
>+ // Convenience function for setting a VT property.
>+-void SetVTSessionProperty(VTSessionRef session,
>++void SetVTSessionProperty(CompressionSessionRef session,
>+                           CFStringRef key,
>+                           CFStringRef value);
>+ 
>+diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm
>+index fafad7466ec..ae45033a570 100644
>+--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm
>++++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/native/src/objc_video_encoder_factory.mm
>+@@ -90,6 +90,11 @@ class ObjCVideoEncoder : public VideoEncoder {
>+     return [encoder_ setBitrate:bitrate framerate:framerate];
>+   }
>+ 
>++  int32_t SetRateAllocation(const VideoBitrateAllocation& allocation, uint32_t framerate) {
>++    auto *rtcAllocation = [[RTCVideoBitrateAllocation alloc] initWithNativeVideoBitrateAllocation:&allocation];
>++    return [encoder_ setRateAllocation: rtcAllocation framerate:framerate];
>++  }
>++
>+   VideoEncoder::EncoderInfo GetEncoderInfo() const {
>+     EncoderInfo info;
>+     info.supports_native_handle = true;
>diff --git a/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj b/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
>index 2a54bdb0078a4b8005a31befed10f9cc63b4de3a..229467017fea6bc5e31ab58c0fb35315c828985d 100644
>--- a/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
>+++ b/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
>@@ -81,7 +81,6 @@
> 		41294093212E128D00AD95E7 /* libvpx_vp8_encoder.h in Headers */ = {isa = PBXBuildFile; fileRef = 4129408F212E128C00AD95E7 /* libvpx_vp8_encoder.h */; };
> 		41294094212E128D00AD95E7 /* libvpx_vp8_decoder.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41294090212E128C00AD95E7 /* libvpx_vp8_decoder.cc */; };
> 		41294095212E128D00AD95E7 /* libvpx_vp8_decoder.h in Headers */ = {isa = PBXBuildFile; fileRef = 41294091212E128C00AD95E7 /* libvpx_vp8_decoder.h */; };
>-		41299B8B2127365100B3414B /* pacer_controller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41E02CD62127363C00C27CD6 /* pacer_controller.cc */; };
> 		41299B8C2127365100B3414B /* send_time_history.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41E02CD82127363D00C27CD6 /* send_time_history.cc */; };
> 		41299B8D2127365100B3414B /* transport_feedback_adapter.h in Headers */ = {isa = PBXBuildFile; fileRef = 41E02CD72127363C00C27CD6 /* transport_feedback_adapter.h */; };
> 		41299B912127367B00B3414B /* isac_vad.c in Sources */ = {isa = PBXBuildFile; fileRef = 41299B8E2127367A00B3414B /* isac_vad.c */; };
>@@ -142,7 +141,6 @@
> 		413A22A11FE18E0600373E99 /* testbase64.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21441FE18D4C00373E99 /* testbase64.h */; };
> 		413A22A21FE18E0600373E99 /* function_view.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21451FE18D4C00373E99 /* function_view.h */; };
> 		413A22A31FE18E0600373E99 /* format_macros.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21471FE18D4D00373E99 /* format_macros.h */; };
>-		413A22A41FE18E0600373E99 /* unixfilesystem.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21481FE18D4D00373E99 /* unixfilesystem.h */; };
> 		413A22A51FE18E0600373E99 /* testechoserver.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21491FE18D4E00373E99 /* testechoserver.cc */; };
> 		413A22A71FE18E0600373E99 /* proxyserver.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A214C1FE18D4F00373E99 /* proxyserver.cc */; };
> 		413A22A81FE18E0600373E99 /* asyncinvoker-inl.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A214D1FE18D5000373E99 /* asyncinvoker-inl.h */; };
>@@ -228,16 +226,13 @@
> 		413A23381FE18E0700373E99 /* macutils.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21E41FE18DB000373E99 /* macutils.cc */; };
> 		413A23391FE18E0700373E99 /* task_queue.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21E51FE18DB100373E99 /* task_queue.h */; };
> 		413A233B1FE18E0700373E99 /* bitbuffer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21E71FE18DB100373E99 /* bitbuffer.cc */; };
>-		413A233C1FE18E0700373E99 /* pathutils.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21E81FE18DB200373E99 /* pathutils.cc */; };
> 		413A233D1FE18E0700373E99 /* physicalsocketserver.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21E91FE18DB200373E99 /* physicalsocketserver.cc */; };
> 		413A233E1FE18E0700373E99 /* byteorder.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21EA1FE18DB300373E99 /* byteorder.h */; };
> 		413A23421FE18E0700373E99 /* thread.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21EE1FE18DB400373E99 /* thread.h */; };
> 		413A23431FE18E0700373E99 /* crc32.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21EF1FE18DB500373E99 /* crc32.cc */; };
>-		413A23451FE18E0700373E99 /* optionsfile.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21F11FE18DB600373E99 /* optionsfile.h */; };
> 		413A23471FE18E0700373E99 /* swap_queue.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21F31FE18DB700373E99 /* swap_queue.h */; };
> 		413A23481FE18E0700373E99 /* dscp.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21F41FE18DB700373E99 /* dscp.h */; };
> 		413A23491FE18E0700373E99 /* timeutils.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21F51FE18DB700373E99 /* timeutils.h */; };
>-		413A234A1FE18E0700373E99 /* optionsfile.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21F61FE18DB800373E99 /* optionsfile.cc */; };
> 		413A234B1FE18E0700373E99 /* type_traits.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21F71FE18DB800373E99 /* type_traits.h */; };
> 		413A234C1FE18E0700373E99 /* random.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21F81FE18DB900373E99 /* random.h */; };
> 		413A234D1FE18E0700373E99 /* socketaddresspair.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21F91FE18DB900373E99 /* socketaddresspair.cc */; };
>@@ -255,7 +250,6 @@
> 		413A235C1FE18E0700373E99 /* bind.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22091FE18DC000373E99 /* bind.h */; };
> 		413A235D1FE18E0700373E99 /* gunit_prod.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A220A1FE18DC000373E99 /* gunit_prod.h */; };
> 		413A235E1FE18E0700373E99 /* asyncsocket.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A220B1FE18DC100373E99 /* asyncsocket.h */; };
>-		413A235F1FE18E0700373E99 /* noop.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A220C1FE18DC100373E99 /* noop.cc */; };
> 		413A23601FE18E0700373E99 /* rate_limiter.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A220D1FE18DC200373E99 /* rate_limiter.h */; };
> 		413A23611FE18E0700373E99 /* socketadapters.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A220E1FE18DC200373E99 /* socketadapters.h */; };
> 		413A23621FE18E0700373E99 /* ratetracker.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A220F1FE18DC200373E99 /* ratetracker.cc */; };
>@@ -287,7 +281,6 @@
> 		413A238D1FE18E0700373E99 /* rate_limiter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A223A1FE18DD800373E99 /* rate_limiter.cc */; };
> 		413A238E1FE18E0700373E99 /* buffer.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A223B1FE18DD900373E99 /* buffer.h */; };
> 		413A238F1FE18E0700373E99 /* event_tracer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A223C1FE18DD900373E99 /* event_tracer.cc */; };
>-		413A23901FE18E0800373E99 /* pathutils.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A223D1FE18DD900373E99 /* pathutils.h */; };
> 		413A23921FE18E0800373E99 /* sslroots.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A223F1FE18DDA00373E99 /* sslroots.h */; };
> 		413A23951FE18E0800373E99 /* httpcommon.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22421FE18DDB00373E99 /* httpcommon.h */; };
> 		413A23961FE18E0800373E99 /* asyncsocket.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22431FE18DDC00373E99 /* asyncsocket.cc */; };
>@@ -306,12 +299,10 @@
> 		413A23AE1FE18E0800373E99 /* sequenced_task_checker_impl.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A225B1FE18DE600373E99 /* sequenced_task_checker_impl.h */; };
> 		413A23AF1FE18E0800373E99 /* sslfingerprint.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A225C1FE18DE600373E99 /* sslfingerprint.h */; };
> 		413A23B11FE18E0800373E99 /* stringize_macros.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A225E1FE18DE700373E99 /* stringize_macros.h */; };
>-		413A23B51FE18E0800373E99 /* fileutils.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22621FE18DE900373E99 /* fileutils.cc */; };
> 		413A23B71FE18E0800373E99 /* event.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22641FE18DEB00373E99 /* event.cc */; };
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> 		413A23BC1FE18E0800373E99 /* flags.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22691FE18DEF00373E99 /* flags.h */; };
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> 		413A23BE1FE18E0800373E99 /* logsinks.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A226B1FE18DF000373E99 /* logsinks.cc */; };
> 		413A23C01FE18E0800373E99 /* nullsocketserver.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A226D1FE18DF100373E99 /* nullsocketserver.h */; };
> 		413A23C11FE18E0800373E99 /* sslstreamadapter.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A226E1FE18DF100373E99 /* sslstreamadapter.h */; };
>@@ -331,7 +322,6 @@
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> 		413A23D91FE18E0800373E99 /* testutils.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22861FE18DFC00373E99 /* testutils.h */; };
> 		413A23DA1FE18E0800373E99 /* filerotatingstream.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22871FE18DFD00373E99 /* filerotatingstream.cc */; };
>-		413A23DD1FE18E0800373E99 /* thread_darwin.mm in Sources */ = {isa = PBXBuildFile; fileRef = 413A228A1FE18DFE00373E99 /* thread_darwin.mm */; };
> 		413A23DE1FE18E0800373E99 /* task_queue_gcd.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A228B1FE18DFE00373E99 /* task_queue_gcd.cc */; };
> 		413A23DF1FE18E0800373E99 /* firewallsocketserver.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A228C1FE18DFE00373E99 /* firewallsocketserver.h */; };
> 		413A23E21FE18E0800373E99 /* cpu_time.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22901FE18E0000373E99 /* cpu_time.h */; };
>@@ -342,7 +332,6 @@
> 		413A23EB1FE18E0800373E99 /* location.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22991FE18E0300373E99 /* location.cc */; };
> 		413A23EC1FE18E0800373E99 /* location.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A229A1FE18E0400373E99 /* location.h */; };
> 		413A23ED1FE18E0800373E99 /* ipaddress.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A229B1FE18E0400373E99 /* ipaddress.cc */; };
>-		413A23EE1FE18E0800373E99 /* noop.mm in Sources */ = {isa = PBXBuildFile; fileRef = 413A229C1FE18E0400373E99 /* noop.mm */; };
> 		413A23EF1FE18E0800373E99 /* race_checker.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A229D1FE18E0500373E99 /* race_checker.h */; };
> 		413A23F11FE18E0800373E99 /* sslfingerprint.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A229F1FE18E0600373E99 /* sslfingerprint.cc */; };
> 		413A23F21FE18E0800373E99 /* bytebuffer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22A01FE18E0600373E99 /* bytebuffer.cc */; };
>@@ -412,11 +401,8 @@
> 		413E676D2169854B00EF37ED /* RTCWrappedNativeVideoEncoder.mm in Sources */ = {isa = PBXBuildFile; fileRef = 417954002169851F0028266B /* RTCWrappedNativeVideoEncoder.mm */; };
> 		413E67722169863B00EF37ED /* temporal_layers_checker.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413E676E2169863A00EF37ED /* temporal_layers_checker.cc */; };
> 		413E67732169863B00EF37ED /* libvpx_interface.h in Headers */ = {isa = PBXBuildFile; fileRef = 413E676F2169863A00EF37ED /* libvpx_interface.h */; };
>-		413E67742169863B00EF37ED /* vp8_temporal_layers.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413E67702169863B00EF37ED /* vp8_temporal_layers.cc */; };
> 		413E67752169863B00EF37ED /* libvpx_interface.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413E67712169863B00EF37ED /* libvpx_interface.cc */; };
> 		413E67762169866100EF37ED /* RTCVideoCodecConstants.mm in Sources */ = {isa = PBXBuildFile; fileRef = 417953FF2169851F0028266B /* RTCVideoCodecConstants.mm */; };
>-		413E67792169867A00EF37ED /* channel_receive_proxy.h in Headers */ = {isa = PBXBuildFile; fileRef = 413E67772169867A00EF37ED /* channel_receive_proxy.h */; };
>-		413E677A2169867A00EF37ED /* channel_receive_proxy.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413E67782169867A00EF37ED /* channel_receive_proxy.cc */; };
> 		413E677D216986AE00EF37ED /* rtp_header_extension_size.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413E677B216986AD00EF37ED /* rtp_header_extension_size.cc */; };
> 		413E677E216986AE00EF37ED /* rtp_header_extension_size.h in Headers */ = {isa = PBXBuildFile; fileRef = 413E677C216986AE00EF37ED /* rtp_header_extension_size.h */; };
> 		413E67812169870F00EF37ED /* svc_config.h in Headers */ = {isa = PBXBuildFile; fileRef = 413E677F2169870F00EF37ED /* svc_config.h */; };
>@@ -434,7 +420,6 @@
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> 		413E67A02169889500EF37ED /* RTCVideoEncoderQpThresholds.h in Headers */ = {isa = PBXBuildFile; fileRef = 413E679C2169889400EF37ED /* RTCVideoEncoderQpThresholds.h */; };
> 		413E67A12169889500EF37ED /* RTCRtpFragmentationHeader.h in Headers */ = {isa = PBXBuildFile; fileRef = 413E679D2169889400EF37ED /* RTCRtpFragmentationHeader.h */; };
>-		413E67A22169889500EF37ED /* RTCRtpFragmentationHeader.m in Sources */ = {isa = PBXBuildFile; fileRef = 413E679E2169889400EF37ED /* RTCRtpFragmentationHeader.m */; };
> 		413E67A32169889500EF37ED /* RTCVideoEncoderQpThresholds.m in Sources */ = {isa = PBXBuildFile; fileRef = 413E679F2169889500EF37ED /* RTCVideoEncoderQpThresholds.m */; };
> 		413E67A5216988DC00EF37ED /* video_stream_encoder_observer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413E67A4216988DC00EF37ED /* video_stream_encoder_observer.cc */; };
> 		4140B8201E4E3383007409E6 /* audio_encoder_pcm.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4140B8181E4E3383007409E6 /* audio_encoder_pcm.cc */; };
>@@ -525,7 +510,6 @@
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> 		4145E4C71EF896D700FCF6E6 /* rtcstats_objects.h in Headers */ = {isa = PBXBuildFile; fileRef = 4145E4C31EF896D200FCF6E6 /* rtcstats_objects.h */; };
> 		4145E4C81EF896D700FCF6E6 /* rtcstats.h in Headers */ = {isa = PBXBuildFile; fileRef = 4145E4C41EF896D700FCF6E6 /* rtcstats.h */; };
>-		4145E4CA1EF8CB3200FCF6E6 /* createpeerconnectionfactory.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4145E4C91EF8CB3100FCF6E6 /* createpeerconnectionfactory.cc */; };
> 		4145E4CC1EF8CB8B00FCF6E6 /* rtptransport.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4145E4CB1EF8CB8A00FCF6E6 /* rtptransport.cc */; };
> 		4145E4CE1EF8CB9500FCF6E6 /* rtptransport.h in Headers */ = {isa = PBXBuildFile; fileRef = 4145E4CD1EF8CB9400FCF6E6 /* rtptransport.h */; };
> 		4145E4D11EF8CC2000FCF6E6 /* webrtcvideoengine.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4145E4CF1EF8CC1700FCF6E6 /* webrtcvideoengine.cc */; };
>@@ -561,7 +545,6 @@
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> 		415F1FAF21272FBA00064CBF /* gain_applier.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41A08BDE21272F84001D5D7B /* gain_applier.cc */; };
> 		415F1FB021272FBA00064CBF /* gain_applier.h in Headers */ = {isa = PBXBuildFile; fileRef = 41A08BE721272F86001D5D7B /* gain_applier.h */; };
>-		415F1FB121272FBA00064CBF /* gain_curve_applier.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41A08BE021272F85001D5D7B /* gain_curve_applier.cc */; };
> 		415F1FB221272FBA00064CBF /* interpolated_gain_curve.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41A08BE121272F85001D5D7B /* interpolated_gain_curve.cc */; };
> 		415F1FB321272FBA00064CBF /* limiter.h in Headers */ = {isa = PBXBuildFile; fileRef = 41A08BEA21272F86001D5D7B /* limiter.h */; };
> 		415F1FB421272FBA00064CBF /* noise_spectrum_estimator.h in Headers */ = {isa = PBXBuildFile; fileRef = 41A08BE221272F85001D5D7B /* noise_spectrum_estimator.h */; };
>@@ -646,9 +629,6 @@
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> 		416D3BE2212731C200775F09 /* down_sampler.h in Headers */ = {isa = PBXBuildFile; fileRef = 41D6B460212731A1008F9353 /* down_sampler.h */; };
>-		416D3BE3212731C200775F09 /* fixed_gain_controller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41D6B465212731A2008F9353 /* fixed_gain_controller.cc */; };
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> 		416D3BE6212731C200775F09 /* limiter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41D6B470212731A6008F9353 /* limiter.cc */; };
> 		416D3BE7212731C200775F09 /* noise_level_estimator.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41D6B461212731A1008F9353 /* noise_level_estimator.cc */; };
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>@@ -683,9 +663,7 @@
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> 		417953D32169834F0028266B /* channel_send.cc in Sources */ = {isa = PBXBuildFile; fileRef = 417953CD2169834E0028266B /* channel_send.cc */; };
> 		417953D42169834F0028266B /* channel_send.h in Headers */ = {isa = PBXBuildFile; fileRef = 417953CE2169834E0028266B /* channel_send.h */; };
>-		417953D52169834F0028266B /* channel_send_proxy.cc in Sources */ = {isa = PBXBuildFile; fileRef = 417953CF2169834E0028266B /* channel_send_proxy.cc */; };
> 		417953D62169834F0028266B /* channel_receive.h in Headers */ = {isa = PBXBuildFile; fileRef = 417953D02169834F0028266B /* channel_receive.h */; };
>-		417953D72169834F0028266B /* channel_send_proxy.h in Headers */ = {isa = PBXBuildFile; fileRef = 417953D12169834F0028266B /* channel_send_proxy.h */; };
> 		417953D82169834F0028266B /* channel_receive.cc in Sources */ = {isa = PBXBuildFile; fileRef = 417953D22169834F0028266B /* channel_receive.cc */; };
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> 		417953DC216983910028266B /* field_trial.cc in Sources */ = {isa = PBXBuildFile; fileRef = 417953DA216983900028266B /* field_trial.cc */; };
>@@ -752,7 +730,6 @@
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> 		419241562127376F00634FCF /* fakecandidatepair.h in Headers */ = {isa = PBXBuildFile; fileRef = 4192414C2127376E00634FCF /* fakecandidatepair.h */; };
>-		419241572127376F00634FCF /* udptransport.h in Headers */ = {isa = PBXBuildFile; fileRef = 4192414D2127376E00634FCF /* udptransport.h */; };
> 		419241582127376F00634FCF /* fakepackettransport.h in Headers */ = {isa = PBXBuildFile; fileRef = 4192414E2127376E00634FCF /* fakepackettransport.h */; };
> 		419241592127376F00634FCF /* fakeicetransport.h in Headers */ = {isa = PBXBuildFile; fileRef = 4192414F2127376E00634FCF /* fakeicetransport.h */; };
> 		4192415A2127376F00634FCF /* icetransportinternal.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241502127376E00634FCF /* icetransportinternal.h */; };
>@@ -777,7 +754,6 @@
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> 		4192418B212749C800634FCF /* send_time_history.h in Headers */ = {isa = PBXBuildFile; fileRef = 41924187212749C700634FCF /* send_time_history.h */; };
> 		4192418C212749C800634FCF /* send_side_congestion_controller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41924188212749C700634FCF /* send_side_congestion_controller.cc */; };
>-		4192418D212749C800634FCF /* pacer_controller.h in Headers */ = {isa = PBXBuildFile; fileRef = 41924189212749C800634FCF /* pacer_controller.h */; };
> 		4192418E212749C800634FCF /* transport_feedback_adapter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4192418A212749C800634FCF /* transport_feedback_adapter.cc */; };
> 		419241902127581000634FCF /* fec_private_tables_bursty.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4192418F2127581000634FCF /* fec_private_tables_bursty.cc */; };
> 		419241982127586500634FCF /* rnn_vad_weights.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241932127586400634FCF /* rnn_vad_weights.h */; };
>@@ -891,8 +867,6 @@
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> 		419C832E1FE245CD0040C30F /* pacer.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83281FE245CB0040C30F /* pacer.h */; };
> 		419C83311FE245CD0040C30F /* interval_budget.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C832B1FE245CC0040C30F /* interval_budget.h */; };
>-		419C83341FE245EA0040C30F /* time_interval.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83321FE245E90040C30F /* time_interval.h */; };
>-		419C83351FE245EA0040C30F /* time_interval.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83331FE245E90040C30F /* time_interval.cc */; };
> 		419C833C1FE246230040C30F /* audio_encoder_g722_config.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83371FE246220040C30F /* audio_encoder_g722_config.h */; };
> 		419C833D1FE246230040C30F /* audio_decoder_g722.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83381FE246220040C30F /* audio_decoder_g722.h */; };
> 		419C833E1FE246230040C30F /* audio_encoder_g722.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83391FE246220040C30F /* audio_encoder_g722.cc */; };
>@@ -965,7 +939,6 @@
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> 		419C83DA1FE247F20040C30F /* packettransportinternal.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83D11FE247F00040C30F /* packettransportinternal.cc */; };
> 		419C83DB1FE247F20040C30F /* portinterface.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83D21FE247F00040C30F /* portinterface.cc */; };
>-		419C83DC1FE247F20040C30F /* udptransport.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83D31FE247F00040C30F /* udptransport.cc */; };
> 		419C83DD1FE247F20040C30F /* dtlstransport.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83D41FE247F10040C30F /* dtlstransport.h */; };
> 		419C83DE1FE247F20040C30F /* dtlstransportinternal.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83D51FE247F10040C30F /* dtlstransportinternal.cc */; };
> 		419C83DF1FE247F20040C30F /* dtlstransportinternal.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83D61FE247F10040C30F /* dtlstransportinternal.h */; };
>@@ -1078,8 +1051,6 @@
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> 		41A08BD321272EFA001D5D7B /* round_robin_packet_queue.h in Headers */ = {isa = PBXBuildFile; fileRef = 41A08BD121272EFA001D5D7B /* round_robin_packet_queue.h */; };
> 		41A08BD421272EFA001D5D7B /* round_robin_packet_queue.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41A08BD221272EFA001D5D7B /* round_robin_packet_queue.cc */; };
>-		41A08BD721272F1D001D5D7B /* congestion_window_pushback_controller.h in Headers */ = {isa = PBXBuildFile; fileRef = 41A08BD521272F1C001D5D7B /* congestion_window_pushback_controller.h */; };
>-		41A08BD821272F1D001D5D7B /* congestion_window_pushback_controller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41A08BD621272F1C001D5D7B /* congestion_window_pushback_controller.cc */; };
> 		41A391731EFC447C00C4516A /* sha1-altivec.c in Sources */ = {isa = PBXBuildFile; fileRef = 41A3914F1EFC446E00C4516A /* sha1-altivec.c */; };
> 		41A391741EFC447C00C4516A /* sha1.c in Sources */ = {isa = PBXBuildFile; fileRef = 41A391501EFC446E00C4516A /* sha1.c */; };
> 		41A391751EFC447C00C4516A /* sha256.c in Sources */ = {isa = PBXBuildFile; fileRef = 41A391511EFC446E00C4516A /* sha256.c */; };
>@@ -1368,7 +1339,6 @@
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> 		41E02CA72127352D00C27CD6 /* probe_bitrate_estimator.h in Headers */ = {isa = PBXBuildFile; fileRef = 41E02C932127352A00C27CD6 /* probe_bitrate_estimator.h */; };
> 		41E02CA82127352D00C27CD6 /* acknowledged_bitrate_estimator.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41E02C942127352A00C27CD6 /* acknowledged_bitrate_estimator.cc */; };
>-		41E02CA92127352D00C27CD6 /* goog_cc_factory.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41E02C952127352A00C27CD6 /* goog_cc_factory.cc */; };
> 		41E02CAA2127352D00C27CD6 /* goog_cc_network_control.h in Headers */ = {isa = PBXBuildFile; fileRef = 41E02C962127352A00C27CD6 /* goog_cc_network_control.h */; };
> 		41E02CAB2127352D00C27CD6 /* delay_based_bwe.h in Headers */ = {isa = PBXBuildFile; fileRef = 41E02C972127352A00C27CD6 /* delay_based_bwe.h */; };
> 		41E02CAC2127352D00C27CD6 /* bitrate_estimator.h in Headers */ = {isa = PBXBuildFile; fileRef = 41E02C982127352B00C27CD6 /* bitrate_estimator.h */; };
>@@ -1593,6 +1563,65 @@
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> 		41F9BFE92051DEEA00ABF0B9 /* turnportfactory.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F9BFE72051DEE900ABF0B9 /* turnportfactory.h */; };
> 		41F9BFEA2051DEEA00ABF0B9 /* turnportfactory.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F9BFE82051DEEA00ABF0B9 /* turnportfactory.cc */; };
>+		41FCBB0E21B1F53700A5DF27 /* create_peerconnection_factory.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41FCBB0921B1F53000A5DF27 /* create_peerconnection_factory.cc */; };
>+		41FCBB0F21B1F53700A5DF27 /* media_transport_interface.h in Headers */ = {isa = PBXBuildFile; fileRef = 41FCBB0A21B1F53100A5DF27 /* media_transport_interface.h */; };
>+		41FCBB1021B1F53700A5DF27 /* scoped_refptr.h in Headers */ = {isa = PBXBuildFile; fileRef = 41FCBB0B21B1F53200A5DF27 /* scoped_refptr.h */; };
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>+		41FCBB7A21B1FFC000A5DF27 /* ascii.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41FCBB7821B1FFC000A5DF27 /* ascii.cc */; };
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>+		41FCBB8021B2001200A5DF27 /* string_view.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41FCBB7F21B2001100A5DF27 /* string_view.cc */; };
>+		41FCBB8421B2007B00A5DF27 /* memutil.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41FCBB8221B2007B00A5DF27 /* memutil.cc */; };
>+		41FCBB8521B2007B00A5DF27 /* memutil.h in Headers */ = {isa = PBXBuildFile; fileRef = 41FCBB8321B2007B00A5DF27 /* memutil.h */; };
>+		41FCBB8921B200FD00A5DF27 /* RTCRtpFragmentationHeader.m in Sources */ = {isa = PBXBuildFile; fileRef = 413E679E2169889400EF37ED /* RTCRtpFragmentationHeader.m */; };
> 		41FE8651215C530700B62C07 /* nasm-listfmt.c in Sources */ = {isa = PBXBuildFile; fileRef = 41FE8650215C530600B62C07 /* nasm-listfmt.c */; };
> 		41FE8653215C532500B62C07 /* nasm-eval.c in Sources */ = {isa = PBXBuildFile; fileRef = 41FE8645215C526000B62C07 /* nasm-eval.c */; };
> 		41FE8654215C532500B62C07 /* nasm-pp.c in Sources */ = {isa = PBXBuildFile; fileRef = 41D12426215C52410036B057 /* nasm-pp.c */; };
>@@ -1685,8 +1714,6 @@
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> 		5C0885341E4A99D200403995 /* rdbx.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C0885221E4A99D200403995 /* rdbx.h */; };
> 		5C0885351E4A99D200403995 /* stat.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C0885231E4A99D200403995 /* stat.h */; };
>-		5C088C101E4AA44400403995 /* bundlefilter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63FA3A1E41761F002CA531 /* bundlefilter.cc */; };
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> 		5C088C121E4AA44400403995 /* channel.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63FA3D1E41761F002CA531 /* channel.cc */; };
> 		5C088C131E4AA44400403995 /* channel.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63FA3E1E41761F002CA531 /* channel.h */; };
> 		5C088C141E4AA44400403995 /* channelmanager.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63FA401E41761F002CA531 /* channelmanager.cc */; };
>@@ -1980,7 +2007,6 @@
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> 		5C4B4C1C1E431F75002651C8 /* incoming_video_stream.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4C131E431F75002651C8 /* incoming_video_stream.cc */; };
> 		5C4B4C1E1E431F75002651C8 /* video_frame_buffer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4C151E431F75002651C8 /* video_frame_buffer.cc */; };
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> 		5C4B4C201E431F75002651C8 /* video_render_frames.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4C171E431F75002651C8 /* video_render_frames.cc */; };
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>@@ -2345,8 +2371,6 @@
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> 		5CDD837B1E439A3500621E92 /* ivf_file_writer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD836D1E439A3500621E92 /* ivf_file_writer.cc */; };
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> 		5CDD83811E439A3500621E92 /* quality_scaler.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD83731E439A3500621E92 /* quality_scaler.cc */; };
> 		5CDD83821E439A3500621E92 /* quality_scaler.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CDD83741E439A3500621E92 /* quality_scaler.h */; };
> 		5CDD83851E439A3500621E92 /* vp8_header_parser.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD83771E439A3500621E92 /* vp8_header_parser.cc */; };
>@@ -2453,8 +2477,6 @@
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> 		5CDD85121E43B1EA00621E92 /* call_statistics.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD84FB1E43B1EA00621E92 /* call_statistics.cc */; };
> 		5CDD85131E43B1EA00621E92 /* call_statistics.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CDD84FC1E43B1EA00621E92 /* call_statistics.h */; };
>-		5CDD85151E43B1EA00621E92 /* codec_manager.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD84FE1E43B1EA00621E92 /* codec_manager.cc */; };
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> 		5CDD85181E43B1EA00621E92 /* rent_a_codec.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD85011E43B1EA00621E92 /* rent_a_codec.cc */; };
> 		5CDD85191E43B1EA00621E92 /* rent_a_codec.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CDD85021E43B1EA00621E92 /* rent_a_codec.h */; };
> 		5CDD852F1E43B39C00621E92 /* bitrate_allocator.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD851C1E43B39C00621E92 /* bitrate_allocator.cc */; };
>@@ -2841,8 +2863,6 @@
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> 		5CDD8A4D1E43BFB300621E92 /* red_payload_splitter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD89E41E43BFB300621E92 /* red_payload_splitter.cc */; };
> 		5CDD8A4E1E43BFB300621E92 /* red_payload_splitter.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CDD89E51E43BFB300621E92 /* red_payload_splitter.h */; };
>-		5CDD8A4F1E43BFB300621E92 /* rtcp.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD89E61E43BFB300621E92 /* rtcp.cc */; };
>-		5CDD8A501E43BFB300621E92 /* rtcp.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CDD89E71E43BFB300621E92 /* rtcp.h */; };
> 		5CDD8A511E43BFB300621E92 /* statistics_calculator.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD89E81E43BFB300621E92 /* statistics_calculator.cc */; };
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> 		5CDD8A541E43BFB300621E92 /* sync_buffer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD89EB1E43BFB300621E92 /* sync_buffer.cc */; };
>@@ -3798,7 +3818,6 @@
> 		413A21441FE18D4C00373E99 /* testbase64.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = testbase64.h; path = rtc_base/testbase64.h; sourceTree = "<group>"; };
> 		413A21451FE18D4C00373E99 /* function_view.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = function_view.h; path = rtc_base/function_view.h; sourceTree = "<group>"; };
> 		413A21471FE18D4D00373E99 /* format_macros.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = format_macros.h; path = rtc_base/format_macros.h; sourceTree = "<group>"; };
>-		413A21481FE18D4D00373E99 /* unixfilesystem.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = unixfilesystem.h; path = rtc_base/unixfilesystem.h; sourceTree = "<group>"; };
> 		413A21491FE18D4E00373E99 /* testechoserver.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = testechoserver.cc; path = rtc_base/testechoserver.cc; sourceTree = "<group>"; };
> 		413A214A1FE18D4E00373E99 /* bitrateallocationstrategy.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = bitrateallocationstrategy.cc; path = rtc_base/bitrateallocationstrategy.cc; sourceTree = "<group>"; };
> 		413A214B1FE18D4E00373E99 /* proxy_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = proxy_unittest.cc; path = rtc_base/proxy_unittest.cc; sourceTree = "<group>"; };
>@@ -3872,7 +3891,6 @@
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> 		413A219B1FE18D8B00373E99 /* socketadapters.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = socketadapters.cc; path = rtc_base/socketadapters.cc; sourceTree = "<group>"; };
>-		413A219D1FE18D8D00373E99 /* pathutils_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = pathutils_unittest.cc; path = rtc_base/pathutils_unittest.cc; sourceTree = "<group>"; };
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>@@ -3907,7 +3925,6 @@
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> 		413A21C51FE18D9F00373E99 /* helpers_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = helpers_unittest.cc; path = rtc_base/helpers_unittest.cc; sourceTree = "<group>"; };
> 		413A21C61FE18DA000373E99 /* natsocketfactory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = natsocketfactory.cc; path = rtc_base/natsocketfactory.cc; sourceTree = "<group>"; };
>-		413A21C71FE18DA100373E99 /* optionsfile_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = optionsfile_unittest.cc; path = rtc_base/optionsfile_unittest.cc; sourceTree = "<group>"; };
> 		413A21C81FE18DA100373E99 /* nat_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = nat_unittest.cc; path = rtc_base/nat_unittest.cc; sourceTree = "<group>"; };
> 		413A21C91FE18DA200373E99 /* platform_thread_types.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = platform_thread_types.h; path = rtc_base/platform_thread_types.h; sourceTree = "<group>"; };
> 		413A21CA1FE18DA300373E99 /* platform_thread.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = platform_thread.cc; path = rtc_base/platform_thread.cc; sourceTree = "<group>"; };
>@@ -3936,18 +3953,15 @@
> 		413A21E41FE18DB000373E99 /* macutils.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = macutils.cc; path = rtc_base/macutils.cc; sourceTree = "<group>"; };
> 		413A21E51FE18DB100373E99 /* task_queue.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = task_queue.h; path = rtc_base/task_queue.h; sourceTree = "<group>"; };
> 		413A21E71FE18DB100373E99 /* bitbuffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = bitbuffer.cc; path = rtc_base/bitbuffer.cc; sourceTree = "<group>"; };
>-		413A21E81FE18DB200373E99 /* pathutils.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = pathutils.cc; path = rtc_base/pathutils.cc; sourceTree = "<group>"; };
> 		413A21E91FE18DB200373E99 /* physicalsocketserver.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = physicalsocketserver.cc; path = rtc_base/physicalsocketserver.cc; sourceTree = "<group>"; };
> 		413A21EA1FE18DB300373E99 /* byteorder.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = byteorder.h; path = rtc_base/byteorder.h; sourceTree = "<group>"; };
> 		413A21EB1FE18DB300373E99 /* sslstreamadapter_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sslstreamadapter_unittest.cc; path = rtc_base/sslstreamadapter_unittest.cc; sourceTree = "<group>"; };
> 		413A21EE1FE18DB400373E99 /* thread.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = thread.h; path = rtc_base/thread.h; sourceTree = "<group>"; };
> 		413A21EF1FE18DB500373E99 /* crc32.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = crc32.cc; path = rtc_base/crc32.cc; sourceTree = "<group>"; };
> 		413A21F01FE18DB500373E99 /* memory_usage_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = memory_usage_unittest.cc; path = rtc_base/memory_usage_unittest.cc; sourceTree = "<group>"; };
>-		413A21F11FE18DB600373E99 /* optionsfile.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = optionsfile.h; path = rtc_base/optionsfile.h; sourceTree = "<group>"; };
> 		413A21F31FE18DB700373E99 /* swap_queue.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = swap_queue.h; path = rtc_base/swap_queue.h; sourceTree = "<group>"; };
> 		413A21F41FE18DB700373E99 /* dscp.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = dscp.h; path = rtc_base/dscp.h; sourceTree = "<group>"; };
> 		413A21F51FE18DB700373E99 /* timeutils.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = timeutils.h; path = rtc_base/timeutils.h; sourceTree = "<group>"; };
>-		413A21F61FE18DB800373E99 /* optionsfile.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = optionsfile.cc; path = rtc_base/optionsfile.cc; sourceTree = "<group>"; };
> 		413A21F71FE18DB800373E99 /* type_traits.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = type_traits.h; path = rtc_base/type_traits.h; sourceTree = "<group>"; };
> 		413A21F81FE18DB900373E99 /* random.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = random.h; path = rtc_base/random.h; sourceTree = "<group>"; };
> 		413A21F91FE18DB900373E99 /* socketaddresspair.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = socketaddresspair.cc; path = rtc_base/socketaddresspair.cc; sourceTree = "<group>"; };
>@@ -3967,7 +3981,6 @@
> 		413A22091FE18DC000373E99 /* bind.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = bind.h; path = rtc_base/bind.h; sourceTree = "<group>"; };
> 		413A220A1FE18DC000373E99 /* gunit_prod.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = gunit_prod.h; path = rtc_base/gunit_prod.h; sourceTree = "<group>"; };
> 		413A220B1FE18DC100373E99 /* asyncsocket.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = asyncsocket.h; path = rtc_base/asyncsocket.h; sourceTree = "<group>"; };
>-		413A220C1FE18DC100373E99 /* noop.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = noop.cc; path = rtc_base/noop.cc; sourceTree = "<group>"; };
> 		413A220D1FE18DC200373E99 /* rate_limiter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rate_limiter.h; path = rtc_base/rate_limiter.h; sourceTree = "<group>"; };
> 		413A220E1FE18DC200373E99 /* socketadapters.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = socketadapters.h; path = rtc_base/socketadapters.h; sourceTree = "<group>"; };
> 		413A220F1FE18DC200373E99 /* ratetracker.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = ratetracker.cc; path = rtc_base/ratetracker.cc; sourceTree = "<group>"; };
>@@ -4002,7 +4015,6 @@
> 		413A222F1FE18DD200373E99 /* bind_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = bind_unittest.cc; path = rtc_base/bind_unittest.cc; sourceTree = "<group>"; };
> 		413A22311FE18DD300373E99 /* stringutils_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = stringutils_unittest.cc; path = rtc_base/stringutils_unittest.cc; sourceTree = "<group>"; };
> 		413A22321FE18DD300373E99 /* compile_assert_c.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = compile_assert_c.h; path = rtc_base/compile_assert_c.h; sourceTree = "<group>"; };
>-		413A22331FE18DD400373E99 /* unixfilesystem.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = unixfilesystem.cc; path = rtc_base/unixfilesystem.cc; sourceTree = "<group>"; };
> 		413A22341FE18DD400373E99 /* ssladapter_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = ssladapter_unittest.cc; path = rtc_base/ssladapter_unittest.cc; sourceTree = "<group>"; };
> 		413A22351FE18DD500373E99 /* criticalsection.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = criticalsection.cc; path = rtc_base/criticalsection.cc; sourceTree = "<group>"; };
> 		413A22361FE18DD500373E99 /* byteorder_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = byteorder_unittest.cc; path = rtc_base/byteorder_unittest.cc; sourceTree = "<group>"; };
>@@ -4012,7 +4024,6 @@
> 		413A223A1FE18DD800373E99 /* rate_limiter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rate_limiter.cc; path = rtc_base/rate_limiter.cc; sourceTree = "<group>"; };
> 		413A223B1FE18DD900373E99 /* buffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = buffer.h; path = rtc_base/buffer.h; sourceTree = "<group>"; };
> 		413A223C1FE18DD900373E99 /* event_tracer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = event_tracer.cc; path = rtc_base/event_tracer.cc; sourceTree = "<group>"; };
>-		413A223D1FE18DD900373E99 /* pathutils.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = pathutils.h; path = rtc_base/pathutils.h; sourceTree = "<group>"; };
> 		413A223F1FE18DDA00373E99 /* sslroots.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = sslroots.h; path = rtc_base/sslroots.h; sourceTree = "<group>"; };
> 		413A22401FE18DDB00373E99 /* openssldigest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = openssldigest.cc; path = rtc_base/openssldigest.cc; sourceTree = "<group>"; };
> 		413A22421FE18DDB00373E99 /* httpcommon.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = httpcommon.h; path = rtc_base/httpcommon.h; sourceTree = "<group>"; };
>@@ -4043,7 +4054,6 @@
> 		413A225D1FE18DE600373E99 /* atomicops_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = atomicops_unittest.cc; path = rtc_base/atomicops_unittest.cc; sourceTree = "<group>"; };
> 		413A225E1FE18DE700373E99 /* stringize_macros.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = stringize_macros.h; path = rtc_base/stringize_macros.h; sourceTree = "<group>"; };
> 		413A22611FE18DE800373E99 /* cpu_time_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = cpu_time_unittest.cc; path = rtc_base/cpu_time_unittest.cc; sourceTree = "<group>"; };
>-		413A22621FE18DE900373E99 /* fileutils.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = fileutils.cc; path = rtc_base/fileutils.cc; sourceTree = "<group>"; };
> 		413A22631FE18DEA00373E99 /* opensslstreamadapter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = opensslstreamadapter.h; path = rtc_base/opensslstreamadapter.h; sourceTree = "<group>"; };
> 		413A22641FE18DEB00373E99 /* event.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = event.cc; path = rtc_base/event.cc; sourceTree = "<group>"; };
> 		413A22651FE18DEB00373E99 /* crc32_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = crc32_unittest.cc; path = rtc_base/crc32_unittest.cc; sourceTree = "<group>"; };
>@@ -4051,7 +4061,6 @@
> 		413A22671FE18DEC00373E99 /* sigslottester.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = sigslottester.h; path = rtc_base/sigslottester.h; sourceTree = "<group>"; };
> 		413A22681FE18DEE00373E99 /* network_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = network_unittest.cc; path = rtc_base/network_unittest.cc; sourceTree = "<group>"; };
> 		413A22691FE18DEF00373E99 /* flags.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = flags.h; path = rtc_base/flags.h; sourceTree = "<group>"; };
>-		413A226A1FE18DF000373E99 /* fileutils.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = fileutils.h; path = rtc_base/fileutils.h; sourceTree = "<group>"; };
> 		413A226B1FE18DF000373E99 /* logsinks.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = logsinks.cc; path = rtc_base/logsinks.cc; sourceTree = "<group>"; };
> 		413A226C1FE18DF000373E99 /* sslidentity_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sslidentity_unittest.cc; path = rtc_base/sslidentity_unittest.cc; sourceTree = "<group>"; };
> 		413A226D1FE18DF100373E99 /* nullsocketserver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = nullsocketserver.h; path = rtc_base/nullsocketserver.h; sourceTree = "<group>"; };
>@@ -4081,7 +4090,6 @@
> 		413A22861FE18DFC00373E99 /* testutils.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = testutils.h; path = rtc_base/testutils.h; sourceTree = "<group>"; };
> 		413A22871FE18DFD00373E99 /* filerotatingstream.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = filerotatingstream.cc; path = rtc_base/filerotatingstream.cc; sourceTree = "<group>"; };
> 		413A22881FE18DFD00373E99 /* rtccertificate_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rtccertificate_unittest.cc; path = rtc_base/rtccertificate_unittest.cc; sourceTree = "<group>"; };
>-		413A228A1FE18DFE00373E99 /* thread_darwin.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; name = thread_darwin.mm; path = rtc_base/thread_darwin.mm; sourceTree = "<group>"; };
> 		413A228B1FE18DFE00373E99 /* task_queue_gcd.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = task_queue_gcd.cc; path = rtc_base/task_queue_gcd.cc; sourceTree = "<group>"; };
> 		413A228C1FE18DFE00373E99 /* firewallsocketserver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = firewallsocketserver.h; path = rtc_base/firewallsocketserver.h; sourceTree = "<group>"; };
> 		413A228E1FE18DFF00373E99 /* timeutils_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = timeutils_unittest.cc; path = rtc_base/timeutils_unittest.cc; sourceTree = "<group>"; };
>@@ -4097,7 +4105,6 @@
> 		413A22991FE18E0300373E99 /* location.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = location.cc; path = rtc_base/location.cc; sourceTree = "<group>"; };
> 		413A229A1FE18E0400373E99 /* location.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = location.h; path = rtc_base/location.h; sourceTree = "<group>"; };
> 		413A229B1FE18E0400373E99 /* ipaddress.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = ipaddress.cc; path = rtc_base/ipaddress.cc; sourceTree = "<group>"; };
>-		413A229C1FE18E0400373E99 /* noop.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; name = noop.mm; path = rtc_base/noop.mm; sourceTree = "<group>"; };
> 		413A229D1FE18E0500373E99 /* race_checker.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = race_checker.h; path = rtc_base/race_checker.h; sourceTree = "<group>"; };
> 		413A229E1FE18E0500373E99 /* rate_limiter_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rate_limiter_unittest.cc; path = rtc_base/rate_limiter_unittest.cc; sourceTree = "<group>"; };
> 		413A229F1FE18E0600373E99 /* sslfingerprint.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sslfingerprint.cc; path = rtc_base/sslfingerprint.cc; sourceTree = "<group>"; };
>@@ -4165,16 +4172,12 @@
> 		413E67642169854500EF37ED /* RTCVideoEncoderVP8.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = RTCVideoEncoderVP8.mm; sourceTree = "<group>"; };
> 		413E676E2169863A00EF37ED /* temporal_layers_checker.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = temporal_layers_checker.cc; path = codecs/vp8/temporal_layers_checker.cc; sourceTree = "<group>"; };
> 		413E676F2169863A00EF37ED /* libvpx_interface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = libvpx_interface.h; path = codecs/vp8/libvpx_interface.h; sourceTree = "<group>"; };
>-		413E67702169863B00EF37ED /* vp8_temporal_layers.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = vp8_temporal_layers.cc; path = codecs/vp8/vp8_temporal_layers.cc; sourceTree = "<group>"; };
> 		413E67712169863B00EF37ED /* libvpx_interface.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = libvpx_interface.cc; path = codecs/vp8/libvpx_interface.cc; sourceTree = "<group>"; };
>-		413E67772169867A00EF37ED /* channel_receive_proxy.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = channel_receive_proxy.h; sourceTree = "<group>"; };
>-		413E67782169867A00EF37ED /* channel_receive_proxy.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = channel_receive_proxy.cc; sourceTree = "<group>"; };
> 		413E677B216986AD00EF37ED /* rtp_header_extension_size.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtp_header_extension_size.cc; sourceTree = "<group>"; };
> 		413E677C216986AE00EF37ED /* rtp_header_extension_size.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtp_header_extension_size.h; sourceTree = "<group>"; };
> 		413E677F2169870F00EF37ED /* svc_config.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = svc_config.h; path = codecs/vp9/svc_config.h; sourceTree = "<group>"; };
> 		413E67802169870F00EF37ED /* svc_config.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = svc_config.cc; path = codecs/vp9/svc_config.cc; sourceTree = "<group>"; };
> 		413E67832169877D00EF37ED /* RTCVideoEncoderSettings.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCVideoEncoderSettings.h; sourceTree = "<group>"; };
>-		413E67842169877D00EF37ED /* RTCVideoEncoderSettings.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = RTCVideoEncoderSettings.mm; sourceTree = "<group>"; };
> 		413E6788216987B900EF37ED /* RTCCVPixelBuffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCCVPixelBuffer.h; sourceTree = "<group>"; };
> 		413E6789216987B900EF37ED /* RTCCVPixelBuffer.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = RTCCVPixelBuffer.mm; sourceTree = "<group>"; };
> 		413E678C216987DB00EF37ED /* RTCVideoFrame.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCVideoFrame.h; sourceTree = "<group>"; };
>@@ -4308,7 +4311,6 @@
> 		4145E4C21EF896CE00FCF6E6 /* rtcstatsreport.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rtcstatsreport.h; path = stats/rtcstatsreport.h; sourceTree = "<group>"; };
> 		4145E4C31EF896D200FCF6E6 /* rtcstats_objects.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rtcstats_objects.h; path = stats/rtcstats_objects.h; sourceTree = "<group>"; };
> 		4145E4C41EF896D700FCF6E6 /* rtcstats.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rtcstats.h; path = stats/rtcstats.h; sourceTree = "<group>"; };
>-		4145E4C91EF8CB3100FCF6E6 /* createpeerconnectionfactory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = createpeerconnectionfactory.cc; sourceTree = "<group>"; };
> 		4145E4CB1EF8CB8A00FCF6E6 /* rtptransport.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtptransport.cc; sourceTree = "<group>"; };
> 		4145E4CD1EF8CB9400FCF6E6 /* rtptransport.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtptransport.h; sourceTree = "<group>"; };
> 		4145E4CF1EF8CC1700FCF6E6 /* webrtcvideoengine.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = webrtcvideoengine.cc; path = engine/webrtcvideoengine.cc; sourceTree = "<group>"; };
>@@ -4455,9 +4457,7 @@
> 		417953B3216982420028266B /* RTCDefaultVideoDecoderFactory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RTCDefaultVideoDecoderFactory.h; sourceTree = "<group>"; };
> 		417953CD2169834E0028266B /* channel_send.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = channel_send.cc; sourceTree = "<group>"; };
> 		417953CE2169834E0028266B /* channel_send.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = channel_send.h; sourceTree = "<group>"; };
>-		417953CF2169834E0028266B /* channel_send_proxy.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = channel_send_proxy.cc; sourceTree = "<group>"; };
> 		417953D02169834F0028266B /* channel_receive.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = channel_receive.h; sourceTree = "<group>"; };
>-		417953D12169834F0028266B /* channel_send_proxy.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = channel_send_proxy.h; sourceTree = "<group>"; };
> 		417953D22169834F0028266B /* channel_receive.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = channel_receive.cc; sourceTree = "<group>"; };
> 		417953D9216983900028266B /* metrics.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = metrics.cc; path = source/metrics.cc; sourceTree = "<group>"; };
> 		417953DA216983900028266B /* field_trial.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = field_trial.cc; path = source/field_trial.cc; sourceTree = "<group>"; };
>@@ -4536,7 +4536,6 @@
> 		4192414A2127376D00634FCF /* basicasyncresolverfactory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = basicasyncresolverfactory.cc; sourceTree = "<group>"; };
> 		4192414B2127376D00634FCF /* regatheringcontroller.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = regatheringcontroller.h; sourceTree = "<group>"; };
> 		4192414C2127376E00634FCF /* fakecandidatepair.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fakecandidatepair.h; sourceTree = "<group>"; };
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> 		4192414E2127376E00634FCF /* fakepackettransport.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fakepackettransport.h; sourceTree = "<group>"; };
> 		4192414F2127376E00634FCF /* fakeicetransport.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fakeicetransport.h; sourceTree = "<group>"; };
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>@@ -4561,7 +4560,6 @@
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> 		41924188212749C700634FCF /* send_side_congestion_controller.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = send_side_congestion_controller.cc; sourceTree = "<group>"; };
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> 		4192418A212749C800634FCF /* transport_feedback_adapter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = transport_feedback_adapter.cc; sourceTree = "<group>"; };
> 		4192418F2127581000634FCF /* fec_private_tables_bursty.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = fec_private_tables_bursty.cc; sourceTree = "<group>"; };
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> 		419C83371FE246220040C30F /* audio_encoder_g722_config.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_encoder_g722_config.h; path = audio_codecs/g722/audio_encoder_g722_config.h; sourceTree = "<group>"; };
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> 		419C83D21FE247F00040C30F /* portinterface.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = portinterface.cc; sourceTree = "<group>"; };
>-		419C83D31FE247F00040C30F /* udptransport.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = udptransport.cc; sourceTree = "<group>"; };
> 		419C83D41FE247F10040C30F /* dtlstransport.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = dtlstransport.h; sourceTree = "<group>"; };
> 		419C83D51FE247F10040C30F /* dtlstransportinternal.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = dtlstransportinternal.cc; sourceTree = "<group>"; };
> 		419C83D61FE247F10040C30F /* dtlstransportinternal.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = dtlstransportinternal.h; sourceTree = "<group>"; };
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> 		41A08BD121272EFA001D5D7B /* round_robin_packet_queue.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = round_robin_packet_queue.h; sourceTree = "<group>"; };
> 		41A08BD221272EFA001D5D7B /* round_robin_packet_queue.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = round_robin_packet_queue.cc; sourceTree = "<group>"; };
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> 		41A08BDA21272F84001D5D7B /* compute_interpolated_gain_curve.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = compute_interpolated_gain_curve.h; sourceTree = "<group>"; };
> 		41A08BDB21272F84001D5D7B /* adaptive_digital_gain_applier.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = adaptive_digital_gain_applier.h; sourceTree = "<group>"; };
> 		41A08BDC21272F84001D5D7B /* adaptive_mode_level_estimator.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = adaptive_mode_level_estimator.h; sourceTree = "<group>"; };
> 		41A08BDD21272F84001D5D7B /* vector_float_frame.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = vector_float_frame.h; sourceTree = "<group>"; };
> 		41A08BDE21272F84001D5D7B /* gain_applier.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = gain_applier.cc; sourceTree = "<group>"; };
> 		41A08BDF21272F84001D5D7B /* fixed_digital_level_estimator.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = fixed_digital_level_estimator.h; sourceTree = "<group>"; };
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> 		41A08BE121272F85001D5D7B /* interpolated_gain_curve.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = interpolated_gain_curve.cc; sourceTree = "<group>"; };
> 		41A08BE221272F85001D5D7B /* noise_spectrum_estimator.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = noise_spectrum_estimator.h; sourceTree = "<group>"; };
> 		41A08BE321272F85001D5D7B /* vad_with_level.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = vad_with_level.h; sourceTree = "<group>"; };
>@@ -5176,11 +5168,8 @@
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> 		41D6B464212731A2008F9353 /* adaptive_agc.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = adaptive_agc.h; sourceTree = "<group>"; };
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> 		41D6B466212731A2008F9353 /* vector_float_frame.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = vector_float_frame.cc; sourceTree = "<group>"; };
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> 		41D6B468212731A3008F9353 /* saturation_protector.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = saturation_protector.cc; sourceTree = "<group>"; };
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> 		41D6B46A212731A3008F9353 /* noise_spectrum_estimator.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = noise_spectrum_estimator.cc; sourceTree = "<group>"; };
> 		41D6B46B212731A3008F9353 /* saturation_protector.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = saturation_protector.h; sourceTree = "<group>"; };
> 		41D6B46C212731A4008F9353 /* vad_with_level.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = vad_with_level.cc; sourceTree = "<group>"; };
>@@ -5231,7 +5220,6 @@
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> 		41E02C932127352A00C27CD6 /* probe_bitrate_estimator.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = probe_bitrate_estimator.h; sourceTree = "<group>"; };
> 		41E02C942127352A00C27CD6 /* acknowledged_bitrate_estimator.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = acknowledged_bitrate_estimator.cc; sourceTree = "<group>"; };
>-		41E02C952127352A00C27CD6 /* goog_cc_factory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = goog_cc_factory.cc; sourceTree = "<group>"; };
> 		41E02C962127352A00C27CD6 /* goog_cc_network_control.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = goog_cc_network_control.h; sourceTree = "<group>"; };
> 		41E02C972127352A00C27CD6 /* delay_based_bwe.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = delay_based_bwe.h; sourceTree = "<group>"; };
> 		41E02C982127352B00C27CD6 /* bitrate_estimator.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = bitrate_estimator.h; sourceTree = "<group>"; };
>@@ -5260,7 +5248,6 @@
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> 		41E02CD12127360600C27CD6 /* sample_counter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = sample_counter.h; path = rtc_base/numerics/sample_counter.h; sourceTree = "<group>"; };
> 		41E02CD22127360700C27CD6 /* sample_counter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sample_counter.cc; path = rtc_base/numerics/sample_counter.cc; sourceTree = "<group>"; };
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> 		41E02CD72127363C00C27CD6 /* transport_feedback_adapter.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = transport_feedback_adapter.h; sourceTree = "<group>"; };
> 		41E02CD82127363D00C27CD6 /* send_time_history.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = send_time_history.cc; sourceTree = "<group>"; };
> 		41EA53A41EFC2BFD002FF04C /* hmac.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; path = hmac.c; sourceTree = "<group>"; };
>@@ -5465,8 +5452,8 @@
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> 		41F9BFC42051CAE400ABF0B9 /* signal_processing_library.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = signal_processing_library.h; path = signal_processing/include/signal_processing_library.h; sourceTree = "<group>"; };
> 		41F9BFC62051DCE800ABF0B9 /* webrtc_vad.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = webrtc_vad.h; path = vad/include/webrtc_vad.h; sourceTree = "<group>"; };
>-		41F9BFC92051DDA200ABF0B9 /* alr_experiment.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = alr_experiment.cc; path = rtc_base/experiments/alr_experiment.cc; sourceTree = "<group>"; };
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>+		41F9BFC92051DDA200ABF0B9 /* alr_experiment.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = alr_experiment.cc; sourceTree = "<group>"; };
>+		41F9BFCA2051DDA200ABF0B9 /* alr_experiment.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = alr_experiment.h; sourceTree = "<group>"; };
> 		41F9BFCD2051DDE400ABF0B9 /* fft_buffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fft_buffer.h; sourceTree = "<group>"; };
> 		41F9BFCE2051DDE400ABF0B9 /* matrix_buffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = matrix_buffer.cc; sourceTree = "<group>"; };
> 		41F9BFCF2051DDE400ABF0B9 /* vector_buffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = vector_buffer.cc; sourceTree = "<group>"; };
>@@ -5481,6 +5468,65 @@
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>+		41FCBB0921B1F53000A5DF27 /* create_peerconnection_factory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = create_peerconnection_factory.cc; sourceTree = "<group>"; };
>+		41FCBB0A21B1F53100A5DF27 /* media_transport_interface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = media_transport_interface.h; sourceTree = "<group>"; };
>+		41FCBB0B21B1F53200A5DF27 /* scoped_refptr.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = scoped_refptr.h; sourceTree = "<group>"; };
>+		41FCBB0C21B1F53200A5DF27 /* media_transport_interface.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = media_transport_interface.cc; sourceTree = "<group>"; };
>+		41FCBB0D21B1F53600A5DF27 /* create_peerconnection_factory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = create_peerconnection_factory.h; sourceTree = "<group>"; };
>+		41FCBB1321B1F7AA00A5DF27 /* moving_average.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = moving_average.cc; path = rtc_base/numerics/moving_average.cc; sourceTree = "<group>"; };
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>+		41FCBB2621B1F87E00A5DF27 /* sent_packet.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = sent_packet.h; sourceTree = "<group>"; };
>+		41FCBB2A21B1F8B600A5DF27 /* cryptooptions.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = cryptooptions.h; sourceTree = "<group>"; };
>+		41FCBB2B21B1F8B600A5DF27 /* framedecryptorinterface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = framedecryptorinterface.h; sourceTree = "<group>"; };
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>+		41FCBB3F21B1FDB800A5DF27 /* congestion_window_pushback_controller.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = congestion_window_pushback_controller.cc; sourceTree = "<group>"; };
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>+		41FCBB4421B1FDEF00A5DF27 /* mdns_message.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = mdns_message.cc; sourceTree = "<group>"; };
>+		41FCBB4521B1FDEF00A5DF27 /* icecredentialsiterator.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = icecredentialsiterator.h; sourceTree = "<group>"; };
>+		41FCBB4B21B1FE4400A5DF27 /* cpu_speed_experiment.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = cpu_speed_experiment.h; sourceTree = "<group>"; };
>+		41FCBB4C21B1FE4400A5DF27 /* jitter_upper_bound_experiment.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = jitter_upper_bound_experiment.cc; sourceTree = "<group>"; };
>+		41FCBB4D21B1FE4500A5DF27 /* cpu_speed_experiment.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = cpu_speed_experiment.cc; sourceTree = "<group>"; };
>+		41FCBB4E21B1FE4500A5DF27 /* jitter_upper_bound_experiment.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = jitter_upper_bound_experiment.h; sourceTree = "<group>"; };
>+		41FCBB4F21B1FE4500A5DF27 /* normalize_simulcast_size_experiment.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = normalize_simulcast_size_experiment.h; sourceTree = "<group>"; };
>+		41FCBB5021B1FE4500A5DF27 /* normalize_simulcast_size_experiment.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = normalize_simulcast_size_experiment.cc; sourceTree = "<group>"; };
>+		41FCBB5721B1FE6D00A5DF27 /* frame_dumping_decoder.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = frame_dumping_decoder.cc; sourceTree = "<group>"; };
>+		41FCBB5821B1FE6E00A5DF27 /* frame_dumping_decoder.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = frame_dumping_decoder.h; sourceTree = "<group>"; };
>+		41FCBB5B21B1FEA300A5DF27 /* bitrate_controller.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = bitrate_controller.cc; path = bitrate_controller/bitrate_controller.cc; sourceTree = "<group>"; };
>+		41FCBB5D21B1FEC400A5DF27 /* goog_cc_factory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = goog_cc_factory.h; sourceTree = "<group>"; };
>+		41FCBB5E21B1FEC400A5DF27 /* goog_cc_factory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = goog_cc_factory.cc; sourceTree = "<group>"; };
>+		41FCBB6121B1FEF500A5DF27 /* create_vp8_temporal_layers.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = create_vp8_temporal_layers.h; path = video_codecs/create_vp8_temporal_layers.h; sourceTree = "<group>"; };
>+		41FCBB6221B1FEF500A5DF27 /* vp8_temporal_layers.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = vp8_temporal_layers.h; path = video_codecs/vp8_temporal_layers.h; sourceTree = "<group>"; };
>+		41FCBB6321B1FEF500A5DF27 /* create_vp8_temporal_layers.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = create_vp8_temporal_layers.cc; path = video_codecs/create_vp8_temporal_layers.cc; sourceTree = "<group>"; };
>+		41FCBB6721B1FF1B00A5DF27 /* loss_based_bandwidth_estimation.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = loss_based_bandwidth_estimation.h; path = bitrate_controller/loss_based_bandwidth_estimation.h; sourceTree = "<group>"; };
>+		41FCBB6821B1FF1B00A5DF27 /* loss_based_bandwidth_estimation.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = loss_based_bandwidth_estimation.cc; path = bitrate_controller/loss_based_bandwidth_estimation.cc; sourceTree = "<group>"; };
>+		41FCBB6B21B1FF3900A5DF27 /* key_derivation.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = key_derivation.cc; path = rtc_base/key_derivation.cc; sourceTree = "<group>"; };
>+		41FCBB6C21B1FF3A00A5DF27 /* key_derivation.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = key_derivation.h; path = rtc_base/key_derivation.h; sourceTree = "<group>"; };
>+		41FCBB6F21B1FF7400A5DF27 /* hdr_metadata.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = hdr_metadata.cc; sourceTree = "<group>"; };
>+		41FCBB7021B1FF7400A5DF27 /* hdr_metadata.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = hdr_metadata.h; sourceTree = "<group>"; };
>+		41FCBB7321B1FFA400A5DF27 /* cocoa_threading.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = cocoa_threading.mm; sourceTree = "<group>"; };
>+		41FCBB7421B1FFA400A5DF27 /* cocoa_threading.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = cocoa_threading.h; sourceTree = "<group>"; };
>+		41FCBB7721B1FFBF00A5DF27 /* ascii.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ascii.h; sourceTree = "<group>"; };
>+		41FCBB7821B1FFC000A5DF27 /* ascii.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = ascii.cc; sourceTree = "<group>"; };
>+		41FCBB7B21B1FFF500A5DF27 /* openssl_key_derivation_hkdf.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = openssl_key_derivation_hkdf.cc; path = rtc_base/openssl_key_derivation_hkdf.cc; sourceTree = "<group>"; };
>+		41FCBB7C21B1FFF500A5DF27 /* openssl_key_derivation_hkdf.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = openssl_key_derivation_hkdf.h; path = rtc_base/openssl_key_derivation_hkdf.h; sourceTree = "<group>"; };
>+		41FCBB7F21B2001100A5DF27 /* string_view.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = string_view.cc; sourceTree = "<group>"; };
>+		41FCBB8221B2007B00A5DF27 /* memutil.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = memutil.cc; sourceTree = "<group>"; };
>+		41FCBB8321B2007B00A5DF27 /* memutil.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = memutil.h; sourceTree = "<group>"; };
>+		41FCBB8721B200DB00A5DF27 /* RTCVideoEncoderSettings.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = RTCVideoEncoderSettings.m; sourceTree = "<group>"; };
> 		41FE8644215C525F00B62C07 /* genversion.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = genversion.c; sourceTree = "<group>"; };
> 		41FE8645215C526000B62C07 /* nasm-eval.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = "nasm-eval.c"; sourceTree = "<group>"; };
> 		41FE8646215C526000B62C07 /* nasm-eval.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = "nasm-eval.h"; sourceTree = "<group>"; };
>@@ -5887,7 +5933,6 @@
> 		5C4B4C121E431F75002651C8 /* i420_buffer_pool.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = i420_buffer_pool.cc; sourceTree = "<group>"; };
> 		5C4B4C131E431F75002651C8 /* incoming_video_stream.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = incoming_video_stream.cc; sourceTree = "<group>"; };
> 		5C4B4C151E431F75002651C8 /* video_frame_buffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = video_frame_buffer.cc; sourceTree = "<group>"; };
>-		5C4B4C161E431F75002651C8 /* video_frame.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = video_frame.cc; sourceTree = "<group>"; };
> 		5C4B4C171E431F75002651C8 /* video_render_frames.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = video_render_frames.cc; sourceTree = "<group>"; };
> 		5C4B4C181E431F75002651C8 /* video_render_frames.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_render_frames.h; sourceTree = "<group>"; };
> 		5C4B4C241E431F9C002651C8 /* audio_converter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = audio_converter.cc; sourceTree = "<group>"; };
>@@ -6028,8 +6073,6 @@
> 		5C63F9D91E4174F6002CA531 /* turnserver.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = turnserver.cc; sourceTree = "<group>"; };
> 		5C63F9DA1E4174F6002CA531 /* turnserver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = turnserver.h; sourceTree = "<group>"; };
> 		5C63F9DB1E4174F6002CA531 /* udpport.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = udpport.h; sourceTree = "<group>"; };
>-		5C63FA3A1E41761F002CA531 /* bundlefilter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = bundlefilter.cc; sourceTree = "<group>"; };
>-		5C63FA3B1E41761F002CA531 /* bundlefilter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = bundlefilter.h; sourceTree = "<group>"; };
> 		5C63FA3D1E41761F002CA531 /* channel.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = channel.cc; sourceTree = "<group>"; };
> 		5C63FA3E1E41761F002CA531 /* channel.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = channel.h; sourceTree = "<group>"; };
> 		5C63FA401E41761F002CA531 /* channelmanager.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = channelmanager.cc; sourceTree = "<group>"; };
>@@ -6234,8 +6277,6 @@
> 		5CDD836C1E439A3500621E92 /* frame_dropper.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = frame_dropper.h; sourceTree = "<group>"; };
> 		5CDD836D1E439A3500621E92 /* ivf_file_writer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = ivf_file_writer.cc; sourceTree = "<group>"; };
> 		5CDD836E1E439A3500621E92 /* ivf_file_writer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ivf_file_writer.h; sourceTree = "<group>"; };
>-		5CDD836F1E439A3500621E92 /* moving_average.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = moving_average.cc; sourceTree = "<group>"; };
>-		5CDD83701E439A3500621E92 /* moving_average.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = moving_average.h; sourceTree = "<group>"; };
> 		5CDD83731E439A3500621E92 /* quality_scaler.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = quality_scaler.cc; sourceTree = "<group>"; };
> 		5CDD83741E439A3500621E92 /* quality_scaler.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = quality_scaler.h; sourceTree = "<group>"; };
> 		5CDD83771E439A3500621E92 /* vp8_header_parser.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = vp8_header_parser.cc; sourceTree = "<group>"; };
>@@ -6346,8 +6387,6 @@
> 		5CDD84F91E43B1EA00621E92 /* audio_coding_module.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = audio_coding_module.cc; sourceTree = "<group>"; };
> 		5CDD84FB1E43B1EA00621E92 /* call_statistics.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = call_statistics.cc; sourceTree = "<group>"; };
> 		5CDD84FC1E43B1EA00621E92 /* call_statistics.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = call_statistics.h; sourceTree = "<group>"; };
>-		5CDD84FE1E43B1EA00621E92 /* codec_manager.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = codec_manager.cc; sourceTree = "<group>"; };
>-		5CDD84FF1E43B1EA00621E92 /* codec_manager.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = codec_manager.h; sourceTree = "<group>"; };
> 		5CDD85011E43B1EA00621E92 /* rent_a_codec.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rent_a_codec.cc; sourceTree = "<group>"; };
> 		5CDD85021E43B1EA00621E92 /* rent_a_codec.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rent_a_codec.h; sourceTree = "<group>"; };
> 		5CDD851C1E43B39C00621E92 /* bitrate_allocator.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = bitrate_allocator.cc; path = call/bitrate_allocator.cc; sourceTree = "<group>"; };
>@@ -6734,8 +6773,6 @@
> 		5CDD89E21E43BFB300621E92 /* random_vector.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = random_vector.h; sourceTree = "<group>"; };
> 		5CDD89E41E43BFB300621E92 /* red_payload_splitter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = red_payload_splitter.cc; sourceTree = "<group>"; };
> 		5CDD89E51E43BFB300621E92 /* red_payload_splitter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = red_payload_splitter.h; sourceTree = "<group>"; };
>-		5CDD89E61E43BFB300621E92 /* rtcp.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtcp.cc; sourceTree = "<group>"; };
>-		5CDD89E71E43BFB300621E92 /* rtcp.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtcp.h; sourceTree = "<group>"; };
> 		5CDD89E81E43BFB300621E92 /* statistics_calculator.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = statistics_calculator.cc; sourceTree = "<group>"; };
> 		5CDD89E91E43BFB300621E92 /* statistics_calculator.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = statistics_calculator.h; sourceTree = "<group>"; };
> 		5CDD89EB1E43BFB300621E92 /* sync_buffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = sync_buffer.cc; sourceTree = "<group>"; };
>@@ -7643,6 +7680,12 @@
> 		4106D56B216C2B91001E3C40 /* strings */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41FCBB8121B2005A00A5DF27 /* internal */,
>+				41FCBB7821B1FFC000A5DF27 /* ascii.cc */,
>+				41FCBB7721B1FFBF00A5DF27 /* ascii.h */,
>+				41FCBB3A21B1FD3F00A5DF27 /* match.cc */,
>+				41FCBB3B21B1FD3F00A5DF27 /* match.h */,
>+				41FCBB7F21B2001100A5DF27 /* string_view.cc */,
> 				4106D56C216C2BA9001E3C40 /* string_view.h */,
> 			);
> 			path = strings;
>@@ -7713,23 +7756,26 @@
> 				41DDB27621267AC100296D47 /* builtin_video_decoder_factory.h */,
> 				41E02CC22127358700C27CD6 /* builtin_video_encoder_factory.cc */,
> 				41DDB27721267AC100296D47 /* builtin_video_encoder_factory.h */,
>+				41FCBB6321B1FEF500A5DF27 /* create_vp8_temporal_layers.cc */,
>+				41FCBB6121B1FEF500A5DF27 /* create_vp8_temporal_layers.h */,
> 				41DDB27A21267AC200296D47 /* sdp_video_format.cc */,
> 				4145F6191FE1F38B00EB9CAF /* sdp_video_format.h */,
> 				41DDB27021267AC000296D47 /* video_codec.cc */,
> 				41DDB27121267AC000296D47 /* video_codec.h */,
>+				41DDB27221267AC000296D47 /* video_decoder.cc */,
>+				4145E48B1EF88B9600FCF6E6 /* video_decoder.h */,
> 				417953F0216984550028266B /* video_decoder_factory.cc */,
> 				4145F61A1FE1F38C00EB9CAF /* video_decoder_factory.h */,
> 				41DDB27421267AC100296D47 /* video_decoder_software_fallback_wrapper.cc */,
> 				41DDB27821267AC100296D47 /* video_decoder_software_fallback_wrapper.h */,
>-				41DDB27221267AC000296D47 /* video_decoder.cc */,
>-				4145E48B1EF88B9600FCF6E6 /* video_decoder.h */,
>+				4145F61C1FE1F38D00EB9CAF /* video_encoder.cc */,
>+				4145E48C1EF88B9D00FCF6E6 /* video_encoder.h */,
> 				41E02CC32127358800C27CD6 /* video_encoder_config.cc */,
> 				41DDB27B21267AC200296D47 /* video_encoder_config.h */,
> 				4145F61B1FE1F38D00EB9CAF /* video_encoder_factory.h */,
> 				41DDB27521267AC100296D47 /* video_encoder_software_fallback_wrapper.cc */,
> 				41DDB27921267AC200296D47 /* video_encoder_software_fallback_wrapper.h */,
>-				4145F61C1FE1F38D00EB9CAF /* video_encoder.cc */,
>-				4145E48C1EF88B9D00FCF6E6 /* video_encoder.h */,
>+				41FCBB6221B1FEF500A5DF27 /* vp8_temporal_layers.h */,
> 			);
> 			name = video_codecs;
> 			sourceTree = "<group>";
>@@ -7790,7 +7836,7 @@
> 			children = (
> 				413A21801FE18D7A00373E99 /* DEPS */,
> 				419241652127387900634FCF /* experiments */,
>-				41F9BFC82051DD8E00ABF0B9 /* experiments */,
>+				41FCBB2421B1F85D00A5DF27 /* network */,
> 				419C83411FE246380040C30F /* numerics */,
> 				413A21FD1FE18DBB00373E99 /* OWNERS */,
> 				41DDB267212679AE00296D47 /* strings */,
>@@ -7881,8 +7927,6 @@
> 				413A22871FE18DFD00373E99 /* filerotatingstream.cc */,
> 				413A218C1FE18D8300373E99 /* filerotatingstream.h */,
> 				413A225A1FE18DE500373E99 /* filerotatingstream_unittest.cc */,
>-				413A22621FE18DE900373E99 /* fileutils.cc */,
>-				413A226A1FE18DF000373E99 /* fileutils.h */,
> 				413A22171FE18DC600373E99 /* firewallsocketserver.cc */,
> 				413A228C1FE18DFE00373E99 /* firewallsocketserver.h */,
> 				413A21611FE18D5B00373E99 /* flags.cc */,
>@@ -7907,6 +7951,8 @@
> 				413A214E1FE18D5000373E99 /* ipaddress.h */,
> 				413A216A1FE18D6700373E99 /* ipaddress_unittest.cc */,
> 				413A22121FE18DC400373E99 /* keep_ref_until_done.h */,
>+				41FCBB6B21B1FF3900A5DF27 /* key_derivation.cc */,
>+				41FCBB6C21B1FF3A00A5DF27 /* key_derivation.h */,
> 				413A22991FE18E0300373E99 /* location.cc */,
> 				413A229A1FE18E0400373E99 /* location.h */,
> 				413A21721FE18D6D00373E99 /* logging.cc */,
>@@ -7947,14 +7993,14 @@
> 				413A22381FE18DD600373E99 /* networkmonitor.cc */,
> 				413A22291FE18DCF00373E99 /* networkmonitor.h */,
> 				413A21691FE18D6500373E99 /* networkroute.h */,
>-				413A220C1FE18DC100373E99 /* noop.cc */,
>-				413A229C1FE18E0400373E99 /* noop.mm */,
> 				413A227F1FE18DF900373E99 /* nullsocketserver.cc */,
> 				413A226D1FE18DF100373E99 /* nullsocketserver.h */,
> 				413A227C1FE18DF800373E99 /* nullsocketserver_unittest.cc */,
> 				413A21E21FE18DAF00373E99 /* onetimeevent.h */,
> 				413A21621FE18D5D00373E99 /* onetimeevent_unittest.cc */,
> 				413A22751FE18DF600373E99 /* openssl.h */,
>+				41FCBB7B21B1FFF500A5DF27 /* openssl_key_derivation_hkdf.cc */,
>+				41FCBB7C21B1FFF500A5DF27 /* openssl_key_derivation_hkdf.h */,
> 				413A219A1FE18D8B00373E99 /* openssladapter.cc */,
> 				413A21DB1FE18DAC00373E99 /* openssladapter.h */,
> 				413A227E1FE18DF900373E99 /* openssladapter_unittest.cc */,
>@@ -7970,12 +8016,6 @@
> 				413A22631FE18DEA00373E99 /* opensslstreamadapter.h */,
> 				41DDB25F212679A200296D47 /* opensslutility.cc */,
> 				41DDB25B2126799F00296D47 /* opensslutility.h */,
>-				413A21F61FE18DB800373E99 /* optionsfile.cc */,
>-				413A21F11FE18DB600373E99 /* optionsfile.h */,
>-				413A21C71FE18DA100373E99 /* optionsfile_unittest.cc */,
>-				413A21E81FE18DB200373E99 /* pathutils.cc */,
>-				413A223D1FE18DD900373E99 /* pathutils.h */,
>-				413A219D1FE18D8D00373E99 /* pathutils_unittest.cc */,
> 				413A21E91FE18DB200373E99 /* physicalsocketserver.cc */,
> 				413A22981FE18E0300373E99 /* physicalsocketserver.h */,
> 				413A227B1FE18DF800373E99 /* physicalsocketserver_unittest.cc */,
>@@ -8100,7 +8140,6 @@
> 				413A21FC1FE18DBB00373E99 /* thread_checker_impl.cc */,
> 				413A215E1FE18D5900373E99 /* thread_checker_impl.h */,
> 				413A21B61FE18D9800373E99 /* thread_checker_unittest.cc */,
>-				413A228A1FE18DFE00373E99 /* thread_darwin.mm */,
> 				413A22261FE18DCD00373E99 /* thread_unittest.cc */,
> 				413A22281FE18DCE00373E99 /* timestampaligner.cc */,
> 				413A22011FE18DBD00373E99 /* timestampaligner.h */,
>@@ -8111,8 +8150,6 @@
> 				413A21AB1FE18D9300373E99 /* trace_event.h */,
> 				413A21F71FE18DB800373E99 /* type_traits.h */,
> 				413A224C1FE18DE000373E99 /* unittest_main.cc */,
>-				413A22331FE18DD400373E99 /* unixfilesystem.cc */,
>-				413A21481FE18D4D00373E99 /* unixfilesystem.h */,
> 				413A21DA1FE18DAC00373E99 /* virtualsocket_unittest.cc */,
> 				413A21BE1FE18D9C00373E99 /* virtualsocketserver.cc */,
> 				413A21631FE18D5E00373E99 /* virtualsocketserver.h */,
>@@ -8205,11 +8242,11 @@
> 		413AD19F21265ADF003F7263 /* absl */ = {
> 			isa = PBXGroup;
> 			children = (
>-				4106D56B216C2B91001E3C40 /* strings */,
> 				413AD1A021265AFA003F7263 /* algorithm */,
> 				413AD1A621265B35003F7263 /* base */,
> 				41DDB24521265B8200296D47 /* container */,
> 				41A08BB621268A6A001D5D7B /* memory */,
>+				4106D56B216C2B91001E3C40 /* strings */,
> 				41DDB24A21265BA800296D47 /* types */,
> 			);
> 			name = absl;
>@@ -8228,6 +8265,7 @@
> 		413AD1A621265B35003F7263 /* base */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41FCBB1F21B1F81400A5DF27 /* internal */,
> 				413AD1A921265B59003F7263 /* casts.h */,
> 				413AD1A821265B59003F7263 /* dynamic_annotations.cc */,
> 				413AD1AA21265B59003F7263 /* macros.h */,
>@@ -8250,9 +8288,9 @@
> 		413E67A62169893600EF37ED /* video_frame_buffer */ = {
> 			isa = PBXGroup;
> 			children = (
>+				413E67A82169894C00EF37ED /* RTCNativeI420Buffer+Private.h */,
> 				413E67A92169894C00EF37ED /* RTCNativeI420Buffer.h */,
> 				413E67A72169894B00EF37ED /* RTCNativeI420Buffer.mm */,
>-				413E67A82169894C00EF37ED /* RTCNativeI420Buffer+Private.h */,
> 				413E67AA2169894C00EF37ED /* RTCNativeMutableI420Buffer.h */,
> 				4144B3D021698966004363AC /* RTCNativeMutableI420Buffer.mm */,
> 			);
>@@ -8370,9 +8408,9 @@
> 			children = (
> 				4144B3F4216B081B004363AC /* RTCEncodedImage+Private.h */,
> 				4144B3F3216B081A004363AC /* RTCEncodedImage+Private.mm */,
>+				41889D74216BB7B6004614DD /* RTCRtpEncodingParameters+Private.h */,
> 				414FB869216BB767001F5492 /* RTCRtpEncodingParameters.h */,
> 				414FB86A216BB768001F5492 /* RTCRtpEncodingParameters.mm */,
>-				41889D74216BB7B6004614DD /* RTCRtpEncodingParameters+Private.h */,
> 				414FB868216BB766001F5492 /* RTCRtpFragmentationHeader+Private.h */,
> 				414FB867216BB766001F5492 /* RTCRtpFragmentationHeader+Private.mm */,
> 				4144B3E72169C5E3004363AC /* RTCVideoCodecInfo+Private.h */,
>@@ -8462,10 +8500,10 @@
> 				416225D02169818100A91C9B /* video_decoder_factory.mm */,
> 				416225CD2169818000A91C9B /* video_encoder_factory.h */,
> 				416225CC2169818000A91C9B /* video_encoder_factory.mm */,
>-				416225CB2169818000A91C9B /* video_frame_buffer.h */,
>-				416225CF2169818100A91C9B /* video_frame_buffer.mm */,
> 				416225CE2169818000A91C9B /* video_frame.h */,
> 				416225D12169818100A91C9B /* video_frame.mm */,
>+				416225CB2169818000A91C9B /* video_frame_buffer.h */,
>+				416225CF2169818100A91C9B /* video_frame_buffer.mm */,
> 			);
> 			path = api;
> 			sourceTree = "<group>";
>@@ -8488,8 +8526,8 @@
> 		416225EA216981BA00A91C9B /* components */ = {
> 			isa = PBXGroup;
> 			children = (
>-				413E6787216987A500EF37ED /* video_frame_buffer */,
> 				416225EB216981CE00A91C9B /* video_codec */,
>+				413E6787216987A500EF37ED /* video_frame_buffer */,
> 			);
> 			name = components;
> 			path = sdk/objc/components;
>@@ -8502,9 +8540,9 @@
> 				416225F5216981F400A91C9B /* helpers.h */,
> 				417953AC2169823D0028266B /* nalu_rewriter.cc */,
> 				411521D521698220000ABAF7 /* nalu_rewriter.h */,
>+				417953B2216982410028266B /* RTCCodecSpecificInfoH264+Private.h */,
> 				416225F1216981F300A91C9B /* RTCCodecSpecificInfoH264.h */,
> 				416225ED216981F200A91C9B /* RTCCodecSpecificInfoH264.mm */,
>-				417953B2216982410028266B /* RTCCodecSpecificInfoH264+Private.h */,
> 				417953B3216982420028266B /* RTCDefaultVideoDecoderFactory.h */,
> 				417953AD2169823E0028266B /* RTCDefaultVideoDecoderFactory.m */,
> 				416225F7216981F400A91C9B /* RTCDefaultVideoEncoderFactory.h */,
>@@ -8545,11 +8583,11 @@
> 		417953DD216983B20028266B /* include */ = {
> 			isa = PBXGroup;
> 			children = (
>-				417953E0216983C90028266B /* module_common_types_public.h */,
>+				417953DF216983C90028266B /* module.h */,
> 				417953E1216983CA0028266B /* module_common_types.cc */,
> 				417953E2216983CA0028266B /* module_common_types.h */,
>+				417953E0216983C90028266B /* module_common_types_public.h */,
> 				417953DE216983C90028266B /* module_fec_types.h */,
>-				417953DF216983C90028266B /* module.h */,
> 			);
> 			path = include;
> 			sourceTree = "<group>";
>@@ -8572,7 +8610,7 @@
> 				413E679C2169889400EF37ED /* RTCVideoEncoderQpThresholds.h */,
> 				413E679F2169889500EF37ED /* RTCVideoEncoderQpThresholds.m */,
> 				413E67832169877D00EF37ED /* RTCVideoEncoderSettings.h */,
>-				413E67842169877D00EF37ED /* RTCVideoEncoderSettings.mm */,
>+				41FCBB8721B200DB00A5DF27 /* RTCVideoEncoderSettings.m */,
> 				413E678C216987DB00EF37ED /* RTCVideoFrame.h */,
> 				413E678D216987DB00EF37ED /* RTCVideoFrame.mm */,
> 				413E678E216987DB00EF37ED /* RTCVideoFrameBuffer.h */,
>@@ -8595,11 +8633,11 @@
> 		417953FC216984F40028266B /* video_codec */ = {
> 			isa = PBXGroup;
> 			children = (
>-				413E67642169854500EF37ED /* RTCVideoEncoderVP8.mm */,
> 				417953FD2169851F0028266B /* RTCVideoCodecConstants.h */,
> 				417953FF2169851F0028266B /* RTCVideoCodecConstants.mm */,
> 				41795402216985200028266B /* RTCVideoDecoderVP8.mm */,
> 				417954012169851F0028266B /* RTCVideoEncoderVP8.h */,
>+				413E67642169854500EF37ED /* RTCVideoEncoderVP8.mm */,
> 				417953FE2169851F0028266B /* RTCWrappedNativeVideoDecoder.h */,
> 				41795403216985200028266B /* RTCWrappedNativeVideoDecoder.mm */,
> 				41795404216985200028266B /* RTCWrappedNativeVideoEncoder.h */,
>@@ -8752,12 +8790,20 @@
> 		419241652127387900634FCF /* experiments */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41F9BFC92051DDA200ABF0B9 /* alr_experiment.cc */,
>+				41F9BFCA2051DDA200ABF0B9 /* alr_experiment.h */,
> 				4192416B2127389B00634FCF /* congestion_controller_experiment.cc */,
> 				419241672127389A00634FCF /* congestion_controller_experiment.h */,
>+				41FCBB4D21B1FE4500A5DF27 /* cpu_speed_experiment.cc */,
>+				41FCBB4B21B1FE4400A5DF27 /* cpu_speed_experiment.h */,
> 				419241682127389A00634FCF /* field_trial_parser.cc */,
> 				419241692127389B00634FCF /* field_trial_parser.h */,
> 				419241E621275A7600634FCF /* field_trial_units.cc */,
> 				419241E521275A7600634FCF /* field_trial_units.h */,
>+				41FCBB4C21B1FE4400A5DF27 /* jitter_upper_bound_experiment.cc */,
>+				41FCBB4E21B1FE4500A5DF27 /* jitter_upper_bound_experiment.h */,
>+				41FCBB5021B1FE4500A5DF27 /* normalize_simulcast_size_experiment.cc */,
>+				41FCBB4F21B1FE4500A5DF27 /* normalize_simulcast_size_experiment.h */,
> 				4192416A2127389B00634FCF /* quality_scaling_experiment.cc */,
> 				419241662127389A00634FCF /* quality_scaling_experiment.h */,
> 				417953E8216983F10028266B /* rtt_mult_experiment.cc */,
>@@ -8938,6 +8984,8 @@
> 				419C834C1FE246640040C30F /* histogram_percentile_counter.h */,
> 				419C834D1FE246640040C30F /* mathutils.h */,
> 				419C834A1FE246640040C30F /* mod_ops.h */,
>+				41FCBB1321B1F7AA00A5DF27 /* moving_average.cc */,
>+				41FCBB1421B1F7AA00A5DF27 /* moving_average.h */,
> 				419C83491FE246640040C30F /* moving_max_counter.h */,
> 				419C83461FE246630040C30F /* moving_median_filter.h */,
> 				419C83421FE246620040C30F /* percentile_filter.h */,
>@@ -9159,14 +9207,15 @@
> 		41A08BD921272F3E001D5D7B /* agc2 */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41299B98212736B200B3414B /* rnn_vad */,
> 				41D6B46E212731A4008F9353 /* adaptive_agc.cc */,
> 				41D6B464212731A2008F9353 /* adaptive_agc.h */,
> 				41A08BE421272F85001D5D7B /* adaptive_digital_gain_applier.cc */,
> 				41A08BDB21272F84001D5D7B /* adaptive_digital_gain_applier.h */,
>-				41A08BE921272F86001D5D7B /* adaptive_mode_level_estimator_agc.cc */,
>-				41D6B462212731A1008F9353 /* adaptive_mode_level_estimator_agc.h */,
> 				41D6B45F212731A1008F9353 /* adaptive_mode_level_estimator.cc */,
> 				41A08BDC21272F84001D5D7B /* adaptive_mode_level_estimator.h */,
>+				41A08BE921272F86001D5D7B /* adaptive_mode_level_estimator_agc.cc */,
>+				41D6B462212731A1008F9353 /* adaptive_mode_level_estimator_agc.h */,
> 				413E679A2169886200EF37ED /* agc2_common.cc */,
> 				41D6B46D212731A4008F9353 /* agc2_common.h */,
> 				41D6B45E212731A0008F9353 /* biquad_filter.cc */,
>@@ -9177,19 +9226,14 @@
> 				41D6B460212731A1008F9353 /* down_sampler.h */,
> 				41A08BE521272F86001D5D7B /* fixed_digital_level_estimator.cc */,
> 				41A08BDF21272F84001D5D7B /* fixed_digital_level_estimator.h */,
>-				41D6B465212731A2008F9353 /* fixed_gain_controller.cc */,
>-				41D6B467212731A3008F9353 /* fixed_gain_controller.h */,
> 				41A08BDE21272F84001D5D7B /* gain_applier.cc */,
> 				41A08BE721272F86001D5D7B /* gain_applier.h */,
>-				41A08BE021272F85001D5D7B /* gain_curve_applier.cc */,
>-				41D6B469212731A3008F9353 /* gain_curve_applier.h */,
> 				41A08BE121272F85001D5D7B /* interpolated_gain_curve.cc */,
> 				41D6B470212731A6008F9353 /* limiter.cc */,
> 				41A08BEA21272F86001D5D7B /* limiter.h */,
> 				41D6B461212731A1008F9353 /* noise_level_estimator.cc */,
> 				41D6B46A212731A3008F9353 /* noise_spectrum_estimator.cc */,
> 				41A08BE221272F85001D5D7B /* noise_spectrum_estimator.h */,
>-				41299B98212736B200B3414B /* rnn_vad */,
> 				41D6B468212731A3008F9353 /* saturation_protector.cc */,
> 				41D6B46B212731A3008F9353 /* saturation_protector.h */,
> 				41D6B46F212731A4008F9353 /* signal_classifier.cc */,
>@@ -9627,10 +9671,11 @@
> 				41E02C992127352B00C27CD6 /* alr_detector.h */,
> 				41E02C9A2127352B00C27CD6 /* bitrate_estimator.cc */,
> 				41E02C982127352B00C27CD6 /* bitrate_estimator.h */,
>+				41FCBB3F21B1FDB800A5DF27 /* congestion_window_pushback_controller.cc */,
>+				41FCBB3E21B1FDB800A5DF27 /* congestion_window_pushback_controller.h */,
> 				41E02C912127352900C27CD6 /* delay_based_bwe.cc */,
> 				41E02C972127352A00C27CD6 /* delay_based_bwe.h */,
> 				41E02CA32127352D00C27CD6 /* delay_increase_detector_interface.h */,
>-				41E02C952127352A00C27CD6 /* goog_cc_factory.cc */,
> 				41E02CA22127352C00C27CD6 /* goog_cc_network_control.cc */,
> 				41E02C962127352A00C27CD6 /* goog_cc_network_control.h */,
> 				41E02C9D2127352B00C27CD6 /* median_slope_estimator.cc */,
>@@ -9650,6 +9695,8 @@
> 			children = (
> 				41E02CC7212735B600C27CD6 /* bitrate_settings.cc */,
> 				41E02CCB212735B700C27CD6 /* bitrate_settings.h */,
>+				41FCBB5E21B1FEC400A5DF27 /* goog_cc_factory.cc */,
>+				41FCBB5D21B1FEC400A5DF27 /* goog_cc_factory.h */,
> 				41E02CCA212735B700C27CD6 /* network_control.h */,
> 				41E02CC8212735B700C27CD6 /* network_types.cc */,
> 				41E02CC9212735B700C27CD6 /* network_types.h */,
>@@ -9660,8 +9707,8 @@
> 		41E02CD52127362300C27CD6 /* rtp */ = {
> 			isa = PBXGroup;
> 			children = (
>-				41E02CD62127363C00C27CD6 /* pacer_controller.cc */,
>-				41924189212749C800634FCF /* pacer_controller.h */,
>+				41FCBB3521B1F8FC00A5DF27 /* control_handler.cc */,
>+				41FCBB3421B1F8FB00A5DF27 /* control_handler.h */,
> 				41924188212749C700634FCF /* send_side_congestion_controller.cc */,
> 				41E02CD82127363D00C27CD6 /* send_time_history.cc */,
> 				41924187212749C700634FCF /* send_time_history.h */,
>@@ -9877,6 +9924,8 @@
> 				41F263AB212680A800274F59 /* arch.h */,
> 				41F263A9212680A700274F59 /* asm_defines.h */,
> 				41F263A8212680A700274F59 /* BUILD.gn */,
>+				41FCBB7421B1FFA400A5DF27 /* cocoa_threading.h */,
>+				41FCBB7321B1FFA400A5DF27 /* cocoa_threading.mm */,
> 				41F263A5212680A700274F59 /* fallthrough.h */,
> 				41F263A3212680A600274F59 /* file_wrapper.cc */,
> 				41F263AA212680A800274F59 /* file_wrapper.h */,
>@@ -9903,9 +9952,9 @@
> 		41F411BD1EF8DB8200343C26 /* vp8 */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4145F6151FE1EF5C00EB9CAF /* include */,
> 				419C842B1FE24E7E0040C30F /* default_temporal_layers.cc */,
> 				419C84311FE24E800040C30F /* default_temporal_layers.h */,
>-				4145F6151FE1EF5C00EB9CAF /* include */,
> 				413E67712169863B00EF37ED /* libvpx_interface.cc */,
> 				413E676F2169863A00EF37ED /* libvpx_interface.h */,
> 				41294090212E128C00AD95E7 /* libvpx_vp8_decoder.cc */,
>@@ -9914,9 +9963,8 @@
> 				4129408F212E128C00AD95E7 /* libvpx_vp8_encoder.h */,
> 				419C84301FE24E7F0040C30F /* screenshare_layers.cc */,
> 				419C842C1FE24E7F0040C30F /* screenshare_layers.h */,
>-				413E676E2169863A00EF37ED /* temporal_layers_checker.cc */,
> 				419C84331FE24E800040C30F /* temporal_layers.h */,
>-				413E67702169863B00EF37ED /* vp8_temporal_layers.cc */,
>+				413E676E2169863A00EF37ED /* temporal_layers_checker.cc */,
> 			);
> 			name = vp8;
> 			sourceTree = "<group>";
>@@ -10037,13 +10085,44 @@
> 			name = include;
> 			sourceTree = "<group>";
> 		};
>-		41F9BFC82051DD8E00ABF0B9 /* experiments */ = {
>+		41FCBB1F21B1F81400A5DF27 /* internal */ = {
> 			isa = PBXGroup;
> 			children = (
>-				41F9BFC92051DDA200ABF0B9 /* alr_experiment.cc */,
>-				41F9BFCA2051DDA200ABF0B9 /* alr_experiment.h */,
>+				41FCBB2021B1F82E00A5DF27 /* throw_delegate.cc */,
>+				41FCBB2121B1F82F00A5DF27 /* throw_delegate.h */,
> 			);
>-			name = experiments;
>+			name = internal;
>+			path = base/internal;
>+			sourceTree = "<group>";
>+		};
>+		41FCBB2421B1F85D00A5DF27 /* network */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41FCBB2521B1F87E00A5DF27 /* sent_packet.cc */,
>+				41FCBB2621B1F87E00A5DF27 /* sent_packet.h */,
>+			);
>+			name = network;
>+			path = rtc_base/network;
>+			sourceTree = "<group>";
>+		};
>+		41FCBB2921B1F8A300A5DF27 /* crypto */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41FCBB2D21B1F8B600A5DF27 /* cryptooptions.cc */,
>+				41FCBB2A21B1F8B600A5DF27 /* cryptooptions.h */,
>+				41FCBB2B21B1F8B600A5DF27 /* framedecryptorinterface.h */,
>+				41FCBB2C21B1F8B600A5DF27 /* frameencryptorinterface.h */,
>+			);
>+			path = crypto;
>+			sourceTree = "<group>";
>+		};
>+		41FCBB8121B2005A00A5DF27 /* internal */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41FCBB8221B2007B00A5DF27 /* memutil.cc */,
>+				41FCBB8321B2007B00A5DF27 /* memutil.h */,
>+			);
>+			path = internal;
> 			sourceTree = "<group>";
> 		};
> 		41FE864E215C52E900B62C07 /* listfmts */ = {
>@@ -10853,12 +10932,12 @@
> 		5C4B4B581E431C49002651C8 /* objc */ = {
> 			isa = PBXGroup;
> 			children = (
>-				4144B3EC2169C61C004363AC /* helpers */,
> 				417953FB216984DC0028266B /* api */,
> 				417953F2216984830028266B /* base */,
> 				416225EA216981BA00A91C9B /* components */,
>-				416225C72169814100A91C9B /* native */,
> 				5C4B4B591E431C55002651C8 /* Framework */,
>+				4144B3EC2169C61C004363AC /* helpers */,
>+				416225C72169814100A91C9B /* native */,
> 			);
> 			name = objc;
> 			sourceTree = "<group>";
>@@ -10894,7 +10973,6 @@
> 				5C4B4C101E431F75002651C8 /* bitrate_adjuster.cc */,
> 				5C4B4C121E431F75002651C8 /* i420_buffer_pool.cc */,
> 				5C4B4C131E431F75002651C8 /* incoming_video_stream.cc */,
>-				5C4B4C161E431F75002651C8 /* video_frame.cc */,
> 				5C4B4C151E431F75002651C8 /* video_frame_buffer.cc */,
> 				5C4B4C171E431F75002651C8 /* video_render_frames.cc */,
> 				5C4B4C181E431F75002651C8 /* video_render_frames.h */,
>@@ -11097,17 +11175,25 @@
> 		5C63F8E21E41732B002CA531 /* api */ = {
> 			isa = PBXGroup;
> 			children = (
>-				41F9BF792051C7FD00ABF0B9 /* array_view.h */,
> 				5CD2853B1E6A61BE0094FDC8 /* audio_codecs */,
>+				5CDD8C451E43C57900621E92 /* call */,
>+				41FCBB2921B1F8A300A5DF27 /* crypto */,
>+				4145E4C01EF8962800FCF6E6 /* stats */,
>+				41E02CC62127359F00C27CD6 /* transport */,
>+				419241D421275A1800634FCF /* units */,
>+				5CD284601E6A57DD0094FDC8 /* video */,
>+				412455581EF88AD900F11809 /* video_codecs */,
>+				41F9BF792051C7FD00ABF0B9 /* array_view.h */,
> 				415F1FCA212730F000064CBF /* audio_frame.cc */,
> 				415F1FCB212730F000064CBF /* audio_frame.h */,
> 				415F1FC9212730F000064CBF /* audio_mixer.h */,
> 				41F2637521267B7600274F59 /* audio_options.cc */,
> 				41F2637821267B7600274F59 /* audio_options.h */,
> 				41F2637921267B7700274F59 /* bitrate_constraints.h */,
>-				5CDD8C451E43C57900621E92 /* call */,
> 				419C830E1FE2427F0040C30F /* candidate.cc */,
> 				41F9BF772051C7FD00ABF0B9 /* candidate.h */,
>+				41FCBB0921B1F53000A5DF27 /* create_peerconnection_factory.cc */,
>+				41FCBB0D21B1F53600A5DF27 /* create_peerconnection_factory.h */,
> 				41F9BF822051C80000ABF0B9 /* cryptoparams.h */,
> 				41F2637721267B7600274F59 /* datachannelinterface.cc */,
> 				5C63F8E71E41737B002CA531 /* datachannelinterface.h */,
>@@ -11119,6 +11205,8 @@
> 				41F2637B21267B7700274F59 /* jsepicecandidate.cc */,
> 				5C63F8F11E41737B002CA531 /* jsepicecandidate.h */,
> 				5C63F8F31E41737B002CA531 /* jsepsessiondescription.h */,
>+				41FCBB0C21B1F53200A5DF27 /* media_transport_interface.cc */,
>+				41FCBB0A21B1F53100A5DF27 /* media_transport_interface.h */,
> 				5C63F8F61E41737B002CA531 /* mediaconstraintsinterface.cc */,
> 				5C63F8F71E41737B002CA531 /* mediaconstraintsinterface.h */,
> 				5CD2863F1E6A683A0094FDC8 /* mediastreaminterface.cc */,
>@@ -11148,16 +11236,12 @@
> 				5C63F91B1E41737B002CA531 /* rtpsenderinterface.h */,
> 				41F2637C21267B7700274F59 /* rtptransceiverinterface.cc */,
> 				41F9BF802051C7FF00ABF0B9 /* rtptransceiverinterface.h */,
>+				41FCBB0B21B1F53200A5DF27 /* scoped_refptr.h */,
> 				41F9BF842051C80000ABF0B9 /* setremotedescriptionobserverinterface.h */,
>-				4145E4C01EF8962800FCF6E6 /* stats */,
> 				5C63F9211E41737B002CA531 /* statstypes.cc */,
> 				5C63F9221E41737B002CA531 /* statstypes.h */,
>-				41E02CC62127359F00C27CD6 /* transport */,
> 				41F9BF832051C80000ABF0B9 /* turncustomizer.h */,
> 				5C63F9241E41737B002CA531 /* umametrics.h */,
>-				419241D421275A1800634FCF /* units */,
>-				5CD284601E6A57DD0094FDC8 /* video */,
>-				412455581EF88AD900F11809 /* video_codecs */,
> 				5C63F9281E41737B002CA531 /* videosourceproxy.h */,
> 			);
> 			path = api;
>@@ -11191,8 +11275,12 @@
> 				4192414F2127376E00634FCF /* fakeicetransport.h */,
> 				4192414E2127376E00634FCF /* fakepackettransport.h */,
> 				5C63F9951E4174F6002CA531 /* fakeportallocator.h */,
>+				41FCBB4321B1FDEE00A5DF27 /* icecredentialsiterator.cc */,
>+				41FCBB4521B1FDEF00A5DF27 /* icecredentialsiterator.h */,
> 				419C83D71FE247F20040C30F /* icetransportinternal.cc */,
> 				419241502127376E00634FCF /* icetransportinternal.h */,
>+				41FCBB4421B1FDEF00A5DF27 /* mdns_message.cc */,
>+				41FCBB4221B1FDEE00A5DF27 /* mdns_message.h */,
> 				5C63F9971E4174F6002CA531 /* p2pconstants.cc */,
> 				5C63F9981E4174F6002CA531 /* p2pconstants.h */,
> 				5C63F99C1E4174F6002CA531 /* p2ptransportchannel.cc */,
>@@ -11242,8 +11330,6 @@
> 				5C63F9D91E4174F6002CA531 /* turnserver.cc */,
> 				5C63F9DA1E4174F6002CA531 /* turnserver.h */,
> 				5C63F9DB1E4174F6002CA531 /* udpport.h */,
>-				419C83D31FE247F00040C30F /* udptransport.cc */,
>-				4192414D2127376E00634FCF /* udptransport.h */,
> 			);
> 			path = base;
> 			sourceTree = "<group>";
>@@ -11253,13 +11339,10 @@
> 			children = (
> 				5CD284C41E6A60570094FDC8 /* audiotrack.cc */,
> 				5CD284C51E6A60570094FDC8 /* audiotrack.h */,
>-				5C63FA3A1E41761F002CA531 /* bundlefilter.cc */,
>-				5C63FA3B1E41761F002CA531 /* bundlefilter.h */,
> 				5C63FA3D1E41761F002CA531 /* channel.cc */,
> 				5C63FA3E1E41761F002CA531 /* channel.h */,
> 				5C63FA401E41761F002CA531 /* channelmanager.cc */,
> 				5C63FA411E41761F002CA531 /* channelmanager.h */,
>-				4145E4C91EF8CB3100FCF6E6 /* createpeerconnectionfactory.cc */,
> 				5CD284C61E6A60570094FDC8 /* datachannel.cc */,
> 				5CD284C71E6A60570094FDC8 /* datachannel.h */,
> 				419C82D61FE20E580040C30F /* dtlssrtptransport.cc */,
>@@ -11451,31 +11534,37 @@
> 		5CD284601E6A57DD0094FDC8 /* video */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41FCBB1721B1F7CC00A5DF27 /* builtin_video_bitrate_allocator_factory.cc */,
>+				41FCBB1A21B1F7CD00A5DF27 /* builtin_video_bitrate_allocator_factory.h */,
> 				4102F6D221273414006AE8D7 /* color_space.cc */,
> 				4102F6D821273416006AE8D7 /* color_space.h */,
> 				41F263C72126818A00274F59 /* encoded_frame.cc */,
> 				41F263C22126818900274F59 /* encoded_frame.h */,
>+				41FCBB1921B1F7CC00A5DF27 /* encoded_image.cc */,
>+				41FCBB1821B1F7CC00A5DF27 /* encoded_image.h */,
>+				41FCBB6F21B1FF7400A5DF27 /* hdr_metadata.cc */,
>+				41FCBB7021B1FF7400A5DF27 /* hdr_metadata.h */,
> 				4102F6D621273415006AE8D7 /* i010_buffer.cc */,
> 				41F263BF2126818900274F59 /* i010_buffer.h */,
> 				5CD284611E6A57F40094FDC8 /* i420_buffer.cc */,
> 				5CD284621E6A57F40094FDC8 /* i420_buffer.h */,
> 				4102F6D921273416006AE8D7 /* video_bitrate_allocation.cc */,
> 				41F263C42126818900274F59 /* video_bitrate_allocation.h */,
>-				4102F6DA21273416006AE8D7 /* video_bitrate_allocator_factory.h */,
> 				41F263C12126818900274F59 /* video_bitrate_allocator.h */,
>+				4102F6DA21273416006AE8D7 /* video_bitrate_allocator_factory.h */,
> 				419C840D1FE249AA0040C30F /* video_content_type.cc */,
> 				419C840F1FE249AB0040C30F /* video_content_type.h */,
>-				4124554A1EF8874300F11809 /* video_frame_buffer.cc */,
>-				5CD284631E6A57F40094FDC8 /* video_frame_buffer.h */,
> 				5CD284641E6A57F40094FDC8 /* video_frame.cc */,
> 				5CD284651E6A57F40094FDC8 /* video_frame.h */,
>+				4124554A1EF8874300F11809 /* video_frame_buffer.cc */,
>+				5CD284631E6A57F40094FDC8 /* video_frame_buffer.h */,
> 				5CD284661E6A57F40094FDC8 /* video_rotation.h */,
> 				4102F6D321273415006AE8D7 /* video_sink_interface.h */,
> 				41F263C62126818A00274F59 /* video_source_interface.cc */,
> 				41F263BE2126818800274F59 /* video_source_interface.h */,
>+				41F263C52126818A00274F59 /* video_stream_decoder.h */,
> 				4102F6D121273414006AE8D7 /* video_stream_decoder_create.cc */,
> 				4102F6D421273415006AE8D7 /* video_stream_decoder_create.h */,
>-				41F263C52126818A00274F59 /* video_stream_decoder.h */,
> 				4102F6D521273415006AE8D7 /* video_stream_encoder_create.cc */,
> 				41F263C02126818900274F59 /* video_stream_encoder_create.h */,
> 				41F263C32126818900274F59 /* video_stream_encoder_interface.h */,
>@@ -11762,8 +11851,6 @@
> 				5CDD836C1E439A3500621E92 /* frame_dropper.h */,
> 				5CDD836D1E439A3500621E92 /* ivf_file_writer.cc */,
> 				5CDD836E1E439A3500621E92 /* ivf_file_writer.h */,
>-				5CDD836F1E439A3500621E92 /* moving_average.cc */,
>-				5CDD83701E439A3500621E92 /* moving_average.h */,
> 				5CDD83731E439A3500621E92 /* quality_scaler.cc */,
> 				5CDD83741E439A3500621E92 /* quality_scaler.h */,
> 				419241EA21275AFA00634FCF /* simulcast_rate_allocator.cc */,
>@@ -11907,8 +11994,6 @@
> 				5CDD84F91E43B1EA00621E92 /* audio_coding_module.cc */,
> 				5CDD84FB1E43B1EA00621E92 /* call_statistics.cc */,
> 				5CDD84FC1E43B1EA00621E92 /* call_statistics.h */,
>-				5CDD84FE1E43B1EA00621E92 /* codec_manager.cc */,
>-				5CDD84FF1E43B1EA00621E92 /* codec_manager.h */,
> 				5CDD85011E43B1EA00621E92 /* rent_a_codec.cc */,
> 				5CDD85021E43B1EA00621E92 /* rent_a_codec.h */,
> 			);
>@@ -12008,6 +12093,8 @@
> 				5CDD855D1E43B5C000621E92 /* call_stats.h */,
> 				5CDD85601E43B5C000621E92 /* encoder_rtcp_feedback.cc */,
> 				5CDD85611E43B5C000621E92 /* encoder_rtcp_feedback.h */,
>+				41FCBB5721B1FE6D00A5DF27 /* frame_dumping_decoder.cc */,
>+				41FCBB5821B1FE6E00A5DF27 /* frame_dumping_decoder.h */,
> 				5CDD85661E43B5C000621E92 /* overuse_frame_detector.cc */,
> 				5CDD85671E43B5C000621E92 /* overuse_frame_detector.h */,
> 				5CD286281E6A66C80094FDC8 /* quality_threshold.cc */,
>@@ -12376,6 +12463,7 @@
> 		5CDD87FA1E43BE2600621E92 /* source */ = {
> 			isa = PBXGroup;
> 			children = (
>+				5CDD88EB1E43BF2500621E92 /* rtcp_packet */,
> 				5CDD87FC1E43BE3C00621E92 /* byte_io.h */,
> 				4102F6EB21273460006AE8D7 /* contributing_sources.cc */,
> 				5CDD87FE1E43BE3C00621E92 /* dtmf_queue.cc */,
>@@ -12390,10 +12478,10 @@
> 				5CDD88061E43BE3C00621E92 /* flexfec_header_reader_writer.h */,
> 				5CDD880A1E43BE3C00621E92 /* flexfec_receiver.cc */,
> 				5CDD880C1E43BE3C00621E92 /* flexfec_sender.cc */,
>-				5CDD880D1E43BE3C00621E92 /* forward_error_correction_internal.cc */,
>-				5CDD880E1E43BE3C00621E92 /* forward_error_correction_internal.h */,
> 				5CDD880F1E43BE3C00621E92 /* forward_error_correction.cc */,
> 				5CDD88101E43BE3C00621E92 /* forward_error_correction.h */,
>+				5CDD880D1E43BE3C00621E92 /* forward_error_correction_internal.cc */,
>+				5CDD880E1E43BE3C00621E92 /* forward_error_correction_internal.h */,
> 				5CDD88151E43BE3C00621E92 /* packet_loss_stats.cc */,
> 				5CDD88161E43BE3C00621E92 /* packet_loss_stats.h */,
> 				5CDD88181E43BE3C00621E92 /* playout_delay_oracle.cc */,
>@@ -12403,19 +12491,20 @@
> 				5CDD881E1E43BE3C00621E92 /* remote_ntp_time_estimator.cc */,
> 				5CD285F81E6A64520094FDC8 /* rtcp_nack_stats.cc */,
> 				5CD285F91E6A64520094FDC8 /* rtcp_nack_stats.h */,
>-				5CDD88EB1E43BF2500621E92 /* rtcp_packet */,
> 				5CDD88211E43BE3C00621E92 /* rtcp_packet.cc */,
> 				5CDD88221E43BE3C00621E92 /* rtcp_packet.h */,
> 				5CDD88241E43BE3C00621E92 /* rtcp_receiver.cc */,
> 				5CDD88251E43BE3C00621E92 /* rtcp_receiver.h */,
> 				5CDD88271E43BE3C00621E92 /* rtcp_sender.cc */,
> 				5CDD88281E43BE3C00621E92 /* rtcp_sender.h */,
>+				419C82F61FE20EFB0040C30F /* rtcp_transceiver.cc */,
>+				419C82FA1FE20EFD0040C30F /* rtcp_transceiver.h */,
> 				419C82F91FE20EFD0040C30F /* rtcp_transceiver_config.cc */,
> 				419C82F81FE20EFC0040C30F /* rtcp_transceiver_config.h */,
> 				419C82F71FE20EFC0040C30F /* rtcp_transceiver_impl.cc */,
> 				419C82FC1FE20EFE0040C30F /* rtcp_transceiver_impl.h */,
>-				419C82F61FE20EFB0040C30F /* rtcp_transceiver.cc */,
>-				419C82FA1FE20EFD0040C30F /* rtcp_transceiver.h */,
>+				5CDD883A1E43BE3C00621E92 /* rtp_format.cc */,
>+				5CDD883B1E43BE3C00621E92 /* rtp_format.h */,
> 				5CDD882E1E43BE3C00621E92 /* rtp_format_h264.cc */,
> 				5CDD882F1E43BE3C00621E92 /* rtp_format_h264.h */,
> 				5CDD88301E43BE3C00621E92 /* rtp_format_video_generic.cc */,
>@@ -12424,35 +12513,33 @@
> 				5CDD88361E43BE3C00621E92 /* rtp_format_vp8.h */,
> 				5CDD88381E43BE3C00621E92 /* rtp_format_vp9.cc */,
> 				5CDD88391E43BE3C00621E92 /* rtp_format_vp9.h */,
>-				5CDD883A1E43BE3C00621E92 /* rtp_format.cc */,
>-				5CDD883B1E43BE3C00621E92 /* rtp_format.h */,
>-				419241AF2127590200634FCF /* rtp_generic_frame_descriptor_extension.cc */,
>-				4102F6E82127345F006AE8D7 /* rtp_generic_frame_descriptor_extension.h */,
> 				4102F6E52127345E006AE8D7 /* rtp_generic_frame_descriptor.cc */,
> 				4102F6E62127345F006AE8D7 /* rtp_generic_frame_descriptor.h */,
>+				419241AF2127590200634FCF /* rtp_generic_frame_descriptor_extension.cc */,
>+				4102F6E82127345F006AE8D7 /* rtp_generic_frame_descriptor_extension.h */,
> 				4145E4DB1EF8CCEE00FCF6E6 /* rtp_header_extension_map.cc */,
> 				413E677B216986AD00EF37ED /* rtp_header_extension_size.cc */,
> 				413E677C216986AE00EF37ED /* rtp_header_extension_size.h */,
> 				5CDD883F1E43BE3C00621E92 /* rtp_header_extensions.cc */,
> 				5CDD88401E43BE3C00621E92 /* rtp_header_extensions.h */,
> 				5CDD88411E43BE3C00621E92 /* rtp_header_parser.cc */,
>+				5CDD88481E43BE3C00621E92 /* rtp_packet.cc */,
>+				5CDD88491E43BE3C00621E92 /* rtp_packet.h */,
> 				5CDD88431E43BE3C00621E92 /* rtp_packet_history.cc */,
> 				5CDD88441E43BE3C00621E92 /* rtp_packet_history.h */,
> 				419C82FE1FE20F010040C30F /* rtp_packet_received.cc */,
> 				5CDD88451E43BE3C00621E92 /* rtp_packet_received.h */,
> 				4102F6E92127345F006AE8D7 /* rtp_packet_to_send.cc */,
> 				5CDD88461E43BE3C00621E92 /* rtp_packet_to_send.h */,
>-				5CDD88481E43BE3C00621E92 /* rtp_packet.cc */,
>-				5CDD88491E43BE3C00621E92 /* rtp_packet.h */,
> 				5CDD88541E43BE3C00621E92 /* rtp_rtcp_config.h */,
> 				5CDD88561E43BE3C00621E92 /* rtp_rtcp_impl.cc */,
> 				5CDD88571E43BE3C00621E92 /* rtp_rtcp_impl.h */,
>+				5CDD885D1E43BE3C00621E92 /* rtp_sender.cc */,
>+				5CDD885E1E43BE3C00621E92 /* rtp_sender.h */,
> 				5CDD88581E43BE3C00621E92 /* rtp_sender_audio.cc */,
> 				5CDD88591E43BE3C00621E92 /* rtp_sender_audio.h */,
> 				5CDD885B1E43BE3C00621E92 /* rtp_sender_video.cc */,
> 				5CDD885C1E43BE3C00621E92 /* rtp_sender_video.h */,
>-				5CDD885D1E43BE3C00621E92 /* rtp_sender.cc */,
>-				5CDD885E1E43BE3C00621E92 /* rtp_sender.h */,
> 				5CDD885F1E43BE3C00621E92 /* rtp_utility.cc */,
> 				5CDD88601E43BE3C00621E92 /* rtp_utility.h */,
> 				4102F6E72127345F006AE8D7 /* rtp_video_header.cc */,
>@@ -12587,8 +12674,6 @@
> 				5CDD89E21E43BFB300621E92 /* random_vector.h */,
> 				5CDD89E41E43BFB300621E92 /* red_payload_splitter.cc */,
> 				5CDD89E51E43BFB300621E92 /* red_payload_splitter.h */,
>-				5CDD89E61E43BFB300621E92 /* rtcp.cc */,
>-				5CDD89E71E43BFB300621E92 /* rtcp.h */,
> 				5CDD89E81E43BFB300621E92 /* statistics_calculator.cc */,
> 				5CDD89E91E43BFB300621E92 /* statistics_calculator.h */,
> 				5CDD89EB1E43BFB300621E92 /* sync_buffer.cc */,
>@@ -12642,6 +12727,7 @@
> 		5CDD8ABB1E43C22200621E92 /* audio */ = {
> 			isa = PBXGroup;
> 			children = (
>+				5CD2854A1E6A62090094FDC8 /* utility */,
> 				4102F69A21273261006AE8D7 /* audio_level.cc */,
> 				4102F69821273260006AE8D7 /* audio_level.h */,
> 				5CDD8ABD1E43C23900621E92 /* audio_receive_stream.cc */,
>@@ -12652,12 +12738,8 @@
> 				5CDD8AC41E43C23900621E92 /* audio_state.h */,
> 				41F9BF992051C84B00ABF0B9 /* audio_transport_impl.cc */,
> 				41F9BF9A2051C84B00ABF0B9 /* audio_transport_impl.h */,
>-				413E67782169867A00EF37ED /* channel_receive_proxy.cc */,
>-				413E67772169867A00EF37ED /* channel_receive_proxy.h */,
> 				417953D22169834F0028266B /* channel_receive.cc */,
> 				417953D02169834F0028266B /* channel_receive.h */,
>-				417953CF2169834E0028266B /* channel_send_proxy.cc */,
>-				417953D12169834F0028266B /* channel_send_proxy.h */,
> 				417953CD2169834E0028266B /* channel_send.cc */,
> 				417953CE2169834E0028266B /* channel_send.h */,
> 				5CDD8AC61E43C23900621E92 /* conversion.h */,
>@@ -12665,11 +12747,8 @@
> 				419C83B21FE2472E0040C30F /* null_audio_poller.h */,
> 				4102F69721273260006AE8D7 /* remix_resample.cc */,
> 				4102F69D21273261006AE8D7 /* remix_resample.h */,
>-				419C83331FE245E90040C30F /* time_interval.cc */,
>-				419C83321FE245E90040C30F /* time_interval.h */,
> 				4102F69F21273262006AE8D7 /* transport_feedback_packet_loss_tracker.cc */,
> 				4102F69621273260006AE8D7 /* transport_feedback_packet_loss_tracker.h */,
>-				5CD2854A1E6A62090094FDC8 /* utility */,
> 			);
> 			path = audio;
> 			sourceTree = "<group>";
>@@ -12827,10 +12906,10 @@
> 		5CDD8C121E43C39200621E92 /* vp9 */ = {
> 			isa = PBXGroup;
> 			children = (
>-				413E67922169881900EF37ED /* svc_rate_allocator.cc */,
>-				413E67932169881900EF37ED /* svc_rate_allocator.h */,
> 				413E67802169870F00EF37ED /* svc_config.cc */,
> 				413E677F2169870F00EF37ED /* svc_config.h */,
>+				413E67922169881900EF37ED /* svc_rate_allocator.cc */,
>+				413E67932169881900EF37ED /* svc_rate_allocator.h */,
> 				5CDD8C131E43C3B400621E92 /* vp9_noop.cc */,
> 			);
> 			name = vp9;
>@@ -13207,8 +13286,6 @@
> 				419241B12127591C00634FCF /* bbr */,
> 				41E02C8F212734E800C27CD6 /* goog_cc */,
> 				41E02CD52127362300C27CD6 /* rtp */,
>-				41A08BD621272F1C001D5D7B /* congestion_window_pushback_controller.cc */,
>-				41A08BD521272F1C001D5D7B /* congestion_window_pushback_controller.h */,
> 				413091FD1EF8D06D00757C55 /* receive_side_congestion_controller.cc */,
> 				413092141EF8D5ED00757C55 /* send_side_congestion_controller.cc */,
> 				5CDD8FB01E43CD0700621E92 /* transport_feedback_adapter.cc */,
>@@ -13305,8 +13382,11 @@
> 		5CDD90511E43D10400621E92 /* bitrate_controller */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41FCBB5B21B1FEA300A5DF27 /* bitrate_controller.cc */,
> 				5CDD90521E43D11200621E92 /* bitrate_controller_impl.cc */,
> 				5CDD90531E43D11200621E92 /* bitrate_controller_impl.h */,
>+				41FCBB6821B1FF1B00A5DF27 /* loss_based_bandwidth_estimation.cc */,
>+				41FCBB6721B1FF1B00A5DF27 /* loss_based_bandwidth_estimation.h */,
> 				5CDD905B1E43D11200621E92 /* send_side_bandwidth_estimation.cc */,
> 				5CDD905C1E43D11200621E92 /* send_side_bandwidth_estimation.h */,
> 			);
>@@ -13543,7 +13623,6 @@
> 			isa = PBXHeadersBuildPhase;
> 			buildActionMask = 2147483647;
> 			files = (
>-				5C088C111E4AA44400403995 /* bundlefilter.h in Headers */,
> 				5C088C131E4AA44400403995 /* channel.h in Headers */,
> 				5C088C151E4AA44400403995 /* channelmanager.h in Headers */,
> 				5C088C191E4AA44400403995 /* externalhmac.h in Headers */,
>@@ -13723,7 +13802,6 @@
> 				416D3BDD212731C200775F09 /* adaptive_mode_level_estimator_agc.h in Headers */,
> 				41F411C81EF97BAF00343C26 /* adm_helpers.h in Headers */,
> 				5CD285A91E6A63430094FDC8 /* aec3_common.h in Headers */,
>-				4144B3D52169896D004363AC /* RTCNativeMutableI420Buffer.h in Headers */,
> 				5CD285AB1E6A63430094FDC8 /* aec3_fft.h in Headers */,
> 				5CDD86B71E43B9C200621E92 /* aec_common.h in Headers */,
> 				5CDD86BD1E43B9C200621E92 /* aec_core.h in Headers */,
>@@ -13758,6 +13836,7 @@
> 				41433D011F79B33400387B4D /* arith_routins.h in Headers */,
> 				41F9BF872051C80100ABF0B9 /* array_view.h in Headers */,
> 				413A23E41FE18E0800373E99 /* arraysize.h in Headers */,
>+				41FCBB7921B1FFC000A5DF27 /* ascii.h in Headers */,
> 				41F263B1212680A800274F59 /* asm_defines.h in Headers */,
> 				5C63F8D91E416D53002CA531 /* assertions.h in Headers */,
> 				413A22A81FE18E0600373E99 /* asyncinvoker-inl.h in Headers */,
>@@ -13776,7 +13855,6 @@
> 				5CDD84161E439B2900621E92 /* audio_decoder.h in Headers */,
> 				5CD285451E6A61D20094FDC8 /* audio_decoder.h in Headers */,
> 				5CD285431E6A61D20094FDC8 /* audio_decoder_factory.h in Headers */,
>-				417953C02169824B0028266B /* RTCCodecSpecificInfoH264.h in Headers */,
> 				41F9BF982051C82100ABF0B9 /* audio_decoder_factory_template.h in Headers */,
> 				419C83E81FE248350040C30F /* audio_decoder_g711.h in Headers */,
> 				4140B8321E4E3396007409E6 /* audio_decoder_g722.h in Headers */,
>@@ -13820,7 +13898,6 @@
> 				419C83691FE246CA0040C30F /* audio_encoder_isac_float.h in Headers */,
> 				5CDD8C0C1E43C34600621E92 /* audio_encoder_isac_t.h in Headers */,
> 				5CDD8C0B1E43C34600621E92 /* audio_encoder_isac_t_impl.h in Headers */,
>-				417953C92169824B0028266B /* RTCVideoEncoderH264.h in Headers */,
> 				419C83F31FE2488D0040C30F /* audio_encoder_L16.h in Headers */,
> 				5CDD8C641E43C60900621E92 /* audio_encoder_opus.h in Headers */,
> 				419C82F01FE20EB50040C30F /* audio_encoder_opus.h in Headers */,
>@@ -13833,9 +13910,7 @@
> 				415F1FCF212730F000064CBF /* audio_frame.h in Headers */,
> 				5CD284931E6A5F410094FDC8 /* audio_frame_manipulator.h in Headers */,
> 				5CD2854E1E6A62130094FDC8 /* audio_frame_operations.h in Headers */,
>-				4144B3DD2169A070004363AC /* RTCVideoDecoder.h in Headers */,
> 				4102F6A221273262006AE8D7 /* audio_level.h in Headers */,
>-				413E67A02169889500EF37ED /* RTCVideoEncoderQpThresholds.h in Headers */,
> 				5CDD8A901E43C00F00621E92 /* audio_loop.h in Headers */,
> 				415F1FCD212730F000064CBF /* audio_mixer.h in Headers */,
> 				5CD284961E6A5F410094FDC8 /* audio_mixer_impl.h in Headers */,
>@@ -13854,7 +13929,6 @@
> 				5CDD8C4F1E43C58E00621E92 /* audio_sink.h in Headers */,
> 				5CDD8A921E43C00F00621E92 /* audio_sink.h in Headers */,
> 				5CDD8AD31E43C23900621E92 /* audio_state.h in Headers */,
>-				417953E3216983CA0028266B /* module_fec_types.h in Headers */,
> 				5CD284B31E6A5F9F0094FDC8 /* audio_state.h in Headers */,
> 				41F9BF9C2051C84C00ABF0B9 /* audio_transport_impl.h in Headers */,
> 				41F9BFA02051C88500ABF0B9 /* audio_util.h in Headers */,
>@@ -13877,7 +13951,6 @@
> 				5C4B4D0A1E4322F6002651C8 /* basicportallocator.h in Headers */,
> 				419241C62127593F00634FCF /* bbr_factory.h in Headers */,
> 				419241C32127593F00634FCF /* bbr_network_controller.h in Headers */,
>-				416225E52169819500A91C9B /* objc_video_encoder_factory.h in Headers */,
> 				413A235C1FE18E0700373E99 /* bind.h in Headers */,
> 				416D3BE1212731C200775F09 /* biquad_filter.h in Headers */,
> 				413A23771FE18E0700373E99 /* bitbuffer.h in Headers */,
>@@ -13886,7 +13959,6 @@
> 				41F2638221267B7700274F59 /* bitrate_constraints.h in Headers */,
> 				5CDD8F7E1E43CBE000621E92 /* bitrate_controller.h in Headers */,
> 				5CDD905E1E43D11200621E92 /* bitrate_controller_impl.h in Headers */,
>-				4144B3EA2169C5E3004363AC /* RTCVideoEncoderSettings+Private.h in Headers */,
> 				41E02CAC2127352D00C27CD6 /* bitrate_estimator.h in Headers */,
> 				5CDD8FA11E43CCBE00621E92 /* bitrate_prober.h in Headers */,
> 				41E02CD0212735B700C27CD6 /* bitrate_settings.h in Headers */,
>@@ -13894,7 +13966,6 @@
> 				41BE71A8215C463700A7B196 /* bitvect.h in Headers */,
> 				5CD285AD1E6A63430094FDC8 /* block_framer.h in Headers */,
> 				5CDD86881E43B93900621E92 /* block_mean_calculator.h in Headers */,
>-				417953C32169824B0028266B /* RTCH264ProfileLevelId.h in Headers */,
> 				5CD285B11E6A63430094FDC8 /* block_processor.h in Headers */,
> 				5CD285AF1E6A63430094FDC8 /* block_processor_metrics.h in Headers */,
> 				413A238E1FE18E0700373E99 /* buffer.h in Headers */,
>@@ -13902,16 +13973,13 @@
> 				413A232B1FE18E0700373E99 /* bufferqueue.h in Headers */,
> 				5CD285491E6A61D20094FDC8 /* builtin_audio_decoder_factory.h in Headers */,
> 				413091FC1EF8CFF800757C55 /* builtin_audio_encoder_factory.h in Headers */,
>-				417953E4216983CA0028266B /* module.h in Headers */,
>-				413E67792169867A00EF37ED /* channel_receive_proxy.h in Headers */,
>+				41FCBB1E21B1F7CD00A5DF27 /* builtin_video_bitrate_allocator_factory.h in Headers */,
> 				41F2636021267ADF00274F59 /* builtin_video_decoder_factory.h in Headers */,
>-				413E67692169854B00EF37ED /* RTCVideoEncoderVP8.h in Headers */,
> 				41F2636121267ADF00274F59 /* builtin_video_encoder_factory.h in Headers */,
> 				5CDD8B751E43C2B500621E92 /* bw_expand.h in Headers */,
> 				5CDD893F1E43BF3A00621E92 /* bye.h in Headers */,
> 				5CDD88761E43BE3C00621E92 /* byte_io.h in Headers */,
> 				413A23A41FE18E0800373E99 /* bytebuffer.h in Headers */,
>-				417953F8216984BE0028266B /* RTCCodecSpecificInfo.h in Headers */,
> 				41BE71AE215C463700A7B196 /* bytecode.h in Headers */,
> 				413A233E1FE18E0700373E99 /* byteorder.h in Headers */,
> 				5CD284B81E6A5F9F0094FDC8 /* call.h in Headers */,
>@@ -13933,16 +14001,16 @@
> 				5CDD8B831E43C2B500621E92 /* cb_update_best_index.h in Headers */,
> 				5C4B4C651E431F9C002651C8 /* channel_buffer.h in Headers */,
> 				5CDD8F811E43CBE000621E92 /* channel_controller.h in Headers */,
>+				417953D62169834F0028266B /* channel_receive.h in Headers */,
>+				417953D42169834F0028266B /* channel_send.h in Headers */,
> 				5CDD8B851E43C2B500621E92 /* chebyshev.h in Headers */,
> 				413A234F1FE18E0700373E99 /* checks.h in Headers */,
> 				5C08852A1E4A99D200403995 /* cipher.h in Headers */,
>-				413E67662169854B00EF37ED /* RTCVideoCodecConstants.h in Headers */,
> 				5CDD86A21E43B99400621E92 /* circular_buffer.h in Headers */,
>+				41FCBB7621B1FFA400A5DF27 /* cocoa_threading.h in Headers */,
> 				5C4B48DC1E42C1E3002651C8 /* codec.h in Headers */,
>-				416225E82169819500A91C9B /* objc_video_decoder_factory.h in Headers */,
> 				5CDD87B01E43BC0500621E92 /* codec.h in Headers */,
> 				41433CF31F79B33400387B4D /* codec.h in Headers */,
>-				5CDD85161E43B1EA00621E92 /* codec_manager.h in Headers */,
> 				5CDD83C91E439A6F00621E92 /* codec_timer.h in Headers */,
> 				4102F6E221273416006AE8D7 /* color_space.h in Headers */,
> 				5CDD8A0E1E43BFB300621E92 /* comfort_noise.h in Headers */,
>@@ -13961,27 +14029,31 @@
> 				5CDD8FEC1E43CDCA00621E92 /* config.h in Headers */,
> 				5C63F8DB1E416D53002CA531 /* config.h in Headers */,
> 				4192416D2127389B00634FCF /* congestion_controller_experiment.h in Headers */,
>-				41A08BD721272F1D001D5D7B /* congestion_window_pushback_controller.h in Headers */,
>+				41FCBB4021B1FDB800A5DF27 /* congestion_window_pushback_controller.h in Headers */,
> 				5CDD8A941E43C00F00621E92 /* constant_pcm_packet_source.h in Headers */,
> 				5C4B48781E42C1BA002651C8 /* constants.h in Headers */,
> 				5CDD8B8A1E43C2B500621E92 /* constants.h in Headers */,
> 				413A236C1FE18E0700373E99 /* constructormagic.h in Headers */,
> 				413AD1A421265B10003F7263 /* container.h in Headers */,
>+				41FCBB3821B1F8FC00A5DF27 /* control_handler.h in Headers */,
> 				5CDD8F861E43CBE000621E92 /* controller.h in Headers */,
> 				5CDD8F841E43CBE000621E92 /* controller_manager.h in Headers */,
> 				5CDD8AD41E43C23900621E92 /* conversion.h in Headers */,
> 				41F9BFB42051C93600ABF0B9 /* convert_legacy_video_factory.h in Headers */,
> 				413A23AC1FE18E0800373E99 /* copyonwritebuffer.h in Headers */,
> 				41BE71BC215C463700A7B196 /* coretype.h in Headers */,
>-				41889D76216BC4EC004614DD /* RTCRtpEncodingParameters.h in Headers */,
>+				41FCBB5121B1FE4600A5DF27 /* cpu_speed_experiment.h in Headers */,
> 				413A23E21FE18E0800373E99 /* cpu_time.h in Headers */,
> 				5CDD87B21E43BC0500621E92 /* crc.h in Headers */,
> 				413A22AC1FE18E0600373E99 /* crc32.h in Headers */,
> 				5CDD8B8C1E43C2B500621E92 /* create_augmented_vec.h in Headers */,
>+				41FCBB1221B1F53700A5DF27 /* create_peerconnection_factory.h in Headers */,
>+				41FCBB6421B1FEF600A5DF27 /* create_vp8_temporal_layers.h in Headers */,
> 				413A23691FE18E0700373E99 /* criticalsection.h in Headers */,
> 				5CDD8A101E43BFB300621E92 /* cross_correlation.h in Headers */,
> 				5C08852B1E4A99D200403995 /* crypto_kernel.h in Headers */,
> 				5C08852C1E4A99D200403995 /* crypto_types.h in Headers */,
>+				41FCBB2E21B1F8B700A5DF27 /* cryptooptions.h in Headers */,
> 				5C4B48DD1E42C1E3002651C8 /* cryptoparams.h in Headers */,
> 				41F9BF902051C80100ABF0B9 /* cryptoparams.h in Headers */,
> 				413A23291FE18E0700373E99 /* cryptstring.h in Headers */,
>@@ -14044,7 +14116,6 @@
> 				5CDD84B11E43AF1300621E92 /* echo_control_mobile_impl.h in Headers */,
> 				5CD285BB1E6A63430094FDC8 /* echo_path_delay_estimator.h in Headers */,
> 				5CD285BD1E6A63430094FDC8 /* echo_path_variability.h in Headers */,
>-				417953BD2169824B0028266B /* helpers.h in Headers */,
> 				5CD285C11E6A63430094FDC8 /* echo_remover.h in Headers */,
> 				5CD285BF1E6A63430094FDC8 /* echo_remover_metrics.h in Headers */,
> 				413A213E1FE0F0EF00373E99 /* ekt.h in Headers */,
>@@ -14053,6 +14124,7 @@
> 				5CDD8A961E43C00F00621E92 /* encode_neteq_input.h in Headers */,
> 				5CDD83CD1E439A6F00621E92 /* encoded_frame.h in Headers */,
> 				41A08BAC212681C8001D5D7B /* encoded_frame.h in Headers */,
>+				41FCBB1C21B1F7CD00A5DF27 /* encoded_image.h in Headers */,
> 				41E02C89212734B900C27CD6 /* encoder_database.h in Headers */,
> 				5CDD85A01E43B5C000621E92 /* encoder_rtcp_feedback.h in Headers */,
> 				41ECEAFC20646664009D5141 /* EncoderUtilities.h in Headers */,
>@@ -14067,7 +14139,6 @@
> 				5C08852E1E4A99D200403995 /* err.h in Headers */,
> 				41BE71AA215C463700A7B196 /* errwarn.h in Headers */,
> 				413A23A51FE18E0800373E99 /* event.h in Headers */,
>-				413E67852169877D00EF37ED /* RTCVideoEncoderSettings.h in Headers */,
> 				5CD286151E6A66130094FDC8 /* event_log_writer.h in Headers */,
> 				413A237E1FE18E0700373E99 /* event_tracer.h in Headers */,
> 				419C83501FE246650040C30F /* exp_filter.h in Headers */,
>@@ -14076,7 +14147,6 @@
> 				41BE71A6215C463700A7B196 /* expr.h in Headers */,
> 				5CDD894B1E43BF3A00621E92 /* extended_jitter_report.h in Headers */,
> 				5CDD894E1E43BF3A00621E92 /* extended_reports.h in Headers */,
>-				413E67A12169889500EF37ED /* RTCRtpFragmentationHeader.h in Headers */,
> 				5CDD8A981E43C00F00621E92 /* fake_decode_from_file.h in Headers */,
> 				41299B972127369100B3414B /* fake_network_pipe.h in Headers */,
> 				419241562127376F00634FCF /* fakecandidatepair.h in Headers */,
>@@ -14084,7 +14154,6 @@
> 				419241532127376F00634FCF /* fakedtlstransport.h in Headers */,
> 				419241592127376F00634FCF /* fakeicetransport.h in Headers */,
> 				5C4B48DF1E42C1E3002651C8 /* fakemediaengine.h in Headers */,
>-				413E676A2169854B00EF37ED /* RTCWrappedNativeVideoDecoder.h in Headers */,
> 				413A22B71FE18E0700373E99 /* fakenetwork.h in Headers */,
> 				5C4B48E01E42C1E3002651C8 /* fakenetworkinterface.h in Headers */,
> 				419241582127376F00634FCF /* fakepackettransport.h in Headers */,
>@@ -14104,7 +14173,6 @@
> 				41E02C8E212734B900C27CD6 /* fec_rate_table.h in Headers */,
> 				5CDD887C1E43BE3C00621E92 /* fec_test_helper.h in Headers */,
> 				41F2638921267F4000274F59 /* fft.h in Headers */,
>-				416225E32169819500A91C9B /* objc_frame_buffer.h in Headers */,
> 				41433D221F79B33400387B4D /* fft.h in Headers */,
> 				41F2636F21267B3E00274F59 /* fft4g.h in Headers */,
> 				41F9BFD32051DDE500ABF0B9 /* fft_buffer.h in Headers */,
>@@ -14119,12 +14187,10 @@
> 				5CDD8C8C1E43C66000621E92 /* file_utils.h in Headers */,
> 				41F263B2212680A800274F59 /* file_wrapper.h in Headers */,
> 				413A22E11FE18E0700373E99 /* filerotatingstream.h in Headers */,
>-				413A23BD1FE18E0800373E99 /* fileutils.h in Headers */,
> 				41433CED1F79B33400387B4D /* filterbank_internal.h in Headers */,
> 				41433D0D1F79B33400387B4D /* filterbank_tables.h in Headers */,
> 				5CDD8BA11E43C2B500621E92 /* filtered_cb_vecs.h in Headers */,
> 				5C11A0041E457400004F0987 /* fine_audio_buffer.h in Headers */,
>-				416225D62169818100A91C9B /* video_frame.h in Headers */,
> 				5CDD89511E43BF3A00621E92 /* fir.h in Headers */,
> 				5C4B4C6E1E431F9C002651C8 /* fir_filter.h in Headers */,
> 				419C831A1FE242B20040C30F /* fir_filter_c.h in Headers */,
>@@ -14133,12 +14199,10 @@
> 				413A23DF1FE18E0800373E99 /* firewallsocketserver.h in Headers */,
> 				41DDB24D21265BD700296D47 /* fixed_array.h in Headers */,
> 				415F1FAE21272FBA00064CBF /* fixed_digital_level_estimator.h in Headers */,
>-				416D3BE4212731C200775F09 /* fixed_gain_controller.h in Headers */,
> 				413A23BC1FE18E0800373E99 /* flags.h in Headers */,
> 				5CDD887F1E43BE3C00621E92 /* flexfec_header_reader_writer.h in Headers */,
> 				5CDD85371E43B39C00621E92 /* flexfec_receive_stream.h in Headers */,
> 				5CD284BA1E6A5F9F0094FDC8 /* flexfec_receive_stream_impl.h in Headers */,
>-				413E67812169870F00EF37ED /* svc_config.h in Headers */,
> 				419C83CB1FE247B40040C30F /* flexfec_receiver.h in Headers */,
> 				419C83CE1FE247B40040C30F /* flexfec_sender.h in Headers */,
> 				413A22A31FE18E0600373E99 /* format_macros.h in Headers */,
>@@ -14152,8 +14216,11 @@
> 				5CDD8BA31E43C2B500621E92 /* frame_classify.h in Headers */,
> 				5CD2849B1E6A5F410094FDC8 /* frame_combiner.h in Headers */,
> 				5CDD837A1E439A3500621E92 /* frame_dropper.h in Headers */,
>+				41FCBB5A21B1FE6E00A5DF27 /* frame_dumping_decoder.h in Headers */,
> 				5CDD8F911E43CBE000621E92 /* frame_length_controller.h in Headers */,
> 				5CDD83D41E439A6F00621E92 /* frame_object.h in Headers */,
>+				41FCBB2F21B1F8B700A5DF27 /* framedecryptorinterface.h in Headers */,
>+				41FCBB3021B1F8B700A5DF27 /* frameencryptorinterface.h in Headers */,
> 				413A22A21FE18E0600373E99 /* function_view.h in Headers */,
> 				41F2638E21267F4900274F59 /* g711.h in Headers */,
> 				4140B8251E4E3383007409E6 /* g711_interface.h in Headers */,
>@@ -14164,8 +14231,6 @@
> 				5CDD84B31E43AF1300621E92 /* gain_control_for_experimental_agc.h in Headers */,
> 				5CDD84B51E43AF1300621E92 /* gain_control_impl.h in Headers */,
> 				41A08BCF21272EE2001D5D7B /* gain_controller2.h in Headers */,
>-				416D3BE5212731C200775F09 /* gain_curve_applier.h in Headers */,
>-				41889D75216BB7B9004614DD /* RTCRtpEncodingParameters+Private.h in Headers */,
> 				5CDD8BA51E43C2B500621E92 /* gain_dequant.h in Headers */,
> 				5CDD87031E43BA7500621E92 /* gain_map_internal.h in Headers */,
> 				5CDD8BA71E43C2B500621E92 /* gain_quant.h in Headers */,
>@@ -14176,6 +14241,7 @@
> 				5CDD8BAD1E43C2B500621E92 /* get_sync_seq.h in Headers */,
> 				413A21411FE0F0EF00373E99 /* getopt_s.h in Headers */,
> 				5CDD872A1E43BABE00621E92 /* gmm.h in Headers */,
>+				41FCBB5F21B1FEC400A5DF27 /* goog_cc_factory.h in Headers */,
> 				41E02CAA2127352D00C27CD6 /* goog_cc_network_control.h in Headers */,
> 				413A23011FE18E0700373E99 /* gtest_prod_util.h in Headers */,
> 				413A22EA1FE18E0700373E99 /* gunit.h in Headers */,
>@@ -14187,6 +14253,8 @@
> 				4145E4BE1EF894F600FCF6E6 /* h264_profile_level_id.h in Headers */,
> 				5CD285EB1E6A639F0094FDC8 /* h264_sprop_parameter_sets.h in Headers */,
> 				5CDD83DA1E439A6F00621E92 /* h264_sps_pps_tracker.h in Headers */,
>+				41FCBB7221B1FF7400A5DF27 /* hdr_metadata.h in Headers */,
>+				417953BD2169824B0028266B /* helpers.h in Headers */,
> 				413A22D81FE18E0700373E99 /* helpers.h in Headers */,
> 				5CDD83DC1E439A6F00621E92 /* histogram.h in Headers */,
> 				419C83581FE246650040C30F /* histogram_percentile_counter.h in Headers */,
>@@ -14196,7 +14264,7 @@
> 				41A08BAD212681C8001D5D7B /* i010_buffer.h in Headers */,
> 				5CD284681E6A57F40094FDC8 /* i420_buffer.h in Headers */,
> 				41109AAD1E5FA19200C0955A /* i420_buffer_pool.h in Headers */,
>-				4144B3DE2169A070004363AC /* RTCVideoEncoder.h in Headers */,
>+				41FCBB4921B1FDEF00A5DF27 /* icecredentialsiterator.h in Headers */,
> 				41A08BC321269577001D5D7B /* icelogger.h in Headers */,
> 				413091F71EF8CFBD00757C55 /* iceserverparsing.h in Headers */,
> 				4192415A2127376F00634FCF /* icetransportinternal.h in Headers */,
>@@ -14231,6 +14299,7 @@
> 				5CDD83E21E439A6F00621E92 /* jitter_buffer.h in Headers */,
> 				5CDD83E01E439A6F00621E92 /* jitter_buffer_common.h in Headers */,
> 				5CDD83E41E439A6F00621E92 /* jitter_estimator.h in Headers */,
>+				41FCBB5421B1FE4600A5DF27 /* jitter_upper_bound_experiment.h in Headers */,
> 				5C63F9411E41737B002CA531 /* jsep.h in Headers */,
> 				5C63F9431E41737B002CA531 /* jsepicecandidate.h in Headers */,
> 				5C63F9451E41737B002CA531 /* jsepsessiondescription.h in Headers */,
>@@ -14239,12 +14308,13 @@
> 				5C63F8DE1E416D53002CA531 /* json.h in Headers */,
> 				413A23651FE18E0700373E99 /* keep_ref_until_done.h in Headers */,
> 				5C0885301E4A99D200403995 /* key.h in Headers */,
>+				41FCBB6E21B1FF3A00A5DF27 /* key_derivation.h in Headers */,
> 				4192419B2127586500634FCF /* kiss_fft.h in Headers */,
> 				5CDD84201E439B2900621E92 /* legacy_encoded_audio_frame.h in Headers */,
> 				5CDD84B91E43AF1300621E92 /* level_estimator_impl.h in Headers */,
>+				413E67732169863B00EF37ED /* libvpx_interface.h in Headers */,
> 				41294095212E128D00AD95E7 /* libvpx_vp8_decoder.h in Headers */,
> 				41294093212E128D00AD95E7 /* libvpx_vp8_encoder.h in Headers */,
>-				417953F7216984BE0028266B /* RTCEncodedImage.h in Headers */,
> 				415F1FB321272FBA00064CBF /* limiter.h in Headers */,
> 				41BE71AB215C463700A7B196 /* linemap.h in Headers */,
> 				41BE71B9215C463700A7B196 /* listfmt.h in Headers */,
>@@ -14253,10 +14323,10 @@
> 				5CDD8C101E43C34600621E92 /* locked_bandwidth_info.h in Headers */,
> 				413A23321FE18E0700373E99 /* logging.h in Headers */,
> 				413A23521FE18E0700373E99 /* logsinks.h in Headers */,
>+				41FCBB6921B1FF1B00A5DF27 /* loss_based_bandwidth_estimation.h in Headers */,
> 				419241C02127593F00634FCF /* loss_rate_filter.h in Headers */,
> 				5CDD87061E43BA7500621E92 /* loudness_histogram.h in Headers */,
> 				5CD2855E1E6A62ED0094FDC8 /* low_cut_filter.h in Headers */,
>-				416225D52169818100A91C9B /* video_encoder_factory.h in Headers */,
> 				419241402127372400634FCF /* lp_residual.h in Headers */,
> 				5CDD87C61E43BC0500621E92 /* lpc_analysis.h in Headers */,
> 				5CDD8BC21E43C2B500621E92 /* lpc_encode.h in Headers */,
>@@ -14274,15 +14344,17 @@
> 				5CDD8BCE1E43C2B500621E92 /* lsp_to_lsf.h in Headers */,
> 				413A23971FE18E0800373E99 /* macutils.h in Headers */,
> 				5CD285CC1E6A63430094FDC8 /* main_filter_update_gain.h in Headers */,
>+				41FCBB3D21B1FD4000A5DF27 /* match.h in Headers */,
> 				5CD285D01E6A63430094FDC8 /* matched_filter.h in Headers */,
>-				4144B3DF2169A070004363AC /* RTCVideoEncoderFactory.h in Headers */,
> 				5CD285CE1E6A63430094FDC8 /* matched_filter_lag_aggregator.h in Headers */,
> 				419C83591FE246650040C30F /* mathutils.h in Headers */,
> 				41F9BFD62051DDE500ABF0B9 /* matrix_buffer.h in Headers */,
> 				41BE71B8215C463700A7B196 /* md5.h in Headers */,
>+				41FCBB4621B1FDEF00A5DF27 /* mdns_message.h in Headers */,
> 				5CDD86A51E43B99400621E92 /* mean_variance_estimator.h in Headers */,
> 				5CDD83E61E439A6F00621E92 /* media_opt_util.h in Headers */,
> 				5CDD83E81E439A6F00621E92 /* media_optimization.h in Headers */,
>+				41FCBB0F21B1F53700A5DF27 /* media_transport_interface.h in Headers */,
> 				5C4B48E51E42C1E3002651C8 /* mediachannel.h in Headers */,
> 				4102F6B221273331006AE8D7 /* mediaconfig.h in Headers */,
> 				5C4B48E71E42C1E3002651C8 /* mediaconstants.h in Headers */,
>@@ -14298,6 +14370,7 @@
> 				5CD286331E6A67BF0094FDC8 /* mediatypes.h in Headers */,
> 				41A08BB821268A7D001D5D7B /* memory.h in Headers */,
> 				413A23091FE18E0700373E99 /* memory_usage.h in Headers */,
>+				41FCBB8521B2007B00A5DF27 /* memutil.h in Headers */,
> 				5CDD8A301E43BFB300621E92 /* merge.h in Headers */,
> 				413A22FA1FE18E0700373E99 /* messagedigest.h in Headers */,
> 				413A22B91FE18E0700373E99 /* messagehandler.h in Headers */,
>@@ -14306,21 +14379,23 @@
> 				5C11A0051E457400004F0987 /* mock_audio_device_buffer.h in Headers */,
> 				5CDD8FED1E43CDCA00621E92 /* mock_audio_processing.h in Headers */,
> 				419C83561FE246650040C30F /* mod_ops.h in Headers */,
>-				5CDD837E1E439A3500621E92 /* moving_average.h in Headers */,
>+				417953E4216983CA0028266B /* module.h in Headers */,
>+				417953E7216983CA0028266B /* module_common_types.h in Headers */,
>+				417953E5216983CA0028266B /* module_common_types_public.h in Headers */,
>+				417953E3216983CA0028266B /* module_fec_types.h in Headers */,
>+				41FCBB1621B1F7AA00A5DF27 /* moving_average.h in Headers */,
> 				5CD286111E6A64C90094FDC8 /* moving_max.h in Headers */,
>-				41889D78216BC4EC004614DD /* RTCRtpFragmentationHeader+Private.h in Headers */,
> 				419C83551FE246650040C30F /* moving_max_counter.h in Headers */,
> 				419C83521FE246650040C30F /* moving_median_filter.h in Headers */,
> 				5CDD8C8F1E43C66000621E92 /* moving_moments.h in Headers */,
> 				5CDD8BD01E43C2B500621E92 /* my_corr.h in Headers */,
> 				5CDD89541E43BF3A00621E92 /* nack.h in Headers */,
> 				5CDD83E91E439A6F00621E92 /* nack_fec_tables.h in Headers */,
>-				417953EE216984210028266B /* vp9_profile.h in Headers */,
> 				5CDD83EB1E439A6F00621E92 /* nack_module.h in Headers */,
> 				5CDD8A331E43BFB300621E92 /* nack_tracker.h in Headers */,
> 				4145E4B71EF8943E00FCF6E6 /* nalu_rewriter.h in Headers */,
>+				417953BF2169824B0028266B /* nalu_rewriter.h in Headers */,
> 				413A22C41FE18E0700373E99 /* natserver.h in Headers */,
>-				4144B3E02169A070004363AC /* RTCVideoDecoderFactory.h in Headers */,
> 				413A22D91FE18E0700373E99 /* natsocketfactory.h in Headers */,
> 				413A22E71FE18E0700373E99 /* nattypes.h in Headers */,
> 				5CDD8BD21E43C2B500621E92 /* nearest_neighbor.h in Headers */,
>@@ -14335,7 +14410,6 @@
> 				41E02CCF212735B700C27CD6 /* network_control.h in Headers */,
> 				41E02CCE212735B700C27CD6 /* network_types.h in Headers */,
> 				413A237C1FE18E0700373E99 /* networkmonitor.h in Headers */,
>-				413E6791216987DB00EF37ED /* RTCVideoFrameBuffer.h in Headers */,
> 				413A22C21FE18E0700373E99 /* networkroute.h in Headers */,
> 				5CDD872B1E43BABE00621E92 /* noise_gmm_tables.h in Headers */,
> 				415F1FB421272FBA00064CBF /* noise_spectrum_estimator.h in Headers */,
>@@ -14343,11 +14417,12 @@
> 				5CDD84EA1E43B0B600621E92 /* noise_suppression_impl.h in Headers */,
> 				5CDD85FD1E43B84E00621E92 /* noise_suppression_x.h in Headers */,
> 				5CDD8A3E1E43BFB300621E92 /* normal.h in Headers */,
>-				417953EB216983F10028266B /* rtt_mult_experiment.h in Headers */,
>+				41FCBB5521B1FE4600A5DF27 /* normalize_simulcast_size_experiment.h in Headers */,
> 				5CDD86A81E43B99400621E92 /* normalized_covariance_estimator.h in Headers */,
> 				5C63F9541E41737B002CA531 /* notifier.h in Headers */,
> 				5CDD86011E43B84E00621E92 /* ns_core.h in Headers */,
> 				41ECEAC120640F28009D5141 /* NSString+StdString.h in Headers */,
>+				4144B3EF2169C63F004363AC /* NSString+StdString.h in Headers */,
> 				5CDD86061E43B84E00621E92 /* nsx_core.h in Headers */,
> 				5CDD86071E43B84E00621E92 /* nsx_defines.h in Headers */,
> 				419C83B41FE2472E0040C30F /* null_audio_poller.h in Headers */,
>@@ -14355,14 +14430,18 @@
> 				5C0885321E4A99D200403995 /* null_cipher.h in Headers */,
> 				413A23C01FE18E0800373E99 /* nullsocketserver.h in Headers */,
> 				5CD285331E6A61110094FDC8 /* nullwebrtcvideoengine.h in Headers */,
>+				416225E32169819500A91C9B /* objc_frame_buffer.h in Headers */,
>+				416225E82169819500A91C9B /* objc_video_decoder_factory.h in Headers */,
>+				416225E52169819500A91C9B /* objc_video_encoder_factory.h in Headers */,
>+				416225E92169819500A91C9B /* objc_video_frame.h in Headers */,
> 				413A23361FE18E0700373E99 /* onetimeevent.h in Headers */,
> 				5CDD86951E43B93900621E92 /* ooura_fft.h in Headers */,
> 				5CDD86921E43B93900621E92 /* ooura_fft_tables_common.h in Headers */,
>+				41FCBB7E21B1FFF600A5DF27 /* openssl_key_derivation_hkdf.h in Headers */,
> 				41DDB266212679A300296D47 /* opensslcertificate.h in Headers */,
> 				41DDB263212679A300296D47 /* opensslsessioncache.h in Headers */,
> 				41DDB261212679A300296D47 /* opensslutility.h in Headers */,
> 				41DDB25121265BE900296D47 /* optional.h in Headers */,
>-				413A23451FE18E0700373E99 /* optionsfile.h in Headers */,
> 				5CDD8C661E43C60900621E92 /* opus_inst.h in Headers */,
> 				5CDD8C681E43C60900621E92 /* opus_interface.h in Headers */,
> 				5CDD87CF1E43BC0500621E92 /* os_specific_inline.h in Headers */,
>@@ -14378,7 +14457,6 @@
> 				41F2639921267F5E00274F59 /* pa_ringbuffer.h in Headers */,
> 				5CDD8FA31E43CCBE00621E92 /* paced_sender.h in Headers */,
> 				419C832E1FE245CD0040C30F /* pacer.h in Headers */,
>-				4192418D212749C800634FCF /* pacer_controller.h in Headers */,
> 				5CDD8BD41E43C2B500621E92 /* pack_bits.h in Headers */,
> 				5CDD83EF1E439A6F00621E92 /* packet.h in Headers */,
> 				5CDD8AB01E43C00F00621E92 /* packet.h in Headers */,
>@@ -14386,7 +14464,6 @@
> 				5CDD8A411E43BFB300621E92 /* packet_buffer.h in Headers */,
> 				5CDD83ED1E439A6F00621E92 /* packet_buffer.h in Headers */,
> 				5CDD888D1E43BE3C00621E92 /* packet_loss_stats.h in Headers */,
>-				413E67952169881900EF37ED /* svc_rate_allocator.h in Headers */,
> 				419241C82127593F00634FCF /* packet_number_indexed_queue.h in Headers */,
> 				4102F6B921273382006AE8D7 /* packet_receiver.h in Headers */,
> 				5CDD8FA51E43CCBE00621E92 /* packet_router.h in Headers */,
>@@ -14395,7 +14472,6 @@
> 				5C63F9F41E4174F6002CA531 /* packetsocketfactory.h in Headers */,
> 				5C63F9F51E4174F6002CA531 /* packettransportinterface.h in Headers */,
> 				419C83D91FE247F20040C30F /* packettransportinternal.h in Headers */,
>-				413A23901FE18E0800373E99 /* pathutils.h in Headers */,
> 				5C4B48841E42C1BA002651C8 /* payload_type_mapper.h in Headers */,
> 				5CDD90011E43CE3A00621E92 /* pcm16b.h in Headers */,
> 				5CD285001E6A60570094FDC8 /* peerconnection.h in Headers */,
>@@ -14405,7 +14481,6 @@
> 				4102F69521273206006AE8D7 /* peerconnectioninternal.h in Headers */,
> 				5C63F95B1E41737B002CA531 /* peerconnectionproxy.h in Headers */,
> 				41F9BFE02051DE1800ABF0B9 /* peerconnectionwrapper.h in Headers */,
>-				417953BB216982420028266B /* RTCDefaultVideoDecoderFactory.h in Headers */,
> 				419C834E1FE246650040C30F /* percentile_filter.h in Headers */,
> 				41BE71C0215C463700A7B196 /* phash.h in Headers */,
> 				413A23EA1FE18E0800373E99 /* physicalsocketserver.h in Headers */,
>@@ -14426,10 +14501,8 @@
> 				413A231D1FE18E0700373E99 /* platform_thread_types.h in Headers */,
> 				5CDD88901E43BE3C00621E92 /* playout_delay_oracle.h in Headers */,
> 				5CDD89571E43BF3A00621E92 /* pli.h in Headers */,
>-				413E676C2169854B00EF37ED /* RTCWrappedNativeVideoEncoder.h in Headers */,
> 				5CDD87341E43BABE00621E92 /* pole_zero_filter.h in Headers */,
> 				5CDD8BD61E43C2B500621E92 /* poly_to_lsf.h in Headers */,
>-				417953BF2169824B0028266B /* nalu_rewriter.h in Headers */,
> 				5CDD8BD81E43C2B500621E92 /* poly_to_lsp.h in Headers */,
> 				5C63F9F81E4174F6002CA531 /* port.h in Headers */,
> 				5C63F9FB1E4174F6002CA531 /* portallocator.h in Headers */,
>@@ -14446,20 +14519,17 @@
> 				413A231F1FE18E0700373E99 /* proxyinfo.h in Headers */,
> 				413A23C51FE18E0800373E99 /* proxyserver.h in Headers */,
> 				5C63F9FF1E4174F6002CA531 /* pseudotcp.h in Headers */,
>-				4106D56D216C2BAA001E3C40 /* string_view.h in Headers */,
> 				5CDD89591E43BF3A00621E92 /* psfb.h in Headers */,
> 				5CDD87561E43BAF500621E92 /* push_sinc_resampler.h in Headers */,
> 				5CDD83821E439A3500621E92 /* quality_scaler.h in Headers */,
> 				4192416C2127389B00634FCF /* quality_scaling_experiment.h in Headers */,
> 				5CD2862B1E6A66C80094FDC8 /* quality_threshold.h in Headers */,
>-				4144B3D42169896D004363AC /* RTCNativeI420Buffer+Private.h in Headers */,
> 				413A23EF1FE18E0800373E99 /* race_checker.h in Headers */,
> 				413A234C1FE18E0700373E99 /* random.h in Headers */,
> 				5CDD8A4B1E43BFB300621E92 /* random_vector.h in Headers */,
> 				5CDD895C1E43BF3A00621E92 /* rapid_resync_request.h in Headers */,
> 				413A23601FE18E0700373E99 /* rate_limiter.h in Headers */,
> 				413A230D1FE18E0700373E99 /* rate_statistics.h in Headers */,
>-				4144B3EF2169C63F004363AC /* NSString+StdString.h in Headers */,
> 				413A23041FE18E0700373E99 /* ratetracker.h in Headers */,
> 				41A08BCB2126961F001D5D7B /* raw_logging.h in Headers */,
> 				5C0885331E4A99D200403995 /* rdb.h in Headers */,
>@@ -14483,7 +14553,6 @@
> 				5C63FA021E4174F6002CA531 /* relayport.h in Headers */,
> 				5C63FA051E4174F6002CA531 /* relayserver.h in Headers */,
> 				5CDD89621E43BF3A00621E92 /* remb.h in Headers */,
>-				413E677E216986AE00EF37ED /* rtp_header_extension_size.h in Headers */,
> 				4102F6A721273262006AE8D7 /* remix_resample.h in Headers */,
> 				5CDD8FDC1E43CD6600621E92 /* remote_bitrate_estimator_abs_send_time.h in Headers */,
> 				5CDD8FDE1E43CD6600621E92 /* remote_bitrate_estimator_single_stream.h in Headers */,
>@@ -14516,7 +14585,6 @@
> 				41F9BFA92051C8E200ABF0B9 /* rtc_event_alr_state.h in Headers */,
> 				419C839D1FE246F90040C30F /* rtc_event_audio_network_adaptation.h in Headers */,
> 				419C83981FE246F90040C30F /* rtc_event_audio_playout.h in Headers */,
>-				413E678A216987B900EF37ED /* RTCCVPixelBuffer.h in Headers */,
> 				419C83A11FE246F90040C30F /* rtc_event_audio_receive_stream_config.h in Headers */,
> 				419C838E1FE246F90040C30F /* rtc_event_audio_send_stream_config.h in Headers */,
> 				419C83931FE246F90040C30F /* rtc_event_bwe_update_delay_based.h in Headers */,
>@@ -14529,7 +14597,6 @@
> 				419C84231FE24BA60040C30F /* rtc_event_log_output_file.h in Headers */,
> 				419C839F1FE246F90040C30F /* rtc_event_probe_cluster_created.h in Headers */,
> 				419C83A01FE246F90040C30F /* rtc_event_probe_result_failure.h in Headers */,
>-				4144B3D22169896D004363AC /* RTCNativeI420Buffer.h in Headers */,
> 				419C839A1FE246F90040C30F /* rtc_event_probe_result_success.h in Headers */,
> 				419C83901FE246F90040C30F /* rtc_event_rtcp_packet_incoming.h in Headers */,
> 				419C83941FE246F90040C30F /* rtc_event_rtcp_packet_outgoing.h in Headers */,
>@@ -14537,7 +14604,6 @@
> 				419C83AB1FE247080040C30F /* rtc_event_rtp_packet_outgoing.h in Headers */,
> 				419C83AE1FE247080040C30F /* rtc_event_video_receive_stream_config.h in Headers */,
> 				419C83B01FE247080040C30F /* rtc_event_video_send_stream_config.h in Headers */,
>-				417953D42169834F0028266B /* channel_send.h in Headers */,
> 				419C841E1FE24AEF0040C30F /* rtc_stream_config.h in Headers */,
> 				413A24621FE1991A00373E99 /* RTCAudioSession.h in Headers */,
> 				413A24581FE1991A00373E99 /* RTCAudioSessionConfiguration.h in Headers */,
>@@ -14545,21 +14611,28 @@
> 				413A24571FE1991A00373E99 /* RTCAudioTrack.h in Headers */,
> 				413A245B1FE1991A00373E99 /* RTCCameraPreviewView.h in Headers */,
> 				413A244D1FE1991A00373E99 /* RTCCameraVideoCapturer.h in Headers */,
>-				417953F9216984BE0028266B /* RTCI420Buffer.h in Headers */,
> 				413A23231FE18E0700373E99 /* rtccertificate.h in Headers */,
> 				413A22DB1FE18E0700373E99 /* rtccertificategenerator.h in Headers */,
>+				417953F8216984BE0028266B /* RTCCodecSpecificInfo.h in Headers */,
>+				417953BA216982420028266B /* RTCCodecSpecificInfoH264+Private.h in Headers */,
>+				417953C02169824B0028266B /* RTCCodecSpecificInfoH264.h in Headers */,
> 				413A24441FE1991A00373E99 /* RTCConfiguration.h in Headers */,
>+				413E678A216987B900EF37ED /* RTCCVPixelBuffer.h in Headers */,
> 				413A24551FE1991A00373E99 /* RTCDataChannel.h in Headers */,
> 				413A24701FE1991A00373E99 /* RTCDataChannelConfiguration.h in Headers */,
>-				416225D22169818100A91C9B /* video_decoder_factory.h in Headers */,
>+				417953BB216982420028266B /* RTCDefaultVideoDecoderFactory.h in Headers */,
>+				417953C22169824B0028266B /* RTCDefaultVideoEncoderFactory.h in Headers */,
> 				413A24521FE1991A00373E99 /* RTCDispatcher.h in Headers */,
> 				413A245D1FE1991A00373E99 /* RTCEAGLVideoView.h in Headers */,
>+				4144B3F6216B081B004363AC /* RTCEncodedImage+Private.h in Headers */,
>+				417953F7216984BE0028266B /* RTCEncodedImage.h in Headers */,
> 				41F411B11EF8DA0200343C26 /* rtcerror.h in Headers */,
>-				417953E5216983CA0028266B /* module_common_types_public.h in Headers */,
> 				41F9BF882051C80100ABF0B9 /* rtceventlogoutput.h in Headers */,
> 				413A24401FE1991A00373E99 /* RTCFieldTrials.h in Headers */,
> 				413A245A1FE1991A00373E99 /* RTCFileLogger.h in Headers */,
> 				413A24561FE1991A00373E99 /* RTCFileVideoCapturer.h in Headers */,
>+				417953C32169824B0028266B /* RTCH264ProfileLevelId.h in Headers */,
>+				417953F9216984BE0028266B /* RTCI420Buffer.h in Headers */,
> 				413A244E1FE1991A00373E99 /* RTCIceCandidate.h in Headers */,
> 				413A245E1FE1991A00373E99 /* RTCIceServer.h in Headers */,
> 				413A24481FE1991A00373E99 /* RTCIntervalRange.h in Headers */,
>@@ -14574,12 +14647,13 @@
> 				413A246A1FE1991A00373E99 /* RTCMetricsSampleInfo.h in Headers */,
> 				413A24671FE1991A00373E99 /* RTCMTLNSVideoView.h in Headers */,
> 				413A24711FE1991A00373E99 /* RTCMTLVideoView.h in Headers */,
>+				4144B3D42169896D004363AC /* RTCNativeI420Buffer+Private.h in Headers */,
>+				4144B3D22169896D004363AC /* RTCNativeI420Buffer.h in Headers */,
>+				4144B3D52169896D004363AC /* RTCNativeMutableI420Buffer.h in Headers */,
> 				413A24591FE1991A00373E99 /* RTCNSGLVideoView.h in Headers */,
>-				5CDD8A501E43BFB300621E92 /* rtcp.h in Headers */,
> 				419C82B21FE20DCD0040C30F /* rtcp_demuxer.h in Headers */,
> 				5CD285FB1E6A64520094FDC8 /* rtcp_nack_stats.h in Headers */,
> 				5CDD88981E43BE3C00621E92 /* rtcp_packet.h in Headers */,
>-				4144B3EB2169C5E3004363AC /* RTCVideoCodecInfo+Private.h in Headers */,
> 				419C82BD1FE20DCD0040C30F /* rtcp_packet_sink_interface.h in Headers */,
> 				5CDD889B1E43BE3C00621E92 /* rtcp_receiver.h in Headers */,
> 				5CDD889E1E43BE3C00621E92 /* rtcp_sender.h in Headers */,
>@@ -14588,15 +14662,17 @@
> 				419C83051FE20F020040C30F /* rtcp_transceiver_impl.h in Headers */,
> 				413A243E1FE1991A00373E99 /* RTCPeerConnection.h in Headers */,
> 				413A24631FE1991A00373E99 /* RTCPeerConnectionFactory.h in Headers */,
>-				417953C22169824B0028266B /* RTCDefaultVideoEncoderFactory.h in Headers */,
> 				413A246C1FE1991A00373E99 /* RTCRtpCodecParameters.h in Headers */,
>+				41889D75216BB7B9004614DD /* RTCRtpEncodingParameters+Private.h in Headers */,
>+				41889D76216BC4EC004614DD /* RTCRtpEncodingParameters.h in Headers */,
> 				413A244C1FE1991A00373E99 /* RTCRtpEncodingParameters.h in Headers */,
>+				41889D78216BC4EC004614DD /* RTCRtpFragmentationHeader+Private.h in Headers */,
>+				413E67A12169889500EF37ED /* RTCRtpFragmentationHeader.h in Headers */,
> 				413A24451FE1991A00373E99 /* RTCRtpParameters.h in Headers */,
> 				413A243B1FE1991A00373E99 /* RTCRtpReceiver.h in Headers */,
> 				413A245C1FE1991A00373E99 /* RTCRtpSender.h in Headers */,
> 				413A244A1FE1991A00373E99 /* RTCSessionDescription.h in Headers */,
> 				413A24661FE1991A00373E99 /* RTCSSLAdapter.h in Headers */,
>-				417953C62169824B0028266B /* RTCVideoDecoderH264.h in Headers */,
> 				4145E4C81EF896D700FCF6E6 /* rtcstats.h in Headers */,
> 				4145E4C71EF896D700FCF6E6 /* rtcstats_objects.h in Headers */,
> 				5CD285081E6A60570094FDC8 /* rtcstatscollector.h in Headers */,
>@@ -14607,19 +14683,37 @@
> 				413A243D1FE1991A00373E99 /* RTCVideoCapturer.h in Headers */,
> 				41ECEAB620630108009D5141 /* RTCVideoCodec+Private.h in Headers */,
> 				413A24431FE1991A00373E99 /* RTCVideoCodec.h in Headers */,
>+				413E67662169854B00EF37ED /* RTCVideoCodecConstants.h in Headers */,
> 				413A24531FE1991A00373E99 /* RTCVideoCodecFactory.h in Headers */,
> 				413A243C1FE1991A00373E99 /* RTCVideoCodecH264.h in Headers */,
>+				4144B3EB2169C5E3004363AC /* RTCVideoCodecInfo+Private.h in Headers */,
>+				413E67992169883900EF37ED /* RTCVideoCodecInfo.h in Headers */,
>+				4144B3DD2169A070004363AC /* RTCVideoDecoder.h in Headers */,
>+				4144B3E02169A070004363AC /* RTCVideoDecoderFactory.h in Headers */,
>+				417953C52169824B0028266B /* RTCVideoDecoderFactoryH264.h in Headers */,
>+				417953C62169824B0028266B /* RTCVideoDecoderH264.h in Headers */,
> 				413A24651FE1991A00373E99 /* RTCVideoDecoderVP8.h in Headers */,
> 				413A24601FE1991A00373E99 /* RTCVideoDecoderVP9.h in Headers */,
>+				4144B3DE2169A070004363AC /* RTCVideoEncoder.h in Headers */,
>+				4144B3DF2169A070004363AC /* RTCVideoEncoderFactory.h in Headers */,
>+				417953C82169824B0028266B /* RTCVideoEncoderFactoryH264.h in Headers */,
>+				417953C92169824B0028266B /* RTCVideoEncoderH264.h in Headers */,
>+				413E67A02169889500EF37ED /* RTCVideoEncoderQpThresholds.h in Headers */,
>+				4144B3EA2169C5E3004363AC /* RTCVideoEncoderSettings+Private.h in Headers */,
>+				413E67852169877D00EF37ED /* RTCVideoEncoderSettings.h in Headers */,
>+				413E67692169854B00EF37ED /* RTCVideoEncoderVP8.h in Headers */,
> 				413A246E1FE1991A00373E99 /* RTCVideoEncoderVP8.h in Headers */,
> 				413A24461FE1991A00373E99 /* RTCVideoEncoderVP9.h in Headers */,
>-				413E67732169863B00EF37ED /* libvpx_interface.h in Headers */,
> 				413A244F1FE1991A00373E99 /* RTCVideoFrame.h in Headers */,
>+				413E678F216987DB00EF37ED /* RTCVideoFrame.h in Headers */,
>+				413E6791216987DB00EF37ED /* RTCVideoFrameBuffer.h in Headers */,
> 				413A24411FE1991A00373E99 /* RTCVideoFrameBuffer.h in Headers */,
> 				413A246F1FE1991A00373E99 /* RTCVideoRenderer.h in Headers */,
> 				413A24611FE1991A00373E99 /* RTCVideoSource.h in Headers */,
> 				413A24501FE1991A00373E99 /* RTCVideoTrack.h in Headers */,
> 				413A243A1FE1991A00373E99 /* RTCVideoViewShading.h in Headers */,
>+				413E676A2169854B00EF37ED /* RTCWrappedNativeVideoDecoder.h in Headers */,
>+				413E676C2169854B00EF37ED /* RTCWrappedNativeVideoEncoder.h in Headers */,
> 				419C82B81FE20DCD0040C30F /* rtp_config.h in Headers */,
> 				419C83C91FE247B40040C30F /* rtp_cvo.h in Headers */,
> 				413092011EF8D0A600757C55 /* rtp_demuxer.h in Headers */,
>@@ -14630,11 +14724,11 @@
> 				5CDD88AC1E43BE3C00621E92 /* rtp_format_vp8.h in Headers */,
> 				5CDD88AF1E43BE3C00621E92 /* rtp_format_vp9.h in Headers */,
> 				5CDD83F71E439A6F00621E92 /* rtp_frame_reference_finder.h in Headers */,
>-				417953D62169834F0028266B /* channel_receive.h in Headers */,
> 				5CDD8AB91E43C00F00621E92 /* rtp_generator.h in Headers */,
> 				41E02C7B2127347400C27CD6 /* rtp_generic_frame_descriptor.h in Headers */,
> 				41E02C792127347400C27CD6 /* rtp_generic_frame_descriptor_extension.h in Headers */,
> 				419C83C41FE247B40040C30F /* rtp_header_extension_map.h in Headers */,
>+				413E677E216986AE00EF37ED /* rtp_header_extension_size.h in Headers */,
> 				5CDD88B61E43BE3C00621E92 /* rtp_header_extensions.h in Headers */,
> 				419C83CC1FE247B40040C30F /* rtp_header_parser.h in Headers */,
> 				41F9BF8D2051C80100ABF0B9 /* rtp_headers.h in Headers */,
>@@ -14642,7 +14736,6 @@
> 				5CDD88BA1E43BE3D00621E92 /* rtp_packet_history.h in Headers */,
> 				5CDD88BB1E43BE3D00621E92 /* rtp_packet_received.h in Headers */,
> 				419C82B61FE20DCD0040C30F /* rtp_packet_sink_interface.h in Headers */,
>-				413E678F216987DB00EF37ED /* RTCVideoFrame.h in Headers */,
> 				5CDD88BC1E43BE3D00621E92 /* rtp_packet_to_send.h in Headers */,
> 				41D6B45A21273159008F9353 /* rtp_payload_params.h in Headers */,
> 				419C83C31FE247B40040C30F /* rtp_rtcp.h in Headers */,
>@@ -14679,6 +14772,7 @@
> 				419C82DB1FE20E590040C30F /* rtptransportinternaladapter.h in Headers */,
> 				5C4B48F21E42C1E3002651C8 /* rtputils.h in Headers */,
> 				5CDD83F91E439A6F00621E92 /* rtt_filter.h in Headers */,
>+				417953EB216983F10028266B /* rtt_mult_experiment.h in Headers */,
> 				419241CB2127593F00634FCF /* rtt_stats.h in Headers */,
> 				419C82C21FE20DCD0040C30F /* rtx_receive_stream.h in Headers */,
> 				419241622127385B00634FCF /* rw_lock_posix.h in Headers */,
>@@ -14690,6 +14784,7 @@
> 				413A23281FE18E0700373E99 /* sanitizer.h in Headers */,
> 				416D3BEA212731C200775F09 /* saturation_protector.h in Headers */,
> 				413A23051FE18E0700373E99 /* scoped_ref_ptr.h in Headers */,
>+				41FCBB1021B1F53700A5DF27 /* scoped_refptr.h in Headers */,
> 				41F9BFB62051C93600ABF0B9 /* scopedvideodecoder.h in Headers */,
> 				41F9BFB72051C93600ABF0B9 /* scopedvideoencoder.h in Headers */,
> 				419C84361FE24E800040C30F /* screenshare_layers.h in Headers */,
>@@ -14704,6 +14799,7 @@
> 				5CDD85BA1E43B5C000621E92 /* send_statistics_proxy.h in Headers */,
> 				4192418B212749C800634FCF /* send_time_history.h in Headers */,
> 				5CDD89731E43BF3A00621E92 /* sender_report.h in Headers */,
>+				41FCBB2821B1F87E00A5DF27 /* sent_packet.h in Headers */,
> 				419241AC212758D300634FCF /* sequence_buffer.h in Headers */,
> 				413A22D41FE18E0700373E99 /* sequenced_task_checker.h in Headers */,
> 				413A23AE1FE18E0800373E99 /* sequenced_task_checker_impl.h in Headers */,
>@@ -14715,7 +14811,6 @@
> 				5CD285E01E6A63430094FDC8 /* shadow_filter_update_gain.h in Headers */,
> 				415F1FB521272FBA00064CBF /* signal_classifier.h in Headers */,
> 				41F9BFC52051CAE500ABF0B9 /* signal_processing_library.h in Headers */,
>-				417953D72169834F0028266B /* channel_send_proxy.h in Headers */,
> 				413A23811FE18E0700373E99 /* signalthread.h in Headers */,
> 				41F263BC2126811500274F59 /* sigslot.h in Headers */,
> 				413A23331FE18E0700373E99 /* sigslotrepeater.h in Headers */,
>@@ -14755,7 +14850,6 @@
> 				413A213F1FE0F0EF00373E99 /* srtp.h in Headers */,
> 				413A213D1FE0F0EF00373E99 /* srtp_priv.h in Headers */,
> 				419C82E11FE20E590040C30F /* srtpsession.h in Headers */,
>-				417953C82169824B0028266B /* RTCVideoEncoderFactoryH264.h in Headers */,
> 				419C82E21FE20E590040C30F /* srtptransport.h in Headers */,
> 				41E02CBD2127356A00C27CD6 /* sslcertificate.h in Headers */,
> 				413A23AF1FE18E0800373E99 /* sslfingerprint.h in Headers */,
>@@ -14766,7 +14860,6 @@
> 				5CDD87371E43BABE00621E92 /* standalone_vad.h in Headers */,
> 				5C0885351E4A99D200403995 /* stat.h in Headers */,
> 				5CDD8BEC1E43C2B500621E92 /* state_construct.h in Headers */,
>-				416225E92169819500A91C9B /* objc_video_frame.h in Headers */,
> 				5CDD8BEE1E43C2B500621E92 /* state_search.h in Headers */,
> 				5CDD8A521E43BFB300621E92 /* statistics_calculator.h in Headers */,
> 				5CDD85BD1E43B5C000621E92 /* stats_counter.h in Headers */,
>@@ -14778,6 +14871,7 @@
> 				5C4B48F51E42C1E3002651C8 /* streamparams.h in Headers */,
> 				41DDB26C212679D200296D47 /* string_builder.h in Headers */,
> 				413A23641FE18E0700373E99 /* string_to_number.h in Headers */,
>+				4106D56D216C2BAA001E3C40 /* string_view.h in Headers */,
> 				413A22EB1FE18E0700373E99 /* stringencode.h in Headers */,
> 				413A23B11FE18E0800373E99 /* stringize_macros.h in Headers */,
> 				413A22CE1FE18E0700373E99 /* stringutils.h in Headers */,
>@@ -14791,13 +14885,14 @@
> 				5CD285E11E6A63430094FDC8 /* subtractor_output.h in Headers */,
> 				5CD285E51E6A63430094FDC8 /* suppression_filter.h in Headers */,
> 				5CD285E71E6A63430094FDC8 /* suppression_gain.h in Headers */,
>+				413E67812169870F00EF37ED /* svc_config.h in Headers */,
>+				413E67952169881900EF37ED /* svc_rate_allocator.h in Headers */,
> 				5CDD8BF01E43C2B500621E92 /* swap_bytes.h in Headers */,
> 				413A23471FE18E0700373E99 /* swap_queue.h in Headers */,
> 				419241A9212758D300634FCF /* symmetric_matrix_buffer.h in Headers */,
> 				41BE71A3215C463700A7B196 /* symrec.h in Headers */,
> 				5CDD8A551E43BFB300621E92 /* sync_buffer.h in Headers */,
> 				5CD284BF1E6A5F9F0094FDC8 /* syncable.h in Headers */,
>-				4144B3F6216B081B004363AC /* RTCEncodedImage+Private.h in Headers */,
> 				5CDD89791E43BF3A00621E92 /* target_bitrate.h in Headers */,
> 				413A23391FE18E0700373E99 /* task_queue.h in Headers */,
> 				413A237D1FE18E0700373E99 /* task_queue_posix.h in Headers */,
>@@ -14816,9 +14911,9 @@
> 				413A22E81FE18E0700373E99 /* thread_checker.h in Headers */,
> 				413A22B81FE18E0700373E99 /* thread_checker_impl.h in Headers */,
> 				5CDD84C41E43AF1300621E92 /* three_band_filter_bank.h in Headers */,
>+				41FCBB2321B1F82F00A5DF27 /* throw_delegate.h in Headers */,
> 				5CDD8A581E43BFB300621E92 /* tick_timer.h in Headers */,
> 				419241DD21275A3000634FCF /* time_delta.h in Headers */,
>-				419C83341FE245EA0040C30F /* time_interval.h in Headers */,
> 				5CDD8A5B1E43BFB300621E92 /* time_stretch.h in Headers */,
> 				5CDD88DB1E43BE3D00621E92 /* time_util.h in Headers */,
> 				419241E021275A3000634FCF /* timestamp.h in Headers */,
>@@ -14856,20 +14951,17 @@
> 				413A234B1FE18E0700373E99 /* type_traits.h in Headers */,
> 				5CDD84C61E43AF1300621E92 /* typing_detection.h in Headers */,
> 				5C63FA311E4174F6002CA531 /* udpport.h in Headers */,
>-				419241572127376F00634FCF /* udptransport.h in Headers */,
> 				413A246B1FE1991A00373E99 /* UIDevice+RTCDevice.h in Headers */,
> 				5CDD88E01E43BE3D00621E92 /* ulpfec_generator.h in Headers */,
> 				5CDD88E31E43BE3D00621E92 /* ulpfec_header_reader_writer.h in Headers */,
> 				419C83C61FE247B40040C30F /* ulpfec_receiver.h in Headers */,
> 				5CDD88E51E43BE3D00621E92 /* ulpfec_receiver_impl.h in Headers */,
> 				5C63F9751E41737B002CA531 /* umametrics.h in Headers */,
>-				413A22A41FE18E0600373E99 /* unixfilesystem.h in Headers */,
> 				5CDD8BF21E43C2B500621E92 /* unpack_bits.h in Headers */,
> 				41F263B0212680A800274F59 /* unused.h in Headers */,
> 				413A21401FE0F0EF00373E99 /* ut_sim.h in Headers */,
> 				5CDD87091E43BA7500621E92 /* utility.h in Headers */,
> 				5CDD87DF1E43BC2E00621E92 /* utility.h in Headers */,
>-				417953BA216982420028266B /* RTCCodecSpecificInfoH264+Private.h in Headers */,
> 				5CDD873B1E43BABE00621E92 /* vad_audio_proc.h in Headers */,
> 				5CDD87381E43BABE00621E92 /* vad_audio_proc_internal.h in Headers */,
> 				5CDD873E1E43BABE00621E92 /* vad_circular_buffer.h in Headers */,
>@@ -14888,19 +14980,21 @@
> 				5CDD84021E439A6F00621E92 /* video_coding_impl.h in Headers */,
> 				419C84131FE249AB0040C30F /* video_content_type.h in Headers */,
> 				4145E48D1EF88B9D00FCF6E6 /* video_decoder.h in Headers */,
>+				416225D22169818100A91C9B /* video_decoder_factory.h in Headers */,
> 				4145F61E1FE1F38D00EB9CAF /* video_decoder_factory.h in Headers */,
> 				41F2636621267ADF00274F59 /* video_decoder_software_fallback_wrapper.h in Headers */,
> 				4145E48E1EF88B9D00FCF6E6 /* video_encoder.h in Headers */,
> 				41F2636821267ADF00274F59 /* video_encoder_config.h in Headers */,
>+				416225D52169818100A91C9B /* video_encoder_factory.h in Headers */,
> 				4145F61F1FE1F38D00EB9CAF /* video_encoder_factory.h in Headers */,
> 				41F2636A21267ADF00274F59 /* video_encoder_software_fallback_wrapper.h in Headers */,
>+				416225D62169818100A91C9B /* video_frame.h in Headers */,
> 				5CD2846B1E6A57F40094FDC8 /* video_frame.h in Headers */,
> 				41109AAE1E5FA19200C0955A /* video_frame_buffer.h in Headers */,
> 				5CD284691E6A57F40094FDC8 /* video_frame_buffer.h in Headers */,
>+				416225D32169818100A91C9B /* video_frame_buffer.h in Headers */,
> 				4102F6CF212733B7006AE8D7 /* video_quality_observer.h in Headers */,
> 				419C82C41FE20DCD0040C30F /* video_receive_stream.h in Headers */,
>-				417953C52169824B0028266B /* RTCVideoDecoderFactoryH264.h in Headers */,
>-				413E67992169883900EF37ED /* RTCVideoCodecInfo.h in Headers */,
> 				5CDD85CB1E43B5C000621E92 /* video_receive_stream.h in Headers */,
> 				5C4B4C211E431F75002651C8 /* video_render_frames.h in Headers */,
> 				5CD2846C1E6A57F40094FDC8 /* video_rotation.h in Headers */,
>@@ -14932,13 +15026,14 @@
> 				5CD285191E6A60570094FDC8 /* videotracksource.h in Headers */,
> 				413A22BC1FE18E0700373E99 /* virtualsocketserver.h in Headers */,
> 				5CDD87411E43BABE00621E92 /* voice_activity_detector.h in Headers */,
>-				417953E7216983CA0028266B /* module_common_types.h in Headers */,
> 				5CDD84C81E43AF1300621E92 /* voice_detection_impl.h in Headers */,
> 				5CDD87421E43BABE00621E92 /* voice_gmm_tables.h in Headers */,
> 				5CFD53811E4BA4F500482908 /* voice_processing_audio_unit.h in Headers */,
> 				4145F6171FE1EFCA00EB9CAF /* vp8.h in Headers */,
> 				419241F421275C3200634FCF /* vp8_encoder_simulcast_proxy.h in Headers */,
> 				5CDD83861E439A3500621E92 /* vp8_header_parser.h in Headers */,
>+				41FCBB6521B1FEF600A5DF27 /* vp8_temporal_layers.h in Headers */,
>+				417953EE216984210028266B /* vp9_profile.h in Headers */,
> 				5CDD8BF41E43C2B500621E92 /* vq3.h in Headers */,
> 				5CDD8BF61E43C2B500621E92 /* vq4.h in Headers */,
> 				5C4B4C811E431F9C002651C8 /* wav_file.h in Headers */,
>@@ -14955,7 +15050,6 @@
> 				5C4B48931E42C1BA002651C8 /* webrtcvideocapturerfactory.h in Headers */,
> 				5C4B48941E42C1BA002651C8 /* webrtcvideodecoderfactory.h in Headers */,
> 				5C4B48961E42C1BA002651C8 /* webrtcvideoencoderfactory.h in Headers */,
>-				416225D32169818100A91C9B /* video_frame_buffer.h in Headers */,
> 				4145E4D21EF8CC2000FCF6E6 /* webrtcvideoengine.h in Headers */,
> 				5C4B489E1E42C1BA002651C8 /* webrtcvoiceengine.h in Headers */,
> 				5CDD8BF81E43C2B500621E92 /* window32_w32.h in Headers */,
>@@ -15611,7 +15705,6 @@
> 			isa = PBXSourcesBuildPhase;
> 			buildActionMask = 2147483647;
> 			files = (
>-				5C088C101E4AA44400403995 /* bundlefilter.cc in Sources */,
> 				5C088C121E4AA44400403995 /* channel.cc in Sources */,
> 				5C088C141E4AA44400403995 /* channelmanager.cc in Sources */,
> 				5C088C181E4AA44400403995 /* externalhmac.cc in Sources */,
>@@ -16107,15 +16200,12 @@
> 				5CDD8B6A1E43C2B500621E92 /* abs_quant_loop.c in Sources */,
> 				5CDD89F81E43BFB300621E92 /* accelerate.cc in Sources */,
> 				41E02CA82127352D00C27CD6 /* acknowledged_bitrate_estimator.cc in Sources */,
>-				416225D82169818100A91C9B /* video_decoder_factory.mm in Sources */,
> 				5CDD85031E43B1EA00621E92 /* acm_codec_database.cc in Sources */,
> 				5CDD85091E43B1EA00621E92 /* acm_receiver.cc in Sources */,
>-				417953D32169834F0028266B /* channel_send.cc in Sources */,
> 				5CDD850B1E43B1EA00621E92 /* acm_resampler.cc in Sources */,
> 				5C4B48D71E42C1E3002651C8 /* adaptedvideotracksource.cc in Sources */,
> 				416D3BDB212731C200775F09 /* adaptive_agc.cc in Sources */,
> 				415F1FA621272FBA00064CBF /* adaptive_digital_gain_applier.cc in Sources */,
>-				413E67942169881900EF37ED /* svc_rate_allocator.cc in Sources */,
> 				5CD285A41E6A63430094FDC8 /* adaptive_fir_filter.cc in Sources */,
> 				416D3BDE212731C200775F09 /* adaptive_mode_level_estimator.cc in Sources */,
> 				415F1FA821272FBA00064CBF /* adaptive_mode_level_estimator_agc.cc in Sources */,
>@@ -16130,6 +16220,7 @@
> 				5CDD85E71E43B81000621E92 /* aecm_core.cc in Sources */,
> 				5CDD85E41E43B81000621E92 /* aecm_core_c.cc in Sources */,
> 				5CDD87011E43BA7500621E92 /* agc.cc in Sources */,
>+				413E679B2169886200EF37ED /* agc2_common.cc in Sources */,
> 				5CDD86FF1E43BA7500621E92 /* agc_manager_direct.cc in Sources */,
> 				5CDD8FD11E43CD6600621E92 /* aimd_rate_control.cc in Sources */,
> 				41F2639E2126800000274F59 /* aligned_malloc.cc in Sources */,
>@@ -16145,6 +16236,7 @@
> 				41433D291F79B33400387B4D /* arith_routines_hist.c in Sources */,
> 				5CDD87A81E43BC0500621E92 /* arith_routines_logist.c in Sources */,
> 				41433D0F1F79B33400387B4D /* arith_routines_logist.c in Sources */,
>+				41FCBB7A21B1FFC000A5DF27 /* ascii.cc in Sources */,
> 				413A238C1FE18E0700373E99 /* asyncinvoker.cc in Sources */,
> 				413A216D1FE18D6A00373E99 /* asyncpacketsocket.cc in Sources */,
> 				413A23551FE18E0700373E99 /* asyncresolverinterface.cc in Sources */,
>@@ -16172,7 +16264,6 @@
> 				4140B8221E4E3383007409E6 /* audio_decoder_pcm.cc in Sources */,
> 				5CDD8FFC1E43CE3A00621E92 /* audio_decoder_pcm16b.cc in Sources */,
> 				5C119FFB1E457400004F0987 /* audio_device_buffer.cc in Sources */,
>-				41889D77216BC4EC004614DD /* RTCRtpEncodingParameters.mm in Sources */,
> 				5C11A01D1E457578004F0987 /* audio_device_dummy.cc in Sources */,
> 				5C119FFE1E457400004F0987 /* audio_device_generic.cc in Sources */,
> 				5C11A00B1E457448004F0987 /* audio_device_mac.cc in Sources */,
>@@ -16222,7 +16313,6 @@
> 				5C4B4C5F1E431F9C002651C8 /* audio_util.cc in Sources */,
> 				5CDD8A041E43BFB300621E92 /* audio_vector.cc in Sources */,
> 				5CD284F01E6A60570094FDC8 /* audiotrack.cc in Sources */,
>-				417953D82169834F0028266B /* channel_receive.cc in Sources */,
> 				5CDD8B721E43C2B500621E92 /* augmented_cb_corr.c in Sources */,
> 				5CDD86401E43B8B500621E92 /* auto_corr_to_refl_coef.c in Sources */,
> 				5CDD86411E43B8B500621E92 /* auto_correlation.c in Sources */,
>@@ -16244,6 +16334,7 @@
> 				5C4B4C191E431F75002651C8 /* bitrate_adjuster.cc in Sources */,
> 				5CDD852F1E43B39C00621E92 /* bitrate_allocator.cc in Sources */,
> 				5CDD8F7D1E43CBE000621E92 /* bitrate_controller.cc in Sources */,
>+				41FCBB5C21B1FEA300A5DF27 /* bitrate_controller.cc in Sources */,
> 				5CDD905D1E43D11200621E92 /* bitrate_controller_impl.cc in Sources */,
> 				41E02CAE2127352D00C27CD6 /* bitrate_estimator.cc in Sources */,
> 				5CDD8FA01E43CCBE00621E92 /* bitrate_prober.cc in Sources */,
>@@ -16257,6 +16348,7 @@
> 				413A239B1FE18E0800373E99 /* bufferqueue.cc in Sources */,
> 				5CD285481E6A61D20094FDC8 /* builtin_audio_decoder_factory.cc in Sources */,
> 				413091FB1EF8CFF800757C55 /* builtin_audio_encoder_factory.cc in Sources */,
>+				41FCBB1B21B1F7CD00A5DF27 /* builtin_video_bitrate_allocator_factory.cc in Sources */,
> 				41F2635F21267ADF00274F59 /* builtin_video_decoder_factory.cc in Sources */,
> 				41E02CC42127358800C27CD6 /* builtin_video_encoder_factory.cc in Sources */,
> 				5CDD8B741E43C2B500621E92 /* bw_expand.c in Sources */,
>@@ -16279,12 +16371,14 @@
> 				5CDD8B821E43C2B500621E92 /* cb_update_best_index.c in Sources */,
> 				5C4B4C641E431F9C002651C8 /* channel_buffer.cc in Sources */,
> 				5CDD8F801E43CBE000621E92 /* channel_controller.cc in Sources */,
>+				417953D82169834F0028266B /* channel_receive.cc in Sources */,
>+				417953D32169834F0028266B /* channel_send.cc in Sources */,
> 				5CDD8B841E43C2B500621E92 /* chebyshev.c in Sources */,
> 				413A23731FE18E0700373E99 /* checks.cc in Sources */,
> 				5CDD86A11E43B99400621E92 /* circular_buffer.cc in Sources */,
> 				5C4B4CC61E4320A9002651C8 /* clock.cc in Sources */,
>+				41FCBB7521B1FFA400A5DF27 /* cocoa_threading.mm in Sources */,
> 				5C4B48DB1E42C1E3002651C8 /* codec.cc in Sources */,
>-				5CDD85151E43B1EA00621E92 /* codec_manager.cc in Sources */,
> 				5CDD83C81E439A6F00621E92 /* codec_timer.cc in Sources */,
> 				4102F6DC21273416006AE8D7 /* color_space.cc in Sources */,
> 				5CDD8A0D1E43BFB300621E92 /* comfort_noise.cc in Sources */,
>@@ -16299,13 +16393,12 @@
> 				415F1FAA21272FBA00064CBF /* compute_interpolated_gain_curve.cc in Sources */,
> 				5CDD8FEB1E43CDCA00621E92 /* config.cc in Sources */,
> 				419241712127389B00634FCF /* congestion_controller_experiment.cc in Sources */,
>-				41A08BD821272F1D001D5D7B /* congestion_window_pushback_controller.cc in Sources */,
>+				41FCBB4121B1FDB800A5DF27 /* congestion_window_pushback_controller.cc in Sources */,
> 				5CDD8A931E43C00F00621E92 /* constant_pcm_packet_source.cc in Sources */,
> 				5CDD8B891E43C2B500621E92 /* constants.c in Sources */,
> 				419C83FB1FE249370040C30F /* constants.cc in Sources */,
> 				41E02C782127347400C27CD6 /* contributing_sources.cc in Sources */,
>-				413E67A22169889500EF37ED /* RTCRtpFragmentationHeader.m in Sources */,
>-				416225E72169819500A91C9B /* objc_video_decoder_factory.mm in Sources */,
>+				41FCBB3921B1F8FC00A5DF27 /* control_handler.cc in Sources */,
> 				5CDD8F851E43CBE000621E92 /* controller.cc in Sources */,
> 				5CDD8F831E43CBE000621E92 /* controller_manager.cc in Sources */,
> 				41F9BFAC2051C91900ABF0B9 /* convert_legacy_video_factory.cc in Sources */,
>@@ -16313,13 +16406,16 @@
> 				413A23151FE18E0700373E99 /* copyonwritebuffer.cc in Sources */,
> 				5C4B4CCA1E4320A9002651C8 /* cpu_features.cc in Sources */,
> 				5C4B4CCB1E4320A9002651C8 /* cpu_info.cc in Sources */,
>+				41FCBB5321B1FE4600A5DF27 /* cpu_speed_experiment.cc in Sources */,
> 				5CDD87B11E43BC0500621E92 /* crc.c in Sources */,
> 				413A23431FE18E0700373E99 /* crc32.cc in Sources */,
> 				5CDD8B8B1E43C2B500621E92 /* create_augmented_vec.c in Sources */,
>-				4145E4CA1EF8CB3200FCF6E6 /* createpeerconnectionfactory.cc in Sources */,
>+				41FCBB0E21B1F53700A5DF27 /* create_peerconnection_factory.cc in Sources */,
>+				41FCBB6621B1FEF600A5DF27 /* create_vp8_temporal_layers.cc in Sources */,
> 				413A23881FE18E0700373E99 /* criticalsection.cc in Sources */,
> 				5CDD864B1E43B8B500621E92 /* cross_correlation.c in Sources */,
> 				5CDD8A0F1E43BFB300621E92 /* cross_correlation.cc in Sources */,
>+				41FCBB3121B1F8B700A5DF27 /* cryptooptions.cc in Sources */,
> 				413A23221FE18E0700373E99 /* cryptstring.cc in Sources */,
> 				419C848E1FE2574D0040C30F /* custom_extensions.cc in Sources */,
> 				419C848C1FE2574D0040C30F /* d1_both.cc in Sources */,
>@@ -16353,7 +16449,6 @@
> 				41E02CA52127352D00C27CD6 /* delay_based_bwe.cc in Sources */,
> 				5CDD868D1E43B93900621E92 /* delay_estimator.cc in Sources */,
> 				5CDD868B1E43B93900621E92 /* delay_estimator_wrapper.cc in Sources */,
>-				417953FA216984BE0028266B /* RTCEncodedImage.m in Sources */,
> 				5CDD8A1D1E43BFB300621E92 /* delay_manager.cc in Sources */,
> 				5CDD8A201E43BFB300621E92 /* delay_peak_detector.cc in Sources */,
> 				5CDD86EE1E43BA6D00621E92 /* digital_agc.c in Sources */,
>@@ -16388,10 +16483,10 @@
> 				5CDD87B71E43BC0500621E92 /* encode.c in Sources */,
> 				41433D121F79B33400387B4D /* encode.c in Sources */,
> 				5CDD87B51E43BC0500621E92 /* encode_lpc_swb.c in Sources */,
>-				417953DC216983910028266B /* field_trial.cc in Sources */,
> 				5CDD8A951E43C00F00621E92 /* encode_neteq_input.cc in Sources */,
> 				5CDD83CC1E439A6F00621E92 /* encoded_frame.cc in Sources */,
> 				41A08BAB212681C8001D5D7B /* encoded_frame.cc in Sources */,
>+				41FCBB1D21B1F7CD00A5DF27 /* encoded_image.cc in Sources */,
> 				41E02C88212734B900C27CD6 /* encoder_database.cc in Sources */,
> 				5CDD859F1E43B5C000621E92 /* encoder_rtcp_feedback.cc in Sources */,
> 				5CDD86511E43B8B500621E92 /* energy.c in Sources */,
>@@ -16414,7 +16509,6 @@
> 				5CDD894A1E43BF3A00621E92 /* extended_jitter_report.cc in Sources */,
> 				5CDD894D1E43BF3A00621E92 /* extended_reports.cc in Sources */,
> 				5CDD8A971E43C00F00621E92 /* fake_decode_from_file.cc in Sources */,
>-				413E67822169870F00EF37ED /* svc_config.cc in Sources */,
> 				41299B962127369100B3414B /* fake_network_pipe.cc in Sources */,
> 				413A23101FE18E0700373E99 /* fakeclock.cc in Sources */,
> 				413A23CC1FE18E0800373E99 /* fakesslidentity.cc in Sources */,
>@@ -16425,11 +16519,11 @@
> 				419241D3212759A100634FCF /* fec_private_tables_random.cc in Sources */,
> 				5CDD887B1E43BE3C00621E92 /* fec_test_helper.cc in Sources */,
> 				41F2638A21267F4000274F59 /* fft.c in Sources */,
>-				417953EA216983F10028266B /* rtt_mult_experiment.cc in Sources */,
> 				41433CEA1F79B33400387B4D /* fft.c in Sources */,
> 				41F2636E21267B3E00274F59 /* fft4g.c in Sources */,
> 				41F9BFD72051DDE500ABF0B9 /* fft_buffer.cc in Sources */,
> 				4192413D2127372400634FCF /* fft_util.cc in Sources */,
>+				417953DC216983910028266B /* field_trial.cc in Sources */,
> 				4192416E2127389B00634FCF /* field_trial_parser.cc in Sources */,
> 				419241E821275A7600634FCF /* field_trial_units.cc in Sources */,
> 				413A22E61FE18E0700373E99 /* file.cc in Sources */,
>@@ -16437,14 +16531,11 @@
> 				5C11A01F1E457578004F0987 /* file_audio_device_factory.cc in Sources */,
> 				413A23D81FE18E0800373E99 /* file_posix.cc in Sources */,
> 				5CDD8C8B1E43C66000621E92 /* file_utils.cc in Sources */,
>-				416225D72169818100A91C9B /* video_frame_buffer.mm in Sources */,
> 				41F263AC212680A800274F59 /* file_wrapper.cc in Sources */,
> 				413A23DA1FE18E0800373E99 /* filerotatingstream.cc in Sources */,
>-				413A23B51FE18E0800373E99 /* fileutils.cc in Sources */,
> 				5CDD86551E43B8B500621E92 /* filter_ar.c in Sources */,
> 				5CDD86541E43B8B500621E92 /* filter_ar_fast_q12.c in Sources */,
> 				5CDD87BC1E43BC0500621E92 /* filter_functions.c in Sources */,
>-				4144B3E82169C5E3004363AC /* RTCVideoCodecInfo+Private.mm in Sources */,
> 				5CDD86561E43B8B500621E92 /* filter_ma_fast_q12.c in Sources */,
> 				41433D271F79B33400387B4D /* filterbank_tables.c in Sources */,
> 				41433D061F79B33400387B4D /* filterbanks.c in Sources */,
>@@ -16457,7 +16548,6 @@
> 				419C831D1FE242B20040C30F /* fir_filter_factory.cc in Sources */,
> 				5C4B4C6A1E431F9C002651C8 /* fir_filter_sse.cc in Sources */,
> 				415F1FAD21272FBA00064CBF /* fixed_digital_level_estimator.cc in Sources */,
>-				416D3BE3212731C200775F09 /* fixed_gain_controller.cc in Sources */,
> 				413A217D1FE18D7800373E99 /* flags.cc in Sources */,
> 				5CDD887E1E43BE3C00621E92 /* flexfec_header_reader_writer.cc in Sources */,
> 				4102F6BC21273382006AE8D7 /* flexfec_receive_stream.cc in Sources */,
>@@ -16471,8 +16561,8 @@
> 				5CDD83D11E439A6F00621E92 /* frame_buffer2.cc in Sources */,
> 				5CDD8BA21E43C2B500621E92 /* frame_classify.c in Sources */,
> 				5CD2849A1E6A5F410094FDC8 /* frame_combiner.cc in Sources */,
>-				417953C72169824B0028266B /* RTCVideoDecoderH264.mm in Sources */,
> 				5CDD83791E439A3500621E92 /* frame_dropper.cc in Sources */,
>+				41FCBB5921B1FE6E00A5DF27 /* frame_dumping_decoder.cc in Sources */,
> 				5CDD8F901E43CBE000621E92 /* frame_length_controller.cc in Sources */,
> 				5CDD83D31E439A6F00621E92 /* frame_object.cc in Sources */,
> 				41F2638D21267F4900274F59 /* g711.c in Sources */,
>@@ -16484,7 +16574,6 @@
> 				5CDD84B21E43AF1300621E92 /* gain_control_for_experimental_agc.cc in Sources */,
> 				5CDD84B41E43AF1300621E92 /* gain_control_impl.cc in Sources */,
> 				41A08BD021272EE2001D5D7B /* gain_controller2.cc in Sources */,
>-				415F1FB121272FBA00064CBF /* gain_curve_applier.cc in Sources */,
> 				5CDD8BA41E43C2B500621E92 /* gain_dequant.c in Sources */,
> 				5CDD8BA61E43C2B500621E92 /* gain_quant.c in Sources */,
> 				5CDD83D51E439A6F00621E92 /* generic_decoder.cc in Sources */,
>@@ -16495,19 +16584,20 @@
> 				5CDD86581E43B8B500621E92 /* get_scaling_square.c in Sources */,
> 				5CDD8BAC1E43C2B500621E92 /* get_sync_seq.c in Sources */,
> 				5CDD87291E43BABE00621E92 /* gmm.cc in Sources */,
>-				41E02CA92127352D00C27CD6 /* goog_cc_factory.cc in Sources */,
>+				41FCBB6021B1FEC400A5DF27 /* goog_cc_factory.cc in Sources */,
> 				41E02CB62127352D00C27CD6 /* goog_cc_network_control.cc in Sources */,
> 				5CDD85581E43B42B00621E92 /* h264.cc in Sources */,
> 				5CDD83541E43257200621E92 /* h264_bitstream_parser.cc in Sources */,
> 				5CDD83561E43257200621E92 /* h264_common.cc in Sources */,
> 				4145E4BF1EF894F600FCF6E6 /* h264_profile_level_id.cc in Sources */,
>-				41889D79216BC4EC004614DD /* RTCRtpFragmentationHeader+Private.mm in Sources */,
> 				5CD285EA1E6A639F0094FDC8 /* h264_sprop_parameter_sets.cc in Sources */,
> 				5CDD83D91E439A6F00621E92 /* h264_sps_pps_tracker.cc in Sources */,
> 				419C84691FE255FA0040C30F /* handshake.cc in Sources */,
> 				419C84891FE2574D0040C30F /* handshake_client.cc in Sources */,
> 				419C848B1FE2574D0040C30F /* handshake_server.cc in Sources */,
>+				41FCBB7121B1FF7400A5DF27 /* hdr_metadata.cc in Sources */,
> 				413A236E1FE18E0700373E99 /* helpers.cc in Sources */,
>+				417953BC2169824B0028266B /* helpers.cc in Sources */,
> 				5CDD83DB1E439A6F00621E92 /* histogram.cc in Sources */,
> 				419C83511FE246650040C30F /* histogram_percentile_counter.cc in Sources */,
> 				5CDD8BAE1E43C2B500621E92 /* hp_input.c in Sources */,
>@@ -16517,6 +16607,7 @@
> 				5CDD90501E43D0E900621E92 /* i420.cc in Sources */,
> 				5CD284671E6A57F40094FDC8 /* i420_buffer.cc in Sources */,
> 				5C4B4C1B1E431F75002651C8 /* i420_buffer_pool.cc in Sources */,
>+				41FCBB4721B1FDEF00A5DF27 /* icecredentialsiterator.cc in Sources */,
> 				41A08BC421269577001D5D7B /* icelogger.cc in Sources */,
> 				413091F81EF8CFBD00757C55 /* iceserverparsing.cc in Sources */,
> 				419C83E01FE247F20040C30F /* icetransportinternal.cc in Sources */,
>@@ -16539,7 +16630,6 @@
> 				415F1FB221272FBA00064CBF /* interpolated_gain_curve.cc in Sources */,
> 				419C829A1FE20CA10040C30F /* interval_budget.cc in Sources */,
> 				5CDD87C01E43BC0500621E92 /* intialize.c in Sources */,
>-				417953CC216982DB0028266B /* RTCCodecSpecificInfoH264.mm in Sources */,
> 				413A23ED1FE18E0800373E99 /* ipaddress.cc in Sources */,
> 				5CDD87C31E43BC0500621E92 /* isac.c in Sources */,
> 				41299B912127367B00B3414B /* isac_vad.c in Sources */,
>@@ -16547,29 +16637,32 @@
> 				5CDD837B1E439A3500621E92 /* ivf_file_writer.cc in Sources */,
> 				5CDD83E11E439A6F00621E92 /* jitter_buffer.cc in Sources */,
> 				5CDD83E31E439A6F00621E92 /* jitter_estimator.cc in Sources */,
>+				41FCBB5221B1FE4600A5DF27 /* jitter_upper_bound_experiment.cc in Sources */,
> 				41F9BF862051C80100ABF0B9 /* jsep.cc in Sources */,
> 				5CD284741E6A5D280094FDC8 /* jsepicecandidate.cc in Sources */,
> 				41F2638421267B7700274F59 /* jsepicecandidate.cc in Sources */,
> 				5CD284721E6A5D080094FDC8 /* jsepsessiondescription.cc in Sources */,
> 				41F9BFE32051DE1800ABF0B9 /* jseptransport.cc in Sources */,
> 				4102F69021273206006AE8D7 /* jseptransportcontroller.cc in Sources */,
>+				41FCBB6D21B1FF3A00A5DF27 /* key_derivation.cc in Sources */,
> 				4192419A2127586500634FCF /* kiss_fft.cc in Sources */,
> 				5CDD87C41E43BC0500621E92 /* lattice.c in Sources */,
>-				413E67682169854B00EF37ED /* RTCVideoDecoderVP8.mm in Sources */,
> 				41433CF21F79B33400387B4D /* lattice.c in Sources */,
>-				416225D42169818100A91C9B /* video_encoder_factory.mm in Sources */,
> 				41433D141F79B33400387B4D /* lattice_c.c in Sources */,
> 				5CDD841F1E439B2900621E92 /* legacy_encoded_audio_frame.cc in Sources */,
> 				5CDD84B81E43AF1300621E92 /* level_estimator_impl.cc in Sources */,
> 				5CDD865A1E43B8B500621E92 /* levinson_durbin.c in Sources */,
>+				413E67752169863B00EF37ED /* libvpx_interface.cc in Sources */,
> 				41294094212E128D00AD95E7 /* libvpx_vp8_decoder.cc in Sources */,
> 				41294092212E128D00AD95E7 /* libvpx_vp8_encoder.cc in Sources */,
> 				416D3BE6212731C200775F09 /* limiter.cc in Sources */,
> 				5CD284F61E6A60570094FDC8 /* localaudiosource.cc in Sources */,
> 				413A23EB1FE18E0800373E99 /* location.cc in Sources */,
> 				5CDD8C0F1E43C34600621E92 /* locked_bandwidth_info.cc in Sources */,
>+				4144B3D821698D3D004363AC /* logging.cc in Sources */,
> 				413A22C01FE18E0700373E99 /* logging_mac.mm in Sources */,
> 				413A23BE1FE18E0800373E99 /* logsinks.cc in Sources */,
>+				41FCBB6A21B1FF1B00A5DF27 /* loss_based_bandwidth_estimation.cc in Sources */,
> 				419241C52127593F00634FCF /* loss_rate_filter.cc in Sources */,
> 				5CDD87051E43BA7500621E92 /* loudness_histogram.cc in Sources */,
> 				5CD2855D1E6A62ED0094FDC8 /* low_cut_filter.cc in Sources */,
>@@ -16578,7 +16671,6 @@
> 				5CDD8BC11E43C2B500621E92 /* lpc_encode.c in Sources */,
> 				5CDD87C71E43BC0500621E92 /* lpc_gain_swb_tables.c in Sources */,
> 				41433CEE1F79B33400387B4D /* lpc_masking_model.c in Sources */,
>-				4144B3F02169C63F004363AC /* NSString+StdString.mm in Sources */,
> 				5CDD87C91E43BC0500621E92 /* lpc_shape_swb12_tables.c in Sources */,
> 				5CDD87CB1E43BC0500621E92 /* lpc_shape_swb16_tables.c in Sources */,
> 				5CDD87CD1E43BC0500621E92 /* lpc_tables.c in Sources */,
>@@ -16593,13 +16685,15 @@
> 				413A232C1FE18E0700373E99 /* macifaddrs_converter.cc in Sources */,
> 				413A23381FE18E0700373E99 /* macutils.cc in Sources */,
> 				5CD285CB1E6A63430094FDC8 /* main_filter_update_gain.cc in Sources */,
>+				41FCBB3C21B1FD4000A5DF27 /* match.cc in Sources */,
> 				5CD285CF1E6A63430094FDC8 /* matched_filter.cc in Sources */,
> 				5CD285CD1E6A63430094FDC8 /* matched_filter_lag_aggregator.cc in Sources */,
> 				41F9BFD42051DDE500ABF0B9 /* matrix_buffer.cc in Sources */,
>+				41FCBB4821B1FDEF00A5DF27 /* mdns_message.cc in Sources */,
> 				5CDD86A41E43B99400621E92 /* mean_variance_estimator.cc in Sources */,
> 				5CDD83E51E439A6F00621E92 /* media_opt_util.cc in Sources */,
> 				5CDD83E71E439A6F00621E92 /* media_optimization.cc in Sources */,
>-				416225E62169819500A91C9B /* objc_video_encoder_factory.mm in Sources */,
>+				41FCBB1121B1F53700A5DF27 /* media_transport_interface.cc in Sources */,
> 				4102F6B321273331006AE8D7 /* mediachannel.cc in Sources */,
> 				5C4B48E61E42C1E3002651C8 /* mediaconstants.cc in Sources */,
> 				5C63F9481E41737B002CA531 /* mediaconstraintsinterface.cc in Sources */,
>@@ -16607,24 +16701,25 @@
> 				41E02CB12127352D00C27CD6 /* median_slope_estimator.cc in Sources */,
> 				5CD284FA1E6A60570094FDC8 /* mediastream.cc in Sources */,
> 				5CD286401E6A683A0094FDC8 /* mediastreaminterface.cc in Sources */,
>-				4144B3D32169896D004363AC /* RTCNativeI420Buffer.mm in Sources */,
> 				5CD284FC1E6A60570094FDC8 /* mediastreamobserver.cc in Sources */,
> 				5CD286321E6A67BF0094FDC8 /* mediatypes.cc in Sources */,
> 				413A23801FE18E0700373E99 /* memory_usage.cc in Sources */,
>+				41FCBB8421B2007B00A5DF27 /* memutil.cc in Sources */,
> 				5CDD8A2F1E43BFB300621E92 /* merge.cc in Sources */,
> 				413A22F91FE18E0700373E99 /* messagedigest.cc in Sources */,
> 				413A236B1FE18E0700373E99 /* messagehandler.cc in Sources */,
> 				413A22B11FE18E0700373E99 /* messagequeue.cc in Sources */,
>+				417953DB216983910028266B /* metrics.cc in Sources */,
> 				5CDD865E1E43B8B500621E92 /* min_max_operations.c in Sources */,
>-				5CDD837D1E439A3500621E92 /* moving_average.cc in Sources */,
>-				413E67A32169889500EF37ED /* RTCVideoEncoderQpThresholds.m in Sources */,
>+				417953E6216983CA0028266B /* module_common_types.cc in Sources */,
>+				41FCBB1521B1F7AA00A5DF27 /* moving_average.cc in Sources */,
> 				5CD286101E6A64C90094FDC8 /* moving_max.cc in Sources */,
> 				5CDD8C8E1E43C66000621E92 /* moving_moments.cc in Sources */,
> 				5CDD8BCF1E43C2B500621E92 /* my_corr.c in Sources */,
> 				5CDD89531E43BF3A00621E92 /* nack.cc in Sources */,
> 				5CDD83EA1E439A6F00621E92 /* nack_module.cc in Sources */,
> 				5CDD8A321E43BFB300621E92 /* nack_tracker.cc in Sources */,
>-				417953B9216982420028266B /* RTCVideoEncoderFactoryH264.m in Sources */,
>+				417953B4216982420028266B /* nalu_rewriter.cc in Sources */,
> 				413A23031FE18E0700373E99 /* nattypes.cc in Sources */,
> 				5CDD8BD11E43C2B500621E92 /* nearest_neighbor.c in Sources */,
> 				5CDD8A3B1E43BFB300621E92 /* neteq.cc in Sources */,
>@@ -16637,46 +16732,44 @@
> 				41E02CCD212735B700C27CD6 /* network_types.cc in Sources */,
> 				413A238B1FE18E0700373E99 /* networkmonitor.cc in Sources */,
> 				416D3BE7212731C200775F09 /* noise_level_estimator.cc in Sources */,
>-				413E6790216987DB00EF37ED /* RTCVideoFrame.mm in Sources */,
> 				416D3BE8212731C200775F09 /* noise_spectrum_estimator.cc in Sources */,
> 				5CDD85FE1E43B84E00621E92 /* noise_suppression.c in Sources */,
> 				5CDD84E91E43B0B600621E92 /* noise_suppression_impl.cc in Sources */,
> 				5CDD85FC1E43B84E00621E92 /* noise_suppression_x.c in Sources */,
>-				417953B8216982420028266B /* RTCVideoDecoderFactoryH264.m in Sources */,
>-				413A235F1FE18E0700373E99 /* noop.cc in Sources */,
>-				413A23EE1FE18E0800373E99 /* noop.mm in Sources */,
> 				5CDD8A3D1E43BFB300621E92 /* normal.cc in Sources */,
>+				41FCBB5621B1FE4600A5DF27 /* normalize_simulcast_size_experiment.cc in Sources */,
> 				5CDD86A71E43B99400621E92 /* normalized_covariance_estimator.cc in Sources */,
> 				5CDD86001E43B84E00621E92 /* ns_core.c in Sources */,
>+				4144B3F02169C63F004363AC /* NSString+StdString.mm in Sources */,
> 				5CDD86051E43B84E00621E92 /* nsx_core.c in Sources */,
> 				5CDD86021E43B84E00621E92 /* nsx_core_c.c in Sources */,
> 				41F411AD1EF8D91F00343C26 /* null_aec_dump_factory.cc in Sources */,
> 				419C83B31FE2472E0040C30F /* null_audio_poller.cc in Sources */,
> 				413A23D21FE18E0800373E99 /* nullsocketserver.cc in Sources */,
>+				416225E42169819500A91C9B /* objc_frame_buffer.mm in Sources */,
>+				416225E72169819500A91C9B /* objc_video_decoder_factory.mm in Sources */,
>+				416225E62169819500A91C9B /* objc_video_encoder_factory.mm in Sources */,
>+				416225E22169819500A91C9B /* objc_video_frame.mm in Sources */,
> 				5CDD86941E43B93900621E92 /* ooura_fft.cc in Sources */,
> 				5CDD86911E43B93900621E92 /* ooura_fft_sse2.cc in Sources */,
>+				41FCBB7D21B1FFF600A5DF27 /* openssl_key_derivation_hkdf.cc in Sources */,
> 				419C84271FE24BD10040C30F /* openssladapter.cc in Sources */,
> 				41DDB264212679A300296D47 /* opensslcertificate.cc in Sources */,
> 				419C84261FE24BCF0040C30F /* openssldigest.cc in Sources */,
> 				419C84251FE24BC80040C30F /* opensslidentity.cc in Sources */,
> 				41DDB262212679A300296D47 /* opensslsessioncache.cc in Sources */,
> 				419C843F1FE250B90040C30F /* opensslstreamadapter.cc in Sources */,
>-				413E679B2169886200EF37ED /* agc2_common.cc in Sources */,
> 				41DDB265212679A300296D47 /* opensslutility.cc in Sources */,
> 				41DDB25221265BE900296D47 /* optional.cc in Sources */,
>-				413A234A1FE18E0700373E99 /* optionsfile.cc in Sources */,
> 				5CDD8C671E43C60900621E92 /* opus_interface.c in Sources */,
> 				5CDD8FD71E43CD6600621E92 /* overuse_detector.cc in Sources */,
>-				417953D52169834F0028266B /* channel_send_proxy.cc in Sources */,
> 				5CDD8FD91E43CD6600621E92 /* overuse_estimator.cc in Sources */,
> 				5CDD85A41E43B5C000621E92 /* overuse_frame_detector.cc in Sources */,
> 				5C63F9ED1E4174F6002CA531 /* p2pconstants.cc in Sources */,
> 				5C63F9F21E4174F6002CA531 /* p2ptransportchannel.cc in Sources */,
> 				41F2639A21267F5E00274F59 /* pa_ringbuffer.c in Sources */,
> 				5CDD8FA21E43CCBE00621E92 /* paced_sender.cc in Sources */,
>-				41299B8B2127365100B3414B /* pacer_controller.cc in Sources */,
> 				5CDD8BD31E43C2B500621E92 /* pack_bits.c in Sources */,
>-				4144B3D121698967004363AC /* RTCNativeMutableI420Buffer.mm in Sources */,
> 				5CDD8A421E43BFB300621E92 /* packet.cc in Sources */,
> 				5CDD83EE1E439A6F00621E92 /* packet.cc in Sources */,
> 				5CDD8AAF1E43C00F00621E92 /* packet.cc in Sources */,
>@@ -16688,7 +16781,6 @@
> 				412455561EF887FB00F11809 /* packetlossestimator.cc in Sources */,
> 				419C83F51FE248CA0040C30F /* packetsocketfactory.cc in Sources */,
> 				419C83DA1FE247F20040C30F /* packettransportinternal.cc in Sources */,
>-				413A233C1FE18E0700373E99 /* pathutils.cc in Sources */,
> 				5C4B48831E42C1BA002651C8 /* payload_type_mapper.cc in Sources */,
> 				5CDD90001E43CE3A00621E92 /* pcm16b.c in Sources */,
> 				5CD284FF1E6A60570094FDC8 /* peerconnection.cc in Sources */,
>@@ -16707,7 +16799,6 @@
> 				5CDD87301E43BABE00621E92 /* pitch_internal.cc in Sources */,
> 				5CDD87D51E43BC0600621E92 /* pitch_lag_tables.c in Sources */,
> 				41433D191F79B33400387B4D /* pitch_lag_tables.c in Sources */,
>-				413E676D2169854B00EF37ED /* RTCWrappedNativeVideoEncoder.mm in Sources */,
> 				419241412127372400634FCF /* pitch_search.cc in Sources */,
> 				419241442127372800634FCF /* pitch_search_internal.cc in Sources */,
> 				413A235A1FE18E0700373E99 /* platform_file.cc in Sources */,
>@@ -16747,7 +16838,6 @@
> 				413A23621FE18E0700373E99 /* ratetracker.cc in Sources */,
> 				41A08BCC21269620001D5D7B /* raw_logging.cc in Sources */,
> 				5CDD86611E43B8B500621E92 /* real_fft.c in Sources */,
>-				413E67752169863B00EF37ED /* libvpx_interface.cc in Sources */,
> 				5C4B4C771E431F9C002651C8 /* real_fourier.cc in Sources */,
> 				5C4B4C721E431F9C002651C8 /* real_fourier_ooura.cc in Sources */,
> 				413091FE1EF8D06D00757C55 /* receive_side_congestion_controller.cc in Sources */,
>@@ -16783,14 +16873,12 @@
> 				5CDD86641E43B8B500621E92 /* resample_by_2_internal.c in Sources */,
> 				5CDD86681E43B8B500621E92 /* resample_fractional.c in Sources */,
> 				5CDD8AB11E43C00F00621E92 /* resample_input_audio_file.cc in Sources */,
>-				413E67982169883900EF37ED /* RTCVideoCodecInfo.m in Sources */,
> 				5CDD87581E43BAF500621E92 /* resampler.cc in Sources */,
> 				5CDD84BD1E43AF1300621E92 /* residual_echo_detector.cc in Sources */,
> 				5CD285DD1E6A63430094FDC8 /* residual_echo_estimator.cc in Sources */,
> 				5C4B4C7A1E431F9C002651C8 /* ring_buffer.c in Sources */,
> 				5CDD84BF1E43AF1300621E92 /* rms_level.cc in Sources */,
> 				419241462127372800634FCF /* rnn.cc in Sources */,
>-				417953BC2169824B0028266B /* helpers.cc in Sources */,
> 				419241992127586500634FCF /* rnn_vad_weights.cc in Sources */,
> 				5CFD538F1E4BD3A300482908 /* rotate_neon.cc in Sources */,
> 				5CFD53901E4BD3A300482908 /* rotate_neon64.cc in Sources */,
>@@ -16804,7 +16892,6 @@
> 				419C839B1FE246F90040C30F /* rtc_event_audio_receive_stream_config.cc in Sources */,
> 				419C839E1FE246F90040C30F /* rtc_event_audio_send_stream_config.cc in Sources */,
> 				419C838B1FE246F90040C30F /* rtc_event_bwe_update_delay_based.cc in Sources */,
>-				417953EF216984210028266B /* vp9_profile.cc in Sources */,
> 				419C838D1FE246F90040C30F /* rtc_event_bwe_update_loss_based.cc in Sources */,
> 				41A08BC021269553001D5D7B /* rtc_event_ice_candidate_pair.cc in Sources */,
> 				41A08BBD21269553001D5D7B /* rtc_event_ice_candidate_pair_config.cc in Sources */,
>@@ -16824,10 +16911,17 @@
> 				419C841D1FE24AEF0040C30F /* rtc_stream_config.cc in Sources */,
> 				413A230F1FE18E0700373E99 /* rtccertificate.cc in Sources */,
> 				413A22BE1FE18E0700373E99 /* rtccertificategenerator.cc in Sources */,
>+				417953CC216982DB0028266B /* RTCCodecSpecificInfoH264.mm in Sources */,
>+				413E678B216987B900EF37ED /* RTCCVPixelBuffer.mm in Sources */,
>+				417953B5216982420028266B /* RTCDefaultVideoDecoderFactory.m in Sources */,
>+				417953B7216982420028266B /* RTCDefaultVideoEncoderFactory.m in Sources */,
>+				4144B3F5216B081B004363AC /* RTCEncodedImage+Private.mm in Sources */,
>+				417953FA216984BE0028266B /* RTCEncodedImage.m in Sources */,
> 				41F411B01EF8DA0100343C26 /* rtcerror.cc in Sources */,
>-				5CDD8A4F1E43BFB300621E92 /* rtcp.cc in Sources */,
>+				4144B3D621698A1A004363AC /* RTCH264ProfileLevelId.mm in Sources */,
>+				4144B3D32169896D004363AC /* RTCNativeI420Buffer.mm in Sources */,
>+				4144B3D121698967004363AC /* RTCNativeMutableI420Buffer.mm in Sources */,
> 				419C82B91FE20DCD0040C30F /* rtcp_demuxer.cc in Sources */,
>-				413E67762169866100EF37ED /* RTCVideoCodecConstants.mm in Sources */,
> 				5CD285FA1E6A64520094FDC8 /* rtcp_nack_stats.cc in Sources */,
> 				5CDD88971E43BE3C00621E92 /* rtcp_packet.cc in Sources */,
> 				5CDD889A1E43BE3C00621E92 /* rtcp_receiver.cc in Sources */,
>@@ -16835,15 +16929,31 @@
> 				419C82FF1FE20F020040C30F /* rtcp_transceiver.cc in Sources */,
> 				419C83021FE20F020040C30F /* rtcp_transceiver_config.cc in Sources */,
> 				419C83001FE20F020040C30F /* rtcp_transceiver_impl.cc in Sources */,
>+				41889D77216BC4EC004614DD /* RTCRtpEncodingParameters.mm in Sources */,
>+				41889D79216BC4EC004614DD /* RTCRtpFragmentationHeader+Private.mm in Sources */,
>+				41FCBB8921B200FD00A5DF27 /* RTCRtpFragmentationHeader.m in Sources */,
> 				5C63FA781E417AED002CA531 /* rtcstats.cc in Sources */,
> 				5C63FA771E417AED002CA531 /* rtcstats_objects.cc in Sources */,
> 				5CD285071E6A60570094FDC8 /* rtcstatscollector.cc in Sources */,
> 				5C63FA791E417AED002CA531 /* rtcstatsreport.cc in Sources */,
> 				4102F69221273206006AE8D7 /* rtcstatstraversal.cc in Sources */,
>+				413E67762169866100EF37ED /* RTCVideoCodecConstants.mm in Sources */,
>+				4144B3E82169C5E3004363AC /* RTCVideoCodecInfo+Private.mm in Sources */,
>+				413E67982169883900EF37ED /* RTCVideoCodecInfo.m in Sources */,
>+				417953B8216982420028266B /* RTCVideoDecoderFactoryH264.m in Sources */,
>+				417953C72169824B0028266B /* RTCVideoDecoderH264.mm in Sources */,
>+				413E67682169854B00EF37ED /* RTCVideoDecoderVP8.mm in Sources */,
>+				417953B9216982420028266B /* RTCVideoEncoderFactoryH264.m in Sources */,
>+				417953CA2169824B0028266B /* RTCVideoEncoderH264.mm in Sources */,
>+				413E67A32169889500EF37ED /* RTCVideoEncoderQpThresholds.m in Sources */,
>+				4144B3F7216B09CB004363AC /* RTCVideoEncoderSettings+Private.mm in Sources */,
>+				413E67652169854600EF37ED /* RTCVideoEncoderVP8.mm in Sources */,
>+				413E6790216987DB00EF37ED /* RTCVideoFrame.mm in Sources */,
>+				413E676B2169854B00EF37ED /* RTCWrappedNativeVideoDecoder.mm in Sources */,
>+				413E676D2169854B00EF37ED /* RTCWrappedNativeVideoEncoder.mm in Sources */,
> 				41D6B45921273159008F9353 /* rtp_bitrate_configurator.cc in Sources */,
> 				419C82B41FE20DCD0040C30F /* rtp_config.cc in Sources */,
> 				413092021EF8D0A600757C55 /* rtp_demuxer.cc in Sources */,
>-				417953B5216982420028266B /* RTCDefaultVideoDecoderFactory.m in Sources */,
> 				5CDD88B01E43BE3C00621E92 /* rtp_format.cc in Sources */,
> 				5CDD88A41E43BE3C00621E92 /* rtp_format_h264.cc in Sources */,
> 				5CDD88A61E43BE3C00621E92 /* rtp_format_video_generic.cc in Sources */,
>@@ -16854,6 +16964,7 @@
> 				41E02C7A2127347400C27CD6 /* rtp_generic_frame_descriptor.cc in Sources */,
> 				419241B02127590200634FCF /* rtp_generic_frame_descriptor_extension.cc in Sources */,
> 				4145E4DC1EF8CCEF00FCF6E6 /* rtp_header_extension_map.cc in Sources */,
>+				413E677D216986AE00EF37ED /* rtp_header_extension_size.cc in Sources */,
> 				5CDD88B51E43BE3C00621E92 /* rtp_header_extensions.cc in Sources */,
> 				5CDD88B71E43BE3C00621E92 /* rtp_header_parser.cc in Sources */,
> 				41F9BF8B2051C80100ABF0B9 /* rtp_headers.cc in Sources */,
>@@ -16867,7 +16978,6 @@
> 				5CDD88CC1E43BE3D00621E92 /* rtp_rtcp_impl.cc in Sources */,
> 				5CDD88D31E43BE3D00621E92 /* rtp_sender.cc in Sources */,
> 				5CDD88CE1E43BE3D00621E92 /* rtp_sender_audio.cc in Sources */,
>-				417953DB216983910028266B /* metrics.cc in Sources */,
> 				5CDD88D11E43BE3D00621E92 /* rtp_sender_video.cc in Sources */,
> 				419C82BC1FE20DCD0040C30F /* rtp_stream_receiver_controller.cc in Sources */,
> 				5CDD85B21E43B5C000621E92 /* rtp_streams_synchronizer.cc in Sources */,
>@@ -16876,26 +16986,22 @@
> 				5CDD88D51E43BE3D00621E92 /* rtp_utility.cc in Sources */,
> 				41E02C7D2127347400C27CD6 /* rtp_video_header.cc in Sources */,
> 				41D6B45C21273159008F9353 /* rtp_video_sender.cc in Sources */,
>-				413E678B216987B900EF37ED /* RTCCVPixelBuffer.mm in Sources */,
>-				413E67742169863B00EF37ED /* vp8_temporal_layers.cc in Sources */,
> 				413091F41EF8CF9200757C55 /* rtp_video_stream_receiver.cc in Sources */,
> 				5C4B48EB1E42C1E3002651C8 /* rtpdataengine.cc in Sources */,
> 				5CDD896C1E43BF3A00621E92 /* rtpfb.cc in Sources */,
> 				419C82C91FE20E2C0040C30F /* rtpmediautils.cc in Sources */,
>-				4144B3D821698D3D004363AC /* logging.cc in Sources */,
> 				419C83151FE242800040C30F /* rtpparameters.cc in Sources */,
> 				4102F69321273206006AE8D7 /* rtpparametersconversion.cc in Sources */,
> 				5CD285091E6A60570094FDC8 /* rtpreceiver.cc in Sources */,
>-				4144B3F5216B081B004363AC /* RTCEncodedImage+Private.mm in Sources */,
>-				416225D92169818100A91C9B /* video_frame.mm in Sources */,
> 				41F2638321267B7700274F59 /* rtpreceiverinterface.cc in Sources */,
> 				5CD2850B1E6A60570094FDC8 /* rtpsender.cc in Sources */,
>+				4144B3E22169B5BD004363AC /* rtpsenderinterface.cc in Sources */,
> 				419C82E61FE20E590040C30F /* rtptransceiver.cc in Sources */,
> 				41F2638521267B7700274F59 /* rtptransceiverinterface.cc in Sources */,
> 				4145E4CC1EF8CB8B00FCF6E6 /* rtptransport.cc in Sources */,
> 				5C4B48F11E42C1E3002651C8 /* rtputils.cc in Sources */,
> 				5CDD83F81E439A6F00621E92 /* rtt_filter.cc in Sources */,
>-				413E67722169863B00EF37ED /* temporal_layers_checker.cc in Sources */,
>+				417953EA216983F10028266B /* rtt_mult_experiment.cc in Sources */,
> 				419241CA2127593F00634FCF /* rtt_stats.cc in Sources */,
> 				419C82B71FE20DCD0040C30F /* rtx_receive_stream.cc in Sources */,
> 				419241642127385B00634FCF /* rw_lock_posix.cc in Sources */,
>@@ -16912,7 +17018,6 @@
> 				41239B3F21476DC400396F81 /* screenshare_layers.cc in Sources */,
> 				5CD2863C1E6A681C0094FDC8 /* sctptransport.cc in Sources */,
> 				5CD2850D1E6A60570094FDC8 /* sctputils.cc in Sources */,
>-				413E67652169854600EF37ED /* RTCVideoEncoderVP8.mm in Sources */,
> 				5CDD896F1E43BF3A00621E92 /* sdes.cc in Sources */,
> 				41F2636221267ADF00274F59 /* sdp_video_format.cc in Sources */,
> 				419C82DC1FE20E590040C30F /* sdputils.cc in Sources */,
>@@ -16923,11 +17028,11 @@
> 				5CDD85B91E43B5C000621E92 /* send_statistics_proxy.cc in Sources */,
> 				41299B8C2127365100B3414B /* send_time_history.cc in Sources */,
> 				5CDD89721E43BF3A00621E92 /* sender_report.cc in Sources */,
>+				41FCBB2721B1F87E00A5DF27 /* sent_packet.cc in Sources */,
> 				413A239D1FE18E0800373E99 /* sequenced_task_checker_impl.cc in Sources */,
> 				5CDD83FB1E439A6F00621E92 /* session_info.cc in Sources */,
> 				41F9BFE42051DE1800ABF0B9 /* sessiondescription.cc in Sources */,
> 				5CD285DF1E6A63430094FDC8 /* shadow_filter_update_gain.cc in Sources */,
>-				413E677A2169867A00EF37ED /* channel_receive_proxy.cc in Sources */,
> 				416D3BEB212731C200775F09 /* signal_classifier.cc in Sources */,
> 				413A235B1FE18E0700373E99 /* signalthread.cc in Sources */,
> 				41F263BB2126811500274F59 /* sigslot.cc in Sources */,
>@@ -16946,18 +17051,14 @@
> 				5C4B4CE51E4320A9002651C8 /* sleep.cc in Sources */,
> 				5CDD8BE51E43C2B500621E92 /* smooth.c in Sources */,
> 				5CDD8BE31E43C2B500621E92 /* smooth_out_data.c in Sources */,
>-				413E67A5216988DC00EF37ED /* video_stream_encoder_observer.cc in Sources */,
>-				417953B4216982420028266B /* nalu_rewriter.cc in Sources */,
> 				5CD285F61E6A63F60094FDC8 /* smoothing_filter.cc in Sources */,
> 				41DDB2562126790A00296D47 /* socket.cc in Sources */,
> 				413A22F01FE18E0700373E99 /* socketadapters.cc in Sources */,
> 				413A23681FE18E0700373E99 /* socketaddress.cc in Sources */,
>-				4144B3D621698A1A004363AC /* RTCH264ProfileLevelId.mm in Sources */,
> 				413A234D1FE18E0700373E99 /* socketaddresspair.cc in Sources */,
> 				413A22D31FE18E0700373E99 /* socketstream.cc in Sources */,
> 				5CDD8BE71E43C2B500621E92 /* sort_sq.c in Sources */,
> 				5C4B4C7D1E431F9C002651C8 /* sparse_fir_filter.cc in Sources */,
>-				4144B3E22169B5BD004363AC /* rtpsenderinterface.cc in Sources */,
> 				419241AD212758D300634FCF /* spectral_features.cc in Sources */,
> 				419241A8212758D300634FCF /* spectral_features_internal.cc in Sources */,
> 				41433D021F79B33400387B4D /* spectrum_ar_model_tables.c in Sources */,
>@@ -16992,7 +17093,6 @@
> 				41E02CBE2127356A00C27CD6 /* sslcertificate.cc in Sources */,
> 				413A23F11FE18E0800373E99 /* sslfingerprint.cc in Sources */,
> 				419C84161FE24A8D0040C30F /* sslidentity.cc in Sources */,
>-				413E677D216986AE00EF37ED /* rtp_header_extension_size.cc in Sources */,
> 				419C84171FE24AA30040C30F /* sslstreamadapter.cc in Sources */,
> 				5CDD87361E43BABE00621E92 /* standalone_vad.cc in Sources */,
> 				5CDD8BEB1E43C2B500621E92 /* state_construct.c in Sources */,
>@@ -17005,8 +17105,8 @@
> 				5CDD85BF1E43B5C000621E92 /* stream_synchronization.cc in Sources */,
> 				5C4B48F41E42C1E3002651C8 /* streamparams.cc in Sources */,
> 				41DDB26E212679D200296D47 /* string_builder.cc in Sources */,
>-				413E676B2169854B00EF37ED /* RTCWrappedNativeVideoDecoder.mm in Sources */,
> 				413A23D61FE18E0800373E99 /* string_to_number.cc in Sources */,
>+				41FCBB8021B2001200A5DF27 /* string_view.cc in Sources */,
> 				413A22B51FE18E0700373E99 /* stringencode.cc in Sources */,
> 				413A23131FE18E0700373E99 /* stringutils.cc in Sources */,
> 				5C63FA0B1E4174F6002CA531 /* stun.cc in Sources */,
>@@ -17016,6 +17116,8 @@
> 				5CD285E21E6A63430094FDC8 /* subtractor.cc in Sources */,
> 				5CD285E41E6A63430094FDC8 /* suppression_filter.cc in Sources */,
> 				5CD285E61E6A63430094FDC8 /* suppression_gain.cc in Sources */,
>+				413E67822169870F00EF37ED /* svc_config.cc in Sources */,
>+				413E67942169881900EF37ED /* svc_rate_allocator.cc in Sources */,
> 				5CDD8BEF1E43C2B500621E92 /* swap_bytes.c in Sources */,
> 				5CDD8A541E43BFB300621E92 /* sync_buffer.cc in Sources */,
> 				5CD284BE1E6A5F9F0094FDC8 /* syncable.cc in Sources */,
>@@ -17025,14 +17127,14 @@
> 				413A23DE1FE18E0800373E99 /* task_queue_gcd.cc in Sources */,
> 				413A232D1FE18E0700373E99 /* task_queue_posix.cc in Sources */,
> 				5C63FA171E4174F6002CA531 /* tcpport.cc in Sources */,
>+				413E67722169863B00EF37ED /* temporal_layers_checker.cc in Sources */,
> 				413A22A51FE18E0600373E99 /* testechoserver.cc in Sources */,
> 				413A22E51FE18E0700373E99 /* thread.cc in Sources */,
> 				413A23501FE18E0700373E99 /* thread_checker_impl.cc in Sources */,
>-				413A23DD1FE18E0800373E99 /* thread_darwin.mm in Sources */,
> 				5CDD84C31E43AF1300621E92 /* three_band_filter_bank.cc in Sources */,
>+				41FCBB2221B1F82F00A5DF27 /* throw_delegate.cc in Sources */,
> 				5CDD8A571E43BFB300621E92 /* tick_timer.cc in Sources */,
> 				419241E221275A3000634FCF /* time_delta.cc in Sources */,
>-				419C83351FE245EA0040C30F /* time_interval.cc in Sources */,
> 				5CDD8A5A1E43BFB300621E92 /* time_stretch.cc in Sources */,
> 				5CDD88DA1E43BE3D00621E92 /* time_util.cc in Sources */,
> 				419241E321275A3000634FCF /* timestamp.cc in Sources */,
>@@ -17072,9 +17174,7 @@
> 				41F9BFEA2051DEEA00ABF0B9 /* turnportfactory.cc in Sources */,
> 				5C63FA2F1E4174F6002CA531 /* turnserver.cc in Sources */,
> 				5C4B48F91E42C1E3002651C8 /* turnutils.cc in Sources */,
>-				416225E42169819500A91C9B /* objc_frame_buffer.mm in Sources */,
> 				5CDD84C51E43AF1300621E92 /* typing_detection.cc in Sources */,
>-				419C83DC1FE247F20040C30F /* udptransport.cc in Sources */,
> 				5CDD88DF1E43BE3D00621E92 /* ulpfec_generator.cc in Sources */,
> 				5CDD88E21E43BE3D00621E92 /* ulpfec_header_reader_writer.cc in Sources */,
> 				5CDD88E41E43BE3D00621E92 /* ulpfec_receiver_impl.cc in Sources */,
>@@ -17087,7 +17187,6 @@
> 				5CDD86D61E43BA2800621E92 /* vad_core.c in Sources */,
> 				5CDD86D91E43BA2800621E92 /* vad_filterbank.c in Sources */,
> 				5CDD86DC1E43BA2800621E92 /* vad_gmm.c in Sources */,
>-				417953E6216983CA0028266B /* module_common_types.cc in Sources */,
> 				5CDD86DF1E43BA2800621E92 /* vad_sp.c in Sources */,
> 				416D3BEC212731C200775F09 /* vad_with_level.cc in Sources */,
> 				41F9BFD52051DDE500ABF0B9 /* vector_buffer.cc in Sources */,
>@@ -17098,17 +17197,20 @@
> 				5CD285351E6A61590094FDC8 /* video_codec_initializer.cc in Sources */,
> 				41E02C8D212734B900C27CD6 /* video_coding_defines.cc in Sources */,
> 				5CDD84011E439A6F00621E92 /* video_coding_impl.cc in Sources */,
>-				416225E22169819500A91C9B /* objc_video_frame.mm in Sources */,
> 				419C84111FE249AB0040C30F /* video_content_type.cc in Sources */,
> 				41F2636721267ADF00274F59 /* video_decoder.cc in Sources */,
>+				417953F1216984550028266B /* video_decoder_factory.cc in Sources */,
>+				416225D82169818100A91C9B /* video_decoder_factory.mm in Sources */,
> 				41F2636521267ADF00274F59 /* video_decoder_software_fallback_wrapper.cc in Sources */,
> 				4145F6201FE1F38D00EB9CAF /* video_encoder.cc in Sources */,
> 				41E02CC52127358800C27CD6 /* video_encoder_config.cc in Sources */,
>+				416225D42169818100A91C9B /* video_encoder_factory.mm in Sources */,
> 				41F2636921267ADF00274F59 /* video_encoder_software_fallback_wrapper.cc in Sources */,
> 				5CD2846A1E6A57F40094FDC8 /* video_frame.cc in Sources */,
>-				5C4B4C1F1E431F75002651C8 /* video_frame.cc in Sources */,
>+				416225D92169818100A91C9B /* video_frame.mm in Sources */,
> 				4124554B1EF8874300F11809 /* video_frame_buffer.cc in Sources */,
> 				5C4B4C1E1E431F75002651C8 /* video_frame_buffer.cc in Sources */,
>+				416225D72169818100A91C9B /* video_frame_buffer.mm in Sources */,
> 				4102F6C9212733B7006AE8D7 /* video_quality_observer.cc in Sources */,
> 				419C82BE1FE20DCD0040C30F /* video_receive_stream.cc in Sources */,
> 				5CDD85CA1E43B5C000621E92 /* video_receive_stream.cc in Sources */,
>@@ -17116,15 +17218,14 @@
> 				5C4B4C201E431F75002651C8 /* video_render_frames.cc in Sources */,
> 				419C82B31FE20DCD0040C30F /* video_send_stream.cc in Sources */,
> 				5CDD85CD1E43B5C000621E92 /* video_send_stream.cc in Sources */,
>-				417953CA2169824B0028266B /* RTCVideoEncoderH264.mm in Sources */,
> 				4102F6CC212733B7006AE8D7 /* video_send_stream_impl.cc in Sources */,
>-				417953F1216984550028266B /* video_decoder_factory.cc in Sources */,
> 				5CDD84041E439A6F00621E92 /* video_sender.cc in Sources */,
> 				41A08BB0212681C8001D5D7B /* video_source_interface.cc in Sources */,
> 				5CDD85CF1E43B5C000621E92 /* video_stream_decoder.cc in Sources */,
> 				4102F6DB21273416006AE8D7 /* video_stream_decoder_create.cc in Sources */,
> 				4102F6CB212733B7006AE8D7 /* video_stream_decoder_impl.cc in Sources */,
> 				419C831F1FE242E60040C30F /* video_stream_encoder.cc in Sources */,
>+				413E67A5216988DC00EF37ED /* video_stream_encoder_observer.cc in Sources */,
> 				419C84101FE249AB0040C30F /* video_timing.cc in Sources */,
> 				5C4B48FC1E42C1E3002651C8 /* videoadapter.cc in Sources */,
> 				5C4B48FF1E42C1E3002651C8 /* videobroadcaster.cc in Sources */,
>@@ -17142,7 +17243,7 @@
> 				419241F521275C3200634FCF /* vp8_encoder_simulcast_proxy.cc in Sources */,
> 				5CDD83851E439A3500621E92 /* vp8_header_parser.cc in Sources */,
> 				5CDD8C141E43C3B400621E92 /* vp9_noop.cc in Sources */,
>-				417953B7216982420028266B /* RTCDefaultVideoEncoderFactory.m in Sources */,
>+				417953EF216984210028266B /* vp9_profile.cc in Sources */,
> 				5CDD8BF31E43C2B500621E92 /* vq3.c in Sources */,
> 				5CDD8BF51E43C2B500621E92 /* vq4.c in Sources */,
> 				5C4B4C801E431F9C002651C8 /* wav_file.cc in Sources */,
>@@ -17165,7 +17266,6 @@
> 				5C4B4C861E431F9C002651C8 /* window_generator.cc in Sources */,
> 				5CDD8C981E43C66000621E92 /* wpd_node.cc in Sources */,
> 				5CDD8C9B1E43C66000621E92 /* wpd_tree.cc in Sources */,
>-				4144B3F7216B09CB004363AC /* RTCVideoEncoderSettings+Private.mm in Sources */,
> 				5CDD8BF91E43C2B500621E92 /* xcorr_coef.c in Sources */,
> 				41DDB2592126792800296D47 /* zero_memory.cc in Sources */,
> 			);
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>index ac6445fa00c71cdd014481202fcbc9e91c1797d3..bc12b6cdf0a9827c09451e6903b06367520a79d2 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>@@ -186,11 +186,6 @@ public:
>         return WEBRTC_VIDEO_CODEC_OK;
>     }
> 
>-    int32_t SetChannelParameters(uint32_t, int64_t) final
>-    {
>-        return WEBRTC_VIDEO_CODEC_OK;
>-    }
>-
>     int32_t Encode(const webrtc::VideoFrame& frame,
>         const webrtc::CodecSpecificInfo* codecInfo,
>         const std::vector<webrtc::FrameType>* frameTypes) final
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp
>index e5dab669d8287ebd30e14ed61517a99fb12c04fe..c24dc0799d14052b7025aaa515fce807bf9bad1c 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp
>@@ -37,6 +37,7 @@ ALLOW_UNUSED_PARAMETERS_BEGIN
> #include <webrtc/api/asyncresolverfactory.h>
> #include <webrtc/api/audio_codecs/builtin_audio_decoder_factory.h>
> #include <webrtc/api/audio_codecs/builtin_audio_encoder_factory.h>
>+#include <webrtc/api/create_peerconnection_factory.h>
> #include <webrtc/api/peerconnectionfactoryproxy.h>
> #include <webrtc/modules/audio_processing/include/audio_processing.h>
> #include <webrtc/p2p/base/basicpacketsocketfactory.h>
>diff --git a/Source/WebKit/NetworkProcess/webrtc/LibWebRTCSocketClient.cpp b/Source/WebKit/NetworkProcess/webrtc/LibWebRTCSocketClient.cpp
>index 73b01967594e0ae4d62cbfe5df66c71d7f00ba5e..c5995bc53255fe2d376fc45fbfb9ee631ee73e71 100644
>--- a/Source/WebKit/NetworkProcess/webrtc/LibWebRTCSocketClient.cpp
>+++ b/Source/WebKit/NetworkProcess/webrtc/LibWebRTCSocketClient.cpp
>@@ -92,7 +92,7 @@ void LibWebRTCSocketClient::signalReadPacket(rtc::AsyncPacketSocket* socket, con
>     auto buffer = WebCore::SharedBuffer::create(value, length);
>     m_rtcProvider.sendFromMainThread([identifier = m_identifier, buffer = WTFMove(buffer), address = RTCNetwork::isolatedCopy(address), packetTime](IPC::Connection& connection) {
>         IPC::DataReference data(reinterpret_cast<const uint8_t*>(buffer->data()), buffer->size());
>-        connection.send(Messages::WebRTCSocket::SignalReadPacket(data, RTCNetwork::IPAddress(address.ipaddr()), address.port(), packetTime.timestamp), identifier);
>+        connection.send(Messages::WebRTCSocket::SignalReadPacket(data, RTCNetwork::IPAddress(address.ipaddr()), address.port(), packetTime), identifier);
>     });
> }
> 
>diff --git a/Source/WebKit/NetworkProcess/webrtc/LibWebRTCSocketClient.h b/Source/WebKit/NetworkProcess/webrtc/LibWebRTCSocketClient.h
>index 360c2e01a4f13c5e808feeed5226771422efd14c..8f205ce029713d8ba213f143ba71bb114752def5 100644
>--- a/Source/WebKit/NetworkProcess/webrtc/LibWebRTCSocketClient.h
>+++ b/Source/WebKit/NetworkProcess/webrtc/LibWebRTCSocketClient.h
>@@ -35,8 +35,8 @@ namespace rtc {
> class AsyncPacketSocket;
> class SocketAddress;
> struct PacketOptions;
>-struct PacketTime;
> struct SentPacket;
>+typedef int64_t PacketTime;
> }
> 
> namespace WebCore {
>diff --git a/Source/WebKit/NetworkProcess/webrtc/NetworkRTCProvider.cpp b/Source/WebKit/NetworkProcess/webrtc/NetworkRTCProvider.cpp
>index ce0f13da7594ad6fab87147e9de2d58b82ebdd93..dd97798daea40ace629ad8f0a3b411e4928d3f82 100644
>--- a/Source/WebKit/NetworkProcess/webrtc/NetworkRTCProvider.cpp
>+++ b/Source/WebKit/NetworkProcess/webrtc/NetworkRTCProvider.cpp
>@@ -37,6 +37,7 @@
> #include "WebRTCSocketMessages.h"
> #include <WebCore/LibWebRTCMacros.h>
> #include <webrtc/rtc_base/asyncpacketsocket.h>
>+#include <webrtc/rtc_base/logging.h>
> #include <wtf/MainThread.h>
> #include <wtf/text/WTFString.h>
> 
>diff --git a/Source/WebKit/NetworkProcess/webrtc/NetworkRTCSocket.h b/Source/WebKit/NetworkProcess/webrtc/NetworkRTCSocket.h
>index d9deaac625503194bf2deef0ab85978c706c6f58..6e4bfe2b16419cbbc51b80d95f759efe2edd92af 100644
>--- a/Source/WebKit/NetworkProcess/webrtc/NetworkRTCSocket.h
>+++ b/Source/WebKit/NetworkProcess/webrtc/NetworkRTCSocket.h
>@@ -42,8 +42,8 @@ namespace rtc {
> class AsyncPacketSocket;
> class SocketAddress;
> struct PacketOptions;
>-struct PacketTime;
> struct SentPacket;
>+typedef int64_t PacketTime;
> }
> 
> namespace WebCore {
>diff --git a/Source/WebKit/WebProcess/Network/webrtc/LibWebRTCSocket.cpp b/Source/WebKit/WebProcess/Network/webrtc/LibWebRTCSocket.cpp
>index 70541573373d269271202b061eb9c15c8ec7656a..e2c5df557c40115ddf48fd4079afb949a6c25569 100644
>--- a/Source/WebKit/WebProcess/Network/webrtc/LibWebRTCSocket.cpp
>+++ b/Source/WebKit/WebProcess/Network/webrtc/LibWebRTCSocket.cpp
>@@ -82,7 +82,7 @@ void LibWebRTCSocket::signalAddressReady(const rtc::SocketAddress& address)
> void LibWebRTCSocket::signalReadPacket(const WebCore::SharedBuffer& buffer, rtc::SocketAddress&& address, int64_t timestamp)
> {
>     m_remoteAddress = WTFMove(address);
>-    SignalReadPacket(this, buffer.data(), buffer.size(), m_remoteAddress, rtc::PacketTime(timestamp, 0));
>+    SignalReadPacket(this, buffer.data(), buffer.size(), m_remoteAddress, timestamp);
> }
> 
> void LibWebRTCSocket::signalSentPacket(int rtcPacketID, int64_t sendTimeMs)

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Attachments on bug 192316: 356390 | 356436 | 356469 | 356793 | 356816 | 356841